1 === release 0.10.28 ===
3 2010-03-08 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6 releasing 0.10.28, "Those Norwegians"
8 2010-03-08 21:57:03 +0100 Benjamin Otte <otte@redhat.com>
10 * ext/theora/gsttheoraenc.c:
11 theora: Fix SIGFPE when using 0/1 framerate
12 libtheora crashes with a 0 framerate, so let's forbid it.
13 https://bugzilla.redhat.com/show_bug.cgi?id=571289
15 2010-03-08 14:50:25 +0000 David Schleef <ds@schleef.org>
17 * ext/ogg/dirac_parse.c:
18 oggdemux: fix dirac header parsing
21 2010-03-08 14:46:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
23 * tests/examples/overlay/Makefile.am:
24 examples: make sure to dist qtgv-xoverlay.h header file
28 2010-03-08 12:11:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
30 * gst-libs/gst/rtp/gstbasertpdepayload.c:
31 basedepay: clarify some documentation
33 2010-03-08 11:25:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
35 * ext/alsa/gstalsasrc.c:
36 alsasrc: return right number of bytes that we wrote
38 2010-03-08 11:20:51 +0100 Dake Gu <gudake@gmail.com>
40 * gst-libs/gst/rtsp/gstrtspconnection.c:
41 rtspconnection: fix handling of x-server-ip-address
42 Fix handling of x-server-ip-address.
44 2010-03-02 11:25:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
46 * docs/design/draft-keyframe-force.txt:
47 docs: update keyframe force event
48 Add field to send all headers.
50 === release 0.10.27 ===
52 2010-03-06 00:09:29 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
58 * docs/plugins/gst-plugins-base-plugins.args:
59 * docs/plugins/inspect/plugin-adder.xml:
60 * docs/plugins/inspect/plugin-alsa.xml:
61 * docs/plugins/inspect/plugin-app.xml:
62 * docs/plugins/inspect/plugin-audioconvert.xml:
63 * docs/plugins/inspect/plugin-audiorate.xml:
64 * docs/plugins/inspect/plugin-audioresample.xml:
65 * docs/plugins/inspect/plugin-audiotestsrc.xml:
66 * docs/plugins/inspect/plugin-cdparanoia.xml:
67 * docs/plugins/inspect/plugin-decodebin.xml:
68 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
69 * docs/plugins/inspect/plugin-gdp.xml:
70 * docs/plugins/inspect/plugin-gio.xml:
71 * docs/plugins/inspect/plugin-gnomevfs.xml:
72 * docs/plugins/inspect/plugin-libvisual.xml:
73 * docs/plugins/inspect/plugin-ogg.xml:
74 * docs/plugins/inspect/plugin-pango.xml:
75 * docs/plugins/inspect/plugin-playback.xml:
76 * docs/plugins/inspect/plugin-subparse.xml:
77 * docs/plugins/inspect/plugin-tcp.xml:
78 * docs/plugins/inspect/plugin-theora.xml:
79 * docs/plugins/inspect/plugin-typefindfunctions.xml:
80 * docs/plugins/inspect/plugin-uridecodebin.xml:
81 * docs/plugins/inspect/plugin-video4linux.xml:
82 * docs/plugins/inspect/plugin-videorate.xml:
83 * docs/plugins/inspect/plugin-videoscale.xml:
84 * docs/plugins/inspect/plugin-videotestsrc.xml:
85 * docs/plugins/inspect/plugin-volume.xml:
86 * docs/plugins/inspect/plugin-vorbis.xml:
87 * docs/plugins/inspect/plugin-ximagesink.xml:
88 * docs/plugins/inspect/plugin-xvimagesink.xml:
89 * gst-plugins-base.doap:
90 * win32/common/_stdint.h:
91 * win32/common/config.h:
94 2010-03-06 00:08:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
130 2010-03-05 15:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
133 configure: first check for QtGui >= 4.6, only then for >= 4.0
134 If we first check for >= 4.0 the second check for >= 4.6 will just
135 short-cut since we are using the same prefix for the variables for
136 both checks, and they've already been set previously. So the examples
137 requiring >= 4.6 were built even in the >= 4.0 case.
139 2010-03-03 20:18:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
142 * win32/common/_stdint.h:
143 * win32/common/config.h:
144 0.10.26.4 pre-release
146 2010-03-03 20:17:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
149 po: update translations
151 2010-03-03 20:15:44 +0000 Josep Torra Valles <n770galaxy@gmail.com>
153 * gst/playback/gstplaysink.c:
154 playsink: avoid g_object_set() on NULL pointers
155 There may not be an overlay element if a text-sink is set.
158 2010-03-01 12:17:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
160 * ext/ogg/gstoggstream.c:
161 oggstream: mark skeleton streams correctly
162 Mark skeleton streams because we need to ignore them for calculating the
163 duration of the stream.
166 2010-02-24 01:10:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
170 * win32/common/_stdint.h:
171 * win32/common/config.h:
172 0.10.26.3 pre-release
174 2010-02-23 16:57:53 +0100 Götz Waschk <waschk@mandriva.org>
176 * tests/examples/overlay/Makefile.am:
177 examples: Dist header file for the Qt graphics view example
180 2010-02-23 11:41:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
182 * ext/ogg/gstoggdemux.c:
183 oggdemux: use the chain begin_time instead of our counter
184 We update the passed begintime argument to narrow our search region in the
185 binary search. This means that it does not always contain the chain begin time
186 after a couple of bisects. Use the real chain->begin_time to bring the
187 granuletime to the time in the chain instead.
190 2010-02-19 18:24:40 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
192 * tests/check/elements/videorate.c:
193 videorate: tests: New unit tests for upstream caps nego
194 Adds unit tests that check videorate's upstream caps
195 negotiation works properly (put passthrough caps
199 2010-01-27 15:07:47 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
201 * gst/videorate/gstvideorate.c:
202 videorate: Improve upstream negotiation
203 Put peer pad caps preferred framerates first, indicating
204 they are videorate's first choices, removing an unnecessary
208 2010-02-21 19:52:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
210 * gst/playback/gstplaybin2.c:
211 * gst/playback/gstplaysink.c:
212 * gst/playback/gstplaysink.h:
213 * gst/playback/gstsubtitleoverlay.c:
214 * gst/playback/gstsubtitleoverlay.h:
215 playbin2, playsink, subtitleoverlay: Set subtitle encoding properly
216 For this add subtitle encoding properties to playsink and subtitleoverlay
217 and update the values in the containing elements.
218 Also update the font description in textoverlay or the used renderer
219 element if it is changed during playback.
222 2010-02-22 13:01:19 +0200 Stefan Kost <ensonic@users.sf.net>
224 * tests/examples/overlay/gtk-xoverlay.c:
225 * tests/examples/overlay/qt-xoverlay.cpp:
226 * tests/examples/overlay/qtgv-xoverlay.cpp:
227 examples: also add sink detection and set title to qt examples
228 Also set a title in the qt examples like it is now done in the gtk example.
229 Fix the newly added find_video_sink in the gtk example and add similar function
232 2010-02-19 14:40:43 +0200 Stefan Kost <ensonic@users.sf.net>
234 * tests/examples/overlay/.gitignore:
235 gitignore: ignore files in new example directroy
237 2010-02-17 14:59:33 +0200 Stefan Kost <ensonic@users.sf.net>
239 * gst-libs/gst/video/Makefile.am:
240 make: fix copy and paste error in git rules (audio<->video)
242 2010-02-19 17:44:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
244 * gst/playback/gstplaysink.c:
245 playsink: Ghost the video sinkpad if a text sinkpad is available
246 Only don't ghost it if no visualizations are need and if
247 no text is needed and no textchain was created yet.
250 2010-02-19 00:22:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
253 * win32/common/_stdint.h:
254 * win32/common/config.h:
255 0.10.26.2 pre-release
257 2010-02-19 00:20:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
291 po: update translation files
293 2010-02-19 00:17:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
295 * tests/examples/overlay/.gitignore:
296 Ignore new overlay examples
298 2010-02-18 23:47:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
300 * tests/examples/overlay/gtk-xoverlay.c:
301 examples: don't hard-code xvimagesink for Gtk+ GstXOverlay example
302 Try to find a working videosink, don't hardcode xvimagesink. Also
303 add some borders to window and give it a title so that it's clear
304 that this is really a Gtk+ window and not a window created by the
307 2010-02-18 11:42:55 -0800 David Schleef <ds@schleef.org>
310 tcp(client/server)src: Fix handling of closed sockets
311 The peer closing the socket should cause an EOS, instead of
312 silently doing nothing. This changes the behavior to be
313 more like fdsrc. Fixes: #610386
315 2010-02-18 12:42:53 +0000 Patrick Radizi <patrick.radizi@axis.com>
317 * gst-libs/gst/rtsp/gstrtspconnection.c:
318 rtspconnection: make sure not to dereference NULL username or password
321 2010-02-17 21:22:54 -0800 David Schleef <ds@schleef.org>
323 * ext/theora/gsttheoradec.c:
324 theoradec: Fix chroma copying for 4:2:2
325 Fix mixup of height/width, causing only half the chroma lines to
326 be copied when outputting buffers. Fixes: #610329.
328 2010-02-16 15:43:26 +0200 Stefan Kost <ensonic@users.sf.net>
331 * gst-libs/gst/interfaces/xoverlay.c:
332 * tests/examples/Makefile.am:
333 * tests/examples/overlay/Makefile.am:
334 * tests/examples/overlay/gtk-xoverlay.c:
335 * tests/examples/overlay/qt-xoverlay.cpp:
336 * tests/examples/overlay/qtgv-xoverlay.cpp:
337 * tests/examples/overlay/qtgv-xoverlay.h:
338 examples: add video overlay examples for gtk, qt and qt graphics view
339 Add simple videotestsrc ! xvimagesink examples using gtk and qt. This patch also
340 adds all boilerplate to configure for using c++. The qt based examples are
341 optional like their gtk counterparts.
343 2010-02-16 17:20:01 +0200 Stefan Kost <ensonic@users.sf.net>
345 * docs/libs/compiling.sgml:
346 * docs/libs/gst-plugins-base-libs-docs.sgml:
347 * docs/libs/gst-plugins-base-libs-sections.txt:
348 docs: cleanup library docs
349 Correct name of included files. Remove files that are not used anymore. Add many
350 new api entries to their sections.
352 2010-02-15 11:11:04 +0200 Stefan Kost <ensonic@users.sf.net>
354 * tests/icles/test-colorkey.c:
355 test-colorkey: remove the XInitThreads()
356 We don't do this is any other example, this should be done for us in gdk it if
359 2010-02-16 10:09:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
361 * gst/playback/gsturidecodebin.c:
362 uridecodebin: use same message string for missing elements as in playbin
363 Use the same translated message string for missing core elements as
364 playbin uses, which is a bit nicer and also indicates that there is
365 something wrong with the user's GStreamer installation (which arguably
366 is the case if elements like typefind or queue2 are missing).
368 2010-02-08 13:54:57 +0200 Kaj-Michael Lang <milang@tal.org>
370 * gst/typefind/gsttypefindfunctions.c:
371 typefind: Handle stm module format
374 2010-02-15 12:10:10 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
376 * ext/vorbis/gstivorbisdec.c:
377 ivorbisdec: set rank to SECONDARY
379 2010-02-15 12:09:53 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
383 * ext/vorbis/Makefile.am:
384 * ext/vorbis/gstivorbisdec.c:
385 * ext/vorbis/gstvorbisdec.c:
386 * ext/vorbis/gstvorbisdec.h:
387 * ext/vorbis/gstvorbisdeclib.c:
388 * ext/vorbis/gstvorbisdeclib.h:
389 vorbisdec: also support ivorbis tremor decoder
390 ... which only needs a bit of refactoring and extracting to support
391 the minor difference in (i)vorbis interface.
394 2010-02-03 14:37:43 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
396 * ext/vorbis/gstvorbisdec.c:
397 * ext/vorbis/gstvorbisdec.h:
398 vorbisdec: reduce some hard-coding
399 ... such as assuming float all over, and base src caps on template caps.
401 2010-02-15 10:23:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
403 * tests/check/elements/playbin.c:
404 playbin: Fix the primary-decoder-missing test with USE_DECODEBIN2
406 2010-02-15 09:04:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
408 * ext/ogg/gstoggparse.c:
409 oggparse: Fix another format string compiler warning
411 2010-02-15 08:56:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
413 * ext/ogg/gstoggdemux.c:
414 oggdemux: Fix format string compiler warnings
416 2010-02-15 08:48:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
418 * gst/playback/gstplaybin2.c:
419 playbin2: Post a missing element message and an error message if no uridecodebin can be found
421 2010-02-15 08:46:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
423 * gst/playback/gstplaysink.c:
424 playsink: Post missing element messages if a core plugin is missing
425 And post a warning in cases where we can still continue to work
426 or an error when the missing element is fatal.
428 2010-02-15 08:28:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
430 * tests/check/elements/playbin2.c:
431 playbin2: Enable all unit tests
432 They're all working and valgrind clean now.
434 2010-02-15 08:26:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
436 * gst/playback/gstdecodebin2.c:
437 decodebin2: First post a missing-plugin message, then emit the unkown-type signal
438 This makes sure that there *always* is a missing plugin message in the bus
439 before any errors or warning messages.
441 2010-02-15 08:20:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
443 * gst/playback/gsturidecodebin.c:
444 uridecodebin: Missing decoder errors should be STREAM CODEC_NOT_FOUND
445 and not CORE MISSING_PLUGIN.
447 2010-02-15 08:18:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
449 * gst/playback/gstplaybin2.c:
450 playbin2: Free the subtitle URI
452 2010-02-15 08:06:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
454 * gst/playback/gsturidecodebin.c:
455 uridecodebin: Post missing plugin messages if a required element can't be created
456 Especially if no suitable URI source can be found.
458 2010-02-15 06:50:29 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
460 * tests/check/elements/.gitignore:
461 tests: Add decodebin2 test to .gitignore
463 2010-02-15 01:18:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
465 * gst/playback/gstdecodebin2.c:
466 decodebin2: Set ghostpad targets to NULL when freeing a decode chain
467 Otherwise the ghostpad will still be linked to the peer and there
468 will still be a reference kept, leading to nothing being unlinked
469 and destroyed until decodebin2 is finalized.
470 This fixes reuse of decodebin2 if a raw stream is connected to
473 2010-02-15 01:17:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
475 * tests/check/Makefile.am:
476 * tests/check/elements/decodebin2.c:
477 decodebin2: Add simple unit test, mainly a copy of the decodebin unit test
478 The only difference between the two unit tests right now is,
479 that the decodebin2 test resets the element to READY before trying
480 to reuse it instead of NULL. decodebin2 guarantees to be reusable
481 without going back to NULL.
483 2010-02-15 00:11:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
485 * ext/ogg/gstoggstream.c:
486 ogg: theora PAR of 0:N, N:0 or 0:0 is allowed and maps to 1:1
489 2010-02-14 23:16:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
492 Automatic update of common submodule
493 From 96dc793 to 44ecce7
495 2010-02-14 23:10:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
497 * tests/check/Makefile.am:
498 playbin2: Enable playbin2 unit test
499 It now contains a single working unit test and can be enabled.
500 The other more useful unit tests still need fixing.
502 2010-02-14 22:16:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
504 * tests/check/elements/playbin.c:
505 playbin: Fix indention in the unit test
507 2010-02-13 01:08:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
509 * gst/volume/gstvolume.c:
510 volume: Replace this variables by self
512 2010-02-12 19:43:13 +0100 Josep Torra Valles <n770galaxy@gmail.com>
514 * gst/playback/gstplaysink.c:
515 playsink: Reset the sink's state to NULL before unreffing it unless it's the same instance again
516 This makes sure that we don't destroy the last reference before the
517 element gets back to NULL state. Fixes assertion failures if a playbin2
518 instance is reused but different sinks are automatically chosen because
521 2010-02-12 18:00:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
523 * gst-libs/gst/app/gstappsrc.c:
524 appsrc: fix Since tag
526 2010-02-12 14:19:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
528 * gst-libs/gst/riff/riff-read.c:
529 riff: treat JUNQ chunks like JUNK chunks
531 2010-02-12 14:29:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
533 * gst-libs/gst/app/gstappsrc.c:
534 appsrc: Update basesrc segment duration and post duration messages from the streaming thread
536 2010-02-11 14:10:02 +0200 Stefan Kost <ensonic@users.sf.net>
538 * gst-libs/gst/tag/tags.c:
539 tags: improve docs about determining the encoding
541 2010-02-11 14:09:05 +0200 Stefan Kost <ensonic@users.sf.net>
543 * gst-libs/gst/tag/gstvorbistag.c:
544 comment: fix wrong header comment
546 2010-02-01 13:50:14 +0200 Stefan Kost <ensonic@users.sf.net>
548 * gst-libs/gst/riff/riff-ids.h:
549 riff: add a variant of the JUNK tag that several adobe products produce
550 JUNQ has same semantics as JUNK.
552 2010-02-01 19:01:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
554 * gst-libs/gst/app/gstappsrc.c:
555 appsrc: add min-percent property
556 Emit need-data when the amount of data in the internal queue drops below
560 2010-02-01 18:56:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
562 * gst-libs/gst/app/gstappsrc.c:
564 Avoid some typechecks.
565 Avoid dereferencing appsrc->priv all the time.
567 2010-02-01 18:55:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
569 * gst-libs/gst/app/gstappsink.c:
571 Avoid some typecasting.
572 Avoid dereferencing appsink->priv all the time.
574 2010-02-01 15:09:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
576 * gst/playback/gsturidecodebin.c:
577 uridecodebin: avoid some typecasts
579 2010-01-29 16:34:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
581 * gst-libs/gst/rtsp/gstrtspconnection.c:
582 rtsp: ignore \n and \r as the first line
583 Be more forgiving for bad servers and ignore \r and \n when we are looking for
584 the response/request line.
587 2010-02-10 16:05:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
589 * gst-libs/gst/rtsp/gstrtspconnection.c:
590 rtsp: fail gracefully on bad Content-Length headers
591 Be careful when allocating the amount of bytes specified in the Content-Length
592 because it can be an insanely huge value. Try to allocate the memory but fail
593 gracefully with a nice error when the allocation failed.
595 2010-02-10 10:12:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
597 * gst/ffmpegcolorspace/imgconvert.c:
598 * gst/ffmpegcolorspace/imgconvert_template.h:
599 ffmpegcolorspace: Add conversions from all ARGB formats to AYUV and back
601 2010-02-09 17:39:21 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
603 * gst-libs/gst/app/gstappsrc.c:
604 appsrc: Update segment duration and post a duration message if the duration changes
607 2010-02-11 10:56:17 +0100 Benjamin Otte <otte@redhat.com>
609 * tests/examples/seek/Makefile.am:
610 build: link to libm in examples that use it
611 This fixes build failure in Fedora 13.
613 2010-02-11 01:11:30 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
616 Update MAINTAINERS, add myself
618 2010-02-11 23:57:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
621 configure: back to development
622 Slushy freeze remains in effect.
624 === release 0.10.26 ===
626 2010-02-10 20:17:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
632 * docs/plugins/gst-plugins-base-plugins.args:
633 * docs/plugins/gst-plugins-base-plugins.hierarchy:
634 * docs/plugins/gst-plugins-base-plugins.interfaces:
635 * docs/plugins/gst-plugins-base-plugins.prerequisites:
636 * docs/plugins/inspect/plugin-adder.xml:
637 * docs/plugins/inspect/plugin-alsa.xml:
638 * docs/plugins/inspect/plugin-app.xml:
639 * docs/plugins/inspect/plugin-audioconvert.xml:
640 * docs/plugins/inspect/plugin-audiorate.xml:
641 * docs/plugins/inspect/plugin-audioresample.xml:
642 * docs/plugins/inspect/plugin-audiotestsrc.xml:
643 * docs/plugins/inspect/plugin-cdparanoia.xml:
644 * docs/plugins/inspect/plugin-decodebin.xml:
645 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
646 * docs/plugins/inspect/plugin-gdp.xml:
647 * docs/plugins/inspect/plugin-gio.xml:
648 * docs/plugins/inspect/plugin-gnomevfs.xml:
649 * docs/plugins/inspect/plugin-libvisual.xml:
650 * docs/plugins/inspect/plugin-ogg.xml:
651 * docs/plugins/inspect/plugin-pango.xml:
652 * docs/plugins/inspect/plugin-playback.xml:
653 * docs/plugins/inspect/plugin-subparse.xml:
654 * docs/plugins/inspect/plugin-tcp.xml:
655 * docs/plugins/inspect/plugin-theora.xml:
656 * docs/plugins/inspect/plugin-typefindfunctions.xml:
657 * docs/plugins/inspect/plugin-uridecodebin.xml:
658 * docs/plugins/inspect/plugin-video4linux.xml:
659 * docs/plugins/inspect/plugin-videorate.xml:
660 * docs/plugins/inspect/plugin-videoscale.xml:
661 * docs/plugins/inspect/plugin-videotestsrc.xml:
662 * docs/plugins/inspect/plugin-volume.xml:
663 * docs/plugins/inspect/plugin-vorbis.xml:
664 * docs/plugins/inspect/plugin-ximagesink.xml:
665 * docs/plugins/inspect/plugin-xvimagesink.xml:
666 * gst-plugins-base.doap:
667 * win32/common/_stdint.h:
668 * win32/common/config.h:
671 2010-02-10 20:16:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
707 2010-02-08 11:21:35 +0100 Benjamin M. Schwartz <bens@alum.mit.edu>
709 * ext/theora/gsttheoradec.c:
710 theoradec: PARs of 0:x, x:0 and 0:0 are all allowed and map to 1:1
713 2010-01-24 12:31:04 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
715 * ext/ogg/gstoggstream.c:
716 oggdemux: use the default granpos functions for kate streams
717 Set timestamps on kate packets. See bug #600929.
719 2010-02-05 01:18:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
722 * win32/common/_stdint.h:
723 * win32/common/config.h:
724 0.10.25.3 pre-release
726 2010-02-04 18:52:59 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
729 po: update translations
731 2010-02-04 18:32:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
733 * gst/playback/gstplaybin2.c:
734 Revert "playbin2: Only allow to set the URIs in states <= READY or from an about-to-finish signal handler"
735 This reverts commit 7335ce5d3e03c126a417a721571cb6f3af136ecf.
736 Support abusing the uri property to configure the next uri to play
737 outside of the about-to-finish handler for the time being after all.
738 We also shouldn't use thread private structures for this, since it
739 should be possible to block the thread that emitted about-to-finish
740 while the main thread sets the uri property. See #607226.
742 2010-02-02 10:18:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
744 * ext/ogg/gstoggdemux.c:
745 oggdemux: Don't leak allocated buffers
746 This can happen if the combined flow return is not OK although the
747 allocation succeeded or if the packet in question is a BOS and we're
748 not going to push headers.
751 2010-02-01 11:44:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
753 * gst/playback/gsturidecodebin.c:
754 uridecodebin: clean up decodebin properties
755 When reusing a decodebin2 element, clear the properties we might have changed,
756 to their default values or else we might end up with old configuration.
759 2010-01-29 13:56:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
761 * gst/playback/gstplaybin2.c:
762 playbin2: when no uri is set, post an error message
763 When no uri is set, don't just return STATE_CHANGE_FAILURE from the
764 state change function, but actually post an error message.
766 2010-01-30 15:18:13 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
769 Automatic update of common submodule
770 From 15d47a6 to 96dc793
772 2010-01-28 17:12:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
774 * gst/adder/gstadder.c:
775 adder: don't hold object lock when calling peer elements
776 Do not hold the object lock while we call methods on peer elements as this can
780 2010-01-27 01:12:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
783 0.10.25.2 pre-release
785 2010-01-27 01:07:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
787 * win32/common/_stdint.h:
788 * win32/common/config.h:
789 * win32/common/gstrtsp-enumtypes.c:
790 * win32/common/interfaces-enumtypes.c:
791 * win32/common/interfaces-enumtypes.h:
792 * win32/common/pbutils-enumtypes.c:
793 * win32/common/video-enumtypes.c:
794 win32: update generated files for non-autotools win32 builds
796 2010-01-27 00:56:00 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
830 po: update translation files
832 2010-01-27 00:41:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
834 * gst-libs/gst/audio/gstaudiosrc.c:
835 audiosrc: add gratuitious FIXME for use of generic G_TYPE_POINTER type
837 2010-01-26 16:47:40 +0100 Edward Hervey <bilboed@bilboed.com>
839 * gst/playback/gstdecodebin2.c:
840 decodebin2: Don't skip an element when getting the topology
843 2010-01-24 14:41:44 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
845 * ext/ogg/gstoggdemux.c:
846 oggdemux: sparse streams aren't timed by end time, and their duration isn't implicit
847 Fixes timestamps and durations on Kate subtitle streams.
848 See http://www.xiph.org/ogg/doc/ogg-multiplex.html section 'start-time and
849 end-time positioning' for some more details, and bug #600929.
851 2010-01-23 20:15:08 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
853 * ext/ogg/gstoggstream.c:
854 oggdemux: properly set up the media type for kate streams
857 2010-01-25 18:57:52 +0100 Julien Moutte <julien@fluendo.com>
859 * gst/playback/gstsubtitleoverlay.c:
860 subtitleoverlay: relax caps template on sink pads
861 Allow any caps on sink pad templates as we could do passthrough with non raw
864 2010-01-25 15:14:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
866 * ext/ogg/gstoggdemux.c:
867 * ext/ogg/gstoggstream.h:
868 oggdemux: use right type for the serialno
869 Use a consistent type for the serialno to avoid problems when comparing between
870 signed and unsigned variants.
873 2010-01-25 14:00:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
875 * ext/ogg/gstoggdemux.c:
876 oggdemux: don't push headers twice
877 Don't push the stream headers twice but only in the activation of a chain.
880 2010-01-25 13:18:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
882 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
884 2010-01-25 12:31:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
886 * ext/ogg/gstoggdemux.c:
887 * ext/ogg/gstoggdemux.h:
888 oggdemux: rename a variable
889 Rename the 'seekable' variable to 'pullmode'. We might be able to seek in push
892 2010-01-25 12:22:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
894 * gst/playback/gstinputselector.c:
895 Revert "inputselector: Protect g_object_notify() with the object's mutex"
896 This reverts commit a37426c41c80fd21e5017fea01a786c05bcd9661, it's
897 causing deadlocks with playbin2.
899 2010-01-24 20:55:26 +0100 Kipp Cannon <kcannon@ligo.caltech.edu>
901 * gst/playback/gstinputselector.c:
902 inputselector: Protect g_object_notify() with the object's mutex
903 This works around the thread unsafety of g_object_notify()
906 2010-01-24 20:46:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
908 * gst/typefind/gsttypefindfunctions.c:
909 typefindfunctions: Add typefinder for ISO MP4 files
912 2010-01-24 13:29:07 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
914 * ext/ogg/gstoggdemux.c:
915 oggdemux: fix crash when freeing headers
916 Use _ogg_packet_free() instead of gst_mini_object_unref in one more
917 place now that the header list contains ogg packets and not buffers.
918 file: Stephen_Fry-Happy_Birthday_GNU-nq_600px_425kbit.ogv
920 2010-01-24 08:57:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
922 * ext/ogg/gstoggdemux.c:
923 oggdemux: Strip trailing \0 for subtitle OGM streams
926 2010-01-23 22:09:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
928 * ext/ogg/gstoggdemux.c:
929 oggdemux: Correctly set DELTA_UNIT flag for OGM streams
931 2010-01-23 22:05:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
933 * ext/ogg/gstoggdemux.c:
934 oggdemux: Don't strip all 0-bytes from the end of OGM packets
935 This fixes broken packets pushed downstream by oggdemux for
936 MPEG4 streams for example.
938 2010-01-23 22:03:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
940 * ext/ogg/gstoggdemux.c:
941 oggdemux: Extract tags from OGM text streams and don't push them downstream
943 2010-01-23 14:46:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
945 * ext/ogg/gstoggdemux.c:
946 oggdemux: Store header/queued packets as ogg_packet and use normal peer chaining functions to pass them downstream
948 2010-01-23 15:25:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
950 * gst/typefind/gsttypefindfunctions.c:
951 typefinding: optimise AC-3 typefinder a bit
952 Make AC-3 typefinder use the DataScanCtx stuff so we don't have to
953 do gst_type_find_peek() in the inner loop all the time. Also return
954 when we've suggested AC3 caps, instead of continuing with the loop.
956 2010-01-23 14:31:15 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
958 * gst/typefind/gsttypefindfunctions.c:
959 Revert "typefind: Reduce number of calls to gst_type_find_peek."
960 This reverts commit c661bfaa991c58f1fbd9fbc0dae90b8b2c27f92b.
961 This breaks AC-3 typefinding for all cases where the first frame
964 2010-01-23 15:35:05 +0100 Edward Hervey <bilboed@bilboed.com>
966 * gst-libs/gst/pbutils/descriptions.c:
967 pbutils: Add description for Zip Block Motion Video
969 2010-01-23 15:34:54 +0100 Edward Hervey <bilboed@bilboed.com>
971 * gst-libs/gst/riff/riff-media.c:
972 riff: Add mapping for Zip Block Motion Video
974 2010-01-23 15:26:37 +0100 Edward Hervey <bilboed@bilboed.com>
976 * gst-libs/gst/riff/riff-media.c:
977 riff: YUNV is a fourcc which is also used for YUY2 raw video
979 2010-01-23 15:13:45 +0100 Edward Hervey <bilboed@bilboed.com>
981 * gst-libs/gst/riff/riff-media.c:
982 riff: vp61 and VP61 are also valid On2 VP6 fourcc
984 2010-01-23 15:10:45 +0100 Edward Hervey <bilboed@bilboed.com>
986 * gst-libs/gst/riff/riff-media.c:
987 riff: Add mapping for On2 VP5
989 2010-01-23 15:04:35 +0100 Edward Hervey <bilboed@bilboed.com>
991 * gst-libs/gst/riff/riff-media.c:
992 riff: Add mapping for Sigma-Designs MPEG4
993 It's actually a xvid-compatible stream. both xviddec and ffmpeg handle it.
995 2010-01-23 14:35:28 +0100 Edward Hervey <bilboed@bilboed.com>
997 * gst-libs/gst/pbutils/descriptions.c:
998 pbutils: Add description for LOCO Lossless codec
1000 2010-01-23 14:35:16 +0100 Edward Hervey <bilboed@bilboed.com>
1002 * gst-libs/gst/riff/riff-media.c:
1003 riff: Add mapping for LOCO Lossless codec
1005 2010-01-23 14:08:39 +0100 Edward Hervey <bilboed@bilboed.com>
1007 * gst-libs/gst/riff/riff-media.c:
1008 riff: Add support for YV12 / Uncompressed packed YVU 4:2:2
1010 2010-01-23 13:50:26 +0100 Edward Hervey <bilboed@bilboed.com>
1012 * gst-libs/gst/pbutils/descriptions.c:
1013 pbutils: add description for Autodesk Animator codec
1015 2010-01-23 13:50:09 +0100 Edward Hervey <bilboed@bilboed.com>
1017 * gst-libs/gst/riff/riff-media.c:
1018 riff: Add mapping for Autodesk Animator Codec
1020 2010-01-23 13:20:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1022 * ext/ogg/gstoggdemux.c:
1023 oggdemux: ...and set caps on queued packet buffers too
1025 2010-01-23 13:19:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1027 * ext/ogg/gstoggdemux.c:
1028 oggdemux: Set caps on header buffers
1030 2010-01-22 16:23:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1032 * gst/playback/gsturidecodebin.c:
1033 uridecodebin: handle raw sources about-to-finish signals
1034 When we are dealing with a source that produces raw audio/video, we don't use a
1035 decodebin2 to decode the data and we thus don't have the drained/about-to-finish
1036 signal emited. To fix this, we add a padprobe on the source pads and emit the
1037 drained signal ourselves. This then makes playbin2 emit the about-to-finish
1038 signal for raw sources such as cdda://
1041 2010-01-22 16:15:54 +0200 Stefan Kost <ensonic@users.sf.net>
1043 * gst/typefind/gsttypefindfunctions.c:
1044 typefind: include stdio.h for sscanf
1046 2010-01-22 01:49:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1048 * gst/typefind/gsttypefindfunctions.c:
1049 typefinding: add PNM typefinder
1050 Add PNM typefinder, so we can remove the one that's in the PNM plugin
1051 in -bad (which btw uses different/wrong media types that don't match
1052 the ones used by gdkpixbufdec) and people don't make fun of us for
1053 loading image decoders when typefinding and playing back audio files.
1055 2010-01-21 19:31:23 +0100 Thijs Vermeir <thijsvermeir@gmail.com>
1057 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
1058 * gst/ffmpegcolorspace/imgconvert.c:
1059 ffmpegcolorspace: rename performance category
1060 rename the performance category to ffmpegcolorspace_performance
1061 as there is already a global GST_CAT_PERFORMANCE in core
1063 2010-01-21 17:32:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1065 * ext/ogg/gstoggdemux.c:
1066 * ext/ogg/gstoggdemux.h:
1067 oggdemux: keep track of added pads
1068 Keep track of the pads we added and removed.
1069 Remove some unused fields.
1070 Don't add pads for which we don't have caps.
1072 2010-01-21 17:31:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1074 * ext/ogg/gstoggstream.c:
1075 oggstream: don't call NULL setup functions
1076 If we find a known mapper but it doesn't have a setup function, simply skip it
1077 instead of crashing.
1079 2010-01-21 17:30:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1081 * ext/ogg/gstoggstream.c:
1082 oggstream: avoid division by 0 on bad annodex streams
1084 2010-01-21 13:47:01 +0100 Edward Hervey <bilboed@bilboed.com>
1086 * gst-libs/gst/pbutils/descriptions.c:
1087 pbutils: Add description for y4m container
1089 2010-01-19 14:31:34 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
1091 * gst-libs/gst/rtp/gstbasertppayload.c:
1092 basertppayload: ptime/maxptime should be unsigned
1093 https://bugzilla.gnome.org/show_bug.cgi?id=607403
1095 2010-01-18 21:16:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
1097 * gst-libs/gst/rtp/gstbasertppayload.c:
1098 * gst-libs/gst/rtp/gstbasertppayload.h:
1099 basertppayload: ptime should be in nanoseconds
1100 https://bugzilla.gnome.org/show_bug.cgi?id=607403
1102 2010-01-20 00:53:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1105 Automatic update of common submodule
1106 From 14cec89 to 15d47a6
1108 2010-01-19 13:33:06 -0800 David Schleef <ds@schleef.org>
1110 * gst/typefind/gsttypefindfunctions.c:
1111 typefind: rewrite h.264 detection
1112 Make detection simpler: check for NALs, check that they make
1113 sense, and report how certain we are that it's a raw H.264 stream.
