1 === release 0.10.26 ===
3 2010-02-10 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6 releasing 0.10.26, "You will know when you get there"
8 2010-02-08 11:21:35 +0100 Benjamin M. Schwartz <bens@alum.mit.edu>
10 * ext/theora/gsttheoradec.c:
11 theoradec: PARs of 0:x, x:0 and 0:0 are all allowed and map to 1:1
14 2010-01-24 12:31:04 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
16 * ext/ogg/gstoggstream.c:
17 oggdemux: use the default granpos functions for kate streams
18 Set timestamps on kate packets. See bug #600929.
20 2010-02-05 01:18:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
23 * win32/common/_stdint.h:
24 * win32/common/config.h:
27 2010-02-04 18:52:59 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
30 po: update translations
32 2010-02-04 18:32:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
34 * gst/playback/gstplaybin2.c:
35 Revert "playbin2: Only allow to set the URIs in states <= READY or from an about-to-finish signal handler"
36 This reverts commit 7335ce5d3e03c126a417a721571cb6f3af136ecf.
37 Support abusing the uri property to configure the next uri to play
38 outside of the about-to-finish handler for the time being after all.
39 We also shouldn't use thread private structures for this, since it
40 should be possible to block the thread that emitted about-to-finish
41 while the main thread sets the uri property. See #607226.
43 2010-02-02 10:18:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
45 * ext/ogg/gstoggdemux.c:
46 oggdemux: Don't leak allocated buffers
47 This can happen if the combined flow return is not OK although the
48 allocation succeeded or if the packet in question is a BOS and we're
49 not going to push headers.
52 2010-02-01 11:44:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
54 * gst/playback/gsturidecodebin.c:
55 uridecodebin: clean up decodebin properties
56 When reusing a decodebin2 element, clear the properties we might have changed,
57 to their default values or else we might end up with old configuration.
60 2010-01-29 13:56:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
62 * gst/playback/gstplaybin2.c:
63 playbin2: when no uri is set, post an error message
64 When no uri is set, don't just return STATE_CHANGE_FAILURE from the
65 state change function, but actually post an error message.
67 2010-01-30 15:18:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
70 Automatic update of common submodule
71 From 15d47a6 to 96dc793
73 2010-01-28 17:12:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
75 * gst/adder/gstadder.c:
76 adder: don't hold object lock when calling peer elements
77 Do not hold the object lock while we call methods on peer elements as this can
81 2010-01-27 01:12:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
86 2010-01-27 01:07:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
88 * win32/common/_stdint.h:
89 * win32/common/config.h:
90 * win32/common/gstrtsp-enumtypes.c:
91 * win32/common/interfaces-enumtypes.c:
92 * win32/common/interfaces-enumtypes.h:
93 * win32/common/pbutils-enumtypes.c:
94 * win32/common/video-enumtypes.c:
95 win32: update generated files for non-autotools win32 builds
97 2010-01-27 00:56:00 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
131 po: update translation files
133 2010-01-27 00:41:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
135 * gst-libs/gst/audio/gstaudiosrc.c:
136 audiosrc: add gratuitious FIXME for use of generic G_TYPE_POINTER type
138 2010-01-26 16:47:40 +0100 Edward Hervey <bilboed@bilboed.com>
140 * gst/playback/gstdecodebin2.c:
141 decodebin2: Don't skip an element when getting the topology
144 2010-01-24 14:41:44 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
146 * ext/ogg/gstoggdemux.c:
147 oggdemux: sparse streams aren't timed by end time, and their duration isn't implicit
148 Fixes timestamps and durations on Kate subtitle streams.
149 See http://www.xiph.org/ogg/doc/ogg-multiplex.html section 'start-time and
150 end-time positioning' for some more details, and bug #600929.
152 2010-01-23 20:15:08 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
154 * ext/ogg/gstoggstream.c:
155 oggdemux: properly set up the media type for kate streams
158 2010-01-25 18:57:52 +0100 Julien Moutte <julien@fluendo.com>
160 * gst/playback/gstsubtitleoverlay.c:
161 subtitleoverlay: relax caps template on sink pads
162 Allow any caps on sink pad templates as we could do passthrough with non raw
165 2010-01-25 15:14:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
167 * ext/ogg/gstoggdemux.c:
168 * ext/ogg/gstoggstream.h:
169 oggdemux: use right type for the serialno
170 Use a consistent type for the serialno to avoid problems when comparing between
171 signed and unsigned variants.
174 2010-01-25 14:00:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
176 * ext/ogg/gstoggdemux.c:
177 oggdemux: don't push headers twice
178 Don't push the stream headers twice but only in the activation of a chain.
181 2010-01-25 13:18:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
183 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
185 2010-01-25 12:31:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
187 * ext/ogg/gstoggdemux.c:
188 * ext/ogg/gstoggdemux.h:
189 oggdemux: rename a variable
190 Rename the 'seekable' variable to 'pullmode'. We might be able to seek in push
193 2010-01-25 12:22:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
195 * gst/playback/gstinputselector.c:
196 Revert "inputselector: Protect g_object_notify() with the object's mutex"
197 This reverts commit a37426c41c80fd21e5017fea01a786c05bcd9661, it's
198 causing deadlocks with playbin2.
200 2010-01-24 20:55:26 +0100 Kipp Cannon <kcannon@ligo.caltech.edu>
202 * gst/playback/gstinputselector.c:
203 inputselector: Protect g_object_notify() with the object's mutex
204 This works around the thread unsafety of g_object_notify()
207 2010-01-24 20:46:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
209 * gst/typefind/gsttypefindfunctions.c:
210 typefindfunctions: Add typefinder for ISO MP4 files
213 2010-01-24 13:29:07 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
215 * ext/ogg/gstoggdemux.c:
216 oggdemux: fix crash when freeing headers
217 Use _ogg_packet_free() instead of gst_mini_object_unref in one more
218 place now that the header list contains ogg packets and not buffers.
219 file: Stephen_Fry-Happy_Birthday_GNU-nq_600px_425kbit.ogv
221 2010-01-24 08:57:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
223 * ext/ogg/gstoggdemux.c:
224 oggdemux: Strip trailing \0 for subtitle OGM streams
227 2010-01-23 22:09:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
229 * ext/ogg/gstoggdemux.c:
230 oggdemux: Correctly set DELTA_UNIT flag for OGM streams
232 2010-01-23 22:05:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
234 * ext/ogg/gstoggdemux.c:
235 oggdemux: Don't strip all 0-bytes from the end of OGM packets
236 This fixes broken packets pushed downstream by oggdemux for
237 MPEG4 streams for example.
239 2010-01-23 22:03:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
241 * ext/ogg/gstoggdemux.c:
242 oggdemux: Extract tags from OGM text streams and don't push them downstream
244 2010-01-23 14:46:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
246 * ext/ogg/gstoggdemux.c:
247 oggdemux: Store header/queued packets as ogg_packet and use normal peer chaining functions to pass them downstream
249 2010-01-23 15:25:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
251 * gst/typefind/gsttypefindfunctions.c:
252 typefinding: optimise AC-3 typefinder a bit
253 Make AC-3 typefinder use the DataScanCtx stuff so we don't have to
254 do gst_type_find_peek() in the inner loop all the time. Also return
255 when we've suggested AC3 caps, instead of continuing with the loop.
257 2010-01-23 14:31:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
259 * gst/typefind/gsttypefindfunctions.c:
260 Revert "typefind: Reduce number of calls to gst_type_find_peek."
261 This reverts commit c661bfaa991c58f1fbd9fbc0dae90b8b2c27f92b.
262 This breaks AC-3 typefinding for all cases where the first frame
265 2010-01-23 15:35:05 +0100 Edward Hervey <bilboed@bilboed.com>
267 * gst-libs/gst/pbutils/descriptions.c:
268 pbutils: Add description for Zip Block Motion Video
270 2010-01-23 15:34:54 +0100 Edward Hervey <bilboed@bilboed.com>
272 * gst-libs/gst/riff/riff-media.c:
273 riff: Add mapping for Zip Block Motion Video
275 2010-01-23 15:26:37 +0100 Edward Hervey <bilboed@bilboed.com>
277 * gst-libs/gst/riff/riff-media.c:
278 riff: YUNV is a fourcc which is also used for YUY2 raw video
280 2010-01-23 15:13:45 +0100 Edward Hervey <bilboed@bilboed.com>
282 * gst-libs/gst/riff/riff-media.c:
283 riff: vp61 and VP61 are also valid On2 VP6 fourcc
285 2010-01-23 15:10:45 +0100 Edward Hervey <bilboed@bilboed.com>
287 * gst-libs/gst/riff/riff-media.c:
288 riff: Add mapping for On2 VP5
290 2010-01-23 15:04:35 +0100 Edward Hervey <bilboed@bilboed.com>
292 * gst-libs/gst/riff/riff-media.c:
293 riff: Add mapping for Sigma-Designs MPEG4
294 It's actually a xvid-compatible stream. both xviddec and ffmpeg handle it.
296 2010-01-23 14:35:28 +0100 Edward Hervey <bilboed@bilboed.com>
298 * gst-libs/gst/pbutils/descriptions.c:
299 pbutils: Add description for LOCO Lossless codec
301 2010-01-23 14:35:16 +0100 Edward Hervey <bilboed@bilboed.com>
303 * gst-libs/gst/riff/riff-media.c:
304 riff: Add mapping for LOCO Lossless codec
306 2010-01-23 14:08:39 +0100 Edward Hervey <bilboed@bilboed.com>
308 * gst-libs/gst/riff/riff-media.c:
309 riff: Add support for YV12 / Uncompressed packed YVU 4:2:2
311 2010-01-23 13:50:26 +0100 Edward Hervey <bilboed@bilboed.com>
313 * gst-libs/gst/pbutils/descriptions.c:
314 pbutils: add description for Autodesk Animator codec
316 2010-01-23 13:50:09 +0100 Edward Hervey <bilboed@bilboed.com>
318 * gst-libs/gst/riff/riff-media.c:
319 riff: Add mapping for Autodesk Animator Codec
321 2010-01-23 13:20:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
323 * ext/ogg/gstoggdemux.c:
324 oggdemux: ...and set caps on queued packet buffers too
326 2010-01-23 13:19:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
328 * ext/ogg/gstoggdemux.c:
329 oggdemux: Set caps on header buffers
331 2010-01-22 16:23:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
333 * gst/playback/gsturidecodebin.c:
334 uridecodebin: handle raw sources about-to-finish signals
335 When we are dealing with a source that produces raw audio/video, we don't use a
336 decodebin2 to decode the data and we thus don't have the drained/about-to-finish
337 signal emited. To fix this, we add a padprobe on the source pads and emit the
338 drained signal ourselves. This then makes playbin2 emit the about-to-finish
339 signal for raw sources such as cdda://
342 2010-01-22 16:15:54 +0200 Stefan Kost <ensonic@users.sf.net>
344 * gst/typefind/gsttypefindfunctions.c:
345 typefind: include stdio.h for sscanf
347 2010-01-22 01:49:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
349 * gst/typefind/gsttypefindfunctions.c:
350 typefinding: add PNM typefinder
351 Add PNM typefinder, so we can remove the one that's in the PNM plugin
352 in -bad (which btw uses different/wrong media types that don't match
353 the ones used by gdkpixbufdec) and people don't make fun of us for
354 loading image decoders when typefinding and playing back audio files.
356 2010-01-21 19:31:23 +0100 Thijs Vermeir <thijsvermeir@gmail.com>
358 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
359 * gst/ffmpegcolorspace/imgconvert.c:
360 ffmpegcolorspace: rename performance category
361 rename the performance category to ffmpegcolorspace_performance
362 as there is already a global GST_CAT_PERFORMANCE in core
364 2010-01-21 17:32:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
366 * ext/ogg/gstoggdemux.c:
367 * ext/ogg/gstoggdemux.h:
368 oggdemux: keep track of added pads
369 Keep track of the pads we added and removed.
370 Remove some unused fields.
371 Don't add pads for which we don't have caps.
373 2010-01-21 17:31:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
375 * ext/ogg/gstoggstream.c:
376 oggstream: don't call NULL setup functions
377 If we find a known mapper but it doesn't have a setup function, simply skip it
380 2010-01-21 17:30:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
382 * ext/ogg/gstoggstream.c:
383 oggstream: avoid division by 0 on bad annodex streams
385 2010-01-21 13:47:01 +0100 Edward Hervey <bilboed@bilboed.com>
387 * gst-libs/gst/pbutils/descriptions.c:
388 pbutils: Add description for y4m container
390 2010-01-19 14:31:34 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
392 * gst-libs/gst/rtp/gstbasertppayload.c:
393 basertppayload: ptime/maxptime should be unsigned
394 https://bugzilla.gnome.org/show_bug.cgi?id=607403
396 2010-01-18 21:16:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
398 * gst-libs/gst/rtp/gstbasertppayload.c:
399 * gst-libs/gst/rtp/gstbasertppayload.h:
400 basertppayload: ptime should be in nanoseconds
401 https://bugzilla.gnome.org/show_bug.cgi?id=607403
403 2010-01-20 00:53:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
406 Automatic update of common submodule
407 From 14cec89 to 15d47a6
409 2010-01-19 13:33:06 -0800 David Schleef <ds@schleef.org>
411 * gst/typefind/gsttypefindfunctions.c:
412 typefind: rewrite h.264 detection
413 Make detection simpler: check for NALs, check that they make
414 sense, and report how certain we are that it's a raw H.264 stream.
417 2010-01-18 14:33:30 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
419 * gst-libs/gst/rtp/gstbasertppayload.c:
420 basertppayload: Reject empty caps
421 https://bugzilla.gnome.org/show_bug.cgi?id=607353
423 2010-01-19 08:39:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
425 * ext/ogg/gstoggdemux.c:
426 oggdemux: No need to subtract begin time
427 Last stop is already based on the chain start and there is no need
428 to subtract the chain start as it may lead to a negative overflow.
429 This was causing seeking issues when the target chain was not
430 the first one (that has chain start = 0)
433 2010-01-19 09:25:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
435 * gst-libs/gst/audio/audio.h:
436 audio: Use rounding scaling functions for GST_CLOCK_TIME_TO_FRAMES and _FRAMES_TO_CLOCK_TIME
439 2010-01-18 15:22:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
441 * ext/ogg/gstoggdemux.c:
442 oggdemux: granulepos is relative to its chain
443 When performing seeks, the granulepos should be offset by
444 its chain start time to avoid using wrong values to
445 update segment's last_stop. A sample file is indicated on
448 2010-01-18 17:57:16 +0100 Edward Hervey <bilboed@bilboed.com>
450 * gst-libs/gst/pbutils/descriptions.c:
451 pbutils: Add description for MXF container format
453 2010-01-18 10:07:30 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
455 * gst/playback/gstplaysink.c:
456 playsink: re-use iterator callback to avoid code duplication
458 2010-01-18 02:08:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
460 * gst/playback/gstplaysink.c:
461 playsink: when looking for sink properties, make sure they have the right type
462 We don't want to end up setting values on elements where the property is of
463 a different type than we expect. Can't transform the value either, since we
464 can't really make assumptions about the scale and transform function.
465 Fixes crashes when using playbin2 with apexsink (#606949).
467 2010-01-18 09:30:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
469 * gst/playback/gstplaybin2.c:
470 playbin2: Only allow to set the URIs in states <= READY or from an about-to-finish signal handler
471 Changing the URIs in a state > READY results in unexpected behaviour,
472 i.e. the new URIs are only used after the current track has finished.
475 2010-01-15 19:52:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
477 * gst/playback/gstdecodebin2.c:
478 decodebin2: sprinkle some more locking
479 ... to avoid races and ensure some data structure consistency.
482 2010-01-14 18:26:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
484 * gst/playback/gstdecodebin2.c:
485 decodebin2: mind blocked pads when shutting down
486 Fix regression in shutdown deadlock handling now that the
487 target of a ghostpad is blocked instead of ghostpad itself.
490 2010-01-14 13:36:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
492 * gst/playback/gstplaysink.c:
493 playsink: Fix disabling of subtitles if subtitles were used before
494 In this case the video still goes through the text chain and
495 subtitles are still going in there, in case subtitles are
496 enabled again. This makes sure that re-enabling subtitles
498 Fixes hanging video when disabling subtitles, caused by an
501 2010-01-14 10:43:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
503 * gst/playback/gstplaybin2.c:
504 playbin2: fix pad ref leak
506 2010-01-12 21:42:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
508 * docs/plugins/Makefile.am:
509 docs: fix out-of-source build
511 2009-04-29 11:50:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
513 * tests/icles/stress-playbin.c:
514 stress-playbin: fix error return check
516 2010-01-14 10:10:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
518 * ext/theora/Makefile.am:
519 * ext/theora/gsttheora.c:
520 * ext/theora/gsttheoradec.c:
521 * ext/theora/gsttheoraenc.c:
522 * ext/theora/gsttheoraparse.c:
523 * ext/theora/theora.c:
524 * ext/theora/theoradec.c:
525 * ext/theora/theoraenc.c:
526 * ext/theora/theoraparse.c:
527 theora: Rename source files to have the same name as the headers
529 2010-01-14 10:07:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
531 * ext/vorbis/Makefile.am:
532 * ext/vorbis/gstvorbis.c:
533 * ext/vorbis/gstvorbisdec.c:
534 * ext/vorbis/gstvorbisenc.c:
535 * ext/vorbis/gstvorbisparse.c:
536 * ext/vorbis/gstvorbistag.c:
537 * ext/vorbis/vorbis.c:
538 * ext/vorbis/vorbisdec.c:
539 * ext/vorbis/vorbisenc.c:
540 * ext/vorbis/vorbisparse.c:
541 * ext/vorbis/vorbistag.c:
542 vorbis: Rename source files to have the same name as the headers
544 2010-01-14 10:05:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
546 * ext/vorbis/Makefile.am:
547 * ext/vorbis/gstvorbiscommon.c:
548 * ext/vorbis/gstvorbiscommon.h:
549 * ext/vorbis/vorbisdec.c:
550 * ext/vorbis/vorbisenc.c:
551 vorbis: Move channel layout definitions into a single separate file
552 ...instead of having two copies.
554 2010-01-14 08:19:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
556 * ext/vorbis/vorbisdec.c:
557 * ext/vorbis/vorbisenc.c:
558 vorbis: Add official 6.1 and 7.1 channel mappings
559 These are in the Vorbis spec since 2010-01-13. Fixes bug #606926.
561 2010-01-13 23:05:45 +0100 Benjamin Otte <otte@redhat.com>
563 * gst-libs/gst/rtsp/gstrtspdefs.c:
564 rtsp: Don't define h_error ourselves
565 It's included from netdb.h and that header might define it differently,
566 which can lead to build failures.
568 2010-01-13 17:36:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
570 * gst/typefind/gsttypefindfunctions.c:
571 typefind: mp4 video is not parsed
573 2010-01-13 12:49:20 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
575 * gst/typefind/gsttypefindfunctions.c:
576 typefind: Add aac stream-format to caps
577 Also add the aac stream-format field on the caps when
580 2010-01-13 09:39:54 +0100 Brijesh Singh <brijesh.ksingh@gmail.com>
582 * gst/playback/gstplaysink.c:
583 playsink: Fix handling of the native audio/video flags
586 2010-01-12 16:35:50 +0100 Edward Hervey <bilboed@bilboed.com>
588 * ext/ogg/gstoggdemux.c:
589 oggdemux: Fix unitialized variable.
590 If the package isn't handled, gracefully return GST_FLOW_OK.
592 2010-01-10 23:50:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
594 * gst-libs/gst/interfaces/xoverlay.c:
595 docs: flesh out GtkXOverlay docs some more and add example for Gtk+ >= 2.18
596 Explain why the whole bus sync handler mess is needed. Add section about
597 how to use GstXOverlay in connection with Gtk+ and mention the Gtk+ API
598 break issue and how to work around it (see #601809).
600 2010-01-10 21:18:04 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
602 * gst-libs/gst/netbuffer/gstnetbuffer.c:
603 docs: minor netbuffer documentation fix
605 2010-01-10 20:41:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
639 po: update translated strings
640 Queue2 moved into core, so remove its strings.
642 2010-01-08 16:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
644 * ext/ogg/gstoggdemux.c:
645 * ext/ogg/gstoggstream.h:
646 oggdemux: push headers when activating chains
647 Keep a list of headers for each stream of a chain. When a chain is activated,
648 push the headers before pushing the data so that decoders can sync.
649 Fix seeking in chains, take the chain start time into account when comparing
653 2010-01-07 15:26:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
655 * gst-libs/gst/tag/Makefile.am:
656 * gst-libs/gst/tag/lang-tables.c:
657 * gst-libs/gst/tag/lang-tables.dat:
658 * gst-libs/gst/tag/lang.c:
659 tag: fix up disting of lang-tables.c more correctly
660 lang-tables.c is included by lang.c and not really a proper source
661 file that should be compiled into its own object, so rename it to
662 lang-tables.dat and put it into EXTRA_DIST instead to ensure it
665 2010-01-07 13:50:03 +0000 Christian Schaller <christian.schaller@collabora.co.uk>
667 * gst-libs/gst/tag/Makefile.am:
668 * gst-plugins-base.spec.in:
669 Add missing source file for tagger to Makefile and update spec file
671 2010-01-06 18:30:57 -0800 Mark Yen <mook@songbirdnest.com>
673 * gst-libs/gst/riff/riff-media.c:
674 riff-media: handle 32 bit raw RGB video.
676 2010-01-06 13:57:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
678 * ext/ogg/gstoggstream.c:
679 oggdemux: decide flac header packet by content rather than count
681 2010-01-06 13:56:26 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
683 * ext/ogg/gstoggdemux.c:
684 oggdemux: reset header packet count at bos page
686 2010-01-06 13:39:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
688 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
689 audiopayload: add support for buffer-lists
691 2010-01-06 11:33:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
693 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
695 2010-01-05 17:17:58 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
697 * ext/pango/gsttextoverlay.c:
698 textoverlay: Ignore zero framerate
699 https://bugzilla.gnome.org/show_bug.cgi?id=606163
701 2009-12-29 18:45:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
703 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
704 basertpaudiopayload: Respect ptime if it is given
705 If the ptime is given in the caps, respect it and force the minimum
706 and maximum sizes to be exactly the requested ptime.
707 https://bugzilla.gnome.org/show_bug.cgi?id=606050
709 2009-12-29 18:36:29 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
711 * gst-libs/gst/rtp/gstbasertppayload.c:
712 * gst-libs/gst/rtp/gstbasertppayload.h:
713 rtpbasepayload: Store ptime from caps
714 https://bugzilla.gnome.org/show_bug.cgi?id=606050
716 2009-12-02 19:40:58 +0530 Olivier Crête <olivier.crete@collabora.co.uk>
718 * gst-libs/gst/rtp/gstbasertppayload.c:
719 basertppayload: Accept maxptime from caps
720 https://bugzilla.gnome.org/show_bug.cgi?id=606050
722 2010-01-05 14:11:06 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
724 * ext/ogg/gstoggstream.c:
725 oggdemux: enhance flac packet duration calculation
727 2010-01-05 10:38:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
729 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
731 2010-01-04 09:49:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
733 * tests/examples/seek/seek.c:
734 * tests/icles/test-colorkey.c:
735 examples: use Gtk+-2.18 API conditionally
736 so the seek example and colorkey test work with older Gtk+ versions
740 2009-12-29 00:53:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
742 * tests/icles/test-colorkey.c:
743 tests: fix colorkey test up for Gtk+ >= 2.18
744 Make test-colorkey work with newer versions of Gtk+.
747 2009-12-29 00:40:27 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
749 * tests/examples/seek/seek.c:
750 examples: make seek example work with Gtk+ >= 2.18
751 Gtk+ broke API slightly with the introduction of
752 client-side windows in Gtk+ 2.18. Fix up seek
753 example to work with newer Gtk+ versions.
756 2009-12-26 23:29:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
758 * tests/icles/stress-xoverlay.c:
759 tests: fix warning and memory leak in stress-overlay test
760 Not all messages have structures and we need to unref messages
761 when returning GST_BUS_DROP in the sync bus handler.
763 2009-12-26 18:46:50 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
765 * gst/audiorate/gstaudiorate.c:
766 audiorate: correctly eat empty and dummy buffers
768 2009-12-24 19:56:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
770 * gst/adder/gstadder.c:
771 adder: be a lot smarter with buffer management
773 Try to reuse one of the input buffer as the output buffer. This usually works
774 and avoids an allocation and a memcpy.
775 Be smarter with GAP buffers so that they don't get mixed or cleared at all. Also
776 try to use a GAP buffer as the output buffer when all input buffers are GAP
779 2009-12-24 16:30:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
781 * gst/adder/Makefile.am:
782 * gst/adder/gstadder.c:
783 * tests/check/elements/adder.c:
784 adder: use collectpads clipping function
785 Install a clipping function in the collectpads and use the audio clipping helper
786 function to perform clipping to the segment boundaries.
789 2009-12-24 13:58:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
791 * gst/adder/gstadder.c:
792 adder: fix juvenile comment
794 2009-12-23 21:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
796 * gst/playback/gstdecodebin2.c:
797 decodebin2: fix typo in debug message
799 2009-12-23 18:18:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
801 * gst/playback/gstdecodebin2.c:
802 decodebin2: avoid some type checks
804 2009-12-23 17:08:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
806 * gst/playback/gstplaybin2.c:
807 playbin2: avoid leaking selector request pads
809 2009-12-23 15:46:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
811 * gst/playback/gsturidecodebin.c:
812 uridecodebin: avoid leaking queue and typefind
813 Don't leak the queue and typefind elements that we might link after the
816 2009-12-23 15:43:52 +0100 Jonathan Matthew <jonathan@d14n.org>
818 * gst/playback/gsturidecodebin.c:
819 uridecodebin: don't name the queue
820 There is no reason to name the queue.
823 2009-12-23 15:30:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
825 * win32/common/libgstrtp.def:
826 defs: update defs with new symbols
828 2009-12-22 20:15:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
830 * docs/libs/gst-plugins-base-libs-sections.txt:
831 * gst-libs/gst/rtp/gstrtcpbuffer.c:
832 * gst-libs/gst/rtp/gstrtcpbuffer.h:
833 rtcpbuffer: add helper functions for SDES types
834 Add functions to convert SDES names to their types and back. Will be used later
835 to set SDES items using a GstStructure.
838 2009-12-21 19:12:02 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
841 Automatic update of common submodule
842 From 47cb23a to 14cec89
844 2009-12-21 18:45:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
846 * gst/audiorate/gstaudiorate.c:
847 audiorate: add Since marker for the new tolerance property
849 2009-12-21 07:57:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
851 * gst-libs/gst/tag/lang.c:
852 docs: use 'Returns: xyz' rather than 'Returns xyz' to make gtk-doc happy
854 2009-12-21 07:50:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
856 * tests/examples/app/appsrc-ra.c:
857 * tests/examples/app/appsrc-seekable.c:
858 * tests/examples/app/appsrc-stream.c:
859 * tests/examples/app/appsrc-stream2.c:
860 tests: don't use deprecated GLib API g_mapped_file_free
863 2009-12-20 17:34:46 -0800 David Schleef <ds@schleef.org>
865 * ext/theora/gsttheoraenc.h:
866 * ext/theora/theoraenc.c:
867 theoraenc: Add encoder controls for libtheora 1.1
868 Added drop-frames, cap-overflow, cap-underflow, and rate-buffer.
870 2009-12-19 21:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
872 * gst-libs/gst/audio/gstbaseaudiosink.c:
873 baseaudiosink: increase default drift tolerance to fix glitches with WMA
874 Increase default drift tolerance to 40ms to avoid glitches with decoders
875 or formats where there's a lot of timestamp jitter for some reason or
876 another (in this case: asf/wma), at least until we implement timestamp
879 2009-12-16 11:43:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
881 * gst/playback/gstdecodebin2.c:
882 decodebin2: add some debugging
884 2009-12-15 18:41:38 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
886 * gst/audiorate/gstaudiorate.c:
887 * gst/audiorate/gstaudiorate.h:
888 audiorate: add a tolerance property
889 It may not be uncommon for the input timestamps to experience some jitter
890 around the 'perfect time'. As such, instead of regularly adding and dropping
891 samples, optionally allow for some tolerance in a more relaxed approach.
892 API: GstAudioRate:tolerance
894 2009-12-15 19:50:56 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
896 * docs/plugins/Makefile.am:
897 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
898 * docs/plugins/gst-plugins-base-plugins-sections.txt:
899 * gst/audiorate/gstaudiorate.c:
900 audiorate: add documentation
902 2009-12-15 16:52:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
904 * gst/audiorate/Makefile.am:
905 * gst/audiorate/gstaudiorate.c:
906 * gst/audiorate/gstaudiorate.h:
907 audiorate: use separate header file
909 2009-12-14 21:17:57 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
911 * gst/audiorate/gstaudiorate.c:
912 audiorate: set DISCONT when resyncing (e.g. newsegment)
914 2009-12-14 18:47:27 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
916 * gst/audiorate/gstaudiorate.c:
917 audiorate: also fill up segments if possible
919 2009-12-15 19:29:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
921 * gst/audiorate/gstaudiorate.c:
922 audiorate: fix segment handling
923 Do not compare a media (buffer) time to a (bogus) running time
924 (or their offset equivalents).
926 2009-12-15 19:22:45 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
928 * gst/audiorate/gstaudiorate.c:
929 audiorate: properly report truncated samples as dropped samples
931 2009-12-13 18:43:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
933 * gst-libs/gst/tag/lang.c:
934 docs: mention that gst_tag_get_language_name() may return NULL
936 2009-12-13 18:42:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
938 * tests/check/libs/tag.c:
939 checks: some more testing for the new language code functions
941 2009-12-12 18:58:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
943 * gst-libs/gst/interfaces/mixer.c:
944 * gst-libs/gst/interfaces/mixeroptions.c:
945 * gst-libs/gst/interfaces/mixertrack.c:
946 docs: misc. mixer docs improvements
948 2009-12-12 18:16:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
950 * gst-libs/gst/app/gstappsink.c:
951 * gst-libs/gst/app/gstappsrc.c:
952 docs: add short descriptions for API reference contents page
954 2009-12-12 17:43:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
956 * gst-libs/gst/tag/lang-tables.c:
957 * gst-libs/gst/tag/mklangtables.c:
958 tag: make internal language names table static
960 2009-12-12 17:41:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
962 * gst-libs/gst/tag/lang.c:
963 * gst-libs/gst/tag/mklangtables.c:
964 tag: don't use GLib 2.22 API
965 g_mapped_file_unref() was introduced in GLib 2.22, but we depend
966 only on GLib 2.18, so use g_mapped_file_free() when compiling
967 against older GLib versions until we bump the GLib dependency.
969 2009-12-11 23:59:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
973 * docs/libs/gst-plugins-base-libs-docs.sgml:
974 * docs/libs/gst-plugins-base-libs-sections.txt:
975 * gst-libs/gst/tag/Makefile.am:
976 * gst-libs/gst/tag/lang-tables.c:
977 * gst-libs/gst/tag/lang.c:
978 * gst-libs/gst/tag/mklangtables.c:
979 * gst-libs/gst/tag/tag.h:
980 * tests/check/libs/tag.c:
981 * win32/common/libgsttag.def:
982 tag: add some utility functions for language codes and tags
983 Add some utility functions for language tags and ISO-639
984 codes. These are useful for both GUIs and elements. The
985 iso-codes package is used for language name translations
987 API: gst_tag_get_language_codes()
988 API: gst_tag_get_language_name()
989 API: gst_tag_get_language_code()
990 API: gst_tag_get_language_code_iso_639_1()
991 API: gst_tag_get_language_code_iso_639_2B()
992 API: gst_tag_get_language_code_iso_639_2T()
994 2009-12-11 12:02:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
996 * ext/ogg/gstoggstream.c:
997 ogg: ogm video has constant packet duration
999 2009-12-10 22:47:53 -0800 David Schleef <ds@schleef.org>
1001 * ext/ogg/gstoggstream.c:
1002 oggdemux: implement old fLaC mapping
1004 2009-12-10 17:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1006 * gst/tcp/gsttcpclientsrc.c:
1007 tcpclientsrc: unset flushing state too
1008 When unlocking, we set the flushing state on the fdset. Implement unlock_stop so
1009 that we can use it to unset the flushing state again.
1012 2009-12-10 16:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1014 * ext/ogg/gstoggdemux.c:
1015 * ext/ogg/gstoggdemux.h:
1016 oggdemux: remove redundant fields
1018 2009-12-09 19:03:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1020 * ext/vorbis/gstvorbisdec.h:
1021 * ext/vorbis/vorbisdec.c:
1022 vorbisdec: adapt to new oggdemux
1023 Remove all granulepos hacks and simply use the timestamps from the new oggdemux
1024 like any other decoder.
1026 2009-12-09 19:04:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1028 * ext/vorbis/vorbisdec.c:
1029 vorbisdec: fix peer query
1031 2009-12-09 17:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1033 * ext/theora/theoradec.c:
1034 theoradec: fix query
1036 2009-12-09 16:55:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1038 * ext/theora/theoradec.c:
1039 theoradec: small cleanups
1041 2009-12-09 16:38:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1043 * ext/vorbis/vorbisdec.c:
1044 vorbisdec: use gst_pad_peer_query()
1046 2009-12-09 12:10:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1048 * gst/playback/gstplaysink.c:
1049 playsink: fix video when subtitles disabled
1050 When we have a source with subtitles but they were disabled with the flags,
1051 still ghostpad the video pad instead of leaving it unlinked.
1053 2009-12-09 09:47:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1055 * ext/pango/gsttextoverlay.c:
1056 textoverlay: Only flush downstream on seeks for flushing seeks
1058 2009-12-09 09:35:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1060 * ext/pango/gsttextoverlay.c:
1061 textoverlay: Proxy buffer allocation on the video sinkpad to the srcpad
1063 2009-12-08 17:30:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1065 * tests/examples/seek/seek.c:
1066 seek: update slider only 25 times a second
1067 don't update the slider a 100 times a second, it's likely higher than the screen
1068 framerate and just wastes cpu.
1070 2009-12-08 17:23:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1072 * ext/theora/gsttheoradec.h:
1073 * ext/theora/theoradec.c:
1074 theora: remove granulepos hacks
1075 Remove the granulepos hacking now that oggdemux outputs timestamps like any
1078 2009-12-08 13:40:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1080 * gst/playback/gstplaybin2.c:
1081 playbin2: Fix stream-changed message list iteration
1082 When iterating the list and removing the current element, first
1083 get the next element and then remove the current one and not
1084 the other way around.
1086 2009-12-07 18:49:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1088 * ext/ogg/gstoggdemux.c:
1089 oggdemux: improve keyframe seeking
1090 Improve keyframe seeking.
1091 Fix reverse playback.
1093 2009-12-07 15:42:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1095 * ext/ogg/gstoggdemux.c:
1096 oggdemux: implement keyframe seeking
1097 Implement keyframe seeking in oggdemux by doing the double seek trick. First
1098 seek to the required position, then read pages for all streams to grab the
1099 granulepos (to know the timing of the keyframe) of each stream, then seek back
1100 to the first keyframe.
1102 2009-12-07 09:13:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1104 * gst/playback/gstplaysink.c:
1105 playsink: Some minor cleanup
1107 2009-12-06 18:05:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1109 * gst/playback/gstplaybin2.c:
1110 playbin2: Reset stream segments on FLUSH_STOP and don't adjust QoS events for non-time segments
1112 2009-12-04 16:35:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1114 * ext/ogg/gstoggdemux.c:
1115 oggdemux: fix timestamps after seek
1116 After a seek, discard all packets before the packet with the granulepos on it so
1117 that the output buffers contain valid timestamps.
1118 Reorder some code so that we check the timestamps before allocating and pushing
1120 Do more checks on valid packets in ogm mode.
1122 2009-12-04 15:39:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1124 * ext/ogg/gstoggdemux.c:
1125 oggdemux: add comment
1127 2009-12-04 14:01:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1129 * ext/ogg/gstoggdemux.c:
1130 oggdemux: don't do math with invalid granulepos
1131 When the current granulepos is unknown and set to -1, don't try to add durations
1134 2009-12-04 13:14:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1136 * ext/ogg/gstoggdemux.c:
1137 * ext/ogg/gstoggdemux.h:
1138 oggdemux: guard against wrong granulepos
1139 Clamp the initial granulepos to 0 instead of going negative for some badly muxed
1142 2009-12-04 12:26:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1144 * ext/theora/theoradec.c:
1145 theoradec: don't fail on bogus granulepos
1146 Do some additional checks on the granulpos timestamp before using it for
1147 calculating the duration because oggdemux generates wrong granulepos now.
1148 Fixes seeking somewhat again.
1150 2009-12-03 20:05:29 -0800 David Schleef <ds@schleef.org>
1152 * ext/ogg/gstoggdemux.c:
1153 * ext/ogg/gstoggstream.c:
1154 * ext/ogg/gstoggstream.h:
1155 oggdemux: reimplement OGM support
1156 OGM demuxing no longer requires helper elements. It's done internally
1157 in oggdemux. Vorbis comments are still not handled because I don't
1158 have anything to test with.
1160 2009-12-03 17:02:11 -0800 David Schleef <ds@schleef.org>
1162 * ext/ogg/gstoggstream.c:
1163 oggdemux: fix for I-frame-only theora
1165 2009-12-03 01:16:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1167 * ext/ogg/gstoggstream.c:
1168 ogg: log when ogg mapper doesn't accept the setup header packet
1170 2009-12-02 02:08:46 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1172 * ext/ogg/gstoggstream.c:
1173 ogg: extract width, height and PAR from theora header and add to caps
1175 2009-12-03 23:43:08 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1177 * ext/ogg/gstoggstream.c:
1178 ogg: extract number of channels from FLAC, speex and vorbis headers
1181 2009-12-03 22:14:34 +0200 Stefan Kost <ensonic@users.sf.net>
1183 * gst/playback/gstplaybin2.c:
1184 build: fix build with debug logging disabled.
1186 2009-12-03 21:07:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1188 * ext/ogg/gstoggdemux.c:
1189 * ext/ogg/gstoggstream.c:
1190 ogg: more print fixes
1191 gstoggstream.c:419: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘gint64’
1192 gstoggdemux.c:2253: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘GstClockTime’
1193 gstoggdemux.c:2333: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘GstClockTime’
1195 2009-12-03 16:57:48 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1197 * ext/ogg/gstoggparse.c:
1198 * ext/ogg/gstoggstream.c:
1199 ogg: Fixing some printf format strings
1200 Fixes some printf format strings to make it build on mac.
1202 2009-12-03 18:08:49 +0200 Stefan Kost <ensonic@users.sf.net>
1204 * gst/playback/gstfactorylists.c:
1205 * gst/playback/gstfactorylists.h:
1206 * gst/playback/gstplaybin2.c:
1207 playbin2: don't iterate the factory lists in non-debug mode
1208 When debugging is disabled, we won't see anything printed anyway.
1210 2009-12-02 23:55:55 -0800 David Schleef <ds@schleef.org>
1212 * gst/videoscale/vs_4tap.c:
1215 2009-12-02 23:27:55 +0200 Stefan Kost <ensonic@users.sf.net>
1217 * gst/subparse/qttextparse.c:
1218 build: add missing includes for sprintf and atoi
1220 2009-12-01 16:42:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1222 * gst/subparse/gstsubparse.c:
1223 * gst/subparse/qttextparse.c:
1224 subparse: Add support for some tags of qttext
1225 Currently supporting timescale, timestamps, font, size,
1226 textColor, backColor, plain, bold and italic
1229 2009-12-01 13:13:24 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1231 * gst/subparse/Makefile.am:
1232 * gst/subparse/gstsubparse.c:
1233 * gst/subparse/gstsubparse.h:
1234 * gst/subparse/qttextparse.c:
1235 * gst/subparse/qttextparse.h:
1236 subparse: add qttext support
1237 Adds basic support for qttext subtitles, still lacks markup tags
1238 to make it prettier, but the plain text already works.
1239 Implemented according to:
1240 http://www.apple.com/quicktime/tutorials/texttracks.html
1241 http://www.apple.com/quicktime/tutorials/textdescriptors.html
1244 2009-12-01 13:22:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1246 * gst/subparse/gstsubparse.c:
1247 subparse: conditionally cleanup sami context
1248 Only cleanup sami context if we are parsing sami subtitles,
1249 otherwise we might have crashes.
1251 2009-12-01 13:19:35 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1253 * gst/subparse/gstsubparse.c:
1254 subparse: Add missing caps to sink caps template
1255 Some caps were missing from the sink caps template when
1258 2009-12-01 15:06:10 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1261 Automatic update of common submodule
1262 From 87bf428 to 47cb23a
1264 2009-12-01 14:14:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1267 Automatic update of common submodule
1268 From da4c75c to 87bf428
1270 2009-11-30 10:22:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1272 * gst/playback/gstsubtitleoverlay.c:
1273 subtitleoverlay: Fix some pad refcount issues
1276 2009-11-27 18:54:57 +0100 Edward Hervey <bilboed@bilboed.com>
1279 Automatic update of common submodule
1280 From 53a2485 to da4c75c
1282 2009-11-25 17:04:41 -0800 David Schleef <ds@schleef.org>
1284 * ext/ogg/gstoggstream.c:
1285 * ext/ogg/gstoggstream.h:
1286 oggdemux: handle theora streams with 0 keyoffset
1288 2009-11-25 16:53:26 -0800 David Schleef <ds@schleef.org>
1290 * ext/ogg/gstoggdemux.c:
1291 oggdemux: Handle unknown streams
1293 2009-11-26 14:30:33 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1295 * ext/pango/gsttextoverlay.c:
1296 Revert "textoverlay: First draw outline text and then the real text"
1297 This reverts commit 60aa09d28c1f9fd29b56876d7ac6c0366d6cef4d.
1298 First drawing the real text and then the outline produces ugly
1299 text in lower resolutions. The outline line width needs to be somehow
1300 changed relative to the resolution. Fixes bug #602924.
1302 2009-11-26 10:30:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1304 * gst-libs/gst/audio/gstaudiofilter.c:
1305 audiofilter: Use G_DEFINE_ABSTRACT_TYPE_WITH_CODE
1306 ...and fix code style a bit.
1308 2009-11-26 10:31:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1310 * gst-libs/gst/audio/gstaudiofilter.h:
1311 audiofilter: Add _CAST variants of the cast macros
1313 2009-11-25 10:26:16 -0600 Wim Taymans <wim.taymans@collabora.co.uk>
1315 * gst-libs/gst/audio/gstbaseaudiosink.c:
1316 audiosink: add adjustement when slaving
1317 Our calibration against the pipeline clock is done with the adjusted
1318 ringbuffer time, so take the adjustement into account. Fixes some audio dropouts
1319 when reusing audio sinks after switching clocks and slaving methods in a
1322 2009-11-25 16:17:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1324 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
1325 ffmpegcolorspace: Prefer transforming alpha formats to alpha formats and the other way around
1326 Fixes bug #602834 and #350748.
1328 2009-11-25 00:46:55 -0800 David Schleef <ds@schleef.org>
1330 * ext/ogg/gstoggdemux.c:
1331 oggdemux: Reset last_granule during seeking
1332 Fix case where we would reconstruct the wrong granulepos for
1333 outgoing streams immediately after a seek.
1335 2009-11-24 22:08:09 -0800 David Schleef <ds@schleef.org>
1337 * ext/ogg/gstoggdemux.c:
1338 * ext/ogg/gstoggdemux.h:
1339 * ext/ogg/gstoggstream.c:
1340 * ext/ogg/gstoggstream.h:
1341 oggdemux: Fix timestamp generation for theora
1342 Timestamp generation was broken by the last commit for formats
1343 with a non-zero granule shift. Also keep track of the last keyframe
1344 so that we can regenerate granulepos for theora.
1346 2009-11-24 21:22:03 -0800 David Schleef <ds@schleef.org>
1348 * ext/ogg/gstoggdemux.c:
1349 * ext/ogg/gstoggstream.c:
1350 * ext/ogg/gstoggstream.h:
1351 * ext/ogg/vorbis_parse.c:
1352 oggdemux: Fix vorbis parsing
1353 Add a granule to granulepos conversion function. Fix the duration
1354 function for vorbis. Handle timestamps on header packets differently
1355 and be more careful about calculating OFFSET and OFFSET_END. After
1356 this change, timestamps for vorbis don't exactly match up with the
1357 timestamps that vorbisparse outputs, but it's unclear if vorbisparse
1358 is actually correct and it would add a lot more code to make oggdemux
1359 match vorbisparse. Fixes #602790.
1361 2009-11-19 19:28:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1363 * gst/playback/gstplaybin2.c:
1364 playbin2: Transform QoS events to be meaningful for upstream elements
1365 This is necessary because the sinks don't notice the group switches
1366 and the decoders/demuxers have a different running time than the
1370 2009-11-21 22:05:34 +0100 David Schleef <ds@schleef.org>
1372 * ext/ogg/gstoggdemux.c:
1373 ogg: Fix generation of timestamps and durations
1374 After changing some internal functions, I forgot to update
1375 the code that puts the values on the buffers.
1377 2009-08-29 10:51:48 -0700 David Schleef <ds@schleef.org>
1379 * ext/ogg/Makefile.am:
1380 * ext/ogg/dirac_parse.c:
1381 * ext/ogg/dirac_parse.h:
1382 * ext/ogg/gstoggdemux.c:
1383 * ext/ogg/gstoggdemux.h:
1384 * ext/ogg/gstoggparse.c:
1385 * ext/ogg/gstoggstream.c:
1386 * ext/ogg/gstoggstream.h:
1387 * ext/ogg/vorbis_parse.c:
1388 ogg: Add ogg stream parsing
1389 Adds code that parses headers of various formats encapsulated in
1390 Ogg in order to calculate timestamps and durations of each buffer.
1391 Removes the creation of helper decoder elements to do this calculation
1392 via conversion queries.
1393 Fixes: #344013, #568014.
1395 2009-09-04 00:11:38 -0700 David Schleef <ds@schleef.org>
1397 * ext/ogg/gstoggmux.c:
1398 oggmux: don't overwrite object properties
1400 2009-11-21 17:54:49 +0200 Stefan Kost <ensonic@users.sf.net>
1402 * ext/theora/theoradec.c:
1403 debug: also cast packet.packetno to gint64 in debug log
1404 We do this already for granulepos to handle ogg_int64_t mismatches.
1406 2009-11-21 17:47:26 +0200 Stefan Kost <ensonic@users.sf.net>
1408 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1409 debug: fix format string that was missing a var
1411 2009-10-10 00:32:04 +0300 Stefan Kost <ensonic@users.sf.net>
1413 * gst/adder/gstadder.c:
1414 * tests/check/elements/adder.c:
1415 adder: make events succeed, if they succed on atleast one pad
1417 2009-11-19 14:51:33 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1419 * gst/playback/gstdecodebin2.c:
1420 decodebin2: error when all streams have no buffers
1421 In some cases (all buffers dropped by a parser) a decodebin2
1422 chain might receive an EOS before it gets enough data to
1423 expose a decoded pad. In the case that no streams can expose
1424 a pad we should error out instead of hang.
1427 2009-11-19 12:23:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1429 * gst/playback/gstplaybin2.c:
1430 playbin2: Fix stupid bug introduced in last commit
1432 2009-11-19 12:10:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1434 * gst/playback/gstplaybin2.c:
1435 playbin2: Aggregate the stream-changed message by looking at the seqnum
1436 Just counting how many messages were sent and how many were received
1437 is not good enough because they might've been duplicated (e.g. by the
1438 visualization audio tee). Comparing the sequence numbers should give
1439 better results in that case.
1441 2009-11-19 10:05:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1443 * gst/playback/gstplaybin2.c:
1444 playbin2: Ignore async state changes of the uridecodebins
1445 Otherwise the async state change from READY->PAUSED of the
1446 uridecodebins will take playbin2 from PLAYING->PAUSED again
1447 during gapless group switches.
1450 2009-11-19 10:30:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1453 Automatic update of common submodule
1454 From 0702fe1 to 53a2485
1456 2009-11-18 14:50:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1458 * gst/playback/gstdecodebin2.c:
1459 decodebin2: set to buffer less on no-more-pads
1460 When a decodebin2 receives no-more-pads of a group it
1461 can set that group's multiqueue buffering thresholds to
1462 'playing' buffering method, avoiding that it buffers
1463 too long and cause problems when using with queue2.
1464 See the associated bug for details.
1467 2009-11-18 17:09:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1469 * gst-libs/gst/audio/gstbaseaudiosink.c:
1470 baseaudiosink: fix initial calibration
1471 When we are calibrating the internal clock against the external clock take into
1472 account the time offset applied to our internal clock because we will subtract
1473 that in the render_function again.
1475 2009-11-18 09:22:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1477 * gst/playback/gstplaybin2.c:
1478 playbin2: Don't handle DURATION queries during group switches
1479 During a group switch return the cached duration of the old group
1480 because the old group still didn't finish playback. If we have no
1481 cached duration return FALSE.
1484 2009-11-15 19:36:21 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1486 * gst/playback/gstplaybin2.c:
1487 playbin2: Post a stream-changed message after activating a group
1488 This is useful to detect when playbin2 has really switched to the next
1489 group after about-to-finish for example.
1492 2009-11-18 12:27:19 +0000 Jan Schmidt <thaytan@noraisin.net>
1494 * win32/common/libgstvideo.def:
1495 win32: Add new still-frame API to the defs
1496 Add gst_video_event_new_still_frame() and
1497 gst_video_event_parse_still_frame() functions to the win32 defs files
1499 2009-11-18 12:37:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1501 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1502 baseaudiosrc: fix 'uninitialized' compiler warning
1504 2009-11-18 10:14:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1507 configure: bump core requirement to 0.10.25.1
1508 We depend on new API that's only in git so far.
1510 2009-11-15 17:34:37 +0000 Jan Schmidt <thaytan@noraisin.net>
1512 * gst-libs/gst/video/video.c:
1513 * gst-libs/gst/video/video.h:
1514 * tests/check/libs/video.c:
1515 video: Add functions to create/parse still frame events.
1516 Add a new video event to mark the start or end of a still-frame
1517 sequence, and a parser function to identify and extract info from
1519 API: gst_video_event_new_still_frame()
1520 API: gst_video_event_parse_still_frame()
1523 2009-11-17 16:39:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1525 * gst/playback/gstplaysink.c:
1526 playsink: make sure we always go to PAUSED async
1527 Set the need_async_start flag before going to PAUSED so that we always post the
1528 ASYNC_START message, even after reusing playsink.
1530 2009-11-17 16:37:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1532 * gst/playback/gstplaysink.c:
1533 playsink: make sure we remain a sink
1534 When we remove our elements, we could lose our sink flag. Make sure we remain a
1535 sink by setting the flag again after removing elements.
1537 2009-11-16 22:47:54 +0200 Stefan Kost <ensonic@users.sf.net>
1539 * gst/audioconvert/gstaudioconvert.c:
1540 audioconvert: remove unused array
1542 2009-11-16 09:57:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1544 * gst/subparse/gstsubparse.c:
1545 subparse: Use new double->fraction transformation function from core
1547 2009-11-14 14:05:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1549 * gst/playback/gstplaybin2.c:
1550 playbin2: Make subtitle error handling more robust and ignore late errors too
1551 Make sure, to only "simulate" subtitle no-more-pads if it was still
1552 pending and also handle errors in the subtitle pipeline as warnings
1553 after the subtitles prerolled.
1554 Don't set the suburidecodebin to READY after errors, handle_message
1555 will usually be called from the streaming thread and doing that
1556 from there is obviously not a good idea.
1558 2009-11-14 13:21:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1560 * gst/playback/gstsubtitleoverlay.c:
1561 * gst/playback/gstsubtitleoverlay.h:
1562 subtitleoverlay: Handle errors from subtitle elements as warning and go into passthrough mode
1564 2009-11-13 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1566 * gst/playback/gstplaybin2.c:
1567 playbin2: Don't leak the GError and debug string when parsing error messages
1569 2009-11-13 11:16:44 +0100 Sreerenj B <bsreerenj@gmail.com>
1571 * gst-libs/gst/rtsp/gstrtspconnection.c:
1572 rtsp: avoid crashing on SIGPIPE
1573 Use send() instead of write() so that we can pass the MSG_NOSIGNAL flags to
1574 avoid crashing with SIGPIPE when the remote end is not listening to us anymore.
1577 2009-11-11 17:35:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1579 * gst/playback/gstplaybin2.c:
1580 playbin2: Improve subtitle passthrough in uridecodebin
1581 Now the caps property isn't set anymore for the subtitle caps
1582 but instead in the autoplug-continue signal it is detected
1583 if the caps belong to a supported subtitle stream.
1584 This makes automatic use of newly installed plugins.
1586 2009-11-11 17:08:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1588 * gst/playback/gstsubtitleoverlay.c:
1589 subtitleoverlay: Only recreate factory caps if necessary and cache them
1591 2009-11-10 18:27:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1593 * gst/playback/gstsubtitleoverlay.c:
1594 * gst/playback/gstsubtitleoverlay.h:
1595 subtitleoverlay: Only update the factory list when the registry has changed
1596 Also don't free the list every time we go to NULL.
1598 2009-11-08 15:04:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1600 * gst/playback/gstsubtitleoverlay.c:
1601 subtitleoverlay: Use gst_pad_get_caps_reffed()
1603 2009-11-07 21:38:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1605 * gst/playback/gstplaybin2.c:
1606 * gst/playback/gstplaysink.c:
1607 playbin2/playsink: Use new "silent" property instead of unlinking
1608 This makes sure that subtitleoverlay still gets segment updates and
1609 everything to pass on downstream. Without this segment problems happen.
1611 2009-11-07 21:10:27 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1613 * gst/playback/gstsubtitleoverlay.c:
1614 * gst/playback/gstsubtitleoverlay.h:
1615 subtitleoverlay: Update segments after pushing the events downstream
1616 This makes sure that we don't apply segments twice downstream. Also
1617 always send our newsegment events downstream.
1619 2009-11-07 21:09:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1621 * gst/playback/gstsubtitleoverlay.c:
1622 * gst/playback/gstsubtitleoverlay.h:
1623 subtitleoverlay: Add silent property to disable subtitles
1624 This tries to disable subtitles in the overlay or renderer
1625 and if that's not possible it goes into passthrough mode.
1627 2009-11-07 11:46:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1629 * gst/playback/gstsubtitleoverlay.c:
1630 * gst/playback/gstsubtitleoverlay.h:
1631 subtitleoverlay: Set the video framerate on parsers if possible
1634 2009-11-07 11:31:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1636 * gst/subparse/gstsubparse.c:
1637 * gst/subparse/gstsubparse.h:
1638 subparse: Make fps a GstFraction typed property and use it properly
1640 2009-11-07 11:08:19 +0100 Iago Toral <itoral@igalia.com>
1642 * gst/subparse/gstsubparse.c:
1643 * gst/subparse/gstsubparse.h:
1644 subparse: Add property for the video framerate
1646 2009-11-06 12:51:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1648 * gst/playback/gstplaybin2.c:
1649 playbin2: Handle external subtitles better
1650 First of all, make sure that suburidecodebin never
1651 errors out because of not-linked in case external subtitles
1652 are used but then subtitles are disabled.
1653 And then make sure that external subtitles always start from
1654 the correct position and are not racing until EOS if they
1655 get unselected and selected again.
1657 2009-11-04 17:29:07 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1659 * gst/playback/gstplaybin2.c:
1660 playbin2: Flush the subtitles before switching to a new subtitle stream
1661 This makes sure that all currently shown subtitles disappear
1662 and new ones can be shown as soon as possible.
1664 2009-11-03 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1666 * gst/playback/gstplaybin2.c:
1667 playbin2: Set subtitle caps as raw caps for the uridecodebins
1668 This will make sure that no subparse is ever plugged and subtitleoverlay,
1669 that subpicture streams are handled the same was as subtitles and that
1670 subtitle renderers are used if available.
1671 Fixes bugs #595123, #570753, #591662, #591706.
1673 2009-11-03 12:33:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1675 * gst/playback/gstplaybin2.c:
1676 * gst/playback/gstplaysink.c:
1677 * gst/playback/gstplaysink.h:
1678 playbin2/playsink: Remove everything related to subpicture streams
1679 These will soon be handled the same way as subtitle streams.
1681 2009-11-02 15:50:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1683 * gst/playback/gstplaysink.c:
1684 playsink: Add a queue before subtitleoverlay
1685 This will improve playback, and the same thing is done
1686 for subpicture streams too.
1688 2009-11-02 15:05:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1690 * gst/playback/gstplaysink.c:
1691 playsink: Use subtitleoverlay for subtitles
1693 2009-11-02 07:43:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1695 * docs/plugins/Makefile.am:
1696 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
1697 * docs/plugins/gst-plugins-base-plugins-sections.txt:
1698 * docs/plugins/gst-plugins-base-plugins.args:
1699 * docs/plugins/gst-plugins-base-plugins.hierarchy:
1700 * docs/plugins/gst-plugins-base-plugins.interfaces:
1701 * docs/plugins/gst-plugins-base-plugins.prerequisites:
1702 * docs/plugins/inspect/plugin-adder.xml:
1703 * docs/plugins/inspect/plugin-alsa.xml:
1704 * docs/plugins/inspect/plugin-app.xml:
1705 * docs/plugins/inspect/plugin-audioconvert.xml:
1706 * docs/plugins/inspect/plugin-audiorate.xml:
1707 * docs/plugins/inspect/plugin-audioresample.xml:
1708 * docs/plugins/inspect/plugin-audiotestsrc.xml:
1709 * docs/plugins/inspect/plugin-cdparanoia.xml:
1710 * docs/plugins/inspect/plugin-decodebin.xml:
1711 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
1712 * docs/plugins/inspect/plugin-gdp.xml:
1713 * docs/plugins/inspect/plugin-gio.xml:
1714 * docs/plugins/inspect/plugin-gnomevfs.xml:
1715 * docs/plugins/inspect/plugin-libvisual.xml:
1716 * docs/plugins/inspect/plugin-ogg.xml:
1717 * docs/plugins/inspect/plugin-pango.xml:
1718 * docs/plugins/inspect/plugin-playback.xml:
1719 * docs/plugins/inspect/plugin-subparse.xml:
1720 * docs/plugins/inspect/plugin-tcp.xml:
1721 * docs/plugins/inspect/plugin-theora.xml:
1722 * docs/plugins/inspect/plugin-typefindfunctions.xml:
1723 * docs/plugins/inspect/plugin-uridecodebin.xml:
1724 * docs/plugins/inspect/plugin-video4linux.xml:
1725 * docs/plugins/inspect/plugin-videorate.xml:
1726 * docs/plugins/inspect/plugin-videoscale.xml:
1727 * docs/plugins/inspect/plugin-videotestsrc.xml:
1728 * docs/plugins/inspect/plugin-volume.xml:
1729 * docs/plugins/inspect/plugin-vorbis.xml:
1730 * docs/plugins/inspect/plugin-ximagesink.xml:
1731 * docs/plugins/inspect/plugin-xvimagesink.xml:
1732 subtitleoverlay: Add to the docs
1734 2009-10-13 16:48:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1736 * gst/playback/Makefile.am:
1737 * gst/playback/gstplayback.c:
1738 * gst/playback/gstsubtitleoverlay.c:
1739 * gst/playback/gstsubtitleoverlay.h:
1740 subtitleoverlay: Add new element for generic subtitle overlaying
1741 This autopluggs the required elements for parsing and rendering
1742 different subtitle formats on a video stream.
1745 2009-11-11 19:32:01 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
1747 * ext/theora/theoradec.c:
1748 theoradec: Keep timestamp from incoming buffer if it is valid
1751 2009-11-11 14:00:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1753 * gst/playback/gstdecodebin2.c:
1754 * gst/playback/gstplaybin2.c:
1755 * gst/playback/gsturidecodebin.c:
1756 playback: Update factories list on every access if the registry has changed
1757 This makes application's simpler because the element doesn't need to
1758 go to NULL first to make use of newly installed plugins.
1761 2009-11-10 18:13:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1763 * gst/playback/gstdecodebin2.c:
1764 * gst/playback/gstplaybin2.c:
1765 * gst/playback/gsturidecodebin.c:
1766 playback: When going from NULL->READY check if the registry has new features
1767 This makes it possible to use newly installed plugins after going back
1768 to NULL instead of requiring a new instance.
1771 2009-11-10 13:55:26 +0000 Jan Schmidt <thaytan@noraisin.net>
1773 * gst-libs/gst/app/gstappsrc.c:
1774 appsrc: Clear the EOS state on a seek.
1775 Allow seeking back into the stream after it hits EOS.
1777 2009-11-10 12:21:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1779 * gst/audioresample/README:
1780 * gst/audioresample/arch.h:
1781 * gst/audioresample/fixed_arm4.h:
1782 * gst/audioresample/fixed_arm5e.h:
1783 * gst/audioresample/fixed_bfin.h:
1784 * gst/audioresample/fixed_debug.h:
1785 * gst/audioresample/resample.c:
1786 * gst/audioresample/resample_sse.h:
1787 * gst/audioresample/speex_resampler.h:
1788 audioresample: Update speex resampler to latest GIT
1790 2009-11-10 00:48:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1792 * gst/playback/gstplaysink.c:
1793 playsink: assign chain->mute before using it
1794 Fixes GObject warnings when starting totem.
1796 2009-10-28 22:10:33 -0700 David Schleef <ds@schleef.org>
1798 * ext/theora/theoradec.c:
1799 theora: Fix alignment of frames when converting
1800 Fix logic inversion in calculating the offset in the theora
1801 frame when copying to a GStreamer frame.
1803 2009-11-09 19:58:20 +0100 Edward Hervey <bilboed@bilboed.com>
1805 * gst/playback/gstfactorylists.c:
1806 playback: Fix the order in strcmp that I broke in previous commit.
1808 2009-11-09 19:16:21 +0100 Edward Hervey <bilboed@bilboed.com>
1810 * gst/typefind/gsttypefindfunctions.c:
1811 typefind: Reduce number of calls to gst_type_find_peek.
1812 Shaves off a couple percents off typefinding
1814 2009-11-09 17:49:51 +0100 Edward Hervey <bilboed@bilboed.com>
1816 * gst/playback/gstfactorylists.c:
1817 playback: Avoid expensive API calls in tight loop.
1818 We know we're dealing with GstPluginFeature.
1820 2009-11-09 18:11:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1822 * tests/check/libs/cddabasesrc.c:
1823 cddabasesrc: Add unit test for property settings
1824 Also includes a regression test for bug #601104.
1826 2009-11-09 18:04:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1828 * gst-libs/gst/cdda/gstcddabasesrc.c:
1829 cddabasesrc: Never return a negative track number in get_uri()
1831 2009-11-09 18:03:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1833 * gst-libs/gst/cdda/gstcddabasesrc.c:
1834 cddabasesrc: Don't set the track to 1 every time a device is set
1837 2009-11-08 11:27:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1839 * gst/playback/gstinputselector.c:
1840 inputselector: Remove useless variables and fix a uninitialized variable compiler warnings
1842 2009-11-06 17:01:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1844 * gst/playback/gstdecodebin2.c:
1845 decodebin2: Add property to disable/enable posting of stream-topology messages
1846 Most people don't need this messages and generating them is quite
1849 2009-11-06 15:12:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1851 * gst/playback/gstdecodebin2.c:
1852 decodebin2: Protect subtitle elements and subtitle encoding by a new mutex
1853 Using the object lock here can and will lead to deadlocks because
1854 of deep-notifies of property changes: the deep-notify handler will
1855 get the parent of objects, which will take the object lock again.
1858 2009-11-06 13:13:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1860 * gst/playback/gstinputselector.c:
1861 inputselector: Make sure that running_time->timestamp calculation never becomes negative
1863 2009-11-06 13:25:05 +0200 Mart Raudsepp <leio@gentoo.org>
1865 * tests/examples/seek/scrubby.c:
1866 * tests/examples/seek/seek.c:
1867 examples: Correct casting of g_signal* funcs first arguments
1868 This completes the deprecated GTK API fix in commits 81a0a986 and
1869 79adfa54 - unlike gtk_signal_connect and co, g_signal_connect and
1870 co take a gpointer, not a GtkObject.
1872 2009-11-06 12:25:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1874 * gst/playback/gsturidecodebin.c:
1875 uridecodebin: Improve all-raw-caps detection for pads
1877 2009-11-06 12:19:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1879 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1880 basesrc: fix startup position in the ringbuffer
1881 When we start and we need to produce the first sample, go to the next sample
1882 that will be written into the ringbuffer instead of trying to go to sample 0.
1883 We relied on rather small ringbuffer sizes to correctly go to the current
1884 sample, which breaks whith large buffers.
1887 2009-11-06 11:26:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1889 * gst/playback/gstinputselector.c:
1890 inputselector: Use the start time (i.e. timestamp) as the last stop
1891 Using the end time makes it impossible to replace buffers, which is
1892 a big problem for subtitles that could have very long durations.
1894 2009-11-06 12:08:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1896 * ext/pango/gsttextoverlay.c:
1897 textoverlay: Synchronize video/text based on the running time
1898 Instead of simply using the buffer timestamps.
1900 2009-11-06 09:30:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1902 * ext/pango/gsttextoverlay.c:
1903 textoverlay: Clip text buffers to the text segment and reset segments properly
1905 2009-11-06 09:01:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1907 * ext/pango/gsttextoverlay.c:
1908 * ext/pango/gsttextoverlay.h:
1909 textoverlay: Put the video segment into the instance struct instead of allocating it separately
1911 2009-11-06 09:05:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1913 * ext/pango/gsttextoverlay.c:
1914 textoverlay: Check if text timestamp/duration is valid before clipping
1916 2009-11-05 23:33:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1918 * ext/theora/theoradec.c:
1919 theoradec: printf format fix
1921 2009-11-05 15:42:09 +0100 Olivier Crête <olivier.crete@collabora.co.uk>
1923 * gst/gdp/gstgdpdepay.c:
1924 gdpdepay: Clear adapter on flush and state change
1927 2009-11-05 13:12:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1929 * gst/playback/gstinputselector.c:
1930 inputselector: use _get_caps_reffed()
1932 2009-11-05 13:00:27 +0200 Stefan Kost <ensonic@users.sf.net>
1934 * gst/playback/gstdecodebin2.c:
1935 * gst/playback/gstplaybin2.c:
1936 * gst/playback/gsturidecodebin.c:
1937 pad: rename new api from _refed to _reffed.
1938 Due to popular demand rename the new api as we still can.
1940 2009-11-04 18:57:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1942 * gst/playback/gstplaybin2.c:
1943 * gst/playback/gsturidecodebin.c:
1944 playbin2: avoid copying caps
1945 Use get_caps_refed() when we can.
1947 2009-11-04 18:31:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1949 * gst/playback/gstdecodebin2.c:
1950 decodebin2: use new getcaps function to avoid copies
1951 Use the gst_pad_get_caps_refed() to avoid some caps copy functions.
1953 2009-11-04 17:50:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1955 * gst/playback/gsturidecodebin.c:
1956 uridecodebin: use faster element_link_pads
1957 Use the faster gst_element_link_pads because we know for sure the sinkpad name
1958 and we don't need to have the function search for a suitable pad anymore.
1960 2009-11-04 16:16:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1962 * gst-libs/gst/audio/gstbaseaudiosink.c:
1963 baseaudiosink: make drift tolerance configurable
1964 Add drift-tolerance property (defaulting to 20ms) to handle resync after clock
1965 drift or timestamp drift instead of relying on the latency-time value for clock
1966 drift and 500ms for timestamp drift.
1967 Remove warning about discont timestamp and simply resync. The warning is in some
1968 cases not correct and is triggered more frequently now that we lower the
1971 2009-11-04 10:52:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1973 * gst/playback/gstplaybin2.c:
1974 playbin2: Return NOT_LINKED for unselected text pads from a demuxer
1975 We want to return NOT_LINKED for unselected pads but only for pads
1976 from the normal uridecodebin. This makes sure that subtitle streams
1977 are not raced past audio/video from decodebin2's multiqueue.
1978 For pads from suburidecodebin OK should always be returned, otherwise
1979 it will most likely stop with an error.
1981 2009-11-04 08:20:59 +0100 Stefan Kost <ensonic@users.sf.net>
1983 * gst/playback/gstinputselector.c:
1984 inputselector: also add inline to the proto to fix the build
1985 Merged from gst-plugins-bad, e1e9be6dbe1bd0df0543f2a72dcf9cc6d644dd78.
1987 2009-11-03 12:01:16 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1989 * gst/playback/gsturidecodebin.c:
1990 uridecodebin: Initialize caps property with the default raw caps
1992 2009-11-03 11:48:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1994 * gst/playback/Makefile.am:
1995 * gst/playback/gstdecodebin2.c:
1996 * gst/playback/gstrawcaps.h:
1997 decodebin2: Use static caps for the default raw caps and put them into a separate header
1998 This way we can use the same default raw caps everywhere.
2000 2009-11-03 08:26:37 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2002 * ext/pango/gsttextoverlay.c:
2003 textoverlay: First draw outline text and then the real text
2004 Improves the output a bit because no parts of the outline are
2007 2009-10-31 14:02:40 +0100 Josep Torra Valles <n770galaxy@gmail.com>
2009 * gst/playback/gstplaybin.c:
2010 playbin: Make sure to keep a reference on the volume element
2011 Fixes null pointer dereferences under certain circumstances.
2014 2009-10-31 09:47:54 +0100 Edward Hervey <bilboed@bilboed.com>
2017 po: queue2 has moved to core
2019 2009-10-30 09:24:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2021 * gst/playback/gstplaysink.c:
2022 playsink: Reset {mute,volume}-changed flags after setting the volume
2023 These flags are there to make sure that the volume is set, if there
2024 is no volume element yet.
2026 2009-10-30 09:24:03 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2028 * gst/playback/gstplaysink.c:
2029 playsink: If notify::{volume,mute} is triggered by the volume element, update our internal state
2031 2009-10-29 14:30:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2033 * gst/playback/gstplaysink.c:
2034 playsink: Proxy notify::volume and notify::mute from the volume/mute elements (or sinks)
2037 2009-10-29 14:19:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2039 * gst/playback/gstplaybin2.c:
2040 playbin2: Proxy notify::volume and notify::mute from the playsink to playbin2
2042 2009-10-29 11:37:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2044 * docs/plugins/inspect/plugin-queue2.xml:
2045 queue2: Remove inspect file
2047 2009-10-29 11:29:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2049 * gst/playback/Makefile.am:
2050 * gst/playback/gstqueue2.c:
2051 queue2: Remove from gst-plugins-base
2052 This is now in coreplugins.
2054 2009-10-28 11:29:36 +0200 Stefan Kost <ensonic@users.sf.net>
2056 * docs/libs/gst-plugins-base-libs-docs.sgml:
2057 docs: include more indexes
2059 2009-10-28 11:13:20 +0200 Stefan Kost <ensonic@users.sf.net>
2061 * docs/libs/gst-plugins-base-libs-docs.sgml:
2062 docs: turn entities into xi:includes
2063 This is faster to process and easier to maintain. Its also less 80s.
2065 2009-10-28 10:17:43 +0200 Stefan Kost <ensonic@users.sf.net>
2067 * gst-libs/gst/rtp/gstrtpbuffer.c:
2068 rtp: dump packets which we reject
2070 2009-10-28 01:01:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2072 * tests/check/pipelines/.gitignore:
2073 .gitignore: ignore basetime unit test binary
2075 2009-10-28 00:59:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2077 * ext/alsa/gstalsasink.c:
2078 * ext/alsa/gstalsasrc.c:
2079 * gst-libs/gst/audio/gstaudiosink.c:
2080 * gst-libs/gst/audio/gstaudiosrc.c:
2081 * gst-libs/gst/audio/gstbaseaudiosink.c:
2082 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2083 * gst-libs/gst/audio/gstringbuffer.c:
2084 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
2085 * gst/adder/gstadder.c:
2086 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
2087 * gst/gdp/gstgdpdepay.c:
2088 * gst/gdp/gstgdppay.c:
2089 * gst/playback/gstdecodebin.c:
2090 * gst/playback/gstdecodebin2.c:
2091 * gst/playback/gstinputselector.c:
2092 * gst/playback/gstplaybasebin.c:
2093 * gst/playback/gstplaybin.c:
2094 * gst/playback/gstplaybin2.c:
2095 * gst/playback/gstplaysink.c:
2096 * gst/playback/gstqueue2.c:
2097 * gst/playback/gststreaminfo.c:
2098 * gst/playback/gststreamselector.c:
2099 * gst/subparse/gstssaparse.c:
2100 Remove GST_DEBUG_FUNCPTR where they're pointless
2101 There's not much point in using GST_DEBUG_FUNCPTR with GObject
2102 virtual functions such as get_property, set_propery, finalize and
2103 dispose, since they'll never be used by anyone anyway. Saves a
2104 few bytes and possibly a sixteenth of a polar bear.
2106 2009-10-27 15:23:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2108 * gst/playback/gstqueue2.c:
2109 queue2: add custom acceptcaps function
2111 2009-10-27 15:22:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2113 * gst/playback/gstdecodebin2.c:
2114 decodebin2: implement low/high watermark property
2116 2009-10-23 14:56:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2118 * tests/examples/seek/seek.c:
2119 seek: add checkbox to enable buffering
2121 2009-10-23 14:54:47 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2123 * gst/playback/gsturidecodebin.c:
2124 uridecodebin: don't use 2 buffering elements
2125 Only use the multiqueue buffering when we don't have a stream (and thus are
2126 using queue2 to do the buffering already).
2128 2009-10-23 14:34:42 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2130 * gst/playback/gstplay-enum.c:
2131 * gst/playback/gstplay-enum.h:
2132 * gst/playback/gstplaybin2.c:
2133 playbin2: add flag to enable decodebin buffering
2134 Add a flag that enables buffering in decodebin.
2136 2009-10-23 14:32:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2138 * gst/playback/gstdecodebin2.c:
2139 decodebin2: buffering is implemented now
2141 2009-10-23 14:30:52 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2143 * gst/playback/gsturidecodebin.c:
2144 uridecodebin: buffering is implemented now
2146 2009-10-23 14:09:17 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2148 * gst/playback/gstdecodebin2.c:
2149 decodebin2: configure use-buffering on multiqueue
2151 2009-10-23 13:58:25 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2153 * gst/playback/gsturidecodebin.c:
2154 uridecodebin: use 0 for max buffer size
2156 2009-10-23 13:53:21 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2158 * gst/playback/gsturidecodebin.c:
2159 uridecodebin: set some reasonable defaults
2161 2009-10-23 13:44:12 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2163 * gst/playback/gsturidecodebin.c:
2164 uridecodebin: set buffering properties on decodebin2
2165 Propagate the buffering properties on decodebin2 but only if we are not already
2166 doing download buffering.
2168 2009-10-23 11:52:09 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2170 * gst/playback/gsturidecodebin.c:
2171 uridecodebin: add use-buffering property
2172 Add a use-buffering property that will perform buffering on the parsed or
2175 2009-10-23 11:31:47 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2177 * gst/playback/gstdecodebin2.c:
2178 decodebin2: refactor queue size configuration.
2179 Refactor the queue size configuration into a new method.
2180 Use the same queue values for buffering as for preroll.
2182 2009-10-23 11:08:50 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2184 * gst/playback/gstdecodebin2.c:
2185 decodebin2: move error path down
2187 2009-10-23 11:02:40 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2189 * gst/playback/gstdecodebin2.c:
2190 decodebin2: implement max queue size properties
2192 2009-10-23 10:42:23 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2194 * gst/playback/gstdecodebin2.c:
2195 decodebin2: add properties for buffering
2196 Add properties that can be used to configure the multiqueue buffers and
2199 2009-10-24 13:19:08 +0200 Edward Hervey <bilboed@bilboed.com>
2201 * tests/examples/app/Makefile.am:
2202 * tests/examples/seek/Makefile.am:
2203 * tests/examples/v4l/Makefile.am:
2204 examples: fix linking order.
2205 the uninstalled wrapper would create a LD_LIBRARY_PATH with system-wide
2206 path before the local ones... resulting in the example applications picking
2207 up the system-wide libraries and not the (potentially modified) uninstalled
2210 2009-10-24 13:08:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2212 * gst/playback/gstplaybin2.c:
2213 playbin2: Don't destroy the suburidecodebin on errors
2214 It can still be reused
2216 2009-10-24 13:07:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2218 * gst/playback/gstplaybin2.c:
2219 playbin2: If setting the state of the suburidecodebin fails just warn, don't error out
2221 2009-10-24 12:12:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2223 * gst/playback/gstplaybin2.c:
2224 playbin2: Don't set uridecodebin states to NULL before reusing them
2225 This makes sure that the internal decodebin2 and everything else can
2226 be reused without reinstantiation.
2228 2009-10-18 17:28:22 +0200 Edward Hervey <bilboed@bilboed.com>
2230 * gst/playback/gsturidecodebin.c:
2231 uridecodebin: Store unused decodebin2 instances for further usage.
2232 This allows faster re-use of uridecodebin.
2233 https://bugzilla.gnome.org/show_bug.cgi?id=599471
2235 2009-10-23 17:49:15 -0700 David Schleef <ds@schleef.org>
2237 * ext/theora/gsttheoraparse.h:
2238 * ext/theora/theoraparse.c:
2239 theora: Convert theoraparse to libtheora 1.0 API
2241 2009-10-21 12:38:59 +0300 Olivier Crête <olivier.crete@collabora.co.uk>
2243 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
2244 rtpaudiopayload: Only sent exact multiple of the frame size
2245 Also align the maximum size with the frame size, not only the minimum
2247 2009-10-22 09:12:03 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
2249 * gst/audiorate/gstaudiorate.c:
2250 audiorate: move debug calculation into debug macro
2251 Remove in_duration and move its calculation to
2252 GST_LOG_OBJECT macro. This way it will only be calculated
2253 if we have debug enabled.
2255 2009-10-22 09:06:02 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
2257 * gst/audiorate/gstaudiorate.c:
2258 audiorate: Removing unused variable
2259 The in_stop variable was never read. Removing it.
2261 2009-10-22 08:40:01 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
2263 * gst/audiorate/gstaudiorate.c:
2264 audiorate: be more accurate on offset math
2265 Replace gst_util_uint64_scale_int for its rounding version
2266 to improve accuracy and avoid inserting samples where
2270 2009-10-22 10:17:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2272 * ext/pango/gsttextoverlay.c:
2273 textoverlay: Optimize a bit more
2274 ...and add a FIXME for bug #598695 and explain
2275 what we should do once Pango supports user fonts.
2277 2009-10-22 10:02:11 +0200 Iago Toral <itoral@igalia.com>
2279 * gst/subparse/gstsubparse.c:
2280 * gst/subparse/gstsubparse.h:
2281 * tests/check/elements/subparse.c:
2282 subparse: Add support for DKS subtitle format
2285 2009-10-22 09:31:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2287 * ext/pango/gsttextoverlay.c:
2288 textoverlay: Do shading as first operation
2290 2009-10-22 09:08:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2292 * ext/pango/gsttextoverlay.c:
2293 textoverlay: Only use a single cairo surface for drawing
2294 ... and comment/optimize what is going on here a bit better.
2296 2009-10-21 16:24:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2298 * gst/playback/gstinputselector.c:
2299 inputselector: set output caps before pushing
2300 Set the output caps on the srcpad before pushing the buffer because else core
2301 will do a rather expensive check to see if we can actually accept those caps on
2304 2009-10-21 15:58:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2306 * gst/playback/gstinputselector.c:
2307 inputselector: install an acceptcaps function
2308 Install a custom acceptcaps function instead of using the default expensive
2309 check. We accept whatever downstream accepts so we pass along the acceptcaps
2310 call to the downstream peer.
2312 2009-10-21 20:35:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2314 * gst/typefind/gsttypefindfunctions.c:
2315 typefind: fix typo in previous mxf typefinder change
2317 2009-10-21 20:44:33 +0200 Edward Hervey <bilboed@bilboed.com>
2319 * gst/typefind/gsttypefindfunctions.c:
2320 typefind: speed up mxf_type_find over 300 times for worst case scenarios
2321 * memcmp is expensive and was being abused, reduce calling it by checking
2323 * iterating one byte at at time over 64 kbites introduces a certain overhead,
2324 therefore we now do it in chunks of 1024 bytes
2325 And I do mean over 300 times. The average instruction call per mxf_type_find
2326 was previously 785685 and it's now down to 2458 :)
2328 2009-10-20 17:13:39 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2330 * gst/playback/gstfactorylists.c:
2331 decodebin2: avoid type checks
2333 2009-10-20 09:00:28 +0200 Edward Hervey <bilboed@bilboed.com>
2335 * gst/playback/gstdecodebin2.c:
2336 gst/decodebin2: Ensure we get fixed caps for topology message
2337 There are some corner cases (like with dvdemux amongst others) where
2338 the caps won't be negotiated, but the pad has fixed caps.
2340 2009-10-20 08:52:36 +0200 Edward Hervey <bilboed@bilboed.com>
2342 * gst/playback/gstdecodebin2.c:
2343 gst/decodebin2: Don't expose chains if we're shutting down.
2344 This avoids adding flushing pads to ourself
2346 2009-10-17 21:16:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2349 * ext/pango/gsttextoverlay.c:
2350 pango: bump pango requirement to stable version and remove ifdefs
2351 Bump pango requirement from an ancient development version to an
2352 ancient stable version.
2354 2009-10-17 21:11:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2356 * gst-libs/gst/rtsp/.gitignore:
2357 .gitignore: update after files got renamed
2359 2009-10-16 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2361 * gst-libs/gst/rtp/gstbasertppayload.c:
2362 basertppayload: small comment fix
2364 2009-10-16 10:50:35 +0200 Peter Kjellerstedt <pkj@axis.com>
2366 * gst-libs/gst/rtp/gstbasertppayload.c:
2367 rtp: Correct timestamping of buffers when buffer_lists are used
2368 The timestamping of buffers when buffer_lists are used failed if
2369 a buffer did not have both a timestamp and an offset.
2371 2009-10-16 10:56:56 +0300 Stefan Kost <ensonic@users.sf.net>
2373 * gst-libs/gst/app/Makefile.am:
2374 * gst-libs/gst/audio/Makefile.am:
2375 * gst-libs/gst/interfaces/Makefile.am:
2376 * gst-libs/gst/pbutils/Makefile.am:
2377 * gst-libs/gst/rtsp/Makefile.am:
2378 * gst-libs/gst/rtsp/gstrtsp-marshal.list:
2379 * gst-libs/gst/rtsp/gstrtspextension.c:
2380 * gst-libs/gst/rtsp/rtsp-marshal.list:
2381 * gst-libs/gst/video/Makefile.am:
2382 * gst/playback/Makefile.am:
2383 * gst/tcp/Makefile.am:
2384 build: fix previous commit to fully accomodate the glib-gen.mak changes
2385 I also renamed glib_enum_prefix to glib_gen_prefix as we also use that for the
2386 marshallers. Also rename the rtsp-marshal.list to work with the unified prefix.
2388 2009-10-16 10:18:45 +0300 Stefan Kost <ensonic@users.sf.net>
2390 * gst-libs/gst/app/Makefile.am:
2391 * gst-libs/gst/audio/Makefile.am:
2392 * gst-libs/gst/interfaces/Makefile.am:
2393 * gst-libs/gst/pbutils/Makefile.am:
2394 * gst-libs/gst/rtsp/Makefile.am:
2395 * gst-libs/gst/video/Makefile.am:
2396 * gst/playback/Makefile.am:
2397 * gst/tcp/Makefile.am:
2398 build: use gst-glib-gen.mak to fix the glib build rules. Fixes #598114
2399 The build rules in glib-gen.mak were using pattern rules in a non save way.
2401 2009-10-16 10:14:36 +0300 Stefan Kost <ensonic@users.sf.net>
2404 Automatic update of common submodule
2405 From 85d1530 to 0702fe1
2407 2009-09-10 11:39:18 +0200 Benjamin Otte <otte@gnome.org>
2409 * ext/theora/theoradec.c:
2410 theora: Make theoradec use gstvideo for image conversion
2411 Vastly simplifies code.
2412 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2414 2009-09-10 09:36:31 +0200 Benjamin Otte <otte@gnome.org>
2416 * ext/theora/theoradec.c:
2417 theora: Don't always round to even width/height
2418 Previously, the code always rounded to even sizes. Now it only ensures
2419 that pic_x and pic_y are multiples of 2 if the output format requires
2421 Also inlcudes fixes to take pic_x/y into account properly when copying
2423 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2425 2009-09-10 00:00:44 +0200 Benjamin Otte <otte@gnome.org>
2428 theora: Don't check for theora.pc anymore
2429 THe new APIs from theoradec and theoraenc are used now.
2430 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2432 2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
2434 * ext/theora/gsttheoradec.h:
2435 * ext/theora/theoradec.c:
2436 theora: Convert theoradec to libtheora 1.0 API
2437 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2439 2009-09-09 23:44:36 +0200 Benjamin Otte <otte@gnome.org>
2441 * ext/theora/Makefile.am:
2442 * ext/theora/gsttheoraenc.h:
2443 * ext/theora/theoraenc.c:
2444 theora: Port encoder to new Theora API
2445 Includes ripping out the old buffer copy code to fill up to frame size.
2446 This is not necesary with the new encoder.
2447 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2449 2009-09-09 21:59:31 +0200 Benjamin Otte <otte@gnome.org>
2451 * ext/theora/gsttheoraenc.h:
2452 * ext/theora/theoraenc.c:
2453 theora: Disable sharpness property
2454 It's ignored by libtheora
2455 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2457 2009-09-09 21:57:08 +0200 Benjamin Otte <otte@gnome.org>
2459 * ext/theora/gsttheoraenc.h:
2460 * ext/theora/theoraenc.c:
2461 theora: Disable noise-sensitivity property
2462 It is ignored by libtheora
2463 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2465 2009-09-09 21:50:57 +0200 Benjamin Otte <otte@gnome.org>
2467 * ext/theora/gsttheoraenc.h:
2468 * ext/theora/theoraenc.c:
2469 theora: Disable keyframe-mindistance property
2470 It's ignored by the current Theora library
2471 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2473 2009-09-09 21:48:08 +0200 Benjamin Otte <otte@gnome.org>
2475 * ext/theora/gsttheoraenc.h:
2476 * ext/theora/theoraenc.c:
2477 theora: Disable keyframe_threshold property
2478 It's ignored by the current theora encoder
2479 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2481 2009-09-09 20:26:47 +0200 Benjamin Otte <otte@gnome.org>
2483 * ext/theora/gsttheoraenc.h:
2484 * ext/theora/theoraenc.c:
2485 theora: Get rid of "quick" property
2486 The proeprty is not used by libtheora at all
2487 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2489 2009-09-08 15:12:23 +0200 Benjamin Otte <otte@gnome.org>
2492 * ext/theora/theoraenc.c:
2493 theora: remove support for outdated granulepos hack
2494 This is in preparation to switching to switching to the new Theora API
2495 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2497 2009-09-08 13:23:04 +0200 Benjamin Otte <otte@gnome.org>
2499 * ext/theora/gsttheoraenc.h:
2500 * ext/theora/theoraenc.c:
2501 theora: Ignore border property
2502 Always make the video use black as padding color.
2503 The output will be identical to previous versions.
2504 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2506 2009-09-08 13:18:26 +0200 Benjamin Otte <otte@gnome.org>
2508 * ext/theora/gsttheoraenc.h:
2509 * ext/theora/theoraenc.c:
2510 theora: Ignore the center property, always set video to top left
2511 This is not a necessary property, the output will be identical no matter
2513 https://bugzilla.gnome.org/show_bug.cgi?id=594729
2515 2009-10-15 16:34:28 +0100 Jan Schmidt <thaytan@noraisin.net>
2518 po: Don't create backup .po files
2519 As well as preventing creation of useless backup files, it works
2520 around a bug in gettext 0.17 on OS/X
2522 2009-10-15 13:13:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2524 * gst/playback/gstdecodebin2.c:
2525 decodebin2: Post a element message on the bus with the stream topology
2528 2009-10-15 13:01:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2530 * gst/playback/gstdecodebin2.c:
2531 decodebin2: Store the "endcaps" of a chain
2532 This are the caps that either resulted in a deadend if
2533 no plugin for them could be found or raw caps.
2535 2009-10-15 11:38:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2537 * gst/playback/gstdecodebin2.c:
2538 decodebin2: Store for every chain, which pad resulted in its creation
2540 2009-10-15 10:28:39 +0100 Jan Schmidt <thaytan@noraisin.net>
2542 * tests/check/pipelines/basetime.c:
2543 check: Don't fail the basetime test when no audiosrc is available
2544 On OS/X the DEFAULT_AUDIOSRC is not going to be available, because
2545 it isn't in gst-plugins-base. Just defer the test, instead of
2548 2009-10-14 10:41:03 +0200 Edward Hervey <bilboed@bilboed.com>
2551 Automatic update of common submodule
2552 From a3e3ce4 to 85d1530
2554 2009-10-14 08:36:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2556 * gst/playback/gstplaybin2.c:
2557 playbin2: Use gst_object_has_ancestor() instead of our own implementation of it
2559 2009-10-13 19:14:41 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
2561 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2562 baseaudiosrc: fix timestamp comparission, Fixes #597407
2564 2009-10-13 13:52:02 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
2566 * tests/check/Makefile.am:
2567 * tests/check/pipelines/basetime.c:
2568 tests: new test for baseaudiosrc base_time comparison
2569 This test reveals a bug in comparison operation between timestamp and
2570 GstElement's base_time in GstBaseAudioSrc.
2572 2009-10-08 19:55:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2574 * gst/playback/gstplaybin2.c:
2575 playbin2: Don't stop completely on initialization errors from subtitle elements
2576 Instead disable the subtitles and play the other parts of the stream.
2579 2009-10-13 16:50:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2581 * gst/playback/gstdecodebin2.c:
2582 decodebin2: Ignore no-more-pads from non-demuxer elements
2583 instead of printing an error that no corresponding group could
2584 be found. no-more-pads from non-demuxer elements doesn't give
2585 any additional information because there can only be a single srcpad.
2588 2009-10-12 21:30:15 +0300 Stefan Kost <ensonic@users.sf.net>
2590 * gst/audioconvert/gstaudioconvert.c:
2591 audioconvert: track active conversion in perf log
2593 2009-10-12 15:48:46 +0200 Patrick Radizi <patrick.radizi at axis.com>
2595 * gst-libs/gst/rtsp/gstrtspconnection.c:
2596 rtsp: handle socket errors
2597 gstrtspconnection.c:gst_rtsp_connection_receive() can hang when an error occured
2598 on a socekt. Fix this problem by checking for error on 'other' socket after poll
2602 2009-10-06 14:08:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2604 * gst-libs/gst/audio/gstaudioclock.c:
2605 audioclock: whitespace fixes
2607 2009-10-06 14:07:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2609 * ext/theora/theoradec.c:
2610 theoradec: avoid confusing error
2612 2009-10-09 22:00:45 +0200 Josep Torra <n770galaxy@gmail.com>
2614 * ext/vorbis/vorbisdec.c:
2615 * ext/vorbis/vorbisenc.c:
2616 vorbis: fixes warings in macosx snow leopard
2618 2009-10-09 18:52:12 +0200 Josep Torra <n770galaxy@gmail.com>
2620 * ext/theora/theoradec.c:
2621 * ext/theora/theoraparse.c:
2622 theora: fixes warnings on macosx snow leopard
2624 2009-10-09 16:56:29 +0200 Josep Torra <n770galaxy@gmail.com>
2626 * ext/ogg/gstoggmux.c:
2627 * ext/ogg/gstoggparse.c:
2628 ogg: fixes warnings on macosx snow leopard
2630 2009-10-09 16:19:17 +0200 Josep Torra <n770galaxy@gmail.com>
2632 * ext/ogg/gstoggdemux.c:
2633 oggdemux: fix a warning in macosx
2635 2009-10-08 14:16:44 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
2637 * gst-libs/gst/tag/tags.c:
2638 tag: use BOM to recognize UTF-16/32 encoding and convert accordingly
2640 2009-10-09 15:11:16 +0100 Jan Schmidt <thaytan@noraisin.net>
2642 * tests/check/gst-plugins-base.supp:
2643 check: Add valgrind suppressions for ALSA and fontconfig bits on Jaunty.
2645 2009-10-09 15:32:45 +0200 Josep Torra <n770galaxy@gmail.com>
2647 * ext/gnomevfs/gstgnomevfssrc.c:
2648 audioconvert: change the format instead of cast as ensonic asked
2650 2009-10-09 15:29:15 +0200 Josep Torra <n770galaxy@gmail.com>
2652 * gst/audioconvert/gstchannelmix.c:
2653 audioconvert: fixes warning: format not a string literal and no format arguments
2654 redo of valid part of my previous revert.
2656 2009-10-09 15:19:42 +0200 Josep Torra <n770galaxy@gmail.com>
2659 * gst/audioconvert/gstchannelmix.c:
2660 Revert "audioconvert: fixes warning: format not a string literal and no format arguments"
2661 Revert this commit as unintentionally I've changed common.
2662 This reverts commit 49ea0138223ec5f9e53780635cbcc70f33778667.
2664 2009-10-09 14:28:42 +0200 Josep Torra <n770galaxy@gmail.com>
2666 * ext/gnomevfs/gstgnomevfssrc.c:
2667 gnomevfssrc: fixes warnings in macosx
2668 warning: format '%llu' expects type 'long long unsigned int', but argument 8 has type 'GnomeVFSFileOffset'
2669 warning: format '%lld' expects type 'long long int', but argument 9 has type 'guint64'
2671 2009-10-09 14:23:36 +0200 Josep Torra <n770galaxy@gmail.com>
2673 * gst/videorate/gstvideorate.c:
2674 videorate: fix warning in macosx
2676 2009-10-09 14:20:47 +0200 Josep Torra <n770galaxy@gmail.com>
2678 * gst/audiorate/gstaudiorate.c:
2679 audiorate: fix warning in macosx
2681 2009-10-09 14:14:15 +0200 Josep Torra <n770galaxy@gmail.com>
2684 * gst/audioconvert/gstchannelmix.c:
2685 audioconvert: fixes warning: format not a string literal and no format arguments
2687 2009-10-09 14:07:24 +0200 Josep Torra <n770galaxy@gmail.com>
2689 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2690 * gst-libs/gst/audio/gstringbuffer.c:
2691 audio: fix warnings building on macosx
2693 2009-10-08 18:08:22 +0300 Stefan Kost <ensonic@users.sf.net>
2695 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
2696 * gst/ffmpegcolorspace/imgconvert.c:
2697 ffmpegcolorspace: chwck formats just once per _chain()
2699 2009-10-08 17:49:39 +0300 Stefan Kost <ensonic@users.sf.net>
2701 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
2702 * gst/ffmpegcolorspace/imgconvert.c:
2703 ffmpegcolorspace: add perf-log-category and log suboptimal operation
2704 Log if we use an intermediate colorspace for conversion.
2706 2009-10-08 10:59:36 +0100 Jan Schmidt <thaytan@noraisin.net>
2709 Automatic update of common submodule
2710 From 19fa4f3 to a3e3ce4
2712 2009-10-08 00:17:21 +0100 Jan Schmidt <jan.schmidt@sun.com>
2714 * gst/playback/gstdecodebin2.c:
2715 decodebin2: Fix type-punning warning
2717 2009-09-26 12:56:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2719 * gst/playback/gstdecodebin2.c:
2720 decodebin2: Chains with an exposed endpad are complete too
2721 This allows partial group changes, i.e. demuxer2 in the example below
2722 goes EOS but has a next group and audio2 stays the same.
2723 /-- >demuxer2---->video
2724 demuxer--- \--->audio1
2727 2009-09-26 12:47:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2729 * gst/playback/gstdecodebin2.c:
2730 decodebin2: Use the iterate internal links function instead of string magic to get multiqueue srcpads
2732 2009-09-24 14:56:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2734 * gst/playback/gsturidecodebin.c:
2735 uridecodebin: Don't post missing plugin messages twice
2736 decodebin2 already posts them after emitting the unknown-type signal,
2737 there's no need to post another one.
2739 2009-09-26 12:17:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2741 * gst/playback/gstdecodebin2.c:
2742 decodebin2: Rewrite autoplugging and how groups of pads are exposed
2743 This now keeps track of everything that is going on, creates
2744 a tree of chains and groups to allow "demuxer after demuxer" scenarios
2745 and allows chained Oggs with multiple streams (needs oggdemux or playbin2 fixes).
2746 Also document everything in detail and give a general overview of what
2747 decodebin2 is doing at the top of the sources.
2748 Fixes bug #596183, #563828 and #591677.
2750 2009-10-07 17:45:33 +0300 Stefan Kost <ensonic@users.sf.net>
2752 * sys/ximage/ximagesink.c:
2753 ximagesink: only start event thread if needed
2754 The event thread is doing 20 wakeups per second to poll the events. If one
2755 runs ximagesink with handle-events=false and handle-expose=false then we can
2756 avoid the extra thread.
2758 2009-10-07 16:56:28 +0200 Edward Hervey <bilboed@bilboed.com>
2760 * ext/theora/theoraenc.c:
2761 theoraenc: Make the default quality property 48.
2762 This guarantees that people who use theoraenc without modifying any
2763 properties will end up with a reasonably good quality output.
2764 48 is also the default of the encoder_example application shipped with
2767 2009-10-07 11:48:37 +0200 Benjamin Otte <otte@gnome.org>
2769 * tests/check/libs/video.c:
2770 tests/check/libs/video.c: Update strides for Y41B
2772 2009-10-07 10:32:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2774 * gst-libs/gst/rtsp/gstrtspconnection.c:
2775 rtspconnection: we can use GLib 2.18 API unconditionally now
2777 2009-10-07 10:13:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2780 configure: bump GLib requirement to 2.18
2781 Bump required GLib version as per the release planning docs.
2783 2009-10-05 00:33:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2785 * gst-libs/gst/interfaces/tuner.c:
2786 docs: clarify GstTuner docs in two places
2788 2009-09-25 15:32:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2790 * sys/v4l/gstv4lelement.c:
2791 v4l: fix compiler warning
2792 Fix 'variable may be used uninitialized' compiler warning (which is
2793 true in theory, but can't actually ever happen, since we always
2794 call the function with check=FALSE).
2797 2009-10-07 11:56:35 +0300 Stefan Kost <ensonic@users.sf.net>
2799 * ext/gnomevfs/gstgnomevfssrc.c:
2800 * ext/ogg/gstogmparse.c:
2801 * gst/subparse/gstsubparse.c:
2802 * gst/subparse/mpl2parse.c:
2803 * gst/subparse/tmplayerparse.c:
2804 build: sprintf, sscanf need stdio.h
2806 2009-09-15 15:26:06 +0300 Stefan Kost <ensonic@users.sf.net>
2808 * sys/xvimage/xvimagesink.c:
2809 xvimagesink: only start event thread if needed
2810 The event thread is doing 20 wakeups per second to poll the events. If one runs
2811 xvimagesink with handle-events=false and handle-expose=false then we can avoid
2814 2009-10-07 09:58:27 +0200 Benjamin Otte <otte@gnome.org>
2816 * gst-libs/gst/video/video.h:
2817 Update Since tags for NV12/NV21
2818 They are added in 0.10.26 now, not 0.10.25
2820 2009-09-23 15:31:50 +0200 Benjamin Otte <otte@gnome.org>
2822 * gst/videotestsrc/videotestsrc.c:
2823 [videotestsrc] Make checkers-8 pattern create 8x8 instead of 16x16 tiles
2825 2009-09-23 11:03:57 +0200 Benjamin Otte <otte@gnome.org>
2827 * gst/ffmpegcolorspace/imgconvert_template.h:
2828 [ffmpegcolorspace] Fix NV12 and NV21 with odd width and height
2830 2009-09-23 10:25:02 +0200 Benjamin Otte <otte@gnome.org>
2832 * gst-libs/gst/video/video.c:
2833 * gst-libs/gst/video/video.h:
2834 Add NV12 and NV21 formats
2836 2009-09-21 18:49:42 +0200 Benjamin Otte <otte@gnome.org>
2838 * gst-libs/gst/video/video.c:
2840 Chroma components should be aligned on 4byte boundaries.
2841 https://bugzilla.gnome.org/show_bug.cgi?id=595849
2843 2009-09-21 18:49:06 +0200 Benjamin Otte <otte@gnome.org>
2845 * gst/videotestsrc/videotestsrc.c:
2846 [videotestsrc] Fix Y41B
2847 Chroma components should be aligned on 4byte boundaries.
2848 https://bugzilla.gnome.org/show_bug.cgi?id=595849
2850 2009-10-07 07:28:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2853 * gst-libs/gst/interfaces/streamvolume.c:
2854 streamvolume: Define cbrt() if it's not available
2855 Fixes build on Win32, bug #597537.
2857 2009-09-24 16:05:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2859 * gst/playback/gstfactorylists.c:
2860 factorylist: Use gst_caps_can_intersect() instead of _intersect()
2861 This is faster and results in less allocations.
2863 2009-09-26 12:10:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2865 * gst/playback/gstdecodebin2.c:
2866 decodebin2: Don't set the external ghostpads blocked but only their targets
2867 Pad blocks should never be done on external pads as outside elements
2868 might want to use their own pad blocks on them and this will lead to
2869 conflicts and deadlocks.
2871 2009-09-26 12:04:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2873 * gst/playback/gstdecodebin2.c:
2874 decodebin2: Only use the object lock for protecting the subtitle elements
2875 Using the decodebin lock will result in deadlocks if the subtitle encoding
2876 is accessed from a pad-added handler.
2878 2009-09-26 18:11:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2880 * gst/playback/gstplaybin2.c:
2881 playbin2: Improve debugging of pad blocks
2883 2009-09-23 16:07:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2885 * gst/playback/gstplaybin2.c:
2886 * gst/playback/gstplaysink.c:
2887 playbin2/playsink: Use gst_object_ref_sink() instead of calling both separately
2889 2009-10-06 19:59:11 -0700 David Schleef <ds@schleef.org>
2892 configure: Add an 'else' to pangocairo check
2893 Otherwise it exits if it fails.
2895 2009-10-06 19:35:50 -0700 David Schleef <ds@schleef.org>
2897 * gst/videotestsrc/gstvideotestsrc.c:
2898 * gst/videotestsrc/gstvideotestsrc.h:
2899 * gst/videotestsrc/videotestsrc.c:
2900 * gst/videotestsrc/videotestsrc.h:
2901 videotestsrc: add pattern with out-of-gamut colors
2902 Adds a pattern with out-of-gamut colors in a checkerboard
2903 pattern with in-gamut neighbors. Useful for checking YCbCr->RGB
2904 color matrixing. Correct matrixing and clamping will cause the
2905 checkerboard pattern to be invisible.
2907 2009-10-06 19:17:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2909 * gst-libs/gst/rtsp/gstrtspconnection.c:
2910 rtsp: use CLOSE_SOCKET() instead of close()
2911 Use CLOSE_SOCKET instead of directly calling close() because it does the right
2915 2009-10-01 14:19:41 +0200 Robert Swain <robert swain gmail com>
2917 * gst/audioresample/gstaudioresample.c:
2918 audioresample: fix printf variable type
2919 Change printf variable type from %lu to %" G_GUINT64_FORMAT " as it
2920 should be for guint64.
2923 2009-09-30 23:22:35 +0100 Jan Schmidt <thaytan@noraisin.net>
2925 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
2926 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
2927 ffmpegcolorspace: Use the ffmpegcolorspace debug category
2928 Move gstffmpegcodecmap debug to the ffmpegcolorspace category
2930 2009-09-22 11:58:26 +0100 Jan Schmidt <thaytan@noraisin.net>
2932 * gst/gdp/gstgdppay.c:
2933 gdppay: Don't repeat tags buffers for every new segment
2934 Only send a tag buffer when one is received, not after every new segment
2937 2009-09-28 20:25:35 -0700 David Schleef <ds@schleef.org>
2939 * gst/typefind/gsttypefindfunctions.c:
2940 typefind: detect 'ftypqt ' as video/quicktime
2942 2009-10-06 19:47:00 +0100 Jan Schmidt <thaytan@noraisin.net>
2945 back to development -> 0.10.25.1
2947 === release 0.10.25 ===
2949 2009-10-05 13:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
2955 * docs/plugins/gst-plugins-base-plugins.args:
2956 * docs/plugins/gst-plugins-base-plugins.hierarchy:
2957 * docs/plugins/gst-plugins-base-plugins.interfaces:
2958 * docs/plugins/gst-plugins-base-plugins.prerequisites:
2959 * docs/plugins/gst-plugins-base-plugins.signals:
2960 * docs/plugins/inspect/plugin-adder.xml:
2961 * docs/plugins/inspect/plugin-alsa.xml:
2962 * docs/plugins/inspect/plugin-app.xml:
2963 * docs/plugins/inspect/plugin-audioconvert.xml:
2964 * docs/plugins/inspect/plugin-audiorate.xml:
2965 * docs/plugins/inspect/plugin-audioresample.xml:
2966 * docs/plugins/inspect/plugin-audiotestsrc.xml:
2967 * docs/plugins/inspect/plugin-cdparanoia.xml:
2968 * docs/plugins/inspect/plugin-decodebin.xml:
2969 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
2970 * docs/plugins/inspect/plugin-gdp.xml:
2971 * docs/plugins/inspect/plugin-gio.xml:
2972 * docs/plugins/inspect/plugin-gnomevfs.xml:
2973 * docs/plugins/inspect/plugin-libvisual.xml:
2974 * docs/plugins/inspect/plugin-ogg.xml:
2975 * docs/plugins/inspect/plugin-pango.xml:
2976 * docs/plugins/inspect/plugin-playback.xml:
2977 * docs/plugins/inspect/plugin-queue2.xml:
2978 * docs/plugins/inspect/plugin-subparse.xml:
2979 * docs/plugins/inspect/plugin-tcp.xml:
2980 * docs/plugins/inspect/plugin-theora.xml:
2981 * docs/plugins/inspect/plugin-typefindfunctions.xml:
2982 * docs/plugins/inspect/plugin-uridecodebin.xml:
2983 * docs/plugins/inspect/plugin-video4linux.xml:
2984 * docs/plugins/inspect/plugin-videorate.xml:
2985 * docs/plugins/inspect/plugin-videoscale.xml:
2986 * docs/plugins/inspect/plugin-videotestsrc.xml:
2987 * docs/plugins/inspect/plugin-volume.xml:
2988 * docs/plugins/inspect/plugin-vorbis.xml:
2989 * docs/plugins/inspect/plugin-ximagesink.xml:
2990 * docs/plugins/inspect/plugin-xvimagesink.xml:
2991 * gst-plugins-base.doap:
2994 2009-10-05 13:49:10 +0100 Jan Schmidt <thaytan@noraisin.net>
3030 2009-10-01 17:17:55 +0100 Jan Schmidt <thaytan@noraisin.net>
3066 0.10.24.4 pre-release
3068 2009-10-01 10:37:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3070 * ext/pango/gsttextoverlay.c:
3071 * ext/pango/gsttextrender.c:
3072 pango: Unpremultiply Cairo's ARGB to match GStreamers ARGB
3074 2009-09-28 22:06:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3076 * gst/playback/gstplaysink.c:
3077 playsink: make the lock recursive for now
3080 2009-09-28 21:54:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3082 * gst/playback/gstplaysink.c:
3083 playsink: fix the vis property getter
3085 2009-09-30 18:06:56 +0100 Christian F.K. Schaller <christian.schaller@collabora.co.uk>
3087 * gst-plugins-base.spec.in:
3088 Add missing file to spec file
3090 2009-09-17 16:57:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3092 * gst-libs/gst/cdda/gstcddabasesrc.c:
3093 * tests/check/libs/cddabasesrc.c:
3094 cddabasesrc: Fix string leaks in the unit test and a leak in cddabasesrc
3096 2009-09-17 23:42:52 +1000 Jonathan Matthew <jonathan@d14n.org>
3098 * gst-libs/gst/cdda/gstcddabasesrc.c:
3099 * tests/check/libs/cddabasesrc.c:
3100 cddabasesrc: ignore URI fragments that look like device paths
3101 Rhythmbox uses cdda:// URIs of the form cdda://track#device, which
3102 worked before the fix for bug #321532.
3103 Also adds a check for negative track numbers and some unit tests for URI
3107 2009-09-17 01:20:45 +0100 Jan Schmidt <thaytan@noraisin.net>
3142 0.10.24.3 pre-release
3144 2009-09-15 15:23:49 -0700 Michael Smith <msmith@songbirdnest.com>
3146 * gst-libs/gst/tag/gstvorbistag.c:
3147 vorbistag: don't ever return NULL in list of strings.
3149 2009-09-14 12:18:33 +0200 Edward Hervey <bilboed@bilboed.com>
3151 * gst/playback/gstplaysink.c:
3152 playsink: Expose mute,volume,vis-plugin and font-desc properties
3153 https://bugzilla.gnome.org/show_bug.cgi?id=594623
3155 2009-09-09 12:42:04 +0200 Edward Hervey <bilboed@bilboed.com>
3157 * gst/playback/gstplaysink.c:
3158 GstPlaySink: Expose 'reconfigure' as an action signal.
3160 2009-09-09 11:17:28 +0200 Edward Hervey <bilboed@bilboed.com>
3162 * gst/playback/gstplaysink.c:
3163 GstPlaySink: Expose flags as a gobject property.
3165 2009-09-08 11:35:20 +0200 Edward Hervey <bilboed@bilboed.com>
3167 * gst/playback/gstplayback.c:
3168 * gst/playback/gstplaysink.c:
3169 * gst/playback/gstplaysink.h:
3170 playback: Register playsink as an element.
3171 This allows using playsink from outside the playback plugin.
3172 Add code to be able to request the sink pads using standard GStreamer API.
3173 TODO : expose GObject properties/signals.
3175 2009-09-12 14:55:06 +0300 Stefan Kost <ensonic@users.sf.net>
3177 * docs/libs/gst-plugins-base-libs.types:
3178 docs: add new gst_stream_volume_get_type to types file
3179 This is needs to get Gobject features to show up in the docs.
3181 2009-09-12 15:48:11 -0700 David Schleef <ds@schleef.org>
3183 * ext/ogg/gstoggdemux.c:
3184 oggdemux: Fix duration calculation for truncated files
3185 If the last page of a stream has a granulepos of -1, that is,
3186 it doesn't complete a packet, we need to continue to search
3187 for the last granulepos.
3189 2009-09-12 14:01:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3192 * gst-libs/gst/app/Makefile.am:
3193 * gst-libs/gst/audio/Makefile.am:
3194 * gst-libs/gst/cdda/Makefile.am:
3195 * gst-libs/gst/fft/Makefile.am:
3196 * gst-libs/gst/interfaces/Makefile.am:
3197 * gst-libs/gst/netbuffer/Makefile.am:
3198 * gst-libs/gst/pbutils/Makefile.am:
3199 * gst-libs/gst/riff/Makefile.am:
3200 * gst-libs/gst/rtp/Makefile.am:
3201 * gst-libs/gst/rtsp/Makefile.am:
3202 * gst-libs/gst/sdp/Makefile.am:
3203 * gst-libs/gst/tag/Makefile.am:
3204 * gst-libs/gst/video/Makefile.am:
3205 introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH
3206 This way g-ir-scanner can find the gstreamer-*-0.10 pkg-config files.
3208 2009-09-12 02:23:07 +0100 Jan Schmidt <thaytan@noraisin.net>
3210 * ext/theora/theoraenc.c:
3211 theoraenc: Fix a string leak in _getcaps()
3213 2009-09-11 23:49:11 +0100 Jan Schmidt <thaytan@noraisin.net>
3250 0.10.24.2 pre-release
3252 2009-09-11 21:44:18 +0100 Jan Schmidt <thaytan@noraisin.net>
3254 * tests/check/elements/audioresample.c:
3255 check: Improve audioresample test
3256 Make the audioresample test work with CK_FORK=no, and
3257 turn a g_print into a GST_INFO.
3259 2009-09-11 22:09:06 +0200 Benjamin Otte <otte@gnome.org>
3261 * gst/videotestsrc/videotestsrc.c:
3262 videotestsrc: Fix crashes with even widths
3263 The fix for green lines introduced by commit
3264 35fdfcc6258c66ba462a4330a35deffb0f2b501d caused invalid memory accesses
3265 for even widths. This patch fixes it.
3267 2009-09-11 15:11:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3269 * gst/playback/gstplaybin2.c:
3270 playbin2: Implement GstStreamVolume interface
3272 2009-09-11 15:04:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3274 * gst/volume/gstvolume.c:
3275 * gst/volume/gstvolume.h:
3276 * tests/check/Makefile.am:
3277 * tests/check/elements/volume.c:
3278 volume: Implement GstStreamVolume interface
3280 2009-09-11 14:54:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3282 * docs/libs/gst-plugins-base-libs-docs.sgml:
3283 * docs/libs/gst-plugins-base-libs-sections.txt:
3284 * gst-libs/gst/interfaces/Makefile.am:
3285 * gst-libs/gst/interfaces/streamvolume.c:
3286 * gst-libs/gst/interfaces/streamvolume.h:
3287 * gst/playback/Makefile.am:
3288 * win32/common/libgstinterfaces.def:
3289 interfaces: API: Add GstStreamVolume interface
3292 2009-09-11 12:20:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3294 * gst-libs/gst/rtsp/gstrtspconnection.c:
3295 rtsp: properly fix the HTTP manual mode
3296 When we're not parsing HTTP, return EPARSE when we get an HTTP
3299 2009-09-11 10:16:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3301 * gst-libs/gst/interfaces/mixertrack.h:
3302 mixertrack: add READONLY and WRITEONLY flags
3303 Should really have been READABLE and WRITABLE, but those are hard to
3304 add whilst maintaining backwards compatibility. See #343615.
3305 API: GST_MIXER_TRACK_READONLY
3306 API: GST_MIXER_TRACK_WRITEONLY
3308 2009-09-11 10:02:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3310 * gst-libs/gst/audio/gstringbuffer.c:
3311 ringbuffer: fix build against core that has debugging disabled
3312 The macro is called GST_DISABLE_GST_DEBUG, not GST_DISABLE_DEBUG.
3314 2009-09-11 07:38:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3316 * gst/videorate/gstvideorate.c:
3317 videorate: Add Since marker for the new skip-to-first property
3319 2009-09-11 07:36:10 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
3321 * gst/videorate/gstvideorate.c:
3322 * gst/videorate/gstvideorate.h:
3323 videorate: Make videorate work with a live source
3324 Add a property that makes videorate skip to the first buffer it
3325 receives instead of padding the stream from segment start to the
3329 2009-09-11 07:20:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3331 * gst-libs/gst/fft/gstfft.h:
3332 * gst-libs/gst/fft/gstfftf32.h:
3333 * gst-libs/gst/fft/gstfftf64.h:
3334 * gst-libs/gst/fft/gstffts16.h:
3335 * gst-libs/gst/fft/gstffts32.h:
3336 fft: Mark one function as const and add notes that the structs should be private in 0.11
3338 2009-09-10 22:28:19 +0300 Stefan Kost <ensonic@users.sf.net>
3340 * gst-libs/gst/audio/gstringbuffer.c:
3341 ringbuffer: add human readable format names when logging
3342 Add string array with human readable names for format and type to be used in log
3345 2009-09-10 18:19:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3347 * gst-libs/gst/rtp/gstbasertppayload.c:
3348 basertppay: don't print RTP timestamps as clocktime
3349 Don't try to print the RTP timestamp as a GstClockTime, it's just a guint32.
3352 2009-09-10 16:55:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3354 * gst/playback/gstplaybin.c:
3355 * gst/playback/gstplaybin2.c:
3356 playbin(2): Document that the volume property uses a linear scale
3359 2009-09-10 14:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3361 * gst-libs/gst/rtsp/gstrtspconnection.c:
3362 rtsp: don't return EPARSE
3363 Don't blindly return EPARSE when http mode is disabled.
3364 Restore old http mode after temporarily setting it to TRUE.
3366 2009-09-10 12:38:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3368 * gst-libs/gst/audio/gstbaseaudiosink.c:
3369 baseaudiosink: add ugly backward compat hack
3370 Check for pulsesink < 0.10.17 because it includes code that is now included in
3371 baseaudiosink. Disable that code in baseaudiosink to be compatible with the
3374 2009-09-10 10:56:29 +0200 Benjamin Otte <otte@gnome.org>
3376 * gst/ffmpegcolorspace/imgconvert.c:
3377 ffmpegcolorspace: Handle YVU9/YUV9 conversion with odd widths
3378 A green border could be visible when converting to Y444 or RGB, because
3379 the last chroma samples weren't copied correctly
3381 2009-09-10 10:43:37 +0200 Benjamin Otte <otte@gnome.org>
3383 * gst/videotestsrc/videotestsrc.c:
3384 videotestsrc: Fix YVU9 and YUV9
3385 - Buffer sizes were computed different from ffmpegcolorspace
3386 - Green bar on right size for widths not divisable by 4
3388 2009-09-10 10:08:28 +0200 Benjamin Otte <otte@gnome.org>
3390 * gst/videotestsrc/videotestsrc.c:
3391 videotestsrc: Fix image for odd widths in some formats
3392 videotestsrc rounds chroma down. This causes it to omit the last chroma
3393 value completely for odd widths when the chroma is downsampled.
3394 This patch special cases the last pixel to not be rounded down.
3396 2009-09-10 10:02:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3398 * ext/ogg/gstoggdemux.c:
3399 oggdemux: Handle kate and cmml as sparse streams too
3401 2009-09-10 10:00:16 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3403 * ext/ogg/gstoggdemux.c:
3404 * ext/ogg/gstoggdemux.h:
3405 oggdemux: Better handling of sparse streams by sending segment updates
3408 2009-09-10 09:43:28 +0300 Stefan Kost <ensonic@users.sf.net>
3410 * gst/playback/gsturidecodebin.c:
3411 docs: tell a biit more about uri-decodebin and buffering
3413 2009-09-09 18:24:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3415 * gst-libs/gst/audio/gstbaseaudiosink.c:
3416 baseaudiosink: take clock time in setcaps
3417 Take the time of the clock so that the last_time field is set. This is important
3418 for sinks that restart their internal ringbuffer after a caps change and need to
3419 know the last know position.
3421 2009-09-09 18:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3423 * gst-libs/gst/audio/gstaudioclock.c:
3424 audioclock: add some more debug
3426 2009-09-09 16:44:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3428 * ext/theora/theoraenc.c:
3429 theoraenc: Print a debug message with supported formats
3431 2009-09-07 17:29:38 +0200 Benjamin Otte <otte@gnome.org>
3433 * ext/theora/theoraenc.c:
3434 theora: Check supported input formats in getcaps function
3435 We want to fail early when an older libtheora release is used that does
3436 not support Y444 or Y42B formats, so use a getcaps function that does
3439 2009-09-04 21:37:04 +0200 Benjamin Otte <otte@gnome.org>
3441 * ext/theora/theoraenc.c:
3442 theora: Implement support in theoraenc for Y444 and Y42B
3445 2009-09-04 20:23:52 +0200 Benjamin Otte <otte@gnome.org>
3447 * ext/theora/theoraenc.c:
3448 theora: Refactor the buffer copy code
3450 2009-09-04 16:59:49 +0200 Benjamin Otte <otte@gnome.org>
3452 * ext/theora/theoraenc.c:
3453 theora: Split yuv_buffer creation into its own function
3455 2009-09-04 16:49:08 +0200 Benjamin Otte <otte@gnome.org>
3457 * ext/theora/theoraenc.c:
3458 theora: Split out buffer resize in its own function
3460 2009-09-04 14:06:09 +0200 Benjamin Otte <otte@gnome.org>
3462 * ext/theora/theoraenc.c:
3463 theora: Add assertions that functions don't fail
3464 Some functions in libtheora can return an error, but that error cannot
3465 ever happen inside theoraenc. In those cases assert that it doesn't.
3467 2009-09-09 16:21:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3469 * tests/examples/seek/seek.c:
3470 seek: make stop state configurable
3471 Make it easy to experiment with different stop states (NULL and READY)
3473 2009-09-09 16:19:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3475 * gst-libs/gst/audio/gstbaseaudiosink.c:
3476 baseaudiosink: correct for clock reset
3477 When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
3478 also make sure that the clock is updated with the elapsed time so that it
3479 alsways increments even when the ringbuffer goes back to 0. When this happened
3480 we need to adjust the sample position for the reset ringbuffer.
3483 2009-09-09 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3485 * gst-libs/gst/audio/gstbaseaudiosink.h:
3486 baseaudiosink: whitespace fixes
3488 2009-09-09 16:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3490 * gst-libs/gst/audio/gstringbuffer.c:
3491 ringbuffer: add more debug
3493 2009-09-09 10:25:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3495 * gst-libs/gst/interfaces/colorbalance.h:
3496 * gst-libs/gst/interfaces/mixer.h:
3499 2009-09-08 17:59:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3501 * gst-libs/gst/video/gstvideosink.c:
3502 * gst-libs/gst/video/gstvideosink.h:
3503 videosink: add "show-preroll-frame" property
3504 Add a property to disable rendering of video frames during preroll. This
3505 will only work for videosinks that use the new ::show_frame() vfunc instead
3506 of overriding basesink's preroll and render vfuncs directly.
3507 API: GstVideoSink:show-preroll-frame
3509 2009-09-08 17:43:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3511 * sys/ximage/ximagesink.c:
3512 * sys/xvimage/xvimagesink.c:
3513 ximagesink, xvimagesink: use new GstVideoSink::show_frame() vfunc
3515 2009-09-08 18:19:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3517 * gst-libs/gst/video/gstvideosink.c:
3518 * gst-libs/gst/video/gstvideosink.h:
3519 video: add GstVideoSinkClass::show_frame()
3520 Add ::show_frame() vfunc which maps to basesink's ::preroll and ::render
3521 vfuncs and add some gtk-doc chunks.
3522 API: GstVideoSinkClass::show_frame()
3524 2009-09-08 16:00:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3526 * gst-libs/gst/interfaces/navigation.c:
3527 navigation: don't do stuff inside g_return_val_if_fail() statements
3528 Or it will all fall apart if someone compiles with -DG_DISABLE_ASSERT.
3530 2009-08-31 20:24:22 +0200 Havard Graff <havard.graff@tandberg.com>
3532 * gst-libs/gst/interfaces/navigation.c:
3533 navigation: Fix compiler warning with MSVC
3536 2009-08-31 20:31:56 +0200 Havard Graff <havard.graff@tandberg.com>
3538 * gst-libs/gst/rtp/gstbasertpdepayload.c:
3539 basertpdepayload: fix event forwarding
3541 2009-08-31 20:36:37 +0200 Havard Graff <havard.graff@tandberg.com>
3543 * gst-libs/gst/rtp/gstrtcpbuffer.c:
3544 rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
3547 2009-09-08 13:02:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3549 * gst/playback/gstplaybin2.c:
3550 * gst/playback/gstplaysink.c:
3551 * gst/playback/gstplaysink.h:
3554 2009-09-08 12:59:20 +0200 Håvard Graff <havard.graff@tandberg.com>
3556 * gst-libs/gst/audio/gstbaseaudiosrc.c:
3557 baseaudiosrc: improve slave skew resync
3558 The old one did the mistake of not actually advancing the ringbuffer, it just
3559 adjusted the segbase, introducing the whole lenght of the ringbuffer as an
3560 extra delay in the pipeline.
3561 Also make sure that the resync can never go back in time, producing the same
3562 timestamps that has already been produced, as this can cause severe problems
3563 for sinks and other synching mechanisms.
3566 2009-09-07 17:13:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3568 * gst/typefind/gsttypefindfunctions.c:
3569 typefinding: disable typefinder for headerless flac
3570 Disable headerless flac typefinder as long as it happily typefinds anything
3571 including /dev/urandom as flac and as long as it's not particularly useful
3572 given that such streams don't really exist in the wild.
3573 Also fix up some comments so that gtk-doc doesn't complain about them.
3575 2009-09-06 15:21:43 +0300 René Stadler <mail@renestadler.de>
3577 * sys/ximage/ximagesink.c:
3578 ximagesink: fix small memory leak when setting window title
3580 2009-09-06 01:42:42 +0300 René Stadler <mail@renestadler.de>
3582 * sys/xvimage/xvimagesink.c:
3583 xvimagesink: fix small memory leak when setting window title
3585 2009-09-05 13:55:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3588 introspection: Add *.gir and *.typelib to .gitignore
3590 2009-09-05 13:46:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3592 * gst-libs/gst/app/Makefile.am:
3593 * gst-libs/gst/audio/Makefile.am:
3594 * gst-libs/gst/interfaces/Makefile.am:
3595 * gst-libs/gst/pbutils/Makefile.am:
3596 * gst-libs/gst/rtsp/Makefile.am:
3597 * gst-libs/gst/video/Makefile.am:
3598 introduction: Fix out-of-tree build
3600 2009-09-05 13:13:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3602 * gst-libs/gst/rtsp/Makefile.am:
3603 rtsp: Fix introspection build by ordering sources/headers in dependency order
3605 2009-09-05 13:09:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3607 * gst-libs/gst/audio/Makefile.am:
3608 audio: Remove debug echo
3610 2009-09-05 13:08:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3612 * gst-libs/gst/audio/Makefile.am:
3613 audio: Fix build of introspection data by using dependency order for the headers/sources
3615 2009-09-05 12:31:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3617 * gst-libs/gst/app/Makefile.am:
3618 * gst-libs/gst/audio/Makefile.am:
3619 * gst-libs/gst/cdda/Makefile.am:
3620 * gst-libs/gst/fft/Makefile.am:
3621 * gst-libs/gst/interfaces/Makefile.am:
3622 * gst-libs/gst/netbuffer/Makefile.am:
3623 * gst-libs/gst/pbutils/Makefile.am:
3624 * gst-libs/gst/riff/Makefile.am:
3625 * gst-libs/gst/rtp/Makefile.am:
3626 * gst-libs/gst/rtsp/Makefile.am:
3627 * gst-libs/gst/sdp/Makefile.am:
3628 * gst-libs/gst/tag/Makefile.am:
3629 * gst-libs/gst/video/Makefile.am:
3630 introspection: Strip Gst prefix from all types/functions
3632 2009-09-05 11:49:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3634 * gst-libs/gst/Makefile.am:
3635 * gst-libs/gst/app/Makefile.am:
3636 * gst-libs/gst/audio/Makefile.am:
3637 * gst-libs/gst/fft/Makefile.am:
3638 * gst-libs/gst/interfaces/Makefile.am:
3639 * gst-libs/gst/netbuffer/Makefile.am:
3640 * gst-libs/gst/pbutils/Makefile.am:
3641 * gst-libs/gst/riff/Makefile.am:
3642 * gst-libs/gst/rtp/Makefile.am:
3643 * gst-libs/gst/rtsp/Makefile.am:
3644 * gst-libs/gst/sdp/Makefile.am:
3645 * gst-libs/gst/tag/Makefile.am:
3646 * gst-libs/gst/video/Makefile.am:
3647 introspection: Fix build if gir-repository is not installed
3649 2009-09-05 11:37:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3651 * gst-libs/gst/video/Makefile.am:
3652 video: Add gobject-introspection support
3654 2009-09-05 11:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3656 * gst-libs/gst/tag/Makefile.am:
3657 tag: Add gobject-introspection support
3659 2009-09-05 11:34:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3661 * gst-libs/gst/sdp/Makefile.am:
3662 sdp: Add gobject-introspection support
3664 2009-09-05 11:31:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3666 * gst-libs/gst/app/Makefile.am:
3667 * gst-libs/gst/audio/Makefile.am:
3668 * gst-libs/gst/interfaces/Makefile.am:
3669 * gst-libs/gst/pbutils/Makefile.am:
3670 libs: Add nodist headers and sources to the introspection files
3672 2009-09-05 11:28:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3674 * gst-libs/gst/rtsp/Makefile.am:
3675 rtsp: Add gobject-introspection support
3677 2009-09-05 11:25:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3679 * gst-libs/gst/rtp/Makefile.am:
3680 rtp: Add gobject-introspection support
3682 2009-09-05 11:23:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3684 * gst-libs/gst/riff/Makefile.am:
3685 riff: Add gobject-introspection support
3687 2009-09-05 11:20:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3689 * gst-libs/gst/pbutils/Makefile.am:
3690 pbutils: Add gobject-introspection support
3692 2009-09-05 11:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3694 * gst-libs/gst/netbuffer/Makefile.am:
3695 netbuffer: Add gobject-introspection support
3697 2009-09-05 11:15:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3699 * gst-libs/gst/interfaces/Makefile.am:
3700 interfaces: Add gobject-introspection support
3702 2009-09-05 11:04:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3704 * gst-libs/gst/fft/Makefile.am:
3705 fft: Add gobject-introspection support
3707 2009-09-05 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3709 * gst-libs/gst/cdda/Makefile.am:
3710 cdda: Add gobject-introspection support
3711 This is disabled for now until gobject-introspection is fixed
3713 2009-09-05 10:50:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3715 * gst-libs/gst/audio/Makefile.am:
3716 audio: Add gobject-introspection support
3718 2009-09-05 10:40:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3721 * gst-libs/gst/app/Makefile.am:
3722 app: Add gobject-introspection support
3724 2009-09-05 10:20:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3727 Automatic update of common submodule
3728 From 00a859e to 19fa4f3
3730 2009-09-04 15:48:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3732 * gst/typefind/gsttypefindfunctions.c:
3733 typefind: fix midi typefinding
3734 We already have a audio/midi typefinder so don't override it with the midi in
3735 RIFF typefinder or else we fail to detect plain midi files.
3737 2009-09-04 11:29:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3739 * gst/playback/gsturidecodebin.c:
3740 uridecodebin: do buffering for more uris
3741 Add ssh://, ftp://, sftp://, myth:// to the list of uris that require
3745 2009-09-04 07:36:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3747 * gst/typefind/gsttypefindfunctions.c:
3748 typefindfunctions: Add typefinder for Midi inside RIFF
3749 This is a standard Midi file format that should be supported by
3750 all Midi decoders and also has the mimetype audio/mid according to
3751 the Midi specification homepage.
3754 2009-09-03 18:53:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3756 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
3757 audiortppay: add some debugging
3759 2009-09-03 17:53:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3761 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
3762 audiortppay: handle gaps
3763 Add various conversion functions between time<->bytes<->rtptime that will be
3765 Refactor the min/max packet length code so that it can be used for both
3766 sample/frame based payloaders. Cache the returned values.
3768 When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
3769 same gap as the GStreamer timestamps gap.
3771 2009-09-03 14:13:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3773 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
3774 audiortppay: fix frame duration calculations
3775 Fix the calculation of the frame duration and rtp timestamps.
3778 2009-09-03 14:13:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3780 * gst-libs/gst/rtp/gstbasertppayload.c:
3781 rtppay: add some debugging
3783 2009-09-02 19:49:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3785 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
3786 audiortppay: use offsets for RTP timestamps
3787 Have a custom sample/frame function to generate an offset that the base class
3788 will use for generating RTP timestamps. This results in perfect RTP timestamps
3789 on the output buffers.
3790 Refactor setting metadata on output buffers.
3791 Add some more functionality to _flush().
3792 Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
3793 the next outgoing buffer.
3794 Flush the pending data on EOS.
3796 2009-09-02 13:13:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3798 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
3799 audiortppay: move function around
3801 2009-09-02 13:12:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3803 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
3804 audiortppay: fix sample duration calculation
3806 2009-09-02 12:24:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3808 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
3809 audiortppay: more refactoring
3810 Unify the sample/frame buffer handling code by making the functions plugable.
3812 2009-09-02 12:03:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3814 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
3815 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
3816 audiortppayload: refactor some more
3817 Refactor getting the packet min/max size and alignment code.
3818 Refactor converting bytes to time.
3819 change some variable to something shorter.
3821 2009-09-02 10:46:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3823 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
3824 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
3825 * win32/common/libgstrtp.def:
3826 audiortppayload: refactor and cleanup
3827 Always use the adapter when we need to fragment the incomming buffer. Use more
3828 modern adapter functions to avoid malloc and memcpy. The overall result is that
3829 the code looks cleaner while it should be equally fast and in some case avoid a
3831 Use the adapter timestamping functions for more precise timestamps in case of
3833 Cache some values instead of recalculating them.
3834 Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
3835 the internal adapter.
3836 API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
3838 2009-09-03 16:56:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3843 2009-09-03 11:29:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3845 * gst-libs/gst/rtp/gstbasertppayload.c:
3846 basertppay: add property to disable perfect RTP time
3847 Add a property to disable the generation of perfect RTP timestamps. By default
3849 API: GstBaseRTPPayload::perfect-rtptime
3851 2009-09-02 19:47:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3853 * gst-libs/gst/rtp/gstbasertppayload.c:
3854 basertppay: allow subclasses to influence RTP time
3855 Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
3856 which RTP timestamps are generated. Usually timestamps are created from the
3857 GStreamer timestamps on the buffer, which could result in imperfect RTP
3860 2009-09-02 19:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3862 * gst-libs/gst/rtp/gstbasertppayload.h:
3863 basertppay: add macro to cast
3865 2009-09-01 18:26:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3867 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
3868 audiopayload: code cleanups
3870 2009-09-01 18:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3872 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
3873 audiortppayload: don't check adapter
3874 the adapter is never NULL so we don't need to check it.
3875 Use _scale functions to avoid overflows.
3877 2009-09-03 00:14:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3880 * gst/typefind/Makefile.am:
3881 * gst/typefind/gsttypefindfunctions.c:
3882 typefinding: move gio-based xdg mime typefinder from -bad to -base
3883 Its purposes is mainly to avoid false positives (e.g. mp3 typefinder
3884 reporting a 20% probability and somesuch). Won't be registered if
3885 the gio plugin has been disabled via ./configure --disable-gio.
3887 2009-09-01 15:06:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3889 * gst/subparse/gstsubparse.c:
3890 subparse: GstAdapter is not a GstObject and should be freed with g_object_unref
3892 2009-09-01 15:02:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3894 * sys/v4l/v4lsrc_calls.c:
3895 v4lsrc: fix timestamping for when we do not have a clock yet
3898 2009-09-01 14:30:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3900 * sys/v4l/v4lsrc_calls.c:
3901 v4lsrc: don't log not-yet-initialised integer value
3903 2009-09-01 14:28:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3905 * sys/v4l/v4lsrc_calls.c:
3906 v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize
3907 And reflow code to be more indent friendly.
3909 2009-09-01 10:39:52 +0200 Jonas Holmberg <jonas.holmberg@axis.com>
3911 * gst-libs/gst/rtp/gstbasertppayload.c:
3912 * gst-libs/gst/rtp/gstbasertppayload.h:
3913 basertppayload: Make instance init faster by not reading /dev/urandom 3 times
3914 ... which is the default seed when creating a new GRand. Because
3915 GLib in older versions used buffered IO this would take a lot of time.
3916 Instead use the global GRand for getting random numbers and keep the
3917 three instance GRand for backward compatibility with a simple seed.
3920 2009-08-31 22:48:01 +0300 Stefan Kost <ensonic@users.sf.net>
3922 * gst/adder/gstadder.c:
3923 adder: improve caps filter functionality. Fixes #590146.
3924 Also use the capsfilter if there is no src-peer as the caps constrain what
3925 we can do. Don't create any_caps as a default, as we check for NULL to skip the
3926 filtering. This is a (small) performance regression as we always intersect
3929 2009-08-31 11:10:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3931 * gst/playback/gstdecodebin2.c:
3932 decodebin2: Post missing plugin messages before any error messages
3934 2009-08-28 19:06:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3936 * gst-libs/gst/cdda/gstcddabasesrc.c:
3937 cddabasesrc: safely handle the indexes
3939 2009-08-28 19:06:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3941 * win32/common/libgstrtsp.def:
3942 def: add new rtsp symbols
3944 2009-08-28 14:08:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3946 * gst-libs/gst/rtp/gstbasertppayload.h:
3947 basertppayload: whitespace fixes.
3949 2009-08-27 18:59:49 +0200 Marc-André Lureau <mlureau@flumotion.com>
3951 * gst/gdp/gstgdppay.c:
3952 Bug 593035 - set IN_CAPS for streamheader buffer
3954 2009-08-26 16:56:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3956 * gst/playback/gstinputselector.c:
3957 * gst/playback/gststreamselector.c:
3958 playbin: The internally linked pad of the selector might be NULL in some cases
3960 2009-08-26 16:45:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3962 * gst/playback/gstinputselector.c:
3963 * gst/playback/gststreamselector.c:
3964 playbin: Fix iterate internal linked pads functions for the stream selectors
3965 This now used the new gst_iterator_new_single() function and as a side effect
3968 2009-08-26 09:08:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3970 * gst-libs/gst/riff/riff-ids.h:
3971 * gst-libs/gst/riff/riff-read.c:
3972 riff: Add support for AVF files
3973 AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF.
3976 2009-08-26 09:08:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3978 * gst/typefind/gsttypefindfunctions.c:
3979 typefindfunctions: Detect AVF files as RIFF files too
3980 AVF is valid RIFF but has AVF0 as first fourcc instead of RIFF.
3981 Partially fixes bug #593117.
3983 2009-08-21 11:51:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3985 * tests/check/elements/audioresample.c:
3986 audioresample: Add unit test for checking for timestamp drifts
3987 This also checks for perfect timestamping and offsetting.
3989 2009-08-21 10:11:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3991 * gst/audioresample/gstaudioresample.c:
3992 audioresample: Fix drain processing
3993 In case we have to convert internally don't process output length input samples
3994 but history length input samples.
3996 2009-08-21 10:02:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3998 * tests/check/elements/audioresample.c:
3999 audioresample: Improve debugging a bit in the unit test
4001 2009-08-21 10:00:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4003 * gst/audioresample/gstaudioresample.c:
4004 audioresample: On the first buffer we need discont handling
4005 Otherwise we won't get upstream timestamps and everything and all
4006 output buffers would have -1 timestamps.
4008 2009-08-21 08:23:39 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
4011 * gst/subparse/gstsubparse.c:
4012 subparse: Remove dependency on regex.h as it's not used anyway
4015 2009-08-21 06:58:31 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
4017 * gst/audioresample/gstaudioresample.c:
4018 audioresample: Fix buffer overflow when pushing the drain
4020 2009-08-21 06:57:58 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
4022 * gst/audioresample/gstaudioresample.c:
4023 * gst/audioresample/gstaudioresample.h:
4024 audioresample: Fix timestamp drift
4027 2009-08-24 11:34:35 -0700 David Schleef <ds@schleef.org>
4029 * ext/gnomevfs/gstgnomevfssrc.c:
4030 * ext/ogg/gstogmparse.c:
4031 * ext/pango/gsttextrender.c:
4032 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
4033 * gst/playback/gstinputselector.c:
4034 * gst/playback/gststreamselector.c:
4035 * gst/subparse/gstsubparse.c:
4036 * sys/v4l/gstv4lmjpegsink.c:
4037 * sys/v4l/gstv4lmjpegsrc.c:
4038 * sys/v4l/gstv4lsrc.c:
4039 Remove Ronald Bultje from Authors field
4040 Replaced with "GStreamer maintainers
4041 <gstreamer-devel@lists.sourceforge.net>" or just removed,
4042 depending on the number of other authors.
4044 2009-08-24 15:06:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4046 * gst/playback/gstplaybin2.c:
4047 playbin2: fix refcounting of _get_sink()
4048 g_value_set_object() increases the refcount of the sink, which is not needed
4049 because the object should already be refcounted. Make sure this is always the
4050 case and use g_value_take_object().
4053 2009-08-24 14:39:16 +0200 Peter Kjellerstedt <pkj@axis.com>
4055 * gst-libs/gst/rtsp/gstrtspdefs.c:
4056 rtsp: Mark Transport as supporting multiple values.
4058 2009-08-24 13:58:17 +0200 Peter Kjellerstedt <pkj@axis.com>
4060 * gst-libs/gst/rtsp/gstrtspconnection.h:
4061 * gst-libs/gst/rtsp/gstrtspdefs.h:
4062 * gst-libs/gst/rtsp/gstrtspmessage.h:
4063 rtsp: Added missing Since tags.
4065 2009-08-24 13:27:55 +0200 Eero Nurkkala <ext-eero.nurkkala at nokia.com>
4067 * gst-libs/gst/audio/gstringbuffer.c:
4068 ringbuffer: Improve audiosink startup performance
4069 When we start the ringbuffer, immediatly continue processing samples if the
4070 writer prepared some for us.
4073 2009-08-17 11:53:43 +0200 Peter Kjellerstedt <pkj@axis.com>
4075 * gst-libs/gst/rtsp/gstrtspconnection.c:
4076 * gst-libs/gst/rtsp/gstrtspconnection.h:
4077 rtsp: Added new API for sending using GstRTSPWatch.
4078 The new API to send messages using GstRTSPWatch will first try to send the
4079 message immediately. Then, if that failed (or the message was not sent
4080 fully), it will queue the remaining message for later delivery. This avoids
4081 unnecessary context switches, and makes it possible to keep track of
4082 whether the connection is blocked (the unblocking of the connection is
4083 indicated by the reception of the message_sent signal).
4084 This also deprecates the old API (gst_rtsp_watch_queue_data() and
4085 gst_rtsp_watch_queue_message().)
4086 API: gst_rtsp_watch_write_data()
4087 API: gst_rtsp_watch_send_message()
4089 2009-08-17 11:46:32 +0200 Peter Kjellerstedt <pkj@axis.com>
4091 * gst-libs/gst/rtsp/gstrtspconnection.c:
4092 rtsp: Made gst_rtsp_watch_queue_data() thread safe.
4094 2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
4096 * gst-libs/gst/rtsp/gstrtspconnection.c:
4097 * gst-libs/gst/rtsp/gstrtspconnection.h:
4098 rtsp: Added gst_rtsp_connection_set_http_mode().
4099 With gst_rtsp_connection_set_http_mode() it is possible to tell the
4100 connection whether to allow HTTP messages to be supported. By enabling HTTP
4101 support the automatic HTTP tunnel support will also be disabled.
4102 API: gst_rtsp_connection_set_http_mode()
4104 2009-06-16 19:35:23 +0200 Peter Kjellerstedt <pkj@axis.com>
4106 * gst-libs/gst/rtsp/gstrtspconnection.c:
4107 rtsp: Allow gst_rtsp_connection_do_tunnel() to just setup decoding context.
4108 If the second connection passed to gst_rtsp_connection_do_tunnel() is NULL
4109 then just setup the base64 decoding context for the first connection.
4111 2009-06-16 19:04:54 +0200 Peter Kjellerstedt <pkj@axis.com>
4113 * gst-libs/gst/rtsp/gstrtspconnection.c:
4114 rtsp: Write as much as possible in gst_rtsp_source_dispatch().
4115 Try to write as much as possible if there are multiple messages queued.
4117 2009-06-16 18:38:02 +0200 Peter Kjellerstedt <pkj@axis.com>
4119 * gst-libs/gst/rtsp/gstrtspconnection.c:
4120 * gst-libs/gst/rtsp/gstrtspconnection.h:
4121 rtsp: Add error_full callback to GstRTSPWatchFuncs.
4122 The error_full callback is similar to the error callback, but allows for
4123 better error handling. For read errors a partial message is provided to
4124 help an RTSP server generate a more correct error response, and for write
4125 errors the write queue id of the failed message is returned.
4127 2009-08-17 18:29:17 +0200 Peter Kjellerstedt <pkj@axis.com>
4129 * gst-libs/gst/rtsp/gstrtspconnection.c:
4130 rtsp: Made read_line() support LWS.
4131 Rewrote read_line() to support LWS (Line White Space), the method used by
4132 RTSP (and HTTP) to break long lines. Also added support for \r and \n as
4133 line endings (in addition to the official \r\n).
4135 2009-08-20 14:12:50 +0200 Peter Kjellerstedt <pkj@axis.com>
4137 * gst-libs/gst/rtsp/gstrtspconnection.c:
4138 * gst-libs/gst/rtsp/gstrtspdefs.c:
4139 * gst-libs/gst/rtsp/gstrtspdefs.h:
4140 rtsp: Do not split headers which should not be split.
4141 From RFC 2068 section 4.2: "Multiple message-header fields with the same
4142 field-name may be present in a message if and only if the entire
4143 field-value for that header field is defined as a comma-separated list
4144 [i.e., #(values)]." This means that we should not split other headers which
4145 may contain a comma, e.g., Range and Date.
4147 2009-08-20 14:12:09 +0200 Peter Kjellerstedt <pkj@axis.com>
4149 * gst-libs/gst/rtsp/gstrtspconnection.c:
4150 rtsp: Parse WWW-Authenticate headers correctly.
4151 Due to the odd syntax for WWW-Authenticate (and Proxy-Authenticate) which
4152 allows commas both to separate between multiple challenges, and within the
4153 challenges themself, we need to take some extra care to split these headers
4156 2009-06-17 21:46:27 +0200 Peter Kjellerstedt <pkj@axis.com>
4158 * gst-libs/gst/rtsp/gstrtspconnection.c:
4159 rtsp: Improve parse_line().
4160 Make parse_line() handle keys with multiple values on one line correctly.
4162 2009-06-17 23:15:23 +0200 Peter Kjellerstedt <pkj@axis.com>
4164 * gst-libs/gst/rtsp/gstrtspconnection.c:
4165 rtsp: Rewrote setup_tunneling().
4166 Rewrote setup_tunneling() to use normal GstRTSPMessages instead of hard
4167 coded strings and duplicates of the message parsing code.
4169 2009-08-24 10:20:16 +0200 Peter Kjellerstedt <pkj@axis.com>
4171 * gst-libs/gst/rtsp/gstrtspconnection.c:
4172 * gst-libs/gst/rtsp/gstrtspdefs.c:
4173 * gst-libs/gst/rtsp/gstrtspdefs.h:
4174 rtsp: Rewrote gen_tunnel_reply().
4175 Rewrote gen_tunnel_reply() to generate a normal GstRTSPMessage rather
4176 than a hard coded string.
4178 2009-08-24 10:19:35 +0200 Peter Kjellerstedt <pkj@axis.com>
4180 * gst-libs/gst/rtsp/gstrtspconnection.c:
4181 rtsp: Ignore the Content-Length for POST requests.
4182 The Content-Length for POST requests with an x-sessioncookie header should
4183 be ignored as the length is bogus and only there to fool proxies.
4185 2009-06-17 20:52:48 +0200 Peter Kjellerstedt <pkj@axis.com>
4187 * gst-libs/gst/rtsp/gstrtspconnection.c:
4188 rtsp: Normalize lines (remove extra whitespace) before parsing.
4190 2009-06-10 13:11:31 +0200 Peter Kjellerstedt <pkj@axis.com>
4192 * gst-libs/gst/rtsp/gstrtspconnection.c:
4193 rtsp: Made parse_string() return a result.
4194 This will catch parsing errors when a too long string is received.
4196 2009-06-10 11:43:31 +0200 Peter Kjellerstedt <pkj@axis.com>
4198 * gst-libs/gst/rtsp/gstrtspconnection.c:
4199 rtsp: Improved parsing of messages.
4200 Do not abort message parsing as soon as there is an error. Instead parse
4201 as much as possible to allow a server to return as meaningful an error as
4204 2009-06-09 17:54:20 +0200 Peter Kjellerstedt <pkj@axis.com>
4206 * gst-libs/gst/rtsp/gstrtspconnection.c:
4207 * gst-libs/gst/rtsp/gstrtspdefs.c:
4208 * gst-libs/gst/rtsp/gstrtspdefs.h:
4209 * gst-libs/gst/rtsp/gstrtspmessage.c:
4210 * gst-libs/gst/rtsp/gstrtspmessage.h:
4211 rtsp: Added support for HTTP messages
4213 2009-06-09 16:22:17 +0200 Peter Kjellerstedt <pkj@axis.com>
4215 * gst-libs/gst/rtsp/gstrtspconnection.c:
4216 * gst-libs/gst/rtsp/gstrtspconnection.h:
4217 rtsp: Added gst_rtsp_connection_create_from_fd().
4218 API: gst_rtsp_connection_create_from_fd()
4220 2009-06-09 15:27:17 +0200 Peter Kjellerstedt <pkj@axis.com>
4222 * gst-libs/gst/rtsp/gstrtspconnection.c:
4223 rtsp: Add initial buffer support.
4224 The initial buffer contains data for a connection which should be used
4225 before starting to actually read anything from the socket.
4227 2009-08-24 13:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4229 * gst-libs/gst/app/gstappsink.c:
4230 appsink: don't block in paused
4231 When we are asked to unlock we should either leave the render function or call
4232 the wait_preroll method to release the stream lock.
4235 2009-08-24 13:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4237 * docs/libs/gst-plugins-base-libs-sections.txt:
4238 docs: fix includes for appsrc/appsink
4240 2009-08-24 11:24:27 +0200 Peter Kjellerstedt <pkj@axis.com>
4242 * gst-libs/gst/rtsp/gstrtspdefs.c:
4243 * gst-libs/gst/rtsp/gstrtspdefs.h:
4244 rtsp: Add support for the Authentication-Info header.
4245 The Authentication-Info header is defined in RFC 2617 (Digest Access
4248 2009-08-20 13:11:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4250 * ext/ogg/gstoggmux.c:
4251 * tests/check/pipelines/oggmux.c:
4252 oggmux: don't drop the streamheader field from the output caps
4253 Revert previous 'fix' for bug #588717 and fix it properly, whilst
4254 maintaining the streamheader field on the output caps. Also make
4255 sure we don't leak header buffers we couldn't push when downstream
4256 is unlinked. Add unit test for the presence of the streamheader
4257 field on the output caps and for the issue from bug #588717.
4259 2009-08-18 21:45:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4261 * gst/playback/gstinputselector.c:
4262 * gst/playback/gststreamselector.c:
4263 streamselector/inputselector: Use iterate internal links instead of deprecated get internal links
4265 2009-08-19 09:31:51 +0200 Peter Kjellerstedt <pkj@axis.com>
4267 * gst-libs/gst/rtsp/gstrtspconnection.c:
4268 rtsp: Avoid duplicated headers.
4269 Remove any existing Session and Date headers before adding new ones
4270 when sending a request. This may happen if the user of this code reuses
4271 a request (rtspsrc does this when resending after authorization fails).
4273 2009-08-18 16:49:58 +0200 Peter Kjellerstedt <pkj@axis.com>
4275 * gst-libs/gst/rtsp/gstrtspconnection.c:
4276 rtsp: Corrected the HTTP digest authorization computation.
4277 Do not use sizeof() on an array passed as an argument to a function and
4278 expect to get anything but the size of a pointer. As a result only the
4279 first 4 (or 8) bytes of the response buffer were initialized to 0 in
4280 auth_digest_compute_response() which caused it to return a string which
4281 was not NUL-terminated...
4283 2009-08-18 11:15:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4285 * gst/playback/gstplaysink.c:
4286 playsink: Also send SEEK events directly to a subpicture sink
4288 2009-08-18 08:39:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4290 * gst/playback/gstplaysink.c:
4291 playsink: If a custom text sink is used, send events to it too
4292 Before, SEEK events would be sent to the video sink, which wouldn't
4293 be linked in any way to the subtitle part of the pipeline and
4294 subparse would never see the SEEK event. This would then seek
4295 the audio/video but the subtitles would continue from the old
4299 2009-08-18 08:20:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4301 * gst/playback/gsturidecodebin.c:
4302 uridecodebin: Make missing plugins emit a warning message, not an error message
4303 The problem with an error message is, that it will stop playback completely
4304 while it could be that only a audio decoder plugin is missing and the video
4305 could be played with the available plugins.
4308 2009-08-13 17:42:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4310 * gst/playback/gsturidecodebin.c:
4311 uridecodebin: Post a correct error message for unknown types
4312 Before we had STREAM/WRONG_TYPE but it's really CORE/MISSING_PLUGIN
4313 because a plugin is missing and nothing else is wrong.
4314 Also make it an error instead of a warning.
4315 Really fixes bug #591677.
4317 2009-08-13 15:48:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4319 * gst/playback/gsturidecodebin.c:
4320 uridecodebin: Post a missing plugin message additional to the error message on unknown types
4323 2009-08-13 10:59:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4325 * gst/playback/gstplaysink.c:
4357 playbin2: fix error message string
4360 2009-08-05 15:38:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4362 * gst-libs/gst/riff/riff-read.c:
4363 riff: align API doc of gst_riff_parse_chunk with reality
4365 2009-08-05 15:36:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4367 * gst/playback/gstdecodebin2.c:
4368 decodebin2: avoid assertion failure on empty/NULL caps
4370 2009-08-12 12:09:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4372 * gst/typefind/gsttypefindfunctions.c:
4373 typefindfunctions: Also detect SVG by the <svg> starting tag
4374 Not all SVG images have the DOCTYPE specified.
4376 2009-08-10 20:18:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4378 * gst-libs/gst/rtsp/gstrtspconnection.c:
4379 rtspconnection: don't use GLib-2.18 function
4380 g_checksum_reset() was added only in GLib 2.18, but we still require
4381 only 2.16, so work around that if we only have 2.16. Fixes #591357.
4383 2009-08-10 15:40:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4385 * tests/check/pipelines/streamheader.c:
4386 streamheader: Fix caps leak in the vorbisenc unit test
4388 2009-08-10 14:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4390 * tests/check/pipelines/streamheader.c:
4391 checks: fix stream header unit test hanging in gst_task_cleanup_all()
4392 Set pipelines to NULL state and unref when done.
4394 2009-08-10 10:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4396 * gst-libs/gst/rtsp/Makefile.am:
4397 * gst-libs/gst/rtsp/gstrtspconnection.c:
4398 * gst-libs/gst/rtsp/md5.c:
4399 * gst-libs/gst/rtsp/md5.h:
4400 rtsp: Use GLib's GChecksum instead of our own MD5 implementation
4402 2009-08-10 03:46:39 +0300 Mart Raudsepp <leio@gentoo.org>
4404 * gst-libs/gst/interfaces/navigation.c:
4405 navigation: Fix doc blurb typo for gst_navigation_send_key_event
4407 2009-08-09 12:13:16 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4409 * gst/subparse/gstsubparse.c:
4410 subparse: Allow . instead of , as millisecond delimiter in srt subtitles
4413 2009-08-08 17:51:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4415 * gst-libs/gst/audio/gstaudiosrc.c:
4416 * gst/playback/gstinputselector.c:
4417 * gst/playback/gststreamselector.c:
4418 Revert inlines that cause compiler warnings and are not needed anyway
4420 2009-08-08 15:54:57 +0200 Edward Hervey <bilboed@bilboed.com>
4422 * gst-libs/gst/audio/gstaudioclock.c:
4423 * gst-libs/gst/audio/gstaudiosink.c:
4424 * gst-libs/gst/audio/gstaudiosrc.c:
4425 * gst-libs/gst/audio/gstbaseaudiosrc.c:
4426 * gst-libs/gst/audio/gstringbuffer.c:
4427 * gst-libs/gst/interfaces/propertyprobe.c:
4428 * gst-libs/gst/riff/riff-media.c:
4429 * gst-libs/gst/rtp/gstbasertpdepayload.c:
4430 * gst-libs/gst/video/gstvideofilter.c:
4431 * gst-libs/gst/video/gstvideosink.c:
4432 gst-libs: Remove dead assignments and resulting unused variables.
4434 2009-08-08 15:54:41 +0200 Edward Hervey <bilboed@bilboed.com>
4436 * ext/alsa/gstalsadeviceprobe.c:
4437 * ext/alsa/gstalsasink.c:
4438 * ext/alsa/gstalsasrc.c:
4439 * ext/gnomevfs/gstgnomevfssrc.c:
4440 * ext/ogg/gstoggaviparse.c:
4441 * ext/ogg/gstoggdemux.c:
4442 * ext/ogg/gstoggmux.c:
4443 * ext/pango/gsttextrender.c:
4444 * ext/vorbis/vorbisenc.c:
4445 ext: Remove dead assignments and resulting unused variables.
4447 2009-08-08 15:54:02 +0200 Edward Hervey <bilboed@bilboed.com>
4449 * gst/adder/gstadder.c:
4450 * gst/audioconvert/gstaudioconvert.c:
4451 * gst/audioresample/gstaudioresample.c:
4452 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
4453 * gst/ffmpegcolorspace/imgconvert.c:
4454 * gst/playback/gstdecodebin.c:
4455 * gst/playback/gstdecodebin2.c:
4456 * gst/playback/gstfactorylists.c:
4457 * gst/playback/gstinputselector.c:
4458 * gst/playback/gstplaysink.c:
4459 * gst/playback/gststreamselector.c:
4460 * gst/tcp/gsttcpclientsink.c:
4461 * gst/videoscale/gstvideoscale.c:
4462 * gst/videoscale/vs_image.c:
4463 * gst/videotestsrc/gstvideotestsrc.c:
4464 gst: Remove dead assignments and resulting unused variables
4466 2009-08-07 13:05:42 +0200 Josep Torra <n770galaxy@gmail.com>
4468 * docs/design/draft-va.txt:
4469 docs: add draft for generic introduction of video acceleration APIs idea
4471 2009-08-07 08:53:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4473 * ext/theora/gsttheoradec.h:
4474 * ext/theora/theoradec.c:
4475 Revert "theora: Convert theoradec to libtheora 1.0 API"
4476 This reverts commit f1e142ac9dcfb754d85357b9077d5aee48559dd9.
4477 Temporarily revert until we have a workaround for debian/ubuntu
4478 packaging failure (see http://bugs.debian.org/528710).
4480 2009-08-07 09:32:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4482 * gst/typefind/gsttypefindfunctions.c:
4483 typefindfunctions: Add typefinders for many game sound console formats supported by gme
4484 These are AY, GBS, GYM, KSS, SAP and VGM. SPC and NSF already had typefinders.
4486 2009-07-16 11:29:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4488 * ext/ogg/gstoggmux.c:
4489 oggmux: fix warning when we're not linked downstream and error out properly
4490 Fix caps warning when there's no element linked downstream, and pass
4491 not-linked flow return value correctly up the chain, so we error out
4492 correctly. Fixes #588717.
4494 2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
4496 * ext/theora/gsttheoradec.h:
4497 * ext/theora/theoradec.c:
4498 theora: Convert theoradec to libtheora 1.0 API
4500 2009-08-06 20:47:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4502 * ext/pango/gsttextrender.c:
4503 textrender: Fix blitting of text over the output buffer and cairo painting
4505 2009-08-06 09:13:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4507 * ext/pango/gsttextrender.c:
4508 textrender: Fix endianness problems (i.e. make it work again on big endian architectures)
4510 2009-07-31 14:27:28 +0300 Stefan Kost <ensonic@users.sf.net>
4512 * tests/icles/test-colorkey.c:
4513 colorkey-test: fix xsync error
4515 2009-07-06 23:06:50 +0300 Siarhei Siamashka <siarhei.siamashka@nokia.com>
4517 * gst/ffmpegcolorspace/imgconvert.c:
4518 * gst/ffmpegcolorspace/imgconvert_template.h:
4519 ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats
4521 2009-07-14 12:33:29 +0300 Stefan Kost <ensonic@users.sf.net>
4523 * gst/playback/gstplaysink.c:
4524 playbin2: smarter sink selection. Fixes #588523
4525 Don't do fallbacks if application specified a sink element. When doing the
4526 fallback use configured default elements instead of hardcoded linux only
4527 elements. Improve error messages accordingly.
4529 2009-08-06 12:18:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4531 * gst/playback/gstqueue2.c:
4532 queue2: post error message when pausing task if so appropriate
4533 If a downstream element returns an error while upstream has already
4534 put all data into queue2 (including EOS), upstream will no longer
4535 chain into queue2, so it is up to queue2 to perform some
4536 EOS handling / message posting in such cases. See #589991.
4538 2009-08-06 12:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4540 * gst-libs/gst/audio/gstbaseaudiosrc.c:
4541 baseaudiosrc: change default slave method
4542 Set the default slave method to the much better skew slaving algortihm.
4544 2009-08-06 12:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4546 * ext/pango/gsttextoverlay.c:
4547 textoverlay: make buffer writable
4548 Make the input buffer writable before changing its contents.
4550 2009-08-06 09:55:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4552 * gst/typefind/gsttypefindfunctions.c:
4553 typefinding: fix postscript typefinder probability
4554 Two bytes for a rare format hardly warrants MAXIMUM typefinding
4555 probability, POSSIBLE seems more appropriate.
4557 2009-08-04 14:55:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4559 * ext/pango/gsttextoverlay.c:
4560 pango: Send queries from the srcpad directly to the video sinkpad
4562 2009-08-04 14:32:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4564 * gst/subparse/gstsubparse.c:
4565 subparse: Implement POSITION query
4567 2009-08-04 14:29:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4569 * gst/subparse/gstsubparse.c:
4570 * gst/subparse/samiparse.c:
4571 subparse: Implement SEEKING query
4573 2009-08-04 14:14:53 +0200 John Millikin <jmillikin@gmail.com>
4576 * gst-libs/gst/tag/gstid3tag.c:
4577 * gst-libs/gst/tag/gstvorbistag.c:
4578 tag: Add support for ALBUM_ARTIST tag in vorbiscomments and ID3v2 tags
4579 Require latest core for this.
4582 2009-08-04 12:46:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4584 * ext/pango/gsttextoverlay.c:
4585 * ext/pango/gsttextoverlay.h:
4586 pango: Add support for xRGB and BGRx formats
4588 2009-08-04 12:22:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4590 * ext/pango/gsttextoverlay.c:
4591 pango: Fix endianness issues from the pangocairo switch
4592 cairo's ARGB is in native endianness, i.e. ARGB on big endian architectures
4593 and BGRA on little endian architectures.
4595 2009-08-04 12:11:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4597 * ext/pango/gsttextoverlay.c:
4598 pango: Re-add shading support which was dropped by a previous patch
4600 2009-08-04 11:58:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4603 * ext/pango/gsttextoverlay.c:
4604 pango: Check if pangocairo supports vertical rendering and fix properties
4606 2009-08-04 11:45:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4608 * ext/pango/gsttextrender.c:
4609 textrender: Use PROP_X instead of ARG_X consistently
4611 2009-08-04 11:42:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4613 * ext/pango/gstclockoverlay.c:
4614 * ext/pango/gsttextoverlay.c:
4615 * ext/pango/gsttextrender.c:
4616 * ext/pango/gsttimeoverlay.c:
4617 pango: Some minor cleanup
4619 2009-08-04 11:36:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4622 pango: Check for pangocairo instead of pangoft2
4624 2009-08-04 11:35:10 +0200 Young-Ho Cha <ganadist@chollian.net>
4626 * ext/pango/gsttextoverlay.c:
4627 * ext/pango/gsttextoverlay.h:
4628 * ext/pango/gsttextrender.c:
4629 * ext/pango/gsttextrender.h:
4630 pango: Use pango-cairo instead of pango-ft2
4631 pango-cairo will always use the native font rendering backend
4632 of the platform and provides better results.
4635 2009-08-04 10:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4637 * gst/typefind/gsttypefindfunctions.c:
4638 typefindfunctions: Add SVG typefinder
4640 2009-08-04 10:29:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4642 * gst/typefind/gsttypefindfunctions.c:
4643 typefindfunctions: Add postscript typefinder
4645 2009-07-30 15:08:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4647 * gst/typefind/gsttypefindfunctions.c:
4648 typefindfunctions: Use static caps again for MPEG4 typefinding
4650 2009-07-30 15:05:28 +0200 Arnout Vandecappelle <arnout@mind.be>
4652 * gst/typefind/gsttypefindfunctions.c:
4653 typefindfunctions: Implement better & more flexible MPEG4 typefinding
4654 This detects more MPEG4 streams as MPEG4.
4657 2009-07-30 14:04:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4659 * gst-libs/gst/cdda/gstcddabasesrc.c:
4660 cddabasesrc: Allow to specify the device name in the URI
4661 The allowed URI scheme is now:
4662 cdda://(device#)?track
4663 Also allow every combination of uppercase and lowercase
4664 characters for the protocol part.
4667 2009-07-30 12:37:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4669 * gst/videoscale/gstvideoscale.c:
4670 videoscale: Restrict width/height to 2^15 - 1
4671 Otherwise integer overflows will happen, resulting in segmentation faults.
4674 2009-07-29 14:55:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4676 * gst/ffmpegcolorspace/imgconvert_template.h:
4677 ffmpegcolorspace: Fix indention of template header
4679 2009-07-29 14:10:35 +0200 Philip Jägenstedt <philipj@opera.com>
4681 * gst-libs/gst/app/gstappsrc.c:
4682 appsrc: Clarify documentation about caps and linkage
4685 2009-07-29 07:42:05 +0200 Benjamin Gaignard <benjamin@gaignard.net>
4687 * gst/typefind/gsttypefindfunctions.c:
4688 typefindfunctions: Fix typefinding of SDP files
4691 2009-07-28 20:50:06 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
4693 * gst/audioresample/gstaudioresample.c:
4694 audioresample: Take the output offsets from the input if possible
4697 2009-07-28 15:54:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4699 * gst/videoscale/gstvideoscale.c:
4700 videoscale: Make sure to allocate enough memory for the temporary buffer
4701 and fix scaling of odd-height interlaced video.
4703 2009-07-28 15:18:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4705 * gst/videoscale/gstvideoscale.c:
4706 videoscale: Fix interlaced scaling for I420
4707 ...and some other minor mistakes in the previous change.
4709 2009-07-28 14:12:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4711 * gst/ffmpegcolorspace/avcodec.h:
4712 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
4713 * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
4714 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
4715 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
4716 * gst/ffmpegcolorspace/imgconvert.c:
4717 ffmpegcolorspace: Include interlacing information in the AVPicture
4718 This later allows to handle interlaced AVPicture different than
4719 progressive ones which is needed for horizontally subsampled YUV
4720 formats, see bug #589242.
4722 2009-07-28 13:55:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4724 * gst/videoscale/gstvideoscale.c:
4725 * gst/videoscale/gstvideoscale.h:
4726 videoscale: Add support for interlaced content
4727 videoscale is not mixing content of two seperate fields anymore
4728 and does scaling on every field separately.
4731 2009-08-06 01:44:24 +0100 Jan Schmidt <thaytan@noraisin.net>
4734 back to development -> 0.10.24.1
4736 2009-08-05 02:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
4738 * gst-plugins-base.doap:
4739 Add 0.10.24 release to the doap file
4741 === release 0.10.24 ===
4743 2009-08-05 00:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
4749 * docs/plugins/gst-plugins-base-plugins.args:
4750 * docs/plugins/gst-plugins-base-plugins.hierarchy:
4751 * docs/plugins/gst-plugins-base-plugins.interfaces:
4752 * docs/plugins/gst-plugins-base-plugins.prerequisites:
4753 * docs/plugins/gst-plugins-base-plugins.signals:
4754 * docs/plugins/inspect/plugin-adder.xml:
4755 * docs/plugins/inspect/plugin-alsa.xml:
4756 * docs/plugins/inspect/plugin-app.xml:
4757 * docs/plugins/inspect/plugin-audioconvert.xml:
4758 * docs/plugins/inspect/plugin-audiorate.xml:
4759 * docs/plugins/inspect/plugin-audioresample.xml:
4760 * docs/plugins/inspect/plugin-audiotestsrc.xml:
4761 * docs/plugins/inspect/plugin-cdparanoia.xml:
4762 * docs/plugins/inspect/plugin-decodebin.xml:
4763 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
4764 * docs/plugins/inspect/plugin-gdp.xml:
4765 * docs/plugins/inspect/plugin-gio.xml:
4766 * docs/plugins/inspect/plugin-gnomevfs.xml:
4767 * docs/plugins/inspect/plugin-libvisual.xml:
4768 * docs/plugins/inspect/plugin-ogg.xml:
4769 * docs/plugins/inspect/plugin-pango.xml:
4770 * docs/plugins/inspect/plugin-playback.xml:
4771 * docs/plugins/inspect/plugin-queue2.xml:
4772 * docs/plugins/inspect/plugin-subparse.xml:
4773 * docs/plugins/inspect/plugin-tcp.xml:
4774 * docs/plugins/inspect/plugin-theora.xml:
4775 * docs/plugins/inspect/plugin-typefindfunctions.xml:
4776 * docs/plugins/inspect/plugin-uridecodebin.xml:
4777 * docs/plugins/inspect/plugin-video4linux.xml:
4778 * docs/plugins/inspect/plugin-videorate.xml:
4779 * docs/plugins/inspect/plugin-videoscale.xml:
4780 * docs/plugins/inspect/plugin-videotestsrc.xml:
4781 * docs/plugins/inspect/plugin-volume.xml:
4782 * docs/plugins/inspect/plugin-vorbis.xml:
4783 * docs/plugins/inspect/plugin-ximagesink.xml:
4784 * docs/plugins/inspect/plugin-xvimagesink.xml:
4787 2009-08-05 00:38:40 +0100 Jan Schmidt <thaytan@noraisin.net>
4822 2009-08-01 17:26:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4824 * gst/typefind/gsttypefindfunctions.c:
4825 * tests/check/gst/typefindfunctions.c:
4826 typefinding: fix detection of fLaC id packet in broken flac-in-ogg
4827 There are flac-in-ogg files without the usual flac packet framing
4828 and these files just have a 4-byte fLaC ID packet as first packet.
4829 We need to recognise the type just from these four bytes if we
4830 want oggdemux to recognise these streams correctly.
4832 2009-07-30 14:40:50 +0100 Jan Schmidt <thaytan@noraisin.net>
4868 0.10.24.5 pre-release
4870 2009-07-29 14:15:53 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
4872 * gst-libs/gst/audio/gstaudiofilter.c:
4873 audiofilter: Don't assert on slightly different caps
4874 Plugins should not assert on incompatible caps, caps negotiation will
4877 2009-07-30 13:42:21 +0300 Stefan Kost <ensonic@users.sf.net>
4879 * gst/adder/gstadder.c:
4880 adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146.
4882 2009-07-30 09:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4885 configure: bump Gtk+ requirement of GUI examples from 2.12 to 2.14
4886 The gio mount example needs GtkMountOperation, which is new in 2.14.
4888 2009-07-27 10:29:27 +0100 Balachandran C <balachandran_c@rediffmail.com>
4890 * ext/alsa/gstalsasrc.c:
4891 alsasrc: set alsasrc->handle back to NULL when closing device
4892 Fixes crashes in gst_alsa_find_device_name() when probing or
4893 reading the device-name property (e.g. when doing a dot-file
4894 dump). Fixes #589797.
4896 2009-07-24 19:26:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4898 * gst/playback/gststreamselector.c:
4899 playbin: rename GType of stream selector pad to avoid clash with input-selector from -bad
4900 Rename the GType of the pads of playbin's internal stream selector
4901 element so they don't use the same type name as input-selector's
4902 pads. Fixes #589622.
4904 2009-07-24 13:39:55 +0100 Jan Schmidt <thaytan@noraisin.net>
4937 0.10.23.4 pre-release
4939 2009-07-24 13:46:15 +0100 Jan Schmidt <thaytan@noraisin.net>
4941 * tests/examples/v4l/.gitignore:
4942 ignores: Ignore v4l probing example binary
4944 2009-07-24 09:35:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4946 * gst/typefind/gsttypefindfunctions.c:
4947 typefind: recognise Kate spu subtitles as well
4948 Recognise spu-subtitles, SUB and K-SPU as valid categories for
4949 Kate subtitles as well.
4951 2009-07-24 00:42:16 +0300 Stefan Kost <ensonic@users.sf.net>
4954 Automatic update of common submodule
4955 From fedaaee to 94f95e3
4957 2009-07-22 14:21:43 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
4959 * gst-plugins-base.spec.in:
4960 Update spec file with latest changes
4962 2009-07-20 17:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
4995 * win32/common/_stdint.h:
4996 * win32/common/audio-enumtypes.c:
4997 * win32/common/config.h:
4998 * win32/common/gstrtsp-enumtypes.c:
4999 * win32/common/interfaces-enumtypes.c:
5000 * win32/common/video-enumtypes.c:
5001 0.10.23.3 pre-release
5003 2009-07-20 12:51:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5005 * gst/audiotestsrc/gstaudiotestsrc.c:
5006 audiotestsrc: call send_event directly
5007 We can't call gst_element_send_event() from a streaming thread as it gets the
5008 state lock. Instead call the send_event method directly until we have a nice API
5009 for this in basesrc.
5012 2009-07-03 04:42:24 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
5014 * gst-libs/gst/audio/gstaudiosink.c:
5015 audiosink: Add stream-status messages
5018 2009-07-03 04:41:05 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
5020 * gst-libs/gst/audio/gstaudiosrc.c:
5021 audiosrc: Add stream-status messages
5024 2009-07-20 10:53:11 +0200 Edward Hervey <bilboed@bilboed.com>
5026 * gst/adder/gstadder.c:
5027 gstadder: Don't forget to free pending events on flush/dispose.
5030 2009-07-12 10:08:12 +0200 Edward Hervey <bilboed@bilboed.com>
5032 * tests/check/elements/adder.c:
5033 tests/adder: Add stream consistency checking. Fixes #588748
5035 2009-07-12 10:07:34 +0200 Edward Hervey <bilboed@bilboed.com>
5037 * gst/audiotestsrc/gstaudiotestsrc.c:
5038 audiotestsrc: Make sure tags are properly serialized. Fixes #588746
5039 We do this by letting the basesrc base class handle the tags.
5041 2009-07-13 09:28:54 +0200 Edward Hervey <bilboed@bilboed.com>
5043 * gst/adder/gstadder.c:
5044 * gst/adder/gstadder.h:
5045 adder: Collect incoming tag events and send them after newsegment. Fixes #588747
5047 2009-07-16 09:32:46 +0200 Edward Hervey <bilboed@bilboed.com>
5049 * ext/vorbis/vorbisdec.c:
5050 vorbisdec: Check for empty tag strings. Fixes #588724
5052 2009-07-14 17:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5054 * gst/playback/gstqueue2.c:
5055 queue2: fix leak and improve buffering
5056 Keep track of the max requested position and compare this to the write position
5057 in the temp file to get the current amount of buffered data.
5058 Fix memleak of all incomming buffers.
5061 2009-07-15 17:40:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5063 * gst/playback/Makefile.am:
5064 * gst/playback/gstinputselector.c:
5065 * gst/playback/gstinputselector.h:
5066 * gst/playback/gstplay-marshal.list:
5067 * gst/playback/gstplaybin2.c:
5068 playbin2: use private copy of input-selector
5069 We shouldn't really depend on elements from -bad for stream
5070 selection in playbin2, so use a private copy of input-selector
5071 until the selector plugin is ready to be moved to -base or -good.
5074 2009-07-15 17:26:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5076 * gst/playback/gstinputselector.c:
5077 * gst/playback/gstinputselector.h:
5078 playback: add private copy of the input-selector from gst-plugins-bad
5079 Not hooked up yet though. See #586356.
5081 2009-07-14 19:00:36 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
5083 * tests/examples/v4l/Makefile.am:
5084 examples: fix v4l probe example build
5087 2009-07-14 19:00:10 +0100 Jan Schmidt <thaytan@noraisin.net>
5121 0.10.23.2 pre-release
5123 2009-07-14 16:24:10 +0100 Jan Schmidt <thaytan@noraisin.net>
5127 Add Turkish translations
5129 2009-07-14 15:31:13 +0100 Jan Schmidt <thaytan@noraisin.net>
5131 * tests/check/elements/adder.c:
5132 adder: One more attempt to fix the adder test
5133 Give up and discard and recreate the alsasrc after checking it can
5134 be opened, due to some strange crash inside alsa when we don't.
5136 2009-07-14 15:06:41 +0100 Jan Schmidt <thaytan@noraisin.net>
5138 * tests/check/elements/adder.c:
5139 adder: Perform get_state() in the unit test
5140 Wait for the alsasrc to return to NULL after setting it to PAUSED for
5141 testing, otherwise it leads to segfaults later on.
5143 2009-07-14 14:39:32 +0100 Jan Schmidt <thaytan@noraisin.net>
5145 * tests/check/elements/adder.c:
5146 adder: Don't fail when alsasrc is unavailable
5147 Make the liveadder test succeed silently when it can't be completed
5148 either because alsasrc is unavailable, or because the device is
5151 2009-07-13 22:51:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5153 * gst-libs/gst/pbutils/descriptions.c:
5154 * gst/typefind/gsttypefindfunctions.c:
5155 typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest
5156 Differentiate subtitle streams and lyrics/cracktastic/complex streams via
5157 the category string in the headers. This seems like a useful distinction
5158 to make, and also seems more future-proof. See #525743.