1116 2010-01-18 14:33:30 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
1118 * gst-libs/gst/rtp/gstbasertppayload.c:
1119 basertppayload: Reject empty caps
1120 https://bugzilla.gnome.org/show_bug.cgi?id=607353
1122 2010-01-19 08:39:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1124 * ext/ogg/gstoggdemux.c:
1125 oggdemux: No need to subtract begin time
1126 Last stop is already based on the chain start and there is no need
1127 to subtract the chain start as it may lead to a negative overflow.
1128 This was causing seeking issues when the target chain was not
1129 the first one (that has chain start = 0)
1132 2010-01-19 09:25:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1134 * gst-libs/gst/audio/audio.h:
1135 audio: Use rounding scaling functions for GST_CLOCK_TIME_TO_FRAMES and _FRAMES_TO_CLOCK_TIME
1138 2010-01-18 15:22:52 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1140 * ext/ogg/gstoggdemux.c:
1141 oggdemux: granulepos is relative to its chain
1142 When performing seeks, the granulepos should be offset by
1143 its chain start time to avoid using wrong values to
1144 update segment's last_stop. A sample file is indicated on
1147 2010-01-18 17:57:16 +0100 Edward Hervey <bilboed@bilboed.com>
1149 * gst-libs/gst/pbutils/descriptions.c:
1150 pbutils: Add description for MXF container format
1152 2010-01-18 10:07:30 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1154 * gst/playback/gstplaysink.c:
1155 playsink: re-use iterator callback to avoid code duplication
1157 2010-01-18 02:08:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1159 * gst/playback/gstplaysink.c:
1160 playsink: when looking for sink properties, make sure they have the right type
1161 We don't want to end up setting values on elements where the property is of
1162 a different type than we expect. Can't transform the value either, since we
1163 can't really make assumptions about the scale and transform function.
1164 Fixes crashes when using playbin2 with apexsink (#606949).
1166 2010-01-18 09:30:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1168 * gst/playback/gstplaybin2.c:
1169 playbin2: Only allow to set the URIs in states <= READY or from an about-to-finish signal handler
1170 Changing the URIs in a state > READY results in unexpected behaviour,
1171 i.e. the new URIs are only used after the current track has finished.
1174 2010-01-15 19:52:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1176 * gst/playback/gstdecodebin2.c:
1177 decodebin2: sprinkle some more locking
1178 ... to avoid races and ensure some data structure consistency.
1181 2010-01-14 18:26:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1183 * gst/playback/gstdecodebin2.c:
1184 decodebin2: mind blocked pads when shutting down
1185 Fix regression in shutdown deadlock handling now that the
1186 target of a ghostpad is blocked instead of ghostpad itself.
1189 2010-01-14 13:36:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1191 * gst/playback/gstplaysink.c:
1192 playsink: Fix disabling of subtitles if subtitles were used before
1193 In this case the video still goes through the text chain and
1194 subtitles are still going in there, in case subtitles are
1195 enabled again. This makes sure that re-enabling subtitles
1197 Fixes hanging video when disabling subtitles, caused by an
1200 2010-01-14 10:43:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1202 * gst/playback/gstplaybin2.c:
1203 playbin2: fix pad ref leak
1205 2010-01-12 21:42:59 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1207 * docs/plugins/Makefile.am:
1208 docs: fix out-of-source build
1210 2009-04-29 11:50:03 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1212 * tests/icles/stress-playbin.c:
1213 stress-playbin: fix error return check
1215 2010-01-14 10:10:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1217 * ext/theora/Makefile.am:
1218 * ext/theora/gsttheora.c:
1219 * ext/theora/gsttheoradec.c:
1220 * ext/theora/gsttheoraenc.c:
1221 * ext/theora/gsttheoraparse.c:
1222 * ext/theora/theora.c:
1223 * ext/theora/theoradec.c:
1224 * ext/theora/theoraenc.c:
1225 * ext/theora/theoraparse.c:
1226 theora: Rename source files to have the same name as the headers
1228 2010-01-14 10:07:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1230 * ext/vorbis/Makefile.am:
1231 * ext/vorbis/gstvorbis.c:
1232 * ext/vorbis/gstvorbisdec.c:
1233 * ext/vorbis/gstvorbisenc.c:
1234 * ext/vorbis/gstvorbisparse.c:
1235 * ext/vorbis/gstvorbistag.c:
1236 * ext/vorbis/vorbis.c:
1237 * ext/vorbis/vorbisdec.c:
1238 * ext/vorbis/vorbisenc.c:
1239 * ext/vorbis/vorbisparse.c:
1240 * ext/vorbis/vorbistag.c:
1241 vorbis: Rename source files to have the same name as the headers
1243 2010-01-14 10:05:35 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1245 * ext/vorbis/Makefile.am:
1246 * ext/vorbis/gstvorbiscommon.c:
1247 * ext/vorbis/gstvorbiscommon.h:
1248 * ext/vorbis/vorbisdec.c:
1249 * ext/vorbis/vorbisenc.c:
1250 vorbis: Move channel layout definitions into a single separate file
1251 ...instead of having two copies.
1253 2010-01-14 08:19:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1255 * ext/vorbis/vorbisdec.c:
1256 * ext/vorbis/vorbisenc.c:
1257 vorbis: Add official 6.1 and 7.1 channel mappings
1258 These are in the Vorbis spec since 2010-01-13. Fixes bug #606926.
1260 2010-01-13 23:05:45 +0100 Benjamin Otte <otte@redhat.com>
1262 * gst-libs/gst/rtsp/gstrtspdefs.c:
1263 rtsp: Don't define h_error ourselves
1264 It's included from netdb.h and that header might define it differently,
1265 which can lead to build failures.
1267 2010-01-13 17:36:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1269 * gst/typefind/gsttypefindfunctions.c:
1270 typefind: mp4 video is not parsed
1272 2010-01-13 12:49:20 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1274 * gst/typefind/gsttypefindfunctions.c:
1275 typefind: Add aac stream-format to caps
1276 Also add the aac stream-format field on the caps when
1279 2010-01-13 09:39:54 +0100 Brijesh Singh <brijesh.ksingh@gmail.com>
1281 * gst/playback/gstplaysink.c:
1282 playsink: Fix handling of the native audio/video flags
1285 2010-01-12 16:35:50 +0100 Edward Hervey <bilboed@bilboed.com>
1287 * ext/ogg/gstoggdemux.c:
1288 oggdemux: Fix unitialized variable.
1289 If the package isn't handled, gracefully return GST_FLOW_OK.
1291 2010-01-10 23:50:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1293 * gst-libs/gst/interfaces/xoverlay.c:
1294 docs: flesh out GtkXOverlay docs some more and add example for Gtk+ >= 2.18
1295 Explain why the whole bus sync handler mess is needed. Add section about
1296 how to use GstXOverlay in connection with Gtk+ and mention the Gtk+ API
1297 break issue and how to work around it (see #601809).
1299 2010-01-10 21:18:04 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1301 * gst-libs/gst/netbuffer/gstnetbuffer.c:
1302 docs: minor netbuffer documentation fix
1304 2010-01-10 20:41:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1338 po: update translated strings
1339 Queue2 moved into core, so remove its strings.
1341 2010-01-08 16:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1343 * ext/ogg/gstoggdemux.c:
1344 * ext/ogg/gstoggstream.h:
1345 oggdemux: push headers when activating chains
1346 Keep a list of headers for each stream of a chain. When a chain is activated,
1347 push the headers before pushing the data so that decoders can sync.
1348 Fix seeking in chains, take the chain start time into account when comparing
1352 2010-01-07 15:26:57 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1354 * gst-libs/gst/tag/Makefile.am:
1355 * gst-libs/gst/tag/lang-tables.c:
1356 * gst-libs/gst/tag/lang-tables.dat:
1357 * gst-libs/gst/tag/lang.c:
1358 tag: fix up disting of lang-tables.c more correctly
1359 lang-tables.c is included by lang.c and not really a proper source
1360 file that should be compiled into its own object, so rename it to
1361 lang-tables.dat and put it into EXTRA_DIST instead to ensure it
1364 2010-01-07 13:50:03 +0000 Christian Schaller <christian.schaller@collabora.co.uk>
1366 * gst-libs/gst/tag/Makefile.am:
1367 * gst-plugins-base.spec.in:
1368 Add missing source file for tagger to Makefile and update spec file
1370 2010-01-06 18:30:57 -0800 Mark Yen <mook@songbirdnest.com>
1372 * gst-libs/gst/riff/riff-media.c:
1373 riff-media: handle 32 bit raw RGB video.
1375 2010-01-06 13:57:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1377 * ext/ogg/gstoggstream.c:
1378 oggdemux: decide flac header packet by content rather than count
1380 2010-01-06 13:56:26 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1382 * ext/ogg/gstoggdemux.c:
1383 oggdemux: reset header packet count at bos page
1385 2010-01-06 13:39:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1387 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
1388 audiopayload: add support for buffer-lists
1390 2010-01-06 11:33:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1392 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
1394 2010-01-05 17:17:58 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
1396 * ext/pango/gsttextoverlay.c:
1397 textoverlay: Ignore zero framerate
1398 https://bugzilla.gnome.org/show_bug.cgi?id=606163
1400 2009-12-29 18:45:32 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
1402 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
1403 basertpaudiopayload: Respect ptime if it is given
1404 If the ptime is given in the caps, respect it and force the minimum
1405 and maximum sizes to be exactly the requested ptime.
1406 https://bugzilla.gnome.org/show_bug.cgi?id=606050
1408 2009-12-29 18:36:29 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
1410 * gst-libs/gst/rtp/gstbasertppayload.c:
1411 * gst-libs/gst/rtp/gstbasertppayload.h:
1412 rtpbasepayload: Store ptime from caps
1413 https://bugzilla.gnome.org/show_bug.cgi?id=606050
1415 2009-12-02 19:40:58 +0530 Olivier Crête <olivier.crete@collabora.co.uk>
1417 * gst-libs/gst/rtp/gstbasertppayload.c:
1418 basertppayload: Accept maxptime from caps
1419 https://bugzilla.gnome.org/show_bug.cgi?id=606050
1421 2010-01-05 14:11:06 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1423 * ext/ogg/gstoggstream.c:
1424 oggdemux: enhance flac packet duration calculation
1426 2010-01-05 10:38:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1428 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
1430 2010-01-04 09:49:25 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1432 * tests/examples/seek/seek.c:
1433 * tests/icles/test-colorkey.c:
1434 examples: use Gtk+-2.18 API conditionally
1435 so the seek example and colorkey test work with older Gtk+ versions
1439 2009-12-29 00:53:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1441 * tests/icles/test-colorkey.c:
1442 tests: fix colorkey test up for Gtk+ >= 2.18
1443 Make test-colorkey work with newer versions of Gtk+.
1446 2009-12-29 00:40:27 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1448 * tests/examples/seek/seek.c:
1449 examples: make seek example work with Gtk+ >= 2.18
1450 Gtk+ broke API slightly with the introduction of
1451 client-side windows in Gtk+ 2.18. Fix up seek
1452 example to work with newer Gtk+ versions.
1455 2009-12-26 23:29:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1457 * tests/icles/stress-xoverlay.c:
1458 tests: fix warning and memory leak in stress-overlay test
1459 Not all messages have structures and we need to unref messages
1460 when returning GST_BUS_DROP in the sync bus handler.
1462 2009-12-26 18:46:50 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1464 * gst/audiorate/gstaudiorate.c:
1465 audiorate: correctly eat empty and dummy buffers
1467 2009-12-24 19:56:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1469 * gst/adder/gstadder.c:
1470 adder: be a lot smarter with buffer management
1472 Try to reuse one of the input buffer as the output buffer. This usually works
1473 and avoids an allocation and a memcpy.
1474 Be smarter with GAP buffers so that they don't get mixed or cleared at all. Also
1475 try to use a GAP buffer as the output buffer when all input buffers are GAP
1478 2009-12-24 16:30:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1480 * gst/adder/Makefile.am:
1481 * gst/adder/gstadder.c:
1482 * tests/check/elements/adder.c:
1483 adder: use collectpads clipping function
1484 Install a clipping function in the collectpads and use the audio clipping helper
1485 function to perform clipping to the segment boundaries.
1488 2009-12-24 13:58:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1490 * gst/adder/gstadder.c:
1491 adder: fix juvenile comment
1493 2009-12-23 21:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1495 * gst/playback/gstdecodebin2.c:
1496 decodebin2: fix typo in debug message
1498 2009-12-23 18:18:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1500 * gst/playback/gstdecodebin2.c:
1501 decodebin2: avoid some type checks
1503 2009-12-23 17:08:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1505 * gst/playback/gstplaybin2.c:
1506 playbin2: avoid leaking selector request pads
1508 2009-12-23 15:46:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1510 * gst/playback/gsturidecodebin.c:
1511 uridecodebin: avoid leaking queue and typefind
1512 Don't leak the queue and typefind elements that we might link after the
1515 2009-12-23 15:43:52 +0100 Jonathan Matthew <jonathan@d14n.org>
1517 * gst/playback/gsturidecodebin.c:
1518 uridecodebin: don't name the queue
1519 There is no reason to name the queue.
1522 2009-12-23 15:30:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1524 * win32/common/libgstrtp.def:
1525 defs: update defs with new symbols
1527 2009-12-22 20:15:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1529 * docs/libs/gst-plugins-base-libs-sections.txt:
1530 * gst-libs/gst/rtp/gstrtcpbuffer.c:
1531 * gst-libs/gst/rtp/gstrtcpbuffer.h:
1532 rtcpbuffer: add helper functions for SDES types
1533 Add functions to convert SDES names to their types and back. Will be used later
1534 to set SDES items using a GstStructure.
1537 2009-12-21 19:12:02 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1540 Automatic update of common submodule
1541 From 47cb23a to 14cec89
1543 2009-12-21 18:45:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1545 * gst/audiorate/gstaudiorate.c:
1546 audiorate: add Since marker for the new tolerance property
1548 2009-12-21 07:57:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1550 * gst-libs/gst/tag/lang.c:
1551 docs: use 'Returns: xyz' rather than 'Returns xyz' to make gtk-doc happy
1553 2009-12-21 07:50:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1555 * tests/examples/app/appsrc-ra.c:
1556 * tests/examples/app/appsrc-seekable.c:
1557 * tests/examples/app/appsrc-stream.c:
1558 * tests/examples/app/appsrc-stream2.c:
1559 tests: don't use deprecated GLib API g_mapped_file_free
1562 2009-12-20 17:34:46 -0800 David Schleef <ds@schleef.org>
1564 * ext/theora/gsttheoraenc.h:
1565 * ext/theora/theoraenc.c:
1566 theoraenc: Add encoder controls for libtheora 1.1
1567 Added drop-frames, cap-overflow, cap-underflow, and rate-buffer.
1569 2009-12-19 21:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1571 * gst-libs/gst/audio/gstbaseaudiosink.c:
1572 baseaudiosink: increase default drift tolerance to fix glitches with WMA
1573 Increase default drift tolerance to 40ms to avoid glitches with decoders
1574 or formats where there's a lot of timestamp jitter for some reason or
1575 another (in this case: asf/wma), at least until we implement timestamp
1578 2009-12-16 11:43:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1580 * gst/playback/gstdecodebin2.c:
1581 decodebin2: add some debugging
1583 2009-12-15 18:41:38 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1585 * gst/audiorate/gstaudiorate.c:
1586 * gst/audiorate/gstaudiorate.h:
1587 audiorate: add a tolerance property
1588 It may not be uncommon for the input timestamps to experience some jitter
1589 around the 'perfect time'. As such, instead of regularly adding and dropping
1590 samples, optionally allow for some tolerance in a more relaxed approach.
1591 API: GstAudioRate:tolerance
1593 2009-12-15 19:50:56 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1595 * docs/plugins/Makefile.am:
1596 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
1597 * docs/plugins/gst-plugins-base-plugins-sections.txt:
1598 * gst/audiorate/gstaudiorate.c:
1599 audiorate: add documentation
1601 2009-12-15 16:52:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1603 * gst/audiorate/Makefile.am:
1604 * gst/audiorate/gstaudiorate.c:
1605 * gst/audiorate/gstaudiorate.h:
1606 audiorate: use separate header file
1608 2009-12-14 21:17:57 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1610 * gst/audiorate/gstaudiorate.c:
1611 audiorate: set DISCONT when resyncing (e.g. newsegment)
1613 2009-12-14 18:47:27 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1615 * gst/audiorate/gstaudiorate.c:
1616 audiorate: also fill up segments if possible
1618 2009-12-15 19:29:29 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1620 * gst/audiorate/gstaudiorate.c:
1621 audiorate: fix segment handling
1622 Do not compare a media (buffer) time to a (bogus) running time
1623 (or their offset equivalents).
1625 2009-12-15 19:22:45 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1627 * gst/audiorate/gstaudiorate.c:
1628 audiorate: properly report truncated samples as dropped samples
1630 2009-12-13 18:43:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1632 * gst-libs/gst/tag/lang.c:
1633 docs: mention that gst_tag_get_language_name() may return NULL
1635 2009-12-13 18:42:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1637 * tests/check/libs/tag.c:
1638 checks: some more testing for the new language code functions
1640 2009-12-12 18:58:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1642 * gst-libs/gst/interfaces/mixer.c:
1643 * gst-libs/gst/interfaces/mixeroptions.c:
1644 * gst-libs/gst/interfaces/mixertrack.c:
1645 docs: misc. mixer docs improvements
1647 2009-12-12 18:16:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1649 * gst-libs/gst/app/gstappsink.c:
1650 * gst-libs/gst/app/gstappsrc.c:
1651 docs: add short descriptions for API reference contents page
1653 2009-12-12 17:43:26 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1655 * gst-libs/gst/tag/lang-tables.c:
1656 * gst-libs/gst/tag/mklangtables.c:
1657 tag: make internal language names table static
1659 2009-12-12 17:41:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1661 * gst-libs/gst/tag/lang.c:
1662 * gst-libs/gst/tag/mklangtables.c:
1663 tag: don't use GLib 2.22 API
1664 g_mapped_file_unref() was introduced in GLib 2.22, but we depend
1665 only on GLib 2.18, so use g_mapped_file_free() when compiling
1666 against older GLib versions until we bump the GLib dependency.
1668 2009-12-11 23:59:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1672 * docs/libs/gst-plugins-base-libs-docs.sgml:
1673 * docs/libs/gst-plugins-base-libs-sections.txt:
1674 * gst-libs/gst/tag/Makefile.am:
1675 * gst-libs/gst/tag/lang-tables.c:
1676 * gst-libs/gst/tag/lang.c:
1677 * gst-libs/gst/tag/mklangtables.c:
1678 * gst-libs/gst/tag/tag.h:
1679 * tests/check/libs/tag.c:
1680 * win32/common/libgsttag.def:
1681 tag: add some utility functions for language codes and tags
1682 Add some utility functions for language tags and ISO-639
1683 codes. These are useful for both GUIs and elements. The
1684 iso-codes package is used for language name translations
1686 API: gst_tag_get_language_codes()
1687 API: gst_tag_get_language_name()
1688 API: gst_tag_get_language_code()
1689 API: gst_tag_get_language_code_iso_639_1()
1690 API: gst_tag_get_language_code_iso_639_2B()
1691 API: gst_tag_get_language_code_iso_639_2T()
1693 2009-12-11 12:02:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1695 * ext/ogg/gstoggstream.c:
1696 ogg: ogm video has constant packet duration
1698 2009-12-10 22:47:53 -0800 David Schleef <ds@schleef.org>
1700 * ext/ogg/gstoggstream.c:
1701 oggdemux: implement old fLaC mapping
1703 2009-12-10 17:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1705 * gst/tcp/gsttcpclientsrc.c:
1706 tcpclientsrc: unset flushing state too
1707 When unlocking, we set the flushing state on the fdset. Implement unlock_stop so
1708 that we can use it to unset the flushing state again.
1711 2009-12-10 16:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1713 * ext/ogg/gstoggdemux.c:
1714 * ext/ogg/gstoggdemux.h:
1715 oggdemux: remove redundant fields
1717 2009-12-09 19:03:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1719 * ext/vorbis/gstvorbisdec.h:
1720 * ext/vorbis/vorbisdec.c:
1721 vorbisdec: adapt to new oggdemux
1722 Remove all granulepos hacks and simply use the timestamps from the new oggdemux
1723 like any other decoder.
1725 2009-12-09 19:04:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1727 * ext/vorbis/vorbisdec.c:
1728 vorbisdec: fix peer query
1730 2009-12-09 17:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1732 * ext/theora/theoradec.c:
1733 theoradec: fix query
1735 2009-12-09 16:55:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1737 * ext/theora/theoradec.c:
1738 theoradec: small cleanups
1740 2009-12-09 16:38:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1742 * ext/vorbis/vorbisdec.c:
1743 vorbisdec: use gst_pad_peer_query()
1745 2009-12-09 12:10:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1747 * gst/playback/gstplaysink.c:
1748 playsink: fix video when subtitles disabled
1749 When we have a source with subtitles but they were disabled with the flags,
1750 still ghostpad the video pad instead of leaving it unlinked.
1752 2009-12-09 09:47:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1754 * ext/pango/gsttextoverlay.c:
1755 textoverlay: Only flush downstream on seeks for flushing seeks
1757 2009-12-09 09:35:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1759 * ext/pango/gsttextoverlay.c:
1760 textoverlay: Proxy buffer allocation on the video sinkpad to the srcpad
1762 2009-12-08 17:30:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1764 * tests/examples/seek/seek.c:
1765 seek: update slider only 25 times a second
1766 don't update the slider a 100 times a second, it's likely higher than the screen
1767 framerate and just wastes cpu.
1769 2009-12-08 17:23:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1771 * ext/theora/gsttheoradec.h:
1772 * ext/theora/theoradec.c:
1773 theora: remove granulepos hacks
1774 Remove the granulepos hacking now that oggdemux outputs timestamps like any
1777 2009-12-08 13:40:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1779 * gst/playback/gstplaybin2.c:
1780 playbin2: Fix stream-changed message list iteration
1781 When iterating the list and removing the current element, first
1782 get the next element and then remove the current one and not
1783 the other way around.
1785 2009-12-07 18:49:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1787 * ext/ogg/gstoggdemux.c:
1788 oggdemux: improve keyframe seeking
1789 Improve keyframe seeking.
1790 Fix reverse playback.
1792 2009-12-07 15:42:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1794 * ext/ogg/gstoggdemux.c:
1795 oggdemux: implement keyframe seeking
1796 Implement keyframe seeking in oggdemux by doing the double seek trick. First
1797 seek to the required position, then read pages for all streams to grab the
1798 granulepos (to know the timing of the keyframe) of each stream, then seek back
1799 to the first keyframe.
1801 2009-12-07 09:13:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1803 * gst/playback/gstplaysink.c:
1804 playsink: Some minor cleanup
1806 2009-12-06 18:05:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1808 * gst/playback/gstplaybin2.c:
1809 playbin2: Reset stream segments on FLUSH_STOP and don't adjust QoS events for non-time segments
1811 2009-12-04 16:35:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1813 * ext/ogg/gstoggdemux.c:
1814 oggdemux: fix timestamps after seek
1815 After a seek, discard all packets before the packet with the granulepos on it so
1816 that the output buffers contain valid timestamps.
1817 Reorder some code so that we check the timestamps before allocating and pushing
1819 Do more checks on valid packets in ogm mode.
1821 2009-12-04 15:39:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1823 * ext/ogg/gstoggdemux.c:
1824 oggdemux: add comment
1826 2009-12-04 14:01:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1828 * ext/ogg/gstoggdemux.c:
1829 oggdemux: don't do math with invalid granulepos
1830 When the current granulepos is unknown and set to -1, don't try to add durations
1833 2009-12-04 13:14:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1835 * ext/ogg/gstoggdemux.c:
1836 * ext/ogg/gstoggdemux.h:
1837 oggdemux: guard against wrong granulepos
1838 Clamp the initial granulepos to 0 instead of going negative for some badly muxed
1841 2009-12-04 12:26:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
1843 * ext/theora/theoradec.c:
1844 theoradec: don't fail on bogus granulepos
1845 Do some additional checks on the granulpos timestamp before using it for
1846 calculating the duration because oggdemux generates wrong granulepos now.
1847 Fixes seeking somewhat again.
1849 2009-12-03 20:05:29 -0800 David Schleef <ds@schleef.org>
1851 * ext/ogg/gstoggdemux.c:
1852 * ext/ogg/gstoggstream.c:
1853 * ext/ogg/gstoggstream.h:
1854 oggdemux: reimplement OGM support
1855 OGM demuxing no longer requires helper elements. It's done internally
1856 in oggdemux. Vorbis comments are still not handled because I don't
1857 have anything to test with.
1859 2009-12-03 17:02:11 -0800 David Schleef <ds@schleef.org>
1861 * ext/ogg/gstoggstream.c:
1862 oggdemux: fix for I-frame-only theora
1864 2009-12-03 01:16:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1866 * ext/ogg/gstoggstream.c:
1867 ogg: log when ogg mapper doesn't accept the setup header packet
1869 2009-12-02 02:08:46 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1871 * ext/ogg/gstoggstream.c:
1872 ogg: extract width, height and PAR from theora header and add to caps
1874 2009-12-03 23:43:08 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1876 * ext/ogg/gstoggstream.c:
1877 ogg: extract number of channels from FLAC, speex and vorbis headers
1880 2009-12-03 22:14:34 +0200 Stefan Kost <ensonic@users.sf.net>
1882 * gst/playback/gstplaybin2.c:
1883 build: fix build with debug logging disabled.
1885 2009-12-03 21:07:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1887 * ext/ogg/gstoggdemux.c:
1888 * ext/ogg/gstoggstream.c:
1889 ogg: more print fixes
1890 gstoggstream.c:419: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘gint64’
1891 gstoggdemux.c:2253: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘GstClockTime’
1892 gstoggdemux.c:2333: error: format ‘%lld’ expects type ‘long long int’, but argument 8 has type ‘GstClockTime’
1894 2009-12-03 16:57:48 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1896 * ext/ogg/gstoggparse.c:
1897 * ext/ogg/gstoggstream.c:
1898 ogg: Fixing some printf format strings
1899 Fixes some printf format strings to make it build on mac.
1901 2009-12-03 18:08:49 +0200 Stefan Kost <ensonic@users.sf.net>
1903 * gst/playback/gstfactorylists.c:
1904 * gst/playback/gstfactorylists.h:
1905 * gst/playback/gstplaybin2.c:
1906 playbin2: don't iterate the factory lists in non-debug mode
1907 When debugging is disabled, we won't see anything printed anyway.
1909 2009-12-02 23:55:55 -0800 David Schleef <ds@schleef.org>
1911 * gst/videoscale/vs_4tap.c:
1914 2009-12-02 23:27:55 +0200 Stefan Kost <ensonic@users.sf.net>
1916 * gst/subparse/qttextparse.c:
1917 build: add missing includes for sprintf and atoi
1919 2009-12-01 16:42:42 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1921 * gst/subparse/gstsubparse.c:
1922 * gst/subparse/qttextparse.c:
1923 subparse: Add support for some tags of qttext
1924 Currently supporting timescale, timestamps, font, size,
1925 textColor, backColor, plain, bold and italic
1928 2009-12-01 13:13:24 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1930 * gst/subparse/Makefile.am:
1931 * gst/subparse/gstsubparse.c:
1932 * gst/subparse/gstsubparse.h:
1933 * gst/subparse/qttextparse.c:
1934 * gst/subparse/qttextparse.h:
1935 subparse: add qttext support
1936 Adds basic support for qttext subtitles, still lacks markup tags
1937 to make it prettier, but the plain text already works.
1938 Implemented according to:
1939 http://www.apple.com/quicktime/tutorials/texttracks.html
1940 http://www.apple.com/quicktime/tutorials/textdescriptors.html
1943 2009-12-01 13:22:57 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1945 * gst/subparse/gstsubparse.c:
1946 subparse: conditionally cleanup sami context
1947 Only cleanup sami context if we are parsing sami subtitles,
1948 otherwise we might have crashes.
1950 2009-12-01 13:19:35 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
1952 * gst/subparse/gstsubparse.c:
1953 subparse: Add missing caps to sink caps template
1954 Some caps were missing from the sink caps template when
1957 2009-12-01 15:06:10 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1960 Automatic update of common submodule
1961 From 87bf428 to 47cb23a
1963 2009-12-01 14:14:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1966 Automatic update of common submodule
1967 From da4c75c to 87bf428
1969 2009-11-30 10:22:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1971 * gst/playback/gstsubtitleoverlay.c:
1972 subtitleoverlay: Fix some pad refcount issues
1975 2009-11-27 18:54:57 +0100 Edward Hervey <bilboed@bilboed.com>
1978 Automatic update of common submodule
1979 From 53a2485 to da4c75c
1981 2009-11-25 17:04:41 -0800 David Schleef <ds@schleef.org>
1983 * ext/ogg/gstoggstream.c:
1984 * ext/ogg/gstoggstream.h:
1985 oggdemux: handle theora streams with 0 keyoffset
1987 2009-11-25 16:53:26 -0800 David Schleef <ds@schleef.org>
1989 * ext/ogg/gstoggdemux.c:
1990 oggdemux: Handle unknown streams
1992 2009-11-26 14:30:33 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1994 * ext/pango/gsttextoverlay.c:
1995 Revert "textoverlay: First draw outline text and then the real text"
1996 This reverts commit 60aa09d28c1f9fd29b56876d7ac6c0366d6cef4d.
1997 First drawing the real text and then the outline produces ugly
1998 text in lower resolutions. The outline line width needs to be somehow
1999 changed relative to the resolution. Fixes bug #602924.
2001 2009-11-26 10:30:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2003 * gst-libs/gst/audio/gstaudiofilter.c:
2004 audiofilter: Use G_DEFINE_ABSTRACT_TYPE_WITH_CODE
2005 ...and fix code style a bit.
2007 2009-11-26 10:31:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2009 * gst-libs/gst/audio/gstaudiofilter.h:
2010 audiofilter: Add _CAST variants of the cast macros
2012 2009-11-25 10:26:16 -0600 Wim Taymans <wim.taymans@collabora.co.uk>
2014 * gst-libs/gst/audio/gstbaseaudiosink.c:
2015 audiosink: add adjustement when slaving
2016 Our calibration against the pipeline clock is done with the adjusted
2017 ringbuffer time, so take the adjustement into account. Fixes some audio dropouts
2018 when reusing audio sinks after switching clocks and slaving methods in a
2021 2009-11-25 16:17:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2023 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
2024 ffmpegcolorspace: Prefer transforming alpha formats to alpha formats and the other way around
2025 Fixes bug #602834 and #350748.
2027 2009-11-25 00:46:55 -0800 David Schleef <ds@schleef.org>
2029 * ext/ogg/gstoggdemux.c:
2030 oggdemux: Reset last_granule during seeking
2031 Fix case where we would reconstruct the wrong granulepos for
2032 outgoing streams immediately after a seek.
2034 2009-11-24 22:08:09 -0800 David Schleef <ds@schleef.org>
2036 * ext/ogg/gstoggdemux.c:
2037 * ext/ogg/gstoggdemux.h:
2038 * ext/ogg/gstoggstream.c:
2039 * ext/ogg/gstoggstream.h:
2040 oggdemux: Fix timestamp generation for theora
2041 Timestamp generation was broken by the last commit for formats
2042 with a non-zero granule shift. Also keep track of the last keyframe
2043 so that we can regenerate granulepos for theora.
2045 2009-11-24 21:22:03 -0800 David Schleef <ds@schleef.org>
2047 * ext/ogg/gstoggdemux.c:
2048 * ext/ogg/gstoggstream.c:
2049 * ext/ogg/gstoggstream.h:
2050 * ext/ogg/vorbis_parse.c:
2051 oggdemux: Fix vorbis parsing
2052 Add a granule to granulepos conversion function. Fix the duration
2053 function for vorbis. Handle timestamps on header packets differently
2054 and be more careful about calculating OFFSET and OFFSET_END. After
2055 this change, timestamps for vorbis don't exactly match up with the
2056 timestamps that vorbisparse outputs, but it's unclear if vorbisparse
2057 is actually correct and it would add a lot more code to make oggdemux
2058 match vorbisparse. Fixes #602790.
2060 2009-11-19 19:28:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2062 * gst/playback/gstplaybin2.c:
2063 playbin2: Transform QoS events to be meaningful for upstream elements
2064 This is necessary because the sinks don't notice the group switches
2065 and the decoders/demuxers have a different running time than the
2069 2009-11-21 22:05:34 +0100 David Schleef <ds@schleef.org>
2071 * ext/ogg/gstoggdemux.c:
2072 ogg: Fix generation of timestamps and durations
2073 After changing some internal functions, I forgot to update
2074 the code that puts the values on the buffers.
2076 2009-08-29 10:51:48 -0700 David Schleef <ds@schleef.org>
2078 * ext/ogg/Makefile.am:
2079 * ext/ogg/dirac_parse.c:
2080 * ext/ogg/dirac_parse.h:
2081 * ext/ogg/gstoggdemux.c:
2082 * ext/ogg/gstoggdemux.h:
2083 * ext/ogg/gstoggparse.c:
2084 * ext/ogg/gstoggstream.c:
2085 * ext/ogg/gstoggstream.h:
2086 * ext/ogg/vorbis_parse.c:
2087 ogg: Add ogg stream parsing
2088 Adds code that parses headers of various formats encapsulated in
2089 Ogg in order to calculate timestamps and durations of each buffer.
2090 Removes the creation of helper decoder elements to do this calculation
2091 via conversion queries.
2092 Fixes: #344013, #568014.
2094 2009-09-04 00:11:38 -0700 David Schleef <ds@schleef.org>
2096 * ext/ogg/gstoggmux.c:
2097 oggmux: don't overwrite object properties
2099 2009-11-21 17:54:49 +0200 Stefan Kost <ensonic@users.sf.net>
2101 * ext/theora/theoradec.c:
2102 debug: also cast packet.packetno to gint64 in debug log
2103 We do this already for granulepos to handle ogg_int64_t mismatches.
2105 2009-11-21 17:47:26 +0200 Stefan Kost <ensonic@users.sf.net>
2107 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2108 debug: fix format string that was missing a var
2110 2009-10-10 00:32:04 +0300 Stefan Kost <ensonic@users.sf.net>
2112 * gst/adder/gstadder.c:
2113 * tests/check/elements/adder.c:
2114 adder: make events succeed, if they succed on atleast one pad
2116 2009-11-19 14:51:33 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
2118 * gst/playback/gstdecodebin2.c:
2119 decodebin2: error when all streams have no buffers
2120 In some cases (all buffers dropped by a parser) a decodebin2
2121 chain might receive an EOS before it gets enough data to
2122 expose a decoded pad. In the case that no streams can expose
2123 a pad we should error out instead of hang.
2126 2009-11-19 12:23:08 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2128 * gst/playback/gstplaybin2.c:
2129 playbin2: Fix stupid bug introduced in last commit
2131 2009-11-19 12:10:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2133 * gst/playback/gstplaybin2.c:
2134 playbin2: Aggregate the stream-changed message by looking at the seqnum
2135 Just counting how many messages were sent and how many were received
2136 is not good enough because they might've been duplicated (e.g. by the
2137 visualization audio tee). Comparing the sequence numbers should give
2138 better results in that case.
2140 2009-11-19 10:05:28 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2142 * gst/playback/gstplaybin2.c:
2143 playbin2: Ignore async state changes of the uridecodebins
2144 Otherwise the async state change from READY->PAUSED of the
2145 uridecodebins will take playbin2 from PLAYING->PAUSED again
2146 during gapless group switches.
2149 2009-11-19 10:30:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2152 Automatic update of common submodule
2153 From 0702fe1 to 53a2485
2155 2009-11-18 14:50:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
2157 * gst/playback/gstdecodebin2.c:
2158 decodebin2: set to buffer less on no-more-pads
2159 When a decodebin2 receives no-more-pads of a group it
2160 can set that group's multiqueue buffering thresholds to
2161 'playing' buffering method, avoiding that it buffers
2162 too long and cause problems when using with queue2.
2163 See the associated bug for details.
2166 2009-11-18 17:09:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2168 * gst-libs/gst/audio/gstbaseaudiosink.c:
2169 baseaudiosink: fix initial calibration
2170 When we are calibrating the internal clock against the external clock take into
2171 account the time offset applied to our internal clock because we will subtract
2172 that in the render_function again.
2174 2009-11-18 09:22:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2176 * gst/playback/gstplaybin2.c:
2177 playbin2: Don't handle DURATION queries during group switches
2178 During a group switch return the cached duration of the old group
2179 because the old group still didn't finish playback. If we have no
2180 cached duration return FALSE.
2183 2009-11-15 19:36:21 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2185 * gst/playback/gstplaybin2.c:
2186 playbin2: Post a stream-changed message after activating a group
2187 This is useful to detect when playbin2 has really switched to the next
2188 group after about-to-finish for example.
2191 2009-11-18 12:27:19 +0000 Jan Schmidt <thaytan@noraisin.net>
2193 * win32/common/libgstvideo.def:
2194 win32: Add new still-frame API to the defs
2195 Add gst_video_event_new_still_frame() and
2196 gst_video_event_parse_still_frame() functions to the win32 defs files
2198 2009-11-18 12:37:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
2200 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2201 baseaudiosrc: fix 'uninitialized' compiler warning
2203 2009-11-18 10:14:41 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2206 configure: bump core requirement to 0.10.25.1
2207 We depend on new API that's only in git so far.
2209 2009-11-15 17:34:37 +0000 Jan Schmidt <thaytan@noraisin.net>
2211 * gst-libs/gst/video/video.c:
2212 * gst-libs/gst/video/video.h:
2213 * tests/check/libs/video.c:
2214 video: Add functions to create/parse still frame events.
2215 Add a new video event to mark the start or end of a still-frame
2216 sequence, and a parser function to identify and extract info from
2218 API: gst_video_event_new_still_frame()
2219 API: gst_video_event_parse_still_frame()
2222 2009-11-17 16:39:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2224 * gst/playback/gstplaysink.c:
2225 playsink: make sure we always go to PAUSED async
2226 Set the need_async_start flag before going to PAUSED so that we always post the
2227 ASYNC_START message, even after reusing playsink.
2229 2009-11-17 16:37:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2231 * gst/playback/gstplaysink.c:
2232 playsink: make sure we remain a sink
2233 When we remove our elements, we could lose our sink flag. Make sure we remain a
2234 sink by setting the flag again after removing elements.
2236 2009-11-16 22:47:54 +0200 Stefan Kost <ensonic@users.sf.net>
2238 * gst/audioconvert/gstaudioconvert.c:
2239 audioconvert: remove unused array
2241 2009-11-16 09:57:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2243 * gst/subparse/gstsubparse.c:
2244 subparse: Use new double->fraction transformation function from core
2246 2009-11-14 14:05:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2248 * gst/playback/gstplaybin2.c:
2249 playbin2: Make subtitle error handling more robust and ignore late errors too
2250 Make sure, to only "simulate" subtitle no-more-pads if it was still
2251 pending and also handle errors in the subtitle pipeline as warnings
2252 after the subtitles prerolled.
2253 Don't set the suburidecodebin to READY after errors, handle_message
2254 will usually be called from the streaming thread and doing that
2255 from there is obviously not a good idea.
2257 2009-11-14 13:21:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2259 * gst/playback/gstsubtitleoverlay.c:
2260 * gst/playback/gstsubtitleoverlay.h:
2261 subtitleoverlay: Handle errors from subtitle elements as warning and go into passthrough mode
2263 2009-11-13 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2265 * gst/playback/gstplaybin2.c:
2266 playbin2: Don't leak the GError and debug string when parsing error messages
2268 2009-11-13 11:16:44 +0100 Sreerenj B <bsreerenj@gmail.com>
2270 * gst-libs/gst/rtsp/gstrtspconnection.c:
2271 rtsp: avoid crashing on SIGPIPE
2272 Use send() instead of write() so that we can pass the MSG_NOSIGNAL flags to
2273 avoid crashing with SIGPIPE when the remote end is not listening to us anymore.
2276 2009-11-11 17:35:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2278 * gst/playback/gstplaybin2.c:
2279 playbin2: Improve subtitle passthrough in uridecodebin
2280 Now the caps property isn't set anymore for the subtitle caps
2281 but instead in the autoplug-continue signal it is detected
2282 if the caps belong to a supported subtitle stream.
2283 This makes automatic use of newly installed plugins.
2285 2009-11-11 17:08:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2287 * gst/playback/gstsubtitleoverlay.c:
2288 subtitleoverlay: Only recreate factory caps if necessary and cache them
2290 2009-11-10 18:27:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2292 * gst/playback/gstsubtitleoverlay.c:
2293 * gst/playback/gstsubtitleoverlay.h:
2294 subtitleoverlay: Only update the factory list when the registry has changed
2295 Also don't free the list every time we go to NULL.
2297 2009-11-08 15:04:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2299 * gst/playback/gstsubtitleoverlay.c:
2300 subtitleoverlay: Use gst_pad_get_caps_reffed()
2302 2009-11-07 21:38:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2304 * gst/playback/gstplaybin2.c:
2305 * gst/playback/gstplaysink.c:
2306 playbin2/playsink: Use new "silent" property instead of unlinking
2307 This makes sure that subtitleoverlay still gets segment updates and
2308 everything to pass on downstream. Without this segment problems happen.
2310 2009-11-07 21:10:27 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2312 * gst/playback/gstsubtitleoverlay.c:
2313 * gst/playback/gstsubtitleoverlay.h:
2314 subtitleoverlay: Update segments after pushing the events downstream
2315 This makes sure that we don't apply segments twice downstream. Also
2316 always send our newsegment events downstream.
2318 2009-11-07 21:09:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2320 * gst/playback/gstsubtitleoverlay.c:
2321 * gst/playback/gstsubtitleoverlay.h:
2322 subtitleoverlay: Add silent property to disable subtitles
2323 This tries to disable subtitles in the overlay or renderer
2324 and if that's not possible it goes into passthrough mode.
2326 2009-11-07 11:46:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2328 * gst/playback/gstsubtitleoverlay.c:
2329 * gst/playback/gstsubtitleoverlay.h:
2330 subtitleoverlay: Set the video framerate on parsers if possible
2333 2009-11-07 11:31:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2335 * gst/subparse/gstsubparse.c:
2336 * gst/subparse/gstsubparse.h:
2337 subparse: Make fps a GstFraction typed property and use it properly
2339 2009-11-07 11:08:19 +0100 Iago Toral <itoral@igalia.com>
2341 * gst/subparse/gstsubparse.c:
2342 * gst/subparse/gstsubparse.h:
2343 subparse: Add property for the video framerate
2345 2009-11-06 12:51:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2347 * gst/playback/gstplaybin2.c:
2348 playbin2: Handle external subtitles better
2349 First of all, make sure that suburidecodebin never
2350 errors out because of not-linked in case external subtitles
2351 are used but then subtitles are disabled.
2352 And then make sure that external subtitles always start from
2353 the correct position and are not racing until EOS if they
2354 get unselected and selected again.
2356 2009-11-04 17:29:07 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2358 * gst/playback/gstplaybin2.c:
2359 playbin2: Flush the subtitles before switching to a new subtitle stream
2360 This makes sure that all currently shown subtitles disappear
2361 and new ones can be shown as soon as possible.
2363 2009-11-03 12:47:55 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2365 * gst/playback/gstplaybin2.c:
2366 playbin2: Set subtitle caps as raw caps for the uridecodebins
2367 This will make sure that no subparse is ever plugged and subtitleoverlay,
2368 that subpicture streams are handled the same was as subtitles and that
2369 subtitle renderers are used if available.
2370 Fixes bugs #595123, #570753, #591662, #591706.
2372 2009-11-03 12:33:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2374 * gst/playback/gstplaybin2.c:
2375 * gst/playback/gstplaysink.c:
2376 * gst/playback/gstplaysink.h:
2377 playbin2/playsink: Remove everything related to subpicture streams
2378 These will soon be handled the same way as subtitle streams.
2380 2009-11-02 15:50:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2382 * gst/playback/gstplaysink.c:
2383 playsink: Add a queue before subtitleoverlay
2384 This will improve playback, and the same thing is done
2385 for subpicture streams too.
2387 2009-11-02 15:05:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2389 * gst/playback/gstplaysink.c:
2390 playsink: Use subtitleoverlay for subtitles
2392 2009-11-02 07:43:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2394 * docs/plugins/Makefile.am:
2395 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
2396 * docs/plugins/gst-plugins-base-plugins-sections.txt:
2397 * docs/plugins/gst-plugins-base-plugins.args:
2398 * docs/plugins/gst-plugins-base-plugins.hierarchy:
2399 * docs/plugins/gst-plugins-base-plugins.interfaces:
2400 * docs/plugins/gst-plugins-base-plugins.prerequisites:
2401 * docs/plugins/inspect/plugin-adder.xml:
2402 * docs/plugins/inspect/plugin-alsa.xml:
2403 * docs/plugins/inspect/plugin-app.xml:
2404 * docs/plugins/inspect/plugin-audioconvert.xml:
2405 * docs/plugins/inspect/plugin-audiorate.xml:
2406 * docs/plugins/inspect/plugin-audioresample.xml:
2407 * docs/plugins/inspect/plugin-audiotestsrc.xml:
2408 * docs/plugins/inspect/plugin-cdparanoia.xml:
2409 * docs/plugins/inspect/plugin-decodebin.xml:
2410 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
2411 * docs/plugins/inspect/plugin-gdp.xml:
2412 * docs/plugins/inspect/plugin-gio.xml:
2413 * docs/plugins/inspect/plugin-gnomevfs.xml:
2414 * docs/plugins/inspect/plugin-libvisual.xml:
2415 * docs/plugins/inspect/plugin-ogg.xml:
2416 * docs/plugins/inspect/plugin-pango.xml:
2417 * docs/plugins/inspect/plugin-playback.xml:
2418 * docs/plugins/inspect/plugin-subparse.xml:
2419 * docs/plugins/inspect/plugin-tcp.xml:
2420 * docs/plugins/inspect/plugin-theora.xml:
2421 * docs/plugins/inspect/plugin-typefindfunctions.xml:
2422 * docs/plugins/inspect/plugin-uridecodebin.xml:
2423 * docs/plugins/inspect/plugin-video4linux.xml:
2424 * docs/plugins/inspect/plugin-videorate.xml:
2425 * docs/plugins/inspect/plugin-videoscale.xml:
2426 * docs/plugins/inspect/plugin-videotestsrc.xml:
2427 * docs/plugins/inspect/plugin-volume.xml:
2428 * docs/plugins/inspect/plugin-vorbis.xml:
2429 * docs/plugins/inspect/plugin-ximagesink.xml:
2430 * docs/plugins/inspect/plugin-xvimagesink.xml:
2431 subtitleoverlay: Add to the docs
2433 2009-10-13 16:48:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2435 * gst/playback/Makefile.am:
2436 * gst/playback/gstplayback.c:
2437 * gst/playback/gstsubtitleoverlay.c:
2438 * gst/playback/gstsubtitleoverlay.h:
2439 subtitleoverlay: Add new element for generic subtitle overlaying
2440 This autopluggs the required elements for parsing and rendering
2441 different subtitle formats on a video stream.
2444 2009-11-11 19:32:01 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
2446 * ext/theora/theoradec.c:
2447 theoradec: Keep timestamp from incoming buffer if it is valid
2450 2009-11-11 14:00:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2452 * gst/playback/gstdecodebin2.c:
2453 * gst/playback/gstplaybin2.c:
2454 * gst/playback/gsturidecodebin.c:
2455 playback: Update factories list on every access if the registry has changed
2456 This makes application's simpler because the element doesn't need to
2457 go to NULL first to make use of newly installed plugins.
2460 2009-11-10 18:13:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2462 * gst/playback/gstdecodebin2.c:
2463 * gst/playback/gstplaybin2.c:
2464 * gst/playback/gsturidecodebin.c:
2465 playback: When going from NULL->READY check if the registry has new features
2466 This makes it possible to use newly installed plugins after going back
2467 to NULL instead of requiring a new instance.
2470 2009-11-10 13:55:26 +0000 Jan Schmidt <thaytan@noraisin.net>
2472 * gst-libs/gst/app/gstappsrc.c:
2473 appsrc: Clear the EOS state on a seek.
2474 Allow seeking back into the stream after it hits EOS.
2476 2009-11-10 12:21:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2478 * gst/audioresample/README:
2479 * gst/audioresample/arch.h:
2480 * gst/audioresample/fixed_arm4.h:
2481 * gst/audioresample/fixed_arm5e.h:
2482 * gst/audioresample/fixed_bfin.h:
2483 * gst/audioresample/fixed_debug.h:
2484 * gst/audioresample/resample.c:
2485 * gst/audioresample/resample_sse.h:
2486 * gst/audioresample/speex_resampler.h:
2487 audioresample: Update speex resampler to latest GIT
2489 2009-11-10 00:48:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2491 * gst/playback/gstplaysink.c:
2492 playsink: assign chain->mute before using it
2493 Fixes GObject warnings when starting totem.
2495 2009-10-28 22:10:33 -0700 David Schleef <ds@schleef.org>
2497 * ext/theora/theoradec.c:
2498 theora: Fix alignment of frames when converting
2499 Fix logic inversion in calculating the offset in the theora
2500 frame when copying to a GStreamer frame.
2502 2009-11-09 19:58:20 +0100 Edward Hervey <bilboed@bilboed.com>
2504 * gst/playback/gstfactorylists.c:
2505 playback: Fix the order in strcmp that I broke in previous commit.
2507 2009-11-09 19:16:21 +0100 Edward Hervey <bilboed@bilboed.com>
2509 * gst/typefind/gsttypefindfunctions.c:
2510 typefind: Reduce number of calls to gst_type_find_peek.
2511 Shaves off a couple percents off typefinding
2513 2009-11-09 17:49:51 +0100 Edward Hervey <bilboed@bilboed.com>
2515 * gst/playback/gstfactorylists.c:
2516 playback: Avoid expensive API calls in tight loop.
2517 We know we're dealing with GstPluginFeature.
2519 2009-11-09 18:11:42 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2521 * tests/check/libs/cddabasesrc.c:
2522 cddabasesrc: Add unit test for property settings
2523 Also includes a regression test for bug #601104.
2525 2009-11-09 18:04:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2527 * gst-libs/gst/cdda/gstcddabasesrc.c:
2528 cddabasesrc: Never return a negative track number in get_uri()
2530 2009-11-09 18:03:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2532 * gst-libs/gst/cdda/gstcddabasesrc.c:
2533 cddabasesrc: Don't set the track to 1 every time a device is set
2536 2009-11-08 11:27:10 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2538 * gst/playback/gstinputselector.c:
2539 inputselector: Remove useless variables and fix a uninitialized variable compiler warnings
2541 2009-11-06 17:01:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2543 * gst/playback/gstdecodebin2.c:
2544 decodebin2: Add property to disable/enable posting of stream-topology messages
2545 Most people don't need this messages and generating them is quite
2548 2009-11-06 15:12:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2550 * gst/playback/gstdecodebin2.c:
2551 decodebin2: Protect subtitle elements and subtitle encoding by a new mutex
2552 Using the object lock here can and will lead to deadlocks because
2553 of deep-notifies of property changes: the deep-notify handler will
2554 get the parent of objects, which will take the object lock again.
2557 2009-11-06 13:13:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2559 * gst/playback/gstinputselector.c:
2560 inputselector: Make sure that running_time->timestamp calculation never becomes negative
2562 2009-11-06 13:25:05 +0200 Mart Raudsepp <leio@gentoo.org>
2564 * tests/examples/seek/scrubby.c:
2565 * tests/examples/seek/seek.c:
2566 examples: Correct casting of g_signal* funcs first arguments
2567 This completes the deprecated GTK API fix in commits 81a0a986 and
2568 79adfa54 - unlike gtk_signal_connect and co, g_signal_connect and
2569 co take a gpointer, not a GtkObject.
2571 2009-11-06 12:25:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2573 * gst/playback/gsturidecodebin.c:
2574 uridecodebin: Improve all-raw-caps detection for pads
2576 2009-11-06 12:19:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2578 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2579 basesrc: fix startup position in the ringbuffer
2580 When we start and we need to produce the first sample, go to the next sample
2581 that will be written into the ringbuffer instead of trying to go to sample 0.
2582 We relied on rather small ringbuffer sizes to correctly go to the current
2583 sample, which breaks whith large buffers.
2586 2009-11-06 11:26:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2588 * gst/playback/gstinputselector.c:
2589 inputselector: Use the start time (i.e. timestamp) as the last stop
2590 Using the end time makes it impossible to replace buffers, which is
2591 a big problem for subtitles that could have very long durations.
2593 2009-11-06 12:08:19 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2595 * ext/pango/gsttextoverlay.c:
2596 textoverlay: Synchronize video/text based on the running time
2597 Instead of simply using the buffer timestamps.
2599 2009-11-06 09:30:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2601 * ext/pango/gsttextoverlay.c:
2602 textoverlay: Clip text buffers to the text segment and reset segments properly
2604 2009-11-06 09:01:34 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2606 * ext/pango/gsttextoverlay.c:
2607 * ext/pango/gsttextoverlay.h:
2608 textoverlay: Put the video segment into the instance struct instead of allocating it separately
2610 2009-11-06 09:05:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2612 * ext/pango/gsttextoverlay.c:
2613 textoverlay: Check if text timestamp/duration is valid before clipping
2615 2009-11-05 23:33:42 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2617 * ext/theora/theoradec.c:
2618 theoradec: printf format fix
2620 2009-11-05 15:42:09 +0100 Olivier Crête <olivier.crete@collabora.co.uk>
2622 * gst/gdp/gstgdpdepay.c:
2623 gdpdepay: Clear adapter on flush and state change
2626 2009-11-05 13:12:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2628 * gst/playback/gstinputselector.c:
2629 inputselector: use _get_caps_reffed()
2631 2009-11-05 13:00:27 +0200 Stefan Kost <ensonic@users.sf.net>
2633 * gst/playback/gstdecodebin2.c:
2634 * gst/playback/gstplaybin2.c:
2635 * gst/playback/gsturidecodebin.c:
2636 pad: rename new api from _refed to _reffed.
2637 Due to popular demand rename the new api as we still can.
2639 2009-11-04 18:57:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2641 * gst/playback/gstplaybin2.c:
2642 * gst/playback/gsturidecodebin.c:
2643 playbin2: avoid copying caps
2644 Use get_caps_refed() when we can.
2646 2009-11-04 18:31:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2648 * gst/playback/gstdecodebin2.c:
2649 decodebin2: use new getcaps function to avoid copies
2650 Use the gst_pad_get_caps_refed() to avoid some caps copy functions.
2652 2009-11-04 17:50:11 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2654 * gst/playback/gsturidecodebin.c:
2655 uridecodebin: use faster element_link_pads
2656 Use the faster gst_element_link_pads because we know for sure the sinkpad name
2657 and we don't need to have the function search for a suitable pad anymore.
2659 2009-11-04 16:16:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2661 * gst-libs/gst/audio/gstbaseaudiosink.c:
2662 baseaudiosink: make drift tolerance configurable
2663 Add drift-tolerance property (defaulting to 20ms) to handle resync after clock
2664 drift or timestamp drift instead of relying on the latency-time value for clock
2665 drift and 500ms for timestamp drift.
2666 Remove warning about discont timestamp and simply resync. The warning is in some
2667 cases not correct and is triggered more frequently now that we lower the
2670 2009-11-04 10:52:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2672 * gst/playback/gstplaybin2.c:
2673 playbin2: Return NOT_LINKED for unselected text pads from a demuxer
2674 We want to return NOT_LINKED for unselected pads but only for pads
2675 from the normal uridecodebin. This makes sure that subtitle streams
2676 are not raced past audio/video from decodebin2's multiqueue.
2677 For pads from suburidecodebin OK should always be returned, otherwise
2678 it will most likely stop with an error.
2680 2009-11-04 08:20:59 +0100 Stefan Kost <ensonic@users.sf.net>
2682 * gst/playback/gstinputselector.c:
2683 inputselector: also add inline to the proto to fix the build
2684 Merged from gst-plugins-bad, e1e9be6dbe1bd0df0543f2a72dcf9cc6d644dd78.
2686 2009-11-03 12:01:16 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2688 * gst/playback/gsturidecodebin.c:
2689 uridecodebin: Initialize caps property with the default raw caps
2691 2009-11-03 11:48:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2693 * gst/playback/Makefile.am:
2694 * gst/playback/gstdecodebin2.c:
2695 * gst/playback/gstrawcaps.h:
2696 decodebin2: Use static caps for the default raw caps and put them into a separate header
2697 This way we can use the same default raw caps everywhere.
2699 2009-11-03 08:26:37 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2701 * ext/pango/gsttextoverlay.c:
2702 textoverlay: First draw outline text and then the real text
2703 Improves the output a bit because no parts of the outline are
2706 2009-10-31 14:02:40 +0100 Josep Torra Valles <n770galaxy@gmail.com>
2708 * gst/playback/gstplaybin.c:
2709 playbin: Make sure to keep a reference on the volume element
2710 Fixes null pointer dereferences under certain circumstances.
2713 2009-10-31 09:47:54 +0100 Edward Hervey <bilboed@bilboed.com>
2716 po: queue2 has moved to core
2718 2009-10-30 09:24:30 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2720 * gst/playback/gstplaysink.c:
2721 playsink: Reset {mute,volume}-changed flags after setting the volume
2722 These flags are there to make sure that the volume is set, if there
2723 is no volume element yet.
2725 2009-10-30 09:24:03 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2727 * gst/playback/gstplaysink.c:
2728 playsink: If notify::{volume,mute} is triggered by the volume element, update our internal state
2730 2009-10-29 14:30:31 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2732 * gst/playback/gstplaysink.c:
2733 playsink: Proxy notify::volume and notify::mute from the volume/mute elements (or sinks)
2736 2009-10-29 14:19:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2738 * gst/playback/gstplaybin2.c:
2739 playbin2: Proxy notify::volume and notify::mute from the playsink to playbin2
2741 2009-10-29 11:37:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2743 * docs/plugins/inspect/plugin-queue2.xml:
2744 queue2: Remove inspect file
2746 2009-10-29 11:29:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2748 * gst/playback/Makefile.am:
2749 * gst/playback/gstqueue2.c:
2750 queue2: Remove from gst-plugins-base
2751 This is now in coreplugins.
2753 2009-10-28 11:29:36 +0200 Stefan Kost <ensonic@users.sf.net>
2755 * docs/libs/gst-plugins-base-libs-docs.sgml:
2756 docs: include more indexes
2758 2009-10-28 11:13:20 +0200 Stefan Kost <ensonic@users.sf.net>
2760 * docs/libs/gst-plugins-base-libs-docs.sgml:
2761 docs: turn entities into xi:includes
2762 This is faster to process and easier to maintain. Its also less 80s.
2764 2009-10-28 10:17:43 +0200 Stefan Kost <ensonic@users.sf.net>
2766 * gst-libs/gst/rtp/gstrtpbuffer.c:
2767 rtp: dump packets which we reject
2769 2009-10-28 01:01:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2771 * tests/check/pipelines/.gitignore:
2772 .gitignore: ignore basetime unit test binary
2774 2009-10-28 00:59:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2776 * ext/alsa/gstalsasink.c:
2777 * ext/alsa/gstalsasrc.c:
2778 * gst-libs/gst/audio/gstaudiosink.c:
2779 * gst-libs/gst/audio/gstaudiosrc.c:
2780 * gst-libs/gst/audio/gstbaseaudiosink.c:
2781 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2782 * gst-libs/gst/audio/gstringbuffer.c:
2783 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
2784 * gst/adder/gstadder.c:
2785 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
2786 * gst/gdp/gstgdpdepay.c:
2787 * gst/gdp/gstgdppay.c:
2788 * gst/playback/gstdecodebin.c:
2789 * gst/playback/gstdecodebin2.c:
2790 * gst/playback/gstinputselector.c:
2791 * gst/playback/gstplaybasebin.c:
2792 * gst/playback/gstplaybin.c:
2793 * gst/playback/gstplaybin2.c:
2794 * gst/playback/gstplaysink.c:
2795 * gst/playback/gstqueue2.c:
2796 * gst/playback/gststreaminfo.c:
2797 * gst/playback/gststreamselector.c:
2798 * gst/subparse/gstssaparse.c:
2799 Remove GST_DEBUG_FUNCPTR where they're pointless
2800 There's not much point in using GST_DEBUG_FUNCPTR with GObject
2801 virtual functions such as get_property, set_propery, finalize and
2802 dispose, since they'll never be used by anyone anyway. Saves a
2803 few bytes and possibly a sixteenth of a polar bear.
2805 2009-10-27 15:23:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2807 * gst/playback/gstqueue2.c:
2808 queue2: add custom acceptcaps function
2810 2009-10-27 15:22:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2812 * gst/playback/gstdecodebin2.c:
2813 decodebin2: implement low/high watermark property
2815 2009-10-23 14:56:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2817 * tests/examples/seek/seek.c:
2818 seek: add checkbox to enable buffering
2820 2009-10-23 14:54:47 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2822 * gst/playback/gsturidecodebin.c:
2823 uridecodebin: don't use 2 buffering elements
2824 Only use the multiqueue buffering when we don't have a stream (and thus are
2825 using queue2 to do the buffering already).
2827 2009-10-23 14:34:42 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2829 * gst/playback/gstplay-enum.c:
2830 * gst/playback/gstplay-enum.h:
2831 * gst/playback/gstplaybin2.c:
2832 playbin2: add flag to enable decodebin buffering
2833 Add a flag that enables buffering in decodebin.
2835 2009-10-23 14:32:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2837 * gst/playback/gstdecodebin2.c:
2838 decodebin2: buffering is implemented now
2840 2009-10-23 14:30:52 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2842 * gst/playback/gsturidecodebin.c:
2843 uridecodebin: buffering is implemented now
2845 2009-10-23 14:09:17 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2847 * gst/playback/gstdecodebin2.c:
2848 decodebin2: configure use-buffering on multiqueue
2850 2009-10-23 13:58:25 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2852 * gst/playback/gsturidecodebin.c:
2853 uridecodebin: use 0 for max buffer size
2855 2009-10-23 13:53:21 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2857 * gst/playback/gsturidecodebin.c:
2858 uridecodebin: set some reasonable defaults
2860 2009-10-23 13:44:12 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2862 * gst/playback/gsturidecodebin.c:
2863 uridecodebin: set buffering properties on decodebin2
2864 Propagate the buffering properties on decodebin2 but only if we are not already
2865 doing download buffering.
2867 2009-10-23 11:52:09 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2869 * gst/playback/gsturidecodebin.c:
2870 uridecodebin: add use-buffering property
2871 Add a use-buffering property that will perform buffering on the parsed or
2874 2009-10-23 11:31:47 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2876 * gst/playback/gstdecodebin2.c:
2877 decodebin2: refactor queue size configuration.
2878 Refactor the queue size configuration into a new method.
2879 Use the same queue values for buffering as for preroll.
2881 2009-10-23 11:08:50 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2883 * gst/playback/gstdecodebin2.c:
2884 decodebin2: move error path down
2886 2009-10-23 11:02:40 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2888 * gst/playback/gstdecodebin2.c:
2889 decodebin2: implement max queue size properties
2891 2009-10-23 10:42:23 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2893 * gst/playback/gstdecodebin2.c:
2894 decodebin2: add properties for buffering
2895 Add properties that can be used to configure the multiqueue buffers and
2898 2009-10-24 13:19:08 +0200 Edward Hervey <bilboed@bilboed.com>
2900 * tests/examples/app/Makefile.am:
2901 * tests/examples/seek/Makefile.am:
2902 * tests/examples/v4l/Makefile.am:
2903 examples: fix linking order.
2904 the uninstalled wrapper would create a LD_LIBRARY_PATH with system-wide
2905 path before the local ones... resulting in the example applications picking
2906 up the system-wide libraries and not the (potentially modified) uninstalled
2909 2009-10-24 13:08:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2911 * gst/playback/gstplaybin2.c:
2912 playbin2: Don't destroy the suburidecodebin on errors
2913 It can still be reused
2915 2009-10-24 13:07:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2917 * gst/playback/gstplaybin2.c:
2918 playbin2: If setting the state of the suburidecodebin fails just warn, don't error out
2920 2009-10-24 12:12:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2922 * gst/playback/gstplaybin2.c:
2923 playbin2: Don't set uridecodebin states to NULL before reusing them
2924 This makes sure that the internal decodebin2 and everything else can
2925 be reused without reinstantiation.
2927 2009-10-18 17:28:22 +0200 Edward Hervey <bilboed@bilboed.com>
2929 * gst/playback/gsturidecodebin.c:
2930 uridecodebin: Store unused decodebin2 instances for further usage.
2931 This allows faster re-use of uridecodebin.
2932 https://bugzilla.gnome.org/show_bug.cgi?id=599471
2934 2009-10-23 17:49:15 -0700 David Schleef <ds@schleef.org>
2936 * ext/theora/gsttheoraparse.h:
2937 * ext/theora/theoraparse.c:
2938 theora: Convert theoraparse to libtheora 1.0 API
2940 2009-10-21 12:38:59 +0300 Olivier Crête <olivier.crete@collabora.co.uk>
2942 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
2943 rtpaudiopayload: Only sent exact multiple of the frame size
2944 Also align the maximum size with the frame size, not only the minimum
2946 2009-10-22 09:12:03 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
2948 * gst/audiorate/gstaudiorate.c:
2949 audiorate: move debug calculation into debug macro
2950 Remove in_duration and move its calculation to
2951 GST_LOG_OBJECT macro. This way it will only be calculated
2952 if we have debug enabled.
2954 2009-10-22 09:06:02 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
2956 * gst/audiorate/gstaudiorate.c:
2957 audiorate: Removing unused variable
2958 The in_stop variable was never read. Removing it.
2960 2009-10-22 08:40:01 -0300 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
2962 * gst/audiorate/gstaudiorate.c:
2963 audiorate: be more accurate on offset math
2964 Replace gst_util_uint64_scale_int for its rounding version
2965 to improve accuracy and avoid inserting samples where
2969 2009-10-22 10:17:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2971 * ext/pango/gsttextoverlay.c:
2972 textoverlay: Optimize a bit more
2973 ...and add a FIXME for bug #598695 and explain
2974 what we should do once Pango supports user fonts.
2976 2009-10-22 10:02:11 +0200 Iago Toral <itoral@igalia.com>
2978 * gst/subparse/gstsubparse.c:
2979 * gst/subparse/gstsubparse.h:
2980 * tests/check/elements/subparse.c:
2981 subparse: Add support for DKS subtitle format
2984 2009-10-22 09:31:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2986 * ext/pango/gsttextoverlay.c:
2987 textoverlay: Do shading as first operation
2989 2009-10-22 09:08:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2991 * ext/pango/gsttextoverlay.c:
2992 textoverlay: Only use a single cairo surface for drawing
2993 ... and comment/optimize what is going on here a bit better.
2995 2009-10-21 16:24:29 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
2997 * gst/playback/gstinputselector.c:
2998 inputselector: set output caps before pushing
2999 Set the output caps on the srcpad before pushing the buffer because else core
3000 will do a rather expensive check to see if we can actually accept those caps on
3003 2009-10-21 15:58:11 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
3005 * gst/playback/gstinputselector.c:
3006 inputselector: install an acceptcaps function
3007 Install a custom acceptcaps function instead of using the default expensive
3008 check. We accept whatever downstream accepts so we pass along the acceptcaps
3009 call to the downstream peer.