5160 2009-02-21 13:18:10 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
5162 * ext/ogg/gstoggmux.c:
5163 oggmux: add Kate caps to the list of accepted types
5166 2009-07-13 21:56:46 +0300 Stefan Kost <ensonic@users.sf.net>
5168 * gst/playback/gsturidecodebin.c:
5169 uridecodebin: treat uri-schemas incasesensitive
5170 Treat uri-schemas incasesensitive. This is mandated in rfc2396 section 3.1.
5171 Fixes not showing buffering messages e.g. for HTTP://...
5173 2009-07-13 21:54:47 +0300 Stefan Kost <ensonic@users.sf.net>
5175 * gst-libs/gst/interfaces/navigation.c:
5176 navigation: simplify docs
5177 Make short-desc short - its used in the toc. Strip uneeded markup.
5179 2009-07-13 18:31:15 +0100 Jan Schmidt <thaytan@noraisin.net>
5181 * win32/common/libgstnetbuffer.def:
5182 * win32/common/libgstvideo.def:
5184 Remove methods from video base classes that have moved to -bad.
5185 Add gst_netaddress_to_string
5187 2009-07-13 17:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
5189 * tests/examples/gio/.gitignore:
5190 ignores: ignore the giosrc-mounting example binary
5192 2009-07-13 17:54:40 +0100 Jan Schmidt <thaytan@noraisin.net>
5194 * gst-libs/gst/interfaces/navigation.c:
5195 navigation: Add some partial documentation
5196 Add a general documentation blurb for the GstNavigation functionality.
5197 Still lacks some example code and detail on how to implement it.
5199 2009-07-13 17:52:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5201 * gst-libs/gst/pbutils/descriptions.c:
5202 pbutils: add description for Siren codec and make two descriptions non-translatable
5204 2009-07-13 12:23:20 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
5207 Automatic update of common submodule
5208 From 5845b63 to fedaaee
5210 2009-07-13 18:21:49 +0200 Elliott Sales de Andrade <quantum.analyst at gmail.com>
5212 * gst-libs/gst/riff/riff-ids.h:
5213 * gst-libs/gst/riff/riff-media.c:
5214 riff: add siren to the RIFF parser
5215 Add siren7 caps to the RIFF parser.
5217 2009-07-13 14:55:59 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
5220 * tests/examples/Makefile.am:
5221 * tests/examples/v4l/Makefile.am:
5222 * tests/examples/v4l/probe.c:
5223 v4lsrc: add a simple test case for device probing
5225 2009-07-03 11:38:01 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
5228 * sys/v4l/Makefile.am:
5229 * sys/v4l/gstv4lelement.c:
5230 v4lsrc: optional support for device probing with gudev
5231 Enumerate v4l devices using gudev if available.
5234 2009-07-10 23:24:36 +0100 Stefan Kost <ensonic@users.sf.net>
5236 * gst/adder/gstadder.c:
5237 adder: add since tags to docs
5239 2009-07-10 21:29:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5241 * tests/examples/seek/seek.c:
5242 seek: don't automatically start pipeline in DB
5243 Keep the pipeline paused when we detect download buffering. The user has to
5244 manually start the pipeline for now because we can't estimate when the buffering
5245 will finish or when we have underrun.
5247 2009-07-10 21:01:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5249 * gst/playback/gstqueue2.c:
5250 queue2: flush differently, avoiding deadlocks
5251 Don't flush the file by closing and opening it but instead use g_freopen. This
5252 avoids a deadlock in shutdown because we emit the temp-location property change
5253 with the wrong lock held.
5255 2009-07-10 20:25:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5257 * tests/examples/seek/seek.c:
5258 seek: add a checkbox for progressive download
5260 2009-07-10 20:24:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5262 * gst/playback/gsturidecodebin.c:
5263 uridecodebin: Fix template construction
5264 Fix the construction of the temporary filename construction as the application
5265 name can be NULL and we don't want a separator between the prgname and the
5268 2009-07-10 20:04:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5270 * gst/playback/gstplay-enum.c:
5271 * gst/playback/gstplay-enum.h:
5272 * gst/playback/gstplaybin2.c:
5273 playbin2: add support for progressive download
5274 Add a new playbin2 flag (initially disabled) to enable progressive download
5275 buffering in uridecodebin.
5277 2009-07-10 19:59:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5279 * gst/playback/gsturidecodebin.c:
5280 uridecodebin: add download property
5281 Add a download property that will attempt to configure queue2 into progressive
5283 Make sure we only enable download buffering for quicktime and flv formats.
5285 2009-07-10 19:49:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5287 * gst/playback/gstqueue2.c:
5288 queue2: add temp-template property
5289 Add a new temp-template property so that queue2 can securely allocate a
5290 temporary filename. Deprecate the temp-location property for setting the
5291 location but still use it to notify the allocated temp file.
5293 2009-07-10 20:06:28 +0100 Stefan Kost <ensonic@users.sf.net>
5295 * gst/adder/gstadder.c:
5296 * gst/adder/gstadder.h:
5297 adder: add a caps-property to avoid to need to plug a capsfilter afterwards
5298 Adder can only handle one common format accross the pads. Thus one needed to add
5299 a capsfilter afterwards and manage the caps. Now one can simply set the caps on
5302 2009-07-10 18:59:05 +0100 Stefan Kost <ensonic@users.sf.net>
5304 * tests/check/elements/adder.c:
5305 adder: skip live-seek text if we have no audiosrc, add new test
5306 The seek-test needs a real audiosrc. Also add a test that checks that adder is
5307 reusable. Finaly handle warnings as warnings to fix a assertion.
5309 2009-07-10 19:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5311 * ext/gio/gstgiosink.c:
5312 gio: Also post a "not-mounted" message from giosink
5314 2009-07-10 17:15:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5316 * tests/examples/gio/giosrc-mounting.c:
5317 gio: Remove workaround for playbin2 bug in the sample application
5318 The playbin2 bug was #588078.
5320 2009-07-10 17:08:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5322 * gst/playback/gstplaybin2.c:
5323 playbin2: Make it possible for READY->PAUSED to succeed after it failed the first time
5324 If READY->PAUSED failed in the source element we would've swapped
5325 the current and next group already. To allow READY->PAUSED to succeed
5326 after the first failure we have to swap the current and next group
5327 back again. This also ensure that we're again in the same state
5328 as before the failed state change and not at the next group.
5329 This was especially a problem for playbin2 pipelines that use the
5330 new mounting support in giosrc as the source would fail for READY->PAUSED
5331 the first time, the application mounts the location and then tries
5332 to go READY->PAUSED again (and this time it would succeed).
5335 2009-07-10 11:42:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5338 * tests/examples/Makefile.am:
5339 * tests/examples/gio/Makefile.am:
5340 * tests/examples/gio/giosrc-mounting.c:
5341 gio: Add example application that shows how to handle the "not-mounted" message
5343 2009-07-10 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5346 gio: Remove the experimental status from the GIO plugin
5349 2009-07-10 11:24:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5351 * ext/gio/gstgiosink.c:
5352 * ext/gio/gstgiosrc.c:
5353 gio: Add documentation for the new "not-mounted" and "file-exists" messages
5355 2009-07-09 13:45:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5357 * ext/gio/gstgiobasesrc.c:
5358 gio: Make sure that we have the correct stream position when starting
5360 2009-07-08 17:24:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5362 * ext/gio/gstgiobasesink.c:
5363 gio: Make sure to flush the output stream if it shouldn't be closed
5364 Otherwise there might still be unwritten data after the element
5367 2009-07-08 17:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5369 * ext/gio/gstgiobasesink.c:
5370 * ext/gio/gstgiobasesink.h:
5371 * ext/gio/gstgiobasesrc.c:
5372 * ext/gio/gstgiobasesrc.h:
5373 * ext/gio/gstgiosink.c:
5374 * ext/gio/gstgiosrc.c:
5375 gio: Don't close the GIO streams for the giostream{src,sink} elements
5376 This makes it possible to do something useful with the streams
5377 after the element has stopped. Fixes bug #587896.
5379 2009-07-08 17:19:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5381 * tests/check/pipelines/gio.c:
5382 gio: Try to reuse the pipeline with the same stream objects
5384 2009-07-08 17:02:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5386 * ext/gio/gstgiobasesink.c:
5387 * ext/gio/gstgiobasesrc.c:
5388 gio: Improve the error message if a stream is already closed before usage
5390 2009-07-08 16:55:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5392 * ext/gio/gstgiosink.c:
5393 gio: Post a custom file-exists message on the bus if the file already exists
5394 An application can handle this message, remove the file in question
5395 and restart the pipeline again without showing an error.
5396 This fixes bug #529300.
5398 2009-07-08 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5400 * ext/gio/gstgiosrc.c:
5401 gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted
5403 2009-07-08 16:50:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5405 * ext/gio/gstgiosink.c:
5406 gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink
5408 2009-07-08 15:52:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5410 * ext/gio/gstgiosrc.c:
5411 gio: Post a custom "not-mounted" message on the bus
5412 This allows applications to mount the GFile if possible and restart
5413 the pipeline instead of simply giving an error.
5415 2009-07-08 15:08:32 +0200 Philip Jägenstedt <philipj@opera.com>
5417 * gst/audioconvert/gstchannelmix.c:
5418 audioconvert: Fix compilation when debugging is disabled
5421 2009-07-07 20:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5423 * ext/gio/gstgiobasesink.c:
5424 * ext/gio/gstgiobasesink.h:
5425 * ext/gio/gstgiobasesrc.h:
5426 * ext/gio/gstgiosink.c:
5427 * ext/gio/gstgiosink.h:
5428 * ext/gio/gstgiostreamsink.c:
5429 * ext/gio/gstgiostreamsink.h:
5430 gio: Add vfunc for requesting the stream for the sinks too
5432 2009-07-07 20:21:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5434 * ext/gio/gstgiobasesink.c:
5435 * ext/gio/gstgiobasesink.h:
5436 * ext/gio/gstgiobasesrc.c:
5437 * ext/gio/gstgiosink.c:
5438 * ext/gio/gstgiosrc.c:
5439 * ext/gio/gstgiostreamsink.c:
5440 * ext/gio/gstgiostreamsrc.c:
5441 gio: Some more random cleanup
5443 2009-07-07 20:20:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5446 * ext/gio/gstgiobasesink.c:
5447 * ext/gio/gstgiobasesrc.c:
5448 * ext/gio/gstgiobasesrc.h:
5449 * ext/gio/gstgiosink.c:
5450 * ext/gio/gstgiosrc.c:
5451 * ext/gio/gstgiosrc.h:
5452 * ext/gio/gstgiostreamsink.c:
5453 * ext/gio/gstgiostreamsrc.c:
5454 * ext/gio/gstgiostreamsrc.h:
5455 gio: Update my mail address and copyright
5457 2009-07-07 20:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5459 * ext/gio/gstgiobasesrc.c:
5460 * ext/gio/gstgiobasesrc.h:
5461 * ext/gio/gstgiosrc.c:
5462 * ext/gio/gstgiostreamsrc.c:
5463 * ext/gio/gstgiostreamsrc.h:
5464 gio: General clean up and simplification
5465 The GInputStreams are now requested by a vfunc from
5466 the subclasses instead of relying that the subclass
5467 sets it until it's needed.
5468 This might also fix bug #587896.
5470 2009-07-06 22:31:12 +0100 Stefan Kost <ensonic@users.sf.net>
5472 * gst/adder/gstadder.c:
5473 adder: keep sending newsegments after seeking
5474 Adder sends with timestamps from 0 upwards. After seeking we need to send
5475 new-segments to get correct positions-queries.
5477 2009-07-06 20:44:00 +0100 Stefan Kost <ensonic@users.sf.net>
5479 * tests/check/elements/adder.c:
5480 adder: make test more robust
5481 Add audioconverts to the live-seeking test to make it negotiate.
5483 2009-06-30 17:19:50 +0300 Stefan Kost <ensonic@users.sf.net>
5485 * sys/xvimage/xvimagesink.c:
5486 xvimagesink: use core performance log category
5488 2009-07-05 21:29:40 +0200 Edward Hervey <bilboed@bilboed.com>
5490 * gst/adder/gstadder.c:
5491 adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped.
5492 This ensures that collectpads' cookie is properly updated so that when the streaming
5493 threads will restart and be checking for the flushing status of all pads there will
5494 be no inconsistent state.
5496 2009-07-05 18:01:38 +0200 Hans-Peter Nilsson <hp@gcc.gnu.org>
5498 * ext/pango/gstclockoverlay.c:
5499 pango: Call tzset() before localtime_r()
5500 POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't
5501 required to set the state variables that define the current timezone. Indeed,
5502 glibc (at least 2.9) doesn't do this for subsequent calls. The effect is that
5503 if the system timezone is changed for a running program between two calls to
5504 gst_clock_overlay_render_time, it won't be noticed. For glibc, changing the
5505 timezone equals /etc/localtime being modified.
5508 2009-07-01 17:33:14 -0700 David Schleef <ds@schleef.org>
5511 build: remove spurious schroedinger reference
5513 2009-07-01 10:25:43 -0700 David Schleef <ds@schleef.org>
5517 * ext/schroedinger/Makefile.am:
5518 * ext/schroedinger/gstschro.c:
5519 * ext/schroedinger/gstschrodec.c:
5520 * ext/schroedinger/gstschroenc.c:
5521 * ext/schroedinger/gstschroparse.c:
5522 * ext/schroedinger/gstschroutils.c:
5523 * ext/schroedinger/gstschroutils.h:
5524 * gst-libs/gst/video/Makefile.am:
5525 * gst-libs/gst/video/gstbasevideocodec.c:
5526 * gst-libs/gst/video/gstbasevideocodec.h:
5527 * gst-libs/gst/video/gstbasevideodecoder.c:
5528 * gst-libs/gst/video/gstbasevideodecoder.h:
5529 * gst-libs/gst/video/gstbasevideoencoder.c:
5530 * gst-libs/gst/video/gstbasevideoencoder.h:
5531 * gst-libs/gst/video/gstbasevideoparse.c:
5532 * gst-libs/gst/video/gstbasevideoparse.h:
5533 * gst-libs/gst/video/gstbasevideoutils.c:
5534 * gst-libs/gst/video/gstbasevideoutils.h:
5535 basevideo: send basevideo back to remedial school
5536 Move basevideo classes and schroedinger plugin to -bad.
5538 2009-07-01 12:54:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5540 * docs/libs/gst-plugins-base-libs-sections.txt:
5541 * gst-libs/gst/netbuffer/gstnetbuffer.h:
5542 netaddress: add constant for max len
5544 2009-07-01 12:48:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5546 * docs/libs/gst-plugins-base-libs-sections.txt:
5547 * gst-libs/gst/netbuffer/gstnetbuffer.c:
5548 * gst-libs/gst/netbuffer/gstnetbuffer.h:
5549 netbuffer: add gst_netaddress_to_string
5550 Add function to serialize a net address to a string.
5551 API: GstNetAddress::gst_netaddress_to_string()
5553 2009-06-30 18:44:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5555 * gst/playback/gsturidecodebin.c:
5556 uridecodebin: make fd:// uri use buffering too
5557 fd:// usually operate in push mode only and are thus suitable for buffering.
5559 2009-06-30 14:46:38 +0300 Stefan Kost <ensonic@users.sf.net>
5561 * gst/playback/gstplaybin2.c:
5562 * gst/volume/gstvolume.c:
5563 volume: include "1.0=100%" in property description
5565 2009-06-30 14:45:51 +0300 Stefan Kost <ensonic@users.sf.net>
5567 * gst/playback/gstplaysink.c:
5568 playsink: remove unused property defs
5570 2009-06-29 17:11:50 +0300 Stefan Kost <ensonic@users.sf.net>
5572 * gst-libs/gst/audio/multichannel.c:
5573 multichannel: rewrite the new doc comment a bit
5574 Its part of the audio lib.
5576 2009-06-29 14:34:02 +0100 Jan Schmidt <thaytan@noraisin.net>
5578 * gst/playback/gstplaysink.c:
5579 playsink: Avoid a segfault when the video sink fails to start
5580 Don't attempt to display the subpictures and segfault when the
5581 video sink failed to start (and hence the videochain is NULL).
5583 2009-06-29 15:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5585 * gst-libs/gst/audio/gstringbuffer.c:
5586 * gst-libs/gst/audio/gstringbuffer.h:
5587 ringbuffer: add vmethod to clear the ringbuffer
5588 Add a vmethod so that subclasses can be notified when they should clear the data
5591 2009-06-29 14:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
5593 * gst-libs/gst/riff/riff-media.c:
5594 riff-media: Fix the fourcc caps property for VC-1/WMVA
5595 The caps property for carrying fourccs is 'format', not 'fourcc'
5597 2009-06-29 12:20:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5599 * gst-libs/gst/rtsp/gstrtspconnection.c:
5600 rtsp: include in.h for FreeBSD compat
5603 2009-06-29 12:20:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5605 * win32/common/libgstapp.def:
5606 defs: add defs for new appsink buffer-list method
5608 2009-06-29 12:14:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5610 * gst-libs/gst/app/gstappsink.c:
5611 * gst-libs/gst/app/gstappsink.h:
5612 appsink: add docs and signals
5613 Add docs for the new callback.
5614 Add signals for the new buffer-list support.
5616 2009-06-29 10:24:36 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
5618 * tests/check/elements/appsink.c:
5619 Added unit tests for buffer list support in appsink.
5621 2009-06-17 11:12:08 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
5623 * gst-libs/gst/app/gstappsink.c:
5624 Added buffer list support.
5626 2009-06-17 09:23:11 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
5628 * gst-libs/gst/app/gstappsink.h:
5629 Added buffer list support.
5631 2009-06-29 09:36:27 +0200 Peter Kjellerstedt <pkj@axis.com>
5633 * gst-libs/gst/sdp/gstsdpmessage.c:
5634 sdp: Include winsock2.h after defining WINVER.
5635 Similar to bug #587080.
5637 2009-06-29 09:31:40 +0200 Peter Kjellerstedt <pkj@axis.com>
5639 * gst-libs/gst/rtsp/gstrtspconnection.c:
5640 rtsp: Moved a comment.
5642 2009-06-27 23:23:02 +0300 Stefan Kost <ensonic@users.sf.net>
5644 * gst-libs/gst/audio/audio.c:
5645 * gst-libs/gst/audio/multichannel.c:
5646 docs: add basic section docs for multichannel and relocate the ones for audio
5647 Add section docs for multichannel, so that it has a short desc in the toc too.
5648 Move the section docs in adio up, so that the follow the copyright like
5651 2009-06-26 21:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
5653 * sys/v4l/gstv4lelement.c:
5654 * sys/v4l/gstv4lsrc.c:
5655 v4l: open/close device in ready.
5656 Simillar change like in v4l2src. This allows probing feature in paused, where
5657 streaming is noit yet started.
5659 2009-06-10 17:05:22 +0300 René Stadler <rene.stadler@nokia.com>
5661 * gst/playback/gstplaysink.c:
5662 playbin2: fix initial volume handling also when reusing the element
5663 This is a follow-up to commit 452988, making it work correctly when the audio
5666 2009-06-26 21:48:58 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
5668 * gst-libs/gst/rtsp/gstrtspconnection.c:
5669 Define WINVER before including any win headers
5672 2009-06-27 00:50:54 +0300 René Stadler <mail@renestadler.de>
5674 * gst-libs/gst/riff/riff-read.c:
5675 riff: prevent crash if rounded up tag size exceeds data size
5676 When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
5677 and an invalid read past the buffer data follows.
5679 2009-06-26 15:17:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5681 * gst-libs/gst/video/gstbasevideocodec.c:
5682 basevideocodec: By default don't allow caps changes on the srcpad
5683 This fixed playback of Dirac files with schrodec when upstream wants
5684 a different width/height, basevideocodec accepts this and then
5685 pushes buffers with new caps but content of the old caps.
5686 In the best case this will just result in wrong unit size and a
5687 failure in basestransform elements.
5689 2009-06-26 14:11:21 +0100 Jan Schmidt <thaytan@noraisin.net>
5692 autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
5693 Check for more automake command variants. Use printf instead of 'echo -n'
5696 2009-06-26 13:41:38 +0100 Jan Schmidt <thaytan@noraisin.net>
5699 Automatic update of common submodule
5700 From f810030 to 5845b63
5702 2009-06-26 13:14:02 +0300 Stefan Kost <ensonic@users.sf.net>
5704 * gst/playback/gstscreenshot.c:
5705 screenshot: don't leak message
5707 2009-06-25 12:04:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5709 * gst/typefind/gsttypefindfunctions.c:
5710 typefinding: lower the h264 typefinder's probability
5711 A NEARLY_CERTAIN is absolutely not warranted given the kind
5712 of things it checks for. Even a LIKELY is probably not entirely
5715 2009-06-24 15:13:56 +0100 Jan Schmidt <jan.schmidt@sun.com>
5718 Automatic update of common submodule
5719 From f3bb51b to f810030
5721 2009-06-24 09:48:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5723 * gst-libs/gst/pbutils/descriptions.c:
5724 pbutils: add description for multipart
5725 So we get slightly nicer error messages when multipartdemux is missing.
5727 2009-06-23 18:07:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5729 * gst/adder/gstadder.c:
5730 adder: only unflush when we flushed before
5731 Ass suggested by Stefan Kost:
5732 Keep track of when the sinkpad was set to flushing and unflush the pad when an
5733 upstream flushing seek failed.
5735 2009-06-23 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5737 * gst/playback/gsturidecodebin.c:
5738 uridecodebin: fix leak when the source fails to change state
5740 2009-06-23 12:40:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5742 * gst/subparse/gstssaparse.c:
5743 ssaparse: avoid leaking all buffers
5745 2009-06-22 22:18:03 +0300 Stefan Kost <ensonic@users.sf.net>
5747 * tests/check/elements/adder.c:
5748 adder: test seek handling in adder
5749 This tests seeking on an adder that has a normal and a live source connected.
5750 Wheter the current behavior is the desired one needs to be discussed still
5753 2009-06-22 16:17:10 +0300 Stefan Kost <ensonic@users.sf.net>
5755 * sys/ximage/ximagesink.c:
5756 * sys/xvimage/xvimagesink.c:
5757 x(v)imagesink: pass the xwindow along to not look at the yet unset var.
5758 When we call this from xwindow_new, x(v)imagesink->xwindow is not yet set.
5760 2009-06-22 11:40:33 +0300 Stefan Kost <ensonic@users.sf.net>
5762 * sys/ximage/ximagesink.c:
5763 * sys/ximage/ximagesink.h:
5764 * sys/xvimage/xvimagesink.c:
5765 * sys/xvimage/xvimagesink.h:
5766 x(v)imagesink: catch tags and show title in own window
5767 Refactor the code that sets the window title. Catch tag-events and use title
5768 metadata for the window title.
5770 2009-06-21 19:42:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5772 * gst/audiotestsrc/gstaudiotestsrc.c:
5773 audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
5774 Also make all the function arrays constant.
5776 2009-06-21 12:27:37 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
5778 * gst/audiotestsrc/gstaudiotestsrc.c:
5779 * gst/audiotestsrc/gstaudiotestsrc.h:
5780 audiotestsrc: Add support for generating gaussian white noise
5781 This patch adds support for stationary white Gaussian noise.
5782 The Box-Muller algorithm is used to generate pairs of independent
5783 normally-distributed random numbers.
5786 2009-06-20 23:46:28 +0100 Jan Schmidt <thaytan@noraisin.net>
5788 * gst/ffmpegcolorspace/imgconvert.c:
5789 * gst/ffmpegcolorspace/imgconvert_template.h:
5790 ffmpegcolorspace: Fix NV12 and NV21 transformations
5791 Fix some stride problems, fix the nv12 to nv21 direct transformation,
5792 and implement a direct conversion to yuv444 to save CPU.
5794 2009-06-20 22:36:21 +0100 Jan Schmidt <thaytan@noraisin.net>
5796 * gst/videotestsrc/videotestsrc.c:
5797 videotestsrc: Fix NV12 painting for odd strides/heights
5799 2009-06-19 22:16:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5801 * ext/cdparanoia/gstcdparanoiasrc.c:
5802 cdparanoia: run-time license is LGPL now that we require cdparanoia 0.10.2
5803 cdparanoia has an LGPL v2.1 license since 0.10.1 and we now require 0.10.2.
5804 Finally fixes #531035.
5806 2009-06-19 21:25:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5808 * ext/cdparanoia/gstcdparanoiasrc.c:
5809 cdparanoia: try to guess a good cache size if it's set to -1
5810 Try to guess from the paranoia-mode setting whether playback or
5811 ripping is wanted, and use a smaller cache size if we're likely
5812 to be doing playback, to avoid a long startup delay. Since this
5813 was the value used in older cdparanoia versions, it should be
5814 fine in any case. See #586331.
5816 2009-06-19 11:27:40 +1000 Jonathan Matthew <jonathan@d14n.org>
5819 * ext/cdparanoia/gstcdparanoiasrc.c:
5820 * ext/cdparanoia/gstcdparanoiasrc.h:
5821 cdparanoia: expose cache size setting
5822 This setting was added in cdparanoia 10.2. The default value is good
5823 for audio extraction, but lower values (previous versions of cdparanoia
5824 used 150) are better for realtime playback.
5827 2009-06-19 17:43:03 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
5829 * gst-plugins-base.spec.in:
5830 Make build of schro plugin conditional
5832 2009-06-19 15:52:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5834 * docs/libs/gst-plugins-base-libs-sections.txt:
5835 * gst-libs/gst/rtp/gstbasertppayload.c:
5836 * gst-libs/gst/rtp/gstbasertppayload.h:
5837 * win32/common/libgstrtp.def:
5838 basertppayload: add support for bufferlists
5839 Based on patch from Ognyan Tonchev.
5842 2009-06-19 15:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5844 * gst-libs/gst/rtp/gstrtpbuffer.c:
5845 rtpbuffer: use new convenience functions
5846 New core convenience functions makes the list getters and setters trivial.
5847 Maybe even too trivial...
5849 2009-06-18 19:07:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5851 * win32/common/libgstrtp.def:
5852 defs: add new symbol to win32 defs file
5853 Based on patches by Ognyan Tonchev.
5856 2009-06-18 19:04:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5858 * docs/libs/gst-plugins-base-libs-sections.txt:
5859 * gst-libs/gst/rtp/gstrtpbuffer.c:
5860 rtp: cleanups, add _list_get_seq() too
5861 Clean up the docs a little.
5862 Add missing _list_get_seq method.
5863 Add new symbols to the docs
5865 2009-06-18 18:47:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5867 * gst-libs/gst/rtp/gstrtpbuffer.c:
5868 * win32/common/libgstrtp.def:
5870 Add Since tags to docs
5871 Move some code around
5874 2009-06-18 17:46:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5876 * gst-libs/gst/rtp/gstrtpbuffer.c:
5877 * gst-libs/gst/rtp/gstrtpbuffer.h:
5878 * tests/check/libs/rtp.c:
5879 rtp: add bufferlist support
5881 2009-06-18 18:03:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5883 * gst-libs/gst/rtp/gstrtpbuffer.c:
5884 rtp: pass data to macros instead of GstBuffer
5886 2009-06-18 17:42:10 +0100 Jan Schmidt <thaytan@noraisin.net>
5888 * win32/common/libgstrtsp.def:
5889 win32: Add gst_rtsp_watch_queue_data() to the exports
5890 Fix the tests by exporting the new symbol from the win32 dlls
5892 2009-06-18 18:13:22 +0300 Stefan Kost <ensonic@users.sf.net>
5894 * sys/xvimage/xvimagesink.c:
5895 xvimagesink: appname might be NULL
5896 Don't set title if appname is unknown.
5898 2009-06-18 17:58:06 +0300 Stefan Kost <ensonic@users.sf.net>
5900 * sys/xvimage/xvimagesink.c:
5901 xvimagesink: set window title from application name
5903 2009-06-09 19:14:00 +0200 Peter Kjellerstedt <pkj@axis.com>
5905 * gst-libs/gst/rtsp/gstrtspurl.c:
5906 rtsp: Made the parsing of the RTSP URL scheme more generic.
5908 2009-06-15 13:58:26 +0200 Peter Kjellerstedt <pkj@axis.com>
5910 * gst-libs/gst/rtsp/gstrtspconnection.c:
5911 * gst-libs/gst/rtsp/gstrtspconnection.h:
5912 rtsp: Added gst_rtsp_watch_queue_data().
5913 gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
5914 but allows for queuing any data block for writing (much like
5915 gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)
5916 API: gst_rtsp_watch_queue_data()
5918 2009-06-09 16:37:09 +0200 Peter Kjellerstedt <pkj@axis.com>
5920 * gst-libs/gst/rtsp/gstrtspconnection.c:
5921 rtsp: Only extract the session ID from RTSP responses.
5923 2009-06-09 19:06:57 +0200 Peter Kjellerstedt <pkj@axis.com>
5925 * gst-libs/gst/rtsp/gstrtspurl.c:
5926 rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
5928 2009-06-09 14:31:18 +0200 Peter Kjellerstedt <pkj@axis.com>
5930 * gst-libs/gst/rtsp/gstrtspconnection.c:
5931 rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
5933 2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
5935 * gst-libs/gst/rtsp/gstrtspconnection.c:
5936 rtsp: Improved base64 decoding in fill_bytes().
5937 The base64 decoding in fill_bytes() expected the size of the read data to
5938 be evenly divisible by four (which is true for the base64 encoded data
5939 itself). This did not, however, take whitespace (especially line breaks)
5940 into account and would fail the decoding if any whitespace was present.
5942 2009-06-17 14:00:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5944 * gst-libs/gst/audio/gstbaseaudiosrc.c:
5945 audiosrc: fix get_offset
5946 When we need to jump to the most recently captured sample, jump to where the
5947 next sample will be written instead of to some old data.
5950 2009-06-17 13:18:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5952 * gst-libs/gst/audio/gstbaseaudiosink.c:
5953 audiosink: free the ringbuffer when going to NULL
5954 Unparent and free the ringbuffer when going to NULL, like we do with the
5955 audiosrc element. We can do this now because we correctly manage the time
5958 2009-06-17 13:17:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5960 * gst-libs/gst/audio/gstaudiosink.c:
5961 * gst-libs/gst/audio/gstaudiosrc.c:
5962 audio: correctly handle short read/writes
5964 2009-05-05 15:37:54 +0300 René Stadler <rene.stadler@nokia.com>
5966 * gst-libs/gst/audio/gstbaseaudiosrc.c:
5967 baseaudiosrc: add some extra logging for buffer timestamps
5969 2009-06-17 11:22:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5971 * gst/adder/gstadder.c:
5972 adder: more seeking fixes.
5973 When a seek failed upstream, make sure the adder sinkpad is set unflushing again
5974 so that streaming can continue.
5975 We only have a pending segment when we flushed.
5976 Set the flush_stop_pending flag inside the appropriate locks and before we
5977 attempt to perform the upstream seek.
5978 Add some more comments.
5979 Use the right lock to protect the flags in flush_stop.
5982 2009-06-17 07:24:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5984 * gst/playback/gstdecodebin2.c:
5985 decodebin2: Free iterator after removing all groups
5987 2009-06-16 19:38:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5989 * gst-libs/gst/video/gstvideofilter.c:
5990 videofilter: Add a default get_unit_size function
5991 This returns the correct values for all formats that are handled by
5992 GstVideoFormat and makes all the custom get_unit_size functions in
5993 many elements unnecessary.
5995 2009-06-16 18:57:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5997 * gst-libs/gst/rtsp/gstrtspdefs.c:
5998 * gst-libs/gst/rtsp/gstrtspdefs.h:
5999 rtsp: add Timestamp header field
6002 2009-06-16 18:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6004 * gst/playback/gstplaybin2.c:
6005 playbin2: set smarter target state on uridecodebin
6006 Set the target state of the newly added uridecodebins to somthing else that
6007 PAUSED so that we keep their state in sync with the playsink state.
6010 2009-06-16 18:13:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6012 * gst/playback/gstplaysink.c:
6013 playsink: set the sink flag on the element
6015 2009-06-16 18:09:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6017 * gst/playback/gsturidecodebin.c:
6018 uridecodebin: add debug message
6020 2009-06-16 14:05:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6022 * gst-libs/gst/audio/gstaudiosink.c:
6023 * gst-libs/gst/audio/gstaudiosrc.c:
6024 audiosink, audiosrc: do the class_ref()s in the right class_init functions
6025 Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
6027 2009-06-15 15:39:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6029 * gst-libs/gst/audio/gstaudiosink.c:
6030 * gst-libs/gst/audio/gstaudiosrc.c:
6031 audiosink,audiosrc: ref the audio ring buffer class and type in class_init
6032 Hack around thread-safety issues in GObject and our racy _get_type()
6033 functions (we could easily fix the _get_type() functions, but we still
6034 need to hack around the GObject class races until we require a newer
6035 GLib version, I think).
6037 2009-06-15 12:57:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6039 * gst-libs/gst/audio/gstbaseaudiosrc.c:
6040 audiosrc: return FALSE when receiving a SEEK event
6041 When receiving a seek event, return FALSE as we don't implement seeking.
6043 2009-06-15 11:06:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6045 * tests/examples/seek/seek.c:
6046 Don't use deprecated GTK API
6049 2009-06-15 11:40:00 +0300 Stefan Kost <ensonic@users.sf.net>
6051 * gst/adder/gstadder.c:
6052 adder: send flush_stop when seeking failed
6053 At least do the fix to sent the flush_stop when seeking failed to ensure we
6054 keep no pads flushing. before it was send when the seeking worked which is just
6055 plain wrong and was not the intention.
6057 2009-06-12 15:17:14 +0200 Peter Kjellerstedt <pkj@axis.com>
6059 * gst-libs/gst/rtsp/gstrtspconnection.c:
6060 rtsp: Use a more consistent naming of GstRTSPRec variables.
6062 2009-06-12 15:11:05 +0200 Peter Kjellerstedt <pkj@axis.com>
6064 * gst-libs/gst/rtsp/gstrtspconnection.c:
6065 * gst-libs/gst/rtsp/gstrtspconnection.h:
6066 rtsp: Call message_sent() callback for all sent messages.
6067 Previously the messages_sent() callback was only called for messages
6068 which had a CSeq, which excluded all data messages. Instead of using the
6069 CSeq as ID, use a simple index counter.
6071 2009-06-14 22:13:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6073 * ext/ogg/gstoggdemux.c:
6074 * ext/theora/theoradec.c:
6075 * ext/vorbis/vorbisdec.c:
6076 oggdemux: post/send tags with the container-format tag
6077 For this to work properly, theoradec and vorbisdec need to put
6078 tag events received from upstream into the pending_events list
6079 so they get pushed out after any newsegment event, not before.
6081 2009-06-14 20:30:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6083 * tests/examples/seek/scrubby.c:
6084 * tests/examples/seek/seek.c:
6085 * tests/old/examples/seek/cdplayer.c:
6086 Don't use deprecated GTK API
6089 2009-06-12 16:31:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6091 * gst/adder/gstadder.c:
6092 adder: send flush-stop earlier
6093 When no flush-stop has been sent by upstream, we have to send one ourselves to
6094 continue playback. Do this as soon as the collect function is called instead of
6095 after we possibly pushed segment events (that got then flushed out)
6097 2009-06-12 13:55:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6099 * tests/examples/seek/seek.c:
6100 seek: add shuttle controls
6102 2009-06-12 13:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6104 * tests/examples/seek/stepping2.c:
6105 example: fix compile
6107 2009-06-12 13:52:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6109 * tests/examples/seek/Makefile.am:
6110 examples: build the stepping2 example
6112 2009-06-12 13:52:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6114 * gst/playback/gstplaysink.c:
6115 playsink: update for new step API
6117 2009-06-12 13:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6119 * ext/ogg/gstoggdemux.c:
6120 oggdemux: do reverse seeks more accurate
6121 For reverse seeking with the accurate flag set, try to be more precise by
6122 seeking a little bit after the requested position.
6124 2009-06-11 22:32:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6126 * ext/ogg/gstogmparse.c:
6127 * gst/subparse/gstssaparse.c:
6128 * gst/subparse/gstssaparse.h:
6129 * gst/subparse/gstsubparse.c:
6130 * gst/subparse/gstsubparse.h:
6131 subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC
6132 Make subtitle parsers post a taglist with codec tags, so the application
6133 knows what kind of subtitle a subtitle stream is. Fixes #576552.
6135 2009-06-11 19:12:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6137 * gst-libs/gst/audio/gstringbuffer.c:
6138 ringbuffer: handle border cases in resampler
6140 2009-06-11 13:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
6143 * docs/libs/Makefile.am:
6144 * docs/plugins/Makefile.am:
6145 docs: Update common. Use upload-doc.mak instead of upload.mak
6147 2009-06-11 12:39:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6149 * gst-libs/gst/rtp/gstbasertppayload.c:
6152 2009-06-11 12:17:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6154 * gst-libs/gst/audio/gstbaseaudiosink.c:
6155 baseaudiosink: reset accum when dropping samples
6156 When we are resampling and we drop samples because we paused, reset the accum
6157 counter because it's now invalid.
6159 2009-06-11 11:16:15 +0100 Jan Schmidt <thaytan@noraisin.net>
6161 * docs/libs/gst-plugins-base-libs-sections.txt:
6162 * gst-libs/gst/interfaces/mixer.h:
6163 * gst-libs/gst/video/gstbasevideodecoder.h:
6164 docs: Fix a couple of warnings from the docs build.
6166 2009-06-10 21:36:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6168 * gst-libs/gst/audio/testchannels.c:
6169 Don't include config.h multiple times when build audio testchannel app.
6170 Fixes build problem on win32 (#585075).
6172 2009-06-10 16:56:51 +0100 Jan Schmidt <thaytan@noraisin.net>
6174 * gst/playback/gstplaybin2.c:
6175 * gst/playback/gsturidecodebin.c:
6176 playbin2/uridecodebin: Fix connection-speed propagation
6177 uridecodebin expects the passed connection-speed value in kbps, so we
6178 need to divide the value stored in bps by 1000. Also, lower the upper
6179 limit on the properties to the value that we can actually store in our
6180 internal guint (which is plenty high enough)
6182 2009-06-10 14:37:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6184 * gst/subparse/gstsubparse.c:
6185 * tests/check/elements/subparse.c:
6186 subparse: recognise more subrip timestamp variants
6187 Be even less restrictive in what we accept for .srt timestamps when
6188 typefinding and parsing subrip subtitles and add a unit test for
6189 the 'new' format. Fixes #585197.
6191 2009-06-09 22:00:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6193 * gst-libs/gst/rtsp/gstrtsptransport.h:
6194 rtsp: add some more docs
6196 2009-06-09 18:24:55 +0200 Peter Kjellerstedt <pkj@axis.com>
6198 * gst-libs/gst/rtsp/gstrtspmessage.c:
6199 rtsp: Avoid a compiler warning.
6201 2009-06-09 18:23:28 +0200 Peter Kjellerstedt <pkj@axis.com>
6203 * gst-libs/gst/rtsp/gstrtspdefs.h:
6204 rtsp: Updated documentation for GstRTSPResult.
6205 Moved GST_RTSP_ELAST to be last in the documentation to match the actual
6208 2009-05-20 17:30:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6211 autogen: remove -Wno-portability from here
6212 as it is in configure.ac now.
6214 2009-06-09 16:28:20 +0200 Peter Kjellerstedt <pkj@axis.com>
6216 * gst-libs/gst/rtsp/gstrtspconnection.c:
6217 rtsp: Plug a memory leak.
6218 Free memory related to any partially read and/or written RTSP messages.
6220 2009-06-09 12:09:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6222 * gst-libs/gst/audio/gstbaseaudiosink.c:
6223 baseaudiosink: no need to cause discont when clipping
6224 Remove the discont-when-clipping hack now that basesink provides us with
6225 correctly clipped samples when stepping.
6227 2009-06-08 17:26:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6229 * gst-libs/gst/audio/gstbaseaudiosink.c:
6230 audiosink: don't align when we clip
6231 Don't align samples when they were clipped. Not entirely correct but better than
6234 2009-06-08 16:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6236 * tests/examples/seek/.gitignore:
6237 * tests/examples/seek/stepping2.c:
6238 examples: add stepping example in PLAYING
6239 Add stepping example in PLAYING, audio is a bit distorted because basesink does
6240 not provide good clipping info yet.
6242 2009-06-08 10:25:00 +0200 Edward Hervey <bilboed@bilboed.com>
6244 * gst-libs/gst/pbutils/descriptions.c:
6245 pbutils: Add description for hdv/aux-* formats.
6247 2009-06-07 22:20:33 +0400 LRN <lrn1986@gmail.com>
6249 * ext/schroedinger/Makefile.am:
6250 Added libgstbase to schro's LIBADD
6253 2009-06-06 02:15:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6255 * gst-libs/gst/tag/gstid3tag.c:
6256 libgsttag: don't extract genres from empty ID3v1 tags
6257 If we don't have any other info, don't try to interpret the
6258 genre field. In particular we don't want to interpret a genre
6259 of 0 as 'Blues' if no other fields are set and the entire tag
6262 2009-06-05 18:13:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6264 * gst/playback/gstdecodebin2.c:
6265 decodebin2: make sure varargs are of right type
6266 Explicitly cast the variables to g_object_set to their right types.
6268 2009-06-05 16:49:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6270 * gst/playback/gstdecodebin2.c:
6271 decodebin2: increase stream probing queues
6272 When we are probing for streams, we want to set the queue size in such a way
6273 that we can scan a maximum amount of data without consuming too much memory.
6274 Therefore, remove the time limit on the queue and only stop scanning after 2MB
6278 2009-06-05 14:06:17 +0200 Peter Kjellerstedt <pkj@axis.com>
6280 * gst-libs/gst/rtsp/gstrtspconnection.c:
6283 2009-06-05 14:05:54 +0200 Peter Kjellerstedt <pkj@axis.com>
6285 * gst-libs/gst/rtsp/gstrtspconnection.c:
6286 rtsp: Remove an unused variable.
6288 2009-06-05 13:59:14 +0200 Peter Kjellerstedt <pkj@axis.com>
6290 * gst-libs/gst/rtsp/gstrtspconnection.c:
6291 rtsp: Removed duplicate initialization of conn->writefd.
6293 2009-06-05 13:55:08 +0200 Peter Kjellerstedt <pkj@axis.com>
6295 * gst-libs/gst/rtsp/gstrtspconnection.c:
6296 rtsp: Use #defined status codes.
6298 2009-06-05 13:53:29 +0200 Peter Kjellerstedt <pkj@axis.com>
6300 * gst-libs/gst/rtsp/gstrtspconnection.c:
6301 rtsp: Correct gen_tunnel_reply().
6302 Prevent gen_tunnel_reply() from generating an incomplete response
6303 in case an error response code is given.
6305 2009-06-05 10:57:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6308 * win32/common/_stdint.h:
6309 * win32/common/config.h:
6310 * win32/common/video-enumtypes.c:
6311 configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
6312 See #584835. Also update win32 files while we're at it.
6314 2009-06-04 08:57:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6316 * gst/playback/gstplaybin2.c:
6317 playbin2: API: Add {audio,video,text}-tags-changed signals
6320 2009-06-03 20:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6322 * ext/vorbis/vorbisdec.c:
6323 vorbisdec: don't put invalid bitrate values into the taglist
6324 Bitrates are stored as 32-bit signed integers in the vorbis
6325 identification headers, but seem to be read incorrectly,
6326 namely as unsigned 32-bit integers, into the vorbis structure
6327 members which are of type long, which makes our check for
6328 values <= 0 fail with files that put -1 in there for unset
6331 2009-06-03 15:52:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6333 * tests/examples/seek/.gitignore:
6334 ignore: add new stepping app to ignore
6336 2009-06-03 15:31:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6338 * tests/examples/seek/Makefile.am:
6339 * tests/examples/seek/stepping.c:
6340 examples: add stepping example.
6341 Add an example of using playbin2 and frame stepping to simulate variable rate
6342 playback based on a sine wave.
6344 2009-06-03 12:45:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6346 * gst/playback/gstplaybin2.c:
6347 * gst/playback/gstplaysink.h:
6348 playbin2: also set custom text and subp sinks
6349 Set the custom subpicture and text sinks along with the custom audio and video
6351 Fix a little docs blurb too.
6353 2009-06-02 12:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6355 * gst-libs/gst/rtsp/gstrtspconnection.c:
6356 * gst-libs/gst/rtsp/gstrtspconnection.h:
6357 rtsp: add G_LIKELY because we can
6359 2009-06-02 09:53:05 +0200 Edward Hervey <bilboed@bilboed.com>
6361 * gst/typefind/gsttypefindfunctions.c:
6362 typefindfunctions: Fix caps for ogg typefinder.
6364 2009-05-29 11:10:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6366 * docs/libs/gst-plugins-base-libs-sections.txt:
6367 docs: remove some cruft from -sections.txt file
6369 2009-06-01 11:31:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6371 * gst/playback/gstplaysink.c:
6372 * tests/examples/seek/seek.c:
6373 add framestepping to playbin2 and seek
6375 2009-06-01 09:59:22 +0200 Peter Kjellerstedt <pkj@axis.com>
6377 * gst-libs/gst/rtsp/gstrtspconnection.c:
6378 rtsp: Avoid compiler warnings with -Wextra.
6380 2009-06-01 09:58:27 +0200 Peter Kjellerstedt <pkj@axis.com>
6382 * gst-libs/gst/rtsp/gstrtspconnection.h:
6383 rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
6385 2009-06-01 09:43:04 +0200 Peter Kjellerstedt <pkj@axis.com>
6387 * gst-libs/gst/sdp/gstsdpmessage.c:
6388 sdp: Remove an unused variable.
6390 2009-05-30 14:17:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6392 * gst/ffmpegcolorspace/imgconvert.c:
6393 * gst/ffmpegcolorspace/imgconvert_template.h:
6394 ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale
6396 2009-05-29 00:09:15 +0100 Jan Schmidt <thaytan@noraisin.net>
6398 * gst/playback/gstplaybin2.c:
6399 playbin2: Have playbin recognise PGS subpicture streams
6400 Recognise PGS subpicture streams and connect them to the SPU pad
6401 in playsink. Unfortunately this fails badly with negotiation errors
6402 if the SPU is not recent enough to support the stream. I'm not sure
6403 how to add format negotiation in yet.
6405 2009-05-21 23:11:29 +0100 Jan Schmidt <thaytan@noraisin.net>
6407 * gst/playback/gstdecodebin2.c:
6408 * gst/playback/gsturidecodebin.c:
6409 decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.
6411 2009-05-28 20:37:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6413 * gst/playback/gstplaysink.c:
6414 playbin2: fix volume handling for audio sinks without "volume" property
6415 When using an audio sink without a "volume" property, volume control
6416 would only work for the first song. For the next song, we'd try to
6417 re-use the existing audio chain, but inadvertently set chain->volume
6418 to NULL instead of to the existing volume element.
6420 2009-05-28 17:05:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6422 * gst/playback/gstplaysink.c:
6423 playbin2: cosmetic change to avoid unnecessary line breaks
6424 Looks nicer and works around gst-indent silliness.
6426 2009-05-28 17:21:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6428 * gst/playback/gstplaysink.c:
6429 playbin2: don't lose the ref to the volume element
6430 Only release the ref to the volume element when it is controled by a sink. For
6431 software volume we never have to fear that it will change.
6433 2009-05-28 15:21:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6435 * gst/playback/gstplaybin2.c:
6436 * gst/playback/gstplaysink.c:
6437 playbin2: actually use configured audio/video sinks
6438 playbin2 inadvertently used autoaudiosink and autovideosink up to now,
6439 since it would overwrite the sinks configured via the "audio-sink"
6440 and "video-sink" properties with the stream-specific group sinks when
6441 configuring the outputs. Those are usually NULL however, so that would
6442 overwrite the configured sinks with NULL which makes playbin2 then
6443 default to the auto sinks. Fix this by keeping a reference to each
6444 configured sink in playbin2 and setting up the right sinks depending
6445 on whether there is a stream-specific sink or not.
6448 2009-05-27 17:37:38 +0300 Stefan Kost <ensonic@users.sf.net>
6450 * tests/examples/seek/seek.c:
6451 seek: add volume label and sync with sink volume
6452 Look at the volume and have the pulsemixer open at same time. Unfortunately
6453 playbin2 does not emit notify on volume right, so this polls for now.
6455 2009-05-27 18:12:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6457 * gst/playback/gstdecodebin2.c:
6458 decodebin2: remove leftover elements
6459 Remove all of the elements inside decodebin2 when goint to READY and NULL.
6460 Makes decodebin2 reusable.
6463 2009-05-27 15:36:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6465 * gst/playback/gstplaysink.c:
6466 playbin2; release refs to volume/mute properties
6467 Release the refs to the volume and mute property elemens before setting the
6468 child elements to READY or NULL.
6471 2009-05-27 12:10:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6473 * gst/gdp/gstgdppay.c:
6474 gdppay: set caps on outgoing buffers
6475 Set caps on outgoing buffers because NULL caps confuse basetransform.
6478 2009-05-27 11:08:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6480 * gst-libs/gst/netbuffer/gstnetbuffer.c:
6481 netbuffer: also note the order of IP4 addresses
6482 IP4 addresses are also stored in network byte order. Make a note of this in the
6485 2009-05-26 22:43:34 +0200 Alessandro Decina <alessandro.d@gmail.com>
6487 * ext/theora/theoraparse.c:
6488 theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.
6490 2009-05-26 11:13:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6492 * gst-libs/gst/rtsp/gstrtspconnection.c:
6493 Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14"
6494 This reverts commit 418760cf740332c12c3fd9cf3244af134fa9534b.
6495 We now require GLib 2.16.
6497 2009-05-26 15:18:09 +0100 Jan Schmidt <thaytan@noraisin.net>
6502 2009-05-26 15:37:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6504 * gst-libs/gst/netbuffer/gstnetbuffer.c:
6505 netbuffer: document that the port is network order
6506 Document the fact that we store the port number in network order in
6507 GstNetAddress and that the caller should byteswap appropriately.
6509 2009-05-26 15:23:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6511 * gst/videoscale/gstvideoscale.c:
6512 * gst/videoscale/vs_4tap.c:
6513 * gst/videoscale/vs_4tap.h:
6514 * gst/videoscale/vs_image.c:
6515 * gst/videoscale/vs_image.h:
6516 * gst/videoscale/vs_scanline.c:
6517 * gst/videoscale/vs_scanline.h:
6518 videoscale: Add support for 16 bit grayscale in native endianness
6520 2009-05-26 14:58:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6522 * gst/ffmpegcolorspace/avcodec.h:
6523 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
6524 * gst/ffmpegcolorspace/imgconvert.c:
6525 ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian
6527 2009-05-26 14:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6529 * gst/videotestsrc/videotestsrc.c:
6530 * gst/videotestsrc/videotestsrc.h:
6531 videotestsrc: Add support for 16 bit grayscale in native endianness
6533 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
6535 add can-activate-pull property to baseaudiosink
6536 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
6539 2009-05-26 13:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6541 * ext/ogg/gstoggdemux.c:
6542 oggdemux: fix boundary case for seeking.
6543 When we have exactly 0 bytes left to search, make sure we stop instead of going
6544 into an infinite loop.
6546 2009-05-26 11:11:03 +0200 Bastien Nocera <hadess at hadess.net>
6548 * gst-libs/gst/cdda/Makefile.am:
6549 * gst-libs/gst/cdda/gstcddabasesrc.c:
6550 * gst-libs/gst/cdda/sha1.c:
6551 * gst-libs/gst/cdda/sha1.h:
6552 cddabasesrc: Remove copy of sha1 digest
6553 Remove our copy of sha1 digest now that we depend on glib 2.16.
6556 2009-05-25 17:54:01 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
6558 * gst-plugins-base.spec.in:
6561 2009-05-23 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6563 * gst-libs/gst/video/gstbasevideodecoder.c:
6564 * gst-libs/gst/video/gstbasevideoparse.c:
6565 * gst-libs/gst/video/gstbasevideoutils.c:
6566 * gst-libs/gst/video/gstbasevideoutils.h:
6567 * win32/common/libgstvideo.def:
6568 video: don't expose internal gst_adapter_get_buffer() helper function
6569 If it's really needed it should go into GstAdapter in core.
6571 2009-05-22 21:29:51 -0700 David Schleef <ds@schleef.org>
6573 * gst-libs/gst/video/gstbasevideodecoder.c:
6574 basevideo: Fix memleak
6576 2009-05-22 21:27:58 -0700 David Schleef <ds@schleef.org>
6578 * ext/schroedinger/gstschrodec.c:
6579 * ext/schroedinger/gstschroparse.c:
6580 schro: Fix usage of adapter_masked_scan_uint32
6581 Because *somebody* changed the API without telling me.
6583 2009-05-22 21:25:06 -0700 David Schleef <ds@schleef.org>
6585 * ext/schroedinger/gstschro.c:
6586 schro: Change package name to GST_PACKAGE_NAME
6588 2009-05-22 17:34:10 -0700 David Schleef <ds@schleef.org>
6590 * gst-libs/gst/video/gstbasevideoencoder.c:
6591 basevideo: Add preset interface to encoder
6593 2009-05-22 17:31:14 -0700 David Schleef <ds@schleef.org>
6595 * gst/audioresample/gstaudioresample.c:
6596 Run liboil benchmark multiple times
6597 The statistics function requires multiple runs, otherwise
6598 it causes a divide by zero error.
6600 2009-05-22 19:36:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6602 * m4/gst-fionread.m4:
6603 m4: fix 'suspicious cache value' warning for gst-fionread.m4
6604 .. here as well (should really be moved to common, but I'm too lazy).
6606 2009-05-22 17:41:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6608 * ext/vorbis/vorbisdec.c:
6609 vorbisdec: detect and report errors better
6610 Check the return values of a couple more libvorbis functions and post an error
6611 when something is wrong instead of continuing and crashing.
6613 2009-05-22 15:49:14 +0300 Stefan Kost <ensonic@users.sf.net>
6615 * gst/playback/gstplaysink.c:
6616 playbin2: fix initial volume and mute handling
6617 Use two flags to remember volume/mute changes at times when we don't have the
6618 audiochain yet (e.g. construction). Only set values when they were actualy
6619 changed. This makes pulseaudio's stream restore functional.
6621 2009-05-22 10:19:51 +0100 Jan Schmidt <thaytan@noraisin.net>
6624 Automatic update of common submodule
6625 From d3a8fab to 888e0a2
6627 2009-05-22 09:03:22 +0100 Jan Schmidt <thaytan@noraisin.net>
6629 * win32/common/libgstvideo.def:
6630 win32: Remove gst_adapter_masked_scan_uint32 from the exports
6632 2009-05-21 10:48:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6634 * gst-libs/gst/audio/gstbaseaudiosink.c:
6635 audiosink: improve debug message
6637 2009-05-19 18:10:55 -0700 Michael Smith <msmith@songbirdnest.com>
6639 * gst-libs/gst/tag/gstid3tag.c:
6640 gstid3tag: Don't extract a track number unless present.
6641 In ID3v1, a track number is present only if byte 125 is null AND
6642 byte 126 is non-null. If the track number is not present, don't add
6643 a track number tag with value 0.
6645 2009-05-20 00:48:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6647 * gst-libs/gst/video/gstbasevideoutils.c:
6648 * gst-libs/gst/video/gstbasevideoutils.h:
6649 videoutils: remove adapter methods
6650 Remove adapter methods now that they are in core.
6652 2009-05-20 00:42:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6654 * win32/common/libgstvideo.def:
6655 defs: add new symbols
6657 2009-05-19 17:47:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6660 autogen: pass -Wno-portability to automake to suppress warnings
6663 2009-05-19 02:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6665 * docs/libs/.gitignore:
6666 gitignore: remove bogus *.sgml wildcard - these files are tracked in git
6668 2009-05-19 18:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6670 * gst/tcp/gsttcpclientsrc.c:
6671 tcpclientsrc: this is not a live source
6672 Don't mark us as a live source because we are not.
6674 2009-05-19 18:41:02 +0300 Stefan Kost <ensonic@users.sf.net>
6676 * gst/adder/gstadder.c:
6677 adder: only send flush_stop when seek failed
6678 This is still not the ultimate fix. Added some comment to explain the troubles.
6680 2009-05-19 17:17:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6682 * gst-libs/gst/audio/gstbaseaudiosink.c:
6683 audiosink: return the return value of wait_preroll
6684 Return the value that _wait_preroll() returned instead of always WRONG_STATE.
6686 2009-05-19 16:45:56 +0300 Stefan Kost <ensonic@users.sf.net>
6688 * gst/adder/gstadder.c:
6689 * gst/adder/gstadder.h:
6690 adder: send flush_stop to match flush_start
6691 Adder was relying that something else sends a flush stop. When using adder with
6692 a livesource it was not getting a flush_stop and thus all pads downstream where
6693 keept flushing. Mark a pending flush_stop and send it when we are working on
6694 the new segment back in the streaming thread.
6696 2009-05-19 16:02:44 +0300 Stefan Kost <ensonic@users.sf.net>
6698 * tests/examples/seek/seek.c:
6699 seek: ui improvements
6700 Repaint the window black on expose, as this looks nicer when resizing or using
6701 the expander. Also show time after slider, as this saves a whole line (nice on
6704 2009-04-29 18:36:17 +0300 Stefan Kost <ensonic@users.sf.net>
6706 * gst/playback/gstdecodebin.c:
6707 decodebin: use iterators instead of list
6708 The list api is deprecated. Use threadsafe iterators instead.
6710 2009-05-19 15:35:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6712 * gst/playback/gsturidecodebin.c:
6713 uridecodebin: configure caps on decodebin2
6714 Implement the caps property by setting the configured caps on new decodebin2
6718 2009-05-19 15:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6720 * gst/playback/gstdecodebin2.c:
6721 decodebin2: avoid some _caps_ref in some cases
6722 Only mess with the caps refcount when we configure different caps.
6724 2009-05-19 15:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6726 * gst/playback/gsturidecodebin.c:
6727 uridecodebin: fix potential caps leak
6728 Free the user-configured caps in finalize.
6730 2009-05-19 15:20:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6732 * gst/playback/gsturidecodebin.c:
6733 uridecodebin: add queue after cdda://
6734 Add a queue2 after the raw output pads of certain sources such as those for uris
6736 No tuning of the queue is done yet as the defaults seem to work fine for me.
6739 2009-05-19 12:45:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6741 * ext/ogg/gstoggdemux.c:
6742 oggdemux: don't loop when at EOS
6743 When we try to read the last page, don't try to read past the upper boundary, as
6744 this might cause endless loops.
6747 2009-05-19 11:20:19 +0200 Edward Hervey <bilboed@bilboed.com>
6749 * gst/audioresample/gstaudioresample.c:
6750 audioresample: Don't drain remaining buffers after a flush.
6751 If we were resetted (due to a flush), we can not drain the remaining
6752 buffers since they would be pushed before a valid new newsegment event.
6754 2009-05-18 22:29:07 -0700 Michael Smith <msmith@syncword.(none)>
6756 * ext/theora/theoradec.c:
6757 theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.
6759 2009-05-19 01:13:34 +0300 Stefan Kost <ensonic@users.sf.net>
6761 * gst/adder/gstadder.c:
6762 adder: add more logging and return value checking
6764 2009-05-19 01:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
6766 * gst/adder/gstadder.c:
6767 adder: handle the return value from iterator_fold
6769 2009-05-19 01:03:44 +0300 Stefan Kost <ensonic@users.sf.net>
6771 * gst/adder/gstadder.c:
6772 adder: use the pad in logging as objects
6773 Helps to differenciate between source and sinks pads.
6775 2009-04-21 22:54:19 +0300 Stefan Kost <ensonic@users.sf.net>
6777 * tests/examples/seek/seek.c:
6778 seek: use parser for mp3 and rename variable
6780 2009-05-18 11:08:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6782 * tests/examples/seek/seek.c:
6783 seek: add playbin2 options in expander
6784 Add the playbin2 stream selection options inside an expander to preserve some
6787 2009-02-10 15:29:10 -0800 David Schleef <ds@schleef.org>
6789 * gst/videotestsrc/videotestsrc.c:
6790 videotestsrc: Add support for v210 and v216 formats
6792 2009-05-15 16:21:15 -0700 David Schleef <ds@schleef.org>
6794 * gst-libs/gst/video/gstbasevideocodec.c:
6795 * gst-libs/gst/video/gstbasevideodecoder.c:
6796 * gst-libs/gst/video/gstbasevideoencoder.c:
6797 * gst-libs/gst/video/gstbasevideoparse.c:
6798 video: remove // comments
6800 2009-05-15 16:18:18 -0700 David Schleef <ds@schleef.org>
6802 * gst-libs/gst/video/video.c:
6803 * gst-libs/gst/video/video.h:
6804 video: Add Y444, v210, v216 formats
6806 2009-05-15 16:12:37 -0700 David Schleef <ds@schleef.org>
6810 * ext/schroedinger/Makefile.am:
6811 * ext/schroedinger/gstschro.c:
6812 * ext/schroedinger/gstschrodec.c:
6813 * ext/schroedinger/gstschroenc.c:
6814 * ext/schroedinger/gstschroparse.c:
6815 * ext/schroedinger/gstschroutils.c:
6816 * ext/schroedinger/gstschroutils.h:
6817 schro: Move schro plugin from Schroedinger
6818 Previous history is in Schroedinger. Depends on, and is an example
6819 of using, GstBaseVideo* base classes.
6820 Code was reindented, and an #ifdef HAVE_ENCODER removed.
6822 2009-05-15 10:23:08 -0700 David Schleef <ds@schleef.org>
6824 * gst-libs/gst/video/Makefile.am:
6825 * gst-libs/gst/video/gstbasevideocodec.c:
6826 * gst-libs/gst/video/gstbasevideocodec.h:
6827 * gst-libs/gst/video/gstbasevideodecoder.c:
6828 * gst-libs/gst/video/gstbasevideodecoder.h:
6829 * gst-libs/gst/video/gstbasevideoencoder.c:
6830 * gst-libs/gst/video/gstbasevideoencoder.h:
6831 * gst-libs/gst/video/gstbasevideoparse.c:
6832 * gst-libs/gst/video/gstbasevideoparse.h:
6833 * gst-libs/gst/video/gstbasevideoutils.c:
6834 * gst-libs/gst/video/gstbasevideoutils.h:
6835 video: Copy BaseVideo classes from Schroedinger
6837 2009-05-15 23:05:45 +0200 Arnout Vandecappelle <arnout@mind.be>
6839 * gst/tcp/gstmultifdsink.c:
6840 multifdsink: add num-fds property
6841 multifdsink::num-fds
6843 2009-05-15 20:36:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6845 * gst-libs/gst/pbutils/descriptions.c:
6846 pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
6848 2009-05-14 11:44:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6850 * ext/vorbis/vorbisenc.c:
6851 vorbisenc: Implement Preset interface
6853 2009-05-14 11:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6855 * ext/theora/theoraenc.c:
6856 theoraenc: Implement Preset interface
6858 2009-05-14 11:41:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6860 * ext/ogg/gstoggmux.c:
6861 oggmux: Implement Preset interface
6863 2009-05-14 21:37:22 +0100 Jan Schmidt <thaytan@noraisin.net>
6865 * gst/playback/gstplaysink.c:
6866 playbin2: Fix cdda:// playback
6867 Don't send async-start when the playsink has already been configured
6868 before changing state.
6870 2009-05-14 01:31:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6873 configure: require core CVS for gst_adapter_prev_timestamp()
6874 which is used in the libvisual plugin.
6876 2009-04-22 18:34:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6879 AUTHORS: fix my email
6881 2009-04-22 18:35:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6883 * gst-libs/gst/audio/gstaudioclock.c:
6884 audioclock: make our internal time monotonic
6885 Make the internal time increase monotonically.
6887 2009-05-13 19:27:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6889 * ext/libvisual/visual.c:
6890 visual: remove next_ts variable
6891 We can remove the next_ts variable as we don't use it anymore.
6893 2009-05-13 19:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6895 * ext/libvisual/visual.c:
6896 visual: use new adapter timestamp code
6897 Use the new adapter timestamp tracking code to make things easier and produce
6898 vastly better output timestamps.
6900 2009-05-13 01:35:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6932 po: avoid conflicts of local *.po files with files in git
6933 Make it so that filenames and line numbers are only stored in the *.pot file
6934 (which is not in git), but not in the individual *.po files. This information
6935 is hardly useful for translators in our case, and it should avoid the constant
6936 conflicts of local *.po files with the ones in git which are caused by the
6937 source files changing and the line numbers being updated. This commit might
6938 cause one last merge conflict for you, which you can work around with
6939 "git checkout po/*.po" before merging or pulling. After that there should
6940 (hopefully) not be any more local modifications of these files (unless
6941 someone committed additions or changes to translated strings and the
6942 *.po files haven't been updated yet, that is).
6944 2009-05-12 23:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6946 * tests/check/elements/.gitignore:
6947 * tests/check/elements/audioresample.c:
6948 tests: fix audioresample unit test on big endian architectures
6949 Don't hardcode endianness=1234 in the filtercaps, it will cause
6950 pad link failures which will result in the test timing out.
6952 2009-05-12 17:18:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6954 * gst/audiotestsrc/gstaudiotestsrc.c:
6955 audiotestsrc: fix broken enum nick - it should have a hyphen
6956 The enum nick should be 'sine-table', not 'sine table'. Technically this is
6957 an API/ABI change I guess, but anyone who was using this and didn't report
6960 2009-05-01 01:04:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6962 * gst/audiotestsrc/gstaudiotestsrc.c:
6963 audiotestsrc: seek to the requested byte offset, not the expected byte offset
6965 2009-05-01 01:03:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6967 * gst/audiotestsrc/gstaudiotestsrc.c:
6968 * gst/audiotestsrc/gstaudiotestsrc.h:
6969 audiotestsrc: support more than just one channel
6971 2009-05-12 15:52:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6973 * gst-libs/gst/interfaces/propertyprobe.h:
6974 propertyprobe: Fix typo in the docs
6976 2009-05-12 12:17:55 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
6978 * ext/ogg/gstoggmux.c:
6979 * ext/theora/theora.c:
6980 * ext/vorbis/vorbis.c:
6981 Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder
6983 2009-04-30 16:37:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6985 * gst/videorate/gstvideorate.c:
6986 * gst/videorate/gstvideorate.h:
6987 videorate: handle invalid timestamps better
6988 Handle buffers with -1 timestamps better by keeping track of the en time of the
6989 previous buffer and assuming the -1 timestamp buffer goes right after the
6991 when we have two buffers that are equally good, output the oldest buffer once to
6993 don't try to calculate latency when the input framerate is unknown.
6995 2009-04-28 11:37:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6997 * ext/ogg/gstoggmux.c:
6998 oggmux: small debug statement in DISCONT
7000 2009-04-28 11:24:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7002 * ext/ogg/gstoggdemux.c:
7003 * ext/ogg/gstoggdemux.h:
7004 oggdemux: fix abuse of ogg API, handle broken oggs
7005 When we feed the ogg sync layer, we need to feed it contiguous data even if the
7006 sync layer did not consume all of it yet. This makes sure that it always finds
7007 the next page even for more corrupted files. Use a different read_offset for
7008 this purpose. since we now keep track of the sync layer, we don't have to reset
7009 after finding a start of a page.
7010 Add some more debug info for the error paths.
7011 Only reset the sync layer when we perform a seek operation.
7012 Avoid failure when the next chain has no bos pages but instead simply ignore it.
7013 when we receive unknown page serial numbers mid stream, don't fail but post a
7014 warning and hope that we get back on track later.
7017 2009-04-30 16:41:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7019 * gst/playback/gstdecodebin2.c:
7020 decodebin2: make subpictures a raw output format
7021 Subpictures are a raw format, we want those pads exposed so that playbin2 can do
7022 the subpicture mixing.
7024 2009-04-27 10:15:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7026 * gst-libs/gst/rtp/gstbasertppayload.c:
7027 * gst-libs/gst/rtp/gstbasertppayload.h:
7028 rtpdepay: add some more comments
7030 2009-04-17 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7032 * gst-libs/gst/audio/gstaudioclock.c:
7033 audioclock: make sure values are ever increasing
7035 2009-05-05 17:17:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7037 * gst/playback/gstplaysink.c:
7038 playbin2: make fallback identity silent
7039 Set the signal-handoffs to FALSE and silent to TRUE for the fallback identity
7040 element so that it consumes less CPU.
7042 2009-04-17 10:57:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7044 * gst/playback/gstplaybin2.c:
7045 * gst/playback/gstplaysink.c:
7046 playbin2: handle custom audiosinks differently
7047 Keep track of the autoplugged custom sinks and configure them in the playsink
7048 element when we have collected all streams.
7049 Also make sure that we only select one custom sink.
7050 When unreffing the internal sink, we don't need to change the state to NULL.
7052 2009-05-12 10:36:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7054 * gst/playback/gstplaybin2.c:
7055 * gst/playback/gstplaysink.c:
7056 * gst/playback/gstplaysink.h:
7057 playbin2: unify custom sink get/set functions
7058 Use one function to set/get all of the different sink types.
7059 cleanup up the subpicture chain too.
7060 Allow setting a custom subpicture sink.
7062 2009-05-11 18:29:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7064 * gst-libs/gst/interfaces/tunernorm.h:
7065 interfaces: Seperate some more struct definitions from typedefs
7067 2009-05-11 15:48:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7069 * gst-libs/gst/interfaces/navigation.h:
7070 * gst-libs/gst/interfaces/videoorientation.h:
7071 * gst-libs/gst/interfaces/xoverlay.h:
7072 interfaces: Seperate some more struct definitions from typedefs
7074 2009-05-10 17:28:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7076 * win32/common/libgstinterfaces.def:
7077 Add new functions to win32 exports
7079 2009-05-10 17:28:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7081 * docs/libs/gst-plugins-base-libs-sections.txt:
7082 Add new functions to the docs
7084 2009-05-10 17:25:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7086 * gst-libs/gst/interfaces/mixer.c:
7087 * gst-libs/gst/interfaces/mixer.h:
7088 interfaces: API: Add gst_mixer_get_mixer_type()
7089 This is a convenience function that returns the mixer_type
7090 of the interface struct.
7092 2009-05-10 17:25:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7094 * gst-libs/gst/interfaces/colorbalance.c:
7095 interfaces: Add docs for gst_color_balance_get_balance_type()
7097 2009-05-10 11:17:19 +0200 Marc-Andre Lureau <marcandre.lureau@gmail.com>
7100 Run libtoolize before aclocal
7101 This unbreaks the build in some cases. Fixes bug #582021
7103 2009-05-07 17:38:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7105 * ext/pango/gsttextrender.c:
7106 textrender: Correctly initialize the background for ARGB too
7108 2009-05-07 16:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7110 * ext/pango/gsttextrender.c:
7111 * ext/pango/gsttextrender.h:
7112 textrender: Use libgstvideo functions to create caps
7113 Also check if downstream wants ARGB always when we get
7116 2009-05-07 16:52:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7118 * ext/pango/gsttextrender.c:
7119 textrender: Don't always use ARGB if downstream supports it but take it's preference
7121 2009-05-07 16:48:08 +0200 Kapil Agrawal <kapil@mediamagictechnologies.com>
7123 * ext/pango/gsttextrender.c:
7124 * ext/pango/gsttextrender.h:
7125 textrender: Add support for ARGB and alignment properties
7128 2009-05-07 16:42:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7130 * ext/pango/gsttextrender.c:
7131 textrender: Add ; after GST_BOILERPLATE to fix indention
7133 2009-05-07 15:10:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7135 * gst-libs/gst/tag/gstvorbistag.c:
7136 vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
7138 2009-05-07 14:59:36 +0200 Arnout Vandecappelle <arnout@mind.be>
7140 * gst/typefind/gsttypefindfunctions.c:
7141 typefindfunctions: made mp3_type_find less aggressive
7142 mp3_type_find could suggest already when only a single valid header
7143 was found, if it ran out of data before the end of the next frame.
7144 Therefore, ignore the last found frame if it was incomplete.
7147 2009-05-07 14:48:29 +0200 John Millikin <jmillikin@gmail.com>
7149 * gst-libs/gst/tag/gstvorbistag.c:
7150 vorbistag: Store cover art in vorbiscomments
7153 2009-05-07 06:14:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7155 * gst-libs/gst/interfaces/colorbalance.c:
7156 * gst-libs/gst/interfaces/colorbalance.h:
7157 interfaces: API: Add gst_color_balance_get_balance_type()
7158 This is a convenience function that returns the balance_type
7159 of the interface struct.
7161 2009-05-06 17:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7163 * gst-libs/gst/interfaces/colorbalance.h:
7164 * gst-libs/gst/interfaces/colorbalancechannel.h:
7165 * gst-libs/gst/interfaces/tuner.h:
7166 * gst-libs/gst/interfaces/tunerchannel.h:
7167 interfaces: Separate struct definitions from typedefs
7169 2009-05-06 14:03:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7171 * pkgconfig/gstreamer-app-uninstalled.pc.in:
7172 Fix libdir for uninstalled gstreamer-app library
7174 2009-05-12 01:59:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7176 * gst-libs/gst/pbutils/descriptions.c:
7177 pbutils: add description for APE tag caps
7179 2009-05-12 01:35:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7182 configure: bump core requirement to last release
7183 as that's more likely to be true than that we need
7186 2009-05-12 01:21:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7190 configure: rename CVS -> git in a couple of places
7192 2009-05-12 01:17:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7195 configure: bump GLib requirement to GLib >= 2.16
7196 as per the New Regime (see wiki).
7198 2009-05-01 00:09:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7200 * gst-libs/gst/tag/gsttagdemux.c:
7201 tagdemux: cache events from upstream and re-send them once we have a source pad
7202 Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
7205 2009-05-07 14:07:44 -0700 Michael Smith <msmith@songbirdnest.com>
7207 * gst-libs/gst/riff/riff-media.c:
7208 riff: support UYVY raw 4:2:2 in riff.
7210 2009-05-11 21:20:07 +0100 Jan Schmidt <thaytan@noraisin.net>
7213 Back to development -> 0.10.23.1
7215 2009-04-27 22:42:55 -0700 Michael Smith <msmith@syncword.(none)>
7217 * ext/theora/theoradec.c:
7218 theoradec: fix buffer overrun on 422 decode.
7220 2009-04-27 21:39:01 -0700 Michael Smith <msmith@syncword.(none)>
7222 * ext/theora/theoradec.c:
7223 theoradec: 444 support.
7225 2009-04-27 21:30:04 -0700 Michael Smith <msmith@syncword.(none)>
7227 * ext/theora/theoradec.c:
7228 theoradec: handle 422 images (as YUY2).
7230 2009-04-27 21:01:51 -0700 Michael Smith <msmith@syncword.(none)>
7232 * ext/theora/gsttheoradec.h:
7233 * ext/theora/theoradec.c:
7234 theoradec: rearrange code in preparation for 422 and 444 support.
7236 === release 0.10.23 ===
7238 2009-05-10 23:57:01 +0100 Jan Schmidt <thaytan@noraisin.net>
7244 * docs/plugins/gst-plugins-base-plugins.args:
7245 * docs/plugins/gst-plugins-base-plugins.hierarchy:
7246 * docs/plugins/gst-plugins-base-plugins.interfaces:
7247 * docs/plugins/gst-plugins-base-plugins.prerequisites:
7248 * docs/plugins/gst-plugins-base-plugins.signals:
7249 * docs/plugins/inspect/plugin-adder.xml:
7250 * docs/plugins/inspect/plugin-alsa.xml:
7251 * docs/plugins/inspect/plugin-app.xml:
7252 * docs/plugins/inspect/plugin-audioconvert.xml:
7253 * docs/plugins/inspect/plugin-audiorate.xml:
7254 * docs/plugins/inspect/plugin-audioresample.xml:
7255 * docs/plugins/inspect/plugin-audiotestsrc.xml:
7256 * docs/plugins/inspect/plugin-cdparanoia.xml:
7257 * docs/plugins/inspect/plugin-decodebin.xml:
7258 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
7259 * docs/plugins/inspect/plugin-gdp.xml:
7260 * docs/plugins/inspect/plugin-gio.xml:
7261 * docs/plugins/inspect/plugin-gnomevfs.xml:
7262 * docs/plugins/inspect/plugin-libvisual.xml:
7263 * docs/plugins/inspect/plugin-ogg.xml:
7264 * docs/plugins/inspect/plugin-pango.xml:
7265 * docs/plugins/inspect/plugin-playback.xml:
7266 * docs/plugins/inspect/plugin-queue2.xml:
7267 * docs/plugins/inspect/plugin-subparse.xml:
7268 * docs/plugins/inspect/plugin-tcp.xml:
7269 * docs/plugins/inspect/plugin-theora.xml:
7270 * docs/plugins/inspect/plugin-typefindfunctions.xml:
7271 * docs/plugins/inspect/plugin-uridecodebin.xml:
7272 * docs/plugins/inspect/plugin-video4linux.xml:
7273 * docs/plugins/inspect/plugin-videorate.xml:
7274 * docs/plugins/inspect/plugin-videoscale.xml:
7275 * docs/plugins/inspect/plugin-videotestsrc.xml:
7276 * docs/plugins/inspect/plugin-volume.xml:
7277 * docs/plugins/inspect/plugin-vorbis.xml:
7278 * docs/plugins/inspect/plugin-ximagesink.xml:
7279 * docs/plugins/inspect/plugin-xvimagesink.xml:
7280 * gst-plugins-base.doap:
7281 * win32/common/_stdint.h:
7282 * win32/common/config.h:
7285 2009-05-10 23:56:05 +0100 Jan Schmidt <thaytan@noraisin.net>
7318 2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
7350 * win32/common/_stdint.h:
7351 * win32/common/config.h:
7352 0.10.22.6 pre-release
7354 2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7356 * gst/playback/gstplaysink.c:
7357 playbin2: fix resume after pause
7358 Don't ignore the state change of the children, they might be doing an ASYNC
7361 2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
7394 0.10.22.5 pre-release
7396 2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7398 * gst/tcp/gstmultifdsink.c:
7399 * gst/tcp/gsttcp-marshal.list:
7400 multifdsink: fix signature of the add-full signal
7401 The second parameter is a GstSyncMethod enum, not a boolean.
7403 2009-05-07 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7405 * gst/playback/gstplaysink.c:
7406 playsink: initialize variable too
7408 2009-05-07 14:28:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7410 * gst/playback/gstplaysink.c:
7411 playbin2: make playsink go ASYNC to PAUSED
7412 Make playsink go async to the PAUSED state instead of relying on uridecodebin
7413 for async behaviour in playbin. This solves some problems (mainly with DVD)
7414 where the pipeline would go to PLAYING before preroll completed, failing to
7415 select the audiosink clock.
7418 2009-05-06 16:09:52 +0100 Jan Schmidt <thaytan@noraisin.net>
7450 * win32/common/_stdint.h:
7451 * win32/common/config.h:
7452 0.10.22.4 pre-release
7454 2009-05-06 13:19:34 +0100 Zaheer Merali <zaheerabbas@merali.org>
7456 * ext/theora/theoraenc.c:
7457 * ext/vorbis/vorbisenc.c:
7458 vorbisenc, theoraenc: Ensure gp is computed consistently + clip to segment
7459 With vorbisenc, compute the granulepos with running time and clip incoming
7461 With theoraenc, drop out of segment buffers.
7463 2009-05-01 16:47:53 +0100 Jan Schmidt <thaytan@noraisin.net>
7465 * gst/audioresample/gstaudioresample.c:
7466 audioresample: Fix buffer size transformations
7467 When calculating the input/output buffer sizes in the transform_size function,
7468 take the number of channels into account, so we don't end up calculating
7469 a buffer size that only contains a partial number of audio frames.
7470 Also, when going from output size to input size, round down rather than
7471 up, so as to calculate the minimum number of samples that *might* yield
7472 a buffer of the intended destination size.
7473 Fixes: #580470 and #580952
7475 2009-04-29 16:45:27 +0100 Jan Schmidt <thaytan@noraisin.net>
7477 * ext/vorbis/gstvorbisenc.h:
7478 * ext/vorbis/vorbisenc.c:
7479 vorbisenc: Ensure output buffers fall within the segment
7480 Add the start position of the first segment to the running time
7481 used to generate buffer timestamps in vorbisenc. This avoids generating
7482 buffers which fall outside the initial segment. The element segment
7483 handling requires more extensive fixing, but this at least prevents
7484 regressions. Fixes: #580020
7486 2009-04-29 11:18:42 +0200 Andy Wingo <wingo@oblong.net>
7488 * gst-libs/gst/audio/gstbaseaudiosink.c:
7489 Revert "add can-activate-pull property to baseaudiosink"
7490 This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.
7492 2009-04-29 11:18:33 +0200 Andy Wingo <wingo@oblong.net>
7494 * gst-libs/gst/audio/gstbaseaudiosink.c:
7495 Revert "[baseaudiosink] add docs for can-activate-pull"
7496 This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.
7498 2009-04-28 18:48:33 +0200 Andy Wingo <wingo@oblong.net>
7500 [baseaudiosink] add docs for can-activate-pull
7501 * gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
7504 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
7506 add can-activate-pull property to baseaudiosink
7507 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
7510 2009-04-28 11:32:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7512 * gst/videorate/gstvideorate.c:
7513 * gst/videorate/gstvideorate.h:
7514 videorate: clear discont on duplicated buffers
7515 When videorate duplicates a buffer with a DISCONT flag, it copies the discont on
7516 the first pushed buffer but fails to clear it for subsequent buffers. This
7517 causes theoraenc!oggmux and possibly other elements to consider this a discont
7519 Fix videorate to produce discont as the first buffer and after a flushing seek.
7522 2009-04-24 18:13:00 +0100 Jan Schmidt <thaytan@noraisin.net>
7524 * tests/check/Makefile.am:
7525 check: Disable the playbin2 for this release, as it is a bit racy.
7526 Disable the test, as per the discussion in #580120. Needs re-enabling
7527 after the release, when playbin2 is fixed.
7529 2009-04-23 08:41:19 +0200 Edward Hervey <bilboed@bilboed.com>
7531 * gst/playback/gstdecodebin2.c:
7532 decodebin2: Don't reduce max-size-time of exposed groups. Fixes #579912
7533 The 2s limit is way too small for a lot of files (which have an interleave
7534 in time of between 3 and 5s). Instead, leave it to the initial 5s value
7535 and reduce the other limits (allowing us to stay memory-efficient).
7537 2009-04-21 21:06:59 +0100 Jan Schmidt <thaytan@noraisin.net>
7569 * win32/common/_stdint.h:
7570 * win32/common/config.h:
7571 0.10.22.3 pre-release
7573 2009-04-21 20:41:23 +0100 René Stadler <mail@renestadler.de>
7575 * gst/audioresample/gstaudioresample.c:
7576 audioresample: Fix unused variable in compilation with --disable-gst-debug
7579 2009-04-21 22:12:28 +0100 Jan Schmidt <thaytan@noraisin.net>
7582 Automatic update of common submodule
7583 From b3941ea to 6ab11d1
7585 2009-04-21 20:57:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7587 * gst/playback/gstplaybasebin.c:
7588 playbin: only use raw_decoding_mode when it's true
7589 First check the pad caps if they are raw before setting the raw_decoding_mode to
7590 TRUE. Fixes playback of transport streams and other streams that require large
7594 2009-04-19 18:15:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7596 * gst-libs/gst/cdda/gstcddabasesrc.c:
7597 * tests/check/libs/cddabasesrc.c:
7598 cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core
7599 Don't use REPLACE_ALL merge mode when that's not really what we want,
7600 as now that REPLACE_ALL actually does what it's supposed to do in
7601 core, we drop tags we wanted to keep, such as the various disc id
7602 tags. Add unit test for this as well. Fixes #579463.
7604 2009-04-17 10:34:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7606 * gst-libs/gst/rtsp/gstrtspconnection.c:
7607 rtspconnection: don't use GLib-2.16 API, we require only 2.14
7610 2009-04-17 10:55:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7612 * gst-libs/gst/audio/gstbaseaudiosink.c:
7613 baseaudiosink: don't unparent the ringbuffer
7614 when going to NULL, don't unparent the ringbuffer because we don't support going
7615 back to 0 very well yet.
7618 2009-04-17 10:53:10 +0200 Olivier Crete <tester at tester.ca>
7620 * gst-libs/gst/rtp/gstrtcpbuffer.c:
7621 RTCP: don't fail when retrieving invalid PT
7622 We can't meaningfully assert on valid packet types so just return the type as it
7623 is. Update the comments to reflect this.
7626 2009-04-16 12:12:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7628 * docs/libs/gst-plugins-base-libs-sections.txt:
7629 * gst-libs/gst/app/gstappsink.h:
7630 * gst-libs/gst/app/gstappsrc.h:
7631 app: add trivial cast macros
7632 Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
7633 and add the macros to the standard macros in the docs.
7636 2009-04-16 12:09:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7638 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
7639 pkgconfig: add the app/ directory to Libs
7640 Add the appsrc/appsink directory to the Libs in the uninstalled
7641 pkgconfig file so that one can build against it.
7644 2009-04-15 22:59:31 +0100 Jan Schmidt <thaytan@noraisin.net>
7647 0.10.22.2 pre-release
7649 2009-04-15 22:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
7652 ChangeLog: regenerate changelog with the gen-changelog script
7654 2009-04-16 00:41:13 +0100 Jan Schmidt <thaytan@noraisin.net>
7685 po: Update po files from TP
7687 2009-04-16 00:40:59 +0100 Jan Schmidt <thaytan@noraisin.net>
7689 * win32/common/_stdint.h:
7690 * win32/common/config.h:
7691 * win32/common/gstrtsp-enumtypes.c:
7692 * win32/common/interfaces-enumtypes.c:
7693 * win32/common/interfaces-enumtypes.h:
7694 * win32/common/video-enumtypes.c:
7695 win32: Update win32 build files
7697 2009-04-16 00:31:55 +0100 Jan Schmidt <thaytan@noraisin.net>
7699 * tests/check/libs/video.c:
7700 check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.
7702 2009-04-16 00:31:00 +0100 Jan Schmidt <thaytan@noraisin.net>
7704 * tests/check/elements/playbin2.c:
7705 check: Fix the input uri in playbin2 test.
7706 Don't try and use a random file in wim's home directory as a test input
7708 2009-04-15 15:35:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7710 * gst-libs/gst/video/video.h:
7711 video: Fix typo in the docs
7713 2009-04-15 14:53:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7715 * gst-libs/gst/video/video.c:
7716 * gst-libs/gst/video/video.h:
7717 video: Add support for YVYU YUV colorspace
7719 2009-04-15 00:17:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7721 * docs/libs/gst-plugins-base-libs-docs.sgml:
7722 * gst-libs/gst/fft/gstfft.c:
7723 docs: fix hyperlink and move fft attribution to the right place
7725 2009-04-15 00:02:39 +0300 Stefan Kost <ensonic@users.sf.net>
7727 * gst-libs/gst/audio/gstbaseaudiosink.c:
7728 log: use G_GUINT64_FORMAT instead of llu
7730 2009-04-14 18:31:52 +0200 Josep Torra <n770galaxy at gmail.com>
7732 * gst-libs/gst/rtsp/gstrtspdefs.c:
7733 * gst-libs/gst/rtsp/gstrtspdefs.h:
7734 RTSP: add missing headers for WMS RTSP
7735 Add missing headers related to Windows Media RTSP extension.
7738 2009-04-14 18:16:37 +0200 Olivier Crete <tester at tester.ca>
7740 * docs/design/draft-keyframe-force.txt:
7741 * ext/theora/gsttheoraenc.h:
7742 * ext/theora/theoraenc.c:
7743 theoraenc: implement upstream keyframe force
7744 Implement handling of upstream keyframe forcing.
7745 Update the design documents too.
7748 2009-04-14 17:31:31 +0200 Olivier Crete <tester at tester.ca>
7750 * ext/theora/theoraenc.c:
7751 theoraenc: factor out keyframe forcing
7754 2009-04-14 17:01:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7757 * gst-libs/gst/fft/gstfft.c:
7758 Give credit to Mark Borgerding (kissfft author)
7759 and add myself to AUTHORS as well. Fixes #575638.
7761 2009-04-14 17:04:06 +0200 Jan Urbanski <j.urbanski at students.mimuw.edu.pl>
7763 * gst/tcp/gstmultifdsink.c:
7764 * gst/tcp/gstmultifdsink.h:
7765 multifdsink: add property to resend streamheaders
7766 Adds a new property in multifdsink, resend-streamheader.
7767 If this property is false, the multifdsink will not send the streamheader if
7768 there's already one set for a particular client.
7769 There are some formats in which every stream needs to start with a certain
7770 blob, but you can't inject this blob at leisure. If the producer wants to
7771 change the blob in question and sets in as the streamheader on the outgoing
7772 buffers' caps, new clients of multifdsink will get the new streamheader, but
7773 old clients will break, because they'll see the blob in the middle of the
7775 The property is true by default, so existing code will not see any difference.
7778 2009-04-14 16:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7780 * gst/tcp/gstmultifdsink.c:
7781 * gst/tcp/gstmultifdsink.h:
7782 multifdsink: add property to handle client write
7783 Add a property to disable listening to client writes. This property is usefull
7784 when other code will deal with reading from the client socket.
7785 API: GstMultiFdSink::handle-read property
7787 2009-04-14 16:45:20 +0200 Johann Prieur <johann.prieur at gmail.com>
7789 * docs/libs/gst-plugins-base-libs-sections.txt:
7790 * gst-libs/gst/rtp/gstrtcpbuffer.c:
7791 * gst-libs/gst/rtp/gstrtcpbuffer.h:
7792 * win32/common/libgstrtp.def:
7793 RTCP: add beginnings of Feedback messages
7794 Add the beginnings of parsing and constructing Feedback messages.
7797 2009-04-14 13:51:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7799 * gst/playback/gstplaysink.c:
7800 playbin2: clear the target
7801 Clear the target of our ghostpads before we remove the pad from the element.
7802 This to make sure that the internal pad is not left linked to whatever pad we
7803 were ghosted to. This should only be a problem when we leak the ghostpads.
7804 Also release our subpicture pads.
7807 2009-04-14 12:10:30 +0100 Hannes Bistry <hannesb@gmx.net>
7809 * sys/ximage/ximagesink.c:
7810 ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
7813 2009-04-14 13:16:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7815 * gst-libs/gst/audio/gstbaseaudiosrc.c:
7816 baseaudiosrc: adjust the internal timestamp
7817 Adjust the internal timestamp before comparing it against the adjusted clock
7821 2009-04-14 13:12:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7823 * gst-libs/gst/audio/gstbaseaudiosink.c:
7824 baseaudiosink: use new clock time methods
7825 Use the unadjusted internal clock times to calculate the internal/external
7826 offset when calibrating the clock.
7827 When going to NULL, unparent and free the ringbuffer, like we do in the source
7831 2009-04-14 13:08:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7833 * gst-libs/gst/audio/gstaudioclock.c:
7834 * gst-libs/gst/audio/gstaudioclock.h:
7835 * win32/common/libgstaudio.def:
7836 audioclock: add methods for the internal offset
7837 Add two methods for getting the unadjusted time of the clock and one for
7838 adjusting an internal time. We will need these methods for correctly handling
7839 the time after a gst_audio_clock_reset().
7840 Add a debug category and some debug lines to the audio clock.
7841 API: gst_audio_clock_get_time()
7842 API: gst_audio_clock_adjust()
7843 API: GST_AUDIO_CLOCK_CAST()
7845 2009-04-14 11:34:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7847 * gst/playback/gstdecodebin2.c:
7848 decodebin2: fix up the debugs and warnings
7849 Use _OBJECT variants because we can. Go over some log statements and put them in
7853 2009-04-12 22:26:33 +0200 Luca Ognibene <luca.ognibene at gmail.com>
7855 * gst/tcp/gstmultifdsink.c:
7856 multifdsink: fix error in sync-method
7857 Multifdsink did not handle sync-method=latest-keyframe correctly when the
7858 soft-limit is set to -1 (unlimited).
7861 2009-04-10 21:49:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7863 * gst-libs/gst/audio/gstbaseaudiosink.c:
7864 baseaudiosink: use the internal clock time
7865 We can't assume that the internal clock time is the same as the function we
7866 installed on our provided clock because somebody might have changed it.
7868 2009-04-10 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7870 * tests/examples/seek/seek.c:
7871 seek: handle clock-lost messages
7872 When we receive a clock-lost message we need to pause and play to select a new
7875 2009-04-10 13:44:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7877 * tests/check/Makefile.am:
7878 * tests/check/elements/playbin2.c:
7879 check: add a unit test for playbin2
7880 Add unit test for playbin2 and include the refcount test in #577794.
7882 2009-04-10 13:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7884 * gst/playback/gstplaysink.c:
7885 playbin2: fix refcounting of visualisations
7888 2009-04-10 13:27:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7890 * gst/playback/gstplaysink.c:
7891 playsink: fix refcounting of custom elements
7892 Sink the custom sinks, let other elements we create be sunken by the bin we add
7896 2009-04-10 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7898 * tests/check/elements/appsink.c:
7899 check: fix appsink test
7900 Fix the appsink test now that the method signature changed.
7902 2009-04-10 12:26:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7904 * gst/playback/gstplaybin2.c:
7905 playbin2: handle missing input-selector
7906 Gracefully degrade and disable stream selection when input-selector is
7909 2009-04-09 23:46:17 +0200 Martin Samuelsson <martin.samuelsson at axis.com>
7911 * gst-libs/gst/app/gstappsink.c:
7912 * gst-libs/gst/app/gstappsink.h:
7913 appsink: make callbacks return GstFlowReturn
7914 Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
7915 errors can be reported properly.
7918 2009-04-09 18:04:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7920 * gst-libs/gst/audio/gstringbuffer.c:
7921 * gst-libs/gst/audio/gstringbuffer.h:
7922 ringbuffer: allow for custom commit functions
7923 Allow subclasses to override the commit method.
7925 2009-04-08 18:04:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7927 * gst-libs/gst/audio/gstbaseaudiosink.c:
7928 baseaudiosink: fix a small glitch after pause
7929 After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
7930 the amount of output samples we consumed. We can't do this reliably with the
7931 current API when we are doing trick modes but we can do the right thing for
7934 2009-04-08 16:43:27 +0300 Stefan Kost <ensonic@users.sf.net>
7936 * gst/playback/gstplaysink.c:
7937 playbin2: better error message on sink failure
7938 If we could create the sinks, but the don't work, don't send the missing plugin
7939 message and report that the state-changed failed.
7941 2009-04-07 22:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
7943 * gst-libs/gst/audio/gstaudiofilter.c:
7944 audiofilter: don't leak pad-template
7945 gst_element_class_add_pad_template() does not take ownership.
7947 2009-04-04 21:18:38 +0300 Felipe Contreras <felipe.contreras@gmail.com>
7950 Automatic update of common submodule
7951 From d0ea89e to b3941ea
7953 2009-04-04 16:28:14 +0200 Edward Hervey <bilboed@bilboed.com>
7955 * gst-libs/gst/interfaces/navigation.c:
7956 * sys/v4l/v4lsrc_calls.c:
7957 navigation/v4l: Don't use g_return_val_if_fail for computed/used values.
7959 2009-03-22 09:46:37 +0100 Edward Hervey <bilboed@bilboed.com>
7961 * ext/theora/theoradec.c:
7962 theoradec: return GST_CLOCK_TIME_NONE for negative framecounts.
7963 This fixes most seeking issues when used with gnonlin.
7966 2009-04-04 14:53:42 +0200 Edward Hervey <bilboed@bilboed.com>
7969 Automatic update of common submodule
7970 From f8b3d91 to d0ea89e
7972 2009-04-03 10:51:42 -0700 Michael Smith <msmith@songbirdnest.com>
7974 * gst/playback/gstplaybin2.c:
7975 playbin2: don't leak selector when getting current stream numbers.
7977 2009-04-02 22:28:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7979 * gst-libs/gst/rtsp/gstrtspconnection.c:
7980 rtsp: use fully qualified urls when using a proxy
7981 Use a fully qualified url when specifying the url for tunneled requests through
7985 2009-03-31 00:54:30 +0100 Jan Schmidt <thaytan@noraisin.net>
7987 * docs/libs/gst-plugins-base-libs-sections.txt:
7988 * gst-libs/gst/interfaces/navigation.c:
7989 * gst-libs/gst/interfaces/navigation.h:
7990 * tests/check/Makefile.am:
7991 * tests/check/libs/.gitignore:
7992 * tests/check/libs/navigation.c:
7993 * win32/common/libgstinterfaces.def:
7994 navigation: Extend the navigation interface
7995 Add support for a set of standard commands that can be queried and executed to
7996 support applications like DVD. Add query construction and parsing functions.
7997 Add new messages that can be sent on the bus to provide notifications related
7998 to commands, multiangle changes, and button highlight activity.
7999 Add some helper functions to parse the existing GstNavigation events that
8000 elements might receive.
8001 Document it all and add unit tests.
8003 2009-02-04 17:03:07 +0000 Jan Schmidt <thaytan@noraisin.net>
8005 * gst/playback/gstplaybasebin.c:
8006 * gst/playback/gstplaybasebin.h:
8007 playbin: Add simple 'raw decoding mode'.
8008 Raw decoding mode removes almost all buffering in video and audio queues
8009 when a source providing already decoded video/audio is detected, on the
8010 possibly bogus assumption that such a source should provide sufficient
8011 internal queueing. Fixes playback on some DVDs, and improves it
8014 2009-04-02 09:27:07 +0100 Jan Schmidt <thaytan@noraisin.net>
8016 * tests/check/elements/.gitignore:
8017 ignores: Ignore the videoscale check binary
8019 2009-04-02 12:13:57 +0100 Jan Schmidt <thaytan@noraisin.net>
8021 * win32/common/libgstrtsp.def:
8022 win32: Add gst_rtsp_connection_set_proxy to the win32 exports
8024 2009-04-02 10:42:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8026 * ext/alsa/gstalsamixer.c:
8027 alsamixer: don't forget to release locks in a few places
8030 2009-04-02 11:10:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8032 * gst/videoscale/vs_4tap.c:
8033 videoscale: Don't read over line ends when taking the last Cr or Cb
8035 2009-04-02 10:52:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8037 * gst/videoscale/vs_4tap.c:
8038 videoscale: Don't write to few pixels and don't mix Cr and Cb
8041 2009-04-01 15:15:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8043 * gst/audioresample/gstaudioresample.c:
8044 * tests/check/elements/audioresample.c:
8045 audioresample: fix negotiation so that upstream can actually fixate to downstream's rate
8046 If one side has a preference for a particular sample rate or set of sample rates, we
8047 should honour this in the caps we advertise and transform to and from, so that elements
8048 actually know about the other side's sample rate preference and can negotiate to it
8049 if supported. Also add unit test for this.
8051 2009-03-26 19:34:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8053 * gst/playback/gstplaybin2.c:
8054 docs: add a blurb about redirect messages to playbin2 docs
8056 2009-04-01 09:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8058 * gst-libs/gst/rtsp/gstrtspconnection.c:
8059 rtsp: fix little typo in the comments
8061 2009-03-31 17:52:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8063 * gst-libs/gst/rtsp/gstrtspconnection.c:
8064 rtspconnection: make gst_rtsp_watch_queue_message() thread-safe
8065 People might queue messages from a thread other than the thread in which
8066 the main context which this watch is attached is iterated from, so use
8067 a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
8068 over list nodes just freed in the other thread. This just fixes issues
8069 I've had with gst-rtsp-server. We might need more locking in various
8072 2009-03-31 18:13:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8074 * gst-libs/gst/rtsp/gstrtspconnection.c:
8075 * gst-libs/gst/rtsp/gstrtspmessage.c:
8076 rtsp: clear the entire builder structure
8077 And use structure instead of variable with sizeof when
8078 clearing the rtsp message structure, for clarity.
8080 2009-03-31 17:56:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8082 * gst-libs/gst/rtsp/gstrtspmessage.c:
8083 docs: fix typo in gst_rtsp_message_unset() API docs
8085 2009-03-31 19:00:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8087 * gst-libs/gst/rtsp/gstrtspconnection.c:
8088 * gst-libs/gst/rtsp/gstrtspconnection.h:
8089 rtsp: add support for proxies
8090 Add suport for proxy servers. Currently only used for tunneled HTTP
8091 connections without authentication.
8093 2009-03-31 18:57:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8095 * gst-libs/gst/rtsp/gstrtspmessage.c:
8096 Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
8097 This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.
8099 2009-03-26 18:54:56 +0200 Stefan Kost <ensonic@users.sf.net>
8101 * sys/xvimage/xvimagesink.c:
8102 xvimagesink: use xcontext->depth instead of bits in attr.max_value for colorkey
8103 According to the drivers in http://cgit.freedesktop.org/xorg/driver/ we should
8104 format the colorkey depending on xcontext->depth. This is what they will use to
8105 interprete the value. The max_value in turn is usualy a constant regardless of
8108 2009-03-31 12:22:14 +0300 Stefan Kost <ensonic@users.sf.net>
8110 * gst-libs/gst/rtsp/gstrtspmessage.c:
8111 rtsp: reset whole message (was sizeof pointer instead of sizeof type)
8113 2009-03-31 00:56:18 +0100 Jan Schmidt <thaytan@noraisin.net>
8115 * gst-libs/gst/interfaces/mixer.c:
8116 doc: Fix a typo in the GstMixer docs
8118 2009-03-29 12:01:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8120 * gst/videoscale/vs_scanline.c:
8121 videoscale: Fix linear scaling for one byte components
8124 2009-03-29 11:53:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8126 * gst/videoscale/vs_4tap.c:
8127 videoscale: Fix 4tap scaling of YUYV and friends
8129 2009-03-28 16:08:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8131 * gst/videoscale/vs_image.c:
8132 * gst/videoscale/vs_scanline.c:
8133 * gst/videoscale/vs_scanline.h:
8134 videoscale: Rewrite YUYV (and friends) scaling and don't read/write over line ends
8135 Partially fixes bug #577054, there's just one issue left now.
8137 2009-03-28 12:48:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8139 * tests/check/elements/videoscale.c:
8140 videoscale: Add some more unit tests
8142 2009-03-28 11:51:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8144 * gst/videoscale/gstvideoscale.c:
8145 videoscale: Use bilinear instead of 4tap scaling for heights < 4
8146 Partially fixes bug #577054.
8148 2009-03-28 11:45:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8150 * gst/videoscale/vs_scanline.c:
8151 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY/RGB/RGBA
8152 This case is for upscaling a frame with width=1
8153 Partially fixes bug #577054.
8155 2009-03-28 11:27:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8157 * gst/videoscale/vs_scanline.c:
8158 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
8159 Partially fixes bug #577054.
8161 2009-03-28 10:40:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8163 * gst/videotestsrc/gstvideotestsrc.c:
8164 videotestsrc: Initialize buffer memory with zeroes
8165 This prevents valgrind warnings when accessing the "x" parts
8166 of xRGB and friends in other elements that handle (and can handle)
8167 xRGB like ARGB (for example videoscale).
8169 2009-03-28 10:25:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8171 * tests/check/Makefile.am:
8172 * tests/check/elements/videoscale.c:
8173 videoscale: Add a lot of unit tests
8175 2009-03-28 10:06:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8177 * gst/videoscale/gstvideoscale.c:
8178 videocale: Add support for video/x-raw-gray with bpp=depth=8
8180 2009-03-28 10:01:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8182 * gst/videotestsrc/videotestsrc.c:
8183 videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8
8185 2009-03-28 09:43:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8187 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
8188 ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format
8190 2009-03-27 19:12:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8192 * gst/videoscale/vs_4tap.c:
8193 videoscale: Take the next luma value instead of every second next when scaling UYVY and friends
8195 2009-03-27 19:09:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8197 * gst/videoscale/gstvideoscale.c:
8198 videoscale: Add support for v308 YUV colorspace
8200 2009-03-27 13:15:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8202 * gst/videoscale/vs_4tap.c:
8203 videoscale: Add my copyright to the 4tap scalers
8205 2009-03-27 13:14:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8207 * gst/videoscale/gstvideoscale.c:
8208 videoscale: Enable 4-tap scaling for all supported formats
8210 2009-03-27 13:14:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8212 * gst/videoscale/vs_4tap.c:
8213 * gst/videoscale/vs_4tap.h:
8214 videoscale: Implement 4-tap scaling for RGB565 and RGB555
8216 2009-03-27 10:47:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8218 * gst/videoscale/vs_4tap.c:
8219 * gst/videoscale/vs_4tap.h:
8220 videoscale: Implement 4-tap scaling for UYVY
8222 2009-03-27 09:33:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8224 * gst/videoscale/vs_4tap.c:
8225 * gst/videoscale/vs_4tap.h:
8226 videoscale: Implement 4-tap scaling for YUY2 and YVYU
8228 2009-03-26 22:14:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8230 * gst/videoscale/vs_4tap.c:
8231 * gst/videoscale/vs_4tap.h:
8232 videoscale: Implement 4-tap scaling for RGB and BGR
8234 2009-03-26 22:08:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8236 * gst/videoscale/vs_4tap.c:
8237 * gst/videoscale/vs_4tap.h:
8238 videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats
8240 2009-03-26 11:02:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8242 * ext/pango/gsttextoverlay.c:
8243 textoverlay: Fix drawing of UYVY text borders
8245 2009-03-26 10:36:27 +0100 Zeeshan Ali <zeeshan.ali@nokia.com>
8247 * ext/pango/gsttextoverlay.c:
8248 * ext/pango/gsttextoverlay.h:
8249 textoverlay: Add support for UYVY colorspace
8252 2009-03-25 19:01:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8254 * gst/playback/gstdecodebin2.c:
8255 decodebin2: do some more cleanup
8256 Free the groups when we go to READY.
8257 Allow for NO_PREROLL elements.
8259 2009-03-25 16:37:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8261 * gst-libs/gst/rtsp/gstrtspconnection.c:
8262 rtsp: start CSeq counting from 1 instead of 0
8263 Start counting from 1 instead of 0 as this is what most other clients
8266 2009-03-25 16:35:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8268 * gst-libs/gst/rtsp/gstrtspdefs.c:
8269 * gst-libs/gst/rtsp/gstrtspdefs.h:
8270 rtsp: add ETag and If-Match headers
8271 Add new headers, we need them for RealMedia support.
8273 2009-03-25 14:16:25 +0200 Stefan Kost <ensonic@users.sf.net>
8275 * sys/xvimage/xvimagesink.c:
8276 xvimagesink: scale the colorkey components in case of 16bit visuals
8277 Use a default that won't be scales to 0,0,0
8279 2009-03-25 11:27:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8281 * gst-libs/gst/audio/gstbaseaudiosrc.c:
8282 audiosrc: improve 'Dropped n samples' warning message
8284 2009-03-24 19:41:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8286 * tests/examples/app/appsrc-ra.c:
8287 * tests/examples/app/appsrc-seekable.c:
8288 examples: use new method to set flags
8289 Use the new core method for setting object enum properties by name.
8291 2009-03-24 18:29:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8293 * gst/playback/gstplaysink.c:
8294 * gst/playback/gstplaysink.h:
8295 playbin2: add more support for subpictures
8297 2009-03-24 17:12:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8299 * gst/playback/gstplaybin2.c:
8300 * gst/playback/gstplaysink.c:
8301 * gst/playback/gstplaysink.h:
8302 playbin2: first support for subpictures
8303 Add beginnings of subpicture support.
8305 2009-03-24 15:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8307 * tests/examples/seek/seek.c:
8308 seek: print tags from the different tracks
8310 2009-03-24 12:22:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8312 * gst/playback/gstplaybin2.c:
8313 playbin2: blacklist subpictures for now
8314 Blacklist the subpictures until we add support for them.
8315 Add some small debug info.
8318 2009-03-24 12:19:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8320 * gst/playback/gsturidecodebin.c:
8321 uridecodebin: expose more media types
8322 Expose more media types from a raw source, such as the subpicture and various
8324 Small cleanups and add some more debugging.
8327 2009-03-24 10:42:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8329 * gst/playback/gstplaysink.c:
8330 playbin2: rescan audio sinks for volume/mute
8331 Rescan the audio sinks for the mute and volume properties.
8334 2009-03-23 19:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8336 * gst/playback/gstplaysink.c:
8337 playbin2: fix reuse of the video chains
8338 When reusing playbin with visualisations, reset the async property on the video
8339 sink because some sinks might dynamically recreate their sinks.
8342 2009-03-23 17:37:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8344 * gst/playback/gstplaysink.c:
8345 playbin2: allow dynamic swtiching of subtitles
8346 When we have the textpad configured, enable and disable the subtitles by setting
8347 the silent flag on the overlay element instead of trying to remove elements.
8350 2009-03-23 16:59:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8352 * tests/icles/playbin-text.c:
8353 tests: print some more info in the text example
8354 Print both the position and the running_time when the subtitle becomes available
8357 2009-03-23 16:04:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8359 * gst/playback/gstplaysink.c:
8360 playbin2: fix dynamic switching of visualisations
8361 Fix the switching of visualisations by requesting and releasing the tee request
8365 2009-03-23 16:19:11 +0200 Stefan Kost <ensonic@users.sf.net>
8368 * gst/tcp/gsttcpclientsink.c:
8369 * gst/tcp/gsttcpclientsrc.c:
8370 * gst/tcp/gsttcpserversink.c:
8371 * gst/tcp/gsttcpserversrc.c:
8372 docs: add examples for tcp elements, also use correct section name. Fixes #564139
8373 Updated the examples in the README to actually work. Add them to api docs. Tests
8374 the api-docs and fix the section names to make the docs actualy show up.
8375 The example for "tcpserversrc" needs review (might be an element bug).
8377 2009-03-17 09:14:02 +0200 Stefan Kost <ensonic@users.sf.net>
8379 * gst/videoscale/gstvideoscale.c:
8380 indent: fix damange that gst-indent did some time ago
8382 2009-03-23 15:27:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8384 * gst/playback/gstplaysink.c:
8385 playbin2: fix linking order
8386 Link after doing the state change and unlink before shutting down. Makes the
8387 window for causing races in toggling the visualisations smaller.
8390 2009-03-23 12:26:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8392 * gst/playback/gsturidecodebin.c:
8393 uridecodebin: reset counter
8394 reset the number of pending dynamic operations back to 0 when we reuse
8398 2009-03-23 11:38:53 +0100 Edward Hervey <bilboed@bilboed.com>
8400 * ext/theora/theoradec.c:
8401 theoradec: Use GST_CLOCK_TIME_NONE for invalid positions. Fixes #543591
8402 The problem was that previously we didn't check whether _theora_granule_frame
8403 returned a negative framecount or not, resulting in bogus timestamps.
8405 2009-03-21 09:46:28 +0100 René Stadler <mail@renestadler.de>
8407 * ext/vorbis/vorbisenc.c:
8408 vorbisenc: Set caps on non-header ouput buffers.
8411 2009-03-20 16:13:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8413 * tests/examples/seek/seek.c:
8414 seek: Add some more debug
8415 Add some more info about the selected streams.
8417 2009-03-20 15:47:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8419 * gst/playback/gstdecodebin2.c:
8420 decodebin2: a pad starts out being not drained.
8421 Mark a new pad as not drained until we get EOS on it.
8423 2009-03-20 14:17:19 +0100 LRN <lrn1986 at gmail dot com>
8425 * gst/playback/gstqueue2.c:
8426 win32: fix seeking in large files
8427 Fix Seeking in large files by using the 64-bit seek functions.
8430 2009-03-19 20:31:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8432 * gst/playback/gstdecodebin2.c:
8433 decodebin2: recover from failing to add a pad
8434 When we cannot add a pad to the decodebin2 for some reason, print a warning but
8435 continue adding the remaining pads.
8437 2009-03-19 19:35:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8439 * gst/playback/gstdecodebin2.c:
8440 decodebin2: more cleanups and docs.
8441 Add some more comments and use g_list_prepend().
8443 2009-03-19 19:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8445 * gst/playback/gstdecodebin2.c:
8446 decodebin2: refactoring and race fixes
8447 Refactor some code so that we can take the right locks and in the right order.
8448 Fixes quite a bit of races already.
8450 2009-03-19 19:03:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8452 * gst/playback/gstplaybin2.c:
8453 playbin2: remove the group cond + cleanups
8454 Remove the group GCond that we used for waiting for groups to finish because we
8455 use pad blocking on the selectors and counters instead for waiting for the
8457 remove the obsolete about_to_finish variable set while emiting the
8458 about-to-finish signal and fix some old comments.
8459 We don't need to take the playbin lock when querying the uridecodebin.
8461 2009-03-18 10:45:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8463 * tests/icles/playbin-text.c:
8464 icles: print better error and warning messages
8467 2009-03-17 22:53:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8469 * gst-libs/gst/rtsp/gstrtspbase64.c:
8470 * gst-libs/gst/rtsp/gstrtspbase64.h:
8471 rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode
8472 This also fixes another instance of CVE-2008-4316.
8474 2009-03-17 19:53:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8476 * ext/ogg/gstoggdemux.c:
8477 oggdemux: report -1 for duration in push mode
8478 In push mode we must return TRUE from the duration query with a value of -1
8479 meaning that we know that we don't know the duration.
8481 2009-03-17 19:09:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8483 * gst/playback/gstdecodebin2.c:
8484 decodebin2: add extra dynamic ref for demuxers
8485 When we make a group connected to a demuxer, keep an extra dynamic refcount for
8486 the group which is only decremented when no_more_pads or a multiqueue overrun is
8487 detected. This way we avoid a race between exposing the group while more dynamic
8488 refs are added from new pads.
8491 2009-03-17 15:39:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8493 * gst/playback/gstplaysink.c:
8494 playbin2: sync state of the sink correctly
8495 Sync the state of the newly added chains to the state of the parent sink element
8496 to avoid lost async-start messages. Fixes cdda:// async-done message storm.
8498 2009-03-17 11:54:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8500 * gst/playback/gstplaybin2.c:
8501 playbin2: return NOT_LINKED for unselected streams
8502 When streams are not selected in the selector, return NOT_LINKED so that
8503 upstream elements can skip decoding. Only do this for audio and video pads
8504 because for text streams the overhead is smaller and they could come from
8507 2009-03-17 11:51:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8509 * gst/playback/gstplaysink.c:
8510 playbin: set custom text sink properties
8511 Set the custom sink async=FALSE to not make it participate in preroll because we
8512 are dealing with sparse streams.
8513 Try to set sync=TRUE on the custom text sink.
8515 2009-03-17 11:30:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8517 * tests/icles/playbin-text.c:
8518 example: use appsink instead of fakesink
8519 Use appsink instead of fakesink to get the subtitles.
8520 Make things more pretty.
8522 2009-03-17 11:24:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8524 * tests/icles/.gitignore:
8525 * tests/icles/Makefile.am:
8526 * tests/icles/playbin-text.c:
8527 examples: add example of intercepting subtitles
8528 Add an example of how to install a custom sink for receiving subtitles in
8531 2009-03-17 11:03:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8533 * tests/check/elements/appsink.c:
8534 tests: fix include in the appsink test
8535 Fix dist by doing the right include.
8537 2009-03-16 16:42:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8539 * gst/playback/gstplaybin2.c:
8540 playbin2: don't try to set invalid stream numbers
8541 Fix a problem with setting the stream numbers because we check for the wrong
8545 2009-03-16 16:16:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8547 * gst/playback/gstplaybin2.c:
8548 playbin2: release the shutdown lock
8549 Release the shutdown lock when we wait for other groups to complete or else we
8550 have a deadlock when the other group completes and tries to grab the shutdown
8554 2009-03-16 15:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8556 * tests/examples/app/appsrc-ra.c:
8557 * tests/examples/app/appsrc-seekable.c:
8558 * tests/examples/app/appsrc-stream.c:
8559 * tests/examples/app/appsrc-stream2.c:
8560 examples: fix g_object_set() value type.
8561 Make sure we cast the length value as a gint64 to the vararg g_object_set() just
8562 incase sizeof(gsize) != sizeof(gint64).
8564 2009-03-15 19:57:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8566 * gst/typefind/gsttypefindfunctions.c:
8567 typefinding: make flac typefinder return lower probability for frame headers
8568 The flac frame header typefinder overstates the likelihood of a match, leading
8569 to false positives with e.g. aac streams and PDF files. Reduce probabilty
8570 returned from LIKELY to POSSIBLE for the frame header matchin code.
8573 2009-03-11 12:59:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8575 * gst/typefind/gsttypefindfunctions.c:
8576 typefinding: improve image/bmp typefinder
8577 Detect more variations and also bail out in more cases where the values
8578 don't make sense. Furthermore, add width/height and bpp to the caps,
8581 2009-03-13 15:22:42 +0000 Jan Schmidt <thaytan@noraisin.net>
8583 * tests/check/Makefile.am:
8584 check: Ignore alsamixer in the states test too
8586 2009-03-13 15:22:11 +0000 Jan Schmidt <thaytan@noraisin.net>
8588 * sys/v4l/v4l_calls.c:
8589 v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.
8591 2009-03-13 16:19:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8593 * gst-libs/gst/rtsp/gstrtspconnection.c:
8594 rtsp: fix resolving of hostnames
8595 We were returning a pointer to a stack variable with the resolved hostname,
8597 return a copy of the resolved ip address instead.
8600 2009-03-13 15:29:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8602 * ext/vorbis/vorbisparse.c:
8603 vorbisparse: be smarter when queueing headers
8604 Look at the first buffer byte to see if a buffer is a header instead of counting
8607 2009-03-13 15:27:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8609 * ext/theora/gsttheoraparse.h:
8610 * ext/theora/theoraparse.c:
8611 theoraparse: be smarter when queuing headers
8612 Look at the first byte of the buffer data (if we can) to decide if the packet is
8613 a header packet or not instead of counting packets.
8615 2009-03-13 15:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8617 * ext/ogg/gstoggdemux.c:
8618 oggdemux: add some debug info
8619 Add some debug info to log when the seek worked.
8621 2009-03-13 15:14:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8623 * gst-libs/gst/app/gstappsrc.c:
8624 appsrc: release lock in _eos flushing case
8625 Release the mutex when we are flushing in gst_app_src_end_of_stream()
8628 2009-03-13 11:49:10 +0000 Jan Schmidt <thaytan@noraisin.net>
8630 * ext/vorbis/vorbisdec.c:
8631 vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.
8633 2009-03-13 11:48:28 +0000 Jan Schmidt <thaytan@noraisin.net>
8635 * ext/theora/theoradec.c:
8636 theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.
8638 2009-03-12 18:27:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8640 * gst/playback/gsturidecodebin.c:
8641 playbin2: fix raw elements like cdda://
8642 Fix a fixme with a one liner and make cd playback work again.
8644 2009-03-12 17:47:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8646 * gst/playback/gstplaybin2.c:
8647 * gst/playback/gstplaysink.c:
8648 * gst/playback/gstplaysink.h:
8649 playbin2: improve subtitle handling
8650 Add property to playbin2 to configure a custom sink that receives the raw
8651 subtitle buffers instead of using a textoverlay.
8652 Improve the property finding code to make it more usable.
8653 Use property find code to find async properties in custom sinks that are bins.
8654 Improve text overlay code to gracefully handle missing elements.
8656 2009-02-24 15:58:42 +0000 Jan Schmidt <thaytan@noraisin.net>
8658 * gst-libs/gst/tag/gstvorbistag.c:
8659 vorbistag: Protect memory allocation calculation from overflow.
8660 Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
8662 2009-03-12 11:34:20 +0000 Jan Urbanski <jurbanski@flumotion.com>
8664 * gst-plugins-base.spec.in:
8667 2009-03-11 18:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8669 * gst-libs/gst/rtsp/gstrtspconnection.c:
8670 rtsp: fix parsing of the timeout parameter
8673 2009-03-11 16:20:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8675 * gst-libs/gst/rtsp/gstrtspmessage.c:
8676 rtsp: fix g_return condition
8677 when parsing a data message, we require a data message.
8679 2009-03-11 13:33:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8681 * gst/typefind/gsttypefindfunctions.c:
8682 typefinding: flac typefinder fixes
8683 Use scan context for initial peek as well. Peek 6 bytes in the initial
8684 peek rather than 5 bytes, to match the length of the memcmp we're doing
8685 on that data later. Return immediately when we found caps from looking
8686 at the beginning of the data - no point in continuing to scan the next
8687 64kB for something matching a frame header.
8689 2009-03-11 14:08:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8691 * gst-libs/gst/rtsp/gstrtspmessage.c:
8692 rtsp: free the right string.
8693 Free the key value before we remove the header item from the array. The item we
8694 retrieved from the array is only valid until we remove it from the array.
8696 2009-03-11 14:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8698 * gst-libs/gst/rtsp/gstrtspconnection.c:
8699 rtsp: keep track of amount of decoded bytes
8700 Keep track of the actual amount of decoded bytes, which can be less than 3 when
8701 we decode the last bits of a base64 message.
8703 2009-03-10 21:00:26 +0200 Stefan Kost <ensonic@users.sf.net>
8705 * gst/adder/gstadder.c:
8706 adder: log details in getcaps like in setcaps
8708 2009-03-10 13:11:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8711 win32: update MANIFEST, fixing 'make dist'
8713 2009-03-09 23:12:00 +0000 Jan Schmidt <thaytan@noraisin.net>
8716 Automatic update of common submodule
8717 From 7032163 to f8b3d91
8719 2009-03-09 16:19:40 +0100 Jonathan Matthew <notverysmart at gmail dot com>
8721 * gst/typefind/gsttypefindfunctions.c:
8722 typefind: add photoshop typefind functions
8723 Add photoshop typefind functions.
8726 2009-03-09 15:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8728 * gst/playback/gstdecodebin2.c:
8729 decodebin2: only remove pads that were added
8730 Flag pads that were added so that we can see if we need to remove them later or
8733 2009-03-09 13:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8735 * gst-libs/gst/rtsp/gstrtsptransport.c:
8736 rtsp: only add ports when not using TCP
8737 Only add the port numbers in the transport string when we are using udp or
8740 2009-03-09 13:53:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8742 * gst-libs/gst/rtsp/gstrtspmessage.c:
8743 rtsp: use gstreamer dump mem
8746 2009-03-09 13:51:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8748 * gst-libs/gst/rtsp/gstrtspconnection.c:
8749 rtsp: use glib base64 encoder
8752 2009-03-06 19:28:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
8754 * gst/playback/gstdecodebin2.c:
8755 Unblock blocked ghostpads when shutting down. Fixes #574293.
8757 2009-03-09 10:03:13 +0100 Edward Hervey <bilboed@bilboed.com>
8759 * gst-libs/gst/riff/riff-media.c:
8760 Riff: Add mapping for Fraps video codec.
8761 Found through insanity testrun. Confirmed mapping in libavformat.
8763 2009-03-09 09:07:13 +0100 Edward Hervey <bilboed@bilboed.com>
8765 * gst-libs/gst/riff/riff-media.c:
8766 riff: Add the 'DVR ' mapping for mpeg2video.
8767 Found this in 3 files from the insanity suite and mapping is also present
8770 2009-03-09 09:06:40 +0100 Edward Hervey <bilboed@bilboed.com>
8772 * gst/typefind/gsttypefindfunctions.c:
8773 typefind: Use the proper data pointer instead of poking random memory.
8775 2009-03-08 18:17:48 +0100 LRN <lrn1986@gmail.com>
8777 * gst-libs/gst/rtsp/gstrtspconnection.c:
8778 rtsp: fix compilation on windows.
8779 Remove unused variable when building for windows.
8782 2009-03-08 12:03:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8785 Automatic update of common submodule
8786 From ffa738d to 7032163
8788 2009-03-08 11:19:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8791 Automatic update of common submodule
8792 From 3f13e4e to ffa738d
8794 2009-03-07 11:44:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8797 Automatic update of common submodule
8798 From 3c7456b to 3f13e4e
8800 2009-03-07 10:44:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8803 Automatic update of common submodule
8804 From 57c83f2 to 3c7456b
8806 2009-03-06 19:02:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8808 * ext/theora/theoradec.c:
8809 theoradec: parse and use codec_data in the caps
8810 Parse the codec_data in the caps and use this as the headers.
8813 2009-03-06 18:53:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8815 * gst-libs/gst/riff/riff-media.c:
8816 riff: add theora mapping
8817 Add theora mappings. See #574169.
8819 2009-03-06 16:31:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8821 * gst-libs/gst/rtsp/gstrtspconnection.c:
8822 * gst-libs/gst/rtsp/gstrtspconnection.h:
8823 * win32/common/libgstrtsp.def:
8824 rtsp: Add methods for getting the read/write fds
8825 API:gst_rtsp_connection_get_readfd()
8826 API:gst_rtsp_connection_get_writefd()
8828 2009-03-06 10:35:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8831 * win32/common/audio-enumtypes.c:
8832 win32: indent copied *-enumtypes.c files in make win32-update
8834 2009-03-06 10:35:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8837 win32: update MANIFEST
8839 2009-03-06 10:30:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8842 * win32/common/config.h:
8843 win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define
8845 2009-03-06 10:05:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8847 * win32/common/_stdint.h:
8848 * win32/common/config.h:
8849 * win32/common/gstrtsp-enumtypes.c:
8850 * win32/common/interfaces-enumtypes.c:
8851 * win32/common/multichannel-enumtypes.c:
8852 * win32/common/pbutils-enumtypes.c:
8853 * win32/common/video-enumtypes.c:
8854 * win32/common/video-enumtypes.h:
8855 win32: update windows files via make win32-update
8856 Updates win32 files using the new system/hook, and defines HAVE_PROCESS_H,
8857 which fixes the build of pbutils on windows (#574319).
8859 2009-03-06 10:03:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8862 gitignore: ignore more
8864 2009-03-06 10:37:38 +0100 Julien Moutte <julien@fluendo.com>
8866 * gst-libs/gst/rtsp/gstrtspconnection.c:
8867 Fix build on Mac OS X
8869 2009-03-05 15:42:23 -0800 Michael Smith <msmith@songbirdnest.com>
8871 * gst/playback/gstdecodebin2.c:
8872 decodebin2: don't stay connected to notify::caps after negotiation
8873 Disconnect the notify::caps signal in our callback (it'll be re-added
8874 if we're not, in fact, finished getting complete caps). Ensures that
8875 caps changes mid-stream (e.g. from an mp3 that changes from
8876 stereo->mono mid-file) don't cause us to try to add a new pad.
8878 2009-03-05 13:48:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8880 * gst-libs/gst/rtsp/gstrtsprange.c:
8881 rtsp: fix parsing of 'now-' ranges.
8884 2009-03-05 12:43:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8886 * tests/examples/dynamic/.gitignore:
8887 * tests/examples/dynamic/Makefile.am:
8888 * tests/examples/dynamic/sprinkle.c:
8889 * tests/examples/dynamic/sprinkle2.c:
8890 * tests/examples/dynamic/sprinkle3.c:
8891 examples: add some more sprinkle examples
8892 Add some more sprinle examples and add some more comments.
8895 2009-03-05 11:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8897 * docs/plugins/gst-plugins-base-plugins-sections.txt:
8898 docs: add appsrc symbols to standard section
8901 2009-03-05 12:27:16 +0200 Stefan Kost <ensonic@users.sf.net>
8903 * gst/adder/gstadder.c:
8904 adder: add variants for unsigned to fix warnings for unneeded check
8905 For unsigned int out+in can't be < 0.
8907 2009-03-05 10:58:12 +0200 Stefan Kost <ensonic@users.sf.net>
8909 * gst/subparse/gstsubparse.c:
8910 subparse: use the right variable in debug log, encoding is not yet initialized
8912 2009-03-05 10:51:25 +0200 Stefan Kost <ensonic@users.sf.net>
8914 * sys/v4l/v4l_calls.c:
8915 v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix
8917 2009-03-05 10:39:33 +0200 Stefan Kost <ensonic@users.sf.net>
8919 * gst/audioresample/gstaudioresample.c:
8920 audioresample: add missing break in event handling, remove dead code
8922 2009-03-04 16:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8924 * gst-libs/gst/rtsp/gstrtspconnection.c:
8925 rtsp: do some more cleanup in _close
8926 Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
8927 unconnected state as it was allocated.
8929 2009-03-04 16:11:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8931 * gst-libs/gst/rtsp/gstrtspconnection.c:
8932 * gst-libs/gst/rtsp/gstrtspconnection.h:
8933 rtsp: fix the memory management of the url
8934 Constify the url parameter in _create.
8935 Make a copy of the url stored in the connection.
8936 Free the url when the connection is freed.
8938 2009-03-04 12:21:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8940 * docs/libs/gst-plugins-base-libs-sections.txt:
8941 * gst-libs/gst/rtsp/gstrtspconnection.c:
8942 * gst-libs/gst/rtsp/gstrtspconnection.h:
8943 * win32/common/libgstrtsp.def:
8944 RTSP: Add support for server tunneling
8945 Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
8946 that a server can store and match the id against other tunnel requests.
8947 Fix the URI in the tunnel requests so that they contain the absolute uri and the
8948 query string if any instead of just the hostname.
8949 Transparently base64 decode the input stream when tunneling.
8950 Add method to set the connection ip address so that it can be included in the
8952 Add method to connect the two tunnel requests.
8953 Add two callbacks for the async mode to notify a tunnel start and tunnel
8955 Add method to reset the watch after the connection has been tunneled.
8956 Various little refactoring to make more stuff reusable.
8957 API: RTSP::gst_rtsp_connection_set_ip()
8958 API: RTSP::gst_rtsp_connection_get_tunnelid()
8959 API: RTSP::gst_rtsp_connection_do_tunnel()
8960 API: RTSP::gst_rtsp_watch_reset()
8962 2009-03-04 12:18:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8964 * gst-libs/gst/rtsp/gstrtspdefs.c:
8965 * gst-libs/gst/rtsp/gstrtspdefs.h:
8966 rtsp: add new defines for tunneling
8967 Add two more result codes for tunneling support.
8969 2009-03-04 12:12:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8971 * gst-libs/gst/rtsp/gstrtspmessage.h:
8972 rtsp: remove , from last enum member
8973 Remove , from last enum member to improve compatibility with other compilers.
8975 2009-02-28 15:23:20 -0800 LRN <lrn1986@gmail.com>
8977 * gst/subparse/gstsubparse.c:
8978 subparse: Convert regex code to GRegex code
8979 Fixes: #572993. Patch author prefers to use an alias, contact
8980 ds if you actually need a real name.
8981 Signed-off-by: David Schleef <ds@schleef.org>
8983 2009-03-02 16:13:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8985 * gst-libs/gst/rtsp/gstrtspconnection.c:
8986 rtsp: remove debugging g_message
8989 2009-03-02 16:03:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8991 * docs/libs/gst-plugins-base-libs-sections.txt:
8992 * gst-libs/gst/rtsp/gstrtspconnection.c:
8993 * gst-libs/gst/rtsp/gstrtspconnection.h:
8994 * win32/common/libgstrtsp.def:
8995 RTSP: add support for Quicktime tunneled RTSP
8996 Add support for tunneling RTSP over HTTP.
8997 Fix documentation some more.
8999 API: RTSP:gst_rtsp_connection_is_tunneled()
9000 API: RTSP:gst_rtsp_connection_set_tunneled()
9002 2009-03-02 15:48:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9004 * gst-libs/gst/rtsp/gstrtsptransport.h:
9005 * gst-libs/gst/rtsp/gstrtspurl.c:
9006 RTSP: parse rtsph uris as RTSP tunneled over HTTP
9007 Add transport define for RTSP tunneled over HTTP.
9008 Parse rtsph:// uris as tunneled HTTP over TCP.
9009 API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
9012 2009-03-02 12:48:18 +0100 Edward Hervey <bilboed@bilboed.com>
9014 * win32/common/libgstrtsp.def:
9015 win32: Add gst_rtsp_connection_get_url definition
9016 No, I'm not wim's buildslave, seriously.
9018 2009-03-02 10:58:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9020 * gst-libs/gst/rtsp/gstrtspconnection.c:
9021 * gst-libs/gst/rtsp/gstrtspconnection.h:
9022 rtsp: add _get_url method and separate sockets
9023 Add gst_rtsp_connection_get_url() method.
9024 Reserve space for 2 sockets, one for reading and one for writing. Use socket
9025 pointers to select the read and write sockets. This should allow us to implement
9026 tunneling over HTTP soon.
9027 API: RTSP::gst_rtsp_connection_get_url()
9029 2009-03-01 18:31:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9031 * gst-libs/gst/app/gstapp-marshal.list:
9032 app: force automatic rebuild of gstapp-marshal.[ch] after previous change
9033 The previous change to appsrc/appsink requires people to 'make clean'
9034 to get the marshallers rebuilt (causing a build failure otherwise).
9035 Change some lines in the .list file around to force a rebuild of
9036 these files automatically.
9038 2009-02-28 11:07:04 -0800 David Schleef <ds@schleef.org>
9041 Bump glib requirement to 2.14
9043 2009-02-28 19:37:53 +0100 LRN <lrn1986@gmail.com>
9045 * ext/gio/gstgiobasesink.c:
9046 gio: Use correct format modifier for size_t
9049 2009-02-28 19:35:33 +0100 LRN <lrn1986@gmail.com>
9051 * gst-libs/gst/rtsp/gstrtspconnection.c:
9052 rtspconnection: Use correct types for some functions on Win32
9055 2009-02-28 13:11:59 +0100 Edward Hervey <bilboed@bilboed.com>
9057 * gst-libs/gst/rtsp/gstrtspconnection.c:
9058 rtspconnection: Fix warning about using unitialized value.
9060 2009-02-28 12:41:28 +0100 Edward Hervey <bilboed@bilboed.com>
9062 * gst-libs/gst/riff/riff-ids.h:
9063 * gst-libs/gst/riff/riff-media.c:
9064 riff: Add more codec mappings.
9065 This comes mostly from a review of ffmpeg/libavformat/riff.c
9067 2009-02-27 11:14:25 +0200 Stefan Kost <ensonic@users.sf.net>
9069 * ext/alsa/gstalsa.c:
9070 alsa: release pcminfo after the strdup
9072 2009-02-26 17:38:47 +0200 Stefan Kost <ensonic@users.sf.net>
9074 * gst-libs/gst/rtsp/gstrtsprange.c:
9075 rtsprange: don't leak the range in case of parsing error.
9076 Free the gstRTSPTimeRange if we don't return it. Also simplify
9077 gst_rtsp_range_free() as it is valid to pass NULL to g_free().
9079 2009-02-26 16:47:39 +0200 Stefan Kost <ensonic@users.sf.net>
9081 * ext/alsa/gstalsa.c:
9082 alsa: cleanup name lookup.
9083 We can break, once we have a name to make sure, we won't read it ever twice.
9085 2009-02-26 16:09:03 +0200 Stefan Kost <ensonic@users.sf.net>
9087 * gst/subparse/gstsubparse.c:
9088 subparse: don't leak line, if flushing
9090 2009-02-26 16:03:39 +0200 Stefan Kost <ensonic@users.sf.net>
9092 * ext/gio/gstgiosink.c:
9093 giosink: reflow error handling to not leak uri
9095 2009-02-26 15:53:10 +0200 Stefan Kost <ensonic@users.sf.net>
9097 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
9098 * gst/ffmpegcolorspace/imgconvert.c:
9099 ffmpegcolorspace: remove unused code/variables
9101 2009-02-26 12:10:47 +0200 Stefan Kost <ensonic@users.sf.net>
9103 * sys/ximage/ximagesink.c:
9104 ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps
9106 2009-02-26 16:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9108 * docs/libs/gst-plugins-base-libs-sections.txt:
9109 * gst-libs/gst/app/gstappsink.c:
9110 * gst-libs/gst/app/gstappsrc.c:
9111 * gst-libs/gst/app/gstappsrc.h:
9112 * win32/common/libgstapp.def:
9113 app: add callbacks to appsrc, cleanups
9114 Add a uri handler to appsink.
9115 don't emit signals when we have installed callbacks on appsink.
9116 Add callbacks to appsrc to replace the signals.
9117 Add property to disable callbacks in appsrc, default to TRUE for backwards
9118 compatibility but disable when callbacks are installed.
9119 API: GstAppSrc::emit-signals
9120 API: GstAppSrc::gst_app_src_set_emit_signals()
9121 API: GstAppSrc::gst_app_src_get_emit_signals()
9122 API: GstAppSrc::gst_app_src_set_callbacks()
9124 2009-02-26 11:42:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9126 * docs/libs/gst-plugins-base-libs-sections.txt:
9127 * gst-libs/gst/app/gstappsink.h:
9128 * tests/check/elements/appsink.c:
9129 Appsink: add padding for callbacks + docs
9130 Add some padding to the callbacks structure just to be safe.
9131 Remove the now invisible marshaller methods from the docs.
9132 Fix a comment in the unit test.
9134 2009-02-26 09:52:59 +0100 Edward Hervey <bilboed@bilboed.com>
9136 * win32/common/libgstapp.def:
9137 win32: Add new libgstapp symbol
9139 2009-02-26 10:07:21 +0200 Stefan Kost <ensonic@users.sf.net>
9141 * docs/plugins/gst-plugins-base-plugins-sections.txt:
9142 docs: clean section.txt file.
9143 Add appsrc/sink symbols to private, as they are covered in the libs docs.
9145 2009-02-26 10:06:23 +0200 Stefan Kost <ensonic@users.sf.net>
9147 * gst/playback/gstplaybasebin.c:
9148 docs: fix random text after since: tag. Also fix class name to make the docs actual appear.
9150 2009-02-26 09:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
9152 * docs/plugins/gst-plugins-base-plugins.args:
9153 * docs/plugins/gst-plugins-base-plugins.hierarchy:
9154 * docs/plugins/gst-plugins-base-plugins.interfaces:
9155 * docs/plugins/gst-plugins-base-plugins.prerequisites:
9156 * docs/plugins/inspect/plugin-adder.xml:
9157 * docs/plugins/inspect/plugin-alsa.xml:
9158 * docs/plugins/inspect/plugin-app.xml:
9159 * docs/plugins/inspect/plugin-audioconvert.xml:
9160 * docs/plugins/inspect/plugin-audiorate.xml:
9161 * docs/plugins/inspect/plugin-audioresample.xml:
9162 * docs/plugins/inspect/plugin-audiotestsrc.xml:
9163 * docs/plugins/inspect/plugin-cdparanoia.xml:
9164 * docs/plugins/inspect/plugin-decodebin.xml:
9165 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
9166 * docs/plugins/inspect/plugin-gdp.xml:
9167 * docs/plugins/inspect/plugin-gio.xml:
9168 * docs/plugins/inspect/plugin-gnomevfs.xml:
9169 * docs/plugins/inspect/plugin-libvisual.xml:
9170 * docs/plugins/inspect/plugin-ogg.xml:
9171 * docs/plugins/inspect/plugin-pango.xml:
9172 * docs/plugins/inspect/plugin-playback.xml:
9173 * docs/plugins/inspect/plugin-queue2.xml:
9174 * docs/plugins/inspect/plugin-subparse.xml:
9175 * docs/plugins/inspect/plugin-tcp.xml:
9176 * docs/plugins/inspect/plugin-theora.xml:
9177 * docs/plugins/inspect/plugin-typefindfunctions.xml:
9178 * docs/plugins/inspect/plugin-uridecodebin.xml:
9179 * docs/plugins/inspect/plugin-video4linux.xml:
9180 * docs/plugins/inspect/plugin-videorate.xml:
9181 * docs/plugins/inspect/plugin-videoscale.xml:
9182 * docs/plugins/inspect/plugin-videotestsrc.xml:
9183 * docs/plugins/inspect/plugin-volume.xml:
9184 * docs/plugins/inspect/plugin-vorbis.xml:
9185 * docs/plugins/inspect/plugin-ximagesink.xml:
9186 * docs/plugins/inspect/plugin-xvimagesink.xml:
9187 * gst/playback/gstplaybin2.c:
9188 docs: playbin2 has no stream-info
9190 2009-02-26 09:53:03 +0200 Stefan Kost <ensonic@users.sf.net>
9192 * gst-libs/gst/video/video.h:
9193 docs: fix newly added interlace constants and plug holes in video format docs
9195 2009-02-26 09:35:43 +0200 Stefan Kost <ensonic@users.sf.net>
9197 * gst-libs/gst/app/gstappsink.c:
9198 * gst-libs/gst/app/gstappsrc.c:
9199 * gst-libs/gst/audio/gstaudiofilter.c:
9200 * gst-libs/gst/audio/gstringbuffer.c:
9201 * gst-libs/gst/rtp/gstrtcpbuffer.c:
9202 docs: don't put random stuff in tags.
9203 Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
9204 tag to append text again to the documentation body.
9206 2009-02-06 11:10:15 +0200 Stefan Kost <ensonic@users.sf.net>
9208 * sys/ximage/ximagesink.c:
9209 ximagsink: do not access uninitialized height variable.
9210 Exit like in xvimagesink, if we have partial caps.
9212 2009-02-25 20:26:05 -0800 David Schleef <ds@schleef.org>
9216 * win32/common/config.h.in:
9217 Change how win32/common/config.h is updated
9218 Generate win32/common/config.h-new directly from config.h.in,
9219 using shell variables in configure and some hard-coded information.
9220 Change top-level makefile so that 'make win32-update' copies the
9221 generated file to win32/common/config.h, which we keep in source
9222 control. It's kept in source control so that the git tree is
9224 This change is similar to the one recently applied to GStreamer,
9225 except that it adds a few -base specific defines.
9227 2009-02-25 19:40:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9229 * gst-libs/gst/app/Makefile.am:
9230 * gst-libs/gst/app/gstappsink.c:
9231 * gst-libs/gst/app/gstappsrc.c:
9232 * win32/common/libgstapp.def:
9233 app: add win32 .def file and only export functions we want exported
9234 Add a .def file for win32 builds (and make check-exports).
9235 Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes #573165).
9236 Make sure private marshaller functions aren't exported by prefixing them with __gst;
9237 also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
9238 a comment why we're not using glib-genmarshal for this one.
9240 2009-02-25 17:08:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9242 * tests/examples/dynamic/.gitignore:
9243 * tests/examples/dynamic/Makefile.am:
9244 * tests/examples/dynamic/sprinkle.c:
9245 sprinkle: Add another example app
9246 Add an example app that dynamically adds and removes audiotestsrc elements from
9249 2009-02-25 16:25:33 +0100 Peter Kjellerstedt <pkj@axis.com>
9251 * gst-libs/gst/rtsp/gstrtspconnection.c:
9254 2009-02-25 11:31:02 +0100 Peter Kjellerstedt <pkj@axis.com>
9256 * gst-libs/gst/rtsp/gstrtspconnection.c:
9257 * gst/tcp/gstmultifdsink.c:
9258 rtsp, multifdsink: Unify the use of union gst_sockaddr.
9260 2009-02-25 14:22:35 +0000 Jan Schmidt <thaytan@noraisin.net>
9264 build: Update shave init statement for changes in common. Bump common.
9266 2009-02-25 13:16:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9268 * sys/xvimage/xvimagesink.c:
9269 * sys/xvimage/xvimagesink.h:
9270 xvimageink: protect buffer_alloc from shutdown
9271 Use the pool_lock in the buffer_alloc function to detect shutdown. Avoids
9272 crashes when the sink is shutdown.
9274 2009-02-25 12:43:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9276 * gst/playback/gstplaybin2.c:
9277 playbin: use flushing pads instead of fakesink
9278 Use the flushing pads on playsink to terminate on shutdown instead of plugging
9279 fakesinks. this should be a little cheaper.
9281 2009-02-25 12:42:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9283 * gst/playback/gstplaysink.c:
9284 * gst/playback/gstplaysink.h:
9285 playsink: Add FLUSHING pad type
9286 Make it possible to request a flushing pad from the playsink. We can eventually
9287 use these flushing pads to quickly terminate the dataflow when we are shutting
9290 2009-02-25 11:31:52 +0000 Jan Schmidt <thaytan@noraisin.net>
9293 Automatic update of common submodule
9294 From 9cf8c9b to a6ce5c6
9296 2009-02-25 09:52:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9298 * gst-libs/gst/riff/riff-media.c:
9299 riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
9302 2009-02-25 12:07:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9304 * tests/icles/stress-playbin.c:
9305 stress-playbin: print the current uri
9306 Print the current uri so that we can more easily see what uri caused a crash or
9309 2009-02-25 11:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9311 * tests/icles/stress-playbin.c:
9312 Print the errors more clearly
9313 Print some more verbose messages when dealing with errors.
9315 2009-02-25 10:08:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9317 * gst/playback/gstplaybin2.c:
9318 Release the group lock when setting states
9319 Release the group lock while we perform the state changes on the uridecodebins
9320 because that might trigger callbacks that we need to handle with the group lock
9321 taken. Avoids a possible deadly embrace in some id3/flac files.
9324 2009-02-25 10:05:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9326 * gst/playback/gstdecodebin2.c:
9327 Combine finding and creating groups
9328 Combine the search for the current group and optionally creating one into one
9329 function so that we can avoid taking the lock multiple times.
9331 2009-02-25 08:22:00 +0100 Edward Hervey <bilboed@bilboed.com>
9333 * gst/playback/gstplaybin2.c:
9334 Playbin2: Don't leave unused parameters in debug statements.
9335 Fixes build on macosx
9337 2009-02-24 10:33:05 +0100 Edward Hervey <bilboed@bilboed.com>
9339 * gst-libs/gst/riff/riff-media.c:
9340 Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)
9342 2009-02-24 18:43:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9344 * gst/playback/gstplaybin2.c:
9345 Add some G_UNLIKELY because we can
9346 Add a G_UNLIKELY when checking the shutdown variable.
9348 2009-02-24 17:23:58 +0000 Garret D'Amore <garrett.damore@sun.com>
9350 * gst-libs/gst/interfaces/mixer.h:
9351 * gst-libs/gst/interfaces/mixertrack.h:
9352 mixer interface: Add flags to enhance mixer interfaces
9353 This patch adds a few flags to the mixer and mixerctrl interface to
9354 better support OSSv4 (and potentially other backends).
9355 Patch By: Garret D'Amore <garrett.damore@sun.com>
9356 Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>
9357 API: GST_MIXER_FLAG_HAS_WHITELIST, GST_MIXER_FLAG_GROUPING,
9358 API: GST_MIXER_TRACK_NO_RECORD, GST_MIXER_TRACK_NO_MUTE,
9359 API: GST_MIXER_TRACK_WHITELIST
9361 2009-02-24 17:03:08 +0000 Jan Schmidt <thaytan@noraisin.net>
9363 * gst/tcp/gstmultifdsink.c:
9364 multifdsink: Fix strict aliasing error using a union
9366 2009-02-24 16:49:40 +0000 Jan Schmidt <thaytan@noraisin.net>
9368 * gst-libs/gst/rtsp/gstrtspconnection.c:
9369 rtsp: Fix a strict aliasing warning
9370 Fix strict aliasing warnings from casting a sockaddr_storage and
9371 using it as a sockaddr_in6. Use a union instead.
9373 2009-02-24 16:08:49 +0000 Jan Schmidt <thaytan@noraisin.net>
9375 * docs/libs/.gitignore:
9376 * docs/libs/tmpl/.gitignore:
9377 * docs/plugins/.gitignore:
9378 * docs/plugins/tmpl/.gitignore:
9379 Remove .gitignore files from the docs tmpl dirs, that are killed by make clean.
9381 2009-02-24 14:36:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9383 * docs/plugins/Makefile.am:
9384 * ext/vorbis/Makefile.am:
9385 * ext/vorbis/gstvorbisdec.h:
9386 * ext/vorbis/gstvorbisenc.h:
9387 * ext/vorbis/gstvorbisparse.h:
9388 * ext/vorbis/gstvorbistag.h:
9389 * ext/vorbis/vorbis.c:
9390 * ext/vorbis/vorbisdec.c:
9391 * ext/vorbis/vorbisdec.h:
9392 * ext/vorbis/vorbisenc.c:
9393 * ext/vorbis/vorbisenc.h:
9394 * ext/vorbis/vorbisparse.c:
9395 * ext/vorbis/vorbisparse.h:
9396 * ext/vorbis/vorbistag.c:
9397 * ext/vorbis/vorbistag.h:
9398 vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
9400 2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9402 * gst/ffmpegcolorspace/avcodec.h:
9403 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
9404 * gst/ffmpegcolorspace/imgconvert.c:
9405 ffmpegcolorspace: Add conversion from/to YVYU colorspace
9408 2009-02-24 13:42:01 +0100 Jonas Danielsson <jonas.danielsson@axis.com>
9410 * gst/ffmpegcolorspace/imgconvert.c:
9411 ffmpegcolorspace: Add direct UYVY->GRAY8 conversion
9412 The conversion from UYVY to RGB24 and then to GRAY8
9413 is quite slow. Fixes bug #569655.
9415 2009-02-19 17:16:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9417 * gst/playback/gstplaybin2.c:
9418 playbin2: fix deadlock when shutting down. Fixes #572577.
9420 2009-02-19 17:15:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9422 * tests/icles/stress-playbin.c:
9423 stress-playbin: make more flexible, e.g. also useful for playbin2
9425 2009-02-24 12:11:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9427 * gst-libs/gst/rtsp/gstrtspconnection.c:
9428 Match WSAStartup and WSACleanup correctly
9429 Don't randomly call WSAStartup and WSACleanup but instead call the startup when
9430 we create a connection and cleanup when we free it again. Because the internal
9431 datastructure is refcounted, this should not cause any refcounting leaks when
9432 the connection is managed correctly.
9435 2009-02-18 11:59:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9437 * gst/playback/gstplaysink.c:
9438 playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105.
9440 2009-02-23 10:57:42 -0800 David Flynn <davidf@rd.bbc.co.uk>
9442 * pkgconfig/gstreamer-app-uninstalled.pc.in:
9443 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
9444 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
9445 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
9446 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
9447 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
9448 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
9449 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
9450 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
9451 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
9452 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
9453 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
9454 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
9455 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
9456 * pkgconfig/gstreamer-video-uninstalled.pc.in:
9457 Add srcdir to includes for out-of-source builds
9458 When you use gstreamer uninstalled and build outside
9459 the source tree, the includes need to be specified for
9460 both the source tree and the build tree.
9461 Signed-off-by: David Schleef <ds@schleef.org>
9463 2009-02-22 17:23:52 +0000 Jan Schmidt <thaytan@noraisin.net>
9466 * docs/libs/Makefile.am:
9467 * docs/plugins/Makefile.am:
9468 Use shave for the build output
9470 2009-02-23 12:17:07 +0100 Edward Hervey <bilboed@bilboed.com>
9472 * win32/common/libgstrtsp.def:
9473 win32: Add new symbol to libgstrtsp.def
9475 2009-02-23 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9477 * gst-libs/gst/rtsp/gstrtspextension.c:
9478 * gst-libs/gst/rtsp/gstrtspextension.h:
9479 Add method for handling server requests
9480 Add a receive_request so that extensions can react to server requests.
9482 2009-02-22 19:20:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9484 * tests/check/libs/netbuffer.c:
9485 Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)
9487 2009-02-22 19:19:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9489 * ext/theora/theoraparse.c:
9490 theoraparse: Use the correct unref functions
9492 2009-02-22 19:18:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9494 * sys/ximage/ximagesink.c:
9495 * sys/xvimage/xvimagesink.c:
9496 x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()
9498 2009-02-22 19:12:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9500 * gst-libs/gst/tag/gsttagdemux.c:
9501 tagdemux: Unref the actual buffer instead of the memory address of the buffer
9503 2009-02-22 15:47:53 +0000 Jan Schmidt <thaytan@noraisin.net>
9506 Automatic update of common submodule
9507 From 5d7c9cc to 9cf8c9b
9509 2009-02-22 14:49:29 +0100 Edward Hervey <bilboed@bilboed.com>
9511 * win32/common/libgstrtsp.def:
9512 * win32/common/libgstvideo.def:
9513 win32/common: Update .def files for recent API addition
9515 2009-02-22 13:43:35 +0100 Edward Hervey <bilboed@bilboed.com>
9517 * tests/check/libs/rtp.c:
9518 tests: Fix indentation
9520 2009-02-22 13:42:33 +0100 Edward Hervey <bilboed@bilboed.com>
9522 * gst-libs/gst/video/video.c:
9523 libs/video: Fix gst_video_format_new_caps* functions.
9524 Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
9527 2009-02-21 11:13:36 -0800 David Schleef <ds@schleef.org>
9530 Automatic update of common submodule
9531 From 80c627d to 5d7c9cc
9533 2009-02-20 17:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9535 * gst-libs/gst/rtsp/gstrtspmessage.c:
9536 Improve key/value parsing
9537 Improve header field parsing by keeping a ref to the key/value instead of
9538 copying it into a local variable.
9540 2009-02-20 12:35:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9542 * gst-libs/gst/rtsp/gstrtspconnection.c:
9543 Add trailing \0 to message length
9544 We always put a trailing 0 at the end of the message body. Reflect this fact in
9545 the length of the message.
9547 2009-02-20 09:50:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9549 * gst-libs/gst/rtsp/gstrtspconnection.c:
9550 Don't parse headers for data messages
9551 Don't try to parse the headers on a data message because they don't have
9554 2009-02-19 12:18:29 -0800 Benjamin M. Schwartz <bens@alum.mit.edu>
9556 * ext/theora/gsttheoraenc.h:
9557 * ext/theora/theoraenc.c:
9558 theoraenc: Add property for speed level control
9559 Add property "speed-level" to control the amount of motion searching
9560 the encoder does. This is only available in libtheora >= 1.0 and
9561 will silently fail with earlier libraries. Fixes: #572275.
9562 Signed-off-by: David Schleef <ds@schleef.org>
9564 2009-02-19 17:40:45 +0100 Edward Hervey <bilboed@bilboed.com>
9566 * gst-libs/gst/video/video.c:
9567 * gst-libs/gst/video/video.h:
9568 video: Fix 'Since' tags
9570 2009-01-26 10:30:53 +0100 Edward Hervey <bilboed@bilboed.com>
9572 * docs/libs/gst-plugins-base-libs-sections.txt:
9573 * gst-libs/gst/video/video.c:
9574 * gst-libs/gst/video/video.h:
9575 video: Add flags for interlaced video along with convenience methods for interlaced caps.
9576 These three flags allow all know combinations of interlaced formats. They should
9577 only be used when the caps contain 'interlaced=True'.
9578 Fixes #163577 (yes, it's a 4 year old bug).
9580 2009-02-19 15:51:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9582 * docs/libs/gst-plugins-base-libs-sections.txt:
9583 * gst-libs/gst/rtsp/gstrtspconnection.c:
9584 * gst-libs/gst/rtsp/gstrtspconnection.h:
9585 Make RTSPConnection opaque and rename RTSPChannel
9586 Make the RTSPConnection object opaque so that we can extend it in the future.
9587 Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
9589 2009-01-26 10:31:14 +0100 Edward Hervey <bilboed@bilboed.com>
9591 * gst-libs/gst/riff/riff-media.c:
9592 Add some more mappings for h264 in riff
9594 2009-02-19 10:49:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9596 * win32/common/libgstrtsp.def:
9597 Add new RTSP symbols to def files
9598 Add the new RTSP symbols to the windows def file.
9600 2009-02-19 10:44:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9602 * docs/libs/gst-plugins-base-libs-sections.txt:
9603 * gst-libs/gst/app/gstappsink.c:
9604 * gst-libs/gst/app/gstappsink.h:
9605 * tests/check/Makefile.am:
9606 * tests/check/elements/.gitignore:
9607 * tests/check/elements/appsink.c:
9608 Add method to install callbacks on appsink
9609 Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
9611 Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
9612 performant alternative to connecting to the signals.
9613 Add a unit test for appsink.
9614 Clean up some of the appsink docs.
9615 API: GstAppSink::gst_app_sink_set_callbacks()
9617 2009-02-18 18:46:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9619 * docs/libs/gst-plugins-base-libs-sections.txt:
9620 * gst-libs/gst/rtsp/gstrtspconnection.c:
9621 * gst-libs/gst/rtsp/gstrtspconnection.h:
9622 Add RTSP accept method
9623 Add a method to accept a connection on a socket and create a GstRTSPConnection
9625 API: gst_rtsp_connection_accept()
9627 2009-02-18 17:42:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9629 * docs/libs/gst-plugins-base-libs-sections.txt:
9630 * gst-libs/gst/rtsp/gstrtspconnection.c:
9631 * gst-libs/gst/rtsp/gstrtspconnection.h:
9632 Add RTSP channel object for async io
9633 Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
9634 that the connection can be monitored from a maincontext. This allows us to
9635 operate in ASYNC mode, which is handy when building a server.
9636 Rework the old code to use the async code under the hood.
9637 API: gst_rtsp_channel_new()
9638 API: gst_rtsp_channel_unref()
9639 API: gst_rtsp_channel_attach()
9640 API: gst_rtsp_channel_queue_message()
9642 2009-02-15 07:30:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9644 * gst/audioresample/gstaudioresample.c:
9645 audioresample: Add locking to protect the resampling context
9646 When setting the quality/filter-length while PLAYING the
9647 resampling context will be destroyed and created again in
9648 some cases, which will cause crashes in the transform function
9649 if it's called at that time.
9651 2009-02-13 10:10:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9653 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
9654 * gst/videotestsrc/videotestsrc.c:
9655 ffmpegcolorspace/videotestsrc: Use v308 instead of V308
9657 2009-02-12 19:02:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9659 * gst/ffmpegcolorspace/avcodec.h:
9660 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
9661 * gst/ffmpegcolorspace/imgconvert.c:
9662 * gst/ffmpegcolorspace/imgconvert_template.h:
9663 ffmpegcolorspace: Add support for packed 4:4:4 YUV (format=V308)
9664 Only conversions from/to are implemented, which
9665 gives (indirect) support for all possible conversions.
9666 Partially fixes bug #571147.
9668 2009-02-12 18:17:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9670 * gst/videotestsrc/videotestsrc.c:
9671 videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
9672 Partially fixes bug #571147.
9674 2009-02-12 09:18:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9676 * gst-libs/gst/tag/gsttagdemux.c:
9677 tagdemux: don't abort when downstream pulls a buffer of size 0
9678 Pulling a 0-sized buffer is allowed, and we should handle this correctly instead of
9679 aborting. Fixes #571009 (wma file with ID3v2 tag).
9681 2009-02-11 16:39:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9683 * gst-libs/gst/riff/riff-read.c:
9684 riff: error out on nonsensical chunk sizes instead of aborting
9685 When encountering a nonsensical chunk size such as (guint)-1, error out cleanly instead of
9686 continuing and trying to g_memdup() 4GB of data that doesn't exist, which will either abort
9687 in g_malloc() or crash.
9688 Fixes #553295, crash with fuzzed AVI file.
9690 2009-02-11 16:39:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9693 Make git ignore backup files.
9695 2009-02-10 20:38:58 -0800 Michael Smith <msmith@syncword.(none)>
9697 * gst/playback/gstplaybin2.c:
9698 Revert "Remove pad-removed handlers after setting the decodebins to NULL."
9699 This reverts commit b36d8f3e119f9edc5993c08025614ee32642972e.
9700 This brought back some deadlocks. A small leak is better, for now. Need to
9701 figure out a way to fix the leak properly.
9703 2009-02-10 17:16:07 -0800 Michael Smith <msmith@songbirdnest.com>
9705 * gst/playback/gstplaybin2.c:
9706 playbin2: Fix segfault on notify after group change.
9707 If our group has been switched, then we get a selector active-pad
9708 notification, we don't need to notify.
9710 2009-02-10 17:10:33 -0800 Michael Smith <msmith@songbirdnest.com>
9712 * gst/playback/gstplaysink.c:
9713 playbin2: Look for volume/mute properties recursively in audio element.
9714 Rather than only checking for volume property on the audio sink
9715 directly, recursively look for it on sinks within it (if it's a bin).
9716 Allows use of sink-as-volume-control where the application has supplied
9717 an audio-sink bin that includes a real audio sink internally.
9719 2009-02-10 18:29:22 +0000 Christian Schaller <cschalle@crazyhorse.localdomain>
9721 * gst-plugins-base.spec.in:
9722 Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base
9724 2009-02-10 17:39:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9726 * gst/videotestsrc/videotestsrc.c:
9727 videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
9728 Partially fixes bug #571147.
9730 2009-02-10 17:37:06 +0100 Peter Kjellerstedt <pkj@axis.com>
9732 * gst-libs/gst/rtsp/gstrtspmessage.c:
9733 gstrtspmessage: Minor documentation correction.
9734 Corrected documentation about what needs to be freed after calling
9735 gst_rtsp_message_new(), gst_rtsp_message_new_request(),
9736 gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
9738 2009-02-10 11:00:12 +0100 Antoine Tremblay <hexa00@gmail.com>
9740 * ext/alsa/gstalsamixer.c:
9741 alsamixer: Fix race condition that made alsamixer not working properly
9742 This is due to race conditions between functions that
9743 modified the mixer like set_volume and
9744 snd_mixer_handle_events since the handle_events
9745 can now be called at any time.
9746 Fixed by adding locking around any snd_mixer call
9747 since even read functions can modify the mixer stucture, since
9748 alsa likes to clear it's values before reading new ones.
9749 The favorite race condition seemed to be that set_volume
9750 called read_elem (in alsalib) that reset the volumes to
9751 0 and then read them with read_x_volume. This read looped
9752 on each channel and as the race condition occured the
9753 channels value could be anything , most of the time
9754 it was 0. Thus no value was read or only the value of
9755 one channel was and the volume was reset to 0.
9758 2009-02-09 12:02:21 +0100 Edward Hervey <bilboed@bilboed.com>
9761 Bump revision to use for common submodule.
9763 2009-02-05 15:47:00 +0200 Stefan Kost <ensonic@users.sf.net>
9765 * sys/xvimage/xvimagesink.c:
9766 xvimagesink: do not call _xwindow_clear on ready->paused.
9767 Calling clear at that transition does things like stopping xvideo (which is not
9768 running at that time) and also clearing anything what the application might have drawn.
9769 This breaks handle-expose and autopaint-colorkey features.
9771 2009-02-04 17:03:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9773 * docs/libs/gst-plugins-base-libs-sections.txt:
9774 * gst-libs/gst/rtsp/gstrtsprange.c:
9775 * gst-libs/gst/rtsp/gstrtsprange.h:
9776 RTSPRange: Add method to serialize ranges
9777 Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
9778 be used by a server.
9779 API: GstRTSPRange::gst_rtsp_range_to_string()
9781 2009-02-04 13:16:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9783 * gst-libs/gst/rtsp/gstrtspurl.c:
9784 * gst-libs/gst/rtsp/gstrtspurl.h:
9785 GstRTSPUrl: Add some const to methods
9786 Add const to the methods that do not modify the object.
9788 2009-02-04 13:53:30 +0200 Stefan Kost <ensonic@users.sf.net>
9790 * gst/playback/gstplaysink.c:
9791 playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
9792 The flags where present but actually not been taken into account.
9794 2009-02-04 12:06:38 +0200 Stefan Kost <ensonic@users.sf.net>
9796 * gst/audioresample/gstaudioresample.c:
9797 audioresample: Add a proper deprecation comment and also drop G_PARAM_CONSTRUCT.
9798 The comment will ensure that is is marked properly in the docs and the
9799 GParamSpecflag was causing a duplicated initialisation of the same value.
9801 2009-02-04 11:18:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9803 * gst-libs/gst/rtsp/gstrtspconnection.c:
9804 Add more g_return_if_fail() calls
9805 Check that we have a valid file descriptor before entering certain functions in
9806 order to avoid undesirable situations.
9807 Add some more debugging in the connect method.
9809 2009-02-04 10:31:21 +0200 Stefan Kost <ensonic@users.sf.net>
9812 * gst/audioresample/Makefile.am:
9813 * gst/audioresample/gstaudioresample.c:
9814 audioresample: Only pull in liboil if its actualy used.
9815 Liboil still has quite significant startup overhead especialy on embedded
9816 platforms. In audioresample it was only used for the profiling timer.
9818 2009-02-03 15:26:08 +0200 Stefan Kost <ensonic@users.sf.net>
9820 * gst/typefind/gsttypefindfunctions.c:
9821 typefind: Make the flac check more tight to not mistace some aac files for flac. Fixes #570356.
9822 Add comments about the flac format. Tighten the check to not allow values that
9825 2009-02-03 10:52:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9827 * win32/common/libgstrtsp.def:
9829 Add new methods to the windows def file.
9831 2009-02-02 17:25:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9833 * gst-libs/gst/pbutils/install-plugins.c:
9834 * tests/check/libs/pbutils.c:
9835 pbutils: remove duplicate detail strings when calling the external codec installer
9836 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.
9838 2009-02-02 18:05:42 +0200 Stefan Kost <ensonic@users.sf.net>
9840 * gst-libs/gst/audio/gstaudiosink.c:
9841 * gst-libs/gst/audio/gstaudiosink.h:
9842 Add a FIXME 0.11. Make the log message a bit more detailed and add comments.
9844 2009-02-02 15:43:03 +0200 Stefan Kost <ensonic@users.sf.net>
9847 * gst/audioresample/gstaudioresample.c:
9848 Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark.
9850 2009-02-02 13:30:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9852 * sys/ximage/ximagesink.c:
9853 Fix buffer_alloc in ximagesink
9854 Remove some useless debug info that reported wrong image sizes.
9855 When upstream does not accept out suggested size, fall back to allocating an
9856 image of the requested width/height instead of the currently configured size.
9857 The problem is that an image is reused from the pool because the width/height
9858 match but the caps on the new buffer are the requested caps with possibly
9859 different height/width resulting in errors.
9861 2009-02-02 12:54:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9863 * gst/playback/gstdecodebin2.c:
9864 * gst/playback/gsturidecodebin.c:
9865 Fix documentation for autoplug-select
9866 fix the documentation strings for the autoplug-select signal.
9869 2009-02-02 10:09:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9871 * gst-libs/gst/rtsp/gstrtspmessage.c:
9872 Fix string leak in rtspmessage
9873 when we remove a header field from a message we must free the value associated
9874 with the key to avoid a memory leak.
9876 2009-01-31 18:45:47 +0200 Stefan Kost <ensonic@users.sf.net>
9878 * docs/libs/gst-plugins-base-libs-docs.sgml:
9879 Its "Base Library" and not just "Library".
9881 2009-01-31 18:44:32 +0200 Stefan Kost <ensonic@users.sf.net>
9883 * gst-libs/gst/audio/gstaudiofilter.c:
9884 Link to the class, as we can't link to the members yet.
9886 2009-01-30 17:48:23 -0800 Michael Smith <msmith@songbirdnest.com>
9888 * gst/playback/gstplaybin2.c:
9889 Remove pad-removed handlers after setting the decodebins to NULL.
9890 They do needed cleanup; without this we leak selector requestpads.
9892 2009-01-30 17:47:07 -0800 Michael Smith <msmith@songbirdnest.com>
9894 * gst/playback/gstplaybin2.c:
9895 Unref selector request pad even if we no longer have a selector.
9896 During destruction, we won't have a selector any more, but we still need
9897 to unref the pad to avoid leaking it.
9899 2009-01-30 15:23:23 -0800 Michael Smith <msmith@songbirdnest.com>
9901 * gst/playback/gstplaybin2.c:
9902 Unref source in playbin2's finalize method
9904 2009-01-30 12:04:01 -0800 Michael Smith <msmith@songbirdnest.com>
9906 * gst/playback/gstplaysink.c:
9907 Fix more leaks of pads and elements in gstplaysink.
9908 Don't keep extra references to volume and mute elements; we don't need
9910 Ensure we unref pads that we have references to, and release request
9913 2009-01-30 11:04:37 -0800 Michael Smith <msmith@songbirdnest.com>
9915 * gst/playback/gstplaysink.c:
9916 Avoid leaking all playsinks. Fix some internal leaks.
9917 Playsink was holding references to itself. Don't do that, it's not cool.
9918 Also, free all chains in dispose.
9920 2009-01-30 10:54:12 -0800 Michael Smith <msmith@songbirdnest.com>
9922 * gst/playback/gstplaybin2.c:
9923 Unref peer request pad after releasing it, since we hold a reference.
9925 2009-01-30 10:52:52 -0800 Michael Smith <msmith@songbirdnest.com>
9927 * gst/playback/gstplaybin2.c:
9928 Fix caps leak in playbin2.
9930 2009-01-30 10:51:11 -0800 Michael Smith <msmith@songbirdnest.com>
9932 * gst/playback/gstplaybin2.c:
9933 Unref active pad from selector when finding active stream.
9935 2009-01-30 10:49:55 -0800 Michael Smith <msmith@songbirdnest.com>
9937 * gst/playback/gstplaybin2.c:
9938 Free uris when finalizing playbin2 instance.
9940 2009-01-30 10:38:17 -0800 Michael Smith <msmith@songbirdnest.com>
9942 * gst/playback/gsturidecodebin.c:
9943 Unref pads when iterating over them in analyse_source.
9944 Fixes leak of source's srcpad when using uridecodebin.
9946 2009-01-30 22:22:07 +0200 Stefan Kost <ensonic@users.sf.net>
9948 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
9949 Add releaseinfo with online url.
9951 2009-01-30 17:58:15 +0000 Jan Schmidt <jan.schmidt@sun.com>
9953 * gst/playback/gstplaybasebin.c:
9954 Fix compilation warning on Forte
9956 2009-01-30 17:16:39 +0000 Jan Schmidt <jan.schmidt@sun.com>
9958 * gst/adder/gstadder.c:
9959 Don't do void pointer arithmetic.
9961 2009-01-30 17:25:51 +0000 Jan Schmidt <thaytan@noraisin.net>
9966 2009-01-30 08:50:53 +0100 Edward Hervey <bilboed@bilboed.com>
9970 Use a symbolic link for the pre-commit client-side hook
9972 2009-01-30 08:12:42 +0100 Edward Hervey <bilboed@bilboed.com>
9975 Add more files/directories to ignore
9977 2009-01-29 14:00:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9979 * gst-libs/gst/rtsp/gstrtspdefs.c:
9981 Fix some typos in the doc string of the new
9982 gst_rtsp_options_as_string() method.
9984 2009-01-29 11:55:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9986 * docs/libs/gst-plugins-base-libs-sections.txt:
9987 * gst-libs/gst/rtsp/gstrtspconnection.c:
9988 * gst-libs/gst/rtsp/gstrtspmessage.c:
9989 * gst-libs/gst/rtsp/gstrtspmessage.h:
9990 Add new RTSP message method to set header
9991 Add gst_rtsp_message_take_header() that takes ownership of the passed header
9992 value. This allows us to avoid an allocations and memory copy in some
9994 API: GstRTSPMessage::gst_rtsp_message_take_header()
9996 2009-01-29 11:51:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9998 * docs/libs/gst-plugins-base-libs-sections.txt:
9999 Add new method to docs
10000 Add the new gst_rtsp_options_as_text() method to the docs.
10002 2009-01-28 11:48:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10004 * gst-libs/gst/rtsp/gstrtspdefs.c:
10005 * gst-libs/gst/rtsp/gstrtspdefs.h:
10006 Add method to serialize RTSP options
10007 Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
10009 API: GstRTSP::gst_rtsp_options_as_text()
10011 2009-01-26 17:59:37 -0800 Michael Smith <msmith@songbirdnest.com>
10013 * gst/typefind/gsttypefindfunctions.c:
10014 Ensure we have sufficient data when using data scan contexts.
10015 Fixes crashes typefinding things that look like they might contain AAC
10016 data (but probably aren't actually AAC).
10018 2009-01-26 23:32:09 +0000 Jan Schmidt <thaytan@noraisin.net>
10020 * ext/gio/Makefile.am:
10021 Fix include order for gio plugin
10023 2009-01-23 23:59:48 +0000 Jan Schmidt <thaytan@noraisin.net>
10025 * win32/common/config.h:
10026 Update win32 config.h for 0.10.22.1 dev cycle
10028 2009-01-23 23:16:11 +0000 Jan Schmidt <thaytan@noraisin.net>
10031 * docs/libs/.gitignore:
10032 * gst-libs/gst/audio/.gitignore:
10033 * gst-libs/gst/video/.gitignore:
10035 * tests/examples/dynamic/.gitignore:
10036 Extend and clean up git ignores
10038 2009-01-23 12:31:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10041 * docs/plugins/Makefile.am:
10042 * docs/plugins/gst-plugins-base-plugins-sections.txt:
10043 * docs/plugins/gst-plugins-base-plugins.args:
10044 * docs/plugins/gst-plugins-base-plugins.hierarchy:
10045 * docs/plugins/gst-plugins-base-plugins.interfaces:
10046 * docs/plugins/gst-plugins-base-plugins.prerequisites:
10047 * docs/plugins/inspect/plugin-adder.xml:
10048 * docs/plugins/inspect/plugin-alsa.xml:
10049 * docs/plugins/inspect/plugin-app.xml:
10050 * docs/plugins/inspect/plugin-audioconvert.xml:
10051 * docs/plugins/inspect/plugin-audiorate.xml:
10052 * docs/plugins/inspect/plugin-audioresample.xml:
10053 * docs/plugins/inspect/plugin-audiotestsrc.xml:
10054 * docs/plugins/inspect/plugin-cdparanoia.xml:
10055 * docs/plugins/inspect/plugin-decodebin.xml:
10056 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
10057 * docs/plugins/inspect/plugin-gdp.xml:
10058 * docs/plugins/inspect/plugin-gio.xml:
10059 * docs/plugins/inspect/plugin-gnomevfs.xml:
10060 * docs/plugins/inspect/plugin-libvisual.xml:
10061 * docs/plugins/inspect/plugin-ogg.xml:
10062 * docs/plugins/inspect/plugin-pango.xml:
10063 * docs/plugins/inspect/plugin-playback.xml:
10064 * docs/plugins/inspect/plugin-queue2.xml:
10065 * docs/plugins/inspect/plugin-subparse.xml:
10066 * docs/plugins/inspect/plugin-tcp.xml:
10067 * docs/plugins/inspect/plugin-theora.xml:
10068 * docs/plugins/inspect/plugin-typefindfunctions.xml:
10069 * docs/plugins/inspect/plugin-uridecodebin.xml:
10070 * docs/plugins/inspect/plugin-video4linux.xml:
10071 * docs/plugins/inspect/plugin-videorate.xml:
10072 * docs/plugins/inspect/plugin-videoscale.xml:
10073 * docs/plugins/inspect/plugin-videotestsrc.xml:
10074 * docs/plugins/inspect/plugin-volume.xml:
10075 * docs/plugins/inspect/plugin-vorbis.xml:
10076 * docs/plugins/inspect/plugin-ximagesink.xml:
10077 * docs/plugins/inspect/plugin-xvimagesink.xml:
10078 * gst/audioresample/Makefile.am:
10079 * gst/audioresample/README:
10080 * gst/audioresample/arch.h:
10081 * gst/audioresample/buffer.c:
10082 * gst/audioresample/buffer.h:
10083 * gst/audioresample/debug.c:
10084 * gst/audioresample/debug.h:
10085 * gst/audioresample/fixed_arm4.h:
10086 * gst/audioresample/fixed_arm5e.h:
10087 * gst/audioresample/fixed_bfin.h:
10088 * gst/audioresample/fixed_debug.h:
10089 * gst/audioresample/fixed_generic.h:
10090 * gst/audioresample/functable.c:
10091 * gst/audioresample/functable.h:
10092 * gst/audioresample/gstaudioresample.c:
10093 * gst/audioresample/gstaudioresample.h:
10094 * gst/audioresample/resample.c:
10095 * gst/audioresample/resample.h:
10096 * gst/audioresample/resample_chunk.c:
10097 * gst/audioresample/resample_functable.c:
10098 * gst/audioresample/resample_ref.c:
10099 * gst/audioresample/resample_sse.h:
10100 * gst/audioresample/speex_resampler.h:
10101 * gst/audioresample/speex_resampler_double.c:
10102 * gst/audioresample/speex_resampler_float.c:
10103 * gst/audioresample/speex_resampler_int.c:
10104 * gst/audioresample/speex_resampler_wrapper.h:
10105 * gst/speexresample/Makefile.am:
10106 * gst/speexresample/README:
10107 * gst/speexresample/arch.h:
10108 * gst/speexresample/fixed_arm4.h:
10109 * gst/speexresample/fixed_arm5e.h:
10110 * gst/speexresample/fixed_bfin.h:
10111 * gst/speexresample/fixed_debug.h:
10112 * gst/speexresample/fixed_generic.h:
10113 * gst/speexresample/gstspeexresample.c:
10114 * gst/speexresample/gstspeexresample.h:
10115 * gst/speexresample/resample.c:
10116 * gst/speexresample/resample_sse.h:
10117 * gst/speexresample/speex_resampler.h:
10118 * gst/speexresample/speex_resampler_double.c:
10119 * gst/speexresample/speex_resampler_float.c:
10120 * gst/speexresample/speex_resampler_int.c:
10121 * gst/speexresample/speex_resampler_wrapper.h:
10122 * gst/typefind/gsttypefindfunctions.c:
10123 * tests/check/Makefile.am:
10124 * tests/check/elements/audioresample.c:
10125 * tests/check/elements/speexresample.c:
10126 Rename files and types from speexresample to audioresample
10127 Rename files and types from speexresample to audioresample
10128 to finish the move and to prevent any confusion.
10130 2009-01-23 11:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10132 * sys/xvimage/xvimagesink.c:
10133 Add some more debugging to the Xv strides
10134 Add some more debugging to the strides as they are received from the server and
10135 the expected strides.
10137 2009-01-23 11:40:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10139 * gst/typefind/gsttypefindfunctions.c:
10140 Add typefind function for gsm
10141 Because core now supports typefindfactories without a typefind function we can
10142 register a factory fo GSM that will --if all else fails-- assume the file is a
10143 GSM file based on the registered extension.
10146 2009-01-23 11:37:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10148 * gst/playback/gsturidecodebin.c:
10149 Use more performant link function
10150 We can use gst_element_link_pads() instead of the more generic
10151 gst_element_link() function because we know the pads. This saves some cycles
10152 because the more generic function needs to search for possible compatible caps
10155 2009-01-23 11:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10157 * gst-libs/gst/riff/riff-ids.h:
10158 * gst-libs/gst/riff/riff-media.c:
10159 Add more codec ids for RIFF formats
10160 Handle codec ID for various other AAC formats.
10161 Sync the list of possible codec ids with that of ffmpeg.
10164 2009-01-23 11:27:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10166 * ext/theora/theoradec.c:
10167 Use rounded values for image strides and sizes
10168 Round up the height before calculating the expected size and
10169 strides of the output image.
10171 2009-01-23 11:23:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10173 * ext/alsa/gstalsasink.c:
10174 Improve debug message
10175 Improve the debug message when alsa returns an error.
10177 2009-01-23 11:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10179 * gst-libs/gst/app/gstappsrc.c:
10180 Reset queued_bytes counter when flushing
10181 Set the amount of queued bytes in the internal queue back to 0 when we clear the
10185 2009-01-23 10:19:27 +0100 Benjamin Gaignard <benjamin@gaignard.net>
10187 * gst/typefind/gsttypefindfunctions.c:
10188 Add typefinder for Mobile XMF. Fixes bug #568707.
10190 2009-01-23 10:00:11 +0100 Brian Cameron <brian.cameron@sun.com>
10193 Fix linking on Solaris. Fixes bug #568482.
10194 Check for nsl and socket libraries and add them to
10195 LIBS if they're found. They're needed for socket()
10196 and gethostbyname() on Solaris.
10198 2009-01-22 22:09:47 +0000 Jan Schmidt <thaytan@noraisin.net>
10200 * gst/playback/gstplaybasebin.c:
10201 Fix use-after-unref problem noticed by Josep Torra Valles, and run gst-indent
10203 2009-01-22 17:46:59 +0200 Stefan Kost <ensonic@users.sf.net>
10206 Update common snapshot.
10208 2009-01-22 13:47:24 +0100 Sebastian Dröge <slomo@circular-chaos.org>
10211 Fix pre-commit hook
10213 2009-01-22 13:12:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10215 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
10217 2009-01-22 10:14:28 +0100 Sebastian Dröge <slomo@circular-chaos.org>
10219 * gst-libs/gst/fft/gstfftf32.c:
10220 * gst-libs/gst/fft/gstfftf64.c:
10221 * gst-libs/gst/fft/gstffts16.c:
10222 * gst-libs/gst/fft/gstffts32.c:
10223 Reduce the number of allocations for creating FFT contexts
10224 Reduce the number of allocations from 2 to 1 for every FFT
10225 context by allocating enough memory for the FFT context
10226 and passing parts of it to the kissfft allocation functions.
10228 2009-01-22 11:32:56 +0000 Jan Schmidt <thaytan@noraisin.net>
10231 Back to devel -> 0.10.22.1
10233 2009-01-22 05:57:53 +0100 Edward Hervey <bilboed@bilboed.com>
10237 Install and use pre-commit indentation hook from common
10239 2009-01-21 13:09:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10241 * gst-libs/gst/rtp/gstrtpbuffer.c:
10242 * tests/check/libs/rtp.c:
10243 Avoid overflows in the padding checks by doing the check slightly differently. Add a unit test to check for correct behaviour.
10245 2009-01-21 04:31:32 +0100 Edward Hervey <bilboed@bilboed.com>
10248 autogen.sh : Use git submodule
10250 === release 0.10.22 ===
10252 2009-01-19 23:10:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10258 * docs/plugins/gst-plugins-base-plugins.hierarchy:
10259 * docs/plugins/gst-plugins-base-plugins.interfaces:
10260 * docs/plugins/gst-plugins-base-plugins.prerequisites:
10261 * docs/plugins/inspect/plugin-adder.xml:
10262 * docs/plugins/inspect/plugin-alsa.xml:
10263 * docs/plugins/inspect/plugin-app.xml:
10264 * docs/plugins/inspect/plugin-audioconvert.xml:
10265 * docs/plugins/inspect/plugin-audiorate.xml:
10266 * docs/plugins/inspect/plugin-audioresample.xml:
10267 * docs/plugins/inspect/plugin-audiotestsrc.xml:
10268 * docs/plugins/inspect/plugin-cdparanoia.xml:
10269 * docs/plugins/inspect/plugin-decodebin.xml:
10270 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
10271 * docs/plugins/inspect/plugin-gdp.xml:
10272 * docs/plugins/inspect/plugin-gnomevfs.xml:
10273 * docs/plugins/inspect/plugin-libvisual.xml:
10274 * docs/plugins/inspect/plugin-ogg.xml:
10275 * docs/plugins/inspect/plugin-pango.xml:
10276 * docs/plugins/inspect/plugin-playback.xml:
10277 * docs/plugins/inspect/plugin-queue2.xml:
10278 * docs/plugins/inspect/plugin-subparse.xml:
10279 * docs/plugins/inspect/plugin-tcp.xml:
10280 * docs/plugins/inspect/plugin-theora.xml:
10281 * docs/plugins/inspect/plugin-typefindfunctions.xml:
10282 * docs/plugins/inspect/plugin-uridecodebin.xml:
10283 * docs/plugins/inspect/plugin-video4linux.xml:
10284 * docs/plugins/inspect/plugin-videorate.xml:
10285 * docs/plugins/inspect/plugin-videoscale.xml:
10286 * docs/plugins/inspect/plugin-videotestsrc.xml:
10287 * docs/plugins/inspect/plugin-volume.xml:
10288 * docs/plugins/inspect/plugin-vorbis.xml:
10289 * docs/plugins/inspect/plugin-ximagesink.xml:
10290 * docs/plugins/inspect/plugin-xvimagesink.xml:
10291 * gst-plugins-base.doap:
10321 * win32/common/config.h:
10323 Original commit message from CVS:
10326 2009-01-19 22:01:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10358 Original commit message from CVS:
10361 2009-01-16 11:44:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10363 gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
10364 Original commit message from CVS:
10365 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
10366 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
10367 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
10368 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
10369 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
10370 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
10371 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
10372 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
10373 Use correct struct alignment everywhere to prevent unaligned
10374 memory accesses, resulting in SIGBUS on sparc and probably others.
10377 2009-01-16 11:40:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10379 gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
10380 Original commit message from CVS:
10381 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
10382 Forward unknown events upstream to allow latency configuration.
10385 2009-01-13 14:47:19 +0000 Wim Taymans <wim.taymans@gmail.com>
10387 gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
10388 Original commit message from CVS:
10389 * gst/playback/gstplaybin2.c: (groups_set_locked_state):
10390 Provide the right arguments to a debug line.
10392 2009-01-13 06:51:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10394 sys/xvimage/xvimagesink.c: Don't reset the colorkey when element is reused. Fixes #567511.
10395 Original commit message from CVS:
10396 * sys/xvimage/xvimagesink.c:
10397 Don't reset the colorkey when element is reused. Fixes #567511.
10399 2009-01-09 23:42:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10401 configure.ac: 0.10.21.3 pre-release
10402 Original commit message from CVS:
10404 0.10.21.3 pre-release
10406 2009-01-09 23:13:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10408 gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
10409 Original commit message from CVS:
10410 * gst-libs/gst/app/gstappsink.c:
10411 Store the returned signal id in the right slot when
10412 registering the pull-buffer signal.
10414 Spotted by: Thomas Vander Stichele <thomas at apestaart dot org>
10416 2009-01-09 17:17:50 +0000 Tim-Philipp Müller <tim@centricular.net>
10418 gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
10419 Original commit message from CVS:
10420 * gst-libs/gst/interfaces/mixer.c:
10421 Small docs addition to clarify that one really mustn't free
10422 the constant GList returned (#566812).
10424 2009-01-08 17:18:24 +0000 Wim Taymans <wim.taymans@gmail.com>
10426 Add GType for GstRTSPUrl and expose a copy function because we can.
10427 Original commit message from CVS:
10428 * docs/libs/gst-plugins-base-libs-sections.txt:
10429 * gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
10430 (gst_rtsp_url_get_type), (gst_rtsp_url_copy):
10431 * gst-libs/gst/rtsp/gstrtspurl.h:
10432 * win32/common/libgstrtsp.def:
10433 Add GType for GstRTSPUrl and expose a copy function because we can.
10434 API: gst_rtsp_url_copy()
10437 2009-01-07 18:36:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10439 Add plugin dependency for the GIO and GVfs modules.
10440 Original commit message from CVS:
10442 * ext/gio/gstgio.c: (plugin_init):
10443 Add plugin dependency for the GIO and GVfs modules.
10446 2009-01-07 18:32:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10448 Add plugin dependency for the gnomevfs modules.
10449 Original commit message from CVS:
10451 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
10452 Add plugin dependency for the gnomevfs modules.
10455 2009-01-07 18:30:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10457 win32/common/libgstcdda.def: Add new symbol to the list of exported symbols.
10458 Original commit message from CVS:
10459 * win32/common/libgstcdda.def:
10460 Add new symbol to the list of exported symbols.
10462 2009-01-07 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
10464 gst/playback/gstplaybin2.c: Fix some comments and docs.
10465 Original commit message from CVS:
10466 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
10467 (gst_play_bin_set_uri), (gst_play_bin_set_suburi),
10468 (no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
10469 (activate_group), (deactivate_group), (groups_set_locked_state),
10470 (gst_play_bin_change_state):
10471 Fix some comments and docs.
10472 Post an error message when we fail to link the selector to the sink.
10473 Remove pushing of EOS, this seems unneeded.
10474 Lock the state of deactivated groups so that they don't accidentally
10475 reactivate when the playbin2 state changes.
10476 Reuse uridecodebins.
10477 Unlock and relock state of groups when playbin goes to NULL.
10480 * gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
10481 Only do something in the pad removed callback when we are dealing with
10482 our sourcepads because the sinkpads don't have a ghostpad.
10484 2009-01-07 10:50:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10486 gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
10487 Original commit message from CVS:
10488 * gst-libs/gst/cdda/gstcddabasesrc.c:
10489 * gst-libs/gst/cdda/gstcddabasesrc.h:
10490 Make the GType of GstCDDABaseSrcMode public for bindings.
10493 2009-01-06 18:03:51 +0000 Tim-Philipp Müller <tim@centricular.net>
10495 Use new core API to make registry re-scan the plugin whenever visualisations are added or removed (see #350477).
10496 Original commit message from CVS:
10498 * ext/libvisual/visual.c: (plugin_init):
10499 Use new core API to make registry re-scan the plugin
10500 whenever visualisations are added or removed (see #350477).
10502 2009-01-06 17:30:31 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
10504 gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
10505 Original commit message from CVS:
10506 Patch by: José Alburquerque <jaalburqu svn gnome org>
10507 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
10508 * gst-libs/gst/audio/gstaudioclock.h:
10509 Make gst_audio_clock_new use const gchar* to ease the wrapping of
10510 C++ bindings. Fixes #566723.
10512 2009-01-06 12:16:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10514 Add pkg-config files for libgstapp. Fixes bug #566761.
10515 Original commit message from CVS:
10517 * pkgconfig/Makefile.am:
10518 * pkgconfig/gstreamer-app-uninstalled.pc.in:
10519 * pkgconfig/gstreamer-app.pc.in:
10520 Add pkg-config files for libgstapp. Fixes bug #566761.
10522 2009-01-06 11:10:29 +0000 Tim-Philipp Müller <tim@centricular.net>
10524 gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
10525 Original commit message from CVS:
10526 * gst-libs/gst/app/gstappsink.c:
10527 * gst-libs/gst/app/gstappsink.h:
10528 * gst-libs/gst/app/gstappsrc.c:
10529 * gst-libs/gst/app/gstappsrc.h:
10530 Make debug categories static. Use _element_class_set_details_simple().
10532 2009-01-06 10:56:45 +0000 Tim-Philipp Müller <tim@centricular.net>
10534 gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
10535 Original commit message from CVS:
10536 * gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
10537 (gst_app_sink_class_init), (gst_app_sink_init),
10538 (gst_app_sink_dispose), (gst_app_sink_finalize),
10539 (gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
10540 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
10541 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
10542 (gst_app_sink_render), (gst_app_sink_getcaps),
10543 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
10544 (gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
10545 (gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
10546 (gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
10547 (gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
10548 (gst_app_sink_pull_buffer)::
10549 * gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
10550 * gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
10551 (gst_app_src_class_init), (gst_app_src_init),
10552 (gst_app_src_flush_queued), (gst_app_src_dispose),
10553 (gst_app_src_finalize), (gst_app_src_set_property),
10554 (gst_app_src_get_property), (gst_app_src_unlock),
10555 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
10556 (gst_app_src_is_seekable), (gst_app_src_check_get_range),
10557 (gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
10558 (gst_app_src_set_caps), (gst_app_src_get_caps),
10559 (gst_app_src_set_size), (gst_app_src_get_size),
10560 (gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
10561 (gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
10562 (gst_app_src_set_latencies), (gst_app_src_set_latency),
10563 (gst_app_src_get_latency), (gst_app_src_push_buffer_full),
10564 (gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
10565 * gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
10566 Move private data into a private instance struct. Add padding to
10567 instance and class structures exposed in public headers. Add
10568 Since markers to the gtk-doc blurbs (#566750).
10570 2009-01-06 10:50:37 +0000 Wim Taymans <wim.taymans@gmail.com>
10572 tests/examples/app/appsrc_ex.c: Some comments.
10573 Original commit message from CVS:
10574 * tests/examples/app/appsrc_ex.c: (main):
10576 When pulling a buffer we can get NULL when the element is EOS, don't try
10577 to unref this NULL buffer.
10579 2009-01-06 10:16:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10581 gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
10582 Original commit message from CVS:
10583 * gst-libs/gst/video/Makefile.am:
10584 * gst-libs/gst/video/video.h:
10585 Fix up build flags and include statement for the new generated
10586 enumtypes files, to fix dist.
10588 2009-01-05 23:04:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10590 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
10591 Original commit message from CVS:
10593 * docs/libs/Makefile.am:
10594 * docs/libs/gst-plugins-base-libs-docs.sgml:
10595 * docs/libs/gst-plugins-base-libs-sections.txt:
10596 * docs/plugins/Makefile.am:
10597 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
10598 * docs/plugins/gst-plugins-base-plugins-sections.txt:
10599 * docs/plugins/gst-plugins-base-plugins.args:
10600 * docs/plugins/gst-plugins-base-plugins.hierarchy:
10601 * docs/plugins/gst-plugins-base-plugins.interfaces:
10602 * docs/plugins/gst-plugins-base-plugins.prerequisites:
10603 * docs/plugins/gst-plugins-base-plugins.signals:
10604 * docs/plugins/inspect/plugin-app.xml:
10605 * gst-libs/gst/Makefile.am:
10606 * gst-libs/gst/app/gstappsink.c:
10607 * gst-libs/gst/app/gstappsrc.c:
10608 * tests/examples/Makefile.am:
10609 * tests/examples/app/Makefile.am:
10610 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
10612 2009-01-05 17:13:13 +0000 Wim Taymans <wim.taymans@gmail.com>
10614 gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
10615 Original commit message from CVS:
10616 * gst-libs/gst/audio/gstbaseaudiosink.c:
10617 (gst_base_audio_sink_change_state):
10618 Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
10619 take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
10620 this because the async_play method is deprecated and usually not called
10623 2009-01-05 12:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
10625 gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
10626 Original commit message from CVS:
10627 * gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
10628 Disconnect signal handlers before destroying a previous decodebin so
10629 that we don't end up causing deadlocks. Fixes #566586.
10631 2009-01-05 10:59:35 +0000 Wim Taymans <wim.taymans@gmail.com>
10633 gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
10634 Original commit message from CVS:
10635 * gst/audiotestsrc/gstaudiotestsrc.c:
10636 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
10637 (gst_audio_test_src_check_get_range),
10638 (gst_audio_test_src_set_property),
10639 (gst_audio_test_src_get_property):
10640 * gst/audiotestsrc/gstaudiotestsrc.h:
10641 Add property to control pull/push based scheduling.
10643 2009-01-02 15:04:13 +0000 Alessandro Decina <alessandro.d@gmail.com>
10645 Make the seek and colorkey examples depend on gtk+-x11 as they use
10646 Original commit message from CVS:
10648 * tests/examples/seek/Makefile.am:
10649 * tests/icles/Makefile.am:
10650 Make the seek and colorkey examples depend on gtk+-x11 as they use
10652 Fixes the build with gtk+-quartz.
10654 2008-12-31 16:04:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10656 win32/common/: Add new exports to win32 files.
10657 Original commit message from CVS:
10658 * win32/common/libgstaudio.def:
10659 * win32/common/libgsttag.def:
10660 * win32/common/libgstvideo.def:
10661 Add new exports to win32 files.
10663 2008-12-31 13:31:55 +0000 Edward Hervey <bilboed@bilboed.com>
10665 gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
10666 Original commit message from CVS:
10667 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
10668 * gst-libs/gst/tag/gsttagdemux.h:
10669 Add GType for GstTagDemuxResult enum.
10671 2008-12-31 13:01:30 +0000 Edward Hervey <bilboed@bilboed.com>
10673 gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
10674 Original commit message from CVS:
10675 * gst-libs/gst/video/Makefile.am:
10676 * gst-libs/gst/video/video.h:
10677 Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
10678 This will help bindings to use it.
10680 2008-12-31 11:20:26 +0000 Edward Hervey <bilboed@bilboed.com>
10682 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
10683 Original commit message from CVS:
10684 * gst-libs/gst/audio/Makefile.am:
10685 * gst-libs/gst/audio/audio.c:
10686 * gst-libs/gst/audio/multichannel.h:
10687 * gst-libs/gst/audio/testchannels.c:
10689 * win32/common/audio-enumtypes.c:
10690 (gst_audio_channel_position_get_type),
10691 (gst_ring_buffer_state_get_type),
10692 (gst_ring_buffer_seg_state_get_type),
10693 (gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
10694 * win32/common/audio-enumtypes.h:
10695 * win32/common/multichannel-enumtypes.c:
10696 * win32/common/multichannel-enumtypes.h:
10697 * win32/vs6/grammar.dsp:
10698 * win32/vs6/libgstaudio.dsp:
10699 * win32/vs7/libgstaudio.vcproj:
10700 * win32/vs8/libgstaudio.vcproj:
10701 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
10702 audio- in order to wrap all enums declarations of that library.
10703 This modification should not matter since that header file is not a
10704 public header (it will be included by public headers).
10705 Modify win32 crap^Wfiles accordingly.
10707 2008-12-30 17:55:07 +0000 Edward Hervey <bilboed@bilboed.com>
10709 gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
10710 Original commit message from CVS:
10711 * gst-libs/gst/audio/gstbaseaudiosrc.h:
10712 * gst-libs/gst/audio/gstbaseaudiosink.h:
10713 Complete Sebastien's commit from the 13th by exporting the
10714 _slave_method_get_type() methods.
10716 2008-12-29 16:45:20 +0000 Wim Taymans <wim.taymans@gmail.com>
10718 gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
10719 Original commit message from CVS:
10720 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
10721 (gst_app_src_init), (gst_app_src_set_property),
10722 (gst_app_src_get_property), (gst_app_src_query),
10723 (gst_app_src_set_latencies), (gst_app_src_set_latency),
10724 (gst_app_src_get_latency), (gst_app_src_push_buffer_full):
10725 * gst-libs/gst/app/gstappsrc.h:
10726 Add properties and methods to configure and retrieve the min and max
10729 2008-12-20 17:38:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10731 ext/: Implement URI query. Fixes bug #562949.
10732 Original commit message from CVS:
10733 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_query):
10734 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_class_init),
10735 (gst_gio_base_src_query):
10736 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_query):
10737 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
10738 (gst_gnome_vfs_src_query):
10739 Implement URI query. Fixes bug #562949.
10741 2008-12-20 12:48:43 +0000 Wim Taymans <wim.taymans@gmail.com>
10743 gst/playback/gstplaybin2.c: Add some debug info.
10744 Original commit message from CVS:
10745 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
10746 Add some debug info.
10747 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
10748 (gst_play_sink_reconfigure), (gst_play_sink_request_pad),
10749 (gst_play_sink_release_pad):
10750 Add some more debug info.
10751 Reconfigure the audio chain when we switch between raw and encoded audio
10752 in gapless playback.
10754 2008-12-20 12:45:03 +0000 Wim Taymans <wim.taymans@gmail.com>
10756 gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
10757 Original commit message from CVS:
10758 * gst-libs/gst/audio/gstbaseaudiosink.c:
10759 (gst_base_audio_sink_setcaps):
10760 Pause the write thread before deactivating and releasing the ringbuffer
10761 to avoid a deadlock when we do gapless playback with different sample
10762 rates in playbin2. Fixes #564929.
10764 2008-12-19 13:03:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10766 gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
10767 Original commit message from CVS:
10768 * gst-libs/gst/audio/gstbaseaudiosrc.c:
10769 Make GstAudioSrcSlaveMethod get_type() function non-static
10770 as it's public now.
10771 * win32/common/libgstaudio.def:
10772 * win32/common/libgstnetbuffer.def:
10773 Add some missing functions to the list of exported symbols.
10775 2008-12-18 12:37:33 +0000 Andrew Feren <acferen@yahoo.com>
10777 gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
10778 Original commit message from CVS:
10779 Patch by: Andrew Feren <acferen at yahoo dot com>
10780 * gst-libs/gst/netbuffer/gstnetbuffer.c:
10781 (gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
10782 (gst_netaddress_get_address_bytes),
10783 (gst_netaddress_set_address_bytes):
10784 * gst-libs/gst/netbuffer/gstnetbuffer.h:
10785 Make gst_netaddress_get_ip4_address fail for v6 addresses.
10786 Make gst_netaddress_get_ip6_address either fail or return the v4
10787 address as a transitional v6 address.
10788 Add two convenience functions:
10789 API: gst_netaddress_get_address_bytes()
10790 API: gst_netaddress_set_address_bytes()
10793 2008-12-17 13:51:46 +0000 Wim Taymans <wim.taymans@gmail.com>
10795 Add appsrc and appsink documentation.
10796 Original commit message from CVS:
10797 * docs/plugins/Makefile.am:
10798 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
10799 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
10800 * gst-libs/gst/app/gstappsink.c:
10801 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
10802 Add appsrc and appsink documentation.
10804 2008-12-17 08:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10806 gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
10807 Original commit message from CVS:
10808 * gst/adder/Makefile.am:
10809 * gst/adder/gstadder.c:
10810 Cleanup variable names to make the adder-loop easier to understand.
10811 Also try to use liboil to spee it up, but ifdef it out as it does not
10812 make any change for me (Intel pentim M (sse,sse2) please try on other
10815 2008-12-16 20:16:17 +0000 Wim Taymans <wim.taymans@gmail.com>
10817 Add minimal docs to make the remaining tcp elements show up.
10818 Original commit message from CVS:
10819 * docs/plugins/Makefile.am:
10820 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
10821 * docs/plugins/gst-plugins-base-plugins-sections.txt:
10822 * gst/tcp/gsttcpclientsink.c:
10823 * gst/tcp/gsttcpclientsrc.c:
10824 * gst/tcp/gsttcpserversrc.c:
10825 Add minimal docs to make the remaining tcp elements show up.
10828 2008-12-15 12:02:26 +0000 Wim Taymans <wim.taymans@gmail.com>
10830 examples/app/: Fix example to unref after emiting the push-buffer action.
10831 Original commit message from CVS:
10832 * examples/app/appsrc-ra.c: (feed_data):
10833 * examples/app/appsrc-seekable.c: (feed_data):
10834 * examples/app/appsrc-stream.c: (read_data):
10835 * examples/app/appsrc-stream2.c: (feed_data):
10836 Fix example to unref after emiting the push-buffer action.
10837 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
10838 (gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
10839 (gst_app_src_push_buffer_action):
10840 Don't take the ref on the buffer in push-buffer action because it's too
10841 awkward for bindings. Fixes #564482.
10843 2008-12-13 19:32:13 +0000 Tim-Philipp Müller <tim@centricular.net>
10845 win32/common/config.h: Update to CVS version.
10846 Original commit message from CVS:
10847 * win32/common/config.h:
10848 Update to CVS version.
10849 * win32/common/config.h.in:
10850 Hardcode path to plugin install helper exe, just like we hardcode
10851 the paths in core. Removes another source of VCS conflicts for
10852 people hacking gst-plugins-base on systems with autotools.
10854 2008-12-13 16:21:12 +0000 Edward Hervey <bilboed@bilboed.com>
10856 m4/Makefile.am: And a couple more .m4 that don't exist anymore with gettext 0.17
10857 Original commit message from CVS:
10859 And a couple more .m4 that don't exist anymore with gettext 0.17
10861 2008-12-13 12:41:56 +0000 Edward Hervey <bilboed@bilboed.com>
10863 m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we...
10864 Original commit message from CVS:
10866 inttypes.m4 hasn't been available since gettext-0.15, and since we now
10867 require gettext >= 0.17 ... we can remove it from the list of files to
10870 2008-12-13 06:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10872 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
10873 Original commit message from CVS:
10874 * gst-libs/gst/audio/gstbaseaudiosink.c:
10875 (gst_base_audio_sink_slave_method_get_type),
10876 (gst_base_audio_sink_class_init):
10877 * gst-libs/gst/audio/gstbaseaudiosink.h:
10878 * gst-libs/gst/audio/gstbaseaudiosrc.c:
10879 (gst_base_audio_src_slave_method_get_type),
10880 (gst_base_audio_src_class_init):
10881 * gst-libs/gst/audio/gstbaseaudiosrc.h:
10882 API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
10883 public API. This is needed for the C++ bindings to be able
10884 to use this base classes. Fixes bug #564200, #564206.
10886 2008-12-12 19:41:28 +0000 Edward Hervey <bilboed@bilboed.com>
10888 gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
10889 Original commit message from CVS:
10890 * gst-libs/gst/cdda/gstcddabasesrc.c:
10891 (gst_cdda_base_src_handle_event):
10892 Remove erroneous gst_buffer_ref().
10893 * tests/check/libs/rtp.c: (GST_START_TEST):
10894 Don't forget to unref the buffer once you're done with it.
10896 2008-12-12 13:06:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10898 gst/playback/: XRef to GstXOverlay.
10899 Original commit message from CVS:
10900 * gst/playback/gstplaybin.c:
10901 * gst/playback/gstplaybin2.c:
10902 XRef to GstXOverlay.
10904 2008-12-12 10:54:45 +0000 Edward Hervey <bilboed@bilboed.com>
10906 gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
10907 Original commit message from CVS:
10908 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
10909 Free the factory array when finalizing.
10910 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
10911 Use a GstStaticPadTemplate since the src pad caps are fixed.
10913 2008-12-12 07:17:21 +0000 Edward Hervey <bilboed@bilboed.com>
10915 ext/vorbis/vorbisenc.c: Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with pad templates.
10916 Original commit message from CVS:
10917 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_base_init),
10918 (gst_vorbis_enc_init):
10919 Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with
10922 2008-12-12 07:15:22 +0000 Edward Hervey <bilboed@bilboed.com>
10924 gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
10925 Original commit message from CVS:
10926 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
10927 (gst_riff_create_video_template_caps):
10928 Add mapping for VP6 in avi/riff.
10930 2008-12-11 15:49:12 +0000 Edward Hervey <bilboed@bilboed.com>
10932 gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
10933 Original commit message from CVS:
10934 * gst/subparse/samiparse.c: (sami_context_push_state),
10935 (sami_context_pop_state), (start_sami_element), (end_sami_element):
10936 Some versions of libxml seem to be very picky as to strict formatting
10937 of the input and never 'close' the final </body> tag.
10938 In order to fix that bad behaviour, we trigger the flushing of
10939 remaining data on both </body> and </sami>.
10942 2008-12-11 12:32:03 +0000 Guillaume Emont <guillaume@fluendo.com>
10944 gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
10945 Original commit message from CVS:
10946 Patch by: Guillaume Emont <guillaume at fluendo dot com>
10947 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
10948 Add typefinders for MS Word files and OS X .DS_Store files to
10949 prevent them to be recognized as MPEG files. Fixes bug #564098.
10951 2008-12-11 11:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
10953 gst/playback/gstplaysink.c: Add some more debug info.
10954 Original commit message from CVS:
10955 * gst/playback/gstplaysink.c: (gen_audio_chain),
10956 (gst_play_sink_reconfigure):
10957 Add some more debug info.
10958 Fix linking of just an encoded sink.
10959 Handle failure to create a sink chain more gracefully than crashing.
10961 2008-12-11 10:33:48 +0000 Wim Taymans <wim.taymans@gmail.com>
10963 tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
10964 Original commit message from CVS:
10965 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
10966 Pushing 10 buffers is enough to run the test.
10968 2008-12-11 10:28:43 +0000 Wim Taymans <wim.taymans@gmail.com>
10970 tests/examples/seek/seek.c: Hook up the SKIP seek flag.
10971 Original commit message from CVS:
10972 * tests/examples/seek/seek.c: (do_seek), (stop_cb),
10973 (skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
10975 Hook up the SKIP seek flag.
10977 2008-12-10 18:43:32 +0000 Wim Taymans <wim.taymans@gmail.com>
10979 gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
10980 Original commit message from CVS:
10981 * gst/playback/gstplaybin2.c: (pad_added_cb):
10982 Error out with a missing-plugin error when the input-selector was not
10984 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
10987 2008-12-10 17:39:32 +0000 Wim Taymans <wim.taymans@gmail.com>
10989 gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
10990 Original commit message from CVS:
10991 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
10992 (gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
10993 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
10994 (gst_play_sink_send_event), (gst_play_sink_change_state):
10996 Try to set the selected sink to READY before using it. This will allow
10997 for detection of incompatible formats sooner.
10998 Don't cause a fatal error when conversion elements are missing but post
10999 a missing-element message and a warning instead because things might
11000 still link and run fine.
11001 Simplyfy the construction of audio and video sink chains.
11003 2008-12-10 14:55:10 +0000 Wim Taymans <wim.taymans@gmail.com>
11005 ext/ogg/gstoggdemux.c: Use G_DEFINE_TYPE for the OggPad to get some threadsafe type init from glib.
11006 Original commit message from CVS:
11007 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
11008 (gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
11009 Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
11012 2008-12-10 08:19:13 +0000 Luis Menina <liberforce@freeside.fr>
11014 gst/: Include glib.h instead of a specific GLib header. Including single
11015 Original commit message from CVS:
11016 Patch by: Luis Menina <liberforce at freeside dot fr>
11017 * gst-libs/gst/floatcast/floatcast.h:
11018 * gst/typefind/gsttypefindfunctions.c:
11019 Include glib.h instead of a specific GLib header. Including single
11020 GLib headers is deprecated. Fixes bug #563904.
11022 2008-12-09 18:30:10 +0000 Julien Moutte <julien@moutte.net>
11024 gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
11025 Original commit message from CVS:
11026 2008-12-09 Julien Moutte <julien@fluendo.com>
11027 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
11028 Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
11030 2008-12-09 17:21:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11032 gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
11033 Original commit message from CVS:
11034 * gst-libs/gst/riff/riff-read.c:
11035 Fix handling of odd chunks in riff metadata.
11037 2008-12-08 18:44:22 +0000 Wim Taymans <wim.taymans@gmail.com>
11039 gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
11040 Original commit message from CVS:
11041 * gst/volume/gstvolume.c: (gst_volume_class_init),
11042 (volume_before_transform), (volume_transform_ip):
11043 Use new basetransform vmethod to reconfigure the dynamic properties and
11044 any pending volume/mute changes. Fixes #563508.
11046 2008-12-08 18:12:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11048 configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...
11049 Original commit message from CVS:
11051 First check for "theoraenc theoradec" and if that failed check
11052 for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
11053 deprecate the latter. Also linking on Windows fails with just "theora"
11054 and the version check would fail for the release candidates.
11057 2008-12-08 15:25:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11059 gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
11060 Original commit message from CVS:
11061 * gst/playback/gstdecodebin.c:
11062 * gst/playback/gstdecodebin2.c:
11063 Add basic docs to decodebin and link to decodebin from decodebin2.
11065 2008-12-08 12:08:32 +0000 Olivier Crete <tester@tester.ca>
11067 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
11068 Original commit message from CVS:
11069 Patch by: Olivier Crete <tester at tester ca>
11070 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
11071 * gst-libs/gst/rtp/gstrtcpbuffer.h:
11072 Implement gst_rtcp_packet_remove(). Fixes #563174.
11073 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
11074 Add unit test for some RTCP functions.
11076 2008-12-04 20:09:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11078 configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.
11079 Original commit message from CVS:
11081 Apparently AC_CONFIG_MACRO_DIR breaks when using more
11082 than one macro directory, reverting last change.
11084 2008-12-04 19:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11086 configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros.
11087 Original commit message from CVS:
11089 Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
11092 2008-12-03 17:47:44 +0000 Edward Hervey <bilboed@bilboed.com>
11094 sys/: Clear all flags on buffers returned from the image pool.
11095 Original commit message from CVS:
11096 * sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
11097 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
11098 Clear all flags on buffers returned from the image pool.
11101 2008-12-01 19:36:35 +0000 이문형 <iwings@gmail.com>
11103 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...
11104 Original commit message from CVS:
11105 Patch by: 이문형 <iwings at gmail dot com>
11106 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
11107 Don't forget to release the lock again if we bail out because some
11108 pad is flushing or we've reached EOS, otherwise things will lock up
11109 next time _push_buffer() is called (#562802).
11111 2008-11-29 13:31:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11113 Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
11114 Original commit message from CVS:
11115 Patch by: Cygwin Ports maintainer
11116 <yselkowitz at users dot sourceforge dot net>
11119 Require gettext 0.17 because older versions don't mix with libtool
11120 2.2. At build time an older gettext version will still work.
11123 2008-11-28 13:30:36 +0000 Christian Schaller <uraeus@gnome.org>
11126 * gst/speexresample/Makefile.am:
11128 Original commit message from CVS:
11131 2008-11-28 09:44:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11133 Update documentation of speexresample for the new element name.
11134 Original commit message from CVS:
11135 * docs/plugins/gst-plugins-base-plugins.args:
11136 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11137 * docs/plugins/gst-plugins-base-plugins.interfaces:
11138 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11139 * docs/plugins/inspect/plugin-videorate.xml:
11140 * gst/speexresample/gstspeexresample.c:
11141 Update documentation of speexresample for the new element name.
11143 2008-11-28 09:04:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11145 gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
11146 Original commit message from CVS:
11147 * gst/speexresample/README:
11148 Update README with the latest diff between the Speex resampler
11151 2008-11-28 08:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11153 gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
11154 Original commit message from CVS:
11155 * gst/speexresample/gstspeexresample.c: (plugin_init):
11156 Update the debug category from speex_resample to audioresample.
11158 2008-11-27 19:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11160 Remove audioresample files.
11161 Original commit message from CVS:
11162 * gst/audioresample/Makefile.am:
11163 * gst/audioresample/buffer.c:
11164 * gst/audioresample/buffer.h:
11165 * gst/audioresample/debug.c:
11166 * gst/audioresample/debug.h:
11167 * gst/audioresample/functable.c:
11168 * gst/audioresample/functable.h:
11169 * gst/audioresample/gstaudioresample.c:
11170 * gst/audioresample/gstaudioresample.h:
11171 * gst/audioresample/resample.c:
11172 * gst/audioresample/resample.h:
11173 * gst/audioresample/resample_chunk.c:
11174 * gst/audioresample/resample_functable.c:
11175 * gst/audioresample/resample_ref.c:
11176 * tests/check/elements/audioresample.c:
11177 Remove audioresample files.
11179 2008-11-27 17:04:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11181 docs/plugins/inspect/plugin-audioresample.xml: Regenerated for library filename change.
11182 Original commit message from CVS:
11183 * docs/plugins/inspect/plugin-audioresample.xml:
11184 Regenerated for library filename change.
11186 2008-11-27 16:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11188 Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
11189 Original commit message from CVS:
11191 * docs/plugins/Makefile.am:
11192 * docs/plugins/gst-plugins-base-plugins-sections.txt:
11193 * docs/plugins/gst-plugins-base-plugins.args:
11194 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11195 * docs/plugins/gst-plugins-base-plugins.interfaces:
11196 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11197 * docs/plugins/inspect/plugin-adder.xml:
11198 * docs/plugins/inspect/plugin-alsa.xml:
11199 * docs/plugins/inspect/plugin-audioconvert.xml:
11200 * docs/plugins/inspect/plugin-audiorate.xml:
11201 * docs/plugins/inspect/plugin-audioresample.xml:
11202 * docs/plugins/inspect/plugin-audiotestsrc.xml:
11203 * docs/plugins/inspect/plugin-cdparanoia.xml:
11204 * docs/plugins/inspect/plugin-decodebin.xml:
11205 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
11206 * docs/plugins/inspect/plugin-gdp.xml:
11207 * docs/plugins/inspect/plugin-gio.xml:
11208 * docs/plugins/inspect/plugin-gnomevfs.xml:
11209 * docs/plugins/inspect/plugin-libvisual.xml:
11210 * docs/plugins/inspect/plugin-ogg.xml:
11211 * docs/plugins/inspect/plugin-pango.xml:
11212 * docs/plugins/inspect/plugin-playback.xml:
11213 * docs/plugins/inspect/plugin-queue2.xml:
11214 * docs/plugins/inspect/plugin-subparse.xml:
11215 * docs/plugins/inspect/plugin-tcp.xml:
11216 * docs/plugins/inspect/plugin-theora.xml:
11217 * docs/plugins/inspect/plugin-typefindfunctions.xml:
11218 * docs/plugins/inspect/plugin-uridecodebin.xml:
11219 * docs/plugins/inspect/plugin-video4linux.xml:
11220 * docs/plugins/inspect/plugin-videorate.xml:
11221 * docs/plugins/inspect/plugin-videoscale.xml:
11222 * docs/plugins/inspect/plugin-videotestsrc.xml:
11223 * docs/plugins/inspect/plugin-volume.xml:
11224 * docs/plugins/inspect/plugin-vorbis.xml:
11225 * docs/plugins/inspect/plugin-ximagesink.xml:
11226 * docs/plugins/inspect/plugin-xvimagesink.xml:
11227 * gst/speexresample/gstspeexresample.c: (plugin_init):
11228 * gst/speexresample/Makefile.am:
11229 * tests/check/Makefile.am:
11230 * tests/check/elements/speexresample.c: (setup_speexresample),
11231 (GST_START_TEST), (test_pipeline):
11232 Rename the moved speexresample to audioresample, integrate into the
11233 build system and remove the old audioresample from the build system.
11234 Fixes bug #558124, #385061, #346218, #116051.
11236 2008-11-27 16:47:41 +0000 Wim Taymans <wim.taymans@gmail.com>
11238 gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
11239 Original commit message from CVS:
11240 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11241 (gst_base_audio_src_get_offset), (gst_base_audio_src_create):
11242 Avoid nasty int overflows after about 12 hours and 25 minutes when these
11243 code paths are triggered.
11244 A free beer to Håvard Graff for finding this!
11246 2008-11-27 11:16:44 +0000 이문형 <iwings@gmail.com>
11248 gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
11249 Original commit message from CVS:
11250 Patch by: 이문형 <iwings at gmail dot com>
11251 * gst-libs/gst/rtsp/gstrtspconnection.c:
11252 (gst_rtsp_connection_connect):
11253 A successful gst_poll_wait() doesn't always mean successful connect() on
11254 Windows. We should check errors by calling gst_poll_fd_has_error().
11257 2008-11-25 16:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11259 tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.
11260 Original commit message from CVS:
11261 * tests/check/elements/speexresample.c: (test_pipeline):
11262 Make unit test again faster to prevent timeouts with valgrind.
11264 2008-11-25 15:33:30 +0000 Wim Taymans <wim.taymans@gmail.com>
11266 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
11267 Original commit message from CVS:
11268 * gst-libs/gst/rtp/gstrtcpbuffer.c:
11269 Fix typo in the docs.
11271 2008-11-25 15:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
11273 ext/ogg/gstoggdemux.c: If no stream was found before receiving EOS, post an error message.
11274 Original commit message from CVS:
11275 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
11276 If no stream was found before receiving EOS, post an error message.
11279 2008-11-25 15:14:30 +0000 Wim Taymans <wim.taymans@gmail.com>
11281 ext/theora/: Parse segment events.
11282 Original commit message from CVS:
11283 * ext/theora/gsttheoraenc.h:
11284 * ext/theora/theoraenc.c: (gst_theora_enc_init),
11285 (theora_buffer_from_packet), (theora_push_packet),
11286 (theora_enc_sink_event), (theora_enc_is_discontinuous),
11287 (theora_enc_chain):
11288 Parse segment events.
11289 Pass incomming buffer timestamps to outgoing buffers.
11290 Use the running_time to construct the granulepos.
11293 2008-11-25 11:00:55 +0000 Wim Taymans <wim.taymans@gmail.com>
11295 gst/playback/gstplaybin2.c: Fix buffer-duration property.
11296 Original commit message from CVS:
11297 * gst/playback/gstplaybin2.c: (activate_group):
11298 Fix buffer-duration property.
11300 2008-11-25 10:32:49 +0000 Wim Taymans <wim.taymans@gmail.com>
11302 gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
11303 Original commit message from CVS:
11304 * gst-libs/gst/audio/gstbaseaudiosink.c:
11305 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
11306 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
11307 (gst_base_audio_sink_change_state):
11308 Really fix audiosink drain handling by keeping track of the running_time
11309 of the last sample.
11311 2008-11-24 20:25:24 +0000 Michael Smith <msmith@xiph.org>
11313 gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
11314 Original commit message from CVS:
11315 * gst/playback/gstplaybin2.c:
11316 Add notification of current stream. Add ability to configure buffer
11318 * gst/playback/gsturidecodebin.c:
11319 Add ability to configure buffer sizes for streaming mode.
11322 2008-11-24 20:11:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11324 gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
11325 Original commit message from CVS:
11326 * gst-libs/gst/audio/gstbaseaudiosink.c:
11327 Time is already in running_time. Remove base_time handling. Fixes
11328 audiosinks not draining and thus chopping some audio in the end.
11330 2008-11-24 19:18:59 +0000 David Schleef <ds@schleef.org>
11332 ext/ogg/gstoggmux.*: If we're muxing a dirac stream, flush the page after every picture.
11333 Original commit message from CVS:
11334 * ext/ogg/gstoggmux.c:
11335 * ext/ogg/gstoggmux.h:
11336 If we're muxing a dirac stream, flush the page after every picture.
11338 2008-11-24 12:56:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11340 gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
11341 Original commit message from CVS:
11342 * gst-libs/gst/audio/gstbaseaudiosink.c:
11343 Add one log message to check for audio_drained. Sync one log message
11344 with the condition. Send EOS after draining audio in pull mode.
11346 2008-11-24 12:07:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11348 ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr...
11349 Original commit message from CVS:
11350 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
11351 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
11352 Use gst_buffer_try_new_and_alloc() and fail properly if the
11353 allocation failed. This prevents abort() if downstream elements
11354 request an insane amount of memory.
11356 2008-11-24 12:03:11 +0000 Jon Trowbridge <trow@ximian.com>
11358 gst/volume/gstvolume.*: Cleanup volume, define and use default values.
11359 Original commit message from CVS:
11360 * gst/volume/gstvolume.c: (volume_choose_func),
11361 (volume_update_volume), (gst_volume_set_volume),
11362 (gst_volume_get_volume), (gst_volume_set_mute),
11363 (gst_volume_class_init), (gst_volume_init),
11364 (volume_process_double), (volume_process_float),
11365 (volume_process_int32), (volume_process_int32_clamp),
11366 (volume_process_int24), (volume_process_int24_clamp),
11367 (volume_process_int16), (volume_process_int16_clamp),
11368 (volume_process_int8), (volume_process_int8_clamp), (volume_setup),
11369 (volume_transform_ip), (volume_set_property),
11370 (volume_get_property):
11371 * gst/volume/gstvolume.h:
11372 Cleanup volume, define and use default values.
11373 Recalculate new volume and mute setup before processing. Fixes #561789.
11374 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
11375 Add controller unit test. Patch by: Jonathan Matthew
11376 Fix bogus test that messed with basetransform's internal state.
11378 2008-11-22 15:02:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11380 tests/check/elements/speexresample.c: Make the unit test a bit faster to prevent timeouts, especially with valgrind.
11381 Original commit message from CVS:
11382 * tests/check/elements/speexresample.c: (GST_START_TEST):
11383 Make the unit test a bit faster to prevent timeouts, especially
11386 2008-11-22 14:44:26 +0000 Wim Taymans <wim.taymans@gmail.com>
11388 gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
11389 Original commit message from CVS:
11390 * gst/videorate/gstvideorate.c:
11391 Add jpeg and png image media types to the caps. Fixes #561436.
11393 2008-11-22 14:31:43 +0000 Wim Taymans <wim.taymans@gmail.com>
11395 gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ...
11396 Original commit message from CVS:
11397 * gst/playback/gstplaysink.c: (gen_audio_chain):
11398 Don't post an error when we can't configure the volume but post a
11399 warning instead. Fixes #561780.
11401 2008-11-21 20:32:56 +0000 Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
11403 gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video...
11404 Original commit message from CVS:
11405 Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
11406 * gst/videotestsrc/gstvideotestsrc.c:
11407 * gst/videotestsrc/gstvideotestsrc.h:
11408 * gst/videotestsrc/videotestsrc.c:
11409 * gst/videotestsrc/videotestsrc.h:
11410 Add a zone plate pattern generator based on BBC R&D Report
11411 1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate
11412 kx2=20 ky2=20 kt=1'.
11414 2008-11-21 15:45:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11416 gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
11417 Original commit message from CVS:
11418 * gst/speexresample/gstspeexresample.c:
11419 (gst_speex_resample_class_init), (gst_speex_resample_set_property),
11420 (gst_speex_resample_get_property):
11421 Add a "filter-length" property that maps to the quality values
11422 for compatibilty with audioresample.
11424 2008-11-21 00:04:48 +0000 Michael Smith <msmith@xiph.org>
11426 gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
11427 Original commit message from CVS:
11428 * gst/playback/gstdecodebin2.c:
11429 Fix random fat-fingering making this not compile.
11431 2008-11-20 22:11:38 +0000 Michael Smith <msmith@xiph.org>
11433 gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh...
11434 Original commit message from CVS:
11435 * gst/playback/gstdecodebin2.c:
11436 If the top-level type of the stream is plain text, don't try to decode
11437 it, matching behaviour of decodebin.
11438 * gst/playback/gstplaysink.c:
11439 If we fail to generate a text chain (e.g. due to missing optional
11440 plugins), don't crash.
11442 2008-11-20 22:06:05 +0000 Michael Smith <msmith@xiph.org>
11444 gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
11445 Original commit message from CVS:
11446 * gst-libs/gst/rtsp/gstrtspdefs.c:
11447 Fix win32 build. Oops.
11449 2008-11-20 21:40:49 +0000 Michael Smith <msmith@xiph.org>
11451 gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
11452 Original commit message from CVS:
11453 * gst-libs/gst/rtsp/gstrtspdefs.c:
11454 Use WSAGetLastError() rather than errno/h_errno on win32.
11456 2008-11-20 21:20:27 +0000 Michael Smith <msmith@xiph.org>
11458 gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
11459 Original commit message from CVS:
11460 * gst-libs/gst/riff/riff-media.c:
11461 Support WMA Lossless properly.
11463 2008-11-19 00:24:44 +0000 David Schleef <ds@schleef.org>
11465 gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect...
11466 Original commit message from CVS:
11467 * gst/videotestsrc/gstvideotestsrc.c:
11468 * gst/videotestsrc/gstvideotestsrc.h:
11469 * gst/videotestsrc/videotestsrc.c:
11470 * gst/videotestsrc/videotestsrc.h:
11471 Add "colorspec" property, specifying whether to generate BT.601
11472 or BT.709 video. This only affects YCbCr values, not RGB, since
11473 if you're generating a 709 test pattern, presumably you want
11474 709 RGB primaries, not 601. Also add "smpte75" pattern, which
11475 uses 75% colors instead of 100%, since this is often more useful
11476 for testing (and also follows the SMPTE EG-1 guideline).
11478 2008-11-18 18:08:42 +0000 Alessandro Decina <alessandro.d@gmail.com>
11480 gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2.
11481 Original commit message from CVS:
11482 * gst/playback/gstdecodebin.c:
11483 Add a "sink-caps" property to decodebin like it's done for decodebin2.
11486 2008-11-14 21:44:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11488 gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
11489 Original commit message from CVS:
11490 * gst/audioresample/gstaudioresample.c:
11491 Guard against a NULL dereference I somehow encountered -
11492 with a FLUSH_STOP arriving either before basetransform _start(),
11494 * gst/typefind/gsttypefindfunctions.c:
11495 Make sure we never jump backwards when typefinding corrupt mov files.
11497 2008-11-14 21:39:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11499 gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
11500 Original commit message from CVS:
11501 * gst-libs/gst/interfaces/propertyprobe.c:
11502 Fix random type causing a docs warning.
11504 2008-11-14 15:40:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11506 sys/v4l/gstv4l.c: Give it a minimal rank for autovideosrc.
11507 Original commit message from CVS:
11508 * sys/v4l/gstv4l.c:
11509 Give it a minimal rank for autovideosrc.
11511 2008-11-13 21:11:13 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
11513 gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types.
11514 Original commit message from CVS:
11515 * gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
11517 Improve typefinding of ISO JPEG2000 mime types.
11519 2008-11-13 18:18:32 +0000 Wim Taymans <wim.taymans@gmail.com>
11521 sys/xvimage/xvimagesink.*: Avoid typechecking when we do trivial casts.
11522 Original commit message from CVS:
11523 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
11524 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps),
11525 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
11526 * sys/xvimage/xvimagesink.h:
11527 Avoid typechecking when we do trivial casts.
11528 Move error handling out of the main program flow.
11529 Sneak in the display-region caps property, not completely correct yet.
11530 Cache the width/height in buffer_alloc instead of parsing it from the
11533 2008-11-13 17:27:37 +0000 Wim Taymans <wim.taymans@gmail.com>
11535 gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
11536 Original commit message from CVS:
11537 * gst/playback/gstplaybin2.c: (deactivate_group):
11538 don't try to unlink the selector sinkpad when we don't have it yet. This
11539 can happen if an error occured before the group was complete.
11541 2008-11-13 15:37:40 +0000 Wim Taymans <wim.taymans@gmail.com>
11543 gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
11544 Original commit message from CVS:
11545 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
11546 (gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
11547 (gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
11548 (gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
11549 (gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
11550 (gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
11551 (gst_rtp_buffer_get_extension_data),
11552 (gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
11553 (gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
11554 (gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
11555 (gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
11556 (gst_rtp_buffer_get_payload_type),
11557 (gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
11558 (gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
11559 (gst_rtp_buffer_set_timestamp),
11560 (gst_rtp_buffer_get_payload_subbuffer),
11561 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
11562 Avoid expensive type checks we already did as part of the
11563 _validate() function that should be called first.
11565 2008-11-11 16:40:50 +0000 Wim Taymans <wim.taymans@gmail.com>
11567 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some cases where a newsegment event was not sent.
11568 Original commit message from CVS:
11569 * gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
11570 (gst_base_rtp_depayload_push_full),
11571 (gst_base_rtp_depayload_set_gst_timestamp):
11572 Fix some cases where a newsegment event was not sent.
11574 2008-11-11 15:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
11576 gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co...
11577 Original commit message from CVS:
11578 * gst/playback/gstplaybin2.c: (activate_group):
11579 Catch state change errors and stop from the uridecodebin elements
11580 instead of trying to continue in vain.
11582 2008-11-10 14:53:45 +0000 Edward Hervey <bilboed@bilboed.com>
11584 gst/: Wim, you're a bad boy. You don't want people to contact you or what?
11585 Original commit message from CVS:
11586 * gst-libs/gst/app/gstappsink.c:
11587 * gst-libs/gst/app/gstappsrc.c:
11588 * gst/h264parse/gsth264parse.c:
11589 Wim, you're a bad boy. You don't want people to contact you or what?
11591 2008-11-10 14:22:09 +0000 Wim Taymans <wim.taymans@gmail.com>
11593 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
11594 Original commit message from CVS:
11595 * gst-libs/gst/audio/gstbaseaudiosink.c:
11596 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
11597 (gst_base_audio_sink_callback):
11598 Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
11599 for the latency to expire, fixes #559567.
11601 2008-11-10 13:55:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
11603 gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
11604 Original commit message from CVS:
11605 * gst/adder/gstadder.c:
11606 Change author string after seeing output of gst-inspector.
11608 2008-11-10 10:33:26 +0000 Wim Taymans <wim.taymans@gmail.com>
11610 gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559...
11611 Original commit message from CVS:
11612 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
11613 Don't try to do crazy things when we only have a text pad without a
11614 video pad. Fixes #559478.
11616 2008-11-07 17:35:46 +0000 Wim Taymans <wim.taymans@gmail.com>
11618 gst-libs/gst/app/gstappsrc.*: Add is-live property.
11619 Original commit message from CVS:
11620 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
11621 (gst_app_src_init), (gst_app_src_set_property),
11622 (gst_app_src_get_property), (gst_app_src_push_buffer):
11623 * gst-libs/gst/app/gstappsrc.h:
11624 Add is-live property.
11625 Add some more docs.
11627 2008-11-06 12:14:51 +0000 Wim Taymans <wim.taymans@gmail.com>
11629 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...
11630 Original commit message from CVS:
11631 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
11632 Fix case where we don't have a range for the rates or channels as is the
11633 case with truespeech.
11635 2008-11-05 19:18:25 +0000 Wim Taymans <wim.taymans@gmail.com>
11637 gst/volume/gstvolume.*: Keep negotiated state in a separate variable.
11638 Original commit message from CVS:
11639 * gst/volume/gstvolume.c: (volume_update_real_volume),
11640 (gst_volume_set_volume), (gst_volume_get_volume),
11641 (gst_volume_set_mute), (gst_volume_init), (volume_setup),
11642 (volume_transform_ip), (volume_update_mute),
11643 (volume_update_volume), (volume_get_property):
11644 * gst/volume/gstvolume.h:
11645 Keep negotiated state in a separate variable.
11646 Protect the volume and mute properties with the object lock.
11647 Protect modifying the transform with the transform lock.
11649 2008-11-05 12:20:21 +0000 Wim Taymans <wim.taymans@gmail.com>
11651 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Only convert caps to string when debug is enabled.
11652 Original commit message from CVS:
11653 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
11654 (gst_ffmpeg_pixfmt_to_caps):
11655 Only convert caps to string when debug is enabled.
11657 2008-11-04 18:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
11659 ext/theora/: Copy seqnum.
11660 Original commit message from CVS:
11661 * ext/theora/gsttheoradec.h:
11662 * ext/theora/theoradec.c: (gst_theora_dec_init),
11663 (gst_theora_dec_reset), (theora_dec_src_event),
11664 (theora_dec_sink_event), (theora_handle_type_packet):
11666 Keep events in a pending list, like vorbisdec, instead of trying
11667 to construct a segment event ourselves.
11668 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
11669 (vorbis_dec_src_event), (vorbis_dec_sink_event):
11670 * ext/vorbis/vorbisdec.h:
11673 2008-11-04 17:24:35 +0000 Wim Taymans <wim.taymans@gmail.com>
11675 ext/ogg/gstoggdemux.*: Copy seqnums around to track playback segments and messages.
11676 Original commit message from CVS:
11677 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
11678 (gst_ogg_demux_deactivate_current_chain),
11679 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page),
11680 (gst_ogg_demux_loop):
11681 * ext/ogg/gstoggdemux.h:
11682 Copy seqnums around to track playback segments and messages.
11684 2008-11-04 12:42:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11686 Don't install static libs for plugins. Fixes #550851 for -bad.
11687 Original commit message from CVS:
11688 * ext/alsaspdif/Makefile.am:
11689 * ext/amrwb/Makefile.am:
11690 * ext/apexsink/Makefile.am:
11691 * ext/arts/Makefile.am:
11692 * ext/artsd/Makefile.am:
11693 * ext/audiofile/Makefile.am:
11694 * ext/audioresample/Makefile.am:
11695 * ext/bz2/Makefile.am:
11696 * ext/cdaudio/Makefile.am:
11697 * ext/celt/Makefile.am:
11698 * ext/dc1394/Makefile.am:
11699 * ext/dirac/Makefile.am:
11700 * ext/directfb/Makefile.am:
11701 * ext/divx/Makefile.am:
11702 * ext/dts/Makefile.am:
11703 * ext/faac/Makefile.am:
11704 * ext/faad/Makefile.am:
11705 * ext/gsm/Makefile.am:
11706 * ext/hermes/Makefile.am:
11707 * ext/ivorbis/Makefile.am:
11708 * ext/jack/Makefile.am:
11709 * ext/jp2k/Makefile.am:
11710 * ext/ladspa/Makefile.am:
11711 * ext/lcs/Makefile.am:
11712 * ext/libfame/Makefile.am:
11713 * ext/libmms/Makefile.am:
11714 * ext/metadata/Makefile.am:
11715 * ext/mpeg2enc/Makefile.am:
11716 * ext/mplex/Makefile.am:
11717 * ext/musepack/Makefile.am:
11718 * ext/musicbrainz/Makefile.am:
11719 * ext/mythtv/Makefile.am:
11720 * ext/nas/Makefile.am:
11721 * ext/neon/Makefile.am:
11722 * ext/ofa/Makefile.am:
11723 * ext/polyp/Makefile.am:
11724 * ext/resindvd/Makefile.am:
11725 * ext/sdl/Makefile.am:
11726 * ext/shout/Makefile.am:
11727 * ext/snapshot/Makefile.am:
11728 * ext/sndfile/Makefile.am:
11729 * ext/soundtouch/Makefile.am:
11730 * ext/spc/Makefile.am:
11731 * ext/swfdec/Makefile.am:
11732 * ext/tarkin/Makefile.am:
11733 * ext/theora/Makefile.am:
11734 * ext/timidity/Makefile.am:
11735 * ext/twolame/Makefile.am:
11736 * ext/x264/Makefile.am:
11737 * ext/xine/Makefile.am:
11738 * ext/xvid/Makefile.am:
11739 * gst-libs/gst/app/Makefile.am:
11740 * gst-libs/gst/dshow/Makefile.am:
11741 * gst/aiffparse/Makefile.am:
11742 * gst/app/Makefile.am:
11743 * gst/audiobuffer/Makefile.am:
11744 * gst/bayer/Makefile.am:
11745 * gst/cdxaparse/Makefile.am:
11746 * gst/chart/Makefile.am:
11747 * gst/colorspace/Makefile.am:
11748 * gst/dccp/Makefile.am:
11749 * gst/deinterlace/Makefile.am:
11750 * gst/deinterlace2/Makefile.am:
11751 * gst/dvdspu/Makefile.am:
11752 * gst/festival/Makefile.am:
11753 * gst/filter/Makefile.am:
11754 * gst/flacparse/Makefile.am:
11755 * gst/flv/Makefile.am:
11756 * gst/games/Makefile.am:
11757 * gst/h264parse/Makefile.am:
11758 * gst/librfb/Makefile.am:
11759 * gst/mixmatrix/Makefile.am:
11760 * gst/modplug/Makefile.am:
11761 * gst/mpeg1sys/Makefile.am:
11762 * gst/mpeg4videoparse/Makefile.am:
11763 * gst/mpegdemux/Makefile.am:
11764 * gst/mpegtsmux/Makefile.am:
11765 * gst/mpegvideoparse/Makefile.am:
11766 * gst/mve/Makefile.am:
11767 * gst/nsf/Makefile.am:
11768 * gst/nuvdemux/Makefile.am:
11769 * gst/overlay/Makefile.am:
11770 * gst/passthrough/Makefile.am:
11771 * gst/pcapparse/Makefile.am:
11772 * gst/playondemand/Makefile.am:
11773 * gst/rawparse/Makefile.am:
11774 * gst/real/Makefile.am:
11775 * gst/rtjpeg/Makefile.am:
11776 * gst/rtpmanager/Makefile.am:
11777 * gst/scaletempo/Makefile.am:
11778 * gst/sdp/Makefile.am:
11779 * gst/selector/Makefile.am:
11780 * gst/smooth/Makefile.am:
11781 * gst/smoothwave/Makefile.am:
11782 * gst/speed/Makefile.am:
11783 * gst/speexresample/Makefile.am:
11784 * gst/stereo/Makefile.am:
11785 * gst/subenc/Makefile.am:
11786 * gst/tta/Makefile.am:
11787 * gst/vbidec/Makefile.am:
11788 * gst/videodrop/Makefile.am:
11789 * gst/videosignal/Makefile.am:
11790 * gst/virtualdub/Makefile.am:
11791 * gst/vmnc/Makefile.am:
11792 * gst/y4m/Makefile.am:
11793 * sys/acmenc/Makefile.am:
11794 * sys/cdrom/Makefile.am:
11795 * sys/dshowdecwrapper/Makefile.am:
11796 * sys/dshowsrcwrapper/Makefile.am:
11797 * sys/dvb/Makefile.am:
11798 * sys/dxr3/Makefile.am:
11799 * sys/fbdev/Makefile.am:
11800 * sys/oss4/Makefile.am:
11801 * sys/qcam/Makefile.am:
11802 * sys/qtwrapper/Makefile.am:
11803 * sys/vcd/Makefile.am:
11804 * sys/wininet/Makefile.am:
11805 * win32/common/config.h:
11806 Don't install static libs for plugins. Fixes #550851 for -bad.
11808 2008-11-03 15:30:14 +0000 Matthias Kretz <kretz@kde.org>
11810 ext/alsa/gstalsasink.c: Make all access non-blocking so that we can better handle unplugging of usb devices. Fixes #5...
11811 Original commit message from CVS:
11812 Based on patch by: Matthias Kretz <kretz at kde dot org>
11813 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
11814 (gst_alsasink_prepare), (gst_alsasink_unprepare),
11815 (gst_alsasink_write):
11816 Make all access non-blocking so that we can better handle unplugging
11817 of usb devices. Fixes #559111
11819 2008-11-03 10:49:24 +0000 Damien Lespiau <damien.lespiau@gmail.com>
11821 gst-libs/gst/rtsp/gstrtspconnection.c: Make the next call to poll not depend on previous calls to poll with or withou...
11822 Original commit message from CVS:
11823 Patch by: Damien Lespiau <damien.lespiau gmail com>
11824 * gst-libs/gst/rtsp/gstrtspconnection.c:
11825 (gst_rtsp_connection_write):
11826 Make the next call to poll not depend on previous calls to poll with or
11827 without reading from the active descriptor. Fixes #544293.
11829 2008-11-03 08:55:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11831 gst/speexresample/gstspeexresample.c: Add TODO at the top of the file for enabling SSE/ARM specific optimizations and...
11832 Original commit message from CVS:
11833 * gst/speexresample/gstspeexresample.c:
11834 (gst_speex_resample_convert_buffer):
11835 Add TODO at the top of the file for enabling SSE/ARM specific
11836 optimizations and choosing the fastest implementation at runtime.
11837 Add g_assert_not_reached() at two places that should really never
11840 2008-11-02 09:19:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11842 gst/speexresample/gstspeexresample.c: Fix format string and arguments.
11843 Original commit message from CVS:
11844 * gst/speexresample/gstspeexresample.c:
11845 (gst_speex_resample_check_discont):
11846 Fix format string and arguments.
11847 * gst/speexresample/resample_sse.h:
11850 2008-11-01 19:38:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11852 gst/speexresample/: Add missing headers to Makefile.am.
11853 Original commit message from CVS:
11854 * gst/speexresample/Makefile.am:
11855 * gst/speexresample/gstspeexresample.c:
11856 (gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
11857 (gst_speex_resample_convert_buffer), (_benchmark_int_float),
11858 (_benchmark_int_int), (_benchmark_integer_resampling),
11860 * gst/speexresample/gstspeexresample.h:
11861 * gst/speexresample/resample.c:
11862 * gst/speexresample/speex_resampler_double.c:
11863 * gst/speexresample/speex_resampler_float.c:
11864 * gst/speexresample/speex_resampler_int.c:
11865 * gst/speexresample/speex_resampler_wrapper.h:
11866 Add missing headers to Makefile.am.
11867 Update copyright, years and my mail address.
11868 Benchmark the integer resampling implementation against the
11869 float implementation and use the faster one for 8/16 bit integer
11870 input. On most recent systems the floating point version is faster.
11872 2008-10-31 09:49:57 +0000 Nick Haddad <nick@haddads.net>
11874 gst-libs/gst/riff/: Add support for other fourcc codes that are commonly used for 'uncompressed RGB', including 'RGB ...
11875 Original commit message from CVS:
11876 Patch by: Nick Haddad <nick at haddads dot net>
11877 * gst-libs/gst/riff/riff-ids.h:
11878 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
11879 Add support for other fourcc codes that are commonly used for
11880 'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
11883 2008-10-30 14:55:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11885 gst/speexresample/gstspeexresample.c: The length for the buffer conversion function is the number of audio frames, i....
11886 Original commit message from CVS:
11887 * gst/speexresample/gstspeexresample.c:
11888 (gst_speex_resample_convert_buffer):
11889 The length for the buffer conversion function is the number of
11890 audio frames, i.e. we need to multiply it by the number of channels
11891 to get the number of values. Also spotted by the unit test after
11892 running in valgrind.
11894 2008-10-30 14:46:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11896 tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
11897 Original commit message from CVS:
11898 * tests/check/elements/speexresample.c: (element_message_cb),
11899 (eos_message_cb), (test_pipeline), (GST_START_TEST),
11900 (speexresample_suite):
11901 Add pipeline unit tests for testing all supported formats with
11902 up/downsampling and different in/outrates.
11903 * gst/speexresample/gstspeexresample.c:
11904 (gst_speex_resample_push_drain), (gst_speex_resample_process):
11905 * gst/speexresample/speex_resampler_wrapper.h:
11906 Fix bugs identified by the testsuite.
11908 2008-10-30 13:44:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11910 gst/speexresample/: Add support for int8, int24 and int32 input by converting internally to/from int16 or double.
11911 Original commit message from CVS:
11912 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
11913 (gst_speex_resample_get_funcs),
11914 (gst_speex_resample_transform_size),
11915 (gst_speex_resample_convert_buffer),
11916 (gst_speex_resample_push_drain), (gst_speex_resample_process):
11917 * gst/speexresample/gstspeexresample.h:
11918 * gst/speexresample/speex_resampler_wrapper.h:
11919 Add support for int8, int24 and int32 input by converting internally
11920 to/from int16 or double.
11922 2008-10-30 12:43:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11924 Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
11925 Original commit message from CVS:
11926 * gst/speexresample/Makefile.am:
11927 * gst/speexresample/arch.h:
11928 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
11929 (gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
11930 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
11931 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
11932 (_gcd), (gst_speex_resample_transform_size),
11933 (gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
11934 (gst_speex_resample_process), (gst_speex_resample_transform),
11935 (gst_speex_resample_query), (gst_speex_resample_set_property):
11936 * gst/speexresample/gstspeexresample.h:
11937 * gst/speexresample/resample.c:
11938 * gst/speexresample/speex_resampler.h:
11939 * gst/speexresample/speex_resampler_double.c:
11940 * gst/speexresample/speex_resampler_wrapper.h:
11941 * tests/check/elements/speexresample.c: (setup_speexresample),
11942 (test_perfect_stream_instance), (GST_START_TEST),
11943 (test_discont_stream_instance):
11944 Add support for double samples as input and refactor the usage
11945 of the different compilation flavors of the speex resampler.
11947 2008-10-30 11:43:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11949 gst/audioresample/gstaudioresample.c: Return the result of parent_class->event().
11950 Original commit message from CVS:
11951 * gst/audioresample/gstaudioresample.c:
11952 Return the result of parent_class->event().
11954 2008-10-29 17:02:55 +0000 Wim Taymans <wim.taymans@gmail.com>
11956 gst-libs/gst/app/gstappsink.c: Fix the docs.
11957 Original commit message from CVS:
11958 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
11961 2008-10-29 12:11:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11963 gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
11964 Original commit message from CVS:
11965 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
11966 (gst_speex_resample_get_unit_size),
11967 (gst_speex_resample_push_drain), (gst_speex_resample_event),
11968 (gst_speex_resample_check_discont), (gst_speex_resample_process),
11969 (gst_speex_resample_transform):
11970 * gst/speexresample/gstspeexresample.h:
11971 Rewrite timestamp tracking to make it more robust and guarantee
11972 a continous stream.
11973 * tests/check/Makefile.am:
11974 * tests/check/elements/speexresample.c: (setup_speexresample),
11975 (cleanup_speexresample), (fail_unless_perfect_stream),
11976 (test_perfect_stream_instance), (GST_START_TEST),
11977 (test_discont_stream_instance), (live_switch_alloc_only_48000),
11978 (live_switch_get_sink_caps), (live_switch_push),
11979 (speexresample_suite):
11980 Add unit tests for speexresample based on the audioresample unit tests.
11982 2008-10-28 19:30:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11984 gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
11985 Original commit message from CVS:
11986 * gst/speexresample/gstspeexresample.c:
11987 (gst_speex_resample_get_unit_size),
11988 (gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
11989 (gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
11990 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
11991 (gst_speex_resample_push_drain), (gst_speex_resample_event),
11992 (gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
11993 (gst_speex_resample_process), (gst_speex_resample_transform),
11994 (gst_speex_resample_query), (gst_speex_resample_set_property):
11995 * gst/speexresample/gstspeexresample.h:
11996 Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
11997 instead of GST_DEBUG, ...
11999 2008-10-28 16:28:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12001 gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
12002 Original commit message from CVS:
12003 * gst/speexresample/gstspeexresample.c:
12004 (gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
12005 (gst_speex_resample_process):
12006 Fixate to the nearest supported rate instead of the first one.
12008 2008-10-28 16:25:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12010 gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
12011 Original commit message from CVS:
12012 * gst/audioresample/gstaudioresample.c:
12013 (gst_audioresample_class_init), (audioresample_fixate_caps):
12014 Fixate the rate to the nearest supported rate instead of
12015 the first one. Fixes bug #549510.
12017 2008-10-28 11:46:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12019 gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
12020 Original commit message from CVS:
12021 * gst/speexresample/README:
12022 * gst/speexresample/arch.h:
12023 * gst/speexresample/fixed_arm4.h:
12024 * gst/speexresample/fixed_arm5e.h:
12025 * gst/speexresample/fixed_bfin.h:
12026 * gst/speexresample/fixed_debug.h:
12027 * gst/speexresample/fixed_generic.h:
12028 * gst/speexresample/resample.c: (compute_func), (main), (sinc),
12029 (cubic_coef), (resampler_basic_direct_single),
12030 (resampler_basic_direct_double),
12031 (resampler_basic_interpolate_single),
12032 (resampler_basic_interpolate_double), (update_filter),
12033 (speex_resampler_init_frac), (speex_resampler_process_native),
12034 (speex_resampler_magic), (speex_resampler_process_float),
12035 (speex_resampler_process_int),
12036 (speex_resampler_process_interleaved_float),
12037 (speex_resampler_process_interleaved_int),
12038 (speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
12039 (speex_resampler_reset_mem):
12040 * gst/speexresample/speex_resampler.h:
12041 Update Speex resampler with latest version from Speex GIT.
12043 2008-10-27 14:57:34 +0000 Wim Taymans <wim.taymans@gmail.com>
12045 win32/common/libgstaudio.def: Add new symbols.
12046 Original commit message from CVS:
12047 * win32/common/libgstaudio.def:
12050 2008-10-23 09:57:06 +0000 Wim Taymans <wim.taymans@gmail.com>
12052 ext/vorbis/vorbisdec.c: Attempt to make obfuscated code clearer.
12053 Original commit message from CVS:
12054 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
12055 Attempt to make obfuscated code clearer.
12057 2008-10-23 07:11:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12059 Move float endianness conversion macros to core. Second part of bug ##555196.
12060 Original commit message from CVS:
12061 * docs/libs/gst-plugins-base-libs-sections.txt:
12062 * gst-libs/gst/floatcast/floatcast.h:
12063 Move float endianness conversion macros to core. Second part of
12066 2008-10-22 12:29:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12068 sys/: Don't mark as gtk-doc docs as they aren't public.
12069 Original commit message from CVS:
12070 * sys/ximage/ximagesink.h:
12071 * sys/xvimage/xvimagesink.h:
12072 Don't mark as gtk-doc docs as they aren't public.
12074 2008-10-22 12:25:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12076 Allow setting colorkey if possible. Implement property probe interface for optional X features (autopaint-colorkey, d...
12077 Original commit message from CVS:
12078 * sys/xvimage/xvimagesink.c:
12079 * sys/xvimage/xvimagesink.h:
12080 * tests/icles/Makefile.am:
12081 * tests/icles/test-colorkey.c:
12082 Allow setting colorkey if possible. Implement property probe interface
12083 for optional X features (autopaint-colorkey, double-buffer and
12084 colorkey). Fixes #554533
12086 2008-10-22 12:01:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12088 gst-libs/gst/tag/tags.c: Remove useless buffer size assignment. It already has this value.
12089 Original commit message from CVS:
12090 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
12091 Remove useless buffer size assignment. It already has this value.
12093 2008-10-20 15:35:37 +0000 Wim Taymans <wim.taymans@gmail.com>
12095 gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
12096 Original commit message from CVS:
12097 * gst-libs/gst/audio/gstaudiosink.c:
12098 (gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
12099 (gst_audioringbuffer_activate), (gst_audioringbuffer_release),
12100 (gst_audioringbuffer_stop):
12101 Implement a separate activate functions to start monitoring the segments
12102 or, in pull mode, pulling in data.
12103 * gst-libs/gst/audio/gstbaseaudiosink.c:
12104 (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
12105 (gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
12106 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
12107 (gst_base_audio_sink_activate_pull),
12108 (gst_base_audio_sink_async_play),
12109 (gst_base_audio_sink_change_state):
12110 Implement pad and element convert query function.
12111 Activate the ringbuffer.
12112 Use the segment last_stop value as the offset to pull.
12113 Use new basesink _do_preroll() method to preroll in the pulling thread.
12114 Take appropriate locking in the pulling thread.
12115 * gst-libs/gst/audio/gstringbuffer.h:
12118 2008-10-20 14:08:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12120 gst/typefind/gsttypefindfunctions.c: Improve MXF typefinding a bit by searching for a header partition pack instead o...
12121 Original commit message from CVS:
12122 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
12123 Improve MXF typefinding a bit by searching for a header partition
12124 pack instead of just a general partition pack and checking more
12125 bytes for valid values.
12127 2008-10-20 13:45:55 +0000 Wim Taymans <wim.taymans@gmail.com>
12129 tests/icles/.cvsignore: update ignore file.
12130 Original commit message from CVS:
12131 * tests/icles/.cvsignore:
12132 update ignore file.
12133 * tests/icles/Makefile.am:
12134 * tests/icles/test-box.c: (make_pipeline), (main):
12135 Add another interactive command line experimentation suite for
12136 dynamically boxing/cropping/saling an input video.
12138 2008-10-17 13:19:05 +0000 Wim Taymans <wim.taymans@gmail.com>
12140 Add methods to more accuratly control the pulling thread of a ringbuffer.
12141 Original commit message from CVS:
12142 * docs/libs/gst-plugins-base-libs-sections.txt:
12143 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
12144 (gst_ring_buffer_activate), (gst_ring_buffer_is_active):
12145 * gst-libs/gst/audio/gstringbuffer.h:
12146 Add methods to more accuratly control the pulling thread of a
12148 Add format conversion helper code to the ringbuffer.
12149 API: GstRingBuffer:gst_ring_buffer_activate()
12150 API: GstRingBuffer:gst_ring_buffer_is_active()
12151 API: GstRingBuffer:gst_ring_buffer_convert()
12153 2008-10-16 15:44:37 +0000 Wim Taymans <wim.taymans@gmail.com>
12155 gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
12156 Original commit message from CVS:
12157 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
12158 (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
12159 (gst_audioringbuffer_stop):
12160 Signal thread startup earlier so that we can immediatly go into pull
12161 mode when we have to and block on preroll.
12163 2008-10-16 15:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
12165 gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
12166 Original commit message from CVS:
12167 * gst-libs/gst/audio/gstringbuffer.c:
12168 (gst_ring_buffer_prepare_read):
12169 In pull mode we want the callback to prepull a buffer we can preroll on
12170 even when we are not yet playing.
12172 2008-10-16 15:07:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12174 Don't install static libs for plugins. Fixes #550851 for base.
12175 Original commit message from CVS:
12176 * ext/alsa/Makefile.am:
12177 * ext/cdparanoia/Makefile.am:
12178 * ext/gio/Makefile.am:
12179 * ext/gnomevfs/Makefile.am:
12180 * ext/libvisual/Makefile.am:
12181 * ext/ogg/Makefile.am:
12182 * ext/pango/Makefile.am:
12183 * ext/theora/Makefile.am:
12184 * ext/vorbis/Makefile.am:
12185 * gst/adder/Makefile.am:
12186 * gst/audioconvert/Makefile.am:
12187 * gst/audiorate/Makefile.am:
12188 * gst/audioresample/Makefile.am:
12189 * gst/audiotestsrc/Makefile.am:
12190 * gst/ffmpegcolorspace/Makefile.am:
12191 * gst/gdp/Makefile.am:
12192 * gst/playback/Makefile.am:
12193 * gst/subparse/Makefile.am:
12194 * gst/tcp/Makefile.am:
12195 * gst/typefind/Makefile.am:
12196 * gst/videorate/Makefile.am:
12197 * gst/videoscale/Makefile.am:
12198 * gst/videotestsrc/Makefile.am:
12199 * gst/volume/Makefile.am:
12200 * sys/v4l/Makefile.am:
12201 * sys/ximage/Makefile.am:
12202 * sys/xvimage/Makefile.am:
12203 Don't install static libs for plugins. Fixes #550851 for base.
12205 2008-10-16 13:50:00 +0000 Wim Taymans <wim.taymans@gmail.com>
12207 gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe...
12208 Original commit message from CVS:
12209 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
12210 Set the default blocksize to -1 because we will then use the configured
12211 samplesperbuffer to create our output buffer.
12213 2008-10-15 15:28:41 +0000 Edward Hervey <bilboed@bilboed.com>
12215 gst-libs/gst/riff/riff-media.c: Add mappping for the KMVC (Karl Morton's Video) Codec.
12216 Original commit message from CVS:
12217 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
12218 (gst_riff_create_video_template_caps):
12219 Add mappping for the KMVC (Karl Morton's Video) Codec.
12221 2008-10-15 14:25:50 +0000 Edward Hervey <bilboed@bilboed.com>
12223 gst/typefind/gsttypefindfunctions.c: Don't forget to advance the offset of what we're matching against, else we end u...
12224 Original commit message from CVS:
12225 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
12226 Don't forget to advance the offset of what we're matching against, else
12227 we end up in a forever loop.
12229 2008-10-15 11:25:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12231 gst/subparse/gstsubparse.c: Improve typefinding a bit. If we don't have a Unicode charset try GST_SUBTITLE_ENCODING a...
12232 Original commit message from CVS:
12233 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
12234 Improve typefinding a bit. If we don't have a Unicode charset
12235 try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
12237 2008-10-14 11:13:59 +0000 Edward Hervey <bilboed@bilboed.com>
12239 ext/theora/theoradec.c: Fix build on macosx.
12240 Original commit message from CVS:
12241 * ext/theora/theoradec.c: (theora_dec_decode_buffer):
12242 Fix build on macosx.
12244 2008-10-13 11:36:13 +0000 Robin Stocker <robin@nibor.org>
12246 ext/theora/: Parse input caps and make the PAR override the encoded PAR when specified by a container. Fixes #555699.
12247 Original commit message from CVS:
12248 Based on patch by: Robin Stocker <robin at nibor dot org>
12249 * ext/theora/gsttheoradec.h:
12250 * ext/theora/theoradec.c: (gst_theora_dec_init),
12251 (theora_dec_setcaps), (theora_handle_type_packet),
12252 (theora_dec_decode_buffer), (theora_dec_change_state):
12253 Parse input caps and make the PAR override the encoded PAR when
12254 specified by a container. Fixes #555699.
12256 2008-10-13 09:16:59 +0000 Wim Taymans <wim.taymans@gmail.com>
12258 gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY
12259 Original commit message from CVS:
12260 * gst-libs/gst/rtp/gstbasertpdepayload.c:
12261 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
12262 (gst_base_rtp_depayload_set_gst_timestamp),
12263 (gst_base_rtp_depayload_change_state):
12264 * gst-libs/gst/rtp/gstbasertpdepayload.h:
12265 Add some more G_LIKELY
12266 Fail when the setcaps function was not called.
12267 * gst-libs/gst/rtp/gstbasertppayload.c:
12268 (gst_basertppayload_set_outcaps):
12269 Propagate return value of setcaps.
12271 2008-10-13 08:58:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12273 gst/subparse/: Add support for UTF16/UTF32 subtitles as long as the first bytes of the first buffer contain the BOM. ...
12274 Original commit message from CVS:
12275 * gst/subparse/Makefile.am:
12276 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
12277 (gst_sub_parse_class_init), (gst_sub_parse_init),
12278 (gst_convert_to_utf8), (detect_encoding), (convert_encoding),
12279 (get_next_line), (gst_sub_parse_data_format_autodetect),
12280 (feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
12281 (gst_subparse_type_find):
12282 * gst/subparse/gstsubparse.h:
12283 Add support for UTF16/UTF32 subtitles as long as the first bytes of
12284 the first buffer contain the BOM. This also adds support for other
12285 encodings that allow NUL bytes via the encoding property.
12286 Fixes bugs #552237 and #456788.
12288 2008-10-13 08:15:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12290 gst-libs/gst/tag/tags.c: Don't drop the last byte of image tags if they're not an URI list.
12291 Original commit message from CVS:
12292 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
12293 Don't drop the last byte of image tags if they're not an URI list.
12296 2008-10-13 08:00:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12298 gst/typefind/gsttypefindfunctions.c: For looking at the 4th byte we have to get 4 bytes of course and not 3.
12299 Original commit message from CVS:
12300 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
12301 For looking at the 4th byte we have to get 4 bytes of course
12304 2008-10-13 07:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12306 gst/typefind/gsttypefindfunctions.c: Improve FLAC-without-headers typefinding by looking at most of the frame header ...
12307 Original commit message from CVS:
12308 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
12309 Improve FLAC-without-headers typefinding by looking at most of the
12310 frame header and checking if invalid values are used. Should prevent
12311 quite some false positives compared to the old version which only
12312 check if the first 14 bits are set.
12314 2008-10-11 16:27:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12316 sys/xvimage/xvimagesink.c: Don't assert on caps==NULL.
12317 Original commit message from CVS:
12318 * sys/xvimage/xvimagesink.c:
12319 Don't assert on caps==NULL.
12321 2008-10-10 17:13:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12323 Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
12324 Original commit message from CVS:
12325 * gst/subparse/gstsubparse.c:
12326 (gst_sub_parse_data_format_autodetect), (handle_buffer),
12327 (gst_sub_parse_change_state):
12328 * gst/subparse/gstsubparse.h:
12329 * tests/check/elements/subparse.c: (GST_START_TEST):
12330 Add support for subtitle files with UTF-8 BOM at the beginning
12331 by simple stripping it from the first line before passing it
12332 to any parsing code. Fixes bug #555257 and playback of files
12333 created by Gnome Subtitles.
12335 2008-10-10 15:45:15 +0000 Wim Taymans <wim.taymans@gmail.com>
12337 gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place.
12338 Original commit message from CVS:
12339 * gst/audiotestsrc/gstaudiotestsrc.c:
12340 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
12341 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
12342 (gst_audio_test_src_start), (gst_audio_test_src_stop),
12343 (gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
12344 (gst_audio_test_src_create):
12345 * gst/audiotestsrc/gstaudiotestsrc.h:
12346 Define the default property values in the usual place.
12347 Implement start/stop to reset values correctly.
12348 Calculate the sample size only once when we negotiate.
12349 Rename some values to make more sense.
12350 Keep track of our byte range.
12351 Add support for pull based scheduling. Disabled for now until we have
12352 the whole stack working.
12353 Set the BUFFER_OFFSET correctly.
12355 2008-10-10 15:32:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12357 Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
12358 Original commit message from CVS:
12359 Based on a patch by: xavierb at gmail dot com
12360 * gst/subparse/gstsubparse.c:
12361 (gst_sub_parse_data_format_autodetect):
12362 * tests/check/elements/subparse.c: (GST_START_TEST):
12363 Make the detection of the used subtitle a bit less strict
12364 for srt subtitles. Fixes bug #555607.
12366 2008-10-10 15:21:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12368 ext/vorbis/vorbisenc.c: Fix discontinuity detection which was broken by last commit.
12369 Original commit message from CVS:
12370 * ext/vorbis/vorbisenc.c:
12371 (gst_vorbis_enc_buffer_check_discontinuous):
12372 Fix discontinuity detection which was broken by last commit.
12374 2008-10-09 11:18:09 +0000 Tim-Philipp Müller <tim@centricular.net>
12376 configure.ac: Require core CVS for ghostpad API additions used by decodebin2.
12377 Original commit message from CVS:
12379 Require core CVS for ghostpad API additions used by decodebin2.
12381 2008-10-08 15:30:33 +0000 Edward Hervey <bilboed@bilboed.com>
12383 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
12384 Original commit message from CVS:
12385 * gst-libs/gst/audio/gstbaseaudiosrc.c:
12386 (gst_base_audio_src_create):
12387 Fix debug statements (space between '%' and actual format).
12389 2008-10-08 14:44:04 +0000 Wim Taymans <wim.taymans@gmail.com>
12391 gst/playback/gstdecodebin2.c: Remove bogus assert, the decodepad could have been created inside an already existing g...
12392 Original commit message from CVS:
12393 * gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
12394 Remove bogus assert, the decodepad could have been created inside an
12395 already existing group.
12397 2008-10-08 14:01:42 +0000 Andy Wingo <wingo@pobox.com>
12401 Original commit message from CVS:
12404 2008-10-08 14:00:07 +0000 Andy Wingo <wingo@pobox.com>
12406 gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
12407 Original commit message from CVS:
12408 2008-10-08 Andy Wingo <wingo@pobox.com>
12409 * gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
12410 target instead of setting it.
12411 (gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
12412 API for a decode pad. The bugfix is that we set the group in
12413 activate(), not when the pad was created because it might be NULL
12415 (gst_decode_group_control_source_pad, gst_decode_group_expose):
12416 Update to use the API.
12418 2008-10-08 12:49:40 +0000 Andy Wingo <wingo@pobox.com>
12420 gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
12421 Original commit message from CVS:
12422 2008-10-08 Andy Wingo <wingo@pobox.com>
12423 * gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
12424 be a subclass of GstGhostPad.
12425 (analyze_new_pad): So, when emitting the signals that determine
12426 how we do autoplugging, already create the ghost pad and use it as
12427 the pad in the signal arguments. This allows applications to make
12428 a connection between the pad passed in e.g. autoplug-continue, and
12429 the pad passed in new-decoded-pad.
12430 (connect_pad, expose_pad): Update to receive the ghosted decode
12431 pad in the args, retargetting it as necessary if we have to plug
12432 the target pad through a multiqueue.
12433 (gst_decode_group_control_source_pad): Adapt to receive an
12434 already-ghosted pad that just needs activation, blocking, and
12435 drain notification.
12436 (sort_end_pads): Adapt for decode pads actually being pads.
12437 (gst_decode_group_expose): Adapt for decode pads actually being
12438 pads. Rewrite the decode pad names so they appear in order. Adds a
12439 new error case if we couldn't set the name.
12440 (gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
12442 (gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
12443 New API for the decode pad, needed because we shouldn't do these
12444 things inside gst_decode_pad_new(), but after.
12445 (gst_decode_pad_new): Change to actually make the real pad, and
12446 delay the blocking/drainage bits.
12448 2008-10-08 12:12:01 +0000 Daniel Drake <dsd@laptop.org>
12450 ext/ogg/gstoggmux.c: Unref all buffers when clearing collectpads. Fixes bug #546955.
12451 Original commit message from CVS:
12452 Patch by: Daniel Drake <dsd at laptop dot org>
12453 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
12454 Unref all buffers when clearing collectpads. Fixes bug #546955.
12456 2008-10-08 12:08:01 +0000 Klaas <klaas@rivercrew.net>
12458 ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
12459 Original commit message from CVS:
12460 Based on a patch by: Klaas <klaas at rivercrew dot net>
12461 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
12462 (gst_vorbis_enc_buffer_check_discontinuous),
12463 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
12464 * ext/vorbis/vorbisenc.h:
12465 Keep track of the upstream segments and use the running time on that
12466 segment instead of the buffer timestamp everywhere. Fixes bug #525807.
12468 2008-10-08 11:50:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12470 gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
12471 Original commit message from CVS:
12472 * gst/audioconvert/audioconvert.c: (audio_convert_convert):
12473 Prevent overflows with big buffer when calculating the size of
12474 the intermediate buffer by using gst_util_uint64_scale() instead of
12475 plain arithmetics. Fixes bug #552801.
12477 2008-10-08 10:49:15 +0000 Pavel Zeldin <pzeldin@gmail.com>
12479 ext/pango/gstclockoverlay.*: API: Add ability to specify format for date/time display by adding a "time-format" prope...
12480 Original commit message from CVS:
12481 Patch by: Pavel Zeldin <pzeldin at gmail dot com>
12482 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
12483 (gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
12484 (gst_clock_overlay_init), (gst_clock_overlay_set_property),
12485 (gst_clock_overlay_get_property):
12486 * ext/pango/gstclockoverlay.h:
12487 API: Add ability to specify format for date/time display by
12488 adding a "time-format" property.
12491 2008-10-08 09:22:26 +0000 Jan Gerber <j@oil21.org>
12493 gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
12494 Original commit message from CVS:
12495 Patch by: Jan Gerber <j at oil21 dot org>
12496 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
12497 (gst_riff_create_video_template_caps):
12498 Add FFV1 fourcc to support playback of FFMPEG lossless video
12499 in AVI. Fixes bug #555319.
12501 2008-10-08 09:12:36 +0000 Håvard Graff <havard.graff@tandberg.com>
12503 gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
12504 Original commit message from CVS:
12505 Patch by: Håvard Graff <havard dot graff at tandberg dot com>
12506 * gst-libs/gst/audio/gstbaseaudiosrc.c:
12507 (gst_base_audio_src_create):
12508 Implement skew clock slaving. Fixes #552559.
12510 2008-10-08 09:10:23 +0000 Wim Taymans <wim.taymans@gmail.com>
12512 gst-libs/gst/audio/: Fix include of config.h
12513 Original commit message from CVS:
12514 * gst-libs/gst/audio/multichannel.c:
12515 * gst-libs/gst/audio/testchannels.c:
12516 Fix include of config.h
12518 2008-10-06 16:36:20 +0000 Tero Saarni <tero.saarni@gmail.com>
12520 gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
12521 Original commit message from CVS:
12522 Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
12523 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
12524 (print_media), (gst_sdp_message_dump):
12525 Fix parsing of the c= field containing multicast addresses.
12527 Add the connection info to the session or streams.
12528 Fix parsing of the bandwidth.
12529 Add debugging for the connections and bandwidths for a media.
12530 Add debugging for the bandwidth of the session.
12532 2008-10-06 16:31:27 +0000 Wim Taymans <wim.taymans@gmail.com>
12534 gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
12535 Original commit message from CVS:
12536 * gst-libs/gst/rtp/gstbasertppayload.c:
12537 (gst_basertppayload_change_state):
12538 Configure the next seqnum and timestamp in the state change so that they
12539 can be queried soon after.
12541 2008-10-06 16:29:33 +0000 Wim Taymans <wim.taymans@gmail.com>
12543 gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
12544 Original commit message from CVS:
12545 * gst-libs/gst/rtp/gstbasertpdepayload.c:
12546 (gst_base_rtp_depayload_chain):
12547 Improve debugging of the rtptime.
12549 2008-10-05 11:33:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12551 configure.ac: Back to development -> 0.10.21.1
12552 Original commit message from CVS:
12554 Back to development -> 0.10.21.1
12556 2008-10-05 08:18:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12560 Original commit message from CVS:
12563 2008-10-05 08:11:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12565 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
12566 Original commit message from CVS:
12567 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
12569 Add typefinder for MXF.
12571 2008-10-05 08:10:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12573 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
12574 Original commit message from CVS:
12575 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
12577 Add typefinder for MXF.
12579 2008-10-03 15:19:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12581 tests/icles/Makefile.am: Only build test-colorkey if GTK+ is available.
12582 Original commit message from CVS:
12583 * tests/icles/Makefile.am:
12584 Only build test-colorkey if GTK+ is available.
12586 === release 0.10.21 ===
12588 2008-10-03 00:03:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12594 * docs/plugins/gst-plugins-base-plugins.args:
12595 * docs/plugins/gst-plugins-base-plugins.hierarchy:
12596 * docs/plugins/gst-plugins-base-plugins.interfaces:
12597 * docs/plugins/gst-plugins-base-plugins.prerequisites:
12598 * docs/plugins/inspect/plugin-adder.xml:
12599 * docs/plugins/inspect/plugin-alsa.xml:
12600 * docs/plugins/inspect/plugin-audioconvert.xml:
12601 * docs/plugins/inspect/plugin-audiorate.xml:
12602 * docs/plugins/inspect/plugin-audioresample.xml:
12603 * docs/plugins/inspect/plugin-audiotestsrc.xml:
12604 * docs/plugins/inspect/plugin-cdparanoia.xml:
12605 * docs/plugins/inspect/plugin-decodebin.xml:
12606 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
12607 * docs/plugins/inspect/plugin-gdp.xml:
12608 * docs/plugins/inspect/plugin-gio.xml:
12609 * docs/plugins/inspect/plugin-gnomevfs.xml:
12610 * docs/plugins/inspect/plugin-libvisual.xml:
12611 * docs/plugins/inspect/plugin-ogg.xml:
12612 * docs/plugins/inspect/plugin-pango.xml:
12613 * docs/plugins/inspect/plugin-playback.xml:
12614 * docs/plugins/inspect/plugin-queue2.xml:
12615 * docs/plugins/inspect/plugin-subparse.xml:
12616 * docs/plugins/inspect/plugin-tcp.xml:
12617 * docs/plugins/inspect/plugin-theora.xml:
12618 * docs/plugins/inspect/plugin-typefindfunctions.xml:
12619 * docs/plugins/inspect/plugin-uridecodebin.xml:
12620 * docs/plugins/inspect/plugin-video4linux.xml:
12621 * docs/plugins/inspect/plugin-videorate.xml:
12622 * docs/plugins/inspect/plugin-videoscale.xml:
12623 * docs/plugins/inspect/plugin-videotestsrc.xml:
12624 * docs/plugins/inspect/plugin-volume.xml:
12625 * docs/plugins/inspect/plugin-vorbis.xml:
12626 * docs/plugins/inspect/plugin-ximagesink.xml:
12627 * docs/plugins/inspect/plugin-xvimagesink.xml:
12628 * gst-plugins-base.doap:
12629 * win32/common/config.h:
12631 Original commit message from CVS:
12634 2008-10-02 23:44:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12665 Original commit message from CVS:
12668 2008-09-28 22:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12670 configure.ac: 0.10.20.4 pre-release
12671 Original commit message from CVS:
12673 0.10.20.4 pre-release
12675 2008-09-25 10:46:00 +0000 ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>
12677 ext/theora/theoraparse.c: Set the BOS flag on the BOS packet. Fixes #553244.
12678 Original commit message from CVS:
12679 Patch by: ogg.k.ogg.k <ogg dot k dot ogg dot k at googlemail dot com>
12680 * ext/theora/theoraparse.c: (theora_parse_set_streamheader):
12681 Set the BOS flag on the BOS packet. Fixes #553244.
12683 2008-09-23 17:48:14 +0000 Wim Taymans <wim.taymans@gmail.com>
12685 gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
12686 Original commit message from CVS:
12687 * gst-libs/gst/rtsp/gstrtspmessage.c:
12688 (gst_rtsp_message_parse_request),
12689 (gst_rtsp_message_parse_response):
12690 Fix the g_return_val_if_fail() statements.
12692 2008-09-22 17:44:14 +0000 Michael Smith <msmith@xiph.org>
12694 gst-libs/gst/tag/gsttagdemux.c: Fail to activate if there's insufficient data in the file to be usable, preventing an...
12695 Original commit message from CVS:
12696 * gst-libs/gst/tag/gsttagdemux.c:
12697 Fail to activate if there's insufficient data in the file to be usable,
12698 preventing an assertion fail later. Fixes #552960
12700 2008-09-16 15:36:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12702 Commit stuff that should have gone in last week when I made the pre-releases:
12703 Original commit message from CVS:
12704 Commit stuff that should have gone in last week when I made the pre-releases:
12705 2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
12707 0.10.20.2 pre-release
12713 2008-09-15 15:11:18 +0000 Tim-Philipp Müller <tim@centricular.net>
12715 gst/: Recognise Kate subtitle streams (#550582).
12716 Original commit message from CVS:
12717 * gst-libs/gst/pbutils/descriptions.c:
12718 * gst/typefind/gsttypefindfunctions.c:
12719 Recognise Kate subtitle streams (#550582).
12721 2008-09-13 11:04:02 +0000 Tim-Philipp Müller <tim@centricular.net>
12723 gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
12724 Original commit message from CVS:
12725 * gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
12726 Remove trailing comma from enum list, which causes problems
12727 with -pendantic (#550729).
12729 2008-09-05 19:04:47 +0000 Tim-Philipp Müller <tim@centricular.net>
12731 gst-libs/gst/interfaces/propertyprobe.c: More sanity checks for our second-favourite interface.
12732 Original commit message from CVS:
12733 * gst-libs/gst/interfaces/propertyprobe.c:
12734 (gst_property_probe_get_properties),
12735 (gst_property_probe_get_property),
12736 (gst_property_probe_probe_property),
12737 (gst_property_probe_probe_property_name),
12738 (gst_property_probe_needs_probe),
12739 (gst_property_probe_needs_probe_name),
12740 (gst_property_probe_get_values),
12741 (gst_property_probe_get_values_name),
12742 (gst_property_probe_probe_and_get_values),
12743 (gst_property_probe_probe_and_get_values_name):
12744 More sanity checks for our second-favourite interface.
12746 2008-09-05 14:12:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12748 gst-libs/gst/interfaces/propertyprobe.c: Check for NULL pointer, in the hope that this fixes #532864.
12749 Original commit message from CVS:
12750 * gst-libs/gst/interfaces/propertyprobe.c:
12751 Check for NULL pointer, in the hope that this fixes #532864.
12753 2008-09-05 10:24:05 +0000 Tim-Philipp Müller <tim@centricular.net>
12755 sys/xvimage/xvimagesink.c: No really, the next release is 0.10.21 (fix Since: tags in docs).
12756 Original commit message from CVS:
12757 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
12758 No really, the next release is 0.10.21 (fix Since: tags in docs).
12760 2008-09-04 16:25:06 +0000 Wim Taymans <wim.taymans@gmail.com>
12762 gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
12763 Original commit message from CVS:
12764 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
12765 Disable a code path that is now called but causes a deadlock for some
12766 reason and is unneeded.
12768 2008-09-04 13:46:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12770 sys/xvimage/xvimagesink.*: Add a "draw-border" property that can be set to false to disable drawing borders.
12771 Original commit message from CVS:
12772 * sys/xvimage/xvimagesink.c:
12773 * sys/xvimage/xvimagesink.h:
12774 Add a "draw-border" property that can be set to false to disable
12776 * tests/icles/test-colorkey.c:
12777 * tests/icles/Makefile.am:
12778 Add new test application for the colorkey handling.
12780 2008-09-03 14:00:06 +0000 Edward Hervey <bilboed@bilboed.com>
12782 gst-libs/gst/riff/riff-media.c: Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
12783 Original commit message from CVS:
12784 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
12785 Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
12786 This will also be fixed for upcoming gst-ffmpeg release so that once
12787 this release of -base is out, it will work with the latest gst-ffmpeg
12790 2008-09-03 13:27:20 +0000 Edward Hervey <bilboed@bilboed.com>
12792 gst-libs/gst/riff/riff-media.c: Add Truespeech mapping for RIFF formats (AVI/WAV).
12793 Original commit message from CVS:
12794 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
12795 (gst_riff_create_audio_template_caps):
12796 Add Truespeech mapping for RIFF formats (AVI/WAV).
12799 2008-09-03 12:23:44 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
12801 gst/typefind/gsttypefindfunctions.c: Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
12802 Original commit message from CVS:
12803 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
12804 Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
12807 2008-09-03 10:12:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12809 Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
12810 Original commit message from CVS:
12812 * gst/subparse/Makefile.am:
12813 * gst/subparse/gstsubparse.c:
12814 * gst/subparse/samiparse.c:
12815 * tests/check/elements/subparse.c:
12816 Rework last change, so that we build subparse, but just disable the
12817 sami parse functionality, if we're configured to not use xml. In the
12818 tests only the sami test is disabled now.
12820 2008-09-02 15:07:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12822 configure.ac: Disable subparse when xml is disabled. It woundn't work anyway. Fixes test runs.
12823 Original commit message from CVS:
12825 Disable subparse when xml is disabled. It woundn't work anyway. Fixes
12828 2008-09-02 09:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
12830 po/POTFILES.in: Add some more files with strings for translation.
12831 Original commit message from CVS:
12833 Add some more files with strings for translation.
12835 2008-09-02 06:37:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12837 Use new geo location tags from core. Fixes #481169
12838 Original commit message from CVS:
12839 * gst-libs/gst/tag/gstvorbistag.c:
12840 * tests/check/libs/tag.c:
12841 Use new geo location tags from core. Fixes #481169
12843 2008-09-01 16:05:45 +0000 Edward Hervey <bilboed@bilboed.com>
12845 tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
12846 Original commit message from CVS:
12847 * tests/check/elements/audioresample.c: (setup_audioresample),
12848 (fail_unless_perfect_stream), (test_perfect_stream_instance),
12849 (test_discont_stream_instance):
12850 Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
12851 Add debugging for coherence.
12853 2008-08-30 15:55:06 +0000 Jonathan Matthew <notverysmart@gmail.com>
12855 gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
12856 Original commit message from CVS:
12857 Patch by: Jonathan Matthew <notverysmart gmail com>
12858 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
12859 Add typefinder for PDF documents (which is nice to have, since it's a
12860 common format, but also helps prevent false positives). Fixes #549814.
12862 2008-08-27 15:30:16 +0000 Wim Taymans <wim.taymans@gmail.com>
12864 gst/playback/gstplaybin2.c: Fix nasty race where multiple decodebins could start pushing data before we manage to con...
12865 Original commit message from CVS:
12866 * gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
12868 Fix nasty race where multiple decodebins could start pushing data before
12869 we manage to configure the sinks, resulting in not-linked errors in
12870 typical RTSP streaming cases.
12872 2008-08-26 17:24:31 +0000 Wim Taymans <wim.taymans@gmail.com>
12874 gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
12875 Original commit message from CVS:
12876 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
12877 Since we now call stop, we trigger this code path that causes a deadlock
12878 is apparently not needed.
12880 2008-08-26 15:45:36 +0000 Wim Taymans <wim.taymans@gmail.com>
12882 gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
12883 Original commit message from CVS:
12884 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
12885 (gst_ring_buffer_stop):
12886 Also allow the case where the ringbuffer was paused when we try to stop
12887 it so that the basesrc stop function is still called.
12889 2008-08-23 15:25:44 +0000 Mike Ruprecht <cmaiku@gmail.com>
12891 sys/v4l/gstv4lelement.c: Reprobe devices again instead of taking a cached list as new devices could've been plugged i...
12892 Original commit message from CVS:
12893 Patch by: Mike Ruprecht <cmaiku at gmail dot com>
12894 * sys/v4l/gstv4lelement.c: (gst_v4l_class_probe_devices):
12895 Reprobe devices again instead of taking a cached list as new
12896 devices could've been plugged in. Fixes bug #549062.
12898 2008-08-23 15:19:59 +0000 Alessandro Dessina <alessandro@nnva.org>
12900 ext/ogg/gstoggdemux.c: Don't add pads and activate them for skeleton streams. These are already handled inside oggdem...
12901 Original commit message from CVS:
12902 Patch by: Alessandro Dessina <alessandro nnva org>
12903 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
12904 (gst_ogg_demux_activate_chain):
12905 Don't add pads and activate them for skeleton streams. These are already
12906 handled inside oggdemux. Fixes bug #537599.
12908 2008-08-22 15:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
12910 ext/vorbis/vorbisdec.c: Reset variable so that query and convert fail after going back to
12911 Original commit message from CVS:
12912 * ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
12913 Reset variable so that query and convert fail after going back to
12914 READY. Fixes #548898.
12916 2008-08-22 07:24:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12918 ext/vorbis/vorbisenc.c: If a buffer arrives with a timestamp before the timestamp+duration of the previous buffer cli...
12919 Original commit message from CVS:
12920 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
12921 If a buffer arrives with a timestamp before the timestamp+duration
12922 of the previous buffer clip it instead of dropping it completely.
12923 Slight improvement for the unfixable bug #548913.
12925 2008-08-21 14:19:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12927 ext/vorbis/vorbisdec.c: Take the current timestamp instead of timestamp+duration for the offset.
12928 Original commit message from CVS:
12929 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
12930 Take the current timestamp instead of timestamp+duration for the offset.
12931 This offset will later be used for calculating the timestamp and
12932 otherwise vorbisdec will interpolate timestamps wrong if upstream
12933 only sends timestamps and no granulepos.
12935 2008-08-21 11:20:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12937 tests/examples/seek/seek.c: Don't crash when having no visualisations.
12938 Original commit message from CVS:
12939 * tests/examples/seek/seek.c:
12940 Don't crash when having no visualisations.
12942 2008-08-16 20:57:27 +0000 David Schleef <ds@schleef.org>
12944 gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
12945 Original commit message from CVS:
12946 * gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
12947 check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
12950 2008-08-15 07:24:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12952 gst-libs/gst/pbutils/missing-plugins.c: When cleaning up the caps fields also remove "depth" for the same reason we r...
12953 Original commit message from CVS:
12954 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
12955 When cleaning up the caps fields also remove "depth" for the same
12956 reason we remove "width".
12958 2008-08-14 17:14:53 +0000 Tim-Philipp Müller <tim@centricular.net>
12960 gst-libs/gst/pbutils/descriptions.c: Add Lead H.264 here as well.
12961 Original commit message from CVS:
12962 * gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
12963 Add Lead H.264 here as well.
12965 2008-08-14 15:17:31 +0000 Julien Moutte <julien@moutte.net>
12967 gst-libs/gst/riff/riff-media.c: Add Lead H.264 variant.
12968 Original commit message from CVS:
12969 2008-08-14 Julien Moutte <julien@fluendo.com>
12970 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
12971 (gst_riff_create_video_template_caps): Add Lead H.264 variant.
12973 2008-08-13 09:17:38 +0000 Wim Taymans <wim.taymans@gmail.com>
12975 gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
12976 Original commit message from CVS:
12977 * gst-libs/gst/audio/gstbaseaudiosrc.c:
12978 (gst_base_audio_src_create):
12979 When not slaved to another clock also subtract the base_time from our
12980 internal clock time to get the running time.
12982 2008-08-13 00:59:07 +0000 David Schleef <ds@schleef.org>
12984 ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate, since it has no basis in libtheora.
12985 Original commit message from CVS:
12986 * ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
12987 since it has no basis in libtheora.
12989 2008-08-12 06:31:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12991 gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
12992 Original commit message from CVS:
12993 * gst-libs/gst/interfaces/propertyprobe.h:
12994 Remove double "interface" from doc-string.
12995 * gst-libs/gst/interfaces/xoverlay.h:
12996 Document interface.
12997 * gst-libs/gst/riff/riff.c:
12998 Add basic doc blobs.
13000 2008-08-11 15:05:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13002 gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
13003 Original commit message from CVS:
13004 * gst-libs/gst/audio/Makefile.am:
13005 Don't try to build that example anymore.
13007 2008-08-11 14:51:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13009 gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
13010 Original commit message from CVS:
13011 * gst-libs/gst/audio/.cvsignore:
13012 * gst-libs/gst/audio/Makefile.am:
13013 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
13014 * gst-libs/gst/audio/make_filter:
13015 Move audiofiltertemplate to gst-template.
13017 2008-08-11 09:20:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13019 More docs and shuffling. What can we do with the hundreds of #defines.
13020 Original commit message from CVS:
13021 * docs/libs/gst-plugins-base-libs-sections.txt:
13022 * gst-libs/gst/audio/gstaudiosrc.h:
13023 More docs and shuffling. What can we do with the hundreds of #defines.
13025 2008-08-11 08:34:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13027 gst-libs/gst/: Reducing number of dundocumented symbols.
13028 Original commit message from CVS:
13029 * gst-libs/gst/audio/audio.h:
13030 * gst-libs/gst/audio/gstaudiofilter.h:
13031 * gst-libs/gst/audio/gstringbuffer.h:
13032 * gst-libs/gst/interfaces/propertyprobe.h:
13033 * gst-libs/gst/tag/gsttagdemux.h:
13034 Reducing number of dundocumented symbols.
13036 2008-08-11 07:16:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13038 gst-libs/gst/audio/audio.c: Fix doc comment syntax.
13039 Original commit message from CVS:
13040 * gst-libs/gst/audio/audio.c:
13041 Fix doc comment syntax.
13042 * gst-libs/gst/interfaces/propertyprobe.c:
13043 Add more doc-comments and a FIXME: for the signal.
13045 2008-08-07 16:11:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13047 ext/ogg/gstoggmux.*: Don't pretend to support NEWSEGMENT events, instead override the
13048 Original commit message from CVS:
13049 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sink_event),
13050 (gst_ogg_mux_request_new_pad):
13051 * ext/ogg/gstoggmux.h:
13052 Don't pretend to support NEWSEGMENT events, instead override the
13053 GstCollectPads event function to return FALSE on NEWSEGMENT events
13054 and do the normal work for other events.
13055 This prevents elements like flacenc to seek to the start and rewrite
13056 some data which then results in a broken Ogg packet.
13058 2008-08-07 15:58:58 +0000 Frederic Crozat <fcrozat@mandriva.org>
13060 Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
13061 Original commit message from CVS:
13062 Patch by: Frederic Crozat <fcrozat@mandriva.org>
13063 * ext/alsa/gstalsaplugin.c: (plugin_init):
13064 * ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
13065 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
13066 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
13067 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
13068 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
13069 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
13070 * gst/playback/gstdecodebin.c: (plugin_init):
13071 * gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
13072 * gst/playback/gstplayback.c: (plugin_init):
13073 * gst/playback/gstqueue2.c: (plugin_init):
13074 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
13075 * sys/v4l/gstv4l.c: (plugin_init):
13076 Make sure gettext returns translations in UTF-8 encoding rather
13077 than in the current locale encoding (#546822).
13079 2008-08-06 13:12:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13081 gst-libs/gst/pbutils/descriptions.c: Add audio/x-qdm for qtdemux.
13082 Original commit message from CVS:
13083 * gst-libs/gst/pbutils/descriptions.c:
13084 Add audio/x-qdm for qtdemux.
13086 2008-08-05 15:38:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13088 ext/vorbis/vorbisdec.c: Do not leak old taglist.
13089 Original commit message from CVS:
13090 * ext/vorbis/vorbisdec.c:
13091 Do not leak old taglist.
13093 2008-08-04 12:35:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13095 tests/icles/test-scale.c: Include <stdlib.h> for atoi().
13096 Original commit message from CVS:
13097 * tests/icles/test-scale.c:
13098 Include <stdlib.h> for atoi().
13100 2008-08-04 09:11:08 +0000 Andy Wingo <wingo@pobox.com>
13102 gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
13103 Original commit message from CVS:
13104 2008-08-04 Andy Wingo <wingo@pobox.com>
13105 * gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
13108 2008-08-01 13:06:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13110 gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
13111 Original commit message from CVS:
13112 * gst/adder/gstadder.c:
13113 Cleanup lots of empty lines that came from gst-indent going havoc
13114 before I added the INDENT_ON/OFF marker some time agao.
13116 2008-08-01 11:55:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13118 Bump requirement to latest core and use new tag for riff formats.
13119 Original commit message from CVS:
13121 * gst-libs/gst/riff/riff-read.c:
13122 Bump requirement to latest core and use new tag for riff formats.
13123 Needed for #520694.
13125 2008-08-01 11:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
13127 tests/examples/dynamic/: Add example app that dynamically switches between 3 'encoders'.
13128 Original commit message from CVS:
13129 * tests/examples/dynamic/Makefile.am:
13130 * tests/examples/dynamic/codec-select.c: (make_encoder),
13131 (make_pipeline), (do_switch), (my_bus_callback), (main):
13132 Add example app that dynamically switches between 3 'encoders'.
13134 2008-07-31 13:06:13 +0000 Wim Taymans <wim.taymans@gmail.com>
13136 gst/playback/gstplaysink.c: Add some more comments.
13137 Original commit message from CVS:
13138 * gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
13139 Add some more comments.
13141 2008-07-31 12:58:44 +0000 Wim Taymans <wim.taymans@gmail.com>
13143 gst/videotestsrc/gstvideotestsrc.c: Discard buffers of the wrong size after renegotiation, this is perfectly possible...
13144 Original commit message from CVS:
13145 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
13146 (gst_video_test_src_create):
13147 Discard buffers of the wrong size after renegotiation, this is perfectly
13148 possible with things like capsfilter that could suggest caps changes
13149 upstream without knowing the size of the buffer.
13151 2008-07-31 11:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
13153 tests/icles/: Add dynamic rescaling tests for the new basetransform.
13154 Original commit message from CVS:
13155 * tests/icles/.cvsignore:
13156 * tests/icles/Makefile.am:
13157 * tests/icles/test-scale.c: (make_pipeline), (main):
13158 Add dynamic rescaling tests for the new basetransform.
13160 2008-07-30 19:51:36 +0000 Tim-Philipp Müller <tim@centricular.net>
13162 gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
13163 Original commit message from CVS:
13164 * gst/audioconvert/Makefile.am:
13165 Dist recently-added gstfastrandom.h.
13167 2008-07-30 15:29:44 +0000 Edward Hervey <bilboed@bilboed.com>
13169 sys/xvimage/xvimagesink.c: Fix a "may be used uninitialized in this function" which weirdly only appears on macosx (?).
13170 Original commit message from CVS:
13171 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
13172 Fix a "may be used uninitialized in this function" which weirdly only
13173 appears on macosx (?).
13175 2008-07-30 09:02:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13177 gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
13178 Original commit message from CVS:
13179 * gst-libs/gst/riff/riff-ids.h:
13180 Adding acid chunk for tempo and loop information.
13182 2008-07-29 13:01:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13184 sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
13185 Original commit message from CVS:
13186 * sys/xvimage/Makefile.am:
13187 floor() needs linking to $(LIBM).
13189 2008-07-29 12:35:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13191 ext/gnomevfs/gstgnomevfssrc.c: Aggregate short reads and add some comments and debug logging.
13192 Original commit message from CVS:
13193 * ext/gnomevfs/gstgnomevfssrc.c:
13194 Aggregate short reads and add some comments and debug logging.
13197 2008-07-29 10:26:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13199 gst/playback/gstplaybasebin.c: Fix property doc markup (its not a signal).
13200 Original commit message from CVS:
13201 * gst/playback/gstplaybasebin.c:
13202 Fix property doc markup (its not a signal).
13203 * sys/xvimage/xvimagesink.c:
13204 Add since tag for new proeprties (also add sice tags fro the last two
13207 2008-07-29 08:59:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13209 sys/xvimage/xvimagesink.*: Add autofill/colorkey properties. Fixes #538656.
13210 Original commit message from CVS:
13211 * sys/xvimage/xvimagesink.c:
13212 * sys/xvimage/xvimagesink.h:
13213 Add autofill/colorkey properties. Fixes #538656.
13215 2008-07-29 01:58:05 +0000 David Schleef <ds@schleef.org>
13217 sys/xvimage/xvimagesink.c: Fix rounding errors when converting colorbalance values between hardware and object proper...
13218 Original commit message from CVS:
13219 * sys/xvimage/xvimagesink.c:
13220 Fix rounding errors when converting colorbalance values
13221 between hardware and object property ranges. Partial
13222 fix for #537889, however, there still seems to be a small
13223 drift problem that could be totem's fault.
13225 2008-07-28 15:34:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13227 ext/ogg/gstoggdemux.c: Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
13228 Original commit message from CVS:
13229 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
13230 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
13231 Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
13232 This fixes a critical warning.
13234 2008-07-28 13:12:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13236 ext/ogg/gstoggmux.c: Allow muxing of CELT into Ogg streams.
13237 Original commit message from CVS:
13238 * ext/ogg/gstoggmux.c:
13239 Allow muxing of CELT into Ogg streams.
13241 2008-07-28 12:47:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13243 gst/typefind/gsttypefindfunctions.c: Add simple typefinder for the CELT codec (www.celt-codec.org).
13244 Original commit message from CVS:
13245 * gst/typefind/gsttypefindfunctions.c: (celt_type_find),
13247 Add simple typefinder for the CELT codec (www.celt-codec.org).
13249 2008-07-27 11:12:41 +0000 Jan Gerber <j@oil21.org>
13251 ext/ogg/gstoggdemux.c: Fix calculation of the start time from skeleton streams.
13252 Original commit message from CVS:
13253 Patch by: Jan Gerber <j at oil21 dot org>
13254 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
13255 Fix calculation of the start time from skeleton streams.
13258 2008-07-24 13:19:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13260 tests/examples/seek/seek.c: Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
13261 Original commit message from CVS:
13262 * tests/examples/seek/seek.c:
13263 Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
13265 2008-07-23 18:34:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13267 gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
13268 Original commit message from CVS:
13269 * gst/audioconvert/audioconvert.h:
13270 * gst/audioconvert/gstaudioquantize.c:
13271 (gst_audio_quantize_setup_dither),
13272 (gst_audio_quantize_free_dither):
13273 * gst/audioconvert/gstfastrandom.h:
13274 Implement a linear congruential generator as pseudo random number
13275 generator for the dither noise. This is about 2 times faster than
13276 using GLib's mersenne twister. Also this uses only integer math for
13277 generating integers while GLib internally uses floating point math.
13279 2008-07-23 18:27:15 +0000 Michael Smith <msmith@xiph.org>
13281 configure.ac: Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
13282 Original commit message from CVS:
13284 Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
13286 2008-07-23 13:17:31 +0000 Damien Lespiau <damien.lespiau@gmail.com>
13288 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
13289 Original commit message from CVS:
13290 Patch by: Damien Lespiau <damien.lespiau gmail com>
13291 * gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
13292 Use GST_STR_NULL to avoid crashes with libcs that don't
13293 like NULL strings in printf args (such as the win32 one).
13296 2008-07-17 14:21:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13298 sys/xvimage/xvimagesink.c: Oops - set the size of the image used for probing back to 1x1, for consistency with ximage...
13299 Original commit message from CVS:
13300 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
13301 Oops - set the size of the image used for probing back to 1x1, for
13302 consistency with ximagesink
13304 2008-07-17 13:57:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13306 sys/: it's not legal to ask the
13307 Original commit message from CVS:
13308 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
13309 (gst_ximagesink_ximage_new):
13310 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
13311 (gst_xvimagesink_xvimage_new):
13312 Apparently on Solaris and OS/X (at least), it's not legal to ask the
13313 X server to attach to a shared memory segment after we've deleted it,
13314 with the result that MIT-SHM is disabled. Instead, remove it only after
13315 X succeeds in attaching too.
13317 2008-07-17 02:30:24 +0000 David Schleef <ds@schleef.org>
13319 gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second.
13320 Original commit message from CVS:
13321 * gst/audiotestsrc/gstaudiotestsrc.c:
13322 * gst/audiotestsrc/gstaudiotestsrc.h:
13323 Add 'ticks', a 1/30 second sine wave pulse every second.
13325 2008-07-15 22:43:16 +0000 David Schleef <ds@schleef.org>
13327 gst-libs/gst/video/video.c: Revert ABI change.
13328 Original commit message from CVS:
13329 * gst-libs/gst/video/video.c: Revert ABI change.
13331 2008-07-15 13:05:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13333 gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
13334 Original commit message from CVS:
13335 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
13336 Make it impossible to have NULL caps at the point where we set
13337 framerate and other things. Also don't return immediately for "3ivd"
13338 video and let framerate, etc be set. Might fix bug #542508.
13340 2008-07-14 17:06:26 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
13342 gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
13343 Original commit message from CVS:
13344 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
13345 Video format can also be conveniently determined from (many)
13348 2008-07-14 08:18:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13350 gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
13351 Original commit message from CVS:
13352 * gst/playback/gstplaybasebin.c:
13353 * gst/playback/gstplaybasebin.h:
13354 * gst/playback/gstplaybin.c:
13355 * gst/playback/gststreamselector.c:
13356 First stab at integrating DVD subpicture overlay into
13357 playbin. Successfully plugs and plays, but the queues need
13358 shrinking - 3 seconds of video is too much buffering.
13360 2008-07-11 18:06:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13362 gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs.
13363 Original commit message from CVS:
13364 * gst/audioconvert/gstaudioconvert.c:
13365 Remove now obsolete note in the docs.
13367 2008-07-11 06:10:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13369 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
13370 Original commit message from CVS:
13371 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
13372 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
13373 * docs/plugins/gst-plugins-base-plugins-sections.txt:
13374 * docs/plugins/gst-plugins-base-plugins.args:
13375 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13376 * docs/plugins/gst-plugins-base-plugins.interfaces:
13377 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13378 * docs/plugins/gst-plugins-base-plugins.signals:
13379 * docs/plugins/inspect/plugin-adder.xml:
13380 * docs/plugins/inspect/plugin-alsa.xml:
13381 * docs/plugins/inspect/plugin-audioconvert.xml:
13382 * docs/plugins/inspect/plugin-audiorate.xml:
13383 * docs/plugins/inspect/plugin-audioresample.xml:
13384 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13385 * docs/plugins/inspect/plugin-cdparanoia.xml:
13386 * docs/plugins/inspect/plugin-decodebin.xml:
13387 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13388 * docs/plugins/inspect/plugin-gdp.xml:
13389 * docs/plugins/inspect/plugin-gnomevfs.xml:
13390 * docs/plugins/inspect/plugin-libvisual.xml:
13391 * docs/plugins/inspect/plugin-ogg.xml:
13392 * docs/plugins/inspect/plugin-pango.xml:
13393 * docs/plugins/inspect/plugin-playback.xml:
13394 * docs/plugins/inspect/plugin-queue2.xml:
13395 * docs/plugins/inspect/plugin-subparse.xml:
13396 * docs/plugins/inspect/plugin-tcp.xml:
13397 * docs/plugins/inspect/plugin-theora.xml:
13398 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13399 * docs/plugins/inspect/plugin-uridecodebin.xml:
13400 * docs/plugins/inspect/plugin-video4linux.xml:
13401 * docs/plugins/inspect/plugin-videorate.xml:
13402 * docs/plugins/inspect/plugin-videoscale.xml:
13403 * docs/plugins/inspect/plugin-videotestsrc.xml:
13404 * docs/plugins/inspect/plugin-volume.xml:
13405 * docs/plugins/inspect/plugin-vorbis.xml:
13406 * docs/plugins/inspect/plugin-ximagesink.xml:
13407 * docs/plugins/inspect/plugin-xvimagesink.xml:
13408 * ext/alsa/gstalsamixer.c:
13409 * ext/alsa/gstalsasink.c:
13410 * ext/alsa/gstalsasrc.c:
13411 * ext/gio/gstgiosink.c:
13412 * ext/gio/gstgiosrc.c:
13413 * ext/gio/gstgiostreamsink.c:
13414 * ext/gio/gstgiostreamsrc.c:
13415 * ext/gnomevfs/gstgnomevfssink.c:
13416 * ext/gnomevfs/gstgnomevfssrc.c:
13417 * ext/ogg/gstoggdemux.c:
13418 * ext/ogg/gstoggmux.c:
13419 * ext/pango/gstclockoverlay.c:
13420 * ext/pango/gsttextoverlay.c:
13421 * ext/pango/gsttextrender.c:
13422 * ext/pango/gsttimeoverlay.c:
13423 * ext/theora/theoradec.c:
13424 * ext/theora/theoraenc.c:
13425 * ext/theora/theoraparse.c:
13426 * ext/vorbis/vorbisdec.c:
13427 * ext/vorbis/vorbisenc.c:
13428 * ext/vorbis/vorbisparse.c:
13429 * ext/vorbis/vorbistag.c:
13430 * gst/adder/gstadder.c:
13431 * gst/audioconvert/gstaudioconvert.c:
13432 * gst/audioresample/gstaudioresample.c:
13433 * gst/audiotestsrc/gstaudiotestsrc.c:
13434 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
13435 * gst/gdp/gstgdpdepay.c:
13436 * gst/gdp/gstgdppay.c:
13437 * gst/playback/gstdecodebin2.c:
13438 * gst/playback/gstplaybin.c:
13439 * gst/playback/gstplaybin2.c:
13440 * gst/playback/gstqueue2.c:
13441 * gst/playback/gsturidecodebin.c:
13442 * gst/tcp/gstmultifdsink.c:
13443 * gst/tcp/gsttcpserversink.c:
13444 * gst/videorate/gstvideorate.c:
13445 * gst/videoscale/gstvideoscale.c:
13446 * gst/videotestsrc/gstvideotestsrc.c:
13447 * gst/volume/gstvolume.c:
13448 * sys/ximage/ximagesink.c:
13449 * sys/xvimage/xvimagesink.c:
13450 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
13451 titles. Drop mentining that all our example pipelines are "simple"
13454 2008-07-10 21:06:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13456 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
13457 Original commit message from CVS:
13458 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
13459 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
13460 * docs/plugins/gst-plugins-base-plugins-sections.txt:
13461 * docs/plugins/gst-plugins-base-plugins.args:
13462 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13463 * docs/plugins/gst-plugins-base-plugins.interfaces:
13464 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13465 * docs/plugins/gst-plugins-base-plugins.signals:
13466 * docs/plugins/inspect/plugin-adder.xml:
13467 * docs/plugins/inspect/plugin-alsa.xml:
13468 * docs/plugins/inspect/plugin-audioconvert.xml:
13469 * docs/plugins/inspect/plugin-audiorate.xml:
13470 * docs/plugins/inspect/plugin-audioresample.xml:
13471 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13472 * docs/plugins/inspect/plugin-cdparanoia.xml:
13473 * docs/plugins/inspect/plugin-decodebin.xml:
13474 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13475 * docs/plugins/inspect/plugin-gdp.xml:
13476 * docs/plugins/inspect/plugin-gnomevfs.xml:
13477 * docs/plugins/inspect/plugin-libvisual.xml:
13478 * docs/plugins/inspect/plugin-ogg.xml:
13479 * docs/plugins/inspect/plugin-pango.xml:
13480 * docs/plugins/inspect/plugin-playback.xml:
13481 * docs/plugins/inspect/plugin-queue2.xml:
13482 * docs/plugins/inspect/plugin-subparse.xml:
13483 * docs/plugins/inspect/plugin-tcp.xml:
13484 * docs/plugins/inspect/plugin-theora.xml:
13485 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13486 * docs/plugins/inspect/plugin-uridecodebin.xml:
13487 * docs/plugins/inspect/plugin-video4linux.xml:
13488 * docs/plugins/inspect/plugin-videorate.xml:
13489 * docs/plugins/inspect/plugin-videoscale.xml:
13490 * docs/plugins/inspect/plugin-videotestsrc.xml:
13491 * docs/plugins/inspect/plugin-volume.xml:
13492 * docs/plugins/inspect/plugin-vorbis.xml:
13493 * docs/plugins/inspect/plugin-ximagesink.xml:
13494 * docs/plugins/inspect/plugin-xvimagesink.xml:
13495 * ext/alsa/gstalsamixer.c:
13496 * ext/alsa/gstalsasink.c:
13497 * ext/alsa/gstalsasrc.c:
13498 * ext/gio/gstgiosink.c:
13499 * ext/gio/gstgiosrc.c:
13500 * ext/gio/gstgiostreamsink.c:
13501 * ext/gio/gstgiostreamsrc.c:
13502 * ext/gnomevfs/gstgnomevfssink.c:
13503 * ext/gnomevfs/gstgnomevfssrc.c:
13504 * ext/ogg/gstoggdemux.c:
13505 * ext/ogg/gstoggmux.c:
13506 * ext/pango/gstclockoverlay.c:
13507 * ext/pango/gsttextoverlay.c:
13508 * ext/pango/gsttextrender.c:
13509 * ext/pango/gsttimeoverlay.c:
13510 * ext/theora/theoradec.c:
13511 * ext/theora/theoraenc.c:
13512 * ext/theora/theoraparse.c:
13513 * ext/vorbis/vorbisdec.c:
13514 * ext/vorbis/vorbisenc.c:
13515 * ext/vorbis/vorbisparse.c:
13516 * ext/vorbis/vorbistag.c:
13517 * gst/adder/gstadder.c:
13518 * gst/audioconvert/gstaudioconvert.c:
13519 * gst/audioresample/gstaudioresample.c:
13520 * gst/audiotestsrc/gstaudiotestsrc.c:
13521 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
13522 * gst/gdp/gstgdpdepay.c:
13523 * gst/gdp/gstgdppay.c:
13524 * gst/playback/gstdecodebin2.c:
13525 * gst/playback/gstplaybin.c:
13526 * gst/playback/gstplaybin2.c:
13527 * gst/playback/gstqueue2.c:
13528 * gst/playback/gsturidecodebin.c:
13529 * gst/tcp/gstmultifdsink.c:
13530 * gst/tcp/gsttcpserversink.c:
13531 * gst/videorate/gstvideorate.c:
13532 * gst/videoscale/gstvideoscale.c:
13533 * gst/videotestsrc/gstvideotestsrc.c:
13534 * gst/volume/gstvolume.c:
13535 * sys/ximage/ximagesink.c:
13536 * sys/xvimage/xvimagesink.c:
13537 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
13538 titles. Drop mentining that all our example pipelines are "simple"
13541 2008-07-07 17:25:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13543 tests/examples/seek/Makefile.am: Fix out of tree build by adding all required CFLAGS.
13544 Original commit message from CVS:
13545 * tests/examples/seek/Makefile.am:
13546 Fix out of tree build by adding all required CFLAGS.
13548 2008-07-07 09:55:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13550 gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
13551 Original commit message from CVS:
13552 * gst/playback/gstdecodebin.c: (add_raw_queue):
13553 And ref the pad before returning it again when linking to the queue
13554 failed. Otherwise we will unref the pad twice later and things break.
13556 2008-07-07 09:48:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13558 gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
13559 Original commit message from CVS:
13560 * gst/playback/gstdecodebin.c: (add_raw_queue):
13561 If linking the raw pad with a queue fails, try it without a queue
13562 instead of failing completely. This should never happen.
13564 2008-07-06 23:22:12 +0000 Evgeniy Stepanov <eugeni.stepanov@gmail.com>
13566 gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
13567 Original commit message from CVS:
13568 Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
13569 * gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
13570 Add a queue after a demuxer if the demuxer outputs raw data. This was
13571 done before only for non-raw data but is required in this case too.
13573 decodebin2 doesn't have this issue because all streams of a group
13574 go through multiqueue.
13576 2008-07-03 09:12:49 +0000 Damien Lespiau <damien.lespiau@gmail.com>
13578 gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
13579 Original commit message from CVS:
13580 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
13581 * gst-libs/gst/sdp/gstsdpmessage.c:
13582 Makes libgstsdp compile with mingw32 by defining the right WINVER so
13583 that getaddrinfo() can be used. Fixes #541358.
13585 2008-07-01 13:22:49 +0000 Wim Taymans <wim.taymans@gmail.com>
13587 gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
13588 Original commit message from CVS:
13589 * gst/videotestsrc/gstvideotestsrc.c:
13590 (gst_video_test_src_class_init), (gst_video_test_src_init),
13591 (gst_video_test_src_set_property),
13592 (gst_video_test_src_get_property), (gst_video_test_src_create):
13593 * gst/videotestsrc/gstvideotestsrc.h:
13594 Cleanups, use default property values as defines.
13595 Add property to enable/disable peer buffer allocation.
13597 2008-06-30 09:46:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13599 tests/check/: Enable unit tests on PPC again as the bugs are now fixed.
13600 Original commit message from CVS:
13601 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
13602 * tests/check/pipelines/streamheader.c: (streamheader_suite):
13603 Enable unit tests on PPC again as the bugs are now fixed.
13605 2008-06-30 09:20:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13607 gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
13608 Original commit message from CVS:
13609 * gst-libs/gst/riff/riff-ids.h:
13610 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
13611 (gst_riff_create_audio_template_caps):
13612 Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
13615 2008-06-30 08:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13617 gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
13618 Original commit message from CVS:
13619 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
13620 (gst_ffmpeg_pixfmt_to_caps):
13621 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
13622 (gst_ffmpegcsp_get_unit_size):
13623 Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
13624 it on other formats. Also adjust the unit size only for that format
13625 to not include the palette. Fixes bug #540497.
13627 2008-06-29 13:45:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13629 gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
13630 Original commit message from CVS:
13631 * gst/adder/gstadder.c:
13632 Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
13634 2008-06-27 07:55:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13636 ChangeLog: ChangeLog surgery.
13637 Original commit message from CVS:
13640 * tests/examples/seek/seek.c:
13641 Move variable into ifdef too.
13643 2008-06-27 07:42:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13645 tests/examples/seek/seek.c: Include config.h and check if we have X. Fixes: #540334.
13646 Original commit message from CVS:
13647 * tests/examples/seek/seek.c:
13648 Include config.h and check if we have X. Fixes: #540334.
13650 2008-06-26 06:03:38 +0000 Sam Morris <sam@robots.org.to.uk>
13652 gst-libs/gst/interfaces/mixertrack.c: API: Add "index" property to GstMixerTrack to differantiate between multiple mi...
13653 Original commit message from CVS:
13654 Patch by: Sam Morris <sam at robots dot org to uk>
13655 * gst-libs/gst/interfaces/mixertrack.c:
13656 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
13657 (gst_mixer_track_set_property):
13658 API: Add "index" property to GstMixerTrack to differantiate between
13659 multiple mixer tracks with the same label.
13660 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
13661 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
13662 Set the "index" property of GstMixerTrack to the index given by ALSA.
13665 2008-06-25 13:15:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13667 tests/examples/seek/: Remove libgstvideo usage. Use gtk_get_option_group instead of gtk_init().
13668 Original commit message from CVS:
13669 * tests/examples/seek/Makefile.am:
13670 * tests/examples/seek/seek.c:
13671 Remove libgstvideo usage. Use gtk_get_option_group instead of
13674 2008-06-24 16:27:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13676 tests/check/Makefile.am: Name the test registry format neutral.
13677 Original commit message from CVS:
13678 * tests/check/Makefile.am:
13679 Name the test registry format neutral.
13681 2008-06-24 16:22:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13683 gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
13684 Original commit message from CVS:
13685 * gst/playback/gstqueue2.c:
13686 Do not double notify. Remove the unsued return value.
13688 2008-06-24 16:15:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13690 ext/alsa/gstalsamixer.c: Also consider "speaker" as a name for master volume. If that doesn't help look for the first...
13691 Original commit message from CVS:
13692 * ext/alsa/gstalsamixer.c:
13693 Also consider "speaker" as a name for master volume. If that doesn't
13694 help look for the first non-mono volume control that also has a
13697 2008-06-24 16:10:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13699 ChangeLog: Forgot to save the ChangeLog :/
13700 Original commit message from CVS:
13702 Forgot to save the ChangeLog :/
13704 2008-06-24 16:05:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13706 tests/examples/seek/: Embedd the xwindow.
13707 Original commit message from CVS:
13708 * tests/examples/seek/Makefile.am:
13709 * tests/examples/seek/seek.c:
13710 Embedd the xwindow.
13712 2008-06-24 01:14:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13714 sys/ximage/ximagesink.h: When the caps change, make sure to re-draw borders in force-aspect-ratio=true mode.
13715 Original commit message from CVS:
13716 * sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
13717 (gst_ximagesink_setcaps):
13718 * sys/ximage/ximagesink.h:
13719 When the caps change, make sure to re-draw borders in
13720 force-aspect-ratio=true mode.
13721 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
13722 Don't clear the border_draw flag until we actually draw the border.
13723 * tests/check/Makefile.am:
13724 Ignore alsasink/src during the states test too, so it doesn't fail
13725 when running without access to the sound device.
13727 2008-06-22 18:35:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13729 tests/examples/seek/seek.c: Fix crasher when playing a parse-launch line the 2nd time.
13730 Original commit message from CVS:
13731 * tests/examples/seek/seek.c:
13732 Fix crasher when playing a parse-launch line the 2nd time.
13734 2008-06-21 18:56:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
13736 tests/check/pipelines/oggmux.c: Properly ifdef tests to fix compilation.
13737 Original commit message from CVS:
13738 * tests/check/pipelines/oggmux.c:
13739 Properly ifdef tests to fix compilation.
13741 2008-06-21 10:25:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
13745 Original commit message from CVS:
13748 2008-06-20 18:24:24 +0000 Michael Smith <msmith@xiph.org>
13750 gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
13751 Original commit message from CVS:
13752 * gst/playback/gstplay-marshal.list:
13753 * gst/playback/gstplaybin2.c:
13754 Add get-video-pad, get-audio-pad, get-text-pad action signals to
13755 playbin2. This allows the user to get to the selector's sinkpads, and
13756 thus inspect a range of things - caps, tags, etc.
13758 2008-06-20 17:27:03 +0000 Michael Smith <msmith@xiph.org>
13760 gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
13761 Original commit message from CVS:
13762 * gst/playback/gstplaybin2.c:
13763 Use a different constant for the convert-frame signal id.
13766 2008-06-20 17:18:55 +0000 Michael Smith <msmith@xiph.org>
13768 gst/playback/: Fix a whole bunch of typos in comments and log statements.
13769 Original commit message from CVS:
13770 * gst/playback/gstplaybin2.c:
13771 * gst/playback/gstplaysink.c:
13772 Fix a whole bunch of typos in comments and log statements.
13774 2008-06-20 17:02:48 +0000 Michael Smith <msmith@xiph.org>
13776 sys/xvimage/xvimagesink.c: Don't set colour balance values on the Xv port if the user hasn't changed them (via proper...
13777 Original commit message from CVS:
13778 * sys/xvimage/xvimagesink.c:
13779 Don't set colour balance values on the Xv port if the user hasn't
13780 changed them (via properties or the interface). Avoids accumulating
13781 rounding errors for the common case.
13782 Partial fix for bug #537889.
13784 2008-06-20 16:56:18 +0000 Michael Smith <msmith@xiph.org>
13786 gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
13787 Original commit message from CVS:
13788 * gst/playback/gstdecodebin2.c:
13789 Ensure decodebin2 emits 'drained' signal once, and only once, when all
13792 2008-06-20 16:12:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
13795 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...
13796 Original commit message from CVS:
13797 apparently it's an error to specify nc -l -p 3000 - though the short usage
13798 does not make it very clear that you can drop the host arg with -l
13800 2008-06-20 09:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
13802 ext/vorbis/vorbisenc.c: Report the encoder latency. Fixes #538232.
13803 Original commit message from CVS:
13804 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
13805 (gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
13806 Report the encoder latency. Fixes #538232.
13808 2008-06-20 09:19:59 +0000 Wim Taymans <wim.taymans@gmail.com>
13810 gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
13811 Original commit message from CVS:
13812 * gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
13813 (notify_source), (activate_group):
13814 Implement the source property, emit notify when it changes in the
13815 underlying uridecodebin.
13817 2008-06-20 09:14:26 +0000 Wim Taymans <wim.taymans@gmail.com>
13819 tests/examples/seek/seek.c: Free and clear the seek element list so that we don't use invalid references when seeking...
13820 Original commit message from CVS:
13821 * tests/examples/seek/seek.c: (stop_cb):
13822 Free and clear the seek element list so that we don't use invalid
13823 references when seeking after recreating a gst-launch line.
13825 2008-06-20 09:09:37 +0000 Wim Taymans <wim.taymans@gmail.com>
13827 gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
13828 Original commit message from CVS:
13829 * gst-libs/gst/audio/gstbaseaudiosink.c:
13830 (gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
13831 (gst_base_audio_sink_render):
13832 Report latency even if we are not live instead of hiding it.
13833 Take ts-offset and render-delay of the basesink into account when
13834 scheduling samples.
13835 Rework the clipping code so that we can take the various offsets into
13836 account and still do correct clipping.
13838 2008-06-20 08:52:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13840 configure.ac: Bump verion back to devel -> 0.10.20.1
13841 Original commit message from CVS:
13843 Bump verion back to devel -> 0.10.20.1
13845 2008-06-20 08:47:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13847 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...
13848 Original commit message from CVS:
13849 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
13850 Don't increase the size of non-string image buffers by one as this
13851 might in theory confuse decoders. Still increase it by one for string
13852 image buffers to append '\0'.
13854 2008-06-20 08:45:13 +0000 Antoine Tremblay <hexa00@gmail.com>
13856 gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
13857 Original commit message from CVS:
13858 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
13859 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
13860 Fix a buffer memleak and remove a confusing and wrong debug output.
13863 2008-06-19 11:25:37 +0000 Wim Taymans <wim.taymans@gmail.com>
13865 examples/app/appsink-src.c: Don't use a buffer after unreffing it.
13866 Original commit message from CVS:
13867 * examples/app/appsink-src.c: (on_new_buffer_from_source):
13868 Don't use a buffer after unreffing it.
13870 === release 0.10.20 ===
13872 2008-06-18 14:36:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13878 * docs/plugins/gst-plugins-base-plugins.args:
13879 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13880 * docs/plugins/gst-plugins-base-plugins.interfaces:
13881 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13882 * docs/plugins/inspect/plugin-adder.xml:
13883 * docs/plugins/inspect/plugin-alsa.xml:
13884 * docs/plugins/inspect/plugin-audioconvert.xml:
13885 * docs/plugins/inspect/plugin-audiorate.xml:
13886 * docs/plugins/inspect/plugin-audioresample.xml:
13887 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13888 * docs/plugins/inspect/plugin-cdparanoia.xml:
13889 * docs/plugins/inspect/plugin-decodebin.xml:
13890 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13891 * docs/plugins/inspect/plugin-gdp.xml:
13892 * docs/plugins/inspect/plugin-gnomevfs.xml:
13893 * docs/plugins/inspect/plugin-libvisual.xml:
13894 * docs/plugins/inspect/plugin-ogg.xml:
13895 * docs/plugins/inspect/plugin-pango.xml:
13896 * docs/plugins/inspect/plugin-playback.xml:
13897 * docs/plugins/inspect/plugin-queue2.xml:
13898 * docs/plugins/inspect/plugin-subparse.xml:
13899 * docs/plugins/inspect/plugin-tcp.xml:
13900 * docs/plugins/inspect/plugin-theora.xml:
13901 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13902 * docs/plugins/inspect/plugin-uridecodebin.xml:
13903 * docs/plugins/inspect/plugin-video4linux.xml:
13904 * docs/plugins/inspect/plugin-videorate.xml:
13905 * docs/plugins/inspect/plugin-videoscale.xml:
13906 * docs/plugins/inspect/plugin-videotestsrc.xml:
13907 * docs/plugins/inspect/plugin-volume.xml:
13908 * docs/plugins/inspect/plugin-vorbis.xml:
13909 * docs/plugins/inspect/plugin-ximagesink.xml:
13910 * docs/plugins/inspect/plugin-xvimagesink.xml:
13911 * gst-plugins-base.doap:
13913 * win32/common/config.h:
13915 Original commit message from CVS:
13918 2008-06-18 14:32:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13947 Original commit message from CVS:
13950 2008-06-18 06:31:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13952 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
13953 Original commit message from CVS:
13954 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
13955 * examples/app/appsrc-ra.c:
13956 * examples/app/appsrc-seekable.c:
13957 * examples/app/appsrc-stream.c:
13958 * examples/app/appsrc-stream2.c:
13959 * ext/directfb/dfbvideosink.h:
13960 * ext/metadata/gstbasemetadata.c:
13961 * ext/metadata/gstbasemetadata.h:
13962 * ext/metadata/metadata.c:
13963 * ext/metadata/metadataexif.c:
13964 * ext/theora/theoradec.h:
13965 * gst/deinterlace2/gstdeinterlace2.h:
13966 * gst/deinterlace2/tvtime/speedy.c:
13967 * gst/deinterlace2/tvtime/speedy.h:
13968 * gst/deinterlace2/tvtime/vfir.c:
13969 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
13972 2008-06-16 14:11:36 +0000 Andy Wingo <wingo@pobox.com>
13974 * gst-libs/gst/app/gstappsrc.c:
13975 gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
13976 Original commit message from CVS:
13977 2008-06-16 Andy Wingo <wingo@pobox.com>
13978 * gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
13979 (gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
13980 G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
13982 2008-06-16 07:30:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13984 Final round of doc updates.
13985 Original commit message from CVS:
13986 * gst/rtpmanager/gstrtpjitterbuffer.c:
13987 * gst/speed/gstspeed.c:
13988 * gst/speexresample/gstspeexresample.c:
13989 * gst/videosignal/gstvideoanalyse.c:
13990 * gst/videosignal/gstvideodetect.c:
13991 * gst/videosignal/gstvideomark.c:
13992 * sys/dvb/gstdvbsrc.c:
13993 * sys/oss4/oss4-mixer.c:
13994 * sys/oss4/oss4-sink.c:
13995 * sys/oss4/oss4-source.c:
13996 * sys/wininet/gstwininetsrc.c:
13997 Final round of doc updates.
13999 2008-06-13 11:59:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14001 docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
14002 Original commit message from CVS:
14003 * docs/plugins/Makefile.am:
14004 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
14005 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
14006 * docs/plugins/gst-plugins-bad-plugins.args:
14007 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
14008 * docs/plugins/gst-plugins-bad-plugins.interfaces:
14009 * docs/plugins/gst-plugins-bad-plugins.prerequisites:
14010 * docs/plugins/gst-plugins-bad-plugins.signals:
14011 * docs/plugins/inspect/plugin-alsaspdif.xml:
14012 * docs/plugins/inspect/plugin-amrwb.xml:
14013 * docs/plugins/inspect/plugin-app.xml:
14014 * docs/plugins/inspect/plugin-bayer.xml:
14015 * docs/plugins/inspect/plugin-bz2.xml:
14016 * docs/plugins/inspect/plugin-cdaudio.xml:
14017 * docs/plugins/inspect/plugin-cdxaparse.xml:
14018 * docs/plugins/inspect/plugin-dtsdec.xml:
14019 * docs/plugins/inspect/plugin-dvb.xml:
14020 * docs/plugins/inspect/plugin-dvdspu.xml:
14021 * docs/plugins/inspect/plugin-faac.xml:
14022 * docs/plugins/inspect/plugin-faad.xml:
14023 * docs/plugins/inspect/plugin-fbdevsink.xml:
14024 * docs/plugins/inspect/plugin-festival.xml:
14025 * docs/plugins/inspect/plugin-filter.xml:
14026 * docs/plugins/inspect/plugin-flvdemux.xml:
14027 * docs/plugins/inspect/plugin-freeze.xml:
14028 * docs/plugins/inspect/plugin-gsm.xml:
14029 * docs/plugins/inspect/plugin-gstinterlace.xml:
14030 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
14031 * docs/plugins/inspect/plugin-h264parse.xml:
14032 * docs/plugins/inspect/plugin-interleave.xml:
14033 * docs/plugins/inspect/plugin-jack.xml:
14034 * docs/plugins/inspect/plugin-ladspa.xml:
14035 * docs/plugins/inspect/plugin-metadata.xml:
14036 * docs/plugins/inspect/plugin-mms.xml:
14037 * docs/plugins/inspect/plugin-modplug.xml:
14038 * docs/plugins/inspect/plugin-mpeg2enc.xml:
14039 * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
14040 * docs/plugins/inspect/plugin-mpegtsparse.xml:
14041 * docs/plugins/inspect/plugin-mpegvideoparse.xml:
14042 * docs/plugins/inspect/plugin-musepack.xml:
14043 * docs/plugins/inspect/plugin-musicbrainz.xml:
14044 * docs/plugins/inspect/plugin-mve.xml:
14045 * docs/plugins/inspect/plugin-mythtv.xml
14046 * docs/plugins/inspect/plugin-nas.xml:
14047 * docs/plugins/inspect/plugin-neon.xml:
14048 * docs/plugins/inspect/plugin-nsfdec.xml:
14049 * docs/plugins/inspect/plugin-nuvdemux.xml:
14050 * docs/plugins/inspect/plugin-oss4.xml
14051 * docs/plugins/inspect/plugin-rawparse.xml:
14052 * docs/plugins/inspect/plugin-real.xml:
14053 * docs/plugins/inspect/plugin-replaygain.xml:
14054 * docs/plugins/inspect/plugin-rfbsrc.xml:
14055 * docs/plugins/inspect/plugin-sdl.xml:
14056 * docs/plugins/inspect/plugin-sdp.xml:
14057 * docs/plugins/inspect/plugin-selector.xml:
14058 * docs/plugins/inspect/plugin-sndfile.xml:
14059 * docs/plugins/inspect/plugin-soundtouch.xml:
14060 * docs/plugins/inspect/plugin-spcdec.xml:
14061 * docs/plugins/inspect/plugin-speed.xml:
14062 * docs/plugins/inspect/plugin-speexresample.xml:
14063 * docs/plugins/inspect/plugin-stereo.xml:
14064 * docs/plugins/inspect/plugin-subenc.xml
14065 * docs/plugins/inspect/plugin-timidity.xml:
14066 * docs/plugins/inspect/plugin-tta.xml:
14067 * docs/plugins/inspect/plugin-vcdsrc.xml:
14068 * docs/plugins/inspect/plugin-videosignal.xml:
14069 * docs/plugins/inspect/plugin-vmnc.xml:
14070 * docs/plugins/inspect/plugin-wildmidi.xml:
14071 * docs/plugins/inspect/plugin-x264.xml:
14072 * docs/plugins/inspect/plugin-xvid.xml:
14073 * docs/plugins/inspect/plugin-y4menc.xml:
14074 * ext/amrwb/gstamrwbdec.c:
14075 * ext/amrwb/gstamrwbenc.c:
14076 * ext/amrwb/gstamrwbparse.c:
14077 * ext/dc1394/gstdc1394.c:
14078 * ext/directfb/dfbvideosink.c:
14079 * ext/ivorbis/vorbisdec.c:
14080 * ext/jack/gstjackaudiosink.c:
14081 * ext/mpeg2enc/gstmpeg2enc.cc:
14082 * ext/mplex/gstmplex.cc:
14083 * ext/musicbrainz/gsttrm.c:
14084 * ext/mythtv/gstmythtvsrc.c:
14085 * ext/theora/theoradec.c:
14086 * ext/timidity/gsttimidity.c:
14087 * ext/timidity/gstwildmidi.c:
14088 * gst-libs/gst/app/gstappsink.c:
14089 * gst/deinterlace/gstdeinterlace.c:
14090 * gst/dvdspu/gstdvdspu.c:
14091 * gst/festival/gstfestival.c:
14092 * gst/freeze/gstfreeze.c:
14093 * gst/interleave/deinterleave.c:
14094 * gst/interleave/interleave.c:
14095 * gst/modplug/gstmodplug.cc:
14096 * gst/nuvdemux/gstnuvdemux.c:
14097 Add missing elements to docs. Fix doc-markup: use convinience syntax
14098 for examples (produces valid docbook), add several refsec2 when we
14099 have several titles. Fix some types.
14101 2008-06-12 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
14103 examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
14104 Original commit message from CVS:
14105 * examples/app/.cvsignore:
14106 * examples/app/Makefile.am:
14107 * examples/app/appsink-src.c: (on_new_buffer_from_source),
14108 (on_source_message), (on_sink_message), (main):
14109 Add beefed up example app from bug #413418. It now also uses appsink
14110 instead of fakesink for more ultimate coolness.
14111 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
14112 (gst_app_src_init), (gst_app_src_set_property),
14113 (gst_app_src_get_property), (gst_app_src_unlock),
14114 (gst_app_src_unlock_stop), (gst_app_src_create),
14115 (gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
14116 (gst_app_src_end_of_stream):
14117 * gst-libs/gst/app/gstappsrc.h:
14118 Add block property to allow push based implementation to block when we
14119 fill up the appsrc queues.
14120 Emit the enough-data signal while releasing our lock.
14122 2008-06-12 14:50:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14124 examples/app/.cvsignore: Ignore more.
14125 Original commit message from CVS:
14126 * examples/app/.cvsignore:
14129 2008-06-12 14:49:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14131 Do not use short_description in section docs for elements. We extract them from element details and there will be war...
14132 Original commit message from CVS:
14133 * ext/dc1394/gstdc1394.c:
14134 * ext/ivorbis/vorbisdec.c:
14135 * ext/jack/gstjackaudiosink.c:
14136 * ext/metadata/gstmetadatademux.c:
14137 * ext/mythtv/gstmythtvsrc.c:
14138 * ext/theora/theoradec.c:
14139 * gst-libs/gst/app/gstappsink.c:
14140 * gst/bayer/gstbayer2rgb.c:
14141 * gst/deinterlace/gstdeinterlace.c:
14142 * gst/rawparse/gstaudioparse.c:
14143 * gst/rawparse/gstvideoparse.c:
14144 * gst/rtpmanager/gstrtpbin.c:
14145 * gst/rtpmanager/gstrtpclient.c:
14146 * gst/rtpmanager/gstrtpjitterbuffer.c:
14147 * gst/rtpmanager/gstrtpptdemux.c:
14148 * gst/rtpmanager/gstrtpsession.c:
14149 * gst/rtpmanager/gstrtpssrcdemux.c:
14150 * gst/selector/gstinputselector.c:
14151 * gst/selector/gstoutputselector.c:
14152 * gst/videosignal/gstvideoanalyse.c:
14153 * gst/videosignal/gstvideodetect.c:
14154 * gst/videosignal/gstvideomark.c:
14155 * sys/oss4/oss4-mixer.c:
14156 * sys/oss4/oss4-sink.c:
14157 * sys/oss4/oss4-source.c:
14158 Do not use short_description in section docs for elements. We extract
14159 them from element details and there will be warnings if they differ.
14160 Also fixing up the ChangeLog order.
14162 2008-06-11 21:17:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14164 configure.ac: 0.10.19.3 pre-release
14165 Original commit message from CVS:
14167 0.10.19.3 pre-release
14169 2008-06-11 20:13:00 +0000 David Schleef <ds@schleef.org>
14171 gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
14172 Original commit message from CVS:
14173 * gst-libs/gst/rtsp/gstrtspconnection.c:
14174 Fix build on win32.
14175 Patch By: David Schleef <ds@schleef.org>
14178 2008-06-11 09:35:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14180 ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
14181 Original commit message from CVS:
14182 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
14183 (gst_gio_base_src_create):
14184 * ext/gio/gstgiobasesrc.h:
14185 Try to read the requested number of bytes, even if the first
14186 read returns less than requested, until nothing is read anymore
14187 or we have the requested amount of bytes. This fixes playback of
14188 files via Samba as Samba only allows to read 64k at once.
14189 Implement a caching algorithm that makes sure that we read at
14190 least 4k of data every time. Some elements will try to read a few
14191 bytes, then seek, read again a few bytes and so on and this is
14192 painfully slow as every operation has to go over DBus if GVfs is
14194 Fixes bug #536849 and #536848.
14195 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
14196 (gst_gio_src_check_get_range):
14197 Override check_get_range() to blacklist http/https URIs
14198 and whitelist file URIs. More to be added on demand.
14200 2008-06-06 16:50:51 +0000 Wim Taymans <wim.taymans@gmail.com>
14202 examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
14203 Original commit message from CVS:
14204 * examples/app/Makefile.am:
14205 * examples/app/appsrc-ra.c: (feed_data), (seek_data),
14206 (found_source), (bus_message), (main):
14207 * examples/app/appsrc-seekable.c: (feed_data), (seek_data),
14208 (found_source), (bus_message), (main):
14209 * examples/app/appsrc-stream2.c: (feed_data), (found_source),
14210 (bus_message), (main):
14211 Added 3 more example application for using appsrc in random-access mode,
14212 pull-mode streaming and pull mode seekable.
14213 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
14214 (gst_app_src_start), (gst_app_src_do_get_size),
14215 (gst_app_src_create):
14216 * gst-libs/gst/app/gstappsrc.h:
14217 Make stream-type property writable.
14218 Unset flushing when starting so that we reuse appsrc.
14219 Inform basesrc about the configured size.
14220 Emit seek-data signal when we are going to a different offset in
14221 random-access mode.
14223 2008-06-06 14:19:54 +0000 Wim Taymans <wim.taymans@gmail.com>
14225 examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
14226 Original commit message from CVS:
14227 * examples/app/appsrc-stream.c: (found_source), (main):
14228 Use deep-notify until we can depend on a playbin2 with support for the
14231 2008-06-05 16:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
14233 examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
14234 Original commit message from CVS:
14235 * examples/app/.cvsignore:
14236 * examples/app/Makefile.am:
14237 * examples/app/appsrc-stream.c: (read_data), (start_feed),
14238 (stop_feed), (found_source), (bus_message), (main):
14239 Added an example on how to use appsrc in playbin in streaming mode from
14241 * examples/app/appsrc_ex.c: (main):
14242 Set pipeline to NULL to free queued buffers.
14243 * gst-libs/gst/app/gstapp-marshal.list:
14244 * gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
14245 (gst_app_src_class_init), (gst_app_src_init),
14246 (gst_app_src_flush_queued), (gst_app_src_dispose),
14247 (gst_app_src_set_property), (gst_app_src_get_property),
14248 (gst_app_src_unlock), (gst_app_src_unlock_stop),
14249 (gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
14250 (gst_app_src_check_get_range), (gst_app_src_do_seek),
14251 (gst_app_src_create), (gst_app_src_set_stream_type),
14252 (gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
14253 (gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
14254 (gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
14255 (gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
14256 (gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
14257 * gst-libs/gst/app/gstappsrc.h:
14258 Measure max queue size in bytes instead.
14259 Add support for 3 modes of operation, streaming, seekable and
14260 random-access, making basesrc handle the scheduling modes for each.
14261 Add appsrc:// uri handler so that automatic plugging can be done from
14262 playbin2 or uridecodebin, for example.
14263 Added support for custom segment formats.
14264 Add support for push and pull based operations from the application.
14265 Expand the methods so that errors can be detected.
14266 Flush the queued buffers on seeks and when shutting down.
14267 Add signals to inform the app that a seek must happen.
14269 2008-06-05 09:47:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14271 configure.ac: 0.10.19.2 pre-release
14272 Original commit message from CVS:
14274 0.10.19.2 pre-release
14276 2008-06-04 21:48:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14278 win32/common/: Add new API functions to the dll exports
14279 Original commit message from CVS:
14280 * win32/common/libgstrtsp.def:
14281 * win32/common/libgsttag.def:
14282 Add new API functions to the dll exports
14284 2008-06-04 17:42:38 +0000 Michael Smith <msmith@xiph.org>
14286 gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
14287 Original commit message from CVS:
14288 * gst/playback/gstplaybasebin.c:
14289 Disconnect signals from decodebins we created before we remove it from
14290 playbin, to avoid crashes if the decodebin is eventually disposed after
14291 the playbin itself (possible if the app takes a reference on the
14295 2008-06-04 17:12:40 +0000 Tim-Philipp Müller <tim@centricular.net>
14297 gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
14298 Original commit message from CVS:
14299 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
14300 (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
14301 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
14302 (h264_video_type_find), (mpeg_video_stream_type_find),
14303 (dv_type_find), (mmsh_type_find):
14304 Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
14305 copy caps for no good reason (this may be desirable to make it easier
14306 to detect leaks, but then it should probably be done for all caps
14307 in the typefinder somewhere).
14309 2008-06-04 16:06:49 +0000 Peter Kjellerstedt <pkj@axis.com>
14311 tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
14312 Original commit message from CVS:
14313 * tests/check/Makefile.am:
14314 Do not try to run the check tests for subparse unless it has been
14317 2008-06-04 16:00:26 +0000 Peter Kjellerstedt <pkj@axis.com>
14319 tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
14320 Original commit message from CVS:
14321 * tests/check/pipelines/streamheader.c: (buffer_probe_cb),
14322 (test_multifdsink_gdp_vorbisenc), (streamheader_suite):
14323 Do not try to run a test which requires vorbisenc unless we have
14326 2008-06-04 11:53:53 +0000 Peter Kjellerstedt <pkj@axis.com>
14328 gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
14329 Original commit message from CVS:
14330 * gst-libs/gst/rtsp/gstrtspconnection.c:
14331 (gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
14332 (gst_rtsp_connection_clear_auth_params),
14333 (gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
14334 * gst-libs/gst/rtsp/gstrtspconnection.h:
14335 Add a couple of missing argument guards.
14336 Add a way of setting the DSCP for an RTSP connection.
14337 Add an accessor method for the ip member of GstRTSPConnection as all
14338 members are supposed to be private.
14340 2008-06-04 11:33:23 +0000 Peter Kjellerstedt <pkj@axis.com>
14342 gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
14343 Original commit message from CVS:
14344 * gst/tcp/gstmultifdsink.c: (setup_dscp_client):
14345 Fixed accidental use of IPv4 options for all IPv6 addresses.
14347 2008-06-04 10:18:42 +0000 Tim-Philipp Müller <tim@centricular.net>
14349 gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
14350 Original commit message from CVS:
14351 * gst-libs/gst/interfaces/mixertrack.h:
14352 Document mixer track flags.
14354 2008-06-04 05:58:38 +0000 Antoine Tremblay <hexa00@gmail.com>
14356 gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
14357 Original commit message from CVS:
14358 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
14359 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
14360 Don't set caps on the buffers that contain a copy of the buffer
14361 including the caps of them resulting in an always increasing refcount
14362 of the caps and insanely large caps. Instead include a buffer without
14363 caps in the new caps. Fixes bug #536475.
14365 2008-06-04 05:44:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14367 gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
14368 Original commit message from CVS:
14369 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
14370 Transform a given PAR to a range on the struct with the generic
14371 height/width instead of the struct with the possibly restricted
14374 2008-06-04 04:24:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14376 gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
14377 Original commit message from CVS:
14378 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
14379 Prefer the given format if it contains something stricter than [1,MAX]
14380 for height or width and only put a structure that requires rescaling
14381 as second. This makes it possible to use videoscale in pipelines where
14382 the source can actually produce the wanted height/width but usually
14383 selects a different one from the requested.
14385 2008-06-03 20:01:58 +0000 John Millikin <jmillikin@gmail.com>
14387 gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
14388 Original commit message from CVS:
14389 Based on patch by: John Millikin <jmillikin gmail com>
14390 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
14391 (gst_vorbis_tag_add_coverart):
14392 Retrieve COVERART tags from vorbis comments (#512333)
14394 2008-06-03 19:44:48 +0000 Tim-Philipp Müller <tim@centricular.net>
14396 gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
14397 Original commit message from CVS:
14398 * gst-libs/gst/tag/tag.h:
14399 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
14400 Don't forget to add new enum value here too (should probably use
14401 glib-mkenums here...).
14403 2008-06-03 19:29:06 +0000 Tim-Philipp Müller <tim@centricular.net>
14405 gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
14406 Original commit message from CVS:
14407 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
14408 * gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
14409 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
14410 (gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
14411 (gst_tag_image_data_to_image_buffer):
14412 Add two utility functions to avoid code duplication (#512333):
14413 API: add gst_tag_image_data_to_image_buffer()
14414 API: add gst_tag_list_add_id3_image()
14416 2008-06-03 08:54:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14418 win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
14419 Original commit message from CVS:
14420 * win32/common/libgstaudio.def:
14421 Add gst_audio_check_channel_positions() to the exported symbols.
14423 2008-06-03 08:48:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14425 API: Make gst_audio_check_channel_positions() public.
14426 Original commit message from CVS:
14427 * docs/libs/gst-plugins-base-libs-sections.txt:
14428 * gst-libs/gst/audio/multichannel.c:
14429 (gst_audio_check_channel_positions):
14430 * gst-libs/gst/audio/multichannel.h:
14431 API: Make gst_audio_check_channel_positions() public.
14432 * tests/check/libs/audio.c: (GST_START_TEST):
14433 Add some simple checks for gst_audio_check_channel_positions().
14435 2008-06-02 20:09:14 +0000 Tim-Philipp Müller <tim@centricular.net>
14437 sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.
14438 Original commit message from CVS:
14439 * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
14440 minrange and maxrange are scaled according to the frequency
14443 2008-06-02 18:37:02 +0000 Tim-Philipp Müller <tim@centricular.net>
14445 ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
14446 Original commit message from CVS:
14447 * ext/pango/Makefile.am:
14448 * ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
14449 (gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
14450 Use gstvideo functions to calculate strides and plane offsets. Fixes
14451 rendering issue ('ghost' images of the text on the chroma planes)
14452 with widths or heights that are not multiples of 8 (#506659 and
14453 probably also #485729).
14454 * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
14456 Test with odd height/width too.
14458 2008-06-02 12:20:35 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14460 gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
14461 Original commit message from CVS:
14462 * gst/adder/gstadder.c: (gst_adder_query_duration),
14463 (gst_adder_query_latency):
14464 When using gst_element_iterate_pads() one has to unref every pad
14467 2008-05-31 19:57:57 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
14469 gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
14470 Original commit message from CVS:
14471 * gst-libs/gst/audio/gstbaseaudiosrc.c:
14472 (gst_base_audio_src_class_init):
14473 Add a gtk-doc chunk for the new properties to have a Since: indication.
14475 2008-05-31 19:50:59 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
14478 ChangeLog surgery, mark API change
14479 Original commit message from CVS:
14480 ChangeLog surgery, mark API change
14482 2008-05-31 18:10:47 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
14484 gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
14485 Original commit message from CVS:
14486 * gst-libs/gst/audio/gstbaseaudiosrc.c:
14487 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
14488 (gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
14489 (gst_base_audio_src_change_state):
14490 Provide readable actual-buffer-time and actual-latency-time properties
14491 that reflect the configured ringbuffer values. Fixes #524724.
14493 2008-05-30 15:29:20 +0000 Wim Taymans <wim.taymans@gmail.com>
14495 gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
14496 Original commit message from CVS:
14497 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
14498 (gst_basertppayload_change_state):
14499 Simply converting the running time into an RTP timestamp by scaling it
14500 based on the clock-rate is good enough for making an RTP timestamp. This
14501 has the added benefit that we can later on expose a property with the
14502 RTP timestamp of running time 0, as is needed for RTSP servers to
14503 generate the response of the PLAY request.
14505 2008-05-30 08:42:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14507 gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
14508 Original commit message from CVS:
14509 * gst/audioconvert/gstaudioconvert.c:
14510 (structure_has_fixed_channel_positions),
14511 (gst_audio_convert_transform_caps):
14512 Allow up to 11 positioned channels now that audioconvert can handle
14513 this but add no default positions for > 8 channels.
14514 * tests/check/elements/audioconvert.c: (GST_START_TEST):
14515 Add some unit tests for the above change: Test conversion of
14516 11 positioned channels to stereo and the other way around, test
14517 conversion of 15 unpositioned channels in different ways.
14519 2008-05-29 19:45:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14521 win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
14522 Original commit message from CVS:
14523 * win32/common/libgstaudio.def:
14524 Add gst_audio_clock_reset to the list of exported symbols.
14526 2008-05-29 19:37:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14528 tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
14529 Original commit message from CVS:
14530 * tests/check/elements/vorbisdec.c: (vorbisdec_suite):
14531 Remove wrong_channels_identification_header unit test as we now
14532 support 7 (and more channels).
14534 2008-05-29 12:17:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14536 gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
14537 Original commit message from CVS:
14538 * gst/audioconvert/gstchannelmix.c:
14539 (gst_channel_mix_fill_one_other):
14540 If mixing left or right to center (or the other way around) only take
14541 the complete value if we don't already have the original position in
14544 2008-05-29 11:34:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14546 gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
14547 Original commit message from CVS:
14548 * gst-libs/gst/audio/multichannel.c:
14549 (gst_audio_check_channel_positions),
14550 (gst_audio_set_structure_channel_positions_list),
14551 (gst_audio_fixate_channel_positions):
14552 Allow rear center together with rear left/right and other previously
14553 conflicting channel positions. The reason why they weren't allowed
14554 was the channel mixing implementation in audioconvert.
14555 Also take this into account when fixing channel layouts.
14556 Allow setting channel positions for 1/2 channels when using
14557 gst_audio_set_structure_channel_position().
14558 * gst/audioconvert/gstchannelmix.c:
14559 (gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
14560 (gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
14561 (gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
14562 Major rewrite of the channel mixing.
14563 We now allow previously conflicting channel positions to appear
14564 together (rear center and rear left/right for example).
14566 Rework the way channels are mixed together to take more possible
14567 channel positions into account, properly mix from/to side channels
14568 and don't assume that either center, left&right or nothing of a
14569 specific position is available anymore.
14570 * tests/check/elements/audioconvert.c: (GST_START_TEST):
14571 Adjust unit tests with non-standard 1/2 channel layouts to the more
14572 correct new behaviour.
14573 Add a unit test for 5.1->Stereo downmixing.
14575 2008-05-29 07:02:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14577 ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
14578 Original commit message from CVS:
14579 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
14580 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
14581 Add sane defaults for the 7 and 8 channel layouts as those are
14582 undefined in the Vorbis spec. Use NONE channel layouts when decoding
14583 more than 8 channels instead of erroring out. Fixes bug #535356.
14585 2008-05-28 16:10:20 +0000 Wim Taymans <wim.taymans@gmail.com>
14587 Add theoraparse to the docs and fix some docs.
14588 Original commit message from CVS:
14589 * docs/plugins/Makefile.am:
14590 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
14591 * docs/plugins/gst-plugins-base-plugins-sections.txt:
14592 * ext/theora/theoraparse.c:
14593 Add theoraparse to the docs and fix some docs.
14595 2008-05-28 15:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
14597 gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
14598 Original commit message from CVS:
14599 * gst-libs/gst/cdda/gstcddabasesrc.c:
14600 (gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
14601 Fix EOS condition and track addition check, the track.end sector is
14602 included in the track. Fixes #533265.
14604 2008-05-28 14:49:24 +0000 Mark Nauwelaerts <manauw@skynet.be>
14606 gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
14607 Original commit message from CVS:
14608 Patch by: Mark Nauwelaerts <manauw at skynet be>
14609 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
14610 (gst_video_rate_flush_prev), (gst_video_rate_event),
14611 (gst_video_rate_chain):
14612 * gst/videorate/gstvideorate.h:
14613 React (more) to NEWSEGMENT
14614 Small adjustment in timestamp calculation to prevent mismatches
14617 2008-05-28 11:31:44 +0000 Tim-Philipp Müller <tim@centricular.net>
14619 tests/examples/seek/seek.c: Initialise error to NULL as we should.
14620 Original commit message from CVS:
14621 * tests/examples/seek/seek.c: (make_parselaunch_pipeline):
14622 Initialise error to NULL as we should.
14624 2008-05-28 08:14:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14626 gst/adder/gstadder.c: Implement latency query.
14627 Original commit message from CVS:
14628 * gst/adder/gstadder.c: (gst_adder_query_duration),
14629 (gst_adder_query_latency), (gst_adder_query):
14630 Implement latency query.
14632 2008-05-27 18:10:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14634 gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
14635 Original commit message from CVS:
14636 * gst/adder/gstadder.c: (gst_adder_query_duration):
14637 Correctly resync the iterator if gst_iterator_next() returns
14638 GST_ITERATOR_RESYNC.
14640 2008-05-27 17:14:07 +0000 Tim-Philipp Müller <tim@centricular.net>
14642 win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
14643 Original commit message from CVS:
14644 * win32/vs6/libgstpbutils.dsp:
14645 Add pbutils-enumtypes.c to sources (#518037).
14647 2008-05-27 16:20:17 +0000 Wim Taymans <wim.taymans@gmail.com>
14649 gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
14650 Original commit message from CVS:
14651 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
14652 (gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
14653 * gst-libs/gst/audio/gstaudioclock.h:
14654 Add method to inform the clock that the time starts from 0 again. We use
14655 this info to calculate a clock offset so that the time we report in
14656 internal_time is monotonically increasing, as required by the clock base
14657 class. Fixes #521761.
14658 API: GstAudioClock::gst_audio_clock_reset()
14659 * gst-libs/gst/audio/gstbaseaudiosink.c:
14660 (gst_base_audio_sink_skew_slaving),
14661 (gst_base_audio_sink_change_state):
14662 * gst-libs/gst/audio/gstbaseaudiosrc.c:
14663 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
14664 Reset reported time when we (re)create the ringbuffer.
14666 2008-05-27 16:11:32 +0000 Tim-Philipp Müller <tim@centricular.net>
14668 ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
14669 Original commit message from CVS:
14670 * ext/alsa/gstalsamixertrack.c:
14671 (gst_alsa_mixer_track_update_alsa_capabilities):
14672 Make sure playback volumes aren't accidentally overwritten by
14673 capture volumes if an alsa mixer track has both playback and
14674 capture capabilities: we create two GstMixerTracks in that
14675 case, so make sure we query only the alsa capabilities that
14676 refer to the type of GstMixerTrack we created from the dual
14677 capability alsa element. Should fix issues with Audigy2 sound
14680 2008-05-27 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.net>
14682 tests/check/pipelines/oggmux.c: Don't use deprecated function.
14683 Original commit message from CVS:
14684 * tests/check/pipelines/oggmux.c: (test_pipeline):
14685 Don't use deprecated function.
14687 2008-05-27 10:35:55 +0000 Wim Taymans <wim.taymans@gmail.com>
14689 gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
14690 Original commit message from CVS:
14691 * gst/playback/gstdecodebin2.c:
14692 (gst_decode_group_control_source_pad), (gst_decode_group_expose):
14693 Check for NULL cases and log them, creating ghostpads can, for example,
14694 fail when the pad returns wrong caps.
14695 * gst/playback/gstplaybin2.c: (perform_eos):
14696 When pushing out the EOS event, collect the return value and warn when
14699 2008-05-26 17:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
14701 gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
14702 Original commit message from CVS:
14703 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
14704 (gst_riff_create_video_template_caps):
14705 Add support for DVCPRO.
14707 2008-05-26 10:29:20 +0000 Tim-Philipp Müller <tim@centricular.net>
14709 gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
14710 Original commit message from CVS:
14711 * gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
14712 Change default scaling method from nearest-neighbour to bilinear.
14714 2008-05-26 10:26:00 +0000 Tim-Philipp Müller <tim@centricular.net>
14716 tests/check/libs/video.c: More checks.
14717 Original commit message from CVS:
14718 * tests/check/libs/video.c:
14721 2008-05-25 20:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
14723 Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
14724 Original commit message from CVS:
14725 * gst/subparse/gstsubparse.c: (parser_state_init),
14726 (gst_sub_parse_format_autodetect), (handle_buffer):
14727 * gst/subparse/gstsubparse.h:
14728 * tests/check/elements/subparse.c: (test_tmplayer_style3b):
14729 Limit duration to a maximum of five seconds for tmplayer format where
14730 we can guess the duration only from the timestamp of the next line of
14731 text. We don't want to show a text for eternities just because nothing
14732 else is being said for a while.
14734 2008-05-23 14:14:28 +0000 Wim Taymans <wim.taymans@gmail.com>
14736 gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
14737 Original commit message from CVS:
14738 * gst-libs/gst/rtp/gstbasertpdepayload.c:
14739 (gst_base_rtp_depayload_chain),
14740 (gst_base_rtp_depayload_handle_sink_event),
14741 (gst_base_rtp_depayload_push_full),
14742 (gst_base_rtp_depayload_change_state):
14743 Check sequence numbers, mark input buffers with a discont flag for the
14744 subclass when we detected a gap, drop duplicate buffers. We do this
14745 because one can use the element without a jitterbuffer in front and we
14746 don't want to feed the subclasses invalid or reordered data.
14747 Do an error when the subclass did not provide a process function instead
14749 Some other small cleanups.
14751 2008-05-22 22:35:40 +0000 Tim-Philipp Müller <tim@centricular.net>
14753 gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
14754 Original commit message from CVS:
14755 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
14756 May just as well use the precalculated uvstride here.
14758 2008-05-22 22:09:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14760 Add some documentation comments, and some new headers to be scanned.
14761 Original commit message from CVS:
14762 * docs/plugins/Makefile.am:
14763 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
14764 * docs/plugins/gst-plugins-base-plugins-sections.txt:
14765 * docs/plugins/gst-plugins-base-plugins.args:
14766 * docs/plugins/gst-plugins-base-plugins.hierarchy:
14767 * docs/plugins/gst-plugins-base-plugins.interfaces:
14768 * docs/plugins/gst-plugins-base-plugins.prerequisites:
14769 * docs/plugins/inspect/plugin-adder.xml:
14770 * docs/plugins/inspect/plugin-alsa.xml:
14771 * docs/plugins/inspect/plugin-audioconvert.xml:
14772 * docs/plugins/inspect/plugin-audiorate.xml:
14773 * docs/plugins/inspect/plugin-audioresample.xml:
14774 * docs/plugins/inspect/plugin-audiotestsrc.xml:
14775 * docs/plugins/inspect/plugin-cdparanoia.xml:
14776 * docs/plugins/inspect/plugin-decodebin.xml:
14777 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
14778 * docs/plugins/inspect/plugin-gdp.xml:
14779 * docs/plugins/inspect/plugin-gio.xml:
14780 * docs/plugins/inspect/plugin-gnomevfs.xml:
14781 * docs/plugins/inspect/plugin-libvisual.xml:
14782 * docs/plugins/inspect/plugin-ogg.xml:
14783 * docs/plugins/inspect/plugin-pango.xml:
14784 * docs/plugins/inspect/plugin-playback.xml:
14785 * docs/plugins/inspect/plugin-queue2.xml:
14786 * docs/plugins/inspect/plugin-subparse.xml:
14787 * docs/plugins/inspect/plugin-tcp.xml:
14788 * docs/plugins/inspect/plugin-theora.xml:
14789 * docs/plugins/inspect/plugin-typefindfunctions.xml:
14790 * docs/plugins/inspect/plugin-uridecodebin.xml:
14791 * docs/plugins/inspect/plugin-video4linux.xml:
14792 * docs/plugins/inspect/plugin-videorate.xml:
14793 * docs/plugins/inspect/plugin-videoscale.xml:
14794 * docs/plugins/inspect/plugin-videotestsrc.xml:
14795 * docs/plugins/inspect/plugin-volume.xml:
14796 * docs/plugins/inspect/plugin-vorbis.xml:
14797 * docs/plugins/inspect/plugin-ximagesink.xml:
14798 * docs/plugins/inspect/plugin-xvimagesink.xml:
14799 * ext/cdparanoia/gstcdparanoiasrc.c:
14800 * ext/ogg/gstoggdemux.c:
14801 * ext/ogg/gstoggdemux.h:
14802 * ext/ogg/gstoggmux.c:
14803 * ext/ogg/gstoggmux.h:
14804 * gst/audioconvert/audioconvert.c:
14805 * gst/audioconvert/audioconvert.h:
14806 * gst/audioconvert/gstaudioconvert.h:
14807 * gst/gdp/gstgdpdepay.h:
14808 * gst/gdp/gstgdppay.h:
14809 * gst/playback/gstdecodebin.c:
14810 * gst/playback/gstdecodebin2.c:
14811 * gst/playback/gstplaybin.c:
14812 * gst/playback/gstplaybin2.c:
14813 * gst/playback/gsturidecodebin.c:
14814 * gst/tcp/gstmultifdsink.c:
14815 * gst/tcp/gstmultifdsink.h:
14816 * gst/tcp/gsttcp.h:
14817 Add some documentation comments, and some new headers to be scanned.
14818 Rename some internal enum declarations (audioconvert's DitherType and
14819 NoiseShapingType, GstUnitType from the TCP elements) to match the
14820 documented GObject type names so that the docs pick them up.