3011 2009-10-21 20:35:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3013 * gst/typefind/gsttypefindfunctions.c:
3014 typefind: fix typo in previous mxf typefinder change
3016 2009-10-21 20:44:33 +0200 Edward Hervey <bilboed@bilboed.com>
3018 * gst/typefind/gsttypefindfunctions.c:
3019 typefind: speed up mxf_type_find over 300 times for worst case scenarios
3020 * memcmp is expensive and was being abused, reduce calling it by checking
3022 * iterating one byte at at time over 64 kbites introduces a certain overhead,
3023 therefore we now do it in chunks of 1024 bytes
3024 And I do mean over 300 times. The average instruction call per mxf_type_find
3025 was previously 785685 and it's now down to 2458 :)
3027 2009-10-20 17:13:39 -0400 Wim Taymans <wim.taymans@collabora.co.uk>
3029 * gst/playback/gstfactorylists.c:
3030 decodebin2: avoid type checks
3032 2009-10-20 09:00:28 +0200 Edward Hervey <bilboed@bilboed.com>
3034 * gst/playback/gstdecodebin2.c:
3035 gst/decodebin2: Ensure we get fixed caps for topology message
3036 There are some corner cases (like with dvdemux amongst others) where
3037 the caps won't be negotiated, but the pad has fixed caps.
3039 2009-10-20 08:52:36 +0200 Edward Hervey <bilboed@bilboed.com>
3041 * gst/playback/gstdecodebin2.c:
3042 gst/decodebin2: Don't expose chains if we're shutting down.
3043 This avoids adding flushing pads to ourself
3045 2009-10-17 21:16:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3048 * ext/pango/gsttextoverlay.c:
3049 pango: bump pango requirement to stable version and remove ifdefs
3050 Bump pango requirement from an ancient development version to an
3051 ancient stable version.
3053 2009-10-17 21:11:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3055 * gst-libs/gst/rtsp/.gitignore:
3056 .gitignore: update after files got renamed
3058 2009-10-16 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3060 * gst-libs/gst/rtp/gstbasertppayload.c:
3061 basertppayload: small comment fix
3063 2009-10-16 10:50:35 +0200 Peter Kjellerstedt <pkj@axis.com>
3065 * gst-libs/gst/rtp/gstbasertppayload.c:
3066 rtp: Correct timestamping of buffers when buffer_lists are used
3067 The timestamping of buffers when buffer_lists are used failed if
3068 a buffer did not have both a timestamp and an offset.
3070 2009-10-16 10:56:56 +0300 Stefan Kost <ensonic@users.sf.net>
3072 * gst-libs/gst/app/Makefile.am:
3073 * gst-libs/gst/audio/Makefile.am:
3074 * gst-libs/gst/interfaces/Makefile.am:
3075 * gst-libs/gst/pbutils/Makefile.am:
3076 * gst-libs/gst/rtsp/Makefile.am:
3077 * gst-libs/gst/rtsp/gstrtsp-marshal.list:
3078 * gst-libs/gst/rtsp/gstrtspextension.c:
3079 * gst-libs/gst/rtsp/rtsp-marshal.list:
3080 * gst-libs/gst/video/Makefile.am:
3081 * gst/playback/Makefile.am:
3082 * gst/tcp/Makefile.am:
3083 build: fix previous commit to fully accomodate the glib-gen.mak changes
3084 I also renamed glib_enum_prefix to glib_gen_prefix as we also use that for the
3085 marshallers. Also rename the rtsp-marshal.list to work with the unified prefix.
3087 2009-10-16 10:18:45 +0300 Stefan Kost <ensonic@users.sf.net>
3089 * gst-libs/gst/app/Makefile.am:
3090 * gst-libs/gst/audio/Makefile.am:
3091 * gst-libs/gst/interfaces/Makefile.am:
3092 * gst-libs/gst/pbutils/Makefile.am:
3093 * gst-libs/gst/rtsp/Makefile.am:
3094 * gst-libs/gst/video/Makefile.am:
3095 * gst/playback/Makefile.am:
3096 * gst/tcp/Makefile.am:
3097 build: use gst-glib-gen.mak to fix the glib build rules. Fixes #598114
3098 The build rules in glib-gen.mak were using pattern rules in a non save way.
3100 2009-10-16 10:14:36 +0300 Stefan Kost <ensonic@users.sf.net>
3103 Automatic update of common submodule
3104 From 85d1530 to 0702fe1
3106 2009-09-10 11:39:18 +0200 Benjamin Otte <otte@gnome.org>
3108 * ext/theora/theoradec.c:
3109 theora: Make theoradec use gstvideo for image conversion
3110 Vastly simplifies code.
3111 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3113 2009-09-10 09:36:31 +0200 Benjamin Otte <otte@gnome.org>
3115 * ext/theora/theoradec.c:
3116 theora: Don't always round to even width/height
3117 Previously, the code always rounded to even sizes. Now it only ensures
3118 that pic_x and pic_y are multiples of 2 if the output format requires
3120 Also inlcudes fixes to take pic_x/y into account properly when copying
3122 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3124 2009-09-10 00:00:44 +0200 Benjamin Otte <otte@gnome.org>
3127 theora: Don't check for theora.pc anymore
3128 THe new APIs from theoradec and theoraenc are used now.
3129 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3131 2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
3133 * ext/theora/gsttheoradec.h:
3134 * ext/theora/theoradec.c:
3135 theora: Convert theoradec to libtheora 1.0 API
3136 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3138 2009-09-09 23:44:36 +0200 Benjamin Otte <otte@gnome.org>
3140 * ext/theora/Makefile.am:
3141 * ext/theora/gsttheoraenc.h:
3142 * ext/theora/theoraenc.c:
3143 theora: Port encoder to new Theora API
3144 Includes ripping out the old buffer copy code to fill up to frame size.
3145 This is not necesary with the new encoder.
3146 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3148 2009-09-09 21:59:31 +0200 Benjamin Otte <otte@gnome.org>
3150 * ext/theora/gsttheoraenc.h:
3151 * ext/theora/theoraenc.c:
3152 theora: Disable sharpness property
3153 It's ignored by libtheora
3154 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3156 2009-09-09 21:57:08 +0200 Benjamin Otte <otte@gnome.org>
3158 * ext/theora/gsttheoraenc.h:
3159 * ext/theora/theoraenc.c:
3160 theora: Disable noise-sensitivity property
3161 It is ignored by libtheora
3162 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3164 2009-09-09 21:50:57 +0200 Benjamin Otte <otte@gnome.org>
3166 * ext/theora/gsttheoraenc.h:
3167 * ext/theora/theoraenc.c:
3168 theora: Disable keyframe-mindistance property
3169 It's ignored by the current Theora library
3170 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3172 2009-09-09 21:48:08 +0200 Benjamin Otte <otte@gnome.org>
3174 * ext/theora/gsttheoraenc.h:
3175 * ext/theora/theoraenc.c:
3176 theora: Disable keyframe_threshold property
3177 It's ignored by the current theora encoder
3178 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3180 2009-09-09 20:26:47 +0200 Benjamin Otte <otte@gnome.org>
3182 * ext/theora/gsttheoraenc.h:
3183 * ext/theora/theoraenc.c:
3184 theora: Get rid of "quick" property
3185 The proeprty is not used by libtheora at all
3186 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3188 2009-09-08 15:12:23 +0200 Benjamin Otte <otte@gnome.org>
3191 * ext/theora/theoraenc.c:
3192 theora: remove support for outdated granulepos hack
3193 This is in preparation to switching to switching to the new Theora API
3194 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3196 2009-09-08 13:23:04 +0200 Benjamin Otte <otte@gnome.org>
3198 * ext/theora/gsttheoraenc.h:
3199 * ext/theora/theoraenc.c:
3200 theora: Ignore border property
3201 Always make the video use black as padding color.
3202 The output will be identical to previous versions.
3203 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3205 2009-09-08 13:18:26 +0200 Benjamin Otte <otte@gnome.org>
3207 * ext/theora/gsttheoraenc.h:
3208 * ext/theora/theoraenc.c:
3209 theora: Ignore the center property, always set video to top left
3210 This is not a necessary property, the output will be identical no matter
3212 https://bugzilla.gnome.org/show_bug.cgi?id=594729
3214 2009-10-15 16:34:28 +0100 Jan Schmidt <thaytan@noraisin.net>
3217 po: Don't create backup .po files
3218 As well as preventing creation of useless backup files, it works
3219 around a bug in gettext 0.17 on OS/X
3221 2009-10-15 13:13:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3223 * gst/playback/gstdecodebin2.c:
3224 decodebin2: Post a element message on the bus with the stream topology
3227 2009-10-15 13:01:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3229 * gst/playback/gstdecodebin2.c:
3230 decodebin2: Store the "endcaps" of a chain
3231 This are the caps that either resulted in a deadend if
3232 no plugin for them could be found or raw caps.
3234 2009-10-15 11:38:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3236 * gst/playback/gstdecodebin2.c:
3237 decodebin2: Store for every chain, which pad resulted in its creation
3239 2009-10-15 10:28:39 +0100 Jan Schmidt <thaytan@noraisin.net>
3241 * tests/check/pipelines/basetime.c:
3242 check: Don't fail the basetime test when no audiosrc is available
3243 On OS/X the DEFAULT_AUDIOSRC is not going to be available, because
3244 it isn't in gst-plugins-base. Just defer the test, instead of
3247 2009-10-14 10:41:03 +0200 Edward Hervey <bilboed@bilboed.com>
3250 Automatic update of common submodule
3251 From a3e3ce4 to 85d1530
3253 2009-10-14 08:36:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3255 * gst/playback/gstplaybin2.c:
3256 playbin2: Use gst_object_has_ancestor() instead of our own implementation of it
3258 2009-10-13 19:14:41 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
3260 * gst-libs/gst/audio/gstbaseaudiosrc.c:
3261 baseaudiosrc: fix timestamp comparission, Fixes #597407
3263 2009-10-13 13:52:02 +0300 Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
3265 * tests/check/Makefile.am:
3266 * tests/check/pipelines/basetime.c:
3267 tests: new test for baseaudiosrc base_time comparison
3268 This test reveals a bug in comparison operation between timestamp and
3269 GstElement's base_time in GstBaseAudioSrc.
3271 2009-10-08 19:55:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3273 * gst/playback/gstplaybin2.c:
3274 playbin2: Don't stop completely on initialization errors from subtitle elements
3275 Instead disable the subtitles and play the other parts of the stream.
3278 2009-10-13 16:50:37 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3280 * gst/playback/gstdecodebin2.c:
3281 decodebin2: Ignore no-more-pads from non-demuxer elements
3282 instead of printing an error that no corresponding group could
3283 be found. no-more-pads from non-demuxer elements doesn't give
3284 any additional information because there can only be a single srcpad.
3287 2009-10-12 21:30:15 +0300 Stefan Kost <ensonic@users.sf.net>
3289 * gst/audioconvert/gstaudioconvert.c:
3290 audioconvert: track active conversion in perf log
3292 2009-10-12 15:48:46 +0200 Patrick Radizi <patrick.radizi at axis.com>
3294 * gst-libs/gst/rtsp/gstrtspconnection.c:
3295 rtsp: handle socket errors
3296 gstrtspconnection.c:gst_rtsp_connection_receive() can hang when an error occured
3297 on a socekt. Fix this problem by checking for error on 'other' socket after poll
3301 2009-10-06 14:08:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3303 * gst-libs/gst/audio/gstaudioclock.c:
3304 audioclock: whitespace fixes
3306 2009-10-06 14:07:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3308 * ext/theora/theoradec.c:
3309 theoradec: avoid confusing error
3311 2009-10-09 22:00:45 +0200 Josep Torra <n770galaxy@gmail.com>
3313 * ext/vorbis/vorbisdec.c:
3314 * ext/vorbis/vorbisenc.c:
3315 vorbis: fixes warings in macosx snow leopard
3317 2009-10-09 18:52:12 +0200 Josep Torra <n770galaxy@gmail.com>
3319 * ext/theora/theoradec.c:
3320 * ext/theora/theoraparse.c:
3321 theora: fixes warnings on macosx snow leopard
3323 2009-10-09 16:56:29 +0200 Josep Torra <n770galaxy@gmail.com>
3325 * ext/ogg/gstoggmux.c:
3326 * ext/ogg/gstoggparse.c:
3327 ogg: fixes warnings on macosx snow leopard
3329 2009-10-09 16:19:17 +0200 Josep Torra <n770galaxy@gmail.com>
3331 * ext/ogg/gstoggdemux.c:
3332 oggdemux: fix a warning in macosx
3334 2009-10-08 14:16:44 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
3336 * gst-libs/gst/tag/tags.c:
3337 tag: use BOM to recognize UTF-16/32 encoding and convert accordingly
3339 2009-10-09 15:11:16 +0100 Jan Schmidt <thaytan@noraisin.net>
3341 * tests/check/gst-plugins-base.supp:
3342 check: Add valgrind suppressions for ALSA and fontconfig bits on Jaunty.
3344 2009-10-09 15:32:45 +0200 Josep Torra <n770galaxy@gmail.com>
3346 * ext/gnomevfs/gstgnomevfssrc.c:
3347 audioconvert: change the format instead of cast as ensonic asked
3349 2009-10-09 15:29:15 +0200 Josep Torra <n770galaxy@gmail.com>
3351 * gst/audioconvert/gstchannelmix.c:
3352 audioconvert: fixes warning: format not a string literal and no format arguments
3353 redo of valid part of my previous revert.
3355 2009-10-09 15:19:42 +0200 Josep Torra <n770galaxy@gmail.com>
3358 * gst/audioconvert/gstchannelmix.c:
3359 Revert "audioconvert: fixes warning: format not a string literal and no format arguments"
3360 Revert this commit as unintentionally I've changed common.
3361 This reverts commit 49ea0138223ec5f9e53780635cbcc70f33778667.
3363 2009-10-09 14:28:42 +0200 Josep Torra <n770galaxy@gmail.com>
3365 * ext/gnomevfs/gstgnomevfssrc.c:
3366 gnomevfssrc: fixes warnings in macosx
3367 warning: format '%llu' expects type 'long long unsigned int', but argument 8 has type 'GnomeVFSFileOffset'
3368 warning: format '%lld' expects type 'long long int', but argument 9 has type 'guint64'
3370 2009-10-09 14:23:36 +0200 Josep Torra <n770galaxy@gmail.com>
3372 * gst/videorate/gstvideorate.c:
3373 videorate: fix warning in macosx
3375 2009-10-09 14:20:47 +0200 Josep Torra <n770galaxy@gmail.com>
3377 * gst/audiorate/gstaudiorate.c:
3378 audiorate: fix warning in macosx
3380 2009-10-09 14:14:15 +0200 Josep Torra <n770galaxy@gmail.com>
3383 * gst/audioconvert/gstchannelmix.c:
3384 audioconvert: fixes warning: format not a string literal and no format arguments
3386 2009-10-09 14:07:24 +0200 Josep Torra <n770galaxy@gmail.com>
3388 * gst-libs/gst/audio/gstbaseaudiosrc.c:
3389 * gst-libs/gst/audio/gstringbuffer.c:
3390 audio: fix warnings building on macosx
3392 2009-10-08 18:08:22 +0300 Stefan Kost <ensonic@users.sf.net>
3394 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
3395 * gst/ffmpegcolorspace/imgconvert.c:
3396 ffmpegcolorspace: chwck formats just once per _chain()
3398 2009-10-08 17:49:39 +0300 Stefan Kost <ensonic@users.sf.net>
3400 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
3401 * gst/ffmpegcolorspace/imgconvert.c:
3402 ffmpegcolorspace: add perf-log-category and log suboptimal operation
3403 Log if we use an intermediate colorspace for conversion.
3405 2009-10-08 10:59:36 +0100 Jan Schmidt <thaytan@noraisin.net>
3408 Automatic update of common submodule
3409 From 19fa4f3 to a3e3ce4
3411 2009-10-08 00:17:21 +0100 Jan Schmidt <jan.schmidt@sun.com>
3413 * gst/playback/gstdecodebin2.c:
3414 decodebin2: Fix type-punning warning
3416 2009-09-26 12:56:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3418 * gst/playback/gstdecodebin2.c:
3419 decodebin2: Chains with an exposed endpad are complete too
3420 This allows partial group changes, i.e. demuxer2 in the example below
3421 goes EOS but has a next group and audio2 stays the same.
3422 /-- >demuxer2---->video
3423 demuxer--- \--->audio1
3426 2009-09-26 12:47:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3428 * gst/playback/gstdecodebin2.c:
3429 decodebin2: Use the iterate internal links function instead of string magic to get multiqueue srcpads
3431 2009-09-24 14:56:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3433 * gst/playback/gsturidecodebin.c:
3434 uridecodebin: Don't post missing plugin messages twice
3435 decodebin2 already posts them after emitting the unknown-type signal,
3436 there's no need to post another one.
3438 2009-09-26 12:17:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3440 * gst/playback/gstdecodebin2.c:
3441 decodebin2: Rewrite autoplugging and how groups of pads are exposed
3442 This now keeps track of everything that is going on, creates
3443 a tree of chains and groups to allow "demuxer after demuxer" scenarios
3444 and allows chained Oggs with multiple streams (needs oggdemux or playbin2 fixes).
3445 Also document everything in detail and give a general overview of what
3446 decodebin2 is doing at the top of the sources.
3447 Fixes bug #596183, #563828 and #591677.
3449 2009-10-07 17:45:33 +0300 Stefan Kost <ensonic@users.sf.net>
3451 * sys/ximage/ximagesink.c:
3452 ximagesink: only start event thread if needed
3453 The event thread is doing 20 wakeups per second to poll the events. If one
3454 runs ximagesink with handle-events=false and handle-expose=false then we can
3455 avoid the extra thread.
3457 2009-10-07 16:56:28 +0200 Edward Hervey <bilboed@bilboed.com>
3459 * ext/theora/theoraenc.c:
3460 theoraenc: Make the default quality property 48.
3461 This guarantees that people who use theoraenc without modifying any
3462 properties will end up with a reasonably good quality output.
3463 48 is also the default of the encoder_example application shipped with
3466 2009-10-07 11:48:37 +0200 Benjamin Otte <otte@gnome.org>
3468 * tests/check/libs/video.c:
3469 tests/check/libs/video.c: Update strides for Y41B
3471 2009-10-07 10:32:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3473 * gst-libs/gst/rtsp/gstrtspconnection.c:
3474 rtspconnection: we can use GLib 2.18 API unconditionally now
3476 2009-10-07 10:13:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3479 configure: bump GLib requirement to 2.18
3480 Bump required GLib version as per the release planning docs.
3482 2009-10-05 00:33:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3484 * gst-libs/gst/interfaces/tuner.c:
3485 docs: clarify GstTuner docs in two places
3487 2009-09-25 15:32:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3489 * sys/v4l/gstv4lelement.c:
3490 v4l: fix compiler warning
3491 Fix 'variable may be used uninitialized' compiler warning (which is
3492 true in theory, but can't actually ever happen, since we always
3493 call the function with check=FALSE).
3496 2009-10-07 11:56:35 +0300 Stefan Kost <ensonic@users.sf.net>
3498 * ext/gnomevfs/gstgnomevfssrc.c:
3499 * ext/ogg/gstogmparse.c:
3500 * gst/subparse/gstsubparse.c:
3501 * gst/subparse/mpl2parse.c:
3502 * gst/subparse/tmplayerparse.c:
3503 build: sprintf, sscanf need stdio.h
3505 2009-09-15 15:26:06 +0300 Stefan Kost <ensonic@users.sf.net>
3507 * sys/xvimage/xvimagesink.c:
3508 xvimagesink: only start event thread if needed
3509 The event thread is doing 20 wakeups per second to poll the events. If one runs
3510 xvimagesink with handle-events=false and handle-expose=false then we can avoid
3513 2009-10-07 09:58:27 +0200 Benjamin Otte <otte@gnome.org>
3515 * gst-libs/gst/video/video.h:
3516 Update Since tags for NV12/NV21
3517 They are added in 0.10.26 now, not 0.10.25
3519 2009-09-23 15:31:50 +0200 Benjamin Otte <otte@gnome.org>
3521 * gst/videotestsrc/videotestsrc.c:
3522 [videotestsrc] Make checkers-8 pattern create 8x8 instead of 16x16 tiles
3524 2009-09-23 11:03:57 +0200 Benjamin Otte <otte@gnome.org>
3526 * gst/ffmpegcolorspace/imgconvert_template.h:
3527 [ffmpegcolorspace] Fix NV12 and NV21 with odd width and height
3529 2009-09-23 10:25:02 +0200 Benjamin Otte <otte@gnome.org>
3531 * gst-libs/gst/video/video.c:
3532 * gst-libs/gst/video/video.h:
3533 Add NV12 and NV21 formats
3535 2009-09-21 18:49:42 +0200 Benjamin Otte <otte@gnome.org>
3537 * gst-libs/gst/video/video.c:
3539 Chroma components should be aligned on 4byte boundaries.
3540 https://bugzilla.gnome.org/show_bug.cgi?id=595849
3542 2009-09-21 18:49:06 +0200 Benjamin Otte <otte@gnome.org>
3544 * gst/videotestsrc/videotestsrc.c:
3545 [videotestsrc] Fix Y41B
3546 Chroma components should be aligned on 4byte boundaries.
3547 https://bugzilla.gnome.org/show_bug.cgi?id=595849
3549 2009-10-07 07:28:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3552 * gst-libs/gst/interfaces/streamvolume.c:
3553 streamvolume: Define cbrt() if it's not available
3554 Fixes build on Win32, bug #597537.
3556 2009-09-24 16:05:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3558 * gst/playback/gstfactorylists.c:
3559 factorylist: Use gst_caps_can_intersect() instead of _intersect()
3560 This is faster and results in less allocations.
3562 2009-09-26 12:10:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3564 * gst/playback/gstdecodebin2.c:
3565 decodebin2: Don't set the external ghostpads blocked but only their targets
3566 Pad blocks should never be done on external pads as outside elements
3567 might want to use their own pad blocks on them and this will lead to
3568 conflicts and deadlocks.
3570 2009-09-26 12:04:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3572 * gst/playback/gstdecodebin2.c:
3573 decodebin2: Only use the object lock for protecting the subtitle elements
3574 Using the decodebin lock will result in deadlocks if the subtitle encoding
3575 is accessed from a pad-added handler.
3577 2009-09-26 18:11:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3579 * gst/playback/gstplaybin2.c:
3580 playbin2: Improve debugging of pad blocks
3582 2009-09-23 16:07:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3584 * gst/playback/gstplaybin2.c:
3585 * gst/playback/gstplaysink.c:
3586 playbin2/playsink: Use gst_object_ref_sink() instead of calling both separately
3588 2009-10-06 19:59:11 -0700 David Schleef <ds@schleef.org>
3591 configure: Add an 'else' to pangocairo check
3592 Otherwise it exits if it fails.
3594 2009-10-06 19:35:50 -0700 David Schleef <ds@schleef.org>
3596 * gst/videotestsrc/gstvideotestsrc.c:
3597 * gst/videotestsrc/gstvideotestsrc.h:
3598 * gst/videotestsrc/videotestsrc.c:
3599 * gst/videotestsrc/videotestsrc.h:
3600 videotestsrc: add pattern with out-of-gamut colors
3601 Adds a pattern with out-of-gamut colors in a checkerboard
3602 pattern with in-gamut neighbors. Useful for checking YCbCr->RGB
3603 color matrixing. Correct matrixing and clamping will cause the
3604 checkerboard pattern to be invisible.
3606 2009-10-06 19:17:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3608 * gst-libs/gst/rtsp/gstrtspconnection.c:
3609 rtsp: use CLOSE_SOCKET() instead of close()
3610 Use CLOSE_SOCKET instead of directly calling close() because it does the right
3614 2009-10-01 14:19:41 +0200 Robert Swain <robert swain gmail com>
3616 * gst/audioresample/gstaudioresample.c:
3617 audioresample: fix printf variable type
3618 Change printf variable type from %lu to %" G_GUINT64_FORMAT " as it
3619 should be for guint64.
3622 2009-09-30 23:22:35 +0100 Jan Schmidt <thaytan@noraisin.net>
3624 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
3625 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
3626 ffmpegcolorspace: Use the ffmpegcolorspace debug category
3627 Move gstffmpegcodecmap debug to the ffmpegcolorspace category
3629 2009-09-22 11:58:26 +0100 Jan Schmidt <thaytan@noraisin.net>
3631 * gst/gdp/gstgdppay.c:
3632 gdppay: Don't repeat tags buffers for every new segment
3633 Only send a tag buffer when one is received, not after every new segment
3636 2009-09-28 20:25:35 -0700 David Schleef <ds@schleef.org>
3638 * gst/typefind/gsttypefindfunctions.c:
3639 typefind: detect 'ftypqt ' as video/quicktime
3641 2009-10-06 19:47:00 +0100 Jan Schmidt <thaytan@noraisin.net>
3644 back to development -> 0.10.25.1
3646 === release 0.10.25 ===
3648 2009-10-05 13:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
3654 * docs/plugins/gst-plugins-base-plugins.args:
3655 * docs/plugins/gst-plugins-base-plugins.hierarchy:
3656 * docs/plugins/gst-plugins-base-plugins.interfaces:
3657 * docs/plugins/gst-plugins-base-plugins.prerequisites:
3658 * docs/plugins/gst-plugins-base-plugins.signals:
3659 * docs/plugins/inspect/plugin-adder.xml:
3660 * docs/plugins/inspect/plugin-alsa.xml:
3661 * docs/plugins/inspect/plugin-app.xml:
3662 * docs/plugins/inspect/plugin-audioconvert.xml:
3663 * docs/plugins/inspect/plugin-audiorate.xml:
3664 * docs/plugins/inspect/plugin-audioresample.xml:
3665 * docs/plugins/inspect/plugin-audiotestsrc.xml:
3666 * docs/plugins/inspect/plugin-cdparanoia.xml:
3667 * docs/plugins/inspect/plugin-decodebin.xml:
3668 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
3669 * docs/plugins/inspect/plugin-gdp.xml:
3670 * docs/plugins/inspect/plugin-gio.xml:
3671 * docs/plugins/inspect/plugin-gnomevfs.xml:
3672 * docs/plugins/inspect/plugin-libvisual.xml:
3673 * docs/plugins/inspect/plugin-ogg.xml:
3674 * docs/plugins/inspect/plugin-pango.xml:
3675 * docs/plugins/inspect/plugin-playback.xml:
3676 * docs/plugins/inspect/plugin-queue2.xml:
3677 * docs/plugins/inspect/plugin-subparse.xml:
3678 * docs/plugins/inspect/plugin-tcp.xml:
3679 * docs/plugins/inspect/plugin-theora.xml:
3680 * docs/plugins/inspect/plugin-typefindfunctions.xml:
3681 * docs/plugins/inspect/plugin-uridecodebin.xml:
3682 * docs/plugins/inspect/plugin-video4linux.xml:
3683 * docs/plugins/inspect/plugin-videorate.xml:
3684 * docs/plugins/inspect/plugin-videoscale.xml:
3685 * docs/plugins/inspect/plugin-videotestsrc.xml:
3686 * docs/plugins/inspect/plugin-volume.xml:
3687 * docs/plugins/inspect/plugin-vorbis.xml:
3688 * docs/plugins/inspect/plugin-ximagesink.xml:
3689 * docs/plugins/inspect/plugin-xvimagesink.xml:
3690 * gst-plugins-base.doap:
3693 2009-10-05 13:49:10 +0100 Jan Schmidt <thaytan@noraisin.net>
3729 2009-10-01 17:17:55 +0100 Jan Schmidt <thaytan@noraisin.net>
3765 0.10.24.4 pre-release
3767 2009-10-01 10:37:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3769 * ext/pango/gsttextoverlay.c:
3770 * ext/pango/gsttextrender.c:
3771 pango: Unpremultiply Cairo's ARGB to match GStreamers ARGB
3773 2009-09-28 22:06:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3775 * gst/playback/gstplaysink.c:
3776 playsink: make the lock recursive for now
3779 2009-09-28 21:54:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3781 * gst/playback/gstplaysink.c:
3782 playsink: fix the vis property getter
3784 2009-09-30 18:06:56 +0100 Christian F.K. Schaller <christian.schaller@collabora.co.uk>
3786 * gst-plugins-base.spec.in:
3787 Add missing file to spec file
3789 2009-09-17 16:57:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3791 * gst-libs/gst/cdda/gstcddabasesrc.c:
3792 * tests/check/libs/cddabasesrc.c:
3793 cddabasesrc: Fix string leaks in the unit test and a leak in cddabasesrc
3795 2009-09-17 23:42:52 +1000 Jonathan Matthew <jonathan@d14n.org>
3797 * gst-libs/gst/cdda/gstcddabasesrc.c:
3798 * tests/check/libs/cddabasesrc.c:
3799 cddabasesrc: ignore URI fragments that look like device paths
3800 Rhythmbox uses cdda:// URIs of the form cdda://track#device, which
3801 worked before the fix for bug #321532.
3802 Also adds a check for negative track numbers and some unit tests for URI
3806 2009-09-17 01:20:45 +0100 Jan Schmidt <thaytan@noraisin.net>
3841 0.10.24.3 pre-release
3843 2009-09-15 15:23:49 -0700 Michael Smith <msmith@songbirdnest.com>
3845 * gst-libs/gst/tag/gstvorbistag.c:
3846 vorbistag: don't ever return NULL in list of strings.
3848 2009-09-14 12:18:33 +0200 Edward Hervey <bilboed@bilboed.com>
3850 * gst/playback/gstplaysink.c:
3851 playsink: Expose mute,volume,vis-plugin and font-desc properties
3852 https://bugzilla.gnome.org/show_bug.cgi?id=594623
3854 2009-09-09 12:42:04 +0200 Edward Hervey <bilboed@bilboed.com>
3856 * gst/playback/gstplaysink.c:
3857 GstPlaySink: Expose 'reconfigure' as an action signal.
3859 2009-09-09 11:17:28 +0200 Edward Hervey <bilboed@bilboed.com>
3861 * gst/playback/gstplaysink.c:
3862 GstPlaySink: Expose flags as a gobject property.
3864 2009-09-08 11:35:20 +0200 Edward Hervey <bilboed@bilboed.com>
3866 * gst/playback/gstplayback.c:
3867 * gst/playback/gstplaysink.c:
3868 * gst/playback/gstplaysink.h:
3869 playback: Register playsink as an element.
3870 This allows using playsink from outside the playback plugin.
3871 Add code to be able to request the sink pads using standard GStreamer API.
3872 TODO : expose GObject properties/signals.
3874 2009-09-12 14:55:06 +0300 Stefan Kost <ensonic@users.sf.net>
3876 * docs/libs/gst-plugins-base-libs.types:
3877 docs: add new gst_stream_volume_get_type to types file
3878 This is needs to get Gobject features to show up in the docs.
3880 2009-09-12 15:48:11 -0700 David Schleef <ds@schleef.org>
3882 * ext/ogg/gstoggdemux.c:
3883 oggdemux: Fix duration calculation for truncated files
3884 If the last page of a stream has a granulepos of -1, that is,
3885 it doesn't complete a packet, we need to continue to search
3886 for the last granulepos.
3888 2009-09-12 14:01:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3891 * gst-libs/gst/app/Makefile.am:
3892 * gst-libs/gst/audio/Makefile.am:
3893 * gst-libs/gst/cdda/Makefile.am:
3894 * gst-libs/gst/fft/Makefile.am:
3895 * gst-libs/gst/interfaces/Makefile.am:
3896 * gst-libs/gst/netbuffer/Makefile.am:
3897 * gst-libs/gst/pbutils/Makefile.am:
3898 * gst-libs/gst/riff/Makefile.am:
3899 * gst-libs/gst/rtp/Makefile.am:
3900 * gst-libs/gst/rtsp/Makefile.am:
3901 * gst-libs/gst/sdp/Makefile.am:
3902 * gst-libs/gst/tag/Makefile.am:
3903 * gst-libs/gst/video/Makefile.am:
3904 introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH
3905 This way g-ir-scanner can find the gstreamer-*-0.10 pkg-config files.
3907 2009-09-12 02:23:07 +0100 Jan Schmidt <thaytan@noraisin.net>
3909 * ext/theora/theoraenc.c:
3910 theoraenc: Fix a string leak in _getcaps()
3912 2009-09-11 23:49:11 +0100 Jan Schmidt <thaytan@noraisin.net>
3949 0.10.24.2 pre-release
3951 2009-09-11 21:44:18 +0100 Jan Schmidt <thaytan@noraisin.net>
3953 * tests/check/elements/audioresample.c:
3954 check: Improve audioresample test
3955 Make the audioresample test work with CK_FORK=no, and
3956 turn a g_print into a GST_INFO.
3958 2009-09-11 22:09:06 +0200 Benjamin Otte <otte@gnome.org>
3960 * gst/videotestsrc/videotestsrc.c:
3961 videotestsrc: Fix crashes with even widths
3962 The fix for green lines introduced by commit
3963 35fdfcc6258c66ba462a4330a35deffb0f2b501d caused invalid memory accesses
3964 for even widths. This patch fixes it.
3966 2009-09-11 15:11:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3968 * gst/playback/gstplaybin2.c:
3969 playbin2: Implement GstStreamVolume interface
3971 2009-09-11 15:04:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3973 * gst/volume/gstvolume.c:
3974 * gst/volume/gstvolume.h:
3975 * tests/check/Makefile.am:
3976 * tests/check/elements/volume.c:
3977 volume: Implement GstStreamVolume interface
3979 2009-09-11 14:54:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3981 * docs/libs/gst-plugins-base-libs-docs.sgml:
3982 * docs/libs/gst-plugins-base-libs-sections.txt:
3983 * gst-libs/gst/interfaces/Makefile.am:
3984 * gst-libs/gst/interfaces/streamvolume.c:
3985 * gst-libs/gst/interfaces/streamvolume.h:
3986 * gst/playback/Makefile.am:
3987 * win32/common/libgstinterfaces.def:
3988 interfaces: API: Add GstStreamVolume interface
3991 2009-09-11 12:20:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3993 * gst-libs/gst/rtsp/gstrtspconnection.c:
3994 rtsp: properly fix the HTTP manual mode
3995 When we're not parsing HTTP, return EPARSE when we get an HTTP
3998 2009-09-11 10:16:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4000 * gst-libs/gst/interfaces/mixertrack.h:
4001 mixertrack: add READONLY and WRITEONLY flags
4002 Should really have been READABLE and WRITABLE, but those are hard to
4003 add whilst maintaining backwards compatibility. See #343615.
4004 API: GST_MIXER_TRACK_READONLY
4005 API: GST_MIXER_TRACK_WRITEONLY
4007 2009-09-11 10:02:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4009 * gst-libs/gst/audio/gstringbuffer.c:
4010 ringbuffer: fix build against core that has debugging disabled
4011 The macro is called GST_DISABLE_GST_DEBUG, not GST_DISABLE_DEBUG.
4013 2009-09-11 07:38:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4015 * gst/videorate/gstvideorate.c:
4016 videorate: Add Since marker for the new skip-to-first property
4018 2009-09-11 07:36:10 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
4020 * gst/videorate/gstvideorate.c:
4021 * gst/videorate/gstvideorate.h:
4022 videorate: Make videorate work with a live source
4023 Add a property that makes videorate skip to the first buffer it
4024 receives instead of padding the stream from segment start to the
4028 2009-09-11 07:20:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4030 * gst-libs/gst/fft/gstfft.h:
4031 * gst-libs/gst/fft/gstfftf32.h:
4032 * gst-libs/gst/fft/gstfftf64.h:
4033 * gst-libs/gst/fft/gstffts16.h:
4034 * gst-libs/gst/fft/gstffts32.h:
4035 fft: Mark one function as const and add notes that the structs should be private in 0.11
4037 2009-09-10 22:28:19 +0300 Stefan Kost <ensonic@users.sf.net>
4039 * gst-libs/gst/audio/gstringbuffer.c:
4040 ringbuffer: add human readable format names when logging
4041 Add string array with human readable names for format and type to be used in log
4044 2009-09-10 18:19:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4046 * gst-libs/gst/rtp/gstbasertppayload.c:
4047 basertppay: don't print RTP timestamps as clocktime
4048 Don't try to print the RTP timestamp as a GstClockTime, it's just a guint32.
4051 2009-09-10 16:55:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4053 * gst/playback/gstplaybin.c:
4054 * gst/playback/gstplaybin2.c:
4055 playbin(2): Document that the volume property uses a linear scale
4058 2009-09-10 14:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4060 * gst-libs/gst/rtsp/gstrtspconnection.c:
4061 rtsp: don't return EPARSE
4062 Don't blindly return EPARSE when http mode is disabled.
4063 Restore old http mode after temporarily setting it to TRUE.
4065 2009-09-10 12:38:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4067 * gst-libs/gst/audio/gstbaseaudiosink.c:
4068 baseaudiosink: add ugly backward compat hack
4069 Check for pulsesink < 0.10.17 because it includes code that is now included in
4070 baseaudiosink. Disable that code in baseaudiosink to be compatible with the
4073 2009-09-10 10:56:29 +0200 Benjamin Otte <otte@gnome.org>
4075 * gst/ffmpegcolorspace/imgconvert.c:
4076 ffmpegcolorspace: Handle YVU9/YUV9 conversion with odd widths
4077 A green border could be visible when converting to Y444 or RGB, because
4078 the last chroma samples weren't copied correctly
4080 2009-09-10 10:43:37 +0200 Benjamin Otte <otte@gnome.org>
4082 * gst/videotestsrc/videotestsrc.c:
4083 videotestsrc: Fix YVU9 and YUV9
4084 - Buffer sizes were computed different from ffmpegcolorspace
4085 - Green bar on right size for widths not divisable by 4
4087 2009-09-10 10:08:28 +0200 Benjamin Otte <otte@gnome.org>
4089 * gst/videotestsrc/videotestsrc.c:
4090 videotestsrc: Fix image for odd widths in some formats
4091 videotestsrc rounds chroma down. This causes it to omit the last chroma
4092 value completely for odd widths when the chroma is downsampled.
4093 This patch special cases the last pixel to not be rounded down.
4095 2009-09-10 10:02:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4097 * ext/ogg/gstoggdemux.c:
4098 oggdemux: Handle kate and cmml as sparse streams too
4100 2009-09-10 10:00:16 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4102 * ext/ogg/gstoggdemux.c:
4103 * ext/ogg/gstoggdemux.h:
4104 oggdemux: Better handling of sparse streams by sending segment updates
4107 2009-09-10 09:43:28 +0300 Stefan Kost <ensonic@users.sf.net>
4109 * gst/playback/gsturidecodebin.c:
4110 docs: tell a biit more about uri-decodebin and buffering
4112 2009-09-09 18:24:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4114 * gst-libs/gst/audio/gstbaseaudiosink.c:
4115 baseaudiosink: take clock time in setcaps
4116 Take the time of the clock so that the last_time field is set. This is important
4117 for sinks that restart their internal ringbuffer after a caps change and need to
4118 know the last know position.
4120 2009-09-09 18:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4122 * gst-libs/gst/audio/gstaudioclock.c:
4123 audioclock: add some more debug
4125 2009-09-09 16:44:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4127 * ext/theora/theoraenc.c:
4128 theoraenc: Print a debug message with supported formats
4130 2009-09-07 17:29:38 +0200 Benjamin Otte <otte@gnome.org>
4132 * ext/theora/theoraenc.c:
4133 theora: Check supported input formats in getcaps function
4134 We want to fail early when an older libtheora release is used that does
4135 not support Y444 or Y42B formats, so use a getcaps function that does
4138 2009-09-04 21:37:04 +0200 Benjamin Otte <otte@gnome.org>
4140 * ext/theora/theoraenc.c:
4141 theora: Implement support in theoraenc for Y444 and Y42B
4144 2009-09-04 20:23:52 +0200 Benjamin Otte <otte@gnome.org>
4146 * ext/theora/theoraenc.c:
4147 theora: Refactor the buffer copy code
4149 2009-09-04 16:59:49 +0200 Benjamin Otte <otte@gnome.org>
4151 * ext/theora/theoraenc.c:
4152 theora: Split yuv_buffer creation into its own function
4154 2009-09-04 16:49:08 +0200 Benjamin Otte <otte@gnome.org>
4156 * ext/theora/theoraenc.c:
4157 theora: Split out buffer resize in its own function
4159 2009-09-04 14:06:09 +0200 Benjamin Otte <otte@gnome.org>
4161 * ext/theora/theoraenc.c:
4162 theora: Add assertions that functions don't fail
4163 Some functions in libtheora can return an error, but that error cannot
4164 ever happen inside theoraenc. In those cases assert that it doesn't.
4166 2009-09-09 16:21:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4168 * tests/examples/seek/seek.c:
4169 seek: make stop state configurable
4170 Make it easy to experiment with different stop states (NULL and READY)
4172 2009-09-09 16:19:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4174 * gst-libs/gst/audio/gstbaseaudiosink.c:
4175 baseaudiosink: correct for clock reset
4176 When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
4177 also make sure that the clock is updated with the elapsed time so that it
4178 alsways increments even when the ringbuffer goes back to 0. When this happened
4179 we need to adjust the sample position for the reset ringbuffer.
4182 2009-09-09 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4184 * gst-libs/gst/audio/gstbaseaudiosink.h:
4185 baseaudiosink: whitespace fixes
4187 2009-09-09 16:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4189 * gst-libs/gst/audio/gstringbuffer.c:
4190 ringbuffer: add more debug
4192 2009-09-09 10:25:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4194 * gst-libs/gst/interfaces/colorbalance.h:
4195 * gst-libs/gst/interfaces/mixer.h:
4198 2009-09-08 17:59:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4200 * gst-libs/gst/video/gstvideosink.c:
4201 * gst-libs/gst/video/gstvideosink.h:
4202 videosink: add "show-preroll-frame" property
4203 Add a property to disable rendering of video frames during preroll. This
4204 will only work for videosinks that use the new ::show_frame() vfunc instead
4205 of overriding basesink's preroll and render vfuncs directly.
4206 API: GstVideoSink:show-preroll-frame
4208 2009-09-08 17:43:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4210 * sys/ximage/ximagesink.c:
4211 * sys/xvimage/xvimagesink.c:
4212 ximagesink, xvimagesink: use new GstVideoSink::show_frame() vfunc
4214 2009-09-08 18:19:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4216 * gst-libs/gst/video/gstvideosink.c:
4217 * gst-libs/gst/video/gstvideosink.h:
4218 video: add GstVideoSinkClass::show_frame()
4219 Add ::show_frame() vfunc which maps to basesink's ::preroll and ::render
4220 vfuncs and add some gtk-doc chunks.
4221 API: GstVideoSinkClass::show_frame()
4223 2009-09-08 16:00:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4225 * gst-libs/gst/interfaces/navigation.c:
4226 navigation: don't do stuff inside g_return_val_if_fail() statements
4227 Or it will all fall apart if someone compiles with -DG_DISABLE_ASSERT.
4229 2009-08-31 20:24:22 +0200 Havard Graff <havard.graff@tandberg.com>
4231 * gst-libs/gst/interfaces/navigation.c:
4232 navigation: Fix compiler warning with MSVC
4235 2009-08-31 20:31:56 +0200 Havard Graff <havard.graff@tandberg.com>
4237 * gst-libs/gst/rtp/gstbasertpdepayload.c:
4238 basertpdepayload: fix event forwarding
4240 2009-08-31 20:36:37 +0200 Havard Graff <havard.graff@tandberg.com>
4242 * gst-libs/gst/rtp/gstrtcpbuffer.c:
4243 rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
4246 2009-09-08 13:02:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4248 * gst/playback/gstplaybin2.c:
4249 * gst/playback/gstplaysink.c:
4250 * gst/playback/gstplaysink.h:
4253 2009-09-08 12:59:20 +0200 Håvard Graff <havard.graff@tandberg.com>
4255 * gst-libs/gst/audio/gstbaseaudiosrc.c:
4256 baseaudiosrc: improve slave skew resync
4257 The old one did the mistake of not actually advancing the ringbuffer, it just
4258 adjusted the segbase, introducing the whole lenght of the ringbuffer as an
4259 extra delay in the pipeline.
4260 Also make sure that the resync can never go back in time, producing the same
4261 timestamps that has already been produced, as this can cause severe problems
4262 for sinks and other synching mechanisms.
4265 2009-09-07 17:13:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4267 * gst/typefind/gsttypefindfunctions.c:
4268 typefinding: disable typefinder for headerless flac
4269 Disable headerless flac typefinder as long as it happily typefinds anything
4270 including /dev/urandom as flac and as long as it's not particularly useful
4271 given that such streams don't really exist in the wild.
4272 Also fix up some comments so that gtk-doc doesn't complain about them.
4274 2009-09-06 15:21:43 +0300 René Stadler <mail@renestadler.de>
4276 * sys/ximage/ximagesink.c:
4277 ximagesink: fix small memory leak when setting window title
4279 2009-09-06 01:42:42 +0300 René Stadler <mail@renestadler.de>
4281 * sys/xvimage/xvimagesink.c:
4282 xvimagesink: fix small memory leak when setting window title
4284 2009-09-05 13:55:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4287 introspection: Add *.gir and *.typelib to .gitignore
4289 2009-09-05 13:46:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4291 * gst-libs/gst/app/Makefile.am:
4292 * gst-libs/gst/audio/Makefile.am:
4293 * gst-libs/gst/interfaces/Makefile.am:
4294 * gst-libs/gst/pbutils/Makefile.am:
4295 * gst-libs/gst/rtsp/Makefile.am:
4296 * gst-libs/gst/video/Makefile.am:
4297 introduction: Fix out-of-tree build
4299 2009-09-05 13:13:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4301 * gst-libs/gst/rtsp/Makefile.am:
4302 rtsp: Fix introspection build by ordering sources/headers in dependency order
4304 2009-09-05 13:09:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4306 * gst-libs/gst/audio/Makefile.am:
4307 audio: Remove debug echo
4309 2009-09-05 13:08:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4311 * gst-libs/gst/audio/Makefile.am:
4312 audio: Fix build of introspection data by using dependency order for the headers/sources
4314 2009-09-05 12:31:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4316 * gst-libs/gst/app/Makefile.am:
4317 * gst-libs/gst/audio/Makefile.am:
4318 * gst-libs/gst/cdda/Makefile.am:
4319 * gst-libs/gst/fft/Makefile.am:
4320 * gst-libs/gst/interfaces/Makefile.am:
4321 * gst-libs/gst/netbuffer/Makefile.am:
4322 * gst-libs/gst/pbutils/Makefile.am:
4323 * gst-libs/gst/riff/Makefile.am:
4324 * gst-libs/gst/rtp/Makefile.am:
4325 * gst-libs/gst/rtsp/Makefile.am:
4326 * gst-libs/gst/sdp/Makefile.am:
4327 * gst-libs/gst/tag/Makefile.am:
4328 * gst-libs/gst/video/Makefile.am:
4329 introspection: Strip Gst prefix from all types/functions
4331 2009-09-05 11:49:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4333 * gst-libs/gst/Makefile.am:
4334 * gst-libs/gst/app/Makefile.am:
4335 * gst-libs/gst/audio/Makefile.am:
4336 * gst-libs/gst/fft/Makefile.am:
4337 * gst-libs/gst/interfaces/Makefile.am:
4338 * gst-libs/gst/netbuffer/Makefile.am:
4339 * gst-libs/gst/pbutils/Makefile.am:
4340 * gst-libs/gst/riff/Makefile.am:
4341 * gst-libs/gst/rtp/Makefile.am:
4342 * gst-libs/gst/rtsp/Makefile.am:
4343 * gst-libs/gst/sdp/Makefile.am:
4344 * gst-libs/gst/tag/Makefile.am:
4345 * gst-libs/gst/video/Makefile.am:
4346 introspection: Fix build if gir-repository is not installed
4348 2009-09-05 11:37:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4350 * gst-libs/gst/video/Makefile.am:
4351 video: Add gobject-introspection support
4353 2009-09-05 11:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4355 * gst-libs/gst/tag/Makefile.am:
4356 tag: Add gobject-introspection support
4358 2009-09-05 11:34:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4360 * gst-libs/gst/sdp/Makefile.am:
4361 sdp: Add gobject-introspection support
4363 2009-09-05 11:31:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4365 * gst-libs/gst/app/Makefile.am:
4366 * gst-libs/gst/audio/Makefile.am:
4367 * gst-libs/gst/interfaces/Makefile.am:
4368 * gst-libs/gst/pbutils/Makefile.am:
4369 libs: Add nodist headers and sources to the introspection files
4371 2009-09-05 11:28:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4373 * gst-libs/gst/rtsp/Makefile.am:
4374 rtsp: Add gobject-introspection support
4376 2009-09-05 11:25:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4378 * gst-libs/gst/rtp/Makefile.am:
4379 rtp: Add gobject-introspection support
4381 2009-09-05 11:23:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4383 * gst-libs/gst/riff/Makefile.am:
4384 riff: Add gobject-introspection support
4386 2009-09-05 11:20:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4388 * gst-libs/gst/pbutils/Makefile.am:
4389 pbutils: Add gobject-introspection support
4391 2009-09-05 11:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4393 * gst-libs/gst/netbuffer/Makefile.am:
4394 netbuffer: Add gobject-introspection support
4396 2009-09-05 11:15:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4398 * gst-libs/gst/interfaces/Makefile.am:
4399 interfaces: Add gobject-introspection support
4401 2009-09-05 11:04:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4403 * gst-libs/gst/fft/Makefile.am:
4404 fft: Add gobject-introspection support
4406 2009-09-05 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4408 * gst-libs/gst/cdda/Makefile.am:
4409 cdda: Add gobject-introspection support
4410 This is disabled for now until gobject-introspection is fixed
4412 2009-09-05 10:50:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4414 * gst-libs/gst/audio/Makefile.am:
4415 audio: Add gobject-introspection support
4417 2009-09-05 10:40:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4420 * gst-libs/gst/app/Makefile.am:
4421 app: Add gobject-introspection support
4423 2009-09-05 10:20:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4426 Automatic update of common submodule
4427 From 00a859e to 19fa4f3
4429 2009-09-04 15:48:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4431 * gst/typefind/gsttypefindfunctions.c:
4432 typefind: fix midi typefinding
4433 We already have a audio/midi typefinder so don't override it with the midi in
4434 RIFF typefinder or else we fail to detect plain midi files.
4436 2009-09-04 11:29:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4438 * gst/playback/gsturidecodebin.c:
4439 uridecodebin: do buffering for more uris
4440 Add ssh://, ftp://, sftp://, myth:// to the list of uris that require
4444 2009-09-04 07:36:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4446 * gst/typefind/gsttypefindfunctions.c:
4447 typefindfunctions: Add typefinder for Midi inside RIFF
4448 This is a standard Midi file format that should be supported by
4449 all Midi decoders and also has the mimetype audio/mid according to
4450 the Midi specification homepage.
4453 2009-09-03 18:53:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4455 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
4456 audiortppay: add some debugging
4458 2009-09-03 17:53:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4460 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
4461 audiortppay: handle gaps
4462 Add various conversion functions between time<->bytes<->rtptime that will be
4464 Refactor the min/max packet length code so that it can be used for both
4465 sample/frame based payloaders. Cache the returned values.
4467 When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
4468 same gap as the GStreamer timestamps gap.
4470 2009-09-03 14:13:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4472 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
4473 audiortppay: fix frame duration calculations
4474 Fix the calculation of the frame duration and rtp timestamps.
4477 2009-09-03 14:13:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4479 * gst-libs/gst/rtp/gstbasertppayload.c:
4480 rtppay: add some debugging
4482 2009-09-02 19:49:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4484 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
4485 audiortppay: use offsets for RTP timestamps
4486 Have a custom sample/frame function to generate an offset that the base class
4487 will use for generating RTP timestamps. This results in perfect RTP timestamps
4488 on the output buffers.
4489 Refactor setting metadata on output buffers.
4490 Add some more functionality to _flush().
4491 Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
4492 the next outgoing buffer.
4493 Flush the pending data on EOS.
4495 2009-09-02 13:13:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4497 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
4498 audiortppay: move function around
4500 2009-09-02 13:12:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4502 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
4503 audiortppay: fix sample duration calculation
4505 2009-09-02 12:24:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4507 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
4508 audiortppay: more refactoring
4509 Unify the sample/frame buffer handling code by making the functions plugable.
4511 2009-09-02 12:03:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4513 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
4514 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
4515 audiortppayload: refactor some more
4516 Refactor getting the packet min/max size and alignment code.
4517 Refactor converting bytes to time.
4518 change some variable to something shorter.
4520 2009-09-02 10:46:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4522 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
4523 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
4524 * win32/common/libgstrtp.def:
4525 audiortppayload: refactor and cleanup
4526 Always use the adapter when we need to fragment the incomming buffer. Use more
4527 modern adapter functions to avoid malloc and memcpy. The overall result is that
4528 the code looks cleaner while it should be equally fast and in some case avoid a
4530 Use the adapter timestamping functions for more precise timestamps in case of
4532 Cache some values instead of recalculating them.
4533 Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
4534 the internal adapter.
4535 API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
4537 2009-09-03 16:56:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4542 2009-09-03 11:29:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4544 * gst-libs/gst/rtp/gstbasertppayload.c:
4545 basertppay: add property to disable perfect RTP time
4546 Add a property to disable the generation of perfect RTP timestamps. By default
4548 API: GstBaseRTPPayload::perfect-rtptime
4550 2009-09-02 19:47:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4552 * gst-libs/gst/rtp/gstbasertppayload.c:
4553 basertppay: allow subclasses to influence RTP time
4554 Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
4555 which RTP timestamps are generated. Usually timestamps are created from the
4556 GStreamer timestamps on the buffer, which could result in imperfect RTP
4559 2009-09-02 19:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4561 * gst-libs/gst/rtp/gstbasertppayload.h:
4562 basertppay: add macro to cast
4564 2009-09-01 18:26:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4566 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
4567 audiopayload: code cleanups
4569 2009-09-01 18:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4571 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
4572 audiortppayload: don't check adapter
4573 the adapter is never NULL so we don't need to check it.
4574 Use _scale functions to avoid overflows.
4576 2009-09-03 00:14:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4579 * gst/typefind/Makefile.am:
4580 * gst/typefind/gsttypefindfunctions.c:
4581 typefinding: move gio-based xdg mime typefinder from -bad to -base
4582 Its purposes is mainly to avoid false positives (e.g. mp3 typefinder
4583 reporting a 20% probability and somesuch). Won't be registered if
4584 the gio plugin has been disabled via ./configure --disable-gio.
4586 2009-09-01 15:06:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4588 * gst/subparse/gstsubparse.c:
4589 subparse: GstAdapter is not a GstObject and should be freed with g_object_unref
4591 2009-09-01 15:02:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4593 * sys/v4l/v4lsrc_calls.c:
4594 v4lsrc: fix timestamping for when we do not have a clock yet
4597 2009-09-01 14:30:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4599 * sys/v4l/v4lsrc_calls.c:
4600 v4lsrc: don't log not-yet-initialised integer value
4602 2009-09-01 14:28:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4604 * sys/v4l/v4lsrc_calls.c:
4605 v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize
4606 And reflow code to be more indent friendly.
4608 2009-09-01 10:39:52 +0200 Jonas Holmberg <jonas.holmberg@axis.com>
4610 * gst-libs/gst/rtp/gstbasertppayload.c:
4611 * gst-libs/gst/rtp/gstbasertppayload.h:
4612 basertppayload: Make instance init faster by not reading /dev/urandom 3 times
4613 ... which is the default seed when creating a new GRand. Because
4614 GLib in older versions used buffered IO this would take a lot of time.
4615 Instead use the global GRand for getting random numbers and keep the
4616 three instance GRand for backward compatibility with a simple seed.
4619 2009-08-31 22:48:01 +0300 Stefan Kost <ensonic@users.sf.net>
4621 * gst/adder/gstadder.c:
4622 adder: improve caps filter functionality. Fixes #590146.
4623 Also use the capsfilter if there is no src-peer as the caps constrain what
4624 we can do. Don't create any_caps as a default, as we check for NULL to skip the
4625 filtering. This is a (small) performance regression as we always intersect
4628 2009-08-31 11:10:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4630 * gst/playback/gstdecodebin2.c:
4631 decodebin2: Post missing plugin messages before any error messages
4633 2009-08-28 19:06:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4635 * gst-libs/gst/cdda/gstcddabasesrc.c:
4636 cddabasesrc: safely handle the indexes
4638 2009-08-28 19:06:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4640 * win32/common/libgstrtsp.def:
4641 def: add new rtsp symbols
4643 2009-08-28 14:08:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4645 * gst-libs/gst/rtp/gstbasertppayload.h:
4646 basertppayload: whitespace fixes.
4648 2009-08-27 18:59:49 +0200 Marc-André Lureau <mlureau@flumotion.com>
4650 * gst/gdp/gstgdppay.c:
4651 Bug 593035 - set IN_CAPS for streamheader buffer
4653 2009-08-26 16:56:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4655 * gst/playback/gstinputselector.c:
4656 * gst/playback/gststreamselector.c:
4657 playbin: The internally linked pad of the selector might be NULL in some cases
4659 2009-08-26 16:45:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4661 * gst/playback/gstinputselector.c:
4662 * gst/playback/gststreamselector.c:
4663 playbin: Fix iterate internal linked pads functions for the stream selectors
4664 This now used the new gst_iterator_new_single() function and as a side effect
4667 2009-08-26 09:08:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4669 * gst-libs/gst/riff/riff-ids.h:
4670 * gst-libs/gst/riff/riff-read.c:
4671 riff: Add support for AVF files
4672 AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF.
4675 2009-08-26 09:08:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4677 * gst/typefind/gsttypefindfunctions.c:
4678 typefindfunctions: Detect AVF files as RIFF files too
4679 AVF is valid RIFF but has AVF0 as first fourcc instead of RIFF.
4680 Partially fixes bug #593117.
4682 2009-08-21 11:51:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4684 * tests/check/elements/audioresample.c:
4685 audioresample: Add unit test for checking for timestamp drifts
4686 This also checks for perfect timestamping and offsetting.
4688 2009-08-21 10:11:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4690 * gst/audioresample/gstaudioresample.c:
4691 audioresample: Fix drain processing
4692 In case we have to convert internally don't process output length input samples
4693 but history length input samples.
4695 2009-08-21 10:02:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4697 * tests/check/elements/audioresample.c:
4698 audioresample: Improve debugging a bit in the unit test
4700 2009-08-21 10:00:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4702 * gst/audioresample/gstaudioresample.c:
4703 audioresample: On the first buffer we need discont handling
4704 Otherwise we won't get upstream timestamps and everything and all
4705 output buffers would have -1 timestamps.
4707 2009-08-21 08:23:39 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
4710 * gst/subparse/gstsubparse.c:
4711 subparse: Remove dependency on regex.h as it's not used anyway
4714 2009-08-21 06:58:31 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
4716 * gst/audioresample/gstaudioresample.c:
4717 audioresample: Fix buffer overflow when pushing the drain
4719 2009-08-21 06:57:58 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
4721 * gst/audioresample/gstaudioresample.c:
4722 * gst/audioresample/gstaudioresample.h:
4723 audioresample: Fix timestamp drift
4726 2009-08-24 11:34:35 -0700 David Schleef <ds@schleef.org>
4728 * ext/gnomevfs/gstgnomevfssrc.c:
4729 * ext/ogg/gstogmparse.c:
4730 * ext/pango/gsttextrender.c:
4731 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
4732 * gst/playback/gstinputselector.c:
4733 * gst/playback/gststreamselector.c:
4734 * gst/subparse/gstsubparse.c:
4735 * sys/v4l/gstv4lmjpegsink.c:
4736 * sys/v4l/gstv4lmjpegsrc.c:
4737 * sys/v4l/gstv4lsrc.c:
4738 Remove Ronald Bultje from Authors field
4739 Replaced with "GStreamer maintainers
4740 <gstreamer-devel@lists.sourceforge.net>" or just removed,
4741 depending on the number of other authors.
4743 2009-08-24 15:06:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4745 * gst/playback/gstplaybin2.c:
4746 playbin2: fix refcounting of _get_sink()
4747 g_value_set_object() increases the refcount of the sink, which is not needed
4748 because the object should already be refcounted. Make sure this is always the
4749 case and use g_value_take_object().
4752 2009-08-24 14:39:16 +0200 Peter Kjellerstedt <pkj@axis.com>
4754 * gst-libs/gst/rtsp/gstrtspdefs.c:
4755 rtsp: Mark Transport as supporting multiple values.
4757 2009-08-24 13:58:17 +0200 Peter Kjellerstedt <pkj@axis.com>
4759 * gst-libs/gst/rtsp/gstrtspconnection.h:
4760 * gst-libs/gst/rtsp/gstrtspdefs.h:
4761 * gst-libs/gst/rtsp/gstrtspmessage.h:
4762 rtsp: Added missing Since tags.
4764 2009-08-24 13:27:55 +0200 Eero Nurkkala <ext-eero.nurkkala at nokia.com>
4766 * gst-libs/gst/audio/gstringbuffer.c:
4767 ringbuffer: Improve audiosink startup performance
4768 When we start the ringbuffer, immediatly continue processing samples if the
4769 writer prepared some for us.
4772 2009-08-17 11:53:43 +0200 Peter Kjellerstedt <pkj@axis.com>
4774 * gst-libs/gst/rtsp/gstrtspconnection.c:
4775 * gst-libs/gst/rtsp/gstrtspconnection.h:
4776 rtsp: Added new API for sending using GstRTSPWatch.
4777 The new API to send messages using GstRTSPWatch will first try to send the
4778 message immediately. Then, if that failed (or the message was not sent
4779 fully), it will queue the remaining message for later delivery. This avoids
4780 unnecessary context switches, and makes it possible to keep track of
4781 whether the connection is blocked (the unblocking of the connection is
4782 indicated by the reception of the message_sent signal).
4783 This also deprecates the old API (gst_rtsp_watch_queue_data() and
4784 gst_rtsp_watch_queue_message().)
4785 API: gst_rtsp_watch_write_data()
4786 API: gst_rtsp_watch_send_message()
4788 2009-08-17 11:46:32 +0200 Peter Kjellerstedt <pkj@axis.com>
4790 * gst-libs/gst/rtsp/gstrtspconnection.c:
4791 rtsp: Made gst_rtsp_watch_queue_data() thread safe.
4793 2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
4795 * gst-libs/gst/rtsp/gstrtspconnection.c:
4796 * gst-libs/gst/rtsp/gstrtspconnection.h:
4797 rtsp: Added gst_rtsp_connection_set_http_mode().
4798 With gst_rtsp_connection_set_http_mode() it is possible to tell the
4799 connection whether to allow HTTP messages to be supported. By enabling HTTP
4800 support the automatic HTTP tunnel support will also be disabled.
4801 API: gst_rtsp_connection_set_http_mode()
4803 2009-06-16 19:35:23 +0200 Peter Kjellerstedt <pkj@axis.com>
4805 * gst-libs/gst/rtsp/gstrtspconnection.c:
4806 rtsp: Allow gst_rtsp_connection_do_tunnel() to just setup decoding context.
4807 If the second connection passed to gst_rtsp_connection_do_tunnel() is NULL
4808 then just setup the base64 decoding context for the first connection.
4810 2009-06-16 19:04:54 +0200 Peter Kjellerstedt <pkj@axis.com>
4812 * gst-libs/gst/rtsp/gstrtspconnection.c:
4813 rtsp: Write as much as possible in gst_rtsp_source_dispatch().
4814 Try to write as much as possible if there are multiple messages queued.
4816 2009-06-16 18:38:02 +0200 Peter Kjellerstedt <pkj@axis.com>
4818 * gst-libs/gst/rtsp/gstrtspconnection.c:
4819 * gst-libs/gst/rtsp/gstrtspconnection.h:
4820 rtsp: Add error_full callback to GstRTSPWatchFuncs.
4821 The error_full callback is similar to the error callback, but allows for
4822 better error handling. For read errors a partial message is provided to
4823 help an RTSP server generate a more correct error response, and for write
4824 errors the write queue id of the failed message is returned.
4826 2009-08-17 18:29:17 +0200 Peter Kjellerstedt <pkj@axis.com>
4828 * gst-libs/gst/rtsp/gstrtspconnection.c:
4829 rtsp: Made read_line() support LWS.
4830 Rewrote read_line() to support LWS (Line White Space), the method used by
4831 RTSP (and HTTP) to break long lines. Also added support for \r and \n as
4832 line endings (in addition to the official \r\n).
4834 2009-08-20 14:12:50 +0200 Peter Kjellerstedt <pkj@axis.com>
4836 * gst-libs/gst/rtsp/gstrtspconnection.c:
4837 * gst-libs/gst/rtsp/gstrtspdefs.c:
4838 * gst-libs/gst/rtsp/gstrtspdefs.h:
4839 rtsp: Do not split headers which should not be split.
4840 From RFC 2068 section 4.2: "Multiple message-header fields with the same
4841 field-name may be present in a message if and only if the entire
4842 field-value for that header field is defined as a comma-separated list
4843 [i.e., #(values)]." This means that we should not split other headers which
4844 may contain a comma, e.g., Range and Date.
4846 2009-08-20 14:12:09 +0200 Peter Kjellerstedt <pkj@axis.com>
4848 * gst-libs/gst/rtsp/gstrtspconnection.c:
4849 rtsp: Parse WWW-Authenticate headers correctly.
4850 Due to the odd syntax for WWW-Authenticate (and Proxy-Authenticate) which
4851 allows commas both to separate between multiple challenges, and within the
4852 challenges themself, we need to take some extra care to split these headers
4855 2009-06-17 21:46:27 +0200 Peter Kjellerstedt <pkj@axis.com>
4857 * gst-libs/gst/rtsp/gstrtspconnection.c:
4858 rtsp: Improve parse_line().
4859 Make parse_line() handle keys with multiple values on one line correctly.
4861 2009-06-17 23:15:23 +0200 Peter Kjellerstedt <pkj@axis.com>
4863 * gst-libs/gst/rtsp/gstrtspconnection.c:
4864 rtsp: Rewrote setup_tunneling().
4865 Rewrote setup_tunneling() to use normal GstRTSPMessages instead of hard
4866 coded strings and duplicates of the message parsing code.
4868 2009-08-24 10:20:16 +0200 Peter Kjellerstedt <pkj@axis.com>
4870 * gst-libs/gst/rtsp/gstrtspconnection.c:
4871 * gst-libs/gst/rtsp/gstrtspdefs.c:
4872 * gst-libs/gst/rtsp/gstrtspdefs.h:
4873 rtsp: Rewrote gen_tunnel_reply().
4874 Rewrote gen_tunnel_reply() to generate a normal GstRTSPMessage rather
4875 than a hard coded string.
4877 2009-08-24 10:19:35 +0200 Peter Kjellerstedt <pkj@axis.com>
4879 * gst-libs/gst/rtsp/gstrtspconnection.c:
4880 rtsp: Ignore the Content-Length for POST requests.
4881 The Content-Length for POST requests with an x-sessioncookie header should
4882 be ignored as the length is bogus and only there to fool proxies.
4884 2009-06-17 20:52:48 +0200 Peter Kjellerstedt <pkj@axis.com>
4886 * gst-libs/gst/rtsp/gstrtspconnection.c:
4887 rtsp: Normalize lines (remove extra whitespace) before parsing.
4889 2009-06-10 13:11:31 +0200 Peter Kjellerstedt <pkj@axis.com>
4891 * gst-libs/gst/rtsp/gstrtspconnection.c:
4892 rtsp: Made parse_string() return a result.
4893 This will catch parsing errors when a too long string is received.
4895 2009-06-10 11:43:31 +0200 Peter Kjellerstedt <pkj@axis.com>
4897 * gst-libs/gst/rtsp/gstrtspconnection.c:
4898 rtsp: Improved parsing of messages.
4899 Do not abort message parsing as soon as there is an error. Instead parse
4900 as much as possible to allow a server to return as meaningful an error as
4903 2009-06-09 17:54:20 +0200 Peter Kjellerstedt <pkj@axis.com>
4905 * gst-libs/gst/rtsp/gstrtspconnection.c:
4906 * gst-libs/gst/rtsp/gstrtspdefs.c:
4907 * gst-libs/gst/rtsp/gstrtspdefs.h:
4908 * gst-libs/gst/rtsp/gstrtspmessage.c:
4909 * gst-libs/gst/rtsp/gstrtspmessage.h:
4910 rtsp: Added support for HTTP messages
4912 2009-06-09 16:22:17 +0200 Peter Kjellerstedt <pkj@axis.com>
4914 * gst-libs/gst/rtsp/gstrtspconnection.c:
4915 * gst-libs/gst/rtsp/gstrtspconnection.h:
4916 rtsp: Added gst_rtsp_connection_create_from_fd().
4917 API: gst_rtsp_connection_create_from_fd()
4919 2009-06-09 15:27:17 +0200 Peter Kjellerstedt <pkj@axis.com>
4921 * gst-libs/gst/rtsp/gstrtspconnection.c:
4922 rtsp: Add initial buffer support.
4923 The initial buffer contains data for a connection which should be used
4924 before starting to actually read anything from the socket.
4926 2009-08-24 13:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4928 * gst-libs/gst/app/gstappsink.c:
4929 appsink: don't block in paused
4930 When we are asked to unlock we should either leave the render function or call
4931 the wait_preroll method to release the stream lock.
4934 2009-08-24 13:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4936 * docs/libs/gst-plugins-base-libs-sections.txt:
4937 docs: fix includes for appsrc/appsink
4939 2009-08-24 11:24:27 +0200 Peter Kjellerstedt <pkj@axis.com>
4941 * gst-libs/gst/rtsp/gstrtspdefs.c:
4942 * gst-libs/gst/rtsp/gstrtspdefs.h:
4943 rtsp: Add support for the Authentication-Info header.
4944 The Authentication-Info header is defined in RFC 2617 (Digest Access
4947 2009-08-20 13:11:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4949 * ext/ogg/gstoggmux.c:
4950 * tests/check/pipelines/oggmux.c:
4951 oggmux: don't drop the streamheader field from the output caps
4952 Revert previous 'fix' for bug #588717 and fix it properly, whilst
4953 maintaining the streamheader field on the output caps. Also make
4954 sure we don't leak header buffers we couldn't push when downstream
4955 is unlinked. Add unit test for the presence of the streamheader
4956 field on the output caps and for the issue from bug #588717.
4958 2009-08-18 21:45:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4960 * gst/playback/gstinputselector.c:
4961 * gst/playback/gststreamselector.c:
4962 streamselector/inputselector: Use iterate internal links instead of deprecated get internal links
4964 2009-08-19 09:31:51 +0200 Peter Kjellerstedt <pkj@axis.com>
4966 * gst-libs/gst/rtsp/gstrtspconnection.c:
4967 rtsp: Avoid duplicated headers.
4968 Remove any existing Session and Date headers before adding new ones
4969 when sending a request. This may happen if the user of this code reuses
4970 a request (rtspsrc does this when resending after authorization fails).
4972 2009-08-18 16:49:58 +0200 Peter Kjellerstedt <pkj@axis.com>
4974 * gst-libs/gst/rtsp/gstrtspconnection.c:
4975 rtsp: Corrected the HTTP digest authorization computation.
4976 Do not use sizeof() on an array passed as an argument to a function and
4977 expect to get anything but the size of a pointer. As a result only the
4978 first 4 (or 8) bytes of the response buffer were initialized to 0 in
4979 auth_digest_compute_response() which caused it to return a string which
4980 was not NUL-terminated...
4982 2009-08-18 11:15:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4984 * gst/playback/gstplaysink.c:
4985 playsink: Also send SEEK events directly to a subpicture sink
4987 2009-08-18 08:39:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4989 * gst/playback/gstplaysink.c:
4990 playsink: If a custom text sink is used, send events to it too
4991 Before, SEEK events would be sent to the video sink, which wouldn't
4992 be linked in any way to the subtitle part of the pipeline and
4993 subparse would never see the SEEK event. This would then seek
4994 the audio/video but the subtitles would continue from the old
4998 2009-08-18 08:20:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5000 * gst/playback/gsturidecodebin.c:
5001 uridecodebin: Make missing plugins emit a warning message, not an error message
5002 The problem with an error message is, that it will stop playback completely
5003 while it could be that only a audio decoder plugin is missing and the video
5004 could be played with the available plugins.
5007 2009-08-13 17:42:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5009 * gst/playback/gsturidecodebin.c:
5010 uridecodebin: Post a correct error message for unknown types
5011 Before we had STREAM/WRONG_TYPE but it's really CORE/MISSING_PLUGIN
5012 because a plugin is missing and nothing else is wrong.
5013 Also make it an error instead of a warning.
5014 Really fixes bug #591677.
5016 2009-08-13 15:48:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5018 * gst/playback/gsturidecodebin.c:
5019 uridecodebin: Post a missing plugin message additional to the error message on unknown types
5022 2009-08-13 10:59:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5024 * gst/playback/gstplaysink.c:
5056 playbin2: fix error message string
5059 2009-08-05 15:38:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
5061 * gst-libs/gst/riff/riff-read.c:
5062 riff: align API doc of gst_riff_parse_chunk with reality
5064 2009-08-05 15:36:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
5066 * gst/playback/gstdecodebin2.c:
5067 decodebin2: avoid assertion failure on empty/NULL caps
5069 2009-08-12 12:09:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5071 * gst/typefind/gsttypefindfunctions.c:
5072 typefindfunctions: Also detect SVG by the <svg> starting tag
5073 Not all SVG images have the DOCTYPE specified.
5075 2009-08-10 20:18:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5077 * gst-libs/gst/rtsp/gstrtspconnection.c:
5078 rtspconnection: don't use GLib-2.18 function
5079 g_checksum_reset() was added only in GLib 2.18, but we still require
5080 only 2.16, so work around that if we only have 2.16. Fixes #591357.
5082 2009-08-10 15:40:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5084 * tests/check/pipelines/streamheader.c:
5085 streamheader: Fix caps leak in the vorbisenc unit test
5087 2009-08-10 14:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5089 * tests/check/pipelines/streamheader.c:
5090 checks: fix stream header unit test hanging in gst_task_cleanup_all()
5091 Set pipelines to NULL state and unref when done.
5093 2009-08-10 10:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5095 * gst-libs/gst/rtsp/Makefile.am:
5096 * gst-libs/gst/rtsp/gstrtspconnection.c:
5097 * gst-libs/gst/rtsp/md5.c:
5098 * gst-libs/gst/rtsp/md5.h:
5099 rtsp: Use GLib's GChecksum instead of our own MD5 implementation
5101 2009-08-10 03:46:39 +0300 Mart Raudsepp <leio@gentoo.org>
5103 * gst-libs/gst/interfaces/navigation.c:
5104 navigation: Fix doc blurb typo for gst_navigation_send_key_event
5106 2009-08-09 12:13:16 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5108 * gst/subparse/gstsubparse.c:
5109 subparse: Allow . instead of , as millisecond delimiter in srt subtitles
5112 2009-08-08 17:51:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5114 * gst-libs/gst/audio/gstaudiosrc.c:
5115 * gst/playback/gstinputselector.c:
5116 * gst/playback/gststreamselector.c:
5117 Revert inlines that cause compiler warnings and are not needed anyway
5119 2009-08-08 15:54:57 +0200 Edward Hervey <bilboed@bilboed.com>
5121 * gst-libs/gst/audio/gstaudioclock.c:
5122 * gst-libs/gst/audio/gstaudiosink.c:
5123 * gst-libs/gst/audio/gstaudiosrc.c:
5124 * gst-libs/gst/audio/gstbaseaudiosrc.c:
5125 * gst-libs/gst/audio/gstringbuffer.c:
5126 * gst-libs/gst/interfaces/propertyprobe.c:
5127 * gst-libs/gst/riff/riff-media.c:
5128 * gst-libs/gst/rtp/gstbasertpdepayload.c:
5129 * gst-libs/gst/video/gstvideofilter.c:
5130 * gst-libs/gst/video/gstvideosink.c:
5131 gst-libs: Remove dead assignments and resulting unused variables.
5133 2009-08-08 15:54:41 +0200 Edward Hervey <bilboed@bilboed.com>
5135 * ext/alsa/gstalsadeviceprobe.c:
5136 * ext/alsa/gstalsasink.c:
5137 * ext/alsa/gstalsasrc.c:
5138 * ext/gnomevfs/gstgnomevfssrc.c:
5139 * ext/ogg/gstoggaviparse.c:
5140 * ext/ogg/gstoggdemux.c:
5141 * ext/ogg/gstoggmux.c:
5142 * ext/pango/gsttextrender.c:
5143 * ext/vorbis/vorbisenc.c:
5144 ext: Remove dead assignments and resulting unused variables.
5146 2009-08-08 15:54:02 +0200 Edward Hervey <bilboed@bilboed.com>
5148 * gst/adder/gstadder.c:
5149 * gst/audioconvert/gstaudioconvert.c:
5150 * gst/audioresample/gstaudioresample.c:
5151 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
5152 * gst/ffmpegcolorspace/imgconvert.c:
5153 * gst/playback/gstdecodebin.c:
5154 * gst/playback/gstdecodebin2.c:
5155 * gst/playback/gstfactorylists.c:
5156 * gst/playback/gstinputselector.c:
5157 * gst/playback/gstplaysink.c:
5158 * gst/playback/gststreamselector.c:
5159 * gst/tcp/gsttcpclientsink.c:
5160 * gst/videoscale/gstvideoscale.c:
5161 * gst/videoscale/vs_image.c:
5162 * gst/videotestsrc/gstvideotestsrc.c:
5163 gst: Remove dead assignments and resulting unused variables
5165 2009-08-07 13:05:42 +0200 Josep Torra <n770galaxy@gmail.com>
5167 * docs/design/draft-va.txt:
5168 docs: add draft for generic introduction of video acceleration APIs idea
5170 2009-08-07 08:53:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5172 * ext/theora/gsttheoradec.h:
5173 * ext/theora/theoradec.c:
5174 Revert "theora: Convert theoradec to libtheora 1.0 API"
5175 This reverts commit f1e142ac9dcfb754d85357b9077d5aee48559dd9.
5176 Temporarily revert until we have a workaround for debian/ubuntu
5177 packaging failure (see http://bugs.debian.org/528710).
5179 2009-08-07 09:32:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5181 * gst/typefind/gsttypefindfunctions.c:
5182 typefindfunctions: Add typefinders for many game sound console formats supported by gme
5183 These are AY, GBS, GYM, KSS, SAP and VGM. SPC and NSF already had typefinders.
5185 2009-07-16 11:29:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5187 * ext/ogg/gstoggmux.c:
5188 oggmux: fix warning when we're not linked downstream and error out properly
5189 Fix caps warning when there's no element linked downstream, and pass
5190 not-linked flow return value correctly up the chain, so we error out
5191 correctly. Fixes #588717.
5193 2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
5195 * ext/theora/gsttheoradec.h:
5196 * ext/theora/theoradec.c:
5197 theora: Convert theoradec to libtheora 1.0 API
5199 2009-08-06 20:47:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5201 * ext/pango/gsttextrender.c:
5202 textrender: Fix blitting of text over the output buffer and cairo painting
5204 2009-08-06 09:13:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5206 * ext/pango/gsttextrender.c:
5207 textrender: Fix endianness problems (i.e. make it work again on big endian architectures)
5209 2009-07-31 14:27:28 +0300 Stefan Kost <ensonic@users.sf.net>
5211 * tests/icles/test-colorkey.c:
5212 colorkey-test: fix xsync error
5214 2009-07-06 23:06:50 +0300 Siarhei Siamashka <siarhei.siamashka@nokia.com>
5216 * gst/ffmpegcolorspace/imgconvert.c:
5217 * gst/ffmpegcolorspace/imgconvert_template.h:
5218 ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats
5220 2009-07-14 12:33:29 +0300 Stefan Kost <ensonic@users.sf.net>
5222 * gst/playback/gstplaysink.c:
5223 playbin2: smarter sink selection. Fixes #588523
5224 Don't do fallbacks if application specified a sink element. When doing the
5225 fallback use configured default elements instead of hardcoded linux only
5226 elements. Improve error messages accordingly.
5228 2009-08-06 12:18:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
5230 * gst/playback/gstqueue2.c:
5231 queue2: post error message when pausing task if so appropriate
5232 If a downstream element returns an error while upstream has already
5233 put all data into queue2 (including EOS), upstream will no longer
5234 chain into queue2, so it is up to queue2 to perform some
5235 EOS handling / message posting in such cases. See #589991.
5237 2009-08-06 12:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5239 * gst-libs/gst/audio/gstbaseaudiosrc.c:
5240 baseaudiosrc: change default slave method
5241 Set the default slave method to the much better skew slaving algortihm.
5243 2009-08-06 12:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5245 * ext/pango/gsttextoverlay.c:
5246 textoverlay: make buffer writable
5247 Make the input buffer writable before changing its contents.
5249 2009-08-06 09:55:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5251 * gst/typefind/gsttypefindfunctions.c:
5252 typefinding: fix postscript typefinder probability
5253 Two bytes for a rare format hardly warrants MAXIMUM typefinding
5254 probability, POSSIBLE seems more appropriate.
5256 2009-08-04 14:55:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5258 * ext/pango/gsttextoverlay.c:
5259 pango: Send queries from the srcpad directly to the video sinkpad
5261 2009-08-04 14:32:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5263 * gst/subparse/gstsubparse.c:
5264 subparse: Implement POSITION query
5266 2009-08-04 14:29:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5268 * gst/subparse/gstsubparse.c:
5269 * gst/subparse/samiparse.c:
5270 subparse: Implement SEEKING query
5272 2009-08-04 14:14:53 +0200 John Millikin <jmillikin@gmail.com>
5275 * gst-libs/gst/tag/gstid3tag.c:
5276 * gst-libs/gst/tag/gstvorbistag.c:
5277 tag: Add support for ALBUM_ARTIST tag in vorbiscomments and ID3v2 tags
5278 Require latest core for this.
5281 2009-08-04 12:46:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5283 * ext/pango/gsttextoverlay.c:
5284 * ext/pango/gsttextoverlay.h:
5285 pango: Add support for xRGB and BGRx formats
5287 2009-08-04 12:22:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5289 * ext/pango/gsttextoverlay.c:
5290 pango: Fix endianness issues from the pangocairo switch
5291 cairo's ARGB is in native endianness, i.e. ARGB on big endian architectures
5292 and BGRA on little endian architectures.
5294 2009-08-04 12:11:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5296 * ext/pango/gsttextoverlay.c:
5297 pango: Re-add shading support which was dropped by a previous patch
5299 2009-08-04 11:58:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5302 * ext/pango/gsttextoverlay.c:
5303 pango: Check if pangocairo supports vertical rendering and fix properties
5305 2009-08-04 11:45:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5307 * ext/pango/gsttextrender.c:
5308 textrender: Use PROP_X instead of ARG_X consistently
5310 2009-08-04 11:42:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5312 * ext/pango/gstclockoverlay.c:
5313 * ext/pango/gsttextoverlay.c:
5314 * ext/pango/gsttextrender.c:
5315 * ext/pango/gsttimeoverlay.c:
5316 pango: Some minor cleanup
5318 2009-08-04 11:36:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5321 pango: Check for pangocairo instead of pangoft2
5323 2009-08-04 11:35:10 +0200 Young-Ho Cha <ganadist@chollian.net>
5325 * ext/pango/gsttextoverlay.c:
5326 * ext/pango/gsttextoverlay.h:
5327 * ext/pango/gsttextrender.c:
5328 * ext/pango/gsttextrender.h:
5329 pango: Use pango-cairo instead of pango-ft2
5330 pango-cairo will always use the native font rendering backend
5331 of the platform and provides better results.
5334 2009-08-04 10:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5336 * gst/typefind/gsttypefindfunctions.c:
5337 typefindfunctions: Add SVG typefinder
5339 2009-08-04 10:29:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5341 * gst/typefind/gsttypefindfunctions.c:
5342 typefindfunctions: Add postscript typefinder
5344 2009-07-30 15:08:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5346 * gst/typefind/gsttypefindfunctions.c:
5347 typefindfunctions: Use static caps again for MPEG4 typefinding
5349 2009-07-30 15:05:28 +0200 Arnout Vandecappelle <arnout@mind.be>
5351 * gst/typefind/gsttypefindfunctions.c:
5352 typefindfunctions: Implement better & more flexible MPEG4 typefinding
5353 This detects more MPEG4 streams as MPEG4.
5356 2009-07-30 14:04:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5358 * gst-libs/gst/cdda/gstcddabasesrc.c:
5359 cddabasesrc: Allow to specify the device name in the URI
5360 The allowed URI scheme is now:
5361 cdda://(device#)?track
5362 Also allow every combination of uppercase and lowercase
5363 characters for the protocol part.
5366 2009-07-30 12:37:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5368 * gst/videoscale/gstvideoscale.c:
5369 videoscale: Restrict width/height to 2^15 - 1
5370 Otherwise integer overflows will happen, resulting in segmentation faults.
5373 2009-07-29 14:55:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5375 * gst/ffmpegcolorspace/imgconvert_template.h:
5376 ffmpegcolorspace: Fix indention of template header
5378 2009-07-29 14:10:35 +0200 Philip Jägenstedt <philipj@opera.com>
5380 * gst-libs/gst/app/gstappsrc.c:
5381 appsrc: Clarify documentation about caps and linkage
5384 2009-07-29 07:42:05 +0200 Benjamin Gaignard <benjamin@gaignard.net>
5386 * gst/typefind/gsttypefindfunctions.c:
5387 typefindfunctions: Fix typefinding of SDP files
5390 2009-07-28 20:50:06 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
5392 * gst/audioresample/gstaudioresample.c:
5393 audioresample: Take the output offsets from the input if possible
5396 2009-07-28 15:54:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5398 * gst/videoscale/gstvideoscale.c:
5399 videoscale: Make sure to allocate enough memory for the temporary buffer
5400 and fix scaling of odd-height interlaced video.
5402 2009-07-28 15:18:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5404 * gst/videoscale/gstvideoscale.c:
5405 videoscale: Fix interlaced scaling for I420
5406 ...and some other minor mistakes in the previous change.
5408 2009-07-28 14:12:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5410 * gst/ffmpegcolorspace/avcodec.h:
5411 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
5412 * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
5413 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
5414 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
5415 * gst/ffmpegcolorspace/imgconvert.c:
5416 ffmpegcolorspace: Include interlacing information in the AVPicture
5417 This later allows to handle interlaced AVPicture different than
5418 progressive ones which is needed for horizontally subsampled YUV
5419 formats, see bug #589242.
5421 2009-07-28 13:55:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5423 * gst/videoscale/gstvideoscale.c:
5424 * gst/videoscale/gstvideoscale.h:
5425 videoscale: Add support for interlaced content
5426 videoscale is not mixing content of two seperate fields anymore
5427 and does scaling on every field separately.
5430 2009-08-06 01:44:24 +0100 Jan Schmidt <thaytan@noraisin.net>
5433 back to development -> 0.10.24.1
5435 2009-08-05 02:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
5437 * gst-plugins-base.doap:
5438 Add 0.10.24 release to the doap file
5440 === release 0.10.24 ===
5442 2009-08-05 00:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
5448 * docs/plugins/gst-plugins-base-plugins.args:
5449 * docs/plugins/gst-plugins-base-plugins.hierarchy:
5450 * docs/plugins/gst-plugins-base-plugins.interfaces:
5451 * docs/plugins/gst-plugins-base-plugins.prerequisites:
5452 * docs/plugins/gst-plugins-base-plugins.signals:
5453 * docs/plugins/inspect/plugin-adder.xml:
5454 * docs/plugins/inspect/plugin-alsa.xml:
5455 * docs/plugins/inspect/plugin-app.xml:
5456 * docs/plugins/inspect/plugin-audioconvert.xml:
5457 * docs/plugins/inspect/plugin-audiorate.xml:
5458 * docs/plugins/inspect/plugin-audioresample.xml:
5459 * docs/plugins/inspect/plugin-audiotestsrc.xml:
5460 * docs/plugins/inspect/plugin-cdparanoia.xml:
5461 * docs/plugins/inspect/plugin-decodebin.xml:
5462 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
5463 * docs/plugins/inspect/plugin-gdp.xml:
5464 * docs/plugins/inspect/plugin-gio.xml:
5465 * docs/plugins/inspect/plugin-gnomevfs.xml:
5466 * docs/plugins/inspect/plugin-libvisual.xml:
5467 * docs/plugins/inspect/plugin-ogg.xml:
5468 * docs/plugins/inspect/plugin-pango.xml:
5469 * docs/plugins/inspect/plugin-playback.xml:
5470 * docs/plugins/inspect/plugin-queue2.xml:
5471 * docs/plugins/inspect/plugin-subparse.xml:
5472 * docs/plugins/inspect/plugin-tcp.xml:
5473 * docs/plugins/inspect/plugin-theora.xml:
5474 * docs/plugins/inspect/plugin-typefindfunctions.xml:
5475 * docs/plugins/inspect/plugin-uridecodebin.xml:
5476 * docs/plugins/inspect/plugin-video4linux.xml:
5477 * docs/plugins/inspect/plugin-videorate.xml:
5478 * docs/plugins/inspect/plugin-videoscale.xml:
5479 * docs/plugins/inspect/plugin-videotestsrc.xml:
5480 * docs/plugins/inspect/plugin-volume.xml:
5481 * docs/plugins/inspect/plugin-vorbis.xml:
5482 * docs/plugins/inspect/plugin-ximagesink.xml:
5483 * docs/plugins/inspect/plugin-xvimagesink.xml:
5486 2009-08-05 00:38:40 +0100 Jan Schmidt <thaytan@noraisin.net>
5521 2009-08-01 17:26:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5523 * gst/typefind/gsttypefindfunctions.c:
5524 * tests/check/gst/typefindfunctions.c:
5525 typefinding: fix detection of fLaC id packet in broken flac-in-ogg
5526 There are flac-in-ogg files without the usual flac packet framing
5527 and these files just have a 4-byte fLaC ID packet as first packet.
5528 We need to recognise the type just from these four bytes if we
5529 want oggdemux to recognise these streams correctly.
5531 2009-07-30 14:40:50 +0100 Jan Schmidt <thaytan@noraisin.net>
5567 0.10.24.5 pre-release
5569 2009-07-29 14:15:53 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
5571 * gst-libs/gst/audio/gstaudiofilter.c:
5572 audiofilter: Don't assert on slightly different caps
5573 Plugins should not assert on incompatible caps, caps negotiation will
5576 2009-07-30 13:42:21 +0300 Stefan Kost <ensonic@users.sf.net>
5578 * gst/adder/gstadder.c:
5579 adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146.
5581 2009-07-30 09:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5584 configure: bump Gtk+ requirement of GUI examples from 2.12 to 2.14
5585 The gio mount example needs GtkMountOperation, which is new in 2.14.
5587 2009-07-27 10:29:27 +0100 Balachandran C <balachandran_c@rediffmail.com>
5589 * ext/alsa/gstalsasrc.c:
5590 alsasrc: set alsasrc->handle back to NULL when closing device
5591 Fixes crashes in gst_alsa_find_device_name() when probing or
5592 reading the device-name property (e.g. when doing a dot-file
5593 dump). Fixes #589797.
5595 2009-07-24 19:26:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5597 * gst/playback/gststreamselector.c:
5598 playbin: rename GType of stream selector pad to avoid clash with input-selector from -bad
5599 Rename the GType of the pads of playbin's internal stream selector
5600 element so they don't use the same type name as input-selector's
5601 pads. Fixes #589622.
5603 2009-07-24 13:39:55 +0100 Jan Schmidt <thaytan@noraisin.net>
5636 0.10.23.4 pre-release
5638 2009-07-24 13:46:15 +0100 Jan Schmidt <thaytan@noraisin.net>
5640 * tests/examples/v4l/.gitignore:
5641 ignores: Ignore v4l probing example binary
5643 2009-07-24 09:35:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5645 * gst/typefind/gsttypefindfunctions.c:
5646 typefind: recognise Kate spu subtitles as well
5647 Recognise spu-subtitles, SUB and K-SPU as valid categories for
5648 Kate subtitles as well.
5650 2009-07-24 00:42:16 +0300 Stefan Kost <ensonic@users.sf.net>
5653 Automatic update of common submodule
5654 From fedaaee to 94f95e3
5656 2009-07-22 14:21:43 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
5658 * gst-plugins-base.spec.in:
5659 Update spec file with latest changes
5661 2009-07-20 17:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
5694 * win32/common/_stdint.h:
5695 * win32/common/audio-enumtypes.c:
5696 * win32/common/config.h:
5697 * win32/common/gstrtsp-enumtypes.c:
5698 * win32/common/interfaces-enumtypes.c:
5699 * win32/common/video-enumtypes.c:
5700 0.10.23.3 pre-release
5702 2009-07-20 12:51:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5704 * gst/audiotestsrc/gstaudiotestsrc.c:
5705 audiotestsrc: call send_event directly
5706 We can't call gst_element_send_event() from a streaming thread as it gets the
5707 state lock. Instead call the send_event method directly until we have a nice API
5708 for this in basesrc.
5711 2009-07-03 04:42:24 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
5713 * gst-libs/gst/audio/gstaudiosink.c:
5714 audiosink: Add stream-status messages
5717 2009-07-03 04:41:05 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
5719 * gst-libs/gst/audio/gstaudiosrc.c:
5720 audiosrc: Add stream-status messages
5723 2009-07-20 10:53:11 +0200 Edward Hervey <bilboed@bilboed.com>
5725 * gst/adder/gstadder.c:
5726 gstadder: Don't forget to free pending events on flush/dispose.
5729 2009-07-12 10:08:12 +0200 Edward Hervey <bilboed@bilboed.com>
5731 * tests/check/elements/adder.c:
5732 tests/adder: Add stream consistency checking. Fixes #588748
5734 2009-07-12 10:07:34 +0200 Edward Hervey <bilboed@bilboed.com>
5736 * gst/audiotestsrc/gstaudiotestsrc.c:
5737 audiotestsrc: Make sure tags are properly serialized. Fixes #588746
5738 We do this by letting the basesrc base class handle the tags.
5740 2009-07-13 09:28:54 +0200 Edward Hervey <bilboed@bilboed.com>
5742 * gst/adder/gstadder.c:
5743 * gst/adder/gstadder.h:
5744 adder: Collect incoming tag events and send them after newsegment. Fixes #588747
5746 2009-07-16 09:32:46 +0200 Edward Hervey <bilboed@bilboed.com>
5748 * ext/vorbis/vorbisdec.c:
5749 vorbisdec: Check for empty tag strings. Fixes #588724
5751 2009-07-14 17:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5753 * gst/playback/gstqueue2.c:
5754 queue2: fix leak and improve buffering
5755 Keep track of the max requested position and compare this to the write position
5756 in the temp file to get the current amount of buffered data.
5757 Fix memleak of all incomming buffers.
5760 2009-07-15 17:40:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5762 * gst/playback/Makefile.am:
5763 * gst/playback/gstinputselector.c:
5764 * gst/playback/gstinputselector.h:
5765 * gst/playback/gstplay-marshal.list:
5766 * gst/playback/gstplaybin2.c:
5767 playbin2: use private copy of input-selector
5768 We shouldn't really depend on elements from -bad for stream
5769 selection in playbin2, so use a private copy of input-selector
5770 until the selector plugin is ready to be moved to -base or -good.
5773 2009-07-15 17:26:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5775 * gst/playback/gstinputselector.c:
5776 * gst/playback/gstinputselector.h:
5777 playback: add private copy of the input-selector from gst-plugins-bad
5778 Not hooked up yet though. See #586356.
5780 2009-07-14 19:00:36 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
5782 * tests/examples/v4l/Makefile.am:
5783 examples: fix v4l probe example build
5786 2009-07-14 19:00:10 +0100 Jan Schmidt <thaytan@noraisin.net>
5820 0.10.23.2 pre-release
5822 2009-07-14 16:24:10 +0100 Jan Schmidt <thaytan@noraisin.net>
5826 Add Turkish translations
5828 2009-07-14 15:31:13 +0100 Jan Schmidt <thaytan@noraisin.net>
5830 * tests/check/elements/adder.c:
5831 adder: One more attempt to fix the adder test
5832 Give up and discard and recreate the alsasrc after checking it can
5833 be opened, due to some strange crash inside alsa when we don't.
5835 2009-07-14 15:06:41 +0100 Jan Schmidt <thaytan@noraisin.net>
5837 * tests/check/elements/adder.c:
5838 adder: Perform get_state() in the unit test
5839 Wait for the alsasrc to return to NULL after setting it to PAUSED for
5840 testing, otherwise it leads to segfaults later on.
5842 2009-07-14 14:39:32 +0100 Jan Schmidt <thaytan@noraisin.net>
5844 * tests/check/elements/adder.c:
5845 adder: Don't fail when alsasrc is unavailable
5846 Make the liveadder test succeed silently when it can't be completed
5847 either because alsasrc is unavailable, or because the device is
5850 2009-07-13 22:51:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5852 * gst-libs/gst/pbutils/descriptions.c:
5853 * gst/typefind/gsttypefindfunctions.c:
5854 typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest
5855 Differentiate subtitle streams and lyrics/cracktastic/complex streams via
5856 the category string in the headers. This seems like a useful distinction
5857 to make, and also seems more future-proof. See #525743.
5859 2009-02-21 13:18:10 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
5861 * ext/ogg/gstoggmux.c:
5862 oggmux: add Kate caps to the list of accepted types
5865 2009-07-13 21:56:46 +0300 Stefan Kost <ensonic@users.sf.net>
5867 * gst/playback/gsturidecodebin.c:
5868 uridecodebin: treat uri-schemas incasesensitive
5869 Treat uri-schemas incasesensitive. This is mandated in rfc2396 section 3.1.
5870 Fixes not showing buffering messages e.g. for HTTP://...
5872 2009-07-13 21:54:47 +0300 Stefan Kost <ensonic@users.sf.net>
5874 * gst-libs/gst/interfaces/navigation.c:
5875 navigation: simplify docs
5876 Make short-desc short - its used in the toc. Strip uneeded markup.
5878 2009-07-13 18:31:15 +0100 Jan Schmidt <thaytan@noraisin.net>
5880 * win32/common/libgstnetbuffer.def:
5881 * win32/common/libgstvideo.def:
5883 Remove methods from video base classes that have moved to -bad.
5884 Add gst_netaddress_to_string
5886 2009-07-13 17:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
5888 * tests/examples/gio/.gitignore:
5889 ignores: ignore the giosrc-mounting example binary
5891 2009-07-13 17:54:40 +0100 Jan Schmidt <thaytan@noraisin.net>
5893 * gst-libs/gst/interfaces/navigation.c:
5894 navigation: Add some partial documentation
5895 Add a general documentation blurb for the GstNavigation functionality.
5896 Still lacks some example code and detail on how to implement it.
5898 2009-07-13 17:52:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5900 * gst-libs/gst/pbutils/descriptions.c:
5901 pbutils: add description for Siren codec and make two descriptions non-translatable
5903 2009-07-13 12:23:20 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
5906 Automatic update of common submodule
5907 From 5845b63 to fedaaee
5909 2009-07-13 18:21:49 +0200 Elliott Sales de Andrade <quantum.analyst at gmail.com>
5911 * gst-libs/gst/riff/riff-ids.h:
5912 * gst-libs/gst/riff/riff-media.c:
5913 riff: add siren to the RIFF parser
5914 Add siren7 caps to the RIFF parser.
5916 2009-07-13 14:55:59 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
5919 * tests/examples/Makefile.am:
5920 * tests/examples/v4l/Makefile.am:
5921 * tests/examples/v4l/probe.c:
5922 v4lsrc: add a simple test case for device probing
5924 2009-07-03 11:38:01 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
5927 * sys/v4l/Makefile.am:
5928 * sys/v4l/gstv4lelement.c:
5929 v4lsrc: optional support for device probing with gudev
5930 Enumerate v4l devices using gudev if available.
5933 2009-07-10 23:24:36 +0100 Stefan Kost <ensonic@users.sf.net>
5935 * gst/adder/gstadder.c:
5936 adder: add since tags to docs
5938 2009-07-10 21:29:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5940 * tests/examples/seek/seek.c:
5941 seek: don't automatically start pipeline in DB
5942 Keep the pipeline paused when we detect download buffering. The user has to
5943 manually start the pipeline for now because we can't estimate when the buffering
5944 will finish or when we have underrun.
5946 2009-07-10 21:01:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5948 * gst/playback/gstqueue2.c:
5949 queue2: flush differently, avoiding deadlocks
5950 Don't flush the file by closing and opening it but instead use g_freopen. This
5951 avoids a deadlock in shutdown because we emit the temp-location property change
5952 with the wrong lock held.
5954 2009-07-10 20:25:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5956 * tests/examples/seek/seek.c:
5957 seek: add a checkbox for progressive download
5959 2009-07-10 20:24:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5961 * gst/playback/gsturidecodebin.c:
5962 uridecodebin: Fix template construction
5963 Fix the construction of the temporary filename construction as the application
5964 name can be NULL and we don't want a separator between the prgname and the
5967 2009-07-10 20:04:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5969 * gst/playback/gstplay-enum.c:
5970 * gst/playback/gstplay-enum.h:
5971 * gst/playback/gstplaybin2.c:
5972 playbin2: add support for progressive download
5973 Add a new playbin2 flag (initially disabled) to enable progressive download
5974 buffering in uridecodebin.
5976 2009-07-10 19:59:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5978 * gst/playback/gsturidecodebin.c:
5979 uridecodebin: add download property
5980 Add a download property that will attempt to configure queue2 into progressive
5982 Make sure we only enable download buffering for quicktime and flv formats.
5984 2009-07-10 19:49:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5986 * gst/playback/gstqueue2.c:
5987 queue2: add temp-template property
5988 Add a new temp-template property so that queue2 can securely allocate a
5989 temporary filename. Deprecate the temp-location property for setting the
5990 location but still use it to notify the allocated temp file.
5992 2009-07-10 20:06:28 +0100 Stefan Kost <ensonic@users.sf.net>
5994 * gst/adder/gstadder.c:
5995 * gst/adder/gstadder.h:
5996 adder: add a caps-property to avoid to need to plug a capsfilter afterwards
5997 Adder can only handle one common format accross the pads. Thus one needed to add
5998 a capsfilter afterwards and manage the caps. Now one can simply set the caps on
6001 2009-07-10 18:59:05 +0100 Stefan Kost <ensonic@users.sf.net>
6003 * tests/check/elements/adder.c:
6004 adder: skip live-seek text if we have no audiosrc, add new test
6005 The seek-test needs a real audiosrc. Also add a test that checks that adder is
6006 reusable. Finaly handle warnings as warnings to fix a assertion.