14821 Name the playbin2 docs markups properly so they get picked up. They'll
14822 need renaming back when/if playbin2 becomes playbin.
14823 100% symbol coverage for the plugin docs, booya.
14825 2008-05-22 18:30:15 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
14827 gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
14828 Original commit message from CVS:
14829 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
14830 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
14831 Fix generation of NV12/NV21 frames. Fixes bug #532454.
14833 2008-05-22 11:59:33 +0000 Sjoerd Simons <sjoerd@luon.net>
14835 gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
14836 Original commit message from CVS:
14837 Patch by: Sjoerd Simons <sjoerd at luon dot net>
14838 * gst/playback/gstdecodebin.c: (remove_fakesink):
14839 Lock the fakesink before setting the state to NULL and removing it from
14840 the bin so that a concurrent state change cannot interfere.
14843 2008-05-21 17:09:42 +0000 Felipe Contreras <felipe.contreras@nokia.com>
14845 docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
14846 Original commit message from CVS:
14847 * docs/Makefile.am:
14848 Fix installing plugin documentation when gtk-doc is disabled.
14850 2008-05-21 17:01:16 +0000 Felipe Contreras <felipe.contreras@nokia.com>
14852 gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
14853 Original commit message from CVS:
14854 * gst-libs/gst/rtsp/Makefile.am:
14855 Distribute, don't install md5.h
14857 2008-05-21 16:47:58 +0000 Julien Moutte <julien@moutte.net>
14859 gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
14860 Original commit message from CVS:
14861 2008-05-21 Julien Moutte <julien@fluendo.com>
14862 * gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
14863 instead of SOL_IP, works on more platforms.
14864 * gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
14867 2008-05-21 16:44:15 +0000 Wim Taymans <wim.taymans@gmail.com>
14869 Some debug and comment fixes.
14870 Original commit message from CVS:
14871 * ext/vorbis/vorbisdec.c:
14872 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
14873 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
14874 Some debug and comment fixes.
14875 * tests/examples/dynamic/addstream.c: (main):
14878 2008-05-21 16:36:50 +0000 Wim Taymans <wim.taymans@gmail.com>
14880 Don't use bad gst_element_get_pad().
14881 Original commit message from CVS:
14882 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
14883 * gst/playback/decodetest.c: (new_decoded_pad_cb):
14884 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
14885 (try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
14886 (cleanup_decodebin):
14887 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
14888 (connect_element), (gst_decode_group_control_demuxer_pad):
14889 * gst/playback/gstplaybasebin.c: (queue_remove_probe),
14890 (queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
14892 * gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
14893 (gst_play_bin_set_property), (handoff), (gen_video_element),
14894 (gen_text_element), (gen_audio_element), (gen_vis_element),
14895 (remove_sinks), (add_sink), (setup_sinks):
14896 * gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
14897 * gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
14898 (gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
14899 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
14900 (gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
14901 (gen_video_chain), (gen_text_chain), (gen_audio_chain),
14902 (gen_vis_chain), (gst_play_sink_reconfigure),
14903 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
14904 (gst_play_sink_request_pad):
14905 * gst/playback/gsturidecodebin.c: (type_found), (setup_source):
14906 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
14908 * gst/playback/test6.c: (new_decoded_pad_cb):
14909 * tests/check/elements/audioconvert.c: (GST_START_TEST):
14910 * tests/check/elements/audiorate.c: (test_injector_chain),
14911 (do_perfect_stream_test):
14912 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
14913 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
14914 * tests/check/elements/gnomevfssink.c:
14915 * tests/check/elements/textoverlay.c:
14916 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
14917 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
14918 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
14919 * tests/check/pipelines/oggmux.c: (test_pipeline):
14920 * tests/check/pipelines/streamheader.c: (GST_START_TEST):
14921 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
14922 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
14923 * tests/examples/seek/scrubby.c: (make_wav_pipeline):
14924 * tests/examples/seek/seek.c: (make_mod_pipeline),
14925 (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
14926 (make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
14927 (make_theora_pipeline), (make_vorbis_theora_pipeline),
14928 (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
14929 (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
14930 (update_fill), (msg_buffering):
14931 Don't use bad gst_element_get_pad().
14933 2008-05-21 14:35:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14935 gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
14936 Original commit message from CVS:
14937 * gst-libs/gst/riff/riff-media.c:
14938 Fix wrong method name in docs. Fix calculation of strf fields for
14940 * gst-libs/gst/riff/riff-read.c:
14941 Whitespace fix and removing double ';'.
14943 2008-05-21 11:52:30 +0000 Wim Taymans <wim.taymans@gmail.com>
14945 docs/design/part-playbin2.txt: Add some leftover doc.
14946 Original commit message from CVS:
14947 * docs/design/part-playbin2.txt:
14948 Add some leftover doc.
14950 2008-05-21 11:36:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14952 gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
14953 Original commit message from CVS:
14954 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
14955 Fix copy & paste error in last commit.
14957 2008-05-21 11:30:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14959 gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
14960 Original commit message from CVS:
14961 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
14962 Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
14963 other channel positions when source has SIDE channels and dest doesn't
14964 or the other way around.
14966 2008-05-21 11:29:25 +0000 Henrik Eriksson <henriken@axis.com>
14968 gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
14969 Original commit message from CVS:
14970 Patch by: Henrik Eriksson <henriken at axis dot com>
14971 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
14972 (gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
14973 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
14974 (gst_multi_fd_sink_get_property):
14975 * gst/tcp/gstmultifdsink.h:
14976 Add support for DSCP QOS. Fixes #469933.
14978 2008-05-21 07:46:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14980 tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
14981 Original commit message from CVS:
14982 * tests/check/elements/audioconvert.c: (GST_START_TEST):
14983 Add another test that checks if conversion between standard 1 and 2
14984 channel layouts with and without positions set is working.
14986 2008-05-21 07:39:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14988 gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
14989 Original commit message from CVS:
14990 * gst-libs/gst/audio/multichannel.c:
14991 (gst_audio_check_channel_positions):
14992 Allow non-standard 2 channel layouts.
14993 * tests/check/elements/audioconvert.c: (GST_START_TEST):
14994 Add some tests for converting and remapping non-standard 1 and 2
14997 2008-05-21 07:28:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14999 gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
15000 Original commit message from CVS:
15001 * gst/audioconvert/gstchannelmix.c:
15002 (gst_channel_mix_fill_normalize):
15003 Prevent division by zero if the channel mix matrix contains only
15006 2008-05-21 06:45:22 +0000 Antoine Tremblay <hexa00@gmail.com>
15008 gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
15009 Original commit message from CVS:
15010 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
15011 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
15012 Close a buffer memory leak. Fixes bug #534071.
15014 2008-05-21 06:39:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15016 gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
15017 Original commit message from CVS:
15018 * gst-libs/gst/rtsp/gstrtsptransport.h:
15019 Make the GstRTSPTransport struct members public as there are no
15020 setters/getters and it's supposed to be changed directly.
15023 2008-05-21 05:48:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15025 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...
15026 Original commit message from CVS:
15027 * gst/adder/gstadder.c:
15028 Adder also doesn't support audio/x-raw-int with width!=depth so don't
15029 claim this on the pad template caps.
15031 2008-05-20 16:26:53 +0000 Wim Taymans <wim.taymans@gmail.com>
15033 gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
15034 Original commit message from CVS:
15035 * gst-libs/gst/audio/gstbaseaudiosink.c:
15036 (gst_base_audio_sink_sync_latency):
15037 We can only use our optimal calibration if we prerolled before the
15040 2008-05-20 14:35:42 +0000 Tim-Philipp Müller <tim@centricular.net>
15042 configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
15043 Original commit message from CVS:
15045 Require core CVS for GstBaseSrc buffer caps setting magic.
15047 2008-05-20 12:26:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15049 gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
15050 Original commit message from CVS:
15051 * gst/audioconvert/gstaudioconvert.c:
15052 (gst_audio_convert_fixate_channels):
15053 Fix logic in last commit.
15055 2008-05-20 12:15:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15057 gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
15058 Original commit message from CVS:
15059 * gst/audioconvert/gstaudioconvert.c:
15060 (gst_audio_convert_fixate_channels):
15061 Passthrough the channel positions if the number of output channels is
15062 the same as the number of input channels, the input had a channel
15063 layout and downstream requests no special one. We did this already for
15064 > 2 channels but now it's also done for 1 channel. Fixes bug #533617.
15066 2008-05-20 11:13:27 +0000 Wim Taymans <wim.taymans@gmail.com>
15068 ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
15069 Original commit message from CVS:
15070 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
15071 (gst_gnome_vfs_src_finalize),
15072 (gst_gnome_vfs_src_received_headers_callback),
15073 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
15074 * ext/gnomevfs/gstgnomevfssrc.h:
15075 Set the ICY caps on the srcpad from where they get picked up by the base
15076 class now and set on the outgoing buffers.
15077 * gst-libs/gst/audio/gstbaseaudiosrc.c:
15078 (gst_base_audio_src_create):
15079 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
15080 BaseSrc now sets the caps on outgoing buffers automatically.
15082 2008-05-20 11:09:06 +0000 Wim Taymans <wim.taymans@gmail.com>
15084 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
15085 Original commit message from CVS:
15086 * gst-libs/gst/audio/gstbaseaudiosink.c:
15087 (gst_base_audio_sink_resample_slaving),
15088 (gst_base_audio_sink_skew_slaving),
15089 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
15090 (gst_base_audio_sink_async_play),
15091 (gst_base_audio_sink_change_state):
15092 Change the way in which the ringbuffer is started when dealing with a
15093 slaved clock and latency. We now sync to the clock until we reach
15094 upstream latency before starting the ringbuffer. This has the effect
15095 that we can accurately align the master and slave clocks and let the
15096 rate correction code take care of the initial drift or rounding errors
15097 instead of leaving them uncorrected with the old approach.
15099 2008-05-20 08:12:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15101 gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
15102 Original commit message from CVS:
15103 * gst/audioconvert/gstaudioconvert.c:
15104 (gst_audio_convert_fixate_channels):
15105 Correctly set the default channel positions when converting to 8
15108 2008-05-19 16:13:25 +0000 Tim-Philipp Müller <tim@centricular.net>
15110 configure.ac: Error out if we don't have the required version of core.
15111 Original commit message from CVS:
15113 Error out if we don't have the required version of core.
15115 2008-05-19 15:59:40 +0000 Tim-Philipp Müller <tim@centricular.net>
15117 gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
15118 Original commit message from CVS:
15119 * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
15120 Use data scan helper in aac typefinder and stop scanning
15121 for headers when we've found a type. Also fix potential invalid
15122 memory access when calculating the frame length.
15124 2008-05-19 14:09:08 +0000 Tim-Philipp Müller <tim@centricular.net>
15126 gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
15127 Original commit message from CVS:
15128 * gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
15129 (mpeg_sys_is_valid_pack):
15130 Don't modify scan context when we return FALSE in ensure_data, so
15131 it's possible to continue scanning, and we don't end up with a NULL
15132 data pointer and a positive size, which might bite us the next time
15133 we're called. Small constification.
15135 2008-05-16 21:12:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15137 gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
15138 Original commit message from CVS:
15139 * gst/adder/gstadder.c:
15140 Adder doesn't support 24 bit samples so don't claim it supports them
15141 in the pad template caps.
15143 2008-05-14 20:28:02 +0000 Wim Taymans <wim.taymans@gmail.com>
15145 gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
15146 Original commit message from CVS:
15147 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15148 (gst_base_rtp_depayload_chain):
15149 Validate the RTP packet before further processing it. It's just too
15150 dangerous to accept random packets and people are not forced to use a
15151 jitterbuffer or session manager to filter out the bad packets.
15152 * gst-libs/gst/rtp/gstrtpbuffer.c:
15153 (gst_rtp_buffer_set_extension_data),
15154 (gst_rtp_buffer_get_payload_subbuffer):
15156 When setting extension data in a buffer that is too small, we fail and
15157 we should not set the extension bit.
15158 Change GST_WARNINGS into g_warning because they really are
15159 programming errors.
15160 * tests/check/libs/rtp.c: (GST_START_TEST):
15161 Catch the g_warnings now in the unit tests and that fact that failing to
15162 set extension data left the extension bit untouched.
15164 2008-05-14 13:57:41 +0000 Tim-Philipp Müller <tim@centricular.net>
15166 gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
15167 Original commit message from CVS:
15168 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
15169 Revert previous change which made basetransform handle buffer_alloc
15170 and which breaks things badly in the non-passthrough case since it
15171 returned buffers with a different (ie. sometimes smaller) size than
15172 the size requested.
15174 2008-05-14 13:43:12 +0000 Bernard B <b-gnome@largestprime.net>
15176 gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
15177 Original commit message from CVS:
15178 Patch by: Bernard B <b-gnome at largestprime dot net>
15179 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
15180 Fix seqnum compare function for bordercase values and fix the docs
15181 again. Fixes #533075.
15182 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
15183 Add a testcase for seqnum compare function.
15185 2008-05-14 10:58:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15187 gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
15188 Original commit message from CVS:
15189 * gst/adder/gstadder.c: (gst_adder_setcaps),
15190 (gst_adder_class_init):
15191 Correctly declare the supported endianness on the pad templates
15192 and check for correct endianness in the set caps function. Adder
15193 only supports native endianness.
15194 Also use gst_element_class_set_details_simple().
15196 2008-05-14 09:12:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15198 sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.
15199 Original commit message from CVS:
15200 * sys/xvimage/xvimagesink.c:
15201 Better debug logging in port value handling. Merging separate port
15202 value loops into one.
15204 2008-05-13 16:02:19 +0000 Hannes Bistry <hannesb@gmx.de>
15206 gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
15207 Original commit message from CVS:
15208 Patch by: Hannes Bistry <hannesb at gmx dot de>
15209 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
15210 * gst/tcp/gsttcpserversink.c:
15211 (gst_tcp_server_sink_handle_server_read),
15212 (gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
15213 Fix regression in clientsrc because we did not add the fd to the poll
15214 set anymore. Fixes #532364.
15215 Do some cleanups here and there.
15217 2008-05-13 13:04:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15219 gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
15220 Original commit message from CVS:
15221 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
15222 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
15223 * gst/playback/gstplay-marshal.list:
15224 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
15225 Use correct marshallers. GstCaps are a boxed type and no GObject
15228 2008-05-13 11:37:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15230 win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.
15231 Original commit message from CVS:
15232 * win32/common/libgstrtsp.def:
15233 Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
15236 2008-05-13 10:59:49 +0000 Sjoerd Simons <sjoerd@luon.net>
15238 tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
15239 Original commit message from CVS:
15240 Patch by: Sjoerd Simons <sjoerd at luon dot net>
15241 * tests/check/elements/audioresample.c:
15242 (live_switch_alloc_only_48000), (live_switch_get_sink_caps),
15243 (live_switch_push), (GST_START_TEST):
15244 Add unit test for the latest basetransform negotiation changes.
15247 2008-05-13 09:14:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15249 gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
15250 Original commit message from CVS:
15251 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
15252 Fix nv12<->nv21 conversion if stride is larger than width.
15254 2008-05-13 07:28:21 +0000 j^ <j@oil21.org>
15256 ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...
15257 Original commit message from CVS:
15258 Patch by: j^ <j at oil21 dot org>
15259 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
15260 (gst_ogg_pad_parse_skeleton_fisbone):
15261 * ext/ogg/gstoggdemux.h:
15262 Parse presentation time from skeleton streams and use it as offset
15263 for the timestamps. Fixes bug #530068.
15265 2008-05-12 08:45:11 +0000 Wim Taymans <wim.taymans@gmail.com>
15267 gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
15268 Original commit message from CVS:
15269 * gst-libs/gst/audio/gstbaseaudiosink.c:
15270 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
15271 Revert previous patch that attempted to more accurately calculate the
15272 initial offset between master and slave clock. The best thing we can do
15273 in general is take the time of both clocks as the diff since we don't
15274 know when the actual preroll happened.
15276 2008-05-11 19:52:59 +0000 Tim-Philipp Müller <tim@centricular.net>
15278 gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
15279 Original commit message from CVS:
15280 * gst-libs/gst/pbutils/install-plugins.c:
15281 Fix docs: type and missing word.
15283 2008-05-10 20:16:21 +0000 Tim-Philipp Müller <tim@centricular.net>
15285 gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
15286 Original commit message from CVS:
15287 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
15288 Don't do lots of 4-byte peeks, but use the 'new' data scan helper
15289 for this instead; don't check if we've found enough markers after
15290 each and every step, it's enough to do that only if we've actually
15291 found a new marker.
15292 Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
15294 2008-05-10 18:19:17 +0000 Tim-Philipp Müller <tim@centricular.net>
15296 gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
15297 Original commit message from CVS:
15298 * gst/typefind/gsttypefindfunctions.c:
15299 (DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
15300 (data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
15301 (mpeg_video_stream_type_find):
15302 Move scan helper thingy to the beginning of the file so we can use
15303 it in other typefind functions. Rename it to something more
15304 generic. Also improve handling of things towards the end of the
15305 typefind data: peek as much as we can if we know the size of the
15306 data, rather than just min_size.
15308 2008-05-09 21:42:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15310 Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
15311 Original commit message from CVS:
15312 * docs/libs/gst-plugins-base-libs-sections.txt:
15313 * gst-libs/gst/interfaces/colorbalance.c:
15314 * gst-libs/gst/interfaces/colorbalance.h:
15315 * gst-libs/gst/interfaces/colorbalancechannel.c:
15316 * gst-libs/gst/interfaces/colorbalancechannel.h:
15317 * gst-libs/gst/interfaces/tuner.c:
15318 * gst-libs/gst/interfaces/tunerchannel.c:
15319 * gst-libs/gst/interfaces/tunerchannel.h:
15320 * gst-libs/gst/interfaces/tunernorm.c:
15321 * gst-libs/gst/interfaces/tunernorm.h:
15322 * gst-libs/gst/video/video.c:
15323 * gst-libs/gst/video/video.h:
15324 Document the GstTuner and GstColorBalance interfaces, and some
15325 other random API functions that needed it. 70% symbol coverage, woo.
15327 2008-05-09 16:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
15329 gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
15330 Original commit message from CVS:
15331 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
15332 Choose to allocate one less segment but require one additional segment
15334 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
15335 No need to increment the number of segments in the source.
15336 * gst-libs/gst/audio/gstbaseaudiosink.c:
15337 (gst_base_audio_sink_get_time), (clock_convert_external),
15338 (gst_base_audio_sink_resample_slaving),
15339 (gst_base_audio_sink_skew_slaving),
15340 (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
15341 (gst_base_audio_sink_async_play):
15342 Remove adding latency when returning the internal time while subtracting
15343 it again when we use the value a little later.
15344 When calculating the end timestamp, we are making a rounding error
15345 with the current algorithm. Ensure that we don't accumulate these
15346 rounding errors when aligning samples by not resampling at all if we
15347 don't need to. Fixes #419351.
15348 Make the initial calibration of the clock slaving a little more
15349 predictable and accurate. Also handle the case where we don't do
15352 2008-05-09 08:34:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15354 gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
15355 Original commit message from CVS:
15356 Based on a patch by:
15357 Björn Benderius <bjoern dot benderius at axis dot com>
15358 * gst/ffmpegcolorspace/avcodec.h:
15359 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
15360 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
15361 (gst_ffmpegcsp_avpicture_fill):
15362 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
15363 * gst/ffmpegcolorspace/imgconvert_template.h:
15364 Add conversions from/to NV12 and NV21 and conversions between those
15365 two formats. Fixes bug #532166.
15367 2008-05-08 17:35:44 +0000 Edward Hervey <bilboed@bilboed.com>
15369 gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
15370 Original commit message from CVS:
15371 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
15372 Abort the h264 typefinding as soon as _peek() doesn't return anything,
15373 which happens for example with files smaller than 128kb.
15375 2008-05-08 14:46:27 +0000 Wouter Cloetens <zombie@e2big.org>
15377 gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
15378 Original commit message from CVS:
15379 Patch by: Wouter Cloetens <zombie at e2big dot org>
15380 * gst-libs/gst/rtsp/Makefile.am:
15381 * gst-libs/gst/rtsp/gstrtspconnection.c:
15382 (gst_rtsp_connection_create), (md5_digest_to_hex_string),
15383 (auth_digest_compute_hex_urp), (auth_digest_compute_response),
15384 (add_auth_header), (gst_rtsp_connection_free),
15385 (gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
15386 (gst_rtsp_connection_set_auth_param),
15387 (gst_rtsp_connection_clear_auth_params):
15388 * gst-libs/gst/rtsp/gstrtspconnection.h:
15389 Add Digest authorization support for RTSP connections. See #532065.
15390 * gst-libs/gst/rtsp/md5.c:
15391 * gst-libs/gst/rtsp/md5.h:
15392 Yeap, another md5 implementation until we can depend on a glib that has
15395 2008-05-08 06:20:42 +0000 Sjoerd Simons <sjoerd@luon.net>
15397 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
15398 Original commit message from CVS:
15399 Patch by: Sjoerd Simons <sjoerd at luon dot net>
15400 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
15401 Let audioresample use the buffer allocation of basetransform instead
15403 * tests/check/elements/audioresample.c: (alloc_only_48000),
15404 (GST_START_TEST), (audioresample_suite):
15405 Add unit test for the recent basetransform bugfix, where upstream
15406 changes caps to something that can't be passed through anymore.
15408 2008-05-07 19:50:27 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15410 win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
15411 Original commit message from CVS:
15412 * win32/common/config.h.in:
15413 Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
15414 use the real thing than having "???" unconditionally.
15416 2008-05-07 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
15418 gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
15419 Original commit message from CVS:
15420 * gst-libs/gst/audio/gstbaseaudiosink.c:
15421 (gst_base_audio_sink_query):
15422 Report the latency with the new seglatency parameter.
15423 * gst-libs/gst/audio/gstringbuffer.c:
15424 (gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
15425 (gst_ring_buffer_acquire):
15426 * gst-libs/gst/audio/gstringbuffer.h:
15427 Add new field to the ringbufferspec to specify the expected latency
15428 between the underlying device read/write pointer, this is needed
15429 when writing sinks that sit a little closer to the hardware.
15430 Add some more docs for other fields.
15432 2008-05-07 10:38:23 +0000 Wim Taymans <wim.taymans@gmail.com>
15434 gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
15435 Original commit message from CVS:
15436 * gst-libs/gst/app/.cvsignore:
15437 * gst-libs/gst/app/Makefile.am:
15438 * gst-libs/gst/app/gstapp-marshal.list:
15439 Add marshal.list, make it compile and add to cvsignore.
15440 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
15441 (gst_app_sink_stop):
15443 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
15444 (gst_app_src_init), (gst_app_src_set_property),
15445 (gst_app_src_get_property), (gst_app_src_unlock),
15446 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
15447 (gst_app_src_create), (gst_app_src_set_caps),
15448 (gst_app_src_get_caps), (gst_app_src_set_size),
15449 (gst_app_src_get_size), (gst_app_src_set_seekable),
15450 (gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
15451 (gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
15452 (gst_app_src_end_of_stream):
15453 * gst-libs/gst/app/gstappsrc.h:
15454 Beat appsrc in shape, add signals and actions.
15456 Add properties for caps, size, seekability and max-buffers.
15457 Fix unlock/stop code.
15459 2008-05-06 12:35:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15461 gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
15462 Original commit message from CVS:
15463 * gst/volume/gstvolume.c: (volume_transform_ip):
15464 Return NOT_NEGOTIATED if we didn't set a process function yet for some
15465 reason instead of crashing later. Might fix bug #509125.
15467 2008-05-06 12:12:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
15469 gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
15470 Original commit message from CVS:
15471 Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
15472 * gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
15473 * gst/audioconvert/audioconvert.h:
15474 * gst/audioconvert/gstaudioconvert.c:
15475 (gst_audio_convert_parse_caps),
15476 (structure_has_fixed_channel_positions),
15477 (gst_audio_convert_transform_caps):
15478 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
15479 Add support for more than 8 channels and NONE channel layouts. For
15480 more than 8 channels no channel conversion is supported yet, only
15481 format conversions are supported. Fixes bug #398033.
15482 * tests/check/elements/audioconvert.c: (verify_convert),
15483 (GST_START_TEST), (audioconvert_suite):
15484 Add some unit tests by Tim for checking the NONE channel layouts
15485 and more than 8 channels and add some more unit tests for channel
15488 2008-05-06 10:16:49 +0000 Wim Taymans <wim.taymans@gmail.com>
15490 gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
15491 Original commit message from CVS:
15492 * gst/playback/gstdecodebin2.c: (connect_pad):
15493 When autoplugging fails, set the element back to NULL before
15496 2008-05-06 09:59:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15498 win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols.
15499 Original commit message from CVS:
15500 * win32/common/libgstaudio.def:
15501 Add gst_base_audio_src_[sg]et_slave_method() to the exported
15504 2008-05-05 12:33:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15506 gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
15507 Original commit message from CVS:
15508 * gst/subparse/samiparse.c: (handle_start_sync),
15509 (end_sami_element), (characters_sami):
15510 Remove trailing, leading and double whitespaces.
15511 Correctly timestamp buffers and output the last buffer too.
15512 * tests/check/elements/subparse.c: (GST_START_TEST),
15514 Add a simple unit test for SAMI parsing.
15516 2008-05-05 11:14:48 +0000 Young-Ho Cha <ganadist@chollian.net>
15518 gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
15519 Original commit message from CVS:
15520 Patch by: Young-Ho Cha <ganadist at chollian dot net>
15521 * gst/subparse/samiparse.c: (handle_start_sync),
15522 (start_sami_element), (end_sami_element), (characters_sami),
15523 (sami_context_reset):
15524 Only output characters inside the "sync" elements. There could be
15525 other elements like "style" that have some content but should
15526 not be printed. Fixes bug #467911.
15528 2008-05-05 10:27:45 +0000 Wim Taymans <wim.taymans@gmail.com>
15530 gst-libs/gst/app/gstappsink.*: Start some docs.
15531 Original commit message from CVS:
15532 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
15533 (gst_app_sink_init), (gst_app_sink_set_property),
15534 (gst_app_sink_get_property), (gst_app_sink_unlock_start),
15535 (gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
15536 (gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
15537 (gst_app_sink_preroll), (gst_app_sink_render),
15538 (gst_app_sink_set_caps), (gst_app_sink_set_drop),
15539 (gst_app_sink_get_drop):
15540 * gst-libs/gst/app/gstappsink.h:
15542 Add property to drop buffers when the queue is filled
15543 Fix unlocking and flushing when the queues are filled.
15545 2008-05-05 10:03:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15547 gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
15548 Original commit message from CVS:
15549 * gst/playback/gstplaybasebin.c: (set_audio_mute),
15550 (set_active_source):
15551 * gst/playback/gstplaybasebin.h:
15552 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
15553 (playbin_set_audio_mute):
15554 Allow setting -1 as current-audio to mute the current audio stream,
15555 similar to what is done for subtitles. Fixes bug #342294.
15557 2008-05-05 07:41:03 +0000 Edward Hervey <bilboed@bilboed.com>
15559 gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
15560 Original commit message from CVS:
15561 * gst-libs/gst/pbutils/descriptions.c: (formats):
15562 It's SorensOn and not SorensEn.
15564 2008-05-04 15:23:36 +0000 Tim-Philipp Müller <tim@centricular.net>
15566 gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
15567 Original commit message from CVS:
15568 * gst-libs/gst/pbutils/descriptions.c: (formats):
15569 Fix description of video/x-flash-video.
15571 2008-05-04 15:02:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15573 Remove some unused code.
15574 Original commit message from CVS:
15575 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
15576 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
15577 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
15578 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
15579 Remove some unused code.
15580 * gst/audioconvert/gstaudioquantize.c:
15581 (gst_audio_quantize_free_noise_shaping):
15582 Don't return before freeing the noise shaping history.
15584 2008-05-03 16:00:04 +0000 Tim-Philipp Müller <tim@centricular.net>
15586 tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
15587 Original commit message from CVS:
15588 * tests/check/elements/subparse.c: (do_test),
15589 (test_tmplayer_style3b), (subparse_suite):
15590 Add unit test for the tmplayer variant from bug #530962.
15592 2008-05-03 15:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
15594 gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
15595 Original commit message from CVS:
15596 * gst/subparse/gstsubparse.c: (handle_buffer),
15597 (gst_sub_parse_sink_event):
15598 * gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
15599 (tmplayer_parse_line):
15600 Fix parsing of tmplayer subtitle variant where every single line contains
15601 text and there isn't an empty line after each line to determine the
15602 duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
15603 making sure that we push out the last line of text without a duration if
15604 there's still text left in the buffer at the end.
15606 2008-05-03 15:39:04 +0000 Tim-Philipp Müller <tim@centricular.net>
15608 gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
15609 Original commit message from CVS:
15610 * gst/subparse/gstsubparse.c: (feed_textbuf):
15611 Fix detection of discontinuities based on the buffer offset (doesn't work
15612 so well if no buffer offset is set) and also check for the DISCONT buffer
15613 flag. This keeps the parser state from being reset after each buffer in
15616 2008-05-03 12:09:16 +0000 Tim-Philipp Müller <tim@centricular.net>
15618 gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
15619 Original commit message from CVS:
15620 * gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
15621 Further fine-tuning: don't absolutely require sequence or GOP headers
15622 (as introduced in the previous commit), but adjust the typefind
15623 probabilities returned accordingly if we don't see them. Also make sure
15624 picture header and first slice are somewhat close to each other (which
15625 is not perfect but still better than requiring a fixed offset or having
15628 2008-05-02 12:13:08 +0000 Wim Taymans <wim.taymans@gmail.com>
15630 gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
15631 Original commit message from CVS:
15632 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
15633 (gst_basertppayload_sink_setcaps),
15634 (gst_basertppayload_sink_getcaps):
15635 Rename the setcaps/getcaps function internally to make it clear that
15636 they are called for the sink pad.
15638 2008-05-02 12:11:07 +0000 Wim Taymans <wim.taymans@gmail.com>
15640 gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
15641 Original commit message from CVS:
15642 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15643 (gst_base_rtp_depayload_class_init),
15644 (gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
15645 (gst_base_rtp_depayload_packet_lost),
15646 (gst_base_rtp_depayload_set_gst_timestamp):
15647 * gst-libs/gst/rtp/gstbasertpdepayload.h:
15648 Catch packet-lost events from the jitterbuffer and convert them into a
15649 vmethod call (lost-packet) so that depayloaders can do something smart.
15650 Also add a default packet-lost function that sends out a segment update
15653 2008-05-02 11:13:05 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15655 gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
15656 Original commit message from CVS:
15657 * gst/playback/test4.c:
15658 * gst/playback/test5.c:
15659 * gst/playback/test6.c:
15660 * gst/playback/test7.c:
15661 Also include config.h when relying on defines from it. Fixes the
15662 build. Its been a please to serve :)
15664 2008-05-02 10:54:51 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
15667 * gst/videotestsrc/videotestsrc.c:
15668 Add support for NV12 and NV21 in videotestsrc
15669 Original commit message from CVS:
15670 * gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
15671 (paint_setup_NV21), (paint_hline_NV12_NV21):
15672 Add support for NV12 and NV21 in videotestsrc
15674 2008-05-02 10:02:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15676 gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
15677 Original commit message from CVS:
15678 * gst/videoscale/gstvideoscale.c:
15679 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
15680 * gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
15681 (vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
15682 (vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
15683 (vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
15684 (vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
15685 (vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
15686 (vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
15687 (vs_image_scale_linear_RGB555):
15688 Support 1x1 images as input and output as for example the BBC HQ new
15689 streams have 1x1 GIFs in the playlists for some reason.
15691 2008-05-01 19:11:42 +0000 Tim-Philipp Müller <tim@centricular.net>
15693 gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
15694 Original commit message from CVS:
15695 * gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
15697 If we can't activate one of the decoders we plugged in (such as,
15698 say, musepackdec) for some reason (it might not support push mode,
15699 for example), remove any pad probes that close_pad_link() might
15700 have set up. This makes sure we later don't try to remove a probe
15701 for a pad that doesn't exist any longer, and avoids nast warnings
15702 and probably other things too.
15704 2008-04-30 20:54:56 +0000 Tim-Philipp Müller <tim@centricular.net>
15706 gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
15707 Original commit message from CVS:
15708 * gst/typefind/gsttypefindfunctions.c:
15709 (mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
15711 Rework mpeg video stream typefinding a bit more: make sure sequence,
15712 GOP, picture and slice headers appear in the order they should and
15713 that we've in fact at least had one of each; fix picture header
15714 detection; decouple picture and slice header check - don't assume
15715 they're at a fixed offset, there may be extra data in between. Also,
15716 announce varying degrees of probability depending on what we found
15717 exactly (multiple pictures, at least one picture, just sequence and
15718 GOP headers). Finally, in _ensure_data(), take into account that we
15719 might be typefinding smaller amounts of data, such as the first
15720 buffer of a stream, so fall back to the minimum size needed as long
15721 as that's available, instead of erroring out if there's less than
15722 2kB of data. Fixes #526173. Conveniently also doesn't recognise the
15723 fuzzed file from #399342 as valid.
15725 2008-04-30 17:06:45 +0000 Michael Smith <msmith@xiph.org>
15727 ext/theora/theoradec.c: Cool kids don't divide by zero.
15728 Original commit message from CVS:
15729 * ext/theora/theoradec.c:
15730 Cool kids don't divide by zero.
15731 Treat PAR of x:0 as 1:1.
15734 2008-04-30 14:37:52 +0000 Tim-Philipp Müller <tim@centricular.net>
15736 gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
15737 Original commit message from CVS:
15738 * gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
15739 (mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
15740 (mpeg_video_stream_type_find):
15741 Refactor a bit: use context structure to track parsing offset and size of
15742 available data and make the code a bit clearer. Fixes bad memory access
15745 2008-04-28 22:18:49 +0000 Michael Smith <msmith@xiph.org>
15747 gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
15748 Original commit message from CVS:
15749 * gst/playback/test4.c:
15750 * gst/playback/test5.c:
15751 * gst/playback/test6.c:
15752 * gst/tcp/gstmultifdsink.c:
15753 Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
15756 2008-04-28 08:51:38 +0000 Wim Taymans <wim.taymans@gmail.com>
15758 gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
15759 Original commit message from CVS:
15760 * gst-libs/gst/audio/gstbaseaudiosink.h:
15762 * gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
15763 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
15764 (gst_base_audio_src_set_slave_method),
15765 (gst_base_audio_src_get_slave_method),
15766 (gst_base_audio_src_set_property),
15767 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
15768 * gst-libs/gst/audio/gstbaseaudiosrc.h:
15769 Add property and methods for selecting the clock slave method in the
15770 source, like in the sink.
15771 We only implement "none" and "re-timestamp" for now.
15772 API: gst_base_audio_src_set_slave_method()
15773 API: gst_base_audio_src_get_slave_method()
15775 2008-04-25 18:18:47 +0000 Wim Taymans <wim.taymans@gmail.com>
15777 gst-libs/gst/app/gstappsink.*: Add more docs.
15778 Original commit message from CVS:
15779 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
15780 (gst_app_sink_init), (gst_app_sink_set_property),
15781 (gst_app_sink_get_property), (gst_app_sink_event),
15782 (gst_app_sink_preroll), (gst_app_sink_render),
15783 (gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
15784 (gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
15785 (gst_app_sink_pull_buffer):
15786 * gst-libs/gst/app/gstappsink.h:
15788 Add signals for when preroll and render buffers are available.
15789 Add property to control signal emission.
15790 Add property to control the max queue size.
15792 2008-04-25 07:37:09 +0000 Wim Taymans <wim.taymans@gmail.com>
15794 gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
15795 Original commit message from CVS:
15796 * gst-libs/gst/rtp/gstrtpbuffer.c:
15797 Fix the docs about the seqnum compare function, it returns a difference.
15799 2008-04-24 09:27:35 +0000 Edward Hervey <bilboed@bilboed.com>
15801 ext/alsa/gstalsadeviceprobe.c: Don't return before freeing up the allocated structures.
15802 Original commit message from CVS:
15803 * ext/alsa/gstalsadeviceprobe.c:
15804 (gst_alsa_get_device_list): Don't return before freeing up
15805 the allocated structures.
15807 2008-04-24 08:19:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15809 gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
15810 Original commit message from CVS:
15811 * gst/playback/gstplaybin.c:
15812 Remove obsolete streaminfo code and fix a leak. Fixes #529546
15814 2008-04-23 13:50:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15816 ext/ogg/gstoggdemux.c: Revert the event part, that should not go in.
15817 Original commit message from CVS:
15818 * ext/ogg/gstoggdemux.c:
15819 Revert the event part, that should not go in.
15821 2008-04-23 13:45:29 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15823 ext/ogg/gstoggdemux.c: Don't leak GstPluginFeatures when filtering.
15824 Original commit message from CVS:
15825 * ext/ogg/gstoggdemux.c:
15826 Don't leak GstPluginFeatures when filtering.
15828 2008-04-23 08:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15830 sys/xvimage/xvimagesink.c: Add some logging for cases when grabbing the xv failed.
15831 Original commit message from CVS:
15832 * sys/xvimage/xvimagesink.c:
15833 Add some logging for cases when grabbing the xv failed.
15835 2008-04-22 06:18:04 +0000 David Schleef <ds@schleef.org>
15837 ext/ogg/gstoggmux.c: Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos packet. Should conform to what we cu...
15838 Original commit message from CVS:
15839 * ext/ogg/gstoggmux.c:
15840 Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos
15841 packet. Should conform to what we currently think is the
15842 final Ogg/Dirac muxing spec.
15844 2008-04-22 06:13:43 +0000 David Schleef <ds@schleef.org>
15846 sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display. Dark g...
15847 Original commit message from CVS:
15848 * sys/xvimage/xvimagesink.c:
15849 Fix typo that causes the overlay keying color to bright green
15850 on a 16-bit display. Dark grey good. Bright green bad.
15852 2008-04-21 13:47:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15854 ext/gnomevfs/gstgnomevfsuri.c: Add FIXME comment about using uri-list for source and sink.
15855 Original commit message from CVS:
15856 * ext/gnomevfs/gstgnomevfsuri.c:
15857 Add FIXME comment about using uri-list for source and sink.
15859 2008-04-20 11:42:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15861 ext/ogg/gstogmparse.c: GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to vaargs functions to gin...
15862 Original commit message from CVS:
15863 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
15864 GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to
15865 vaargs functions to gint. Otherwise the fractions will get 0 set
15866 instead of the correct value on big endian systems. Fixes bug #529018.
15868 2008-04-20 10:17:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15870 ext/gnomevfs/: Get the list of supported URI schemes in a threadsafe way and use the same list for the source and sink.
15871 Original commit message from CVS:
15872 * ext/gnomevfs/gstgnomevfssink.c:
15873 (gst_gnome_vfs_sink_uri_get_protocols):
15874 * ext/gnomevfs/gstgnomevfssrc.c:
15875 (gst_gnome_vfs_src_uri_get_protocols):
15876 * ext/gnomevfs/gstgnomevfsuri.c: (_internal_get_supported_uris),
15877 (gst_gnomevfs_get_supported_uris):
15878 Get the list of supported URI schemes in a threadsafe way and use the
15879 same list for the source and sink.
15881 2008-04-20 10:11:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15883 ext/gio/gstgio.c: Don't generate a new supported protocols list on each call but cache it. It's supposed to be static...
15884 Original commit message from CVS:
15885 * ext/gio/gstgio.c: (_internal_get_supported_protocols),
15886 (gst_gio_get_supported_protocols):
15887 Don't generate a new supported protocols list on each call but cache
15888 it. It's supposed to be static anyway, this way we only leak it once
15890 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
15891 (gst_gio_sink_class_init), (gst_gio_sink_finalize),
15892 (gst_gio_sink_set_property), (gst_gio_sink_get_property),
15893 (gst_gio_sink_start):
15894 * ext/gio/gstgiosink.h:
15895 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
15896 (gst_gio_src_class_init), (gst_gio_src_finalize),
15897 (gst_gio_src_set_property), (gst_gio_src_get_property),
15898 (gst_gio_src_start):
15899 * ext/gio/gstgiosrc.h:
15900 API: Add "file" properties where one can set a GFile as source/destination.
15901 Add locking to the properties and use gst_element_class_set_details_simple()
15902 instead of a static GstElementDetails struct.
15904 2008-04-19 20:06:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15906 gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
15907 Original commit message from CVS:
15908 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
15910 Add "mpp" and "mp+" as possible extensions for MusePack files.
15911 Add typefinding for MusePack StreamVersion 8 files and include the
15912 stream version in the caps.
15914 2008-04-19 16:33:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15916 gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
15917 Original commit message from CVS:
15918 * gst-libs/gst/rtp/gstrtppayloads.c:
15919 (gst_rtp_payload_info_for_name):
15920 Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
15922 2008-04-18 17:10:43 +0000 Tim-Philipp Müller <tim@centricular.net>
15924 configure.ac: Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level (NB: this only affects compilation of some...
15925 Original commit message from CVS:
15927 Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level
15928 (NB: this only affects compilation of some of the examples).
15929 Remove some configure.ac cruft that's not needed any longer.
15931 2008-04-18 14:54:01 +0000 Edward Hervey <bilboed@bilboed.com>
15933 gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
15934 Original commit message from CVS:
15935 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
15936 Don't validate the payload if there isn't any.
15939 2008-04-17 07:33:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15941 gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
15942 Original commit message from CVS:
15943 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
15944 Use g_atomic_int_set() instead of gst_atomic_int_set().
15946 2008-04-17 07:29:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15948 ext/gio/gstgio.c: Return NULL instead of a gchar * array with one NULL element if we don't get any supported URI sche...
15949 Original commit message from CVS:
15950 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
15951 Return NULL instead of a gchar * array with one NULL element if we
15952 don't get any supported URI schemes from GIO.
15954 2008-04-15 19:06:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15956 gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
15957 Original commit message from CVS:
15958 * gst/audiotestsrc/gstaudiotestsrc.c:
15959 Remove cpp style commented old code.
15961 2008-04-15 19:02:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15963 gst/playback/gstdecodebin2.c: Fix signal docs.
15964 Original commit message from CVS:
15965 * gst/playback/gstdecodebin2.c:
15968 2008-04-14 17:58:19 +0000 Tim-Philipp Müller <tim@centricular.net>
15970 ext/pango/gsttextoverlay.c: Fix textoverlay unit test again by making the supposed default value for the wait-text pr...
15971 Original commit message from CVS:
15972 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
15973 (gst_text_overlay_init):
15974 Fix textoverlay unit test again by making the supposed default
15975 value for the wait-text property the actual default value.
15976 Also fix Since: tag for new property.
15978 2008-04-11 17:13:52 +0000 Tim-Philipp Müller <tim@centricular.net>
15980 gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
15981 Original commit message from CVS:
15982 * gst-libs/gst/video/video.c: (gst_video_format_new_caps),
15983 (gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
15984 (gst_video_format_get_pixel_stride),
15985 (gst_video_format_get_component_width),
15986 (gst_video_format_get_component_height),
15987 (gst_video_format_get_component_offset), (gst_video_format_get_size),
15988 (gst_video_format_convert):
15989 Add guards to these functions to ensure sane input values.
15990 * tests/check/libs/video.c:
15991 Fix unit test not to create caps with width=0 and height=0.
15993 2008-04-11 01:25:01 +0000 Wim Taymans <wim.taymans@gmail.com>
15995 docs/design/draft-keyframe-force.txt: Fix typo.
15996 Original commit message from CVS:
15997 * docs/design/draft-keyframe-force.txt:
15999 * gst/playback/gstqueue2.c: (update_buffering),
16000 (gst_queue_handle_src_query):
16001 Set buffering mode in the messages.
16002 Set buffering percent in the query.
16003 * tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
16004 (do_stream_buffering), (do_download_buffering), (msg_buffering):
16005 Do some more fancy things based on the buffering method in use.
16007 2008-04-09 21:42:24 +0000 Wim Taymans <wim.taymans@gmail.com>
16009 tests/examples/seek/seek.c: Add basic download reports to seek using the new buffering API.
16010 Original commit message from CVS:
16011 * tests/examples/seek/seek.c: (update_fill), (set_update_fill),
16012 (play_cb), (pause_cb), (stop_cb), (msg_state_changed),
16013 (msg_buffering), (main):
16014 Add basic download reports to seek using the new buffering API.
16016 2008-04-09 21:40:17 +0000 Wim Taymans <wim.taymans@gmail.com>
16018 gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
16019 Original commit message from CVS:
16020 * gst/playback/gstqueue2.c: (update_buffering),
16021 (gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
16022 (gst_queue_src_checkgetrange_function):
16023 Include extra buffering stats in the buffering message.
16024 Implement BUFFERING query.
16025 * gst/playback/gsturidecodebin.c: (do_async_start),
16026 (do_async_done), (type_found), (setup_streaming), (setup_source),
16027 (gst_uri_decode_bin_change_state):
16028 Only add decodebin2 when the type is found in streaming mode.
16029 Make uridecodebin async to PAUSED even when we don't have decodebin2
16032 2008-04-09 08:38:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16034 ext/gio/gstgio.c: Filter cdda from the supported URI schemes. We can't support musicbrainz tags and everything else o...
16035 Original commit message from CVS:
16036 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
16037 Filter cdda from the supported URI schemes. We can't support
16038 musicbrainz tags and everything else one expects from a cdda source
16039 with GIO. Fixes bug #526794.
16041 2008-04-07 22:37:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16043 * sys/xvimage/xvimagesink.c:
16044 Fix calculation of 'expected size' for YV12 buffers.
16045 Original commit message from CVS:
16046 2008-04-07 Jan Schmidt <jan.schmidt@sun.com>
16047 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
16048 (gst_xvimagesink_buffer_alloc):
16049 Fix calculation of 'expected size' for YV12 buffers.
16050 Be a little more verbose in the debug output for buffer-alloc'ed
16051 buffers which turn out to have the wrong size.
16053 2008-04-07 22:26:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16056 Fix calculation of 'expected size' for YV12 buffers.
16057 Original commit message from CVS:
16058 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
16059 (gst_xvimagesink_buffer_alloc):
16060 Fix calculation of 'expected size' for YV12 buffers.
16061 Be a little more verbose in the debug output for buffer-alloc'ed
16062 buffers which turn out to have the wrong size.
16064 2008-04-07 10:50:11 +0000 Tim-Philipp Müller <tim@centricular.net>
16066 Merge other changes from 0.10.19 release branch.
16067 Original commit message from CVS:
16070 * gst-plugins-base.doap:
16071 Merge other changes from 0.10.19 release branch.
16073 2008-04-06 20:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
16075 gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
16076 Original commit message from CVS:
16077 * gst-libs/gst/audio/gstbaseaudiosink.c:
16078 (gst_base_audio_sink_class_init):
16079 * gst-libs/gst/audio/gstbaseaudiosrc.c:
16080 (gst_base_audio_src_class_init):
16081 * gst/playback/gstplayback.c: (plugin_init):
16082 * gst/volume/gstvolume.c: (plugin_init):
16083 Work around missing bits of thread-safety on older GLibs some
16084 more to avoid assertions when starting up multiple playbin
16085 objects concurrently (see #512382).
16087 2008-04-06 17:19:39 +0000 Tim-Philipp Müller <tim@centricular.net>
16089 gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
16090 Original commit message from CVS:
16091 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
16092 Remove some more fields.
16094 2008-04-06 08:56:07 +0000 Damien Lespiau <damien.lespiau@gmail.com>
16096 configure.ac: Actually build dlls when cross-compiling with mingw32.
16097 Original commit message from CVS:
16098 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
16100 Actually build dlls when cross-compiling with mingw32.
16103 2008-04-03 23:01:11 +0000 Tim-Philipp Müller <tim@centricular.net>
16105 configure.ac: Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
16106 Original commit message from CVS:
16108 Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
16110 2008-04-03 16:10:53 +0000 Wim Taymans <wim.taymans@gmail.com>
16112 tests/examples/seek/seek.c: Add statusbar.
16113 Original commit message from CVS:
16114 * tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
16115 (msg_buffering), (connect_bus_signals), (main):
16117 Add buffering support with feedback in the statusbar.
16119 2008-04-03 15:58:37 +0000 Tim-Philipp Müller <tim@centricular.net>
16121 ext/ogg/gstoggmux.c: Fix sample pipeline description.
16122 Original commit message from CVS:
16123 * ext/ogg/gstoggmux.c:
16124 Fix sample pipeline description.
16126 2008-04-03 14:58:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16128 docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
16129 Original commit message from CVS:
16130 * docs/plugins/Makefile.am:
16131 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
16132 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
16133 * docs/plugins/gst-plugins-base-plugins-sections.txt:
16134 Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
16135 * docs/plugins/gst-plugins-base-plugins.args:
16136 * docs/plugins/gst-plugins-base-plugins.hierarchy:
16137 * docs/plugins/gst-plugins-base-plugins.interfaces:
16138 * docs/plugins/gst-plugins-base-plugins.prerequisites:
16139 * docs/plugins/inspect/plugin-adder.xml:
16140 * docs/plugins/inspect/plugin-alsa.xml:
16141 * docs/plugins/inspect/plugin-audioconvert.xml:
16142 * docs/plugins/inspect/plugin-audiorate.xml:
16143 * docs/plugins/inspect/plugin-audioresample.xml:
16144 * docs/plugins/inspect/plugin-audiotestsrc.xml:
16145 * docs/plugins/inspect/plugin-cdparanoia.xml:
16146 * docs/plugins/inspect/plugin-decodebin.xml:
16147 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
16148 * docs/plugins/inspect/plugin-gdp.xml:
16149 * docs/plugins/inspect/plugin-gnomevfs.xml:
16150 * docs/plugins/inspect/plugin-libvisual.xml:
16151 * docs/plugins/inspect/plugin-ogg.xml:
16152 * docs/plugins/inspect/plugin-pango.xml:
16153 * docs/plugins/inspect/plugin-playback.xml:
16154 * docs/plugins/inspect/plugin-queue2.xml:
16155 * docs/plugins/inspect/plugin-subparse.xml:
16156 * docs/plugins/inspect/plugin-tcp.xml:
16157 * docs/plugins/inspect/plugin-theora.xml:
16158 * docs/plugins/inspect/plugin-typefindfunctions.xml:
16159 * docs/plugins/inspect/plugin-uridecodebin.xml:
16160 * docs/plugins/inspect/plugin-video4linux.xml:
16161 * docs/plugins/inspect/plugin-videorate.xml:
16162 * docs/plugins/inspect/plugin-videoscale.xml:
16163 * docs/plugins/inspect/plugin-videotestsrc.xml:
16164 * docs/plugins/inspect/plugin-volume.xml:
16165 * docs/plugins/inspect/plugin-vorbis.xml:
16166 * docs/plugins/inspect/plugin-ximagesink.xml:
16167 * docs/plugins/inspect/plugin-xvimagesink.xml:
16168 Update introspection data.
16169 * ext/ogg/gstoggmux.c:
16171 * gst/playback/gstdecodebin2.c:
16172 Don't use gtk-doc style comment start for private stuff, but make it
16173 formatted like this for consistency.
16175 2008-04-03 12:16:04 +0000 Wim Taymans <wim.taymans@gmail.com>
16177 gst/playback/gstdecodebin2.c: Remove fakesink hack, we can now implement this more elegantly.
16178 Original commit message from CVS:
16179 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
16180 (gst_decode_bin_init), (gst_decode_bin_dispose),
16181 (gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
16182 (gst_decode_bin_set_property), (gst_decode_bin_get_property),
16183 (analyze_new_pad), (connect_pad), (expose_pad),
16184 (gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
16185 (gst_decode_group_expose), (gst_decode_group_free),
16186 (do_async_start), (do_async_done), (gst_decode_bin_change_state):
16187 Remove fakesink hack, we can now implement this more elegantly.
16188 Added property to bypass typefinding.
16189 Removed underrun callback and demuxer pad probe, we now use the srcpad
16190 probe to expose groups.
16191 API::sink-caps property
16192 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
16193 Guard against multiple emissions of the no_more_pads signal, which
16194 happens when we are dealing with chained oggs.
16195 * gst/playback/gsturidecodebin.c: (remove_decoders),
16196 (make_decoder), (type_found), (setup_streaming), (source_new_pad),
16198 For streams, use our own typefind element and plug our queue after it.
16199 We will need this to determine the type of buffering to use for the
16202 2008-04-03 10:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
16204 gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
16205 Original commit message from CVS:
16206 * gst-libs/gst/audio/gstbaseaudiosink.c:
16207 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
16208 Guard against over and underflows because of clock slaving.
16209 When we are using our own clock, still compensate for any calibrations
16210 that we might have done to our clock.
16212 2008-04-03 10:22:33 +0000 Wim Taymans <wim.taymans@gmail.com>
16214 ext/theora/theoradec.c: Don't try to do anything fancy with the return code from pushing an event, it does not have e...
16215 Original commit message from CVS:
16216 * ext/theora/theoradec.c: (theora_handle_type_packet),
16217 (theora_dec_chain):
16218 Don't try to do anything fancy with the return code from pushing an
16219 event, it does not have enough information to turn it into a
16222 2008-04-03 10:19:43 +0000 Wim Taymans <wim.taymans@gmail.com>
16224 ext/ogg/gstoggdemux.c: Add small debug line.
16225 Original commit message from CVS:
16226 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_reset),
16227 (gst_ogg_demux_chain_elem_pad):
16228 Add small debug line.
16229 Pass return code from the internal decoder instead of the too generic
16232 2008-04-03 06:39:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16234 gst-libs/gst/cdda/: Use GLib's base64 implementation instead of our own.
16235 Original commit message from CVS:
16236 * gst-libs/gst/cdda/Makefile.am:
16237 * gst-libs/gst/cdda/base64.c:
16238 * gst-libs/gst/cdda/base64.h:
16239 * gst-libs/gst/cdda/gstcddabasesrc.c:
16240 (gst_cddabasesrc_calculate_musicbrainz_discid):
16241 Use GLib's base64 implementation instead of our own.
16243 2008-04-02 15:41:50 +0000 Wim Taymans <wim.taymans@gmail.com>
16245 ext/ogg/gstoggdemux.c: Refix oggdemux, we only have a problem if we failed to find a chain and we are not EOF.
16246 Original commit message from CVS:
16247 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
16248 (gst_ogg_demux_read_chain):
16249 Refix oggdemux, we only have a problem if we failed to find a chain and
16252 2008-04-02 15:07:01 +0000 Victor STINNER <victor.stinner@haypocalc.com>
16254 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...
16255 Original commit message from CVS:
16256 Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
16257 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
16258 (gst_ogg_demux_read_chain):
16259 When we fail to find a BOS page and we and up with no chain, error out
16260 properly instead of segfaulting. Fixes #525665.
16262 2008-04-02 14:58:05 +0000 Wim Taymans <wim.taymans@gmail.com>
16264 ext/ogg/gstoggdemux.c: The new-pad-group sequence is add-pads, no-more-pads, add-pads, no-more-pads...
16265 Original commit message from CVS:
16266 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
16267 (gst_ogg_demux_read_chain), (gst_ogg_demux_handle_page):
16268 The new-pad-group sequence is add-pads, no-more-pads, add-pads,
16271 2008-04-02 11:08:05 +0000 Wim Taymans <wim.taymans@gmail.com>
16273 gst/playback/gstqueue2.c: Update the estimated input data when we push out a buffer.
16274 Original commit message from CVS:
16275 * gst/playback/gstqueue2.c: (update_out_rates),
16276 (gst_queue_open_temp_location_file),
16277 (gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
16278 (gst_queue_handle_src_query), (gst_queue_set_property):
16279 Update the estimated input data when we push out a buffer.
16280 Add some debug info about the temp file.
16281 Only forward src events when we are not using a temp file.
16282 Don't block the duration query, we need to find something better.
16283 Don't leak the temp filename.
16285 2008-04-01 14:01:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16287 configure.ac: Require GLib 2.12 and liboil 0.3.14.
16288 Original commit message from CVS:
16290 Require GLib 2.12 and liboil 0.3.14.
16291 * gst/volume/gstvolume.c: (volume_process_double):
16292 Unconditionally use liboil 0.3.14 function.
16294 2008-03-31 16:08:45 +0000 Wim Taymans <wim.taymans@gmail.com>
16296 gst-libs/gst/riff/riff-media.c: ms-gsm can have arbitrarty sample rates. See #481354.
16297 Original commit message from CVS:
16298 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
16299 ms-gsm can have arbitrarty sample rates. See #481354.
16301 2008-03-28 16:22:35 +0000 Wim Taymans <wim.taymans@gmail.com>
16303 gst-libs/gst/riff/riff-media.c: MP4S is generic MPEG-4, not a microsoft variant.
16304 Original commit message from CVS:
16305 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
16306 MP4S is generic MPEG-4, not a microsoft variant.
16308 2008-03-27 15:26:38 +0000 Michael Smith <msmith@xiph.org>
16310 gst/gdp/gstgdpdepay.c: Check the body CRC (if set) when depayloading.
16311 Original commit message from CVS:
16312 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
16313 Check the body CRC (if set) when depayloading.
16316 2008-03-24 17:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
16318 ext/pango/gsttextoverlay.c: Fix Since: version for new property.
16319 Original commit message from CVS:
16320 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
16321 Fix Since: version for new property.
16323 2008-03-24 16:40:08 +0000 Wim Taymans <wim.taymans@gmail.com>
16325 gst-libs/gst/rtsp/gstrtspconnection.c: Don't error when poll_wait returns EAGAIN.
16326 Original commit message from CVS:
16327 * gst-libs/gst/rtsp/gstrtspconnection.c:
16328 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
16329 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
16330 Don't error when poll_wait returns EAGAIN.
16332 2008-03-24 14:08:22 +0000 Wim Taymans <wim.taymans@gmail.com>
16334 gst/playback/gstqueue2.c: The queue is never filled when there are no buffers in the queue at all.
16335 Original commit message from CVS:
16336 * gst/playback/gstqueue2.c: (gst_queue_is_filled):
16337 The queue is never filled when there are no buffers in the queue at all.
16340 2008-03-24 12:26:30 +0000 Wim Taymans <wim.taymans@gmail.com>
16342 gst/playback/gstplaybin2.c: Update some docs.
16343 Original commit message from CVS:
16344 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
16345 (init_group), (free_group), (gst_play_bin_init),
16346 (gst_play_bin_finalize), (gst_play_bin_set_uri),
16347 (gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
16348 (gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
16349 (gst_play_bin_set_current_video_stream),
16350 (gst_play_bin_set_current_audio_stream),
16351 (gst_play_bin_set_current_text_stream),
16352 (gst_play_bin_set_encoding), (gst_play_bin_set_property),
16353 (gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
16354 (no_more_pads_cb), (perform_eos), (autoplug_select_cb),
16355 (activate_group), (deactivate_group), (setup_next_source),
16356 (save_current_group), (gst_play_bin_change_state):
16358 Add new locks and conds to protect pipeline creation and group
16360 Implement the sub-uri property.
16361 Keep track of pending uridecodebin creation and configure the output
16362 pipeline after all streams are configured.
16363 Propagate subtitle encoding to the uridecodebins.
16364 Implement getting the video/audio/visualisation elements.
16365 Use input-selector for stream switching.
16366 If we are asked to do visualisation, prefer to autoplug raw sinks
16367 instead of sinks that accept encoded data.
16369 2008-03-24 12:15:26 +0000 Wim Taymans <wim.taymans@gmail.com>
16371 gst/playback/gstplaysink.*: Add methods to get audio/video/vis elements.
16372 Original commit message from CVS:
16373 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
16374 (gst_play_sink_init), (gst_play_sink_dispose),
16375 (gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
16376 (gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
16377 (gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
16378 (gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
16379 (gst_play_sink_set_volume), (gst_play_sink_get_volume),
16380 (gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
16381 (gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
16382 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
16383 (gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
16384 * gst/playback/gstplaysink.h:
16385 Add methods to get audio/video/vis elements.
16386 Add methods to set the font description for the overlay.
16387 Remove properties, we're using this element with its methods only.
16388 Add support for subtitles.
16389 Rearrange the locking a bit to not use the object lock for protecting
16390 the pipeline construction.
16391 Try to use the volume and mute property on the sink when its available.
16392 Implement the mute option with volume when the sink does not have a mute
16394 Only add volume element when the sink has no volume property.
16395 Only do visualisations with raw audio pads.
16397 2008-03-24 12:03:02 +0000 Wim Taymans <wim.taymans@gmail.com>
16399 ext/pango/gsttextoverlay.*: Add property to configure waiting for text on the textpad or not, with the default behavi...
16400 Original commit message from CVS:
16401 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
16402 (gst_text_overlay_init), (gst_text_overlay_set_property),
16403 (gst_text_overlay_get_property), (gst_text_overlay_src_event),
16404 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
16405 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
16406 (gst_text_overlay_change_state):
16407 * ext/pango/gsttextoverlay.h:
16408 Add property to configure waiting for text on the textpad or not, with
16409 the default behaviour being the old one (always wait for text before
16410 rendering the video). This default behaviour is usually not the best one
16411 because the text stream can very sparse and could require queueing a lot
16413 Fix the flushing and EOS handing so that we don't mix up their meaning.
16415 2008-03-24 11:54:02 +0000 Wim Taymans <wim.taymans@gmail.com>
16417 gst/playback/gsturidecodebin.c: Add a readonly source property and notify.
16418 Original commit message from CVS:
16419 * gst/playback/gsturidecodebin.c:
16420 (gst_uri_decode_bin_autoplug_factories),
16421 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
16422 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
16423 (gst_uri_decode_bin_set_property),
16424 (gst_uri_decode_bin_get_property), (no_more_pads_full),
16425 (new_decoded_pad_cb), (gen_source_element), (remove_decoders),
16426 (proxy_autoplug_factories_signal), (make_decoder),
16427 (source_new_pad), (setup_source):
16428 Add a readonly source property and notify.
16429 Add new lock for protecting the construction of the pipeline.
16430 Keep track of the decodebins we plugged.
16431 Correctly proxy the autoplug signal so that it actually continues.
16432 Proxy subtitle-encoding to the decodebins.
16434 2008-03-24 11:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
16436 tests/examples/seek/seek.c: Rearrange some buttons in playbin2 and make some other boxes insensitive when needed.
16437 Original commit message from CVS:
16438 * tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
16439 (text_toggle_cb), (update_streams), (main):
16440 Rearrange some buttons in playbin2 and make some other boxes insensitive
16442 Add language codes to subtitle selection boxes when we gind the right
16443 tags for the streams.
16445 2008-03-24 11:36:08 +0000 Wim Taymans <wim.taymans@gmail.com>
16447 gst/playback/gstdecodebin2.c: Protect caps property with the object lock.
16448 Original commit message from CVS:
16449 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
16450 (gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
16451 (gst_decode_bin_set_subs_encoding),
16452 (gst_decode_bin_get_subs_encoding),
16453 (gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
16454 (deactivate_free_recursive):
16455 Protect caps property with the object lock.
16456 Protect encoding property with the object lock.
16457 Keep list of elements we added that have the subtitle-encoding property.
16458 Distribute the subtitle-encoding to all of the elements when it
16461 2008-03-24 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
16463 gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
16464 Original commit message from CVS:
16465 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
16466 Small debug improvement.
16467 * gst-libs/gst/audio/gstbaseaudiosink.c:
16468 (gst_base_audio_sink_render):
16469 Fix bug in determining the sample start/stop position, we want to base
16470 this decision on the fact that we are going forwards or backwards, not
16471 slower or faster. This fixes some ugly resync warnings when playing at
16474 2008-03-23 13:41:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16476 ext/gio/gstgio.c: Correctly set the supported URI schemes and don't leave some schemes in the middle or at the start ...
16477 Original commit message from CVS:
16478 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
16479 Correctly set the supported URI schemes and don't leave
16480 some schemes in the middle or at the start at NULL.
16482 2008-03-23 13:12:41 +0000 Tim-Philipp Müller <tim@centricular.net>
16484 tests/check/elements/gdpdepay.c: Make test compile without unused function/variable warnings on PPC.
16485 Original commit message from CVS:
16486 * tests/check/elements/gdpdepay.c:
16487 Make test compile without unused function/variable warnings on PPC.
16489 2008-03-22 15:00:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16491 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
16492 Original commit message from CVS:
16494 * ext/alsa/gstalsamixerelement.c:
16495 (gst_alsa_mixer_element_class_init):
16496 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
16497 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
16498 * ext/cdparanoia/gstcdparanoiasrc.c:
16499 (gst_cd_paranoia_src_class_init):
16500 * ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
16501 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
16502 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
16503 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
16504 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
16505 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
16506 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
16507 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
16508 * ext/pango/gsttextrender.c: (gst_text_render_class_init):
16509 * ext/theora/theoradec.c: (gst_theora_dec_class_init):
16510 * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
16511 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
16512 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
16513 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
16514 (gst_audio_filter_template_class_init):
16515 * gst-libs/gst/audio/gstbaseaudiosink.c:
16516 (gst_base_audio_sink_class_init):
16517 * gst-libs/gst/audio/gstbaseaudiosrc.c:
16518 (gst_base_audio_src_class_init):
16519 * gst-libs/gst/cdda/gstcddabasesrc.c:
16520 (gst_cdda_base_src_class_init):
16521 * gst-libs/gst/interfaces/mixertrack.c:
16522 (gst_mixer_track_class_init):
16523 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16524 (gst_base_rtp_depayload_class_init):
16525 * gst-libs/gst/rtp/gstbasertppayload.c:
16526 (gst_basertppayload_class_init):
16527 * gst/audioconvert/gstaudioconvert.c:
16528 (gst_audio_convert_class_init):
16529 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
16530 * gst/audioresample/gstaudioresample.c:
16531 (gst_audioresample_class_init):
16532 * gst/audiotestsrc/gstaudiotestsrc.c:
16533 (gst_audio_test_src_class_init):
16534 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
16535 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
16536 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
16537 (preroll_unlinked):
16538 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
16539 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
16540 * gst/playback/gstplaysink.c: (gst_play_sink_class_init):
16541 * gst/playback/gstqueue2.c: (gst_queue_class_init):
16542 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
16543 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
16544 (gst_stream_selector_class_init):
16545 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
16546 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
16547 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
16548 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
16549 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
16550 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
16551 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
16552 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
16553 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
16554 * gst/videotestsrc/gstvideotestsrc.c:
16555 (gst_video_test_src_class_init):
16556 * gst/volume/gstvolume.c: (gst_volume_class_init):
16557 * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
16558 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
16559 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
16560 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
16561 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
16562 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
16563 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
16564 static strings (i.e. all). This gives us less memory usage,
16565 fewer allocations and thus less memory defragmentation. Depend
16566 on core CVS for this. Fixes bug #523806.
16568 2008-03-22 14:13:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16570 ext/gio/gstgio.c: Filter http and https protocols. GIO/GVfs handles them but it's impossible to implement iradio/icec...
16571 Original commit message from CVS:
16572 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
16573 Filter http and https protocols. GIO/GVfs handles them but it's
16574 impossible to implement iradio/icecast with it. Better use
16575 souphttpsrc or something else for this.
16576 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
16577 If getting the file informations by a query fails try it with the
16578 seek-to-end trick too.
16580 2008-03-21 16:46:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16582 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...
16583 Original commit message from CVS:
16584 * gst/volume/gstvolume.c: (gst_volume_interface_supported),
16585 (gst_volume_base_init), (gst_volume_class_init),
16586 (volume_process_double), (volume_process_float),
16587 (volume_transform_ip), (plugin_init):
16588 memset buffers to zero if we get a GAP buffer. We usually see a
16589 buffer as one unit so let's handle it as one and don't care about
16590 volume changes while processing one buffer.
16591 Also clean up some stuff a bit.
16593 2008-03-21 15:58:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16595 gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
16596 Original commit message from CVS:
16597 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
16598 (gst_audio_convert_create_silence_buffer),
16599 (gst_audio_convert_transform):
16600 Make audioconvert GAP-aware by outputting silence buffers when the
16601 input has the GAP flag set. This is up to 8x faster.
16602 Based on a patch by Stefan Kost. Fixes bug #517813.
16604 2008-03-21 15:54:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16606 gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
16607 Original commit message from CVS:
16608 * gst/volume/gstvolume.c: (volume_process_double):
16609 Use oil_scalarmultiply_f64_ns() for double processing when it's
16610 available at compile time.
16612 2008-03-21 13:27:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16614 configure.ac: Fix lrint/lrintf checks to actually work. These functions are in libm on Linux at least so try to link ...
16615 Original commit message from CVS:
16617 Fix lrint/lrintf checks to actually work. These functions are
16618 in libm on Linux at least so try to link to it.
16620 2008-03-21 00:36:20 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16622 configure.ac: Back to development - 0.10.18.1
16623 Original commit message from CVS:
16625 Back to development - 0.10.18.1
16627 === release 0.10.18 ===
16629 2008-03-21 00:26:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16635 * docs/plugins/gst-plugins-base-plugins.args:
16636 * docs/plugins/gst-plugins-base-plugins.hierarchy:
16637 * docs/plugins/gst-plugins-base-plugins.interfaces:
16638 * docs/plugins/gst-plugins-base-plugins.prerequisites:
16639 * docs/plugins/gst-plugins-base-plugins.signals:
16640 * docs/plugins/inspect/plugin-adder.xml:
16641 * docs/plugins/inspect/plugin-alsa.xml:
16642 * docs/plugins/inspect/plugin-audioconvert.xml:
16643 * docs/plugins/inspect/plugin-audiorate.xml:
16644 * docs/plugins/inspect/plugin-audioresample.xml:
16645 * docs/plugins/inspect/plugin-audiotestsrc.xml:
16646 * docs/plugins/inspect/plugin-cdparanoia.xml:
16647 * docs/plugins/inspect/plugin-decodebin.xml:
16648 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
16649 * docs/plugins/inspect/plugin-gdp.xml:
16650 * docs/plugins/inspect/plugin-gnomevfs.xml:
16651 * docs/plugins/inspect/plugin-libvisual.xml:
16652 * docs/plugins/inspect/plugin-ogg.xml:
16653 * docs/plugins/inspect/plugin-pango.xml:
16654 * docs/plugins/inspect/plugin-playback.xml:
16655 * docs/plugins/inspect/plugin-queue2.xml:
16656 * docs/plugins/inspect/plugin-subparse.xml:
16657 * docs/plugins/inspect/plugin-tcp.xml:
16658 * docs/plugins/inspect/plugin-theora.xml:
16659 * docs/plugins/inspect/plugin-typefindfunctions.xml:
16660 * docs/plugins/inspect/plugin-uridecodebin.xml:
16661 * docs/plugins/inspect/plugin-video4linux.xml:
16662 * docs/plugins/inspect/plugin-videorate.xml:
16663 * docs/plugins/inspect/plugin-videoscale.xml:
16664 * docs/plugins/inspect/plugin-videotestsrc.xml:
16665 * docs/plugins/inspect/plugin-volume.xml:
16666 * docs/plugins/inspect/plugin-vorbis.xml:
16667 * docs/plugins/inspect/plugin-ximagesink.xml:
16668 * docs/plugins/inspect/plugin-xvimagesink.xml:
16669 * gst-plugins-base.doap:
16671 * win32/common/config.h:
16673 Original commit message from CVS:
16676 2008-03-21 00:16:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16703 Original commit message from CVS:
16706 2008-03-18 12:19:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16708 0.10.17.4 pre-release
16709 Original commit message from CVS:
16711 * win32/common/config.h:
16712 0.10.17.4 pre-release
16714 2008-03-18 11:20:05 +0000 Wim Taymans <wim.taymans@gmail.com>
16716 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL when trying to print strings that could be NULL because this might...
16717 Original commit message from CVS:
16718 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
16719 Use GST_STR_NULL when trying to print strings that could be NULL because
16720 this might crash on some platforms. See #520808.
16722 2008-03-18 11:10:12 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
16724 gst-libs/gst/rtsp/gstrtspconnection.c: Generic Windows fixes that makes libgstrtsp work on Windows when coupled with ...
16725 Original commit message from CVS:
16726 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
16727 * gst-libs/gst/rtsp/gstrtspconnection.c:
16728 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
16729 (read_line), (gst_rtsp_connection_read_internal):
16730 Generic Windows fixes that makes libgstrtsp work on Windows when
16731 coupled with the new GstPoll API. See #520808.
16733 2008-03-17 22:06:56 +0000 Milosz Derezynski <internalerror@gmail.com>
16735 ext/gio/gstgiobasesrc.c: If seeking to a new position succeeds don't simply return from create() without creating a b...
16736 Original commit message from CVS:
16737 Patch by: Milosz Derezynski <internalerror at gmail dot com>
16738 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
16739 If seeking to a new position succeeds don't simply return from
16740 create() without creating a buffer. Do this only in the case
16741 seeking to the new position fails. Fixes bug #523054.
16743 2008-03-17 10:32:28 +0000 Tim-Philipp Müller <tim@centricular.net>
16745 gst-libs/gst/video/video.c: Fix gst_video_format_parse_caps() for RGB caps with alpha channel (#522635).
16746 Original commit message from CVS:
16747 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
16748 (gst_video_format_from_rgba32_masks):
16749 Fix gst_video_format_parse_caps() for RGB caps with alpha channel
16751 * tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
16752 Add unit test for the RGB caps parsing and creation, checking for
16753 internal consistency of the new API and consistency of the API with
16754 the old GST_VIDEO_CAPS_* defines.
16756 2008-03-14 18:42:35 +0000 David Schleef <ds@schleef.org>
16758 gst/videotestsrc/videotestsrc.c: Oops, revert last change because -base is in freeze.
16759 Original commit message from CVS:
16760 * gst/videotestsrc/videotestsrc.c: Oops, revert last change
16761 because -base is in freeze.
16763 2008-03-14 17:33:09 +0000 William M. Brack <wbrack@mmm.hk>
16765 gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
16766 Original commit message from CVS:
16767 Patch by: William M. Brack
16768 * gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
16770 2008-03-14 09:54:44 +0000 Wim Taymans <wim.taymans@gmail.com>
16772 gst/playback/gststreamselector.*: Revert change that caused regression until a real fix is found.
16773 Original commit message from CVS:
16774 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
16775 (gst_selector_pad_chain):
16776 * gst/playback/gststreamselector.h:
16777 Revert change that caused regression until a real fix is found.
16780 2008-03-12 12:39:13 +0000 Michael Smith <msmith@xiph.org>
16782 gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
16783 Original commit message from CVS:
16784 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
16785 * gst-libs/gst/audio/gstringbuffer.h:
16786 Rename recently added buffer types to make more sense.
16787 * ext/alsa/gstalsasink.c: (alsasink_parse_spec),
16788 (gst_alsasink_write):
16789 Adapt for above API changes.
16792 2008-03-11 13:23:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16794 win32/common/libgstnetbuffer.def: Add new symbol gst_netaddress_equal. Fixes bug #521743.
16795 Original commit message from CVS:
16796 * win32/common/libgstnetbuffer.def:
16797 Add new symbol gst_netaddress_equal. Fixes bug #521743.
16799 2008-03-11 00:25:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16801 0.10.17.3 pre-release
16802 Original commit message from CVS:
16804 * win32/common/config.h:
16805 0.10.17.3 pre-release
16807 2008-03-10 17:19:56 +0000 Wim Taymans <wim.taymans@gmail.com>
16809 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
16810 Original commit message from CVS:
16811 * gst-libs/gst/audio/gstbaseaudiosrc.c:
16812 (gst_base_audio_src_create):
16813 Fix duration when no clock was provided. Fixes #520300.
16815 2008-03-07 18:17:44 +0000 Olivier Crete <tester@tester.ca>
16817 Add trivial function to compare GstNetAddress. See #520626.
16818 Original commit message from CVS:
16819 Patch by: Olivier Crete <tester at tester ca>
16820 * docs/libs/gst-plugins-base-libs-sections.txt:
16821 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
16822 * gst-libs/gst/netbuffer/gstnetbuffer.h:
16823 Add trivial function to compare GstNetAddress. See #520626.
16824 API: GstNetBuffer::gst_netaddress_equal
16826 2008-03-07 16:10:51 +0000 Wim Taymans <wim.taymans@gmail.com>
16828 gst/tcp/gstmultifdsink.c: Update mode property docs, it's deprecated now.
16829 Original commit message from CVS:
16830 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
16831 Update mode property docs, it's deprecated now.
16833 2008-03-07 15:48:51 +0000 Wim Taymans <wim.taymans@gmail.com>
16835 gst/: Remove GstPollMode from gstpoll constructor.
16836 Original commit message from CVS:
16837 * gst-libs/gst/rtsp/gstrtspconnection.c:
16838 (gst_rtsp_connection_create):
16839 * gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
16840 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
16841 * gst/tcp/gstmultifdsink.h:
16842 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
16843 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
16844 Remove GstPollMode from gstpoll constructor.
16846 2008-03-04 00:26:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16848 0.10.17.2 pre-release
16849 Original commit message from CVS:
16851 * win32/common/config.h:
16852 0.10.17.2 pre-release
16854 2008-03-03 23:59:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16856 gst/Makefile.am: GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean them twice
16857 Original commit message from CVS:
16859 GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
16861 * win32/common/libgstinterfaces.def:
16862 * win32/common/libgstrtp.def:
16863 Add new API to the defs
16865 2008-03-03 16:11:50 +0000 Mersad Jelacic <mersad@axis.com>
16867 gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
16868 Original commit message from CVS:
16869 Patch by: Mersad Jelacic <mersad at axis dot com>
16870 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
16871 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
16872 API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
16873 possible to specify the sample size in bits. (#509637)
16875 2008-03-03 13:59:19 +0000 Tim-Philipp Müller <tim@centricular.net>
16877 tests/check/libs/mixer.c: Add a few simple checks for the new message types.
16878 Original commit message from CVS:
16879 * tests/check/libs/mixer.c:
16880 Add a few simple checks for the new message types.
16882 2008-03-03 13:56:38 +0000 Tim-Philipp Müller <tim@centricular.net>
16884 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
16885 Original commit message from CVS:
16886 * docs/libs/gst-plugins-base-libs-sections.txt:
16887 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
16888 (gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
16889 (gst_mixer_message_get_type),
16890 (gst_mixer_message_parse_option_changed),
16891 (gst_mixer_message_parse_options_list_changed):
16892 * gst-libs/gst/interfaces/mixer.h: (GstMixerType),
16893 (GST_MIXER_MESSAGE_OPTION_CHANGED),
16894 (GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
16895 (GST_MIXER_MESSAGE_MIXER_CHANGED):
16896 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
16897 and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
16899 2008-03-03 13:50:18 +0000 Tim-Philipp Müller <tim@centricular.net>
16901 gst-libs/gst/interfaces/mixeroptions.*: API: add GstMixerOptions::get_values vfunc (#519906)
16902 Original commit message from CVS:
16903 * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
16904 (gst_mixer_options_get_values):
16905 * gst-libs/gst/interfaces/mixeroptions.h:
16906 (GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
16907 (_GstMixerOptions), (_GstMixerOptionsClass):
16908 API: add GstMixerOptions::get_values vfunc (#519906)
16910 2008-03-03 12:01:15 +0000 Peter Kjellerstedt <pkj@axis.com>
16912 configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#4...
16913 Original commit message from CVS:
16915 Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
16916 plug-ins are included/excluded. (#498222)
16918 2008-03-03 06:22:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16920 gst/typefind/gsttypefindfunctions.c: Add typefinder for IMelody files, using audio/x-imelody.
16921 Original commit message from CVS:
16922 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
16923 Add typefinder for IMelody files, using audio/x-imelody.
16926 2008-03-03 06:04:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16928 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
16929 Original commit message from CVS:
16930 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
16931 * ext/alsa/gstalsasink.c: (set_hwparams):
16932 * ext/alsa/gstalsasrc.c: (set_hwparams):
16933 * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
16934 * ext/ogg/gstoggmux.h:
16935 * ext/ogg/gstogmparse.c:
16936 * gst-libs/gst/audio/audio.c:
16937 * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
16938 * gst-libs/gst/pbutils/missing-plugins.c:
16939 (gst_missing_uri_sink_message_new),
16940 (gst_missing_element_message_new),
16941 (gst_missing_decoder_message_new),
16942 (gst_missing_encoder_message_new):
16943 * gst-libs/gst/rtp/gstbasertppayload.c:
16944 * gst-libs/gst/rtp/gstrtcpbuffer.c:
16945 (gst_rtcp_packet_bye_get_reason):
16946 * gst/audioconvert/gstaudioconvert.c:
16947 * gst/audioresample/gstaudioresample.c:
16948 * gst/ffmpegcolorspace/imgconvert.c:
16949 * gst/playback/test.c: (gen_video_element), (gen_audio_element):
16950 * gst/typefind/gsttypefindfunctions.c:
16951 * gst/videoscale/vs_4tap.c:
16952 * gst/videoscale/vs_4tap.h:
16953 * sys/v4l/gstv4lelement.c:
16954 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
16955 * sys/v4l/v4l_calls.c:
16956 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
16957 (gst_v4lsrc_try_capture):
16958 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
16959 (gst_ximagesink_ximage_new):
16960 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
16961 (gst_xvimagesink_xvimage_new):
16962 * tests/check/elements/audioconvert.c:
16963 * tests/check/elements/audioresample.c:
16964 (fail_unless_perfect_stream):
16965 * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
16966 * tests/check/elements/decodebin.c:
16967 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
16968 (setup_gdpdepay_streamheader):
16969 * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
16970 (setup_gdppay_streamheader):
16971 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
16972 * tests/check/elements/multifdsink.c: (setup_multifdsink):
16973 * tests/check/elements/textoverlay.c:
16974 * tests/check/elements/videorate.c: (setup_videorate):
16975 * tests/check/elements/videotestsrc.c: (setup_videotestsrc):
16976 * tests/check/elements/volume.c: (setup_volume):
16977 * tests/check/elements/vorbisdec.c: (setup_vorbisdec):
16978 * tests/check/elements/vorbistag.c:
16979 * tests/check/generic/clock-selection.c:
16980 * tests/check/generic/states.c: (setup), (teardown):
16981 * tests/check/libs/cddabasesrc.c:
16982 * tests/check/libs/video.c:
16983 * tests/check/pipelines/gio.c:
16984 * tests/check/pipelines/oggmux.c:
16985 * tests/check/pipelines/simple-launch-lines.c:
16986 (simple_launch_lines_suite):
16987 * tests/check/pipelines/streamheader.c:
16988 * tests/check/pipelines/theoraenc.c:
16989 * tests/check/pipelines/vorbisdec.c:
16990 * tests/check/pipelines/vorbisenc.c:
16991 * tests/examples/seek/scrubby.c:
16992 * tests/examples/seek/seek.c: (query_positions_elems),
16993 (query_positions_pads):
16994 * tests/icles/stress-xoverlay.c: (myclock):
16995 Correct all relevant warnings found by the sparse semantic code
16996 analyzer. This include marking several symbols static, using
16997 NULL instead of 0 for pointers and using "foo (void)" instead
16998 of "foo ()" for declarations.
16999 * win32/common/libgstrtp.def:
17000 Add gst_rtp_buffer_set_extension_data to the symbol definition file.
17002 2008-03-02 18:43:15 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
17004 gst/playback/gstplaybin2.c: Make the function signature of the _get_*_tags() functions match the signature of the vfu...
17005 Original commit message from CVS:
17006 Patch by: José Alburquerque <jaalburqu svn gnome org>
17007 * gst/playback/gstplaybin2.c:
17008 Make the function signature of the _get_*_tags() functions match
17009 the signature of the vfuncs they implement, ie. return a
17010 GstTagList rather than a GstStructure, which is more correct,
17011 even if one is typedef'ed to the other (#518940).
17013 2008-03-02 18:32:36 +0000 Tim-Philipp Müller <tim@centricular.net>
17015 gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
17016 Original commit message from CVS:
17017 * gst-libs/gst/rtsp/gstrtspconnection.c:
17018 Don't include unix headers unconditionally (fixes #518037).
17020 2008-03-02 18:24:37 +0000 Tim-Philipp Müller <tim@centricular.net>
17022 tests/check/libs/video.c: Add unit test that makes sure that the strides, offsets and sizes returned for the various ...
17023 Original commit message from CVS:
17024 * tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
17025 (fourcc_list_struct), (fourcc_list), (fourcc_get_size),
17026 (paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
17027 (paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
17028 (paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
17029 (paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
17030 (gst_video_format_is_packed), (video_format_is_packed):
17031 Add unit test that makes sure that the strides, offsets and
17032 sizes returned for the various YUV formats by the new video API
17033 match the old reference implementation in videotestsrc.
17035 2008-03-02 18:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
17037 gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
17038 Original commit message from CVS:
17039 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
17040 (gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
17041 (gst_video_format_is_rgb), (gst_video_format_is_yuv),
17042 (gst_video_format_has_alpha), (gst_video_format_get_row_stride),
17043 (gst_video_format_get_pixel_stride),
17044 (gst_video_format_get_component_width),
17045 (gst_video_format_get_component_height),
17046 (gst_video_format_get_component_offset), (gst_video_format_get_size):
17047 * gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
17048 (GST_VIDEO_FORMAT_Y42B):
17049 API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
17051 2008-03-02 18:07:10 +0000 Tim-Philipp Müller <tim@centricular.net>
17053 gst-libs/gst/video/video.c: YV12 is I420 with swapped components 1 and 2, so the offset of component 1 for I420 shoul...
17054 Original commit message from CVS:
17055 * gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
17056 YV12 is I420 with swapped components 1 and 2, so the offset of
17057 component 1 for I420 should be the offset for component 2 for YV12
17060 2008-02-29 21:48:00 +0000 Rene Stadler <mail@renestadler.de>
17062 sys/v4l/gstv4lelement.c: Add missing semicolon to fix indentation.
17063 Original commit message from CVS:
17064 * sys/v4l/gstv4lelement.c:
17065 Add missing semicolon to fix indentation.
17067 2008-02-29 18:44:36 +0000 Julien Moutte <julien@moutte.net>
17069 ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
17070 Original commit message from CVS:
17071 2008-02-29 Julien Moutte <julien@fluendo.com>
17072 * ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
17073 (gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
17075 if we can do SPDIF output.
17076 * ext/alsa/gstalsa.h:
17077 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
17078 (gst_alsasink_prepare), (gst_alsasink_close),
17079 (gst_alsasink_write):
17080 * ext/alsa/gstalsasink.h: Initial support for SPDIF.
17081 * gst-libs/gst/audio/gstringbuffer.c:
17082 (gst_ring_buffer_parse_caps):
17083 * gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
17085 to support AC3, EC3 and IEC958 buffers.
17087 2008-02-29 17:59:16 +0000 Tim-Philipp Müller <tim@centricular.net>
17089 gst-libs/gst/interfaces/mixer.c: De-cruft and fix message type assertions (NULL is not a really valid mixer message t...
17090 Original commit message from CVS:
17091 * gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
17092 (gst_mixer_message_parse_mute_toggled),
17093 (gst_mixer_message_parse_record_toggled),
17094 (gst_mixer_message_parse_volume_changed),
17095 (gst_mixer_message_parse_option_changed):
17096 De-cruft and fix message type assertions (NULL is not a really
17097 valid mixer message type string).
17099 2008-02-29 14:52:02 +0000 Wim Taymans <wim.taymans@gmail.com>
17101 ext/libvisual/visual.c: When negotiating, actually start from a format that we can support instead of from the too ge...
17102 Original commit message from CVS:
17103 * ext/libvisual/visual.c: (gst_vis_src_negotiate):
17104 When negotiating, actually start from a format that we can support
17105 instead of from the too generic template.
17107 2008-02-29 12:26:48 +0000 Wim Taymans <wim.taymans@gmail.com>
17109 gst/playback/gstplaybin2.c: Enable vis setting.
17110 Original commit message from CVS:
17111 * gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
17112 Enable vis setting.
17113 * gst/playback/gstplaysink.c: (gst_play_sink_init),
17114 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
17115 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
17117 Implement vis switching while playing.
17119 2008-02-29 00:04:57 +0000 David Schleef <ds@schleef.org>
17121 gst-libs/gst/riff/riff-media.c: Add Dirac mapping
17122 Original commit message from CVS:
17123 * gst-libs/gst/riff/riff-media.c: Add Dirac mapping
17125 2008-02-28 10:54:14 +0000 Peter Kjellerstedt <pkj@axis.com>
17127 gst/tcp/: Removed fdset and stress test, they are now known as GstPoll in core.
17128 Original commit message from CVS:
17129 Patch by: Peter Kjellerstedt <pkj at axis com>
17130 * gst/tcp/Makefile.am:
17131 * gst/tcp/fdsetstress.c:
17132 * gst/tcp/gstfdset.c:
17133 * gst/tcp/gstfdset.h:
17134 Removed fdset and stress test, they are now known as GstPoll in
17136 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
17137 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
17138 (gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
17139 (gst_multi_fd_sink_handle_client_write),
17140 (gst_multi_fd_sink_queue_buffer),
17141 (gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
17142 (gst_multi_fd_sink_stop):
17143 * gst/tcp/gstmultifdsink.h:
17144 * gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
17145 (gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
17146 (gst_tcp_gdp_read_caps):
17147 * gst/tcp/gsttcp.h:
17148 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
17149 (gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
17150 (gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
17151 * gst/tcp/gsttcpclientsink.h:
17152 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
17153 (gst_tcp_client_src_create), (gst_tcp_client_src_start),
17154 (gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
17155 * gst/tcp/gsttcpclientsrc.h:
17156 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
17157 (gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
17158 * gst/tcp/gsttcpserversink.h:
17159 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
17160 (gst_tcp_server_src_create), (gst_tcp_server_src_start),
17161 (gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
17162 * gst/tcp/gsttcpserversrc.h:
17163 Port to GstPoll. See #505417.
17165 2008-02-28 09:54:14 +0000 Wim Taymans <wim.taymans@gmail.com>
17168 Patch Changelog a bit to give credit and refer to the relevant bug.
17169 Original commit message from CVS:
17170 Patch Changelog a bit to give credit and refer to the
17173 2008-02-28 09:50:52 +0000 Wim Taymans <wim.taymans@gmail.com>
17175 gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection.
17176 Original commit message from CVS:
17177 * gst-libs/gst/rtsp/gstrtspconnection.c:
17178 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
17179 (gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
17180 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
17181 (gst_rtsp_connection_free), (gst_rtsp_connection_poll),
17182 (gst_rtsp_connection_flush):
17183 * gst-libs/gst/rtsp/gstrtspconnection.h:
17184 Use GstPoll for the rtsp connection.
17186 2008-02-27 12:19:31 +0000 Wim Taymans <wim.taymans@gmail.com>
17188 tests/examples/seek/seek.c: Add combo box for visualisations, populate it with a factory list of all visualisation pl...
17189 Original commit message from CVS:
17190 * tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
17191 (init_visualization_features), (vis_combo_cb), (shot_cb), (main):
17192 Add combo box for visualisations, populate it with a factory list
17193 of all visualisation plugins, configure vis plugin instance in
17196 2008-02-27 10:55:03 +0000 Wim Taymans <wim.taymans@gmail.com>
17198 tests/check/libs/rtp.c: Add check for RTP buffer defaults, padding and marker bit API.
17199 Original commit message from CVS:
17200 * tests/check/libs/rtp.c: (GST_START_TEST):
17201 Add check for RTP buffer defaults, padding and marker bit API.
17203 2008-02-27 10:42:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17205 gst-libs/gst/cdda/sha1.c: Use memcpy() instead of upcasting a byte array to long *. This fixes an unaligned memory ac...
17206 Original commit message from CVS:
17207 * gst-libs/gst/cdda/sha1.c: (sha_transform):
17208 Use memcpy() instead of upcasting a byte array to long *. This
17209 fixes an unaligned memory access, resulting in SIGBUS on IA64.
17210 This should be ported to GCheckSum once we can use GLib 2.16.
17211 Partially fixes bug #500833.
17213 2008-02-27 10:23:27 +0000 Tim-Philipp Müller <tim@centricular.net>
17215 gst-libs/gst/tag/gsttagdemux.c: Push tag event after the newsegment event. Log the pointer of the buffer we're actual...
17216 Original commit message from CVS:
17217 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
17218 Push tag event after the newsegment event. Log the pointer of
17219 the buffer we're actually going to push rather than the buffer
17220 we're feeding to _make_metadata_writable().
17222 2008-02-25 07:21:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17224 gst/typefind/gsttypefindfunctions.c: Comment smoke typefinder for now. The smokedec plugin needs one frame per buffer...
17225 Original commit message from CVS:
17226 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
17227 Comment smoke typefinder for now. The smokedec plugin needs one
17228 frame per buffer but we have no parser yet, thus it simply crashes
17229 in most situations.
17231 2008-02-25 06:48:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17233 gst/typefind/gsttypefindfunctions.c: Add typefinder for the smoke video codec. Copied from the jpeg plugin.
17234 Original commit message from CVS:
17235 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
17236 Add typefinder for the smoke video codec. Copied from the jpeg plugin.
17238 2008-02-25 06:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17240 gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
17241 Original commit message from CVS:
17242 * gst/typefind/gsttypefindfunctions.c: (mid_type_find),
17244 Add midi typefinder, copied from the timidity plugin.
17246 2008-02-23 09:51:26 +0000 Tomasz Sałaciński <tsalacinski@gmail.com>
17248 Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
17249 Original commit message from CVS:
17250 Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
17251 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
17252 * tests/check/elements/subparse.c: (test_microdvd_with_italics),
17254 Forward slashes at the beginning and end of a line also signify
17255 italics (Fixes: #518162).
17257 2008-02-22 06:38:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17259 tests/check/gst-plugins-base.supp: Add a suppression for a cached value in GIO that wasn't moved while moving gio fro...
17260 Original commit message from CVS:
17261 * tests/check/gst-plugins-base.supp:
17262 Add a suppression for a cached value in GIO that wasn't moved
17263 while moving gio from -bad to -base.
17265 2008-02-22 05:27:24 +0000 Brian Cameron <brian.cameron@sun.com>
17267 configure.ac: Don't hardcode -Wall and -Werror for configure checks, this fails with non-GCC compilers. Fixes bug #51...
17268 Original commit message from CVS:
17269 Patch by: Brian Cameron <brian dot cameron at sun dot com>
17271 Don't hardcode -Wall and -Werror for configure checks, this fails
17272 with non-GCC compilers. Fixes bug #517991.
17274 2008-02-21 08:05:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17276 gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
17277 Original commit message from CVS:
17278 * gst/audiotestsrc/gstaudiotestsrc.c:
17279 Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
17281 2008-02-20 15:37:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17283 ext/gnomevfs/gstgnomevfssink.c: Return FALSE when seeking for a new segment fails instead of silently ignoring the fa...
17284 Original commit message from CVS:
17285 * ext/gnomevfs/gstgnomevfssink.c:
17286 (gst_gnome_vfs_sink_handle_event):
17287 Return FALSE when seeking for a new segment fails instead
17288 of silently ignoring the failure and appending every buffer
17289 that comes for the new segment.
17291 2008-02-20 11:52:28 +0000 Wim Taymans <wim.taymans@gmail.com>
17293 gst/playback/gstplaysink.c: Recursively search the sink element for a last-frame property so that we can also find th...
17294 Original commit message from CVS:
17295 * gst/playback/gstplaysink.c: (find_property),
17296 (gst_play_sink_find_property), (gen_video_chain),
17297 (gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
17298 Recursively search the sink element for a last-frame property so that we
17299 can also find the property in autovideosink and friends that don't
17300 always proxy the internal sink properties.
17302 2008-02-19 20:42:09 +0000 Tim-Philipp Müller <tim@centricular.net>
17304 gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
17305 Original commit message from CVS:
17306 * gst-libs/gst/audio/multichannel.c:
17307 (GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
17308 (gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
17309 (gst_audio_set_structure_channel_positions_list),
17310 (add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
17311 (gst_audio_fixate_channel_positions):
17312 Fix confusing terminology in docs and code: structure fields are
17313 'fields' and not 'properties'.
17315 2008-02-19 20:36:58 +0000 Tim-Philipp Müller <tim@centricular.net>
17317 gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
17318 Original commit message from CVS:
17319 * gst-libs/gst/audio/multichannel.c:
17320 (gst_audio_check_channel_positions), (add_list_to_struct):
17321 Give more useful warning messages if one of the channel
17322 layout enums passed to us is invalid and if the "channels"
17323 field in the caps has a GType we don't expect.
17325 2008-02-19 20:22:09 +0000 Tim-Philipp Müller <tim@centricular.net>
17327 gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
17328 Original commit message from CVS:
17329 * gst-libs/gst/audio/multichannel.c:
17330 Fix typo in docs blurb.
17332 2008-02-19 16:16:55 +0000 Josep Torra Valles <josep@fluendo.com>
17334 gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS typefind lookup to fix typefinding on HD clips.
17335 Original commit message from CVS:
17336 2008-02-19 Julien Moutte <julien@fluendo.com>
17337 Patch by: Josep Torra Valles <josep@fluendo.com>
17338 * gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
17339 typefind lookup to fix typefinding on HD clips.
17341 2008-02-19 15:50:37 +0000 Tim-Philipp Müller <tim@centricular.net>
17343 gst/playback/gstscreenshot.*: Fix up copyright (I rewrote the GStreamer-0.10 code for this from scratch back in the d...
17344 Original commit message from CVS:
17345 * gst/playback/gstscreenshot.c:
17346 * gst/playback/gstscreenshot.h:
17347 Fix up copyright (I rewrote the GStreamer-0.10 code for
17348 this from scratch back in the days).
17350 2008-02-19 15:02:33 +0000 Wim Taymans <wim.taymans@gmail.com>
17352 gst/playback/: Add screenshot conversion code from totem.
17353 Original commit message from CVS:
17354 * gst/playback/Makefile.am:
17355 * gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
17356 (create_element), (gst_play_frame_conv_convert):
17357 * gst/playback/gstscreenshot.h:
17358 Add screenshot conversion code from totem.
17359 * gst/playback/gstplay-marshal.list:
17360 * gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
17361 (gst_play_bin_class_init), (gst_play_bin_convert_frame),
17362 (gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
17363 Implement frame property to get a color-unconverted snapshot.
17364 Implement convert-frame action signal to get a converted snapshot image.
17365 Configure connection speed in uridecodebin.
17366 Document some more properties.
17367 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
17368 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
17369 (gst_play_sink_get_last_frame):
17370 * gst/playback/gstplaysink.h:
17371 Use last-buffer property of the video sink to get a video snapshot.
17372 * tests/examples/seek/seek.c: (shot_cb), (main):
17373 Add snapshot button for playbin2 and use the frame property to save the
17374 frame as a png in the current directory.
17376 2008-02-19 11:45:56 +0000 Josep Torra Valles <josep@fluendo.com>
17378 gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
17379 Original commit message from CVS:
17380 Patch by: Josep Torra Valles <josep at fluendo dot com>
17381 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
17383 Add typefinding support for h264 elementary streams.
17386 2008-02-18 13:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17388 configure.ac: Require CVS of core for new API in collectpads.
17389 Original commit message from CVS:
17391 Require CVS of core for new API in collectpads.
17392 * gst/adder/gstadder.c:
17393 Use new API to make adder sparse stream aware.
17395 2008-02-18 11:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
17397 gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
17398 Original commit message from CVS:
17399 * gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
17401 Get the object data correct so that we can remove our channels
17403 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
17404 (gen_vis_chain), (gst_play_sink_reconfigure),
17405 (gst_play_sink_request_pad):
17406 Add option to disable async behaviour in the sinks when possible. This
17407 makes it possible to avoid an audio queue when dealing with
17409 Add option to add a queue for the audio path.
17410 * tests/examples/seek/seek.c: (clear_streams), (update_streams),
17412 Disable the vis checkbox to match the defaults of playbin2.
17413 Only get the stream info when we need to.
17415 2008-02-17 05:15:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17417 ext/gio/: Don't use async operations as they require a running main loop.
17418 Original commit message from CVS:
17419 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
17420 (gst_gio_base_sink_set_stream):
17421 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
17422 (gst_gio_base_src_set_stream):
17423 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
17424 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
17425 Don't use async operations as they require a running main loop.
17426 This makes us block again when closing streams and unable
17427 to mount the enclosing volume of an URI if it isn't yet.
17429 2008-02-15 18:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
17431 gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
17432 Original commit message from CVS:
17433 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
17434 (gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
17435 (gen_vis_chain), (gst_play_sink_reconfigure),
17436 (gst_play_sink_request_pad):
17437 Move tee in front of the audio and vis pipelines.
17438 Add queue for audio for now.
17439 Add visualisation support.
17440 * tests/examples/seek/seek.c: (main):
17441 Visualisation is by default disabled.
17443 2008-02-15 11:58:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17445 ext/gio/: Improve debugging a bit.
17446 Original commit message from CVS:
17447 * ext/gio/gstgiobasesink.c: (close_stream_cb):
17448 * ext/gio/gstgiobasesrc.c: (close_stream_cb):
17449 Improve debugging a bit.
17450 * ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
17451 * ext/gio/gstgiosink.h:
17452 * ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
17453 * ext/gio/gstgiosrc.h:
17454 Try to mount the enclosing volume of a GFile if it isn't mounted
17455 yet. This requires us to wait for an async operation to finish, done
17456 with an nested GMainLoop. Authentication is not supported yet, will
17459 2008-02-14 18:24:42 +0000 Wim Taymans <wim.taymans@gmail.com>
17461 gst/playback/: Add mute property.
17462 Original commit message from CVS:
17463 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
17464 (gst_play_bin_set_property), (gst_play_bin_get_property),
17465 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
17466 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
17467 (gst_play_sink_get_mute), (gen_audio_chain):
17468 * gst/playback/gstplaysink.h:
17470 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
17471 (gst_selector_pad_chain):
17472 * gst/playback/gststreamselector.h:
17473 Make sure we forward the event only once.
17474 * tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
17475 Add and implement the mute button for playbin2.
17477 2008-02-13 14:34:55 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
17479 ext/alsa/gstalsasink.c: Add some more debug info.
17480 Original commit message from CVS:
17481 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
17482 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
17483 Add some more debug info.
17484 Make sure we never return a negative delay. Fixes #516246.
17486 2008-02-12 20:09:07 +0000 Tim-Philipp Müller <tim@centricular.net>
17488 ext/alsa/gstalsasink.c: Revert patch that makes the sink hold the object lock when calling snd_pcm_delay(), since it ...
17489 Original commit message from CVS:
17490 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
17491 Revert patch that makes the sink hold the object lock when
17492 calling snd_pcm_delay(), since it breaks playback for me.
17494 2008-02-12 19:50:36 +0000 Julien Moutte <julien@moutte.net>
17496 tests/examples/seek/seek.c: Add some seek flags when changing rate.
17497 Original commit message from CVS:
17498 2008-02-12 Julien Moutte <julien@fluendo.com>
17499 * tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
17500 some seek flags when changing rate.
17502 2008-02-12 14:51:26 +0000 Wim Taymans <wim.taymans@gmail.com>
17504 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
17505 Original commit message from CVS:
17506 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
17507 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
17508 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
17509 Fix potential leaks.
17510 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
17511 Fix leak when there is no function configured.
17513 2008-02-12 11:36:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17515 sys/v4l/v4lsrc_calls.c: Correctly chain up the finalize method.
17516 Original commit message from CVS:
17517 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
17518 (gst_v4lsrc_buffer_finalize):
17519 Correctly chain up the finalize method.
17521 2008-02-12 09:24:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17523 ext/gio/: Add documentation and example code for giostreamsink/giostreamsrc.
17524 Original commit message from CVS:
17525 * ext/gio/gstgiostreamsink.c:
17526 * ext/gio/gstgiostreamsrc.c:
17527 Add documentation and example code for giostreamsink/giostreamsrc.
17528 * tests/check/pipelines/gio.c: (GST_START_TEST):
17529 Ask the GMemoryOutputStream for the data instead of assuming that
17530 the pointer to the data stayed the same. It could've been realloc'ed.
17532 2008-02-12 08:55:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17534 ext/gio/: Make the documentation of giosink/giosrc complete, large parts are based on the gnomevfssink/gnomevfssrc docs.
17535 Original commit message from CVS:
17536 * ext/gio/gstgiosink.c:
17537 * ext/gio/gstgiosrc.c:
17538 Make the documentation of giosink/giosrc complete, large parts
17539 are based on the gnomevfssink/gnomevfssrc docs.
17541 2008-02-12 08:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17543 docs/plugins/: Add the GIO documentation again and while at that run make update.
17544 Original commit message from CVS:
17545 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
17546 * docs/plugins/gst-plugins-base-plugins-sections.txt:
17547 * docs/plugins/gst-plugins-base-plugins.args:
17548 * docs/plugins/gst-plugins-base-plugins.hierarchy:
17549 * docs/plugins/gst-plugins-base-plugins.interfaces:
17550 * docs/plugins/gst-plugins-base-plugins.prerequisites:
17551 * docs/plugins/gst-plugins-base-plugins.signals:
17552 * docs/plugins/inspect/plugin-adder.xml:
17553 * docs/plugins/inspect/plugin-audioconvert.xml:
17554 * docs/plugins/inspect/plugin-audiorate.xml:
17555 * docs/plugins/inspect/plugin-audioresample.xml:
17556 * docs/plugins/inspect/plugin-decodebin.xml:
17557 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
17558 * docs/plugins/inspect/plugin-gdp.xml:
17559 * docs/plugins/inspect/plugin-gio.xml:
17560 * docs/plugins/inspect/plugin-gnomevfs.xml:
17561 * docs/plugins/inspect/plugin-libvisual.xml:
17562 * docs/plugins/inspect/plugin-ogg.xml:
17563 * docs/plugins/inspect/plugin-pango.xml:
17564 * docs/plugins/inspect/plugin-playback.xml:
17565 * docs/plugins/inspect/plugin-queue2.xml:
17566 * docs/plugins/inspect/plugin-subparse.xml:
17567 * docs/plugins/inspect/plugin-theora.xml:
17568 * docs/plugins/inspect/plugin-uridecodebin.xml:
17569 * docs/plugins/inspect/plugin-videorate.xml:
17570 * docs/plugins/inspect/plugin-videoscale.xml:
17571 * docs/plugins/inspect/plugin-volume.xml:
17572 * docs/plugins/inspect/plugin-vorbis.xml:
17573 Add the GIO documentation again and while at that run make update.
17575 2008-02-11 20:23:44 +0000 Tim-Philipp Müller <tim@centricular.net>
17577 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...
17578 Original commit message from CVS:
17579 * ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
17580 * ext/alsa/gstalsasink.c: (set_swparams):
17581 * ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
17582 Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
17583 against libasound >= 1.0.16, since it's been deprecated in
17584 0.10.16, and alignment is always 1 then, apparently. (#512899)
17586 2008-02-11 18:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
17588 gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
17589 Original commit message from CVS:
17590 * gst/playback/gstplaybin.c: (gen_audio_element):
17591 * gst/playback/gstplaysink.c: (gen_audio_chain):
17592 Handle case where we can't create the volume element a bit
17595 2008-02-11 18:02:13 +0000 Tim-Philipp Müller <tim@centricular.net>
17597 ext/gnomevfs/: Add support for https protocol. Fixes #510229.
17598 Original commit message from CVS:
17599 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
17600 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
17601 Add support for https protocol. Fixes #510229.
17603 2008-02-11 17:03:18 +0000 Alan Peevers <peeves@pacbell.net>
17605 ext/alsa/gstalsasink.c: Take appropriate lock when calling alsa methods.
17606 Original commit message from CVS:
17607 2008-02-11 Julien Moutte <julien@fluendo.com>
17608 Patch by: Alan Peevers <peeves@pacbell.net>
17609 * ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
17610 lock when calling alsa methods.
17612 2008-02-11 13:03:13 +0000 Tim-Philipp Müller <tim@centricular.net>
17614 gst/typefind/gsttypefindfunctions.c: Bump rank of jpeg and png typefinders, which will return maximum probability in ...
17615 Original commit message from CVS:
17616 * gst/typefind/gsttypefindfunctions.c:
17617 Bump rank of jpeg and png typefinders, which will return maximum
17618 probability in the most common cases (thus short-circuiting more
17619 expensive typefinders like the mp3 one for these two quite common
17622 2008-02-11 09:48:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17624 ext/theora/theoraparse.c: Fix long description of the theora parser to be more verbose than just the type name.
17625 Original commit message from CVS:
17626 * ext/theora/theoraparse.c:
17627 Fix long description of the theora parser to be more verbose than just
17630 2008-02-11 06:47:50 +0000 Branko Čibej <brane@xbc.nu>
17632 sys/xvimage/xvimagesink.c: Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
17633 Original commit message from CVS:
17634 Patch by: Branko Čibej <brane at xbc dot nu>
17635 * sys/xvimage/xvimagesink.c:
17636 Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
17639 2008-02-09 10:41:36 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
17641 gst/playback/gstplaybasebin.c: Set is_dynamic as True if there are elements with both request and sometimes src pad t...
17642 Original commit message from CVS:
17643 * gst/playback/gstplaybasebin.c:
17644 Set is_dynamic as True if there are elements with both request
17645 and sometimes src pad templates instead of breaking out when it
17646 finds the first pad template that is a src.
17648 2008-02-08 18:17:51 +0000 Wim Taymans <wim.taymans@gmail.com>
17650 tests/examples/seek/seek.c: Add some stream switching and volume gui for playbin2.
17651 Original commit message from CVS:
17652 * tests/examples/seek/seek.c: (stop_cb), (clear_streams),
17653 (update_streams), (video_combo_cb), (audio_combo_cb),
17654 (text_combo_cb), (volume_spinbutton_changed_cb), (main):
17655 Add some stream switching and volume gui for playbin2.
17657 2008-02-08 17:47:37 +0000 Wim Taymans <wim.taymans@gmail.com>
17659 gst/playback/gstplay-marshal.list: Added marshal for streamselector Tags.
17660 Original commit message from CVS:
17661 * gst/playback/gstplay-marshal.list:
17662 Added marshal for streamselector Tags.
17663 * gst/playback/gstplaybasebin.c: (set_active_source):
17664 Streamselector now selects pads based on the pad object instead of its
17666 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
17667 (init_group), (gst_play_bin_init), (get_group), (get_tags),
17668 (gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
17669 (gst_play_bin_get_text_tags),
17670 (gst_play_bin_set_current_video_stream),
17671 (gst_play_bin_set_current_audio_stream),
17672 (gst_play_bin_set_current_text_stream),
17673 (gst_play_bin_set_property), (gst_play_bin_get_property),
17674 (pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
17675 Remove option to mute streams with the current-a/v/t property, we have
17676 this functionality in the flags.
17677 Add signals to notify when the number of A/V/T channels changed.
17678 Add action signals to get tags for the A/V/T streams.
17679 Implement setting the current A/V/T stream.
17680 Rearrange some things to simplify stream selection.
17682 * gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
17683 (gst_play_sink_get_volume), (gst_play_sink_set_property),
17684 (gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
17685 (activate_vis), (gst_play_sink_reconfigure):
17686 * gst/playback/gstplaysink.h:
17687 Add and implement volume setting methods.
17688 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
17689 (gst_selector_pad_finalize), (gst_selector_pad_get_property),
17690 (gst_selector_pad_event), (gst_stream_selector_class_init),
17691 (gst_stream_selector_init), (gst_stream_selector_finalize),
17692 (gst_stream_selector_set_property),
17693 (gst_stream_selector_get_property),
17694 (gst_stream_selector_get_linked_pad),
17695 (gst_stream_selector_request_new_pad):
17696 * gst/playback/gststreamselector.h:
17697 Add pad properties for tags and status of pads.
17699 Make active pad selection based on pad object instead of name.
17701 2008-02-08 16:10:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17703 configure.ac: Revert last change as we now check in gtk-doc.m4 for sed.
17704 Original commit message from CVS:
17706 Revert last change as we now check in gtk-doc.m4 for sed.
17708 2008-02-08 14:54:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17710 configure.ac: Find and subst SED when building the docs.
17711 Original commit message from CVS:
17713 Find and subst SED when building the docs.
17715 2008-02-08 14:34:41 +0000 Julien Moutte <julien@moutte.net>
17717 tests/examples/seek/seek.c: Make sure bus signals are reconnected when pressing STOP and then PLAY again for a parse ...
17718 Original commit message from CVS:
17719 2008-02-08 Julien Moutte <julien@fluendo.com>
17720 * tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
17721 (main): Make sure bus signals are reconnected when pressing STOP
17722 and then PLAY again for a parse launch pipeline. Fix a ref leak
17724 * win32/common/config.h: Updated.
17726 2008-02-08 00:57:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17728 configure.ac: Make DISABLE_DEPRECATED defined *only* during CVS, not during pre-releases or releases.
17729 Original commit message from CVS:
17731 Make DISABLE_DEPRECATED defined *only* during CVS, not during
17732 pre-releases or releases.
17734 2008-02-08 00:45:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17736 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is reporting
17737 Original commit message from CVS:
17739 * ext/gio/Makefile.am:
17740 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
17743 2008-02-07 23:40:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17745 docs/plugins/Makefile.am: Add the headers which need scanning for the GIO plugin. The rest of the docs still need mig...
17746 Original commit message from CVS:
17747 * docs/plugins/Makefile.am:
17748 Add the headers which need scanning for the GIO plugin. The rest of
17749 the docs still need migrating.
17751 2008-02-07 23:22:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17753 Add gio in a few more places.
17754 Original commit message from CVS:
17756 * tests/check/Makefile.am:
17757 * tests/check/pipelines/.cvsignore:
17758 Add gio in a few more places.
17760 2008-02-07 23:18:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17762 Move gio plugin from -bad and mark as experimental.
17763 Original commit message from CVS:
17766 * tests/check/Makefile.am:
17767 Move gio plugin from -bad and mark as experimental.
17769 2008-02-07 22:39:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17771 gst-libs/gst/interfaces/: Comment out a couple of other things which break the build when
17772 Original commit message from CVS:
17773 * gst-libs/gst/interfaces/mixeroptions.c:
17774 * gst-libs/gst/interfaces/mixertrack.c:
17775 Comment out a couple of other things which break the build when
17776 GST_DISABLE_DEPRECATED isn't on but -Werror is.
17778 2008-02-07 18:28:29 +0000 Tim-Philipp Müller <tim@centricular.net>
17780 docs/libs/gst-plugins-base-libs-sections.txt: Fix pbutils header.
17781 Original commit message from CVS:
17782 * docs/libs/gst-plugins-base-libs-sections.txt:
17783 Fix pbutils header.
17785 2008-02-07 18:07:41 +0000 Christian Schaller <uraeus@gnome.org>
17787 * gst-plugins-base.spec.in:
17788 commit spec file update which includes all the split .pc files
17789 Original commit message from CVS:
17790 commit spec file update which includes all the split .pc files
17792 2008-02-07 12:17:49 +0000 Wim Taymans <wim.taymans@gmail.com>
17794 gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
17795 Original commit message from CVS:
17796 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
17797 Fix compiler warning.
17799 2008-02-07 11:00:45 +0000 Peter Kjellerstedt <pkj@axis.com>
17801 gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
17802 Original commit message from CVS:
17803 Patch by: Peter Kjellerstedt <pkj at axis com>
17804 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
17805 Clear the addrinfo struct using memset. Fixes #514937.
17807 2008-02-06 15:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
17809 gst/tcp/gstfdset.h: Remove unused field to same some memory.
17810 Original commit message from CVS:
17811 * gst/tcp/gstfdset.h:
17812 Remove unused field to same some memory.
17813 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
17814 Mark action signals as such.
17816 2008-02-06 13:35:58 +0000 Michael Smith <msmith@xiph.org>
17818 ext/theora/theoradec.c: Increment granulepos for new-bitstream versions appropriately.
17819 Original commit message from CVS:
17820 * ext/theora/theoradec.c: (_theora_granule_frame),
17822 Increment granulepos for new-bitstream versions appropriately.
17825 2008-02-04 11:51:31 +0000 Wim Taymans <wim.taymans@gmail.com>
17827 tests/examples/seek/seek.c: Remove obsolete stream_time reset after flushing seek, core does that automatically now.
17828 Original commit message from CVS:
17829 * tests/examples/seek/seek.c: (do_seek),
17830 (rate_spinbutton_changed_cb), (update_streams), (main):
17831 Remove obsolete stream_time reset after flushing seek, core does that
17833 Improve accuracy of speed spinbutton.
17834 Only do playbin2 stuff when we actually use it.
17836 2008-02-02 17:29:32 +0000 Tim-Philipp Müller <tim@centricular.net>
17838 tests/check/Makefile.am: Revert previous change of the test environment's GST_PLUGIN_PATH.
17839 Original commit message from CVS:
17840 * tests/check/Makefile.am:
17841 Revert previous change of the test environment's GST_PLUGIN_PATH.
17842 The problem is not with the plugins, but with element factories
17843 and only occurs if elements are split out from existing plugins
17844 or if plugins change name (see #512740).
17846 2008-02-02 15:32:23 +0000 Tim-Philipp Müller <tim@centricular.net>
17848 tests/check/Makefile.am: Fix the tests environment's GST_PLUGIN_PATH: we want the directory with the core's plugins f...
17849 Original commit message from CVS:
17850 * tests/check/Makefile.am:
17851 Fix the tests environment's GST_PLUGIN_PATH: we want the directory
17852 with the core's plugins first and our local build directories last,
17853 since we might be building against an installed core, and that
17854 core's plugin directory may contain older or other versions of
17855 our own -base plugins, but we really do want to test our local
17856 ones (if there are multiple plugins or element factories with the
17857 same name, those inspected last will trump those read in earlier).
17858 Fixes #512740 for the most part.
17860 2008-02-02 07:13:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17862 Use gmtime_r if available as gmtime is not MT-safe.
17863 Original commit message from CVS:
17865 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
17866 Use gmtime_r if available as gmtime is not MT-safe.
17869 2008-02-02 06:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17871 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
17872 Original commit message from CVS:
17873 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
17874 Cast glong to time_t as time_t might have a different type on
17875 other platforms, like FreeBSD, and we get a compiler warning
17876 otherwise. Fixes bug #511825.
17878 2008-02-01 16:44:21 +0000 Wim Taymans <wim.taymans@gmail.com>
17880 gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
17881 Original commit message from CVS:
17882 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
17883 (get_group), (get_n_pads), (gst_play_bin_get_property),
17884 (pad_added_cb), (no_more_pads_cb), (perform_eos),
17885 (autoplug_select_cb), (deactivate_group):
17886 Remove stream-info, we going for something easier.
17887 Refactor getting the current group.
17888 Implement getting the number of audio/video/text streams.
17889 * gst/playback/gststreamselector.c:
17890 (gst_stream_selector_class_init), (gst_stream_selector_init),
17891 (gst_stream_selector_get_property),
17892 (gst_stream_selector_request_new_pad),
17893 (gst_stream_selector_release_pad):
17894 * gst/playback/gststreamselector.h:
17895 Add property for number of pads.
17896 * tests/examples/seek/seek.c: (set_scale), (update_flag),
17897 (vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
17898 (text_toggle_cb), (update_streams), (msg_async_done),
17899 (msg_state_changed), (main):
17900 Block slider callback when updating the slider position.
17901 Add gui elements for controlling playbin2.
17902 Add callback for async_done that updates position/duration.
17904 2008-02-01 12:56:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17906 docs/plugins/: First round of plugin docs cleansups.
17907 Original commit message from CVS:
17908 * docs/plugins/Makefile.am:
17909 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
17910 * docs/plugins/gst-plugins-base-plugins-sections.txt:
17911 * docs/plugins/gst-plugins-base-plugins.hierarchy:
17912 * docs/plugins/gst-plugins-base-plugins.interfaces:
17913 * docs/plugins/gst-plugins-base-plugins.prerequisites:
17914 First round of plugin docs cleansups.
17915 * docs/plugins/inspect/plugin-adder.xml:
17916 * docs/plugins/inspect/plugin-alsa.xml:
17917 * docs/plugins/inspect/plugin-audioconvert.xml:
17918 * docs/plugins/inspect/plugin-audiorate.xml:
17919 * docs/plugins/inspect/plugin-audioresample.xml:
17920 * docs/plugins/inspect/plugin-audiotestsrc.xml:
17921 * docs/plugins/inspect/plugin-cdparanoia.xml:
17922 * docs/plugins/inspect/plugin-decodebin.xml:
17923 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
17924 * docs/plugins/inspect/plugin-gdp.xml:
17925 * docs/plugins/inspect/plugin-gnomevfs.xml:
17926 * docs/plugins/inspect/plugin-libvisual.xml:
17927 * docs/plugins/inspect/plugin-ogg.xml:
17928 * docs/plugins/inspect/plugin-pango.xml:
17929 * docs/plugins/inspect/plugin-subparse.xml:
17930 * docs/plugins/inspect/plugin-tcp.xml:
17931 * docs/plugins/inspect/plugin-theora.xml:
17932 * docs/plugins/inspect/plugin-typefindfunctions.xml:
17933 * docs/plugins/inspect/plugin-video4linux.xml:
17934 * docs/plugins/inspect/plugin-videorate.xml:
17935 * docs/plugins/inspect/plugin-videoscale.xml:
17936 * docs/plugins/inspect/plugin-videotestsrc.xml:
17937 * docs/plugins/inspect/plugin-volume.xml:
17938 * docs/plugins/inspect/plugin-vorbis.xml:
17939 * docs/plugins/inspect/plugin-ximagesink.xml:
17940 * docs/plugins/inspect/plugin-xvimagesink.xml:
17942 * ext/ogg/Makefile.am:
17943 * ext/ogg/gstoggmux.c:
17944 * ext/ogg/gstoggmux.h:
17945 Add header for oggmux. the c-file needs a doc blob still.
17947 2008-02-01 11:09:16 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
17949 Add gst_rtp_buffer_set_extension_data()
17950 Original commit message from CVS:
17951 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
17952 * gst-libs/gst/rtp/gstrtpbuffer.c:
17953 (gst_rtp_buffer_set_extension_data):
17954 * gst-libs/gst/rtp/gstrtpbuffer.h:
17955 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
17956 Add gst_rtp_buffer_set_extension_data()
17957 Add a unit test for this addition. Fixes #511478.
17958 API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
17960 2008-01-31 17:18:46 +0000 Wim Taymans <wim.taymans@gmail.com>
17962 gst-libs/gst/app/gstappsink.c: Really clean up the queue instead of just unreffing all buffers in it.
17963 Original commit message from CVS:
17964 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
17965 Really clean up the queue instead of just unreffing all buffers
17967 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
17968 (gst_app_src_class_init), (gst_app_src_init),
17969 (gst_app_src_dispose), (gst_app_src_finalize):
17970 Fix dispose/finalize.
17972 2008-01-30 15:34:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17974 ext/gio/: Use async variants of the close stream functions to prevent blocking for a long time there and add some mor...
17975 Original commit message from CVS:
17976 * ext/gio/gstgiobasesink.c: (close_stream_cb),
17977 (gst_gio_base_sink_stop), (gst_gio_base_sink_event),
17978 (gst_gio_base_sink_render), (gst_gio_base_sink_set_stream):
17979 * ext/gio/gstgiobasesrc.c: (close_stream_cb),
17980 (gst_gio_base_src_stop), (gst_gio_base_src_create),
17981 (gst_gio_base_src_set_stream):
17982 Use async variants of the close stream functions to prevent blocking
17983 for a long time there and add some more sanity checks for a correct
17986 2008-01-30 14:42:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17988 configure.ac: Back to CVS
17989 Original commit message from CVS:
17993 === release 0.10.17 ===
17995 2008-01-30 14:19:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18001 * docs/plugins/gst-plugins-base-plugins.hierarchy:
18002 * docs/plugins/inspect/plugin-adder.xml:
18003 * docs/plugins/inspect/plugin-alsa.xml:
18004 * docs/plugins/inspect/plugin-audioconvert.xml:
18005 * docs/plugins/inspect/plugin-audiorate.xml:
18006 * docs/plugins/inspect/plugin-audioresample.xml:
18007 * docs/plugins/inspect/plugin-audiotestsrc.xml:
18008 * docs/plugins/inspect/plugin-cdparanoia.xml:
18009 * docs/plugins/inspect/plugin-decodebin.xml:
18010 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
18011 * docs/plugins/inspect/plugin-gdp.xml:
18012 * docs/plugins/inspect/plugin-gnomevfs.xml:
18013 * docs/plugins/inspect/plugin-libvisual.xml:
18014 * docs/plugins/inspect/plugin-ogg.xml:
18015 * docs/plugins/inspect/plugin-pango.xml:
18016 * docs/plugins/inspect/plugin-subparse.xml:
18017 * docs/plugins/inspect/plugin-tcp.xml:
18018 * docs/plugins/inspect/plugin-theora.xml:
18019 * docs/plugins/inspect/plugin-typefindfunctions.xml:
18020 * docs/plugins/inspect/plugin-video4linux.xml:
18021 * docs/plugins/inspect/plugin-videorate.xml:
18022 * docs/plugins/inspect/plugin-videoscale.xml:
18023 * docs/plugins/inspect/plugin-videotestsrc.xml:
18024 * docs/plugins/inspect/plugin-volume.xml:
18025 * docs/plugins/inspect/plugin-vorbis.xml:
18026 * docs/plugins/inspect/plugin-ximagesink.xml:
18027 * docs/plugins/inspect/plugin-xvimagesink.xml:
18028 * gst-plugins-base.doap:
18029 * win32/common/config.h:
18031 Original commit message from CVS:
18034 2008-01-30 13:45:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18036 gst-libs/gst/interfaces/: Also remove the conditional registration of the signals that disappeared with the ABI chang...
18037 Original commit message from CVS:
18038 * gst-libs/gst/interfaces/mixeroptions.c:
18039 * gst-libs/gst/interfaces/mixertrack.c:
18040 Also remove the conditional registration of the signals
18041 that disappeared with the ABI change in 0.10.14
18043 2008-01-30 12:28:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18045 gst-libs/gst/rtsp/gstrtspconnection.c: Revert patch to gstrtspconnection.c for brown paper bag release of -base. Re-o...
18046 Original commit message from CVS:
18047 * gst-libs/gst/rtsp/gstrtspconnection.c:
18048 Revert patch to gstrtspconnection.c for brown paper bag
18049 release of -base. Re-opens: #511825
18051 2008-01-30 12:20:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18053 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
18054 Original commit message from CVS:
18055 * gst-libs/gst/interfaces/mixeroptions.h:
18056 * gst-libs/gst/interfaces/mixertrack.h:
18057 Change the way these deprecated function pointers are removed
18058 so that the compiled ABI is unconditionally smaller. This
18059 sets in stone an ABI break that actually occurred when the
18060 things were deprecated in 0.10.14, which seems to be the best
18061 fix as the only known users are oss-mixer and sunaudio-mixer in
18065 2008-01-30 12:19:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18067 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
18068 Original commit message from CVS:
18069 * gst-libs/gst/interfaces/mixeroptions.h:
18070 * gst-libs/gst/interfaces/mixertrack.h:
18071 Change the way these deprecated function pointers are removed
18072 so that the compiled ABI is unconditionally smaller. This
18073 sets in stone an ABI break that actually occurred when the
18074 things were deprecated in 0.10.14, which seems to be the best
18075 fix as the only known users are oss-mixer and sunaudio-mixer in
18078 2008-01-30 11:43:53 +0000 Tim-Philipp Müller <tim@centricular.net>
18080 win32/common/libgstpbutils.def: Export the two new _get_type() functions which are needed by the python bindings.
18081 Original commit message from CVS:
18082 * win32/common/libgstpbutils.def:
18083 Export the two new _get_type() functions which are needed
18084 by the python bindings.
18086 2008-01-29 09:59:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18088 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
18089 Original commit message from CVS:
18090 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
18091 Cast glong to time_t as time_t might have a different type on
18092 other platforms, like FreeBSD, and we get a compiler warning
18093 otherwise. Fixes bug #511825.
18095 2008-01-29 09:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18097 gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
18098 Original commit message from CVS:
18099 * gst-libs/gst/audio/gstaudiofilter.c:
18100 (gst_audio_filter_class_init):
18101 Initialize the GstRingerBuffer class to get it's debug category
18102 initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
18103 category and otherwise we get some g_critical(). Fixes bug #512334.
18105 2008-01-28 23:35:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18107 configure.ac: Back to CVS
18108 Original commit message from CVS:
18112 === release 0.10.16 ===
18114 2008-01-28 23:31:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18120 * docs/plugins/gst-plugins-base-plugins.args:
18121 * docs/plugins/gst-plugins-base-plugins.hierarchy:
18122 * docs/plugins/gst-plugins-base-plugins.interfaces:
18123 * docs/plugins/gst-plugins-base-plugins.prerequisites:
18124 * docs/plugins/gst-plugins-base-plugins.signals:
18125 * docs/plugins/inspect/plugin-adder.xml:
18126 * docs/plugins/inspect/plugin-alsa.xml:
18127 * docs/plugins/inspect/plugin-audioconvert.xml:
18128 * docs/plugins/inspect/plugin-audiorate.xml:
18129 * docs/plugins/inspect/plugin-audioresample.xml:
18130 * docs/plugins/inspect/plugin-audiotestsrc.xml:
18131 * docs/plugins/inspect/plugin-cdparanoia.xml:
18132 * docs/plugins/inspect/plugin-decodebin.xml:
18133 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
18134 * docs/plugins/inspect/plugin-gdp.xml:
18135 * docs/plugins/inspect/plugin-gnomevfs.xml:
18136 * docs/plugins/inspect/plugin-libvisual.xml:
18137 * docs/plugins/inspect/plugin-ogg.xml:
18138 * docs/plugins/inspect/plugin-pango.xml:
18139 * docs/plugins/inspect/plugin-subparse.xml:
18140 * docs/plugins/inspect/plugin-tcp.xml:
18141 * docs/plugins/inspect/plugin-theora.xml:
18142 * docs/plugins/inspect/plugin-typefindfunctions.xml:
18143 * docs/plugins/inspect/plugin-video4linux.xml:
18144 * docs/plugins/inspect/plugin-videorate.xml:
18145 * docs/plugins/inspect/plugin-videoscale.xml:
18146 * docs/plugins/inspect/plugin-videotestsrc.xml:
18147 * docs/plugins/inspect/plugin-volume.xml:
18148 * docs/plugins/inspect/plugin-vorbis.xml:
18149 * docs/plugins/inspect/plugin-ximagesink.xml:
18150 * docs/plugins/inspect/plugin-xvimagesink.xml:
18151 * gst-plugins-base.doap:
18152 * win32/common/config.h:
18154 Original commit message from CVS:
18157 2008-01-28 22:15:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18183 Original commit message from CVS:
18186 2008-01-22 15:37:49 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
18188 gst-libs/gst/rtp/gstrtpbuffer.c: Fix typos and wrong extension check. Fixes #511274.
18189 Original commit message from CVS:
18190 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
18191 * gst-libs/gst/rtp/gstrtpbuffer.c:
18192 (gst_rtp_buffer_get_extension_data):
18193 Fix typos and wrong extension check. Fixes #511274.
18195 2008-01-18 00:03:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18197 po/sk.po: Oops - add new sk.po mentioned in the LINGUAS I just committed
18198 Original commit message from CVS:
18200 Oops - add new sk.po mentioned in the LINGUAS I just committed
18202 2008-01-17 22:31:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18204 po/LINGUAS: Add ca translation to the disted list.
18205 Original commit message from CVS:
18207 Add ca translation to the disted list.
18208 * win32/vs6/libgstsdp.dsp:
18209 Convert line endings to CRLF
18211 2008-01-17 21:58:53 +0000 Sébastien Moutte <sebastien@moutte.net>
18213 win32/MANIFEST: Add win32/vs6/libgstrtsp.dsp to MANIFEST
18214 Original commit message from CVS:
18216 Add win32/vs6/libgstrtsp.dsp to MANIFEST
18218 2008-01-16 05:40:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18220 Update for API changes in GIO and require GIO 2.15.2 for this.
18221 Original commit message from CVS:
18223 * tests/check/pipelines/gio.c: (GST_START_TEST):
18224 Update for API changes in GIO and require GIO 2.15.2 for this.
18226 2008-01-14 22:20:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18228 win32/common/: Add new API declarations
18229 Original commit message from CVS:
18230 * win32/common/libgstsdp.def:
18231 * win32/common/libgstvideo.def:
18232 Add new API declarations
18234 2008-01-14 17:00:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18236 ext/theora/: Take a 2nd stab at handling libtheora granulepos changes in the decoder and parser by inspecting the bit...
18237 Original commit message from CVS:
18238 * ext/theora/gsttheoradec.h:
18239 * ext/theora/gsttheoraparse.h:
18240 * ext/theora/theoradec.c:
18241 * ext/theora/theoraparse.c:
18242 Take a 2nd stab at handling libtheora granulepos changes in the decoder
18243 and parser by inspecting the bitstream version of the incoming data.
18245 2008-01-14 13:11:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18247 Provide one pkg-config file for every gst-plugins-base library.
18248 Original commit message from CVS:
18250 * pkgconfig/Makefile.am:
18251 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
18252 * pkgconfig/gstreamer-audio.pc.in:
18253 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
18254 * pkgconfig/gstreamer-cdda.pc.in:
18255 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
18256 * pkgconfig/gstreamer-fft.pc.in:
18257 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
18258 * pkgconfig/gstreamer-floatcast.pc.in:
18259 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
18260 * pkgconfig/gstreamer-interfaces.pc.in:
18261 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
18262 * pkgconfig/gstreamer-netbuffer.pc.in:
18263 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
18264 * pkgconfig/gstreamer-pbutils.pc.in:
18265 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
18266 * pkgconfig/gstreamer-riff.pc.in:
18267 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
18268 * pkgconfig/gstreamer-rtp.pc.in:
18269 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
18270 * pkgconfig/gstreamer-rtsp.pc.in:
18271 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
18272 * pkgconfig/gstreamer-sdp.pc.in:
18273 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
18274 * pkgconfig/gstreamer-tag.pc.in:
18275 * pkgconfig/gstreamer-video-uninstalled.pc.in:
18276 * pkgconfig/gstreamer-video.pc.in:
18277 Provide one pkg-config file for every gst-plugins-base library.
18278 This makes linking to those libraries much more intuitive and
18279 provides standard pkg-config behaviour for them. Fixes bug #499697.
18281 2008-01-14 01:19:34 +0000 David Schleef <ds@schleef.org>
18283 gst/videoscale/vs_4tap.c: Fix valgrind error on 4tap scaling method.
18284 Original commit message from CVS:
18285 * gst/videoscale/vs_4tap.c:
18286 Fix valgrind error on 4tap scaling method.
18288 2008-01-13 21:40:45 +0000 Sébastien Moutte <sebastien@moutte.net>
18290 gst-libs/gst/sdp/gstsdpmessage.c: Include Winsock2.h for VS6 and use a different way initialize hints structure so it...
18291 Original commit message from CVS:
18292 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
18293 Include Winsock2.h for VS6 and use a different way initialize
18294 hints structure so it can build with VS6.
18296 * win32/vs6/libgstsdp.dsp:
18297 * win32/common/libgstsdp.def:
18298 Add new files for libgstsdp.
18299 * win32/vs6/grammar.dsp:
18300 Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
18301 * win32/vs6/gst_plugins_base.dsw:
18302 * win32/vs6/libgstdecodebin.dsp:
18303 * win32/vs6/libgstdecodebin2.dsp:
18304 * win32/vs6/libgstplaybin.dsp:
18305 * win32/vs6/libgstvolume.dsp:
18306 Add new dependencies to the link list.
18308 2008-01-13 17:24:49 +0000 Julien Moutte <julien@moutte.net>
18310 win32/common/: Update/Add generated files in the win32 build directory.
18311 Original commit message from CVS:
18312 2008-01-13 Julien Moutte <julien@fluendo.com>
18313 * win32/common/config.h:
18314 * win32/common/gstrtsp-enumtypes.c: (gst_rtsp_result_get_type),
18315 (gst_rtsp_event_get_type), (gst_rtsp_family_get_type),
18316 (gst_rtsp_state_get_type), (gst_rtsp_version_get_type),
18317 (gst_rtsp_method_get_type), (gst_rtsp_auth_method_get_type),
18318 (gst_rtsp_header_field_get_type),
18319 (gst_rtsp_status_code_get_type):
18320 * win32/common/interfaces-enumtypes.c:
18321 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
18322 (gst_mixer_message_type_get_type), (gst_mixer_flags_get_type),
18323 (gst_mixer_track_flags_get_type),
18324 (gst_tuner_channel_flags_get_type):
18325 * win32/common/multichannel-enumtypes.c:
18326 (gst_audio_channel_position_get_type):
18327 * win32/common/pbutils-enumtypes.c:
18328 (gst_install_plugins_return_get_type):
18329 * win32/common/pbutils-enumtypes.h: Update/Add generated files
18330 in the win32 build directory.
18332 2008-01-12 23:24:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18334 tests/check/Makefile.am: Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
18335 Original commit message from CVS:
18336 * tests/check/Makefile.am:
18337 Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
18338 * tests/check/elements/audiorate.c: (do_perfect_stream_test):
18339 * tests/check/elements/playbin.c:
18340 * tests/check/libs/mixer.c: (test_element_interface_supported),
18341 (gst_implements_interface_init):
18342 * tests/check/libs/rtp.c: (GST_START_TEST):
18343 Fix various assignment type mismatches.
18345 2008-01-12 23:08:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18347 Add test to see if hstrerror is available or if we need libresolv (Solaris) for it, then use it in libgstrtsp.
18348 Original commit message from CVS:
18350 * gst-libs/gst/rtsp/Makefile.am:
18351 Add test to see if hstrerror is available or if we need libresolv
18352 (Solaris) for it, then use it in libgstrtsp.
18354 2008-01-12 14:54:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18356 gst-libs/gst/tag/Makefile.am: Fix include path order
18357 Original commit message from CVS:
18358 * gst-libs/gst/tag/Makefile.am:
18359 Fix include path order
18361 2008-01-11 17:15:23 +0000 Tim-Philipp Müller <tim@centricular.net>
18363 * gst-libs/gst/pbutils/.gitignore:
18364 Ignore more and make buildbot happy
18365 Original commit message from CVS:
18366 Ignore more and make buildbot happy
18368 2008-01-11 16:18:10 +0000 Edward Hervey <bilboed@bilboed.com>
18370 gst-libs/gst/pbutils/install-plugins.*: Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping for bi...
18371 Original commit message from CVS:
18372 * gst-libs/gst/pbutils/install-plugins.c:
18373 (gst_install_plugins_context_copy),
18374 (gst_install_plugins_context_get_type):
18375 * gst-libs/gst/pbutils/install-plugins.h:
18376 Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
18379 2008-01-11 15:48:11 +0000 Michael Smith <msmith@xiph.org>
18381 ext/theora/theoradec.c: Adapt for post-alpha meaning of granulepos, when we have a newer version of libtheora.
18382 Original commit message from CVS:
18383 * ext/theora/theoradec.c: (gst_theora_dec_class_init),
18384 (_theora_granule_frame), (_theora_granule_start_time),
18385 (theora_dec_sink_convert), (theora_dec_decode_buffer):
18386 Adapt for post-alpha meaning of granulepos, when we
18387 have a newer version of libtheora.
18388 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
18389 (theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
18390 (theora_enc_is_discontinuous), (theora_enc_chain):
18392 * tests/check/Makefile.am:
18393 Link libtheora into theoraenc test so we can check which version of
18394 libtheora we're testing against.
18395 * tests/check/pipelines/theoraenc.c: (check_libtheora),
18396 (check_buffer_granulepos),
18397 (check_buffer_granulepos_from_starttime), (GST_START_TEST),
18399 Adapt tests to check the values that are now defined for theora; make
18400 the tests backwards-adapt the passed values if we're running against an
18404 2008-01-10 17:55:53 +0000 Tim-Philipp Müller <tim@centricular.net>
18406 gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
18407 Original commit message from CVS:
18408 * gst-libs/gst/audio/gstbaseaudiosink.c:
18409 (gst_base_audio_sink_class_init):
18410 * gst-libs/gst/audio/gstbaseaudiosrc.c:
18411 (gst_base_audio_src_class_init):
18412 Ref audio clock class from a thread-safe context to make sure
18413 we're not bit by GObjects lack of thread-safety here (#349410),
18414 however unlikely that may be in practice.
18416 2008-01-10 12:22:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18418 autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We...
18419 Original commit message from CVS:
18421 Add -Wno-portability to the automake parameters to stop warnings
18422 about GNU make extensions being used. We require GNU make in almost
18423 every Makefile anyway.
18425 Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
18426 at the same time is required for per target flags.
18428 2008-01-08 21:10:02 +0000 Tim-Philipp Müller <tim@centricular.net>
18430 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...
18431 Original commit message from CVS:
18432 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
18433 Post an error message if we can't pull as many bytes as we need
18434 for the tag. This makes sure the user gets to see a proper error
18435 message if a file with a partial ID3 tag is fed to decodebin, and
18436 not a 'no ID3 tag demuxer' error, which would be confusing
18439 2008-01-08 20:59:20 +0000 Tim-Philipp Müller <tim@centricular.net>
18441 gst-libs/gst/pbutils/descriptions.c: Add description strings for ID3, APE, and ICY tags.
18442 Original commit message from CVS:
18443 * gst-libs/gst/pbutils/descriptions.c: (formats):
18444 Add description strings for ID3, APE, and ICY tags.
18446 2008-01-08 20:48:00 +0000 Tim-Philipp Müller <tim@centricular.net>
18448 gst/playback/gstdecodebin.c: Make sure we error out correctly if we can't activate one of the elements we've added. ...
18449 Original commit message from CVS:
18450 * gst/playback/gstdecodebin.c: (try_to_link_1):
18451 Make sure we error out correctly if we can't activate one of
18452 the elements we've added. Fixes #508138.
18454 2008-01-07 13:59:43 +0000 Bastien Nocera <hadess@hadess.net>
18456 ext/alsa/gstalsamixer.c: Use snd_mixer_selem_set_{playback|capture}_volume_all() if the volume is the same for all ch...
18457 Original commit message from CVS:
18458 Patch by: Bastien Nocera <hadess at hadess net>
18459 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
18460 (check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
18461 Use snd_mixer_selem_set_{playback|capture}_volume_all() if
18462 the volume is the same for all channels. This works around
18463 some problem in alsa that leaves us with inconsistent state
18464 for some reason (#486840).
18466 2008-01-07 13:19:50 +0000 Jerone Young <jerone@gmail.com>
18468 ext/alsa/gstalsamixer.c: If there's no mixer track by the name of 'Master' or 'Front', check if there's one called 'P...
18469 Original commit message from CVS:
18470 Patch by: Jerone Young <jerone at gmail com>
18471 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
18472 If there's no mixer track by the name of 'Master' or 'Front',
18473 check if there's one called 'PCM' before trying the generic
18474 fallback logic (fixes #506928, where we pick 'Mic' as master
18475 track for the AD1984 card in a Thinkpad T61/X61 laptop).
18477 2008-01-07 11:40:04 +0000 Wim Taymans <wim.taymans@gmail.com>
18479 gst/playback/gstplay-enum.*: Add enums for configuration flags.
18480 Original commit message from CVS:
18481 * gst/playback/gstplay-enum.c:
18482 (register_gst_autoplug_select_result),
18483 (gst_autoplug_select_result_get_type), (register_gst_play_flags),
18484 (gst_play_flags_get_type):
18485 * gst/playback/gstplay-enum.h:
18486 Add enums for configuration flags.
18487 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
18488 (init_group), (gst_play_bin_init), (gst_play_bin_set_property),
18489 (gst_play_bin_get_property), (no_more_pads_cb),
18490 (autoplug_select_cb), (gst_play_bin_change_state):
18491 Merge mode with flags.
18492 Add more property getters/setters, defaults and docs.
18493 Add properties to get number of audio/video/text streams.
18494 Create sink object in _init so that we can always rely on it being
18496 * gst/playback/gstplaysink.c: (gst_play_sink_init),
18497 (gen_video_chain), (gen_audio_chain), (gen_vis_chain),
18498 (activate_vis), (gst_play_sink_reconfigure),
18499 (gst_play_sink_set_flags), (gst_play_sink_get_flags),
18500 (gst_play_sink_change_state):
18501 * gst/playback/gstplaysink.h:
18502 Use flags to configure the sink pipelines.
18503 Add tee before audio pipeline so that we can use it for visualisations.
18504 Start working on integrating visualisations.
18505 Remove mode, we can do everything with the flags now.
18506 Add method to configue the sink pipeline.
18508 2008-01-06 16:36:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18510 Update to GMemoryInputStream API changes in GLib SVN and require gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
18511 Original commit message from CVS:
18513 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
18514 * tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
18515 Update to GMemoryInputStream API changes in GLib SVN and require
18516 gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
18517 We can also report the duration for every GSeekable, not only
18518 GFileInputStream and GMemoryInputStream.
18520 2008-01-06 14:39:19 +0000 Tim-Philipp Müller <tim@centricular.net>
18522 tests/check/pipelines/theoraenc.c: Turn these functions into macros so we can see right away where the failure occured.
18523 Original commit message from CVS:
18524 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
18525 (check_buffer_timestamp), (check_buffer_duration):
18526 Turn these functions into macros so we can see right away
18527 where the failure occured.
18529 2008-01-05 22:25:05 +0000 Julien Moutte <julien@moutte.net>
18531 sys/xvimage/xvimagesink.c: Add debugging information to understand how X calculates the stride for XvImages.
18532 Original commit message from CVS:
18533 2008-01-05 Julien Moutte <julien@fluendo.com>
18534 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
18535 debugging information to understand how X calculates the stride
18538 2008-01-03 20:33:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18540 gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
18541 Original commit message from CVS:
18542 * gst/volume/Makefile.am:
18543 * gst/volume/gstvolume.c: (volume_choose_func),
18544 (gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
18546 * gst/volume/gstvolume.h:
18547 Use GstAudioFilter as base class for the volume element instead of
18548 plain GstBaseTransform.
18550 2008-01-03 07:17:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18552 gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
18553 Original commit message from CVS:
18554 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
18555 Don't set element details for the abstract GstAudioFilter class.
18557 2008-01-02 12:09:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18559 gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
18560 Original commit message from CVS:
18561 * gst-libs/gst/audio/gstaudiofilter.c:
18562 (gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
18563 Implement get_unit_size() vmethod of GstBaseTransform.
18565 2008-01-01 12:53:48 +0000 Edward Hervey <bilboed@bilboed.com>
18567 gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
18568 Original commit message from CVS:
18569 * gst-libs/gst/pbutils/Makefile.am:
18570 * gst-libs/gst/pbutils/pbutils.h:
18571 Use glib-enum generator to have a proper enum GType for
18572 GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
18574 2008-01-01 01:21:47 +0000 David Schleef <ds@schleef.org>
18576 tests/check/: Reenable theoraenc test, which fails on the buildbot but not locally.
18577 Original commit message from CVS:
18578 * tests/check/Makefile.am:
18579 * tests/check/pipelines/theoraenc.c:
18580 Reenable theoraenc test, which fails on the buildbot but
18583 2007-12-31 21:31:01 +0000 David Schleef <ds@schleef.org>
18585 docs/: Add *-undeclared.txt to fix buildbot.
18586 Original commit message from CVS:
18587 * docs/libs/.cvsignore:
18588 * docs/plugins/.cvsignore:
18589 Add *-undeclared.txt to fix buildbot.
18591 2007-12-31 20:45:28 +0000 David Schleef <ds@schleef.org>
18593 tests/check/Makefile.am: Second attempt at disabling theoraenc test long enough to get buildbot to compile -base.
18594 Original commit message from CVS:
18595 * tests/check/Makefile.am:
18596 Second attempt at disabling theoraenc test long enough to
18597 get buildbot to compile -base.
18599 2007-12-31 20:21:20 +0000 David Schleef <ds@schleef.org>
18601 tests/check/pipelines/theoraenc.c: Disable theoraenc test long enough to get the buildbot to compile a recent -base.
18602 Original commit message from CVS:
18603 * tests/check/pipelines/theoraenc.c:
18604 Disable theoraenc test long enough to get the buildbot to
18605 compile a recent -base.
18607 2007-12-31 13:17:29 +0000 Wim Taymans <wim.taymans@gmail.com>
18609 tests/examples/seek/seek.c: Make sure we reset the slider value to 0.0 without racing against a possible g_idle that ...
18610 Original commit message from CVS:
18611 * tests/examples/seek/seek.c: (stop_cb):
18612 Make sure we reset the slider value to 0.0 without racing against a
18613 possible g_idle that sets it to something else.
18615 2007-12-31 00:32:53 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
18617 sys/ximage/ximagesink.c: fix typo
18618 Original commit message from CVS:
18619 * sys/ximage/ximagesink.c:
18622 2007-12-30 19:21:16 +0000 Wim Taymans <wim.taymans@gmail.com>
18624 gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
18625 Original commit message from CVS:
18626 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
18627 * gst-libs/gst/rtsp/gstrtspdefs.h:
18628 Add Location header so that we can start implementing redirects.
18631 2007-12-29 20:55:39 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
18633 gst/subparse/gstssaparse.c: combine if's
18634 Original commit message from CVS:
18635 * gst/subparse/gstssaparse.c:
18638 2007-12-29 19:23:59 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
18640 gst/subparse/gstssaparse.c: remove duplicate log message
18641 Original commit message from CVS:
18642 * gst/subparse/gstssaparse.c:
18643 remove duplicate log message
18645 2007-12-29 17:29:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18647 Update to latest API changes in GLib/GIO and require at least gio-2.0 2.15.0 for this.
18648 Original commit message from CVS:
18650 * ext/gio/gstgio.c:
18651 * ext/gio/gstgio.h:
18652 * ext/gio/gstgiobasesink.h:
18653 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
18654 * ext/gio/gstgiobasesrc.h:
18655 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
18656 * ext/gio/gstgiosink.h:
18657 * ext/gio/gstgiosrc.h:
18658 * ext/gio/gstgiostreamsink.h:
18659 * ext/gio/gstgiostreamsrc.h:
18660 * tests/check/pipelines/gio.c:
18661 Update to latest API changes in GLib/GIO and require at least
18662 gio-2.0 2.15.0 for this.
18663 * ext/gio/Makefile.am:
18664 Add GST_PLUGIN_LDFLAGS to LDFLAGS.
18666 2007-12-29 16:23:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18668 ext/libvisual/visual.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached()...
18669 Original commit message from CVS:
18670 * ext/libvisual/visual.c: (gst_visual_chain):
18671 Fix 'xyz may be used uninitialized' compiler warnings caused
18672 by broken g_assert_not_reached() macro in GLib-2.15.x and don't
18673 abort() in any case but properly report the error.
18675 2007-12-28 09:00:27 +0000 Wim Taymans <wim.taymans@gmail.com>
18677 gst/playback/gstplaybin2.c: Code cleanups.
18678 Original commit message from CVS:
18679 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
18680 (gst_play_bin_finalize), (gst_play_bin_set_uri),
18681 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
18682 (gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
18683 (autoplug_select_cb), (activate_group), (deactivate_group),
18684 (setup_next_source), (save_current_group),
18685 (gst_play_bin_change_state):
18687 Remove next-uri, we can use the uri property just fine.
18689 Unref uridecodebin when switching.
18690 Fix going to READY.
18691 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
18692 (gst_play_sink_init), (gst_play_sink_dispose),
18693 (gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
18694 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
18695 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
18696 (gst_play_sink_set_property), (gst_play_sink_get_property),
18697 (gen_video_chain), (gen_text_element), (gen_audio_chain),
18698 (gen_vis_element), (gst_play_sink_get_mode),
18699 (gst_play_sink_set_mode), (gst_play_sink_set_flags),
18700 (gst_play_sink_get_flags), (gst_play_sink_request_pad),
18701 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
18702 (gst_play_sink_change_state):
18703 * gst/playback/gstplaysink.h:
18704 Add some locking to make things threadsafe.
18705 * gst/playback/test7.c: (about_to_finish_cb):
18708 2007-12-22 12:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
18710 gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
18711 Original commit message from CVS:
18712 * gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
18713 (gst_video_scale_get_property), (gst_video_scale_transform_caps),
18714 (gst_video_scale_transform):
18715 Don't claim to be able to handle/transform caps that can't really
18716 be handled by the currently selected scaling method (here: RGB or
18717 packed YUV with 4-tap method). Also add locking to method property.
18718 * tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
18719 (test_basetransform_based):
18720 Some test pipelines for the above (not entirely valgrind clean yet
18723 2007-12-22 05:19:00 +0000 David Schleef <ds@schleef.org>
18725 gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
18726 Original commit message from CVS:
18727 * gst-libs/gst/video/video.c:
18728 * gst-libs/gst/video/video.h:
18729 Add additional RGBA and RGB-24 video formats.
18731 2007-12-21 22:46:56 +0000 Tim-Philipp Müller <tim@centricular.net>
18733 tests/check/: Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be deprecated in the future (see #498924).
18734 Original commit message from CVS:
18735 * tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
18736 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
18737 (test_suburi_error_wrongproto), (test_missing_primary_decoder):
18738 * tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
18739 (cddabasesrc_suite):
18740 Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
18741 deprecated in the future (see #498924).
18743 2007-12-21 22:26:47 +0000 Tim-Philipp Müller <tim@centricular.net>
18745 gst/playback/gststreamselector.c: Don't leak event.
18746 Original commit message from CVS:
18747 * gst/playback/gststreamselector.c: (gst_selector_pad_event):
18750 2007-12-20 19:43:25 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
18752 gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
18753 Original commit message from CVS:
18754 * gst-libs/gst/riff/riff-read.c:
18755 Use GST_ROUND_UP_2 macro
18757 2007-12-20 17:13:37 +0000 Tim-Philipp Müller <tim@centricular.net>
18759 gst/playback/.cvsignore: Ignore more.
18760 Original commit message from CVS:
18761 * gst/playback/.cvsignore:
18764 2007-12-20 10:41:29 +0000 Tim-Philipp Müller <tim@centricular.net>
18766 Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
18767 Original commit message from CVS:
18768 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
18769 * gst/playback/gstplaybasebin.c: (set_subtitles_visible),
18770 (set_active_source):
18771 * gst/playback/gstplaybasebin.h:
18772 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
18773 (setup_sinks), (playbin_set_subtitles_visible):
18774 Make switching off of subtitles work. To avoid all kind of
18775 problems with unlinking of the subtitle input, we just keep
18776 the subtitle inputs linked as they are and tell textoverlay
18777 not to render them. Fixes #373011.
18778 Other subtitle switching issues (esp. when there are both
18779 external and in-stream subtitles) remain. They'll be solved
18782 2007-12-18 16:21:35 +0000 Wim Taymans <wim.taymans@gmail.com>
18784 gst/playback/gststreamselector.c: Init the pad segment too.
18785 Original commit message from CVS:
18786 * gst/playback/gststreamselector.c: (gst_selector_pad_init):
18787 Init the pad segment too.
18789 2007-12-18 15:56:51 +0000 Wim Taymans <wim.taymans@gmail.com>
18791 gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
18792 Original commit message from CVS:
18793 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
18794 (gst_audioringbuffer_open_device),
18795 (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
18796 (gst_audioringbuffer_release), (gst_audioringbuffer_start),
18797 (gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
18798 (gst_audio_sink_create_ringbuffer):
18799 Improve debug output.
18800 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
18801 (gst_ring_buffer_pause), (gst_ring_buffer_delay):
18802 Prevent some functions from doing things and failing when the
18803 ringbuffer is not yet acquired.
18805 2007-12-18 15:32:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18807 gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
18808 Original commit message from CVS:
18809 * gst-libs/gst/interfaces/interfaces.h:
18810 Also remove interfaces.h from CVS as it is not needed anymore.
18812 2007-12-18 15:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18814 gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
18815 Original commit message from CVS:
18816 * gst-libs/gst/interfaces/Makefile.am:
18817 interfaces.h is not used anymore so remove it from the build
18820 2007-12-18 01:01:23 +0000 David Schleef <ds@schleef.org>
18822 gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
18823 Original commit message from CVS:
18824 * gst/videotestsrc/gstvideotestsrc.c:
18825 * gst/videotestsrc/gstvideotestsrc.h:
18826 Add a "blink" pattern. Turn on the pain. Apologies. It's useful
18827 for testing vertical refresh synchronization.
18829 2007-12-18 00:13:26 +0000 David Schleef <ds@schleef.org>
18831 Add new GstVideFormat enum and write a bunch of helper functions based around it.
18832 Original commit message from CVS:
18833 * docs/libs/gst-plugins-base-libs-sections.txt:
18834 * gst-libs/gst/video/video.c:
18835 * gst-libs/gst/video/video.h:
18836 Add new GstVideFormat enum and write a bunch of helper functions
18839 2007-12-17 23:41:14 +0000 Tim-Philipp Müller <tim@centricular.net>
18841 Makefile.am: Use new common/win32.mak.
18842 Original commit message from CVS:
18844 Use new common/win32.mak.
18846 2007-12-17 16:44:51 +0000 Wim Taymans <wim.taymans@gmail.com>
18848 gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
18849 Original commit message from CVS:
18850 * gst-libs/gst/audio/gstbaseaudiosrc.c:
18851 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
18853 When going from PLAYING to PAUSED, pause the ringbuffer before calling
18854 the parent state change function, just like the audiosink, because the
18855 parent waits for the element to finish its processing before completing
18856 the state change. This makes going to PAUSED a lot snappier.
18857 When going from READY to PAUSED, don't allow the ringbuffer to start
18860 2007-12-17 00:01:00 +0000 Edward Hervey <bilboed@bilboed.com>
18862 gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
18863 Original commit message from CVS:
18864 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
18865 Yet another fix for broken software that produce files with an empty
18866 blockalign field. Instead of completely failing, make a second attempt
18867 at guessing the width/depth by looking at strf->size.
18869 2007-12-16 23:52:58 +0000 Tim-Philipp Müller <tim@centricular.net>
18871 gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
18872 Original commit message from CVS:
18873 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
18874 (gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
18875 * gst-libs/gst/pbutils/install-plugins.c:
18876 (gst_install_plugins_spawn_child), (gst_install_plugins_supported):
18877 * gst-libs/gst/pbutils/missing-plugins.c:
18878 (gst_missing_plugin_message_get_installer_detail),
18879 (gst_missing_encoder_installer_detail_new):
18880 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
18881 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
18882 Turn a few g_assert_not_reached() into g_return_val_if_reached() to
18883 avoid compiler warnings (#503930).
18885 2007-12-16 23:46:16 +0000 Edward Hervey <bilboed@bilboed.com>
18887 gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
18888 Original commit message from CVS:
18889 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
18890 Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
18891 for jpeg video streams.
18892 Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
18893 for the above modification.
18895 2007-12-15 17:27:48 +0000 Tim-Philipp Müller <tim@centricular.net>
18897 gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
18898 Original commit message from CVS:
18899 * gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
18900 (gst_x_overlay_handle_events):
18901 More guards (we don't want klass to end up being NULL).
18903 2007-12-15 03:40:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18905 Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
18906 Original commit message from CVS:
18908 * gst/volume/gstvolume.c: (gst_volume_init):
18909 Use new gst_base_transform_set_gap_aware() function as volume
18910 correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
18913 2007-12-14 19:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
18915 tests/examples/seek/seek.c: Don't go to READY on EOS as this avoids testing of seeking and restarting after EOS, use ...
18916 Original commit message from CVS:
18917 * tests/examples/seek/seek.c: (msg_segment_done), (main):
18918 Don't go to READY on EOS as this avoids testing of seeking and
18919 restarting after EOS, use the stop button when you want to READY.
18920 Don't try to do a flushing seek in segment-done, it does not make
18921 sense to use this for gapless playback and is not needed.
18923 2007-12-14 18:46:12 +0000 Wim Taymans <wim.taymans@gmail.com>
18925 gst/playback/gstqueue2.c: Use separate timers for input and output rates.
18926 Original commit message from CVS:
18927 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
18928 (reset_rate_timer), (update_in_rates), (update_out_rates),
18929 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
18930 (gst_queue_chain), (gst_queue_loop):
18931 Use separate timers for input and output rates.
18932 Pause measuring the output rate when we block for more data.
18935 2007-12-14 16:23:06 +0000 Christian Schaller <uraeus@gnome.org>
18937 * gst/speexresample/Makefile.am:
18938 update spec file and add two missing files for disting
18939 Original commit message from CVS:
18940 update spec file and add two missing files for disting
18942 2007-12-14 09:24:55 +0000 Wim Taymans <wim.taymans@gmail.com>
18944 gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
18945 Original commit message from CVS:
18946 * gst/playback/gstqueue2.c: (gst_queue_chain):
18947 Pause the timer to measure the input rate when we block because the
18948 queue is filled. See #503262.
18950 2007-12-13 15:54:00 +0000 Peter Kjellerstedt <pkj@axis.com>
18952 gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
18953 Original commit message from CVS:
18954 Patch by: Peter Kjellerstedt <pkj at axis com>
18955 * gst-libs/gst/rtsp/gstrtspconnection.c:
18956 (gst_rtsp_connection_free):
18957 Close control sockets. Fixes #503440.
18959 2007-12-13 12:31:38 +0000 Wim Taymans <wim.taymans@gmail.com>
18961 gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
18962 Original commit message from CVS:
18963 * gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
18964 Expose the right pad in the right place with the right element.
18966 2007-12-13 11:40:10 +0000 Tim-Philipp Müller <tim@centricular.net>
18968 gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
18969 Original commit message from CVS:
18970 * gst-libs/gst/pbutils/descriptions.c: (formats):
18971 Add description for 'private' dts caps (who come up with that name?).
18973 2007-12-13 10:10:35 +0000 Tim-Philipp Müller <tim@centricular.net>
18975 Makefile.am: Add check-exports target and run it with 'make check'.
18976 Original commit message from CVS:
18978 Add check-exports target and run it with 'make check'.
18980 Be stricter about what we export in our libraries: change regexp so that
18981 we only export _gst_foo(), but not __gst_foo().
18982 * gst-libs/gst/cdda/base64.h: (rfc822_binary):
18983 * gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
18984 Change internal functions to __gst_foo so they dont' get exported.
18985 * win32/common/libgstaudio.def:
18986 Add missing symbols.
18988 2007-12-11 21:18:57 +0000 David Schleef <ds@schleef.org>
18991 ChangeLog: remove conflict markers
18992 Original commit message from CVS:
18993 ChangeLog: remove conflict markers
18995 2007-12-11 17:14:13 +0000 Tim-Philipp Müller <tim@centricular.net>
18997 ext/gnomevfs/: Use gst_tag_freeform_string_to_utf8() here, which also takes into account any character sets specified...
18998 Original commit message from CVS:
18999 * ext/gnomevfs/Makefile.am:
19000 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
19001 Use gst_tag_freeform_string_to_utf8() here, which also takes
19002 into account any character sets specified by the user via
19003 environment variables.
19005 2007-12-10 15:21:41 +0000 Wim Taymans <wim.taymans@gmail.com>
19007 gst/audioconvert/Makefile.am: Also link to libm.
19008 Original commit message from CVS:
19009 * gst/audioconvert/Makefile.am:
19012 2007-12-10 15:13:55 +0000 Wim Taymans <wim.taymans@gmail.com>
19014 gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
19015 Original commit message from CVS:
19016 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19017 No need for floating point operations here. avoids having to link
19018 against the math library too.
19020 2007-12-10 11:16:25 +0000 Tim-Philipp Müller <tim@centricular.net>
19022 Add one or two missing formats. Generate ADPCM description dynamically depending on layout/format.
19023 Original commit message from CVS:
19024 * gst-libs/gst/pbutils/descriptions.c: (formats),
19025 (format_info_get_desc):
19026 * tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
19028 Add one or two missing formats. Generate ADPCM description
19029 dynamically depending on layout/format.
19031 2007-12-09 04:28:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19033 configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
19034 Original commit message from CVS:
19036 Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
19038 2007-12-08 18:38:39 +0000 Robin Stocker <robin.stocker@gmx.ch>
19040 gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
19041 Original commit message from CVS:
19042 Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
19043 * gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
19044 Some .srt files start with chunk number 0 and not chunk number 1,
19045 recognise and accept those as well (fixes #502497).
19046 * tests/check/elements/subparse.c: (srt_input), (srt_input0),
19048 Add unit test for the above.
19050 2007-12-06 12:08:21 +0000 Wim Taymans <wim.taymans@gmail.com>
19052 gst/playback/gstplay-enum.*: Add missing files.
19053 Original commit message from CVS:
19054 * gst/playback/gstplay-enum.c:
19055 (register_gst_autoplug_select_result),
19056 (gst_autoplug_select_result_get_type):
19057 * gst/playback/gstplay-enum.h:
19060 2007-12-05 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
19062 gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
19063 Original commit message from CVS:
19064 * gst/playback/Makefile.am:
19065 Group decodebin2 and uridecodebin into the same plugin so that they
19066 can share the GEnumType.
19067 * gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
19068 (_gst_select_accumulator), (gst_decode_bin_class_init),
19069 (gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
19070 (gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
19071 (analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
19072 Add signal to sort factories instead of the more awkward autoplug-select
19074 Modify autoplug_select so that we can try, skip or expose the
19075 autopluggin of an element on a pad.
19076 * gst/playback/gstfactorylists.c: (compare_ranks),
19077 (decoders_filter), (sinks_filter), (gst_factory_list_is_type),
19078 (element_filter), (gst_factory_list_get_elements),
19079 (gst_factory_list_debug), (gst_factory_list_filter):
19080 * gst/playback/gstfactorylists.h:
19081 Simplify the API, allow getting elements based on mask.
19082 * gst/playback/gstplay-marshal.list:
19083 Add some more marshallers.
19084 * gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
19085 (gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
19086 (autoplug_select_cb), (activate_group):
19087 Add support for managing non-raw sinks by providing a custom element and
19088 sink list to decodebin2.
19089 Try to plug non-raw sinks when decodebin2 using autoplug-select of
19091 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
19092 (gst_play_sink_set_mode), (gst_play_sink_request_pad):
19093 * gst/playback/gstplaysink.h:
19094 Add support for raw and non-raw sinks.
19095 Add support to force sinks selected by playbin2.
19096 Don't plug raw converters for non-raw sinks.
19097 * gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
19098 (_gst_select_accumulator), (gst_uri_decode_bin_class_init),
19099 (proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
19101 Use right accumulators.
19104 2007-12-03 13:47:00 +0000 Wim Taymans <wim.taymans@gmail.com>
19106 gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
19107 Original commit message from CVS:
19108 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
19109 Use runnning time as the base time instead of the timestamp.
19110 Spotted by Saur on IRC.
19112 2007-12-03 11:32:30 +0000 Edward Hervey <bilboed@bilboed.com>
19114 gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
19115 Original commit message from CVS:
19116 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
19117 Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
19119 2007-12-03 10:58:14 +0000 Wim Taymans <wim.taymans@gmail.com>
19121 ext/ogg/gstoggdemux.c: If we find a new serial number but it does not contain a BOS page, make sure we initialize the...
19122 Original commit message from CVS:
19123 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
19124 (gst_ogg_demux_read_chain):
19125 If we find a new serial number but it does not contain a BOS page, make
19126 sure we initialize the chain to NULL because else we will try to scan it
19127 and crash. Fixes #500763
19129 2007-11-30 17:47:15 +0000 Wim Taymans <wim.taymans@gmail.com>
19131 gst/playback/: Refactor some common code to filter factories and check caps compat.
19132 Original commit message from CVS:
19133 * gst/playback/Makefile.am:
19134 * gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
19135 (get_feature_array), (decoders_filter), (sinks_filter),
19136 (gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
19137 (gst_factory_list_filter):
19138 * gst/playback/gstfactorylists.h:
19139 Refactor some common code to filter factories and check caps compat.
19140 * gst/playback/gstdecodebin.c:
19141 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
19142 (gst_decode_bin_init), (gst_decode_bin_dispose),
19143 (gst_decode_bin_autoplug_continue),
19144 (gst_decode_bin_autoplug_factories),
19145 (gst_decode_bin_autoplug_select), (analyze_new_pad),
19146 (find_compatibles):
19147 * gst/playback/gstplaybin.c:
19148 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
19149 (gst_play_bin_init), (gst_play_bin_finalize),
19150 (autoplug_factories_cb), (activate_group):
19151 * gst/playback/gstqueue2.c:
19152 * gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
19153 (proxy_autoplug_continue_signal),
19154 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
19155 (proxy_drained_signal):
19156 Add some more debug info and use factor filtering code.
19158 2007-11-26 13:19:46 +0000 Julien Moutte <julien@moutte.net>
19160 configure.ac: Add QuickTime Wrapper plug-in.
19161 Original commit message from CVS:
19162 2007-11-26 Julien Moutte <julien@fluendo.com>
19163 * configure.ac: Add QuickTime Wrapper plug-in.
19164 * gst/speexresample/gstspeexresample.c:
19165 (gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
19166 build on Mac OS X Leopard. Incorrect printf format arguments.
19168 * sys/qtwrapper/Makefile.am:
19169 * sys/qtwrapper/audiodecoders.c:
19170 (qtwrapper_audio_decoder_base_init),
19171 (qtwrapper_audio_decoder_class_init),
19172 (qtwrapper_audio_decoder_init),
19173 (clear_AudioStreamBasicDescription), (fill_indesc_mp3),
19174 (fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
19175 (make_samr_magic_cookie), (open_decoder),
19176 (qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
19177 (qtwrapper_audio_decoder_chain),
19178 (qtwrapper_audio_decoder_sink_event),
19179 (qtwrapper_audio_decoders_register):
19180 * sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
19182 * sys/qtwrapper/codecmapping.h:
19183 * sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
19184 (image_description_for_mp4v), (image_description_from_stsd_buffer),
19185 (image_description_from_codec_data):
19186 * sys/qtwrapper/imagedescription.h:
19187 * sys/qtwrapper/qtutils.c: (get_name_info_from_component),
19188 (get_output_info_from_component), (dump_avcc_atom),
19189 (dump_image_description), (dump_codec_decompress_params),
19190 (addSInt32ToDictionary), (dump_cvpixel_buffer),
19191 (DestroyAudioBufferList), (AllocateAudioBufferList):
19192 * sys/qtwrapper/qtutils.h:
19193 * sys/qtwrapper/qtwrapper.c: (plugin_init):
19194 * sys/qtwrapper/qtwrapper.h:
19195 * sys/qtwrapper/videodecoders.c:
19196 (qtwrapper_video_decoder_base_init),
19197 (qtwrapper_video_decoder_class_init),
19198 (qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
19199 (fill_image_description), (new_image_description), (close_decoder),
19200 (open_decoder), (qtwrapper_video_decoder_sink_setcaps),
19201 (decompressCb), (qtwrapper_video_decoder_chain),
19202 (qtwrapper_video_decoder_sink_event),
19203 (qtwrapper_video_decoders_register): Initial import of QuickTime
19204 wrapper jointly developped by Songbird authors (Pioneers of the
19205 Inevitable) and Fluendo.
19207 2007-11-26 12:25:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19209 gst/: Add GAP-flag support.
19210 Original commit message from CVS:
19211 * gst/audiotestsrc/gstaudiotestsrc.c:
19212 * gst/volume/gstvolume.c:
19213 * gst/volume/gstvolume.h:
19214 Add GAP-flag support.
19216 2007-11-26 08:43:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19218 gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
19219 Original commit message from CVS:
19220 * gst/speexresample/README:
19221 * gst/speexresample/arch.h:
19222 * gst/speexresample/resample.c: (resampler_basic_direct_single),
19223 (resampler_basic_direct_double),
19224 (resampler_basic_interpolate_single),
19225 (resampler_basic_interpolate_double),
19226 (speex_resampler_process_native), (speex_resampler_process_float),
19227 (speex_resampler_process_int),
19228 (speex_resampler_process_interleaved_float),
19229 (speex_resampler_process_interleaved_int),
19230 (speex_resampler_get_input_latency),
19231 (speex_resampler_get_output_latency):
19232 * gst/speexresample/speex_resampler.h:
19233 Update speex resampler to latest SVN. We're now down to only the
19234 changes noted in README again.
19235 * gst/speexresample/speex_resampler_wrapper.h:
19236 * gst/speexresample/gstspeexresample.c:
19237 (gst_speex_resample_push_drain), (gst_speex_resample_query):
19238 Adjust to API changes.
19240 2007-11-24 15:02:01 +0000 Julien Moutte <julien@moutte.net>
19242 tests/examples/seek/seek.c: Increase the range of the rate selector as I would like to test QOS behavior at higher fo...
19243 Original commit message from CVS:
19244 2007-11-24 Julien MOUTTE <julien@moutte.net>
19245 * tests/examples/seek/seek.c: (main): Increase the range of the
19246 rate selector as I would like to test QOS behavior at higher
19247 forward and reverse playback speed like say 64x.
19249 2007-11-23 10:21:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19251 gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
19252 Original commit message from CVS:
19253 * gst/speexresample/gstspeexresample.c:
19254 (gst_speex_resample_update_state):
19255 Only post the latency message if we have a resampler state already.
19257 2007-11-23 10:21:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19259 gst/audioresample/gstaudioresample.c: Implement latency query.
19260 Original commit message from CVS:
19261 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
19262 (audioresample_query), (audioresample_query_type),
19263 (gst_audioresample_set_property):
19264 Implement latency query.
19266 2007-11-23 10:01:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19268 gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
19269 Original commit message from CVS:
19270 * gst/speexresample/gstspeexresample.c:
19271 (gst_speex_resample_update_state):
19272 Also post GST_MESSAGE_LATENCY if the latency changes.
19274 2007-11-23 08:48:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19276 gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
19277 Original commit message from CVS:
19278 * gst/speexresample/resample.c: (speex_resampler_get_latency),
19279 (speex_resampler_drain_float), (speex_resampler_drain_int),
19280 (speex_resampler_drain_interleaved_float),
19281 (speex_resampler_drain_interleaved_int):
19282 * gst/speexresample/speex_resampler.h:
19283 * gst/speexresample/speex_resampler_wrapper.h:
19284 Add functions to push the remaining samples and to get the latency
19285 of the resampler. These will get added to Speex SVN in this or a
19286 slightly changed form at some point too and should get merged then
19288 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
19289 (gst_speex_resample_init_state),
19290 (gst_speex_resample_transform_size),
19291 (gst_speex_resample_push_drain), (gst_speex_resample_event),
19292 (gst_speex_fix_output_buffer), (gst_speex_resample_process),
19293 (gst_speex_resample_query), (gst_speex_resample_query_type):
19294 Drop the prepending zeroes and output the remaining samples on EOS.
19295 Also properly implement the latency query for this. speexresample
19296 should be completely ready for production use now.
19298 2007-11-21 18:02:21 +0000 Wim Taymans <wim.taymans@gmail.com>
19300 gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
19301 Original commit message from CVS:
19302 * gst-libs/gst/audio/gstbaseaudiosink.c:
19303 (gst_base_audio_sink_drain):
19304 Our EOS time contains the base_time, _wait_eos() expects a running_time
19305 so we have to subtract the base_time again before calling the function.
19306 This fixes an EOS regression where the base_time was added twice and EOS
19307 took longer and longer in certain situations.
19310 2007-11-21 13:04:17 +0000 Wim Taymans <wim.taymans@gmail.com>
19312 Expose methods for some object properties so that subclasses can more easily configure them.
19313 Original commit message from CVS:
19314 * docs/libs/gst-plugins-base-libs-sections.txt:
19315 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
19316 (gst_base_audio_sink_set_provide_clock),
19317 (gst_base_audio_sink_get_provide_clock),
19318 (gst_base_audio_sink_set_slave_method),
19319 (gst_base_audio_sink_get_slave_method),
19320 (gst_base_audio_sink_set_property),
19321 (gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
19322 (gst_base_audio_sink_none_slaving),
19323 (gst_base_audio_sink_handle_slaving):
19324 * gst-libs/gst/audio/gstbaseaudiosink.h:
19325 Expose methods for some object properties so that subclasses can more
19326 easily configure them.
19327 Added slave method none, that completely disables slaving to the
19329 API: gst_base_audio_sink_set_provide_clock()
19330 API: gst_base_audio_sink_get_provide_clock()
19331 API: gst_base_audio_sink_set_slave_method()
19332 API: gst_base_audio_sink_get_slave_method()
19333 * gst-libs/gst/audio/gstbaseaudiosrc.c:
19334 (gst_base_audio_src_set_provide_clock),
19335 (gst_base_audio_src_get_provide_clock),
19336 (gst_base_audio_src_set_property),
19337 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
19338 * gst-libs/gst/audio/gstbaseaudiosrc.h:
19339 Expose methods for some object properties so that subclasses can more
19340 easily configure them.
19341 API: gst_base_audio_src_set_provide_clock()
19342 API: gst_base_audio_src_get_provide_clock()
19344 2007-11-21 10:18:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19346 gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
19347 Original commit message from CVS:
19348 * gst/speexresample/README:
19349 Add README explaining where the resampling code was taken from
19350 and which changes were done.
19351 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
19353 Use g_malloc() and friends instead of malloc() to achieve higher
19354 portability and define the functions inline.
19355 * gst/speexresample/speex_resampler.h:
19356 Add back some useless preprocessor stuff to keep the diff between
19357 our version and the one from the Speex SVN repository lower.
19359 2007-11-20 20:23:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19361 gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
19362 Original commit message from CVS:
19363 * gst/speexresample/gstspeexresample.c:
19364 (gst_speex_fix_output_buffer), (gst_speex_resample_transform):
19365 Some small cleanup and addition of a TODO item.
19367 2007-11-20 12:56:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19369 gst/speexresample/Makefile.am: Add missing file.
19370 Original commit message from CVS:
19371 * gst/speexresample/Makefile.am:
19374 2007-11-20 07:53:56 +0000 Joe Peterson <lavajoe@gentoo.org>
19376 gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
19377 Original commit message from CVS:
19378 Patch by: Joe Peterson <lavajoe at gentoo dot org>
19379 * gst-libs/gst/sdp/gstsdpmessage.c:
19380 Fix compilation on FreeBSD (Gentoo). Fixes #498228.
19382 2007-11-20 07:47:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19384 Add speexresample to the docs and while at that do a make update.
19385 Original commit message from CVS:
19386 * docs/plugins/Makefile.am:
19387 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
19388 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
19389 * docs/plugins/gst-plugins-bad-plugins.args:
19390 * docs/plugins/gst-plugins-bad-plugins.signals:
19391 * docs/plugins/inspect/plugin-bz2.xml:
19392 * docs/plugins/inspect/plugin-cdxaparse.xml:
19393 * docs/plugins/inspect/plugin-dtsdec.xml:
19394 * docs/plugins/inspect/plugin-equalizer.xml:
19395 * docs/plugins/inspect/plugin-faac.xml:
19396 * docs/plugins/inspect/plugin-faad.xml:
19397 * docs/plugins/inspect/plugin-filter.xml:
19398 * docs/plugins/inspect/plugin-freeze.xml:
19399 * docs/plugins/inspect/plugin-gio.xml:
19400 * docs/plugins/inspect/plugin-gsm.xml:
19401 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
19402 * docs/plugins/inspect/plugin-h264parse.xml:
19403 * docs/plugins/inspect/plugin-modplug.xml:
19404 * docs/plugins/inspect/plugin-mpeg2enc.xml:
19405 * docs/plugins/inspect/plugin-musepack.xml:
19406 * docs/plugins/inspect/plugin-musicbrainz.xml:
19407 * docs/plugins/inspect/plugin-nsfdec.xml:
19408 * docs/plugins/inspect/plugin-replaygain.xml:
19409 * docs/plugins/inspect/plugin-soundtouch.xml:
19410 * docs/plugins/inspect/plugin-spcdec.xml:
19411 * docs/plugins/inspect/plugin-spectrum.xml:
19412 * docs/plugins/inspect/plugin-speed.xml:
19413 * docs/plugins/inspect/plugin-tta.xml:
19414 * docs/plugins/inspect/plugin-videosignal.xml:
19415 * docs/plugins/inspect/plugin-xingheader.xml:
19416 * docs/plugins/inspect/plugin-xvid.xml:
19417 * gst/speexresample/gstspeexresample.h:
19418 Add speexresample to the docs and while at that do a make update.
19420 2007-11-20 07:30:30 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19422 gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
19423 Original commit message from CVS:
19424 * gst/speexresample/gstspeexresample.c:
19425 (gst_speex_fix_output_buffer), (gst_speex_resample_process):
19426 If the resampler gives less output samples than expected
19427 adjust the output buffer and print a warning.
19429 2007-11-20 07:02:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19431 Add resample element based on the Speex resampling algorithm.
19432 Original commit message from CVS:
19434 * gst/speexresample/arch.h:
19435 * gst/speexresample/fixed_generic.h:
19436 * gst/speexresample/gstspeexresample.c:
19437 (gst_speex_resample_base_init), (gst_speex_resample_class_init),
19438 (gst_speex_resample_init), (gst_speex_resample_start),
19439 (gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
19440 (gst_speex_resample_transform_caps),
19441 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
19442 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
19443 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
19444 (gst_speex_resample_event), (gst_speex_resample_check_discont),
19445 (gst_speex_resample_process), (gst_speex_resample_transform),
19446 (gst_speex_resample_set_property),
19447 (gst_speex_resample_get_property), (plugin_init):
19448 * gst/speexresample/gstspeexresample.h:
19449 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
19450 (speex_free), (compute_func), (main), (sinc), (cubic_coef),
19451 (resampler_basic_direct_single), (resampler_basic_direct_double),
19452 (resampler_basic_interpolate_single),
19453 (resampler_basic_interpolate_double), (update_filter),
19454 (speex_resampler_init), (speex_resampler_init_frac),
19455 (speex_resampler_destroy), (speex_resampler_process_native),
19456 (speex_resampler_process_float), (speex_resampler_process_int),
19457 (speex_resampler_process_interleaved_float),
19458 (speex_resampler_process_interleaved_int),
19459 (speex_resampler_set_rate), (speex_resampler_get_rate),
19460 (speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
19461 (speex_resampler_set_quality), (speex_resampler_get_quality),
19462 (speex_resampler_set_input_stride),
19463 (speex_resampler_get_input_stride),
19464 (speex_resampler_set_output_stride),
19465 (speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
19466 (speex_resampler_reset_mem), (speex_resampler_strerror):
19467 * gst/speexresample/speex_resampler.h:
19468 * gst/speexresample/speex_resampler_float.c:
19469 * gst/speexresample/speex_resampler_int.c:
19470 * gst/speexresample/speex_resampler_wrapper.h:
19471 Add resample element based on the Speex resampling algorithm.
19473 2007-11-19 12:30:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19475 tests/check/libs/fft.c: Fix scaling to really have dB instead of something else.
19476 Original commit message from CVS:
19477 * tests/check/libs/fft.c: (GST_START_TEST):
19478 Fix scaling to really have dB instead of something else.
19480 2007-11-19 12:08:16 +0000 Julien Moutte <julien@moutte.net>
19482 tests/examples/seek/seek.c: There's a nice macro to check
19483 Original commit message from CVS:
19484 2007-11-19 Julien MOUTTE <julien@moutte.net>
19485 * tests/examples/seek/seek.c: (main): There's a nice macro to
19487 GTK version, use it.
19489 2007-11-19 11:59:20 +0000 Julien Moutte <julien@moutte.net>
19491 tests/examples/seek/seek.c: Try to support stable version of GTK.
19492 Original commit message from CVS:
19493 2007-11-19 Julien MOUTTE <julien@moutte.net>
19494 * tests/examples/seek/seek.c: (main): Try to support stable version
19497 2007-11-17 15:25:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19499 gst/playback/: Fix the build + little README update.
19500 Original commit message from CVS:
19501 * gst/playback/README:
19502 * gst/playback/test7.c:
19503 Fix the build + little README update.
19505 2007-11-16 16:02:45 +0000 Wim Taymans <wim.taymans@gmail.com>
19507 tests/examples/seek/seek.c: Add playbin2 seek pipeline.
19508 Original commit message from CVS:
19509 * tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
19510 Add playbin2 seek pipeline.
19512 2007-11-16 15:44:48 +0000 Wim Taymans <wim.taymans@gmail.com>
19514 gst/playback/: Add playbin2.
19515 Original commit message from CVS:
19516 * gst/playback/Makefile.am:
19517 * gst/playback/gstplayback.c: (plugin_init):
19518 * gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
19519 (eos_cb), (about_to_finish_cb), (main):
19521 Added gapless playback example.
19522 * gst/playback/gstplaybasebin.c:
19523 * gst/playback/gstplaybasebin.h:
19524 * gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
19525 * gst/playback/gstqueue2.c:
19526 * gst/playback/test.c:
19527 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
19529 * gst/playback/gststreaminfo.h:
19531 * gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
19532 (gst_play_bin_class_init), (init_group), (gst_play_bin_init),
19533 (gst_play_bin_dispose), (gst_play_bin_set_uri),
19534 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
19535 (gst_play_bin_get_property), (gst_play_bin_handle_message),
19536 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
19537 (drained_cb), (unlink_group), (activate_group),
19538 (setup_next_source), (gst_play_bin_change_state),
19539 (gst_play_bin2_plugin_init):
19540 Added raw first version of playbin2. Does chained oggs and gapless
19541 playback fine. No support for raw sinks yet. No visualisations or
19543 * gst/playback/gstplaysink.c: (gst_play_sink_get_type),
19544 (gst_play_sink_class_init), (gst_play_sink_init),
19545 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
19546 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
19547 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
19548 (gst_play_sink_set_property), (gst_play_sink_get_property),
19549 (post_missing_element_message), (free_chain), (add_chain),
19550 (activate_chain), (gen_video_chain), (gen_text_element),
19551 (gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
19552 (gst_play_sink_set_mode), (gst_play_sink_request_pad),
19553 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
19554 (gst_play_sink_send_event), (gst_play_sink_change_state):
19555 * gst/playback/gstplaysink.h:
19556 Added Element that abstracts the sinks and their pipelines for playbin2.
19558 2007-11-16 15:05:07 +0000 Wim Taymans <wim.taymans@gmail.com>
19560 gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
19561 Original commit message from CVS:
19562 * gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
19563 (gst_selector_pad_class_init), (gst_selector_pad_init),
19564 (gst_selector_pad_finalize), (gst_selector_pad_reset),
19565 (gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
19566 (gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
19567 (gst_selector_pad_chain), (gst_stream_selector_get_type),
19568 (gst_stream_selector_base_init), (gst_stream_selector_class_init),
19569 (gst_stream_selector_init), (gst_stream_selector_set_property),
19570 (gst_stream_selector_get_linked_pad),
19571 (gst_stream_selector_getcaps),
19572 (gst_stream_selector_is_active_sinkpad),
19573 (gst_stream_selector_activate_sinkpad),
19574 (gst_stream_selector_get_linked_pads),
19575 (gst_stream_selector_request_new_pad),
19576 (gst_stream_selector_release_pad):
19577 * gst/playback/gststreamselector.h:
19578 Improve streamselector, make it select and unselect the current pad more
19580 Subclass GstPad for the sinkpads of the selector.
19581 Handle segments more correctly.
19582 Fix caps negotiation.
19583 Implement release_pad.
19585 2007-11-16 12:51:44 +0000 Wim Taymans <wim.taymans@gmail.com>
19587 gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
19588 Original commit message from CVS:
19589 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
19590 (gst_decode_group_check_if_drained), (source_pad_event_probe),
19592 Add drained signal fired when decodebin finishes decoding the data.
19593 Remove deprecated STATE_DIRTY message.
19594 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
19595 (unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
19596 (analyse_source), (proxy_drained_signal), (make_decoder),
19597 (source_new_pad), (value_list_append_structure_list),
19598 (handle_redirect_message), (handle_message):
19599 Proxy the new drained signal.
19600 Handle pad removed from decodebin.
19601 Handle redirect messages by sorting multiple redirections based on the
19604 2007-11-16 11:22:09 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
19606 gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
19607 Original commit message from CVS:
19608 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
19609 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
19610 Fix leaking headers. Fixes #496761.
19612 2007-11-16 11:16:58 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
19614 sys/: Don't leak the PAR on errors. Fixes #496731.
19615 Original commit message from CVS:
19616 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
19617 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
19618 (gst_ximagesink_change_state):
19619 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
19620 Don't leak the PAR on errors. Fixes #496731.
19622 2007-11-16 10:14:34 +0000 Tim-Philipp Müller <tim@centricular.net>
19624 gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
19625 Original commit message from CVS:
19626 * gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
19627 (gst_tag_from_id3_user_tag):
19628 Add mapping for audio cd discid tags, so we can extract
19629 them from tags as well (see #347848). Also compare identifiers
19630 in ID3v2 TXXX frames in a case-insensitive way to increase
19631 compatibility when reading tags (discid vs. DiscID vs. DiscId).
19633 2007-11-16 01:21:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19635 gst-plugins-base.doap: Oops, fix the release name.
19636 Original commit message from CVS:
19637 * gst-plugins-base.doap:
19638 Oops, fix the release name.
19640 2007-11-16 00:44:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19642 gst-plugins-base.doap: Add 0.10.15 release
19643 Original commit message from CVS:
19644 * gst-plugins-base.doap:
19645 Add 0.10.15 release
19647 2007-11-16 00:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19649 configure.ac: Back to CVS
19650 Original commit message from CVS:
19654 === release 0.10.15 ===
19656 2007-11-16 00:14:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19658 configure.ac: releasing 0.10.15, "No need to argue"
19659 Original commit message from CVS:
19660 === release 0.10.15 ===
19661 2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
19663 releasing 0.10.15, "No need to argue"
19665 2007-11-16 00:04:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19690 Original commit message from CVS:
19693 2007-11-15 21:40:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19695 win32/vs6/libgstfft.dsp: Convert line endings to DOS.
19696 Original commit message from CVS:
19697 * win32/vs6/libgstfft.dsp:
19698 Convert line endings to DOS.
19700 2007-11-15 21:14:04 +0000 Sébastien Moutte <sebastien@moutte.net>
19702 win32/: Add a project file for fft plugin and remove socket based plugin which don't build from the workspace.* win32...
19703 Original commit message from CVS:
19704 * win32/vs6/gst_plugins_base.dsw:
19705 * win32/vs6/libgstfft.dsp:
19707 Add a project file for fft plugin and remove socket
19708 based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
19709 * win32/vs6/libgstrtp.dsp:
19710 * win32/vs6/libgsttag.dsp:
19711 Convert line endings back to DOS.
19714 2007-11-14 12:27:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19716 win32/vs6/: Convert line endings back to DOS
19717 Original commit message from CVS:
19718 * win32/vs6/libgstinterfaces.dsp:
19719 * win32/vs6/libgstrtsp.dsp:
19720 Convert line endings back to DOS
19722 2007-11-14 11:08:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19724 gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
19725 Original commit message from CVS:
19726 * gst-libs/gst/fft/kiss_fft_f32.h:
19727 * gst-libs/gst/fft/kiss_fft_f64.h:
19728 * gst-libs/gst/fft/kiss_fft_s16.h:
19729 * gst-libs/gst/fft/kiss_fft_s32.h:
19730 Don't include malloc.h which doesn't exist on Mac OSX.
19731 Instead, pull in glib.h and use g_malloc/g_free for
19732 consistency. Fixes: #496548
19734 2007-11-09 15:54:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19736 gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
19737 Original commit message from CVS:
19738 * gst/playback/gstdecodebin2.c:
19739 Dont leak ghostpad. Fixes #475451.
19741 2007-11-09 12:21:52 +0000 Wim Taymans <wim.taymans@gmail.com>
19743 Update some more docs and comments.
19744 Original commit message from CVS:
19745 * docs/design/design-decodebin.txt:
19746 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
19747 Update some more docs and comments.
19749 2007-11-07 16:47:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19751 Require GIO >= 0.1.2 and adjust unit test for an API change.
19752 Original commit message from CVS:
19754 * tests/check/pipelines/gio.c: (GST_START_TEST):
19755 Require GIO >= 0.1.2 and adjust unit test for an API change.
19757 2007-11-07 15:18:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19759 ext/gio/gstgio.h: Add macro to check if a stream supports seeking.
19760 Original commit message from CVS:
19761 * ext/gio/gstgio.h:
19762 Add macro to check if a stream supports seeking.
19763 * ext/gio/Makefile.am:
19764 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
19765 (gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
19766 (gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
19767 (gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
19768 (gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
19769 (gst_gio_base_sink_render), (gst_gio_base_sink_query),
19770 (gst_gio_base_sink_set_stream):
19771 * ext/gio/gstgiobasesink.h:
19772 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
19773 (gst_gio_base_src_class_init), (gst_gio_base_src_init),
19774 (gst_gio_base_src_finalize), (gst_gio_base_src_start),
19775 (gst_gio_base_src_stop), (gst_gio_base_src_get_size),
19776 (gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
19777 (gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
19778 (gst_gio_base_src_create), (gst_gio_base_src_set_stream):
19779 * ext/gio/gstgiobasesrc.h:
19780 Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
19781 base classes that only require a GInputStream or GOutputStream to
19783 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
19784 (gst_gio_sink_class_init), (gst_gio_sink_init),
19785 (gst_gio_sink_finalize), (gst_gio_sink_start):
19786 * ext/gio/gstgiosink.h:
19787 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
19788 (gst_gio_src_class_init), (gst_gio_src_init),
19789 (gst_gio_src_finalize), (gst_gio_src_start):
19790 * ext/gio/gstgiosrc.h:
19791 Use the newly created base classes here.
19792 * ext/gio/gstgio.c: (plugin_init):
19793 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
19794 (gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
19795 (gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
19796 (gst_gio_stream_sink_get_property):
19797 * ext/gio/gstgiostreamsink.h:
19798 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
19799 (gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
19800 (gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
19801 (gst_gio_stream_src_get_property):
19802 * ext/gio/gstgiostreamsrc.h:
19803 Implement GstGioStreamSink and GstGioStreamSrc that have a property
19804 to set the GInputStream/GOutputStream that should be used.
19805 * tests/check/Makefile.am:
19806 * tests/check/pipelines/.cvsignore:
19807 * tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
19808 (gio_testsuite), (main):
19809 Add unit test for giostreamsrc and giostreamsink.
19811 2007-11-07 11:48:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19813 ext/gio/gstgio.c: Remove nowadays unnecessary workaround for a crash.
19814 Original commit message from CVS:
19815 * ext/gio/gstgio.c: (plugin_init):
19816 Remove nowadays unnecessary workaround for a crash.
19817 * ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
19818 (gst_gio_sink_start), (gst_gio_sink_stop),
19819 (gst_gio_sink_unlock_stop):
19820 * ext/gio/gstgiosink.h:
19821 * ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
19822 (gst_gio_src_stop), (gst_gio_src_unlock_stop):
19823 * ext/gio/gstgiosrc.h:
19824 Make the finalize function safer, clean up everything that could stay
19826 Reset the cancellable instead of creating a new one after cancelling
19828 Don't store the GFile in the element, it's only necessary for creating
19831 2007-11-06 23:35:39 +0000 Sebastien Moutte <sebastien@moutte.net>
19833 gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
19834 Original commit message from CVS:
19835 Patch by: Sebastien Moutte <sebastien moutte net>
19836 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
19837 (gst_rtcp_unix_to_ntp):
19838 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
19839 Fix some C99-isms and and a missing function that some versions of
19840 MSVC don't like too much (#494346).
19841 * win32/vs6/gst_plugins_base.dsw:
19842 * win32/vs6/libgstaudio.dsp:
19843 * win32/vs6/libgstrtp.dsp:
19844 * win32/vs6/libgsttag.dsp:
19845 Update vs6 projects files (#494346).
19847 2007-11-06 16:38:49 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
19849 win32/common/: More missing symbols to export (fixes #493986).
19850 Original commit message from CVS:
19851 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
19852 * win32/common/libgstaudio.def:
19853 * win32/common/libgstcdda.def:
19854 * win32/common/libgstinterfaces.def:
19855 * win32/common/libgstnetbuffer.def:
19856 * win32/common/libgstpbutils.def:
19857 * win32/common/libgstrtp.def:
19858 * win32/common/libgstrtsp.def:
19859 * win32/common/libgsttag.def:
19860 * win32/common/libgstvideo.def:
19861 More missing symbols to export (fixes #493986).
19863 2007-11-06 11:58:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19865 Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
19866 Original commit message from CVS:
19867 * docs/libs/gst-plugins-base-libs-sections.txt:
19868 * gst-libs/gst/fft/gstfftf32.c:
19869 * gst-libs/gst/fft/gstfftf32.h:
19870 * gst-libs/gst/fft/gstfftf64.c:
19871 * gst-libs/gst/fft/gstfftf64.h:
19872 * gst-libs/gst/fft/gstffts16.c:
19873 * gst-libs/gst/fft/gstffts16.h:
19874 * gst-libs/gst/fft/gstffts32.c:
19875 * gst-libs/gst/fft/gstffts32.h:
19876 * tests/check/libs/fft.c: (GST_START_TEST):
19877 Remove the magnitude and phase calculation functions as these have
19878 very special use cases and can't even be used for the spectrum
19879 element. Also adjust the docs to mention some properties of the used
19880 FFT implemention, i.e. how the values are scaled. Fixes #492098.
19882 2007-11-06 11:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
19884 gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
19885 Original commit message from CVS:
19886 * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
19888 Avoid crash when there are external subtitles (fixes #491722).
19890 2007-11-03 10:39:21 +0000 Tim-Philipp Müller <tim@centricular.net>
19892 ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
19893 Original commit message from CVS:
19894 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
19895 * ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
19896 'Could not open resource for writing' is not an acceptable
19897 error message when we can't open the audio device (see #492334),
19898 even less so when we're trying to open it to record something.
19900 2007-11-02 21:03:01 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
19902 win32/common/libgstrtp.def: Add some more missing symbols (#492813).
19903 Original commit message from CVS:
19904 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
19905 * win32/common/libgstrtp.def:
19906 Add some more missing symbols (#492813).
19908 2007-11-02 14:59:06 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
19910 tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
19911 Original commit message from CVS:
19912 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
19913 * tests/check/elements/audioconvert.c: (verify_convert):
19914 Add check to make sure that the out caps have a channel layout
19915 set on them where they should have one.
19917 2007-11-01 13:28:59 +0000 Vincent Torri <vtorri@univ-evry.fr>
19919 gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
19920 Original commit message from CVS:
19921 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
19922 * gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
19923 * gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
19924 Include our own _stdint.h instead of sys/types.h, makes MingW happy
19926 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
19927 Use _pipe directly, GLib doesn't have a pipe() macro any longer
19928 (it disappeared in GLib 2.14.0) (#492306).
19929 * gst-libs/gst/sdp/Makefile.am:
19930 * gst-libs/gst/sdp/gstsdpmessage.c:
19931 Fix includes and LIBS for win32/Mingw (#492306).
19932 * tests/examples/dynamic/addstream.c (pause_play_stream):
19933 Use more portable g_usleep() instead of sleep() (#492306).
19935 2007-11-01 12:51:57 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
19937 gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
19938 Original commit message from CVS:
19939 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
19940 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
19941 (gst_ring_buffer_parse_caps):
19942 Return NULL instead of an enum that happens to be 0, fixes warning
19944 * gst-libs/gst/audio/gstringbuffer.h:
19945 No trailing commas in enum list (for gcc-2.9x).
19946 * gst/videotestsrc/videotestsrc.c: (random_char):
19947 Make information loss explicit instead of implicitly truncating to
19948 eight bits via the return value. Fixes runtime error on MSVC when
19949 using the debug CRT (#492114).
19950 * win32/common/config.h.in:
19951 Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
19952 * win32/common/libgstinterfaces.def:
19953 * win32/common/libgstrtp.def:
19954 Export a few more symbols (#492114).
19956 2007-11-01 08:06:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19958 gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
19959 Original commit message from CVS:
19960 * gst-libs/gst/audio/audio.c:
19961 * gst-libs/gst/audio/audio.h:
19962 Readd the deprecation guards, but preserve compilability.
19964 2007-10-31 17:54:48 +0000 Tim-Philipp Müller <tim@centricular.net>
19966 gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
19967 Original commit message from CVS:
19968 * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
19969 (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
19970 Preserve channel layout when fixating the number of channels in the
19971 output caps, or make sure there's a suitable channel position layout
19972 set on the caps if required. Fixes #430677.
19974 2007-10-31 17:32:22 +0000 Tim-Philipp Müller <tim@centricular.net>
19976 tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
19977 Original commit message from CVS:
19978 * tests/check/elements/decodebin.c: (test_text_plain_streams):
19979 Make sure the pipeline really operates in push mode as it should
19982 2007-10-31 15:30:15 +0000 Tim-Philipp Müller <tim@centricular.net>
19984 gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
19985 Original commit message from CVS:
19986 * gst-libs/gst/audio/audio.h:
19987 Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
19988 compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
19989 (ie. normal cvs builds) will fail.
19991 2007-10-31 12:47:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19993 tell gtk-doc about the deprecation guard. Apply more doc fixes.
19994 Original commit message from CVS:
19995 * docs/libs/Makefile.am:
19996 * gst-libs/gst/audio/audio.c:
19997 * gst-libs/gst/audio/audio.h:
19998 * gst-libs/gst/interfaces/mixer.c:
19999 tell gtk-doc about the deprecation guard. Apply more doc fixes.
20001 2007-10-31 12:30:28 +0000 Tim-Philipp Müller <tim@centricular.net>
20003 tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
20004 Original commit message from CVS:
20005 * tests/check/libs/audio.c: (init_value_to_channel_layout),
20006 (test_channel_layout_value_intersect), (audio_suite):
20007 Add simple unit test to make sure GstValue intersection
20008 of channel layouts works the way I think it does.
20010 2007-10-30 20:32:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20012 Fix the docs according to what gtk-doc complained about.
20013 Original commit message from CVS:
20014 * docs/libs/gst-plugins-base-libs-sections.txt:
20015 * gst-libs/gst/audio/gstaudiofilter.h:
20016 * gst-libs/gst/interfaces/mixer.h:
20017 * gst-libs/gst/rtp/gstbasertpdepayload.c:
20018 * gst-libs/gst/rtp/gstbasertpdepayload.h:
20019 * gst-libs/gst/sdp/gstsdpmessage.c:
20020 Fix the docs according to what gtk-doc complained about.
20022 2007-10-30 19:46:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20024 tests/icles/stress-playbin.c: Fix the build.
20025 Original commit message from CVS:
20026 * tests/icles/stress-playbin.c:
20029 2007-10-30 15:54:46 +0000 Tim-Philipp Müller <tim@centricular.net>
20031 gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
20032 Original commit message from CVS:
20033 * gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
20034 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
20035 Post nice/more useful error message if we don't have a decoder for
20038 2007-10-30 15:07:58 +0000 Wim Taymans <wim.taymans@gmail.com>
20040 gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
20041 Original commit message from CVS:
20042 * gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
20043 Be a bit more useful, unblock the pads after we fired the no-more-pads
20044 signal so that we can use the signal to inspect and connect all pads
20045 without having to keep extra state outside of decodebin.
20047 2007-10-30 15:00:06 +0000 Wim Taymans <wim.taymans@gmail.com>
20049 gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
20050 Original commit message from CVS:
20051 * gst/playback/gsturidecodebin.c:
20052 (gst_uri_decode_bin_autoplug_continue),
20053 (gst_uri_decode_bin_class_init), (no_more_pads_full):
20054 Implement default signal handler so that we return TRUE when nothing is
20057 2007-10-28 11:53:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20059 gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
20060 Original commit message from CVS:
20061 * gst-libs/gst/riff/riff-media.c:
20062 (gst_riff_wavext_add_channel_layout),
20063 (gst_riff_wave_add_default_channel_layout),
20064 (gst_riff_wavext_get_default_channel_mask),
20065 (gst_riff_create_audio_caps):
20066 Use the ALSA channel layout as default for wav files without channel
20067 layout information. This fixes playback of chan-id.wav on 5.1 systems
20068 for example. Also refactor the channel layout setting a bit and add
20069 more default channel orders. Fixes #489010.
20071 2007-10-28 11:46:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20074 Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
20075 Original commit message from CVS:
20076 (gst_riff_wavext_add_channel_layout),
20077 (gst_riff_wave_add_default_channel_layout),
20078 (gst_riff_wavext_get_default_channel_mask),
20079 (gst_riff_create_audio_caps):
20080 Use the ALSA channel layout as default for wav files without channel
20081 layout information. This fixes playback of chan-id.wav on 5.1 systems
20082 for example. Also refactor the channel layout setting a bit and add
20083 more default channel orders. Fixes #489010.
20085 2007-10-26 18:57:33 +0000 Tim-Philipp Müller <tim@centricular.net>
20087 tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
20088 Original commit message from CVS:
20089 * tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
20090 GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
20091 -DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
20094 2007-10-26 12:07:14 +0000 Christian Schaller <uraeus@gnome.org>
20096 * gst-plugins-base.spec.in:
20098 Original commit message from CVS:
20101 2007-10-25 17:36:49 +0000 Wim Taymans <wim.taymans@gmail.com>
20103 gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
20104 Original commit message from CVS:
20105 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
20106 (gst_decode_bin_dispose), (gst_decode_bin_set_caps),
20107 (gst_decode_bin_set_subs_encoding),
20108 (gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
20109 (gst_decode_bin_get_property), (analyze_new_pad):
20110 Move subtitle encoding property to decodebin2 so that it can set the
20111 property value on all elements that it autoplugs and that require it.
20112 Make caps refcounting more consistent in get/set.
20113 * gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
20114 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
20115 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
20116 (gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
20117 (proxy_autoplug_continue_signal),
20118 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
20120 Proxy properties and relevant signals from the internal decodebin.
20121 Make properties MT safe.
20123 2007-10-25 15:10:59 +0000 Tim-Philipp Müller <tim@centricular.net>
20125 gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
20126 Original commit message from CVS:
20127 * gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
20128 * gst-libs/gst/tag/tags.c:
20129 Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
20130 GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
20131 * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
20132 Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
20133 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
20134 (gst_tag_to_vorbis_comments):
20135 Map new SORTNAME tags (these tags aren't even semi-official, so I'm
20136 just mapping everything I found in the wild) (#414539).
20138 2007-10-24 11:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
20140 gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
20141 Original commit message from CVS:
20142 Inspired by patch of: René Stadler <mail at renestadler dot de>
20143 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
20144 (gst_decode_bin_autoplug_continue),
20145 (gst_decode_bin_autoplug_factories),
20146 (gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
20147 (find_compatibles):
20148 * gst/playback/gstplay-marshal.list:
20149 Remove the autoplug-sort signal and replace it with a binding friendly
20150 autoplug-select signal.
20151 Add an autoplug-factories signal that can be used to generate a list of
20152 factories to try to autoplug.
20153 Add the GstPad to the autoplugging signal args as it might be needed to
20154 make a good factory selection.
20155 Fix up the marshallers for this. Fixes #407282.
20157 2007-10-23 14:23:14 +0000 Tim-Philipp Müller <tim@centricular.net>
20159 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...
20160 Original commit message from CVS:
20161 * gst-libs/gst/tag/gsttagdemux.c:
20162 Don't abort with an assertion if we receive a seek event with
20163 a start type of NONE (see launchpad bug #155878).
20165 2007-10-22 10:21:46 +0000 Wim Taymans <wim.taymans@gmail.com>
20167 sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
20168 Original commit message from CVS:
20169 * sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
20170 (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
20171 (gst_ximagesink_change_state), (gst_ximagesink_reset):
20172 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
20173 (gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
20174 (gst_xvimagesink_change_state), (gst_xvimagesink_reset):
20175 Make sure that before we clean up the X resources, we shutdown and join
20177 Also make sure the event thread does not shut down immediatly after
20178 startup because the running variable is not yet correctly set.
20181 2007-10-16 16:48:38 +0000 Wim Taymans <wim.taymans@gmail.com>
20183 gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
20184 Original commit message from CVS:
20185 * gst/playback/gstdecodebin.c: (new_pad), (type_found):
20186 Make the window for a race in typefind and shutting down smaller until
20187 we figure out the right locking here. Avoids #485753 usually.
20188 * gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
20189 Remove unneeded lock causing a race in typefind and shutting down.
20191 * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
20192 Also remove sinks when going to NULL because we might not complete the
20193 state change to PAUSED, causing the PAUSED->READY state change not to
20196 2007-10-16 15:33:31 +0000 Wim Taymans <wim.taymans@gmail.com>
20198 gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
20199 Original commit message from CVS:
20200 * gst-libs/gst/audio/gstbaseaudiosink.c:
20201 (gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
20202 Also explicitly release the ringbuffer when going to NULL because it
20203 is required in the setcaps function, before the state change to PAUSED
20206 2007-10-16 14:58:53 +0000 Tim-Philipp Müller <tim@centricular.net>
20208 tests/icles/: Does what it says on the tin.
20209 Original commit message from CVS:
20210 * tests/icles/.cvsignore:
20211 * tests/icles/Makefile.am:
20212 * tests/icles/stress-playbin.c:
20213 Does what it says on the tin.
20215 2007-10-15 11:38:39 +0000 Wim Taymans <wim.taymans@gmail.com>
20217 gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
20218 Original commit message from CVS:
20219 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
20220 Fix queue negotiation. See #486758.
20222 2007-10-12 10:52:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20224 Actual code change to go along with:
20225 Original commit message from CVS:
20226 Actual code change to go along with:
20227 2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
20228 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
20229 (gst_xvimagesink_xwindow_new),
20230 (gst_xvimagesink_update_colorbalance),
20231 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
20232 Fix handling of some of the X atoms. If the last parameter is True,
20233 XInternAtom won't create the atom if it doesn't exist, and therefore
20234 might return None. This causes X errors on Xv implementations that
20235 don't provide the colour balance attributes.
20237 2007-10-12 10:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20240 Remove stray character from the changelog.
20241 Original commit message from CVS:
20242 Remove stray character from the changelog.
20244 2007-10-12 10:33:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20247 I'm too lazy to comment this
20248 Original commit message from CVS:
20249 *** empty log message ***
20251 2007-10-11 18:24:09 +0000 Tim-Philipp Müller <tim@centricular.net>
20253 Extract vorbis comment LICENSE tags correctly.
20254 Original commit message from CVS:
20255 * gst-libs/gst/tag/gstvorbistag.c:
20256 * tests/check/libs/tag.c:
20257 Extract vorbis comment LICENSE tags correctly.
20259 2007-10-11 16:12:21 +0000 Jason Kivlighn <jkivlighn@gmail.com>
20261 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
20262 Original commit message from CVS:
20263 Patch by: Jason Kivlighn <jkivlighn gmail com>
20264 * gst-libs/gst/tag/gstid3tag.c:
20265 * tests/check/libs/tag.c:
20266 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
20268 2007-10-10 17:01:51 +0000 Tim-Philipp Müller <tim@centricular.net>
20270 gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
20271 Original commit message from CVS:
20272 * gst-libs/gst/tag/gsttagdemux.c:
20273 Don't error out when a buggy downstream element doesn't
20274 handle the newsegment event we send properly (especially
20275 not without posting a meaningful error message on the
20276 bus). See bug #471370 and launchpad bug #136264.
20278 2007-10-10 15:36:56 +0000 Wim Taymans <wim.taymans@gmail.com>
20280 gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
20281 Original commit message from CVS:
20282 * gst-libs/gst/audio/gstbaseaudiosink.c:
20283 (gst_base_audio_sink_drain):
20284 Use new basesink method to make our EOS drain interruptable.
20286 2007-10-10 09:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20288 gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
20289 Original commit message from CVS:
20290 * gst-libs/gst/rtp/gstrtppayloads.c:
20291 Fix silly search-replace oversight.
20293 2007-10-09 09:57:17 +0000 Laurent Glayal <spglegle@yahoo.fr>
20295 gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
20296 Original commit message from CVS:
20297 Patch by: Laurent Glayal <spglegle at yahoo dot fr>
20298 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
20299 (gst_basertppayload_set_outcaps):
20300 Fix caps memleak. Fixes #484989.
20302 2007-10-08 18:04:34 +0000 Wim Taymans <wim.taymans@gmail.com>
20304 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
20305 Original commit message from CVS:
20306 * gst-libs/gst/rtp/gstbasertpdepayload.c:
20307 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
20310 2007-10-08 18:02:53 +0000 Wim Taymans <wim.taymans@gmail.com>
20312 gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
20313 Original commit message from CVS:
20314 * gst-libs/gst/audio/gstbaseaudiosrc.c:
20315 (gst_base_audio_src_create):
20316 Also handle the case where there is no clock set on the audio source,
20317 like in the unit tests.
20319 2007-10-08 17:40:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20321 gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
20322 Original commit message from CVS:
20323 * gst-libs/gst/rtp/gstrtppayloads.c:
20324 Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
20325 to avoid compiler warnings
20327 2007-10-08 17:12:32 +0000 Wim Taymans <wim.taymans@gmail.com>
20329 gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
20330 Original commit message from CVS:
20331 * gst/playback/gstdecodebin.c: (type_found),
20332 (gst_decode_bin_change_state):
20333 * gst/playback/gstdecodebin2.c: (type_found),
20334 (gst_decode_bin_change_state):
20335 Don't disconnect the have_type signal because we never reconnect it
20336 later on. Instead keep a variable to see if we already detected a type.
20338 2007-10-08 10:47:26 +0000 Wim Taymans <wim.taymans@gmail.com>
20340 gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
20341 Original commit message from CVS:
20342 * gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
20343 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
20345 Unlink the signal handler when we found the type, we're not going to do
20346 anything sensible with more type_found signals anyway.
20348 2007-10-08 06:07:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20350 ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
20351 Original commit message from CVS:
20352 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
20353 Use GIO function to get a list of supported URI schemes instead of
20354 hard coding something.
20356 2007-10-06 16:49:55 +0000 Tim-Philipp Müller <tim@centricular.net>
20358 gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
20359 Original commit message from CVS:
20360 * gst-libs/gst/tag/gsttagdemux.c:
20363 2007-10-06 15:04:53 +0000 Tim-Philipp Müller <tim@centricular.net>
20365 gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
20366 Original commit message from CVS:
20367 * gst-libs/gst/tag/Makefile.am:
20368 * gst-libs/gst/tag/gsttagdemux.c:
20369 * gst-libs/gst/tag/gsttagdemux.h:
20370 API: add GstTagDemux base class for simple tag demuxers.
20371 * docs/libs/gst-plugins-base-libs-docs.sgml:
20372 * docs/libs/gst-plugins-base-libs-sections.txt:
20373 Add GstTagDemux to docs.
20375 2007-10-05 07:49:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20377 gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
20378 Original commit message from CVS:
20379 * gst-libs/gst/rtp/gstrtpbuffer.c:
20380 (gst_rtp_buffer_get_payload_subbuffer):
20381 Fix bug introduced with last commit which inverted the logic and
20382 caused all buffers to be dropped. Fixes #483620.
20383 Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
20385 2007-10-04 06:50:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20387 gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
20388 Original commit message from CVS:
20389 * gst-libs/gst/rtp/gstrtpbuffer.c:
20390 Replace g_return_if_val (as it could be disabled), with regular return
20393 2007-10-03 14:51:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20395 tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
20396 Original commit message from CVS:
20397 * tests/check/pipelines/simple-launch-lines.c:
20398 Print message name and not just number.
20400 2007-10-02 11:11:13 +0000 Wim Taymans <wim.taymans@gmail.com>
20402 gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
20403 Original commit message from CVS:
20404 * gst-libs/gst/audio/gstbaseaudiosink.c:
20405 (gst_base_audio_sink_async_play):
20406 When slaved to the clock, don't try to align a sample with the previous
20407 one when going to PLAYING again.
20409 2007-10-02 09:04:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20411 tests/examples/snapshot/snapshot.c: Fix the build.
20412 Original commit message from CVS:
20413 * tests/examples/snapshot/snapshot.c:
20416 2007-10-02 07:43:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20418 ext/gio/gstgiosink.c: Update to API changes in GIO.
20419 Original commit message from CVS:
20420 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
20421 Update to API changes in GIO.
20423 2007-10-01 16:33:00 +0000 Wim Taymans <wim.taymans@gmail.com>
20425 gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
20426 Original commit message from CVS:
20427 * gst-libs/gst/sdp/gstsdpmessage.h:
20428 Add RFC 3556 bandwidth modifiers.
20430 2007-10-01 13:37:31 +0000 Wim Taymans <wim.taymans@gmail.com>
20432 Update documentation.
20433 Original commit message from CVS:
20434 * docs/libs/gst-plugins-base-libs-docs.sgml:
20435 * docs/libs/gst-plugins-base-libs-sections.txt:
20436 * gst-libs/gst/rtp/gstrtppayloads.c:
20437 Update documentation.
20439 2007-10-01 13:22:14 +0000 Wim Taymans <wim.taymans@gmail.com>
20441 gst-libs/gst/rtp/: Added new file and header to deal with payload info.
20442 Original commit message from CVS:
20443 * gst-libs/gst/rtp/Makefile.am:
20444 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
20445 (gst_rtp_payload_info_for_name):
20446 * gst-libs/gst/rtp/gstrtppayloads.h:
20447 Added new file and header to deal with payload info.
20448 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
20449 (gst_rtp_buffer_default_clock_rate):
20450 * gst-libs/gst/rtp/gstrtpbuffer.h:
20451 Payload specific stuff is move to new headers.
20452 Implement _default_clock rate using the new payload function.
20453 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
20454 (gst_sdp_parse_line):
20455 * gst-libs/gst/sdp/gstsdpmessage.h:
20456 Add some more comments.
20458 2007-10-01 10:22:46 +0000 Wim Taymans <wim.taymans@gmail.com>
20460 gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
20461 Original commit message from CVS:
20462 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
20463 (sdp_check_header), (sdp_type_find), (plugin_init):
20464 Add typefind function for application/sdp.
20465 Remove some old dirac typefind code that was ifdeffed out.
20467 2007-09-29 12:04:02 +0000 Sébastien Moutte <sebastien@moutte.net>
20469 win32/common/libgstaudio.def: Add new exported functions.
20470 Original commit message from CVS:
20471 * win32/common/libgstaudio.def:
20472 Add new exported functions.
20473 * win32/vs6/grammar.dsp:
20474 Add autogeneration and copy of some autegenerated files from win32/common
20476 * win32/vs6/libgstaudioconvert.dsp:
20477 Add gstaudioquantize.c to the build.
20478 * win32/vs6/libgstinterfaces.dsp:
20479 Add videoorientation.c to the build.
20480 * win32/vs6/libgstriff.dsp:
20481 Add libgsttag to the link libraries list.
20482 * win32/vs6/libgstvolume.dsp:
20483 Add liboil to the link.
20484 * win32/vs6/gst_plugins_base.dsw:
20485 * win32/vs6/libgstrtsp.dsp:
20486 * win32/common/libgstrtsp.def:
20487 Add files to build libgstrtsp library.
20489 2007-09-29 07:01:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20491 ext/gio/: Some minor cleanup and allow setting the location only when the element is not playing or paused.
20492 Original commit message from CVS:
20493 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
20494 (gst_gio_sink_set_property), (gst_gio_sink_render):
20495 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
20496 (gst_gio_src_set_property):
20497 Some minor cleanup and allow setting the location only when the
20498 element is not playing or paused.
20500 2007-09-26 15:14:37 +0000 Wim Taymans <wim.taymans@gmail.com>
20502 tests/examples/snapshot/snapshot.c: Print error when pipeline failed to construct.
20503 Original commit message from CVS:
20504 * tests/examples/snapshot/snapshot.c: (main):
20505 Print error when pipeline failed to construct.
20507 2007-09-25 19:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
20509 Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
20510 Original commit message from CVS:
20512 * gst-libs/gst/tag/gstid3tag.c:
20513 * gst-libs/gst/tag/gstvorbistag.c:
20514 Add mappings for the new GST_TAG_COMPOSER for vorbis comments
20517 2007-09-25 11:54:09 +0000 Tim-Philipp Müller <tim@centricular.net>
20519 gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
20520 Original commit message from CVS:
20521 * gst-libs/gst/floatcast/floatcast.h:
20522 Don't include config.h in an installed public header, this
20523 might break compilation of applications that don't have such
20524 a header and doesn't necessarily do what it's supposed to do
20525 anyway (ie. check for the lrint/lrintf defines) (#442065).
20526 Add docs for the various macros and document how this header
20527 has to be used (link against libm, etc.); add a few FIXMEs;
20528 include math.h for non-c99 code path. Based on patch by
20531 2007-09-25 07:50:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20533 configure.ac: Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead of duplicating these macros in confi...
20534 Original commit message from CVS:
20536 Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
20537 of duplicating these macros in configure.ac.
20539 2007-09-22 17:58:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20541 po/: Updated translations to 0.10.14
20542 Original commit message from CVS:
20546 Updated translations to 0.10.14
20548 2007-09-22 17:57:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20552 Original commit message from CVS:
20555 2007-09-22 17:56:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20557 po/pl.po: Added Polish translation.
20558 Original commit message from CVS:
20559 translated by: Jakub Bogusz <qboosh@pld-linux.org>
20561 Added Polish translation.
20563 2007-09-22 17:55:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20565 po/fi.po: Added Finnish translation.
20566 Original commit message from CVS:
20567 translated by: Ilkka Tuohela <hile@iki.fi>
20569 Added Finnish translation.
20571 2007-09-22 17:54:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20573 po/es.po: Added Spanish translation.
20574 Original commit message from CVS:
20575 translated by: Jorge González González <aloriel@gmail.com>
20577 Added Spanish translation.
20579 2007-09-22 17:53:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20581 po/da.po: Added Danish translation.
20582 Original commit message from CVS:
20583 translated by: Mogens Jaeger <mogens@jaeger.tf>
20585 Added Danish translation.
20587 2007-09-22 17:52:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20589 po/zh_CN.po: Added Chinese (simplified) translation.
20590 Original commit message from CVS:
20591 translated by: Funda Wang <fundawang@linux.net.cn>
20593 Added Chinese (simplified) translation.
20595 2007-09-22 17:51:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20597 po/bg.po: Added Bulgarian translation.
20598 Original commit message from CVS:
20599 translated by: Alexander Shopov <ash@contact.bg>
20601 Added Bulgarian translation.
20603 2007-09-21 18:00:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20605 docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy.
20606 Original commit message from CVS:
20607 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
20609 * ext/gio/gstgiosink.h:
20610 * ext/gio/gstgiosrc.h:
20611 Mark private fields of the instance structs private.
20613 2007-09-21 17:31:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20615 docs/plugins/: Add the GIO plugin to the docs and do a make update while doing that.
20616 Original commit message from CVS:
20617 * docs/plugins/Makefile.am:
20618 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
20619 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
20620 * docs/plugins/gst-plugins-bad-plugins.args:
20621 * docs/plugins/gst-plugins-bad-plugins.signals:
20622 * docs/plugins/inspect/plugin-bz2.xml:
20623 * docs/plugins/inspect/plugin-cdxaparse.xml:
20624 * docs/plugins/inspect/plugin-dfbvideosink.xml:
20625 * docs/plugins/inspect/plugin-dtsdec.xml:
20626 * docs/plugins/inspect/plugin-equalizer.xml:
20627 * docs/plugins/inspect/plugin-faac.xml:
20628 * docs/plugins/inspect/plugin-faad.xml:
20629 * docs/plugins/inspect/plugin-filter.xml:
20630 * docs/plugins/inspect/plugin-freeze.xml:
20631 * docs/plugins/inspect/plugin-gio.xml:
20632 * docs/plugins/inspect/plugin-gsm.xml:
20633 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
20634 * docs/plugins/inspect/plugin-h264parse.xml:
20635 * docs/plugins/inspect/plugin-modplug.xml:
20636 * docs/plugins/inspect/plugin-mpeg2enc.xml:
20637 * docs/plugins/inspect/plugin-musepack.xml:
20638 * docs/plugins/inspect/plugin-musicbrainz.xml:
20639 * docs/plugins/inspect/plugin-nsfdec.xml:
20640 * docs/plugins/inspect/plugin-replaygain.xml:
20641 * docs/plugins/inspect/plugin-soundtouch.xml:
20642 * docs/plugins/inspect/plugin-spcdec.xml:
20643 * docs/plugins/inspect/plugin-spectrum.xml:
20644 * docs/plugins/inspect/plugin-speed.xml:
20645 * docs/plugins/inspect/plugin-tta.xml:
20646 * docs/plugins/inspect/plugin-videosignal.xml:
20647 * docs/plugins/inspect/plugin-xingheader.xml:
20648 * docs/plugins/inspect/plugin-xvid.xml:
20649 Add the GIO plugin to the docs and do a make update
20651 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
20652 Fix a small memleak.
20654 2007-09-21 17:07:56 +0000 René Stadler <mail@renestadler.de>
20656 Add a GIO/GVFS plugin with source and sink elements. This will only be enabled when --enable-experimental is given to...
20657 Original commit message from CVS:
20658 Patch by: René Stadler <mail at renestadler dot de>
20661 * ext/gio/Makefile.am:
20662 * ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek),
20663 (gst_gio_get_supported_protocols),
20664 (gst_gio_uri_handler_get_type_sink),
20665 (gst_gio_uri_handler_get_type_src),
20666 (gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri),
20667 (gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init),
20668 (gst_gio_uri_handler_do_init), (plugin_init):
20669 * ext/gio/gstgio.h:
20670 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
20671 (gst_gio_sink_class_init), (gst_gio_sink_init),
20672 (gst_gio_sink_finalize), (gst_gio_sink_set_property),
20673 (gst_gio_sink_get_property), (gst_gio_sink_start),
20674 (gst_gio_sink_stop), (gst_gio_sink_unlock),
20675 (gst_gio_sink_unlock_stop), (gst_gio_sink_event),
20676 (gst_gio_sink_render), (gst_gio_sink_query):
20677 * ext/gio/gstgiosink.h:
20678 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
20679 (gst_gio_src_class_init), (gst_gio_src_init),
20680 (gst_gio_src_finalize), (gst_gio_src_set_property),
20681 (gst_gio_src_get_property), (gst_gio_src_start),
20682 (gst_gio_src_stop), (gst_gio_src_get_size),
20683 (gst_gio_src_is_seekable), (gst_gio_src_unlock),
20684 (gst_gio_src_unlock_stop), (gst_gio_src_check_get_range),
20685 (gst_gio_src_create):
20686 * ext/gio/gstgiosrc.h:
20687 Add a GIO/GVFS plugin with source and sink elements. This will
20688 only be enabled when --enable-experimental is given to configure
20689 for now as the GIO API is not stable yet. Fixes #476916.
20691 2007-09-21 14:37:26 +0000 Wim Taymans <wim.taymans@gmail.com>
20693 gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
20694 Original commit message from CVS:
20695 * gst/playback/gstqueue2.c: (gst_queue_push_one):
20696 Fix compilation wrt printf arguments.
20698 2007-09-20 17:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
20700 examples/app/appsrc_ex.c: Fix compilation after changing the name of a method.
20701 Original commit message from CVS:
20702 * examples/app/appsrc_ex.c: (main):
20703 Fix compilation after changing the name of a method.
20705 2007-09-20 14:09:24 +0000 Wim Taymans <wim.taymans@gmail.com>
20707 Add simple snapshot example program using appsink.
20708 Original commit message from CVS:
20710 * tests/examples/Makefile.am:
20711 * tests/examples/snapshot/.cvsignore:
20712 * tests/examples/snapshot/Makefile.am:
20713 * tests/examples/snapshot/snapshot.c: (main):
20714 Add simple snapshot example program using appsink.
20716 2007-09-20 13:59:50 +0000 Wim Taymans <wim.taymans@gmail.com>
20718 gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
20719 Original commit message from CVS:
20720 * gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
20721 (gst_app_sink_class_init), (gst_app_sink_init),
20722 (gst_app_sink_dispose), (gst_app_sink_finalize),
20723 (gst_app_sink_set_property), (gst_app_sink_get_property),
20724 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
20725 (gst_app_sink_event), (gst_app_sink_getcaps),
20726 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
20727 (gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
20728 (gst_app_sink_pull_buffer):
20729 * gst-libs/gst/app/gstappsink.h:
20730 Add properties, signals and actions to access the element even without
20731 linking to the library.
20732 Fix some method names and signatures.
20734 2007-09-20 10:37:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20736 tests/check/generic/states.c: Improved state change unit test.
20737 Original commit message from CVS:
20738 * tests/check/generic/states.c:
20739 Improved state change unit test.
20741 2007-09-19 18:16:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20743 Ignore registries in any format.
20744 Original commit message from CVS:
20745 * docs/plugins/.cvsignore:
20746 * tests/check/.cvsignore:
20747 Ignore registries in any format.
20749 2007-09-19 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
20751 gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
20752 Original commit message from CVS:
20753 * gst-libs/gst/rtp/gstbasertpdepayload.c:
20754 (gst_base_rtp_depayload_chain),
20755 (gst_base_rtp_depayload_set_gst_timestamp):
20756 Only copy timestamp on outgoing packets if the depayloader did not set
20758 Also copy duration on outgoing packets.
20760 2007-09-19 15:55:08 +0000 Wim Taymans <wim.taymans@gmail.com>
20762 gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
20763 Original commit message from CVS:
20764 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
20765 (gst_basertppayload_set_outcaps):
20766 Fix compilation because of missing %d in printf.
20767 When fixating caps, fixate what we can and throw away all remaining
20768 unfixed caps, subclasses should do something smart if they need to.
20770 2007-09-19 12:04:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20772 ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
20773 Original commit message from CVS:
20774 * ext/gnomevfs/gstgnomevfssrc.c:
20775 Improve debug logs a bit and be more verbose if things go wrong.
20777 2007-09-17 17:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20779 Fix a bunch of compile warnings shown with Forte.
20780 Original commit message from CVS:
20781 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
20782 (gst_text_overlay_set_property):
20783 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
20784 * gst-libs/gst/audio/gstbaseaudiosink.c:
20785 (gst_base_audio_sink_render):
20786 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
20787 (gst_rtcp_unix_to_ntp):
20788 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
20789 * gst/playback/gstqueue2.c:
20790 * tests/examples/seek/seek.c: (set_scale):
20791 Fix a bunch of compile warnings shown with Forte.
20792 * gst/audiorate/gstaudiorate.c:
20793 Always pull in config.h before including any system headers.
20795 2007-09-17 16:22:17 +0000 Wim Taymans <wim.taymans@gmail.com>
20797 gst/playback/gstqueue2.c: Also fix #476514 for queue2.
20798 Original commit message from CVS:
20799 * gst/playback/gstqueue2.c: (update_buffering),
20800 (gst_queue_locked_flush), (gst_queue_locked_enqueue),
20801 (gst_queue_handle_sink_event), (gst_queue_chain),
20802 (gst_queue_push_one), (gst_queue_sink_activate_push),
20803 (gst_queue_src_activate_push), (gst_queue_src_activate_pull):
20804 Also fix #476514 for queue2.
20806 2007-09-16 19:31:06 +0000 Wim Taymans <wim.taymans@gmail.com>
20808 gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
20809 Original commit message from CVS:
20810 * gst-libs/gst/rtp/gstbasertpdepayload.c:
20811 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
20812 (gst_base_rtp_depayload_chain),
20813 (gst_base_rtp_depayload_handle_sink_event),
20814 (gst_base_rtp_depayload_push_full),
20815 (gst_base_rtp_depayload_set_gst_timestamp),
20816 (gst_base_rtp_depayload_change_state):
20817 Remove code to deal with RTP to GST time conversion, we now just copy
20818 the GST timestamp we receive to the outgoing buffers.
20819 Handle segment and flushes correctly.
20820 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
20821 When we have no valid input timestamp, use the previous rtp timestamp on
20822 the outgoing RTP packet instead of the RTP base time.
20824 2007-09-16 01:56:21 +0000 David Schleef <ds@schleef.org>
20826 ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
20827 Original commit message from CVS:
20828 * ext/alsa/gstalsa.c:
20829 * ext/alsa/gstalsadeviceprobe.c:
20830 * ext/alsa/gstalsamixer.c:
20831 * ext/alsa/gstalsasink.c:
20832 * ext/alsa/gstalsasrc.c:
20833 Change alsa alloca's to malloc to fix warnings on gcc-4.2.
20835 2007-09-15 18:41:27 +0000 Wim Taymans <wim.taymans@gmail.com>
20837 gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
20838 Original commit message from CVS:
20839 * gst-libs/gst/rtp/gstbasertppayload.c:
20840 (gst_basertppayload_set_outcaps), (gst_basertppayload_push):
20841 Add some debug info when negotiating caps.
20843 2007-09-15 00:29:11 +0000 Wim Taymans <wim.taymans@gmail.com>
20845 gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
20846 Original commit message from CVS:
20847 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
20848 A buffer with an empty payload is also a valid buffer.
20850 2007-09-14 20:52:00 +0000 Wim Taymans <wim.taymans@gmail.com>
20852 gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
20853 Original commit message from CVS:
20854 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
20855 (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
20856 (gst_basertppayload_change_state):
20857 Make sure we start our RTP timestamp from the random base RTP
20858 timestamp even if the buffer timestamp starts from some random value.
20860 2007-09-14 16:56:16 +0000 Wim Taymans <wim.taymans@gmail.com>
20862 Add simple exmple app to demonstrate starting and pausing live and non-live bins in a PLAYING pipeline.
20863 Original commit message from CVS:
20865 * tests/examples/Makefile.am:
20866 * tests/examples/dynamic/.cvsignore:
20867 * tests/examples/dynamic/Makefile.am:
20868 * tests/examples/dynamic/addstream.c: (create_stream),
20869 (pause_play_stream), (message_received), (eos_message_received),
20870 (perform_step), (main):
20871 Add simple exmple app to demonstrate starting and pausing live and
20872 non-live bins in a PLAYING pipeline.
20874 2007-09-14 10:42:00 +0000 Julien Moutte <julien@moutte.net>
20876 gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
20877 Original commit message from CVS:
20878 2007-09-14 Julien MOUTTE <julien@moutte.net>
20879 * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
20880 typefind for QCP files (RFC #3625)
20882 2007-09-13 22:52:09 +0000 Wim Taymans <wim.taymans@gmail.com>
20884 gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
20885 Original commit message from CVS:
20886 * gst-libs/gst/audio/gstbaseaudiosink.c:
20887 (gst_base_audio_sink_init):
20888 Disable pull mode scheduling, we're not ready for it yet and it subtly
20889 breaks a lot of things.
20891 2007-09-12 17:35:52 +0000 Tim-Philipp Müller <tim@centricular.net>
20893 tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
20894 Original commit message from CVS:
20895 * tests/check/elements/libvisual.c:
20896 Test all libvisual plugins, not just the first one; this reproduces
20897 bug #450336 quite easily. Looks like a problem with the 'jess'
20900 2007-09-12 17:15:12 +0000 Tim-Philipp Müller <tim@centricular.net>
20902 tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
20903 Original commit message from CVS:
20904 * tests/check/Makefile.am:
20905 * tests/check/elements/.cvsignore:
20906 * tests/check/elements/libvisual.c:
20907 Add basic libvisual test case in an attempt to reproduce bug #450336.
20908 Doesn't reproduce that bug, but some other crasher instead (invalid
20909 free), at least with make elements/libvisual.forever and the bumscope
20910 plugin on x86-64/gutsy. Leaving test disabled for now.
20912 2007-09-12 08:38:21 +0000 Peter Kjellerstedt <pkj@axis.com>
20914 gst/: Printf format fixes (#476128).
20915 Original commit message from CVS:
20916 Patch by: Peter Kjellerstedt <pkj at axis com>
20917 * gst-libs/gst/app/gstappsink.c:
20918 * gst/flv/gstflvdemux.c:
20919 * gst/flv/gstflvparse.c:
20920 * gst/interleave/deinterleave.c:
20921 * gst/switch/gstswitch.c:
20922 Printf format fixes (#476128).
20924 2007-09-11 19:07:57 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
20926 gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
20927 Original commit message from CVS:
20928 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
20929 * gst-libs/gst/rtsp/gstrtspconnection.c:
20930 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
20931 (read_body), (gst_rtsp_connection_receive):
20932 Make sure we can not cancel in the middle of receiving a message.
20935 2007-09-11 11:29:12 +0000 Josep Torra Valles <josep@fluendo.com>
20937 gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
20938 Original commit message from CVS:
20939 Patch by: Josep Torra Valles <josep@fluendo.com>
20940 * gst/playback/gstplaybasebin.c:
20941 Increase upper limit for audio queue a bit; fixes preroll problem
20942 with playbin and decodebin2 when playing a quicktime trailer with
20943 multichannel audio via http (#464666).
20945 2007-09-10 22:10:54 +0000 Wim Taymans <wim.taymans@gmail.com>
20947 gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
20948 Original commit message from CVS:
20949 * gst-libs/gst/audio/gstbaseaudiosrc.c:
20950 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
20951 (gst_base_audio_src_provide_clock),
20952 (gst_base_audio_src_set_property),
20953 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
20954 * gst-libs/gst/audio/gstbaseaudiosrc.h:
20955 Allow othe clocks than the internal clock to be used for the pipeline.
20956 Add property to disable clock provide.
20957 API: GstBaseAudioSrc::provide-clock
20959 2007-09-10 12:05:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20961 gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
20962 Original commit message from CVS:
20963 * gst/playback/gstdecodebin2.c:
20964 Don't leak request pads. Fixes #475395.
20966 2007-09-09 10:25:43 +0000 René Stadler <mail@renestadler.de>
20968 sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
20969 Original commit message from CVS:
20970 Patch by: René Stadler <mail at renestadler dot de>
20971 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
20972 (gst_ximage_buffer_class_init):
20973 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
20974 (gst_xvimage_buffer_class_init):
20975 Correctly chain up finalize with the parent class to prevent
20976 memory leaks. Fixes #474880.
20978 2007-09-09 04:08:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20980 Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
20981 Original commit message from CVS:
20982 * gst/volume/gstvolume.c: (volume_choose_func):
20983 * tests/check/elements/volume.c: (GST_START_TEST):
20984 Revert the latest change: floating point samples are allowed to
20985 have any value, not only values in the range [-1,1]. Thanks to Andy
20986 Wingo for noticing.
20987 Also fix processing of int32 samples with volumes > 4 by making the
20988 unity value smaller which prevents overflows.
20990 2007-09-07 17:37:03 +0000 Tim-Philipp Müller <tim@centricular.net>
20992 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
20993 Original commit message from CVS:
20994 * gst-libs/gst/rtp/gstrtpbuffer.c:
20995 * tests/check/libs/rtp.c:
20996 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
20998 2007-09-07 16:46:05 +0000 Haakon Sporsheim <haakon.sporsheim@tandberg.com>
21000 gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
21001 Original commit message from CVS:
21002 Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
21003 * gst-libs/gst/rtp/gstrtpbuffer.c:
21004 Fix up GstRTPHeader helper struct so that compilers will not under
21005 any circumstances add padding in between our fields, as currently
21006 happens with MSVC on win32, because that would lead to us sending
21007 out RTP payloads with broken RTP headers (#471194).
21008 Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
21009 * tests/check/Makefile.am:
21010 * tests/check/libs/.cvsignore:
21011 * tests/check/libs/rtp.c:
21012 Add some simple unit tests for GstRTPBuffer. Some are disabled
21013 because the code tested still needs fixing (set_csrc() does not work).
21015 2007-09-07 15:05:24 +0000 Christian Schaller <uraeus@gnome.org>
21017 * gst-plugins-base.spec.in:
21018 update spec file to include latest RTSP libraries and headers and more
21019 Original commit message from CVS:
21020 update spec file to include latest RTSP libraries and headers and more
21022 2007-09-07 12:41:01 +0000 Tim-Philipp Müller <tim@centricular.net>
21024 win32/: Add rtsp enumtypes (#474384) and update others.
21025 Original commit message from CVS:
21027 * win32/common/gstrtsp-enumtypes.c:
21028 * win32/common/gstrtsp-enumtypes.h:
21029 * win32/common/interfaces-enumtypes.c:
21030 * win32/common/interfaces-enumtypes.h:
21031 * win32/common/multichannel-enumtypes.c:
21032 Add rtsp enumtypes (#474384) and update others.
21034 2007-09-06 20:31:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21036 configure.ac: Fix configure check for HAVE_LIBXML_HTML.
21037 Original commit message from CVS:
21039 Fix configure check for HAVE_LIBXML_HTML.
21041 2007-09-06 12:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
21043 tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
21044 Original commit message from CVS:
21045 * tests/check/libs/.cvsignore:
21046 Ignore more, in case the build bots work again one day.
21048 2007-09-06 07:00:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21050 Add libgstfft, a FFT library based on Kiss FFT which is
21051 Original commit message from CVS:
21052 Reviewed by: Stefan Kost <ensonic@users.sf.net>
21054 * gst-libs/gst/Makefile.am:
21055 * gst-libs/gst/fft/Makefile.am:
21056 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
21057 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
21058 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
21059 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
21060 * gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
21061 * gst-libs/gst/fft/gstfft.h:
21062 * gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
21063 (gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
21064 (gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
21065 * gst-libs/gst/fft/gstfftf32.h:
21066 * gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
21067 (gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
21068 (gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
21069 * gst-libs/gst/fft/gstfftf64.h:
21070 * gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
21071 (gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
21072 (gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
21073 * gst-libs/gst/fft/gstffts16.h:
21074 * gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
21075 (gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
21076 (gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
21077 * gst-libs/gst/fft/gstffts32.h:
21078 * gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
21079 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
21080 (kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
21081 (kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
21082 * gst-libs/gst/fft/kiss_fft_f32.h:
21083 * gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
21084 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
21085 (kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
21086 (kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
21087 * gst-libs/gst/fft/kiss_fft_f64.h:
21088 * gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
21089 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
21090 (kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
21091 (kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
21092 * gst-libs/gst/fft/kiss_fft_s16.h:
21093 * gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
21094 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
21095 (kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
21096 (kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
21097 * gst-libs/gst/fft/kiss_fft_s32.h:
21098 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
21099 (kiss_fftr_f32), (kiss_fftri_f32):
21100 * gst-libs/gst/fft/kiss_fftr_f32.h:
21101 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
21102 (kiss_fftr_f64), (kiss_fftri_f64):
21103 * gst-libs/gst/fft/kiss_fftr_f64.h:
21104 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
21105 (kiss_fftr_s16), (kiss_fftri_s16):
21106 * gst-libs/gst/fft/kiss_fftr_s16.h:
21107 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
21108 (kiss_fftr_s32), (kiss_fftri_s32):
21109 * gst-libs/gst/fft/kiss_fftr_s32.h:
21110 * gst-libs/gst/fft/kiss_version:
21111 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
21112 * pkgconfig/gstreamer-plugins-base.pc.in:
21113 Add libgstfft, a FFT library based on Kiss FFT which is
21114 BSD licensed. Supported sample formats are int16, int32,
21115 float and double. For those formats a real FFT and IFFT
21116 can be done, different windowing functions can be applied
21117 and functions for extracting the magnitude and phase exist.
21119 * docs/libs/Makefile.am:
21120 * docs/libs/gst-plugins-base-libs-docs.sgml:
21121 * docs/libs/gst-plugins-base-libs-sections.txt:
21122 Integrate libgstfft into the docs.
21123 * tests/check/Makefile.am:
21124 * tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
21125 Add unit tests for libgstfft, currently only testing the FFT.
21126 Unit tests for IFFT will follow soon.
21128 2007-09-05 23:07:40 +0000 Peter Kjellerstedt <pkj@axis.com>
21130 gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
21131 Original commit message from CVS:
21132 Patch by: Peter Kjellerstedt <pkj at axis com>
21133 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
21134 (gst_sdp_message_init), (gst_sdp_message_uninit),
21135 (is_multicast_address), (gst_sdp_message_as_text),
21136 (gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
21137 (gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
21138 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
21139 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
21140 (gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
21141 (gst_sdp_media_init), (gst_sdp_media_uninit),
21142 (gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
21143 (gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
21144 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
21145 (gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
21146 (gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
21147 * gst-libs/gst/sdp/gstsdpmessage.h:
21148 Separate INIT_ARRAY() and related macros into two versions, one for
21149 structures and one for pointers (e.g., INIT_ARRAY() and
21150 INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
21151 lists of emails and phone numbers.
21152 Add missing const as appropriate.
21153 Change all gint to guint since they all actually represent unsigned
21155 Do not use time as a variable name as it shadows the global time().
21156 Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
21157 Actually implement gst_sdp_message_add_time().
21158 Make gst_sdp_message_add_time() take repeat times as an argument.
21159 Store repeat times in GstSDPTime as a GArray rather than as gchar**.
21160 Corrected the definition of gst_sdp_media_get_bandwidth() (was
21161 misspelled as badwidth).
21162 gst-indented and a little clean up. Fixes #471067.
21164 2007-09-05 21:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21166 gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
21167 Original commit message from CVS:
21168 * gst/volume/gstvolume.c: (volume_choose_func),
21169 (volume_process_double), (volume_process_double_clamp),
21170 (volume_process_float_clamp):
21171 Correctly clamp float/double samples in the [-1.0,1.0] range to
21172 prevent weird effects.
21173 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
21174 Add unit tests for all samples types that had none before.
21176 2007-09-05 14:09:15 +0000 Tim-Philipp Müller <tim@centricular.net>
21178 gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
21179 Original commit message from CVS:
21180 * gst-libs/gst/rtp/gstrtpbuffer.c:
21181 Need to include stdlib.h for abs() here too.
21183 2007-09-05 14:01:25 +0000 Tim-Philipp Müller <tim@centricular.net>
21185 gst/playback/gststreaminfo.c: Fix build.
21186 Original commit message from CVS:
21187 * gst/playback/gststreaminfo.c:
21190 2007-09-05 10:32:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21192 gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
21193 Original commit message from CVS:
21194 * gst/playback/gststreaminfo.c:
21195 Clean up some half-disabled code and comment.
21197 2007-09-04 16:18:48 +0000 Wim Taymans <wim.taymans@gmail.com>
21199 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
21200 Original commit message from CVS:
21201 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
21202 (gst_base_rtp_payload_audio_handle_event):
21203 Return FALSE from the event handler to let the parent class handle the
21205 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21206 (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
21207 Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
21208 * gst-libs/gst/rtp/gstbasertppayload.c:
21209 Bump the MTU to 1400.
21211 2007-09-04 01:50:55 +0000 Johan Dahlin <johan@gnome.org>
21213 gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
21214 Original commit message from CVS:
21215 2007-09-03 Johan Dahlin <jdahlin@async.com.br>
21216 * gst/typefind/gsttypefindfunctions.c (plugin_init):
21217 Add an audio/x-nsf typefind function for the nsfdec element.
21219 2007-09-03 20:46:38 +0000 Renato Filho <renato.filho@indt.org.br>
21221 gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
21222 Original commit message from CVS:
21223 * gst/playback/gstplaybasebin.c:
21224 Included "myth://" on stream_uris list for enable buffering to mythtv files
21226 2007-09-03 19:31:11 +0000 Wim Taymans <wim.taymans@gmail.com>
21228 Fix parsing of RB blocks.
21229 Original commit message from CVS:
21230 * docs/libs/gst-plugins-base-libs-sections.txt:
21231 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
21232 (gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
21233 (gst_rtcp_unix_to_ntp):
21234 * gst-libs/gst/rtp/gstrtcpbuffer.h:
21235 Fix parsing of RB blocks.
21237 Added helper functions to convert to/from UNIX and NTP time.
21238 API: gst_rtcp_ntp_to_unix()
21239 API: gst_rtcp_unix_to_ntp()
21240 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
21241 (gst_rtp_buffer_get_header_len),
21242 (gst_rtp_buffer_get_extension_data),
21243 (gst_rtp_buffer_get_payload_subbuffer),
21244 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
21245 (gst_rtp_buffer_ext_timestamp):
21246 * gst-libs/gst/rtp/gstrtpbuffer.h:
21247 Fix some more docs.
21248 Implement handling of packets with extensions.
21249 Fix padding check in _validate().
21250 Added function to get extension data.
21251 API: gst_rtp_buffer_get_header_len()
21252 API: gst_rtp_buffer_get_extension_data()
21254 2007-09-03 19:19:35 +0000 Wim Taymans <wim.taymans@gmail.com>
21256 gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
21257 Original commit message from CVS:
21258 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21259 (gst_base_rtp_depayload_class_init),
21260 (gst_base_rtp_depayload_set_gst_timestamp):
21261 Add some more docs for the queue-delay property and fix a typo in a
21263 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
21266 2007-09-03 19:17:33 +0000 Wim Taymans <wim.taymans@gmail.com>
21268 gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
21269 Original commit message from CVS:
21270 * gst-libs/gst/audio/gstbaseaudiosink.c:
21271 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
21272 (gst_base_audio_sink_change_state):
21273 When skew slaving, try to hover around the middle of a segment so that
21274 we at most drift by half a segment.
21275 If we are aligning in the oposite direction of the clock skew, we don't
21278 2007-08-31 21:07:20 +0000 Wim Taymans <wim.taymans@gmail.com>
21280 gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
21281 Original commit message from CVS:
21282 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21283 (gst_base_rtp_depayload_setcaps),
21284 (gst_base_rtp_depayload_set_gst_timestamp):
21285 Be less silly with the segment start, just apply the clock-base to the
21288 2007-08-31 15:58:30 +0000 Wim Taymans <wim.taymans@gmail.com>
21290 gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
21291 Original commit message from CVS:
21292 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21293 (gst_base_rtp_depayload_class_init),
21294 (gst_base_rtp_depayload_finalize),
21295 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
21296 (gst_base_rtp_depayload_handle_sink_event),
21297 (gst_base_rtp_depayload_set_gst_timestamp),
21298 (gst_base_rtp_depayload_change_state):
21299 * gst-libs/gst/rtp/gstbasertpdepayload.h:
21300 Deprecate the queue handling thread thing and remove the code.
21301 Use new method to calculate the extended timestamp.
21303 2007-08-31 15:21:13 +0000 Wim Taymans <wim.taymans@gmail.com>
21305 gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
21306 Original commit message from CVS:
21307 * gst-libs/gst/rtp/gstrtcpbuffer.c:
21308 (gst_rtcp_packet_sdes_copy_entry):
21309 Use g_strndup which does exactly what we want.
21310 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
21311 (gst_rtp_buffer_ext_timestamp):
21312 * gst-libs/gst/rtp/gstrtpbuffer.h:
21313 Add helper function to compare seqnums.
21314 Add helper function to calculate extended timestamps.
21315 API: gst_rtp_buffer_compare_seqnum()
21316 API: gst_rtp_buffer_ext_timestamp()
21318 2007-08-30 21:59:23 +0000 Wim Taymans <wim.taymans@gmail.com>
21320 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
21321 Original commit message from CVS:
21322 * gst-libs/gst/rtp/gstrtcpbuffer.c:
21323 (gst_rtcp_packet_sdes_get_entry),
21324 (gst_rtcp_packet_sdes_copy_entry):
21325 * gst-libs/gst/rtp/gstrtcpbuffer.h:
21326 Fix and document SDES item data function.
21327 Add new function that makes a proper copy of SDES item data.
21328 API: gst_rtcp_packet_sdes_copy_entry()
21330 2007-08-30 07:29:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21332 The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
21333 Original commit message from CVS:
21336 The tcp and subparse plugins are under gst, but not totaly free of
21337 dependencies. Handle selection inconfigure.ac, so that they show up
21338 on the final list of what is build and what is not. Maybe they should
21339 better be moved to ext.
21341 2007-08-30 06:58:46 +0000 Daniel Díaz <yosoy@danieldiaz.org>
21343 Check if libxml provides HTML parser which subparse needs.
21344 Original commit message from CVS:
21345 Patch by: Daniel Díaz <yosoy@danieldiaz.org>
21348 Check if libxml provides HTML parser which subparse needs.
21351 2007-08-29 14:22:04 +0000 Tim-Philipp Müller <tim@centricular.net>
21353 ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
21354 Original commit message from CVS:
21355 * ext/alsa/gstalsa.c:
21356 Fix typo and compilation on big endian systems.
21358 2007-08-29 12:16:46 +0000 Tim-Philipp Müller <tim@centricular.net>
21360 gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
21361 Original commit message from CVS:
21362 * gst/subparse/gstssaparse.c:
21363 Convert SSA newline codes into actual newline characters (#470766).
21365 2007-08-28 14:58:17 +0000 Tim-Philipp Müller <tim@centricular.net>
21367 API: also add gst_install_plugins_supported() while we're at it (see #470456).
21368 Original commit message from CVS:
21369 * docs/libs/gst-plugins-base-libs-sections.txt:
21370 * gst-libs/gst/pbutils/install-plugins.c:
21371 * gst-libs/gst/pbutils/install-plugins.h:
21372 * tests/check/libs/pbutils.c:
21373 API: also add gst_install_plugins_supported() while we're at it
21376 2007-08-28 14:23:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21378 API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
21379 Original commit message from CVS:
21380 * docs/libs/gst-plugins-base-libs-sections.txt:
21381 * gst-libs/gst/pbutils/missing-plugins.c:
21382 * gst-libs/gst/pbutils/missing-plugins.h:
21383 * tests/check/libs/pbutils.c:
21384 API: add gst_missing_*_installer_detail_new() convenience API so
21385 that applications that know exactly what they're missing can request
21386 installer detail strings for those items directly instead of having
21387 to first create a dummy missing-plugin message and then get the
21388 installer detail string from that. Fixes #470456.
21390 2007-08-27 11:59:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21392 gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
21393 Original commit message from CVS:
21394 * gst/playback/gstdecodebin.c: (close_pad_link):
21395 We need to set up delayed-linking whenever the caps are non-fixed,
21396 not just when there are multiple types - use gst_pad_is_fixed()
21399 2007-08-26 14:14:33 +0000 Tim-Philipp Müller <tim@centricular.net>
21401 gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
21402 Original commit message from CVS:
21403 * gst-libs/gst/pbutils/missing-plugins.c:
21404 (gst_missing_plugin_message_get_installer_detail):
21405 Add missing separator in PID fallback case.
21407 2007-08-24 15:28:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21409 ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
21410 Original commit message from CVS:
21411 * ext/alsa/Makefile.am:
21412 There is no GST_PLUGINS_BASE_LIBS defined.
21413 * ext/alsa/gstalsa.c:
21414 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
21415 * ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
21416 Add support for ALSA 24-bit formats.
21417 snd_pcm_delay can return an error code, especially
21418 during XRUNS. In that case, the best we can do is assume
21420 * gst/audioconvert/Makefile.am:
21421 Add flags from -base before any more-remote dependencies.
21423 2007-08-23 20:45:45 +0000 Davyd <davyd@madeley.id.au>
21425 gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
21426 Original commit message from CVS:
21427 Based on a patch by: Davyd <davyd at madeley dot id dot au>
21428 * gst/volume/gstvolume.c: (volume_choose_func),
21429 (volume_update_real_volume), (gst_volume_set_volume),
21430 (gst_volume_init), (volume_process_int32),
21431 (volume_process_int32_clamp), (volume_process_int24),
21432 (volume_process_int24_clamp), (volume_process_int16),
21433 (volume_process_int16_clamp), (volume_process_int8),
21434 (volume_process_int8_clamp), (volume_update_volume), (plugin_init):
21435 * gst/volume/gstvolume.h:
21436 Add support for int32, int24 and int8 to the volume element.
21439 2007-08-23 12:37:42 +0000 Tim-Philipp Müller <tim@centricular.net>
21441 tests/examples/Makefile.am: Fix even more.
21442 Original commit message from CVS:
21443 * tests/examples/Makefile.am:
21446 2007-08-23 10:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21448 Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
21449 Original commit message from CVS:
21451 * docs/libs/Makefile.am:
21452 * docs/libs/gst-plugins-base-libs-docs.sgml:
21453 * docs/libs/gst-plugins-base-libs-sections.txt:
21454 * ext/gnomevfs/gstgnomevfssrc.c:
21455 * ext/gnomevfs/gstgnomevfssrc.h:
21456 * gst-libs/gst/Makefile.am:
21457 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
21458 * pkgconfig/gstreamer-plugins-base.pc.in:
21459 * sys/v4l/v4lsrc_calls.c:
21460 * tests/examples/Makefile.am:
21461 * win32/common/config.h:
21462 Revert unwanted commit. many thanks to moap. I want a fix for
21463 https://thomas.apestaart.org/moap/trac/ticket/239
21465 2007-08-23 08:33:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21469 * docs/libs/Makefile.am:
21470 * docs/libs/gst-plugins-base-libs-docs.sgml:
21471 * docs/libs/gst-plugins-base-libs-sections.txt:
21472 * ext/gnomevfs/gstgnomevfssrc.c:
21473 * ext/gnomevfs/gstgnomevfssrc.h:
21474 * gst-libs/gst/Makefile.am:
21475 * gst-libs/gst/audio/gstaudiofilter.h:
21476 * gst/typefind/gsttypefindfunctions.c:
21477 * gst/volume/gstvolume.c:
21478 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
21479 * pkgconfig/gstreamer-plugins-base.pc.in:
21480 * sys/v4l/v4lsrc_calls.c:
21481 * tests/examples/Makefile.am:
21482 * win32/common/config.h:
21483 Original commit message from CVS: reviewed by: <delete if not using a buddy> patch by: <delete if not someone else's patch> * configure.ac: * docs/libs/Makefile.am: * docs/libs/gst-plugins-base-libs-docs.sgml: * docs/libs/gst-plugins-base-libs-sections.txt: * ext/gnomevfs/gstgnomevfssrc.c: * ext/gnomevfs/gstgnomevfssrc.h: * gst-libs/gst/Makefile.am: * gst-libs/gst/audio/gstaudiofilter.h: * gst/typefind/gsttypefindfunctions.c: * gst/volume/gstvolume.c: * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: * pkgconfig/gstreamer-plugins-base.pc.in: * sys/v4l/v4lsrc_calls.c: * tests/examples/Makefile.am: * win32/common/config.h:
21485 2007-08-22 15:29:04 +0000 Wim Taymans <wim.taymans@gmail.com>
21487 gst-libs/gst/audio/audio.c: Clarify the docs a little.
21488 Original commit message from CVS:
21489 * gst-libs/gst/audio/audio.c:
21490 Clarify the docs a little.
21492 2007-08-22 11:20:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21494 gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
21495 Original commit message from CVS:
21496 * gst/volume/gstvolume.c:
21497 Enable liboil for float and add more details about problems with
21500 2007-08-21 15:43:24 +0000 Wim Taymans <wim.taymans@gmail.com>
21502 sys/v4l/gstv4lsrc.c: Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
21503 Original commit message from CVS:
21504 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
21505 Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
21507 2007-08-21 12:08:43 +0000 Wim Taymans <wim.taymans@gmail.com>
21509 ext/vorbis/vorbisdec.c: When calculating the first timestamp of the buffers, don't go below 0 and clip the samples be...
21510 Original commit message from CVS:
21511 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
21512 When calculating the first timestamp of the buffers, don't go below 0
21513 and clip the samples because the offset was on the eos page.
21516 2007-08-21 11:42:39 +0000 Wim Taymans <wim.taymans@gmail.com>
21518 ext/ogg/gstoggdemux.c: Also submit the eos page when trying to find the first timestamp.
21519 Original commit message from CVS:
21520 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
21521 (gst_ogg_demux_collect_chain_info):
21522 Also submit the eos page when trying to find the first timestamp.
21525 2007-08-17 15:24:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21527 gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
21528 Original commit message from CVS:
21529 * gst-libs/gst/audio/audio.h:
21530 Use gst_util_uint64_scale() instead of doing the math
21531 with double for GST_FRAMES_TO_CLOCK_TIME() and
21532 GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
21533 prevents rounding errors. Fixes #467667.
21535 2007-08-17 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
21537 gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
21538 Original commit message from CVS:
21539 * gst-libs/gst/rtsp/gstrtspconnection.c:
21540 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
21541 (gst_rtsp_connection_read), (gst_rtsp_connection_poll):
21542 * gst-libs/gst/rtsp/gstrtspconnection.h:
21544 On shutdown, don't read the control socket yet.
21545 Set timeout value correctly in all cases.
21546 Add function to check if the server accepts reads or writes.
21547 API: gst_rtsp_connection_poll()
21548 * gst-libs/gst/rtsp/gstrtspdefs.h:
21549 Fix compilation with -pedantic.
21550 Add enum for _poll.
21552 2007-08-16 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
21554 gst-libs/gst/app/gstappsink.c: Override the preroll vmethod instead of overriding the render method twice.
21555 Original commit message from CVS:
21556 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
21557 Override the preroll vmethod instead of overriding the render method
21560 2007-08-16 16:06:21 +0000 Olivier Crete <tester@tester.ca>
21562 gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
21563 Original commit message from CVS:
21564 Patch by: Olivier Crete <tester at tester ca>
21565 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
21566 (gst_basertppayload_getcaps):
21567 * gst-libs/gst/rtp/gstbasertppayload.h:
21568 Add getcaps vfunc to basertppayload. See #465146.
21570 2007-08-16 11:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
21572 gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
21573 Original commit message from CVS:
21574 * gst/playback/gstplaybasebin.c: (queue_threshold_reached):
21575 Only post buffering messages when we are a stream.
21577 2007-08-15 17:05:45 +0000 Tim-Philipp Müller <tim@centricular.net>
21579 gst-libs/gst/pbutils/: Small docs fix and addition.
21580 Original commit message from CVS:
21581 * gst-libs/gst/pbutils/install-plugins.c:
21582 * gst-libs/gst/pbutils/missing-plugins.c:
21583 Small docs fix and addition.
21585 2007-08-14 17:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
21587 gst-libs/gst/app/gstappsink.c: Don't use new API.
21588 Original commit message from CVS:
21589 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
21592 2007-08-14 17:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
21594 gst-libs/gst/app/gstappsink.*: Make love to appsink.
21595 Original commit message from CVS:
21596 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
21597 (gst_app_sink_class_init), (gst_app_sink_dispose),
21598 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
21599 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
21600 (gst_app_sink_render), (gst_app_sink_get_caps),
21601 (gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
21602 (gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
21603 * gst-libs/gst/app/gstappsink.h:
21604 Make love to appsink.
21605 Make it support pulling of the preroll buffer.
21606 Add docs and debug statements.
21607 Fix some races wrt to EOS handling and stopping.
21609 Implement FLUSHING.
21610 API: gst_app_sink_pull_preroll()
21612 2007-08-13 15:37:29 +0000 Tim-Philipp Müller <tim@centricular.net>
21614 tests/icles/: Add a dumb little test for textoverlay alignments.
21615 Original commit message from CVS:
21616 * tests/icles/.cvsignore:
21617 * tests/icles/Makefile.am:
21618 * tests/icles/test-textoverlay.c:
21619 Add a dumb little test for textoverlay alignments.
21621 2007-08-13 15:26:54 +0000 Dan Williams <dcbw@redhat.com>
21623 ext/pango/gsttextoverlay.*: API: add "line-alignment" property (#459334). Add gtk-doc blurb for "silent" property so ...
21624 Original commit message from CVS:
21625 Patch by: Dan Williams <dcbw redhat com>
21626 * ext/pango/gsttextoverlay.c:
21627 * ext/pango/gsttextoverlay.h:
21628 API: add "line-alignment" property (#459334). Add gtk-doc blurb for
21629 "silent" property so there's a Since tag in the API reference.
21631 2007-08-13 11:21:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21635 Original commit message from CVS:
21638 2007-08-12 16:30:36 +0000 Wim Taymans <wim.taymans@gmail.com>
21640 gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
21641 Original commit message from CVS:
21642 * gst-libs/gst/rtp/gstbasertppayload.c:
21643 (gst_basertppayload_set_outcaps):
21644 * gst-libs/gst/rtp/gstbasertppayload.h:
21645 Improve caps negotiation so that downstream elements can confiure
21646 certain RTP properties by fixing them on the caps. See #465146.
21649 2007-08-11 12:39:51 +0000 Tim-Philipp Müller <tim@centricular.net>
21651 Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
21652 Original commit message from CVS:
21653 * docs/libs/gst-plugins-base-libs-sections.txt:
21654 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21655 * gst-libs/gst/rtp/gstbasertpdepayload.h:
21656 Mark as deprecated some macros which were presumably meant to be
21657 private API and accidentally exposed in the public header file.
21658 Also actually _init() lock (only works at the moment because the
21659 struct is zeroed out when created and the initial values in the
21660 mutex struct are zeroes too). (#459585)
21662 2007-08-10 17:35:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21664 docs/libs/Makefile.am: Remove cruft and do some cleanups.
21665 Original commit message from CVS:
21666 * docs/libs/Makefile.am:
21667 Remove cruft and do some cleanups.
21668 * docs/libs/gst-plugins-base-libs-docs.sgml:
21669 Prepare for comming gtkdoc features (rebase against online docs).
21671 2007-08-10 13:55:44 +0000 Michael Smith <msmith@xiph.org>
21673 gst/audiorate/gstaudiorate.c: Debug output fixes.
21674 Original commit message from CVS:
21675 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
21676 Debug output fixes.
21677 * tests/check/elements/audiorate.c: (do_perfect_stream_test),
21679 Change the number of buffers used; 500 is too many and leads to
21682 2007-08-10 10:08:05 +0000 Tim-Philipp Müller <tim@centricular.net>
21684 gst/: Printf format fixes (#465028).
21685 Original commit message from CVS:
21686 * gst/playback/gstqueue2.c:
21687 * gst/videorate/gstvideorate.c:
21688 Printf format fixes (#465028).
21690 2007-08-09 15:44:02 +0000 Michael Smith <msmith@xiph.org>
21692 gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
21693 Original commit message from CVS:
21694 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
21695 If we have a large (> 1 second) discontinuity, push a series of
21696 smaller buffers rather than a single very large buffer. Avoids
21697 unreasonably large single buffer allocations when encountering a
21699 * tests/check/elements/audiorate.c: (GST_START_TEST),
21701 Add a test for this.
21703 2007-08-09 12:06:43 +0000 Josep Torra Valles <josep@fluendo.com>
21705 gst/playback/gstplaybasebin.c: Fixes: #465015
21706 Original commit message from CVS:
21707 * gst/playback/gstplaybasebin.c: (group_commit),
21708 (queue_remove_probe), (queue_threshold_reached):
21709 Patch by: Josep Torra Valles <josep@fluendo.com>
21711 Make sure we remove the check_queues buffer probe from the
21712 correct queue to avoid racily going back to "buffering 99%" when
21713 buffering is actually complete.
21714 Also, fix the spelling of Josep's surname in the ChangeLog.
21716 2007-08-09 11:37:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21718 ext/ogg/gstoggmux.c: Do not leak oggmux instance.
21719 Original commit message from CVS:
21720 * ext/ogg/gstoggmux.c:
21721 Do not leak oggmux instance.
21722 * ext/vorbis/vorbisenc.c:
21725 2007-08-09 10:51:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21727 po/: Updated translations.
21728 Original commit message from CVS:
21734 Updated translations.
21736 2007-08-08 16:07:21 +0000 Yang Hong <hongyang@redflag-linux.com>
21738 ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
21739 Original commit message from CVS:
21740 patch by: Yang Hong <hongyang@redflag-linux.com>
21741 * ext/pango/gsttextoverlay.c:
21742 * ext/pango/gsttextoverlay.h:
21743 Add 'silent' property to GstTimeOverlay. Fixes #462979
21745 2007-08-08 15:05:22 +0000 Josep Torre Valles <josep@fluendo.com>
21747 Add connection-speed property. Fixes #464690.
21748 Original commit message from CVS:
21749 Patch by: Josep Torre Valles <josep@fluendo.com>
21750 * docs/plugins/gst-plugins-base-plugins.args:
21751 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
21752 (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
21753 (gst_uri_decode_bin_get_property), (gen_source_element):
21754 Add connection-speed property. Fixes #464690.
21756 2007-08-07 15:13:46 +0000 Damien Lespiau <damien.lespiau@gmail.com>
21758 Fix compilation on windows. Fixes #464320.
21759 Original commit message from CVS:
21760 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
21762 * gst-libs/gst/rtsp/Makefile.am:
21763 * gst-libs/gst/rtsp/gstrtspconnection.c:
21764 (gst_rtsp_connection_connect):
21765 Fix compilation on windows. Fixes #464320.
21767 2007-08-07 14:14:54 +0000 Josep Torre Valles <josep@fluendo.com>
21769 gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
21770 Original commit message from CVS:
21771 Patch by: Josep Torre Valles <josep@fluendo.com>
21772 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
21773 (gst_play_base_bin_init), (queue_threshold_reached),
21774 (gen_source_element), (setup_substreams),
21775 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
21776 (gst_play_base_bin_get_streaminfo_value_array):
21777 * gst/playback/gstplaybasebin.h:
21778 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
21779 (gst_play_bin_set_property), (gst_play_bin_get_property),
21780 (gst_play_bin_handle_redirect_message):
21781 Move connection-speed property from playbin to playbasebin so that we
21782 can also configure it in source elements that have the connection-speed
21783 property. Fixes #464028.
21784 Add some debug info here and there.
21786 2007-08-06 16:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21788 gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
21789 Original commit message from CVS:
21790 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
21791 Properly respond to conversion queries. Fixes #464079.
21793 2007-08-03 19:53:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21795 gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
21796 Original commit message from CVS:
21797 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
21798 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
21799 (gst_audio_test_src_init_sine_table),
21800 (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
21801 * gst/audiotestsrc/gstaudiotestsrc.h:
21802 Add float/double and int32 support to audiotestsrc. Fixes #460422.
21803 Also set the default volume to the default value specified in the
21806 2007-08-03 19:40:14 +0000 Jens Granseuer <jensgr@gmx.net>
21808 gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
21809 Original commit message from CVS:
21810 Patch by: Jens Granseuer <jensgr at gmx dot net>
21811 * gst/audioconvert/gstaudioquantize.c:
21812 Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
21814 2007-08-03 15:44:01 +0000 Wim Taymans <wim.taymans@gmail.com>
21816 gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
21817 Original commit message from CVS:
21818 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
21819 Add rdt manager for rdt transport.
21820 Fix parsing of RDT transport.
21822 2007-08-03 14:43:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21824 configure.ac: Back to CVS
21825 Original commit message from CVS:
21829 === release 0.10.14 ===
21831 2007-08-03 14:41:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21837 * docs/plugins/gst-plugins-base-plugins.args:
21838 * docs/plugins/inspect/plugin-adder.xml:
21839 * docs/plugins/inspect/plugin-alsa.xml:
21840 * docs/plugins/inspect/plugin-audioconvert.xml:
21841 * docs/plugins/inspect/plugin-audiorate.xml:
21842 * docs/plugins/inspect/plugin-audioresample.xml:
21843 * docs/plugins/inspect/plugin-audiotestsrc.xml:
21844 * docs/plugins/inspect/plugin-cdparanoia.xml:
21845 * docs/plugins/inspect/plugin-decodebin.xml:
21846 * docs/plugins/inspect/plugin-decodebin2.xml:
21847 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
21848 * docs/plugins/inspect/plugin-gdp.xml:
21849 * docs/plugins/inspect/plugin-gnomevfs.xml:
21850 * docs/plugins/inspect/plugin-libvisual.xml:
21851 * docs/plugins/inspect/plugin-ogg.xml:
21852 * docs/plugins/inspect/plugin-pango.xml:
21853 * docs/plugins/inspect/plugin-playbin.xml:
21854 * docs/plugins/inspect/plugin-subparse.xml:
21855 * docs/plugins/inspect/plugin-tcp.xml:
21856 * docs/plugins/inspect/plugin-theora.xml:
21857 * docs/plugins/inspect/plugin-typefindfunctions.xml:
21858 * docs/plugins/inspect/plugin-video4linux.xml:
21859 * docs/plugins/inspect/plugin-videorate.xml:
21860 * docs/plugins/inspect/plugin-videoscale.xml:
21861 * docs/plugins/inspect/plugin-videotestsrc.xml:
21862 * docs/plugins/inspect/plugin-volume.xml:
21863 * docs/plugins/inspect/plugin-vorbis.xml:
21864 * docs/plugins/inspect/plugin-ximagesink.xml:
21865 * docs/plugins/inspect/plugin-xvimagesink.xml:
21866 * gst-plugins-base.doap:
21867 * win32/common/config.h:
21869 Original commit message from CVS:
21872 2007-08-03 14:24:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21890 Original commit message from CVS:
21893 2007-07-27 17:37:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21895 tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
21896 Original commit message from CVS:
21897 * tests/check/libs/audio.c: (GST_START_TEST):
21898 Fix the test to reflect the behaviour of gst_audio_clip_buffer.
21900 2007-07-27 17:10:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21902 gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
21903 Original commit message from CVS:
21904 * gst-libs/gst/audio/audio.c:
21905 When clipping a buffer with no timestamp, assume it is
21906 within the segment without warnings.
21909 2007-07-27 11:16:23 +0000 Wim Taymans <wim.taymans@gmail.com>
21911 gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
21912 Original commit message from CVS:
21913 * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
21914 Fire the signal on the object, not the interface.
21916 2007-07-27 09:17:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21918 gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
21919 Original commit message from CVS:
21920 * gst-libs/gst/rtsp/.cvsignore:
21921 Ber. Don't include the full path, idiot.
21923 2007-07-27 08:29:29 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21925 gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
21926 Original commit message from CVS:
21927 * gst-libs/gst/rtsp/.cvsignore:
21928 Ignore generated files.
21930 2007-07-26 19:57:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21932 gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
21933 Original commit message from CVS:
21934 * gst-libs/gst/interfaces/Makefile.am:
21935 * gst-libs/gst/interfaces/interfaces-marshal.list:
21936 * gst-libs/gst/interfaces/rtspextension.c:
21937 * gst-libs/gst/interfaces/rtspextension.h:
21938 * gst-libs/gst/rtsp/Makefile.am:
21939 * gst-libs/gst/rtsp/gstrtsp.h:
21940 * gst-libs/gst/rtsp/gstrtspextension.c:
21941 (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
21942 (gst_rtsp_extension_detect_server),
21943 (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
21944 (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
21945 (gst_rtsp_extension_configure_stream),
21946 (gst_rtsp_extension_get_transports),
21947 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
21948 * gst-libs/gst/rtsp/gstrtspextension.h:
21949 * gst-libs/gst/rtsp/rtsp-marshal.list:
21950 Move the rtspextension.h interface into gstrtspextension.h
21951 as part of libgstrtsp instead of libgstinterfaces, because it's
21952 only for use within plugins, not applications.
21953 Add stuff to do the enum & marshal generation needed in libgstrtsp now.
21954 Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
21955 signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
21958 2007-07-26 15:48:01 +0000 Wim Taymans <wim.taymans@gmail.com>
21960 gst-libs/gst/interfaces/: Fix marshaller for the send signal.
21961 Original commit message from CVS:
21962 * gst-libs/gst/interfaces/Makefile.am:
21963 * gst-libs/gst/interfaces/interfaces-marshal.list:
21964 * gst-libs/gst/interfaces/rtspextension.c:
21965 (gst_rtsp_extension_iface_init),
21966 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
21967 * gst-libs/gst/interfaces/rtspextension.h:
21968 Fix marshaller for the send signal.
21969 Add URL to stream selection interface method.
21971 2007-07-26 15:35:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21973 gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
21974 Original commit message from CVS:
21975 * gst-libs/gst/riff/Makefile.am:
21976 Pull in our dependencies from -base before those from outside.
21978 2007-07-26 14:33:01 +0000 Wim Taymans <wim.taymans@gmail.com>
21980 API: gst_rtsp_base64_decode_ip()
21981 Original commit message from CVS:
21982 * docs/libs/gst-plugins-base-libs-sections.txt:
21983 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
21984 * gst-libs/gst/rtsp/gstrtspbase64.h:
21985 API: gst_rtsp_base64_decode_ip()
21986 Added function to decode Base64 in-place.
21988 2007-07-26 14:08:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21990 tests/check/libs/.cvsignore: Ignore the mixer test binary.
21991 Original commit message from CVS:
21992 * tests/check/libs/.cvsignore:
21993 Ignore the mixer test binary.
21995 2007-07-26 10:00:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21997 ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
21998 Original commit message from CVS:
21999 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
22000 Gratuitous comment change to trigger a rebuild on the buildbots.
22002 2007-07-25 18:20:36 +0000 Wim Taymans <wim.taymans@gmail.com>
22004 gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
22005 Original commit message from CVS:
22006 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
22007 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
22008 (gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
22009 (gst_sdp_media_get_format), (gst_sdp_media_get_information),
22010 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
22011 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
22012 (gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
22013 (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
22014 (gst_sdp_media_get_attribute_val):
22015 * gst-libs/gst/sdp/gstsdpmessage.h:
22016 Constify args where we can.
22018 2007-07-25 18:18:49 +0000 Wim Taymans <wim.taymans@gmail.com>
22020 gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
22021 Original commit message from CVS:
22022 * gst-libs/gst/interfaces/Makefile.am:
22023 * gst-libs/gst/interfaces/rtspextension.c:
22024 (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
22025 (gst_rtsp_extension_detect_server),
22026 (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
22027 (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
22028 (gst_rtsp_extension_configure_stream),
22029 (gst_rtsp_extension_get_transports),
22030 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
22031 * gst-libs/gst/interfaces/rtspextension.h:
22032 Move interface for RTSP extensions from -good to here.
22033 Added helper methods to invoke interface methods.
22035 2007-07-25 11:22:30 +0000 Wim Taymans <wim.taymans@gmail.com>
22037 Fix some more RTSP docs.
22038 Original commit message from CVS:
22039 * docs/libs/gst-plugins-base-libs-sections.txt:
22040 * gst-libs/gst/rtsp/gstrtspdefs.h:
22041 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
22042 (gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
22043 (gst_rtsp_message_init_response),
22044 (gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
22045 (gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
22046 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
22047 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
22048 (gst_rtsp_message_get_body), (dump_key_value):
22049 * gst-libs/gst/rtsp/gstrtspmessage.h:
22050 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
22051 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
22052 (gst_rtsp_range_parse):
22053 * gst-libs/gst/rtsp/gstrtsprange.h:
22054 * gst-libs/gst/rtsp/gstrtsptransport.c:
22055 * gst-libs/gst/rtsp/gstrtspurl.c:
22056 Fix some more RTSP docs.
22057 Add some missing methods for dealing with messages.
22059 2007-07-24 19:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
22061 Added beginnings of RTSP documentation.
22062 Original commit message from CVS:
22063 * docs/libs/gst-plugins-base-libs-docs.sgml:
22064 * docs/libs/gst-plugins-base-libs-sections.txt:
22065 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
22066 * gst-libs/gst/rtsp/gstrtspbase64.h:
22067 * gst-libs/gst/rtsp/gstrtspconnection.c:
22068 (gst_rtsp_connection_connect), (add_auth_header),
22069 (gst_rtsp_connection_write), (gst_rtsp_connection_send),
22070 (read_body), (gst_rtsp_connection_receive),
22071 (gst_rtsp_connection_next_timeout),
22072 (gst_rtsp_connection_reset_timeout),
22073 (gst_rtsp_connection_set_auth):
22074 * gst-libs/gst/rtsp/gstrtspconnection.h:
22075 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
22076 * gst-libs/gst/rtsp/gstrtspdefs.h:
22077 * gst-libs/gst/rtsp/gstrtspmessage.h:
22078 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
22079 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
22080 (gst_rtsp_range_parse):
22081 * gst-libs/gst/rtsp/gstrtspurl.h:
22082 Added beginnings of RTSP documentation.
22084 2007-07-24 17:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
22086 Document the SDP library.
22087 Original commit message from CVS:
22088 * docs/libs/Makefile.am:
22089 * docs/libs/gst-plugins-base-libs-docs.sgml:
22090 * docs/libs/gst-plugins-base-libs-sections.txt:
22091 * gst-libs/gst/sdp/gstsdp.h:
22092 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
22093 (gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
22094 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
22095 (gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
22096 (gst_sdp_message_get_attribute_val),
22097 (gst_sdp_message_add_attribute), (gst_sdp_media_new),
22098 (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
22099 (gst_sdp_media_get_media), (gst_sdp_media_set_media),
22100 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
22101 (gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
22102 (gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
22103 (gst_sdp_media_get_format), (gst_sdp_media_add_format),
22104 (gst_sdp_media_get_information), (gst_sdp_media_set_information),
22105 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
22106 (gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
22107 (gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
22108 (gst_sdp_media_set_key), (gst_sdp_media_get_key),
22109 (gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
22110 (gst_sdp_media_get_attribute_val_n),
22111 (gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
22112 (print_media), (gst_sdp_message_dump):
22113 * gst-libs/gst/sdp/gstsdpmessage.h:
22114 Document the SDP library.
22115 Add some of the missing SDPMedia methods.
22117 2007-07-24 11:52:56 +0000 Wim Taymans <wim.taymans@gmail.com>
22119 Move SDP and RTSP from helper objects in -good to a reusable library.
22120 Original commit message from CVS:
22122 * gst-libs/gst/Makefile.am:
22123 * gst-libs/gst/rtsp/Makefile.am:
22124 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
22125 * gst-libs/gst/rtsp/gstrtspbase64.h:
22126 * gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
22127 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
22128 (add_auth_header), (add_date_header), (gst_rtsp_connection_write),
22129 (gst_rtsp_connection_send), (read_line), (read_string), (read_key),
22130 (parse_response_status), (parse_request_line), (parse_line),
22131 (gst_rtsp_connection_read), (read_body),
22132 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
22133 (gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
22134 (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
22135 (gst_rtsp_connection_set_auth):
22136 * gst-libs/gst/rtsp/gstrtspconnection.h:
22137 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
22138 (gst_rtsp_strresult), (gst_rtsp_method_as_text),
22139 (gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
22140 (gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
22141 (gst_rtsp_find_method):
22142 * gst-libs/gst/rtsp/gstrtspdefs.h:
22143 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
22144 (gst_rtsp_message_new), (gst_rtsp_message_init),
22145 (gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
22146 (gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
22147 (gst_rtsp_message_init_data), (gst_rtsp_message_unset),
22148 (gst_rtsp_message_free), (gst_rtsp_message_add_header),
22149 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
22150 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
22151 (gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
22152 (gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
22153 (gst_rtsp_message_dump):
22154 * gst-libs/gst/rtsp/gstrtspmessage.h:
22155 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
22156 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
22157 (gst_rtsp_range_parse), (gst_rtsp_range_free):
22158 * gst-libs/gst/rtsp/gstrtsprange.h:
22159 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
22160 (gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
22161 (gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
22162 (range_as_text), (rtsp_transport_mode_as_text),
22163 (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
22164 (gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
22165 (gst_rtsp_transport_free):
22166 * gst-libs/gst/rtsp/gstrtsptransport.h:
22167 * gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
22168 (gst_rtsp_url_free), (gst_rtsp_url_set_port),
22169 (gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
22170 * gst-libs/gst/rtsp/gstrtspurl.h:
22171 * gst-libs/gst/sdp/Makefile.am:
22172 * gst-libs/gst/sdp/gstsdp.h:
22173 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
22174 (gst_sdp_connection_init), (gst_sdp_bandwidth_init),
22175 (gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
22176 (gst_sdp_attribute_init), (gst_sdp_message_new),
22177 (gst_sdp_message_init), (gst_sdp_message_uninit),
22178 (gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
22179 (gst_sdp_media_uninit), (gst_sdp_media_free),
22180 (gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
22181 (gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
22182 (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
22183 (gst_sdp_message_add_zone), (gst_sdp_message_set_key),
22184 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
22185 (gst_sdp_message_get_attribute_val),
22186 (gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
22187 (gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
22188 (gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
22189 (gst_sdp_media_get_attribute_val_n),
22190 (gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
22191 (read_string), (read_string_del), (gst_sdp_parse_line),
22192 (gst_sdp_message_parse_buffer), (print_media),
22193 (gst_sdp_message_dump):
22194 * gst-libs/gst/sdp/gstsdpmessage.h:
22195 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
22196 Move SDP and RTSP from helper objects in -good to a reusable library.
22197 Use a proper gst_ namespace.
22199 2007-07-23 18:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
22201 ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
22202 Original commit message from CVS:
22203 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
22204 (vorbis_dec_flush_decode):
22205 Use the new buffer clipping function from gstaudio here.
22207 2007-07-23 18:26:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
22209 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
22210 Original commit message from CVS:
22211 * docs/libs/gst-plugins-base-libs-sections.txt:
22212 * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
22213 * gst-libs/gst/audio/audio.h:
22214 * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
22215 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
22216 Also add deprecation guards for gst_audio_structure_set_int() to the
22219 2007-07-23 14:45:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22221 docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
22222 Original commit message from CVS:
22223 * docs/libs/gst-plugins-base-libs-sections.txt:
22226 2007-07-23 11:18:35 +0000 Dan Williams <dcbw@redhat.com>
22228 gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
22229 Original commit message from CVS:
22230 Patch by: Dan Williams <dcbw at redhat dot com>
22231 * gst/playback/gstplaybasebin.c:
22232 (gst_play_base_bin_get_streaminfo_value_array):
22233 Don't return NULL when querying the stream info value array but instead
22234 return an empty array. Fixes #459204.
22236 2007-07-23 10:41:18 +0000 Tim-Philipp Müller <tim@centricular.net>
22238 gst/playback/gsturidecodebin.c: Init debug category before using it.
22239 Original commit message from CVS:
22240 * gst/playback/gsturidecodebin.c:
22241 Init debug category before using it.
22243 2007-07-21 09:56:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22245 gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
22246 Original commit message from CVS:
22247 * gst-libs/gst/interfaces/mixer.h:
22248 Add padding vars in place of the signal pointers
22249 when building with DISABLE_DEPRECATED so that the
22250 interface structure doesn't change size.
22252 2007-07-21 09:21:12 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
22255 Original commit message from CVS:
22256 * docs/libs/gst-plugins-base-libs-sections.txt:
22257 * ext/alsa/gstalsamixer.c:
22258 * ext/alsa/gstalsamixer.h:
22259 * ext/alsa/gstalsamixerelement.c:
22260 * ext/alsa/gstalsamixertrack.c:
22261 * gst-libs/gst/interfaces/mixer.c:
22262 * gst-libs/gst/interfaces/mixer.h:
22263 * gst-libs/gst/interfaces/mixeroptions.c:
22264 * gst-libs/gst/interfaces/mixeroptions.h:
22265 * gst-libs/gst/interfaces/mixertrack.c:
22266 * gst-libs/gst/interfaces/mixertrack.h:
22267 * tests/check/Makefile.am:
22268 * tests/check/libs/mixer.c:
22269 Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
22271 Add support for notifying mixer changes on the message bus, and
22272 implement it in alsamixer.
22273 API: gst_mixer_get_mixer_flags
22274 API: gst_mixer_message_parse_mute_toggled
22275 API: gst_mixer_message_parse_record_toggled
22276 API: gst_mixer_message_parse_volume_changed
22277 API: gst_mixer_message_parse_option_changed
22278 API: GstMixerMessageType
22281 2007-07-20 16:09:03 +0000 Michael Smith <msmith@xiph.org>
22283 sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
22284 Original commit message from CVS:
22285 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
22286 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
22287 xcontext->im_format is only for testing XShm support (as the header
22288 file comments document). Use xvimage->im_format for everything else.
22289 Avoids spurious warnings on buffer allocation before setcaps.
22291 2007-07-20 07:22:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22293 tests/: We should use $(LIBM).
22294 Original commit message from CVS:
22295 * tests/examples/volume/Makefile.am:
22296 * tests/icles/Makefile.am:
22297 We should use $(LIBM).
22299 2007-07-20 06:13:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22301 tests/icles/Makefile.am: This needs -lm.
22302 Original commit message from CVS:
22303 * tests/icles/Makefile.am:
22306 2007-07-18 07:35:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22308 Add stdlib include (free, atoi, exit).
22309 Original commit message from CVS:
22310 * examples/app/appsrc_ex.c:
22311 * examples/switch/switcher.c:
22312 * ext/neon/gstneonhttpsrc.c:
22313 * ext/timidity/gstwildmidi.c:
22314 * ext/x264/gstx264enc.c:
22315 * gst/mve/mveaudioenc.c: (mve_compress_audio):
22316 * gst/rtpmanager/gstrtpclient.c:
22317 * gst/rtpmanager/gstrtpjitterbuffer.c:
22318 * gst/spectrum/demo-audiotest.c:
22319 * gst/spectrum/demo-osssrc.c:
22320 * sys/dvb/gstdvbsrc.c:
22321 Add stdlib include (free, atoi, exit).
22323 2007-07-16 10:10:28 +0000 Wim Taymans <wim.taymans@gmail.com>
22325 gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
22326 Original commit message from CVS:
22327 * gst-libs/gst/rtp/gstbasertppayload.c:
22328 (gst_basertppayload_class_init), (gst_basertppayload_init),
22329 (gst_basertppayload_set_property),
22330 (gst_basertppayload_get_property):
22331 Don't break ABI, restore previous ranges. Keep the default random
22332 selection of timestamp and seqnum offset but as soon as the app sets a
22333 specific value, use that one.
22335 2007-07-14 18:33:15 +0000 Bastien Nocera <hadess@hadess.net>
22337 sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
22338 Original commit message from CVS:
22339 Patch by: Bastien Nocera <hadess at hadess dot net>
22340 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
22341 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
22342 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
22343 * sys/xvimage/xvimagesink.h:
22344 Add option to turn off double-buffering for debugging purposes.
22347 2007-07-14 18:20:41 +0000 Jorn Baayen <jorn@openedhand.com>
22349 sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
22350 Original commit message from CVS:
22351 Patch by: Jorn Baayen <jorn at openedhand dot com>
22352 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
22353 (gst_ximagesink_set_property), (gst_ximagesink_get_property),
22354 (gst_ximagesink_init), (gst_ximagesink_class_init):
22355 * sys/ximage/ximagesink.h:
22356 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
22357 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
22358 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
22359 * sys/xvimage/xvimagesink.h:
22360 add 'handle-expose' property. Useful for video widgets which may want to
22361 be in control of Expose behaviour. Fixes #380625
22363 2007-07-14 17:23:42 +0000 Wim Taymans <wim.taymans@gmail.com>
22365 gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
22366 Original commit message from CVS:
22367 * gst-libs/gst/rtp/gstbasertppayload.c:
22368 (gst_basertppayload_class_init), (gst_basertppayload_init),
22369 (gst_basertppayload_event), (gst_basertppayload_push),
22370 (gst_basertppayload_set_property),
22371 (gst_basertppayload_get_property),
22372 (gst_basertppayload_change_state):
22373 * gst-libs/gst/rtp/gstbasertppayload.h:
22374 Fix ranges of rtp payloader properties so that the full range can be
22375 used in addition to -1 (random).
22376 Fix wrong seqnum reporting in caps.
22379 2007-07-13 18:12:19 +0000 Wim Taymans <wim.taymans@gmail.com>
22381 gst/videorate/gstvideorate.c: Use boilerplate.
22382 Original commit message from CVS:
22383 * gst/videorate/gstvideorate.c: (gst_video_rate_init),
22384 (gst_video_rate_query):
22386 Add latency query, might not be perfect yet but already works a lot
22387 better. Fixes #442557.
22389 2007-07-13 16:05:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22391 sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
22392 Original commit message from CVS:
22393 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
22394 (gst_xvimagesink_setcaps):
22395 * sys/xvimage/xvimagesink.h:
22396 After a caps change, redraw our borders to avoid garbage left there
22397 when the image format changes to a smaller size, like 16:9 -> 4:3
22398 Also, hold the flow_lock a bit longer in the set_caps while we're
22399 fiddling with the xcontext.
22401 2007-07-13 16:02:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22403 Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
22404 Original commit message from CVS:
22407 * tests/Makefile.am:
22408 Remove bogus check for libcheck, since we check for
22409 gstreamer-check and it pulls in the required info from there, and we
22410 weren't actually _using_ the information for libcheck ourselves
22413 2007-07-13 15:52:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22415 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
22416 Original commit message from CVS:
22417 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
22418 (gst_ffmpeg_caps_to_pixfmt):
22419 Fix the r_mask test for RGBA32 on little-endian.
22420 Fix a stupid typo that would have obviously broken
22421 compilation on big-endian, if anyone was testing.
22423 2007-07-12 15:02:43 +0000 Wim Taymans <wim.taymans@gmail.com>
22425 gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
22426 Original commit message from CVS:
22427 * gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
22428 (paint_hline_str4):
22429 * gst/videotestsrc/videotestsrc.h:
22430 Add alpha to the color struct.
22431 Use a default alpha value of 255 instead of 128.
22433 2007-07-12 12:01:20 +0000 Wim Taymans <wim.taymans@gmail.com>
22435 gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
22436 Original commit message from CVS:
22437 * gst/playback/gstplaybasebin.c: (no_more_pads_full),
22439 Clear the dynamic pads counter when starting a new uri. This makes
22440 reusing playbin work again.
22443 2007-07-12 11:13:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22445 configure.ac: Use pkg-config to locate check.
22446 Original commit message from CVS:
22448 Use pkg-config to locate check.
22450 2007-07-11 23:12:12 +0000 Tim-Philipp Müller <tim@centricular.net>
22452 Fix 'make check' build against core CVS.
22453 Original commit message from CVS:
22455 * tests/check/elements/volume.c: (GST_START_TEST):
22456 Fix 'make check' build against core CVS.
22458 2007-07-10 20:46:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22460 gst-libs/gst/: Make gtk-doc happy.
22461 Original commit message from CVS:
22462 * gst-libs/gst/interfaces/propertyprobe.c:
22463 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
22464 * gst-libs/gst/tag/gstvorbistag.c:
22465 Make gtk-doc happy.
22467 2007-07-08 13:07:38 +0000 Tim-Philipp Müller <tim@centricular.net>
22469 gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
22470 Original commit message from CVS:
22471 * gst-libs/gst/audio/gstbaseaudiosink.c:
22472 (gst_base_audio_sink_callback):
22473 Quick hack to make audiosinks stop at EOS when operating in
22474 pull-mode; needs to be fixed properly some day.
22476 2007-07-06 18:19:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22478 docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
22479 Original commit message from CVS:
22480 * docs/libs/gst-plugins-base-libs-sections.txt:
22481 Fix location of includes in the docs.
22483 2007-07-06 11:40:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22485 gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
22486 Original commit message from CVS:
22487 * gst/ffmpegcolorspace/avcodec.h:
22488 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
22489 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
22490 (gst_ffmpegcsp_avpicture_fill):
22491 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
22492 (img_get_alpha_info):
22493 Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
22494 of the existing BGRA32 and RGBA32 formats with the alpha at the other
22495 end of the word. Partially fixes #451908
22497 2007-07-05 08:43:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22499 docs/: Simplify --extra-dir as gtkdoc scans recursively.
22500 Original commit message from CVS:
22501 * docs/libs/Makefile.am:
22502 * docs/plugins/Makefile.am:
22503 Simplify --extra-dir as gtkdoc scans recursively.
22505 2007-07-03 11:52:47 +0000 Wim Taymans <wim.taymans@gmail.com>
22507 gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
22508 Original commit message from CVS:
22509 * gst/adder/gstadder.c: (gst_adder_sink_getcaps),
22510 (gst_adder_request_new_pad):
22511 Make getcaps more robust by not using the proxycaps function. This makes
22512 sure that we don't end up recursively calling getcaps upstream.
22515 2007-06-29 17:21:18 +0000 Wim Taymans <wim.taymans@gmail.com>
22517 gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
22518 Original commit message from CVS:
22519 * gst/audioconvert/audioconvert.c:
22520 Include math.h to fix compilation.
22522 2007-06-29 14:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22524 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
22525 Original commit message from CVS:
22526 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
22527 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
22528 Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
22529 format, as produced by some dc1394 cameras like the iSight.
22530 See http://www.fourcc.org/yuv.php#IYU1
22532 2007-06-28 20:37:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
22534 gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
22535 Original commit message from CVS:
22536 * gst/audioconvert/Makefile.am:
22537 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
22538 (check_default), (audio_convert_prepare_context),
22539 (audio_convert_clean_context), (audio_convert_convert):
22540 * gst/audioconvert/audioconvert.h:
22541 * gst/audioconvert/gstaudioconvert.c:
22542 (gst_audio_convert_dithering_get_type),
22543 (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
22544 (gst_audio_convert_init), (gst_audio_convert_set_caps),
22545 (gst_audio_convert_set_property), (gst_audio_convert_get_property):
22546 * gst/audioconvert/gstaudioconvert.h:
22547 * gst/audioconvert/gstaudioquantize.c:
22548 (gst_audio_quantize_setup_noise_shaping),
22549 (gst_audio_quantize_free_noise_shaping),
22550 (gst_audio_quantize_setup_dither),
22551 (gst_audio_quantize_free_dither),
22552 (gst_audio_quantize_setup_quantize_func),
22553 (gst_audio_quantize_setup), (gst_audio_quantize_free):
22554 * gst/audioconvert/gstaudioquantize.h:
22555 Implement dithering and noise shaping in audioconvert. By default now
22556 TPDF dithering (and no noise shaping) will be used when converting
22557 from a higher bit depth to 20 bit depth or smaller, otherwise
22558 everything will be as it is now.
22559 For the last audioconvert in a pipeline it would make sense to
22560 use some kind of noise shaping, enabling it by default for all
22561 conversions would give undesired results though. Fixes #360246.
22562 * tests/check/elements/audioconvert.c: (setup_audioconvert),
22564 Adjust unit test for the new audioconvert.
22566 2007-06-28 11:06:56 +0000 Wim Taymans <wim.taymans@gmail.com>
22568 gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
22569 Original commit message from CVS:
22570 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
22571 Use other metrics as well when estimating the buffer level.
22573 2007-06-28 10:21:19 +0000 Wim Taymans <wim.taymans@gmail.com>
22575 gst/playback/gstplaybasebin.c: Small debug improvement.
22576 Original commit message from CVS:
22577 * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
22578 Small debug improvement.
22579 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
22581 Tweak the rate estimation period.
22582 When calculating the buffer filledness in rate estimation mode, don't
22583 mix it with other metrics.
22585 2007-06-28 09:46:11 +0000 Wim Taymans <wim.taymans@gmail.com>
22587 gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
22588 Original commit message from CVS:
22589 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
22590 (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
22591 When creating the groups, allow for a 5 second, unlimited buffers
22592 preroll phase after which we expose the group.
22593 When the group is exposed, use a small number of buffers up to a 2
22594 second limit. Also disconnect the overrun signal from multiqueue when we
22595 exposed the group because it is not needed anymore.
22597 2007-06-27 22:30:19 +0000 Tim-Philipp Müller <tim@centricular.net>
22599 gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
22600 Original commit message from CVS:
22601 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
22602 Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
22603 to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
22604 (#451707); also, output some debugging info when dealing with
22606 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
22607 Add unit test for the above.
22609 2007-06-27 12:55:20 +0000 Tim-Philipp Müller <tim@centricular.net>
22611 gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
22612 Original commit message from CVS:
22613 * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
22614 Add description for Windows Media RTP caps.
22615 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
22616 Remove RTP fields that don't define the format from caps.
22618 2007-06-27 10:14:03 +0000 Tim-Philipp Müller <tim@centricular.net>
22620 ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
22621 Original commit message from CVS:
22622 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
22623 Skip empty buffers, but not empty header buffers. That way the original
22624 vorbisdec unit test still passes (#451145); also, take into account
22625 that those empty packets might carry a granulepos.
22626 * tests/check/Makefile.am:
22627 * tests/check/elements/vorbisdec.c:
22628 (_create_codebook_header_buffer), (_create_audio_buffer),
22629 (GST_START_TEST), (vorbisdec_suite):
22630 Add unit test that sends an empty packet.
22632 2007-06-27 09:49:51 +0000 Wim Taymans <wim.taymans@gmail.com>
22634 ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
22635 Original commit message from CVS:
22636 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
22637 Don't error out on 0-sized packets, just emit a warning because this is
22638 not a fatal error. Fixes #451145.
22640 2007-06-25 12:43:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22642 docs/plugins/: Update docs with caps info.
22643 Original commit message from CVS:
22644 * docs/plugins/gst-plugins-base-plugins.args:
22645 * docs/plugins/gst-plugins-base-plugins.signals:
22646 * docs/plugins/inspect/plugin-adder.xml:
22647 * docs/plugins/inspect/plugin-alsa.xml:
22648 * docs/plugins/inspect/plugin-audioconvert.xml:
22649 * docs/plugins/inspect/plugin-audiorate.xml:
22650 * docs/plugins/inspect/plugin-audioresample.xml:
22651 * docs/plugins/inspect/plugin-audiotestsrc.xml:
22652 * docs/plugins/inspect/plugin-cdparanoia.xml:
22653 * docs/plugins/inspect/plugin-decodebin.xml:
22654 * docs/plugins/inspect/plugin-decodebin2.xml:
22655 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
22656 * docs/plugins/inspect/plugin-gdp.xml:
22657 * docs/plugins/inspect/plugin-gnomevfs.xml:
22658 * docs/plugins/inspect/plugin-libvisual.xml:
22659 * docs/plugins/inspect/plugin-ogg.xml:
22660 * docs/plugins/inspect/plugin-pango.xml:
22661 * docs/plugins/inspect/plugin-playbin.xml:
22662 * docs/plugins/inspect/plugin-subparse.xml:
22663 * docs/plugins/inspect/plugin-tcp.xml:
22664 * docs/plugins/inspect/plugin-theora.xml:
22665 * docs/plugins/inspect/plugin-typefindfunctions.xml:
22666 * docs/plugins/inspect/plugin-video4linux.xml:
22667 * docs/plugins/inspect/plugin-videorate.xml:
22668 * docs/plugins/inspect/plugin-videoscale.xml:
22669 * docs/plugins/inspect/plugin-videotestsrc.xml:
22670 * docs/plugins/inspect/plugin-volume.xml:
22671 * docs/plugins/inspect/plugin-vorbis.xml:
22672 * docs/plugins/inspect/plugin-ximagesink.xml:
22673 * docs/plugins/inspect/plugin-xvimagesink.xml:
22674 Update docs with caps info.
22676 2007-06-25 12:04:15 +0000 Tim-Philipp Müller <tim@centricular.net>
22678 po/POTFILES.in: Add more files with translatable strings (#450875).
22679 Original commit message from CVS:
22681 Add more files with translatable strings (#450875).
22683 2007-06-23 14:44:07 +0000 Edward Hervey <bilboed@bilboed.com>
22685 ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
22686 Original commit message from CVS:
22687 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
22688 The chain should be freed if we error out here, else it will leak.
22689 * gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
22690 (cleanup_decodebin):
22691 Don't forget to *properly* remove the signals, else it will leak.
22693 2007-06-22 14:25:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22695 MAINTAINERS: Updating all the maintainers files
22696 Original commit message from CVS:
22698 Updating all the maintainers files
22700 2007-06-21 08:34:46 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22702 tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
22703 Original commit message from CVS:
22704 * tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
22706 Destroy and recreate parse-launch based pipeline after stop to be able
22707 to play again. Reorder some code and add more comments.
22709 2007-06-20 11:09:03 +0000 Wim Taymans <wim@fluendo.com>
22711 gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
22712 Original commit message from CVS:
22713 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
22714 When handling a delayed-caps notification case, mark
22715 the group as dynamic so that the nbdynamic count is
22716 incremented and decremented correctly. Fixes: #449156
22717 Patch by: Wim Taymans <wim@fluendo.com>
22719 2007-06-19 19:13:04 +0000 Andy Wingo <wingo@pobox.com>
22722 * gst-libs/gst/audio/gstbaseaudiosink.c:
22723 * win32/common/config.h:
22724 gst-libs/gst/audio/gstbaseaudiosink.c
22725 Original commit message from CVS:
22726 2007-06-19 Andy Wingo <wingo@pobox.com>
22727 * gst-libs/gst/audio/gstbaseaudiosink.c
22728 (gst_base_audio_sink_init): Enable pull-mode operation.
22730 2007-06-19 09:34:35 +0000 Michael Smith <msmith@xiph.org>
22732 gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
22733 Original commit message from CVS:
22734 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
22735 Change minimum rate back to 1000 to allow low-sample-rate wav files
22738 2007-06-17 17:27:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
22740 po/vi.po: Update translations.
22741 Original commit message from CVS:
22743 Update translations.
22745 2007-06-16 03:42:14 +0000 David Schleef <ds@schleef.org>
22747 gst/playback/gstqueue2.c: Fix compile error from ignored return value.
22748 Original commit message from CVS:
22749 * gst/playback/gstqueue2.c:
22750 Fix compile error from ignored return value.
22752 2007-06-15 15:23:36 +0000 Michael Smith <msmith@xiph.org>
22754 gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
22755 Original commit message from CVS:
22756 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
22757 Update tmpbuf for all neccesary rows, not just one, as is required
22761 2007-06-15 11:15:28 +0000 Michael Smith <msmith@xiph.org>
22763 tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
22764 Original commit message from CVS:
22765 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
22766 (eos_buffer_probe):
22767 Add a test that ensures we set DELTA_UNIT on all non-header,
22768 non-video buffers, if we have a video stream.
22769 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
22770 (gst_ogg_mux_process_best_pad):
22771 Move setting delta_pad to earlier, where we inspect all pads, so
22772 that leading audio pages don't get DELTA_UNIT unset if they come
22773 before the first DELTA_UNIT from video pages. Fixes the newly-added
22774 test. Fixes #385527.
22776 2007-06-14 19:53:27 +0000 Tim-Philipp Müller <tim@centricular.net>
22778 tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
22779 Original commit message from CVS:
22780 * tests/check/pipelines/streamheader.c: (streamheader_suite):
22781 Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
22782 fails on the p5-ppc64 build bot and the failure looks like it is due
22783 to the same issue as #348114, ie. a compiler bug.
22785 2007-06-13 18:20:57 +0000 Edward Hervey <bilboed@bilboed.com>
22787 gst/playback/gstqueue2.c: Fix build on MacOSX.
22788 Original commit message from CVS:
22789 * gst/playback/gstqueue2.c: (gst_queue_create_read):
22790 Fix build on MacOSX.
22792 2007-06-13 09:01:32 +0000 Wim Taymans <wim.taymans@gmail.com>
22794 ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
22795 Original commit message from CVS:
22796 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
22797 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
22798 Fix compilation on mingw. Fixes #446972.
22800 2007-06-12 08:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
22802 gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
22803 Original commit message from CVS:
22804 Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
22805 * gst/playback/gstqueue2.c: (update_buffering),
22806 (gst_queue_locked_enqueue):
22807 Fix a division by zero when the max percent is <= 0. Fixes #446572.
22808 also update the buffering status when receiving events. Fixes #446551.
22810 2007-06-11 11:32:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
22812 gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
22813 Original commit message from CVS:
22814 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
22815 * gst/playback/gstqueue2.c: (gst_queue_peer_query),
22816 (gst_queue_handle_src_query):
22817 Wait for preroll before attempting to forward a duration query upstream.
22820 2007-06-07 21:08:38 +0000 Sébastien Moutte <sebastien@moutte.net>
22822 gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
22823 Original commit message from CVS:
22824 * gst-libs/gst/rtp/gstbasertpdepayload.c:
22825 (gst_base_rtp_depayload_set_gst_timestamp):
22826 Use G_GINT64_CONSTANT macro for int64 constant.
22827 * win32/common/libgstinterfaces.def:
22828 * win32/common/libgsttag.def:
22829 Add new exported functions.
22831 2007-06-07 14:25:32 +0000 Tim-Philipp Müller <tim@centricular.net>
22833 ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
22834 Original commit message from CVS:
22835 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
22836 The BOS page of the first Dirac video stream needs to come before
22837 the BOS page of any Vorbis streams or other audio streams, just like
22840 2007-06-07 09:11:27 +0000 Wim Taymans <wim.taymans@gmail.com>
22842 gst/playback/gstqueue2.c: Fix compilation.
22843 Original commit message from CVS:
22844 * gst/playback/gstqueue2.c: (gst_queue_get_range):
22847 2007-06-06 13:36:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
22849 gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
22850 Original commit message from CVS:
22851 Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
22852 * gst/playback/gstqueue2.c: (gst_queue_init),
22853 (gst_queue_handle_sink_event), (gst_queue_chain),
22854 (gst_queue_get_range), (gst_queue_src_checkgetrange_function),
22855 (gst_queue_sink_activate_push), (gst_queue_src_activate_push),
22856 (gst_queue_src_activate_pull):
22857 Add pull based scheduling and fix some deadlocks. Fixes #444523.
22858 Does not yet completely work because duration queries upstream won't
22861 2007-06-06 09:08:50 +0000 Wim Taymans <wim.taymans@gmail.com>
22863 Some more fseeko checks.
22864 Original commit message from CVS:
22866 * gst/playback/gstqueue2.c: (gst_queue_create_read):
22867 Some more fseeko checks.
22869 2007-06-06 08:01:42 +0000 Wim Taymans <wim.taymans@gmail.com>
22871 configure.ac: check for large file support.
22872 Original commit message from CVS:
22874 check for large file support.
22876 2007-06-05 21:36:11 +0000 Sven Arvidsson <sa@whiz.se>
22878 gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
22879 Original commit message from CVS:
22880 Based on a patch by Sven Arvidsson <sa at whiz dot se>:
22881 * gst/subparse/gstsubparse.c: (parse_subrip),
22882 (subviewer_unescape_newlines), (parse_subviewer),
22883 (gst_sub_parse_data_format_autodetect),
22884 (gst_sub_parse_format_autodetect), (gst_subparse_type_find):
22885 * gst/subparse/gstsubparse.h:
22886 Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
22887 * tests/check/elements/subparse.c: (GST_START_TEST),
22889 Add a unit test for both SubViewer formats.
22891 2007-06-05 17:08:04 +0000 Michael Smith <msmith@xiph.org>
22893 gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
22894 Original commit message from CVS:
22895 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
22896 Don't overflow intermediate values when seeking to large time values
22899 2007-06-05 17:02:13 +0000 Wim Taymans <wim.taymans@gmail.com>
22901 gst/playback/gstqueue2.c: Include stdio to define fseeko.
22902 Original commit message from CVS:
22903 * gst/playback/gstqueue2.c: (gst_queue_have_data),
22904 (gst_queue_create_read), (gst_queue_read_item_from_file),
22905 (gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
22906 Include stdio to define fseeko.
22908 2007-06-05 16:37:09 +0000 Edward Hervey <edward@fluendo.com>
22910 sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
22911 Original commit message from CVS:
22912 Patch by: Edward Hervey <edward@fluendo.com>
22913 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
22914 (gst_v4lsrc_query):
22915 Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
22917 2007-06-05 16:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
22919 gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
22920 Original commit message from CVS:
22921 * gst-libs/gst/riff/Makefile.am:
22922 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
22923 Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
22924 our own implementation.
22926 2007-06-05 16:19:30 +0000 Wim Taymans <wim.taymans@gmail.com>
22928 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
22929 Original commit message from CVS:
22930 * gst-libs/gst/rtp/gstbasertpdepayload.c:
22931 (gst_base_rtp_depayload_setcaps),
22932 (gst_base_rtp_depayload_set_gst_timestamp),
22933 (gst_base_rtp_depayload_change_state):
22934 Handle timestamp wraparound.
22936 2007-06-05 16:17:30 +0000 Wim Taymans <wim.taymans@gmail.com>
22938 gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
22939 Original commit message from CVS:
22940 * gst/playback/gsturidecodebin.c: (no_more_pads_full),
22941 (new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
22942 (gst_uri_decode_bin_change_state):
22943 Make sure we name srcpads uniquely even when using different internal
22945 Signal no-more-pads when no more dynamic elements exist.
22946 Remove pads on cleanup.
22948 2007-06-05 16:14:23 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
22950 gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
22951 Original commit message from CVS:
22952 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
22953 * gst/playback/gstqueue2.c: (gst_queue_class_init),
22954 (gst_queue_init), (gst_queue_finalize),
22955 (gst_queue_write_buffer_to_file), (gst_queue_have_data),
22956 (gst_queue_create_read), (gst_queue_read_item_from_file),
22957 (gst_queue_open_temp_location_file),
22958 (gst_queue_close_temp_location_file), (gst_queue_locked_flush),
22959 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
22960 (gst_queue_is_empty), (gst_queue_is_filled),
22961 (gst_queue_change_state), (gst_queue_set_temp_location),
22962 (gst_queue_set_property):
22963 Add support for filebased buffering. Fixes #441264.
22965 2007-06-05 16:05:19 +0000 Wim Taymans <wim.taymans@gmail.com>
22967 gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
22968 Original commit message from CVS:
22969 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
22970 (analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
22971 (caps_notify_group_cb), (gst_decode_group_new),
22972 (gst_decode_group_free):
22973 Add support for delayed caps fixation when autoplugging.
22974 Optimize cases where a multiqueue is not needed/wanted, like right after
22975 anything that is not a demuxer.
22977 2007-06-05 16:02:57 +0000 Wim Taymans <wim.taymans@gmail.com>
22979 ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
22980 Original commit message from CVS:
22981 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
22982 (gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
22983 (gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
22984 consideratly speedup ogg chain detection by not trying to find a base
22985 timestamp for skeleton streams.
22987 2007-06-05 16:00:33 +0000 Wim Taymans <wim.taymans@gmail.com>
22989 gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
22990 Original commit message from CVS:
22991 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
22992 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
22993 (gst_multi_fd_sink_remove_flush),
22994 (gst_multi_fd_sink_remove_client_link),
22995 (gst_multi_fd_sink_handle_client_write),
22996 (gst_multi_fd_sink_handle_clients):
22997 * gst/tcp/gstmultifdsink.h:
22998 Add support for remuve_flush.
23000 2007-06-05 15:59:00 +0000 Wim Taymans <wim.taymans@gmail.com>
23002 Add draft design for forcing keyframes in encoders and implement in theoraenc.
23003 Original commit message from CVS:
23004 * docs/design/draft-keyframe-force.txt:
23005 * ext/theora/theoraenc.c: (theora_enc_sink_event),
23006 (theora_enc_chain):
23007 Add draft design for forcing keyframes in encoders and implement in
23010 2007-06-05 13:22:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23012 configure.ac: Back to CVS
23013 Original commit message from CVS:
23017 === release 0.10.13 ===
23019 2007-06-05 12:50:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23025 * docs/plugins/gst-plugins-base-plugins.args:
23026 * docs/plugins/inspect/plugin-adder.xml:
23027 * docs/plugins/inspect/plugin-alsa.xml:
23028 * docs/plugins/inspect/plugin-audioconvert.xml:
23029 * docs/plugins/inspect/plugin-audiorate.xml:
23030 * docs/plugins/inspect/plugin-audioresample.xml:
23031 * docs/plugins/inspect/plugin-audiotestsrc.xml:
23032 * docs/plugins/inspect/plugin-cdparanoia.xml:
23033 * docs/plugins/inspect/plugin-decodebin.xml:
23034 * docs/plugins/inspect/plugin-decodebin2.xml:
23035 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
23036 * docs/plugins/inspect/plugin-gdp.xml:
23037 * docs/plugins/inspect/plugin-gnomevfs.xml:
23038 * docs/plugins/inspect/plugin-libvisual.xml:
23039 * docs/plugins/inspect/plugin-ogg.xml:
23040 * docs/plugins/inspect/plugin-pango.xml:
23041 * docs/plugins/inspect/plugin-playbin.xml:
23042 * docs/plugins/inspect/plugin-subparse.xml:
23043 * docs/plugins/inspect/plugin-tcp.xml:
23044 * docs/plugins/inspect/plugin-theora.xml:
23045 * docs/plugins/inspect/plugin-typefindfunctions.xml:
23046 * docs/plugins/inspect/plugin-video4linux.xml:
23047 * docs/plugins/inspect/plugin-videorate.xml:
23048 * docs/plugins/inspect/plugin-videoscale.xml:
23049 * docs/plugins/inspect/plugin-videotestsrc.xml:
23050 * docs/plugins/inspect/plugin-volume.xml:
23051 * docs/plugins/inspect/plugin-vorbis.xml:
23052 * docs/plugins/inspect/plugin-ximagesink.xml:
23053 * docs/plugins/inspect/plugin-xvimagesink.xml:
23054 * gst-plugins-base.doap:
23055 * win32/common/config.h:
23056 * win32/vs6/grammar.dsp:
23057 * win32/vs6/gst_plugins_base.dsw:
23058 * win32/vs6/libgstadder.dsp:
23059 * win32/vs6/libgstaudio.dsp:
23060 * win32/vs6/libgstaudioconvert.dsp:
23061 * win32/vs6/libgstaudiorate.dsp:
23062 * win32/vs6/libgstaudioresample.dsp:
23063 * win32/vs6/libgstaudioscale.dsp:
23064 * win32/vs6/libgstaudiotestsrc.dsp:
23065 * win32/vs6/libgstcdda.dsp:
23066 * win32/vs6/libgstdecodebin.dsp:
23067 * win32/vs6/libgstdecodebin2.dsp:
23068 * win32/vs6/libgstdirectsound.dsp:
23069 * win32/vs6/libgstffmpegcolorspace.dsp:
23070 * win32/vs6/libgstgdp.dsp:
23071 * win32/vs6/libgstinterfaces.dsp:
23072 * win32/vs6/libgstnetbuffer.dsp:
23073 * win32/vs6/libgstogg.dsp:
23074 * win32/vs6/libgstpbutils.dsp:
23075 * win32/vs6/libgstplaybin.dsp:
23076 * win32/vs6/libgstriff.dsp:
23077 * win32/vs6/libgstrtp.dsp:
23078 * win32/vs6/libgstsinesrc.dsp:
23079 * win32/vs6/libgstsubparse.dsp:
23080 * win32/vs6/libgsttag.dsp:
23081 * win32/vs6/libgsttheora.dsp:
23082 * win32/vs6/libgsttypefindfunctions.dsp:
23083 * win32/vs6/libgstutils.dsp:
23084 * win32/vs6/libgstvideo.dsp:
23085 * win32/vs6/libgstvideorate.dsp:
23086 * win32/vs6/libgstvideoscale.dsp:
23087 * win32/vs6/libgstvideotestsrc.dsp:
23088 * win32/vs6/libgstvolume.dsp:
23089 * win32/vs6/libgstvorbis.dsp:
23090 Release 0.10.13 "What's going on?"
23091 Original commit message from CVS:
23092 Release 0.10.13 "What's going on?"
23094 2007-06-05 12:32:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23112 Original commit message from CVS:
23115 2007-05-31 17:08:58 +0000 Wim Taymans <wim@fluendo.com>
23117 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...
23118 Original commit message from CVS:
23119 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
23120 In riff, the depth is stored in the size field but it just means that
23121 the least significant bits are cleared. We can therefore just play
23122 the sample as if it had a depth == width. Fixes: #440997
23123 Patch by: Wim Taymans <wim@fluendo.com>
23124 Patch by: Sebastian Dröge <slomo@circular-chaos.org>
23126 2007-05-31 16:36:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23128 gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
23129 Original commit message from CVS:
23130 * gst-libs/gst/floatcast/floatcast.h:
23131 Define inline when needed on win32 builds. Fixes: #441295
23133 2007-05-29 13:38:35 +0000 Wim Taymans <wim.taymans@gmail.com>
23135 gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
23136 Original commit message from CVS:
23137 * gst/playback/gstplaybasebin.c: (queue_overrun),
23138 (no_more_pads_full):
23139 Stop buffering when the group is commited because the queues filled up.
23142 2007-05-25 10:07:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23144 Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
23145 Original commit message from CVS:
23146 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
23147 (gst_alsa_mixer_free), (gst_alsa_mixer_update),
23148 (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
23149 (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
23150 (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
23151 * ext/alsa/gstalsamixer.h:
23152 * ext/alsa/gstalsamixerelement.c:
23153 (gst_alsa_mixer_element_interface_supported),
23154 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
23155 (gst_alsa_mixer_element_set_property),
23156 (gst_alsa_mixer_element_get_property),
23157 (gst_alsa_mixer_element_change_state):
23158 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
23159 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
23160 (gst_mixer_option_changed):
23161 * gst-libs/gst/interfaces/mixer.h:
23162 Revert commits towards #152864 made so far. We'll pick it up again
23163 after the 0.10.13 release.
23165 2007-05-24 16:22:23 +0000 Wim Taymans <wim.taymans@gmail.com>
23167 gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
23168 Original commit message from CVS:
23169 * gst-libs/gst/audio/gstbaseaudiosink.c:
23170 (gst_base_audio_sink_render):
23171 After an interrupt (PAUSED/flush) assume that the next sample should not
23172 be aligned to the previous sample. Fixes #417992.
23174 2007-05-24 15:16:59 +0000 Tim-Philipp Müller <tim@centricular.net>
23176 gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
23177 Original commit message from CVS:
23178 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
23179 Don't add channels and rate fields to the template caps for
23180 audio/x-dts, as wavparse might not always be able to set them,
23181 which would then lead to 'caps are not a real subset of the
23182 template caps' warnings.
23184 2007-05-24 11:15:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23186 gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
23187 Original commit message from CVS:
23188 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
23189 Handle unknown or invalid pads without crashing, as might occur if
23190 a media file like an mp3 is specified as a subtitle file.
23193 2007-05-24 10:19:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23195 gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
23196 Original commit message from CVS:
23197 * gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
23199 Block the subtitle bin output queue before ghosting it and linking,
23200 then unblock after. This avoids spurious not-linked errors caused
23201 by the queue starting up (because it gets linked when it is ghosted).
23204 2007-05-23 15:54:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23206 tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
23207 Original commit message from CVS:
23208 * tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
23209 Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
23210 file. Avoids flukes where the input gets typefound to some valid but
23213 2007-05-22 15:45:19 +0000 Tim-Philipp Müller <tim@centricular.net>
23215 tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
23216 Original commit message from CVS:
23217 * tests/check/Makefile.am:
23218 * tests/check/elements/.cvsignore:
23219 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
23220 (cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
23221 Add unit test for gnomevfssink seeking and position reporting for
23224 2007-05-22 15:30:26 +0000 Mark Nauwelaerts <manauw@skynet.be>
23226 ext/gnomevfs/gstgnomevfssink.*: see #412648.
23227 Original commit message from CVS:
23228 Patch by: Mark Nauwelaerts <manauw at skynet be>
23229 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
23230 (gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
23231 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
23232 * ext/gnomevfs/gstgnomevfssink.h:
23233 Fix position reporting, especially after a seek (from upstream),
23236 2007-05-22 15:04:41 +0000 Tim-Philipp Müller <tim@centricular.net>
23238 ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
23239 Original commit message from CVS:
23240 * ext/cdparanoia/gstcdparanoiasrc.c:
23243 2007-05-22 11:40:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23245 gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
23246 Original commit message from CVS:
23247 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
23248 Specify the full valid range for MP3 samplerates. Fixes a regression
23249 caused by extra header checks since the last release.
23251 2007-05-21 15:32:42 +0000 Mike Smith <msmith@xiph.org>
23253 sys/: Fix a locking-order bug I introduced with my changes the other day.
23254 Original commit message from CVS:
23255 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
23256 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
23257 Fix a locking-order bug I introduced with my changes the other day.
23258 Patch by Mike Smith.
23260 2007-05-21 15:24:21 +0000 Michael Smith <msmith@xiph.org>
23262 ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
23263 Original commit message from CVS:
23264 * ext/theora/theoradec.c: (theora_handle_data_packet):
23265 Don't look inside 0-length packets (which indicate duplicated
23268 2007-05-21 10:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
23271 Original commit message from CVS:
23272 * ext/cdparanoia/gstcdparanoiasrc.c:
23273 (gst_cd_paranoia_src_read_sector):
23274 * gst-libs/gst/audio/gstbaseaudiosrc.c:
23275 (gst_base_audio_src_create):
23277 * ext/theora/theoradec.c: (theora_dec_sink_event):
23279 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23280 (gst_base_rtp_depayload_set_gst_timestamp):
23282 * gst/playback/gstdecodebin.c: (queue_underrun_cb):
23283 And some debug info when a FIXME path is hit.
23285 2007-05-21 09:45:28 +0000 Wim Taymans <wim.taymans@gmail.com>
23287 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
23288 Original commit message from CVS:
23289 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23290 (gst_base_rtp_audio_payload_class_init),
23291 (gst_base_rtp_audio_payload_init),
23292 (gst_base_rtp_audio_payload_finalize),
23293 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
23294 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
23295 (gst_base_rtp_payload_audio_handle_event):
23296 Some cleanups, remove minptime property as it is now in the parent
23298 Override parent class event function.
23299 * gst-libs/gst/rtp/gstbasertppayload.c:
23300 (gst_basertppayload_class_init), (gst_basertppayload_init),
23301 (gst_basertppayload_event), (gst_basertppayload_set_property),
23302 (gst_basertppayload_get_property):
23303 * gst-libs/gst/rtp/gstbasertppayload.h:
23304 Add min-ptime property.
23305 Add handle-event vmethod. Fixes #415001.
23307 2007-05-18 17:10:03 +0000 Christian Schaller <uraeus@gnome.org>
23309 * gst-plugins-base.spec.in:
23311 Original commit message from CVS:
23314 2007-05-18 15:23:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23316 gst-libs/gst/audio/gstbaseaudiosink.c
23317 Original commit message from CVS:
23318 * gst-libs/gst/audio/gstbaseaudiosink.c
23319 (gst_base_audio_sink_change_state):
23320 Fix typo in comment.
23321 * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
23322 free_dynamics, pad_probe, close_pad_link, try_to_link_1,
23323 get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
23325 * gst/playback/gstplaybin.c (gst_play_bin_set_property,
23326 gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
23327 Remove trailing whitespaces in comments.
23328 * gst/volume/Makefile.am:
23331 2007-05-18 15:10:08 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
23334 * gst-libs/gst/interfaces/mixer.h:
23335 gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
23336 Original commit message from CVS:
23337 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
23338 * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
23339 set_option, get_option, _gst_reserved):
23340 Revert reordering functions (keep ABI).
23342 2007-05-17 17:35:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23344 sys/: When we create our own window, indicate that we handle the
23345 Original commit message from CVS:
23346 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
23347 (gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
23348 (gst_ximagesink_show_frame):
23349 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
23350 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
23351 (gst_xvimagesink_show_frame):
23352 When we create our own window, indicate that we handle the
23353 WM_DELETE client message from the window manager, so that it won't
23354 kill our window (and our app) along with it. Handle ClientMessage,
23355 post an error on the bus, and close the window. Further buffers
23356 arriving will result in a FlowError because the window has been
23359 Clean up the X event handling loop and make them the same for
23360 both xvimagesink and ximagesink while I'm at it.
23362 2007-05-17 16:27:32 +0000 Wim Taymans <wim.taymans@gmail.com>
23364 gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
23365 Original commit message from CVS:
23366 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
23367 Make decodebin2 autoplug depayloaders too.
23368 * gst/playback/gsturidecodebin.c: (source_new_pad):
23369 Set the newly created decoder in a usable state when autoplugging a
23370 dynamic source such as RTSP.
23372 2007-05-17 16:11:03 +0000 Tim-Philipp Müller <tim@centricular.net>
23374 gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
23375 Original commit message from CVS:
23376 * gst/playback/gststreaminfo.c: (cb_probe):
23377 Ignore video-codec tag for audio streams and ignore audio-codec tags
23378 for video streams. Should make codec name collection a bit more
23379 robust against sloppy demuxers that send tag events containing both
23380 tags down each pad.
23382 2007-05-17 15:22:44 +0000 Wim Taymans <wim.taymans@gmail.com>
23384 gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
23385 Original commit message from CVS:
23386 * gst/playback/gstqueue2.c: (update_rates):
23387 Tweak the buffering thresholds a little.
23388 Update the buffer size with the previously calculate rate instead of
23389 only when we calculate a new rate so that we get smoother buffering
23391 * gst/playback/Makefile.am:
23392 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
23393 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
23394 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
23395 (gst_uri_decode_bin_get_property), (unknown_type),
23396 (add_element_stream), (no_more_pads_full), (no_more_pads),
23397 (source_no_more_pads), (new_decoded_pad), (array_has_value),
23398 (gen_source_element), (has_all_raw_caps), (analyse_source),
23399 (remove_decoders), (make_decoder), (remove_source),
23400 (source_new_pad), (setup_source), (decoder_query_init),
23401 (decoder_query_duration_fold), (decoder_query_duration_done),
23402 (decoder_query_position_fold), (decoder_query_position_done),
23403 (decoder_query_latency_fold), (decoder_query_latency_done),
23404 (decoder_query_seeking_fold), (decoder_query_seeking_done),
23405 (decoder_query_generic_fold), (gst_uri_decode_bin_query),
23406 (gst_uri_decode_bin_change_state), (plugin_init):
23407 New element that intergrates a source, optional buffering element and
23410 2007-05-17 14:17:17 +0000 Tim-Philipp Müller <tim@centricular.net>
23412 configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
23413 Original commit message from CVS:
23415 Bump libtheora requirement to 1.0alpha5 for the pixformat check
23416 (also has a .pc file, so we don't need the fallback check any
23417 longer). Fixes #438840.
23419 2007-05-17 13:36:11 +0000 Wim Taymans <wim.taymans@gmail.com>
23421 gst/playback/gstqueue2.c: fix build.
23422 Original commit message from CVS:
23423 * gst/playback/gstqueue2.c: (gst_queue_get_type),
23424 (gst_queue_class_init), (gst_queue_finalize), (update_time_level),
23425 (apply_segment), (apply_buffer), (update_buffering),
23426 (reset_rate_timer), (update_rates), (gst_queue_locked_flush),
23427 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
23428 (gst_queue_handle_sink_event), (gst_queue_is_filled),
23429 (gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
23433 2007-05-17 11:57:44 +0000 Wim Taymans <wim.taymans@gmail.com>
23435 gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
23436 Original commit message from CVS:
23437 * gst/playback/Makefile.am:
23438 * gst/playback/gstqueue2.c: (gst_queue_get_type),
23439 (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
23440 (gst_queue_getcaps), (gst_queue_bufferalloc),
23441 (gst_queue_acceptcaps), (update_time_level), (apply_segment),
23442 (apply_buffer), (update_buffering), (reset_rate_timer),
23443 (update_rates), (gst_queue_locked_flush),
23444 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
23445 (gst_queue_handle_sink_event), (gst_queue_is_empty),
23446 (gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
23447 (gst_queue_loop), (gst_queue_handle_src_event),
23448 (gst_queue_handle_src_query), (gst_queue_sink_activate_push),
23449 (gst_queue_src_activate_push), (gst_queue_change_state),
23450 (gst_queue_set_property), (gst_queue_get_property), (plugin_init):
23451 On our way to playbin2 this is the new network queue that does buffering
23452 all by itself using high and low watermarks. It can also measure up and
23453 downstream bandwidth to optimally size the queue.
23455 2007-05-17 11:16:14 +0000 Michael Smith <msmith@xiph.org>
23457 gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
23458 Original commit message from CVS:
23459 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
23460 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
23461 Use the segment->last_stop value to calculate the next timestamp to
23462 generate after a seek; not the segment->start value.
23464 2007-05-15 20:14:06 +0000 David Schleef <ds@schleef.org>
23466 docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
23467 Original commit message from CVS:
23468 * docs/Makefile.am: Install docs even when --disable-gtk-doc
23469 is disabled. This matches the behavior of gtk+. Fixes #349099.
23471 2007-05-15 17:11:09 +0000 Wim Taymans <wim.taymans@gmail.com>
23473 ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
23474 Original commit message from CVS:
23475 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
23476 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
23477 Some more chained streaming ogg timestamp fixes.
23479 2007-05-15 16:46:10 +0000 Wim Taymans <wim.taymans@gmail.com>
23481 ext/ogg/gstoggdemux.c: Add some FIXMEs.
23482 Original commit message from CVS:
23483 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
23484 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
23485 (gst_ogg_demux_handle_page):
23487 Fix chain start/stop segment handling based on patch by
23488 <ahalda at cs dot mcgill dot ca> see #320984.
23490 2007-05-15 15:33:54 +0000 Michael Smith <msmith@xiph.org>
23492 configure.ac: We don't require a C++ compiler. So don't require one.
23493 Original commit message from CVS:
23495 We don't require a C++ compiler. So don't require one.
23497 2007-05-15 15:29:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23500 * ext/alsa/gstalsamixer.c:
23501 ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
23502 Original commit message from CVS:
23503 * ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
23504 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
23505 gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
23506 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
23507 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
23508 gst_alsa_mixer_update_track):
23509 Apply some of the cleanup Tim suggested in #152864 afterwards.
23511 2007-05-15 14:01:26 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
23513 ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
23514 Original commit message from CVS:
23515 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
23516 * ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
23517 _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
23518 gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
23519 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
23520 gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
23521 gst_alsa_mixer_handle_source_callback,
23522 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
23523 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
23524 gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
23525 gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
23526 gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
23527 gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
23528 * ext/alsa/gstalsamixer.h (handle_source, interface, dir):
23529 * ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
23530 gst_alsa_mixer_element_interface_supported,
23531 gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
23532 gst_alsa_mixer_element_set_property,
23533 gst_alsa_mixer_element_get_property,
23534 gst_alsa_mixer_element_change_state):
23535 * ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
23536 * gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
23537 gst_mixer_option_changed):
23538 * gst-libs/gst/interfaces/mixer.h (set_option, get_option,
23539 volume_changed, option_changed, _gst_reserved):
23540 Implement notification for alsamixer. Fixes #152864
23542 2007-05-15 03:53:11 +0000 David Schleef <ds@schleef.org>
23544 gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
23545 Original commit message from CVS:
23546 * gst/videotestsrc/videotestsrc.c:
23547 * gst/videotestsrc/videotestsrc.h:
23548 Add support for video/x-raw-bayer.
23550 2007-05-13 01:06:19 +0000 David Schleef <ds@schleef.org>
23552 sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
23553 Original commit message from CVS:
23554 * sys/xvimage/xvimagesink.c:
23555 Add some sanity checking for the XVImage size returned by X.
23556 Related to #377400.
23558 2007-05-12 16:18:39 +0000 Wim Taymans <wim.taymans@gmail.com>
23560 gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
23561 Original commit message from CVS:
23562 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23563 (gst_base_rtp_depayload_setcaps),
23564 (gst_base_rtp_depayload_set_gst_timestamp):
23565 Parse and use additional caps fields as described in updated
23566 application/x-rtp caps spec.
23568 2007-05-12 16:16:22 +0000 Wim Taymans <wim.taymans@gmail.com>
23570 ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
23571 Original commit message from CVS:
23572 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
23573 (gst_ogg_demux_collect_chain_info):
23574 If there is a stream in a chain without any data packets, ignore the
23575 stream in the total length calculations. Might be related to #436820.
23577 2007-05-11 17:33:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23579 gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
23580 Original commit message from CVS:
23581 * gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
23582 (mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
23583 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
23584 (mpeg_video_type_find), (mpeg_video_stream_type_find),
23586 Consolidate and re-work our mpeg system stream detection to probe
23587 more packets and produce a higher confidence result. Fixes a
23588 regression caused by lowering the typefind probability last year
23589 - related to bug #397810. Remove the redundant MPEG-1 specific
23590 typefind function, as the new one detects both MPEG-1 & MPEG-2
23592 Also cleanup the MPEG elementary and MPEG-TS detection functions a
23594 Tested against my media test directory, with some improvements and
23597 2007-05-10 15:28:13 +0000 Wim Taymans <wim.taymans@gmail.com>
23599 gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
23600 Original commit message from CVS:
23601 * gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
23602 (queue_out_of_data):
23603 Connect to the new queue "pushing" signal instead of the broken
23606 2007-05-09 21:17:40 +0000 Sébastien Moutte <sebastien@moutte.net>
23608 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
23609 Original commit message from CVS:
23610 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23611 (gst_base_rtp_audio_payload_handle_frame_based_buffer):
23612 Move variable declaration before the first instruction.
23613 * gst/videotestsrc/videotestsrc.c:
23614 Define M_PI if it's not defined yet.
23615 * win32/common/libgstrtp.def:
23616 Add new exported functions.
23618 2007-05-09 11:54:32 +0000 Michael Smith <msmith@xiph.org>
23620 ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
23621 Original commit message from CVS:
23622 * ext/theora/theoradec.c: (theora_handle_type_packet):
23623 gst_pad_push_event() does not return a GstFlowReturn!
23625 2007-05-09 11:25:34 +0000 Wim Taymans <wim.taymans@gmail.com>
23627 tests/examples/seek/: Some small cosmetic changes.
23628 Original commit message from CVS:
23629 * tests/examples/seek/scrubby.c: (stop_cb), (main):
23630 * tests/examples/seek/seek.c: (do_seek):
23631 Some small cosmetic changes.
23633 2007-05-08 19:24:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23636 * gst/adder/gstadder.c:
23637 * gst/adder/gstadder.h:
23638 gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
23639 Original commit message from CVS:
23640 * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
23641 gst_adder_change_state):
23642 * gst/adder/gstadder.h (bps, offset, collect_event, segment,
23643 segment_pending, segment_position, segment_rate):
23644 Handle playback-rate on adder.
23646 2007-05-07 11:43:31 +0000 Michael Smith <msmith@xiph.org>
23648 ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
23649 Original commit message from CVS:
23650 * ext/theora/gsttheoradec.h:
23651 * ext/theora/theoradec.c: (gst_theora_dec_reset),
23652 (theora_dec_sink_event), (theora_handle_comment_packet),
23653 (theora_handle_type_packet), (theora_dec_change_state):
23654 Don't push events (newsegment, tags) before initialising the
23656 This is neccesary for seeking to work correctly in gnonlin.
23658 2007-05-04 13:10:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23660 gst/: gst/audiotestsrc/gstaudiotestsrc.c
23661 Original commit message from CVS:
23662 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23663 * gst/adder/gstadder.c:
23664 * gst/audiotestsrc/gstaudiotestsrc.c
23665 (gst_audio_test_src_create_white_noise):
23666 * gst/videotestsrc/gstvideotestsrc.c:
23667 * gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
23668 VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
23669 volume_sink_template, volume_src_template, gst_volume_init,
23670 volume_process_double, volume_process_int16,
23671 volume_process_int16_clamp):
23672 Doc fixes and formatting.
23674 2007-05-04 12:41:21 +0000 Tim-Philipp Müller <tim@centricular.net>
23676 tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
23677 Original commit message from CVS:
23678 * tests/check/Makefile.am:
23679 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
23680 Minimal check for volume's GstController usability; also another
23683 2007-05-04 09:06:38 +0000 Tim-Philipp Müller <tim@centricular.net>
23685 gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
23686 Original commit message from CVS:
23687 * gst-libs/gst/cdda/gstcddabasesrc.c:
23688 (gst_cdda_base_src_add_track):
23689 Fix it so that it (a) makes sense and (b) doesn't break
23690 everything cdda-related including the unit test.
23692 2007-05-04 08:46:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23694 gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
23695 Original commit message from CVS:
23696 * gst-libs/gst/cdda/gstcddabasesrc.c:
23697 (gst_cdda_base_src_add_track):
23698 Fix build when disabling asserts.
23700 2007-05-03 16:29:10 +0000 Tim-Philipp Müller <tim@centricular.net>
23702 sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
23703 Original commit message from CVS:
23704 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
23705 When XShm is not available, we might get row strides that are not
23706 rounded up to multiples of four; this is bad, because virtually
23707 every RGB-processing element in GStreamer assumes rowstrides are
23708 rounded up to multiples of four, so let's allocate at least enough
23709 memory to avoid crashes in this case. The image will still be
23710 displayed distorted though if this happens, so that still needs
23711 fixing (maybe by allocating a bigger image with an 'even' width
23712 and then clipping it appropriately when rendering - something for
23713 Xlib aficionados in any case).
23715 2007-05-03 13:16:21 +0000 Michael Smith <msmith@xiph.org>
23717 gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
23718 Original commit message from CVS:
23719 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
23720 If a buffer doesn't have a timestamp, assume it's contiguous with
23721 the previous buffer, and synthesise timestamps appropriately.
23723 2007-05-03 11:24:00 +0000 Edward Hervey <bilboed@bilboed.com>
23725 tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
23726 Original commit message from CVS:
23727 * tests/check/elements/videorate.c: (GST_START_TEST):
23728 Set buffer timestamp to a valid value in order to test the buffer
23729 really does stay in videorate.
23731 2007-05-03 10:47:22 +0000 Edward Hervey <bilboed@bilboed.com>
23733 gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
23734 Original commit message from CVS:
23735 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
23736 There is no sensible way to handle incoming buffers which don't have a
23737 valid timestamp. We therefore discard them and wait for the next one.
23739 2007-05-01 18:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
23741 gst/playback/: Better error message for text files.
23742 Original commit message from CVS:
23743 * gst/playback/gstdecodebin.c: (type_found), (plugin_init):
23744 * gst/playback/gstdecodebin2.c: (plugin_init):
23745 Better error message for text files.
23747 2007-04-29 14:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
23749 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
23750 Original commit message from CVS:
23751 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
23752 Fix offset bug in generation RR packets.
23754 2007-04-27 15:33:46 +0000 Julien Moutte <julien@moutte.net>
23756 ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
23757 Original commit message from CVS:
23758 2007-04-27 Julien MOUTTE <julien@moutte.net>
23759 * ext/theora/theoradec.c: (_theora_granule_time),
23760 (theora_dec_push_forward), (theora_handle_data_packet),
23761 (theora_dec_decode_buffer): Calculate buffer duration correctly
23762 to generate a perfect stream (#433888).
23763 * gst/audioresample/gstaudioresample.c:
23764 (audioresample_check_discont): Glib provides ABS.
23766 2007-04-27 15:01:40 +0000 Wim Taymans <wim.taymans@gmail.com>
23768 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
23769 Original commit message from CVS:
23770 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
23771 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
23772 (gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
23773 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
23774 (gst_rtcp_packet_bye_set_reason):
23775 * gst-libs/gst/rtp/gstrtcpbuffer.h:
23776 Fix RB block parsing and writing.
23777 Add support for constructing BYE packets.
23779 2007-04-25 08:54:34 +0000 Tim-Philipp Müller <tim@centricular.net>
23781 When posting a warning message because samples were dropped, post something more intelligible than he default error m...
23782 Original commit message from CVS:
23783 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
23784 (gst_base_audio_src_create):
23786 When posting a warning message because samples were dropped, post
23787 something more intelligible than he default error message for clock
23788 errors which is just confusing in this context (#432984).
23790 2007-04-25 08:10:26 +0000 Wim Taymans <wim.taymans@gmail.com>
23792 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
23793 Original commit message from CVS:
23794 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
23795 (gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
23796 (read_packet_header), (gst_rtcp_packet_move_to_next),
23797 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
23798 (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
23799 (gst_rtcp_packet_sdes_get_item_count),
23800 (gst_rtcp_packet_sdes_first_item),
23801 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
23802 (gst_rtcp_packet_sdes_first_entry),
23803 (gst_rtcp_packet_sdes_next_entry),
23804 (gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
23805 (gst_rtcp_packet_sdes_add_entry):
23806 * gst-libs/gst/rtp/gstrtcpbuffer.h:
23807 Implement code to write SR, RR and SDES packets.
23809 2007-04-24 20:45:24 +0000 Christian Kirbach <Christian.Kirbach@googlemail.com>
23811 sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
23812 Original commit message from CVS:
23813 Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
23814 * sys/ximage/ximagesink.c:
23815 Fix build if XShm is not available (#432362).
23817 2007-04-24 18:58:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
23819 gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
23820 Original commit message from CVS:
23821 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
23822 Initalize the AudioConvertCtx with zeroes, otherwise it will contain
23823 pointers to random memory which are passed to g_free() when
23824 audio_convert_prepare_context() is called the first time.
23826 2007-04-24 15:00:07 +0000 Dan Williams <dcbw@redhat.com>
23828 gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
23829 Original commit message from CVS:
23830 Patch by: Dan Williams <dcbw redhat com>
23831 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
23832 Don't leak incoming buffer if gst_pad_push() returns a
23833 non-OK flow. Fixes #432755.
23834 * tests/check/elements/videorate.c: (GST_START_TEST),
23836 Unit test for the above by Yours Truly.
23838 2007-04-23 20:04:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23840 gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
23841 Original commit message from CVS:
23842 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
23843 (gst_adder_sink_event), (gst_adder_collected):
23844 Fix non-flushing segmented seeks, Fixes #340060 for me
23846 2007-04-21 15:29:27 +0000 Tim-Philipp Müller <tim@centricular.net>
23849 ChangeLog surgery: add API keyword
23850 Original commit message from CVS:
23851 ChangeLog surgery: add API keyword
23853 2007-04-21 15:25:22 +0000 Olivier Crete <tester@tester.ca>
23855 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
23856 Original commit message from CVS:
23857 Patch by: Olivier Crete <tester at tester ca>
23858 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23859 (gst_base_rtp_audio_payload_class_init),
23860 (gst_base_rtp_audio_payload_init),
23861 (gst_base_rtp_audio_payload_dispose):
23862 Chain up to parent class in dispose function; get rid of
23863 unnecessary 'diposed' flag in private structure (#415001).
23865 2007-04-21 15:10:25 +0000 Tim-Philipp Müller <tim@centricular.net>
23867 Some minor docs fixes and additions; also add missing 'Since' bits.
23868 Original commit message from CVS:
23869 * docs/libs/gst-plugins-base-libs.types:
23870 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23871 (gst_base_rtp_audio_payload_class_init):
23872 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23873 * gst-libs/gst/rtp/gstbasertppayload.c:
23874 Some minor docs fixes and additions; also add missing 'Since' bits.
23876 2007-04-21 14:40:45 +0000 Zeeshan Ali <zeenix@gmail.com>
23878 gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
23879 Original commit message from CVS:
23880 Patch by: Zeeshan Ali <zeenix gmail com>
23881 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23882 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
23883 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
23884 (gst_base_rtp_audio_payload_push):
23885 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
23886 The recently-added gst_base_rtp_audio_payload_push() should take an
23887 object of type GstBaseRTPAudioPayload as first argument (#431672).
23889 2007-04-21 14:14:24 +0000 Tim-Philipp Müller <tim@centricular.net>
23891 gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
23892 Original commit message from CVS:
23893 * gst/audioresample/gstaudioresample.c:
23894 Make more functions static, just because we can.
23896 2007-04-21 13:54:39 +0000 Tim-Philipp Müller <tim@centricular.net>
23898 tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
23899 Original commit message from CVS:
23900 * tests/check/elements/audioresample.c:
23901 Add unit test for audioresample shutdown crasher (#420106).
23903 2007-04-20 10:42:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23905 gst/subparse/: Use GST_DISABLE_XML here
23906 Original commit message from CVS:
23907 * gst/subparse/gstsubparse.c:
23908 * gst/subparse/samiparse.c:
23909 Use GST_DISABLE_XML here
23910 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
23911 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
23912 (gst_xvimagesink_buffer_alloc),
23913 (gst_xvimagesink_navigation_send_event):
23914 * sys/xvimage/xvimagesink.h:
23915 Include stdlib.h when using atoi.
23916 * tests/check/elements/playbin.c: (playbin_suite):
23917 Use GST_DISABLE_REGISTRY here
23919 2007-04-19 16:58:53 +0000 Michael Smith <msmith@xiph.org>
23921 ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
23922 Original commit message from CVS:
23923 * ext/theora/gsttheoraenc.h:
23924 * ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
23925 (theora_enc_sink_event), (theora_enc_change_state):
23926 Track initialisation state; don't try to use encoder state if we're
23927 not initialised (it'll segfault).
23929 2007-04-18 11:06:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23931 tests/check/pipelines/.cvsignore: Fix build.
23932 Original commit message from CVS:
23933 * tests/check/pipelines/.cvsignore:
23936 2007-04-17 10:56:37 +0000 Tim-Philipp Müller <tim@centricular.net>
23938 gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
23939 Original commit message from CVS:
23940 * gst/app/Makefile.am:
23941 Fix CFLAGS and hopefully #430594.
23943 2007-04-17 02:53:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
23945 gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
23946 Original commit message from CVS:
23947 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
23948 Allow random depths between 1 and 32 instead of only multiplies of 8.
23950 2007-04-17 02:04:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
23952 gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
23953 Original commit message from CVS:
23954 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
23955 Set the maximum number of channels for PCM and float in the correct
23956 place to have it also used when creating the template caps.
23958 2007-04-17 01:56:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
23960 gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
23961 Original commit message from CVS:
23962 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
23963 Correctly support 4, 6 and 8 channels with normal PCM and float
23965 Fix the depth and signedness calculation in extensible wav files and
23966 also handle 1, 2, 4, 6, 8 channels here when a file without channel
23968 Add support for float, alaw and mulaw in extensible wav files.
23969 This allows correct playback of all but 5 files from
23970 http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
23971 (gst_riff_create_audio_template_caps):
23972 Add voxware and float formats to the template caps.
23974 2007-04-16 22:20:03 +0000 Vincent Torri <vtorri@univ-evry.fr>
23976 ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
23977 Original commit message from CVS:
23978 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
23979 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
23980 Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
23981 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
23982 * gst/audioresample/gstaudioresample.c: (audioresample_do_output):
23983 Use the correct format strings for integer formats.
23985 2007-04-16 21:44:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23987 * gst-plugins-base.doap:
23989 Original commit message from CVS:
23992 2007-04-16 21:42:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23994 * gst-plugins-base.doap:
23996 Original commit message from CVS:
23999 2007-04-15 14:35:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24001 ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
24002 Original commit message from CVS:
24003 * ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
24004 Don't use pad_alloc_buffer_and_set_caps to create a small header
24005 packet, or, worse, to create a big temporary video buffer using the
24008 2007-04-14 12:34:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24010 gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
24011 Original commit message from CVS:
24012 * gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
24013 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
24014 GST_START_TEST, buffer_probe_cb, GST_START_TEST):
24015 Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
24017 2007-04-13 22:10:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24019 * gst/tcp/gstmultifdsink.c:
24021 Original commit message from CVS:
24024 2007-04-13 21:55:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24027 * tests/check/pipelines/streamheader.c:
24028 tests/check/pipelines/streamheader.c (tag_event_probe_cb,
24029 Original commit message from CVS:
24030 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
24031 GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
24032 streamheader_suite):
24033 Add another test set up for failure
24035 2007-04-13 21:09:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24037 * ext/ogg/gstoggmux.c:
24038 * gst/gdp/gstgdpdepay.c:
24040 Original commit message from CVS:
24043 2007-04-13 21:08:11 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24045 tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
24046 Original commit message from CVS:
24047 * tests/check/Makefile.am:
24048 * tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
24049 GST_START_TEST, streamheader_suite, main):
24050 Add a test for the streamheader bug Wim fixed.
24052 2007-04-13 11:42:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24054 ext/theora/theoradec.c: Fix misleading comment.
24055 Original commit message from CVS:
24056 * ext/theora/theoradec.c: (theora_dec_sink_event):
24057 Fix misleading comment.
24059 2007-04-13 06:17:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24061 gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
24062 Original commit message from CVS:
24063 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24064 More sanity checks for the header fields.
24066 2007-04-12 16:36:36 +0000 Tim-Philipp Müller <tim@centricular.net>
24068 gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
24069 Original commit message from CVS:
24070 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
24071 Try encodings from all environment variables, not just those in the
24072 first environment variable that is set.
24074 2007-04-12 15:00:03 +0000 Wim Taymans <wim.taymans@gmail.com>
24076 gst/videorate/gstvideorate.c: Add some debug.
24077 Original commit message from CVS:
24078 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
24079 (gst_video_rate_chain):
24081 * tests/check/elements/videorate.c: (GST_START_TEST),
24083 Added check for videorate changing caps handling. Closes #421834.
24085 2007-04-12 12:57:33 +0000 Michael Smith <msmith@xiph.org>
24087 ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
24088 Original commit message from CVS:
24089 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
24090 Use scale functions to avoid overflow when calculating duration of
24093 2007-04-12 12:19:20 +0000 Tim-Philipp Müller <tim@centricular.net>
24095 API: add gst_tag_freeform_string_to_utf8() (#405072).
24096 Original commit message from CVS:
24097 * docs/libs/gst-plugins-base-libs-sections.txt:
24098 * gst-libs/gst/tag/tag.h:
24099 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
24100 API: add gst_tag_freeform_string_to_utf8() (#405072).
24101 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
24102 Use gst_tag_freeform_string_to_utf8() here.
24104 2007-04-12 10:38:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24106 * gst/tcp/gstmultifdsink.c:
24108 Original commit message from CVS:
24111 2007-04-12 10:03:22 +0000 Wim Taymans <wim.taymans@gmail.com>
24113 gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
24114 Original commit message from CVS:
24115 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
24116 (gst_gdp_pay_sink_event):
24117 Make sure we set the IN_CAPS flag correctly.
24118 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
24119 Get the IN_CAPS flag before we call functions that mess with the flags.
24121 2007-04-10 20:37:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24124 * gst/gdp/gstgdppay.c:
24125 gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
24126 Original commit message from CVS:
24127 * gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
24128 gst_gdp_pay_chain, gst_gdp_pay_sink_event):
24129 Only stamp buffers with offset/offset_end right before they get
24130 pushed. This ensures offset continuity, which was not the case
24132 gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
24134 2007-04-10 20:25:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24136 * gst/gdp/gstgdpdepay.c:
24137 * gst/gdp/gstgdppay.c:
24139 Original commit message from CVS:
24142 2007-04-10 11:23:18 +0000 Christian Schaller <uraeus@gnome.org>
24145 * gst-plugins-base.spec.in:
24146 update spec file for RTP changes
24147 Original commit message from CVS:
24148 update spec file for RTP changes
24150 2007-04-06 12:58:06 +0000 Wim Taymans <wim.taymans@gmail.com>
24152 gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
24153 Original commit message from CVS:
24154 * gst/playback/gstplaybin.c: (add_sink),
24155 (gst_play_bin_change_state):
24156 Activate sync in playbin, we are ready to handle it for live streams.
24158 2007-04-06 09:56:18 +0000 Tim-Philipp Müller <tim@centricular.net>
24160 tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
24161 Original commit message from CVS:
24162 * tests/check/elements/playbin.c:
24163 (test_sink_usage_video_only_stream), (playbin_suite):
24164 Add small test for stream-info-value-array code paths.
24166 2007-04-05 15:44:40 +0000 Wim Taymans <wim.taymans@gmail.com>
24168 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
24169 Original commit message from CVS:
24170 * gst-libs/gst/audio/gstbaseaudiosink.c:
24171 (gst_base_audio_sink_skew_slaving):
24172 Don't try to create invalid calibration parameters by making the
24173 internal time go backwards, instead make external time go forward.