6008 2009-07-10 19:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6010 * ext/gio/gstgiosink.c:
6011 gio: Also post a "not-mounted" message from giosink
6013 2009-07-10 17:15:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6015 * tests/examples/gio/giosrc-mounting.c:
6016 gio: Remove workaround for playbin2 bug in the sample application
6017 The playbin2 bug was #588078.
6019 2009-07-10 17:08:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6021 * gst/playback/gstplaybin2.c:
6022 playbin2: Make it possible for READY->PAUSED to succeed after it failed the first time
6023 If READY->PAUSED failed in the source element we would've swapped
6024 the current and next group already. To allow READY->PAUSED to succeed
6025 after the first failure we have to swap the current and next group
6026 back again. This also ensure that we're again in the same state
6027 as before the failed state change and not at the next group.
6028 This was especially a problem for playbin2 pipelines that use the
6029 new mounting support in giosrc as the source would fail for READY->PAUSED
6030 the first time, the application mounts the location and then tries
6031 to go READY->PAUSED again (and this time it would succeed).
6034 2009-07-10 11:42:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6037 * tests/examples/Makefile.am:
6038 * tests/examples/gio/Makefile.am:
6039 * tests/examples/gio/giosrc-mounting.c:
6040 gio: Add example application that shows how to handle the "not-mounted" message
6042 2009-07-10 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6045 gio: Remove the experimental status from the GIO plugin
6048 2009-07-10 11:24:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6050 * ext/gio/gstgiosink.c:
6051 * ext/gio/gstgiosrc.c:
6052 gio: Add documentation for the new "not-mounted" and "file-exists" messages
6054 2009-07-09 13:45:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6056 * ext/gio/gstgiobasesrc.c:
6057 gio: Make sure that we have the correct stream position when starting
6059 2009-07-08 17:24:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6061 * ext/gio/gstgiobasesink.c:
6062 gio: Make sure to flush the output stream if it shouldn't be closed
6063 Otherwise there might still be unwritten data after the element
6066 2009-07-08 17:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6068 * ext/gio/gstgiobasesink.c:
6069 * ext/gio/gstgiobasesink.h:
6070 * ext/gio/gstgiobasesrc.c:
6071 * ext/gio/gstgiobasesrc.h:
6072 * ext/gio/gstgiosink.c:
6073 * ext/gio/gstgiosrc.c:
6074 gio: Don't close the GIO streams for the giostream{src,sink} elements
6075 This makes it possible to do something useful with the streams
6076 after the element has stopped. Fixes bug #587896.
6078 2009-07-08 17:19:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6080 * tests/check/pipelines/gio.c:
6081 gio: Try to reuse the pipeline with the same stream objects
6083 2009-07-08 17:02:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6085 * ext/gio/gstgiobasesink.c:
6086 * ext/gio/gstgiobasesrc.c:
6087 gio: Improve the error message if a stream is already closed before usage
6089 2009-07-08 16:55:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6091 * ext/gio/gstgiosink.c:
6092 gio: Post a custom file-exists message on the bus if the file already exists
6093 An application can handle this message, remove the file in question
6094 and restart the pipeline again without showing an error.
6095 This fixes bug #529300.
6097 2009-07-08 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6099 * ext/gio/gstgiosrc.c:
6100 gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted
6102 2009-07-08 16:50:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6104 * ext/gio/gstgiosink.c:
6105 gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink
6107 2009-07-08 15:52:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6109 * ext/gio/gstgiosrc.c:
6110 gio: Post a custom "not-mounted" message on the bus
6111 This allows applications to mount the GFile if possible and restart
6112 the pipeline instead of simply giving an error.
6114 2009-07-08 15:08:32 +0200 Philip Jägenstedt <philipj@opera.com>
6116 * gst/audioconvert/gstchannelmix.c:
6117 audioconvert: Fix compilation when debugging is disabled
6120 2009-07-07 20:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6122 * ext/gio/gstgiobasesink.c:
6123 * ext/gio/gstgiobasesink.h:
6124 * ext/gio/gstgiobasesrc.h:
6125 * ext/gio/gstgiosink.c:
6126 * ext/gio/gstgiosink.h:
6127 * ext/gio/gstgiostreamsink.c:
6128 * ext/gio/gstgiostreamsink.h:
6129 gio: Add vfunc for requesting the stream for the sinks too
6131 2009-07-07 20:21:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6133 * ext/gio/gstgiobasesink.c:
6134 * ext/gio/gstgiobasesink.h:
6135 * ext/gio/gstgiobasesrc.c:
6136 * ext/gio/gstgiosink.c:
6137 * ext/gio/gstgiosrc.c:
6138 * ext/gio/gstgiostreamsink.c:
6139 * ext/gio/gstgiostreamsrc.c:
6140 gio: Some more random cleanup
6142 2009-07-07 20:20:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6145 * ext/gio/gstgiobasesink.c:
6146 * ext/gio/gstgiobasesrc.c:
6147 * ext/gio/gstgiobasesrc.h:
6148 * ext/gio/gstgiosink.c:
6149 * ext/gio/gstgiosrc.c:
6150 * ext/gio/gstgiosrc.h:
6151 * ext/gio/gstgiostreamsink.c:
6152 * ext/gio/gstgiostreamsrc.c:
6153 * ext/gio/gstgiostreamsrc.h:
6154 gio: Update my mail address and copyright
6156 2009-07-07 20:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6158 * ext/gio/gstgiobasesrc.c:
6159 * ext/gio/gstgiobasesrc.h:
6160 * ext/gio/gstgiosrc.c:
6161 * ext/gio/gstgiostreamsrc.c:
6162 * ext/gio/gstgiostreamsrc.h:
6163 gio: General clean up and simplification
6164 The GInputStreams are now requested by a vfunc from
6165 the subclasses instead of relying that the subclass
6166 sets it until it's needed.
6167 This might also fix bug #587896.
6169 2009-07-06 22:31:12 +0100 Stefan Kost <ensonic@users.sf.net>
6171 * gst/adder/gstadder.c:
6172 adder: keep sending newsegments after seeking
6173 Adder sends with timestamps from 0 upwards. After seeking we need to send
6174 new-segments to get correct positions-queries.
6176 2009-07-06 20:44:00 +0100 Stefan Kost <ensonic@users.sf.net>
6178 * tests/check/elements/adder.c:
6179 adder: make test more robust
6180 Add audioconverts to the live-seeking test to make it negotiate.
6182 2009-06-30 17:19:50 +0300 Stefan Kost <ensonic@users.sf.net>
6184 * sys/xvimage/xvimagesink.c:
6185 xvimagesink: use core performance log category
6187 2009-07-05 21:29:40 +0200 Edward Hervey <bilboed@bilboed.com>
6189 * gst/adder/gstadder.c:
6190 adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped.
6191 This ensures that collectpads' cookie is properly updated so that when the streaming
6192 threads will restart and be checking for the flushing status of all pads there will
6193 be no inconsistent state.
6195 2009-07-05 18:01:38 +0200 Hans-Peter Nilsson <hp@gcc.gnu.org>
6197 * ext/pango/gstclockoverlay.c:
6198 pango: Call tzset() before localtime_r()
6199 POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't
6200 required to set the state variables that define the current timezone. Indeed,
6201 glibc (at least 2.9) doesn't do this for subsequent calls. The effect is that
6202 if the system timezone is changed for a running program between two calls to
6203 gst_clock_overlay_render_time, it won't be noticed. For glibc, changing the
6204 timezone equals /etc/localtime being modified.
6207 2009-07-01 17:33:14 -0700 David Schleef <ds@schleef.org>
6210 build: remove spurious schroedinger reference
6212 2009-07-01 10:25:43 -0700 David Schleef <ds@schleef.org>
6216 * ext/schroedinger/Makefile.am:
6217 * ext/schroedinger/gstschro.c:
6218 * ext/schroedinger/gstschrodec.c:
6219 * ext/schroedinger/gstschroenc.c:
6220 * ext/schroedinger/gstschroparse.c:
6221 * ext/schroedinger/gstschroutils.c:
6222 * ext/schroedinger/gstschroutils.h:
6223 * gst-libs/gst/video/Makefile.am:
6224 * gst-libs/gst/video/gstbasevideocodec.c:
6225 * gst-libs/gst/video/gstbasevideocodec.h:
6226 * gst-libs/gst/video/gstbasevideodecoder.c:
6227 * gst-libs/gst/video/gstbasevideodecoder.h:
6228 * gst-libs/gst/video/gstbasevideoencoder.c:
6229 * gst-libs/gst/video/gstbasevideoencoder.h:
6230 * gst-libs/gst/video/gstbasevideoparse.c:
6231 * gst-libs/gst/video/gstbasevideoparse.h:
6232 * gst-libs/gst/video/gstbasevideoutils.c:
6233 * gst-libs/gst/video/gstbasevideoutils.h:
6234 basevideo: send basevideo back to remedial school
6235 Move basevideo classes and schroedinger plugin to -bad.
6237 2009-07-01 12:54:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6239 * docs/libs/gst-plugins-base-libs-sections.txt:
6240 * gst-libs/gst/netbuffer/gstnetbuffer.h:
6241 netaddress: add constant for max len
6243 2009-07-01 12:48:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6245 * docs/libs/gst-plugins-base-libs-sections.txt:
6246 * gst-libs/gst/netbuffer/gstnetbuffer.c:
6247 * gst-libs/gst/netbuffer/gstnetbuffer.h:
6248 netbuffer: add gst_netaddress_to_string
6249 Add function to serialize a net address to a string.
6250 API: GstNetAddress::gst_netaddress_to_string()
6252 2009-06-30 18:44:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6254 * gst/playback/gsturidecodebin.c:
6255 uridecodebin: make fd:// uri use buffering too
6256 fd:// usually operate in push mode only and are thus suitable for buffering.
6258 2009-06-30 14:46:38 +0300 Stefan Kost <ensonic@users.sf.net>
6260 * gst/playback/gstplaybin2.c:
6261 * gst/volume/gstvolume.c:
6262 volume: include "1.0=100%" in property description
6264 2009-06-30 14:45:51 +0300 Stefan Kost <ensonic@users.sf.net>
6266 * gst/playback/gstplaysink.c:
6267 playsink: remove unused property defs
6269 2009-06-29 17:11:50 +0300 Stefan Kost <ensonic@users.sf.net>
6271 * gst-libs/gst/audio/multichannel.c:
6272 multichannel: rewrite the new doc comment a bit
6273 Its part of the audio lib.
6275 2009-06-29 14:34:02 +0100 Jan Schmidt <thaytan@noraisin.net>
6277 * gst/playback/gstplaysink.c:
6278 playsink: Avoid a segfault when the video sink fails to start
6279 Don't attempt to display the subpictures and segfault when the
6280 video sink failed to start (and hence the videochain is NULL).
6282 2009-06-29 15:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6284 * gst-libs/gst/audio/gstringbuffer.c:
6285 * gst-libs/gst/audio/gstringbuffer.h:
6286 ringbuffer: add vmethod to clear the ringbuffer
6287 Add a vmethod so that subclasses can be notified when they should clear the data
6290 2009-06-29 14:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
6292 * gst-libs/gst/riff/riff-media.c:
6293 riff-media: Fix the fourcc caps property for VC-1/WMVA
6294 The caps property for carrying fourccs is 'format', not 'fourcc'
6296 2009-06-29 12:20:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6298 * gst-libs/gst/rtsp/gstrtspconnection.c:
6299 rtsp: include in.h for FreeBSD compat
6302 2009-06-29 12:20:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6304 * win32/common/libgstapp.def:
6305 defs: add defs for new appsink buffer-list method
6307 2009-06-29 12:14:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6309 * gst-libs/gst/app/gstappsink.c:
6310 * gst-libs/gst/app/gstappsink.h:
6311 appsink: add docs and signals
6312 Add docs for the new callback.
6313 Add signals for the new buffer-list support.
6315 2009-06-29 10:24:36 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
6317 * tests/check/elements/appsink.c:
6318 Added unit tests for buffer list support in appsink.
6320 2009-06-17 11:12:08 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
6322 * gst-libs/gst/app/gstappsink.c:
6323 Added buffer list support.
6325 2009-06-17 09:23:11 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
6327 * gst-libs/gst/app/gstappsink.h:
6328 Added buffer list support.
6330 2009-06-29 09:36:27 +0200 Peter Kjellerstedt <pkj@axis.com>
6332 * gst-libs/gst/sdp/gstsdpmessage.c:
6333 sdp: Include winsock2.h after defining WINVER.
6334 Similar to bug #587080.
6336 2009-06-29 09:31:40 +0200 Peter Kjellerstedt <pkj@axis.com>
6338 * gst-libs/gst/rtsp/gstrtspconnection.c:
6339 rtsp: Moved a comment.
6341 2009-06-27 23:23:02 +0300 Stefan Kost <ensonic@users.sf.net>
6343 * gst-libs/gst/audio/audio.c:
6344 * gst-libs/gst/audio/multichannel.c:
6345 docs: add basic section docs for multichannel and relocate the ones for audio
6346 Add section docs for multichannel, so that it has a short desc in the toc too.
6347 Move the section docs in adio up, so that the follow the copyright like
6350 2009-06-26 21:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
6352 * sys/v4l/gstv4lelement.c:
6353 * sys/v4l/gstv4lsrc.c:
6354 v4l: open/close device in ready.
6355 Simillar change like in v4l2src. This allows probing feature in paused, where
6356 streaming is noit yet started.
6358 2009-06-10 17:05:22 +0300 René Stadler <rene.stadler@nokia.com>
6360 * gst/playback/gstplaysink.c:
6361 playbin2: fix initial volume handling also when reusing the element
6362 This is a follow-up to commit 452988, making it work correctly when the audio
6365 2009-06-26 21:48:58 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
6367 * gst-libs/gst/rtsp/gstrtspconnection.c:
6368 Define WINVER before including any win headers
6371 2009-06-27 00:50:54 +0300 René Stadler <mail@renestadler.de>
6373 * gst-libs/gst/riff/riff-read.c:
6374 riff: prevent crash if rounded up tag size exceeds data size
6375 When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
6376 and an invalid read past the buffer data follows.
6378 2009-06-26 15:17:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6380 * gst-libs/gst/video/gstbasevideocodec.c:
6381 basevideocodec: By default don't allow caps changes on the srcpad
6382 This fixed playback of Dirac files with schrodec when upstream wants
6383 a different width/height, basevideocodec accepts this and then
6384 pushes buffers with new caps but content of the old caps.
6385 In the best case this will just result in wrong unit size and a
6386 failure in basestransform elements.
6388 2009-06-26 14:11:21 +0100 Jan Schmidt <thaytan@noraisin.net>
6391 autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
6392 Check for more automake command variants. Use printf instead of 'echo -n'
6395 2009-06-26 13:41:38 +0100 Jan Schmidt <thaytan@noraisin.net>
6398 Automatic update of common submodule
6399 From f810030 to 5845b63
6401 2009-06-26 13:14:02 +0300 Stefan Kost <ensonic@users.sf.net>
6403 * gst/playback/gstscreenshot.c:
6404 screenshot: don't leak message
6406 2009-06-25 12:04:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6408 * gst/typefind/gsttypefindfunctions.c:
6409 typefinding: lower the h264 typefinder's probability
6410 A NEARLY_CERTAIN is absolutely not warranted given the kind
6411 of things it checks for. Even a LIKELY is probably not entirely
6414 2009-06-24 15:13:56 +0100 Jan Schmidt <jan.schmidt@sun.com>
6417 Automatic update of common submodule
6418 From f3bb51b to f810030
6420 2009-06-24 09:48:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6422 * gst-libs/gst/pbutils/descriptions.c:
6423 pbutils: add description for multipart
6424 So we get slightly nicer error messages when multipartdemux is missing.
6426 2009-06-23 18:07:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6428 * gst/adder/gstadder.c:
6429 adder: only unflush when we flushed before
6430 Ass suggested by Stefan Kost:
6431 Keep track of when the sinkpad was set to flushing and unflush the pad when an
6432 upstream flushing seek failed.
6434 2009-06-23 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6436 * gst/playback/gsturidecodebin.c:
6437 uridecodebin: fix leak when the source fails to change state
6439 2009-06-23 12:40:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6441 * gst/subparse/gstssaparse.c:
6442 ssaparse: avoid leaking all buffers
6444 2009-06-22 22:18:03 +0300 Stefan Kost <ensonic@users.sf.net>
6446 * tests/check/elements/adder.c:
6447 adder: test seek handling in adder
6448 This tests seeking on an adder that has a normal and a live source connected.
6449 Wheter the current behavior is the desired one needs to be discussed still
6452 2009-06-22 16:17:10 +0300 Stefan Kost <ensonic@users.sf.net>
6454 * sys/ximage/ximagesink.c:
6455 * sys/xvimage/xvimagesink.c:
6456 x(v)imagesink: pass the xwindow along to not look at the yet unset var.
6457 When we call this from xwindow_new, x(v)imagesink->xwindow is not yet set.
6459 2009-06-22 11:40:33 +0300 Stefan Kost <ensonic@users.sf.net>
6461 * sys/ximage/ximagesink.c:
6462 * sys/ximage/ximagesink.h:
6463 * sys/xvimage/xvimagesink.c:
6464 * sys/xvimage/xvimagesink.h:
6465 x(v)imagesink: catch tags and show title in own window
6466 Refactor the code that sets the window title. Catch tag-events and use title
6467 metadata for the window title.
6469 2009-06-21 19:42:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6471 * gst/audiotestsrc/gstaudiotestsrc.c:
6472 audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
6473 Also make all the function arrays constant.
6475 2009-06-21 12:27:37 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
6477 * gst/audiotestsrc/gstaudiotestsrc.c:
6478 * gst/audiotestsrc/gstaudiotestsrc.h:
6479 audiotestsrc: Add support for generating gaussian white noise
6480 This patch adds support for stationary white Gaussian noise.
6481 The Box-Muller algorithm is used to generate pairs of independent
6482 normally-distributed random numbers.
6485 2009-06-20 23:46:28 +0100 Jan Schmidt <thaytan@noraisin.net>
6487 * gst/ffmpegcolorspace/imgconvert.c:
6488 * gst/ffmpegcolorspace/imgconvert_template.h:
6489 ffmpegcolorspace: Fix NV12 and NV21 transformations
6490 Fix some stride problems, fix the nv12 to nv21 direct transformation,
6491 and implement a direct conversion to yuv444 to save CPU.
6493 2009-06-20 22:36:21 +0100 Jan Schmidt <thaytan@noraisin.net>
6495 * gst/videotestsrc/videotestsrc.c:
6496 videotestsrc: Fix NV12 painting for odd strides/heights
6498 2009-06-19 22:16:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6500 * ext/cdparanoia/gstcdparanoiasrc.c:
6501 cdparanoia: run-time license is LGPL now that we require cdparanoia 0.10.2
6502 cdparanoia has an LGPL v2.1 license since 0.10.1 and we now require 0.10.2.
6503 Finally fixes #531035.
6505 2009-06-19 21:25:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6507 * ext/cdparanoia/gstcdparanoiasrc.c:
6508 cdparanoia: try to guess a good cache size if it's set to -1
6509 Try to guess from the paranoia-mode setting whether playback or
6510 ripping is wanted, and use a smaller cache size if we're likely
6511 to be doing playback, to avoid a long startup delay. Since this
6512 was the value used in older cdparanoia versions, it should be
6513 fine in any case. See #586331.
6515 2009-06-19 11:27:40 +1000 Jonathan Matthew <jonathan@d14n.org>
6518 * ext/cdparanoia/gstcdparanoiasrc.c:
6519 * ext/cdparanoia/gstcdparanoiasrc.h:
6520 cdparanoia: expose cache size setting
6521 This setting was added in cdparanoia 10.2. The default value is good
6522 for audio extraction, but lower values (previous versions of cdparanoia
6523 used 150) are better for realtime playback.
6526 2009-06-19 17:43:03 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
6528 * gst-plugins-base.spec.in:
6529 Make build of schro plugin conditional
6531 2009-06-19 15:52:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6533 * docs/libs/gst-plugins-base-libs-sections.txt:
6534 * gst-libs/gst/rtp/gstbasertppayload.c:
6535 * gst-libs/gst/rtp/gstbasertppayload.h:
6536 * win32/common/libgstrtp.def:
6537 basertppayload: add support for bufferlists
6538 Based on patch from Ognyan Tonchev.
6541 2009-06-19 15:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6543 * gst-libs/gst/rtp/gstrtpbuffer.c:
6544 rtpbuffer: use new convenience functions
6545 New core convenience functions makes the list getters and setters trivial.
6546 Maybe even too trivial...
6548 2009-06-18 19:07:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6550 * win32/common/libgstrtp.def:
6551 defs: add new symbol to win32 defs file
6552 Based on patches by Ognyan Tonchev.
6555 2009-06-18 19:04:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6557 * docs/libs/gst-plugins-base-libs-sections.txt:
6558 * gst-libs/gst/rtp/gstrtpbuffer.c:
6559 rtp: cleanups, add _list_get_seq() too
6560 Clean up the docs a little.
6561 Add missing _list_get_seq method.
6562 Add new symbols to the docs
6564 2009-06-18 18:47:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6566 * gst-libs/gst/rtp/gstrtpbuffer.c:
6567 * win32/common/libgstrtp.def:
6569 Add Since tags to docs
6570 Move some code around
6573 2009-06-18 17:46:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6575 * gst-libs/gst/rtp/gstrtpbuffer.c:
6576 * gst-libs/gst/rtp/gstrtpbuffer.h:
6577 * tests/check/libs/rtp.c:
6578 rtp: add bufferlist support
6580 2009-06-18 18:03:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6582 * gst-libs/gst/rtp/gstrtpbuffer.c:
6583 rtp: pass data to macros instead of GstBuffer
6585 2009-06-18 17:42:10 +0100 Jan Schmidt <thaytan@noraisin.net>
6587 * win32/common/libgstrtsp.def:
6588 win32: Add gst_rtsp_watch_queue_data() to the exports
6589 Fix the tests by exporting the new symbol from the win32 dlls
6591 2009-06-18 18:13:22 +0300 Stefan Kost <ensonic@users.sf.net>
6593 * sys/xvimage/xvimagesink.c:
6594 xvimagesink: appname might be NULL
6595 Don't set title if appname is unknown.
6597 2009-06-18 17:58:06 +0300 Stefan Kost <ensonic@users.sf.net>
6599 * sys/xvimage/xvimagesink.c:
6600 xvimagesink: set window title from application name
6602 2009-06-09 19:14:00 +0200 Peter Kjellerstedt <pkj@axis.com>
6604 * gst-libs/gst/rtsp/gstrtspurl.c:
6605 rtsp: Made the parsing of the RTSP URL scheme more generic.
6607 2009-06-15 13:58:26 +0200 Peter Kjellerstedt <pkj@axis.com>
6609 * gst-libs/gst/rtsp/gstrtspconnection.c:
6610 * gst-libs/gst/rtsp/gstrtspconnection.h:
6611 rtsp: Added gst_rtsp_watch_queue_data().
6612 gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
6613 but allows for queuing any data block for writing (much like
6614 gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)
6615 API: gst_rtsp_watch_queue_data()
6617 2009-06-09 16:37:09 +0200 Peter Kjellerstedt <pkj@axis.com>
6619 * gst-libs/gst/rtsp/gstrtspconnection.c:
6620 rtsp: Only extract the session ID from RTSP responses.
6622 2009-06-09 19:06:57 +0200 Peter Kjellerstedt <pkj@axis.com>
6624 * gst-libs/gst/rtsp/gstrtspurl.c:
6625 rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
6627 2009-06-09 14:31:18 +0200 Peter Kjellerstedt <pkj@axis.com>
6629 * gst-libs/gst/rtsp/gstrtspconnection.c:
6630 rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
6632 2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
6634 * gst-libs/gst/rtsp/gstrtspconnection.c:
6635 rtsp: Improved base64 decoding in fill_bytes().
6636 The base64 decoding in fill_bytes() expected the size of the read data to
6637 be evenly divisible by four (which is true for the base64 encoded data
6638 itself). This did not, however, take whitespace (especially line breaks)
6639 into account and would fail the decoding if any whitespace was present.
6641 2009-06-17 14:00:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6643 * gst-libs/gst/audio/gstbaseaudiosrc.c:
6644 audiosrc: fix get_offset
6645 When we need to jump to the most recently captured sample, jump to where the
6646 next sample will be written instead of to some old data.
6649 2009-06-17 13:18:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6651 * gst-libs/gst/audio/gstbaseaudiosink.c:
6652 audiosink: free the ringbuffer when going to NULL
6653 Unparent and free the ringbuffer when going to NULL, like we do with the
6654 audiosrc element. We can do this now because we correctly manage the time
6657 2009-06-17 13:17:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6659 * gst-libs/gst/audio/gstaudiosink.c:
6660 * gst-libs/gst/audio/gstaudiosrc.c:
6661 audio: correctly handle short read/writes
6663 2009-05-05 15:37:54 +0300 René Stadler <rene.stadler@nokia.com>
6665 * gst-libs/gst/audio/gstbaseaudiosrc.c:
6666 baseaudiosrc: add some extra logging for buffer timestamps
6668 2009-06-17 11:22:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6670 * gst/adder/gstadder.c:
6671 adder: more seeking fixes.
6672 When a seek failed upstream, make sure the adder sinkpad is set unflushing again
6673 so that streaming can continue.
6674 We only have a pending segment when we flushed.
6675 Set the flush_stop_pending flag inside the appropriate locks and before we
6676 attempt to perform the upstream seek.
6677 Add some more comments.
6678 Use the right lock to protect the flags in flush_stop.
6681 2009-06-17 07:24:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6683 * gst/playback/gstdecodebin2.c:
6684 decodebin2: Free iterator after removing all groups
6686 2009-06-16 19:38:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6688 * gst-libs/gst/video/gstvideofilter.c:
6689 videofilter: Add a default get_unit_size function
6690 This returns the correct values for all formats that are handled by
6691 GstVideoFormat and makes all the custom get_unit_size functions in
6692 many elements unnecessary.
6694 2009-06-16 18:57:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6696 * gst-libs/gst/rtsp/gstrtspdefs.c:
6697 * gst-libs/gst/rtsp/gstrtspdefs.h:
6698 rtsp: add Timestamp header field
6701 2009-06-16 18:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6703 * gst/playback/gstplaybin2.c:
6704 playbin2: set smarter target state on uridecodebin
6705 Set the target state of the newly added uridecodebins to somthing else that
6706 PAUSED so that we keep their state in sync with the playsink state.
6709 2009-06-16 18:13:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6711 * gst/playback/gstplaysink.c:
6712 playsink: set the sink flag on the element
6714 2009-06-16 18:09:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6716 * gst/playback/gsturidecodebin.c:
6717 uridecodebin: add debug message
6719 2009-06-16 14:05:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6721 * gst-libs/gst/audio/gstaudiosink.c:
6722 * gst-libs/gst/audio/gstaudiosrc.c:
6723 audiosink, audiosrc: do the class_ref()s in the right class_init functions
6724 Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
6726 2009-06-15 15:39:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6728 * gst-libs/gst/audio/gstaudiosink.c:
6729 * gst-libs/gst/audio/gstaudiosrc.c:
6730 audiosink,audiosrc: ref the audio ring buffer class and type in class_init
6731 Hack around thread-safety issues in GObject and our racy _get_type()
6732 functions (we could easily fix the _get_type() functions, but we still
6733 need to hack around the GObject class races until we require a newer
6734 GLib version, I think).
6736 2009-06-15 12:57:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6738 * gst-libs/gst/audio/gstbaseaudiosrc.c:
6739 audiosrc: return FALSE when receiving a SEEK event
6740 When receiving a seek event, return FALSE as we don't implement seeking.
6742 2009-06-15 11:06:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6744 * tests/examples/seek/seek.c:
6745 Don't use deprecated GTK API
6748 2009-06-15 11:40:00 +0300 Stefan Kost <ensonic@users.sf.net>
6750 * gst/adder/gstadder.c:
6751 adder: send flush_stop when seeking failed
6752 At least do the fix to sent the flush_stop when seeking failed to ensure we
6753 keep no pads flushing. before it was send when the seeking worked which is just
6754 plain wrong and was not the intention.
6756 2009-06-12 15:17:14 +0200 Peter Kjellerstedt <pkj@axis.com>
6758 * gst-libs/gst/rtsp/gstrtspconnection.c:
6759 rtsp: Use a more consistent naming of GstRTSPRec variables.
6761 2009-06-12 15:11:05 +0200 Peter Kjellerstedt <pkj@axis.com>
6763 * gst-libs/gst/rtsp/gstrtspconnection.c:
6764 * gst-libs/gst/rtsp/gstrtspconnection.h:
6765 rtsp: Call message_sent() callback for all sent messages.
6766 Previously the messages_sent() callback was only called for messages
6767 which had a CSeq, which excluded all data messages. Instead of using the
6768 CSeq as ID, use a simple index counter.
6770 2009-06-14 22:13:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6772 * ext/ogg/gstoggdemux.c:
6773 * ext/theora/theoradec.c:
6774 * ext/vorbis/vorbisdec.c:
6775 oggdemux: post/send tags with the container-format tag
6776 For this to work properly, theoradec and vorbisdec need to put
6777 tag events received from upstream into the pending_events list
6778 so they get pushed out after any newsegment event, not before.
6780 2009-06-14 20:30:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6782 * tests/examples/seek/scrubby.c:
6783 * tests/examples/seek/seek.c:
6784 * tests/old/examples/seek/cdplayer.c:
6785 Don't use deprecated GTK API
6788 2009-06-12 16:31:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6790 * gst/adder/gstadder.c:
6791 adder: send flush-stop earlier
6792 When no flush-stop has been sent by upstream, we have to send one ourselves to
6793 continue playback. Do this as soon as the collect function is called instead of
6794 after we possibly pushed segment events (that got then flushed out)
6796 2009-06-12 13:55:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6798 * tests/examples/seek/seek.c:
6799 seek: add shuttle controls
6801 2009-06-12 13:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6803 * tests/examples/seek/stepping2.c:
6804 example: fix compile
6806 2009-06-12 13:52:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6808 * tests/examples/seek/Makefile.am:
6809 examples: build the stepping2 example
6811 2009-06-12 13:52:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6813 * gst/playback/gstplaysink.c:
6814 playsink: update for new step API
6816 2009-06-12 13:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6818 * ext/ogg/gstoggdemux.c:
6819 oggdemux: do reverse seeks more accurate
6820 For reverse seeking with the accurate flag set, try to be more precise by
6821 seeking a little bit after the requested position.
6823 2009-06-11 22:32:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6825 * ext/ogg/gstogmparse.c:
6826 * gst/subparse/gstssaparse.c:
6827 * gst/subparse/gstssaparse.h:
6828 * gst/subparse/gstsubparse.c:
6829 * gst/subparse/gstsubparse.h:
6830 subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC
6831 Make subtitle parsers post a taglist with codec tags, so the application
6832 knows what kind of subtitle a subtitle stream is. Fixes #576552.
6834 2009-06-11 19:12:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6836 * gst-libs/gst/audio/gstringbuffer.c:
6837 ringbuffer: handle border cases in resampler
6839 2009-06-11 13:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
6842 * docs/libs/Makefile.am:
6843 * docs/plugins/Makefile.am:
6844 docs: Update common. Use upload-doc.mak instead of upload.mak
6846 2009-06-11 12:39:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6848 * gst-libs/gst/rtp/gstbasertppayload.c:
6851 2009-06-11 12:17:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6853 * gst-libs/gst/audio/gstbaseaudiosink.c:
6854 baseaudiosink: reset accum when dropping samples
6855 When we are resampling and we drop samples because we paused, reset the accum
6856 counter because it's now invalid.
6858 2009-06-11 11:16:15 +0100 Jan Schmidt <thaytan@noraisin.net>
6860 * docs/libs/gst-plugins-base-libs-sections.txt:
6861 * gst-libs/gst/interfaces/mixer.h:
6862 * gst-libs/gst/video/gstbasevideodecoder.h:
6863 docs: Fix a couple of warnings from the docs build.
6865 2009-06-10 21:36:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6867 * gst-libs/gst/audio/testchannels.c:
6868 Don't include config.h multiple times when build audio testchannel app.
6869 Fixes build problem on win32 (#585075).
6871 2009-06-10 16:56:51 +0100 Jan Schmidt <thaytan@noraisin.net>
6873 * gst/playback/gstplaybin2.c:
6874 * gst/playback/gsturidecodebin.c:
6875 playbin2/uridecodebin: Fix connection-speed propagation
6876 uridecodebin expects the passed connection-speed value in kbps, so we
6877 need to divide the value stored in bps by 1000. Also, lower the upper
6878 limit on the properties to the value that we can actually store in our
6879 internal guint (which is plenty high enough)
6881 2009-06-10 14:37:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6883 * gst/subparse/gstsubparse.c:
6884 * tests/check/elements/subparse.c:
6885 subparse: recognise more subrip timestamp variants
6886 Be even less restrictive in what we accept for .srt timestamps when
6887 typefinding and parsing subrip subtitles and add a unit test for
6888 the 'new' format. Fixes #585197.
6890 2009-06-09 22:00:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6892 * gst-libs/gst/rtsp/gstrtsptransport.h:
6893 rtsp: add some more docs
6895 2009-06-09 18:24:55 +0200 Peter Kjellerstedt <pkj@axis.com>
6897 * gst-libs/gst/rtsp/gstrtspmessage.c:
6898 rtsp: Avoid a compiler warning.
6900 2009-06-09 18:23:28 +0200 Peter Kjellerstedt <pkj@axis.com>
6902 * gst-libs/gst/rtsp/gstrtspdefs.h:
6903 rtsp: Updated documentation for GstRTSPResult.
6904 Moved GST_RTSP_ELAST to be last in the documentation to match the actual
6907 2009-05-20 17:30:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6910 autogen: remove -Wno-portability from here
6911 as it is in configure.ac now.
6913 2009-06-09 16:28:20 +0200 Peter Kjellerstedt <pkj@axis.com>
6915 * gst-libs/gst/rtsp/gstrtspconnection.c:
6916 rtsp: Plug a memory leak.
6917 Free memory related to any partially read and/or written RTSP messages.
6919 2009-06-09 12:09:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6921 * gst-libs/gst/audio/gstbaseaudiosink.c:
6922 baseaudiosink: no need to cause discont when clipping
6923 Remove the discont-when-clipping hack now that basesink provides us with
6924 correctly clipped samples when stepping.