24175 2007-04-05 10:27:06 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
24177 gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
24178 Original commit message from CVS:
24179 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
24180 * gst/playback/gstplaybasebin.c: (add_stream):
24181 Fix leak in add_stream(), when g_value_set_object() increases the
24182 refcount of streaminfo object. Fixes #426250.
24184 2007-04-04 02:45:03 +0000 David Schleef <ds@schleef.org>
24186 gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
24187 Original commit message from CVS:
24188 * gst/videotestsrc/gstvideotestsrc.c:
24189 * gst/videotestsrc/gstvideotestsrc.h:
24190 * gst/videotestsrc/videotestsrc.c:
24191 * gst/videotestsrc/videotestsrc.h:
24192 Add a test pattern called "circular", which has concentric
24193 rings with varying radial frequency. The main purpose of this
24194 pattern is to test fidelity loss in a filter or scaler element.
24195 Notably, this pattern is scale invariant, and is optimally viewed
24196 with a width (and height) of 400.
24198 2007-04-03 11:10:52 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
24200 gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
24201 Original commit message from CVS:
24202 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
24203 * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
24204 (deactivate_free_recursive):
24205 Decodebin2 doesn't unref pads it obtains in some occasions:
24206 - multiqueue src pads, when either connecting further or exposing
24207 - sink pads of new autoplugged elements
24208 - peer pads when recursively freeing elements
24211 2007-03-30 17:05:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
24213 gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
24214 Original commit message from CVS:
24215 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24216 Add audio/x-raw-float support, now that audioconvert support
24217 non-native endianness floats.
24219 2007-03-30 15:00:49 +0000 Tim-Philipp Müller <tim@centricular.net>
24221 docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
24222 Original commit message from CVS:
24223 * docs/libs/gst-plugins-base-libs-docs.sgml:
24224 gstreamer-plugins-base.pc doesn't exist, it's
24225 gstreamer-plugins-base-0.10.pc.
24227 2007-03-29 18:42:34 +0000 René Stadler <mail@renestadler.de>
24229 with some minor changes
24230 Original commit message from CVS:
24231 Patch by: René Stadler <mail at renestadler dot de>
24232 with some minor changes
24233 * gst-libs/gst/floatcast/floatcast.h:
24234 Use more efficient float endianness conversion functions that don't
24235 involve 2 function calls per value.
24236 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
24237 (check_default), (audio_convert_prepare_context):
24238 * gst/audioconvert/gstaudioconvert.c:
24239 (gst_audio_convert_parse_caps), (make_lossless_changes):
24240 Support non-native endianness floats as input and output.
24242 * tests/check/elements/audioconvert.c: (verify_convert),
24244 Add unit tests for the non-native endianness float conversions.
24246 2007-03-29 16:23:53 +0000 Wim Taymans <wim.taymans@gmail.com>
24248 gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
24249 Original commit message from CVS:
24250 * gst-libs/gst/rtp/gstbasertpdepayload.c:
24251 (gst_base_rtp_depayload_base_init),
24252 (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
24253 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
24254 (gst_base_rtp_depayload_set_gst_timestamp),
24255 (gst_base_rtp_depayload_change_state),
24256 (gst_base_rtp_depayload_set_property),
24257 (gst_base_rtp_depayload_get_property):
24258 * gst-libs/gst/rtp/gstbasertpdepayload.h:
24259 Add Private structure.
24260 Bring element code to 2007.
24261 Parse clock-base caps param and use it when generating the
24263 Reset variables before going to PAUSED.
24266 2007-03-29 16:20:31 +0000 Wim Taymans <wim.taymans@gmail.com>
24269 Original commit message from CVS:
24270 * docs/libs/gst-plugins-base-libs-docs.sgml:
24271 * docs/libs/gst-plugins-base-libs-sections.txt:
24272 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
24273 (gst_base_rtp_audio_payload_get_adapter):
24275 Fix some more docs.
24276 * gst-libs/gst/rtp/Makefile.am:
24277 * gst-libs/gst/rtp/gstrtcpbuffer.c:
24278 (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
24279 (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
24280 (gst_rtcp_buffer_get_packet_count), (read_packet_header),
24281 (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
24282 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
24283 (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
24284 (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
24285 (gst_rtcp_packet_sr_get_sender_info),
24286 (gst_rtcp_packet_sr_set_sender_info),
24287 (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
24288 (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
24289 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
24290 (gst_rtcp_packet_sdes_get_chunk_count),
24291 (gst_rtcp_packet_sdes_first_chunk),
24292 (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
24293 (gst_rtcp_packet_sdes_first_item),
24294 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
24295 (gst_rtcp_packet_bye_get_ssrc_count),
24296 (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
24297 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
24298 (gst_rtcp_packet_bye_get_reason_len),
24299 (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
24300 * gst-libs/gst/rtp/gstrtcpbuffer.h:
24301 Add new helper object for parsing and creating RTCP messages.
24303 2007-03-29 12:07:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
24305 gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
24306 Original commit message from CVS:
24307 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24308 PCM samples with width=8 must be always unsigned, no matter what
24311 2007-03-29 11:24:47 +0000 Andy Wingo <wingo@pobox.com>
24313 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
24314 Original commit message from CVS:
24315 2007-03-29 Andy Wingo <wingo@pobox.com>
24316 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
24317 perfect offsets also, not just timestamps.
24318 * tests/check/elements/videorate.c (test_more): Test that given
24319 any incoming offsets, that videorate produces perfect offsets.
24321 2007-03-29 10:19:45 +0000 Wim Taymans <wim.taymans@gmail.com>
24323 gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
24324 Original commit message from CVS:
24325 * gst-libs/gst/riff/riff-ids.h:
24326 Add some more RIFF formats.
24328 2007-03-29 10:17:52 +0000 Wim Taymans <wim.taymans@gmail.com>
24330 gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
24331 Original commit message from CVS:
24332 * gst-libs/gst/rtp/gstrtpbuffer.c:
24333 (gst_rtp_buffer_default_clock_rate):
24334 * gst-libs/gst/rtp/gstrtpbuffer.h:
24335 Fix fixed payload names and docs.
24336 Added method to get the default clock rates of fixed payload types.
24337 API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
24339 2007-03-28 15:24:40 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
24341 tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
24342 Original commit message from CVS:
24343 * tests/check/pipelines/.cvsignore:
24344 Add new vorbisdec test to cvsignore.
24346 2007-03-28 14:50:47 +0000 Wim Taymans <wim.taymans@gmail.com>
24348 gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
24349 Original commit message from CVS:
24350 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
24351 (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
24352 (gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
24353 (gst_base_audio_sink_set_property),
24354 (gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
24355 (clock_convert_external), (gst_base_audio_sink_resample_slaving),
24356 (gst_base_audio_sink_skew_slaving),
24357 (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
24358 (gst_base_audio_sink_async_play):
24359 * gst-libs/gst/audio/gstbaseaudiosink.h:
24360 Store private stuff in GstBaseAudioSinkPrivate.
24361 Add configurable clock slaving modes property.
24362 API:: GstBaseAudioSink::slave-method property
24363 Some more latency reporting tweaks.
24364 Added skew based clock slaving correction and make it the default until
24365 the resampling method is more robust.
24367 2007-03-27 12:44:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
24369 gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
24370 Original commit message from CVS:
24371 * gst/audioconvert/audioconvert.c:
24372 Add docs to the integer pack functions and implement proper
24373 rounding. Before we had rounding towards negative infinity, i.e.
24374 always the smaller number was taken. Now we use natural rounding,
24375 i.e. rounding to the nearest integer and to the one with the largest
24376 absolute value for X.5. The old rounding introduced some minor
24377 distortions. Fixes #420079
24378 * tests/check/elements/audioconvert.c: (GST_START_TEST):
24379 Fix one unit test that assumed the old rounding and added unit tests
24380 for checking signed/unsigned int16 <-> signed/unsigned int16 with
24381 depth 8, one for signed int16 <-> unsigned int16 and one for the new
24382 rounding from signed int32 to signed/unsigned int16.
24384 2007-03-27 11:31:17 +0000 Michael Smith <msmith@xiph.org>
24386 gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
24387 Original commit message from CVS:
24388 * gst/audioconvert/gstaudioconvert.c: (strip_width_64),
24389 (gst_audio_convert_transform_caps):
24390 Fix typo in debug line introduced recently, as pointed out on irc.
24392 2007-03-27 10:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
24394 Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
24395 Original commit message from CVS:
24396 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
24397 * tests/check/libs/tag.c: (GST_START_TEST):
24398 Make sure we parse floating-point numbers in vorbis comments
24399 correctly with either '.' or ',' as separator, no matter what
24400 the current locale is. Add unit test for this too.
24402 2007-03-27 09:37:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24404 * tests/check/pipelines/vorbisdec.c:
24406 Original commit message from CVS:
24409 2007-03-26 22:38:19 +0000 René Stadler <mail@renestadler.de>
24411 gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
24412 Original commit message from CVS:
24413 Patch by: René Stadler <mail at renestadler de>
24414 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
24415 When writing out floating-point numbers to vorbis comment tags, always
24416 use the same character as separator no matter what the current locale is
24418 * tests/check/libs/tag.c: (GST_START_TEST):
24419 Add unit tests for replaygain tags in vorbis comments (closes #423055).
24421 2007-03-26 20:56:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24423 ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
24424 Original commit message from CVS:
24425 * ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
24426 vorbis_handle_data_packet):
24427 Correctly set DURATION to generate a timestamp-continuous stream.
24428 One bug left at the end; see
24429 ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
24430 * tests/check/Makefile.am:
24431 * tests/check/pipelines/vorbisenc.c (GST_START_TEST):
24432 Add a test to check this. Without the above patch this test fails.
24434 2007-03-26 11:44:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24436 gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
24437 Original commit message from CVS:
24438 * gst-libs/gst/rtp/Makefile.am:
24439 The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
24441 2007-03-23 15:43:24 +0000 Christian Schaller <uraeus@gnome.org>
24443 * gst-plugins-base.spec.in:
24445 Original commit message from CVS:
24448 2007-03-23 12:32:33 +0000 Michael Smith <msmith@xiph.org>
24450 gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
24451 Original commit message from CVS:
24452 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
24453 (gst_video_rate_reset), (gst_video_rate_chain):
24454 If videorate changes caps, we can no longer use the old buffer
24455 (which may have a different size, incompatible with our caps).
24456 So don't do that; just duplicate the new frame more times.
24458 2007-03-22 17:43:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24460 gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
24461 Original commit message from CVS:
24462 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
24463 Remove playbin's override of the set_clock vmethod. It's irrelevant
24464 after Wim's commit on the 19th.
24466 2007-03-22 14:37:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24468 gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
24469 Original commit message from CVS:
24470 * gst-libs/gst/app/Makefile.am:
24471 Use GST_ALL_LDFLAGS, which actually exists, but maybe David
24472 can confirm that was what he wanted.
24474 2007-03-22 09:26:02 +0000 Wim Taymans <wim.taymans@gmail.com>
24476 ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
24477 Original commit message from CVS:
24478 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
24479 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
24480 * ext/gnomevfs/gstgnomevfssrc.h:
24481 Don't cache file sizes. Fixes #341078.
24483 2007-03-21 11:03:23 +0000 Tim-Philipp Müller <tim@centricular.net>
24485 gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
24486 Original commit message from CVS:
24487 * gst/playback/gstplaybin.c: (add_sink):
24488 Use GST_PTR_FORMAT to log caps.
24490 2007-03-21 10:23:11 +0000 Young-Ho Cha <ganadist@chollian.net>
24492 gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
24493 Original commit message from CVS:
24494 Patch by: Young-Ho Cha <ganadist at chollian net>
24495 * gst/subparse/samiparse.c: (handle_start_font):
24496 Special-case some more colour names that pango doesn't handle by
24497 default. Fixes #420578.
24499 2007-03-20 11:49:55 +0000 Michael Smith <msmith@xiph.org>
24501 ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
24502 Original commit message from CVS:
24503 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
24504 If we get a zero-sized input buffer, don't pass it to libvorbis, as
24505 that marks EOS internally. After that, libvorbis will buffer all
24506 input data, and encode none of it, eventually leading to memory
24509 2007-03-19 10:52:50 +0000 Wim Taymans <wim.taymans@gmail.com>
24511 gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
24512 Original commit message from CVS:
24513 * gst/playback/gstdecodebin.c: (remove_fakesink):
24514 Don't post STATE_DIRTY anymore.
24515 * gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
24516 (gst_play_bin_change_state):
24517 Remove stream_time reset in seek handling, core does that now.
24518 Disable clocking for live pipelines by forcing a NULL clock to the
24519 complete pipeline, core is too smart now for our previous hack.
24520 We can always autoplug in PAUSED now.
24522 2007-03-18 03:14:01 +0000 David Schleef <ds@schleef.org>
24524 REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
24525 Original commit message from CVS:
24526 * REQUIREMENTS: Update this file, change the formatting to make
24527 it more consistent, plus more machine readable.
24529 2007-03-16 17:29:09 +0000 Michael Smith <msmith@xiph.org>
24531 gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
24532 Original commit message from CVS:
24533 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
24534 (strip_width_64), (append_with_other_format):
24535 Previous fix was too simplistic, and broke the tests. Use a better
24536 approach; only strip 64 from widths for integer audio.
24538 2007-03-16 16:42:23 +0000 Michael Smith <msmith@xiph.org>
24540 gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
24541 Original commit message from CVS:
24542 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
24543 (gst_audio_convert_transform_caps):
24544 We don't support 64 bit integer audio, so don't try to claim we can.
24545 Stops us producing caps don't match our template caps.
24548 2007-03-15 10:52:21 +0000 Michael Smith <msmith@xiph.org>
24550 gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
24551 Original commit message from CVS:
24552 * gst/audioresample/gstaudioresample.c:
24553 (audioresample_check_discont), (audioresample_transform):
24554 Don't trigger discontinuities for very small imperfections; a filter
24555 flush will sound bad, and many plugins have rounding errors leading
24558 2007-03-14 21:11:18 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
24560 gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
24561 Original commit message from CVS:
24562 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
24563 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
24564 Add min-ptime property to RTP base audio payloader. Patch by
24565 olivier.crete@collabora.co.uk.
24567 Indentation/whitespace/documentation fixes.
24569 2007-03-14 17:16:30 +0000 Julien Moutte <julien@moutte.net>
24571 gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
24572 Original commit message from CVS:
24573 2007-03-14 Julien MOUTTE <julien@moutte.net>
24574 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
24575 (audioresample_transform_size), (audioresample_do_output),
24576 (audioresample_transform), (audioresample_pushthrough): Handle
24577 discontinuous streams.
24578 * gst/audioresample/gstaudioresample.h:
24579 * tests/check/elements/audioresample.c:
24580 (test_discont_stream_instance), (GST_START_TEST),
24581 (audioresample_suite): Add a test for discontinuous streams.
24582 * win32/common/config.h: Updated.
24584 2007-03-14 15:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24586 po/: Update translations from translation project.
24587 Original commit message from CVS:
24601 Update translations from translation project.
24603 2007-03-14 15:05:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24605 * gst/gdp/gstgdpdepay.c:
24607 Original commit message from CVS:
24610 2007-03-14 14:48:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24612 gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
24613 Original commit message from CVS:
24614 * gst/audioresample/debug.h:
24615 * gst/audioresample/resample.c: (resample_init):
24616 Since I really am not interested in a debug line for each sample
24617 being processed, move the library's debugging to its own category,
24620 2007-03-14 14:09:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24622 * gst/audioresample/gstaudioresample.c:
24623 add debugging and reformat docs
24624 Original commit message from CVS:
24625 add debugging and reformat docs
24627 2007-03-12 23:29:07 +0000 Michael Smith <msmith@xiph.org>
24629 ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
24630 Original commit message from CVS:
24631 * ext/theora/theoradec.c: (theora_handle_type_packet):
24632 Since the plugin doesn't support anything other than 4:2:0 right
24633 now, post an error and fail if we get something else. Won't matter
24634 until libtheora supports the other pixel formats, but hopefully
24637 2007-03-12 15:50:35 +0000 Alex Lancaster <alexlan@fedoraproject.org>
24640 I'm too lazy to comment this
24641 Original commit message from CVS:
24642 Mention Patch by: Alex Lancaster in a recent commit.
24644 2007-03-12 11:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24646 examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.
24647 Original commit message from CVS:
24648 * examples/app/.cvsignore:
24649 The buildbot demands .cvsignore files, and I comply.
24651 2007-03-11 00:48:26 +0000 David Schleef <ds@schleef.org>
24653 Add appsrc/appsink example.
24654 Original commit message from CVS:
24656 * examples/Makefile.am:
24657 * examples/app/Makefile.am:
24658 * examples/app/appsrc_ex.c:
24659 Add appsrc/appsink example.
24660 * gst-libs/gst/app/Makefile.am:
24661 * gst-libs/gst/app/gstapp.c:
24662 * gst-libs/gst/app/gstappsink.c:
24663 * gst-libs/gst/app/gstappsink.h:
24664 * gst/app/gstapp.c:
24667 2007-03-10 15:59:33 +0000 Sébastien Moutte <sebastien@moutte.net>
24669 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
24670 Original commit message from CVS:
24671 * gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
24672 Use gst_guint64_to_gdouble for conversion.
24674 Add new files to the win32 MANIFEST.
24675 * win32/common/libgstaudio.def:
24676 * win32/common/libgstpbutils.def:
24677 Add new exported functions.
24678 * win32/vs6/gst_plugins_base.dsw:
24679 * win32/vs6/libgstdecodebin.dsp:
24680 * win32/vs6/libgstplaybin.dsp:
24681 Change the link to libgstpbutils.lib.
24682 * win32/vs6/libgstdecodebin2.dsp:
24683 Add a new project for decodebin2.
24684 * win32/vs6/libgstpbutils.dsp:
24685 Add a new project for pbutils.
24687 2007-03-10 12:18:58 +0000 Tim-Philipp Müller <tim@centricular.net>
24689 gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
24690 Original commit message from CVS:
24691 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
24692 Also accept partial dates with only year and month,
24693 like 1999-12-00 (fixes #410396 even more).
24694 * tests/check/libs/tag.c: (GST_START_TEST):
24695 Add unit test for the above.
24697 2007-03-10 11:21:08 +0000 Tim-Philipp Müller <tim@centricular.net>
24699 tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
24700 Original commit message from CVS:
24701 * tests/check/elements/subparse.c: (GST_START_TEST),
24703 Add unit test for MPL2 subtitle format (#413799).
24705 2007-03-10 11:17:52 +0000 Kamil Pawlowski <kamilpe@gmail.com>
24707 gst/subparse/: Add support for MPL2 subtitle format (#413799).
24708 Original commit message from CVS:
24709 Patch by: Kamil Pawlowski <kamilpe gmail com>
24710 * gst/subparse/Makefile.am:
24711 * gst/subparse/gstsubparse.c:
24712 (gst_sub_parse_data_format_autodetect),
24713 (gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
24714 (gst_subparse_type_find):
24715 * gst/subparse/gstsubparse.h:
24716 * gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
24717 * gst/subparse/mpl2parse.h:
24718 Add support for MPL2 subtitle format (#413799).
24720 2007-03-09 17:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
24722 configure.ac: We require core CVS for the new buffer metadata copy functions.
24723 Original commit message from CVS:
24725 We require core CVS for the new buffer metadata copy functions.
24727 2007-03-09 16:51:13 +0000 Wim Taymans <wim.taymans@gmail.com>
24729 gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
24730 Original commit message from CVS:
24731 * gst-libs/gst/tag/gstid3tag.c:
24732 Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
24735 2007-03-09 16:46:35 +0000 Wim Taymans <wim.taymans@gmail.com>
24737 ext/libvisual/visual.c: Improve adapter usage and comments.
24738 Original commit message from CVS:
24739 * ext/libvisual/visual.c: (gst_visual_sink_setcaps),
24740 (gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
24741 Improve adapter usage and comments.
24743 2007-03-09 16:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
24745 Use new metadata copy function.
24746 Original commit message from CVS:
24747 * ext/pango/gsttextrender.c: (gst_text_render_chain):
24748 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
24749 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
24750 Use new metadata copy function.
24751 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
24752 (gst_ffmpegcsp_transform):
24753 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
24754 Basetransform copied the metadata for us.
24756 2007-03-09 16:28:04 +0000 Tim-Philipp Müller <tim@centricular.net>
24758 ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
24759 Original commit message from CVS:
24760 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
24761 (gst_text_overlay_video_event):
24762 Some more logging. Only accept newsegment events in TIME format and
24763 send a WARNING message if they are not in TIME format.
24764 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
24765 (gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
24766 (gst_sub_parse_chain), (gst_sub_parse_sink_event):
24767 * gst/subparse/gstsubparse.h:
24768 No need to allocate GstSegment structure dynamically, just put it
24769 into the instance structure; ignore newsegment events in BYTE
24770 format and in particular don't let it overwrite our saved TIME
24771 segment from the last seek.
24773 2007-03-09 13:05:04 +0000 Michael Smith <msmith@xiph.org>
24775 gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
24776 Original commit message from CVS:
24777 * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
24778 Replace AC3 typefinder with one that isn't terrible, and actually
24781 2007-03-09 12:22:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24783 gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
24784 Original commit message from CVS:
24785 * gst/audioconvert/gstaudioconvert.c:
24786 (gst_audio_convert_transform):
24787 fix error category and translatable string
24789 2007-03-09 11:23:32 +0000 Tim-Philipp Müller <tim@centricular.net>
24791 pkgconfig/: Fix up utils => pbutils here too.
24792 Original commit message from CVS:
24793 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
24794 * pkgconfig/gstreamer-plugins-base.pc.in:
24795 Fix up utils => pbutils here too.
24797 2007-03-09 10:49:53 +0000 Tim-Philipp Müller <tim@centricular.net>
24799 gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
24800 Original commit message from CVS:
24801 * gst/subparse/gstsubparse.c: (handle_buffer):
24802 Break out of loop in chain function as soon as possible if we get
24803 a non-OK flow return.
24805 2007-03-08 18:26:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24807 tests/check/elements/alsa.c: Unref the mixer if the state change fails too (if the alsa devices are inaccessible, for...
24808 Original commit message from CVS:
24809 * tests/check/elements/alsa.c: (GST_START_TEST):
24810 Unref the mixer if the state change fails too (if the
24811 alsa devices are inaccessible, for example)
24813 2007-03-08 17:49:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24815 tests/check/Makefile.am: Don't test libvisual elements in the states check, because libvisual seems to leak internally.
24816 Original commit message from CVS:
24817 * tests/check/Makefile.am:
24818 Don't test libvisual elements in the states check, because libvisual
24819 seems to leak internally.
24820 Re-enable the alsa and states tests now that there's new suppressions
24822 * tests/check/elements/alsa.c: (GST_START_TEST):
24823 Don't leak the alsamixer we instantiated.
24825 2007-03-08 15:22:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24827 sys/: Move some cleanup stuff from the state change handler into a _reset() function that can be called from _finaliz...
24828 Original commit message from CVS:
24829 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
24830 (gst_ximagesink_change_state), (gst_ximagesink_reset),
24831 (gst_ximagesink_finalize):
24832 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
24833 (gst_xvimagesink_reset), (gst_xvimagesink_finalize):
24834 Move some cleanup stuff from the state change handler into a _reset()
24835 function that can be called from _finalize(). This ensures that things
24836 get freed even if (for some reason) the NULL->READY state transition
24837 fails in the parent class.
24838 Even if a parent state change fails, process our downward state change
24839 logic instead of bailing out early.
24840 Free the correct xcontext pointer in ximagesink's xcontext_clear.
24842 2007-03-08 12:53:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24844 ext/alsa/gstalsasink.c: Extra log line.
24845 Original commit message from CVS:
24846 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
24848 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
24849 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
24850 Use pango_font_description_set_family_static instead of
24851 pango_font_description_set_family to save a string copy (it was
24852 leaking due to the strdup anyway)
24853 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
24854 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
24855 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
24856 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
24857 Chain up in finalize.
24859 2007-03-07 18:50:10 +0000 Tim-Philipp Müller <tim@centricular.net>
24861 gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
24862 Original commit message from CVS:
24863 * gst-libs/gst/interfaces/mixertrack.c:
24864 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
24865 (gst_mixer_track_set_property):
24866 API: add "untranslated-label" property which should be set by
24867 implementations at construct time (#414645).
24868 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
24869 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
24870 Set "untranslated-label" when constructing mixer track objects.
24871 * tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
24872 Unit test to check the above.
24874 2007-03-07 17:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
24876 ext/ogg/gstoggdemux.c: Fix confusing debug message.
24877 Original commit message from CVS:
24878 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
24879 Fix confusing debug message.
24881 2007-03-07 17:12:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24883 gst-plugins-base.doap: update doap file with new version
24884 Original commit message from CVS:
24885 * gst-plugins-base.doap:
24886 update doap file with new version
24888 2007-03-07 17:05:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24890 * gst/tcp/gstmultifdsink.c:
24892 Original commit message from CVS:
24895 2007-03-07 16:56:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24897 configure.ac: Back to CVS
24898 Original commit message from CVS:
24902 === release 0.10.12 ===
24904 2007-03-07 16:46:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24910 * docs/plugins/gst-plugins-base-plugins.args:
24911 * docs/plugins/inspect/plugin-adder.xml:
24912 * docs/plugins/inspect/plugin-alsa.xml:
24913 * docs/plugins/inspect/plugin-audioconvert.xml:
24914 * docs/plugins/inspect/plugin-audiorate.xml:
24915 * docs/plugins/inspect/plugin-audioresample.xml:
24916 * docs/plugins/inspect/plugin-audiotestsrc.xml:
24917 * docs/plugins/inspect/plugin-cdparanoia.xml:
24918 * docs/plugins/inspect/plugin-decodebin.xml:
24919 * docs/plugins/inspect/plugin-decodebin2.xml:
24920 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
24921 * docs/plugins/inspect/plugin-gdp.xml:
24922 * docs/plugins/inspect/plugin-gnomevfs.xml:
24923 * docs/plugins/inspect/plugin-libvisual.xml:
24924 * docs/plugins/inspect/plugin-ogg.xml:
24925 * docs/plugins/inspect/plugin-pango.xml:
24926 * docs/plugins/inspect/plugin-playbin.xml:
24927 * docs/plugins/inspect/plugin-subparse.xml:
24928 * docs/plugins/inspect/plugin-tcp.xml:
24929 * docs/plugins/inspect/plugin-theora.xml:
24930 * docs/plugins/inspect/plugin-typefindfunctions.xml:
24931 * docs/plugins/inspect/plugin-video4linux.xml:
24932 * docs/plugins/inspect/plugin-videorate.xml:
24933 * docs/plugins/inspect/plugin-videoscale.xml:
24934 * docs/plugins/inspect/plugin-videotestsrc.xml:
24935 * docs/plugins/inspect/plugin-volume.xml:
24936 * docs/plugins/inspect/plugin-vorbis.xml:
24937 * docs/plugins/inspect/plugin-ximagesink.xml:
24938 * docs/plugins/inspect/plugin-xvimagesink.xml:
24939 * win32/common/config.h:
24941 Original commit message from CVS:
24944 2007-03-07 15:35:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24963 Original commit message from CVS:
24966 2007-03-06 12:31:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24968 configure.ac: Bump version to 0.10.11.4 pre-release
24969 Original commit message from CVS:
24971 Bump version to 0.10.11.4 pre-release
24973 2007-03-06 12:10:08 +0000 Wim Taymans <wim.taymans@gmail.com>
24975 gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
24976 Original commit message from CVS:
24977 * gst-libs/gst/audio/gstbaseaudiosink.c:
24978 (gst_base_audio_sink_async_play):
24979 Fix regression that made GStreamer skip the first samples of audio.
24982 2007-03-05 11:21:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24984 configure.ac: Bump version to 0.10.11.3 pre-release
24985 Original commit message from CVS:
24987 Bump version to 0.10.11.3 pre-release
24989 2007-03-05 09:35:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
24991 po/POTFILES.in: Update paths for the rename from utils to pbutils to fix the build.
24992 Original commit message from CVS:
24994 Update paths for the rename from utils to pbutils to fix the build.
24996 2007-03-05 09:27:55 +0000 Tim-Philipp Müller <tim@centricular.net>
24998 gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
24999 Original commit message from CVS:
25000 * gst-libs/gst/pbutils/Makefile.am:
25001 Change directory to install headers in from gst/utils to gst/pbutils
25004 2007-03-04 23:41:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25006 * tests/check/libs/.gitignore:
25008 Original commit message from CVS:
25011 2007-03-04 23:41:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25013 * win32/common/config.h:
25014 * win32/common/libgstutils.def:
25016 Original commit message from CVS:
25019 2007-03-04 23:39:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25021 rename utils to pbutils
25022 Original commit message from CVS:
25024 * docs/libs/gst-plugins-base-libs-docs.sgml:
25025 * docs/libs/gst-plugins-base-libs-sections.txt:
25026 * gst-libs/gst/Makefile.am:
25027 * gst-libs/gst/interfaces/mixer.c:
25028 * gst-libs/gst/pbutils/Makefile.am:
25029 * gst-libs/gst/pbutils/descriptions.c:
25030 (gst_pb_utils_get_source_description),
25031 (gst_pb_utils_get_sink_description),
25032 (gst_pb_utils_get_decoder_description),
25033 (gst_pb_utils_get_encoder_description),
25034 (gst_pb_utils_get_element_description),
25035 (gst_pb_utils_add_codec_description_to_tag_list),
25036 (gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
25037 * gst-libs/gst/pbutils/descriptions.h:
25038 * gst-libs/gst/pbutils/install-plugins.c:
25039 * gst-libs/gst/pbutils/install-plugins.h:
25040 * gst-libs/gst/pbutils/missing-plugins.c:
25041 (gst_missing_uri_source_message_new),
25042 (gst_missing_uri_sink_message_new),
25043 (gst_missing_element_message_new),
25044 (gst_missing_decoder_message_new),
25045 (gst_missing_encoder_message_new),
25046 (gst_missing_plugin_message_get_description):
25047 * gst-libs/gst/pbutils/missing-plugins.h:
25048 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
25049 * gst-libs/gst/pbutils/pbutils.h:
25050 * gst-libs/gst/utils/Makefile.am:
25051 * gst-libs/gst/utils/base-utils.c:
25052 * gst-libs/gst/utils/base-utils.h:
25053 * gst-libs/gst/utils/descriptions.c:
25054 * gst-libs/gst/utils/descriptions.h:
25055 * gst-libs/gst/utils/install-plugins.c:
25056 * gst-libs/gst/utils/install-plugins.h:
25057 * gst-libs/gst/utils/missing-plugins.c:
25058 * gst-libs/gst/utils/missing-plugins.h:
25059 * gst-plugins-base.spec.in:
25060 * gst/playback/Makefile.am:
25061 * gst/playback/gstdecodebin.c:
25062 * gst/playback/gstdecodebin2.c:
25063 * gst/playback/gstplaybasebin.c: (setup_subtitle),
25064 (gen_source_element):
25065 * gst/playback/gstplaybin.c: (plugin_init):
25066 * tests/check/Makefile.am:
25067 * tests/check/libs/pbutils.c: (GST_START_TEST),
25068 (test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
25069 * tests/check/libs/utils.c:
25070 rename utils to pbutils
25072 2007-03-03 10:23:03 +0000 David Schleef <ds@schleef.org>
25074 gst-libs/gst/app/Makefile.am: Install the headers.
25075 Original commit message from CVS:
25076 * gst-libs/gst/app/Makefile.am:
25077 Install the headers.
25079 2007-03-03 10:10:30 +0000 David Schleef <ds@schleef.org>
25081 gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.
25082 Original commit message from CVS:
25083 * gst-libs/gst/app/Makefile.am:
25084 * gst-libs/gst/app/gstappbuffer.c:
25085 * gst-libs/gst/app/gstappbuffer.h:
25086 * gst-libs/gst/app/gstappsrc.c:
25087 Add GstAppBuffer that includes a callback and closure for
25088 proper handling of data chunks.
25090 2007-03-03 09:06:06 +0000 David Schleef <ds@schleef.org>
25092 gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.
25093 Original commit message from CVS:
25094 * gst-libs/gst/app/gstappsrc.c:
25095 * gst-libs/gst/app/gstappsrc.h:
25096 Hacking to address issues in 413418.
25098 2007-03-03 08:16:57 +0000 David Schleef <ds@schleef.org>
25100 Move the app library to gst-libs/gst/app (duh!)
25101 Original commit message from CVS:
25105 * gst-libs/gst/Makefile.am:
25106 * gst-libs/gst/app/Makefile.am:
25107 * gst-libs/gst/app/gstapp.c:
25108 * gst-libs/gst/app/gstappsrc.c:
25109 * gst-libs/gst/app/gstappsrc.h:
25110 * gst/app/Makefile.am:
25111 * gst/app/gstapp.c:
25112 * gst/app/gstappsrc.c:
25113 * gst/app/gstappsrc.h:
25114 Move the app library to gst-libs/gst/app (duh!)
25116 2007-03-02 12:59:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25118 Add documentation for decodebin2 that indicates that the API is still unstable.
25119 Original commit message from CVS:
25120 * docs/plugins/Makefile.am:
25121 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
25122 * docs/plugins/gst-plugins-base-plugins-sections.txt:
25123 * docs/plugins/inspect/plugin-decodebin2.xml:
25124 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
25125 Add documentation for decodebin2 that indicates that the API
25128 2007-03-01 18:50:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25130 configure.ac: Update to 0.10.11.2 (0.10.12 pre-release)
25131 Original commit message from CVS:
25133 Update to 0.10.11.2 (0.10.12 pre-release)
25135 2007-03-01 17:29:55 +0000 Wim Taymans <wim.taymans@gmail.com>
25137 gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
25138 Original commit message from CVS:
25139 * gst-libs/gst/audio/gstbaseaudiosink.c:
25140 (gst_base_audio_sink_async_play):
25141 base time is irrelevant here.
25143 2007-03-01 17:01:43 +0000 Wim Taymans <wim.taymans@gmail.com>
25145 gst-libs/gst/audio/: Improve debugging.
25146 Original commit message from CVS:
25147 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
25148 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
25150 * gst-libs/gst/audio/gstbaseaudiosink.c:
25151 (gst_base_audio_sink_query), (gst_base_audio_sink_event),
25152 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
25153 Improve latency and clock slaving calculations.
25154 Improve slave clock calibration.
25155 * gst-libs/gst/audio/gstringbuffer.c:
25156 (gst_ring_buffer_commit_full):
25157 When we are asked to render N sample to 0 bytes, return N.
25159 2007-03-01 16:48:45 +0000 Wim Taymans <wim.taymans@gmail.com>
25161 ext/alsa/gstalsasink.*: Remove unused dispose function.
25162 Original commit message from CVS:
25163 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
25164 (gst_alsasink_write), (gst_alsasink_reset):
25165 * ext/alsa/gstalsasink.h:
25166 Remove unused dispose function.
25167 Rename lock to not interfere with alsasrc lock.
25168 * ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
25169 (gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
25170 (gst_alsasrc_read), (gst_alsasrc_reset):
25171 * ext/alsa/gstalsasrc.h:
25172 Implement finalize function.
25173 Use lock to protect alsa access.
25175 Fine tune sw params.
25177 2007-03-01 10:20:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25182 Original commit message from CVS:
25185 2007-02-28 19:27:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25187 configure.ac: Convert to new AG_GST style.
25188 Original commit message from CVS:
25190 Convert to new AG_GST style.
25192 2007-02-28 15:17:20 +0000 Ed Catmur <ed@catmur.co.uk>
25194 gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
25195 Original commit message from CVS:
25196 Patch by: Ed Catmur <ed at catmur dot co dot uk>
25197 * gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
25198 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
25199 Fix race condition when rapidly switching visualisations in playbin.
25202 2007-02-28 15:11:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25204 tests/check/Makefile.am: Include local stuff before system installed things in LDFLAGS and
25205 Original commit message from CVS:
25206 * tests/check/Makefile.am:
25207 Include local stuff before system installed things in LDFLAGS and
25210 2007-02-28 15:10:06 +0000 Wim Taymans <wim.taymans@gmail.com>
25212 ext/ogg/gstoggdemux.c: Improve debugging.
25213 Original commit message from CVS:
25214 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
25217 2007-02-28 15:05:03 +0000 Wim Taymans <wim.taymans@gmail.com>
25219 sys/v4l/: Fix duration and timestamping, taking latency into account.
25220 Original commit message from CVS:
25221 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
25222 (gst_v4lsrc_fixate), (gst_v4lsrc_query):
25223 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
25224 Fix duration and timestamping, taking latency into account.
25225 Implement latency query.
25227 2007-02-28 15:02:25 +0000 Wim Taymans <wim.taymans@gmail.com>
25229 gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
25230 Original commit message from CVS:
25231 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
25232 (gst_audio_clock_new):
25234 * gst-libs/gst/audio/gstbaseaudiosink.c:
25235 (gst_base_audio_sink_init), (gst_base_audio_sink_query):
25236 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
25237 (gst_base_audio_src_query), (gst_base_audio_src_get_offset),
25238 (gst_base_audio_src_create):
25239 Improve latency query code.
25240 Use proper clock names.
25242 2007-02-28 12:57:46 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25244 * tests/check/generic/states.c:
25246 Original commit message from CVS:
25249 2007-02-28 12:44:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25251 tests/check/generic/states.c: Copy the states.c test from core again
25252 Original commit message from CVS:
25253 * tests/check/generic/states.c: (GST_START_TEST):
25254 Copy the states.c test from core again
25255 * tests/check/Makefile.am:
25256 ignore cdio and cdparanoiasrc
25258 2007-02-28 12:08:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25260 gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
25261 Original commit message from CVS:
25262 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
25263 (double_hq), (audio_convert_get_func_index), (check_default),
25264 (audio_convert_prepare_context), (audio_convert_convert):
25265 Also make valgrind happy and avoid copying data in some cases.
25267 2007-02-28 11:58:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25269 * tests/check/generic/states.c:
25271 Original commit message from CVS:
25274 2007-02-28 11:47:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25276 Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
25277 Original commit message from CVS:
25278 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
25279 (double_hq), (audio_convert_get_func_index),
25280 (audio_convert_prepare_context), (audio_convert_convert):
25281 * gst/audioconvert/gstaudioconvert.c:
25282 (gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
25283 (gst_audio_convert_transform_caps):
25284 * tests/check/elements/audioconvert.c: (GST_START_TEST),
25285 (audioconvert_suite):
25286 Don't run inplace if that overwrites source data as we go. Add more
25287 tests. Fixes #339837 even more.
25289 2007-02-27 18:45:37 +0000 Julien Moutte <julien@moutte.net>
25291 tests/examples/seek/seek.c: Fix various seeking bugs (Slider was not updating when doing a non flushing seek, Reverse...
25292 Original commit message from CVS:
25293 2007-02-27 Julien MOUTTE <julien@moutte.net>
25294 * tests/examples/seek/seek.c: (do_seek), (set_update_scale),
25295 (msg_segment_done): Fix various seeking bugs (Slider was not
25296 updating when doing a non flushing seek, Reverse playback
25297 on segment seek was wrong).
25299 2007-02-26 21:01:03 +0000 David Schleef <ds@schleef.org>
25301 Add a new plugin/library to make it easy for apps to shove data into a pipeline.
25302 Original commit message from CVS:
25304 * gst/app/Makefile.am:
25305 * gst/app/gstapp.c:
25306 * gst/app/gstappsrc.c:
25307 * gst/app/gstappsrc.h:
25308 Add a new plugin/library to make it easy for apps to shove
25309 data into a pipeline.
25311 2007-02-26 11:48:49 +0000 Wim Taymans <wim.taymans@gmail.com>
25313 tests/examples/seek/seek.c: When we stop scrubbing, don't leave the pipeline PLAYING when we requested a PAUSED state.
25314 Original commit message from CVS:
25315 * tests/examples/seek/seek.c: (stop_seek):
25316 When we stop scrubbing, don't leave the pipeline PLAYING when we
25317 requested a PAUSED state.
25319 2007-02-25 23:51:03 +0000 René Stadler <mail@renestadler.de>
25321 gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
25322 Original commit message from CVS:
25323 Patch by: René Stadler <mail at renestadler de>
25324 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
25325 Parse date strings in vorbis comments that have an invalid (zero)
25326 month or day (#410396).
25327 * tests/check/libs/tag.c: (GST_START_TEST):
25328 Test case for the above.
25330 2007-02-24 20:12:49 +0000 Loïc Minier <lool+gnome@via.ecp.fr>
25332 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
25333 Original commit message from CVS:
25334 Patch by: Loïc Minier <lool+gnome at via ecp fr>
25336 * ext/alsa/Makefile.am:
25337 * gst/audiotestsrc/Makefile.am:
25338 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
25340 2007-02-23 18:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
25342 gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
25343 Original commit message from CVS:
25344 * gst/playback/gstplaybin.c:
25345 Improve docs: point out that the application needs to assist playbin
25348 2007-02-23 13:10:50 +0000 Tim-Philipp Müller <tim@centricular.net>
25350 Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
25351 Original commit message from CVS:
25352 * gst-libs/gst/utils/install-plugins.c:
25353 * gst-libs/gst/utils/missing-plugins.c:
25354 * tests/check/libs/utils.c: (missing_msg_check_getters):
25355 Change GStreamer marker prefix in detail string from 'gstreamer.net'
25356 to just 'gstreamer'. Document the caps string component of the
25357 decoder/encoder detail a bit better, since not everyone will be
25358 familiar with the GStreamer media type/caps system (but they better
25359 enjoy nested itemized lists).
25361 2007-02-22 12:57:47 +0000 Tim-Philipp Müller <tim@centricular.net>
25363 gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
25364 Original commit message from CVS:
25365 * gst-libs/gst/netbuffer/gstnetbuffer.c:
25366 (notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
25367 Fix copying of GstNetBuffer (would crash before, or at least lead to
25368 invalid memory access, #410772), for now by copying the GstBuffer copy
25369 code from the core over here so we can copy the GstBuffer fields on a
25370 provided buffer instance (of type GstNetBuffer in this case). Would be
25371 better to fix this with some support by the core though (and in the long
25372 run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
25373 * tests/check/Makefile.am:
25374 Enable unit test for GstNetBuffer.
25376 2007-02-22 11:04:10 +0000 Andy Wingo <wingo@pobox.com>
25379 * gst-libs/gst/audio/gstbaseaudiosink.c:
25380 gst-libs/gst/audio/gstbaseaudiosink.c
25381 Original commit message from CVS:
25382 2007-02-22 Andy Wingo <wingo@pobox.com>
25383 * gst-libs/gst/audio/gstbaseaudiosink.c
25384 (gst_base_audio_sink_init): Disable pull-mode activation until we
25385 figure out how to make audio sinks go to PLAYING.
25387 2007-02-22 09:04:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25389 Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
25390 Original commit message from CVS:
25391 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
25392 (double_hq), (audio_convert_get_func_index),
25393 (audio_convert_prepare_context), (audio_convert_convert):
25394 * gst/audioconvert/audioconvert.h:
25395 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
25396 (gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
25397 * gst/audioconvert/gstchannelmix.h:
25398 * tests/check/elements/audioconvert.c: (GST_START_TEST):
25399 Add float as an intermediate format, as well as float mixing. Enable
25400 test that was failing before. Fixes #339837
25402 2007-02-21 16:12:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25404 tests/examples/seek/seek.c: Undo the previous commit: -1 as a stop time implies that the stop time is the end of file...
25405 Original commit message from CVS:
25406 * tests/examples/seek/seek.c: (do_seek):
25407 Undo the previous commit: -1 as a stop time implies that the stop
25408 time is the end of file, clearing any previously configured segment.
25410 2007-02-21 15:36:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25412 tests/examples/seek/seek.c: Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
25413 Original commit message from CVS:
25414 * tests/examples/seek/seek.c: (do_seek):
25415 Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
25417 2007-02-21 13:55:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25419 gst/volume/gstvolume.c: Unbreak volume, value remains gint.
25420 Original commit message from CVS:
25421 * gst/volume/gstvolume.c: (volume_process_int16),
25422 (volume_process_int16_clamp), (volume_set_caps):
25423 Unbreak volume, value remains gint.
25425 2007-02-21 13:08:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25427 gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
25428 Original commit message from CVS:
25429 * gst/volume/gstvolume.c: (volume_choose_func),
25430 (volume_update_real_volume), (gst_volume_set_volume),
25431 (gst_volume_init), (volume_process_double), (volume_process_float),
25432 (volume_process_int16), (volume_process_int16_clamp),
25433 (volume_set_caps), (volume_transform_ip), (volume_update_volume):
25434 * gst/volume/gstvolume.h:
25435 Extend float audio support (double) and some int->uint cleanups.
25437 2007-02-20 15:44:32 +0000 Edward Hervey <bilboed@bilboed.com>
25439 gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
25440 Original commit message from CVS:
25441 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
25442 (multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
25443 (sort_end_pads), (gst_decode_group_expose),
25444 (gst_decode_group_hide):
25445 Don't free groups from the streaming threads. Just put them aside and
25446 free them in dispose.
25448 2007-02-20 11:20:52 +0000 Edward Hervey <bilboed@bilboed.com>
25450 gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
25451 Original commit message from CVS:
25452 * gst/playback/gstdecodebin2.c: (connect_element),
25453 (pad_added_group_cb), (gst_decode_group_check_if_blocked),
25454 (sort_end_pads), (gst_decode_group_expose):
25455 Handle dynamic pads within groups.
25456 Sort pads before exposing them in order to make playbin happy.
25457 There still is a race with the multiqueue filling up. This should be
25461 2007-02-18 21:02:36 +0000 Tim-Philipp Müller <tim@centricular.net>
25463 gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
25464 Original commit message from CVS:
25465 * gst-libs/gst/utils/base-utils.c:
25466 * gst-libs/gst/utils/descriptions.c:
25467 * gst-libs/gst/utils/install-plugins.c:
25468 * gst-libs/gst/utils/missing-plugins.c:
25469 Some more docs (and descriptions for two subtitle formats).
25471 2007-02-16 10:19:45 +0000 Tim-Philipp Müller <tim@centricular.net>
25473 gst-libs/gst/audio/audio.c: Fix documentation.
25474 Original commit message from CVS:
25475 * gst-libs/gst/audio/audio.c:
25478 2007-02-16 10:15:46 +0000 Yves Lefebvre <ivanohe@abacom.com>
25480 gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
25481 Original commit message from CVS:
25482 Patch by: Yves Lefebvre <ivanohe abacom com>
25483 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
25484 Don't leak caps. Fixes #408278.
25486 2007-02-15 15:17:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25488 More docs coverage and some ChangeLog surgery (add missing names)
25489 Original commit message from CVS:
25490 * ext/cdparanoia/gstcdparanoiasrc.h:
25491 * ext/ogg/gstoggdemux.h:
25492 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
25493 (gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
25494 (gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
25495 * gst-libs/gst/audio/audio.h:
25496 * gst-libs/gst/audio/gstaudiofilter.h:
25497 * gst-libs/gst/interfaces/videoorientation.h:
25498 * gst/adder/gstadder.h:
25499 More docs coverage and some ChangeLog surgery (add missing names)
25501 2007-02-15 12:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
25503 sys/: Small constifications.
25504 Original commit message from CVS:
25505 * sys/ximage/ximagesink.c:
25506 (gst_ximagesink_calculate_pixel_aspect_ratio):
25507 * sys/xvimage/xvimagesink.c:
25508 (gst_xvimagesink_calculate_pixel_aspect_ratio):
25509 Small constifications.
25511 2007-02-15 12:06:25 +0000 Wim Taymans <wim.taymans@gmail.com>
25513 gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
25514 Original commit message from CVS:
25515 * gst-libs/gst/audio/gstbaseaudiosink.c:
25516 (gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
25517 (gst_base_audio_sink_render), (gst_base_audio_sink_callback),
25518 (gst_base_audio_sink_async_play),
25519 (gst_base_audio_sink_change_state):
25520 Answer latency query.
25521 Use configured latency when syncing.
25523 * gst-libs/gst/audio/gstbaseaudiosrc.c:
25524 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
25525 (gst_base_audio_src_query), (gst_base_audio_src_change_state):
25526 Fix possible memleak.
25527 Implement latency query.
25530 2007-02-15 11:59:41 +0000 Wim Taymans <wim.taymans@gmail.com>
25532 ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
25533 Original commit message from CVS:
25534 * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
25535 Ignore errors in reset, these are not fatal. They also grab the element
25536 lock which is already taking when this function is called. Fixes
25539 2007-02-13 13:50:56 +0000 Christian Schaller <uraeus@gnome.org>
25541 * gst-plugins-base.spec.in:
25542 add header file for easy codec install
25543 Original commit message from CVS:
25544 add header file for easy codec install
25546 2007-02-13 10:24:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25548 configure.ac: Remove 'tests/examples/xerror/Makefile' from output files again.
25549 Original commit message from CVS:
25551 Remove 'tests/examples/xerror/Makefile' from output files again.
25553 2007-02-13 09:12:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25555 Also crossref against gst-plugins-base-libs.
25556 Original commit message from CVS:
25558 * docs/plugins/Makefile.am:
25559 Also crossref against gst-plugins-base-libs.
25561 2007-02-12 20:42:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25563 Add crossreferences to glib/gobject/gstream docs.
25564 Original commit message from CVS:
25566 * docs/libs/Makefile.am:
25567 * docs/plugins/Makefile.am:
25568 Add crossreferences to glib/gobject/gstream docs.
25569 * gst-libs/gst/audio/audio.h:
25571 * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
25572 Add own debug category.
25574 2007-02-12 11:01:04 +0000 René Stadler <mail@renestadler.de>
25576 gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
25577 Original commit message from CVS:
25578 Patch by: René Stadler <mail at renestadler de>
25579 * gst-libs/gst/tag/gstvorbistag.c:
25580 Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
25583 2007-02-12 10:33:40 +0000 Tim-Philipp Müller <tim@centricular.net>
25585 gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
25586 Original commit message from CVS:
25587 * gst/playback/gstplaybasebin.c: (setup_source):
25588 When we have external subtitles and wait for the subtitle decodebin
25589 to get up and running, we set up a (sync) bus handler for the
25590 subtitle decodebin, so we can stop waiting when it posts an error
25591 message. However, we should do that before we set the subtitle
25592 decodebin's state to playing, otherwise things are racy and we might
25593 miss error messages posted before we had a chance to set up the bus.
25594 This should finally fix totem hanging on .txt pseudo-subtitle files.
25596 2007-02-10 19:27:48 +0000 Sébastien Moutte <sebastien@moutte.net>
25598 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
25599 Original commit message from CVS:
25600 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
25601 Use gst_gdouble_to_guint64 for conversions.
25602 * win32/common/config.h.in:
25603 Add a define for GST_INSTALL_PLUGINS_HELPER
25604 * win32/common/libgstaudio.def:
25605 * win32/common/libgstcdda.def:
25606 * win32/common/libgstnetbuffer.def:
25607 * win32/common/libgstrtp.def:
25608 * win32/common/libgutils.def:
25609 Add new exported functions.
25610 * win32/vs6/gst_plugins_base.dsw:
25611 * win32/vs6/libgstdecodebin.dsp:
25612 * win32/vs6/libgstnetbuffer.dsp:
25613 * win32/vs6/libgstplaybin.dsp:
25614 * win32/vs6/libgstrtp.dsp:
25615 * win32/vs6/libgstvorbis.dsp:
25616 * win32/vs6/libgstcdda.dsp:
25617 * win32/vs6/libgstgdp.dsp:
25618 * win32/vs6/libgstutils.dsp:
25619 Update and add new project files.
25621 2007-02-10 18:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
25623 gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
25624 Original commit message from CVS:
25625 * gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
25626 (subrip_remove_unhandled_tags), (parse_subrip):
25627 For SubRip (.srt) subtitles, ignore all markup tags we don't
25628 handle (like font tags, for example).
25629 * tests/check/elements/subparse.c:
25632 2007-02-09 13:28:01 +0000 Tim-Philipp Müller <tim@centricular.net>
25636 Original commit message from CVS:
25639 2007-02-09 13:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
25641 gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
25642 Original commit message from CVS:
25643 * gst/playback/gstdecodebin.c: (add_fakesink),
25644 (gst_decode_bin_change_state):
25645 * gst/playback/gstdecodebin2.c: (add_fakesink),
25646 (gst_decode_bin_change_state):
25647 Don't error out if there is no fakesink in the READY to NULL state
25648 change, since when decodebin is re-used, we're only adding the
25649 fakesink element in READY to PAUSED.
25650 * tests/check/elements/decodebin.c:
25651 (new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
25653 Minimal unit test to make sure we can use the same decodebin
25654 instance twice (at least with audiotestsrc input).
25656 2007-02-09 09:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
25658 ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
25659 Original commit message from CVS:
25660 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
25661 Try to get devic-name from device string first, and from handle only
25662 as fallback (seems to yield better results and is more robust
25663 against buggy probing code on the application side).
25665 2007-02-08 15:43:26 +0000 Julien Puydt <julien.puydt@laposte.net>
25667 ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
25668 Original commit message from CVS:
25669 Based on patch by: Julien Puydt <julien.puydt at laposte net>
25670 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
25671 (gst_alsa_find_device_name):
25672 * ext/alsa/gstalsa.h:
25673 * ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
25674 * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
25675 Improve device-name detection a bit, especially in the case where
25676 the device is not actually open (#405020, #405024). Move common code
25677 into gstalsa.c instead of duplicating it.
25679 2007-02-07 13:05:01 +0000 Tim-Philipp Müller <tim@centricular.net>
25681 gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
25682 Original commit message from CVS:
25683 * gst/audioconvert/gstaudioconvert.c:
25684 Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
25686 2007-02-06 17:47:32 +0000 Julien Moutte <julien@moutte.net>
25688 sys/xvimage/xvimagesink.*: Implement PropertyProbe Interface for XVAdaptors so that one can choose the adaptor to use...
25689 Original commit message from CVS:
25690 2007-02-06 Julien MOUTTE <julien@moutte.net>
25691 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
25692 (gst_xvimagesink_get_xv_support),
25693 (gst_xvimagesink_xcontext_clear),
25694 (gst_xvimagesink_interface_supported),
25695 (gst_xvimagesink_probe_get_properties),
25696 (gst_xvimagesink_probe_probe_property),
25697 (gst_xvimagesink_probe_needs_probe),
25698 (gst_xvimagesink_probe_get_values),
25699 (gst_xvimagesink_property_probe_interface_init),
25700 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
25701 (gst_xvimagesink_init), (gst_xvimagesink_class_init),
25702 (gst_xvimagesink_get_type):
25703 * sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
25704 for XVAdaptors so that one can choose the adaptor to use with
25705 gstreamer-properties.
25707 2007-02-06 14:00:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25709 gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
25710 Original commit message from CVS:
25711 * gst/audioconvert/gstaudioconvert.c:
25712 Also mention that a conversion from double to float is suboptimal still.
25714 2007-02-06 09:42:05 +0000 Tim-Philipp Müller <tim@centricular.net>
25716 gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
25717 Original commit message from CVS:
25718 * gst-libs/gst/audio/gstaudiofilter.c:
25719 (gst_audio_filter_class_init), (gst_audio_filter_change_state):
25720 Clear our formats structure and free the caps contained in it when
25723 2007-02-05 18:39:51 +0000 Andy Wingo <wingo@pobox.com>
25726 * gst-libs/gst/audio/gstbaseaudiosink.c:
25727 gst-libs/gst/audio/gstbaseaudiosink.c
25728 Original commit message from CVS:
25729 2007-02-05 Andy Wingo <wingo@pobox.com>
25730 * gst-libs/gst/audio/gstbaseaudiosink.c
25731 (gst_base_audio_sink_callback): Update basesink->offset so that we
25732 pull monotonically increasing offsets instead of, um, seeking back
25733 to 0 each time. Fixes alsasrc ! alsasink!
25735 2007-02-05 11:44:52 +0000 Tim-Philipp Müller <tim@centricular.net>
25737 gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
25738 Original commit message from CVS:
25739 * gst/videoscale/gstvideoscale.c:
25740 A width and height of 1 makes us crash, so increase minimum size to
25741 2x2 pixels until someone feels like fixing this (#404512).
25743 2007-02-04 16:23:37 +0000 Tim-Philipp Müller <tim@centricular.net>
25745 tests/check/pipelines/oggmux.c: Add small test to make sure request pads are cleaned up properly even if oggmux never...
25746 Original commit message from CVS:
25747 * tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
25748 Add small test to make sure request pads are cleaned up properly
25749 even if oggmux never changes state out of NULL.
25751 2007-02-04 14:11:51 +0000 Tim-Philipp Müller <tim@centricular.net>
25753 tests/check/libs/utils.c: Fix unit test. Turns out things work much better when you
25754 Original commit message from CVS:
25755 * tests/check/libs/utils.c: (GST_START_TEST):
25756 Fix unit test. Turns out things work much better when you
25757 NULL-terminate string arrays. Should make p5 build bot happy again.
25759 2007-02-03 23:28:45 +0000 Tim-Philipp Müller <tim@centricular.net>
25761 gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
25762 Original commit message from CVS:
25763 * gst-libs/gst/audio/Makefile.am:
25764 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
25765 (gst_audio_filter_template_base_init),
25766 (gst_audio_filter_template_class_init),
25767 (gst_audio_filter_template_init),
25768 (gst_audio_filter_template_set_property),
25769 (gst_audio_filter_template_get_property),
25770 (gst_audio_filter_template_setup),
25771 (gst_audio_filter_template_filter),
25772 (gst_audio_filter_template_filter_inplace), (plugin_init):
25773 Oops, forgot to commit fixed-up example.
25775 2007-02-03 20:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
25777 Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
25778 Original commit message from CVS:
25779 * docs/libs/gst-plugins-base-libs-sections.txt:
25780 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
25781 (gst_audio_filter_class_init), (gst_audio_filter_init),
25782 (gst_audio_filter_set_caps),
25783 (gst_audio_filter_class_add_pad_templates):
25784 * gst-libs/gst/audio/gstaudiofilter.h:
25785 Port GstAudioFilter to 0.10. This change technically breaks
25786 API and ABI (and thus also every library developer's heart),
25787 but seems justifiable on the grounds that the base class was
25788 completely unusable before (ie. would crash immediately when
25789 actually used). Fixes #403963 (and eventually also #403572).
25790 Also document all of this a bit.
25792 2007-02-03 14:26:54 +0000 Tim-Philipp Müller <tim@centricular.net>
25794 Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
25795 Original commit message from CVS:
25796 * gst-libs/gst/utils/install-plugins.c:
25797 (gst_install_plugins_spawn_child):
25798 * tests/check/libs/utils.c:
25799 (test_base_utils_install_plugins_do_callout):
25800 Lowering log level to see why things fail on the p5 build bot;
25801 fix some typos in unit test messages.
25803 2007-02-03 13:59:27 +0000 Tim-Philipp Müller <tim@centricular.net>
25805 tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
25806 Original commit message from CVS:
25807 * tests/check/libs/utils.c:
25808 (test_base_utils_install_plugins_do_callout):
25809 Don't hard-code temp directory for test helper; use GLib functions
25810 to write out file and do error checking etc.
25812 2007-02-02 20:42:08 +0000 Tim-Philipp Müller <tim@centricular.net>
25814 gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
25815 Original commit message from CVS:
25816 * gst-libs/gst/utils/Makefile.am:
25817 * gst-libs/gst/utils/base-utils.h:
25818 * gst-libs/gst/utils/install-plugins.c:
25819 (gst_install_plugins_context_set_xid),
25820 (gst_install_plugins_context_new),
25821 (gst_install_plugins_context_free),
25822 (gst_install_plugins_get_helper),
25823 (gst_install_plugins_spawn_child),
25824 (gst_install_plugins_return_from_status),
25825 (gst_install_plugins_installer_exited),
25826 (gst_install_plugins_async), (gst_install_plugins_sync),
25827 (gst_install_plugins_return_get_name),
25828 (gst_install_plugins_installation_in_progress):
25829 * gst-libs/gst/utils/install-plugins.h:
25830 API: add API for applications to initiate installation of missing
25831 plugins, ie. gst_install_plugins_async() primarily.
25832 Based on libgimme-codec by Ryan Lortie.
25834 Add --with-install-plugins-helper configure option so distros can specify
25835 the path of the helper script or program to call when plugin installation
25836 is requested (distros: please do any argument munging in this helper
25837 script instead of patching GStreamer to pass arguments differently
25838 to another program directly).
25839 * docs/libs/gst-plugins-base-libs-docs.sgml:
25840 * docs/libs/gst-plugins-base-libs-sections.txt:
25841 Build and document new API.
25842 * tests/check/libs/utils.c: (result_cb),
25843 (test_base_utils_install_plugins_do_callout), (GST_START_TEST),
25844 (libgstbaseutils_suite):
25845 Some simple checks for the new API.
25847 2007-02-02 14:44:29 +0000 Tim-Philipp Müller <tim@centricular.net>
25849 tests/check/elements/audioconvert.c: Add small test for 32bit float <=> 64bit float conversion (works only one way so...
25850 Original commit message from CVS:
25851 * tests/check/elements/audioconvert.c: (test_float_conversion):
25852 Add small test for 32bit float <=> 64bit float conversion (works
25853 only one way so far, 32=>64 produces structured noise).
25855 2007-02-02 11:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
25857 gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
25858 Original commit message from CVS:
25859 * gst/audioconvert/gstaudioconvert.c:
25860 (set_structure_widths_32_and_64), (make_lossless_changes):
25861 We don't support floats with a width of 40, 48 or 56 bits.
25863 2007-02-02 09:48:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25865 gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
25866 Original commit message from CVS:
25867 * gst/audioconvert/audioconvert.c: (float), (double),
25868 (audio_convert_get_func_index):
25869 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
25870 (make_lossless_changes):
25871 Support for 64-bit float audio in audioconvert (#339837)
25873 2007-02-01 18:50:08 +0000 Holger Wansing <linux@wansing-online.de>
25875 po/: Add German translation (#352069).
25876 Original commit message from CVS:
25877 Patch by: Holger Wansing <linux wansing-online de>
25880 Add German translation (#352069).
25882 2007-02-01 17:52:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
25884 ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
25885 Original commit message from CVS:
25886 reviewed by: Wim Taymans <wim@fluendo.com>
25887 * ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
25888 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
25889 Use newly added GstCollectPads API to free the allocated resources in
25890 the GstOggPad structures (#402393).
25892 2007-01-31 15:58:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25894 gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
25895 Original commit message from CVS:
25896 * gst/playback/gstplaybin.c: (gen_vis_element):
25897 Add audioresample+audioconvert in front of the visualisation
25898 element, so that elements like libvisual 0.4 that don't support all
25899 samplerates can work.
25902 2007-01-30 19:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
25904 gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
25905 Original commit message from CVS:
25906 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
25907 (gst_play_base_bin_get_streaminfo_value_array):
25908 Take some locks and make a copy of the streaminfo value array we
25909 maintain while holding the lock, so that the application can
25910 retrieve the stream-info as a value array in a thread-safe way.
25912 2007-01-30 11:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
25914 gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
25915 Original commit message from CVS:
25916 * gst/audioconvert/gstaudioconvert.c:
25917 Don't fail on 0 sized buffers. Fixes #396835.
25919 2007-01-29 21:13:07 +0000 David Schleef <ds@schleef.org>
25921 gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
25922 Original commit message from CVS:
25923 * gst/typefind/gsttypefindfunctions.c:
25924 Detect BBCD as video/x-dirac, so we can play raw dirac
25927 2007-01-29 18:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
25929 ext/theora/theoraenc.c: Check return value of theora_encode_header(), or we might try to allocate a random number of ...
25930 Original commit message from CVS:
25931 * ext/theora/theoraenc.c: (theora_enc_chain):
25932 Check return value of theora_encode_header(), or we might try to
25933 allocate a random number of bytes. theora_encode_header() can fail
25934 if libtheora has been compiled with encoding support disabled.
25937 2007-01-29 10:53:06 +0000 Wim Taymans <wim.taymans@gmail.com>
25939 tests/check/gst/.cvsignore: Do as buildbot says.
25940 Original commit message from CVS:
25941 * tests/check/gst/.cvsignore:
25942 Do as buildbot says.
25944 2007-01-29 10:25:11 +0000 Wim Taymans <wim.taymans@gmail.com>
25946 ext/libvisual/visual.c: Fix strides in libvisual. Gst uses X strides.
25947 Original commit message from CVS:
25948 * ext/libvisual/visual.c: (gst_visual_src_setcaps):
25949 Fix strides in libvisual. Gst uses X strides.
25950 Inspired by: <ed at catmur dot co dot uk> and
25951 <tim at centricular dot net>
25954 2007-01-27 13:32:24 +0000 Wim Taymans <wim.taymans@gmail.com>
25956 ext/ogg/gstoggdemux.*: Properly propagate streaming errors when we are scanning the file for chains so that we don't ...
25957 Original commit message from CVS:
25958 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
25959 (gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
25960 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
25961 (gst_ogg_demux_perform_seek),
25962 (gst_ogg_demux_bisect_forward_serialno),
25963 (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
25964 (gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
25965 (gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
25966 (gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
25967 * ext/ogg/gstoggdemux.h:
25968 Properly propagate streaming errors when we are scanning the file for
25969 chains so that we don't crash when shut down. Might fix some crashers
25970 when quickly switching oggs in RB such as #332503 and #378436.
25972 2007-01-26 12:44:46 +0000 Tim-Philipp Müller <tim@centricular.net>
25974 ext/gnomevfs/gstgnomevfssrc.c: Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND error code as well.
25975 Original commit message from CVS:
25976 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
25977 Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
25978 error code as well.
25980 2007-01-25 16:02:41 +0000 Wim Taymans <wim.taymans@gmail.com>
25982 gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
25983 Original commit message from CVS:
25984 * gst/playback/gstplaybasebin.c: (remove_source):
25985 Don't try to disconnect a signal from a finalized object.
25987 2007-01-25 14:29:21 +0000 Tim-Philipp Müller <tim@centricular.net>
25989 gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
25990 Original commit message from CVS:
25991 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
25992 Cast lock macro parameters to make sure we're actually accessing the
25993 lock member at the right class level. Free list itself in _dispose()
25994 as well and NULL it in case dispose gets called multiple times.
25996 2007-01-25 14:02:37 +0000 Edward Hervey <bilboed@bilboed.com>
25998 gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
25999 Original commit message from CVS:
26000 * gst/playback/gstdecodebin2.c:
26001 (gst_decode_bin_dispose),(gst_decode_bin_finalize):
26002 Free GstDecodeGroups no longer used.
26003 (gst_decode_group_expose):
26004 Don't unlock too many times !
26005 (deactivate_free_recursive):
26006 Free iterator once we're done with it.
26007 Fix for recursively deactivating elements (stop at ghostpads).
26009 2007-01-25 12:24:18 +0000 Tim-Philipp Müller <tim@centricular.net>
26011 gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
26012 Original commit message from CVS:
26013 * gst/playback/gstplaybin.c: (handoff):
26014 Fix up caps on the frame buffer before we save it and potentially
26015 make it accessible to other threads via g_object_get; also use
26016 gst_buffer_replace() instead of gst_mini_object_replace().
26018 2007-01-25 12:06:59 +0000 Tim-Philipp Müller <tim@centricular.net>
26020 gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
26021 Original commit message from CVS:
26022 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
26023 Make getting the current frame thread-safe.
26025 2007-01-25 11:48:10 +0000 Edward Hervey <bilboed@bilboed.com>
26027 gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
26028 Original commit message from CVS:
26029 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
26030 (gst_decode_group_new), (gst_decode_group_free):
26031 Set queues to bigger sizes to cope with HD contents.
26032 Fix some mutex freeing and add comment about MT safe methods.
26034 2007-01-24 12:51:20 +0000 Tim-Philipp Müller <tim@centricular.net>
26036 ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
26037 Original commit message from CVS:
26038 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
26039 (gst_text_overlay_text_event):
26040 Don't unnecessarily ref (and then leak) upstream events if the text
26041 pad is not linked. Fixes #399948.
26042 * tests/check/gst-plugins-base.supp:
26043 Add suppression for pango on edgy/x86 for textoverlay test.
26045 2007-01-24 12:10:56 +0000 Wim Taymans <wim.taymans@gmail.com>
26047 gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
26048 Original commit message from CVS:
26049 * gst-libs/gst/rtp/gstrtpbuffer.h:
26050 Add some more fixed payloads.
26052 2007-01-23 18:39:45 +0000 Tim-Philipp Müller <tim@centricular.net>
26054 ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
26055 Original commit message from CVS:
26056 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
26057 Error out properly if we get an error from libogg while reading the
26058 BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).
26060 2007-01-23 17:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
26062 gst/playback/gstdecodebin2.c: Don't leak mutex.
26063 Original commit message from CVS:
26064 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
26066 * tests/check/elements/playbin.c:
26067 (test_sink_usage_video_only_stream),
26068 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
26069 (test_suburi_error_wrongproto), (test_missing_urisource_handler),
26070 (test_missing_suburisource_handler),
26071 (test_missing_primary_decoder), (playbin_suite):
26072 Run all tests once with decodebin and once with decodebin2.
26073 One test does not pass yet with decodebin2.
26075 2007-01-23 14:30:28 +0000 Edward Hervey <bilboed@bilboed.com>
26077 ext/ogg/gstoggmux.c: Fix the cases where oggmux doesn't properly figure out that all sinkpads have gone EOS, and ther...
26078 Original commit message from CVS:
26079 * ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
26080 Fix the cases where oggmux doesn't properly figure out that all
26081 sinkpads have gone EOS, and therefore doesn't push out the remaining
26082 buffers and the final EOS event.
26085 2007-01-23 13:19:19 +0000 Julien Moutte <julien@moutte.net>
26087 sys/: Don't lock on navigation event push, just on keysym to string.
26088 Original commit message from CVS:
26089 2007-01-23 Julien MOUTTE <julien@moutte.net>
26090 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
26091 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
26092 Don't lock on navigation event push, just on keysym to string.
26093 Fixes #397673 again.
26095 2007-01-22 17:37:38 +0000 Edward Hervey <bilboed@bilboed.com>
26097 gst/playback/gstdecodebin2.c: Cleanups.
26098 Original commit message from CVS:
26099 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
26100 (get_current_group), (group_demuxer_event_probe),
26101 (gst_decode_group_expose), (deactivate_free_recursive),
26102 (gst_decode_group_free):
26104 Don't forget to emit 'no-more-pads' once a group is exposed.
26105 Cleanup elements from a DecodeGroup once we remove it.
26106 Protect call to gst_decode_group_expose() with the decodebin lock.
26108 2007-01-22 13:16:42 +0000 Julien Moutte <julien@moutte.net>
26110 sys/: Looking at Xorg code i can't figure out if that XKeysymToString function is thread sensible or not. Lock it jus...
26111 Original commit message from CVS:
26112 2007-01-22 Julien MOUTTE <julien@moutte.net>
26113 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
26114 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
26115 Looking at Xorg code i can't figure out if that XKeysymToString
26116 function is thread sensible or not. Lock it just in case as
26117 recommended by Radek Doulik <rodo at ximian dot com>.
26119 2007-01-22 13:10:13 +0000 Julien Moutte <julien@moutte.net>
26121 sys/: Lock that X Call as well. Fixes #397673.
26122 Original commit message from CVS:
26123 2007-01-22 Julien MOUTTE <julien@moutte.net>
26124 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
26125 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
26126 Lock that X Call as well. Fixes #397673.
26128 2007-01-22 12:03:27 +0000 Tim-Philipp Müller <tim@centricular.net>
26130 gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
26131 Original commit message from CVS:
26132 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
26133 Don't go into an endless loop if the file starts with 00 00 01 2X,
26134 like quicktime redirect files might. Fixes #396042.
26135 * tests/check/Makefile.am:
26136 * tests/check/gst/.cvsignore:
26137 * tests/check/gst/typefindfunctions.c: (GST_START_TEST),
26138 (typefindfunctions_suite):
26139 Add unit test for the above.
26141 2007-01-22 10:27:26 +0000 Tim-Philipp Müller <tim@centricular.net>
26143 gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
26144 Original commit message from CVS:
26145 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
26146 On second thought, use "depth" field rather than "bpp" field.
26148 2007-01-22 09:23:01 +0000 Tim-Philipp Müller <tim@centricular.net>
26150 gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
26151 Original commit message from CVS:
26152 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
26153 Camtasia caps apparently need a bpp field (#398875).
26155 2007-01-19 19:09:05 +0000 Tim-Philipp Müller <tim@centricular.net>
26157 gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
26158 Original commit message from CVS:
26159 * gst/playback/gstplaybasebin.c: (setup_subtitle),
26160 (gen_source_element), (gst_play_base_bin_change_state):
26161 Attempt at a better error message in case we don't have the required
26162 URI handler installed; post missing-plugin message also when we're
26163 missing an URI handler for the subtitle URI; clean up properly also
26164 when an error occurs and we never made it to PAUSED state.
26165 * tests/check/elements/playbin.c: (GST_START_TEST),
26167 Check that we're also getting a missing-plugin messsage for a
26168 missing subtitle URI handler (and clean up properly).
26170 2007-01-19 18:47:30 +0000 Tim-Philipp Müller <tim@centricular.net>
26172 gst/playback/gstplaybasebin.c: Plug a few reference leaks.
26173 Original commit message from CVS:
26174 * gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
26175 Plug a few reference leaks.
26177 2007-01-19 12:23:06 +0000 Tim-Philipp Müller <tim@centricular.net>
26179 gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
26180 Original commit message from CVS:
26181 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
26182 Lower probability a bit if the marker isn't right at the start,
26183 to decrease the chance of false positives.
26185 2007-01-19 11:31:50 +0000 Tim-Philipp Müller <tim@centricular.net>
26187 gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
26188 Original commit message from CVS:
26189 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
26190 Small mpeg2 system stream typefinding improvement: make typefinder
26191 probe a bit into the stream instead of just looking for a marker
26192 at the beginning. Fixes #397810.
26194 2007-01-18 16:23:35 +0000 Tim-Philipp Müller <tim@centricular.net>
26196 gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
26197 Original commit message from CVS:
26198 * gst/audioconvert/gstchannelmix.c:
26199 Remove compatibility cruft for prehistoric GLib versions.
26201 2007-01-17 16:11:14 +0000 Tim-Philipp Müller <tim@centricular.net>
26203 gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
26204 Original commit message from CVS:
26205 * gst/playback/Makefile.am:
26206 * gst/playback/gstdecodebin.c: (close_pad_link):
26207 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
26208 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
26209 (gst_play_base_bin_handle_message_func), (unknown_type):
26210 Let decodebin be the element to post missing-plugin messages for
26211 missing decoders (rather than playbin); make playbin implement
26212 GstBin::handle_message so we can suppress missing-plugin messages
26213 for types we're not handling on purpose (don't want to bring up an
26214 installer in those cases).
26216 2007-01-16 19:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
26218 gst/: Fix potentially unaligned access (#397207).
26219 Original commit message from CVS:
26220 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
26221 * gst-libs/gst/tag/gstvorbistag.c:
26222 (gst_tag_list_to_vorbiscomment_buffer):
26223 * gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
26224 Fix potentially unaligned access (#397207).
26226 2007-01-16 12:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26228 tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
26229 Original commit message from CVS:
26230 * tests/examples/seek/seek.c: (set_scale), (update_scale),
26231 (do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
26232 (rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
26234 Allow to toggle looping while it plays. Fix callback prototype. Clean
26235 up code a bit more. Add copyright header.
26237 2007-01-16 11:41:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26239 sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
26240 Original commit message from CVS:
26241 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
26242 Red and blue mask was swapped (spotted by Dan Williams).
26244 2007-01-15 13:58:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26246 gst-libs/gst/tag/: Use new beats-per-minute tag from core.
26247 Original commit message from CVS:
26248 * gst-libs/gst/tag/gstid3tag.c:
26249 * gst-libs/gst/tag/gstvorbistag.c:
26250 Use new beats-per-minute tag from core.
26252 2007-01-15 11:30:53 +0000 Tim-Philipp Müller <tim@centricular.net>
26254 po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
26255 Original commit message from CVS:
26257 Add new files with translatable strings, so they actually make it
26258 into the template file one day.
26260 2007-01-12 21:19:35 +0000 Andy Wingo <wingo@pobox.com>
26263 * gst-libs/gst/audio/gstbaseaudiosink.c:
26264 * gst-libs/gst/audio/gstbaseaudiosrc.c:
26265 gst-libs/gst/audio/gstbaseaudiosink.c
26266 Original commit message from CVS:
26267 2007-01-12 Andy Wingo <wingo@pobox.com>
26268 * gst-libs/gst/audio/gstbaseaudiosink.c
26269 (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
26270 (gst_base_audio_sink_activate_pull): Remove the handwavey nego
26271 stuff, as the base class handles this now. Actually tell the ring
26273 (gst_base_audio_sink_callback): Cast the ring buffer correctly.
26274 How did this work before? Maybe I'm not as awesome a programmer as
26276 * gst-libs/gst/audio/gstbaseaudiosrc.c
26277 (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
26280 2007-01-12 18:08:23 +0000 Tim-Philipp Müller <tim@centricular.net>
26282 gst-libs/gst/utils/missing-plugins.c: Remove more fields so that the application can better blacklist formats that ha...
26283 Original commit message from CVS:
26284 * gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
26285 Remove more fields so that the application can better blacklist
26286 formats that have been tried before.
26288 2007-01-12 17:43:40 +0000 Christian Schaller <uraeus@gnome.org>
26290 * gst-plugins-base.spec.in:
26292 Original commit message from CVS:
26295 2007-01-12 12:47:29 +0000 Tim-Philipp Müller <tim@centricular.net>
26297 gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
26298 Original commit message from CVS:
26299 * gst-libs/gst/audio/mixerutils.h:
26300 Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
26301 used when compiling with c++ compilers as well.
26303 2007-01-12 09:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
26305 gst/typefind/gsttypefindfunctions.c: Fix comment.
26306 Original commit message from CVS:
26307 * gst/typefind/gsttypefindfunctions.c:
26310 2007-01-11 13:12:17 +0000 Tim-Philipp Müller <tim@centricular.net>
26312 gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
26313 Original commit message from CVS:
26314 * gst/playback/gstplaybin.c: (post_missing_element_message),
26315 (gen_video_element), (gen_text_element), (gen_audio_element),
26317 Post missing-plugin messages also when we error out because
26318 converters, textoverlay or auto*sinks are missing (#161922).
26320 2007-01-10 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
26322 gst/playback/: Fix the case where we try to ref a NULL element when we delay a link because of unfixed caps.
26323 Original commit message from CVS:
26324 * gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
26325 (is_demuxer_element), (new_caps):
26326 * gst/playback/gstplaybasebin.c: (source_new_pad):
26327 Fix the case where we try to ref a NULL element when we delay a link
26328 because of unfixed caps.
26329 Set the state of autoplugged decodebins to PAUSED.
26330 RTSP now works in playbin, we can remove it from the blacklist.
26332 2007-01-09 14:33:24 +0000 Tim-Philipp Müller <tim@centricular.net>
26334 gst/playback/: Post missing-plugin messages on the bus for missing sources and missing decoders/demuxers/depayloaders...
26335 Original commit message from CVS:
26336 * gst/playback/Makefile.am:
26337 * gst/playback/gstplaybasebin.c: (string_arr_has_str),
26338 (unknown_type), (setup_subtitle), (gen_source_element):
26339 * gst/playback/gstplaybin.c: (plugin_init):
26340 Post missing-plugin messages on the bus for missing sources and
26341 missing decoders/demuxers/depayloaders; fix error code used when
26342 we're missing an URI handler source; for media types that we are not
26343 handling on purpose at the moment, don't print "don't know how to
26344 handle xyz" messages to the terminal or post missing-plugin
26345 messages on the bus.
26346 * tests/check/elements/playbin.c: (create_playbin),
26347 (GST_START_TEST), (gst_codec_src_uri_get_type),
26348 (gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
26349 (gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
26350 (gst_codec_src_init_type), (gst_codec_src_base_init),
26351 (gst_codec_src_create), (gst_codec_src_class_init),
26352 (gst_codec_src_init), (plugin_init), (playbin_suite):
26353 Add some tests for the missing-plugin stuff.
26355 2007-01-09 14:20:08 +0000 Tim-Philipp Müller <tim@centricular.net>
26357 API: add new libgstbaseutils library with functions
26358 Original commit message from CVS:
26360 * gst-libs/gst/Makefile.am:
26361 * gst-libs/gst/utils/Makefile.am:
26362 * gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
26363 * gst-libs/gst/utils/base-utils.h:
26364 * gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
26365 (find_format_info), (caps_are_rtp_caps),
26366 (gst_base_utils_get_source_description),
26367 (gst_base_utils_get_sink_description),
26368 (gst_base_utils_get_decoder_description),
26369 (gst_base_utils_get_encoder_description),
26370 (gst_base_utils_get_element_description),
26371 (gst_base_utils_add_codec_description_to_tag_list),
26372 (gst_base_utils_get_codec_description), (gst_base_utils_list_all):
26373 * gst-libs/gst/utils/descriptions.h:
26374 * gst-libs/gst/utils/missing-plugins.c:
26375 (missing_structure_get_type), (copy_and_clean_caps),
26376 (gst_missing_uri_source_message_new),
26377 (gst_missing_uri_sink_message_new),
26378 (gst_missing_element_message_new),
26379 (gst_missing_decoder_message_new),
26380 (gst_missing_encoder_message_new),
26381 (missing_structure_get_string_detail),
26382 (missing_structure_get_caps_detail),
26383 (gst_missing_plugin_message_get_installer_detail),
26384 (gst_missing_plugin_message_get_description),
26385 (gst_is_missing_plugin_message):
26386 * gst-libs/gst/utils/missing-plugins.h:
26387 API: add new libgstbaseutils library with functions
26388 - to create and parse missing-plugins messages
26389 - that provide (translated) descriptions for caps/decoders/sources/etc.
26391 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
26392 * pkgconfig/gstreamer-plugins-base.pc.in:
26394 * docs/libs/gst-plugins-base-libs-docs.sgml:
26395 * docs/libs/gst-plugins-base-libs-sections.txt:
26396 Generate docs for new lib and API.
26397 * tests/check/Makefile.am:
26398 * tests/check/libs/.cvsignore:
26399 * tests/check/libs/utils.c: (missing_msg_check_getters),
26400 (GST_START_TEST), (libgstbaseutils_suite):
26401 Add some basic unit tests.
26403 2007-01-09 13:35:08 +0000 Tim-Philipp Müller <tim@centricular.net>
26405 ext/ogg/Makefile.am: Dist gstoggdemux.h to fix 'make distcheck'.
26406 Original commit message from CVS:
26407 * ext/ogg/Makefile.am:
26408 Dist gstoggdemux.h to fix 'make distcheck'.
26409 * sys/v4l/Makefile.am:
26410 Fix 'make distcheck' even more.
26412 2007-01-09 12:30:46 +0000 Wim Taymans <wim.taymans@gmail.com>
26415 Original commit message from CVS:
26416 * docs/plugins/Makefile.am:
26417 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
26418 * docs/plugins/gst-plugins-base-plugins-sections.txt:
26419 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
26420 (gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
26421 (gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
26422 (gst_ogg_demux_perform_seek):
26423 * ext/ogg/gstoggdemux.h:
26425 Add some more comments.
26428 2007-01-09 11:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
26430 Small documentation updates/fixes
26431 Original commit message from CVS:
26432 * ext/theora/theoradec.c:
26433 * ext/vorbis/vorbisdec.c:
26434 * gst-libs/gst/audio/gstringbuffer.c:
26435 (gst_ring_buffer_commit_full):
26436 * gst-libs/gst/audio/gstringbuffer.h:
26437 * gst-libs/gst/rtp/gstrtpbuffer.c:
26438 * gst-libs/gst/tag/gstvorbistag.c:
26439 Small documentation updates/fixes
26441 2007-01-09 10:37:01 +0000 Tim-Philipp Müller <tim@centricular.net>
26443 configure.ac: Require core CVS HEAD for Andy's basesrc/sink API additions.
26444 Original commit message from CVS:
26446 Require core CVS HEAD for Andy's basesrc/sink API additions.
26448 2007-01-08 14:01:23 +0000 Günter Thelen <daedalus.inc@gmx.net>
26450 gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
26451 Original commit message from CVS:
26452 Patch by: Günter Thelen <daedalus dot inc at gmx net>
26453 * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
26455 Add typefinder for flac-in-ogg in conformance with the ogg-mapping
26456 on flac.sf.net (there appear to be other versions of the first
26457 ogg page in the wild) (#391365).
26459 2007-01-08 13:32:32 +0000 Tim-Philipp Müller <tim@centricular.net>
26461 configure.ac: Check if localtime_r() is available.
26462 Original commit message from CVS:
26464 Check if localtime_r() is available.
26465 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
26466 If localtime_r() is not available, fall back to localtime(). Should
26467 fix build on MingW (#393310).
26469 2007-01-08 12:30:03 +0000 Tim-Philipp Müller <tim@centricular.net>
26471 gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
26472 Original commit message from CVS:
26473 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
26474 * gst/subparse/gstsubparse.h:
26475 Remove spurious 1000 subtrahend when calculating the timestamp from
26476 the frame number and the frame rate . Also, use the frames/second
26477 value specified in the first line of the file, if one is specified
26478 there. Should fix #357503.
26479 * tests/check/elements/subparse.c: (do_test),
26480 (test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
26482 Add some basic unit tests for the microdvd subtitle format.
26484 2007-01-07 21:53:38 +0000 Young-Ho Cha <ganadist@chollian.net>
26486 sys/xvimage/xvimagesink.c: Fixes : #390076.
26487 Original commit message from CVS:
26488 2007-01-07 Julien MOUTTE <julien@moutte.net>
26489 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
26490 (gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
26491 (gst_xvimagesink_xvimage_put),
26492 (gst_lookup_xv_port_from_adaptor),
26493 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
26494 (gst_xvimagesink_set_xwindow_id),
26495 (gst_xvimagesink_set_event_handling),
26496 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
26497 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
26498 Patch by : Young-Ho Cha <ganadist at chollian dot net>
26500 Add an adaptor property to select a specific XV adaptor.
26501 * sys/xvimage/xvimagesink.h:
26503 2007-01-07 18:50:13 +0000 Julien Moutte <julien@moutte.net>
26505 sys/: Use flow_lock much more to protect every access to xwindow.
26506 Original commit message from CVS:
26507 2007-01-07 Julien MOUTTE <julien@moutte.net>
26508 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
26509 (gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
26510 (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
26511 (gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
26512 (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
26513 (gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
26514 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
26515 (gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
26516 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
26517 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
26518 (gst_xvimagesink_change_state),
26519 (gst_xvimagesink_set_xwindow_id),
26520 (gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
26521 Use flow_lock much more to protect every access to xwindow.
26522 Try to catch erros while creating images in case some drivers
26524 just generating an XError when the requested image is too big.
26525 Should fix : #354698, #384008, #384060.
26526 * tests/icles/stress-xoverlay.c: (cycle_window),
26528 Implement some stress testing of setting window xid.
26530 2007-01-07 10:33:55 +0000 Sébastien Moutte <sebastien@moutte.net>
26532 win32/common/libgsaudio.def: Add new exported function.
26533 Original commit message from CVS:
26534 * win32/common/libgsaudio.def:
26535 Add new exported function.
26536 * win32/common/libgstogg.dsp:
26537 Add gstoggaviparse.c to the build.
26538 * win32/common/libgstvideoscale.dsp:
26539 Add vs_4tap.c to the build.
26540 * win32/common/libgstvorbis.dsp:
26541 Add vorbistag.c to the build.
26543 2007-01-06 17:28:40 +0000 Andy Wingo <wingo@pobox.com>
26546 * gst-libs/gst/audio/gstbaseaudiosink.c:
26547 gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
26548 Original commit message from CVS:
26549 2007-01-06 Andy Wingo <wingo@pobox.com>
26550 * gst-libs/gst/audio/gstbaseaudiosink.c
26551 (gst_base_audio_sink_class_init)
26552 (gst_base_audio_sink_init):
26553 (gst_base_audio_sink_activate_pull): Add an activate_pull function
26554 to baseaudiosink, and tell basesink that we can work in pull mode.
26555 This way the ring buffer thread drives the pipeline directly, if
26556 pull mode is possible. There is some lingering nastiness regarding
26558 (gst_base_audio_sink_callback): Implement the callback to pull
26559 data. This interface is a bit light, though -- it should get a
26560 GstFlowReturn return value at least.
26562 2007-01-05 19:43:55 +0000 Tim-Philipp Müller <tim@centricular.net>
26564 Printf format and missing argument fixes.
26565 Original commit message from CVS:
26566 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
26567 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
26568 * gst/playback/gstdecodebin2.c:
26569 (gst_decode_group_check_if_blocked):
26570 Printf format and missing argument fixes.
26572 2007-01-05 18:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26574 ext/ogg/gstogmparse.c: Activate pads before adding them to the element.
26575 Original commit message from CVS:
26576 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
26577 (gst_ogm_parse_change_state):
26578 Activate pads before adding them to the element.
26580 2007-01-05 16:02:50 +0000 Tim-Philipp Müller <tim@centricular.net>
26582 tests/examples/seek/: Call g_thread_init() first thing in main() (see #391278).
26583 Original commit message from CVS:
26584 * tests/examples/seek/scrubby.c: (main):
26585 * tests/examples/seek/seek.c: (main):
26586 Call g_thread_init() first thing in main() (see #391278).
26588 2007-01-05 12:19:34 +0000 Tim-Philipp Müller <tim@centricular.net>
26590 tests/check/: Add test for GstNetBuffer + gst_buffer_copy(). Disabled for the time being, since it's broken, see #393...
26591 Original commit message from CVS:
26592 * tests/check/Makefile.am:
26593 * tests/check/libs/.cvsignore:
26594 * tests/check/libs/netbuffer.c: (GST_START_TEST),
26596 Add test for GstNetBuffer + gst_buffer_copy(). Disabled
26597 for the time being, since it's broken, see #393099.
26599 2007-01-05 12:13:24 +0000 Tim-Philipp Müller <tim@centricular.net>
26601 tests/check/Makefile.am: Update to use GST_PLUGINS_BASE_CFLAGS as well.
26602 Original commit message from CVS:
26603 * tests/check/Makefile.am:
26604 Update to use GST_PLUGINS_BASE_CFLAGS as well.
26606 2007-01-04 12:49:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26608 configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
26609 Original commit message from CVS:
26611 split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
26612 so that GST_BASE_CFLAGS can go inbetween them, making sure
26613 we use uninstalled gst-libs headers
26614 * docs/libs/Makefile.am:
26615 * ext/alsa/Makefile.am:
26616 * ext/cdparanoia/Makefile.am:
26617 * ext/gnomevfs/Makefile.am:
26618 * ext/libvisual/Makefile.am:
26619 * ext/ogg/Makefile.am:
26620 * ext/theora/Makefile.am:
26621 * ext/vorbis/Makefile.am:
26622 * gst-libs/gst/audio/Makefile.am:
26623 * gst-libs/gst/cdda/Makefile.am:
26624 * gst-libs/gst/interfaces/Makefile.am:
26625 * gst-libs/gst/riff/Makefile.am:
26626 * gst-libs/gst/rtp/Makefile.am:
26627 * gst-libs/gst/tag/Makefile.am:
26628 * gst/adder/Makefile.am:
26629 * gst/audioconvert/Makefile.am:
26630 * gst/audiorate/Makefile.am:
26631 * gst/audioresample/Makefile.am:
26632 * gst/playback/Makefile.am:
26633 * gst/tcp/Makefile.am:
26634 * gst/videoscale/Makefile.am:
26635 * gst/volume/Makefile.am:
26636 * sys/ximage/Makefile.am:
26637 * sys/xvimage/Makefile.am:
26638 * tests/icles/Makefile.am:
26641 2007-01-04 11:30:53 +0000 Julien Moutte <julien@moutte.net>
26643 Add a method to the XOverlay interface to allow disabling of event handling in x[v]imagesink elements. This will let ...
26644 Original commit message from CVS:
26645 2007-01-04 Julien MOUTTE <julien@moutte.net>
26646 * gst-libs/gst/interfaces/xoverlay.c:
26647 (gst_x_overlay_handle_events):
26648 * gst-libs/gst/interfaces/xoverlay.h:
26649 * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
26650 (gst_ximagesink_set_xwindow_id),
26651 (gst_ximagesink_set_event_handling),
26652 (gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
26653 (gst_ximagesink_get_property), (gst_ximagesink_init),
26654 (gst_ximagesink_class_init):
26655 * sys/ximage/ximagesink.h:
26656 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
26657 (gst_xvimagesink_set_xwindow_id),
26658 (gst_xvimagesink_set_event_handling),
26659 (gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
26660 (gst_xvimagesink_get_property), (gst_xvimagesink_init),
26661 (gst_xvimagesink_class_init):
26662 * sys/xvimage/xvimagesink.h:
26663 * tests/icles/stress-xoverlay.c: (toggle_events),
26665 Add a method to the XOverlay interface to allow disabling of
26666 event handling in x[v]imagesink elements. This will let X events
26667 propagate to parent windows which can be usefull in some cases.
26668 Be carefull that the application is then responsible of pushing
26669 navigation events and expose events to the video sink.
26672 2007-01-03 15:45:06 +0000 Tim-Philipp Müller <tim@centricular.net>
26674 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION (fixes #392070).
26675 Original commit message from CVS:
26676 * gst-libs/gst/tag/gstvorbistag.c:
26677 * tests/check/libs/tag.c: (GST_START_TEST):
26678 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
26681 2007-01-01 18:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
26684 Original commit message from CVS:
26686 * docs/Makefile.am:
26687 * docs/design/Makefile.am:
26690 2006-12-27 17:15:35 +0000 Julien Moutte <julien@moutte.net>
26692 docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation typo. Fixes: #390063.
26693 Original commit message from CVS:
26694 2006-12-27 Julien MOUTTE <julien@moutte.net>
26695 * docs/libs/gst-plugins-base-libs-sections.txt: Fix a
26697 typo. Fixes: #390063.
26699 2006-12-27 12:08:13 +0000 Julien Moutte <julien@moutte.net>
26701 sys/: Plug a caps leak.
26702 Original commit message from CVS:
26703 2006-12-27 Julien MOUTTE <julien@moutte.net>
26704 * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
26705 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
26707 * win32/common/config.h: Updated.
26709 2006-12-22 12:10:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26711 tests/check/elements/: Fix the dp tests, but activating the pads for the streamheader tests too and cleaning up condi...
26712 Original commit message from CVS:
26713 * tests/check/elements/gdpdepay.c: (cleanup_gdpdepay),
26714 (setup_gdpdepay_streamheader):
26715 * tests/check/elements/gdppay.c: (cleanup_gdppay),
26716 (setup_gdppay_streamheader):
26717 Fix the dp tests, but activating the pads for the streamheader tests
26718 too and cleaning up conditionaly
26720 2006-12-22 11:09:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26722 gst/ffmpegcolorspace/: Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the other end of the wo...
26723 Original commit message from CVS:
26724 * gst/ffmpegcolorspace/avcodec.h:
26725 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
26726 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
26727 (gst_ffmpegcsp_avpicture_fill):
26728 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
26729 (img_get_alpha_info):
26730 Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
26731 other end of the word. Fixes: #387073.
26732 Add some inconsequential branch hints in a couple of places.
26734 2006-12-21 12:30:11 +0000 Tim-Philipp Müller <tim@centricular.net>
26736 gst/ffmpegcolorspace/gstffmpegcodecmap.c: The "signed" field in raw audio caps is of boolean type, trying to extract ...
26737 Original commit message from CVS:
26738 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
26739 (gst_ffmpeg_caps_to_smpfmt):
26740 The "signed" field in raw audio caps is of boolean type, trying to
26741 extract the value with _get_int() will fail (fix to keep in sync with
26742 the copy in gst-ffmpeg)
26744 2006-12-21 08:12:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26746 tests/check/elements/: consistent pad (de)activation
26747 Original commit message from CVS:
26748 * tests/check/elements/audioresample.c: (cleanup_audioresample):
26749 * tests/check/elements/audiotestsrc.c: (cleanup_audiotestsrc):
26750 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
26751 (cleanup_gdpdepay):
26752 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay):
26753 * tests/check/elements/subparse.c: (teardown_subparse):
26754 * tests/check/elements/textoverlay.c: (cleanup_textoverlay):
26755 * tests/check/elements/videorate.c: (cleanup_videorate):
26756 * tests/check/elements/videotestsrc.c: (cleanup_videotestsrc):
26757 * tests/check/elements/volume.c: (cleanup_volume):
26758 * tests/check/elements/vorbisdec.c: (setup_vorbisdec),
26759 (cleanup_vorbisdec):
26760 * tests/check/elements/vorbistag.c: (setup_vorbistag),
26761 (cleanup_vorbistag):
26762 consistent pad (de)activation
26764 2006-12-20 10:29:58 +0000 Tim-Philipp Müller <tim@centricular.net>
26766 gst/typefind/gsttypefindfunctions.c: Forgot to register the extensions.
26767 Original commit message from CVS:
26768 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
26769 Forgot to register the extensions.
26771 2006-12-20 09:25:55 +0000 Tim-Philipp Müller <tim@centricular.net>
26773 gst/typefind/gsttypefindfunctions.c: Add typefinder for VIVO files (my christmas present to the 90s).
26774 Original commit message from CVS:
26775 * gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
26777 Add typefinder for VIVO files (my christmas present to the 90s).
26779 2006-12-16 13:59:09 +0000 Tim-Philipp Müller <tim@centricular.net>
26781 gst/playback/gstdecodebin.c: Special-case the text/plain media type: we only want to recognise it as a 'raw' decoded ...
26782 Original commit message from CVS:
26783 * gst/playback/gstdecodebin.c: (type_found):
26784 Special-case the text/plain media type: we only want to recognise it
26785 as a 'raw' decoded media type if it comes from a demuxer or subtitle
26786 parser, but not if the entire stream is of text/plain type. If the
26787 entire stream is text/plain, we should just error out.
26788 This fixes playback of audio files with lyrics in totem. Totem can't
26789 distinguish between text files and subtitle files and passes any
26790 .txt file with the same basename as the main file to playbin as
26791 suburi, and playbin will then throw a 'subtitle found, but no video
26792 stream' error, which isn't entirely helpful. See #380342.
26793 Also, with this change we'll show a slightly more correct error
26794 message in case totem passes a playlist file to us (although a
26795 custom error message wording instead of the default text would
26796 probably not be a bad idea either).
26797 Same problem also needs to be fixed for playbin+decodebin2.
26798 * tests/check/Makefile.am:
26799 * tests/check/elements/decodebin.c: (src_handoff_cb),
26800 (decodebin_new_decoded_pad_cb), (GST_START_TEST),
26802 Add simple unit test for decodebin for the above.
26804 2006-12-16 12:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
26806 gst/playback/: Refuse to change state to READY when we failed to create any of the required elements in our instance ...
26807 Original commit message from CVS:
26808 * gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
26809 * gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
26810 Refuse to change state to READY when we failed to create any of the
26811 required elements in our instance init function.
26813 2006-12-15 10:52:23 +0000 Tim-Philipp Müller <tim@centricular.net>
26815 docs/libs/gst-plugins-base-libs-sections.txt: Small docs fixes/updates.
26816 Original commit message from CVS:
26817 * docs/libs/gst-plugins-base-libs-sections.txt:
26818 Small docs fixes/updates.
26819 * gst-libs/gst/video/gstvideosink.h:
26820 Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
26821 from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
26822 removed from the base sink API between 0.9.6 and 0.9.7).
26823 API: add GST_VIDEO_SINK_CAST and use it for the height/width
26824 accessor macros, so we don't do a runtime GObject type check every
26827 2006-12-15 00:20:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26830 Original commit message from CVS:
26832 * gst-plugins-base.doap:
26833 * gst-plugins-base.spec.in:
26836 2006-12-09 15:12:38 +0000 Jens Granseuer <jensgr@gmx.net>
26838 Declare variables at the beginning of a block. Fixes #383195.
26839 Original commit message from CVS:
26840 Patch by: Jens Granseuer <jensgr at gmx net>
26841 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
26842 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
26843 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
26844 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
26845 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
26846 Declare variables at the beginning of a block. Fixes #383195.
26848 2006-12-07 02:38:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26850 configure.ac: Bump version nano - back to CVS.
26851 Original commit message from CVS:
26853 Bump version nano - back to CVS.
26855 === release 0.10.11 ===
26857 2006-12-07 02:30:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26859 configure.ac: releasing 0.10.11, "Dumb things"
26860 Original commit message from CVS:
26861 === release 0.10.11 ===
26862 2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
26864 releasing 0.10.11, "Dumb things"
26866 2006-12-05 12:44:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26868 gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
26869 Original commit message from CVS:
26870 * gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
26871 (close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
26872 Handle the case where an element has multiple pads with
26873 unfixed caps as well as still possibly producing more dynamic
26874 pads by storing each case as a distinct entry in the dynamic list.
26875 Fixes #38223 again.
26877 2006-12-04 13:02:37 +0000 Wim Taymans <wim.taymans@gmail.com>
26879 gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
26880 Original commit message from CVS:
26881 * gst/playback/gstdecodebin.c: (close_pad_link):
26882 Fix #382223, add more dynamic caps handling.
26884 2006-12-01 11:35:57 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
26887 Ignore all pot files
26888 Original commit message from CVS:
26889 Ignore all pot files
26891 2006-12-01 10:36:50 +0000 Michael Smith <msmith@xiph.org>
26893 gst/audiorate/gstaudiorate.c: Delete bad debug code.
26894 Original commit message from CVS:
26895 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
26896 Delete bad debug code.
26899 2006-12-01 10:27:54 +0000 Sergey Scobich <sergey.scobich@gmail.com>
26901 Fix compilation on win32 under VS8
26902 Original commit message from CVS:
26903 * gst/videoscale/vs_4tap.c:
26905 * win32/common/config.h:
26906 * win32/vs8/libgstvideoscale.vcproj:
26907 Fix compilation on win32 under VS8
26908 Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
26909 Partially fixes #381175
26911 2006-11-30 23:46:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26928 Original commit message from CVS:
26931 2006-11-30 12:50:42 +0000 Michael Smith <msmith@xiph.org>
26933 tests/check/pipelines/theoraenc.c: It would be very bad if, after a discont buffer, we thought every single following...
26934 Original commit message from CVS:
26935 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
26937 It would be very bad if, after a discont buffer, we thought every
26938 single following buffer was also discont. So, add to the test to
26939 ensure that this isn't the case.
26940 * ext/theora/theoraenc.c: (theora_enc_is_discontinuous):
26941 ... it was the case. So fix it.
26943 2006-11-28 16:43:18 +0000 Wim Taymans <wim.taymans@gmail.com>
26945 gst/playback/gstplaybasebin.c: Improve debug.
26946 Original commit message from CVS:
26947 * gst/playback/gstplaybasebin.c: (check_queue_event):
26949 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
26950 Fix width and height range from 16 - 4096 to 1 - MAXINT, just like the
26951 padtemplate caps. Refixes #357577.
26953 2006-11-28 16:21:27 +0000 Wim Taymans <wim.taymans@gmail.com>
26955 gst/playback/gstplaybasebin.c: Add event probe to see when EOS is in a queue and we can disable the underrun signals....
26956 Original commit message from CVS:
26957 * gst/playback/gstplaybasebin.c: (check_queue_event),
26958 (queue_threshold_reached), (queue_out_of_data),
26959 (gen_preroll_element):
26960 Add event probe to see when EOS is in a queue and we can disable the
26961 underrun signals. Fixes #357577.
26963 2006-11-28 14:40:39 +0000 Edward Hervey <bilboed@bilboed.com>
26965 gst/playback/: New decodebin2 element.
26966 Original commit message from CVS:
26967 * gst/playback/Makefile.am:
26968 * gst/playback/gstdecodebin2.c: (gst_decode_bin_get_type),
26969 (_gst_boolean_accumulator), (gst_decode_bin_class_init),
26970 (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
26971 (gst_decode_bin_init), (gst_decode_bin_dispose),
26972 (gst_decode_bin_finalize), (gst_decode_bin_set_property),
26973 (gst_decode_bin_get_property), (gst_decode_bin_set_caps),
26974 (gst_decode_bin_get_caps), (gst_decode_bin_autoplug_continue),
26975 (gst_decode_bin_autoplug_sort), (analyze_new_pad), (connect_pad),
26976 (connect_element), (expose_pad), (type_found),
26977 (pad_added_group_cb), (pad_removed_group_cb),
26978 (no_more_pads_group_cb), (pad_added_cb), (pad_removed_cb),
26979 (no_more_pads_cb), (find_compatibles), (is_demuxer_element),
26980 (are_raw_caps), (multi_queue_overrun_cb),
26981 (multi_queue_underrun_cb), (gst_decode_group_new),
26982 (get_current_group), (group_demuxer_event_probe),
26983 (gst_decode_group_control_demuxer_pad),
26984 (gst_decode_group_control_source_pad),
26985 (gst_decode_group_check_if_blocked),
26986 (gst_decode_group_check_if_drained), (gst_decode_group_expose),
26987 (gst_decode_group_hide), (gst_decode_group_free),
26988 (gst_decode_group_set_complete), (source_pad_blocked_cb),
26989 (source_pad_event_probe), (gst_decode_pad_new), (add_fakesink),
26990 (remove_fakesink), (find_sink_pad), (gst_decode_bin_change_state),
26992 New decodebin2 element.
26994 * gst/playback/gstplay-marshal.list:
26995 Added marshallers for new signals in decodebin2
26996 * gst/playback/gstplaybasebin.c: (setup_subtitle), (make_decoder):
26997 Use decodebin2 if *and only if* the USE_DECODEBIN2 environment variable
27000 2006-11-28 10:45:40 +0000 Wim Taymans <wim.taymans@gmail.com>
27002 gst/playback/gstplaybasebin.c: Disable rtsp:// uris for the release, it's not good enough yet.
27003 Original commit message from CVS:
27004 * gst/playback/gstplaybasebin.c: (setup_source),
27005 (gst_play_base_bin_change_state):
27006 Disable rtsp:// uris for the release, it's not good enough yet.
27009 2006-11-26 16:39:41 +0000 Wim Taymans <wim.taymans@gmail.com>
27011 ext/theora/theoradec.c: Implement reverse playback.
27012 Original commit message from CVS:
27013 * ext/theora/theoradec.c: (gst_theora_dec_reset),
27014 (theora_dec_push_forward), (theora_dec_push_reverse),
27015 (theora_handle_data_packet), (theora_dec_decode_buffer),
27016 (theora_dec_flush_decode), (theora_dec_chain_reverse),
27017 (theora_dec_chain_forward), (theora_dec_chain):
27018 Implement reverse playback.
27019 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
27020 (vorbis_dec_decode_buffer), (vorbis_dec_flush_decode),
27021 (vorbis_dec_chain_forward):
27022 Clear buffers used for reverse playback in _reset.
27023 No need to set the eos flag, we clip samples using the segment.
27025 2006-11-24 15:40:58 +0000 Wim Taymans <wim.taymans@gmail.com>
27027 ext/ogg/gstoggdemux.c: Some cleanups.
27028 Original commit message from CVS:
27029 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
27030 (gst_ogg_pad_init), (gst_ogg_pad_dispose), (gst_ogg_pad_reset),
27031 (gst_ogg_pad_stream_out), (gst_ogg_pad_submit_page),
27032 (gst_ogg_chain_reset), (gst_ogg_demux_perform_seek):
27034 Handle continued pages in reverse mode.
27036 2006-11-24 15:39:03 +0000 Wim Taymans <wim.taymans@gmail.com>
27038 ext/vorbis/vorbisdec.c: Small cleanups.
27039 Original commit message from CVS:
27040 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
27041 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
27042 (vorbis_dec_flush_decode):
27044 Don't try to add invalid timestamps.
27045 Clipping will unref the buffer.
27047 2006-11-24 08:56:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27049 gst/: remove obsolete _factory_init protos
27050 Original commit message from CVS:
27051 * gst/adder/gstadder.h:
27052 * gst/audiotestsrc/gstaudiotestsrc.h:
27053 remove obsolete _factory_init protos
27055 2006-11-24 08:35:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27057 sys/xvimage/xvimagesink.c: Fix spacing in debug message.
27058 Original commit message from CVS:
27059 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
27060 Fix spacing in debug message.
27062 2006-11-23 11:07:23 +0000 Wim Taymans <wim.taymans@gmail.com>
27064 ext/ogg/gstoggdemux.c: Don't just ignore return values from _pad_push().
27065 Original commit message from CVS:
27066 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
27067 (gst_ogg_demux_chain):
27068 Don't just ignore return values from _pad_push().
27069 Small debug improvements.
27071 2006-11-23 11:02:11 +0000 Michael Smith <msmith@xiph.org>
27073 ext/ogg/gstoggmux.c: If our incoming buffer is marked as DISCONT, then increment the page number (so that the discont...
27074 Original commit message from CVS:
27075 * ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
27076 If our incoming buffer is marked as DISCONT, then increment the page
27077 number (so that the discontinuity is marked in the final ogg
27078 bitstream) and flush the previous page.
27080 2006-11-22 14:34:03 +0000 Michael Smith <msmith@xiph.org>
27082 ext/theora/: Mark discontinuities of > 3/4 of a frame, reinit encoder.
27083 Original commit message from CVS:
27084 * ext/theora/gsttheoraenc.h:
27085 * ext/theora/theoraenc.c: (gst_theora_enc_init),
27086 (theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
27087 (theora_buffer_from_packet), (theora_enc_is_discontinuous),
27088 (theora_enc_chain), (theora_enc_change_state):
27089 Mark discontinuities of > 3/4 of a frame, reinit encoder.
27090 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
27091 (GST_START_TEST), (theoraenc_suite):
27092 Enable discontinuity test, fix it.
27094 2006-11-21 18:39:34 +0000 Tim-Philipp Müller <tim@centricular.net>
27096 ext/pango/gsttextoverlay.*: Some textoverlay fixes: for one, in the video chain function, actually wait for a text bu...
27097 Original commit message from CVS:
27098 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
27099 (gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
27100 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
27101 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
27102 (gst_text_overlay_change_state):
27103 * ext/pango/gsttextoverlay.h:
27104 Some textoverlay fixes: for one, in the video chain function,
27105 actually wait for a text buffer to come in if there is none at the
27106 moment and there should be one; also, deal more gracefully with
27107 incoming buffers that do not have a timestamp or duration; discard
27108 text buffer when not needed any longer. Fixes #341681.
27109 * tests/check/Makefile.am:
27110 * tests/check/elements/.cvsignore:
27111 * tests/check/elements/textoverlay.c:
27112 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
27113 (setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
27114 (create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
27115 (test_video_waits_for_text_send_text_newsegment_thread),
27116 (test_video_waits_for_text_shutdown_element),
27117 (test_render_continuity_push_video_buffers_thread),
27118 (textoverlay_suite):
27119 Add some unit tests for textoverlay.
27121 2006-11-21 09:29:56 +0000 Tim-Philipp Müller <tim@centricular.net>
27123 gst/typefind/gsttypefindfunctions.c: Avoid integer underflow when the found probability for mp3 is smaller than the '...
27124 Original commit message from CVS:
27125 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
27126 Avoid integer underflow when the found probability for mp3 is
27127 smaller than the 'penalty' we subtract if there's not a clean
27128 mp3 header sync at offset 0.
27130 2006-11-21 08:17:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27132 docs/libs/gst-plugins-base-libs-sections.txt: Add some new symbols to the docs
27133 Original commit message from CVS:
27134 * docs/libs/gst-plugins-base-libs-sections.txt:
27135 Add some new symbols to the docs
27137 2006-11-20 16:44:28 +0000 Tim-Philipp Müller <tim@centricular.net>
27139 tests/check/: Enable ffmpegcolorspace test now that the RGBA32 issue is fixed (for now not for valgrinding though, si...
27140 Original commit message from CVS:
27141 * tests/check/Makefile.am:
27142 * tests/check/elements/ffmpegcolorspace.c:
27143 (ffmpegcolorspace_suite):
27144 Enable ffmpegcolorspace test now that the RGBA32 issue is fixed
27145 (for now not for valgrinding though, since it takes too long).
27147 2006-11-20 15:01:09 +0000 Wim Taymans <wim.taymans@gmail.com>
27149 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix RGBA32 caps. Fixes #357038.
27150 Original commit message from CVS:
27151 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
27152 (gst_ffmpeg_pixfmt_to_caps):
27153 Fix RGBA32 caps. Fixes #357038.
27155 2006-11-20 12:20:39 +0000 Tim-Philipp Müller <tim@centricular.net>
27157 gst-libs/gst/interfaces/mixertrack.h: Add FIXME so we can add some padding here in 0.11
27158 Original commit message from CVS:
27159 * gst-libs/gst/interfaces/mixertrack.h:
27160 Add FIXME so we can add some padding here in 0.11
27162 2006-11-19 17:07:34 +0000 Tim-Philipp Müller <tim@centricular.net>
27164 gst-libs/gst/rtp/gstbasertpaudiopayload.h: Fix GstBaseRTPAudioPayload structure so the whole GObject inheritance busi...
27165 Original commit message from CVS:
27166 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
27167 Fix GstBaseRTPAudioPayload structure so the whole GObject
27168 inheritance business actually works (parent class instance structure
27169 must always come first in the derived class instance structure).
27171 2006-11-16 14:35:30 +0000 Tim-Philipp Müller <tim@centricular.net>
27173 Make sure our checks and the videotestsrc plugin link against the local uninstalled gst libs and not any installed gs...
27174 Original commit message from CVS:
27175 * gst/videotestsrc/Makefile.am:
27176 * tests/check/Makefile.am:
27177 Make sure our checks and the videotestsrc plugin link against the
27178 local uninstalled gst libs and not any installed gst libs that
27179 might happen to exist as well.
27180 * tests/check/elements/adder.c: (message_received),
27181 (test_event_message_received), (test_play_twice_message_received):
27182 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
27183 Fix compiler warnings when compiling against core with disabled
27186 2006-11-16 12:55:08 +0000 Michael Smith <msmith@xiph.org>
27188 gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
27189 Original commit message from CVS:
27190 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
27191 (gst_audio_rate_sink_event), (gst_audio_rate_chain):
27192 Fix audiorate, so that it accurately sets offsets and timestamps.
27193 Doesn't change the fundamental algorithmic decisions; so should be
27195 * tests/check/Makefile.am:
27196 Enable audiorate test now that it passes.
27198 2006-11-15 10:05:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27200 sys/xvimage/xvimagesink.c: clear xv when going to NULL, remove // commented non-existant proto
27201 Original commit message from CVS:
27202 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
27203 clear xv when going to NULL, remove // commented non-existant proto
27204 * tests/examples/seek/seek.c: (main):
27205 add missing tooltip description for scrub and play_scrub
27207 2006-11-14 23:34:19 +0000 David Schleef <ds@schleef.org>
27209 configure.ac: Bump liboil requirement to 0.3.8.
27210 Original commit message from CVS:
27212 Bump liboil requirement to 0.3.8.
27213 * gst-libs/gst/riff/riff-media.c:
27215 * gst/videoscale/vs_image.h:
27216 * gst/videoscale/vs_scanline.h:
27217 Use liboil's stdint.h.
27218 * gst/videotestsrc/videotestsrc.c:
27219 Remove liboil related ifdef's, since they aren't needed now, and
27220 won't work with future versions.
27222 2006-11-14 23:08:38 +0000 David Schleef <ds@schleef.org>
27224 gst/videoscale/: Add a 4-tap image scaler. Theoretically looks much prettier.
27225 Original commit message from CVS:
27226 * gst/videoscale/Makefile.am:
27227 * gst/videoscale/gstvideoscale.c:
27228 * gst/videoscale/gstvideoscale.h:
27229 * gst/videoscale/vs_4tap.c:
27230 * gst/videoscale/vs_4tap.h:
27231 * gst/videoscale/vs_image.c:
27232 * gst/videoscale/vs_image.h:
27233 * gst/videoscale/vs_scanline.c:
27234 * gst/videoscale/vs_scanline.h:
27235 Add a 4-tap image scaler. Theoretically looks much prettier.
27236 The tap calculation could use some improvement.
27238 2006-11-14 11:54:14 +0000 Jan David Mol <j.j.d.mol@tudelft.nl>
27240 Various gsize and gssize printf fixes. Fixes #372507.
27241 Original commit message from CVS:
27242 Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
27243 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
27244 (gst_riff_parse_strf_iavs):
27245 * gst/subparse/gstsubparse.c: (convert_encoding):
27246 * gst/tcp/gstmultifdsink.c:
27247 (gst_multi_fd_sink_handle_client_write):
27248 * gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
27249 (gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
27250 (gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
27251 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
27252 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
27253 (gst_ximagesink_ximage_new):
27254 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
27255 Various gsize and gssize printf fixes. Fixes #372507.
27257 2006-11-13 18:14:48 +0000 Wim Taymans <wim.taymans@gmail.com>
27259 ext/vorbis/vorbisdec.*: First stab at vorbis reverse playback.
27260 Original commit message from CVS:
27261 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
27262 (vorbis_dec_push_forward), (vorbis_dec_push_reverse),
27263 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
27264 (vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
27265 (vorbis_dec_chain_forward), (vorbis_dec_chain):
27266 * ext/vorbis/vorbisdec.h:
27267 First stab at vorbis reverse playback.
27269 2006-11-13 17:30:17 +0000 Wim Taymans <wim.taymans@gmail.com>
27271 gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
27272 Original commit message from CVS:
27273 * gst-libs/gst/audio/gstbaseaudiosink.c:
27274 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
27275 * gst-libs/gst/audio/gstbaseaudiosink.h:
27276 Make the clock sync code more accurate wrt resampling and playback
27277 at different rates.
27278 * gst-libs/gst/audio/gstringbuffer.c:
27279 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
27280 * gst-libs/gst/audio/gstringbuffer.h:
27281 Use better algorithm to interpolate sample rates.
27283 2006-11-13 15:31:01 +0000 Michael Smith <msmith@xiph.org>
27285 ext/ogg/gstoggdemux.c: Improve a debug line slightly.
27286 Original commit message from CVS:
27287 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
27288 Improve a debug line slightly.
27289 * ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
27290 Call gst_riff_init() in plugin_init, to avoid getting errors from
27291 the debug system (unrelated changes to another plugin made this turn
27294 2006-11-10 19:20:21 +0000 Sergey Scobich <sergery.scobich@gmail.com>
27296 win32/common/libgsttag.def: Add missing symbol (#366492).
27297 Original commit message from CVS:
27298 Patch by: Sergey Scobich <sergery.scobich at gmail com>
27299 * win32/common/libgsttag.def:
27300 Add missing symbol (#366492).
27302 2006-11-10 00:52:55 +0000 Tim-Philipp Müller <tim@centricular.net>
27304 gst/playback/gststreamselector.c: Don't unref a NULL pad.
27305 Original commit message from CVS:
27306 * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
27307 Don't unref a NULL pad.
27309 2006-11-09 00:50:00 +0000 Christian Schaller <uraeus@gnome.org>
27311 ext/ogg/gstoggdemux.c: Implement first stab at reverse playback.
27312 Original commit message from CVS:
27313 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
27314 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_perform_seek),
27315 (gst_ogg_demux_handle_page), (gst_ogg_demux_chain),
27316 (gst_ogg_demux_loop_forward), (gst_ogg_demux_loop_reverse),
27317 (gst_ogg_demux_loop):
27318 Implement first stab at reverse playback.
27320 2006-11-07 07:22:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27322 gst-libs/gst/riff/riff-media.c: add h263/h264 variants to the caps, Fixes #363118
27323 Original commit message from CVS:
27324 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
27325 (gst_riff_create_video_template_caps):
27326 add h263/h264 variants to the caps, Fixes #363118
27328 2006-11-06 18:24:59 +0000 Tim-Philipp Müller <tim@centricular.net>
27330 gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
27331 Original commit message from CVS:
27332 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
27333 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
27334 Use g_strerror instead of strerror so we get UTF-8.
27336 2006-11-04 07:25:58 +0000 David Schleef <ds@schleef.org>
27338 ext/ogg/: Add/remove KW-DIRAC header here, since it is ogg-specific.
27339 Original commit message from CVS:
27340 * ext/ogg/gstoggdemux.c:
27341 * ext/ogg/gstoggmux.c:
27342 Add/remove KW-DIRAC header here, since it is ogg-specific.
27344 2006-11-03 15:44:31 +0000 Michael Smith <msmith@xiph.org>
27346 gst/typefind/gsttypefindfunctions.c: Recognise more mpeg4 elementary video streams.
27347 Original commit message from CVS:
27348 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
27349 Recognise more mpeg4 elementary video streams.
27351 2006-11-02 17:26:03 +0000 Edward Hervey <bilboed@bilboed.com>
27353 gst/typefind/gsttypefindfunctions.c: Lower the probability of mp3 typefinding functions if we don't find a valid mp3 ...
27354 Original commit message from CVS:
27355 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
27356 Lower the probability of mp3 typefinding functions if we don't find a
27357 valid mp3 header at the start of the file.
27360 2006-11-02 15:06:36 +0000 Wim Taymans <wim.taymans@gmail.com>
27362 ext/theora/: Document and partially implement an algorithm for doing reverse playback of theora video.
27363 Original commit message from CVS:
27364 * ext/theora/gsttheoradec.h:
27365 * ext/theora/theoradec.c: (gst_theora_dec_init),
27366 (theora_dec_sink_event), (theora_dec_chain_forward),
27367 (theora_dec_flush_decode), (theora_dec_chain_reverse),
27368 (theora_dec_chain):
27369 Document and partially implement an algorithm for doing reverse playback
27372 2006-11-02 14:18:45 +0000 Sergey Scobich <sergey.scobich@gmail.com>
27374 win32/: Misc. VS8 build fixes: fix syntax in config.h, add missing entries to libgsttag.def; add missing dependencies...
27375 Original commit message from CVS:
27376 Patch by: Sergey Scobich <sergey.scobich at gmail com>
27377 * win32/common/config.h:
27378 * win32/common/interfaces-enumtypes.c:
27379 * win32/common/libgsttag.def:
27380 * win32/vs8/gst-plugins-base.sln:
27381 * win32/vs8/libgstaudioresample.vcproj:
27382 * win32/vs8/libgstinterfaces.vcproj:
27383 * win32/vs8/libgstogg.vcproj:
27384 * win32/vs8/libgstriff.vcproj:
27385 * win32/vs8/libgsttag.vcproj:
27386 * win32/vs8/libgsttheora.vcproj:
27387 * win32/vs8/libgstvideoscale.vcproj:
27388 * win32/vs8/libgstvorbis.vcproj:
27389 Misc. VS8 build fixes: fix syntax in config.h, add missing entries
27390 to libgsttag.def; add missing dependencies for some vs8 projects;
27391 re-arrange placement of .def files in vs8 projects (#366334).
27393 2006-11-01 14:08:31 +0000 Tim-Philipp Müller <tim@centricular.net>
27395 ext/ogg/gstogg.c: Remove unused variable.
27396 Original commit message from CVS:
27397 * ext/ogg/gstogg.c:
27398 Remove unused variable.
27399 * ext/ogg/gstoggdemux.c:
27400 Fix Wim's surname in plugin description.
27402 2006-10-31 15:05:33 +0000 Wim Taymans <wim.taymans@gmail.com>
27404 gst-plugins-base.spec.in: spec new .h file. Fixes #368310.
27405 Original commit message from CVS:
27406 * gst-plugins-base.spec.in:
27407 spec new .h file. Fixes #368310.
27409 2006-10-31 14:19:07 +0000 Michael Smith <msmith@xiph.org>
27411 gst/tcp/gstmultifdsink.*: Make using the remove or clear signals threadsafe.
27412 Original commit message from CVS:
27413 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
27414 (gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
27415 (gst_multi_fd_sink_get_stats),
27416 (gst_multi_fd_sink_remove_client_link),
27417 (gst_multi_fd_sink_queue_buffer),
27418 (gst_multi_fd_sink_handle_clients):
27419 * gst/tcp/gstmultifdsink.h:
27420 Make using the remove or clear signals threadsafe.
27421 Make calling get-stats with an invalid fd not segfault.
27424 2006-10-31 10:49:19 +0000 Wim Taymans <wim.taymans@gmail.com>
27426 gst-libs/gst/rtp/: Fix and activate base audio payloader.
27427 Original commit message from CVS:
27428 * gst-libs/gst/rtp/Makefile.am:
27429 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
27430 (gst_base_rtp_audio_payload_init):
27431 Fix and activate base audio payloader.
27433 2006-10-28 17:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
27435 gst/typefind/gsttypefindfunctions.c: Add typefinder for QuickTime Image Files (see #366156).
27436 Original commit message from CVS:
27437 * gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
27439 Add typefinder for QuickTime Image Files (see #366156).
27441 2006-10-28 16:00:51 +0000 Tim-Philipp Müller <tim@centricular.net>
27443 gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
27444 Original commit message from CVS:
27445 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
27446 Another typo fix (#366212).
27448 2006-10-27 17:13:35 +0000 Wim Taymans <wim.taymans@gmail.com>
27450 gst/volume/gstvolume.c: Use stream time to synchronize volume property instead of rather random timestamps. This is n...
27451 Original commit message from CVS:
27452 * gst/volume/gstvolume.c: (volume_transform_ip):
27453 Use stream time to synchronize volume property instead of rather random
27454 timestamps. This is needed when gnonlin does its time shifting.
27456 2006-10-27 16:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
27459 I'm too lazy to comment this
27460 Original commit message from CVS:
27461 *** empty log message ***
27463 2006-10-27 16:45:30 +0000 Mark Nauwelaerts <manauw@skynet.be>
27465 ext/ogg/gstoggmux.c: Remove the pad from the element in release_pad.
27466 Original commit message from CVS:
27467 Patch by: Mark Nauwelaerts <manauw at skynet dot be>
27468 * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
27469 Remove the pad from the element in release_pad.
27471 2006-10-27 11:57:18 +0000 Tim-Philipp Müller <tim@centricular.net>
27473 sys/: Explicitly create our custom buffer classes at a thread-safe location as well, since g_type_class_ref() doesn't...
27474 Original commit message from CVS:
27475 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
27476 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
27477 Explicitly create our custom buffer classes at a thread-safe
27478 location as well, since g_type_class_ref() doesn't seem to be
27479 entirely thread-safe either (#365501; also see #349410).
27481 2006-10-26 10:49:00 +0000 Tim-Philipp Müller <tim@centricular.net>
27483 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...
27484 Original commit message from CVS:
27485 * gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
27486 (gst_riff_parse_info):
27487 If strings in INFO chunk are not UTF-8, do something similar to
27488 what we do for ID3v1 tags: check a number of environment variables
27489 (GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
27490 character sets to try, otherwise try the current locale and/or fall
27491 back on ISO-8859-1. Fixes #360552.
27493 2006-10-23 12:46:41 +0000 Tim-Philipp Müller <tim@centricular.net>
27495 gst/videotestsrc/: Add a bunch of exciting new checkers patterns.
27496 Original commit message from CVS:
27497 * gst/videotestsrc/gstvideotestsrc.c:
27498 (gst_video_test_src_pattern_get_type),
27499 (gst_video_test_src_set_pattern):
27500 * gst/videotestsrc/gstvideotestsrc.h:
27501 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_checkers1),
27502 (gst_video_test_src_checkers2), (gst_video_test_src_checkers4),
27503 (gst_video_test_src_checkers8):
27504 * gst/videotestsrc/videotestsrc.h:
27505 Add a bunch of exciting new checkers patterns.
27507 2006-10-23 12:06:44 +0000 Tim-Philipp Müller <tim@centricular.net>
27509 gst/subparse/: Add support for TMPlayer-type subtitles (#362845).
27510 Original commit message from CVS:
27511 * gst/subparse/Makefile.am:
27512 * gst/subparse/gstsubparse.c:
27513 (gst_sub_parse_data_format_autodetect),
27514 (gst_sub_parse_format_autodetect), (handle_buffer),
27515 (gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
27516 * gst/subparse/gstsubparse.h:
27517 * gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
27519 * gst/subparse/tmplayerparse.h:
27520 Add support for TMPlayer-type subtitles (#362845).
27521 * tests/check/elements/subparse.c: (test_tmplayer_do_test),
27522 (GST_START_TEST), (subparse_suite):
27523 Add some basic unit tests for the above.
27525 2006-10-23 11:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
27527 tests/check/elements/audiorate.c: More tests for audiorate: inject buffers to check behaviour when buffers overlap.
27528 Original commit message from CVS:
27529 * tests/check/elements/audiorate.c: (test_injector_base_init),
27530 (test_injector_class_init), (test_injector_chain),
27531 (test_injector_init), (probe_cb), (do_perfect_stream_test),
27532 (GST_START_TEST), (audiorate_suite):
27533 More tests for audiorate: inject buffers to check behaviour when
27536 2006-10-21 16:39:54 +0000 Tim-Philipp Müller <tim@centricular.net>
27538 tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...
27539 Original commit message from CVS:
27540 * tests/check/Makefile.am:
27541 * tests/check/elements/.cvsignore:
27542 * tests/check/elements/audiorate.c: (probe_cb), (got_buf),
27543 (do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
27544 Add some basic unit tests for audiorate. Disabled at the moment
27545 since it doesn't pass yet (see bug #363119).
27547 2006-10-20 17:02:19 +0000 Tim-Philipp Müller <tim@centricular.net>
27549 gst/subparse/gstsubparse.c: Add missing closing tags for markup and fix broken markup, otherwise pango won't render a...
27550 Original commit message from CVS:
27551 * gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
27552 (parse_subrip), (handle_buffer):
27553 Add missing closing tags for markup and fix broken markup,
27554 otherwise pango won't render anything (fixes #357531). Also,
27555 make sure the text we send out is always NUL-terminated
27556 (better safe than sorry etc.).
27557 * tests/check/elements/subparse.c: (test_srt_do_test),
27559 Some more tests for .srt incl. tests for the above stuff.
27561 2006-10-20 13:56:55 +0000 Stefan Kost <ensonic@users.sf.net>
27563 sys/: Try to redraw borders only when needed. Apparently this consumes resources on small devices... :-O (#363607)
27564 Original commit message from CVS:
27565 2006-10-20 Julien MOUTTE <julien@moutte.net>
27566 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
27567 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
27568 Patch by: Stefan Kost <ensonic@users.sf.net>
27569 Try to redraw borders only when needed. Apparently this consumes
27570 resources on small devices... :-O (#363607)
27572 2006-10-20 13:54:19 +0000 Michael Smith <msmith@xiph.org>
27574 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...
27575 Original commit message from CVS:
27576 * gst/tcp/gstmultifdsink.c:
27577 (gst_multi_fd_sink_client_queue_buffer):
27578 If caps change, then update the client's idea of the caps so that we
27579 don't end up re-sending streamheaders for every single buffer after
27582 2006-10-20 12:31:02 +0000 Michael Smith <msmith@xiph.org>
27584 ext/ogg/gstoggparse.c: Set caps on pushed buffers; fix up refcounting of caps objects.
27585 Original commit message from CVS:
27586 * ext/ogg/gstoggparse.c: (gst_ogg_parse_dispose),
27587 (gst_ogg_parse_append_header), (gst_ogg_parse_chain):
27588 Set caps on pushed buffers; fix up refcounting of caps objects.
27590 2006-10-19 14:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
27592 gst/typefind/gsttypefindfunctions.c: Typefind mmsh header data packet to application/x-mmsh (#362625).
27593 Original commit message from CVS:
27594 * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
27596 Typefind mmsh header data packet to application/x-mmsh (#362625).
27598 2006-10-19 09:17:48 +0000 Tim-Philipp Müller <tim@centricular.net>
27600 tests/check/: Add very simple unit test for subparse.
27601 Original commit message from CVS:
27602 * tests/check/Makefile.am:
27603 * tests/check/elements/.cvsignore:
27604 * tests/check/elements/subparse.c: (buffer_from_static_string),
27605 (setup_subparse), (teardown_subparse), (test_srt_do_test),
27606 (GST_START_TEST), (subparse_suite):
27607 Add very simple unit test for subparse.
27609 2006-10-19 09:00:21 +0000 Tim-Philipp Müller <tim@centricular.net>
27611 gst/subparse/gstsubparse.c: Strip trailing newlines from subtitle text output.
27612 Original commit message from CVS:
27613 * gst/subparse/gstsubparse.c: (strip_trailing_newlines),
27615 Strip trailing newlines from subtitle text output.
27617 2006-10-18 18:40:12 +0000 Tim-Philipp Müller <tim@centricular.net>
27619 gst/subparse/gstsubparse.c: Fix memleak; clear subparse->textbuf n state change function.
27620 Original commit message from CVS:
27621 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
27622 (gst_sub_parse_change_state):
27623 Fix memleak; clear subparse->textbuf n state change function.
27625 2006-10-18 15:13:59 +0000 Tim-Philipp Müller <tim@centricular.net>
27627 gst/subparse/gstsubparse.c: Don't require subrip (.srt) files to start with a chunk number of 1.
27628 Original commit message from CVS:
27629 * gst/subparse/gstsubparse.c:
27630 (gst_sub_parse_data_format_autodetect):
27631 Don't require subrip (.srt) files to start with a chunk number of 1.
27633 2006-10-18 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
27635 gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
27636 Original commit message from CVS:
27637 * gst-libs/gst/audio/gstbaseaudiosink.c:
27638 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
27639 * gst-libs/gst/audio/gstbaseaudiosink.h:
27640 Extract rate from the NEWSEGMENT event.
27641 Use commit_full to also take rate adjustment into account when writing
27642 samples to the ringbuffer.
27643 * gst-libs/gst/audio/gstringbuffer.c:
27644 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
27645 (gst_ring_buffer_read):
27646 * gst-libs/gst/audio/gstringbuffer.h:
27647 Added _commit_full() to also take rate into account.
27648 Use simple interpolation algorithm to resample audio.
27649 API: gst_ring_buffer_commit_full()
27650 * tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
27651 * tests/examples/seek/seek.c: (segment_done):
27652 Don't try to seek with 0.0 rate, just pause instead.
27653 Remove bogus debug line.
27655 2006-10-18 12:57:54 +0000 Tim-Philipp Müller <tim@centricular.net>
27657 gst/playback/gstplaybasebin.c: Catch async errors when starting up the subtitle bin, so we can stop waiting and conti...
27658 Original commit message from CVS:
27659 * gst/playback/gstplaybasebin.c: (subbin_startup_sync_msg),
27661 Catch async errors when starting up the subtitle bin, so we can
27662 stop waiting and continue with the main film instead of hanging
27663 forever. Fixes #339366.
27664 * tests/check/elements/playbin.c: (playbin_suite):
27665 Enable unit test for the above.
27667 2006-10-18 09:53:03 +0000 Tim-Philipp Müller <tim@centricular.net>
27669 tests/check/: Some small and basic unit tests for playbin; not very useful yet, but at least a start.
27670 Original commit message from CVS:
27671 * tests/check/Makefile.am:
27672 * tests/check/elements/.cvsignore:
27673 * tests/check/elements/playbin.c: (GST_START_TEST),
27674 (gst_red_video_src_uri_get_type),
27675 (gst_red_video_src_uri_get_protocols),
27676 (gst_red_video_src_uri_get_uri), (gst_red_video_src_uri_set_uri),
27677 (gst_red_video_src_uri_handler_init),
27678 (gst_red_video_src_init_type), (gst_red_video_src_base_init),
27679 (gst_red_video_src_create), (gst_red_video_src_class_init),
27680 (gst_red_video_src_init), (plugin_init), (playbin_suite):
27681 Some small and basic unit tests for playbin; not very useful yet,
27682 but at least a start.
27684 2006-10-18 09:46:35 +0000 Tim-Philipp Müller <tim@centricular.net>
27686 gst/playback/gstplaybin.c: The old pad activation spiel.
27687 Original commit message from CVS:
27688 * gst/playback/gstplaybin.c: (setup_sinks):
27689 The old pad activation spiel.
27691 2006-10-18 09:31:49 +0000 Tim-Philipp Müller <tim@centricular.net>
27693 gst/playback/gstplaybasebin.c: Don't hang forever if the subbin already fails to start up in the state change to PAUS...
27694 Original commit message from CVS:
27695 * gst/playback/gstplaybasebin.c: (setup_source):
27696 Don't hang forever if the subbin already fails to start up in
27697 the state change to PAUSED (#339366).
27699 2006-10-17 17:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
27701 gst-libs/gst/interfaces/tuner.c: Fix some function guards, add some more function guards.
27702 Original commit message from CVS:
27703 * gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
27704 (gst_tuner_set_channel), (gst_tuner_get_channel),
27705 (gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
27706 (gst_tuner_set_frequency), (gst_tuner_get_frequency),
27707 (gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
27708 (gst_tuner_find_channel_by_name):
27709 Fix some function guards, add some more function guards.
27711 2006-10-17 11:34:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27713 gst/playback/gstdecodebin.c: Don't return a pad from get_our_ghost_pad unless it is actually the one we want.
27714 Original commit message from CVS:
27715 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
27716 (remove_element_chain):
27717 Don't return a pad from get_our_ghost_pad unless it is actually the
27719 Change a cast in remove_element_chain slightly.
27721 2006-10-13 15:20:29 +0000 Julien Moutte <julien@moutte.net>
27723 tests/examples/seek/seek.c: Segment seeking needs to use the rate and set stop to -1.
27724 Original commit message from CVS:
27725 2006-10-13 Julien MOUTTE <julien@moutte.net>
27726 * tests/examples/seek/seek.c: (do_seek), (start_seek),
27727 (rate_spinbutton_changed_cb), (segment_done),
27728 (msg_state_changed):
27729 Segment seeking needs to use the rate and set stop to -1.
27731 2006-10-13 14:15:42 +0000 Ville Syrjala <ville.syrjala@movial.fi>
27733 gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
27734 Original commit message from CVS:
27735 * gst-libs/gst/audio/gstbaseaudiosink.c:
27736 (gst_base_audio_sink_setcaps):
27737 Don't crash when ringbuffer is not yet created.
27738 Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
27740 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
27741 * gst/playback/gststreamselector.c:
27742 (gst_stream_selector_request_new_pad):
27743 Activate pads befre adding them to running elements.
27745 2006-10-13 11:25:10 +0000 Julien Moutte <julien@moutte.net>
27747 tests/examples/seek/seek.c: Stop the scale updater when we start grabing the slider. Don't wait for the pipeline to b...
27748 Original commit message from CVS:
27749 2006-10-13 Julien MOUTTE <julien@moutte.net>
27750 * tests/examples/seek/seek.c: (do_seek), (start_seek),
27751 (rate_spinbutton_changed_cb), (msg_state_changed): Stop the
27753 updater when we start grabing the slider. Don't wait for the
27754 pipeline to be PAUSED.
27756 2006-10-13 08:57:52 +0000 Tim-Philipp Müller <tim@centricular.net>
27758 gst-libs/gst/interfaces/mixer.c: Guard mixer interface functions against bogus arguments.
27759 Original commit message from CVS:
27760 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
27761 (gst_mixer_set_volume), (gst_mixer_get_volume),
27762 (gst_mixer_set_mute), (gst_mixer_set_option),
27763 (gst_mixer_get_option), (gst_mixer_mute_toggled),
27764 (gst_mixer_record_toggled), (gst_mixer_volume_changed),
27765 (gst_mixer_option_changed):
27766 Guard mixer interface functions against bogus arguments.
27768 2006-10-12 19:39:07 +0000 Julien Moutte <julien@moutte.net>
27770 tests/examples/seek/seek.c: Use state-changed messages to trigger start/stop of scale update timer. Indeed the scale ...
27771 Original commit message from CVS:
27772 2006-10-12 Julien MOUTTE <julien@moutte.net>
27773 * tests/examples/seek/seek.c: (do_seek), (start_seek),
27775 (play_cb), (pause_cb), (stop_cb),
27776 (rate_spinbutton_changed_cb),
27777 (msg_state_changed), (main): Use state-changed messages to
27779 start/stop of scale update timer. Indeed the scale slider was
27780 jumping here and there because the update timer was activated
27781 before seek completed. This fixes instant applying of rate
27783 by pressing the spinbutton like a crazy man !
27785 2006-10-12 19:09:06 +0000 Sebastien Cote <sebas642@yahoo.ca>
27787 gst-libs/gst/rtp/gstbasertppayload.c: Fix two small memory leaks (#361456).
27788 Original commit message from CVS:
27789 Patch by: Sebastien Cote <sebas642 at yahoo.ca>
27790 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
27791 (gst_basertppayload_finalize):
27792 Fix two small memory leaks (#361456).
27794 2006-10-10 18:56:01 +0000 Julien Moutte <julien@moutte.net>
27796 tests/examples/seek/seek.c: When changing spinbutton we try to change the rate on the fly.
27797 Original commit message from CVS:
27798 2006-10-10 Julien MOUTTE <julien@moutte.net>
27799 * tests/examples/seek/seek.c: (do_seek),
27800 (rate_spinbutton_changed_cb): When changing spinbutton we try
27801 to change the rate on the fly.
27803 2006-10-10 16:50:06 +0000 Wim Taymans <wim.taymans@gmail.com>
27805 gst-libs/gst/riff/: Add WMS caps.
27806 Original commit message from CVS:
27807 * gst-libs/gst/riff/riff-ids.h:
27808 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
27809 (gst_riff_create_audio_template_caps):
27812 2006-10-10 12:49:03 +0000 Josep Torre Valles <josep@fluendo.com>
27814 ext/gnomevfs/: Fix URI interface implementation return type.
27815 Original commit message from CVS:
27816 2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
27817 Patch by: Josep Torre Valles <josep@fluendo.com>
27818 * ext/gnomevfs/gstgnomevfssink.c:
27819 * ext/gnomevfs/gstgnomevfssrc.c:
27820 Fix URI interface implementation return type.
27821 * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
27822 Fix what looks like a copy/paste issue when assigning values.
27823 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
27824 (gst_audio_filter_template_get_type):
27825 Cast to prevent Forte warnings.
27826 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
27827 Fix URI interface implementation return type.
27828 gst_pad_query_position requires a signed integer pointer as
27829 3rd parameter, GstClockTime is unsigned.
27830 * gst/audioconvert/audioconvert.c:
27831 Fix integer overflow when treated as signed.
27832 * gst/audioresample/resample.c: (resample_add_input_data):
27833 Cast to prevent warnings on Forte.
27834 * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
27835 Fix integer overflow when treated as signed.
27836 * gst/ffmpegcolorspace/imgconvert_template.h:
27837 Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
27838 * gst/playback/gstdecodebin.c: (queue_filled_cb),
27839 (cleanup_decodebin):
27840 Who initialises a guint to -1!
27841 Cast function pointers to prevent warnings on Forte.
27842 * gst/playback/gstplaybasebin.c: (queue_deadlock_check),
27843 (queue_threshold_reached):
27844 Cast function pointers correctly to prevent warnings on Forte.
27845 * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
27846 Cast function pointers correctly to prevent warnings on Forte.
27847 * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
27848 Obvious change to unsigned, 0xEF > max signed char.
27849 * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
27850 GstClockTime is unsigned, initialise correctly.
27851 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
27852 Cast so pointer arithemetic doesn't cause warnings on Forte.
27853 * gst/videorate/gstvideorate.c:
27854 Use correct return value.
27855 * tests/examples/seek/scrubby.c:
27856 GstClockTime is unsigned, initialise correctly.
27858 2006-10-10 11:20:03 +0000 Ferenc Gerlits <fgerlits@gmail.com>
27860 gst/typefind/gsttypefindfunctions.c: Recognise XML files and XML-like files shorter than 256 bytes as well (fixes #35...
27861 Original commit message from CVS:
27862 Patch by: Ferenc Gerlits <fgerlits at gmail com>
27863 * gst/typefind/gsttypefindfunctions.c:
27864 Recognise XML files and XML-like files shorter than 256 bytes as
27865 well (fixes #359237).
27867 2006-10-09 15:01:30 +0000 Edgard Lima <edgard.lima@indt.org.br>
27871 * gst/typefind/gsttypefindfunctions.c:
27872 Added typefind functions to video/x-nuv media.
27873 Original commit message from CVS:
27874 Added typefind functions to video/x-nuv media.
27876 2006-10-08 16:59:31 +0000 Tim-Philipp Müller <tim@centricular.net>
27878 gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
27879 Original commit message from CVS:
27880 * gst-libs/gst/interfaces/xoverlay.c:
27881 (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
27882 Some more guards against invalid input.
27884 2006-10-07 18:35:39 +0000 Julien Moutte <julien@moutte.net>
27886 ext/pango/gsttextoverlay.c: Useless goto.
27887 Original commit message from CVS:
27888 2006-10-07 Julien MOUTTE <julien@moutte.net>
27889 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
27891 * tests/examples/seek/seek.c: (do_seek),
27892 (rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
27893 seek example to experiment with rates != 1.0 (reverse playback
27896 2006-10-06 19:20:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27898 gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
27899 Original commit message from CVS:
27900 * gst-libs/gst/interfaces/xoverlay.c:
27901 Unref message in doc-example (spotted by Robert McQueen)
27903 2006-10-06 17:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
27905 gst/typefind/gsttypefindfunctions.c: printf fix.
27906 Original commit message from CVS:
27907 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
27908 (mpeg1_parse_header), (mpeg1_sys_type_find):
27911 2006-10-06 14:37:43 +0000 Wim Taymans <wim.taymans@gmail.com>
27913 gst/playback/: Activate dynamic pads before adding them to the element.
27914 Original commit message from CVS:
27915 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
27917 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
27918 Activate dynamic pads before adding them to the element.
27920 2006-10-06 14:04:53 +0000 Michael Smith <msmith@xiph.org>
27922 gst-libs/gst/floatcast/floatcast.h: Fix obviously-bogus macros; use the correct types.
27923 Original commit message from CVS:
27924 * gst-libs/gst/floatcast/floatcast.h:
27925 Fix obviously-bogus macros; use the correct types.
27927 2006-10-06 13:34:46 +0000 Wim Taymans <wim.taymans@gmail.com>
27929 gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
27930 Original commit message from CVS:
27931 * gst-libs/gst/rtp/gstbasertpdepayload.c:
27932 (gst_base_rtp_depayload_change_state):
27933 Also call parent state change function to activate pads.
27934 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
27935 (mpeg1_parse_header), (mpeg1_sys_type_find):
27936 Add some more debug info in mpeg typefinding.
27938 2006-10-06 12:57:10 +0000 Michael Smith <msmith@xiph.org>
27940 ext/theora/theoradec.c: Zero byte theora packets are valid and well-defined; don't warn on them.
27941 Original commit message from CVS:
27942 * ext/theora/theoradec.c: (theora_dec_chain):
27943 Zero byte theora packets are valid and well-defined; don't warn on
27946 2006-10-06 10:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27948 gst/tcp/gstmultifdsink.c: API: add dropped_buffers to the get-stats GValueArray
27949 Original commit message from CVS:
27950 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
27951 (gst_multi_fd_sink_get_stats), (find_limits),
27952 (gst_multi_fd_sink_queue_buffer):
27953 API: add dropped_buffers to the get-stats GValueArray
27955 2006-10-05 15:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
27957 Printf format fixes.
27958 Original commit message from CVS:
27959 * ext/alsa/gstalsadeviceprobe.c:
27960 (gst_alsa_device_property_probe_get_values):
27961 * ext/alsa/gstalsasink.c: (set_hwparams):
27962 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
27963 (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
27964 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
27965 (gst_ogg_mux_process_best_pad):
27966 * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
27967 (gst_ogg_parse_chain):
27968 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
27969 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
27970 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
27971 (gst_vorbis_enc_buffer_check_discontinuous):
27972 * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
27973 * gst-libs/gst/audio/gstbaseaudiosink.c:
27974 (gst_base_audio_sink_render):
27975 * gst-libs/gst/cdda/gstcddabasesrc.c:
27976 (gst_cdda_base_src_handle_track_seek):
27977 * gst-libs/gst/rtp/gstbasertpdepayload.c:
27978 (gst_base_rtp_depayload_push_full):
27979 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
27980 * gst/audioresample/resample.c: (resample_input_pushthrough):
27981 * gst/playback/gstplaybasebin.c: (queue_out_of_data):
27982 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
27983 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
27984 (wavpack_type_find):
27985 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
27986 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
27987 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
27988 * tests/check/elements/volume.c: (GST_START_TEST):
27989 Printf format fixes.
27991 2006-10-04 13:18:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27993 gst/tcp/gsttcp.c: Fix a simple mistake (see the docs)
27994 Original commit message from CVS:
27995 * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
27996 Fix a simple mistake (see the docs)
27999 2006-10-04 13:15:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28001 * win32/common/config.h:
28003 Original commit message from CVS:
28006 2006-10-03 12:11:45 +0000 Tim-Philipp Müller <tim@centricular.net>
28008 docs/plugins/: Add vorbistag element to docs; update version numbers to 0.10.10.1.
28009 Original commit message from CVS:
28010 * docs/plugins/Makefile.am:
28011 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
28012 * docs/plugins/gst-plugins-base-plugins-sections.txt:
28013 * docs/plugins/gst-plugins-base-plugins.args:
28014 * docs/plugins/gst-plugins-base-plugins.hierarchy:
28015 * docs/plugins/inspect/plugin-adder.xml:
28016 * docs/plugins/inspect/plugin-alsa.xml:
28017 * docs/plugins/inspect/plugin-audioconvert.xml:
28018 * docs/plugins/inspect/plugin-audiorate.xml:
28019 * docs/plugins/inspect/plugin-audioresample.xml:
28020 * docs/plugins/inspect/plugin-audiotestsrc.xml:
28021 * docs/plugins/inspect/plugin-cdparanoia.xml:
28022 * docs/plugins/inspect/plugin-decodebin.xml:
28023 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
28024 * docs/plugins/inspect/plugin-gdp.xml:
28025 * docs/plugins/inspect/plugin-gnomevfs.xml:
28026 * docs/plugins/inspect/plugin-libvisual.xml:
28027 * docs/plugins/inspect/plugin-ogg.xml:
28028 * docs/plugins/inspect/plugin-pango.xml:
28029 * docs/plugins/inspect/plugin-playbin.xml:
28030 * docs/plugins/inspect/plugin-subparse.xml:
28031 * docs/plugins/inspect/plugin-tcp.xml:
28032 * docs/plugins/inspect/plugin-theora.xml:
28033 * docs/plugins/inspect/plugin-typefindfunctions.xml:
28034 * docs/plugins/inspect/plugin-video4linux.xml:
28035 * docs/plugins/inspect/plugin-videorate.xml:
28036 * docs/plugins/inspect/plugin-videoscale.xml:
28037 * docs/plugins/inspect/plugin-videotestsrc.xml:
28038 * docs/plugins/inspect/plugin-volume.xml:
28039 * docs/plugins/inspect/plugin-vorbis.xml:
28040 * docs/plugins/inspect/plugin-ximagesink.xml:
28041 * docs/plugins/inspect/plugin-xvimagesink.xml:
28042 Add vorbistag element to docs; update version numbers to 0.10.10.1.
28044 2006-10-03 11:51:48 +0000 James Doc Livingston <doclivingston@gmail.com>
28046 ext/vorbis/: Add new vorbistag element which derives from vorbisparse and is essentially the same as well, only that ...
28047 Original commit message from CVS:
28048 Patch by: James "Doc" Livingston <doclivingston at gmail com>
28049 * ext/vorbis/Makefile.am:
28050 * ext/vorbis/vorbis.c: (plugin_init):
28051 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_class_init),
28052 (vorbis_parse_parse_packet), (vorbis_parse_chain):
28053 * ext/vorbis/vorbisparse.h:
28054 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_base_init),
28055 (gst_vorbis_tag_class_init), (gst_vorbis_tag_init),
28056 (gst_vorbis_tag_parse_packet):
28057 * ext/vorbis/vorbistag.h:
28058 Add new vorbistag element which derives from vorbisparse
28059 and is essentially the same as well, only that it implements
28060 the GstTagSetter interface and can modify the stream's
28061 vorbiscomment on the fly (#335635).
28062 * tests/check/Makefile.am:
28063 * tests/check/elements/.cvsignore:
28064 * tests/check/elements/vorbistag.c: (setup_vorbistag),
28065 (cleanup_vorbistag), (buffer_probe), (start_pipeline),
28066 (get_buffer), (stop_pipeline), (_create_codebook_header_buffer),
28067 (_create_audio_buffer), (GST_START_TEST), (vorbistag_suite):
28068 Add unit test for new vorbistag element.
28070 2006-10-03 10:36:38 +0000 Tim-Philipp Müller <tim@centricular.net>
28072 ext/vorbis/vorbisparse.c: Set BOS flag in packet structure to fix 'jump depends on unitialized value' errors in valgr...
28073 Original commit message from CVS:
28074 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init),
28075 (vorbis_parse_push_headers), (vorbis_parse_chain):
28076 Set BOS flag in packet structure to fix 'jump depends
28077 on unitialized value' errors in valgrind; various minor
28080 2006-09-30 15:30:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28082 gst/playback/gstdecodebin.c: Fix typo in a debug statement.
28083 Original commit message from CVS:
28084 * gst/playback/gstdecodebin.c: (close_pad_link):
28085 Fix typo in a debug statement.
28086 * gst/playback/gstplaybasebin.c: (probe_triggered),
28087 (new_decoded_pad_full), (new_decoded_pad), (subs_new_decoded_pad),
28088 (gen_source_element), (source_new_pad), (analyse_source),
28090 When handling no_more_pads in new_decoded_pad, make sure to treat
28091 subtitle pads correctly. Fixes playback with subtitle files.
28092 Move a recurring message to LOG level.
28093 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
28094 The maximum value for the Xv colorkey on this Radeon is 0xFFFFFFFF,
28095 which ends up as -1 when cast to an int. Make the logic handle the
28096 max value as an unsigned mask and only change the colorkey when it's
28097 a value we recognise.
28099 2006-09-30 00:14:20 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28101 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
28102 Original commit message from CVS:
28103 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
28104 Removed empty * between paragraphs
28106 2006-09-29 23:50:53 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28108 gst-libs/gst/rtp/: Moved some documentation into .c file
28109 Original commit message from CVS:
28110 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
28111 * gst-libs/gst/rtp/README:
28112 Moved some documentation into .c file
28114 2006-09-29 17:35:01 +0000 Wim Taymans <wim.taymans@gmail.com>
28116 gst/playback/gstdecodebin.c: Fix compilation.
28117 Original commit message from CVS:
28118 * gst/playback/gstdecodebin.c: (no_more_pads):
28121 2006-09-29 16:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
28123 gst/playback/gstdecodebin.c: Remove g_print
28124 Original commit message from CVS:
28125 * gst/playback/gstdecodebin.c: (new_caps):
28127 * gst/playback/gstplaybin.c:
28130 2006-09-29 15:16:32 +0000 Tim-Philipp Müller <tim@centricular.net>
28132 tests/check/Makefile.am: Re-enable cddabasesrc test to see if it works again now.
28133 Original commit message from CVS:
28134 * tests/check/Makefile.am:
28135 Re-enable cddabasesrc test to see if it works again
28138 2006-09-29 13:46:45 +0000 Wim Taymans <wim.taymans@gmail.com>
28140 gst/playback/gstplaybasebin.c: Handle invalid URIs a bit more gracefully.
28141 Original commit message from CVS:
28142 * gst/playback/gstplaybasebin.c: (setup_subtitle),
28143 (gen_source_element):
28144 Handle invalid URIs a bit more gracefully.
28146 2006-09-29 12:54:28 +0000 Tim-Philipp Müller <tim@centricular.net>
28148 tests/check/pipelines/oggmux.c: Remove obsolete comment.
28149 Original commit message from CVS:
28150 * tests/check/pipelines/oggmux.c:
28151 Remove obsolete comment.
28153 2006-09-29 10:43:05 +0000 James Doc Livingston <doclivingston@gmail.com>
28155 ext/ogg/gstoggmux.c: Commit patch from James "Doc" Livingston, adds proper EOS handling in oggmux. GStreamer can, for...
28156 Original commit message from CVS:
28157 * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
28158 (gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
28159 (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
28160 (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
28161 (gst_ogg_mux_collected):
28162 Commit patch from James "Doc" Livingston, adds proper EOS handling
28163 in oggmux. GStreamer can, for the first time ever, create a valid
28165 * tests/check/pipelines/oggmux.c: (check_chain_final_state),
28167 Reenable tests now that they pass.
28169 2006-09-29 08:20:03 +0000 Wim Taymans <wim.taymans@gmail.com>
28171 gst/tcp/gstmultifdsink.c: Stop reading commands when EOF (we read 0) as well.
28172 Original commit message from CVS:
28173 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
28174 Stop reading commands when EOF (we read 0) as well.
28176 2006-09-28 15:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
28178 gst/playback/gstdecodebin.c: Implement delayed caps linking needed for element with a lot of different caps on the sr...
28179 Original commit message from CVS:
28180 * gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
28181 (close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
28182 (find_dynamic), (unlinked), (close_link):
28183 Implement delayed caps linking needed for element with a lot of
28184 different caps on the src pads that get fixed at runtime.
28185 Improve management of dynamic elements.
28186 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
28187 (group_destroy), (group_commit), (check_queue), (queue_overrun),
28188 (gen_preroll_element), (remove_groups), (unknown_type),
28189 (add_element_stream), (no_more_pads_full), (no_more_pads),
28190 (sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
28191 (new_decoded_pad), (setup_subtitle), (array_has_value),
28192 (gen_source_element), (source_new_pad), (has_all_raw_caps),
28193 (analyse_source), (remove_decoders), (make_decoder),
28194 (remove_source), (setup_source), (finish_source), (prepare_output),
28195 (gst_play_base_bin_change_state):
28196 * gst/playback/gstplaybasebin.h:
28197 Use more _CAST instead of full type checking casts.
28198 Small cleanups, plug some leaks.
28199 Handle dynamic sources.
28200 Add some helper functions to create lists of strings used for
28201 blacklisting and other stuff.
28202 Refactor some code dealing with analysing the source.
28203 Re-enable sources without pads (like cd:// or other selfcontained
28206 2006-09-28 15:08:15 +0000 Wim Taymans <wim.taymans@gmail.com>
28208 gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
28209 Original commit message from CVS:
28210 * gst-libs/gst/audio/gstbaseaudiosink.c:
28211 (gst_base_audio_sink_render):
28212 When we have a timestamp, we can still perform clipping.
28213 When we have no clock, we must play the sample ASAP.
28215 2006-09-28 11:46:26 +0000 Wim Taymans <wim.taymans@gmail.com>
28217 gst/audiorate/gstaudiorate.c: Set caps on outgoing buffers.
28218 Original commit message from CVS:
28219 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
28220 Set caps on outgoing buffers.
28221 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
28222 (gst_video_rate_event), (gst_video_rate_chain):
28223 * gst/videorate/gstvideorate.h:
28224 Fix videorate some more. Fixes #357977
28226 2006-09-28 11:34:05 +0000 Tim-Philipp Müller <tim@centricular.net>
28228 tests/check/elements/adder.c: Don't set timeout to 6 seconds when we're running in valgrind ... (and how is 6 seconds...
28229 Original commit message from CVS:
28230 * tests/check/elements/adder.c: (adder_suite):
28231 Don't set timeout to 6 seconds when we're running
28232 in valgrind ... (and how is 6 seconds longer than
28233 the default anyway?)
28235 2006-09-28 10:49:56 +0000 Wim Taymans <wim.taymans@gmail.com>
28237 gst/audiorate/gstaudiorate.c: Keep sink and src segment to keep track of time and support more input formats.
28238 Original commit message from CVS:
28239 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
28240 (gst_audio_rate_sink_event), (gst_audio_rate_convert),
28241 (gst_audio_rate_convert_segments), (gst_audio_rate_chain):
28242 Keep sink and src segment to keep track of time and support more
28244 Fix bogus next_offset and run_time calculation, don't understand how
28245 this could have worked before. Fixes #357976.
28246 Remove some unneeded vars.
28248 2006-09-28 09:41:20 +0000 Tim-Philipp Müller <tim@centricular.net>
28250 gst/playback/gstplaybin.c: Only remove visualisation from visbin if there is a visbin (or: don't throw warnings when ...
28251 Original commit message from CVS:
28252 * gst/playback/gstplaybin.c: (remove_sinks):
28253 Only remove visualisation from visbin if there is a visbin (or:
28254 don't throw warnings when closing totem without playing a file).
28256 2006-09-27 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
28258 gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
28259 Original commit message from CVS:
28260 * gst-libs/gst/audio/gstbaseaudiosink.c:
28261 (gst_base_audio_sink_render):
28262 Add some more info in a WARNING.
28263 * gst-libs/gst/audio/gstbaseaudiosrc.c:
28264 (gst_base_audio_src_create):
28265 Handle PAUSE in create function, use new -core addition to
28266 wait for playing. Fixes pausing and resuming capture from an
28268 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
28269 (gst_ring_buffer_read):
28270 Constify some more.
28271 Caller supports interrupted reads now.
28273 2006-09-27 13:29:49 +0000 Christian Schaller <uraeus@gnome.org>
28275 * gst-plugins-base.spec.in:
28276 add new header file to spec
28277 Original commit message from CVS:
28278 add new header file to spec
28280 2006-09-27 12:55:45 +0000 Tim-Philipp Müller <tim@centricular.net>
28282 tests/check/Makefile.am: Another attempt to make the gen64 buildbot happy.
28283 Original commit message from CVS:
28284 * tests/check/Makefile.am:
28285 Another attempt to make the gen64 buildbot happy.
28287 2006-09-27 11:58:17 +0000 Jonathan Matthew <jonathan@kaolin.wh9.net>
28289 ext/libvisual/visual.c: Libvisual plugin was not passing audio data to libvisual 0.4.0 correctly. Fixes #357800
28290 Original commit message from CVS:
28291 Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
28292 * ext/libvisual/visual.c: (gst_visual_clear_actors),
28293 (gst_visual_chain), (gst_visual_change_state):
28294 Libvisual plugin was not passing audio data to libvisual 0.4.0
28295 correctly. Fixes #357800
28297 2006-09-27 11:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
28299 tests/check/pipelines/simple-launch-lines.c: Add timeout to _get_state() so we see which pipeline it is that causes t...
28300 Original commit message from CVS:
28301 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
28302 Add timeout to _get_state() so we see which pipeline it is
28303 that causes trouble on the gen64 build bot.
28305 2006-09-27 11:06:54 +0000 Wim Taymans <wim.taymans@gmail.com>
28307 gst-libs/gst/rtp/gstbasertpdepayload.c: the source pad always uses fixed caps.
28308 Original commit message from CVS:
28309 * gst-libs/gst/rtp/gstbasertpdepayload.c:
28310 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
28311 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
28312 (gst_base_rtp_depayload_set_gst_timestamp):
28313 the source pad always uses fixed caps.
28315 2006-09-27 11:05:08 +0000 Wim Taymans <wim.taymans@gmail.com>
28317 Added docs for the audio libs.
28318 Original commit message from CVS:
28319 * docs/libs/gst-plugins-base-libs-docs.sgml:
28320 * docs/libs/gst-plugins-base-libs-sections.txt:
28321 * gst-libs/gst/audio/gstaudioclock.c:
28322 * gst-libs/gst/audio/gstaudioclock.h:
28323 * gst-libs/gst/audio/gstaudiosink.c:
28324 * gst-libs/gst/audio/gstaudiosink.h:
28325 * gst-libs/gst/audio/gstaudiosrc.c:
28326 * gst-libs/gst/audio/gstbaseaudiosink.c:
28327 (gst_base_audio_sink_render):
28328 * gst-libs/gst/audio/gstbaseaudiosink.h:
28329 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
28330 * gst-libs/gst/audio/gstbaseaudiosrc.h:
28331 * gst-libs/gst/audio/gstringbuffer.h:
28332 Added docs for the audio libs.
28334 2006-09-27 10:59:24 +0000 Tim-Philipp Müller <tim@centricular.net>
28336 tests/check/Makefile.am: Temporarily disable test that fails on the bots for unknown reasons.
28337 Original commit message from CVS:
28338 * tests/check/Makefile.am:
28339 Temporarily disable test that fails on the bots for unknown reasons.
28341 2006-09-27 00:13:29 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28343 gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
28344 Original commit message from CVS:
28345 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
28346 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
28347 Moved AudioCodecType into priv
28348 Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
28350 2006-09-25 15:47:25 +0000 Wim Taymans <wim.taymans@gmail.com>
28352 gst/playback/gstdecodebin.c: Cleanups and small leak fixes.
28353 Original commit message from CVS:
28354 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
28355 (add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
28356 (is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
28358 Cleanups and small leak fixes.
28359 Added Depayloaders to valid list of autopluggable elements.
28361 2006-09-25 13:24:59 +0000 Wim Taymans <wim.taymans@gmail.com>
28363 gst/playback/gstplaybin.c: Detect NO_PREROLL state change returns and disable clock distribution to the sinks so that...
28364 Original commit message from CVS:
28365 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
28366 (gst_play_bin_vis_blocked), (gst_play_bin_set_property),
28367 (gen_video_element), (gen_text_element), (gen_audio_element),
28368 (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
28369 (gst_play_bin_set_clock_func), (gst_play_bin_change_state):
28370 Detect NO_PREROLL state change returns and disable clock distribution to
28371 the sinks so that sync is disabled.
28372 Avoid some type checking and do simple casts instead.
28373 Small cleanups, fix some FIXMEs.
28374 Be more robust when linking user specified elements, catch an report
28375 errors. Fixes #357404.
28376 Fix some leaks in the error paths.
28378 2006-09-25 12:55:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28381 ChangeLog surgery for missing bug-number
28382 Original commit message from CVS:
28383 ChangeLog surgery for missing bug-number
28385 2006-09-25 11:28:15 +0000 Peter Kjellerstedt <pkj@axis.com>
28387 gst/playback/test.c: Fix compilation with uClibc and -Werror (#357591).
28388 Original commit message from CVS:
28389 Patch by: Peter Kjellerstedt <pkj at axis com>
28390 * gst/playback/test.c:
28391 Fix compilation with uClibc and -Werror (#357591).
28393 2006-09-25 10:21:31 +0000 Tim-Philipp Müller <tim@centricular.net>
28395 gst-libs/gst/tag/gstvorbistag.c: Parse dates that are followed by a time as well (#357532).
28396 Original commit message from CVS:
28397 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
28398 Parse dates that are followed by a time as well (#357532).
28399 * tests/check/libs/tag.c: (test_vorbis_tags):
28400 Add unit test for this.
28402 2006-09-23 15:24:55 +0000 Tim-Philipp Müller <tim@centricular.net>
28404 gst/: A few array const-ifications.
28405 Original commit message from CVS:
28406 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
28407 (gst_audio_convert_transform_caps):
28408 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
28409 * gst/videotestsrc/videotestsrc.h:
28410 A few array const-ifications.
28412 2006-09-23 15:02:51 +0000 Tim-Philipp Müller <tim@centricular.net>
28414 tests/check/Makefile.am: See if this makes the build bots happy.
28415 Original commit message from CVS:
28416 * tests/check/Makefile.am:
28417 See if this makes the build bots happy.
28418 * tests/check/libs/cddabasesrc.c:
28421 2006-09-23 14:30:53 +0000 Young-Ho Cha <ganadist@chollian.net>
28423 gst/subparse/samiparse.c: More case-insensitivity for certain tags; recognise entities with decimal codes as special ...
28424 Original commit message from CVS:
28425 Patch by: Young-Ho Cha <ganadist at chollian dot net>
28426 * gst/subparse/samiparse.c: (handle_start_font),
28427 (fix_invalid_entities):
28428 More case-insensitivity for certain tags; recognise entities with
28429 decimal codes as special entities as well (#357330).
28431 2006-09-23 13:32:07 +0000 Tim-Philipp Müller <tim@centricular.net>
28433 gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
28434 Original commit message from CVS:
28435 * gst-libs/gst/Makefile.am:
28436 Need to build tag directory before cdda.
28438 2006-09-23 13:21:07 +0000 Tim-Philipp Müller <tim@centricular.net>
28440 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
28441 Original commit message from CVS:
28442 * docs/libs/gst-plugins-base-libs-sections.txt:
28443 * gst-libs/gst/cdda/Makefile.am:
28444 * gst-libs/gst/cdda/gstcddabasesrc.c:
28445 (gst_cdda_base_src_base_init):
28446 * gst-libs/gst/cdda/gstcddabasesrc.h:
28447 * gst-libs/gst/tag/tag.h:
28448 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
28449 (gst_tag_register_musicbrainz_tags):
28450 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
28451 depend on libgsttag. This is required so we can extract/read tags like
28452 DISCID without depending on libgstcddabasesrc (which used to register
28454 * gst-libs/gst/tag/gstvorbistag.c:
28455 Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
28456 tags (also see #347848).
28457 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
28458 Log vorbis comments we are actually writing. Const-ify array.
28460 2006-09-23 08:53:30 +0000 Wim Taymans <wim.taymans@gmail.com>
28462 gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
28463 Original commit message from CVS:
28464 * gst/playback/gstplaybasebin.c: (gen_preroll_element):
28465 Improve buffering a bit by avoiding a deadlock because we cannot assume
28466 the underrun is always called.
28468 2006-09-23 08:51:14 +0000 Young-Ho Cha <ganadist@chollian.net>
28470 gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
28471 Original commit message from CVS:
28472 Patch by: Young-Ho Cha <ganadist at chollian dot net>
28473 * gst-libs/gst/riff/riff-ids.h:
28474 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
28475 (gst_riff_create_audio_template_caps):
28476 Added MPEG-4 AAC and id and caps. Fixes #357289
28477 Added WMA9 Lossless id.
28479 2006-09-22 14:50:01 +0000 Tim-Philipp Müller <tim@centricular.net>
28481 ext/gnomevfs/gstgnomevfssrc.c: Fix misleading docs addition.
28482 Original commit message from CVS:
28483 * ext/gnomevfs/gstgnomevfssrc.c:
28484 Fix misleading docs addition.
28485 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
28486 Get rid of compiler warning the right way.
28488 2006-09-22 14:13:34 +0000 Wim Taymans <wim.taymans@gmail.com>
28490 gst-libs/gst/rtp/gstbasertpdepayload.*: Small cleanups.
28491 Original commit message from CVS:
28492 * gst-libs/gst/rtp/gstbasertpdepayload.c:
28493 (gst_base_rtp_depayload_finalize),
28494 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
28495 (gst_base_rtp_depayload_push_full),
28496 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
28497 (gst_base_rtp_depayload_process),
28498 (gst_base_rtp_depayload_set_gst_timestamp),
28499 (gst_base_rtp_depayload_queue_release):
28500 * gst-libs/gst/rtp/gstbasertpdepayload.h:
28503 Refactored the process method and added methods to push from the process
28505 Use _scale functions.
28506 API: gst_base_rtp_depayload_push_ts
28507 API: gst_base_rtp_depayload_push
28508 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
28509 timestamps are uint.
28511 2006-09-22 11:59:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28513 gst-libs/gst/interfaces/xoverlay.c: Remove unused statement from doc example.
28514 Original commit message from CVS:
28515 * gst-libs/gst/interfaces/xoverlay.c:
28516 Remove unused statement from doc example.
28518 2006-09-22 09:52:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28520 * gst/videorate/gstvideorate.c:
28522 Original commit message from CVS:
28525 2006-09-21 13:49:47 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28527 gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
28528 Original commit message from CVS:
28529 * gst-libs/gst/interfaces/videoorientation.c:
28530 (gst_video_orientation_iface_init),
28531 (gst_video_orientation_get_hflip),
28532 (gst_video_orientation_get_vflip),
28533 (gst_video_orientation_get_hcenter),
28534 (gst_video_orientation_get_vcenter),
28535 (gst_video_orientation_set_hflip),
28536 (gst_video_orientation_set_vflip),
28537 (gst_video_orientation_set_hcenter),
28538 (gst_video_orientation_set_vcenter):
28539 Add since tags to new API docs, ChangeLog surgery (forgot API keyword
28542 2006-09-21 09:27:47 +0000 Tim-Philipp Müller <tim@centricular.net>
28544 tests/check/: but disable for now since it doesn't pass (something wrong with
28545 Original commit message from CVS:
28546 * tests/check/Makefile.am:
28547 * tests/check/elements/.cvsignore:
28548 * tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
28549 (create_rgb_conversions), (rgb_conversion_free),
28550 (right_shift_colour), (fix_expected_colour), (check_rgb_buf),
28551 (got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
28552 Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
28553 but disable for now since it doesn't pass (something wrong with
28556 2006-09-21 07:01:48 +0000 Wim Taymans <wim.taymans@gmail.com>
28558 gst/playback/gstplaybasebin.c: Refactor handling of overrun detection.
28559 Original commit message from CVS:
28560 * gst/playback/gstplaybasebin.c: (group_commit),
28561 (queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
28562 (queue_out_of_data), (gen_preroll_element),
28563 (preroll_remove_overrun), (probe_triggered):
28564 Refactor handling of overrun detection.
28565 Separate handling of group completion and deadlock detection when doing
28566 network buffering. This should fix some deadlocks that were not detected
28567 because the group was completed.
28568 Add more comments, improve debugging.
28570 2006-09-21 05:31:00 +0000 Wim Taymans <wim.taymans@gmail.com>
28572 tests/check/: Some more compilation fixes.
28573 Original commit message from CVS:
28574 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
28575 * tests/check/libs/audio.c:
28576 Some more compilation fixes.
28578 2006-09-21 05:12:18 +0000 Wim Taymans <wim.taymans@gmail.com>
28580 gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
28581 Original commit message from CVS:
28582 * gst-libs/gst/audio/gstringbuffer.c:
28583 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
28584 (gst_ring_buffer_read):
28585 Early morning compilation fix.
28587 2006-09-20 18:09:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28591 Original commit message from CVS:
28594 2006-09-20 17:04:57 +0000 Wim Taymans <wim.taymans@gmail.com>
28596 tests/check/: Fix some warnings.
28597 Original commit message from CVS:
28598 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
28599 * tests/check/elements/multifdsink.c: (GST_START_TEST):
28600 * tests/check/elements/videorate.c: (GST_START_TEST):
28601 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
28602 * tests/check/pipelines/oggmux.c: (eos_buffer_probe):
28605 2006-09-20 10:59:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28607 sys/xvimage/xvimagesink.c: change colorkey behaviour back according to #354773 comment 6/7
28608 Original commit message from CVS:
28609 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
28610 (gst_xvimagesink_get_times):
28611 change colorkey behaviour back according to #354773 comment 6/7
28613 2006-09-20 10:42:34 +0000 Tim-Philipp Müller <tim@centricular.net>
28616 ChangeLog surgery: remove junk
28617 Original commit message from CVS:
28618 ChangeLog surgery: remove junk
28620 2006-09-19 11:31:06 +0000 Michael Smith <msmith@xiph.org>
28622 gst/tcp/gstmultifdsink.*: Implement stubbed out properties unit-type, units-soft-max, units-max, to allow specifying ...
28623 Original commit message from CVS:
28624 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
28625 (gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
28626 (gst_multi_fd_sink_recover_client),
28627 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
28628 (gst_multi_fd_sink_get_property):
28629 * gst/tcp/gstmultifdsink.h:
28630 Implement stubbed out properties unit-type, units-soft-max,
28631 units-max, to allow specifying maximum sizes in units other than
28635 2006-09-19 10:23:03 +0000 Wim Taymans <wim.taymans@gmail.com>
28637 gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
28638 Original commit message from CVS:
28639 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
28640 (gst_riff_create_audio_template_caps):
28641 Reorder the audio formats a bit for clarity.
28642 Detect and create caps for MSGSM and MSN (WAV49).
28644 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
28645 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
28646 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
28647 Small cleanups, move error handling out of normal flow for clarity.
28649 2006-09-18 15:59:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28651 Add new interface to control video orientation (fixes #354908)
28652 Original commit message from CVS:
28653 * docs/libs/gst-plugins-base-libs-docs.sgml:
28654 * docs/libs/gst-plugins-base-libs.types:
28655 * gst-libs/gst/interfaces/Makefile.am:
28656 * gst-libs/gst/interfaces/videoorientation.c:
28657 (gst_video_orientation_get_type),
28658 (gst_video_orientation_iface_init),
28659 (gst_video_orientation_get_hflip),
28660 (gst_video_orientation_get_vflip),
28661 (gst_video_orientation_get_hcenter),
28662 (gst_video_orientation_get_vcenter),
28663 (gst_video_orientation_set_hflip),
28664 (gst_video_orientation_set_vflip),
28665 (gst_video_orientation_set_hcenter),
28666 (gst_video_orientation_set_vcenter):
28667 * gst-libs/gst/interfaces/videoorientation.h:
28668 Add new interface to control video orientation (fixes #354908)
28670 2006-09-18 15:48:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28672 gst/videotestsrc/gstvideotestsrc.c: Use G_UNLIKELY in _create and log one more detail.
28673 Original commit message from CVS:
28674 * gst/videotestsrc/gstvideotestsrc.c:
28675 Use G_UNLIKELY in _create and log one more detail.
28676 (gst_video_test_src_get_times), (gst_video_test_src_create):
28677 * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
28678 Use gst_util_uint64_scale_int in _get_times().
28680 2006-09-18 15:00:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28682 sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
28683 Original commit message from CVS:
28684 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
28685 Give better warning message (add object and detail).
28687 2006-09-18 14:42:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28689 sys/xvimage/xvimagesink.c: xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes #354773), use gst_util...
28690 Original commit message from CVS:
28691 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
28692 (gst_xvimagesink_get_times):
28693 xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
28694 #354773), use gst_util_uint64_scale_int in _get_times()
28696 2006-09-18 14:21:45 +0000 Michael Smith <msmith@xiph.org>
28698 ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
28699 Original commit message from CVS:
28700 * ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
28701 Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
28702 always true, leading to dropping all timestamps.
28704 2006-09-18 11:40:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28706 ext/libvisual/visual.c: update to work also with libvisual 0.4 API
28707 Original commit message from CVS:
28708 * ext/libvisual/visual.c: (gst_vis_src_negotiate),
28709 (gst_visual_chain), (gst_visual_change_state):
28710 update to work also with libvisual 0.4 API
28711 * tools/gst-launch-ext.1.in:
28712 * tools/gst-visualise.1.in:
28713 remove references to old man-pages
28714 * tests/examples/seek/seek.c: (main):
28715 add real meadi-buttons, add tool-tips for the seek-options, arrange
28716 seek options in a table
28718 2006-09-18 10:57:28 +0000 Michael Smith <msmith@xiph.org>
28720 ext/ogg/gstoggmux.c: Don't generate out-of-order timestamps from oggmux, instead clamp output timestamps to be >= the...
28721 Original commit message from CVS:
28722 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
28723 (gst_ogg_mux_push_buffer):
28724 Don't generate out-of-order timestamps from oggmux, instead clamp
28725 output timestamps to be >= the previously output ts.
28728 2006-09-18 10:18:22 +0000 Michael Smith <msmith@xiph.org>
28730 gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
28731 Original commit message from CVS:
28732 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
28733 (gst_multi_fd_sink_class_init):
28734 Updates, fixes, and typo corrections for multifdsink. No functional
28737 2006-09-17 21:58:06 +0000 Michael Smith <msmith@xiph.org>
28739 gst/typefind/gsttypefindfunctions.c: Don't crash on truncated files - check that we got an 8 byte buffer before tryin...
28740 Original commit message from CVS:
28741 * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
28742 Don't crash on truncated files - check that we got an 8 byte buffer
28743 before trying to memcmp it.
28745 2006-09-17 20:32:09 +0000 Tim-Philipp Müller <tim@centricular.net>
28747 gst/playback/gstplaybasebin.c: Make stream-switching appear instant to the application (ie. make sure that a g_object...
28748 Original commit message from CVS:
28749 * gst/playback/gstplaybasebin.c: (get_active_source):
28750 Make stream-switching appear instant to the application
28751 (ie. make sure that a g_object_get on 'current-foo' returns
28752 the stream previously set with g_object_set(). Totem needs
28753 this to update stream-related meta-info (like audio-codec)
28754 correctly when switching streams.
28756 2006-09-17 20:14:43 +0000 Tim-Philipp Müller <tim@centricular.net>
28758 ext/alsa/gstalsamixer.c: Try harder to guess which mixer track is the master mixer track (instead of just taking the ...
28759 Original commit message from CVS:
28760 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
28761 (gst_alsa_mixer_ensure_track_list):
28762 Try harder to guess which mixer track is the master mixer
28763 track (instead of just taking the first one that has a pvolume).
28766 2006-09-17 11:24:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28768 gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
28769 Original commit message from CVS:
28770 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
28771 (gst_audio_convert_transform_caps):
28772 Get structure-name just once.
28774 2006-09-16 22:30:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28776 tests/check/: Fix big batch of compiler warnings.
28777 Original commit message from CVS:
28778 * tests/check/elements/audioresample.c: (GST_START_TEST):
28779 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
28780 * tests/check/elements/volume.c: (GST_START_TEST):
28781 * tests/check/elements/vorbisdec.c: (GST_START_TEST):
28782 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
28783 (test_pipeline), (GST_START_TEST):
28784 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
28785 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
28786 Fix big batch of compiler warnings.
28788 2006-09-16 21:54:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28790 ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
28791 Original commit message from CVS:
28792 * ext/gnomevfs/gstgnomevfssrc.c:
28793 Add docs about icydemux usage in connection with gnomevfssrc
28794 * ext/libvisual/visual.c:
28795 * ext/ogg/gstoggaviparse.c:
28796 * ext/ogg/gstoggdemux.c:
28797 * ext/ogg/gstoggmux.c:
28798 * ext/ogg/gstoggparse.c:
28799 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
28800 * gst-libs/gst/audio/gstaudiosink.c:
28801 * gst-libs/gst/audio/gstaudiosrc.c:
28802 * gst/audiorate/gstaudiorate.c:
28803 More G_OBJECT macro fixing.
28804 * gst/audiotestsrc/gstaudiotestsrc.h:
28805 Fix wrong info in header due to copy & paste
28807 2006-09-15 14:53:44 +0000 Wim Taymans <wim.taymans@gmail.com>
28809 gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
28810 Original commit message from CVS:
28811 * gst-libs/gst/audio/gstbaseaudiosink.c:
28812 (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
28813 * gst-libs/gst/audio/gstbaseaudiosrc.c:
28814 (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
28815 (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
28816 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
28817 Do the delay calculation in the source/sink base classes as this is
28818 specific for the capture/playback mode.
28819 Try to fixate a bit better, like round depth up to a multiple of 8
28821 Handle underruns correctly by marking DISCONT on buffers and adjusting
28822 timestamps to handle the gap.
28823 Set offset/offset_end correctly on buffers.
28824 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
28825 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
28826 (gst_ring_buffer_read):
28827 Remove resync and underrun recovery from the ringbuffer.
28828 Fix ringbuffer read code on under/overrun.
28830 2006-09-15 11:17:02 +0000 Wim Taymans <wim.taymans@gmail.com>
28832 gst/playback/gstplaybasebin.*: Don't use a 0 low watermark when buffering, it is catching starvation way too late. In...
28833 Original commit message from CVS:
28834 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
28835 (gst_play_base_bin_init), (fill_buffer), (check_queue),
28836 (queue_threshold_reached), (gst_play_base_bin_set_property),
28837 (gst_play_base_bin_get_property):
28838 * gst/playback/gstplaybasebin.h:
28839 Don't use a 0 low watermark when buffering, it is catching starvation
28840 way too late. Instead, use a 3 second queue with 30 and 95
28841 percent low/high watermarks.
28842 Added queue-min-threshold property to configure low watermark.
28843 Use new _buffering message API.
28844 Make queue_threshold variable big enough to store a uint64 time value.
28845 API: playbin::queue-min-threshold property.
28847 2006-09-15 09:13:50 +0000 Wim Taymans <wim.taymans@gmail.com>
28849 configure.ac: We require 0.10.10.1 now because of _wait_preroll().
28850 Original commit message from CVS:
28852 We require 0.10.10.1 now because of _wait_preroll().
28853 * gst-libs/gst/audio/gstbaseaudiosink.c:
28854 (gst_base_audio_sink_render):
28855 Use gst_base_sink_wait_preroll().
28857 2006-09-15 09:09:00 +0000 Wim Taymans <wim.taymans@gmail.com>
28859 ext/alsa/: Use DEBUG_OBJECT more.
28860 Original commit message from CVS:
28861 * ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
28862 * ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
28863 Use DEBUG_OBJECT more.
28865 === release 0.10.10 ===
28867 2006-09-14 20:09:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28874 * docs/plugins/gst-plugins-base-plugins.args:
28875 * docs/plugins/inspect/plugin-adder.xml:
28876 * docs/plugins/inspect/plugin-alsa.xml:
28877 * docs/plugins/inspect/plugin-audioconvert.xml:
28878 * docs/plugins/inspect/plugin-audiorate.xml:
28879 * docs/plugins/inspect/plugin-audioresample.xml:
28880 * docs/plugins/inspect/plugin-audiotestsrc.xml:
28881 * docs/plugins/inspect/plugin-cdparanoia.xml:
28882 * docs/plugins/inspect/plugin-decodebin.xml:
28883 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
28884 * docs/plugins/inspect/plugin-gdp.xml:
28885 * docs/plugins/inspect/plugin-gnomevfs.xml:
28886 * docs/plugins/inspect/plugin-libvisual.xml:
28887 * docs/plugins/inspect/plugin-ogg.xml:
28888 * docs/plugins/inspect/plugin-pango.xml:
28889 * docs/plugins/inspect/plugin-playbin.xml:
28890 * docs/plugins/inspect/plugin-subparse.xml:
28891 * docs/plugins/inspect/plugin-tcp.xml:
28892 * docs/plugins/inspect/plugin-theora.xml:
28893 * docs/plugins/inspect/plugin-typefindfunctions.xml:
28894 * docs/plugins/inspect/plugin-video4linux.xml:
28895 * docs/plugins/inspect/plugin-videorate.xml:
28896 * docs/plugins/inspect/plugin-videoscale.xml:
28897 * docs/plugins/inspect/plugin-videotestsrc.xml:
28898 * docs/plugins/inspect/plugin-volume.xml:
28899 * docs/plugins/inspect/plugin-vorbis.xml:
28900 * docs/plugins/inspect/plugin-ximagesink.xml:
28901 * docs/plugins/inspect/plugin-xvimagesink.xml:
28902 * ext/theora/theoraparse.c:
28903 * gst-libs/gst/rtp/gstrtpbuffer.c:
28904 * gst/playback/gstplaybin.c:
28905 * tests/check/Makefile.am:
28906 * win32/common/config.h:
28908 Original commit message from CVS:
28911 2006-09-09 16:08:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28914 * win32/common/config.h:
28916 Original commit message from CVS:
28919 2006-09-07 19:01:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28922 update bug in changelog
28923 Original commit message from CVS:
28924 update bug in changelog
28926 2006-09-07 19:00:33 +0000 Michael Smith <msmith@fluendo.com>
28928 Fix implementation of sync-method 'next-keyframe'
28929 Original commit message from CVS:
28930 patch by: Michael Smith <msmith at fluendo dot com>
28931 * gst/tcp/gstmultifdsink.c: (is_sync_frame),
28932 (gst_multi_fd_sink_client_queue_buffer),
28933 (gst_multi_fd_sink_new_client):
28934 * tests/check/elements/multifdsink.c: (GST_START_TEST),
28935 (multifdsink_suite):
28936 Fix implementation of sync-method 'next-keyframe'
28938 2006-09-07 15:00:08 +0000 Wim Taymans <wim@fluendo.com>
28940 ext/gnomevfs/gstgnomevfssrc.c: This patch removes the RANDOM flag that was incorrectly introduced with revision 1.91....
28941 Original commit message from CVS:
28942 patch by: Wim Taymans <wim at fluendo dot com>
28943 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
28944 This patch removes the RANDOM flag that was incorrectly introduced with
28945 revision 1.91. Fixes #354590
28947 2006-09-07 14:56:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28950 * win32/common/config.h:
28952 Original commit message from CVS:
28955 2006-09-07 14:56:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28972 Original commit message from CVS:
28975 2006-09-05 09:12:25 +0000 Tim-Philipp Müller <tim@centricular.net>
28977 tests/check/Makefile.am: Random variation in Makefile line to see if it makes the gen64-base-full bot any happier.
28978 Original commit message from CVS:
28979 * tests/check/Makefile.am:
28980 Random variation in Makefile line to see if it makes the
28981 gen64-base-full bot any happier.
28983 2006-09-04 19:04:35 +0000 Tim-Philipp Müller <tim@centricular.net>
28985 tests/check/pipelines/oggmux.c: Disable test that fails at the moment (killed after timeout).
28986 Original commit message from CVS:
28987 * tests/check/pipelines/oggmux.c: (oggmux_suite):
28988 Disable test that fails at the moment (killed after timeout).
28990 2006-09-04 18:19:06 +0000 James Livingston <doclivingston@gmail.com>
28992 tests/check/: Add simple unit test for oggmux from #337026 with checking for the
28993 Original commit message from CVS:
28994 Patch by: James Livingston <doclivingston at gmail.com>
28995 * tests/check/Makefile.am:
28996 * tests/check/pipelines/.cvsignore:
28997 * tests/check/pipelines/oggmux.c: (get_page_codec),
28998 (check_chain_final_state), (fail_if_audio), (validate_ogg_page),
28999 (eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
29000 (test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
29001 (test_theora_vorbis), (oggmux_suite):
29002 Add simple unit test for oggmux from #337026 with checking for the
29003 EOS flags disabled for the time being.
29005 2006-09-04 09:13:01 +0000 Alessandro Dessina <alessandro@nnva.org>
29007 ext/ogg/gstoggmux.c: Add cmml caps to oggmux. Fixes #353912
29008 Original commit message from CVS:
29009 patch by: Alessandro Dessina <alessandro nnva org>
29010 * ext/ogg/gstoggmux.c:
29011 Add cmml caps to oggmux. Fixes #353912
29013 2006-09-02 13:20:59 +0000 Tim-Philipp Müller <tim@centricular.net>
29015 tests/check/elements/videotestsrc.c: Returning a return value often helps. In this case, we don't need the return val...
29016 Original commit message from CVS:
29017 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
29018 Returning a return value often helps. In this case, we
29019 don't need the return value anyway, so just get rid of it.
29020 Should make build bots much happier.
29022 2006-09-02 12:59:48 +0000 Tim-Philipp Müller <tim@centricular.net>
29024 gst/videotestsrc/videotestsrc.*: Add support for AYUV and the various RGBA formats. Initialise fields of paintinfo st...
29025 Original commit message from CVS:
29026 * gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
29027 (paint_get_structure), (gst_video_test_src_get_size),
29028 (gst_video_test_src_smpte), (gst_video_test_src_snow),
29029 (gst_video_test_src_unicolor), (paint_setup_AYUV),
29030 (paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
29031 (paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
29032 * gst/videotestsrc/videotestsrc.h:
29033 Add support for AYUV and the various RGBA formats. Initialise
29034 fields of paintinfo structs allocated on the stack.
29035 * tests/check/elements/videotestsrc.c: (right_shift_colour),
29036 (fix_expected_colour), (check_rgb_buf), (got_buf_cb),
29037 (GST_START_TEST), (videotestsrc_suite):
29038 Add unit tests for videotestsrc's RGB output.
29040 2006-09-01 16:12:35 +0000 Tim-Philipp Müller <tim@centricular.net>
29042 gst/videotestsrc/: Add more uni-colour patterns ("white", "red", "green", and "blue").
29043 Original commit message from CVS:
29044 * gst/videotestsrc/gstvideotestsrc.c:
29045 (gst_video_test_src_pattern_get_type),
29046 (gst_video_test_src_set_pattern):
29047 * gst/videotestsrc/gstvideotestsrc.h:
29048 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
29049 (gst_video_test_src_black), (gst_video_test_src_white),
29050 (gst_video_test_src_red), (gst_video_test_src_green),
29051 (gst_video_test_src_blue):
29052 * gst/videotestsrc/videotestsrc.h:
29053 Add more uni-colour patterns ("white", "red", "green", and "blue").
29055 2006-09-01 10:07:05 +0000 Tim-Philipp Müller <tim@centricular.net>
29057 gst/videotestsrc/videotestsrc.c: Fix stride for YVYU, should be word-aligned (#353658).
29058 Original commit message from CVS:
29059 * gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
29060 Fix stride for YVYU, should be word-aligned (#353658).
29062 2006-08-31 14:37:33 +0000 Tim-Philipp Müller <tim@centricular.net>
29064 gst/adder/gstadder.c: Fix build.
29065 Original commit message from CVS:
29066 * gst/adder/gstadder.c: (gst_adder_src_event):
29069 2006-08-31 12:39:17 +0000 Edward Hervey <bilboed@bilboed.com>
29071 gst/adder/gstadder.*: Remember the start position asked in the incoming seeks, so we can output GST_EVENT_NEW_SEGMENT...
29072 Original commit message from CVS:
29073 * gst/adder/gstadder.c: (forward_event_func),
29074 (gst_adder_src_event), (gst_adder_collected),
29075 (gst_adder_change_state):
29076 * gst/adder/gstadder.h:
29077 Remember the start position asked in the incoming seeks, so we can
29078 output GST_EVENT_NEW_SEGMENT with a correct position value (instead
29079 of assuming it will always be 0).
29081 2006-08-31 12:31:00 +0000 Edward Hervey <bilboed@bilboed.com>
29083 ext/ogg/gstoggdemux.c: Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
29084 Original commit message from CVS:
29085 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
29086 (gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
29087 (gst_ogg_demux_loop):
29088 Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
29090 2006-08-30 17:22:27 +0000 Tim-Philipp Müller <tim@centricular.net>
29092 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Return FALSE instead of returning a random false unit size when the forma...
29093 Original commit message from CVS:
29094 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
29095 (gst_ffmpegcsp_get_unit_size):
29096 Return FALSE instead of returning a random false unit
29097 size when the format isn't known/supported (even if
29098 this shouldn't happen under normal circumstances).
29100 2006-08-29 15:23:46 +0000 Tim-Philipp Müller <tim@centricular.net>
29102 ext/gnomevfs/gstgnomevfssrc.c: Try harder to get the size from a uri by using _info_uri() when _info_from_handle() do...
29103 Original commit message from CVS:
29104 Patch by: Tim-Philipp Müller <tim at centricular dot net>
29105 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
29106 (gst_gnome_vfs_src_start):
29107 Try harder to get the size from a uri by using _info_uri() when
29108 _info_from_handle() does not give us enough info.
29109 Also follow symlinks when getting the size.
29110 Partially Fixes #332864.
29112 2006-08-29 11:50:51 +0000 Viktor Peters <viktor.peters@gmail.com>
29114 ext/alsa/: Improve and fix mixer track handling, in particular better handling of alsa's pvolume/pswitch/cvolume/cswi...
29115 Original commit message from CVS:
29116 Patch by: Viktor Peters <viktor dot peters at gmail dot com>
29117 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
29118 (gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
29119 (gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
29120 (gst_alsa_mixer_set_record):
29121 * ext/alsa/gstalsamixertrack.c:
29122 (gst_alsa_mixer_track_update_alsa_capabilities),
29123 (alsa_track_has_cap), (gst_alsa_mixer_track_new),
29124 (gst_alsa_mixer_track_update):
29125 * ext/alsa/gstalsamixertrack.h:
29126 Improve and fix mixer track handling, in particular better handling
29127 of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
29128 track objects for tracks that have both capture and playback volume
29129 (and label them differently as well so they're not mistakenly
29130 assumed to be duplicates); classify mixer tracks that only affect
29131 the audible volume of something (rather than the capture volume)
29132 as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
29133 for capture tracks to correspond to alsa-pswitch alsa-cswitch
29134 (following the meaning documented in the mixer interface header
29135 file); add support for alsa's exclusive cswitch groups; update/sync
29136 state/flags better if mixer settings are changed by another
29137 application. Fixes #336075.
29139 2006-08-29 10:58:43 +0000 Tim-Philipp Müller <tim@centricular.net>
29141 gst/playback/gstplaybin.c: Improve docs: add section about BUFFERING messages sent by playbin.
29142 Original commit message from CVS:
29143 * gst/playback/gstplaybin.c:
29144 Improve docs: add section about BUFFERING messages sent by playbin.
29146 2006-08-29 10:51:12 +0000 Michael Smith <msmith@xiph.org>
29148 ext/vorbis/vorbisenc.c: Ignore explicit DISCONT marked on buffers (which is often spurious, particularly when using m...
29149 Original commit message from CVS:
29150 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
29151 (gst_vorbis_enc_buffer_check_discontinuous),
29152 (gst_vorbis_enc_chain):
29153 Ignore explicit DISCONT marked on buffers (which is often spurious,
29154 particularly when using multiple segments), in favour of solely
29155 using the timestamps/durations.
29157 2006-08-29 10:32:34 +0000 Edward Hervey <bilboed@bilboed.com>
29159 gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
29160 Original commit message from CVS:
29161 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
29162 Don't rely on incoming buffers offset anymore, since it is completely
29163 broken when using multiple segments.
29164 Instead convert the incoming buffers timestamp to running time, and
29165 then convert that value to the offsets.
29166 Also inform GstSegment of the last outputted stop position, which is
29167 needed if we received several segments with an unknown stop value.
29169 2006-08-29 08:03:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29171 ext/vorbis/vorbisenc.c: fix buffer unreffing on a header push failure
29172 Original commit message from CVS:
29173 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
29174 fix buffer unreffing on a header push failure
29176 2006-08-28 16:17:13 +0000 Wim Taymans <wim.taymans@gmail.com>
29178 gst/audiorate/gstaudiorate.c: Make the metadata of the buffer writable before changing its flags.
29179 Original commit message from CVS:
29180 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
29181 (gst_audio_rate_chain):
29182 Make the metadata of the buffer writable before changing its
29185 2006-08-28 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
29188 Fix changelog with bugzilla bug it fixed.
29189 Original commit message from CVS:
29190 Fix changelog with bugzilla bug it fixed.
29192 2006-08-28 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
29194 gst/audiorate/gstaudiorate.c: Fix audiorate some more.
29195 Original commit message from CVS:
29196 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
29197 (gst_audio_rate_setcaps), (gst_audio_rate_init),
29198 (gst_audio_rate_sink_event), (gst_audio_rate_src_event),
29199 (gst_audio_rate_chain), (gst_audio_rate_change_state):
29200 Fix audiorate some more.
29201 Reset and resync counters on flush and READY.
29202 Handle the DISCONT flag correctly.
29203 Use GstSegment to track position.
29204 Fail when not negotiated.
29206 2006-08-25 16:48:28 +0000 Michael Smith <msmith@xiph.org>
29208 gst/tcp/gstmultifdsink.c: Fix spelling.
29209 Original commit message from CVS:
29210 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
29212 Remove accidently included debug line.
29214 2006-08-25 16:39:38 +0000 Wim Taymans <wim.taymans@gmail.com>
29216 gst/tcp/gstmultifdsink.c: Small cleanups.
29217 Original commit message from CVS:
29218 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
29220 If a buffer is received with no caps, make the buffer metadata
29221 writable and set the caps, making sure that we don't screw up the
29224 2006-08-25 16:19:55 +0000 Michael Smith <msmith@xiph.org>
29226 gst/gdp/gstgdppay.c: Fix memory leaks and misleading debug messages, add a couple of comments.
29227 Original commit message from CVS:
29228 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
29229 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
29230 Fix memory leaks and misleading debug messages, add a couple of
29232 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
29233 (gst_multi_fd_sink_render):
29234 Do not use gst_buffer_make_writable() in a basesink render method,
29235 as it may incorrectly unref the buffer. Instead, use convoluted
29236 dance to avoid copying the buffer except when we need to.
29238 2006-08-25 09:54:56 +0000 Michael Smith <msmith@xiph.org>
29240 ext/vorbis/vorbisenc.c: Allow very small discontinuities in the timestamps. These we can't do anything useful with an...
29241 Original commit message from CVS:
29242 * ext/vorbis/vorbisenc.c:
29243 (gst_vorbis_enc_buffer_check_discontinuous):
29244 Allow very small discontinuities in the timestamps. These we can't
29245 do anything useful with anyway (because vorbis's timestamps have
29246 only sample granularity), and are commonly produced by elements with
29247 minor bugs. Allow up to 1/2 a sample out.
29250 2006-08-24 11:18:56 +0000 Wim Taymans <wim.taymans@gmail.com>
29252 tests/examples/seek/seek.c: Add a checkbox to enable play scrubbing. Makes it possible to disable normal scrubbing.
29253 Original commit message from CVS:
29254 * tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
29255 (play_scrub_toggle_cb), (main):
29256 Add a checkbox to enable play scrubbing. Makes it possible to disable
29259 2006-08-23 19:37:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29261 tests/check/elements/.cvsignore: make buildbot happy
29262 Original commit message from CVS:
29263 * tests/check/elements/.cvsignore:
29264 make buildbot happy
29266 2006-08-23 16:43:03 +0000 Tim-Philipp Müller <tim@centricular.net>
29268 ext/ogg/gstogmparse.c: Refactor ogm parse, do better input checking, misc. clean-ups.
29269 Original commit message from CVS:
29270 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
29271 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
29272 (gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
29273 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
29274 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
29275 (gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
29276 (gst_ogm_text_parse_strip_trailing_zeroes),
29277 (gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
29278 (gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
29279 Refactor ogm parse, do better input checking, misc. clean-ups.
29280 Cache incoming events and push them once the source pad has
29281 been created. Don't pass unterminated strings to sscanf().
29282 Strip trailing zeroes from subtitle text output, since they
29283 are not valid UTF-8. Don't push vorbiscomment packets on
29284 the subtitle text pad. Output perfect streams if possible.
29286 2006-08-23 15:27:38 +0000 Wim Taymans <wim.taymans@gmail.com>
29288 tests/check/libs/cddabasesrc.c: Waits for tasks to settle down so that we clean up correctly for valgrind.
29289 Original commit message from CVS:
29290 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
29291 Waits for tasks to settle down so that we clean up correctly for
29294 2006-08-23 15:11:56 +0000 Tim-Philipp Müller <tim@centricular.net>
29296 tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
29297 Original commit message from CVS:
29298 * tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
29299 Unit test fixes: \377 is more likely to fit into 8 bits than \777;
29300 actually return return value in taglists_are_equal.
29302 2006-08-23 12:14:20 +0000 Tim-Philipp Müller <tim@centricular.net>
29304 ext/ogg/gstogmparse.c: Fix crash due to broken bitstream parsing on x86-64: can't make any assumptions about sizeof(s...
29305 Original commit message from CVS:
29306 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
29307 Fix crash due to broken bitstream parsing on x86-64: can't make
29308 any assumptions about sizeof(struct) due to alignment/packing
29309 differences on different architectures. Fixes #351790.
29311 2006-08-22 16:31:47 +0000 Wim Taymans <wim.taymans@gmail.com>
29313 gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
29314 Original commit message from CVS:
29315 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
29316 (gst_riff_parse_chunk), (gst_riff_parse_file_header),
29317 (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
29318 (gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
29319 (gst_riff_parse_info):
29320 Protect public functions against bad input.
29324 2006-08-22 15:50:36 +0000 Tim-Philipp Müller <tim@centricular.net>
29326 gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
29327 Original commit message from CVS:
29328 * gst-libs/gst/riff/riff-ids.h:
29329 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
29330 Add voxware audio IDs (even if we can't play it) (#351795).
29332 2006-08-22 15:11:52 +0000 Tim-Philipp Müller <tim@centricular.net>
29334 gst-libs/gst/riff/riff-media.c: Const-ify some arrays and use G_N_ELEMENTS instead of wasting oodles of RAM on termin...
29335 Original commit message from CVS:
29336 * gst-libs/gst/riff/riff-media.c:
29337 (gst_riff_create_video_template_caps),
29338 (gst_riff_create_audio_template_caps),
29339 (gst_riff_create_iavs_template_caps):
29340 Const-ify some arrays and use G_N_ELEMENTS instead
29341 of wasting oodles of RAM on terminator bits.
29343 2006-08-22 08:27:07 +0000 Tim-Philipp Müller <tim@centricular.net>
29345 And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
29346 Original commit message from CVS:
29347 * gst-libs/gst/tag/gstvorbistag.c:
29348 (gst_tag_list_to_vorbiscomment_buffer):
29349 * tests/check/libs/tag.c: (GST_START_TEST):
29350 And the same for _to_vorbiscomment_buffer(): allow
29351 id_data_len == 0 for speex.
29353 2006-08-21 19:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29357 Original commit message from CVS:
29360 2006-08-21 19:01:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29362 Move GDP plugin to -base from -bad. Closes #347783.
29363 Original commit message from CVS:
29365 * docs/plugins/Makefile.am:
29366 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29367 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29368 * docs/plugins/inspect/plugin-gdp.xml:
29369 * gst/gdp/Makefile.am:
29370 * tests/check/Makefile.am:
29371 Move GDP plugin to -base from -bad. Closes #347783.
29373 2006-08-21 18:34:46 +0000 Tim-Philipp Müller <tim@centricular.net>
29375 gst-libs/gst/tag/gstvorbistag.c: Allow id_data_len == 0 (needed for vorbis comments in Speex files).
29376 Original commit message from CVS:
29377 * gst-libs/gst/tag/gstvorbistag.c:
29378 (gst_tag_list_from_vorbiscomment_buffer):
29379 Allow id_data_len == 0 (needed for vorbis comments in Speex files).
29380 Also add some checks to make sure we don't memcmp() beyond the end of
29381 vorbiscomment buffer if the ID to check for is larger than the buffer.
29382 * tests/check/libs/tag.c: (GST_START_TEST):
29383 Some more tests for gst_tag_list_from_vorbiscomment_buffer().
29385 2006-08-21 16:39:25 +0000 Tim-Philipp Müller <tim@centricular.net>
29387 ext/vorbis/vorbisenc.c: Use vorbis comment utility functions from libgsttag instead of re-inventing the wheel (partia...
29388 Original commit message from CVS:
29389 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
29390 (gst_vorbis_enc_set_metadata):
29391 Use vorbis comment utility functions from libgsttag
29392 instead of re-inventing the wheel (partially fixes #347091).
29394 2006-08-21 11:42:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29396 tests/check/elements/audioconvert.c: Fix leaks. Wait for state transitions that might happen ASYNC, as well as some t...
29397 Original commit message from CVS:
29398 * tests/check/elements/audioconvert.c: (GST_START_TEST):
29399 Fix leaks. Wait for state transitions that might happen ASYNC, as well
29400 as some that won't.
29402 2006-08-21 10:32:51 +0000 Wim Taymans <wim.taymans@gmail.com>
29404 docs/libs/: Don't try to GObject scan the netbuffer as it's not a GObject.
29405 Original commit message from CVS:
29406 * docs/libs/Makefile.am:
29407 * docs/libs/gst-plugins-base-libs-sections.txt:
29408 * docs/libs/gst-plugins-base-libs.types:
29409 Don't try to GObject scan the netbuffer as it's not a GObject.
29411 * gst-libs/gst/netbuffer/gstnetbuffer.c:
29412 * gst-libs/gst/netbuffer/gstnetbuffer.h:
29413 Document GstNetBuffer.
29415 2006-08-21 08:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29417 tests/check/elements/audioconvert.c: Add testcase for caps-size-explosion
29418 Original commit message from CVS:
29419 * tests/check/elements/audioconvert.c: (GST_START_TEST),
29420 (audioconvert_suite):
29421 Add testcase for caps-size-explosion
29423 2006-08-20 13:05:44 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29425 gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size
29426 Original commit message from CVS:
29427 * gst/audioconvert/gstaudioconvert.c:
29428 (gst_audio_convert_get_unit_size), (set_structure_widths):
29429 Lower debug, use g_assert in _get_unit_size
29430 * gst/audioresample/gstaudioresample.c:
29431 (audioresample_get_unit_size):
29432 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
29433 (gst_ffmpegcsp_get_unit_size):
29434 * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
29435 use g_assert in _get_unit_size
29437 2006-08-18 21:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
29440 ChangeLog surgery: fix bug number
29441 Original commit message from CVS:
29442 ChangeLog surgery: fix bug number
29444 2006-08-18 16:43:26 +0000 Wim Taymans <wim.taymans@gmail.com>
29446 Document GstRTPBuffer.
29447 Original commit message from CVS:
29448 * docs/libs/gst-plugins-base-libs-sections.txt:
29449 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
29450 (gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
29451 (gst_rtp_buffer_get_payload_buffer):
29452 * gst-libs/gst/rtp/gstrtpbuffer.h:
29453 Document GstRTPBuffer.
29454 Added function to efficiently strip payload headers.
29455 API: gst_rtp_buffer_get_payload_subbuffer()
29457 2006-08-17 16:52:06 +0000 Tim-Philipp Müller <tim@centricular.net>
29459 gst-libs/gst/tag/gstvorbistag.c: Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT tags and deserialise...
29460 Original commit message from CVS:
29461 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
29462 (gst_tag_to_vorbis_comments):
29463 Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
29464 tags and deserialise them properly as well (#351768).
29465 Add some more gtk-doc blurbs and also some g_return_if_fail().
29466 * tests/check/libs/tag.c: (GST_START_TEST),
29467 (back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
29470 2006-08-17 15:43:40 +0000 Wim Taymans <wim.taymans@gmail.com>
29472 ext/ogg/: Added ogg-in-avi parser element. Fixes #140139.
29473 Original commit message from CVS:
29474 * ext/ogg/Makefile.am:
29475 * ext/ogg/gstogg.c: (plugin_init):
29476 * ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
29477 (gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
29478 (gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
29479 (gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
29480 (gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
29481 (gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
29482 Added ogg-in-avi parser element. Fixes #140139.
29483 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
29484 Fixed a bug in oggdemux debug code.
29485 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
29486 (gst_riff_create_audio_template_caps):
29487 Recognise Ogg in the AVI extensible wave format.
29489 2006-08-17 10:00:00 +0000 Tim-Philipp Müller <tim@centricular.net>
29491 gst-libs/gst/cdda/gstcddabasesrc.c: Make buffer durations add up (duration should be next_ts-ts for perfect streams)....
29492 Original commit message from CVS:
29493 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
29494 Make buffer durations add up (duration should be next_ts-ts for
29495 perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
29497 * tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
29498 (test_buffer_timestamps), (cddabasesrc_suite):
29499 Add unit test for the above.
29500 * tests/check/Makefile.am:
29501 Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
29502 to see what happens.
29504 2006-08-16 11:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
29506 ext/alsa/: Avoid setting and using a NULL device name.
29507 Original commit message from CVS:
29508 * ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
29509 (gst_alsasink_open):
29510 * ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
29511 (gst_alsasrc_open):
29512 Avoid setting and using a NULL device name.
29513 Print more info when we fail to open a device.
29515 2006-08-16 11:28:57 +0000 Tim-Philipp Müller <tim@centricular.net>
29517 API: add gst_tag_parse_extended_comment() (#351426).
29518 Original commit message from CVS:
29519 * docs/libs/gst-plugins-base-libs-sections.txt:
29520 * gst-libs/gst/tag/tag.h:
29521 * gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
29522 API: add gst_tag_parse_extended_comment() (#351426).
29523 * tests/check/Makefile.am:
29524 * tests/check/libs/.cvsignore:
29525 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
29526 Add unit test for gst_tag_parse_extended_comment().
29528 2006-08-15 19:20:16 +0000 Tim-Philipp Müller <tim@centricular.net>
29530 sys/: Fix leak (#351502).
29531 Original commit message from CVS:
29532 * sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
29533 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
29534 Fix leak (#351502).
29536 2006-08-15 17:21:33 +0000 Tim-Philipp Müller <tim@centricular.net>
29539 Original commit message from CVS:
29540 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29541 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29542 * docs/plugins/gst-plugins-base-plugins.args:
29543 * gst/playback/gstplaybin.c:
29545 * docs/plugins/inspect/plugin-adder.xml:
29546 * docs/plugins/inspect/plugin-alsa.xml:
29547 * docs/plugins/inspect/plugin-audioconvert.xml:
29548 * docs/plugins/inspect/plugin-audiorate.xml:
29549 * docs/plugins/inspect/plugin-audioresample.xml:
29550 * docs/plugins/inspect/plugin-audiotestsrc.xml:
29551 * docs/plugins/inspect/plugin-cdparanoia.xml:
29552 * docs/plugins/inspect/plugin-decodebin.xml:
29553 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
29554 * docs/plugins/inspect/plugin-gnomevfs.xml:
29555 * docs/plugins/inspect/plugin-ogg.xml:
29556 * docs/plugins/inspect/plugin-pango.xml:
29557 * docs/plugins/inspect/plugin-playbin.xml:
29558 * docs/plugins/inspect/plugin-subparse.xml:
29559 * docs/plugins/inspect/plugin-tcp.xml:
29560 * docs/plugins/inspect/plugin-theora.xml:
29561 * docs/plugins/inspect/plugin-typefindfunctions.xml:
29562 * docs/plugins/inspect/plugin-video4linux.xml:
29563 * docs/plugins/inspect/plugin-videorate.xml:
29564 * docs/plugins/inspect/plugin-videoscale.xml:
29565 * docs/plugins/inspect/plugin-videotestsrc.xml:
29566 * docs/plugins/inspect/plugin-volume.xml:
29567 * docs/plugins/inspect/plugin-vorbis.xml:
29568 * docs/plugins/inspect/plugin-ximagesink.xml:
29569 * docs/plugins/inspect/plugin-xvimagesink.xml:
29570 Update to CVS version.
29572 2006-08-14 17:54:01 +0000 Tim-Philipp Müller <tim@centricular.net>
29574 gst/playback/gstplaybin.c: Add "connection-speed" property; re-order redirect messages with multiple redirect locatio...
29575 Original commit message from CVS:
29576 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
29577 (gst_play_bin_set_property), (gst_play_bin_get_property),
29578 (value_list_append_structure_list),
29579 (gst_play_bin_handle_redirect_message),
29580 (gst_play_bin_handle_message):
29581 Add "connection-speed" property; re-order redirect messages with
29582 multiple redirect locations depending on the minimum bitrate if
29583 that information is available and a connection speed is set
29586 2006-08-14 11:41:04 +0000 Tim-Philipp Müller <tim@centricular.net>
29588 gst/playback/gstplaybin.c: Update max volume to the same value that the volume element uses.
29589 Original commit message from CVS:
29590 * gst/playback/gstplaybin.c:
29591 Update max volume to the same value that the volume element uses.
29593 2006-08-14 10:50:15 +0000 Wim Taymans <wim.taymans@gmail.com>
29595 ext/alsa/gstalsamixer.c: Less uglyness..
29596 Original commit message from CVS:
29597 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
29600 2006-08-14 10:49:10 +0000 Wim Taymans <wim.taymans@gmail.com>
29602 ext/ogg/gstoggdemux.c: Add some more debug info.
29603 Original commit message from CVS:
29604 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
29605 (gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
29606 (gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
29607 Add some more debug info.
29608 Don't crash when a seek failed.
29609 Actually return the result of the seek instead of TRUE.
29610 Ignore multiple BOS pages with the same serial so that we don't create
29611 the same stream multiple times.
29612 Post an error when we fail to do the initial seek.
29614 2006-08-13 14:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
29616 ext/alsa/gstalsa.c: Small code cleanup.
29617 Original commit message from CVS:
29618 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
29619 (gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
29620 Small code cleanup.
29621 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
29622 (gst_alsa_mixer_new):
29623 Remove hack that always set the device to hw:0*.
29624 Properly find the card name for whatever device was configured.
29625 Do some better debugging.
29627 * ext/alsa/gstalsamixerelement.c:
29628 (gst_alsa_mixer_element_set_property),
29629 (gst_alsa_mixer_element_change_state):
29631 Handle setting of a NULL device name better.
29633 2006-08-11 15:53:43 +0000 Wim Taymans <wim.taymans@gmail.com>
29635 gst/adder/gstadder.c: Don't clip float values. Fixes #350900.
29636 Original commit message from CVS:
29637 * gst/adder/gstadder.c:
29638 Don't clip float values. Fixes #350900.
29640 2006-08-11 15:33:17 +0000 Andy Wingo <wingo@pobox.com>
29642 gst/tcp/gsttcp.c: Really fix the build?
29643 Original commit message from CVS:
29644 2006-08-11 Andy Wingo <wingo@pobox.com>
29645 * gst/tcp/gsttcp.c: Really fix the build?
29647 2006-08-11 15:29:56 +0000 Andy Wingo <wingo@pobox.com>
29649 gst/tcp/gsttcp.h: For now, always disable deprecation here -- fixes the build.
29650 Original commit message from CVS:
29651 2006-08-11 Andy Wingo <wingo@pobox.com>
29652 * gst/tcp/gsttcp.h: For now, always disable deprecation here --
29655 2006-08-10 13:01:31 +0000 Tim-Philipp Müller <tim@centricular.net>
29657 gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field.
29658 Original commit message from CVS:
29659 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
29660 Float caps shouldn't have a "signed" field.
29662 2006-08-10 08:56:22 +0000 Tim-Philipp Müller <tim@centricular.net>
29664 ext/ogg/gstoggdemux.c: Implement SEEKING query in its most basic form, so that we can at least check if we're seekabl...
29665 Original commit message from CVS:
29666 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
29667 Implement SEEKING query in its most basic form, so that we can
29668 at least check if we're seekable or not (#350655).
29670 2006-08-09 14:42:58 +0000 Tim-Philipp Müller <tim@centricular.net>
29672 gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
29673 Original commit message from CVS:
29674 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
29675 The checks here are not even close to anything that would
29676 justify MAXIMUM probability, lowering to POSSIBLE until someone
29677 fixes the checks (case at hand: quicktime redirection files
29678 might start with 00 00 01 XX and pass the checks here just
29679 fine, see #350399).
29681 2006-08-08 13:57:29 +0000 Edward Hervey <bilboed@bilboed.com>
29683 tests/check/elements/gdpdepay.c: I forgot to include the file containing the #define :)
29684 Original commit message from CVS:
29685 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
29686 I forgot to include the file containing the #define :)
29687 Now includes "config.h"
29689 2006-08-08 13:45:44 +0000 Edward Hervey <bilboed@bilboed.com>
29691 tests/check/elements/gdpdepay.c: Ignore test known to fail on PPC64. See #348114.
29692 Original commit message from CVS:
29693 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
29694 Ignore test known to fail on PPC64. See #348114.
29696 2006-08-08 08:41:13 +0000 Sjoerd Simons <sjoerd@luon.net>
29698 gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
29699 Original commit message from CVS:
29700 Patch by: Sjoerd Simons <sjoerd at luon net>
29701 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
29702 Better detection for multipart/x-mixed-replace: accept leading
29703 whitespaces before the boundary marker as well (as our very own
29704 multipartmux used to produce) (#349068).
29706 2006-08-07 08:26:03 +0000 Young-Ho Cha <ganadist@chollian.net>
29708 gst-libs/gst/riff/: Detect DTS audio streams (#350157).
29709 Original commit message from CVS:
29710 Patch by: Young-Ho Cha <ganadist at chollian net>
29711 * gst-libs/gst/riff/riff-ids.h:
29712 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
29713 (gst_riff_create_audio_template_caps):
29714 Detect DTS audio streams (#350157).
29716 2006-08-05 17:08:05 +0000 Andy Wingo <wingo@pobox.com>
29718 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (gst_theora_parse_class_init) (theora_parse_dispose, theora_par...
29719 Original commit message from CVS:
29720 2006-08-05 Andy Wingo <wingo@pobox.com>
29721 * ext/theora/gsttheoraparse.h:
29722 * ext/theora/theoraparse.c (gst_theora_parse_class_init)
29723 (theora_parse_dispose, theora_parse_set_property)
29724 (theora_parse_get_property, theora_parse_munge_granulepos)
29725 (theora_parse_push_buffer, theora_parse_change_state): Add a
29726 property 'synchronization-points' to fix badly synchronized oggs.
29728 2006-08-04 13:20:23 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
29730 gst/gdp/gstgdpdepay.c: Fix event parsing by gdpdepay. Fixes #349916.
29731 Original commit message from CVS:
29732 2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
29733 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
29734 Fix event parsing by gdpdepay. Fixes #349916.
29736 2006-08-03 15:04:42 +0000 Tim-Philipp Müller <tim@centricular.net>
29738 tests/check/: Add a few tests for the channel position stuff in libgstaudio.
29739 Original commit message from CVS:
29740 * tests/check/Makefile.am:
29741 * tests/check/libs/.cvsignore:
29742 * tests/check/libs/audio.c: (structure_contains_channel_positions),
29743 (fixed_caps_have_channel_positions), (GST_START_TEST),
29744 (audio_suite), (main):
29745 Add a few tests for the channel position stuff in libgstaudio.
29747 2006-08-03 14:16:06 +0000 Tim-Philipp Müller <tim@centricular.net>
29749 ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
29750 Original commit message from CVS:
29751 * ext/alsa/gstalsa.c: (caps_add_channel_configuration),
29752 (gst_alsa_detect_channels):
29753 * ext/alsa/gstalsasink.c:
29754 Add support for cards that (only) do more than 8 channels,
29755 like the Delta 44 (#345188).
29756 * gst-libs/gst/audio/multichannel.c:
29757 (gst_audio_check_channel_positions):
29758 * gst-libs/gst/audio/multichannel.h:
29759 API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
29760 unspecified channel position and cannot be combined with any
29761 of the other audio channel positions; adjust position layout
29762 checks accordingly (#345188).
29764 2006-08-03 11:15:53 +0000 Tim-Philipp Müller <tim@centricular.net>
29766 gst/typefind/gsttypefindfunctions.c: Recognise ancient RealAudio files (see #349779).
29767 Original commit message from CVS:
29768 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29769 Recognise ancient RealAudio files (see #349779).
29771 2006-08-03 09:01:25 +0000 Jens Granseuer <jensgr@gmx.net>
29773 gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
29774 Original commit message from CVS:
29775 Patch by: Jens Granseuer <jensgr at gmx net>
29776 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29777 Add typefinder for Interplay's MVE format (#348973).
29779 2006-08-02 17:03:29 +0000 Marcel Moreaux <marcelm@luon.net>
29781 gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
29782 Original commit message from CVS:
29783 Patch by: Marcel Moreaux <marcelm at luon dot net>
29784 * gst-libs/gst/rtp/gstbasertpdepayload.c:
29785 (gst_base_rtp_depayload_add_to_queue):
29786 * gst-libs/gst/rtp/gstbasertpdepayload.h:
29787 Handle RTP sequence number rollover.
29788 Disable jitterbuffer by default.
29790 2006-08-02 16:56:19 +0000 Wim Taymans <wim.taymans@gmail.com>
29792 gst/gdp/gstgdpdepay.c: Disable seeking.
29793 Original commit message from CVS:
29794 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
29795 (gst_gdp_depay_finalize), (gst_gdp_depay_sink_event),
29796 (gst_gdp_depay_src_event), (gst_gdp_depay_chain),
29797 (gst_gdp_depay_change_state):
29800 Clear adapter on disconts.
29801 Clear caps when going to READY instead of NULL
29802 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
29803 (gst_gdp_pay_init), (gst_gdp_pay_finalize), (gst_gdp_pay_reset),
29804 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
29805 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
29806 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
29807 (gst_gdp_pay_sink_event), (gst_gdp_pay_src_event),
29808 (gst_gdp_pay_change_state):
29809 * gst/gdp/gstgdppay.h:
29810 Reset payloader when going to READY.
29811 Fix leaked buffers in ->queue on push errors.
29814 Create packetizer in _init, free in _finalize.
29816 2006-07-31 08:48:36 +0000 Wim Taymans <wim.taymans@gmail.com>
29818 gst/gdp/gstgdpdepay.c: Consume all events except EOS because we generate events from the gdp payload instead. Fixes #...
29819 Original commit message from CVS:
29820 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
29821 (gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
29822 Consume all events except EOS because we generate events from
29823 the gdp payload instead. Fixes #349204
29825 2006-07-28 17:17:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29827 gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
29828 Original commit message from CVS:
29829 * gst/audioresample/gstaudioresample.c: (audioresample_stop),
29830 (audioresample_set_caps):
29831 Don't leak references to the incoming caps. Clean them up when
29833 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
29834 (gst_video_scale_finalize):
29835 Don't leak our temporary pixel buffer.
29836 * tests/check/Makefile.am:
29837 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
29838 (GST_START_TEST), (simple_launch_lines_suite):
29839 Fix leaks and re-enable the test for valgrind checking.
29841 2006-07-28 16:39:31 +0000 Sjoerd Simons <sjoerd@luon.net>
29843 gst/typefind/gsttypefindfunctions.c: Add typefind function for multipart/x-mixed-replace (#348916).
29844 Original commit message from CVS:
29845 Patch by: Sjoerd Simons <sjoerd at luon net>
29846 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
29848 Add typefind function for multipart/x-mixed-replace (#348916).
29850 2006-07-28 14:14:58 +0000 Wim Taymans <wim.taymans@gmail.com>
29852 gst/adder/gstadder.c: Fix leak in duration query.
29853 Original commit message from CVS:
29854 * gst/adder/gstadder.c: (gst_adder_setcaps),
29855 (gst_adder_query_duration):
29856 Fix leak in duration query.
29857 Reflow some docs and notes.
29859 2006-07-28 13:42:48 +0000 Michael Smith <msmith@xiph.org>
29861 tests/check/pipelines/vorbisenc.c: Enable Andy's extra vorbisenc test, now that it passes. Also fix one aspect of it.
29862 Original commit message from CVS:
29863 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
29865 Enable Andy's extra vorbisenc test, now that it passes. Also fix one
29868 2006-07-28 12:48:21 +0000 Michael Smith <msmith@xiph.org>
29870 ext/vorbis/vorbisenc.*: Handle discontinuities in the input vorbis stream correctly, so that the output is properly t...
29871 Original commit message from CVS:
29872 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
29873 (gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
29874 (gst_vorbis_enc_push_buffer),
29875 (gst_vorbis_enc_buffer_check_discontinuous),
29876 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
29877 * ext/vorbis/vorbisenc.h:
29878 Handle discontinuities in the input vorbis stream correctly,
29879 so that the output is properly timestamped (and has good granulepos
29880 values). Needs some oggmux fixes too.
29882 2006-07-27 10:52:52 +0000 Kai Vehmanen <kv2004@eca.cx>
29884 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
29885 Original commit message from CVS:
29886 patch by: Kai Vehmanen <kv2004 eca cx>
29887 * gst-libs/gst/rtp/gstbasertpdepayload.c:
29888 (gst_base_rtp_depayload_chain),
29889 (gst_base_rtp_depayload_handle_sink_event),
29890 (gst_base_rtp_depayload_change_state):
29891 Don't send multiple newsegments with different formats.
29894 2006-07-26 15:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
29896 ext/ogg/gstoggdemux.c: Make seeking in ogg more accurate again by doing the more correct granuletime to stream time c...
29897 Original commit message from CVS:
29898 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
29899 (gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
29900 Make seeking in ogg more accurate again by doing the more correct
29901 granuletime to stream time conversion.
29903 2006-07-26 10:59:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29905 gst/tcp/gstmultifdsink.c: debug a little more understandably do not use goto as a substitute for break, especially if...
29906 Original commit message from CVS:
29907 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
29908 (gst_multi_fd_sink_new_client):
29909 debug a little more understandably
29910 do not use goto as a substitute for break, especially if
29911 break is also being used
29913 2006-07-26 10:55:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29915 * gst/tcp/gsttcp.c:
29916 move a recurring normal event to LOG, where it should be
29917 Original commit message from CVS:
29918 move a recurring normal event to LOG, where it should be
29920 2006-07-26 10:54:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29922 * ext/vorbis/vorbisdec.c:
29924 Original commit message from CVS:
29927 2006-07-26 10:52:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29929 gst/gdp/gstgdpdepay.c: proxying get/set caps is the wrong thing to do, since we really do change caps quite fundament...
29930 Original commit message from CVS:
29931 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
29932 proxying get/set caps is the wrong thing to do, since we really
29933 do change caps quite fundamentally
29934 * tests/check/elements/gdpdepay.c:
29935 * tests/check/elements/gdppay.c:
29936 remove declaration of buffers, it's already done in gstcheck.h
29938 2006-07-26 10:31:06 +0000 Tim-Philipp Müller <tim@centricular.net>
29940 gst/playback/: Remove GLib-2.6 compatibility cruft.
29941 Original commit message from CVS:
29942 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
29943 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
29944 Remove GLib-2.6 compatibility cruft.
29946 2006-07-24 16:47:10 +0000 Wim Taymans <wim.taymans@gmail.com>
29948 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
29949 Original commit message from CVS:
29950 * gst-libs/gst/audio/gstbaseaudiosink.c:
29951 (gst_base_audio_sink_render):
29952 Don't try to align a sample to an unknown value.
29954 2006-07-24 15:14:17 +0000 Wim Taymans <wim.taymans@gmail.com>
29956 gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
29957 Original commit message from CVS:
29958 * gst-libs/gst/audio/gstbaseaudiosink.c:
29959 (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
29960 When the audio clock is slaved to another clock, never try to align
29961 samples but trust the rate interpolation algorithm.
29963 2006-07-24 14:34:42 +0000 Wim Taymans <wim.taymans@gmail.com>
29965 ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
29966 Original commit message from CVS:
29967 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
29968 Don't try to calculate silence samples, base class does this much
29970 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
29971 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
29972 (gst_ring_buffer_acquire):
29973 Calculate silence samples correctly.
29974 * gst-libs/gst/audio/gstringbuffer.h:
29977 2006-07-22 17:01:12 +0000 Tim-Philipp Müller <tim@centricular.net>
29979 gst/typefind/gsttypefindfunctions.c: Limit search for the first markup tag to the first few kB of the file. If we don...
29980 Original commit message from CVS:
29981 * gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
29982 Limit search for the first markup tag to the first few kB of
29983 the file. If we don't find one there, it's highly unlikely that
29984 this is an XML(-ish) file.
29986 2006-07-21 17:04:06 +0000 Andy Wingo <wingo@pobox.com>
29988 tests/check/pipelines/theoraenc.c (test_discontinuity): Similar test to the one in vorbisenc. Also commented out.
29989 Original commit message from CVS:
29990 2006-07-21 Andy Wingo <wingo@pobox.com>
29991 * tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
29992 test to the one in vorbisenc. Also commented out.
29994 2006-07-21 16:54:19 +0000 Andy Wingo <wingo@pobox.com>
29996 tests/check/pipelines/vorbisenc.c: New test, commented out until Mike lands some elite vorbisenc patches.
29997 Original commit message from CVS:
29998 2006-07-21 Andy Wingo <wingo@pobox.com>
29999 * tests/check/pipelines/vorbisenc.c:
30000 (test_discontinuity): New test, commented out until Mike lands
30001 some elite vorbisenc patches.
30003 2006-07-21 15:59:24 +0000 Andy Wingo <wingo@pobox.com>
30005 tests/check/pipelines/: Port to bufferstraw.
30006 Original commit message from CVS:
30007 2006-07-21 Andy Wingo <wingo@pobox.com>
30008 * tests/check/pipelines/vorbisenc.c:
30009 * tests/check/pipelines/theoraenc.c: Port to bufferstraw.
30010 Bufferstraw was actually factored out of these tests. Now we share
30013 2006-07-21 11:03:28 +0000 Wim Taymans <wim.taymans@gmail.com>
30015 ext/theora/theoradec.c: Better clipping.
30016 Original commit message from CVS:
30017 * ext/theora/theoradec.c: (clip_buffer):
30020 2006-07-21 10:43:54 +0000 Wim Taymans <wim.taymans@gmail.com>
30022 gst-libs/gst/audio/gstaudiosink.c: Fix leak.
30023 Original commit message from CVS:
30024 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
30025 (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
30026 (gst_audioringbuffer_release), (gst_audioringbuffer_stop):
30028 Avoid type casting when we can.
30029 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
30032 2006-07-20 16:57:29 +0000 Tim-Philipp Müller <tim@centricular.net>
30034 ext/alsa/gstalsamixerelement.c: Make state change fail if the specified device can't be opened for some reason.
30035 Original commit message from CVS:
30036 * ext/alsa/gstalsamixerelement.c:
30037 (gst_alsa_mixer_element_change_state):
30038 Make state change fail if the specified device can't be opened
30041 2006-07-20 10:42:21 +0000 Wim Taymans <wim.taymans@gmail.com>
30043 gst/playback/test.c: Example of a small audio/video player using decodebin.
30044 Original commit message from CVS:
30045 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
30046 (cb_newpad), (main):
30047 Example of a small audio/video player using decodebin.
30049 2006-07-20 05:56:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30051 gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
30052 Original commit message from CVS:
30053 * gst-libs/gst/riff/riff-ids.h:
30054 Add 'fact' chunk id
30056 2006-07-19 18:20:43 +0000 Wim Taymans <wim.taymans@gmail.com>
30058 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't assert when not negotiated but post a meaningfull error message. Fixes ...
30059 Original commit message from CVS:
30060 * gst-libs/gst/rtp/gstbasertpdepayload.c:
30061 (gst_base_rtp_depayload_chain),
30062 (gst_base_rtp_depayload_change_state):
30063 Don't assert when not negotiated but post a meaningfull
30064 error message. Fixes #347918.
30065 * gst-libs/gst/rtp/gstbasertppayload.c:
30066 Add comment about better default MTU size.
30067 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
30068 Small cleanups, start docs.
30070 2006-07-19 14:46:36 +0000 Martin Szulecki <compiz@sukimashita.com>
30072 sys/v4l/gstv4lelement.c: If "device-name" is requested and the device is not open, try to temporarily open it to obta...
30073 Original commit message from CVS:
30074 Patch by: Martin Szulecki
30075 * sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
30076 If "device-name" is requested and the device is not
30077 open, try to temporarily open it to obtain this
30078 information (#342494).
30080 2006-07-19 12:25:00 +0000 Tim-Philipp Müller <tim@centricular.net>
30082 gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
30083 Original commit message from CVS:
30084 * gst-libs/gst/tag/gstid3tag.c:
30085 Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
30086 * gst-libs/gst/tag/gsttageditingprivate.h:
30087 * gst-libs/gst/tag/gstvorbistag.c:
30088 Some more random const-ifications.
30090 2006-07-18 19:48:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30092 gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
30093 Original commit message from CVS:
30094 * gst-libs/gst/riff/riff-ids.h:
30095 * gst-libs/gst/riff/riff-media.c:
30096 (gst_riff_create_video_template_caps):
30097 Add more FOURCCs (sort list to make stuff easier to find),
30098 add comment what those 16 bytes in struct _gst_riff_strh according to
30101 2006-07-17 14:17:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30103 gst/gdp/: remove parent_class setting, BOILERPLATE does this fix typo in comment
30104 Original commit message from CVS:
30105 2006-07-17 Thomas Vander Stichele <thomas at apestaart dot org>
30106 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
30107 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
30108 remove parent_class setting, BOILERPLATE does this
30109 (gst_gdp_pay_reset_streamheader):
30110 fix typo in comment
30112 2006-07-17 13:48:10 +0000 Tim-Philipp Müller <tim@centricular.net>
30114 gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
30115 Original commit message from CVS:
30116 * gst-libs/gst/audio/multichannel.c:
30117 (gst_audio_check_channel_positions),
30118 (gst_audio_fixate_channel_positions):
30119 Const-ify two arrays.
30121 2006-07-17 12:33:42 +0000 Tim-Philipp Müller <tim@centricular.net>
30123 ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
30124 Original commit message from CVS:
30125 * ext/alsa/gstalsa.c: (caps_add_channel_configuration):
30126 Fix typo, so that alsasink also advertises 8 channels
30127 if that's supported (tags: can, worms, open, alsa, ph34r).
30129 2006-07-17 12:01:04 +0000 Wim Taymans <wim.taymans@gmail.com>
30131 ext/ogg/gstoggdemux.c: *sigh*, when is the compiler going to warn when the comments are out-of-sync with the code.. R...
30132 Original commit message from CVS:
30133 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
30134 (gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
30135 *sigh*, when is the compiler going to warn when the comments
30136 are out-of-sync with the code.. Refix case of busted theora
30137 headers with 0 granule pos.
30139 2006-07-14 17:56:59 +0000 Wim Taymans <wim.taymans@gmail.com>
30141 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix 99% cpu load by waiting for absolute times on the clock. Fixes #347300.
30142 Original commit message from CVS:
30143 * gst-libs/gst/rtp/gstbasertpdepayload.c:
30144 (gst_base_rtp_depayload_wait),
30145 (gst_base_rtp_depayload_change_state),
30146 (gst_base_rtp_depayload_set_property),
30147 (gst_base_rtp_depayload_get_property):
30148 Fix 99% cpu load by waiting for absolute times on the
30149 clock. Fixes #347300.
30151 2006-07-14 17:07:08 +0000 Andy Wingo <wingo@pobox.com>
30153 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (theora_parse_drain_event_queue) (theora_parse_push_headers, th...
30154 Original commit message from CVS:
30155 2006-07-14 Andy Wingo <wingo@pobox.com>
30156 * ext/theora/gsttheoraparse.h:
30157 * ext/theora/theoraparse.c (theora_parse_drain_event_queue)
30158 (theora_parse_push_headers, theora_parse_clear_queue)
30159 (theora_parse_drain_queue_prematurely, )
30160 (theora_parse_sink_event, theora_parse_change_state): Queue events
30161 until we initialized our state, like in vorbisparse.
30163 2006-07-14 16:45:17 +0000 Iain * <iaingnome@gmail.com>
30165 ext/vorbis/vorbisparse.h: ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue) (vorbis_parse_push_headers, vorbi...
30166 Original commit message from CVS:
30167 2006-07-14 Andy Wingo <wingo@pobox.com>
30168 * ext/vorbis/vorbisparse.h:
30169 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
30170 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
30171 (vorbis_parse_drain_queue_prematurely, )
30172 (vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
30173 until we have initialized our state. Fixes seeking after an
30175 2006-07-14 Andy Wingo <wingo@pobox.com>
30176 Patch by: Iain * <iaingnome@gmail.com>
30177 * ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.
30179 2006-07-14 15:52:39 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30181 configure.ac: Bump nano back to CVS
30182 Original commit message from CVS:
30184 Bump nano back to CVS
30186 === release 0.10.9 ===
30188 2006-07-14 15:51:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30190 configure.ac: releasing 0.10.9, "I walk the line"
30191 Original commit message from CVS:
30192 2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
30194 releasing 0.10.9, "I walk the line"
30196 2006-07-14 14:12:40 +0000 Michael Smith <msmith@xiph.org>
30198 tests/check/pipelines/vorbisenc.c: Move a g_cond_signal to earlier to avoid sometimes deadlocking (commonly happens w...
30199 Original commit message from CVS:
30200 * tests/check/pipelines/vorbisenc.c: (stop_pipeline):
30201 Move a g_cond_signal to earlier to avoid sometimes deadlocking
30202 (commonly happens when running this test under valgrind) when trying
30203 to remove the buffer probe.
30205 2006-07-14 10:34:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30207 * gst/gdp/Makefile.am:
30208 build as a plugin, not a lib
30209 Original commit message from CVS:
30210 build as a plugin, not a lib
30212 2006-07-13 16:43:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30214 sys/xvimage/xvimagesink.c: Fix missing g_unlock from the previous commit
30215 Original commit message from CVS:
30216 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
30217 Fix missing g_unlock from the previous commit
30219 2006-07-13 16:34:04 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30221 sys/: Implement a locking order to ensure we always take the object lock before the x_lock and never vice-versa.
30222 Original commit message from CVS:
30223 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
30224 (gst_ximagesink_change_state):
30225 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
30226 (gst_xvimagesink_change_state):
30227 Implement a locking order to ensure we always take the object lock
30228 before the x_lock and never vice-versa.
30230 2006-07-13 15:25:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30232 docs/plugins/: add more plugins and elements to docs
30233 Original commit message from CVS:
30234 * docs/plugins/Makefile.am:
30235 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
30236 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
30237 add more plugins and elements to docs
30238 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
30239 fix segfaults due to wrong g_free
30241 * gst/gdp/gstgdppay.c:
30244 2006-07-13 14:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30246 gst/playback/gstdecodebin.c: Fix a caps leak when linking (#347304)
30247 Original commit message from CVS:
30248 * gst/playback/gstdecodebin.c: (find_compatibles):
30249 Fix a caps leak when linking (#347304)
30250 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
30251 (gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
30252 (gst_ximagesink_change_state):
30253 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
30254 (gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
30255 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
30256 (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
30257 Don't leak shared memory resources. Use the object lock to protect
30258 against the xcontext disappearing while returning a buffer from the
30259 pipeline. (#347304)
30261 2006-07-12 14:20:43 +0000 Edward Hervey <bilboed@bilboed.com>
30263 ext/vorbis/vorbisdec.c: gst_tag_list_merge() returns a new object. Take that into account when using it. This avoids ...
30264 Original commit message from CVS:
30265 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
30266 (vorbis_handle_comment_packet):
30267 gst_tag_list_merge() returns a new object. Take that into account when
30268 using it. This avoids memleak.
30269 Revert previous commit which is not needed.
30271 2006-07-12 13:30:20 +0000 Edward Hervey <bilboed@bilboed.com>
30273 ext/vorbis/vorbisdec.c: Reset the decoder in finalize so that all fields get cleared.
30274 Original commit message from CVS:
30275 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
30276 Reset the decoder in finalize so that all fields get cleared.
30278 2006-07-12 13:24:19 +0000 Wim Taymans <wim.taymans@gmail.com>
30280 gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
30281 Original commit message from CVS:
30282 * gst-libs/gst/audio/gstbaseaudiosrc.c:
30283 (gst_base_audio_src_set_clock),
30284 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
30285 Don't try to post an error message when setting the clock fails
30286 as this can happen when adding an element to a bin which will then
30287 deadlock. Fixes #347296.
30289 2006-07-12 13:04:15 +0000 Edward Hervey <bilboed@bilboed.com>
30291 ext/vorbis/vorbisdec.c: Post tag messages on the bus even if we're not initialized.
30292 Original commit message from CVS:
30293 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
30294 (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
30295 (vorbis_handle_type_packet):
30296 Post tag messages on the bus even if we're not initialized.
30297 If we're not initialized, we still postpone the event pushing of tags.
30299 2006-07-12 11:28:37 +0000 Wim Taymans <wim.taymans@gmail.com>
30301 Revert last two changes that broke the freeze.
30302 Original commit message from CVS:
30303 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
30304 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
30305 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
30306 Revert last two changes that broke the freeze.
30308 2006-07-12 10:59:55 +0000 Wim Taymans <wim.taymans@gmail.com>
30310 ext/alsa/gstalsasink.c: basesink calculates silence sample correctly for us.
30311 Original commit message from CVS:
30312 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
30313 basesink calculates silence sample correctly for us.
30315 2006-07-12 10:58:42 +0000 Wim Taymans <wim.taymans@gmail.com>
30317 gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
30318 Original commit message from CVS:
30319 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
30320 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
30321 Calculate correct silence samples so we don't fill our ringbuffer
30324 2006-07-12 10:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
30326 ext/vorbis/vorbisdec.*: Delay sending events (newsegment, tags) until the decoder is properly initialized.
30327 Original commit message from CVS:
30328 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
30329 (gst_vorbis_dec_reset), (vorbis_dec_sink_event),
30330 (vorbis_handle_comment_packet), (vorbis_handle_type_packet):
30331 * ext/vorbis/vorbisdec.h:
30332 Delay sending events (newsegment, tags) until the decoder is properly
30336 2006-07-11 22:40:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30353 Original commit message from CVS:
30356 2006-07-11 21:04:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30358 tests/check/elements/audioconvert.c: Patch from #347221 adding a test for audioconvert channel remappings.
30359 Original commit message from CVS:
30360 * tests/check/elements/audioconvert.c: (get_float_mc_caps),
30361 (get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
30362 Patch from #347221 adding a test for audioconvert
30363 channel remappings.
30365 2006-07-11 12:03:25 +0000 Tim-Philipp Müller <tim@centricular.net>
30367 gst/subparse/gstssaparse.c: Don't include the terminating NUL in the buffer size, it's only there for extra paranoia ...
30368 Original commit message from CVS:
30369 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
30370 (gst_ssa_parse_parse_line):
30371 Don't include the terminating NUL in the buffer size,
30372 it's only there for extra paranoia (would add random
30373 '*' characters at the end of each subtitle since the
30374 terminator itself is not valid UTF-8 technically).
30375 Also fix indenting after boilerplate macro.
30377 2006-07-10 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.net>
30379 gst/playback/gstdecodebin.c: Also emit 'unknown-type' signal (which should really be called unhandled-type) if we fou...
30380 Original commit message from CVS:
30381 * gst/playback/gstdecodebin.c: (close_pad_link):
30382 Also emit 'unknown-type' signal (which should really be
30383 called unhandled-type) if we found potential decoders/demuxers
30384 in the registry but none of them worked in the end (as in the
30385 case where the plugins don't exist any longer but are still
30386 listed in the registry). Fixes #329798.
30388 2006-07-08 13:48:58 +0000 Andy Wingo <wingo@pobox.com>
30391 * ext/theora/theoraparse.c:
30392 theoraparse.c (theora_parse_push_buffer)
30393 Original commit message from CVS:
30394 2006-07-08 Andy Wingo <wingo@pobox.com>
30395 * theoraparse.c (theora_parse_push_buffer)
30396 (theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
30397 Add some more debugging. Fix granulepos reconstruction in the face
30398 of discontinuities.
30400 2006-07-06 15:54:50 +0000 Wim Taymans <wim.taymans@gmail.com>
30402 gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
30403 Original commit message from CVS:
30404 * gst-libs/gst/audio/gstbaseaudiosink.c:
30405 (gst_base_audio_sink_class_init),
30406 (gst_base_audio_sink_provide_clock):
30407 Use gobject_class instead of G_OBJECT_CLASS (klass)
30408 * gst-libs/gst/audio/gstbaseaudiosrc.c:
30409 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
30410 (gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
30411 (gst_base_audio_src_get_time),
30412 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
30413 (gst_base_audio_src_create_ringbuffer):
30414 Fix latency and buffer-time constants and properties ala basesink.
30415 Implement pull based scheduling. Fixes #346527.
30416 Set default blocksize in GstBaseSrc to 0, we default to pushing out
30418 Refuse slaving to another clock instead of silently not working.
30419 Only provide a clock when we are actually able to do so.
30420 Various small cleanups and compiler hints.
30422 2006-07-06 13:23:07 +0000 Lutz Mueller <lutz@topfrose.de>
30424 gst/typefind/gsttypefindfunctions.c: Add typefinding for text/html (#346581).
30425 Original commit message from CVS:
30426 Patch by: Lutz Mueller <lutz at topfrose de>
30427 * gst/typefind/gsttypefindfunctions.c: (html_type_find),
30429 Add typefinding for text/html (#346581).
30431 2006-07-06 13:12:02 +0000 Tim-Philipp Müller <tim@centricular.net>
30433 gst/typefind/gsttypefindfunctions.c: Fix SMIL typefinding, make xml_check_first_element() more useful.
30434 Original commit message from CVS:
30435 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
30436 (xml_check_first_element), (xml_type_find), (smil_type_find):
30437 Fix SMIL typefinding, make xml_check_first_element() more
30440 2006-07-06 13:04:24 +0000 Tim-Philipp Müller <tim@centricular.net>
30442 gst/playback/gstplaybasebin.*: Protect list of elements with a subtitle-encoding property and the subtitle encoding m...
30443 Original commit message from CVS:
30444 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
30445 (gst_play_base_bin_finalize), (decodebin_element_added_cb),
30446 (decodebin_element_removed_cb), (gst_play_base_bin_set_property):
30447 * gst/playback/gstplaybasebin.h:
30448 Protect list of elements with a subtitle-encoding property and
30449 the subtitle encoding member itself with a lock of their own
30450 instead of using the object lock. This prevents a dead-lock in
30451 the element-remove callback in some circumstances when shutting
30454 2006-07-05 20:11:13 +0000 Sébastien Moutte <sebastien@moutte.net>
30456 win32/common/libgsttag.def: Export some new functions.
30457 Original commit message from CVS:
30458 * win32/common/libgsttag.def:
30459 Export some new functions.
30460 * win32/vs6/libgstogg.dsp:
30461 Add a link to libgsttag-0.10.lib.
30463 2006-07-04 16:50:21 +0000 Tim-Philipp Müller <tim@centricular.net>
30465 ext/alsa/gstalsamixertrack.c: Some const-ification.
30466 Original commit message from CVS:
30467 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
30468 Some const-ification.
30470 2006-07-04 14:06:03 +0000 Wim Taymans <wim.taymans@gmail.com>
30472 gst/playback/gstplaybasebin.c: Improve checking if we are dealing with a stream. Added some more uris that need buffe...
30473 Original commit message from CVS:
30474 * gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
30475 Improve checking if we are dealing with a stream. Added some
30476 more uris that need buffering.
30478 2006-07-03 10:43:31 +0000 Edward Hervey <bilboed@bilboed.com>
30480 ext/vorbis/vorbisdec.c: Remove unused variable.
30481 Original commit message from CVS:
30482 * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
30483 Remove unused variable.
30485 2006-07-02 21:48:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30487 Makefile.am: include lcov.mak
30488 Original commit message from CVS:
30492 add GCOV_LIBS to GST_LIBS
30494 2006-07-02 11:08:58 +0000 Michael Sheldon <webmaster@mikeasoft.com>
30496 ext/alsa/gstalsasrc.c: Add 32 bps to template caps and increase channels range from [1,2] to [1,MAX]. See #346326.
30497 Original commit message from CVS:
30498 Patch by: Michael Sheldon <webmaster at mikeasoft com>
30499 * ext/alsa/gstalsasrc.c:
30500 Add 32 bps to template caps and increase channels range
30501 from [1,2] to [1,MAX]. See #346326.
30503 2006-06-30 12:04:51 +0000 Tim-Philipp Müller <tim@centricular.net>
30505 gst-libs/gst/riff/riff-media.c: Recognise 'WMVA' video codec fourcc (#345879).
30506 Original commit message from CVS:
30507 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
30508 Recognise 'WMVA' video codec fourcc (#345879).
30510 2006-06-29 12:21:06 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
30512 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
30513 Original commit message from CVS:
30514 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
30515 Fixed nasty memory leak
30517 2006-06-26 13:19:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30519 gst/tcp/gsttcp.c: fix logging
30520 Original commit message from CVS:
30521 * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
30522 (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
30525 2006-06-23 16:45:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30527 gst/playback/gstdecodebin.c: Protect remove_fakesink using a mutex, so that we don't try and remove the fakesink simu...
30528 Original commit message from CVS:
30529 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
30530 (gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
30531 (remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
30532 Protect remove_fakesink using a mutex, so that we don't try and
30533 remove the fakesink simultaneously from multiple threads.
30534 When going from READY to PAUSED, restore the fakesink, so that
30535 it is there when decodebin gets reused.
30537 2006-06-23 09:53:09 +0000 Tim-Philipp Müller <tim@centricular.net>
30539 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
30540 Original commit message from CVS:
30541 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
30542 * gst-libs/gst/rtp/gstbasertpdepayload.c:
30543 * gst-libs/gst/rtp/gstbasertppayload.c:
30544 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30545 * gst/tcp/gstmultifdsink.c:
30546 * gst/tcp/gsttcpclientsink.c:
30547 * gst/tcp/gsttcpclientsrc.c:
30548 * gst/tcp/gsttcpserversink.c:
30549 * gst/tcp/gsttcpserversrc.c:
30550 * gst/videorate/gstvideorate.c:
30551 * gst/videotestsrc/gstvideotestsrc.c:
30552 * sys/v4l/gstv4ljpegsrc.c:
30553 * sys/v4l/gstv4lmjpegsink.c:
30554 * sys/v4l/gstv4lsrc.c:
30555 * tests/examples/seek/scrubby.c:
30556 * tests/examples/seek/seek.c:
30557 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
30559 2006-06-23 09:09:44 +0000 Tim-Philipp Müller <tim@centricular.net>
30561 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) and fix one GObject boilerplate macro.
30562 Original commit message from CVS:
30563 * ext/directfb/dfbvideosink.c:
30564 * ext/gsm/gstgsmdec.c:
30565 * ext/gsm/gstgsmenc.c:
30566 * ext/libmms/gstmms.c:
30567 * ext/neon/gstneonhttpsrc.c:
30568 * ext/theora/theoradec.c:
30569 * gst/freeze/gstfreeze.c:
30570 * gst/gdp/gstgdpdepay.c:
30571 * gst/gdp/gstgdppay.c:
30572 * sys/glsink/glimagesink.c:
30573 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
30574 and fix one GObject boilerplate macro.
30576 2006-06-22 12:13:31 +0000 Tim-Philipp Müller <tim@centricular.net>
30578 gst-libs/gst/tag/tags.c: Second field in GEnumValue shouldn't be a description, but a stringified version of the enum...
30579 Original commit message from CVS:
30580 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
30581 Second field in GEnumValue shouldn't be a description,
30582 but a stringified version of the enum value.
30584 2006-06-22 12:03:14 +0000 Wim Taymans <wim.taymans@gmail.com>
30586 sys/ximage/ximagesink.c: Avoid type checking in buffer casts.
30587 Original commit message from CVS:
30588 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
30589 (gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
30590 (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
30591 Avoid type checking in buffer casts.
30592 Avoid caps copy in buffer_alloc when we can.
30593 Use pad_peer_accept.
30595 2006-06-22 11:01:41 +0000 Tim-Philipp Müller <tim@centricular.net>
30597 gst-libs/gst/tag/tag.h: Oops, make that 'Since: 0.10.9'.
30598 Original commit message from CVS:
30599 * gst-libs/gst/tag/tag.h:
30600 Oops, make that 'Since: 0.10.9'.
30602 2006-06-22 10:55:05 +0000 Tim-Philipp Müller <tim@centricular.net>
30604 API: add GstTagImageType enum to describe images contained in image tags (#345641).
30605 Original commit message from CVS:
30606 * docs/libs/gst-plugins-base-libs-sections.txt:
30607 * gst-libs/gst/tag/tag.h:
30608 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
30609 (gst_tag_image_type_get_type):
30610 API: add GstTagImageType enum to describe images contained
30611 in image tags (#345641).
30613 2006-06-22 10:31:22 +0000 Tim-Philipp Müller <tim@centricular.net>
30615 gst/tcp/gstmultifdsink.c: Fix warnings with gst-inspect: "buffers-min" property should be of G_TYPE_INT and not G_TYP...
30616 Original commit message from CVS:
30617 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
30618 Fix warnings with gst-inspect: "buffers-min" property
30619 should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
30620 typo in property description.
30622 2006-06-22 10:10:51 +0000 Cody Russell <bratsche@gnome.org>
30624 gst/: Avoid unnecessary class cast check in class_init functions (#337747).
30625 Original commit message from CVS:
30626 Patch by: Cody Russell <bratsche at gnome org>
30627 * gst/audioresample/gstaudioresample.c:
30628 (gst_audioresample_class_init):
30629 * gst/playback/gststreamselector.c:
30630 (gst_stream_selector_class_init):
30631 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
30632 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
30633 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
30634 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
30635 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
30636 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
30637 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
30638 * gst/videotestsrc/gstvideotestsrc.c:
30639 (gst_video_test_src_class_init):
30640 * gst/volume/gstvolume.c: (gst_volume_class_init):
30641 Avoid unnecessary class cast check in class_init
30642 functions (#337747).
30644 2006-06-21 18:39:07 +0000 Tim-Philipp Müller <tim@centricular.net>
30646 ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
30647 Original commit message from CVS:
30648 * ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
30649 (gst_text_overlay_video_chain):
30650 g_markup_escape_text() REALLY doesn't like non-UTF8 input
30651 and doesn't validate its input either (and neither did
30652 textoverlay it seems). Let's do that then and fix #345206.
30654 2006-06-19 17:12:57 +0000 Wim Taymans <wim.taymans@gmail.com>
30656 gst/tcp/gstmultifdsink.*: Added shiny new burst-on-connect methods.
30657 Original commit message from CVS:
30658 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
30659 (gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
30660 (gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
30661 (gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
30662 (find_syncframe), (find_limits), (assign_value),
30663 (count_burst_unit), (gst_multi_fd_sink_new_client),
30664 (gst_multi_fd_sink_handle_client_write),
30665 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
30666 (gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
30667 (gst_multi_fd_sink_change_state):
30668 * gst/tcp/gstmultifdsink.h:
30669 Added shiny new burst-on-connect methods.
30670 Add properties to control the minimal amount of data queued.
30672 API: bytes-min property
30673 API: time-min property
30674 API: buffers-min property
30675 API: burst-unit property
30676 API: burst-value property
30677 API: add-full signal
30678 * gst/tcp/gsttcp-marshal.list:
30679 Added new marshaller code for the new signal.
30680 * tests/check/elements/multifdsink.c: (GST_START_TEST),
30681 (multifdsink_suite):
30682 Added testcases for new burst methods.
30684 2006-06-19 11:35:47 +0000 Christian Schaller <uraeus@gnome.org>
30686 * gst-plugins-base.spec.in:
30687 update for latest changes
30688 Original commit message from CVS:
30689 update for latest changes
30691 2006-06-19 09:57:50 +0000 Edward Hervey <bilboed@bilboed.com>
30693 ext/theora/theoradec.c: Implement clipping for accurate seeking.
30694 Original commit message from CVS:
30695 * ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
30696 Implement clipping for accurate seeking.
30699 2006-06-19 09:08:05 +0000 Philip Jaegenstedt <philip@lysator.liu.se>
30701 gst/videoscale/gstvideoscale.c: Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
30702 Original commit message from CVS:
30703 Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
30704 * gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
30705 (gst_video_scale_transform):
30706 Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
30708 2006-06-17 14:18:41 +0000 Tim-Philipp Müller <tim@centricular.net>
30712 Original commit message from CVS:
30715 2006-06-17 14:13:03 +0000 Tim-Philipp Müller <tim@centricular.net>
30717 configure.ac: Fix --disable-extern (can't set conditionals conditionally, #343602).
30718 Original commit message from CVS:
30720 Fix --disable-extern (can't set conditionals conditionally,
30723 2006-06-16 15:43:23 +0000 Tim-Philipp Müller <tim@centricular.net>
30725 tests/check/elements/audioresample.c: Add test case for bug #342789 fixed below.
30726 Original commit message from CVS:
30727 * tests/check/elements/audioresample.c: (test_reuse),
30728 (audioresample_suite):
30729 Add test case for bug #342789 fixed below.
30731 2006-06-16 15:17:44 +0000 Tim-Philipp Müller <tim@centricular.net>
30733 gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
30734 Original commit message from CVS:
30735 * gst/audioresample/gstaudioresample.c:
30736 (gst_audioresample_class_init), (gst_audioresample_init),
30737 (audioresample_start), (audioresample_stop),
30738 (gst_audioresample_set_property), (gst_audioresample_get_property):
30739 Implement GstBaseTransform::start and ::stop so that audioresample
30740 can clear its internal state properly and be reused insted of
30741 causing non-negotiated errors with playbin under some circumstances
30743 * tests/check/elements/audioresample.c: (setup_audioresample),
30744 (cleanup_audioresample):
30745 Need to set element state here so that ::start and ::stop are
30748 2006-06-16 13:59:29 +0000 Young-Ho Cha <ganadist@chollian.net>
30750 gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
30751 Original commit message from CVS:
30752 Patch by: Young-Ho Cha <ganadist at chollian dot net>
30753 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
30754 Parse extra data better, apparently it's right behind
30755 the normal strf header size. Fixes #343500.
30757 2006-06-16 11:04:21 +0000 Wim Taymans <wim.taymans@gmail.com>
30759 ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
30760 Original commit message from CVS:
30761 * ext/alsa/gstalsasink.c: (set_hwparams):
30762 If we fail to set the buffer_time and period_time alsa
30763 parameters, post a warning and leave alsa select a
30764 default instead of failing. Fixes #342085
30766 2006-06-16 10:30:25 +0000 Tim-Philipp Müller <tim@centricular.net>
30769 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
30770 Original commit message from CVS:
30771 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
30773 2006-06-16 10:20:10 +0000 Tim-Philipp Müller <tim@centricular.net>
30775 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.
30776 Original commit message from CVS:
30777 * docs/libs/gst-plugins-base-libs-sections.txt:
30778 * gst-libs/gst/cdda/gstcddabasesrc.h:
30779 Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
30780 out in the header file and shouldn't be listed in the docs.
30781 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
30782 Fix it so that it doesn't crash in the debug statement.
30784 2006-06-16 10:02:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30786 docs/libs/: add remaining symbols into correct setions
30787 Original commit message from CVS:
30788 * docs/libs/Makefile.am:
30789 * docs/libs/gst-plugins-base-libs-docs.sgml:
30790 * docs/libs/gst-plugins-base-libs-sections.txt:
30791 * docs/libs/gst-plugins-base-libs.types:
30792 add remaining symbols into correct setions
30793 * gst-libs/gst/audio/gstringbuffer.c:
30794 fix incomplete docs
30795 * gst-libs/gst/audio/gstringbuffer.h:
30796 comment out not yet implemented function
30797 * gst-libs/gst/floatcast/floatcast.h:
30798 * gst-libs/gst/netbuffer/gstnetbuffer.c:
30799 add short descriptions
30800 * gst-libs/gst/interfaces/propertyprobe.c:
30801 fix return value docs
30802 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
30803 simplify debug logging
30804 * gst-libs/gst/riff/riff-read.h:
30805 sync function prototype and docs
30806 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
30807 remove left over symbol
30809 2006-06-16 09:56:41 +0000 Tim-Philipp Müller <tim@centricular.net>
30811 Use GST_PLUGIN_DOCS macro in configure.ac, add
30812 Original commit message from CVS:
30815 * docs/Makefile.am:
30816 Use GST_PLUGIN_DOCS macro in configure.ac, add
30817 --enable-plugin-docs default to autogen.sh and use
30818 ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
30820 2006-06-15 15:27:49 +0000 Wim Taymans <wim.taymans@gmail.com>
30822 ext/ogg/gstoggdemux.c: Combine GstFlowReturn from the source pads to give a meaningfull result to the upstream peer o...
30823 Original commit message from CVS:
30824 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
30825 (gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
30826 (gst_ogg_demux_loop):
30827 Combine GstFlowReturn from the source pads to give a
30828 meaningfull result to the upstream peer or to stop the
30829 processing task in case of errors.
30831 2006-06-14 14:49:33 +0000 Tim-Philipp Müller <tim@centricular.net>
30833 gst/playback/gststreaminfo.c: Try GST_TAG_CODEC as fallback when extracting the codec name; more debug info.
30834 Original commit message from CVS:
30835 * gst/playback/gststreaminfo.c: (cb_probe):
30836 Try GST_TAG_CODEC as fallback when extracting the
30837 codec name; more debug info.
30839 2006-06-14 14:34:28 +0000 Tim-Philipp Müller <tim@centricular.net>
30841 ext/ogg/: Extract language tags from ogm subtitle streams, so that the subtitle menu choices are labelled correctly in
30842 Original commit message from CVS:
30843 * ext/ogg/Makefile.am:
30844 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
30845 Extract language tags from ogm subtitle streams, so that
30846 the subtitle menu choices are labelled correctly in
30847 Totem (fixes #344708).
30849 2006-06-14 09:13:54 +0000 Alessandro Decina <alessandro@nnva.org>
30851 ext/ogg/gstoggmux.c: Fix various leaks. Fixes #343699.
30852 Original commit message from CVS:
30853 Patch by: Alessandro Decina <alessandro at nnva dot org>
30854 * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
30855 (gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
30856 (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
30857 (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
30858 Fix various leaks. Fixes #343699.
30859 Add x-smoke mime type.
30861 2006-06-14 08:17:45 +0000 Tim-Philipp Müller <tim@centricular.net>
30863 gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
30864 Original commit message from CVS:
30865 * gst-libs/gst/riff/riff-ids.h:
30866 Add IDs for 'bext' chunks (see #343837).
30868 2006-06-12 12:44:38 +0000 Young-Ho Cha <ganadist@chollian.net>
30870 gst/subparse/samiparse.c: Honour font face tags in SAMI subtitles (#344503).
30871 Original commit message from CVS:
30872 Patch by: Young-Ho Cha <ganadist at chollian net>
30873 * gst/subparse/samiparse.c: (sami_context_pop_state),
30874 (handle_start_font), (end_sami_element):
30875 Honour font face tags in SAMI subtitles (#344503).
30877 2006-06-11 20:41:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30879 po/POTFILES.in: add missing files containing translatable strings
30880 Original commit message from CVS:
30882 add missing files containing translatable strings
30884 2006-06-11 19:55:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30886 docs/libs/tmpl/.cvsignore: we don't want those *.sgml files in CVS either
30887 Original commit message from CVS:
30888 * docs/libs/tmpl/.cvsignore:
30889 we don't want those *.sgml files in CVS either
30891 2006-06-11 19:44:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30894 Original commit message from CVS:
30895 * docs/libs/.cvsignore:
30896 * tests/check/elements/.cvsignore:
30897 * tests/check/libs/.cvsignore:
30900 2006-06-11 18:33:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30902 docs/libs/Makefile.am: also commiting the changed Makefile.am (added more libs to the doc-build)
30903 Original commit message from CVS:
30904 * docs/libs/Makefile.am:
30905 also commiting the changed Makefile.am (added more libs to the
30908 2006-06-11 17:08:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30910 docs/libs/: first batch of reordering things, add index & hierarchy
30911 Original commit message from CVS:
30912 * docs/libs/gst-plugins-base-libs-docs.sgml:
30913 * docs/libs/gst-plugins-base-libs-sections.txt:
30914 * docs/libs/gst-plugins-base-libs.types:
30915 first batch of reordering things, add index & hierarchy
30917 2006-06-11 14:08:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30920 * ext/alsa/Makefile.am:
30921 * ext/cdparanoia/Makefile.am:
30922 * ext/gnomevfs/Makefile.am:
30923 * ext/libvisual/Makefile.am:
30924 * ext/ogg/Makefile.am:
30925 * ext/pango/Makefile.am:
30926 * ext/theora/Makefile.am:
30927 * ext/vorbis/Makefile.am:
30928 * sys/v4l/Makefile.am:
30929 * sys/ximage/Makefile.am:
30930 * sys/xvimage/Makefile.am:
30931 further clean up build
30932 Original commit message from CVS:
30933 further clean up build
30935 2006-06-11 12:14:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30937 configure.ac: use GST_PKG_CHECK_MODULES, cleans up output
30938 Original commit message from CVS:
30940 use GST_PKG_CHECK_MODULES, cleans up output
30942 2006-06-11 12:10:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30945 * win32/common/config.h:
30947 Original commit message from CVS:
30950 2006-06-10 18:52:03 +0000 Tim-Philipp Müller <tim@centricular.net>
30952 ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
30953 Original commit message from CVS:
30954 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
30955 Add support for burn:// URIs (#343385); const-ify things a bit,
30956 use G_N_ELEMENTS instead of hard-coded array size.
30958 2006-06-10 18:25:07 +0000 Young-Ho Cha <ganadist@chollian.net>
30960 gst/subparse/samiparse.c: Fix up broken entities before passing them to libxml *sigh*. (#343303).
30961 Original commit message from CVS:
30962 Patch by: Young-Ho Cha <ganadist at chollian net>
30963 * gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
30964 Fix up broken entities before passing them to libxml *sigh*.
30967 2006-06-09 18:52:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30972 Original commit message from CVS:
30975 === release 0.10.8 ===
30977 2006-06-09 18:49:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30983 * docs/plugins/gst-plugins-base-plugins.args:
30984 * docs/plugins/inspect/plugin-adder.xml:
30985 * docs/plugins/inspect/plugin-alsa.xml:
30986 * docs/plugins/inspect/plugin-audioconvert.xml:
30987 * docs/plugins/inspect/plugin-audiorate.xml:
30988 * docs/plugins/inspect/plugin-audioresample.xml:
30989 * docs/plugins/inspect/plugin-audiotestsrc.xml:
30990 * docs/plugins/inspect/plugin-cdparanoia.xml:
30991 * docs/plugins/inspect/plugin-decodebin.xml:
30992 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
30993 * docs/plugins/inspect/plugin-gnomevfs.xml:
30994 * docs/plugins/inspect/plugin-libvisual.xml:
30995 * docs/plugins/inspect/plugin-ogg.xml:
30996 * docs/plugins/inspect/plugin-pango.xml:
30997 * docs/plugins/inspect/plugin-playbin.xml:
30998 * docs/plugins/inspect/plugin-subparse.xml:
30999 * docs/plugins/inspect/plugin-tcp.xml:
31000 * docs/plugins/inspect/plugin-theora.xml:
31001 * docs/plugins/inspect/plugin-typefindfunctions.xml:
31002 * docs/plugins/inspect/plugin-video4linux.xml:
31003 * docs/plugins/inspect/plugin-videorate.xml:
31004 * docs/plugins/inspect/plugin-videoscale.xml:
31005 * docs/plugins/inspect/plugin-videotestsrc.xml:
31006 * docs/plugins/inspect/plugin-volume.xml:
31007 * docs/plugins/inspect/plugin-vorbis.xml:
31008 * docs/plugins/inspect/plugin-ximagesink.xml:
31009 * docs/plugins/inspect/plugin-xvimagesink.xml:
31010 * win32/common/config.h:
31012 Original commit message from CVS:
31015 2006-06-07 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31017 0.10.7.2 prerelease
31018 Original commit message from CVS:
31034 * win32/common/config.h:
31035 0.10.7.2 prerelease
31037 2006-06-07 11:03:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31039 move last template doc snippets to source code and delete them
31040 Original commit message from CVS:
31041 * docs/libs/tmpl/gstaudio.sgml:
31042 * docs/libs/tmpl/gstcolorbalance.sgml:
31043 * docs/libs/tmpl/gstmixer.sgml:
31044 * docs/libs/tmpl/gstringbuffer.sgml:
31045 * docs/libs/tmpl/gsttuner.sgml:
31046 * docs/libs/tmpl/gstxoverlay.sgml:
31047 * gst-libs/gst/audio/audio.c:
31048 * gst-libs/gst/audio/gstringbuffer.c:
31049 * gst-libs/gst/interfaces/colorbalance.c:
31050 * gst-libs/gst/interfaces/mixer.c:
31051 * gst-libs/gst/interfaces/tuner.c:
31052 * gst-libs/gst/interfaces/xoverlay.c:
31053 move last template doc snippets to source code and delete them
31055 2006-06-06 16:26:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31057 * gst/gdp/gstgdppay.c:
31059 Original commit message from CVS:
31062 2006-06-06 14:39:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31064 configure.ac: enable building of GDP elements
31065 Original commit message from CVS:
31067 enable building of GDP elements
31068 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
31069 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
31070 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
31071 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
31072 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
31073 (gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
31074 (gst_gdp_pay_change_state):
31075 * gst/gdp/gstgdppay.h:
31078 2006-06-06 11:13:18 +0000 Michael Smith <msmith@xiph.org>
31080 ext/theora/theoraparse.c: Mark DELTA_UNIT on non-keyframes.
31081 Original commit message from CVS:
31082 * ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
31083 (theora_parse_drain_queue):
31084 Mark DELTA_UNIT on non-keyframes.
31086 2006-06-03 21:06:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31088 gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
31089 Original commit message from CVS:
31090 * gst-libs/gst/audio/gstbaseaudiosink.c:
31091 (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
31092 * gst-libs/gst/audio/gstbaseaudiosink.h:
31093 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
31094 (gst_ring_buffer_samples_done):
31095 * gst-libs/gst/audio/gstringbuffer.h:
31096 Document better the fact that latency_time and buffer_time are values
31097 stored in microseconds, and not the usual GStreamer nanoseconds.
31098 Change the variables (compatibly) that store them from GstClockTime
31099 to guint64 to make it more clear that they're not storing clock times.
31100 Also, remove the bogus property description that says the user can
31101 specify -1 to get the default value, since that's never been the case.
31102 When computing the default segment size for the ring buffer, make it
31103 an integer number of samples.
31104 When the sub-class indicates a delay greater than the number of
31105 samples we've written return 0 from the audio sink get_time method.
31107 2006-06-02 17:01:02 +0000 Michael Smith <msmith@xiph.org>
31109 tests/check/: Don't busy-wait in tests; this was causing test timeouts very frequently when running under valgrind.
31110 Original commit message from CVS:
31111 * tests/check/elements/audioconvert.c: (set_channel_positions),
31112 (get_float_mc_caps), (get_int_mc_caps):
31113 * tests/check/elements/audioresample.c:
31114 * tests/check/elements/audiotestsrc.c: (GST_START_TEST):
31115 * tests/check/elements/videorate.c:
31116 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
31117 * tests/check/elements/volume.c:
31118 * tests/check/elements/vorbisdec.c:
31119 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
31120 Don't busy-wait in tests; this was causing test timeouts very
31121 frequently when running under valgrind.
31123 2006-06-02 16:45:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31125 * gst/gdp/gstgdpdepay.c:
31126 * gst/gdp/gstgdppay.h:
31128 Original commit message from CVS:
31131 2006-06-02 16:35:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31133 * tests/check/elements/multifdsink.c:
31134 fail_if_can_read is racy
31135 Original commit message from CVS:
31136 fail_if_can_read is racy
31138 2006-06-02 16:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31140 gst/tcp/: make multifdsink properly deal with streamheader:
31141 Original commit message from CVS:
31143 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
31144 (gst_multi_fd_sink_remove_client_link),
31145 (gst_multi_fd_sink_client_queue_caps),
31146 (gst_multi_fd_sink_client_queue_buffer),
31147 (gst_multi_fd_sink_handle_client_write),
31148 (gst_multi_fd_sink_render):
31149 * gst/tcp/gstmultifdsink.h:
31150 make multifdsink properly deal with streamheader:
31151 - streamheader is taken from caps
31152 - buffers marked with IN_CAPS are not sent
31153 - streamheaders are sent, on connection, from the caps of the
31154 buffer where the client gets positioned to
31155 - further streamheader changes are done every time the client
31156 will receive a buffer with different caps
31157 * tests/check/elements/multifdsink.c: (GST_START_TEST),
31158 (gst_multifdsink_create_streamheader):
31161 2006-06-02 15:06:59 +0000 Michael Smith <msmith@xiph.org>
31163 ext/vorbis/vorbisdec.c: Reinstate limit on channel count. Vorbis does not define the meaning of > 6 channels, so they...
31164 Original commit message from CVS:
31165 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
31166 Reinstate limit on channel count. Vorbis does not define the meaning
31167 of > 6 channels, so they're just independent channels. Gstreamer
31168 currently has no mechanism to represent N independent channels.
31170 2006-06-02 14:23:34 +0000 Michael Smith <msmith@xiph.org>
31172 ext/vorbis/vorbisdec.c: Don't arbitrarily restrict channel counts and rate in vorbis.
31173 Original commit message from CVS:
31174 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
31175 Don't arbitrarily restrict channel counts and rate in vorbis.
31176 In terms of effects likely on real-world files, this fixes 96kHz
31177 playback of vorbis.
31179 2006-06-02 14:19:18 +0000 Michael Smith <msmith@xiph.org>
31181 gst/audioconvert/audioconvert.c: More correct float->int conversion.
31182 Original commit message from CVS:
31183 * gst/audioconvert/audioconvert.c: (float):
31184 More correct float->int conversion.
31186 2006-06-02 14:07:42 +0000 Michael Smith <msmith@xiph.org>
31188 ext/ogg/gstoggdemux.c: Don't accidently send GST_CLOCK_TIME_NONE as a new segment start value. Fixes g-critical on tr...
31189 Original commit message from CVS:
31190 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
31191 Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
31192 value. Fixes g-critical on trying to play back ogg containing
31195 2006-06-02 10:34:12 +0000 Wim Taymans <wim.taymans@gmail.com>
31197 gst/playback/gstplaybasebin.*: Make the subtitle detection work from any thread so we don't deadlock. Fixes #343397.
31198 Original commit message from CVS:
31199 * gst/playback/gstplaybasebin.c: (group_create), (group_commit),
31201 * gst/playback/gstplaybasebin.h:
31202 Make the subtitle detection work from any thread so we don't
31203 deadlock. Fixes #343397.
31205 2006-06-02 10:28:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31207 gst/gdp/gstgdppay.c: add crc-header and crc-payload properties don't error out on some things that are recoverable
31208 Original commit message from CVS:
31209 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
31210 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
31211 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
31212 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
31213 (gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
31214 (gst_gdp_pay_get_property):
31215 add crc-header and crc-payload properties
31216 don't error out on some things that are recoverable
31217 * tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
31220 2006-06-02 09:17:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31222 * gst/tcp/gsttcp.c:
31223 show type number when packet is of the wrong type
31224 Original commit message from CVS:
31225 show type number when packet is of the wrong type
31227 2006-06-01 23:04:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31229 gst/volume/Makefile.am: Seriously, it's not *that* hard to get compilation right. Even a drunk can do it ! Add LIBOI...
31230 Original commit message from CVS:
31231 * gst/volume/Makefile.am:
31232 Seriously, it's not *that* hard to get compilation right. Even
31233 a drunk can do it ! Add LIBOIL CFLAGS and LIBS
31235 2006-06-01 22:00:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31237 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
31238 Original commit message from CVS:
31239 * ext/alsaspdif/alsaspdifsink.h:
31240 * ext/amrwb/gstamrwbdec.h:
31241 * ext/amrwb/gstamrwbenc.h:
31242 * ext/amrwb/gstamrwbparse.h:
31243 * ext/arts/gst_arts.h:
31244 * ext/artsd/gstartsdsink.h:
31245 * ext/audiofile/gstafparse.h:
31246 * ext/audiofile/gstafsink.h:
31247 * ext/audiofile/gstafsrc.h:
31248 * ext/audioresample/gstaudioresample.h:
31249 * ext/bz2/gstbz2dec.h:
31250 * ext/bz2/gstbz2enc.h:
31251 * ext/dirac/gstdiracdec.h:
31252 * ext/directfb/dfbvideosink.h:
31253 * ext/divx/gstdivxdec.h:
31254 * ext/divx/gstdivxenc.h:
31255 * ext/dts/gstdtsdec.h:
31256 * ext/faac/gstfaac.h:
31257 * ext/gsm/gstgsmdec.h:
31258 * ext/gsm/gstgsmenc.h:
31259 * ext/ivorbis/vorbisenc.h:
31260 * ext/libfame/gstlibfame.h:
31261 * ext/nas/nassink.h:
31262 * ext/neon/gstneonhttpsrc.h:
31263 * ext/polyp/polypsink.h:
31264 * ext/sdl/sdlaudiosink.h:
31265 * ext/sdl/sdlvideosink.h:
31266 * ext/shout/gstshout.h:
31267 * ext/snapshot/gstsnapshot.h:
31268 * ext/sndfile/gstsf.h:
31269 * ext/swfdec/gstswfdec.h:
31270 * ext/tarkin/gsttarkindec.h:
31271 * ext/tarkin/gsttarkinenc.h:
31272 * ext/theora/theoradec.h:
31273 * ext/wavpack/gstwavpackdec.h:
31274 * ext/wavpack/gstwavpackparse.h:
31275 * ext/xine/gstxine.h:
31276 * ext/xvid/gstxviddec.h:
31277 * ext/xvid/gstxvidenc.h:
31278 * gst/cdxaparse/gstcdxaparse.h:
31279 * gst/cdxaparse/gstcdxastrip.h:
31280 * gst/colorspace/gstcolorspace.h:
31281 * gst/festival/gstfestival.h:
31282 * gst/freeze/gstfreeze.h:
31283 * gst/gdp/gstgdpdepay.h:
31284 * gst/gdp/gstgdppay.h:
31285 * gst/modplug/gstmodplug.h:
31286 * gst/mpeg1sys/gstmpeg1systemencode.h:
31287 * gst/mpeg1videoparse/gstmp1videoparse.h:
31288 * gst/mpeg2sub/gstmpeg2subt.h:
31289 * gst/mpegaudioparse/gstmpegaudioparse.h:
31290 * gst/multifilesink/gstmultifilesink.h:
31291 * gst/overlay/gstoverlay.h:
31292 * gst/playondemand/gstplayondemand.h:
31293 * gst/qtdemux/qtdemux.h:
31294 * gst/rtjpeg/gstrtjpegdec.h:
31295 * gst/rtjpeg/gstrtjpegenc.h:
31296 * gst/smooth/gstsmooth.h:
31297 * gst/smoothwave/gstsmoothwave.h:
31298 * gst/spectrum/gstspectrum.h:
31299 * gst/speed/gstspeed.h:
31300 * gst/stereo/gststereo.h:
31301 * gst/switch/gstswitch.h:
31302 * gst/tta/gstttadec.h:
31303 * gst/tta/gstttaparse.h:
31304 * gst/videodrop/gstvideodrop.h:
31305 * gst/xingheader/gstxingmux.h:
31306 * sys/directdraw/gstdirectdrawsink.h:
31307 * sys/directsound/gstdirectsoundsink.h:
31308 * sys/dxr3/dxr3audiosink.h:
31309 * sys/dxr3/dxr3spusink.h:
31310 * sys/dxr3/dxr3videosink.h:
31311 * sys/qcam/gstqcamsrc.h:
31312 * sys/vcd/vcdsrc.h:
31313 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
31315 2006-06-01 20:39:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31317 gst/volume/gstvolume.*: rewrite the passthrough check, split _int16 and _int16_clamp, fix another property desc., rem...
31318 Original commit message from CVS:
31319 * gst/volume/gstvolume.c: (volume_choose_func),
31320 (volume_update_real_volume), (gst_volume_class_init),
31321 (gst_volume_init), (volume_process_float), (volume_process_int16),
31322 (volume_process_int16_clamp), (volume_set_caps),
31323 (volume_transform_ip), (plugin_init):
31324 * gst/volume/gstvolume.h:
31325 rewrite the passthrough check, split _int16 and _int16_clamp, fix
31326 another property desc., remove unused param from process function
31327 * tests/check/elements/volume.c: (volume_suite):
31328 reactivate the passthrough test
31330 2006-06-01 19:19:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31332 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
31333 Original commit message from CVS:
31334 * ext/alsa/gstalsamixerelement.h:
31335 * ext/alsa/gstalsamixeroptions.h:
31336 * ext/alsa/gstalsamixertrack.h:
31337 * ext/gnomevfs/gstgnomevfssink.h:
31338 * ext/gnomevfs/gstgnomevfssrc.h:
31339 * ext/theora/gsttheoradec.h:
31340 * ext/theora/gsttheoraenc.h:
31341 * ext/theora/gsttheoraparse.h:
31342 * ext/vorbis/vorbisparse.h:
31343 * gst-libs/gst/audio/gstaudioclock.h:
31344 * gst-libs/gst/audio/gstaudiofilter.h:
31345 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
31346 * gst/audioconvert/gstaudioconvert.h:
31347 * gst/audioresample/gstaudioresample.h:
31348 * gst/audiotestsrc/gstaudiotestsrc.h:
31349 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
31350 * gst/playback/gststreamselector.h:
31351 * gst/tcp/gstmultifdsink.h:
31352 * gst/tcp/gsttcpclientsink.h:
31353 * gst/tcp/gsttcpclientsrc.h:
31354 * gst/tcp/gsttcpserversink.h:
31355 * gst/tcp/gsttcpserversrc.h:
31356 * gst/videorate/gstvideorate.h:
31357 * gst/videoscale/gstvideoscale.h:
31358 * gst/videotestsrc/gstvideotestsrc.h:
31359 * gst/volume/gstvolume.h:
31360 * sys/v4l/gstv4ljpegsrc.h:
31361 * sys/v4l/gstv4lmjpegsink.h:
31362 * sys/v4l/gstv4lmjpegsrc.h:
31363 * sys/v4l/gstv4lsrc.h:
31364 * sys/ximage/ximagesink.h:
31365 * sys/xvimage/xvimagesink.h:
31366 * tests/old/testsuite/alsa/sinesrc.h:
31367 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
31369 2006-05-31 16:56:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31371 * tests/check/elements/multifdsink.c:
31372 remove wrong commit
31373 Original commit message from CVS:
31374 remove wrong commit
31376 2006-05-31 16:21:48 +0000 Wim Taymans <wim.taymans@gmail.com>
31378 ext/libvisual/visual.c: Handle DISCONT.
31379 Original commit message from CVS:
31380 * ext/libvisual/visual.c: (gst_visual_reset),
31381 (gst_visual_sink_setcaps), (gst_visual_sink_event),
31382 (gst_visual_src_event), (get_buffer), (gst_visual_chain):
31384 Use running time before doing QoS.
31387 2006-05-31 14:17:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31389 docs/libs/Makefile.am: set a magic variable to indicate we know the docs are incomplete
31390 Original commit message from CVS:
31391 * docs/libs/Makefile.am:
31392 set a magic variable to indicate we know the docs are incomplete
31394 2006-05-30 20:33:59 +0000 Sébastien Moutte <sebastien@moutte.net>
31396 win32/common/libgstvideo.def: export gst_video_calculate_display_ratio
31397 Original commit message from CVS:
31398 * win32/common/libgstvideo.def:
31399 export gst_video_calculate_display_ratio
31400 * win32/vs6/libgstvideoscale.dsp:
31401 add link to libgstvideo-0.10.lib
31403 2006-05-30 19:00:39 +0000 Tim-Philipp Müller <tim@centricular.net>
31405 gst/playback/gstplaybasebin.c: Throw a more comprehensible error for rtsp:// URIs (rather than erroring out with a ne...
31406 Original commit message from CVS:
31407 * gst/playback/gstplaybasebin.c: (gen_source_element):
31408 Throw a more comprehensible error for rtsp:// URIs (rather
31409 than erroring out with a negotiation error later on) until
31410 we fix playbin to handle rtspsrc etc.
31412 2006-05-30 16:09:36 +0000 Wim Taymans <wim.taymans@gmail.com>
31414 ext/pango/gsttextoverlay.c: Added some FIXMEs.
31415 Original commit message from CVS:
31416 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
31417 (gst_text_overlay_text_event):
31420 2006-05-30 16:07:50 +0000 Wim Taymans <wim.taymans@gmail.com>
31422 gst/adder/gstadder.*: Implement release_request_pad.
31423 Original commit message from CVS:
31424 * gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
31425 (gst_adder_request_new_pad), (gst_adder_release_pad):
31426 * gst/adder/gstadder.h:
31427 Implement release_request_pad.
31428 Make padcounter atomic.
31429 * tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
31430 Added check for release_pad in adder.
31432 2006-05-30 16:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
31434 ext/ogg/gstoggdemux.c: Fix build again.
31435 Original commit message from CVS:
31436 * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
31439 2006-05-30 14:59:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31441 ext/ogg/gstoggdemux.c: add more debugging clean up printf formats for granulepos and serialno
31442 Original commit message from CVS:
31443 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
31444 (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
31445 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
31446 (gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
31447 (gst_ogg_demux_seek), (gst_ogg_demux_get_data),
31448 (gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
31449 (gst_ogg_demux_bisect_forward_serialno),
31450 (gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
31451 (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
31453 clean up printf formats for granulepos and serialno
31455 2006-05-30 14:31:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31457 * tests/check/elements/multifdsink.c:
31458 * tests/check/generic/states.c:
31459 properly fail if we can't make an element
31460 Original commit message from CVS:
31461 properly fail if we can't make an element
31463 2006-05-30 13:22:58 +0000 Michael Smith <msmith@xiph.org>
31465 ext/vorbis/vorbisenc.*: Multi-channel caps negotiation, so we can do proper multichannel vorbis encoding, negotiated ...
31466 Original commit message from CVS:
31467 * ext/vorbis/vorbisenc.c: (raw_caps_factory),
31468 (gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
31469 (gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
31470 (gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
31471 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
31472 * ext/vorbis/vorbisenc.h:
31473 Multi-channel caps negotiation, so we can do proper multichannel
31474 vorbis encoding, negotiated through audioconvert.
31476 2006-05-30 11:45:52 +0000 Wim Taymans <wim.taymans@gmail.com>
31478 tests/check/elements/adder.c: Added check to show that #339935 is fixed with ongoing adder and collectpads fixes.
31479 Original commit message from CVS:
31480 * tests/check/elements/adder.c: (test_event_message_received),
31481 (test_play_twice_message_received), (GST_START_TEST),
31483 Added check to show that #339935 is fixed with ongoing
31484 adder and collectpads fixes.
31486 2006-05-29 17:19:48 +0000 Wim Taymans <wim.taymans@gmail.com>
31488 gst/adder/gstadder.c: Don't leak pad name.
31489 Original commit message from CVS:
31490 * gst/adder/gstadder.c: (gst_adder_request_new_pad):
31491 Don't leak pad name.
31493 2006-05-29 15:49:53 +0000 Wim Taymans <wim.taymans@gmail.com>
31495 gst/adder/gstadder.c: Fix adder seeking.
31496 Original commit message from CVS:
31497 * gst/adder/gstadder.c: (gst_adder_query_duration),
31498 (forward_event_func), (forward_event), (gst_adder_src_event):
31500 Make query/seeking code threadsafe.
31501 * tests/check/Makefile.am:
31502 * tests/check/elements/adder.c: (test_event_message_received),
31503 (GST_START_TEST), (test_play_twice_message_received):
31504 Fix adder test case.
31506 2006-05-29 13:21:00 +0000 Young-Ho Cha <ganadist@chollian.net>
31508 gst/playback/gstplaybasebin.*: Add 'subtitle-encoding' property to playbin, so applications can force a subtitle enco...
31509 Original commit message from CVS:
31510 Patch by: Young-Ho Cha <ganadist at chollian net>
31511 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
31512 (gst_play_base_bin_init), (gst_play_base_bin_dispose),
31513 (set_encoding_element), (decodebin_element_added_cb),
31514 (decodebin_element_removed_cb), (setup_subtitle), (setup_source),
31515 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
31516 * gst/playback/gstplaybasebin.h:
31517 Add 'subtitle-encoding' property to playbin, so applications can
31518 force a subtitle encoding for non-UTF8 subtitles (#342268).
31519 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
31520 (gst_sub_parse_set_property):
31521 Rename recently-added 'encoding' property to 'subtitle-encoding'
31522 (so it can be proxied by playbin/decodebin in a generic way
31523 with less danger of false positives).
31525 2006-05-29 11:04:48 +0000 Michael Smith <msmith@xiph.org>
31527 gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf...
31528 Original commit message from CVS:
31529 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
31530 (append_with_other_format), (set_structure_widths),
31531 (gst_audio_convert_transform_caps):
31532 Patch from #341562: give more specific audio caps in get_caps, so
31533 that basetransform can make better decisions on what caps to
31536 2006-05-28 20:04:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31538 tests/check/elements/volume.c: make it compile again
31539 Original commit message from CVS:
31540 * tests/check/elements/volume.c:
31541 make it compile again
31543 2006-05-28 19:56:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31545 tests/check/elements/volume.c: disable test until #343196 gets resolved
31546 Original commit message from CVS:
31547 * tests/check/elements/volume.c: (volume_suite):
31548 disable test until #343196 gets resolved
31550 2006-05-28 19:42:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31552 gst/adder/gstadder.c: Make it easier to copy&paste
31553 Original commit message from CVS:
31554 * gst/adder/gstadder.c: (gst_adder_get_type):
31555 Make it easier to copy&paste
31556 * gst/volume/Makefile.am:
31557 * gst/volume/gstvolume.c: (volume_update_real_volume),
31558 (gst_volume_set_volume), (gst_volume_set_mute),
31559 (gst_volume_class_init), (volume_process_int16), (volume_set_caps),
31560 (volume_transform_ip), (volume_update_mute),
31561 (volume_update_volume):
31562 * gst/volume/gstvolume.h:
31563 Add own debug category, move duplicate code to helper function, fix
31564 property texts, add more comments and prepare ffor liboil-goodness
31565 * tests/check/Makefile.am:
31566 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
31567 add test for mute and passtrough case, be a bit more verbose to track
31569 * tests/check/generic/states.c: (GST_START_TEST):
31570 catch elements that fail to instantiate
31572 2006-05-28 09:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
31574 tests/check/pipelines/: Comment out tests using parse_launch() if core was built without parsing capabilities.
31575 Original commit message from CVS:
31576 * tests/check/pipelines/simple-launch-lines.c:
31577 * tests/check/pipelines/theoraenc.c:
31578 * tests/check/pipelines/vorbisenc.c:
31579 Comment out tests using parse_launch() if core was built without
31580 parsing capabilities.
31582 2006-05-27 13:34:03 +0000 Edward Hervey <bilboed@bilboed.com>
31584 tests/check/Makefile.am: Extra bonus points for whoever explains to ensonic that you are meant to test unit tests tho...
31585 Original commit message from CVS:
31586 * tests/check/Makefile.am:
31587 Extra bonus points for whoever explains to ensonic that you are meant
31588 to test unit tests thoroughly before commiting them, especially if
31589 you know it's going to break.
31590 De-activated element/adder tests.
31592 2006-05-27 13:09:16 +0000 Edward Hervey <bilboed@bilboed.com>
31594 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Marking caps conversion issues as GST_WARNING is way too verbose,
31595 Original commit message from CVS:
31596 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
31597 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
31598 Marking caps conversion issues as GST_WARNING is way too verbose,
31599 Moving them to GST_LOG.
31601 2006-05-27 11:26:18 +0000 Tim-Philipp Müller <tim@centricular.net>
31603 README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from...
31604 Original commit message from CVS:
31606 Replace current README (containing the release notes from
31607 some 0.9.x version) with a proper README taken from the core.
31609 2006-05-26 15:52:23 +0000 Wim Taymans <wim.taymans@gmail.com>
31611 ext/vorbis/vorbisdec.c: Small cleanups.
31612 Original commit message from CVS:
31613 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
31614 (vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
31615 (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
31616 (vorbis_dec_change_state):
31619 Clip output samples to segment boundaries.
31621 2006-05-26 11:17:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31623 sys/ximage/ximagesink.c: Improve the errors produced on bad output, including some human readable description strings.
31624 Original commit message from CVS:
31625 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
31626 (gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
31627 Improve the errors produced on bad output, including some human
31628 readable description strings.
31629 Handle the (theoretical for ximagesink) case where the XServer
31630 has a different idea about the size required for a particular
31631 frame and gives us too small a memory allocation.
31633 2006-05-26 10:18:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31636 Mention bugs fixed by previous commit
31637 Original commit message from CVS:
31638 Mention bugs fixed by previous commit
31640 2006-05-26 09:40:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31642 sys/xvimage/xvimagesink.c: Improve the errors produced on bad output, including some human readable description strings.
31643 Original commit message from CVS:
31644 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
31645 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
31646 (gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
31647 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
31648 Improve the errors produced on bad output, including some human
31649 readable description strings.
31650 Handle RGB Xv formats properly by transforming them into our
31651 big-endian caps description.
31652 Use gst_caps_truncate to ensure that we never try and choose a
31653 non-fixed caps in buffer_alloc.
31654 Handle the case where the XServer has a different idea about the size
31655 required for a particular frame and gives us too small a memory
31657 Use -1 to indicate 'no image format', because 0 is a valid XServer
31658 image format number.
31659 Put RGB Xv formats at the end of the caps, so that we always prefer
31661 Iterate the available Xv Encodings to determine the maximum width and
31662 height, and then return that in our caps.
31664 2006-05-25 16:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31666 gst/playback/gstdecodebin.c: When there is only one unfinished pad and it receives an event that doesn't match our re...
31667 Original commit message from CVS:
31668 * gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
31669 When there is only one unfinished pad and it receives an event that
31670 doesn't match our requirements, we need to set alldone=FALSE so that
31671 the fakesink is not removed yet.
31673 2006-05-25 09:32:31 +0000 Tim-Philipp Müller <tim@centricular.net>
31675 ext/ogg/gstoggdemux.c: Use gst_type_find_helper_for_buffer() to find the type of stream from the first packet.
31676 Original commit message from CVS:
31677 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
31678 Use gst_type_find_helper_for_buffer() to find the type
31679 of stream from the first packet.
31681 Bump requirements to core CVS (needed for vorbis
31682 typefinding to work).
31684 2006-05-24 08:34:53 +0000 Edward Hervey <bilboed@bilboed.com>
31686 gst/typefind/gsttypefindfunctions.c: Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
31687 Original commit message from CVS:
31688 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
31689 Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
31690 Else they play perfectly fine with qtdemux.
31692 2006-05-23 20:38:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31694 make more debug catagories static
31695 Original commit message from CVS:
31696 * ext/theora/theoradec.c:
31697 * ext/theora/theoraenc.c:
31698 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
31699 * gst/audiorate/gstaudiorate.c:
31700 make more debug catagories static
31701 * tests/check/Makefile.am:
31702 * tests/check/elements/adder.c: (message_received),
31703 (test_event_message_received), (GST_START_TEST),
31704 (test_play_twice_message_received), (adder_suite):
31705 added test case for using element twice, extra bonus points for anyone
31706 who can make these test run reliably
31708 2006-05-23 15:18:40 +0000 Tim-Philipp Müller <tim@centricular.net>
31710 ext/theora/theoradec.c: Make work with time-stamped input buffers that do not have a granulepos in BUFFER_OFFSET_END ...
31711 Original commit message from CVS:
31712 * ext/theora/theoradec.c: (theora_dec_chain):
31713 Make work with time-stamped input buffers that do not
31714 have a granulepos in BUFFER_OFFSET_END (like theora
31715 buffers coming from matroskademux). Fixes #342448.
31717 2006-05-22 15:53:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31719 gst/gdp/: Handle error cases when calling functions do downwards state change after parent's change_state
31720 Original commit message from CVS:
31721 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain),
31722 (gst_gdp_depay_change_state):
31723 * gst/gdp/gstgdpdepay.h:
31724 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader),
31725 (gst_gdp_pay_chain), (gst_gdp_pay_sink_event),
31726 (gst_gdp_pay_change_state):
31727 * gst/gdp/gstgdppay.h:
31728 Handle error cases when calling functions
31729 do downwards state change after parent's change_state
31730 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
31731 * tests/check/elements/gdppay.c: (GST_START_TEST):
31734 2006-05-22 13:25:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31736 adding GDP payloader and depayloader. Build integration will follow later when the GDP issues for core are sorted out.
31737 Original commit message from CVS:
31738 * gst/gdp/Makefile.am:
31739 * gst/gdp/gstgdp.c: (plugin_init):
31740 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_base_init),
31741 (gst_gdp_depay_class_init), (gst_gdp_depay_init),
31742 (gst_gdp_depay_finalize), (gst_gdp_depay_chain),
31743 (gst_gdp_depay_change_state), (gst_gdp_depay_plugin_init):
31744 * gst/gdp/gstgdpdepay.h:
31745 * gst/gdp/gstgdppay.c: (gst_gdp_pay_base_init),
31746 (gst_gdp_pay_class_init), (gst_gdp_pay_init),
31747 (gst_gdp_pay_dispose), (gst_gdp_stamp_buffer),
31748 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
31749 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
31750 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
31751 (gst_gdp_pay_sink_event), (gst_gdp_pay_change_state),
31752 (gst_gdp_pay_plugin_init):
31753 * gst/gdp/gstgdppay.h:
31754 * tests/check/Makefile.am:
31755 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
31756 (cleanup_gdpdepay), (gdpdepay_push_per_byte), (GST_START_TEST),
31757 (setup_gdpdepay_streamheader), (gdpdepay_suite), (main):
31758 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay),
31759 (GST_START_TEST), (setup_gdppay_streamheader), (gdppay_suite),
31761 adding GDP payloader and depayloader. Build integration will
31762 follow later when the GDP issues for core are sorted out.
31764 2006-05-22 11:42:03 +0000 Peter Kjellerstedt <pkj@axis.com>
31766 gst/tcp/Makefile.am: fdstresstest doesn't need Gtk+, fix compilation if gtk is not available (#342566).
31767 Original commit message from CVS:
31768 Patch by: Peter Kjellerstedt <pkj at axis com>
31769 * gst/tcp/Makefile.am:
31770 fdstresstest doesn't need Gtk+, fix compilation if
31771 gtk is not available (#342566).
31773 2006-05-19 17:57:56 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
31775 gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
31776 Original commit message from CVS:
31777 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
31779 Removed redundant floor()
31781 2006-05-19 15:00:43 +0000 Tim-Philipp Müller <tim@centricular.net>
31783 gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
31784 Original commit message from CVS:
31785 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
31786 On second thought, just skip JUNK chunks automatically, so
31787 the caller doesn't have to handle this. Fixes #342345.
31788 Also, return GST_FLOW_UNEXPECTED if we get a short read,
31789 not GST_FLOW_ERROR.
31791 2006-05-19 13:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
31793 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...
31794 Original commit message from CVS:
31795 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
31796 Don't bail out on JUNK chunks with a size of 0 (would try to
31797 pull_range 0 bytes before, which sources don't like too much).
31800 2006-05-19 13:02:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31802 Use the gstutil scaling function to preserve 64 bits while calculating output width and height from the display-aspec...
31803 Original commit message from CVS:
31804 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
31805 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
31806 Use the gstutil scaling function to preserve 64 bits while calculating
31807 output width and height from the display-aspect-ratio. (A continuation
31810 2006-05-19 11:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31812 sys/xvimage/xvimagesink.*: When performing buffer allocations, remember the caps and image format we return so that i...
31813 Original commit message from CVS:
31814 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
31815 (gst_xvimagesink_buffer_alloc):
31816 * sys/xvimage/xvimagesink.h:
31817 When performing buffer allocations, remember the caps and image format
31818 we return so that if the same caps are asked for next time we can
31819 return them immediately without doing any caps intersections.
31821 2006-05-18 23:00:02 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
31823 gst-libs/gst/rtp/README: Some new documentation
31824 Original commit message from CVS:
31825 2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
31826 * gst-libs/gst/rtp/README:
31827 Some new documentation
31828 * gst-libs/gst/rtp/gstrtpbuffer.h:
31829 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
31830 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
31831 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
31832 New RTP audio base payloader class. Supports frame or sample based codecs.
31833 Not enabled in Makefile.am until approved.
31835 2006-05-18 20:30:26 +0000 Tim-Philipp Müller <tim@centricular.net>
31837 tests/check/elements/alsa.c: Fix test case: don't try to free NULL GValueArray when there are no devices.
31838 Original commit message from CVS:
31839 * tests/check/elements/alsa.c: (test_device_property_probe):
31840 Fix test case: don't try to free NULL GValueArray when there
31843 2006-05-18 19:21:53 +0000 Tim-Philipp Müller <tim@centricular.net>
31845 tests/check/: Add simple test that runs a device property probe on alsasrc, alsasink and alsamixer. Disable valgrind ...
31846 Original commit message from CVS:
31847 * tests/check/Makefile.am:
31848 * tests/check/elements/alsa.c: (test_device_property_probe),
31849 (alsa_suite), (main):
31850 Add simple test that runs a device property probe on alsasrc,
31851 alsasink and alsamixer. Disable valgrind check for now (too
31852 many leaks in libasound, and valgrind ignored my suppressions
31855 2006-05-18 17:19:39 +0000 Martin Szulecki <gnomebugzilla@sukimashita.com>
31857 ext/alsa/: Clean up and simplify alsa device probing. Make it actually work for multiple classes. Don't cache results...
31858 Original commit message from CVS:
31859 * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
31860 (gst_alsa_device_property_probe_probe_property),
31861 (gst_alsa_device_property_probe_needs_probe),
31862 (gst_alsa_device_property_probe_get_values),
31863 (gst_alsa_type_add_device_property_probe_interface):
31864 * ext/alsa/gstalsadeviceprobe.h:
31865 * ext/alsa/gstalsamixerelement.c:
31866 (gst_alsa_mixer_element_init_interfaces):
31867 * ext/alsa/gstalsamixerelement.h:
31868 Clean up and simplify alsa device probing. Make it actually work
31869 for multiple classes. Don't cache results any longer.
31870 * ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
31871 (gst_alsasink_init):
31872 * ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
31873 (gst_alsasrc_interface_supported), (gst_implements_interface_init),
31874 (gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
31875 Make alsasink and alsasrc implement the GstPropertyProbe interface
31876 for device probing (#342181).
31877 Patch by: Martin Szulecki <gnomebugzilla at sukimashita com>
31879 2006-05-18 10:05:23 +0000 Tim-Philipp Müller <tim@centricular.net>
31881 gst/subparse/samiparse.c: Don't ignore return value of strtol (++compiler_happiness).
31882 Original commit message from CVS:
31883 * gst/subparse/samiparse.c: (handle_start_font):
31884 Don't ignore return value of strtol (++compiler_happiness).
31886 2006-05-17 17:49:10 +0000 Young-Ho Cha <ganadist@chollian.net>
31888 gst/subparse/gstsubparse.*: Add 'encoding' property (#341681).
31889 Original commit message from CVS:
31890 Patch by: Young-Ho Cha <ganadist chollian net>
31891 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
31892 (gst_sub_parse_class_init), (gst_sub_parse_init),
31893 (gst_sub_parse_set_property), (gst_sub_parse_get_property),
31894 (convert_encoding):
31895 * gst/subparse/gstsubparse.h:
31896 Add 'encoding' property (#341681).
31897 * gst/subparse/samiparse.c: (characters_sami):
31898 Output is pango markup, so we need to escape text
31899 between tags (#342143).
31901 2006-05-16 17:34:14 +0000 Tim-Philipp Müller <tim@centricular.net>
31903 gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
31904 Original commit message from CVS:
31905 * gst-libs/gst/audio/multichannel.c:
31906 (gst_audio_check_channel_positions):
31907 It's okay to have caps with channels=1 and a channel position
31908 different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
31909 (deinterleavers might want to keep the position in the caps,
31910 so that they can be re-interleaved again properly later).
31911 Leave check for unexpected 2-channel layouts intact for now.
31913 2006-05-16 16:28:10 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
31915 gst/tcp/gsttcp.c: Return GST_FLOW_UNEXPECTED when we have an eos on the socket so basesrc can do its job correctly.
31916 Original commit message from CVS:
31917 2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
31918 * gst/tcp/gsttcp.c: (gst_tcp_socket_read):
31919 Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
31920 basesrc can do its job correctly.
31922 2006-05-16 15:52:17 +0000 Tim-Philipp Müller <tim@centricular.net>
31924 ext/alsa/: Refactor and improve caps probing code: probe signedness when we probe the supported formats/widths; set e...
31925 Original commit message from CVS:
31926 * ext/alsa/Makefile.am:
31927 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
31928 (gst_alsa_detect_formats), (get_channel_free_structure),
31929 (caps_add_channel_configuration), (gst_alsa_detect_channels),
31930 (gst_alsa_probe_supported_formats):
31931 * ext/alsa/gstalsa.h:
31932 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
31933 Refactor and improve caps probing code: probe signedness
31934 when we probe the supported formats/widths; set endianness
31935 to the one we actually probed for (ie. cpu endianness).
31936 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
31937 (gst_alsasrc_close):
31938 * ext/alsa/gstalsasrc.h:
31939 Implement caps probing for alsasrc.
31941 2006-05-15 17:42:19 +0000 Wim Taymans <wim.taymans@gmail.com>
31943 ext/theora/theoradec.c: Cleanups, add some G_LIKELY.
31944 Original commit message from CVS:
31945 * ext/theora/theoradec.c: (gst_theora_dec_reset),
31946 (theora_dec_src_query), (theora_dec_src_event),
31947 (theora_dec_sink_event), (theora_handle_comment_packet),
31948 (theora_handle_data_packet), (theora_dec_change_state):
31949 Cleanups, add some G_LIKELY.
31950 Use segment helpers instead of our own wrong code.
31951 Clear queued buffers on seek and READY.
31952 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
31953 (vorbis_dec_convert), (vorbis_dec_src_query),
31954 (vorbis_dec_src_event), (vorbis_dec_sink_event),
31955 (vorbis_handle_comment_packet), (vorbis_dec_push),
31956 (vorbis_handle_data_packet), (vorbis_dec_chain),
31957 (vorbis_dec_change_state):
31958 * ext/vorbis/vorbisdec.h:
31959 Remove old useless packetno variable.
31960 Do position query properly.
31962 Do cleanup of queued buffers in new helper function
31965 2006-05-15 17:17:22 +0000 Tim-Philipp Müller <tim@centricular.net>
31967 ext/alsa/gstalsasink.c: Query supported sample rates. Fixes #341732.
31968 Original commit message from CVS:
31969 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
31970 Query supported sample rates. Fixes #341732.
31972 2006-05-15 17:01:02 +0000 Julien Moutte <julien@moutte.net>
31974 gst/playback/gstdecodebin.c: Make decodebin reusable when going from PAUSE_TO_READY and then back to PAUSED.
31975 Original commit message from CVS:
31976 2006-05-15 Julien MOUTTE <julien@moutte.net>
31977 * gst/playback/gstdecodebin.c: (cleanup_decodebin),
31978 (gst_decode_bin_change_state): Make decodebin reusable
31979 when going from PAUSE_TO_READY and then back to PAUSED.
31982 2006-05-15 16:49:31 +0000 Wim Taymans <wim.taymans@gmail.com>
31984 ext/vorbis/vorbisdec.c: Cleanups. Use refcounting and DEBUG_OBJECT.
31985 Original commit message from CVS:
31986 * ext/vorbis/vorbisdec.c: (vorbis_get_query_types),
31987 (vorbis_dec_convert), (vorbis_dec_src_query),
31988 (vorbis_dec_sink_query), (vorbis_dec_src_event),
31989 (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
31990 (vorbis_dec_clean_queued), (vorbis_dec_push),
31991 (vorbis_handle_data_packet), (vorbis_dec_change_state):
31992 Cleanups. Use refcounting and DEBUG_OBJECT.
31993 Reset segment on flush, use code methods instead of our
31995 Fix potential memleak.
31997 2006-05-15 16:46:44 +0000 Tim-Philipp Müller <tim@centricular.net>
31999 ext/alsa/gstalsasink.*: Don't leak allocated snd_output_t structure if there's more than one alsasink instance at a t...
32000 Original commit message from CVS:
32001 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
32002 (gst_alsasink_init):
32003 * ext/alsa/gstalsasink.h:
32004 Don't leak allocated snd_output_t structure if there's
32005 more than one alsasink instance at a time (#341873).
32006 Also fix GObject macros in header file.
32008 2006-05-15 15:31:30 +0000 Tim-Philipp Müller <tim@centricular.net>
32010 gst/subparse/gstsubparse.c: Don't use libxml functions in the typefinding code.
32011 Original commit message from CVS:
32012 * gst/subparse/gstsubparse.c:
32013 (gst_sub_parse_data_format_autodetect):
32014 Don't use libxml functions in the typefinding code.
32016 2006-05-15 15:01:08 +0000 Wim Taymans <wim.taymans@gmail.com>
32018 ext/ogg/gstoggdemux.c: Fix seeking performance in the case where a non-header packet has a 0 granulepos (busted theor...
32019 Original commit message from CVS:
32020 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
32021 Fix seeking performance in the case where a non-header
32022 packet has a 0 granulepos (busted theora case).
32025 2006-05-15 14:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
32027 gst/subparse/gstsubparse.c: Improve SAMI typefinding: handle case where there are whitespaces or newlines in front of...
32028 Original commit message from CVS:
32029 * gst/subparse/gstsubparse.c:
32030 (gst_sub_parse_data_format_autodetect):
32031 Improve SAMI typefinding: handle case where there are
32032 whitespaces or newlines in front of the first <SAMI>
32035 2006-05-15 12:18:13 +0000 Tim-Philipp Müller <tim@centricular.net>
32037 configure.ac: Build video4linux plugin even if there's no XVIDEO, just without implementing the GstXOverlay interface...
32038 Original commit message from CVS:
32040 Build video4linux plugin even if there's no XVIDEO, just
32041 without implementing the GstXOverlay interface (#334002).
32043 2006-05-15 10:17:04 +0000 Tim-Philipp Müller <tim@centricular.net>
32045 Add tentative support for libvisual-0.4 (#336881).
32046 Original commit message from CVS:
32048 * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
32050 Add tentative support for libvisual-0.4 (#336881).
32052 2006-05-15 09:41:03 +0000 Young-Ho Cha <ganadist@chollian.net>
32054 gst/subparse/samiparse.c: Need to map "silver" colour explicitly (#169936).
32055 Original commit message from CVS:
32056 Patch by: Young-Ho Cha <ganadist at chollian net>
32057 * gst/subparse/samiparse.c: (handle_start_font):
32058 Need to map "silver" colour explicitly (#169936).
32060 2006-05-15 09:14:35 +0000 Young-Ho Cha <ganadist@chollian.net>
32062 gst/subparse/: Add support for SAMI subtitles (#169936).
32063 Original commit message from CVS:
32064 Patch by: Young-Ho Cha <ganadist at chollian net>
32065 * gst/subparse/Makefile.am:
32066 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
32067 (parser_state_dispose), (gst_sub_parse_data_format_autodetect),
32068 (gst_sub_parse_format_autodetect), (feed_textbuf),
32069 (gst_subparse_type_find), (plugin_init):
32070 * gst/subparse/gstsubparse.h:
32071 * gst/subparse/samiparse.c:
32072 * gst/subparse/samiparse.h:
32073 Add support for SAMI subtitles (#169936).
32075 2006-05-14 21:18:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32077 * win32/common/config.h:
32079 Original commit message from CVS:
32082 2006-05-14 21:18:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32085 fix mistakes in README
32086 Original commit message from CVS:
32087 fix mistakes in README
32089 2006-05-14 18:15:17 +0000 Michael Smith <msmith@xiph.org>
32091 gst/audioconvert/gstchannelmix.c: Fix #341696: crash when mixing L+R+C to mono or stereo.
32092 Original commit message from CVS:
32093 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
32094 Fix #341696: crash when mixing L+R+C to mono or stereo.
32095 * tests/check/Makefile.am:
32096 * tests/check/elements/audioconvert.c: (set_channel_positions),
32097 (get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
32098 (audioconvert_suite):
32099 Add test for the above, including some generic framework bits for
32100 testing multichannel things.
32102 2006-05-14 16:05:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32106 Original commit message from CVS:
32109 === release 0.10.7 ===
32111 2006-05-14 16:00:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32113 configure.ac: releasing 0.10.7, "Leave the gun"
32114 Original commit message from CVS:
32115 2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
32117 releasing 0.10.7, "Leave the gun"
32119 2006-05-14 15:55:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32137 Original commit message from CVS:
32140 2006-05-12 22:22:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32143 Original commit message from CVS:
32144 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
32145 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
32148 2006-05-12 21:30:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32150 Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
32151 Original commit message from CVS:
32152 * docs/libs/gst-plugins-base-libs-docs.sgml:
32153 * docs/libs/gst-plugins-base-libs-sections.txt:
32154 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
32155 * gst-libs/gst/video/video.h:
32156 * gst/videoscale/Makefile.am:
32157 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
32158 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
32159 * tests/check/Makefile.am:
32160 * tests/check/libs/video.c: (GST_START_TEST), (video_suite),
32162 Fix integer overflow problem with pixel-aspect-ratio calculations
32163 in videoscale and xvimagesink (#341542)
32165 2006-05-12 16:56:52 +0000 Tim-Philipp Müller <tim@centricular.net>
32167 gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
32168 Original commit message from CVS:
32169 * gst-libs/gst/tag/gstid3tag.c:
32170 Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
32172 2006-05-12 10:39:08 +0000 Sébastien Moutte <sebastien@moutte.net>
32174 win32/MANIFEST: update win32 files listing
32175 Original commit message from CVS:
32177 update win32 files listing
32179 2006-05-11 21:47:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32181 * tests/check/elements/multifdsink.c:
32182 disable failing check on gentoo64
32183 Original commit message from CVS:
32184 disable failing check on gentoo64
32186 2006-05-11 21:35:44 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32188 * tests/check/elements/multifdsink.c:
32189 disable failing check on gentoo64
32190 Original commit message from CVS:
32191 disable failing check on gentoo64
32193 2006-05-11 21:20:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32195 * tests/check/elements/multifdsink.c:
32196 macros show the correct line
32197 Original commit message from CVS:
32198 macros show the correct line
32200 2006-05-11 21:04:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32202 * tests/check/elements/multifdsink.c:
32203 macros show the correct line
32204 Original commit message from CVS:
32205 macros show the correct line
32207 2006-05-11 21:01:05 +0000 Sjoerd Simons <sjoerd@luon.net>
32209 gst/playback/gstplaybasebin.*: API: GstPlayBaseBin::stream-info-value-array property use a more bindings-friendly way...
32210 Original commit message from CVS:
32211 2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
32212 patch by: Sjoerd Simons (sjoerd@luon.net)
32213 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
32214 (group_create), (group_destroy), (add_stream),
32215 (gst_play_base_bin_get_property),
32216 (gst_play_base_bin_get_streaminfo_value_array):
32217 * gst/playback/gstplaybasebin.h:
32218 API: GstPlayBaseBin::stream-info-value-array property
32219 use a more bindings-friendly way of exposing streaminfo
32220 using a GValueArray. Tested in ipython.
32223 2006-05-11 19:44:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32225 * tests/check/elements/multifdsink.c:
32226 fix some type warnings
32227 Original commit message from CVS:
32228 fix some type warnings
32230 2006-05-11 19:38:22 +0000 Wim Taymans <wim.taymans@gmail.com>
32232 gst/playback/gstdecodebin.c: Also catch queue underruns but don't do anything yet.
32233 Original commit message from CVS:
32234 * gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
32235 (queue_underrun_cb), (queue_filled_cb):
32236 Also catch queue underruns but don't do anything yet.
32237 Refactor and comment queue enlarging code a bit.
32238 * gst/playback/gstplaybasebin.c: (queue_overrun),
32239 (queue_threshold_reached), (queue_out_of_data),
32240 (gen_preroll_element):
32241 If a queue over/underruns check that we don't create nasty
32242 deadlocks when the min-threshold is not reached but the
32243 max-bytes is. In those cases disable max-bytes when we
32244 know that the queue is fed timed data.
32247 2006-05-11 18:06:18 +0000 Tim-Philipp Müller <tim@centricular.net>
32249 gst/playback/gstplaybin.c: Make playbin automatically plug an 'audioresample' element before the audio sink as well. ...
32250 Original commit message from CVS:
32251 * gst/playback/gstplaybin.c: (gen_audio_element):
32252 Make playbin automatically plug an 'audioresample'
32253 element before the audio sink as well. This solves
32254 problems with sinks that only accept a very specific
32255 sample rate, like esdsink (e.g. #340379).
32257 2006-05-11 16:04:28 +0000 Tim-Philipp Müller <tim@centricular.net>
32259 gst/playback/gstplaybasebin.c: Make http sources send special headers so that we receive icecast metadata if the http...
32260 Original commit message from CVS:
32261 * gst/playback/gstplaybasebin.c: (gen_source_element):
32262 Make http sources send special headers so that we receive
32263 icecast metadata if the http stream is an icecast stream
32264 (otherwise the server will just ignore them). This also
32265 means that from now on users will need the 'icydemux'
32266 element from gst-plugins-good installed if they want to
32267 listen to icecast radio streams. (#341432, #333657).
32269 2006-05-11 12:34:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32271 * gst/tcp/gstmultifdsink.c:
32273 Original commit message from CVS:
32276 2006-05-11 11:40:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32278 gst/tcp/gstmultifdsink.c: remove stupid example from docs - it should come with a simple
32279 Original commit message from CVS:
32280 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
32281 (gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
32282 remove stupid example from docs - it should come with a simple
32285 * tests/check/elements/multifdsink.c: (wait_bytes_served),
32286 (fail_if_can_read), (GST_START_TEST),
32287 (gst_multifdsink_create_streamheader), (multifdsink_suite):
32288 add a test for changing streamheader which exposes a bug in
32291 2006-05-11 10:33:46 +0000 Michael Smith <msmith@xiph.org>
32293 ext/gnomevfs/gstgnomevfssrc.*: Don't set icy-caps unless we have a sane interval value. Move interval to a local vari...
32294 Original commit message from CVS:
32295 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
32296 (gst_gnome_vfs_src_received_headers_callback):
32297 * ext/gnomevfs/gstgnomevfssrc.h:
32298 Don't set icy-caps unless we have a sane interval value. Move
32299 interval to a local variable; we never use it outside this function.
32301 2006-05-11 10:14:20 +0000 Wim Taymans <wim.taymans@gmail.com>
32303 sys/: Register special buffer types along with the objects so that they are not registered at runtime from N differen...
32304 Original commit message from CVS:
32305 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
32306 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
32307 Register special buffer types along with the objects so
32308 that they are not registered at runtime from N different
32309 streaming threads since they are not threadsafe.
32311 2006-05-10 18:31:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32313 * tests/check/elements/multifdsink.c:
32314 set caps and plug leaks
32315 Original commit message from CVS:
32316 set caps and plug leaks
32318 2006-05-10 18:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32320 tests/check/elements/multifdsink.c: add two more tests, one doing streamheader
32321 Original commit message from CVS:
32322 * tests/check/elements/multifdsink.c: (wait_bytes_served),
32323 (GST_START_TEST), (fail_unless_read), (multifdsink_suite):
32324 add two more tests, one doing streamheader
32326 2006-05-10 16:34:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32328 gst/tcp/gstmultifdsink.c: clean up the bufqueue when shutting down
32329 Original commit message from CVS:
32330 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
32331 clean up the bufqueue when shutting down
32332 * tests/check/Makefile.am:
32333 * tests/check/elements/multifdsink.c: (setup_multifdsink),
32334 (cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
32336 add a test for the leak that was just fixed
32338 2006-05-10 15:16:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32340 * gst/tcp/gstmultifdsink.c:
32342 Original commit message from CVS:
32345 2006-05-10 15:14:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32347 * gst/tcp/gstmultifdsink.c:
32348 * gst/tcp/gstmultifdsink.h:
32350 Original commit message from CVS:
32353 2006-05-10 11:54:36 +0000 Wim Taymans <wim.taymans@gmail.com>
32355 gst/adder/gstadder.*: Updated some docs. Added comments and FIXMEs all over the place.
32356 Original commit message from CVS:
32357 * gst/adder/gstadder.c: (gst_adder_setcaps),
32358 (gst_adder_query_duration), (gst_adder_query), (forward_event),
32359 (gst_adder_src_event), (gst_adder_sink_event),
32360 (gst_adder_class_init), (gst_adder_finalize),
32361 (gst_adder_request_new_pad), (gst_adder_collected):
32362 * gst/adder/gstadder.h:
32363 Updated some docs. Added comments and FIXMEs all over the place.
32364 Improve debugging info.
32365 Fix leak on finalize by not calling the parent.
32366 Implement duration query.
32367 Make event forwarding threadsafe.
32368 Correctly send NEWSEGMENT at start and after flush.
32369 Handle EOS correctly.
32370 Post error when not negotiated.
32371 * tests/check/elements/adder.c: (GST_START_TEST):
32372 Added FIXME in the test.
32374 2006-05-09 19:24:46 +0000 Tim-Philipp Müller <tim@centricular.net>
32376 Const-ify GEnumValue and GFlagsValue arrays. Use
32377 Original commit message from CVS:
32378 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
32379 (gst_text_overlay_halign_get_type),
32380 (gst_text_overlay_wrap_mode_get_type):
32381 * ext/theora/theoradec.c: (theora_handle_type_packet),
32382 (theora_handle_data_packet):
32383 * ext/theora/theoraenc.c: (gst_border_mode_get_type),
32384 (theora_enc_sink_setcaps), (theora_enc_chain):
32385 * gst-libs/gst/cdda/gstcddabasesrc.c:
32386 (gst_cdda_base_src_mode_get_type):
32387 * gst/audiotestsrc/gstaudiotestsrc.c:
32388 (gst_audiostestsrc_wave_get_type):
32389 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
32390 * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
32391 * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
32392 (gst_sync_method_get_type), (gst_unit_type_get_type),
32393 (gst_client_status_get_type):
32394 * gst/videoscale/gstvideoscale.c:
32395 (gst_video_scale_method_get_type):
32396 * gst/videotestsrc/gstvideotestsrc.c:
32397 (gst_video_test_src_pattern_get_type):
32398 * gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
32399 (paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
32400 (paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
32401 (paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
32402 (paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
32403 (paint_setup_RGB565), (paint_setup_xRGB1555):
32404 Const-ify GEnumValue and GFlagsValue arrays. Use
32405 GST_ROUND_UP_* macros instead of home-made ones.
32407 2006-05-09 17:40:41 +0000 Tim-Philipp Müller <tim@centricular.net>
32409 configure.ac: Require core CVS for the new newsegment stuff.
32410 Original commit message from CVS:
32412 Require core CVS for the new newsegment stuff.
32414 2006-05-09 17:30:48 +0000 Sjoerd Simons <sjoerd@luon.net>
32416 gst/tcp/gstmultifdsink.c: Register nick for enum value (#341160).
32417 Original commit message from CVS:
32418 Patch by: Sjoerd Simons <sjoerd at luon net>
32419 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
32420 Register nick for enum value (#341160).
32422 2006-05-09 16:46:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32424 gst/typefind/gsttypefindfunctions.c: backout typefind patch #340375
32425 Original commit message from CVS:
32426 * gst/typefind/gsttypefindfunctions.c: (m4a_type_find),
32428 backout typefind patch #340375
32429 * tests/check/elements/adder.c: (message_received),
32430 (GST_START_TEST), (adder_suite):
32431 redo, signal-handling of test
32433 2006-05-09 16:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
32435 gst/adder/gstadder.*: Remove bogus segment merging and forwarding, we don't care about timestamps anyway and we just ...
32436 Original commit message from CVS:
32437 * gst/adder/gstadder.c: (gst_adder_request_new_pad),
32438 (gst_adder_collected):
32439 * gst/adder/gstadder.h:
32440 Remove bogus segment merging and forwarding, we don't
32441 care about timestamps anyway and we just produce a
32443 Also create a nice NEWSEGMENT event when we start.
32444 Use _scale_int some more.
32446 2006-05-09 11:59:13 +0000 Edward Hervey <bilboed@bilboed.com>
32448 tests/icles/stress-xoverlay.c: Fix if core was built without parsing support.
32449 Original commit message from CVS:
32450 * tests/icles/stress-xoverlay.c:
32451 Fix if core was built without parsing support.
32453 2006-05-09 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
32455 gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
32456 Original commit message from CVS:
32457 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
32458 Add SEDG (Samsung MPEG-4) fourcc.
32460 2006-05-09 11:31:47 +0000 Edward Hervey <bilboed@bilboed.com>
32462 tests/examples/volume/volume.c: Fox if core was built without parsing support.
32463 Original commit message from CVS:
32464 * tests/examples/volume/volume.c:
32465 Fox if core was built without parsing support.
32466 * tests/examples/seek/seek.c:
32467 Disable the parse_launch example if core was built without parsing
32470 2006-05-09 11:21:24 +0000 Edward Hervey <bilboed@bilboed.com>
32472 tests/examples/seek/seek.c: Disable the parse_launch example if core was built without parsing support.
32473 Original commit message from CVS:
32474 * tests/examples/seek/seek.c:
32475 Disable the parse_launch example if core was built without parsing
32478 2006-05-08 15:51:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32480 * docs/libs/tmpl/gstcolorbalance.sgml:
32481 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
32482 * gst/tcp/gstmultifdsink.c:
32483 * gst/videoscale/gstvideoscale.c:
32484 doc reparagraphing and DEBUG_FUNCPTRing
32485 Original commit message from CVS:
32486 doc reparagraphing and DEBUG_FUNCPTRing
32488 2006-05-08 11:51:43 +0000 Edward Hervey <bilboed@bilboed.com>
32490 autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize
32491 Original commit message from CVS:
32492 * autogen.sh: (CONFIGURE_DEF_OPT):
32493 libtoolize on Darwin/MacOSX is called glibtoolize
32495 2006-05-07 17:39:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32497 tests/check/: Disable the adder test, until the build-slaves posses the kindness to either like it or to give valid r...
32498 Original commit message from CVS:
32499 * tests/check/Makefile.am:
32500 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
32501 Disable the adder test, until the build-slaves posses the kindness to
32502 either like it or to give valid reason for not doing so
32504 2006-05-07 17:25:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32506 tests/check/elements/adder.c: Shuffle NULL state change around and raise timeout more
32507 Original commit message from CVS:
32508 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
32510 Shuffle NULL state change around and raise timeout more
32512 2006-05-07 17:07:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32514 gst/typefind/gsttypefindfunctions.c: Add typefind to distinguish between "audio/x-m4a" and new type "video/mp4". Fixe...
32515 Original commit message from CVS:
32516 * gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
32517 (mp4_type_find), (plugin_init):
32518 Add typefind to distinguish between "audio/x-m4a" and new type
32519 "video/mp4". Fixes #340375
32520 * tests/check/elements/adder.c: (adder_suite):
32521 Raise timeout to make buildbot happy
32523 2006-05-07 16:39:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32525 Add sink-event handling to adder. It tries to merge incomming newsegment-events. Added test to check if segment_done ...
32526 Original commit message from CVS:
32527 * gst/adder/gstadder.c: (gst_adder_sink_event),
32528 (gst_adder_request_new_pad), (gst_adder_change_state):
32529 * gst/adder/gstadder.h:
32530 * tests/check/Makefile.am:
32531 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
32532 (adder_suite), (main):
32533 Add sink-event handling to adder. It tries to merge incomming
32534 newsegment-events. Added test to check if segment_done is comming
32537 2006-05-05 16:34:15 +0000 Andy Wingo <wingo@pobox.com>
32540 * ext/theora/theoraparse.c:
32541 * ext/vorbis/vorbisparse.c:
32542 ext/theora/theoraparse.c (gst_theora_parse_init) ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
32543 Original commit message from CVS:
32544 2006-05-05 Andy Wingo <wingo@pobox.com>
32545 * ext/theora/theoraparse.c (gst_theora_parse_init)
32546 (theora_parse_src_convert, theora_parse_src_query):
32547 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
32548 (vorbis_parse_convert, vorbis_parse_src_query): Add convert and
32549 query functions on the source pads of the theora and vorbis parse
32550 elements. Fixes position querying when doing a remux.
32552 2006-05-05 13:46:37 +0000 Michael Smith <msmith@xiph.org>
32554 ext/theora/theoraparse.c: Fix flushing.
32555 Original commit message from CVS:
32556 * ext/theora/theoraparse.c: (parse_granulepos),
32557 (theora_parse_drain_queue_prematurely),
32558 (theora_parse_queue_buffer), (theora_parse_sink_event):
32560 Fix invalid granulepos outputs when starting with a non-keyframe.
32562 2006-05-05 12:37:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32564 gst/typefind/gsttypefindfunctions.c: Rearrange MPEG system stream detection, fixing some memleaks in the process.
32565 Original commit message from CVS:
32566 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
32567 (mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
32568 Rearrange MPEG system stream detection, fixing some memleaks in the
32570 Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
32571 they clean up their data correctly.
32572 Remove unused ogganx caps and move the 'is_annodex' check to inside
32573 the 'is_ogg' if statement.
32575 2006-05-05 11:33:37 +0000 Wim Taymans <wim.taymans@gmail.com>
32577 gst/playback/gstdecodebin.c: Properly remove ghostpads. Fixes #340392
32578 Original commit message from CVS:
32579 * gst/playback/gstdecodebin.c: (cleanup_decodebin):
32580 Properly remove ghostpads. Fixes #340392
32582 2006-05-04 18:43:58 +0000 David Schleef <ds@schleef.org>
32584 gst/typefind/gsttypefindfunctions.c:
32585 Original commit message from CVS:
32586 * gst/typefind/gsttypefindfunctions.c:
32588 2006-05-03 16:32:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32590 gst/typefind/gsttypefindfunctions.c: When typefinding an MP3 in push-based mode, don't penalise the probability down ...
32591 Original commit message from CVS:
32592 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
32593 (mpeg_ts_probe_headers), (mpeg_ts_type_find):
32594 When typefinding an MP3 in push-based mode, don't penalise the
32595 probability down to 74% when we found 5 valid frames just because we
32596 can't peek the end of the file.
32597 Make the probability for detecting MPEG Transport Streams based on the
32598 number of sequential headers we successfully detected.
32600 2006-05-03 15:52:46 +0000 Wim Taymans <wim.taymans@gmail.com>
32602 ext/vorbis/vorbisdec.c: Still produce an error when we receive an empty packet.
32603 Original commit message from CVS:
32604 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
32605 (vorbis_dec_push), (vorbis_dec_chain):
32606 Still produce an error when we receive an empty packet.
32608 2006-05-03 15:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
32610 ext/ogg/gstoggdemux.c: Mark buffers with DISCONT after seek and after activating new chains.
32611 Original commit message from CVS:
32612 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
32613 (gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
32614 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
32615 Mark buffers with DISCONT after seek and after activating new
32617 * ext/theora/gsttheoradec.h:
32618 * ext/theora/theoradec.c: (gst_theora_dec_reset),
32619 (theora_get_query_types), (theora_dec_sink_event),
32620 (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
32621 (theora_dec_change_state):
32623 Detect and mark DISCONT buffers.
32624 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
32625 (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
32626 (vorbis_dec_change_state):
32627 * ext/vorbis/vorbisdec.h:
32629 Detect and mark DISCONT buffers.
32630 Don't crash on 0 sized buffers.
32632 2006-05-03 08:58:13 +0000 Wim Taymans <wim.taymans@gmail.com>
32634 gst/volume/gstvolume.c: Increase "volume" property to 10.0. Fixes #340369.
32635 Original commit message from CVS:
32636 * gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
32637 (volume_transform_ip):
32638 Increase "volume" property to 10.0. Fixes #340369.
32639 Set the process function to NULL when capsnego fails so that
32640 we properly error out.
32642 2006-05-02 18:15:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32644 gst/playback/: free cpas using gst_caps_unref, don't leak caps-strings
32645 Original commit message from CVS:
32646 * gst/playback/gstplaybin.c: (add_sink):
32647 * gst/playback/test.c: (main):
32648 * gst/playback/test5.c: (dump_element_stats):
32649 * gst/playback/test6.c: (main):
32650 free cpas using gst_caps_unref, don't leak caps-strings
32652 2006-05-02 06:33:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32654 * gst-libs/gst/rtp/gstbasertppayload.c:
32656 Original commit message from CVS:
32659 2006-05-01 19:08:40 +0000 Tim-Philipp Müller <tim@centricular.net>
32661 gst/typefind/gsttypefindfunctions.c: Refine musepack typefinding a bit. Return MAXIMUM probability when we detect str...
32662 Original commit message from CVS:
32663 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
32665 Refine musepack typefinding a bit. Return MAXIMUM
32666 probability when we detect stream version 7 to make
32667 sure the mpeg audio typefinder doesn't trump us.
32669 2006-04-29 16:25:58 +0000 Tim-Philipp Müller <tim@centricular.net>
32671 gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
32672 Original commit message from CVS:
32673 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
32674 Protect against unexpected NULL strf_data buffer.
32676 2006-04-29 13:09:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32678 tests/check/elements/audioconvert.c: interpret the out[] buffer in the order the bytes are actually put in, which is ...
32679 Original commit message from CVS:
32680 * tests/check/elements/audioconvert.c: (verify_convert),
32682 interpret the out[] buffer in the order the bytes are actually
32683 put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
32684 Other tests should use BYTE_ORDER since the array is filled in
32687 2006-04-29 12:10:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32689 * tests/check/elements/audioconvert.c:
32690 dump expected data when audioconvert test fails
32691 Original commit message from CVS:
32692 dump expected data when audioconvert test fails
32694 2006-04-29 11:55:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32696 tests/check/elements/audioconvert.c: when a test fails, give an indication of which it is
32697 Original commit message from CVS:
32698 * tests/check/elements/audioconvert.c: (verify_convert),
32700 when a test fails, give an indication of which it is
32702 2006-04-29 09:48:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32704 * ext/ogg/gstoggmux.c:
32705 * ext/theora/theoraenc.c:
32706 add another include
32707 Original commit message from CVS:
32708 add another include
32710 2006-04-29 01:24:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32712 * gst/subparse/gstssaparse.c:
32713 atoi() needs stdlib.h
32714 Original commit message from CVS:
32715 atoi() needs stdlib.h
32717 2006-04-29 01:18:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32719 * gst/playback/test4.c:
32720 * gst/playback/test5.c:
32721 * gst/playback/test6.c:
32722 exit needs stdlib.h
32723 Original commit message from CVS:
32724 exit needs stdlib.h
32726 2006-04-29 01:10:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32728 gst-libs/gst/cdda/gstcddabasesrc.c: compile fix; strtol() needs <stdlib.h>
32729 Original commit message from CVS:
32730 * gst-libs/gst/cdda/gstcddabasesrc.c:
32731 compile fix; strtol() needs <stdlib.h>
32733 2006-04-29 01:04:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32737 * docs/Makefile.am:
32738 * docs/libs/Makefile.am:
32739 * docs/libs/tmpl/gstcolorbalance.sgml:
32740 * docs/plugins/Makefile.am:
32742 use common upload.mak
32743 Original commit message from CVS:
32744 use common upload.mak
32746 2006-04-28 19:46:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32748 make GstElementDetails const
32749 Original commit message from CVS:
32750 * ext/alsa/gstalsamixerelement.c:
32751 * ext/alsa/gstalsasrc.c:
32752 * ext/cdparanoia/gstcdparanoiasrc.c:
32753 * ext/gnomevfs/gstgnomevfssink.c:
32754 * ext/gnomevfs/gstgnomevfssrc.c:
32755 * ext/ogg/gstoggdemux.c:
32756 * ext/ogg/gstoggmux.c:
32757 * ext/ogg/gstoggparse.c:
32758 * ext/ogg/gstogmparse.c:
32759 * ext/pango/gstclockoverlay.c:
32760 * ext/pango/gsttextoverlay.c:
32761 * ext/pango/gsttextrender.c:
32762 * ext/pango/gsttimeoverlay.c:
32763 * ext/theora/theoradec.c:
32764 * ext/theora/theoraenc.c:
32765 * ext/vorbis/vorbisdec.c:
32766 * ext/vorbis/vorbisenc.c:
32767 * gst-libs/gst/audio/gstaudiofilter.c:
32768 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
32769 * gst/audioconvert/gstaudioconvert.c:
32770 * gst/audiorate/gstaudiorate.c:
32771 * gst/audioresample/gstaudioresample.c:
32772 * gst/audiotestsrc/gstaudiotestsrc.c:
32773 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
32774 * gst/playback/gstdecodebin.c:
32775 * gst/playback/gstplaybin.c:
32776 * gst/playback/gststreamselector.c:
32777 * gst/subparse/gstsubparse.c:
32778 * gst/tcp/gstmultifdsink.c:
32779 * gst/tcp/gsttcpclientsink.c:
32780 * gst/tcp/gsttcpclientsrc.c:
32781 * gst/tcp/gsttcpserversink.c:
32782 * gst/tcp/gsttcpserversrc.c:
32783 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
32784 * gst/videorate/gstvideorate.c:
32785 * gst/videoscale/gstvideoscale.c:
32786 * gst/videotestsrc/gstvideotestsrc.c:
32787 * gst/volume/gstvolume.c:
32788 * sys/v4l/gstv4ljpegsrc.c:
32789 * sys/v4l/gstv4lmjpegsink.c:
32790 * sys/v4l/gstv4lmjpegsrc.c:
32791 * sys/v4l/gstv4lsrc.c:
32792 * sys/ximage/ximagesink.c:
32793 * sys/xvimage/xvimagesink.c:
32794 * tests/check/libs/cddabasesrc.c:
32795 make GstElementDetails const
32797 2006-04-28 19:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32799 gst/adder/gstadder.c: send events from src-pad to all sink-pads fixes #338657
32800 Original commit message from CVS:
32801 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
32803 send events from src-pad to all sink-pads fixes #338657
32805 2006-04-28 19:08:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32807 ext/alsa/gstalsasink.c: query witdh capabilities from alsa, fixes #338919
32808 Original commit message from CVS:
32809 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
32810 (alsasink_parse_spec):
32811 query witdh capabilities from alsa, fixes #338919
32813 2006-04-28 15:31:28 +0000 Wim Taymans <wim.taymans@gmail.com>
32815 gst/tcp/gstmultifdsink.*: Fix race condition in multifdsink that can lead to spurious duplicate clients. this patch a...
32816 Original commit message from CVS:
32817 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
32818 (gst_multi_fd_sink_remove_client_link):
32819 * gst/tcp/gstmultifdsink.h:
32820 Fix race condition in multifdsink that can lead to spurious
32821 duplicate clients. this patch adds a new signal that is fired when
32822 multifdsink has removed all references to the fd.
32824 Updated documentation.
32825 API: client-fd-removed signal added
32827 2006-04-28 15:24:00 +0000 Michael Smith <msmith@xiph.org>
32829 gst/tcp/gstmultifdsink.c: When asking g_value_array_new to prealloc elements, we may as well ask for the right number...
32830 Original commit message from CVS:
32831 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
32832 When asking g_value_array_new to prealloc elements, we may as well
32833 ask for the right number of elements.
32835 2006-04-28 15:08:09 +0000 Wim Taymans <wim.taymans@gmail.com>
32837 gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
32838 Original commit message from CVS:
32839 * gst-libs/gst/audio/gstbaseaudiosink.c:
32840 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
32841 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
32842 patch to make timestamp checking more tollerant to rounding
32843 errors given that real discontinuities are to be marked on
32844 buffers. Fixes some asf files and #338778.
32845 Also avoid some crashers when we receive an event in the
32848 2006-04-28 15:01:58 +0000 Michael Smith <msmith@xiph.org>
32850 ext/gnomevfs/gstgnomevfssrc.*: Remove ICY handling (mostly) from gnomevfssrc, in favour of proper shared support with...
32851 Original commit message from CVS:
32852 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
32853 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
32854 (gst_gnome_vfs_src_get_property),
32855 (gst_gnome_vfs_src_send_additional_headers_callback),
32856 (gst_gnome_vfs_src_received_headers_callback),
32857 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
32858 (gst_gnome_vfs_src_stop):
32859 * ext/gnomevfs/gstgnomevfssrc.h:
32860 Remove ICY handling (mostly) from gnomevfssrc, in favour of
32861 proper shared support within icydemux.
32863 2006-04-28 14:49:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32865 gst/videorate/gstvideorate.c: fix up docs fix a leak when no caps negotiated fix counting of input frames
32866 Original commit message from CVS:
32867 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
32868 (gst_video_rate_swap_prev), (gst_video_rate_chain):
32870 fix a leak when no caps negotiated
32871 fix counting of input frames
32872 * tests/check/elements/.cvsignore:
32873 * tests/check/elements/videorate.c: (assert_videorate_stats),
32874 (GST_START_TEST), (videorate_suite):
32875 add tests for these
32877 2006-04-28 14:48:11 +0000 Wim Taymans <wim.taymans@gmail.com>
32879 gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
32880 Original commit message from CVS:
32881 * gst-libs/gst/audio/gstringbuffer.c:
32882 (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
32883 (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
32884 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
32885 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
32886 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
32887 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
32888 (gst_ring_buffer_commit), (gst_ring_buffer_read),
32889 (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
32890 (gst_ring_buffer_clear), (gst_ring_buffer_may_start):
32891 Check arguments passed to public functions instead of
32894 2006-04-28 14:37:46 +0000 Wim Taymans <wim.taymans@gmail.com>
32896 gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
32897 Original commit message from CVS:
32898 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
32899 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
32900 GstBaseAudioSrc must be live or it does not work.
32901 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
32902 Don't set live to TRUE as this is the default in the parentclass.
32904 2006-04-28 14:37:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32906 * win32/common/config.h:
32908 Original commit message from CVS:
32911 2006-04-28 14:33:45 +0000 Wim Taymans <wim.taymans@gmail.com>
32913 gst/videoscale/gstvideoscale.c: Videoscale doesn't pass on pixel-aspect ratio. Handle all fixation cases better. Fixe...
32914 Original commit message from CVS:
32915 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps),
32916 (gst_video_scale_fixate_caps), (gst_video_scale_src_event):
32917 Videoscale doesn't pass on pixel-aspect ratio. Handle all
32918 fixation cases better. Fixes #338991
32920 2006-04-28 14:24:38 +0000 Wim Taymans <wim.taymans@gmail.com>
32922 gst/videotestsrc/gstvideotestsrc.c: Handle 0/1 framerate correctly Fixes #331901.
32923 Original commit message from CVS:
32924 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
32925 Handle 0/1 framerate correctly Fixes #331901.
32927 2006-04-28 14:22:16 +0000 Wim Taymans <wim.taymans@gmail.com>
32929 tests/check/elements/audioconvert.c: Added check for correct clipping when doing float samples in audioconvert.
32930 Original commit message from CVS:
32931 * tests/check/elements/audioconvert.c: (get_float_caps),
32932 (GST_START_TEST), (audioconvert_suite):
32933 Added check for correct clipping when doing float samples
32936 2006-04-28 14:19:49 +0000 Wim Taymans <wim.taymans@gmail.com>
32938 gst/videorate/gstvideorate.c: Print more debugging info.
32939 Original commit message from CVS:
32940 * gst/videorate/gstvideorate.c: (gst_video_rate_event),
32941 (gst_video_rate_chain):
32942 Print more debugging info.
32944 2006-04-28 14:17:00 +0000 Wim Taymans <wim.taymans@gmail.com>
32946 gst/audioresample/gstaudioresample.c: Add support for other formats audioresample can handle such as 32 bits in and f...
32947 Original commit message from CVS:
32948 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
32949 (resample_set_state_from_caps):
32950 Add support for other formats audioresample can handle such as
32951 32 bits in and float and 64 bits float. Fixes #301759
32953 2006-04-28 14:12:28 +0000 Wim Taymans <wim.taymans@gmail.com>
32955 gst/audioconvert/audioconvert.c: correctly clip float samples > 1.0. Fixes #338718
32956 Original commit message from CVS:
32957 * gst/audioconvert/audioconvert.c: (float):
32958 correctly clip float samples > 1.0. Fixes #338718
32960 2006-04-28 13:35:34 +0000 Young-Ho Cha <ganadist@chollian.net>
32962 ext/pango/gsttextoverlay.c: Don't strip newlines from the text. Also, center lines within multi-line paragraphs (#339...
32963 Original commit message from CVS:
32964 Patch by: Young-Ho Cha <ganadist at chollian net>
32965 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
32966 (gst_text_overlay_render_text):
32967 Don't strip newlines from the text. Also, center lines
32968 within multi-line paragraphs (#339405).
32970 2006-04-28 12:15:33 +0000 Tim-Philipp Müller <tim@centricular.net>
32972 gst/typefind/gsttypefindfunctions.c: Fix wavpack typefinding to work in more cases (don't peek for chunks of multiple...
32973 Original commit message from CVS:
32974 * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
32975 Fix wavpack typefinding to work in more cases (don't peek
32976 for chunks of multiple hundred kBs at once, but process
32977 things step-by-step in smaller units). Fixes #339786.
32979 2006-04-28 10:58:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32984 Original commit message from CVS:
32987 === release 0.10.6 ===
32989 2006-04-28 10:53:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32995 * docs/plugins/gst-plugins-base-plugins.signals:
32996 * docs/plugins/inspect/plugin-adder.xml:
32997 * docs/plugins/inspect/plugin-alsa.xml:
32998 * docs/plugins/inspect/plugin-audioconvert.xml:
32999 * docs/plugins/inspect/plugin-audiorate.xml:
33000 * docs/plugins/inspect/plugin-audioresample.xml:
33001 * docs/plugins/inspect/plugin-audiotestsrc.xml:
33002 * docs/plugins/inspect/plugin-cdparanoia.xml:
33003 * docs/plugins/inspect/plugin-decodebin.xml:
33004 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
33005 * docs/plugins/inspect/plugin-gnomevfs.xml:
33006 * docs/plugins/inspect/plugin-libvisual.xml:
33007 * docs/plugins/inspect/plugin-ogg.xml:
33008 * docs/plugins/inspect/plugin-pango.xml:
33009 * docs/plugins/inspect/plugin-playbin.xml:
33010 * docs/plugins/inspect/plugin-subparse.xml:
33011 * docs/plugins/inspect/plugin-tcp.xml:
33012 * docs/plugins/inspect/plugin-theora.xml:
33013 * docs/plugins/inspect/plugin-typefindfunctions.xml:
33014 * docs/plugins/inspect/plugin-video4linux.xml:
33015 * docs/plugins/inspect/plugin-videorate.xml:
33016 * docs/plugins/inspect/plugin-videoscale.xml:
33017 * docs/plugins/inspect/plugin-videotestsrc.xml:
33018 * docs/plugins/inspect/plugin-volume.xml:
33019 * docs/plugins/inspect/plugin-vorbis.xml:
33020 * docs/plugins/inspect/plugin-ximagesink.xml:
33021 * docs/plugins/inspect/plugin-xvimagesink.xml:
33024 Original commit message from CVS:
33027 2006-04-28 10:42:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33030 * win32/common/config.h:
33031 dist more win32 files
33032 Original commit message from CVS:
33033 dist more win32 files
33035 2006-04-28 10:41:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33052 Original commit message from CVS:
33055 2006-04-27 00:19:29 +0000 David Schleef <ds@schleef.org>
33057 gst/videoscale/gstvideoscale.c: Add call to oil_init().
33058 Original commit message from CVS:
33059 * gst/videoscale/gstvideoscale.c: Add call to oil_init().
33062 2006-04-26 17:20:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33066 * win32/common/config.h:
33068 Original commit message from CVS:
33071 2006-04-26 17:17:39 +0000 Wim Taymans <wim.taymans@gmail.com>
33073 ext/ogg/gstoggdemux.c: make sure correct newsegments are sent, so that the decoder and the demuxer agree on timestamp...
33074 Original commit message from CVS:
33075 2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
33076 patch by: Wim Taymans
33077 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
33078 (gst_ogg_demux_perform_seek):
33079 make sure correct newsegments are sent, so that the decoder
33080 and the demuxer agree on timestamps. Fixes playback of a lot
33081 of Ogg files that do not start from 0. Fixes #339833.
33083 2006-04-26 16:44:20 +0000 Edward Hervey <edward@fluendo.com>
33085 Fix an infinite loop if frames are passed in with wrongly ordered timestamps. Fixes #339013.
33086 Original commit message from CVS:
33087 Patch by: Edward Hervey <edward@fluendo.com>
33088 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
33089 * tests/check/Makefile.am:
33090 * tests/check/elements/videorate.c: (assert_videorate_stats),
33091 (setup_videorate), (cleanup_videorate), (GST_START_TEST),
33092 (videorate_suite), (main):
33093 Fix an infinite loop if frames are passed in with wrongly ordered
33094 timestamps. Fixes #339013.
33096 2006-04-26 13:55:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33099 * win32/common/config.h:
33101 Original commit message from CVS:
33104 2006-04-22 21:25:41 +0000 Tim-Philipp Müller <tim@centricular.net>
33106 gst/typefind/gsttypefindfunctions.c: fix typefinding on some ISO files. Fixes #339212.
33107 Original commit message from CVS:
33108 Patch by: Tim-Philipp Müller <tim at centricular dot net>
33109 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
33110 fix typefinding on some ISO files. Fixes #339212.
33112 2006-04-22 21:19:06 +0000 Tim-Philipp Müller <tim@centricular.net>
33114 gst-libs/gst/riff/riff-media.c: add another H264 fourcc. Fixes #339047.
33115 Original commit message from CVS:
33116 Patch by: Tim-Philipp Müller <tim at centricular dot net>
33117 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
33118 add another H264 fourcc. Fixes #339047.
33120 2006-04-22 21:12:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33122 gst/playback/gststreamselector.c: Restore old StreamSelector behaviour.
33123 Original commit message from CVS:
33124 Patch by: Jan Schmidt
33125 * gst/playback/gststreamselector.c:
33126 (gst_stream_selector_bufferalloc):
33127 Restore old StreamSelector behaviour.
33130 2006-04-13 09:26:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33133 * gst-libs/gst/rtp/Makefile.am:
33134 * gst-libs/gst/rtp/gstrtpbuffer.h:
33135 reverting rtp patches to fix freeze break on -base as explained on the list
33136 Original commit message from CVS:
33137 reverting rtp patches to fix freeze break on -base as explained on the list
33139 2006-04-13 03:55:12 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
33141 gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
33142 Original commit message from CVS:
33143 2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
33144 * gst-libs/gst/rtp/gstrtpbuffer.h:
33145 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
33146 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
33147 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
33148 New RTP audio base payloader class. Supports frame or sample based codecs
33150 2006-04-12 11:04:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33168 update libtool versioning
33169 Original commit message from CVS:
33170 update libtool versioning
33172 2006-04-12 10:58:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33176 * win32/common/config.h:
33178 Original commit message from CVS:
33181 2006-04-11 17:31:29 +0000 Antoine Tremblay <hexa00@gmail.com>
33183 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
33184 Original commit message from CVS:
33185 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
33186 * gst-libs/gst/rtp/gstbasertpdepayload.c:
33187 (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
33188 Fix some memory leaks: on finalize, free buffers left in the queue
33189 before destroying the queue; in _push(), unref rtp_buf even if
33190 the process vfunc returned a NULL buffer as output buffer (#337548);
33191 demote some recuring debug messages to LOG level.
33193 2006-04-11 15:01:51 +0000 Christian Schaller <uraeus@gnome.org>
33195 * gst-plugins-base.spec.in:
33196 fix version number macro
33197 Original commit message from CVS:
33198 fix version number macro
33200 2006-04-11 14:42:33 +0000 Wim Taymans <wim.taymans@gmail.com>
33202 ext/ogg/gstoggdemux.c: More cleanups.
33203 Original commit message from CVS:
33204 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
33205 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
33206 (gst_ogg_chain_free), (gst_ogg_demux_sink_event),
33207 (gst_ogg_demux_loop):
33209 Respect segment stop when emiting EOS or SEGMENT_DONE.
33212 2006-04-11 10:45:32 +0000 Tim-Philipp Müller <tim@centricular.net>
33214 gst/playback/gststreamselector.c: Don't leak pad name.
33215 Original commit message from CVS:
33216 * gst/playback/gststreamselector.c:
33217 (gst_stream_selector_get_property):
33218 Don't leak pad name.
33220 2006-04-11 09:42:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33223 Mention bug #336617 closed by recent commit
33224 Original commit message from CVS:
33225 Mention bug #336617 closed by recent commit
33227 2006-04-10 20:32:46 +0000 Michael Smith <msmith@xiph.org>
33229 tests/check/: so that FC4 buildslaves can pass.
33230 Original commit message from CVS:
33231 * tests/check/Makefile.am:
33232 * tests/check/gst-plugins-base.supp:
33233 Suppress an old libtheora bug (fixed in more recent versions), so
33234 that FC4 buildslaves can pass.
33236 2006-04-10 19:13:30 +0000 Wim Taymans <wim.taymans@gmail.com>
33238 ext/ogg/gstoggdemux.c: Don't leak events.
33239 Original commit message from CVS:
33240 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
33241 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
33242 (gst_ogg_demux_init), (gst_ogg_demux_finalize),
33243 (gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
33244 (gst_ogg_demux_loop):
33246 Remember what error we got when finding chains, if we
33247 were shutdown, that would not be an error.