6926 2009-06-08 17:26:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6928 * gst-libs/gst/audio/gstbaseaudiosink.c:
6929 audiosink: don't align when we clip
6930 Don't align samples when they were clipped. Not entirely correct but better than
6933 2009-06-08 16:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6935 * tests/examples/seek/.gitignore:
6936 * tests/examples/seek/stepping2.c:
6937 examples: add stepping example in PLAYING
6938 Add stepping example in PLAYING, audio is a bit distorted because basesink does
6939 not provide good clipping info yet.
6941 2009-06-08 10:25:00 +0200 Edward Hervey <bilboed@bilboed.com>
6943 * gst-libs/gst/pbutils/descriptions.c:
6944 pbutils: Add description for hdv/aux-* formats.
6946 2009-06-07 22:20:33 +0400 LRN <lrn1986@gmail.com>
6948 * ext/schroedinger/Makefile.am:
6949 Added libgstbase to schro's LIBADD
6952 2009-06-06 02:15:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6954 * gst-libs/gst/tag/gstid3tag.c:
6955 libgsttag: don't extract genres from empty ID3v1 tags
6956 If we don't have any other info, don't try to interpret the
6957 genre field. In particular we don't want to interpret a genre
6958 of 0 as 'Blues' if no other fields are set and the entire tag
6961 2009-06-05 18:13:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6963 * gst/playback/gstdecodebin2.c:
6964 decodebin2: make sure varargs are of right type
6965 Explicitly cast the variables to g_object_set to their right types.
6967 2009-06-05 16:49:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6969 * gst/playback/gstdecodebin2.c:
6970 decodebin2: increase stream probing queues
6971 When we are probing for streams, we want to set the queue size in such a way
6972 that we can scan a maximum amount of data without consuming too much memory.
6973 Therefore, remove the time limit on the queue and only stop scanning after 2MB
6977 2009-06-05 14:06:17 +0200 Peter Kjellerstedt <pkj@axis.com>
6979 * gst-libs/gst/rtsp/gstrtspconnection.c:
6982 2009-06-05 14:05:54 +0200 Peter Kjellerstedt <pkj@axis.com>
6984 * gst-libs/gst/rtsp/gstrtspconnection.c:
6985 rtsp: Remove an unused variable.
6987 2009-06-05 13:59:14 +0200 Peter Kjellerstedt <pkj@axis.com>
6989 * gst-libs/gst/rtsp/gstrtspconnection.c:
6990 rtsp: Removed duplicate initialization of conn->writefd.
6992 2009-06-05 13:55:08 +0200 Peter Kjellerstedt <pkj@axis.com>
6994 * gst-libs/gst/rtsp/gstrtspconnection.c:
6995 rtsp: Use #defined status codes.
6997 2009-06-05 13:53:29 +0200 Peter Kjellerstedt <pkj@axis.com>
6999 * gst-libs/gst/rtsp/gstrtspconnection.c:
7000 rtsp: Correct gen_tunnel_reply().
7001 Prevent gen_tunnel_reply() from generating an incomplete response
7002 in case an error response code is given.
7004 2009-06-05 10:57:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7007 * win32/common/_stdint.h:
7008 * win32/common/config.h:
7009 * win32/common/video-enumtypes.c:
7010 configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
7011 See #584835. Also update win32 files while we're at it.
7013 2009-06-04 08:57:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7015 * gst/playback/gstplaybin2.c:
7016 playbin2: API: Add {audio,video,text}-tags-changed signals
7019 2009-06-03 20:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7021 * ext/vorbis/vorbisdec.c:
7022 vorbisdec: don't put invalid bitrate values into the taglist
7023 Bitrates are stored as 32-bit signed integers in the vorbis
7024 identification headers, but seem to be read incorrectly,
7025 namely as unsigned 32-bit integers, into the vorbis structure
7026 members which are of type long, which makes our check for
7027 values <= 0 fail with files that put -1 in there for unset
7030 2009-06-03 15:52:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7032 * tests/examples/seek/.gitignore:
7033 ignore: add new stepping app to ignore
7035 2009-06-03 15:31:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7037 * tests/examples/seek/Makefile.am:
7038 * tests/examples/seek/stepping.c:
7039 examples: add stepping example.
7040 Add an example of using playbin2 and frame stepping to simulate variable rate
7041 playback based on a sine wave.
7043 2009-06-03 12:45:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7045 * gst/playback/gstplaybin2.c:
7046 * gst/playback/gstplaysink.h:
7047 playbin2: also set custom text and subp sinks
7048 Set the custom subpicture and text sinks along with the custom audio and video
7050 Fix a little docs blurb too.
7052 2009-06-02 12:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7054 * gst-libs/gst/rtsp/gstrtspconnection.c:
7055 * gst-libs/gst/rtsp/gstrtspconnection.h:
7056 rtsp: add G_LIKELY because we can
7058 2009-06-02 09:53:05 +0200 Edward Hervey <bilboed@bilboed.com>
7060 * gst/typefind/gsttypefindfunctions.c:
7061 typefindfunctions: Fix caps for ogg typefinder.
7063 2009-05-29 11:10:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7065 * docs/libs/gst-plugins-base-libs-sections.txt:
7066 docs: remove some cruft from -sections.txt file
7068 2009-06-01 11:31:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7070 * gst/playback/gstplaysink.c:
7071 * tests/examples/seek/seek.c:
7072 add framestepping to playbin2 and seek
7074 2009-06-01 09:59:22 +0200 Peter Kjellerstedt <pkj@axis.com>
7076 * gst-libs/gst/rtsp/gstrtspconnection.c:
7077 rtsp: Avoid compiler warnings with -Wextra.
7079 2009-06-01 09:58:27 +0200 Peter Kjellerstedt <pkj@axis.com>
7081 * gst-libs/gst/rtsp/gstrtspconnection.h:
7082 rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
7084 2009-06-01 09:43:04 +0200 Peter Kjellerstedt <pkj@axis.com>
7086 * gst-libs/gst/sdp/gstsdpmessage.c:
7087 sdp: Remove an unused variable.
7089 2009-05-30 14:17:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7091 * gst/ffmpegcolorspace/imgconvert.c:
7092 * gst/ffmpegcolorspace/imgconvert_template.h:
7093 ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale
7095 2009-05-29 00:09:15 +0100 Jan Schmidt <thaytan@noraisin.net>
7097 * gst/playback/gstplaybin2.c:
7098 playbin2: Have playbin recognise PGS subpicture streams
7099 Recognise PGS subpicture streams and connect them to the SPU pad
7100 in playsink. Unfortunately this fails badly with negotiation errors
7101 if the SPU is not recent enough to support the stream. I'm not sure
7102 how to add format negotiation in yet.
7104 2009-05-21 23:11:29 +0100 Jan Schmidt <thaytan@noraisin.net>
7106 * gst/playback/gstdecodebin2.c:
7107 * gst/playback/gsturidecodebin.c:
7108 decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.
7110 2009-05-28 20:37:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7112 * gst/playback/gstplaysink.c:
7113 playbin2: fix volume handling for audio sinks without "volume" property
7114 When using an audio sink without a "volume" property, volume control
7115 would only work for the first song. For the next song, we'd try to
7116 re-use the existing audio chain, but inadvertently set chain->volume
7117 to NULL instead of to the existing volume element.
7119 2009-05-28 17:05:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7121 * gst/playback/gstplaysink.c:
7122 playbin2: cosmetic change to avoid unnecessary line breaks
7123 Looks nicer and works around gst-indent silliness.
7125 2009-05-28 17:21:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7127 * gst/playback/gstplaysink.c:
7128 playbin2: don't lose the ref to the volume element
7129 Only release the ref to the volume element when it is controled by a sink. For
7130 software volume we never have to fear that it will change.
7132 2009-05-28 15:21:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7134 * gst/playback/gstplaybin2.c:
7135 * gst/playback/gstplaysink.c:
7136 playbin2: actually use configured audio/video sinks
7137 playbin2 inadvertently used autoaudiosink and autovideosink up to now,
7138 since it would overwrite the sinks configured via the "audio-sink"
7139 and "video-sink" properties with the stream-specific group sinks when
7140 configuring the outputs. Those are usually NULL however, so that would
7141 overwrite the configured sinks with NULL which makes playbin2 then
7142 default to the auto sinks. Fix this by keeping a reference to each
7143 configured sink in playbin2 and setting up the right sinks depending
7144 on whether there is a stream-specific sink or not.
7147 2009-05-27 17:37:38 +0300 Stefan Kost <ensonic@users.sf.net>
7149 * tests/examples/seek/seek.c:
7150 seek: add volume label and sync with sink volume
7151 Look at the volume and have the pulsemixer open at same time. Unfortunately
7152 playbin2 does not emit notify on volume right, so this polls for now.
7154 2009-05-27 18:12:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7156 * gst/playback/gstdecodebin2.c:
7157 decodebin2: remove leftover elements
7158 Remove all of the elements inside decodebin2 when goint to READY and NULL.
7159 Makes decodebin2 reusable.
7162 2009-05-27 15:36:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7164 * gst/playback/gstplaysink.c:
7165 playbin2; release refs to volume/mute properties
7166 Release the refs to the volume and mute property elemens before setting the
7167 child elements to READY or NULL.
7170 2009-05-27 12:10:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7172 * gst/gdp/gstgdppay.c:
7173 gdppay: set caps on outgoing buffers
7174 Set caps on outgoing buffers because NULL caps confuse basetransform.
7177 2009-05-27 11:08:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7179 * gst-libs/gst/netbuffer/gstnetbuffer.c:
7180 netbuffer: also note the order of IP4 addresses
7181 IP4 addresses are also stored in network byte order. Make a note of this in the
7184 2009-05-26 22:43:34 +0200 Alessandro Decina <alessandro.d@gmail.com>
7186 * ext/theora/theoraparse.c:
7187 theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.
7189 2009-05-26 11:13:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7191 * gst-libs/gst/rtsp/gstrtspconnection.c:
7192 Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14"
7193 This reverts commit 418760cf740332c12c3fd9cf3244af134fa9534b.
7194 We now require GLib 2.16.
7196 2009-05-26 15:18:09 +0100 Jan Schmidt <thaytan@noraisin.net>
7201 2009-05-26 15:37:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7203 * gst-libs/gst/netbuffer/gstnetbuffer.c:
7204 netbuffer: document that the port is network order
7205 Document the fact that we store the port number in network order in
7206 GstNetAddress and that the caller should byteswap appropriately.
7208 2009-05-26 15:23:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7210 * gst/videoscale/gstvideoscale.c:
7211 * gst/videoscale/vs_4tap.c:
7212 * gst/videoscale/vs_4tap.h:
7213 * gst/videoscale/vs_image.c:
7214 * gst/videoscale/vs_image.h:
7215 * gst/videoscale/vs_scanline.c:
7216 * gst/videoscale/vs_scanline.h:
7217 videoscale: Add support for 16 bit grayscale in native endianness
7219 2009-05-26 14:58:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7221 * gst/ffmpegcolorspace/avcodec.h:
7222 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
7223 * gst/ffmpegcolorspace/imgconvert.c:
7224 ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian
7226 2009-05-26 14:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7228 * gst/videotestsrc/videotestsrc.c:
7229 * gst/videotestsrc/videotestsrc.h:
7230 videotestsrc: Add support for 16 bit grayscale in native endianness
7232 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
7234 add can-activate-pull property to baseaudiosink
7235 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
7238 2009-05-26 13:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7240 * ext/ogg/gstoggdemux.c:
7241 oggdemux: fix boundary case for seeking.
7242 When we have exactly 0 bytes left to search, make sure we stop instead of going
7243 into an infinite loop.
7245 2009-05-26 11:11:03 +0200 Bastien Nocera <hadess at hadess.net>
7247 * gst-libs/gst/cdda/Makefile.am:
7248 * gst-libs/gst/cdda/gstcddabasesrc.c:
7249 * gst-libs/gst/cdda/sha1.c:
7250 * gst-libs/gst/cdda/sha1.h:
7251 cddabasesrc: Remove copy of sha1 digest
7252 Remove our copy of sha1 digest now that we depend on glib 2.16.
7255 2009-05-25 17:54:01 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
7257 * gst-plugins-base.spec.in:
7260 2009-05-23 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7262 * gst-libs/gst/video/gstbasevideodecoder.c:
7263 * gst-libs/gst/video/gstbasevideoparse.c:
7264 * gst-libs/gst/video/gstbasevideoutils.c:
7265 * gst-libs/gst/video/gstbasevideoutils.h:
7266 * win32/common/libgstvideo.def:
7267 video: don't expose internal gst_adapter_get_buffer() helper function
7268 If it's really needed it should go into GstAdapter in core.
7270 2009-05-22 21:29:51 -0700 David Schleef <ds@schleef.org>
7272 * gst-libs/gst/video/gstbasevideodecoder.c:
7273 basevideo: Fix memleak
7275 2009-05-22 21:27:58 -0700 David Schleef <ds@schleef.org>
7277 * ext/schroedinger/gstschrodec.c:
7278 * ext/schroedinger/gstschroparse.c:
7279 schro: Fix usage of adapter_masked_scan_uint32
7280 Because *somebody* changed the API without telling me.
7282 2009-05-22 21:25:06 -0700 David Schleef <ds@schleef.org>
7284 * ext/schroedinger/gstschro.c:
7285 schro: Change package name to GST_PACKAGE_NAME
7287 2009-05-22 17:34:10 -0700 David Schleef <ds@schleef.org>
7289 * gst-libs/gst/video/gstbasevideoencoder.c:
7290 basevideo: Add preset interface to encoder
7292 2009-05-22 17:31:14 -0700 David Schleef <ds@schleef.org>
7294 * gst/audioresample/gstaudioresample.c:
7295 Run liboil benchmark multiple times
7296 The statistics function requires multiple runs, otherwise
7297 it causes a divide by zero error.
7299 2009-05-22 19:36:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7301 * m4/gst-fionread.m4:
7302 m4: fix 'suspicious cache value' warning for gst-fionread.m4
7303 .. here as well (should really be moved to common, but I'm too lazy).
7305 2009-05-22 17:41:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7307 * ext/vorbis/vorbisdec.c:
7308 vorbisdec: detect and report errors better
7309 Check the return values of a couple more libvorbis functions and post an error
7310 when something is wrong instead of continuing and crashing.
7312 2009-05-22 15:49:14 +0300 Stefan Kost <ensonic@users.sf.net>
7314 * gst/playback/gstplaysink.c:
7315 playbin2: fix initial volume and mute handling
7316 Use two flags to remember volume/mute changes at times when we don't have the
7317 audiochain yet (e.g. construction). Only set values when they were actualy
7318 changed. This makes pulseaudio's stream restore functional.
7320 2009-05-22 10:19:51 +0100 Jan Schmidt <thaytan@noraisin.net>
7323 Automatic update of common submodule
7324 From d3a8fab to 888e0a2
7326 2009-05-22 09:03:22 +0100 Jan Schmidt <thaytan@noraisin.net>
7328 * win32/common/libgstvideo.def:
7329 win32: Remove gst_adapter_masked_scan_uint32 from the exports
7331 2009-05-21 10:48:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7333 * gst-libs/gst/audio/gstbaseaudiosink.c:
7334 audiosink: improve debug message
7336 2009-05-19 18:10:55 -0700 Michael Smith <msmith@songbirdnest.com>
7338 * gst-libs/gst/tag/gstid3tag.c:
7339 gstid3tag: Don't extract a track number unless present.
7340 In ID3v1, a track number is present only if byte 125 is null AND
7341 byte 126 is non-null. If the track number is not present, don't add
7342 a track number tag with value 0.
7344 2009-05-20 00:48:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7346 * gst-libs/gst/video/gstbasevideoutils.c:
7347 * gst-libs/gst/video/gstbasevideoutils.h:
7348 videoutils: remove adapter methods
7349 Remove adapter methods now that they are in core.
7351 2009-05-20 00:42:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7353 * win32/common/libgstvideo.def:
7354 defs: add new symbols
7356 2009-05-19 17:47:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7359 autogen: pass -Wno-portability to automake to suppress warnings
7362 2009-05-19 02:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7364 * docs/libs/.gitignore:
7365 gitignore: remove bogus *.sgml wildcard - these files are tracked in git
7367 2009-05-19 18:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7369 * gst/tcp/gsttcpclientsrc.c:
7370 tcpclientsrc: this is not a live source
7371 Don't mark us as a live source because we are not.
7373 2009-05-19 18:41:02 +0300 Stefan Kost <ensonic@users.sf.net>
7375 * gst/adder/gstadder.c:
7376 adder: only send flush_stop when seek failed
7377 This is still not the ultimate fix. Added some comment to explain the troubles.
7379 2009-05-19 17:17:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7381 * gst-libs/gst/audio/gstbaseaudiosink.c:
7382 audiosink: return the return value of wait_preroll
7383 Return the value that _wait_preroll() returned instead of always WRONG_STATE.
7385 2009-05-19 16:45:56 +0300 Stefan Kost <ensonic@users.sf.net>
7387 * gst/adder/gstadder.c:
7388 * gst/adder/gstadder.h:
7389 adder: send flush_stop to match flush_start
7390 Adder was relying that something else sends a flush stop. When using adder with
7391 a livesource it was not getting a flush_stop and thus all pads downstream where
7392 keept flushing. Mark a pending flush_stop and send it when we are working on
7393 the new segment back in the streaming thread.
7395 2009-05-19 16:02:44 +0300 Stefan Kost <ensonic@users.sf.net>
7397 * tests/examples/seek/seek.c:
7398 seek: ui improvements
7399 Repaint the window black on expose, as this looks nicer when resizing or using
7400 the expander. Also show time after slider, as this saves a whole line (nice on
7403 2009-04-29 18:36:17 +0300 Stefan Kost <ensonic@users.sf.net>
7405 * gst/playback/gstdecodebin.c:
7406 decodebin: use iterators instead of list
7407 The list api is deprecated. Use threadsafe iterators instead.
7409 2009-05-19 15:35:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7411 * gst/playback/gsturidecodebin.c:
7412 uridecodebin: configure caps on decodebin2
7413 Implement the caps property by setting the configured caps on new decodebin2
7417 2009-05-19 15:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7419 * gst/playback/gstdecodebin2.c:
7420 decodebin2: avoid some _caps_ref in some cases
7421 Only mess with the caps refcount when we configure different caps.
7423 2009-05-19 15:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7425 * gst/playback/gsturidecodebin.c:
7426 uridecodebin: fix potential caps leak
7427 Free the user-configured caps in finalize.
7429 2009-05-19 15:20:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7431 * gst/playback/gsturidecodebin.c:
7432 uridecodebin: add queue after cdda://
7433 Add a queue2 after the raw output pads of certain sources such as those for uris
7435 No tuning of the queue is done yet as the defaults seem to work fine for me.
7438 2009-05-19 12:45:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7440 * ext/ogg/gstoggdemux.c:
7441 oggdemux: don't loop when at EOS
7442 When we try to read the last page, don't try to read past the upper boundary, as
7443 this might cause endless loops.
7446 2009-05-19 11:20:19 +0200 Edward Hervey <bilboed@bilboed.com>
7448 * gst/audioresample/gstaudioresample.c:
7449 audioresample: Don't drain remaining buffers after a flush.
7450 If we were resetted (due to a flush), we can not drain the remaining
7451 buffers since they would be pushed before a valid new newsegment event.
7453 2009-05-18 22:29:07 -0700 Michael Smith <msmith@syncword.(none)>
7455 * ext/theora/theoradec.c:
7456 theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.
7458 2009-05-19 01:13:34 +0300 Stefan Kost <ensonic@users.sf.net>
7460 * gst/adder/gstadder.c:
7461 adder: add more logging and return value checking
7463 2009-05-19 01:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
7465 * gst/adder/gstadder.c:
7466 adder: handle the return value from iterator_fold
7468 2009-05-19 01:03:44 +0300 Stefan Kost <ensonic@users.sf.net>
7470 * gst/adder/gstadder.c:
7471 adder: use the pad in logging as objects
7472 Helps to differenciate between source and sinks pads.
7474 2009-04-21 22:54:19 +0300 Stefan Kost <ensonic@users.sf.net>
7476 * tests/examples/seek/seek.c:
7477 seek: use parser for mp3 and rename variable
7479 2009-05-18 11:08:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7481 * tests/examples/seek/seek.c:
7482 seek: add playbin2 options in expander
7483 Add the playbin2 stream selection options inside an expander to preserve some
7486 2009-02-10 15:29:10 -0800 David Schleef <ds@schleef.org>
7488 * gst/videotestsrc/videotestsrc.c:
7489 videotestsrc: Add support for v210 and v216 formats
7491 2009-05-15 16:21:15 -0700 David Schleef <ds@schleef.org>
7493 * gst-libs/gst/video/gstbasevideocodec.c:
7494 * gst-libs/gst/video/gstbasevideodecoder.c:
7495 * gst-libs/gst/video/gstbasevideoencoder.c:
7496 * gst-libs/gst/video/gstbasevideoparse.c:
7497 video: remove // comments
7499 2009-05-15 16:18:18 -0700 David Schleef <ds@schleef.org>
7501 * gst-libs/gst/video/video.c:
7502 * gst-libs/gst/video/video.h:
7503 video: Add Y444, v210, v216 formats
7505 2009-05-15 16:12:37 -0700 David Schleef <ds@schleef.org>
7509 * ext/schroedinger/Makefile.am:
7510 * ext/schroedinger/gstschro.c:
7511 * ext/schroedinger/gstschrodec.c:
7512 * ext/schroedinger/gstschroenc.c:
7513 * ext/schroedinger/gstschroparse.c:
7514 * ext/schroedinger/gstschroutils.c:
7515 * ext/schroedinger/gstschroutils.h:
7516 schro: Move schro plugin from Schroedinger
7517 Previous history is in Schroedinger. Depends on, and is an example
7518 of using, GstBaseVideo* base classes.
7519 Code was reindented, and an #ifdef HAVE_ENCODER removed.
7521 2009-05-15 10:23:08 -0700 David Schleef <ds@schleef.org>
7523 * gst-libs/gst/video/Makefile.am:
7524 * gst-libs/gst/video/gstbasevideocodec.c:
7525 * gst-libs/gst/video/gstbasevideocodec.h:
7526 * gst-libs/gst/video/gstbasevideodecoder.c:
7527 * gst-libs/gst/video/gstbasevideodecoder.h:
7528 * gst-libs/gst/video/gstbasevideoencoder.c:
7529 * gst-libs/gst/video/gstbasevideoencoder.h:
7530 * gst-libs/gst/video/gstbasevideoparse.c:
7531 * gst-libs/gst/video/gstbasevideoparse.h:
7532 * gst-libs/gst/video/gstbasevideoutils.c:
7533 * gst-libs/gst/video/gstbasevideoutils.h:
7534 video: Copy BaseVideo classes from Schroedinger
7536 2009-05-15 23:05:45 +0200 Arnout Vandecappelle <arnout@mind.be>
7538 * gst/tcp/gstmultifdsink.c:
7539 multifdsink: add num-fds property
7540 multifdsink::num-fds
7542 2009-05-15 20:36:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7544 * gst-libs/gst/pbutils/descriptions.c:
7545 pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
7547 2009-05-14 11:44:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7549 * ext/vorbis/vorbisenc.c:
7550 vorbisenc: Implement Preset interface
7552 2009-05-14 11:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7554 * ext/theora/theoraenc.c:
7555 theoraenc: Implement Preset interface
7557 2009-05-14 11:41:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7559 * ext/ogg/gstoggmux.c:
7560 oggmux: Implement Preset interface
7562 2009-05-14 21:37:22 +0100 Jan Schmidt <thaytan@noraisin.net>
7564 * gst/playback/gstplaysink.c:
7565 playbin2: Fix cdda:// playback
7566 Don't send async-start when the playsink has already been configured
7567 before changing state.
7569 2009-05-14 01:31:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7572 configure: require core CVS for gst_adapter_prev_timestamp()
7573 which is used in the libvisual plugin.
7575 2009-04-22 18:34:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7578 AUTHORS: fix my email
7580 2009-04-22 18:35:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7582 * gst-libs/gst/audio/gstaudioclock.c:
7583 audioclock: make our internal time monotonic
7584 Make the internal time increase monotonically.
7586 2009-05-13 19:27:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7588 * ext/libvisual/visual.c:
7589 visual: remove next_ts variable
7590 We can remove the next_ts variable as we don't use it anymore.
7592 2009-05-13 19:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7594 * ext/libvisual/visual.c:
7595 visual: use new adapter timestamp code
7596 Use the new adapter timestamp tracking code to make things easier and produce
7597 vastly better output timestamps.
7599 2009-05-13 01:35:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7631 po: avoid conflicts of local *.po files with files in git
7632 Make it so that filenames and line numbers are only stored in the *.pot file
7633 (which is not in git), but not in the individual *.po files. This information
7634 is hardly useful for translators in our case, and it should avoid the constant
7635 conflicts of local *.po files with the ones in git which are caused by the
7636 source files changing and the line numbers being updated. This commit might
7637 cause one last merge conflict for you, which you can work around with
7638 "git checkout po/*.po" before merging or pulling. After that there should
7639 (hopefully) not be any more local modifications of these files (unless
7640 someone committed additions or changes to translated strings and the
7641 *.po files haven't been updated yet, that is).
7643 2009-05-12 23:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7645 * tests/check/elements/.gitignore:
7646 * tests/check/elements/audioresample.c:
7647 tests: fix audioresample unit test on big endian architectures
7648 Don't hardcode endianness=1234 in the filtercaps, it will cause
7649 pad link failures which will result in the test timing out.
7651 2009-05-12 17:18:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7653 * gst/audiotestsrc/gstaudiotestsrc.c:
7654 audiotestsrc: fix broken enum nick - it should have a hyphen
7655 The enum nick should be 'sine-table', not 'sine table'. Technically this is
7656 an API/ABI change I guess, but anyone who was using this and didn't report
7659 2009-05-01 01:04:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7661 * gst/audiotestsrc/gstaudiotestsrc.c:
7662 audiotestsrc: seek to the requested byte offset, not the expected byte offset
7664 2009-05-01 01:03:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7666 * gst/audiotestsrc/gstaudiotestsrc.c:
7667 * gst/audiotestsrc/gstaudiotestsrc.h:
7668 audiotestsrc: support more than just one channel
7670 2009-05-12 15:52:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7672 * gst-libs/gst/interfaces/propertyprobe.h:
7673 propertyprobe: Fix typo in the docs
7675 2009-05-12 12:17:55 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
7677 * ext/ogg/gstoggmux.c:
7678 * ext/theora/theora.c:
7679 * ext/vorbis/vorbis.c:
7680 Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder
7682 2009-04-30 16:37:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7684 * gst/videorate/gstvideorate.c:
7685 * gst/videorate/gstvideorate.h:
7686 videorate: handle invalid timestamps better
7687 Handle buffers with -1 timestamps better by keeping track of the en time of the
7688 previous buffer and assuming the -1 timestamp buffer goes right after the
7690 when we have two buffers that are equally good, output the oldest buffer once to
7692 don't try to calculate latency when the input framerate is unknown.
7694 2009-04-28 11:37:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7696 * ext/ogg/gstoggmux.c:
7697 oggmux: small debug statement in DISCONT
7699 2009-04-28 11:24:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7701 * ext/ogg/gstoggdemux.c:
7702 * ext/ogg/gstoggdemux.h:
7703 oggdemux: fix abuse of ogg API, handle broken oggs
7704 When we feed the ogg sync layer, we need to feed it contiguous data even if the
7705 sync layer did not consume all of it yet. This makes sure that it always finds
7706 the next page even for more corrupted files. Use a different read_offset for
7707 this purpose. since we now keep track of the sync layer, we don't have to reset
7708 after finding a start of a page.
7709 Add some more debug info for the error paths.
7710 Only reset the sync layer when we perform a seek operation.
7711 Avoid failure when the next chain has no bos pages but instead simply ignore it.
7712 when we receive unknown page serial numbers mid stream, don't fail but post a
7713 warning and hope that we get back on track later.
7716 2009-04-30 16:41:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7718 * gst/playback/gstdecodebin2.c:
7719 decodebin2: make subpictures a raw output format
7720 Subpictures are a raw format, we want those pads exposed so that playbin2 can do
7721 the subpicture mixing.
7723 2009-04-27 10:15:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7725 * gst-libs/gst/rtp/gstbasertppayload.c:
7726 * gst-libs/gst/rtp/gstbasertppayload.h:
7727 rtpdepay: add some more comments
7729 2009-04-17 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7731 * gst-libs/gst/audio/gstaudioclock.c:
7732 audioclock: make sure values are ever increasing
7734 2009-05-05 17:17:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7736 * gst/playback/gstplaysink.c:
7737 playbin2: make fallback identity silent
7738 Set the signal-handoffs to FALSE and silent to TRUE for the fallback identity
7739 element so that it consumes less CPU.
7741 2009-04-17 10:57:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7743 * gst/playback/gstplaybin2.c:
7744 * gst/playback/gstplaysink.c:
7745 playbin2: handle custom audiosinks differently
7746 Keep track of the autoplugged custom sinks and configure them in the playsink
7747 element when we have collected all streams.
7748 Also make sure that we only select one custom sink.
7749 When unreffing the internal sink, we don't need to change the state to NULL.
7751 2009-05-12 10:36:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7753 * gst/playback/gstplaybin2.c:
7754 * gst/playback/gstplaysink.c:
7755 * gst/playback/gstplaysink.h:
7756 playbin2: unify custom sink get/set functions
7757 Use one function to set/get all of the different sink types.
7758 cleanup up the subpicture chain too.
7759 Allow setting a custom subpicture sink.
7761 2009-05-11 18:29:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7763 * gst-libs/gst/interfaces/tunernorm.h:
7764 interfaces: Seperate some more struct definitions from typedefs
7766 2009-05-11 15:48:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7768 * gst-libs/gst/interfaces/navigation.h:
7769 * gst-libs/gst/interfaces/videoorientation.h:
7770 * gst-libs/gst/interfaces/xoverlay.h:
7771 interfaces: Seperate some more struct definitions from typedefs
7773 2009-05-10 17:28:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7775 * win32/common/libgstinterfaces.def:
7776 Add new functions to win32 exports
7778 2009-05-10 17:28:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7780 * docs/libs/gst-plugins-base-libs-sections.txt:
7781 Add new functions to the docs
7783 2009-05-10 17:25:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7785 * gst-libs/gst/interfaces/mixer.c:
7786 * gst-libs/gst/interfaces/mixer.h:
7787 interfaces: API: Add gst_mixer_get_mixer_type()
7788 This is a convenience function that returns the mixer_type
7789 of the interface struct.
7791 2009-05-10 17:25:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7793 * gst-libs/gst/interfaces/colorbalance.c:
7794 interfaces: Add docs for gst_color_balance_get_balance_type()
7796 2009-05-10 11:17:19 +0200 Marc-Andre Lureau <marcandre.lureau@gmail.com>
7799 Run libtoolize before aclocal
7800 This unbreaks the build in some cases. Fixes bug #582021
7802 2009-05-07 17:38:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7804 * ext/pango/gsttextrender.c:
7805 textrender: Correctly initialize the background for ARGB too
7807 2009-05-07 16:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7809 * ext/pango/gsttextrender.c:
7810 * ext/pango/gsttextrender.h:
7811 textrender: Use libgstvideo functions to create caps
7812 Also check if downstream wants ARGB always when we get
7815 2009-05-07 16:52:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7817 * ext/pango/gsttextrender.c:
7818 textrender: Don't always use ARGB if downstream supports it but take it's preference
7820 2009-05-07 16:48:08 +0200 Kapil Agrawal <kapil@mediamagictechnologies.com>
7822 * ext/pango/gsttextrender.c:
7823 * ext/pango/gsttextrender.h:
7824 textrender: Add support for ARGB and alignment properties
7827 2009-05-07 16:42:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7829 * ext/pango/gsttextrender.c:
7830 textrender: Add ; after GST_BOILERPLATE to fix indention
7832 2009-05-07 15:10:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7834 * gst-libs/gst/tag/gstvorbistag.c:
7835 vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
7837 2009-05-07 14:59:36 +0200 Arnout Vandecappelle <arnout@mind.be>
7839 * gst/typefind/gsttypefindfunctions.c:
7840 typefindfunctions: made mp3_type_find less aggressive
7841 mp3_type_find could suggest already when only a single valid header
7842 was found, if it ran out of data before the end of the next frame.
7843 Therefore, ignore the last found frame if it was incomplete.
7846 2009-05-07 14:48:29 +0200 John Millikin <jmillikin@gmail.com>
7848 * gst-libs/gst/tag/gstvorbistag.c:
7849 vorbistag: Store cover art in vorbiscomments
7852 2009-05-07 06:14:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7854 * gst-libs/gst/interfaces/colorbalance.c:
7855 * gst-libs/gst/interfaces/colorbalance.h:
7856 interfaces: API: Add gst_color_balance_get_balance_type()
7857 This is a convenience function that returns the balance_type
7858 of the interface struct.
7860 2009-05-06 17:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7862 * gst-libs/gst/interfaces/colorbalance.h:
7863 * gst-libs/gst/interfaces/colorbalancechannel.h:
7864 * gst-libs/gst/interfaces/tuner.h:
7865 * gst-libs/gst/interfaces/tunerchannel.h:
7866 interfaces: Separate struct definitions from typedefs
7868 2009-05-06 14:03:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7870 * pkgconfig/gstreamer-app-uninstalled.pc.in:
7871 Fix libdir for uninstalled gstreamer-app library
7873 2009-05-12 01:59:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7875 * gst-libs/gst/pbutils/descriptions.c:
7876 pbutils: add description for APE tag caps
7878 2009-05-12 01:35:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7881 configure: bump core requirement to last release
7882 as that's more likely to be true than that we need
7885 2009-05-12 01:21:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7889 configure: rename CVS -> git in a couple of places
7891 2009-05-12 01:17:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7894 configure: bump GLib requirement to GLib >= 2.16
7895 as per the New Regime (see wiki).
7897 2009-05-01 00:09:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7899 * gst-libs/gst/tag/gsttagdemux.c:
7900 tagdemux: cache events from upstream and re-send them once we have a source pad
7901 Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
7904 2009-05-07 14:07:44 -0700 Michael Smith <msmith@songbirdnest.com>
7906 * gst-libs/gst/riff/riff-media.c:
7907 riff: support UYVY raw 4:2:2 in riff.
7909 2009-05-11 21:20:07 +0100 Jan Schmidt <thaytan@noraisin.net>
7912 Back to development -> 0.10.23.1
7914 2009-04-27 22:42:55 -0