33249 2006-04-10 17:05:46 +0000 Wim Taymans <wim.taymans@gmail.com>
33251 gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
33252 Original commit message from CVS:
33253 * gst-libs/gst/audio/gstbaseaudiosink.c:
33254 (gst_base_audio_sink_event):
33255 Starting the ringbuffer when we did not acquire it can cause
33256 a deadlock, is pointless and causes nasty things for
33258 Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
33260 2006-04-10 15:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
33262 ext/ogg/gstoggdemux.c: Add some more debugging.
33263 Original commit message from CVS:
33264 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
33265 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
33266 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
33267 (gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
33268 (gst_ogg_demux_deactivate_current_chain),
33269 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
33270 (gst_ogg_demux_bisect_forward_serialno),
33271 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
33272 Add some more debugging.
33274 2006-04-10 14:52:10 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33277 * ext/theora/theoraenc.c:
33279 Original commit message from CVS:
33282 2006-04-10 10:29:21 +0000 Wim Taymans <wim.taymans@gmail.com>
33284 ext/theora/theoradec.c: Some more debug info.
33285 Original commit message from CVS:
33286 * ext/theora/theoradec.c: (theora_dec_src_event),
33287 (theora_handle_data_packet):
33288 Some more debug info.
33289 * tests/examples/seek/seek.c: (start_seek), (main):
33290 Print element messages too.
33292 2006-04-09 17:14:22 +0000 Sébastien Moutte <sebastien@moutte.net>
33294 gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer sta...
33295 Original commit message from CVS:
33296 * gst/audioresample/debug.h:
33297 replace debug macros with variable number of parameters
33298 by a simple alias to gstreamer standard debug macros
33299 (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
33300 supported by MSVC 6.0 and 7.1)
33301 * gst/audioresample/resample.h:
33302 define M_PI and rint for WIN32
33303 * win32/common/libgstaudio.def:
33304 * win32/common/libgstriff.def:
33305 * win32/common/libgsttag.def:
33306 * win32/common/libgstvideo.def:
33307 add new exported functions
33309 update project files
33311 2006-04-08 21:02:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33313 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
33314 Original commit message from CVS:
33315 * ext/alsa/gstalsamixeroptions.c:
33316 (gst_alsa_mixer_options_class_init):
33317 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
33318 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
33319 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
33320 * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
33321 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
33322 * gst-libs/gst/audio/gstaudiofilter.c:
33323 (gst_audio_filter_class_init):
33324 * gst-libs/gst/audio/gstaudiosink.c:
33325 (gst_audioringbuffer_class_init):
33326 * gst-libs/gst/audio/gstaudiosrc.c:
33327 (gst_audioringbuffer_class_init):
33328 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
33329 * gst-libs/gst/interfaces/colorbalancechannel.c:
33330 (gst_color_balance_channel_class_init):
33331 * gst-libs/gst/interfaces/mixeroptions.c:
33332 (gst_mixer_options_class_init):
33333 * gst-libs/gst/interfaces/mixertrack.c:
33334 (gst_mixer_track_class_init):
33335 * gst-libs/gst/interfaces/tunerchannel.c:
33336 (gst_tuner_channel_class_init):
33337 * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
33338 * gst-libs/gst/netbuffer/gstnetbuffer.c:
33339 (gst_netbuffer_class_init):
33340 * gst-libs/gst/rtp/gstbasertppayload.c:
33341 (gst_basertppayload_class_init):
33342 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
33343 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
33344 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
33345 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
33346 * gst/playback/gststreamselector.c:
33347 (gst_stream_selector_class_init):
33348 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
33349 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
33350 * sys/v4l/gstv4lcolorbalance.c:
33351 (gst_v4l_color_balance_channel_class_init):
33352 * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
33353 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
33354 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
33355 * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
33356 (gst_v4l_tuner_norm_class_init):
33357 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
33358 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
33359 * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
33360 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
33362 2006-04-08 18:09:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33364 Fix broken GObject macros
33365 Original commit message from CVS:
33366 * ext/pango/gsttextrender.h:
33367 * gst-libs/gst/audio/gstaudiosink.h:
33368 * gst-libs/gst/audio/gstaudiosrc.h:
33369 * gst-libs/gst/audio/gstbaseaudiosink.h:
33370 * gst-libs/gst/audio/gstbaseaudiosrc.h:
33371 * gst-libs/gst/audio/gstringbuffer.h:
33372 * gst-libs/gst/rtp/gstbasertpdepayload.h:
33373 * gst-libs/gst/rtp/gstbasertppayload.h:
33374 * gst-libs/gst/video/gstvideofilter.h:
33375 * gst-libs/gst/video/gstvideosink.h:
33376 * gst/playback/gstplaybasebin.h:
33377 * gst/tcp/gstmultifdsink.h:
33378 * sys/v4l/gstv4lelement.h:
33379 Fix broken GObject macros
33381 2006-04-08 16:21:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33383 ext/alsa/gstalsasink.c: More debug to trace why my USB headset is not working with gst
33384 Original commit message from CVS:
33385 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
33386 More debug to trace why my USB headset is not working with gst
33388 2006-04-07 17:18:11 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33390 gst/playback/gstplaybasebin.c: Clean up our group elements properly in the case where it never got committed - it sti...
33391 Original commit message from CVS:
33392 * gst/playback/gstplaybasebin.c: (group_destroy):
33393 Clean up our group elements properly in the case where it never
33394 got committed - it still got added unconditionally to the bin.
33396 2006-04-07 15:14:32 +0000 Wim Taymans <wim.taymans@gmail.com>
33398 ext/theora/theoradec.c: Unref unhandled events.
33399 Original commit message from CVS:
33400 * ext/theora/theoradec.c: (theora_dec_sink_event),
33401 (theora_handle_data_packet), (theora_dec_chain):
33402 Unref unhandled events.
33403 Protect against empty buffers.
33404 Perform QoS on running time.
33406 2006-04-07 13:24:54 +0000 Michael Smith <msmith@xiph.org>
33408 ext/vorbis/vorbisenc.c: Remove leaks from vorbisenc.
33409 Original commit message from CVS:
33410 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
33411 (gst_vorbis_enc_chain):
33412 Remove leaks from vorbisenc.
33413 Mostly minor changes, the only significant one is that now the
33414 buffers we set as 'streamheader' on the caps are copies of the
33415 original buffers, to avoid circular refcounting problems.
33417 2006-04-07 09:51:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33419 gst/playback/gstplaybasebin.c: Don't remove our mute-probe if someone else already did so.
33420 Original commit message from CVS:
33421 * gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
33422 Don't remove our mute-probe if someone else already did so.
33423 Don't set a 2nd one if there is already one pending on the pad.
33424 * gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
33426 When a seek fails, ensure that playbin is still set back to playing.
33427 * gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
33428 (mpeg_ts_type_find), (plugin_init):
33429 Add a typefind function for mpeg-ts streams.
33431 2006-04-06 11:40:45 +0000 Andy Wingo <wingo@pobox.com>
33434 * gst/audiotestsrc/gstaudiotestsrc.c:
33435 * gst/videorate/gstvideorate.c:
33436 gst/videorate/gstvideorate.c (gst_video_rate_reset)
33437 Original commit message from CVS:
33438 2006-04-06 Andy Wingo <wingo@pobox.com>
33439 * gst/videorate/gstvideorate.c (gst_video_rate_reset)
33440 (gst_video_rate_init): Caps-related parameters should not be reset
33441 by a flush -- move their inits to the instance init function.
33442 (gst_video_rate_flush_prev): Don't complain if gst_pad_push
33443 is not OK, just return the result.
33444 * gst/audiotestsrc/gstaudiotestsrc.c
33445 (gst_audio_test_src_class_init)
33446 (gst_audio_test_src_get_times): Re-enable is-live=true, as was
33447 broken by Stefan's commit on 24 March.
33449 2006-04-06 10:50:14 +0000 Andy Wingo <wingo@pobox.com>
33451 ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on buffers being pushed out. Fixes oggmux ! multifdsink.
33452 Original commit message from CVS:
33453 2006-04-06 Andy Wingo <wingo@pobox.com>
33454 * ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
33455 buffers being pushed out. Fixes oggmux ! multifdsink.
33457 2006-04-05 13:05:25 +0000 Tim-Philipp Müller <tim@centricular.net>
33459 ext/vorbis/: Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make vorbisenc adhere to the official nomenclature; u...
33460 Original commit message from CVS:
33461 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
33462 (gst_vorbis_dec_init), (vorbis_dec_finalize):
33463 * ext/vorbis/vorbisdec.h:
33464 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_add_interfaces),
33465 (gst_vorbis_enc_base_init), (gst_vorbis_enc_class_init),
33466 (gst_vorbis_enc_sink_setcaps), (gst_vorbis_enc_convert_src),
33467 (gst_vorbis_enc_convert_sink), (gst_vorbis_enc_get_query_types),
33468 (gst_vorbis_enc_src_query), (gst_vorbis_enc_sink_query),
33469 (gst_vorbis_enc_init), (gst_vorbis_enc_get_tag_value),
33470 (gst_vorbis_enc_metadata_set1), (gst_vorbis_enc_set_metadata),
33471 (gst_vorbis_enc_setup), (gst_vorbis_enc_clear),
33472 (gst_vorbis_enc_buffer_from_packet),
33473 (gst_vorbis_enc_buffer_from_header_packet),
33474 (gst_vorbis_enc_push_buffer), (gst_vorbis_enc_push_packet),
33475 (gst_vorbis_enc_set_header_on_caps), (gst_vorbis_enc_sink_event),
33476 (gst_vorbis_enc_chain), (gst_vorbis_enc_output_buffers),
33477 (gst_vorbis_enc_get_property), (gst_vorbis_enc_set_property),
33478 (gst_vorbis_enc_change_state):
33479 * ext/vorbis/vorbisenc.h:
33480 Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make
33481 vorbisenc adhere to the official nomenclature; use boilerplate
33484 2006-04-04 11:20:24 +0000 Andy Wingo <wingo@pobox.com>
33486 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Whoops, fix bug introduced. Bad hacker!
33487 Original commit message from CVS:
33488 2006-04-04 Andy Wingo <wingo@pobox.com>
33489 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
33490 Whoops, fix bug introduced. Bad hacker!
33492 2006-04-04 11:15:00 +0000 Andy Wingo <wingo@pobox.com>
33494 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Properly handle the case where you get EOS before any buffe...
33495 Original commit message from CVS:
33496 2006-04-04 Andy Wingo <wingo@pobox.com>
33497 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
33498 Properly handle the case where you get EOS before any buffers are
33499 received. Use gst_buffer_make_metadata_writable where appropriate.
33501 2006-04-04 10:16:46 +0000 Andy Wingo <wingo@pobox.com>
33503 ext/theora/theoradec.c (theora_handle_data_packet): This value is often negative -- make it signed so as not to wrap ...
33504 Original commit message from CVS:
33505 2006-04-04 Andy Wingo <wingo@pobox.com>
33506 * ext/theora/theoradec.c (theora_handle_data_packet): This value
33507 is often negative -- make it signed so as not to wrap around.
33508 Fixes segfaults introduced on 9 March.
33510 2006-04-03 16:43:10 +0000 Wim Taymans <wim.taymans@gmail.com>
33512 ext/theora/: Don't try to store a gdouble in a gboolean.
33513 Original commit message from CVS:
33514 * ext/theora/gsttheoradec.h:
33515 * ext/theora/theoradec.c: (theora_dec_src_event):
33516 Don't try to store a gdouble in a gboolean.
33519 2006-04-03 12:55:18 +0000 Michael Smith <msmith@xiph.org>
33521 ext/ogg/gstoggmux.c: Oggmux sucks.
33522 Original commit message from CVS:
33523 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
33525 Make it suck slightly less by writing out the final page.
33526 Still can't encode a vorbis-in-ogg file correctly, though.
33528 2006-04-03 08:49:06 +0000 Andy Wingo <wingo@pobox.com>
33530 ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove a g_print.
33531 Original commit message from CVS:
33532 2006-04-03 Andy Wingo <wingo@pobox.com>
33533 * ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
33536 2006-04-03 08:32:21 +0000 Andy Wingo <wingo@pobox.com>
33538 ext/theora/theora.c (plugin_init): Register theoraparse.
33539 Original commit message from CVS:
33540 2006-04-03 Andy Wingo <wingo@pobox.com>
33541 * ext/theora/theora.c (plugin_init): Register theoraparse.
33542 * ext/theora/gsttheoraparse.h:
33543 * ext/theora/theoraparse.c: New files implementing a theora
33544 parser. Now we can properly remux ogg/theora+vorbis, yay.
33546 2006-04-03 08:28:58 +0000 Andy Wingo <wingo@pobox.com>
33548 ext/vorbis/vorbisparse.c: Add some docs and a copyright.
33549 Original commit message from CVS:
33550 2006-04-03 Andy Wingo <wingo@pobox.com>
33551 * ext/vorbis/vorbisparse.c: Add some docs and a copyright.
33553 2006-04-01 15:34:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33557 don't use AS_LIBTOOL_TAGS, it doesn't work
33558 Original commit message from CVS:
33559 don't use AS_LIBTOOL_TAGS, it doesn't work
33561 2006-04-01 11:41:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33564 * ext/pango/gsttextoverlay.c:
33565 * sys/v4l/gstv4lsrc.c:
33566 remove BT8x8 from description, works for more devices
33567 Original commit message from CVS:
33568 remove BT8x8 from description, works for more devices
33570 2006-04-01 11:21:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33572 gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798)
33573 Original commit message from CVS:
33574 * gst/audiotestsrc/gstaudiotestsrc.c:
33575 Fixed the sample pipeline (see #323798)
33577 2006-04-01 09:50:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33579 use AS_VERSION and AS_NANO more cleanups
33580 Original commit message from CVS:
33582 * win32/common/config.h:
33583 * win32/common/config.h.in:
33584 use AS_VERSION and AS_NANO
33587 2006-03-31 17:08:41 +0000 Andy Wingo <wingo@pobox.com>
33589 ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix uninitialized variable return that would happen.
33590 Original commit message from CVS:
33591 2006-03-31 Andy Wingo <wingo@pobox.com>
33592 * ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
33593 uninitialized variable return that would happen.
33595 2006-03-31 16:57:47 +0000 Andy Wingo <wingo@pobox.com>
33597 ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix uninitialized variable return that would never happen.
33598 Original commit message from CVS:
33599 2006-03-31 Andy Wingo <wingo@pobox.com>
33600 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
33601 uninitialized variable return that would never happen.
33603 2006-03-31 16:43:43 +0000 Andy Wingo <wingo@pobox.com>
33605 ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
33606 Original commit message from CVS:
33607 2006-03-31 Andy Wingo <wingo@pobox.com>
33608 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
33609 (vorbis_parse_sink_event): Add an event function to flush our
33610 state on a seek, and to drain buffers on a premature EOS.
33611 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
33612 (vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
33613 (vorbis_parse_chain, vorbis_parse_queue_buffer)
33614 (vorbis_parse_drain_queue): Queue up buffers until we can set
33615 their timestamps and granulepos values.
33616 * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
33617 and keep track of data needed for deriving granulepos and
33618 timestamps for buffers.
33620 2006-03-30 11:05:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33622 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
33623 * pkgconfig/gstreamer-plugins-base.pc.in:
33624 expose pluginsdir so gonlin can use it for tests
33625 Original commit message from CVS:
33626 expose pluginsdir so gonlin can use it for tests
33628 2006-03-30 10:03:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33630 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
33631 * pkgconfig/gstreamer-plugins-base.pc.in:
33632 add ccda to libraries
33633 Original commit message from CVS:
33634 add ccda to libraries
33636 2006-03-29 14:00:08 +0000 j^ <j@bootlab.org>
33638 better/unified long descriptions
33639 Original commit message from CVS:
33640 Patch by: j^ <j at bootlab dot org>
33641 * ext/alsa/gstalsamixerelement.c:
33642 (gst_alsa_mixer_element_class_init):
33643 * ext/alsa/gstalsasink.c:
33644 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
33645 * ext/ogg/gstoggdemux.c:
33646 * ext/ogg/gstoggmux.c:
33647 * ext/ogg/gstoggparse.c:
33648 * ext/pango/gstclockoverlay.c:
33649 * ext/pango/gsttextoverlay.c:
33650 * ext/pango/gsttextrender.c:
33651 * ext/pango/gsttimeoverlay.c:
33652 * ext/theora/theoradec.c:
33653 * ext/theora/theoraenc.c:
33654 * ext/vorbis/vorbisdec.c:
33655 * ext/vorbis/vorbisenc.c:
33656 * gst/audioconvert/gstaudioconvert.c:
33657 * gst/subparse/gstsubparse.c:
33658 * gst/tcp/gstmultifdsink.c:
33659 * gst/tcp/gsttcpclientsink.c:
33660 * gst/tcp/gsttcpclientsrc.c:
33661 * gst/tcp/gsttcpserversink.c:
33662 * gst/tcp/gsttcpserversrc.c:
33663 better/unified long descriptions
33666 2006-03-29 13:54:24 +0000 Wim Taymans <wim.taymans@gmail.com>
33668 tests/examples/seek/seek.c: Don't let double and tripple clicks mess up our state.
33669 Original commit message from CVS:
33670 * tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
33672 Don't let double and tripple clicks mess up our state.
33674 2006-03-28 13:13:43 +0000 Tim-Philipp Müller <tim@centricular.net>
33676 gst/playback/gstplaybin.c: Error out gracefully when we can't create any of the usual conversion elements for some re...
33677 Original commit message from CVS:
33678 * gst/playback/gstplaybin.c: (gen_video_element),
33679 (gen_text_element), (gen_audio_element), (gen_vis_element):
33680 Error out gracefully when we can't create any of the usual
33681 conversion elements for some reason. Also, don't try to
33682 create an audioscale (sic) element that's not used anyway.
33684 2006-03-28 10:21:52 +0000 Tim-Philipp Müller <tim@centricular.net>
33686 gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
33687 Original commit message from CVS:
33688 * gst/playback/gstplaybasebin.c: (setup_source):
33689 Don't post RESOURCE_NOT_FOUND error when we can't find a source
33690 element for a particular protocol, that's confusing for users.
33691 Instead, post a RESOURCE_FAILED error, so that our own error
33692 message is actually shown in totem etc. (#336303).
33694 2006-03-27 16:36:46 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
33696 ext/gnomevfs/gstgnomevfssrc.c: Fix some minor memory leaks (#336194).
33697 Original commit message from CVS:
33698 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
33699 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
33700 (gst_gnome_vfs_src_get_icy_metadata):
33701 Fix some minor memory leaks (#336194).
33703 2006-03-27 16:15:00 +0000 Tim-Philipp Müller <tim@centricular.net>
33705 ext/gnomevfs/: Make gnomevfssink accept filenames as well as URIs for the "location" property, just like gnomevfssrc ...
33706 Original commit message from CVS:
33707 * ext/gnomevfs/gstgnomevfs.c:
33708 (gst_gnome_vfs_location_to_uri_string):
33709 * ext/gnomevfs/gstgnomevfs.h:
33710 * ext/gnomevfs/gstgnomevfssink.c:
33711 (gst_gnome_vfs_sink_set_property):
33712 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
33713 Make gnomevfssink accept filenames as well as URIs for the
33714 "location" property, just like gnomevfssrc does (and
33715 filesrc/filesink do) (#336190).
33717 2006-03-24 20:35:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33719 tests/check/generic/clock-selection.c: set to NULL before unreffing, fixes a valgrind leak.
33720 Original commit message from CVS:
33721 * tests/check/generic/clock-selection.c: (GST_START_TEST):
33722 set to NULL before unreffing, fixes a valgrind leak.
33723 Why was this not triggering the error that an object needs to
33724 be NULL before unreffing ?
33725 * win32/common/config.h:
33728 2006-03-24 17:57:39 +0000 Tim-Philipp Müller <tim@centricular.net>
33730 gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
33731 Original commit message from CVS:
33732 * gst/subparse/gstsubparse.c: (convert_encoding),
33733 (gst_sub_parse_change_state):
33734 * gst/subparse/gstsubparse.h:
33735 Text subtitle files may or may not be UTF-8. If it's not, we
33736 don't really want to see '?' characters in place of non-ASCII
33737 characters like accented characters. So let's assume the input
33738 is UTF-8 until we come across text that is clearly not. If it's
33739 not UTF-8, we don't really know what it is, so try the following:
33740 (a) see whether the GST_SUBTITLE_ENCODING environment variable
33741 is set; if not, check (b) if the current locale encoding is
33742 non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
33743 the current locale encoding is UTF-8 and the environment variable
33744 was not set to any particular encoding. Not perfect, but better
33745 than nothing (and better than before, I think) (fixes #172848).
33747 2006-03-24 17:39:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33749 * docs/plugins/tmpl/.gitignore:
33750 * tests/check/libs/.gitignore:
33751 * tests/check/pipelines/.gitignore:
33752 * tests/examples/volume/.gitignore:
33754 Original commit message from CVS:
33757 2006-03-24 17:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33759 configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
33760 Original commit message from CVS:
33761 2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
33763 update core requirement to 0.10.4.1 because of async_playback
33764 vmethod on GstBaseSink
33766 2006-03-24 17:11:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33768 use DEBUG_FUNCPTR for collectpads
33769 Original commit message from CVS:
33770 * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
33771 * gst/adder/gstadder.c: (gst_adder_init):
33772 use DEBUG_FUNCPTR for collectpads
33774 2006-03-24 14:11:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33777 don't go through check-torture if no check installed
33778 Original commit message from CVS:
33779 don't go through check-torture if no check installed
33781 2006-03-24 10:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33783 Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
33784 Original commit message from CVS:
33785 * docs/plugins/Makefile.am:
33786 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
33787 * docs/plugins/gst-plugins-base-plugins-sections.txt:
33788 * ext/cdparanoia/gstcdparanoiasrc.c:
33789 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
33790 (gst_gnome_vfs_sink_class_init):
33791 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
33792 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
33793 * ext/ogg/gstoggmux.c:
33794 * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
33795 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
33796 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
33797 * ext/pango/gsttextoverlay.c:
33798 * ext/pango/gsttextrender.c:
33799 * ext/theora/theoradec.c:
33800 * ext/theora/theoraenc.c:
33801 * ext/vorbis/vorbisdec.c:
33802 * ext/vorbis/vorbisenc.c:
33803 * gst-libs/gst/audio/gstaudiofilter.c:
33804 (gst_audio_filter_base_init):
33805 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
33806 (gst_audio_filter_template_base_init):
33807 * gst/adder/gstadder.c: (gst_adder_get_type):
33808 * gst/adder/gstadder.h:
33809 * gst/audioconvert/gstaudioconvert.c:
33810 * gst/audiotestsrc/gstaudiotestsrc.c:
33811 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
33812 (gst_audio_test_src_create):
33813 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
33814 * gst/playback/gstdecodebin.c:
33815 * gst/playback/gstplaybin.c:
33816 * gst/playback/gststreamselector.c:
33817 (gst_stream_selector_base_init):
33818 * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
33819 * gst/volume/gstvolume.c:
33820 * sys/v4l/gstv4lmjpegsink.c:
33821 * sys/v4l/gstv4lmjpegsrc.c:
33822 * tests/check/libs/cddabasesrc.c:
33823 * tests/old/examples/gob/gst-identity2.gob:
33824 Add docs for adder, use GST_ELEMENT_DETAILS macro,
33825 define GstElementDetails at the top
33827 2006-03-23 21:48:18 +0000 Sébastien Moutte <sebastien@moutte.net>
33829 win32/common/libgstinterfaces.def: Add a lot of export functions for gst-python
33830 Original commit message from CVS:
33831 * win32/common/libgstinterfaces.def:
33832 Add a lot of export functions for gst-python
33833 * win32/common/libgstinterfaces.dsp:
33834 Add a missing include folder in the project configuration
33836 2006-03-23 16:58:03 +0000 Wim Taymans <wim.taymans@gmail.com>
33838 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
33839 Original commit message from CVS:
33840 * gst-libs/gst/audio/gstbaseaudiosrc.c:
33841 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
33842 (gst_base_audio_src_change_state):
33843 Fix audio sources, forgot to make the ringbuffer
33846 2006-03-23 16:29:58 +0000 Wim Taymans <wim.taymans@gmail.com>
33848 gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
33849 Original commit message from CVS:
33850 * gst-libs/gst/audio/gstbaseaudiosrc.c:
33851 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
33852 (gst_base_audio_src_change_state):
33853 unparent instead of unref the ringbuffer.
33855 2006-03-23 16:24:23 +0000 Wim Taymans <wim.taymans@gmail.com>
33857 gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
33858 Original commit message from CVS:
33859 * gst-libs/gst/audio/gstbaseaudiosink.c:
33860 (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
33861 (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
33862 Implement new async_play vmethod to start slaving and allow
33863 playback start in case of async PLAY state changes.
33864 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
33865 Enable QoS with new method in base class.
33867 2006-03-23 11:18:19 +0000 Julien MOUTTE <julien@moutte.net>
33869 gst/videotestsrc/gstvideotestsrc.c: Partially handle 0 framerate, only EOS after the first frame is missing.
33870 Original commit message from CVS:
33871 Patch by: Julien MOUTTE <julien at moutte dot net>
33872 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
33873 (gst_video_test_src_do_seek), (gst_video_test_src_create):
33874 Partially handle 0 framerate, only EOS after the first frame
33877 2006-03-23 09:38:59 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
33879 gst/: Patch for support of YVU9 AVI files (#334822)
33880 Original commit message from CVS:
33881 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
33882 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
33883 (gst_riff_create_video_template_caps):
33884 * gst/ffmpegcolorspace/avcodec.h:
33885 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
33886 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
33887 (gst_ffmpegcsp_avpicture_fill):
33888 * gst/ffmpegcolorspace/imgconvert.c:
33889 Patch for support of YVU9 AVI files (#334822)
33891 2006-03-22 15:29:25 +0000 Edward Hervey <bilboed@bilboed.com>
33893 docs/design/design-decodebin.txt: Added design document for new decodebin text/x-pango-markup is also a default targe...
33894 Original commit message from CVS:
33895 * docs/design/design-decodebin.txt:
33896 Added design document for new decodebin
33897 (Target Caps): text/x-pango-markup is also a default target caps.
33899 2006-03-22 15:11:47 +0000 Edward Hervey <bilboed@bilboed.com>
33901 docs/design/design-decodebin.txt: Added design document for new decodebin
33902 Original commit message from CVS:
33903 * docs/design/design-decodebin.txt:
33904 Added design document for new decodebin
33906 2006-03-22 12:33:09 +0000 Wim Taymans <wim.taymans@gmail.com>
33908 gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
33909 Original commit message from CVS:
33910 * gst-libs/gst/audio/gstbaseaudiosink.c:
33911 (gst_base_audio_sink_dispose):
33912 Since we _parent the ringbuffer, we also need to
33913 _unparent instead of a plain _unref.
33915 2006-03-22 12:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
33917 tests/examples/seek/seek.c: Add scrub checkbox.
33918 Original commit message from CVS:
33919 * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
33920 (stop_seek), (scrub_toggle_cb), (main):
33921 Add scrub checkbox.
33923 2006-03-21 17:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
33925 ext/ogg/gstoggparse.c: Fix very inefficient usage of linked lists (#335365).
33926 Original commit message from CVS:
33927 * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
33928 (gst_ogg_parse_chain):
33929 Fix very inefficient usage of linked lists (#335365).
33931 2006-03-21 14:26:01 +0000 Edward Hervey <bilboed@bilboed.com>
33933 gcc 4.1 unreferenced pointer fixes.
33934 Original commit message from CVS:
33935 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
33936 * gst/playback/gstplaybin.c: (handoff):
33937 * gst/playback/gststreamselector.c:
33938 (gst_stream_selector_set_property):
33939 gcc 4.1 unreferenced pointer fixes.
33940 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
33941 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
33942 gst_buffer_ref() now takes a GstBuffer*.
33944 2006-03-20 18:09:41 +0000 Julien Moutte <julien@moutte.net>
33946 sys/xvimage/xvimagesink.c: Fix a memleak reported by Jan Schmidt.
33947 Original commit message from CVS:
33948 2006-03-20 Julien MOUTTE <julien@moutte.net>
33949 * sys/xvimage/xvimagesink.c:
33950 (gst_xvimagesink_get_format_from_caps): Fix a memleak reported
33953 2006-03-19 11:37:46 +0000 Tim-Philipp Müller <tim@centricular.net>
33955 gst/typefind/gsttypefindfunctions.c: Can't do tag preferences via probability, as tags would then lose against types ...
33956 Original commit message from CVS:
33957 * gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
33958 (id3v1_type_find), (apetag_type_find), (plugin_init):
33959 Can't do tag preferences via probability, as tags would then
33960 lose against types that are recognised with MAXIMUM probability
33961 (like .wav); so let all tag typefinders return MAXIMUM themselves
33962 and order them via the rank. Split ID3v1 and ID3v2 typefinders so
33963 that we can prefer APE to ID3v1 (fixes #335028).
33965 2006-03-17 17:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
33967 gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
33968 Original commit message from CVS:
33969 * gst-libs/gst/audio/gstbaseaudiosink.c:
33970 (gst_base_audio_sink_change_state):
33971 * gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
33972 (gst_ring_buffer_may_start):
33973 * gst-libs/gst/audio/gstringbuffer.h:
33974 Only start playback if we are playing.
33975 should fix #330748.
33977 2006-03-17 13:11:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33979 Revert accidental commits to these files.
33980 Original commit message from CVS:
33981 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
33982 * win32/common/config.h:
33983 Revert accidental commits to these files.
33985 2006-03-16 20:01:03 +0000 Michal Benes <michal.benes@xeris.cz>
33987 tests/Makefile.am: Don't try to build tests in tests/icles if we don't have X (#323852)
33988 Original commit message from CVS:
33989 Patch by: Michal Benes <michal dot benes at xeris dot cz>
33990 * tests/Makefile.am:
33991 Don't try to build tests in tests/icles if we
33992 don't have X (#323852)
33994 2006-03-16 13:08:01 +0000 Tim-Philipp Müller <tim@centricular.net>
33996 gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
33997 Original commit message from CVS:
33998 * gst-libs/gst/tag/gstid3tag.c:
33999 Add TXXX frame identifiers for replaygain stuff as used
34000 by some taggers (see #323721).
34002 2006-03-16 10:22:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
34004 gst/playback/gststreamselector.c: Preserve the existing buggy streamselector behaviour by performing a fallback buffe...
34005 Original commit message from CVS:
34006 * gst/playback/gststreamselector.c:
34007 (gst_stream_selector_set_property),
34008 (gst_stream_selector_bufferalloc):
34009 Preserve the existing buggy streamselector behaviour by performing
34010 a fallback buffer allocation when downstream isn't linked yet.
34011 This should really be fixed in playbin by blocking pads until it's
34013 Also, use gst_pad_alloc_buffer instead of
34014 gst_pad_alloc_buffer_and_set.
34016 2006-03-15 22:40:08 +0000 Tim-Philipp Müller <tim@centricular.net>
34018 gst-libs/gst/tag/gstid3tag.c: Don't crash on unknown ID3v2 TXXX frames.
34019 Original commit message from CVS:
34020 * gst-libs/gst/tag/gstid3tag.c:
34021 Don't crash on unknown ID3v2 TXXX frames.
34023 2006-03-15 17:59:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
34025 ext/alsa/gstalsasink.c: Chain up to the parent finalize method.
34026 Original commit message from CVS:
34027 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
34028 Chain up to the parent finalize method.
34029 Add 32-bit sample size to the template caps.
34030 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
34031 (gst_riff_create_video_template_caps):
34032 Add the fourcc that the VMWare codec uses.
34033 * gst/playback/gststreamselector.c:
34034 (gst_stream_selector_set_property),
34035 (gst_stream_selector_bufferalloc),
34036 (gst_stream_selector_request_new_pad):
34037 For the active pad, forward buffer-alloc requests, otherwise
34038 return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
34039 having to memcpy every frame when used by playbin.
34040 * gst/tcp/gstmultifdsink.c:
34041 (gst_multi_fd_sink_handle_client_write):
34042 Get negotiated caps from the sink pad, rather than the sink
34045 2006-03-15 17:11:34 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
34047 ext/gnomevfs/gstgnomevfssrc.c: Don't forget to set src->callbacks_pushed to FALSE again when popping them, otherwise ...
34048 Original commit message from CVS:
34049 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
34050 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_pop_callbacks):
34051 Don't forget to set src->callbacks_pushed to FALSE again when
34052 popping them, otherwise re-activation in a different mode won't
34055 2006-03-15 11:30:29 +0000 Sebastien Moutte <sebastien@moutte.net>
34057 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
34058 Original commit message from CVS:
34059 Patch by: Sebastien Moutte <sebastien moutte net>
34060 * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
34061 (gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
34062 (gst_ffmpeg_smpfmt_to_caps):
34063 Replace __VA_ARGS__ caps creation macros with varargs functions.
34064 Makes things compile on MSVC (#320765), looks nicer, and we can
34065 tell the compiler to check for the NULL terminator.
34067 2006-03-14 15:13:04 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
34069 gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
34070 Original commit message from CVS:
34071 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
34072 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
34073 Make sure the buffer we copy into is really always big
34074 enough, this time for real (#333488).
34076 2006-03-14 13:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
34078 gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
34079 Original commit message from CVS:
34080 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
34081 Add support for 24bpp DIB (#305279).
34083 2006-03-14 11:11:59 +0000 Wim Taymans <wim.taymans@gmail.com>
34085 gst/: Re-enable QoS after the release.
34086 Original commit message from CVS:
34087 * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
34088 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
34089 * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
34090 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
34091 (gst_video_scale_init), (gst_video_scale_src_event):
34092 Re-enable QoS after the release.
34093 Rework videoscale to use the base class src_event handler.
34095 2006-03-14 09:51:01 +0000 Tim-Philipp Müller <tim@centricular.net>
34097 configure.ac: back to CVS.
34098 Original commit message from CVS:
34102 === release 0.10.5 ===
34104 2006-03-13 19:50:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34110 * docs/plugins/inspect/plugin-adder.xml:
34111 * docs/plugins/inspect/plugin-alsa.xml:
34112 * docs/plugins/inspect/plugin-audioconvert.xml:
34113 * docs/plugins/inspect/plugin-audiorate.xml:
34114 * docs/plugins/inspect/plugin-audioresample.xml:
34115 * docs/plugins/inspect/plugin-audiotestsrc.xml:
34116 * docs/plugins/inspect/plugin-cdparanoia.xml:
34117 * docs/plugins/inspect/plugin-decodebin.xml:
34118 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
34119 * docs/plugins/inspect/plugin-gnomevfs.xml:
34120 * docs/plugins/inspect/plugin-libvisual.xml:
34121 * docs/plugins/inspect/plugin-ogg.xml:
34122 * docs/plugins/inspect/plugin-pango.xml:
34123 * docs/plugins/inspect/plugin-playbin.xml:
34124 * docs/plugins/inspect/plugin-subparse.xml:
34125 * docs/plugins/inspect/plugin-tcp.xml:
34126 * docs/plugins/inspect/plugin-theora.xml:
34127 * docs/plugins/inspect/plugin-typefindfunctions.xml:
34128 * docs/plugins/inspect/plugin-video4linux.xml:
34129 * docs/plugins/inspect/plugin-videorate.xml:
34130 * docs/plugins/inspect/plugin-videoscale.xml:
34131 * docs/plugins/inspect/plugin-videotestsrc.xml:
34132 * docs/plugins/inspect/plugin-volume.xml:
34133 * docs/plugins/inspect/plugin-vorbis.xml:
34134 * docs/plugins/inspect/plugin-ximagesink.xml:
34135 * docs/plugins/inspect/plugin-xvimagesink.xml:
34136 * win32/common/config.h:
34138 Original commit message from CVS:
34141 2006-03-13 17:28:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34158 Original commit message from CVS:
34161 2006-03-13 11:17:19 +0000 Tim-Philipp Müller <tim@centricular.net>
34163 docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
34164 Original commit message from CVS:
34165 * docs/plugins/Makefile.am:
34166 Part of previous cdparanoiasrc docs fixes, forgot to commit.
34168 2006-03-12 14:56:31 +0000 Tim-Philipp Müller <tim@centricular.net>
34170 docs/plugins/: Add cdparanoiasrc to docs.
34171 Original commit message from CVS:
34172 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
34173 * docs/plugins/gst-plugins-base-plugins-sections.txt:
34174 * docs/plugins/gst-plugins-base-plugins.hierarchy:
34175 Add cdparanoiasrc to docs.
34176 * gst-libs/gst/cdda/gstcddabasesrc.c:
34177 More GstCddaBaseSrc docs.
34179 2006-03-12 13:47:22 +0000 Tim-Philipp Müller <tim@centricular.net>
34181 Add new API to libgsttag: gst_tag_from_id3_user_tag().
34182 Original commit message from CVS:
34183 * docs/libs/gst-plugins-base-libs-sections.txt:
34184 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
34185 * gst-libs/gst/tag/tag.h:
34186 Add new API to libgsttag: gst_tag_from_id3_user_tag().
34188 2006-03-11 19:47:16 +0000 Tim-Philipp Müller <tim@centricular.net>
34190 gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
34191 Original commit message from CVS:
34192 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
34193 NULL-terminate array of mpeg4 video file extensions.
34194 Fixes crash on PPC (#334226).
34196 2006-03-11 16:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
34198 ext/gnomevfs/gstgnomevfssrc.c: gnome_vfs_uri_is_local() alone is not a good indicator whether we can operate in pull-...
34199 Original commit message from CVS:
34200 * ext/gnomevfs/gstgnomevfssrc.c:
34201 (gst_gnome_vfs_src_check_get_range):
34202 gnome_vfs_uri_is_local() alone is not a good indicator
34203 whether we can operate in pull-mode with a specific URI,
34204 as it returns FALSE for file:// URIs that point to an
34205 NFS-mounted path. Be more conservative here: whitelist
34206 local files, blacklist http URIs and use the old
34207 mechanism for anything else (fixes #334216).
34209 2006-03-10 19:15:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34211 configure.ac: back to trunk
34212 Original commit message from CVS:
34216 === release 0.10.4 ===
34218 2006-03-10 19:05:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34224 * docs/plugins/gst-plugins-base-plugins.args:
34225 * docs/plugins/inspect/plugin-adder.xml:
34226 * docs/plugins/inspect/plugin-alsa.xml:
34227 * docs/plugins/inspect/plugin-audioconvert.xml:
34228 * docs/plugins/inspect/plugin-audiorate.xml:
34229 * docs/plugins/inspect/plugin-audioresample.xml:
34230 * docs/plugins/inspect/plugin-audiotestsrc.xml:
34231 * docs/plugins/inspect/plugin-cdparanoia.xml:
34232 * docs/plugins/inspect/plugin-decodebin.xml:
34233 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
34234 * docs/plugins/inspect/plugin-gnomevfs.xml:
34235 * docs/plugins/inspect/plugin-libvisual.xml:
34236 * docs/plugins/inspect/plugin-ogg.xml:
34237 * docs/plugins/inspect/plugin-pango.xml:
34238 * docs/plugins/inspect/plugin-playbin.xml:
34239 * docs/plugins/inspect/plugin-subparse.xml:
34240 * docs/plugins/inspect/plugin-tcp.xml:
34241 * docs/plugins/inspect/plugin-theora.xml:
34242 * docs/plugins/inspect/plugin-typefindfunctions.xml:
34243 * docs/plugins/inspect/plugin-video4linux.xml:
34244 * docs/plugins/inspect/plugin-videorate.xml:
34245 * docs/plugins/inspect/plugin-videoscale.xml:
34246 * docs/plugins/inspect/plugin-videotestsrc.xml:
34247 * docs/plugins/inspect/plugin-volume.xml:
34248 * docs/plugins/inspect/plugin-vorbis.xml:
34249 * docs/plugins/inspect/plugin-ximagesink.xml:
34250 * docs/plugins/inspect/plugin-xvimagesink.xml:
34252 * win32/common/config.h:
34254 Original commit message from CVS:
34257 2006-03-10 12:37:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
34259 gst-libs/gst/video/gstvideosink.c: Disable max-lateness by setting it to -1 for now, so that we can bed QoS stuff in ...
34260 Original commit message from CVS:
34261 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
34262 Disable max-lateness by setting it to -1 for now, so that
34263 we can bed QoS stuff in thoroughly between now and the next
34266 2006-03-10 11:09:23 +0000 Fabrizio <fabrizio.ge@tiscali.it>
34268 gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
34269 Original commit message from CVS:
34270 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
34271 Make sure we don't read beyond the palette buffer in case of
34272 broken or manipulated files (#333488, patch by: Fabrizio
34275 2006-03-10 10:44:02 +0000 Edward Hervey <bilboed@bilboed.com>
34277 gst/typefind/gsttypefindfunctions.c: Fix for variable not initialized.
34278 Original commit message from CVS:
34279 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
34280 Fix for variable not initialized.
34282 2006-03-09 19:02:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34285 * docs/libs/tmpl/gstringbuffer.sgml:
34300 * win32/common/config.h:
34302 Original commit message from CVS:
34305 2006-03-09 17:58:00 +0000 Wim Taymans <wim.taymans@gmail.com>
34307 ext/libvisual/visual.c: Small cleanups.
34308 Original commit message from CVS:
34309 * ext/libvisual/visual.c: (gst_visual_get_type),
34310 (gst_visual_src_setcaps), (gst_vis_src_negotiate),
34311 (gst_visual_chain):
34313 * ext/theora/gsttheoradec.h:
34314 * ext/theora/theoradec.c: (gst_theora_dec_init),
34315 (gst_theora_dec_reset), (_theora_granule_time),
34316 (theora_dec_src_convert), (theora_dec_sink_convert),
34317 (theora_dec_src_query), (theora_dec_src_event),
34318 (theora_dec_sink_event), (theora_handle_comment_packet),
34319 (theora_handle_header_packet), (theora_dec_push),
34320 (theora_handle_data_packet), (theora_dec_chain),
34321 (theora_dec_change_state):
34324 2006-03-09 17:50:59 +0000 Wim Taymans <wim.taymans@gmail.com>
34326 ext/gnomevfs/gstgnomevfssrc.c: Some cleanups.
34327 Original commit message from CVS:
34328 * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
34329 (audiocast_register_listener), (gst_gnome_vfs_src_start):
34332 2006-03-09 17:45:39 +0000 Wim Taymans <wim.taymans@gmail.com>
34334 ext/ogg/gstoggdemux.c: Don't try to activate NULL chains.
34335 Original commit message from CVS:
34336 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
34337 Don't try to activate NULL chains.
34339 2006-03-09 16:30:41 +0000 Tim-Philipp Müller <tim@centricular.net>
34341 gst/typefind/gsttypefindfunctions.c: Fix invalid memory access to region before peek'd data (#332964).
34342 Original commit message from CVS:
34343 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
34344 Fix invalid memory access to region before peek'd data (#332964).
34346 2006-03-09 15:05:03 +0000 Christophe Fergeau <teuf@gnome.org>
34349 Original commit message from CVS:
34350 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init):
34351 * ext/pango/gsttextrender.c: (gst_text_render_init):
34352 * gst/adder/gstadder.c: (gst_adder_init):
34353 Don't leak padtemplates, patch by Christophe Fergeau,
34356 2006-03-09 12:56:35 +0000 Tim-Philipp Müller <tim@centricular.net>
34358 gst/subparse/gstsubparse.c: Fix invalid memory access: make sure string passed to regexec() is NUL-termianted.
34359 Original commit message from CVS:
34360 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
34361 Fix invalid memory access: make sure string passed to
34362 regexec() is NUL-termianted.
34364 2006-03-09 12:37:59 +0000 Tim-Philipp Müller <tim@centricular.net>
34366 gst/typefind/gsttypefindfunctions.c: Refactor mpeg/audio typefinding to make it more maintainable and easier to fine-...
34367 Original commit message from CVS:
34368 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
34370 Refactor mpeg/audio typefinding to make it more maintainable
34371 and easier to fine-tune. Make probing into middle of the file
34372 work properly (fixes #333900, also see #152688).
34374 2006-03-09 11:10:03 +0000 Tim-Philipp Müller <tim@centricular.net>
34376 gst/typefind/gsttypefindfunctions.c: Remove part from previous commit that was bogus: g_utf8_validate() does in fact ...
34377 Original commit message from CVS:
34378 * gst/typefind/gsttypefindfunctions.c:
34379 (utf8_type_find_have_valid_utf8_at_offset):
34380 Remove part from previous commit that was bogus:
34381 g_utf8_validate() does in fact not accept embedded
34382 zeroes, so we don't need to check for those (thanks
34383 to Mike for the hint).
34385 2006-03-08 17:11:29 +0000 Tim-Philipp Müller <tim@centricular.net>
34387 gst/typefind/gsttypefindfunctions.c: Make plain/text typefinder more conservative: firstly, check for embedded zeroes...
34388 Original commit message from CVS:
34389 * gst/typefind/gsttypefindfunctions.c:
34390 (utf8_type_find_count_embedded_zeroes),
34391 (utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
34392 Make plain/text typefinder more conservative: firstly, check
34393 for embedded zeroes, which are perfectly valid UTF-8 characters,
34394 but also a fairly good sign that something is not a plain text
34395 file; secondly, probe into the middle of the file if possible.
34396 If we can't probe into the middle, limit the probability value
34397 to be returned to TYPE_FIND_POSSIBLE (see #333900).
34399 2006-03-08 11:34:45 +0000 Michael Smith <msmith@xiph.org>
34401 gst/typefind/gsttypefindfunctions.c: Make typefind function name for mpeg4 video unique.
34402 Original commit message from CVS:
34403 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
34404 Make typefind function name for mpeg4 video unique.
34406 2006-03-08 09:53:31 +0000 Wim Taymans <wim.taymans@gmail.com>
34408 ext/libvisual/visual.c: Cleanups, post nice errors.
34409 Original commit message from CVS:
34410 * ext/libvisual/visual.c: (gst_visual_init),
34411 (gst_visual_clear_actors), (gst_visual_dispose),
34412 (gst_visual_reset), (gst_visual_src_setcaps),
34413 (gst_visual_sink_setcaps), (gst_vis_src_negotiate),
34414 (gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
34415 (gst_visual_chain), (gst_visual_change_state):
34416 Cleanups, post nice errors.
34417 Handle sink and src events.
34418 Implement simple QoS.
34419 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
34420 Use new basesink methods to configure max-lateness.
34422 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
34423 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
34424 Debug statement cleanups.
34425 * gst/volume/gstvolume.c: (gst_volume_class_init):
34428 2006-03-08 09:50:23 +0000 Tim-Philipp Müller <tim@centricular.net>
34430 ext/pango/gsttextoverlay.c: Revert API/ABI break from March 1. Keep 'halign' and 'valign' as string type properties, ...
34431 Original commit message from CVS:
34432 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
34433 (gst_text_overlay_init), (gst_text_overlay_set_property),
34434 (gst_text_overlay_get_property):
34435 Revert API/ABI break from March 1. Keep 'halign' and 'valign'
34436 as string type properties, but mark them deprecated. Add
34437 'halignment' and 'valignment' properties that use enums
34438 instead of strings.
34440 2006-03-08 09:37:12 +0000 Fabrizio <fabrizio.ge@tiscali.it>
34442 gst-libs/gst/riff/riff-media.c: Allow palettes with less than 256 colours in AVI files
34443 Original commit message from CVS:
34444 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
34445 Allow palettes with less than 256 colours in AVI files
34446 (#333488, patch by: Fabrizio Gennari).
34448 2006-03-07 21:56:09 +0000 Julien Moutte <julien@moutte.net>
34450 ext/pango/gsttextoverlay.c: Fix wrong EOS handling on text pad. We were releasing the queued text buffer when we shou...
34451 Original commit message from CVS:
34452 2006-03-07 Julien MOUTTE <julien@moutte.net>
34453 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
34454 (gst_text_overlay_video_event): Fix wrong EOS handling on text
34455 pad. We were releasing the queued text buffer when we should keep
34456 it until video pad gets EOS or discard the text buffer because it's
34457 too old. That was eating the last subtitle buffer. Add some more
34460 2006-03-07 17:28:36 +0000 Tim-Philipp Müller <tim@centricular.net>
34462 ext/pango/gsttextoverlay.c: Fix invalid memory access (we can't access a buffer after it's been pushed downstream wit...
34463 Original commit message from CVS:
34464 * ext/pango/gsttextoverlay.c: (gst_text_overlay_render_text),
34465 (gst_text_overlay_video_chain):
34466 Fix invalid memory access (we can't access a buffer after it's been
34467 pushed downstream without taking a reference); fix memory leak (if
34468 there's no text to render, bail out before allocating stuff).
34470 2006-03-07 15:08:15 +0000 Tim-Philipp Müller <tim@centricular.net>
34472 ext/pango/gsttextoverlay.*: If input is plain text, escape it before passing it to pango_layout_set_markup().
34473 Original commit message from CVS:
34474 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
34475 (gst_text_overlay_setcaps_txt), (gst_text_overlay_video_chain):
34476 * ext/pango/gsttextoverlay.h:
34477 If input is plain text, escape it before passing it to
34478 pango_layout_set_markup().
34480 2006-03-07 13:01:21 +0000 Tim-Philipp Müller <tim@centricular.net>
34482 gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
34483 Original commit message from CVS:
34484 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
34485 Don't ignore flow return from gst_pad_push().
34487 2006-03-07 12:49:03 +0000 Christophe Fergeau <teuf@gnome.org>
34489 Don't leak references returned by gst_pad_get_parent()
34490 Original commit message from CVS:
34491 * ext/libvisual/visual.c: (gst_visual_getcaps),
34492 (gst_visual_src_setcaps), (gst_visual_sink_setcaps):
34493 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
34494 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
34495 (gst_vorbisenc_convert_sink):
34496 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
34497 (gst_audio_duration_from_pad_buffer):
34498 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
34499 (gst_audio_filter_chain):
34500 * gst-libs/gst/rtp/gstbasertpdepayload.c:
34501 (gst_base_rtp_depayload_setcaps):
34502 * gst-libs/gst/video/video.c: (gst_video_frame_rate),
34503 (gst_video_get_size):
34504 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
34505 Don't leak references returned by gst_pad_get_parent()
34506 (#333663, based on patch by: Christophe Fergeau).
34508 2006-03-06 20:52:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
34510 ext/gnomevfs/gstgnomevfssink.c: change location param details
34511 Original commit message from CVS:
34512 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
34513 change location param details
34514 * gst/volume/gstvolume.c: (plugin_init):
34515 correct plugin description
34517 2006-03-06 20:07:55 +0000 Tim-Philipp Müller <tim@centricular.net>
34519 ext/gnomevfs/gstgnomevfssrc.c: Override GstBaseSrc::check_get_range() in order to avoid opening the resource just to ...
34520 Original commit message from CVS:
34521 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
34522 (gst_gnome_vfs_src_check_get_range):
34523 Override GstBaseSrc::check_get_range() in order to avoid opening
34524 the resource just to check whether we can operate in pull-mode or
34525 not - we can predict that pretty well from the URI alone. Should
34526 fix problems with last.fm (#331690). (Requires latest core CVS).
34528 2006-03-06 16:18:51 +0000 Wim Taymans <wim.taymans@gmail.com>
34530 gst-libs/gst/video/gstvideosink.c: Throw away frames that are later than 20 ms.
34531 Original commit message from CVS:
34532 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
34533 (gst_video_sink_class_init):
34534 Throw away frames that are later than 20 ms.
34536 2006-03-06 14:14:47 +0000 Fabrizio <fabrizio.ge@tiscali.it>
34538 gst-libs/gst/riff/riff-media.c:
34539 Original commit message from CVS:
34540 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
34541 Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).
34543 2006-03-05 23:39:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34545 ext/ogg/gstoggmux.c: put Theora BOS pages before others. This hardcodes the Ogg/Theora I profile, but hey.
34546 Original commit message from CVS:
34547 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
34548 (gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
34549 put Theora BOS pages before others. This hardcodes
34550 the Ogg/Theora I profile, but hey.
34552 2006-03-05 23:06:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34554 * ext/ogg/gstoggmux.c:
34555 changed more than 5 lines
34556 Original commit message from CVS:
34557 changed more than 5 lines
34559 2006-03-05 22:57:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34561 ogg muxing of vorbis and theora now has pages ordered correctly again, even with delays.
34562 Original commit message from CVS:
34563 ogg muxing of vorbis and theora now has pages ordered correctly again,
34566 updated with some examples
34567 * ext/theora/theoraenc.c: (granulepos_to_timestamp),
34568 (granulepos_add), (theora_buffer_from_packet):
34569 * ext/vorbis/vorbisenc.c: (granulepos_to_timestamp_offset),
34570 (granulepos_to_timestamp), (gst_vorbisenc_buffer_from_packet),
34571 (gst_vorbisenc_chain):
34572 implement strategy from ext/ogg/README
34573 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
34574 (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
34575 (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_compare_pads),
34576 (gst_ogg_mux_queue_pads), (gst_ogg_mux_collected):
34577 Fix muxer so that oggz-validate is happy with all streams;
34578 except for no eos mark, and the BOS page ordering
34579 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
34580 (check_buffer_granulepos):
34581 * tests/check/pipelines/vorbisenc.c: (check_buffer_granulepos):
34582 update tests to check for OFFSET being set as requested
34583 fixed type of granulepos, it's not a ClockTime
34585 2006-03-05 21:34:23 +0000 Julien Moutte <julien@moutte.net>
34587 sys/xvimage/xvimagesink.c: Check that the xvimage we are creating has a correct size before returning it. (#3...
34588 Original commit message from CVS:
34589 2006-03-05 Julien MOUTTE <julien@moutte.net>
34590 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
34591 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
34592 Check that the xvimage we are creating has a correct size before returning it. (#314897)
34594 2006-03-05 13:44:05 +0000 Tim-Philipp Müller <tim@centricular.net>
34596 gst/typefind/gsttypefindfunctions.c: Give id3 and ape tag typefinders a rank slightly higher than PRIMARY to ensure t...
34597 Original commit message from CVS:
34598 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
34599 Give id3 and ape tag typefinders a rank slightly higher
34600 than PRIMARY to ensure they're always run before any of
34601 the other typefinders (in particular wav and mp3) (#324186).
34603 2006-03-05 13:08:37 +0000 Tim-Philipp Müller <tim@centricular.net>
34605 gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
34606 Original commit message from CVS:
34607 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
34608 Add support for '3IVD' fourcc (#333403).
34610 2006-03-04 14:35:10 +0000 Tim-Philipp Müller <tim@centricular.net>
34612 configure.ac: Bump requirements to GStreamer CVS for the new error enum.
34613 Original commit message from CVS:
34615 Bump requirements to GStreamer CVS for the new error enum.
34616 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
34617 Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
34618 space left on the device (fixes #333352).
34620 2006-03-03 23:53:50 +0000 Sébastien Moutte <sebastien@moutte.net>
34622 win32/vs6: add a project file for libgstvolume update the workspace
34623 Original commit message from CVS:
34625 add a project file for libgstvolume
34626 update the workspace
34628 2006-03-03 15:26:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34631 * ext/ogg/gstoggmux.c:
34633 Original commit message from CVS:
34636 2006-03-03 15:22:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34638 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
34639 Original commit message from CVS:
34640 2006-03-03 Thomas Vander Stichele <thomas at apestaart dot org>
34641 * ext/theora/theoraenc.c: (theora_set_header_on_caps):
34642 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
34644 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
34645 Set IN_CAPS on header buffers
34647 2006-03-02 18:23:55 +0000 Wim Taymans <wim.taymans@gmail.com>
34649 docs/plugins/: Add audioresample to docs.
34650 Original commit message from CVS:
34651 * docs/plugins/Makefile.am:
34652 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
34653 * docs/plugins/gst-plugins-base-plugins-sections.txt:
34654 Add audioresample to docs.
34655 * gst/audioconvert/gstaudioconvert.c:
34657 * gst/audioresample/gstaudioresample.c:
34658 (gst_audioresample_base_init), (gst_audioresample_class_init),
34659 (gst_audioresample_init), (gst_audioresample_dispose),
34660 (audioresample_get_unit_size), (audioresample_transform_caps),
34661 (resample_set_state_from_caps), (audioresample_transform_size),
34662 (audioresample_set_caps), (audioresample_event),
34663 (audioresample_do_output), (audioresample_transform),
34664 (audioresample_pushthrough), (gst_audioresample_set_property),
34665 (gst_audioresample_get_property), (plugin_init):
34666 * gst/audioresample/gstaudioresample.h:
34668 Small code cleanups.
34670 2006-03-02 18:12:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34672 * gst/videorate/Makefile.am:
34674 Original commit message from CVS:
34677 2006-03-02 17:48:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34679 * ext/ogg/gstoggmux.c:
34680 debug using the actual GstPad, that allows us to see the serialno in the padname
34681 Original commit message from CVS:
34682 debug using the actual GstPad, that allows us to see the serialno in the padname
34684 2006-03-02 17:46:36 +0000 Wim Taymans <wim.taymans@gmail.com>
34686 docs/plugins/: Added videoscale to docs.
34687 Original commit message from CVS:
34688 * docs/plugins/Makefile.am:
34689 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
34690 * docs/plugins/gst-plugins-base-plugins-sections.txt:
34691 Added videoscale to docs.
34692 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
34693 (gst_video_rate_swap_prev), (gst_video_rate_event),
34694 (gst_video_rate_chain):
34696 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
34697 (gst_video_scale_init), (gst_video_scale_prepare_size),
34698 (gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
34699 (gst_video_scale_fixate_caps), (gst_video_scale_transform):
34700 * gst/videoscale/gstvideoscale.h:
34701 Added docs, examples.
34702 Some code cleanups.
34703 Post errors instead of g_warning.
34705 2006-03-02 17:30:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34707 * ext/ogg/gstoggmux.c:
34708 clean up debug messages
34709 Original commit message from CVS:
34710 clean up debug messages
34712 2006-03-02 17:15:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34714 * ext/ogg/gstoggmux.c:
34715 extra debugging from older version, makes it easier to compare
34716 Original commit message from CVS:
34717 extra debugging from older version, makes it easier to compare
34719 2006-03-02 17:04:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34721 * ext/ogg/gstoggmux.c:
34722 some space cleanup and debug fixes
34723 Original commit message from CVS:
34724 some space cleanup and debug fixes
34726 2006-03-02 16:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
34728 docs/: Added some more docs to libs and plugins.
34729 Original commit message from CVS:
34730 * docs/libs/gst-plugins-base-libs-docs.sgml:
34731 * docs/libs/gst-plugins-base-libs-sections.txt:
34732 * docs/libs/gst-plugins-base-libs.types:
34733 * docs/plugins/Makefile.am:
34734 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
34735 * docs/plugins/gst-plugins-base-plugins-sections.txt:
34736 Added some more docs to libs and plugins.
34737 * gst-libs/gst/audio/gstringbuffer.c:
34738 (gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
34739 * gst-libs/gst/audio/gstringbuffer.h:
34740 Document ringbuffer some more.
34741 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
34742 (gst_video_rate_setcaps), (gst_video_rate_reset),
34743 (gst_video_rate_init), (gst_video_rate_flush_prev),
34744 (gst_video_rate_swap_prev), (gst_video_rate_event),
34745 (gst_video_rate_chain), (gst_video_rate_change_state):
34746 * gst/videorate/gstvideorate.h:
34747 Fix videorate to use segments.
34748 Make it work with 0/1 framerates (closes #331903)
34749 Handle EOS correctly.
34752 2006-03-02 13:13:00 +0000 Tim-Philipp Müller <tim@centricular.net>
34754 ext/ogg/gstogmparse.c: In state change function, first chain up to parent class, then handle downwards state change s...
34755 Original commit message from CVS:
34756 * ext/ogg/gstogmparse.c: (gst_ogm_parse_class_init),
34757 (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
34758 (gst_ogm_text_parse_init), (gst_ogm_parse_change_state):
34759 In state change function, first chain up to parent class,
34760 then handle downwards state change stuff. Remove some
34761 commented out cruft from 0.8 code.
34763 2006-03-02 12:35:59 +0000 Tim-Philipp Müller <tim@centricular.net>
34765 ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
34766 Original commit message from CVS:
34767 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
34768 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
34769 (gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
34770 (gst_ogm_parse_chain):
34771 Don't remove/re-add source pad if the new caps are the same as
34772 the old caps anyway (#333042). When removing source pad, don't
34773 unref it afterwards - we didn't ref it when adding. Sprinkle some
34774 GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
34775 after using gst_pad_get_parent(). Return downstream flow return
34776 value in chain function.
34778 2006-03-02 11:28:23 +0000 Wim Taymans <wim.taymans@gmail.com>
34780 docs/plugins/: Fix hierarchy, added some more elements to the docs.
34781 Original commit message from CVS:
34782 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
34783 * docs/plugins/gst-plugins-base-plugins.args:
34784 * docs/plugins/gst-plugins-base-plugins.hierarchy:
34785 * docs/plugins/gst-plugins-base-plugins.interfaces:
34786 * docs/plugins/gst-plugins-base-plugins.signals:
34787 Fix hierarchy, added some more elements to the docs.
34788 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
34789 (gst_ffmpegcsp_get_type):
34790 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
34791 Fix docs for ffmpegcolorspace.
34793 2006-03-01 19:24:44 +0000 Tim-Philipp Müller <tim@centricular.net>
34795 gst/typefind/gsttypefindfunctions.c: Some typefinding fine-tuning:
34796 Original commit message from CVS:
34797 * gst/typefind/gsttypefindfunctions.c: (id3_type_find),
34798 (apetag_type_find), (ape_type_find), (plugin_init):
34799 Some typefinding fine-tuning:
34800 - rank ID3/APE tags in order of preference via probabilities, so that
34801 ID3v2 > APEv2 > APEv1 > ID3v1.
34802 - three or four bytes don't really justify MAXIMUM probability,
34803 change those to 'very likely' (musepack and monkeysaudio).
34805 2006-03-01 18:25:18 +0000 Wim Taymans <wim.taymans@gmail.com>
34808 Original commit message from CVS:
34809 * docs/plugins/Makefile.am:
34810 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
34811 * docs/plugins/gst-plugins-base-plugins-sections.txt:
34812 * ext/alsa/gstalsamixer.c:
34813 * ext/alsa/gstalsamixer.h:
34814 * ext/alsa/gstalsamixerelement.c:
34815 (gst_alsa_mixer_element_class_init), (gst_alsa_mixer_element_init):
34816 * ext/alsa/gstalsamixerelement.h:
34817 * ext/alsa/gstalsasink.c:
34818 * ext/alsa/gstalsasink.h:
34819 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
34820 (gst_alsasrc_init):
34821 * ext/alsa/gstalsasrc.h:
34823 Small code cleanups.
34825 2006-03-01 17:52:45 +0000 Wim Taymans <wim.taymans@gmail.com>
34827 ext/theora/Makefile.am: Dist new header too,
34828 Original commit message from CVS:
34829 * ext/theora/Makefile.am:
34830 Dist new header too,
34832 2006-03-01 17:39:28 +0000 Wim Taymans <wim.taymans@gmail.com>
34834 Fix some more docs.
34835 Original commit message from CVS:
34836 * docs/plugins/Makefile.am:
34837 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
34838 * docs/plugins/gst-plugins-base-plugins-sections.txt:
34839 * ext/gnomevfs/gstgnomevfssink.h:
34840 * ext/gnomevfs/gstgnomevfssrc.h:
34841 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
34842 * ext/vorbis/vorbisdec.h:
34843 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
34844 * ext/vorbis/vorbisenc.h:
34845 * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
34846 (vorbis_parse_chain), (vorbis_parse_change_state):
34847 * ext/vorbis/vorbisparse.h:
34848 * gst/audioconvert/gstaudioconvert.h:
34849 * gst/tcp/gsttcpserversink.h:
34850 * gst/videotestsrc/gstvideotestsrc.c:
34851 * gst/videotestsrc/gstvideotestsrc.h:
34852 * gst/volume/gstvolume.c:
34853 * gst/volume/gstvolume.h:
34854 Fix some more docs.
34855 Added docs for vorbisdec and vorbisparse.
34858 2006-03-01 16:24:37 +0000 Wim Taymans <wim.taymans@gmail.com>
34860 Updated/added documentation.
34861 Original commit message from CVS:
34862 * docs/plugins/Makefile.am:
34863 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
34864 * docs/plugins/gst-plugins-base-plugins-sections.txt:
34865 * ext/pango/gstclockoverlay.h:
34866 * ext/pango/gsttextoverlay.h:
34867 * ext/pango/gsttextrender.h:
34868 * ext/pango/gsttimeoverlay.h:
34869 * ext/theora/gsttheoradec.h:
34870 * ext/theora/gsttheoraenc.h:
34871 * ext/theora/theoradec.c:
34872 * ext/theora/theoraenc.c:
34873 * gst/audioconvert/gstaudioconvert.h:
34874 * gst/audiotestsrc/gstaudiotestsrc.h:
34875 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
34876 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
34877 * gst/tcp/gstmultifdsink.h:
34878 Updated/added documentation.
34879 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
34880 (gst_text_overlay_halign_get_type),
34881 (gst_text_overlay_wrap_mode_get_type),
34882 (gst_text_overlay_base_init), (gst_text_overlay_class_init),
34883 (gst_text_overlay_init), (gst_text_overlay_set_property),
34884 (gst_text_overlay_get_property):
34885 Fix up properties to be enums instead of string to make bindings,
34886 introspection and automatic GUI creation possible.
34887 Add getters for the properties.
34889 2006-02-28 21:21:07 +0000 Sébastien Moutte <sebastien@moutte.net>
34891 gst/audiotestsrc/gstaudiotestsrc.c: added defines of M_PI and M_PI_2
34892 Original commit message from CVS:
34893 * gst/audiotestsrc/gstaudiotestsrc.c:
34894 added defines of M_PI and M_PI_2
34895 * gst/ffmpegcolorspace/avcodec.h:
34896 removed #include "stdint.h" for win32 as _stdint.h is
34897 autogenerated to win32/common
34898 * win32/common/libgstaudio.def:
34899 * win32/common/libgsttag.def:
34902 some project files bugs corrected
34904 project files are reset to the default vs7 configuration
34905 (they link to msvcr71.dll using default optimizations)
34907 2006-02-28 19:08:12 +0000 Wim Taymans <wim.taymans@gmail.com>
34909 ext/gnomevfs/gstgnomevfssink.c: Fix some docs.
34910 Original commit message from CVS:
34911 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
34914 2006-02-28 13:52:04 +0000 Edward Hervey <bilboed@bilboed.com>
34916 ext/alsa/gstalsasrc.c: Set proper class on the ElementDetails:
34917 Original commit message from CVS:
34918 * ext/alsa/gstalsasrc.c:
34919 Set proper class on the ElementDetails:
34920 Source/Audio instead of Src/Audio
34922 2006-02-28 12:19:11 +0000 Edward Hervey <bilboed@bilboed.com>
34924 gst/videoscale/vs_scanline.c: Revert optimization in videoscale. It should go in liboil and have an appropriate liboi...
34925 Original commit message from CVS:
34926 * gst/videoscale/vs_scanline.c:
34927 (vs_scanline_resample_nearest_RGBA):
34928 Revert optimization in videoscale. It should go in liboil and have
34929 an appropriate liboil function.
34931 2006-02-28 11:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
34933 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
34934 Original commit message from CVS:
34935 * gst-libs/gst/audio/gstbaseaudiosink.c:
34936 (gst_base_audio_sink_provide_clock):
34937 Don't try to provide a clock in the NULL state.
34939 2006-02-28 11:04:47 +0000 Wim Taymans <wim.taymans@gmail.com>
34941 ext/ogg/gstoggdemux.c: Use GstSegment infrastructure to remove duplicated code and handle more seek cases correctly.
34942 Original commit message from CVS:
34943 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
34944 (gst_ogg_pad_event), (gst_ogg_pad_internal_chain),
34945 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
34946 (gst_ogg_demux_deactivate_current_chain),
34947 (gst_ogg_demux_activate_chain), (gst_ogg_demux_do_seek),
34948 (gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_info),
34949 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
34950 (gst_ogg_demux_loop), (gst_ogg_demux_change_state):
34951 Use GstSegment infrastructure to remove duplicated code
34952 and handle more seek cases correctly.
34954 2006-02-28 10:39:19 +0000 Wim Taymans <wim.taymans@gmail.com>
34956 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Don't ignore return code from ffmpeg convert function.
34957 Original commit message from CVS:
34958 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
34959 (gst_ffmpegcsp_transform):
34960 Don't ignore return code from ffmpeg convert function.
34961 * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
34962 Split out some long statements to ease debugging.
34964 2006-02-27 12:08:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
34966 ext/libvisual/visual.c: Don't use gst_pad_use_fixed_caps, because it prevents downstream from being able to renegotia...
34967 Original commit message from CVS:
34968 * ext/libvisual/visual.c: (gst_visual_init),
34969 (gst_vis_src_negotiate), (get_buffer), (plugin_init):
34970 Don't use gst_pad_use_fixed_caps, because it prevents downstream from
34971 being able to renegotiate the size. Instead, use the negotiation
34972 algorithm from the goom plugin to pick an initial output caps.
34973 Also, allow theoretical libvisual plugins that might support non-GL
34974 output even if they also do GL.
34976 2006-02-26 21:05:46 +0000 Julien Moutte <julien@moutte.net>
34978 ext/libvisual/visual.c: Load only non GL plugins. Fix some memleaks and possible negotiation issues.
34979 Original commit message from CVS:
34980 2006-02-26 Julien MOUTTE <julien@moutte.net>
34981 * ext/libvisual/visual.c: (gst_visual_init),
34982 (gst_visual_src_setcaps), (get_buffer), (gst_visual_chain),
34983 (plugin_init): Load only non GL plugins. Fix some memleaks and
34984 possible negotiation issues.
34986 2006-02-24 23:19:44 +0000 Julien Moutte <julien@moutte.net>
34988 gst-libs/gst/tag/tag.h: Adding Annodex tags here.
34989 Original commit message from CVS:
34990 2006-02-25 Julien MOUTTE <julien@moutte.net>
34991 * gst-libs/gst/tag/tag.h: Adding Annodex tags here.
34993 2006-02-24 18:55:27 +0000 Michael Smith <msmith@xiph.org>
34995 gst/typefind/gsttypefindfunctions.c: Fix CMML type find function to not require a specific minor version of the CMML ...
34996 Original commit message from CVS:
34997 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
34998 (cmml_type_find), (plugin_init):
34999 Fix CMML type find function to not require a specific minor version
35000 of the CMML header.
35001 Add an MPEG4 video elementary stream typefind function.
35003 2006-02-24 17:31:53 +0000 Michael Smith <msmith@xiph.org>
35005 ext/ogg/gstoggdemux.c: Annodex support in ogg demuxer. Doesn't do very much without the other annodex patches (to come).
35006 Original commit message from CVS:
35007 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
35008 (gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
35009 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
35010 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
35011 (gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
35012 (gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
35013 Annodex support in ogg demuxer. Doesn't do very much without the
35014 other annodex patches (to come).
35016 2006-02-24 16:21:34 +0000 Tim-Philipp Müller <tim@centricular.net>
35018 gst-libs/gst/riff/riff-media.c:
35019 Original commit message from CVS:
35020 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
35021 Pick up palette for MS video v1 (#327028, patch by:
35022 Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
35024 2006-02-24 13:54:04 +0000 Tim-Philipp Müller <tim@centricular.net>
35026 gst/ffmpegcolorspace/gstffmpegcolorspace.c: The 'palette_data' field from incoming RGB caps shouldn't be proxied on o...
35027 Original commit message from CVS:
35028 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
35029 (gst_ffmpegcsp_caps_remove_format_info),
35030 (gst_ffmpegcsp_get_unit_size):
35031 The 'palette_data' field from incoming RGB caps shouldn't be
35032 proxied on outgoing YUV caps; also, restrict unit size
35033 adjustment in case of paletted data only to the unit that
35034 actually has a palette. Fixes #330711.
35036 2006-02-24 12:18:14 +0000 Tim-Philipp Müller <tim@centricular.net>
35038 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Plug some memory leaks.
35039 Original commit message from CVS:
35040 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
35041 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
35042 (gst_ffmpegcsp_finalize), (gst_ffmpegcsp_class_init),
35043 (gst_ffmpegcsp_get_unit_size):
35044 Plug some memory leaks.
35046 2006-02-24 10:18:52 +0000 Tim-Philipp Müller <tim@centricular.net>
35048 sys/: Add some _CFLAGS and _LIBS that seem to be missing and/or required for Cygwin (see #317048).
35049 Original commit message from CVS:
35050 * sys/ximage/Makefile.am:
35051 * sys/xvimage/Makefile.am:
35052 Add some _CFLAGS and _LIBS that seem to be missing
35053 and/or required for Cygwin (see #317048).
35055 2006-02-24 00:07:18 +0000 Tim-Philipp Müller <tim@centricular.net>
35058 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
35059 Original commit message from CVS:
35060 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
35062 2006-02-22 18:46:46 +0000 Wim Taymans <wim.taymans@gmail.com>
35064 ext/alsa/gstalsasrc.c: Fix description as pointed out by caugier.
35065 Original commit message from CVS:
35066 * ext/alsa/gstalsasrc.c:
35067 Fix description as pointed out by caugier.
35069 2006-02-22 10:29:22 +0000 Edward Hervey <bilboed@bilboed.com>
35071 gst/typefind/gsttypefindfunctions.c: Better 3gp typefinding.
35072 Original commit message from CVS:
35073 Reviewed by : Edward Hervey <edward@fluendo.com>
35074 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
35076 Better 3gp typefinding.
35078 2006-02-21 12:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
35080 ext/gnomevfs/gstgnomevfssrc.c: Don't send EOS event here, the base class will send one for us.
35081 Original commit message from CVS:
35082 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
35083 Don't send EOS event here, the base class will send one for us.
35084 * gst/playback/gstplaybasebin.c: (prepare_output):
35085 Subpictures without video stream aren't allowed either.
35086 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
35087 Fix debug statement copy'n'paste-o.
35089 2006-02-21 12:05:18 +0000 Tim-Philipp Müller <tim@centricular.net>
35091 ext/alsa/gstalsamixer.c: Fix issues with mixer keeping state when muting/unmuting and when changing the volume whilst...
35092 Original commit message from CVS:
35093 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
35094 Fix issues with mixer keeping state when muting/unmuting
35095 and when changing the volume whilst muted (see #331763
35098 2006-02-20 18:27:06 +0000 Tim-Philipp Müller <tim@centricular.net>
35100 gst/subparse/gstsubparse.c: Set right caps given that we send escaped text. Also, honour <i></i>, <b></b> and <u></u>...
35101 Original commit message from CVS:
35102 * gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
35103 (parse_subrip), (gst_sub_parse_format_autodetect):
35104 Set right caps given that we send escaped text. Also,
35105 honour <i></i>, <b></b> and <u></u> markers that can be found
35106 in .srt files (fixes #310202).
35108 2006-02-20 16:21:14 +0000 Tim-Philipp Müller <tim@centricular.net>
35110 gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
35111 Original commit message from CVS:
35112 * gst-libs/gst/audio/mixerutils.c:
35113 (element_factory_rank_compare_func):
35114 Make order in which elements are tried more determinable.
35116 2006-02-20 15:57:51 +0000 Julien Moutte <julien@moutte.net>
35118 gst/playback/gstdecodebin.c: Make decodebin reusable by fixing remove_element_chain first and then introduce a cleane...
35119 Original commit message from CVS:
35120 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
35121 (remove_element_chain), (cleanup_decodebin),
35122 (gst_decode_bin_change_state): Make decodebin reusable by
35123 fixing remove_element_chain first and then introduce a
35124 cleaner in state change to ->NULL. (Closes #331678)
35125 ------------------------------------------------------
35127 2006-02-19 14:32:35 +0000 Wim Taymans <wim.taymans@gmail.com>
35129 ext/gnomevfs/gstgnomevfssink.c: use 0666 mask when creating files so umask gets applied correctly. Fixes #331295.
35130 Original commit message from CVS:
35131 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
35132 use 0666 mask when creating files so umask gets applied
35133 correctly. Fixes #331295.
35135 2006-02-19 14:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
35137 gst/subparse/: Add very basic parser for SSA subtitle streams (as often found in matroska files).
35138 Original commit message from CVS:
35139 * gst/subparse/Makefile.am:
35140 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
35141 (gst_ssa_parse_dispose), (gst_ssa_parse_init),
35142 (gst_ssa_parse_class_init), (gst_ssa_parse_src_event),
35143 (gst_ssa_parse_sink_event), (gst_ssa_parse_setcaps),
35144 (gst_ssa_parse_remove_override_codes), (gst_ssa_parse_parse_line),
35145 (gst_ssa_parse_chain), (gst_ssa_parse_change_state):
35146 * gst/subparse/gstssaparse.h:
35147 * gst/subparse/gstsubparse.c: (plugin_init):
35148 Add very basic parser for SSA subtitle streams (as often
35149 found in matroska files).
35151 2006-02-19 14:09:40 +0000 Tim-Philipp Müller <tim@centricular.net>
35153 gst/playback/gstdecodebin.c: That should be text/x-pango-markup, not text/x-pango-layout.
35154 Original commit message from CVS:
35155 * gst/playback/gstdecodebin.c: (mimetype_is_raw):
35156 That should be text/x-pango-markup, not text/x-pango-layout.
35158 2006-02-19 12:41:03 +0000 Julien Moutte <julien@moutte.net>
35160 ext/pango/gsttextoverlay.c: Polishing.
35161 Original commit message from CVS:
35162 2006-02-19 Julien MOUTTE <julien@moutte.net>
35163 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize):
35166 2006-02-19 12:05:23 +0000 Julien Moutte <julien@moutte.net>
35168 ext/pango/gsttextoverlay.c: Fix state change deadlock.
35169 Original commit message from CVS:
35170 2006-02-19 Julien MOUTTE <julien@moutte.net>
35171 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
35172 (gst_text_overlay_finalize), (gst_text_overlay_init),
35173 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
35174 (gst_text_overlay_render_text),
35175 (gst_text_overlay_text_pad_link),
35176 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
35177 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
35178 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
35179 Fix state change deadlock.
35181 2006-02-19 11:56:28 +0000 Julien Moutte <julien@moutte.net>
35183 ext/pango/gsttextoverlay.*: Fix seeking both for muxed formats and subtitles files.
35184 Original commit message from CVS:
35185 2006-02-19 Julien MOUTTE <julien@moutte.net>
35186 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
35187 (gst_text_overlay_finalize), (gst_text_overlay_init),
35188 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
35189 (gst_text_overlay_render_text),
35190 (gst_text_overlay_text_pad_link),
35191 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
35192 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
35193 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
35194 * ext/pango/gsttextoverlay.h: Fix seeking both for muxed formats
35195 and subtitles files.
35197 2006-02-19 00:40:38 +0000 Julien Moutte <julien@moutte.net>
35199 gst/playback/gstdecodebin.c: pango layout should be considered as row.
35200 Original commit message from CVS:
35201 2006-02-19 Julien MOUTTE <julien@moutte.net>
35202 * gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
35203 should be considered as row.
35205 2006-02-19 00:25:16 +0000 Julien Moutte <julien@moutte.net>
35207 gst/playback/gststreaminfo.*: Introduce language informations.
35208 Original commit message from CVS:
35209 2006-02-19 Julien MOUTTE <julien@moutte.net>
35210 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
35212 * gst/playback/gststreaminfo.h: Introduce language informations.
35214 2006-02-18 22:41:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35216 sys/: Set shared memory segments to be deleted as soon as we have attached, that way they get cleaned up automaticall...
35217 Original commit message from CVS:
35218 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
35219 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
35220 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
35221 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
35222 Set shared memory segments to be deleted as soon as we have attached,
35223 that way they get cleaned up automatically if we crash.
35225 2006-02-18 19:53:48 +0000 Julien Moutte <julien@moutte.net>
35227 ext/pango/: Those functions are called with lock held.
35228 Original commit message from CVS:
35229 2006-02-18 Julien MOUTTE <julien@moutte.net>
35230 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_get_text):
35231 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_get_text): Those
35232 functions are called with lock held.
35234 2006-02-18 19:51:47 +0000 Julien Moutte <julien@moutte.net>
35238 Original commit message from CVS:
35241 2006-02-18 19:10:35 +0000 Julien Moutte <julien@moutte.net>
35243 ext/pango/gsttextoverlay.c: Refactoring of textoverlay without collectpads. This now supports sparse subtitles coming...
35244 Original commit message from CVS:
35245 2006-02-18 Julien MOUTTE <julien@moutte.net>
35246 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
35247 (gst_text_overlay_finalize), (gst_text_overlay_init),
35248 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
35249 (gst_text_overlay_render_text),
35250 (gst_text_overlay_text_pad_link),
35251 (gst_text_overlay_text_pad_unlink),
35252 (gst_text_overlay_text_event),
35253 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
35254 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
35255 (gst_text_overlay_change_state): Refactoring of textoverlay
35256 without collectpads. This now supports sparse subtitles coming
35257 from a demuxer instead of a sub file. Seeking is still broken
35258 though. Need to discuss with wtay some more on how to handle
35260 * ext/pango/gsttextoverlay.h:
35261 * gst/playback/gstplaybin.c: (setup_sinks): Support linking with
35262 subtitles coming from the demuxer.
35264 2006-02-17 19:31:12 +0000 Wim Taymans <wim.taymans@gmail.com>
35266 ext/vorbis/vorbisenc.c: Use some more scaling functions.
35267 Original commit message from CVS:
35268 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
35269 (gst_vorbisenc_convert_sink):
35270 Use some more scaling functions.
35272 2006-02-17 16:12:11 +0000 Tim-Philipp Müller <tim@centricular.net>
35274 ext/cdparanoia/gstcdparanoiasrc.*: Add back 'transport-error' and 'uncorrected-error' signals and make them actually ...
35275 Original commit message from CVS:
35276 * ext/cdparanoia/gstcdparanoiasrc.c:
35277 (gst_cd_paranoia_src_class_init), (gst_cd_paranoia_dummy_callback),
35278 (gst_cd_paranoia_paranoia_callback),
35279 (gst_cd_paranoia_src_signal_is_being_watched),
35280 (gst_cd_paranoia_src_read_sector):
35281 * ext/cdparanoia/gstcdparanoiasrc.h:
35282 Add back 'transport-error' and 'uncorrected-error' signals and
35283 make them actually be fired when bad stuff happens (#319340).
35285 2006-02-17 14:07:01 +0000 Wim Taymans <wim.taymans@gmail.com>
35287 gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
35288 Original commit message from CVS:
35289 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
35290 (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
35291 (gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
35292 (gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
35293 (gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
35294 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
35295 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
35296 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
35297 (gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
35298 (gst_ring_buffer_clear):
35300 Added some G_LIKELY.
35302 2006-02-17 10:15:52 +0000 Wim Taymans <wim.taymans@gmail.com>
35304 gst-libs/gst/audio/TODO: Update TODO
35305 Original commit message from CVS:
35306 * gst-libs/gst/audio/TODO:
35308 * gst-libs/gst/audio/gstbaseaudiosink.c:
35309 (gst_base_audio_sink_get_offset):
35310 When trying to play samples ASAP and we don't have a
35311 previous sample, try to play at position 0 instead of
35312 an invalid position.
35314 2006-02-17 09:24:56 +0000 Wim Taymans <wim.taymans@gmail.com>
35316 ext/alsa/gstalsasink.c: Also release lock when we get an error in _reset(); fix an error message.
35317 Original commit message from CVS:
35318 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
35319 (gst_alsasink_reset):
35320 Also release lock when we get an error in _reset();
35321 fix an error message.
35323 2006-02-16 21:01:23 +0000 Tim-Philipp Müller <tim@centricular.net>
35325 ext/alsa/gstalsasink.*: Add support for more than 2 channels (#326720).
35326 Original commit message from CVS:
35327 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
35328 (gst_alsasink_init), (get_channel_free_structure),
35329 (caps_add_channel_configuration), (gst_alsasink_getcaps),
35330 (gst_alsasink_close):
35331 * ext/alsa/gstalsasink.h:
35332 Add support for more than 2 channels (#326720).
35334 2006-02-16 20:19:51 +0000 Tim-Philipp Müller <tim@centricular.net>
35336 gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
35337 Original commit message from CVS:
35338 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
35339 Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
35340 with 4 or 6 channels, assume a default channel layout to make things
35341 work (not sure there's anything else we can do in those cases).
35343 2006-02-16 19:18:46 +0000 Tim-Philipp Müller <tim@centricular.net>
35345 gst-libs/gst/audio/multichannel.c: Minor docs fix.
35346 Original commit message from CVS:
35347 * gst-libs/gst/audio/multichannel.c:
35349 * gst-libs/gst/riff/Makefile.am:
35350 * gst-libs/gst/riff/riff-ids.h:
35351 * gst-libs/gst/riff/riff-media.c:
35352 (gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
35353 Add support for WAVEFORMATEX, eg. PCM audio with more than two
35354 channels and a channel layout map.
35356 2006-02-16 17:06:46 +0000 Edward Hervey <bilboed@bilboed.com>
35358 gst/videoscale/vs_scanline.c: C-level optimization of the RGBA nearest neighbour function.
35359 Original commit message from CVS:
35360 Reviewed by Edward Hervey <edward@fluendo.com>
35361 * gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
35362 C-level optimization of the RGBA nearest neighbour function.
35363 Eventually this might end up in liboil with vectorized versions.
35365 2006-02-16 11:44:43 +0000 Tim-Philipp Müller <tim@centricular.net>
35367 gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
35368 Original commit message from CVS:
35369 * gst-libs/gst/audio/multichannel.c:
35370 (gst_audio_get_channel_positions):
35371 When we have more than 2 channels, but no channel layout is
35372 specified in the caps, return some default channel layout
35373 to the caller and warn about about a possibly buggy element
35374 (could be buggy filtercaps as well of course) (#317038).
35376 2006-02-16 09:29:38 +0000 Tim-Philipp Müller <tim@centricular.net>
35378 pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: Add gst-libs/gst/cdda to list of lib search paths.
35379 Original commit message from CVS:
35380 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
35381 Add gst-libs/gst/cdda to list of lib search paths.
35383 2006-02-15 12:20:47 +0000 Andy Wingo <wingo@pobox.com>
35385 ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating timestamp, update timestamp_end as well. Fixes a bugaboo. ...
35386 Original commit message from CVS:
35387 2006-02-15 Andy Wingo <wingo@pobox.com>
35388 * ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating
35389 timestamp, update timestamp_end as well. Fixes a bugaboo. I hope
35390 to the Lord Jesus that I do not have to touch the ogg muxer ever
35393 2006-02-15 12:07:57 +0000 Edward Hervey <bilboed@bilboed.com>
35395 gst/typefind/gsttypefindfunctions.c: quicktime movie files can also contain 'uuid' atoms.
35396 Original commit message from CVS:
35397 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
35398 quicktime movie files can also contain 'uuid' atoms.
35400 2006-02-14 18:52:52 +0000 Tim-Philipp Müller <tim@centricular.net>
35402 gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun...
35403 Original commit message from CVS:
35404 * gst/audioconvert/plugin.c: (plugin_init):
35405 Register the GstAudioChannelPosition enum type with the type
35406 system in the plugin_init function, so that it is known before
35407 any element actually makes use of multi-channel stuff. This is
35408 required for example if one wants to be able to deserialise/use
35409 a caps string with channel positions before any pipeline has
35410 been setup and started, like with gst-launch.
35412 2006-02-14 13:45:35 +0000 Wim Taymans <wim.taymans@gmail.com>
35414 gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
35415 Original commit message from CVS:
35416 * gst-libs/gst/audio/gstringbuffer.c:
35417 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
35418 (gst_ring_buffer_samples_done), (wait_segment),
35419 (gst_ring_buffer_commit), (gst_ring_buffer_clear):
35420 Add some compiler G_(UN_)LIKELY help.
35421 SIGNAL the ringbuffer waiters when going to PAUSED as well to
35422 make sure they can exit their functions. Should fix #330748
35424 2006-02-13 20:49:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35426 Windows does not have long long; copy the generated _stdint.h
35427 Original commit message from CVS:
35431 * win32/common/_stdint.h:
35432 Windows does not have long long; copy the generated _stdint.h
35433 * win32/common/interfaces-enumtypes.c:
35434 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
35435 (gst_mixer_track_flags_get_type),
35436 (gst_tuner_channel_flags_get_type):
35437 * win32/common/multichannel-enumtypes.c:
35438 (gst_audio_channel_position_get_type):
35441 2006-02-13 18:49:02 +0000 Wim Taymans <wim.taymans@gmail.com>
35443 gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
35444 Original commit message from CVS:
35445 * gst-libs/gst/audio/gstbaseaudiosink.c:
35446 (gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
35447 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
35448 Always sync on first sample we receive when starting.
35450 2006-02-13 15:59:48 +0000 Wim Taymans <wim.taymans@gmail.com>
35452 gst/playback/gstplaybin.c: Update vis bin docs.
35453 Original commit message from CVS:
35454 * gst/playback/gstplaybin.c: (gen_vis_element):
35455 Update vis bin docs.
35456 Move queue after tee so we don't queue video buffers but
35457 audio samples instead. Fixes problems where the video queue
35458 is filled and the audio queue empty.
35460 2006-02-13 15:17:34 +0000 Tim-Philipp Müller <tim@centricular.net>
35462 gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
35463 Original commit message from CVS:
35464 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
35465 No need to push an EOS event here, GstBaseSrc will do that for us
35466 when we return FLOW_UNEXPECTED.
35468 2006-02-12 14:54:55 +0000 Wim Taymans <wim.taymans@gmail.com>
35470 gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
35471 Original commit message from CVS:
35472 * gst-libs/gst/audio/gstbaseaudiosink.c:
35473 (gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
35474 (gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
35475 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
35476 Use scale functions when possible.
35477 Fix error messages.
35478 Free clockid when after waiting for EOS.
35479 Use G_(UN_)LIKLY when it makes sense.
35480 Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
35482 2006-02-12 14:26:55 +0000 Edward Hervey <bilboed@bilboed.com>
35484 gst/playback/gstplaybasebin.c: Remove stray semi-colon (fixes #330888).
35485 Original commit message from CVS:
35486 * gst/playback/gstplaybasebin.c: (prepare_output):
35487 Remove stray semi-colon (fixes #330888).
35489 2006-02-11 23:35:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35491 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...
35492 Original commit message from CVS:
35493 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
35494 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
35495 Fix up the XShm call testing so that we catch errors, and don't
35496 cause new ones by attempting to detach from a segment we failed
35497 to attach to. Fixes #312439.
35499 2006-02-10 11:29:55 +0000 Edward Hervey <bilboed@bilboed.com>
35501 gst/typefind/gsttypefindfunctions.c: Added flv file typefind (video/x-flv).
35502 Original commit message from CVS:
35503 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
35504 Added flv file typefind (video/x-flv).
35506 2006-02-10 10:53:33 +0000 Edward Hervey <bilboed@bilboed.com>
35508 gst-libs/gst/riff/riff-media.c: Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
35509 Original commit message from CVS:
35510 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
35511 (gst_riff_create_video_template_caps):
35512 Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
35513 Also added the caps to the default set of riff video caps.
35515 2006-02-09 19:05:23 +0000 Andy Wingo <wingo@pobox.com>
35517 ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start time and the end time of the last packet in the page.
35518 Original commit message from CVS:
35519 2006-02-09 Andy Wingo <wingo@pobox.com>
35520 * ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start
35521 time and the end time of the last packet in the page.
35522 (gst_ogg_mux_pad_queue_page): In addition to setting the timestamp
35523 on the pages in our queue, set the duration as well. Reflow a
35525 (gst_ogg_mux_collected): Keep track of GstOggPad->timestamp_end.
35526 Fixes bad muxing order.
35528 2006-02-09 17:04:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35530 gst-libs/gst/rtp/gstbasertppayload.c: update seqnum before setting it on the packet; this makes sure that the timesta...
35531 Original commit message from CVS:
35532 * gst-libs/gst/rtp/gstbasertppayload.c:
35533 (gst_basertppayload_setcaps), (gst_basertppayload_push):
35534 update seqnum before setting it on the packet; this makes sure
35535 that the timestamp and seqnum properties match after pushing
35538 2006-02-09 12:16:35 +0000 Andy Wingo <wingo@pobox.com>
35542 Original commit message from CVS:
35545 2006-02-09 11:46:03 +0000 Andy Wingo <wingo@pobox.com>
35547 * gst-libs/gst/audio/gstringbuffer.c:
35548 * win32/common/config.h:
35550 Original commit message from CVS:
35553 2006-02-09 11:36:18 +0000 Andy Wingo <wingo@pobox.com>
35555 gst-libs/gst/audio/gstringbuffer.c
35556 Original commit message from CVS:
35557 2006-02-09 Andy Wingo <wingo@pobox.com>
35558 * gst-libs/gst/audio/gstringbuffer.c
35559 (gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
35560 overflow after 13.5 hours of recording. Kapow!
35561 * ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
35562 the buffer size -- we don't care about underrun/overrun reporting
35563 right now, just need to return a useful value.
35565 2006-02-09 11:21:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35567 configure.ac: Back to CVS
35568 Original commit message from CVS:
35572 === release 0.10.3 ===
35574 2006-02-09 11:18:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35580 * docs/plugins/inspect/plugin-adder.xml:
35581 * docs/plugins/inspect/plugin-alsa.xml:
35582 * docs/plugins/inspect/plugin-audioconvert.xml:
35583 * docs/plugins/inspect/plugin-audiorate.xml:
35584 * docs/plugins/inspect/plugin-audioresample.xml:
35585 * docs/plugins/inspect/plugin-audiotestsrc.xml:
35586 * docs/plugins/inspect/plugin-cdparanoia.xml:
35587 * docs/plugins/inspect/plugin-decodebin.xml:
35588 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
35589 * docs/plugins/inspect/plugin-gnomevfs.xml:
35590 * docs/plugins/inspect/plugin-libvisual.xml:
35591 * docs/plugins/inspect/plugin-ogg.xml:
35592 * docs/plugins/inspect/plugin-pango.xml:
35593 * docs/plugins/inspect/plugin-playbin.xml:
35594 * docs/plugins/inspect/plugin-subparse.xml:
35595 * docs/plugins/inspect/plugin-tcp.xml:
35596 * docs/plugins/inspect/plugin-theora.xml:
35597 * docs/plugins/inspect/plugin-typefindfunctions.xml:
35598 * docs/plugins/inspect/plugin-video4linux.xml:
35599 * docs/plugins/inspect/plugin-videorate.xml:
35600 * docs/plugins/inspect/plugin-videoscale.xml:
35601 * docs/plugins/inspect/plugin-videotestsrc.xml:
35602 * docs/plugins/inspect/plugin-volume.xml:
35603 * docs/plugins/inspect/plugin-vorbis.xml:
35604 * docs/plugins/inspect/plugin-ximagesink.xml:
35605 * docs/plugins/inspect/plugin-xvimagesink.xml:
35606 * win32/common/config.h:
35608 Original commit message from CVS:
35611 2006-02-08 18:37:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35613 configure.ac: Drat. Bump libtool version number for new API.
35614 Original commit message from CVS:
35616 Drat. Bump libtool version number for new API.
35617 Prelease 0.10.2.3 (of 0.10.3)
35619 2006-02-08 15:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35621 0.10.2.2 prerelease (of 0.10.3).
35622 Original commit message from CVS:
35624 * win32/common/config.h:
35625 0.10.2.2 prerelease (of 0.10.3).
35627 2006-02-08 15:50:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35629 gst/tcp/gsttcpclientsrc.c: Revert Andy's newsegment change pending a more correct fix.
35630 Original commit message from CVS:
35631 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
35632 Revert Andy's newsegment change pending a more correct
35635 2006-02-08 12:46:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35652 Original commit message from CVS:
35655 2006-02-08 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35657 * gst/tcp/gstmultifdsink.c:
35659 Original commit message from CVS:
35662 2006-02-08 09:20:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35664 gst/typefind/gsttypefindfunctions.c: detect more files as 3gp group and reorder the iso file formats
35665 Original commit message from CVS:
35667 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
35668 (qt_type_find), (plugin_init):
35669 detect more files as 3gp
35670 group and reorder the iso file formats
35672 2006-02-07 18:32:00 +0000 Tim-Philipp Müller <tim@centricular.net>
35674 ext/vorbis/vorbis.c: Register musicbrainz tags, so apps don't have to.
35675 Original commit message from CVS:
35676 * ext/vorbis/vorbis.c: (plugin_init):
35677 Register musicbrainz tags, so apps don't have to.
35679 2006-02-07 17:44:53 +0000 Tim-Philipp Müller <tim@centricular.net>
35681 gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
35682 Original commit message from CVS:
35683 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
35684 (gst_tag_to_vorbis_tag):
35685 Make sure we called gst_tag_register_musicbrainz_tags()
35686 before possibly mapping a vorbiscomment string from/to a
35689 2006-02-07 16:16:41 +0000 Tim-Philipp Müller <tim@centricular.net>
35691 gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
35692 Original commit message from CVS:
35693 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
35694 In case we can't find the required number of consecutive
35695 mpeg audio frames to positively identify an MPEG audio
35696 stream, check if there's at least a valid mpeg audio
35697 frame right at offset 0 and if so suggest mpeg/audio
35698 caps with a very low probability (#153004).
35700 2006-02-07 15:52:26 +0000 Andy Wingo <wingo@pobox.com>
35702 gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to a TIME segment if we get timestamped buffers. Requir...
35703 Original commit message from CVS:
35704 2006-02-07 Andy Wingo <wingo@pobox.com>
35705 * gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
35706 a TIME segment if we get timestamped buffers. Requires recent
35707 fixes in core to work properly.
35709 2006-02-07 14:57:46 +0000 Tim-Philipp Müller <tim@centricular.net>
35711 gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
35712 Original commit message from CVS:
35713 * gst/playback/gstplaybasebin.c: (prepare_output):
35714 Don't print the URI as part of the error message, it
35715 makes error dialogs look rather ugly, especially if
35716 the URI is very long or has characters in it that
35719 2006-02-07 13:11:31 +0000 Tim-Philipp Müller <tim@centricular.net>
35721 gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
35722 Original commit message from CVS:
35723 * gst/playback/gstplaybasebin.c: (prepare_output):
35724 Error out if we have only text or subtitles, but nothing
35725 else. Also error out if we have subtitles but no video
35728 2006-02-07 11:44:39 +0000 Tim-Philipp Müller <tim@centricular.net>
35730 ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
35731 Original commit message from CVS:
35732 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
35733 Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
35734 Post an error message on the bus when we encounter an
35735 error, which will hopefully be more meaningful than the
35736 'Internal Flow Error' message users get to see if we
35737 just return GST_FLOW_ERROR.
35739 2006-02-07 11:28:04 +0000 Andy Wingo <wingo@pobox.com>
35741 configure.ac (GST_MAJORMINOR): Update core version req to 0.10.2.2, for the collectpads API addition (#330244).
35742 Original commit message from CVS:
35743 2006-02-07 Andy Wingo <wingo@pobox.com>
35744 * configure.ac (GST_MAJORMINOR): Update core version req to
35745 0.10.2.2, for the collectpads API addition (#330244).
35747 2006-02-06 19:09:26 +0000 Tim-Philipp Müller <tim@centricular.net>
35749 ext/gnomevfs/gstgnomevfs.c: Return FALSE from plugin_init() when GnomeVFS can't be initialised for some reason (#3284...
35750 Original commit message from CVS:
35751 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
35752 Return FALSE from plugin_init() when GnomeVFS can't
35753 be initialised for some reason (#328423).
35755 2006-02-06 13:26:54 +0000 Julien Moutte <julien@moutte.net>
35757 ext/pango/gsttextoverlay.c: Stick to seeking theory until i find the bug.
35758 Original commit message from CVS:
35759 2006-02-06 Julien MOUTTE <julien@moutte.net>
35760 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
35761 Stick to seeking theory until i find the bug.
35762 * gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
35764 2006-02-06 12:38:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35766 Make theoraenc and the tests leak free. Like, really.
35767 Original commit message from CVS:
35768 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
35769 (theora_enc_finalize), (theora_enc_sink_setcaps),
35770 (theora_set_header_on_caps), (theora_enc_chain),
35771 (theora_enc_change_state):
35772 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
35773 Make theoraenc and the tests leak free. Like, really.
35775 2006-02-05 23:31:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35777 Add a finalize method to ensure we clean up state even if someone omitted the state change back to NULL.
35778 Original commit message from CVS:
35779 (theora_enc_finalize), (theora_enc_sink_setcaps):
35780 Add a finalize method to ensure we clean up state even if
35781 someone omitted the state change back to NULL.
35782 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
35783 (gst_vorbisenc_chain):
35784 Free some more leaked bits.
35785 * tests/check/pipelines/theoraenc.c: (start_pipeline),
35787 Wait for state changes to happen if they're ASYNC.
35788 This ought to teach those fancy pants buildbots a lesson.
35790 2006-02-05 22:47:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35792 gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
35793 Original commit message from CVS:
35794 * gst-libs/gst/tag/gstid3tag.c:
35795 Add mapping for ID3 International Standard Recording Code
35798 2006-02-05 22:44:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35800 ext/vorbis/vorbisenc.c: Don't leak tag names.
35801 Original commit message from CVS:
35802 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
35803 Don't leak tag names.
35805 2006-02-05 18:22:01 +0000 Tim-Philipp Müller <tim@centricular.net>
35807 Split libgsttag docs into multiple sections.
35808 Original commit message from CVS:
35809 * docs/libs/gst-plugins-base-libs-docs.sgml:
35810 * docs/libs/gst-plugins-base-libs-sections.txt:
35811 * gst-libs/gst/tag/gstid3tag.c:
35812 * gst-libs/gst/tag/gstvorbistag.c:
35813 * gst-libs/gst/tag/tags.c:
35814 Split libgsttag docs into multiple sections.
35816 2006-02-05 18:01:33 +0000 Tim-Philipp Müller <tim@centricular.net>
35818 Add libgsttag to the docs.
35819 Original commit message from CVS:
35820 * docs/libs/Makefile.am:
35821 * docs/libs/gst-plugins-base-libs-docs.sgml:
35822 * docs/libs/gst-plugins-base-libs-sections.txt:
35823 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
35824 * gst-libs/gst/tag/gstvorbistag.c:
35825 * gst-libs/gst/tag/tag.h:
35826 * gst-libs/gst/tag/tags.c:
35827 Add libgsttag to the docs.
35829 2006-02-05 17:21:23 +0000 Julien Moutte <julien@moutte.net>
35831 ext/pango/gsttextoverlay.c: Fix clockoverlay.
35832 Original commit message from CVS:
35833 2006-02-05 Julien MOUTTE <julien@moutte.net>
35834 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
35835 (gst_text_overlay_init), (gst_text_overlay_src_event),
35836 (gst_text_overlay_collected): Fix clockoverlay.
35838 2006-02-05 17:15:17 +0000 Tim-Philipp Müller <tim@centricular.net>
35840 docs/libs/compiling.sgml: Fix typo: it's pkg-config, not pkg-gconfig
35841 Original commit message from CVS:
35842 * docs/libs/compiling.sgml:
35843 Fix typo: it's pkg-config, not pkg-gconfig
35844 * docs/libs/gst-plugins-base-libs-docs.sgml:
35845 * docs/libs/gst-plugins-base-libs-sections.txt:
35846 * docs/libs/tmpl/gstgconf.sgml:
35847 There is no libgstgconf in 0.10, remove it
35850 2006-02-05 16:03:48 +0000 Julien Moutte <julien@moutte.net>
35852 docs/libs/tmpl/gstcolorbalance.sgml: Updated.
35853 Original commit message from CVS:
35854 2006-02-05 Julien MOUTTE <julien@moutte.net>
35855 * docs/libs/tmpl/gstcolorbalance.sgml: Updated.
35856 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
35857 (gst_text_overlay_src_event), (gst_text_overlay_collected):
35858 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
35859 (gst_sub_parse_class_init), (gst_sub_parse_init),
35860 (gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
35861 (parse_mpsub), (parser_state_init), (handle_buffer),
35862 (gst_sub_parse_chain), (gst_sub_parse_sink_event),
35864 * gst/subparse/gstsubparse.h: Introduce seeking code.
35866 2006-02-05 15:14:06 +0000 Tim-Philipp Müller <tim@centricular.net>
35868 gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
35869 Original commit message from CVS:
35870 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
35871 Add comment about LANGUAGE tag inconsistency (we want
35872 ISO-639-1, but extract three-letter identifiers?)
35874 Add two translatable files.
35876 2006-02-05 14:59:28 +0000 Tim-Philipp Müller <tim@centricular.net>
35878 gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
35879 Original commit message from CVS:
35880 * gst-libs/gst/tag/Makefile.am:
35881 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
35882 * gst-libs/gst/tag/tag.h:
35883 * gst-libs/gst/tag/tags.c:
35884 (gst_tag_register_musicbrainz_tags_internal),
35885 (gst_tag_register_musicbrainz_tags):
35886 Forward-port some tags stuff from the 0.8 branch. This is
35887 mostly the addition of musicbrainz tags and their mapping
35888 to vorbistags, and a vorbistag mapping of the language tag.
35890 2006-02-05 12:06:25 +0000 Julien Moutte <julien@moutte.net>
35892 gst/playback/gstplaybin.c: Fix broken code refactoring.
35893 Original commit message from CVS:
35894 2006-02-05 Julien MOUTTE <julien@moutte.net>
35895 * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
35898 2006-02-05 03:05:41 +0000 David Schleef <ds@schleef.org>
35900 Add Dirac typefinding and add dirac format to oggmux.
35901 Original commit message from CVS:
35902 * ext/ogg/gstoggmux.c:
35903 * gst/typefind/gsttypefindfunctions.c:
35904 Add Dirac typefinding and add dirac format to oggmux.
35906 2006-02-04 07:49:03 +0000 Michael Smith <msmith@xiph.org>
35909 Improve error message for liboil missingness.
35910 Original commit message from CVS:
35911 Improve error message for liboil missingness.
35913 2006-02-03 19:23:41 +0000 Tim-Philipp Müller <tim@centricular.net>
35915 gst/playback/gstdecodebin.c: Don't put essential function call into g_return_*() macro, otherwise it'll all be replac...
35916 Original commit message from CVS:
35917 * gst/playback/gstdecodebin.c: (try_to_link_1):
35918 Don't put essential function call into
35919 g_return_*() macro, otherwise it'll all be
35920 replaced by NOOPs when compiling with
35921 G_DISABLE_CHECKS defined.
35923 2006-02-03 17:45:44 +0000 Edgard Lima <edgard.lima@indt.org.br>
35926 * ext/ogg/gstoggdemux.c:
35927 * ext/ogg/gstoggparse.c:
35928 * gst/tcp/gsttcpserversink.c:
35929 * sys/v4l/v4lsrc_calls.c:
35930 * sys/v4l/v4lsrc_calls.h:
35931 Just make it compile with --disable-gst-debug.
35932 Original commit message from CVS:
35933 Just make it compile with --disable-gst-debug.
35935 2006-02-03 12:51:47 +0000 Wim Taymans <wim.taymans@gmail.com>
35937 ext/alsa/gstalsasink.*: Add lock to protect alsa calls.
35938 Original commit message from CVS:
35939 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
35940 (gst_alsasink_class_init), (gst_alsasink_init),
35941 (gst_alsasink_write), (gst_alsasink_reset):
35942 * ext/alsa/gstalsasink.h:
35943 Add lock to protect alsa calls.
35944 Implement reset to flush samples ASAP, does not work
35947 2006-02-02 18:18:31 +0000 Wim Taymans <wim.taymans@gmail.com>
35949 gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
35950 Original commit message from CVS:
35951 * gst-libs/gst/audio/gstbaseaudiosink.c:
35952 (gst_base_audio_sink_provide_clock):
35953 Ugh.. getting late I guess...
35955 2006-02-02 18:13:26 +0000 Wim Taymans <wim.taymans@gmail.com>
35957 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
35958 Original commit message from CVS:
35959 * gst-libs/gst/audio/gstbaseaudiosink.c:
35960 (gst_base_audio_sink_provide_clock),
35961 (gst_base_audio_sink_set_property),
35962 (gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
35963 Don't try to provide a clock when we are not negotiated since
35964 we might not be able to make it run.
35966 2006-02-02 17:51:48 +0000 Tim-Philipp Müller <tim@centricular.net>
35968 gst/playback/gstdecodebin.c: Unlinking two source pads is ... hard.
35969 Original commit message from CVS:
35970 * gst/playback/gstdecodebin.c: (try_to_link_1):
35971 Unlinking two source pads is ... hard.
35973 2006-02-02 12:14:35 +0000 Wim Taymans <wim.taymans@gmail.com>
35975 gst-libs/gst/audio/TODO: Updated.
35976 Original commit message from CVS:
35977 * gst-libs/gst/audio/TODO:
35979 * gst-libs/gst/audio/gstbaseaudiosink.c:
35980 (gst_base_audio_sink_drain), (gst_base_audio_sink_event):
35981 On EOS, wait till the last sample is played before posting EOS.
35983 2006-02-02 08:53:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35985 * tests/check/pipelines/theoraenc.c:
35986 comment on my understanding
35987 Original commit message from CVS:
35988 comment on my understanding
35990 2006-02-02 08:47:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35993 * tests/check/pipelines/theoraenc.c:
35994 reformat to fit 80 chars
35995 Original commit message from CVS:
35996 reformat to fit 80 chars
35998 2006-02-02 00:04:37 +0000 Kai Vehmanen <kv2004@eca.cx>
36000 gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
36001 Original commit message from CVS:
36002 2006-02-01 Philippe Kalaf <burger at speedy dot org>
36003 * gst-libs/gst/rtp/gstbasertpdepayload.c:
36004 Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
36005 setting queue_delay to zero. Also avoid thread being started if
36006 queue_delay is zero.
36008 2006-02-01 14:51:29 +0000 Tim-Philipp Müller <tim@centricular.net>
36010 gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
36011 Original commit message from CVS:
36012 * gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
36013 Make test work again by connecting fakesinks to each decoded pad,
36014 which makes the pipeline wait until each fakesink has a buffer
36015 queued before going to PAUSED state. At that point we know the
36016 decodebin pads are negotiated.
36018 2006-02-01 11:59:47 +0000 Tim-Philipp Müller <tim@centricular.net>
36020 gst/: Pass unhandled queries to the parent class's query function.
36021 Original commit message from CVS:
36022 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
36023 (gst_cdda_base_src_handle_event):
36024 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
36025 Pass unhandled queries to the parent class's query function.
36027 2006-02-01 11:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
36029 Pass unhandled queries upstream instead of just dropping them (#326447). Also, fix supported query types list for som...
36030 Original commit message from CVS:
36031 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
36032 (gst_ogg_pad_src_query):
36033 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
36034 * ext/theora/theoradec.c: (theora_dec_src_query),
36035 (theora_dec_sink_query):
36036 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
36037 (vorbis_dec_sink_query):
36038 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
36039 (gst_vorbisenc_sink_query):
36040 * gst/adder/gstadder.c: (gst_adder_query):
36041 Pass unhandled queries upstream instead of just
36042 dropping them (#326447). Also, fix supported
36043 query types list for some elements.
36045 2006-02-01 09:58:15 +0000 Tim-Philipp Müller <tim@centricular.net>
36047 gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
36048 Original commit message from CVS:
36049 * gst/typefind/gsttypefindfunctions.c: (au_type_find),
36050 (paris_type_find), (ilbc_type_find), (plugin_init):
36051 Fix typefinding for audio/x-au, audio/x-paris and
36052 audio/iLBC-sh. We cannot use the START_WITH macros
36053 here, because there can only be one typefind factory
36054 with the same name (caps), so the second one would
36055 replace the first one and the first one would never
36056 be called when doing typefinding (see #161712).
36058 2006-01-31 19:25:10 +0000 Wim Taymans <wim.taymans@gmail.com>
36060 ext/vorbis/vorbisdec.c: Use scale_int when we can, add some more scaling.
36061 Original commit message from CVS:
36062 * ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
36063 (vorbis_handle_header_packet), (vorbis_dec_push),
36064 (vorbis_handle_data_packet):
36065 Use scale_int when we can, add some more scaling.
36066 Check packettype before parsing it.
36068 2006-01-31 17:44:35 +0000 Wim Taymans <wim.taymans@gmail.com>
36070 ext/theora/theoradec.c: Call right _scale functions.
36071 Original commit message from CVS:
36072 * ext/theora/theoradec.c: (_theora_granule_time),
36073 (theora_dec_src_convert), (theora_dec_sink_convert):
36074 Call right _scale functions.
36075 Use parameter instead of some other random value.
36077 2006-01-31 17:27:00 +0000 Wim Taymans <wim.taymans@gmail.com>
36079 ext/theora/theoradec.c: Use higher precision timestamps calculation.
36080 Original commit message from CVS:
36081 * ext/theora/theoradec.c: (_theora_granule_frame),
36082 (_theora_granule_time), (_inc_granulepos),
36083 (theora_dec_src_convert), (theora_dec_sink_convert),
36084 (theora_handle_type_packet), (theora_handle_data_packet),
36085 (theora_dec_chain):
36086 Use higher precision timestamps calculation.
36087 Convert some other conversions to _scale.
36089 2006-01-31 17:19:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
36091 gst/: initialize gst_controller before using
36092 Original commit message from CVS:
36093 * gst/audiotestsrc/gstaudiotestsrc.c:
36094 (gst_audio_test_src_create_sine_table), (plugin_init):
36095 * gst/volume/gstvolume.c: (plugin_init):
36096 initialize gst_controller before using
36098 2006-01-31 16:26:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36100 tests/check/pipelines/: Define constant using G_GINT64_CONSTANT to avoid errors when passing it around - otherwise it...
36101 Original commit message from CVS:
36102 * tests/check/pipelines/theoraenc.c:
36103 * tests/check/pipelines/vorbisenc.c:
36104 Define constant using G_GINT64_CONSTANT to avoid errors when
36105 passing it around - otherwise it gets truncated to 32 bits.
36106 Fixes failing tests.
36108 2006-01-31 15:36:13 +0000 Andy Wingo <wingo@pobox.com>
36110 sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the caps being set doesn't have a framerate value. Basic...
36111 Original commit message from CVS:
36112 2006-01-31 Andy Wingo <wingo@pobox.com>
36113 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
36114 caps being set doesn't have a framerate value. Basically a stopgap
36116 * ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
36117 technically correct enough to put into core though.
36118 (gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
36119 DURATION. Fixes theoraenc ! oggmux.
36120 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
36121 fraction, not double.
36123 2006-01-31 12:23:35 +0000 Christian Schaller <uraeus@gnome.org>
36125 * gst-plugins-base.spec.in:
36126 update with latest files
36127 Original commit message from CVS:
36128 update with latest files
36130 2006-01-30 23:42:54 +0000 Sébastien Moutte <sebastien@moutte.net>
36132 win32/vs7: add vs7 project files created by Sergey Scobich
36133 Original commit message from CVS:
36135 add vs7 project files created by Sergey Scobich
36137 2006-01-30 22:18:53 +0000 Sébastien Moutte <sebastien@moutte.net>
36139 win32/vs8: add vs8 project files created by Sergey Scobich
36140 Original commit message from CVS:
36142 add vs8 project files created by Sergey Scobich
36144 2006-01-30 19:22:22 +0000 Andy Wingo <wingo@pobox.com>
36146 ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare timestamp + duration, not just timestamp -- ogg pages should ...
36147 Original commit message from CVS:
36148 2006-01-30 Andy Wingo <wingo@pobox.com>
36149 * ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
36150 timestamp + duration, not just timestamp -- ogg pages should be
36151 ordered by stop time. Necessary fix given the change in vorbis
36154 2006-01-30 19:21:07 +0000 Andy Wingo <wingo@pobox.com>
36157 * ext/theora/gsttheoraenc.h:
36158 * ext/theora/theoraenc.c:
36159 * tests/check/pipelines/theoraenc.c:
36160 ext/theora/theoraenc.c (theora_enc_sink_setcaps)
36161 Original commit message from CVS:
36162 2006-01-30 Andy Wingo <wingo@pobox.com>
36163 * ext/theora/theoraenc.c (theora_enc_sink_setcaps)
36164 (gst_theora_enc_init): Pull the granule shift out of the encoder.
36165 (granulepos_add): New function, handles the messiness of adjusting
36167 (theora_buffer_from_packet):
36168 (theora_enc_chain):
36169 (theora_enc_sink_event): Use granulepos_add, not +.
36170 * tests/check/pipelines/theoraenc.c
36171 (check_buffer_granulepos_from_starttime): Just check the frame
36172 count, not the actual granulepos -- we can't dictate to the
36173 encoder when it should be placing keyframes.
36175 2006-01-30 18:17:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36177 ext/gnomevfs/gstgnomevfssrc.c: SERVICE_NOT_AVAILABLE happens for example when you're trying to play an http:// stream...
36178 Original commit message from CVS:
36179 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
36180 SERVICE_NOT_AVAILABLE happens for example when you're trying to
36181 play an http:// stream from a server that's not serving
36183 2006-01-30 17:08:11 +0000 Andy Wingo <wingo@pobox.com>
36185 tests/check/pipelines/: Totally remove the UINT64_CONSTANT macro, doesn't appear to be needed or available.
36186 Original commit message from CVS:
36187 2006-01-30 Andy Wingo <wingo@pobox.com>
36188 * tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
36189 * tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
36190 remove the UINT64_CONSTANT macro, doesn't appear to be needed or
36193 2006-01-30 17:01:54 +0000 Andy Wingo <wingo@pobox.com>
36195 ext/theora/: Same changes as were done to vorbisenc, although theoraenc was timestamping correctly. Added handling of...
36196 Original commit message from CVS:
36197 2006-01-30 Andy Wingo <wingo@pobox.com>
36198 * ext/theora/gsttheoraenc.h:
36199 * ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
36200 although theoraenc was timestamping correctly. Added handling of
36201 streams that start with nonzero timestamps.
36202 * tests/check/Makefile.am:
36203 * tests/check/pipelines/theoraenc.c: New file, basically does same
36204 tests as vorbisenc.
36205 * tests/check/pipelines/vorbisenc.c: I claim these bugs.
36207 2006-01-30 16:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
36209 gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
36210 Original commit message from CVS:
36211 * gst-libs/gst/audio/gstaudiosink.c:
36212 (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
36213 (gst_audioringbuffer_pause):
36214 Implement pause that does not wait for completion.
36215 * gst-libs/gst/audio/gstbaseaudiosink.c:
36216 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
36217 Don't drop buffers when going to PAUSED but perform preroll on
36218 remaining samples now that core base class supports this.
36219 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
36220 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
36221 (gst_ring_buffer_commit):
36222 Pause should not signal waiters.
36223 Implement return value of _commit correctly.
36225 2006-01-30 15:01:28 +0000 Andy Wingo <wingo@pobox.com>
36227 tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
36228 Original commit message from CVS:
36229 2006-01-30 Andy Wingo <wingo@pobox.com>
36230 * tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
36231 * ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
36232 updated to timestamp from the first sample, not the last.
36233 (gst_vorbisenc_buffer_from_header_packet): New function, takes
36234 special care of granulepos and timestamp for header packets.
36235 (gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
36236 when the first buffer has a nonzero timestamp.
36237 * ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
36238 (GstVorbisEnc.subgranule_offset): New members. Take care of the
36239 case when the first audio buffer we get has a nonzero timestamp.
36240 (GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
36241 properly timestamp vorbis buffers with the time of the first
36242 sample, not the last.
36243 * ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
36244 vorbis_granule_time_copy -- now it takes the granule/subgranule
36245 offset into account.
36246 * tests/check/pipelines/vorbisenc.c: New test for correctness of
36247 timestamps, durations, and granulepos on buffers produced by
36250 2006-01-30 14:42:28 +0000 Eric Jonas <jonas@mit.edu>
36252 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Patch from Eric Jonas to support conversions to/from UYVY (Fixes: #324626)
36253 Original commit message from CVS:
36254 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
36255 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
36256 Patch from Eric Jonas to support conversions to/from UYVY
36259 2006-01-30 08:11:14 +0000 Julien Moutte <julien@moutte.net>
36261 gst/playback/: Implement subtitles.
36262 Original commit message from CVS:
36263 2006-01-30 Julien MOUTTE <julien@moutte.net>
36264 * gst/playback/gstplaybasebin.c: (group_commit),
36266 (setup_subtitle), (setup_source), (set_active_source):
36267 * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
36268 (gen_text_element), (gen_audio_element), (gen_vis_element),
36269 (remove_sinks), (add_sink), (setup_sinks): Implement subtitles.
36271 2006-01-29 19:13:39 +0000 Sébastien Moutte <sebastien@moutte.net>
36273 gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
36274 Original commit message from CVS:
36275 * gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
36276 * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
36277 use of gst_guint64_to_gdouble to be compliant with vs6
36278 * gst/playback/gstdecodebin.c: (try_to_link_1)
36279 * gst/videorate/videorate.c: (gst_video_rate_blank_data)
36280 use of G_GINT64_CONSTANT for int64 constants
36281 * win32/common/libgstinterfaces.def:
36282 export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
36284 update and add new project files
36286 2006-01-29 18:21:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36288 add a win32-update rule like in core, and copy over enumtypes files
36289 Original commit message from CVS:
36292 * win32/common/interfaces-enumtypes.c:
36293 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
36294 (gst_mixer_track_flags_get_type),
36295 (gst_tuner_channel_flags_get_type):
36296 * win32/common/interfaces-enumtypes.h:
36297 * win32/common/multichannel-enumtypes.c:
36298 (gst_audio_channel_position_get_type):
36299 * win32/common/multichannel-enumtypes.h:
36300 add a win32-update rule like in core, and copy over enumtypes files
36302 2006-01-29 18:07:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36305 generate win32/common/config.h
36306 Original commit message from CVS:
36307 generate win32/common/config.h
36309 2006-01-29 18:05:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36311 win32/: add config files just like in core
36312 Original commit message from CVS:
36314 * win32/common/config.h:
36315 * win32/common/config.h.in:
36316 add config files just like in core
36318 2006-01-28 18:22:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36320 ext/alsa/: Update all error messages. All of them should either use the default translated message, or actually prov...
36321 Original commit message from CVS:
36322 * ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
36323 (set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
36324 (gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
36325 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
36326 (set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
36327 (gst_alsasrc_unprepare), (gst_alsasrc_read):
36328 Update all error messages. All of them should either use
36329 the default translated message, or actually provide a
36330 translatable string.
36331 Make the string for channel count problems meaningful.
36333 2006-01-28 18:19:18 +0000 Tim-Philipp Müller <tim@centricular.net>
36335 gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
36336 Original commit message from CVS:
36337 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
36338 Make gcc-4.1 happy (part of #327357).
36340 2006-01-28 16:35:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36342 sys/v4l/v4l_calls.c: check for and throw RESOURCE_BUSY
36343 Original commit message from CVS:
36344 * sys/v4l/v4l_calls.c: (gst_v4l_open):
36345 check for and throw RESOURCE_BUSY
36347 2006-01-28 02:13:14 +0000 David Schleef <ds@schleef.org>
36349 gst/videoscale/vs_scanline.c: Oops, *that's* why I never checked in this change -- it requires liboil features not in...
36350 Original commit message from CVS:
36351 * gst/videoscale/vs_scanline.c: Oops, *that's* why I never
36352 checked in this change -- it requires liboil features not
36353 in 0.3.6. Revert parts.
36355 2006-01-27 23:40:19 +0000 David Schleef <ds@schleef.org>
36357 update liboil requirement to 0.3.6
36358 Original commit message from CVS:
36360 * configure.ac: update liboil requirement to 0.3.6
36361 * gst/videoscale/Makefile.am:
36362 * gst/videoscale/vs_scanline.c: liboilify
36364 2006-01-27 17:00:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36366 ext/libvisual/visual.c: When pad_alloc returns a GstFlowReturn other than GST_FLOW_OK, make sure it is passed upstream.
36367 Original commit message from CVS:
36368 * ext/libvisual/visual.c: (get_buffer):
36369 When pad_alloc returns a GstFlowReturn other
36370 than GST_FLOW_OK, make sure it is passed upstream.
36372 2006-01-27 01:36:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36374 ext/alsa/gstalsasink.c: Free the device name string.
36375 Original commit message from CVS:
36376 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
36377 (gst_alsasink_class_init):
36378 Free the device name string.
36379 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
36380 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
36381 (gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
36382 Don't remove a pad from the collectpads structure until it
36383 is released - it's a request pad, and may receive data again
36384 if the element gets moved back to PLAYING state.
36385 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
36386 Ensure we turn on double buffering on the Xv port, and
36387 set the colour key to something dark and mysterious that
36390 2006-01-27 01:06:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36392 ext/: - a library should not call setlocale. see Libraries node in gettext manual
36393 Original commit message from CVS:
36394 * ext/alsa/gstalsaplugin.c: (plugin_init):
36395 * ext/cdparanoia/gstcdparanoiasrc.c:
36396 (gst_cd_paranoia_src_base_init), (plugin_init):
36397 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
36398 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
36399 - a library should not call setlocale. see Libraries node in
36401 - make sure all plugins that use translation do bindtextdomain
36402 to point to the localedir
36403 * gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
36404 (setup_sinks), (plugin_init):
36405 all this, and check for NULL when creating sinks
36407 2006-01-26 23:21:31 +0000 Julien Moutte <julien@moutte.net>
36409 gst/subparse/gstsubparse.c: Make typefinding of subtitles work again.
36410 Original commit message from CVS:
36411 2006-01-27 Julien MOUTTE <julien@moutte.net>
36412 * gst/subparse/gstsubparse.c: (gst_subparse_type_find),
36413 (plugin_init): Make typefinding of subtitles work again.
36415 2006-01-26 20:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
36417 gst/typefind/gsttypefindfunctions.c: Backport a bunch of typefinding fixes from the 0.8 branch.
36418 Original commit message from CVS:
36419 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
36420 (mp3_type_frame_length_from_header), (mp3_type_find),
36421 (wavpack_type_find), (m4a_type_find), (ircam_type_find),
36423 Backport a bunch of typefinding fixes from the 0.8 branch.
36424 Also, improve wavpack typefinding: if we can't peek the
36425 entire wavpack block, try to parse the bits we can get and
36426 see if we find what we're looking for in those.
36428 2006-01-26 19:17:38 +0000 Julien Moutte <julien@moutte.net>
36430 sys/: Handle some more cases of pixel aspect ratio.
36431 Original commit message from CVS:
36432 2006-01-26 Julien MOUTTE <julien@moutte.net>
36433 * sys/ximage/ximagesink.c:
36434 (gst_ximagesink_calculate_pixel_aspect_ratio):
36435 * sys/xvimage/xvimagesink.c:
36436 (gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
36437 more cases of pixel aspect ratio.
36439 2006-01-26 13:09:24 +0000 Edward Hervey <bilboed@bilboed.com>
36441 gst/playback/gstdecodebin.c: Also consider the flush-start and tag events as unblockers for the pad probes.
36442 Original commit message from CVS:
36443 * gst/playback/gstdecodebin.c: (pad_probe):
36444 Also consider the flush-start and tag events as unblockers
36445 for the pad probes.
36447 2006-01-26 12:32:58 +0000 Julien Moutte <julien@moutte.net>
36449 gst/playback/gstplaybin.c: On the fly visualisation switch, works disabling, enabling as well but it won't be able to...
36450 Original commit message from CVS:
36451 2006-01-26 Julien MOUTTE <julien@moutte.net>
36452 * gst/playback/gstplaybin.c: (gst_play_bin_init),
36453 (gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
36454 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
36455 On the fly visualisation switch, works disabling, enabling as
36456 well but it won't be able to enable vis in a playbin that was
36457 created with no visualisation.
36459 2006-01-25 10:50:32 +0000 Wim Taymans <wim.taymans@gmail.com>
36461 gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
36462 Original commit message from CVS:
36463 * gst-libs/gst/audio/gstbaseaudiosink.c:
36464 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
36465 Undo previous commit, it breaks resume after pause.
36467 2006-01-25 09:27:01 +0000 Wim Taymans <wim.taymans@gmail.com>
36469 gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
36470 Original commit message from CVS:
36471 * gst-libs/gst/audio/gstbaseaudiosink.c:
36472 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
36473 (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
36475 Post error when caps cannot be parsed.
36476 Resync on discontinuity in the stream.
36477 Clip samples to segment boundaries.
36478 return WRONG_STATE sooner when we are flushing.
36479 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
36480 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
36481 Make audiosrc operate in TIME.
36482 Set TIMESTAMP and DURATION on buffers.
36484 2006-01-24 21:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
36486 tests/examples/seek/seek.c: Output tag messages as well.
36487 Original commit message from CVS:
36488 * tests/examples/seek/seek.c: (main):
36489 Output tag messages as well.
36491 2006-01-23 15:05:24 +0000 Edward Hervey <bilboed@bilboed.com>
36493 gst/playback/gstdecodebin.c: Replace GstPadBlockCallback with pad probes that detect first buffer AND eos before remo...
36494 Original commit message from CVS:
36495 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
36496 (free_pad_probes), (remove_fakesink), (pad_probe),
36497 (close_pad_link), (gst_decode_bin_change_state):
36498 Replace GstPadBlockCallback with pad probes that detect
36499 first buffer AND eos before removing fakesink.
36500 Fixes hang with demuxers doing EOS while pre-rolling.
36503 2006-01-23 10:10:36 +0000 Jens Granseuer <jensgr@gmx.net>
36505 GCC 2.95 fixes (#328263).
36506 Original commit message from CVS:
36507 2006-01-23 Andy Wingo <wingo@pobox.com>
36508 * ext/alsa/gstalsasink.c:
36509 * gst-libs/gst/rtp/gstbasertpdepayload.c:
36510 (gst_base_rtp_depayload_setcaps),
36511 (gst_base_rtp_depayload_add_to_queue),
36512 (gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
36513 Patch by: Jens Granseuer <jensgr at gmx dot net>
36515 2006-01-22 17:24:02 +0000 Julien Moutte <julien@moutte.net>
36517 sys/: Playbin keeps some ref to some frames. We might get a frame destroyed after changing state to
36518 Original commit message from CVS:
36519 2006-01-22 Julien MOUTTE <julien@moutte.net>
36520 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
36521 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
36522 (gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
36523 frames. We might get a frame destroyed after changing state to
36524 NULL, adding a safety check on xcontext.
36526 2006-01-22 14:50:53 +0000 Tim-Philipp Müller <tim@centricular.net>
36528 gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
36529 Original commit message from CVS:
36530 * gst-libs/gst/interfaces/xoverlay.c:
36531 Fix prepare-xwindow-id code example in the docs - we need to
36532 ignore all messages that aren't element messages as well.
36534 2006-01-21 22:40:03 +0000 Julien Moutte <julien@moutte.net>
36536 sys/xvimage/xvimagesink.c: I think one day i'll completely undestand how caps negotiation is supposed to work. This r...
36537 Original commit message from CVS:
36538 2006-01-21 Julien MOUTTE <julien@moutte.net>
36539 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
36540 I think one day i'll completely undestand how caps negotiation
36541 is supposed to work. This refactoring handles buffer_alloc
36542 called with caps we can't handle. We definitely don't want a
36543 set_caps with those caps, so we define and allocate a buffer
36544 we would like to receive.
36546 2006-01-20 19:10:26 +0000 Christian Schaller <uraeus@gnome.org>
36550 up automake requirement to 1.7
36551 Original commit message from CVS:
36552 up automake requirement to 1.7
36554 2006-01-19 10:59:51 +0000 Tim-Philipp Müller <tim@centricular.net>
36556 gst/playback/gstplaybasebin.c: Free iterator when done.
36557 Original commit message from CVS:
36558 * gst/playback/gstplaybasebin.c: (setup_source):
36559 Free iterator when done.
36561 2006-01-17 11:43:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36563 gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
36564 Original commit message from CVS:
36565 * gst-libs/gst/audio/gstbaseaudiosink.c:
36566 (gst_base_audio_sink_render):
36567 Fix playback of non-synchronised streams by assuming a rate
36568 of 1.0 instead of a random one.
36569 Makes this work again:
36570 gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
36571 endianness=(int)4321, signed=(boolean)true, width=(int)16,
36572 depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
36573 audioresample ! alsasink
36575 2006-01-16 21:01:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36579 Original commit message from CVS:
36582 === release 0.10.2 ===
36584 2006-01-16 20:59:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36590 * docs/plugins/gst-plugins-base-plugins.args:
36591 * docs/plugins/inspect/plugin-adder.xml:
36592 * docs/plugins/inspect/plugin-alsa.xml:
36593 * docs/plugins/inspect/plugin-audioconvert.xml:
36594 * docs/plugins/inspect/plugin-audiorate.xml:
36595 * docs/plugins/inspect/plugin-audioresample.xml:
36596 * docs/plugins/inspect/plugin-audiotestsrc.xml:
36597 * docs/plugins/inspect/plugin-cdparanoia.xml:
36598 * docs/plugins/inspect/plugin-decodebin.xml:
36599 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
36600 * docs/plugins/inspect/plugin-gnomevfs.xml:
36601 * docs/plugins/inspect/plugin-libvisual.xml:
36602 * docs/plugins/inspect/plugin-ogg.xml:
36603 * docs/plugins/inspect/plugin-pango.xml:
36604 * docs/plugins/inspect/plugin-playbin.xml:
36605 * docs/plugins/inspect/plugin-subparse.xml:
36606 * docs/plugins/inspect/plugin-tcp.xml:
36607 * docs/plugins/inspect/plugin-theora.xml:
36608 * docs/plugins/inspect/plugin-typefindfunctions.xml:
36609 * docs/plugins/inspect/plugin-video4linux.xml:
36610 * docs/plugins/inspect/plugin-videorate.xml:
36611 * docs/plugins/inspect/plugin-videoscale.xml:
36612 * docs/plugins/inspect/plugin-videotestsrc.xml:
36613 * docs/plugins/inspect/plugin-volume.xml:
36614 * docs/plugins/inspect/plugin-vorbis.xml:
36615 * docs/plugins/inspect/plugin-ximagesink.xml:
36616 * docs/plugins/inspect/plugin-xvimagesink.xml:
36618 Original commit message from CVS:
36621 2006-01-16 16:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36623 gst/playback/: Comment out broken code that connects to the state-changed signal.
36624 Original commit message from CVS:
36625 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
36626 * gst/playback/gststreamselector.c:
36627 (gst_stream_selector_set_property):
36628 Comment out broken code that connects to the state-changed signal.
36629 At this point, changing current stream selection is broken, but
36630 stuff like gst-launch playbin current-audio=1 works and filters
36631 to the chosen stream.
36633 2006-01-16 15:31:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36635 ext/vorbis/vorbisdec.c: Fix #327216 (null dereference in vorbisdec)
36636 Original commit message from CVS:
36637 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
36638 Fix #327216 (null dereference in vorbisdec)
36640 2006-01-16 15:19:55 +0000 Tim-Philipp Müller <tim@centricular.net>
36642 ext/theora/theoradec.c: Post taglist actually on bus instead of just freeing it (fixes #327114 and totem bug #327080).
36643 Original commit message from CVS:
36644 * ext/theora/theoradec.c: (theora_handle_comment_packet):
36645 Post taglist actually on bus instead of just freeing it
36646 (fixes #327114 and totem bug #327080).
36647 * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
36648 Use gst_element_found_tags_for_pad(), so that the tags
36649 are sent downstream as an event as well.
36651 2006-01-15 10:06:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36653 sys/: move all regularly occurring messages to GST_LOG level add some more object logs
36654 Original commit message from CVS:
36655 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
36656 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
36657 (gst_ximagesink_buffer_alloc):
36658 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
36659 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
36660 (gst_xvimagesink_buffer_alloc):
36661 move all regularly occurring messages to GST_LOG level
36662 add some more object logs
36664 2006-01-14 22:59:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36682 Original commit message from CVS:
36685 2006-01-14 20:46:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36687 ext/ogg/gstoggmux.c: fix a silly segfault
36688 Original commit message from CVS:
36689 2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
36690 * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
36691 fix a silly segfault
36693 2006-01-14 12:52:22 +0000 Tim-Philipp Müller <tim@centricular.net>
36695 Add docs for mixerutils stuff.
36696 Original commit message from CVS:
36697 * docs/libs/gst-plugins-base-libs-docs.sgml:
36698 * docs/libs/gst-plugins-base-libs-sections.txt:
36699 * gst-libs/gst/audio/mixerutils.c:
36700 * gst-libs/gst/audio/mixerutils.h:
36701 Add docs for mixerutils stuff.
36703 2006-01-13 17:17:07 +0000 Tim-Philipp Müller <tim@centricular.net>
36705 gst/playback/gstplaybasebin.c: Fix playback for sources that emit raw audio or raw video streams (e.g.: cd audio sour...
36706 Original commit message from CVS:
36707 * gst/playback/gstplaybasebin.c: (setup_source):
36708 Fix playback for sources that emit raw audio or
36709 raw video streams (e.g.: cd audio sources) (#325984).
36711 2006-01-13 16:45:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36713 gst-libs/gst/audio/mixerutils.c: actually save the element we create
36714 Original commit message from CVS:
36715 * gst-libs/gst/audio/mixerutils.c:
36716 (gst_audio_mixer_filter_do_filter):
36717 actually save the element we create
36719 2006-01-13 16:17:50 +0000 Christian Schaller <uraeus@gnome.org>
36721 * gst-plugins-base.spec.in:
36722 remove version suffix
36723 Original commit message from CVS:
36724 remove version suffix
36726 2006-01-12 14:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
36728 gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
36729 Original commit message from CVS:
36730 * gst-libs/gst/cdda/gstcddabasesrc.c:
36731 (gst_cdda_base_src_handle_track_seek):
36732 No need to post a tag message on the bus when seeking
36733 within the same track, only post it when the current
36736 2006-01-11 18:30:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36738 gst/playback/: Reenable stream selection. These mechanisms need a complete overhaul in the face of 0.8->0.10 changes ...
36739 Original commit message from CVS:
36740 * gst/playback/gstplaybasebin.c: (group_destroy),
36741 (probe_triggered), (new_decoded_pad), (mute_group_type),
36742 (set_active_source):
36743 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
36744 * gst/playback/gststreamselector.c:
36745 (gst_stream_selector_base_init),
36746 (gst_stream_selector_set_property),
36747 (gst_stream_selector_request_new_pad):
36748 Reenable stream selection. These mechanisms need a complete overhaul
36749 in the face of 0.8->0.10 changes though.
36751 2006-01-11 18:03:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36753 ext/ogg/gstoggdemux.c: Change the pad template to src_%d to match the pads that are created from it. decodebin needs ...
36754 Original commit message from CVS:
36755 * ext/ogg/gstoggdemux.c:
36756 Change the pad template to src_%d to match the pads that
36757 are created from it. decodebin needs this information in order
36758 to decide that oggdemux is capable of producing multiple pads
36759 (and hence needs queues inserted).
36760 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
36761 (gst_ogg_mux_collected):
36762 Make debug output more useful by using GST_PTR_FORMAT.
36764 2006-01-11 17:38:35 +0000 Christian Schaller <uraeus@gnome.org>
36766 * gst-plugins-base.spec.in:
36767 update spec.in file
36768 Original commit message from CVS:
36769 update spec.in file
36771 2006-01-11 15:11:20 +0000 Tim-Philipp Müller <tim@centricular.net>
36773 gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
36774 Original commit message from CVS:
36775 Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
36776 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
36777 Set depth and width for alaw/mulaw (fixes #326601).
36779 2006-01-10 23:58:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36781 tests/icles/Makefile.am: don't build the tests if we don't have the libs
36782 Original commit message from CVS:
36783 * tests/icles/Makefile.am:
36784 don't build the tests if we don't have the libs
36786 2006-01-10 18:06:56 +0000 Tim-Philipp Müller <tim@centricular.net>
36788 ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
36789 Original commit message from CVS:
36790 * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
36791 (gst_cd_paranoia_paranoia_callback):
36792 Don't try to free NULL pointers.
36794 2006-01-10 15:47:48 +0000 Edward Hervey <bilboed@bilboed.com>
36796 gst/audiorate/gstaudiorate.c: Add debugging category.
36797 Original commit message from CVS:
36798 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
36799 (gst_audio_rate_change_state), (plugin_init):
36800 Add debugging category.
36802 Add case for incoming buffers without valid offset/offset_end.
36804 2006-01-10 12:25:59 +0000 Michael Smith <msmith@xiph.org>
36806 gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
36807 Original commit message from CVS:
36808 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
36809 Don't leak GCond in audio sources.
36811 2006-01-10 11:49:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36813 gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
36814 Original commit message from CVS:
36815 * gst/playback/gstplaybin.c: (gen_audio_element):
36816 Don't leak an autoaudiosink/alsasink when we generate
36817 a new audio element. (old code, I guess)
36819 2006-01-10 11:04:21 +0000 Michael Smith <msmith@xiph.org>
36821 gst/audiorate/gstaudiorate.c: Support float audio in audiorate.
36822 Original commit message from CVS:
36823 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
36824 Support float audio in audiorate.
36825 Use width rather than depth for selecting sample width.
36827 2006-01-10 10:06:53 +0000 Tim-Philipp Müller <tim@centricular.net>
36829 gst/videotestsrc/videotestsrc.h: Use GLib types here (that way we don't have to include the generated _stdint.h heade...
36830 Original commit message from CVS:
36831 * gst/videotestsrc/videotestsrc.h:
36832 Use GLib types here (that way we don't have to include the
36833 generated _stdint.h header, which makes life easier for win32
36834 folks that don't use autotools for the build) (#325990, patch
36835 by: Sergey Scobich).
36837 2006-01-10 09:38:44 +0000 Tim-Philipp Müller <tim@centricular.net>
36839 gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
36840 Original commit message from CVS:
36841 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
36842 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
36843 (gst_ring_buffer_pause), (wait_segment):
36844 * gst-libs/gst/audio/gstringbuffer.h:
36845 Name (private) union, makes Forte compiler happy (this time
36846 for real) (#324900).
36848 2006-01-09 10:52:33 +0000 Tim-Philipp Müller <tim@centricular.net>
36850 gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
36851 Original commit message from CVS:
36852 * gst-libs/gst/audio/Makefile.am:
36853 Link against libgstinterfaces, needed for mixer
36854 and property probe stuff.
36856 2006-01-09 10:46:52 +0000 Edward Hervey <bilboed@bilboed.com>
36858 gst-libs/gst/Makefile.am:
36859 Original commit message from CVS:
36860 * gst-libs/gst/Makefile.am:
36862 2006-01-09 09:38:34 +0000 Tim-Philipp Müller <tim@centricular.net>
36864 gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
36865 Original commit message from CVS:
36866 * gst-libs/gst/audio/Makefile.am:
36867 * gst-libs/gst/audio/mixerutils.c:
36868 (gst_audio_mixer_filter_do_filter),
36869 (gst_audio_mixer_filter_check_element),
36870 (gst_audio_mixer_filter_probe_feature),
36871 (element_factory_rank_compare_func),
36872 (gst_audio_default_registry_mixer_filter):
36873 * gst-libs/gst/audio/mixerutils.h:
36874 Add gst_audio_default_registry_mixer_filter() utility
36877 2006-01-03 17:33:16 +0000 Michael Smith <msmith@xiph.org>
36879 gst/audioresample/resample.h: As before, but for o_buf
36880 Original commit message from CVS:
36881 * gst/audioresample/resample.h:
36882 As before, but for o_buf
36884 2006-01-03 17:27:13 +0000 Michael Smith <msmith@xiph.org>
36886 gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
36887 Original commit message from CVS:
36888 * gst/audioresample/resample.h:
36889 Declare struct _ResampleState.buffer as unsigned char *, not void *,
36890 since we do arithmetic on it.
36892 2006-01-02 23:37:38 +0000 Tim-Philipp Müller <tim@centricular.net>
36894 gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
36895 Original commit message from CVS:
36896 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
36897 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
36898 (gst_ring_buffer_pause), (wait_segment):
36899 * gst-libs/gst/audio/gstringbuffer.h:
36900 Sun's Forte compiler doesn't seem to like anonymous structs,
36901 so use same setup as in GstBaseSrc (fixes #324900).
36903 2005-12-30 14:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
36905 move old example to tests/examples/volume/volune.c
36906 Original commit message from CVS:
36908 * gst/volume/Makefile.am:
36909 * gst/volume/demo.c:
36910 move old example to tests/examples/volume/volune.c
36911 * tests/examples/Makefile.am:
36912 * tests/examples/seek/seek.c: (main):
36913 change window-close event from "delete-event" to "destroy"
36914 * tests/examples/volume/Makefile.am:
36915 * tests/examples/volume/volume.c: (value_changed_callback),
36916 (setup_gui), (message_received), (eos_message_received), (main):
36917 fix event handling and bus usage
36919 2005-12-29 20:37:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
36921 gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...
36922 Original commit message from CVS:
36923 * gst/audiotestsrc/gstaudiotestsrc.c:
36924 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
36925 (gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
36926 (gst_audio_test_src_query), (gst_audio_test_src_create_sine),
36927 (gst_audio_test_src_create_square),
36928 (gst_audio_test_src_create_saw),
36929 (gst_audio_test_src_create_triangle),
36930 (gst_audio_test_src_create_silence),
36931 (gst_audio_test_src_create_white_noise),
36932 (gst_audio_test_src_create_pink_noise),
36933 (gst_audio_test_src_init_sine_table),
36934 (gst_audio_test_src_create_sine_table),
36935 (gst_audio_test_src_change_wave),
36936 (gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
36937 (gst_audio_test_src_create), (gst_audio_test_src_set_property):
36938 * gst/audiotestsrc/gstaudiotestsrc.h:
36939 update to basesrc changes, implement segmented seeking and eos handling,
36940 add a 'sine-tab' waveform for performance critical playback
36942 2005-12-29 16:17:55 +0000 Tim-Philipp Müller <tim@centricular.net>
36944 po/POTFILES.in: ... and this time the other modified file that I missed last time.
36945 Original commit message from CVS:
36947 ... and this time the other modified file that I missed last time.
36949 2005-12-29 14:31:49 +0000 Michael Smith <msmith@xiph.org>
36951 gst/playback/gstdecodebin.c: Fix non-C89 variable declaration not at the start of a block. Should help some compilers.
36952 Original commit message from CVS:
36953 * gst/playback/gstdecodebin.c: (new_pad):
36954 Fix non-C89 variable declaration not at the start of a block. Should
36955 help some compilers.
36957 2005-12-29 12:43:22 +0000 Tim-Philipp Müller <tim@centricular.net>
36959 tests/check/Makefile.am: And now fix 'make distcheck' (builddir != srcdir)
36960 Original commit message from CVS:
36961 * tests/check/Makefile.am:
36962 And now fix 'make distcheck' (builddir != srcdir)
36964 2005-12-29 12:22:24 +0000 Tim-Philipp Müller <tim@centricular.net>
36966 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia plugin again (there are still fixes required to pla...
36967 Original commit message from CVS:
36969 * ext/cdparanoia/Makefile.am:
36970 * ext/cdparanoia/gstcdparanoia.c:
36971 * ext/cdparanoia/gstcdparanoia.h:
36972 * ext/cdparanoia/gstcdparanoiasrc.c:
36973 (gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
36974 (gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
36975 (gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
36976 (gst_cd_paranoia_paranoia_callback),
36977 (gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
36978 (gst_cd_paranoia_src_set_property),
36979 (gst_cd_paranoia_src_get_property), (plugin_init):
36980 * ext/cdparanoia/gstcdparanoiasrc.h:
36981 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
36982 plugin again (there are still fixes required to playbin to make
36983 cdda:// uris work there).
36985 2005-12-29 12:13:57 +0000 Tim-Philipp Müller <tim@centricular.net>
36987 tests/check/Makefile.am: Fix test case compilation.
36988 Original commit message from CVS:
36989 * tests/check/Makefile.am:
36990 Fix test case compilation.
36992 2005-12-29 11:49:11 +0000 Tim-Philipp Müller <tim@centricular.net>
36994 gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
36995 Original commit message from CVS:
36996 * gst-libs/gst/cdda/gstcddabasesrc.c:
36997 (gst_cdda_base_src_update_duration),
36998 (gst_cdda_base_src_calculate_cddb_id):
36999 An integer is not a string. Fix access to uninitialised variable.
37000 * tests/check/Makefile.am:
37001 Add cddabasesrc unit test; also actually enable the vorbis test.
37002 * tests/check/generic/states.c:
37003 Blacklist new cd audio elements as well.
37004 * tests/check/libs/cddabasesrc.c:
37005 Unit test for GstCddaBaseSrc (discid calculation mostly).
37007 2005-12-28 18:19:25 +0000 Tim-Philipp Müller <tim@centricular.net>
37009 docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
37010 Original commit message from CVS:
37011 * docs/libs/Makefile.am:
37012 * docs/libs/gst-plugins-base-libs-docs.sgml:
37013 * docs/libs/gst-plugins-base-libs-sections.txt:
37014 * docs/libs/gst-plugins-base-libs.types:
37015 Add docs for libgstcdda/GstCddaBaseSrc.
37016 * gst-libs/gst/interfaces/mixertrack.h:
37017 Do one struct member per line with a semicolon at the end, that way
37018 even gtk-doc might parse it without complaining.
37020 2005-12-28 18:06:50 +0000 Tim-Philipp Müller <tim@centricular.net>
37022 Add new libgstcdda with GstCddaBaseSrc class.
37023 Original commit message from CVS:
37025 * gst-libs/gst/Makefile.am:
37026 * gst-libs/gst/cdda/Makefile.am:
37027 * gst-libs/gst/cdda/base64.c:
37028 * gst-libs/gst/cdda/base64.h:
37029 * gst-libs/gst/cdda/gstcddabasesrc.c:
37030 (gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
37031 (gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
37032 (gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
37033 (gst_cdda_base_src_get_property),
37034 (gst_cdda_base_src_get_track_from_sector),
37035 (gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
37036 (gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
37037 (gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
37038 (gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
37039 (gst_cdda_base_src_uri_get_protocols),
37040 (gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
37041 (gst_cdda_base_src_uri_handler_init),
37042 (gst_cdda_base_src_setup_interfaces),
37043 (gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
37044 (cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
37045 (lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
37046 (gst_cdda_base_src_add_tags),
37047 (gst_cdda_base_src_add_index_associations),
37048 (gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
37049 (gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
37050 (gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
37051 (gst_cdda_base_src_create):
37052 * gst-libs/gst/cdda/gstcddabasesrc.h:
37053 * gst-libs/gst/cdda/sha1.c:
37054 * gst-libs/gst/cdda/sha1.h:
37055 Add new libgstcdda with GstCddaBaseSrc class.
37057 2005-12-28 14:59:41 +0000 Tim-Philipp Müller <tim@centricular.net>
37059 ext/gnomevfs/gstgnomevfssink.h: Use GstBaseSinkClass as parent_class member for class struct, not
37060 Original commit message from CVS:
37061 * ext/gnomevfs/gstgnomevfssink.h:
37062 Use GstBaseSinkClass as parent_class member for class struct, not
37065 2005-12-27 22:29:43 +0000 Tim-Philipp Müller <tim@centricular.net>
37067 gst/videotestsrc/gstvideotestsrc.c: Add start method to reset running time and number of frames sent
37068 Original commit message from CVS:
37069 * gst/videotestsrc/gstvideotestsrc.c:
37070 (gst_video_test_src_class_init), (gst_video_test_src_start):
37071 Add start method to reset running time and number of frames sent
37072 when starting up (fixes #324696; patch by: Michal Benes).
37074 2005-12-27 21:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
37076 docs/plugins/: Add docs stuff for gnomevfssrc and gnomevfssink.
37077 Original commit message from CVS:
37078 * docs/plugins/Makefile.am:
37079 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
37080 * docs/plugins/gst-plugins-base-plugins-sections.txt:
37081 * docs/plugins/gst-plugins-base-plugins.args:
37082 * docs/plugins/gst-plugins-base-plugins.hierarchy:
37083 * docs/plugins/gst-plugins-base-plugins.signals:
37084 Add docs stuff for gnomevfssrc and gnomevfssink.
37085 * ext/gnomevfs/gstgnomevfssrc.c:
37086 Fix example pipeline in gtk-doc blurb.
37088 2005-12-27 21:42:23 +0000 Tim-Philipp Müller <tim@centricular.net>
37090 ext/gnomevfs/: Port gnomevfssink; add gtk-doc blurb.
37091 Original commit message from CVS:
37092 * ext/gnomevfs/Makefile.am:
37093 * ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
37094 (gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
37095 (gst_gnome_vfs_handle_get_type), (plugin_init):
37096 * ext/gnomevfs/gstgnomevfs.h:
37097 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
37098 (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
37099 (gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
37100 (gst_gnome_vfs_sink_set_property),
37101 (gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
37102 (gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
37103 (gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
37104 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
37105 (gst_gnome_vfs_sink_uri_get_type),
37106 (gst_gnome_vfs_sink_uri_get_protocols),
37107 (gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
37108 (gst_gnome_vfs_sink_uri_handler_init):
37109 * ext/gnomevfs/gstgnomevfssink.h:
37110 Port gnomevfssink; add gtk-doc blurb.
37111 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
37112 (gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
37113 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
37114 (gst_gnome_vfs_src_uri_get_type),
37115 (gst_gnome_vfs_src_uri_get_protocols),
37116 (gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
37117 (gst_gnome_vfs_src_uri_handler_init),
37118 (gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
37119 (gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
37120 (gst_gnome_vfs_src_send_additional_headers_callback),
37121 (gst_gnome_vfs_src_received_headers_callback),
37122 (gst_gnome_vfs_src_push_callbacks),
37123 (gst_gnome_vfs_src_pop_callbacks),
37124 (gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
37125 (gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
37126 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
37127 * ext/gnomevfs/gstgnomevfssrc.h:
37128 s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
37129 file; add gtk-doc blurb with example pipelines.
37131 2005-12-23 18:16:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37135 Original commit message from CVS:
37138 === release 0.10.1 ===
37140 2005-12-23 18:08:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37146 * docs/libs/tmpl/gstcolorbalance.sgml:
37147 * docs/plugins/gst-plugins-base-plugins.args:
37148 * docs/plugins/gst-plugins-base-plugins.signals:
37149 * docs/plugins/inspect/plugin-adder.xml:
37150 * docs/plugins/inspect/plugin-alsa.xml:
37151 * docs/plugins/inspect/plugin-audioconvert.xml:
37152 * docs/plugins/inspect/plugin-audiorate.xml:
37153 * docs/plugins/inspect/plugin-audioresample.xml:
37154 * docs/plugins/inspect/plugin-audiotestsrc.xml:
37155 * docs/plugins/inspect/plugin-decodebin.xml:
37156 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
37157 * docs/plugins/inspect/plugin-gnomevfs.xml:
37158 * docs/plugins/inspect/plugin-libvisual.xml:
37159 * docs/plugins/inspect/plugin-ogg.xml:
37160 * docs/plugins/inspect/plugin-pango.xml:
37161 * docs/plugins/inspect/plugin-playbin.xml:
37162 * docs/plugins/inspect/plugin-subparse.xml:
37163 * docs/plugins/inspect/plugin-tcp.xml:
37164 * docs/plugins/inspect/plugin-theora.xml:
37165 * docs/plugins/inspect/plugin-typefindfunctions.xml:
37166 * docs/plugins/inspect/plugin-video4linux.xml:
37167 * docs/plugins/inspect/plugin-videorate.xml:
37168 * docs/plugins/inspect/plugin-videoscale.xml:
37169 * docs/plugins/inspect/plugin-videotestsrc.xml:
37170 * docs/plugins/inspect/plugin-volume.xml:
37171 * docs/plugins/inspect/plugin-vorbis.xml:
37172 * docs/plugins/inspect/plugin-ximagesink.xml:
37173 * docs/plugins/inspect/plugin-xvimagesink.xml:
37175 Original commit message from CVS:
37178 2005-12-21 20:59:52 +0000 Edgard Lima <edgard.lima@indt.org.br>
37181 * gst/typefind/gsttypefindfunctions.c:
37182 iLBC30 and iLBC20 added to typefind.
37183 Original commit message from CVS:
37184 iLBC30 and iLBC20 added to typefind.
37186 2005-12-20 15:57:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37190 * docs/libs/tmpl/gstcolorbalance.sgml:
37206 Original commit message from CVS:
37209 2005-12-20 12:24:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37211 * gst-libs/gst/audio/gstbaseaudiosink.c:
37212 * gst-libs/gst/audio/gstbaseaudiosrc.c:
37213 stop making fun of older compilers
37214 Original commit message from CVS:
37215 stop making fun of older compilers
37217 2005-12-20 12:00:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37219 gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
37220 Original commit message from CVS:
37221 * gst-libs/gst/audio/gstbaseaudiosink.c:
37222 (gst_base_audio_sink_class_init):
37223 * gst-libs/gst/audio/gstbaseaudiosrc.c:
37224 (gst_base_audio_src_class_init):
37225 update strings, values are in microseconds
37226 change the default sink buffer time to something that is smaller
37227 (to help software volume mixing have a slightly lower delay) but
37228 still be acceptable on Wim's laptop
37230 2005-12-20 10:13:05 +0000 Edward Hervey <bilboed@bilboed.com>
37232 gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
37233 Original commit message from CVS:
37234 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
37235 Made a quack, forgot to add DUCK to the riff video template.
37237 2005-12-19 15:00:38 +0000 Edward Hervey <bilboed@bilboed.com>
37239 ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
37240 Original commit message from CVS:
37241 * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
37242 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
37243 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
37244 (gst_ogm_parse_chain):
37245 Make sure pads are initialized correctly.
37246 * gst-libs/gst/riff/riff-ids.h:
37247 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
37248 (gst_riff_create_video_template_caps):
37249 Add a whole bunch of FOURCC <=> MimeType.
37250 Extend the riff video pad template to support the newly added fourcc.
37252 2005-12-18 15:04:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
37254 ext/ogg/gstoggdemux.c: Extra debug output when activating/deactivating chains.
37255 Original commit message from CVS:
37256 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
37257 (gst_ogg_demux_activate_chain):
37258 Extra debug output when activating/deactivating chains.
37259 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
37260 (is_demuxer_element), (try_to_link_1), (remove_element_chain),
37262 Remove a queue from our list when it becomes unlinked.
37263 Don't add queues to elements in class 'Demux' if they
37264 can only produce one pad
37266 2005-12-18 10:54:33 +0000 Julien Moutte <julien@moutte.net>
37268 gst-libs/gst/video/gstvideosink.c: Add a debug category.
37269 Original commit message from CVS:
37270 2005-12-18 Julien MOUTTE <julien@moutte.net>
37271 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
37272 (gst_video_sink_get_type): Add a debug category.
37274 2005-12-18 00:56:07 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
37276 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
37277 Original commit message from CVS:
37278 2005-12-17 Philippe Khalaf <burger@speedy.org>
37279 * gst-libs/gst/rtp/gstbasertpdepayload.c:
37280 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
37281 Handle downstream newsegment by sending our own newsegment before the
37282 next buffer to be released. (#323900)
37284 2005-12-18 00:41:10 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
37286 gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
37287 Original commit message from CVS:
37288 2005-12-17 Philippe Khalaf <burger@speedy.org>
37289 * gst-libs/gst/rtp/gstbasertpdepayload.c:
37290 (gst_base_rtp_depayload_set_gst_timestamp):
37291 add queue delay to new segment as well (as opposed to just the first
37292 buffer). (bug #322347)
37294 2005-12-16 22:00:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
37296 ext/libvisual/visual.c: change some char* into char[]
37297 Original commit message from CVS:
37298 * ext/libvisual/visual.c: (make_valid_name):
37299 change some char* into char[]
37300 * gst/audiotestsrc/gstaudiotestsrc.c:
37301 (gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
37302 (gst_audio_test_src_create):
37303 * gst/audiotestsrc/gstaudiotestsrc.h:
37304 prepare to handle EOS and SEGMENT_DONE
37306 2005-12-16 12:32:37 +0000 Tim-Philipp Müller <tim@centricular.net>
37308 tests/check/generic/states.c: Blacklist cdparanoia element in state test.
37309 Original commit message from CVS:
37310 * tests/check/generic/states.c: (GST_START_TEST):
37311 Blacklist cdparanoia element in state test.
37313 2005-12-16 11:25:51 +0000 Benjamin Pineau <ben.pineau@gmail.com>
37315 gst/tcp/: Add <string.h> includes for memset and FD_ZERO (fixes #323878;
37316 Original commit message from CVS:
37317 * gst/tcp/gsttcp.c:
37318 * gst/tcp/gsttcpclientsink.c:
37319 * gst/tcp/gsttcpserversink.c:
37320 * gst/tcp/gsttcpserversrc.c:
37321 Add <string.h> includes for memset and FD_ZERO (fixes #323878;
37322 patch by: Benjamin Pineau).
37324 2005-12-15 14:43:38 +0000 Michael Smith <msmith@xiph.org>
37326 gst/videorate/gstvideorate.c: Fix timestamping for videorate when the first buffer it sees has a non-zero timestamp. ...
37327 Original commit message from CVS:
37328 * gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
37329 (gst_video_rate_chain):
37330 Fix timestamping for videorate when the first buffer it sees has a
37331 non-zero timestamp. Fix some misleading debug output.
37333 2005-12-15 10:30:14 +0000 Michael Smith <msmith@xiph.org>
37335 gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
37336 Original commit message from CVS:
37337 * gst/audioresample/gstaudioresample.c:
37338 Don't leak all input buffers to audioresample.
37340 2005-12-15 10:15:10 +0000 Tim-Philipp Müller <tim@centricular.net>
37342 ext/pango/gsttextoverlay.c: Don't operate on empty text buffers. Strip newlines and tabs only from the end of the tex...
37343 Original commit message from CVS:
37344 * ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
37345 Don't operate on empty text buffers. Strip newlines and
37346 tabs only from the end of the text, but leave them intact
37347 in the middle. Fix typo in gtk-doc description.
37349 2005-12-15 09:48:19 +0000 Tim-Philipp Müller <tim@centricular.net>
37351 gst/playback/: Make sure the video frame buffer we return to apps via the "frame" property always has caps set on it....
37352 Original commit message from CVS:
37353 * gst/playback/gstplaybasebin.c:
37354 * gst/playback/gstplaybin.c: (handoff):
37355 Make sure the video frame buffer we return to apps via the
37356 "frame" property always has caps set on it. Modify
37357 _gst_gvalue_set_object() macro to handle NULL objects
37360 2005-12-14 20:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
37362 gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
37363 Original commit message from CVS:
37364 * gst/audiotestsrc/gstaudiotestsrc.c:
37365 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
37366 (gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
37367 (gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
37368 (gst_audio_test_src_create):
37369 * gst/audiotestsrc/gstaudiotestsrc.h:
37370 Adjust to some recent api changes and add wtays new cool seeking
37373 2005-12-14 17:58:48 +0000 Tim-Philipp Müller <tim@centricular.net>
37375 ext/alsa/: Helper functions to add device probing via the GstPropertyProbe interface to a class.
37376 Original commit message from CVS:
37377 * ext/alsa/Makefile.am:
37378 * ext/alsa/gstalsadeviceprobe.c:
37379 * ext/alsa/gstalsadeviceprobe.h:
37380 Helper functions to add device probing via the GstPropertyProbe
37381 interface to a class.
37382 * ext/alsa/gstalsamixer.h:
37383 Comment out GST_ALSA_MIXER, it returns a struct that's not
37385 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
37386 Add some debug info.
37387 * ext/alsa/gstalsamixerelement.c:
37388 (gst_alsa_mixer_element_interface_supported),
37389 (gst_implements_interface_init),
37390 (gst_alsa_mixer_element_init_interfaces),
37391 (gst_alsa_mixer_element_class_init),
37392 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
37393 (gst_alsa_mixer_element_set_property),
37394 (gst_alsa_mixer_element_get_property),
37395 (gst_alsa_mixer_element_change_state):
37396 * ext/alsa/gstalsamixerelement.h:
37397 Add 'device' and 'device-name' properties. Add GstPropertyProbe
37398 for device handling (gnome-volume-control will need that).
37400 2005-12-12 20:31:24 +0000 Christian Schaller <uraeus@gnome.org>
37404 * gst-plugins-base.spec.in:
37405 updates to activate cdparanoia plugin
37406 Original commit message from CVS:
37407 updates to activate cdparanoia plugin
37409 2005-12-12 19:13:09 +0000 Michael Smith <msmith@xiph.org>
37411 ext/ogg/gstoggdemux.c: Use the correct function to free list of typefind factories.
37412 Original commit message from CVS:
37413 * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
37414 Use the correct function to free list of typefind factories.
37416 2005-12-12 15:09:55 +0000 Wim Taymans <wim.taymans@gmail.com>
37418 gst/videotestsrc/gstvideotestsrc.*: Implement seeking in videotestsrc.
37419 Original commit message from CVS:
37420 * gst/videotestsrc/gstvideotestsrc.c:
37421 (gst_video_test_src_class_init), (gst_video_test_src_init),
37422 (gst_video_test_src_parse_caps), (gst_video_test_src_query),
37423 (gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
37424 (gst_video_test_src_create):
37425 * gst/videotestsrc/gstvideotestsrc.h:
37426 Implement seeking in videotestsrc.
37429 2005-12-12 15:06:46 +0000 Wim Taymans <wim.taymans@gmail.com>
37431 ext/cdparanoia/: Partially ported cdparanoia now that basesrc can support a plugin like this..
37432 Original commit message from CVS:
37433 * ext/cdparanoia/Makefile.am:
37434 * ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
37435 (gst_paranoia_endian_get_type), (_do_init),
37436 (cdparanoia_class_init), (cdparanoia_init),
37437 (cdparanoia_set_property), (cdparanoia_get_property),
37438 (cdparanoia_do_seek), (cdparanoia_is_seekable),
37439 (cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
37440 (cdparanoia_convert), (cdparanoia_get_query_types),
37441 (cdparanoia_query), (cdparanoia_set_index),
37442 (cdparanoia_uri_set_uri):
37443 * ext/cdparanoia/gstcdparanoia.h:
37444 Partially ported cdparanoia now that basesrc can support a
37447 2005-12-12 13:03:50 +0000 Wim Taymans <wim.taymans@gmail.com>
37449 tests/examples/seek/scrubby.c: Set higher priority for bus events so they don't get reordered with gtk gui events.
37450 Original commit message from CVS:
37451 * tests/examples/seek/scrubby.c: (main):
37452 Set higher priority for bus events so they don't get reordered with
37454 * tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
37455 (flush_toggle_cb), (main):
37456 Added checkbox do disable flushing seeks.
37457 Disable scrubbing when doing non flushing seeks.
37459 2005-12-12 09:52:37 +0000 Tim-Philipp Müller <tim@centricular.net>
37461 gst/subparse/gstsubparse.c: Implement some sort of event handling that doesn't rely on g_return_if_fail; make sure we...
37462 Original commit message from CVS:
37463 * gst/subparse/gstsubparse.c: (gst_sub_parse_init),
37464 (gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
37465 (parser_state_init), (handle_buffer), (gst_sub_parse_chain),
37466 (gst_sub_parse_sink_event), (gst_sub_parse_change_state):
37467 Implement some sort of event handling that doesn't rely on
37468 g_return_if_fail; make sure we always push the last chunk of an
37469 .srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
37470 state change function; remove some old cruft. Seeking is still
37471 rather unlikely to work though.
37472 * tools/.cvsignore:
37475 2005-12-11 22:26:08 +0000 Julien Moutte <julien@moutte.net>
37477 sys/xvimage/xvimagesink.c: Fixed a leak of the current image reference when cleaning up.
37478 Original commit message from CVS:
37479 2005-12-11 Julien MOUTTE <julien@moutte.net>
37480 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
37481 Fixed a leak of the current image reference when cleaning up.
37482 Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
37484 2005-12-09 10:23:42 +0000 Michael Smith <msmith@xiph.org>
37486 tools/: Remove gst-launch-ext. It doesn't work, and is no longer particularly useful.
37487 Original commit message from CVS:
37488 * tools/Makefile.am:
37489 * tools/gst-launch-ext-m.m:
37490 Remove gst-launch-ext. It doesn't work, and is no longer
37491 particularly useful.
37493 2005-12-08 18:53:57 +0000 Luca Ognibene <luogni@tin.it>
37495 ext/ogg/gstogmparse.c: don't pass random values to ogmparse convert function.
37496 Original commit message from CVS:
37497 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
37498 don't pass random values to ogmparse convert function.
37499 Make seeking possible in the exile1.ogm file.
37501 2005-12-07 18:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
37503 gst/playback/: Work around refcount problem with g_value_set_object() that occur if the core has been compiled agains...
37504 Original commit message from CVS:
37505 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
37506 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
37507 Work around refcount problem with g_value_set_object() that occur
37508 if the core has been compiled against GLib-2.6 (g_value_set_object()
37509 will only g_object_ref() the element, but the caller will
37510 gst_object_unref() it and bad things will happen due to the way
37511 GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
37512 totem for people on FC4 using Thomas's 0.10 RPMs.
37514 2005-12-07 11:34:37 +0000 Edward Hervey <bilboed@bilboed.com>
37516 Time to welcome ogm to 0.10 :)
37517 Original commit message from CVS:
37518 Time to welcome ogm to 0.10 :)
37519 * ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
37520 (gst_ogg_pad_typefind):
37521 Oggdemux can now properly typefind elements with dynamic pads.
37522 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
37523 Properly set caps on src pad, and set caps on outgoing buffers.
37525 2005-12-06 19:42:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37528 * ext/alsa/gstalsamixer.h:
37529 * ext/alsa/gstalsamixerelement.h:
37530 * ext/alsa/gstalsamixeroptions.h:
37531 * ext/alsa/gstalsamixertrack.h:
37532 * ext/alsa/gstalsasink.c:
37533 * ext/alsa/gstalsasink.h:
37534 * ext/alsa/gstalsasrc.c:
37535 * ext/alsa/gstalsasrc.h:
37536 * ext/cdparanoia/gstcdparanoia.h:
37537 * ext/gnomevfs/gstgnomevfsuri.h:
37538 * ext/ogg/gstoggdemux.c:
37539 * ext/ogg/gstoggmux.c:
37540 * ext/pango/gsttextoverlay.h:
37541 * ext/theora/theoradec.c:
37542 * ext/theora/theoraenc.c:
37543 * ext/vorbis/vorbisdec.h:
37544 * ext/vorbis/vorbisenc.c:
37545 * ext/vorbis/vorbisenc.h:
37546 * ext/vorbis/vorbisparse.h:
37547 * gst-libs/gst/audio/gstaudioclock.h:
37548 * gst-libs/gst/audio/gstaudiosink.c:
37549 * gst-libs/gst/audio/gstaudiosink.h:
37550 * gst-libs/gst/audio/gstaudiosrc.c:
37551 * gst-libs/gst/audio/gstaudiosrc.h:
37552 * gst-libs/gst/audio/gstbaseaudiosink.c:
37553 * gst-libs/gst/audio/gstbaseaudiosink.h:
37554 * gst-libs/gst/audio/gstbaseaudiosrc.c:
37555 * gst-libs/gst/audio/gstbaseaudiosrc.h:
37556 * gst-libs/gst/audio/gstringbuffer.h:
37557 * gst-libs/gst/audio/multichannel.h:
37558 * gst-libs/gst/floatcast/floatcast.h:
37559 * gst-libs/gst/interfaces/colorbalance.c:
37560 * gst-libs/gst/interfaces/colorbalance.h:
37561 * gst-libs/gst/interfaces/colorbalancechannel.h:
37562 * gst-libs/gst/interfaces/mixer.h:
37563 * gst-libs/gst/interfaces/mixeroptions.h:
37564 * gst-libs/gst/interfaces/mixertrack.h:
37565 * gst-libs/gst/interfaces/navigation.h:
37566 * gst-libs/gst/interfaces/propertyprobe.h:
37567 * gst-libs/gst/interfaces/tuner.h:
37568 * gst-libs/gst/interfaces/tunerchannel.h:
37569 * gst-libs/gst/interfaces/tunernorm.h:
37570 * gst-libs/gst/interfaces/xoverlay.h:
37571 * gst-libs/gst/netbuffer/gstnetbuffer.h:
37572 * gst-libs/gst/riff/riff-ids.h:
37573 * gst-libs/gst/riff/riff-media.h:
37574 * gst-libs/gst/riff/riff-read.h:
37575 * gst-libs/gst/rtp/gstbasertpdepayload.h:
37576 * gst-libs/gst/rtp/gstbasertppayload.c:
37577 * gst-libs/gst/rtp/gstbasertppayload.h:
37578 * gst-libs/gst/rtp/gstrtpbuffer.c:
37579 * gst-libs/gst/rtp/gstrtpbuffer.h:
37580 * gst-libs/gst/tag/gsttageditingprivate.h:
37581 * gst-libs/gst/tag/gstvorbistag.c:
37582 * gst-libs/gst/tag/tag.h:
37583 * gst-libs/gst/video/video.h:
37584 * gst/adder/gstadder.c:
37585 * gst/adder/gstadder.h:
37586 * gst/audioconvert/audioconvert.c:
37587 * gst/audioconvert/audioconvert.h:
37588 * gst/audioconvert/gstaudioconvert.c:
37589 * gst/audioconvert/gstchannelmix.c:
37590 * gst/audioconvert/gstchannelmix.h:
37591 * gst/audiorate/gstaudiorate.c:
37592 * gst/audioresample/buffer.h:
37593 * gst/audioresample/functable.h:
37594 * gst/audioresample/gstaudioresample.c:
37595 * gst/audioresample/resample.h:
37596 * gst/ffmpegcolorspace/avcodec.h:
37597 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
37598 * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
37599 * gst/ffmpegcolorspace/imgconvert.c:
37600 * gst/ffmpegcolorspace/imgconvert_template.h:
37601 * gst/playback/gstdecodebin.c:
37602 * gst/playback/gstplaybasebin.h:
37603 * gst/playback/gstplaybin.c:
37604 * gst/playback/gststreaminfo.h:
37605 * gst/tcp/gstfdset.c:
37606 * gst/tcp/gstfdset.h:
37607 * gst/tcp/gstmultifdsink.c:
37608 * gst/tcp/gstmultifdsink.h:
37609 * gst/tcp/gsttcp.h:
37610 * gst/tcp/gsttcpclientsrc.c:
37611 * gst/tcp/gsttcpclientsrc.h:
37612 * gst/tcp/gsttcpplugin.h:
37613 * gst/tcp/gsttcpserversink.c:
37614 * gst/tcp/gsttcpserversrc.c:
37615 * gst/typefind/gsttypefindfunctions.c:
37616 * gst/videorate/gstvideorate.c:
37617 * gst/videotestsrc/gstvideotestsrc.h:
37618 * gst/videotestsrc/videotestsrc.h:
37619 * sys/v4l/gstv4lcolorbalance.h:
37620 * sys/v4l/gstv4ltuner.h:
37621 * sys/v4l/gstv4lxoverlay.h:
37622 * sys/v4l/v4l_calls.h:
37623 * sys/v4l/videodev_mjpeg.h:
37624 * tests/check/elements/audioconvert.c:
37625 * tests/check/elements/audioresample.c:
37626 * tests/check/elements/audiotestsrc.c:
37627 * tests/check/elements/videotestsrc.c:
37628 * tests/check/elements/volume.c:
37629 * tests/examples/seek/scrubby.c:
37630 * tests/examples/seek/seek.c:
37632 Original commit message from CVS:
37635 2005-12-06 19:28:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37637 * docs/libs/tmpl/gstaudio.sgml:
37638 * docs/libs/tmpl/gstcolorbalance.sgml:
37639 * docs/libs/tmpl/gstgconf.sgml:
37640 * docs/libs/tmpl/gstmixer.sgml:
37641 * docs/libs/tmpl/gstringbuffer.sgml:
37642 * docs/libs/tmpl/gsttuner.sgml:
37643 * docs/libs/tmpl/gstxoverlay.sgml:
37644 put back stability level
37645 Original commit message from CVS:
37646 put back stability level
37648 2005-12-05 18:11:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37652 Original commit message from CVS:
37655 === release 0.10.0 ===
37657 2005-12-05 18:02:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37663 * docs/libs/tmpl/gstcolorbalance.sgml:
37664 * docs/plugins/inspect/plugin-adder.xml:
37665 * docs/plugins/inspect/plugin-alsa.xml:
37666 * docs/plugins/inspect/plugin-audioconvert.xml:
37667 * docs/plugins/inspect/plugin-audiorate.xml:
37668 * docs/plugins/inspect/plugin-audioresample.xml:
37669 * docs/plugins/inspect/plugin-audiotestsrc.xml:
37670 * docs/plugins/inspect/plugin-decodebin.xml:
37671 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
37672 * docs/plugins/inspect/plugin-gnomevfs.xml:
37673 * docs/plugins/inspect/plugin-libvisual.xml:
37674 * docs/plugins/inspect/plugin-ogg.xml:
37675 * docs/plugins/inspect/plugin-pango.xml:
37676 * docs/plugins/inspect/plugin-playbin.xml:
37677 * docs/plugins/inspect/plugin-subparse.xml:
37678 * docs/plugins/inspect/plugin-tcp.xml:
37679 * docs/plugins/inspect/plugin-theora.xml:
37680 * docs/plugins/inspect/plugin-typefindfunctions.xml:
37681 * docs/plugins/inspect/plugin-video4linux.xml:
37682 * docs/plugins/inspect/plugin-videorate.xml:
37683 * docs/plugins/inspect/plugin-videoscale.xml:
37684 * docs/plugins/inspect/plugin-videotestsrc.xml:
37685 * docs/plugins/inspect/plugin-volume.xml:
37686 * docs/plugins/inspect/plugin-vorbis.xml:
37687 * docs/plugins/inspect/plugin-ximagesink.xml:
37688 * docs/plugins/inspect/plugin-xvimagesink.xml:
37690 Original commit message from CVS: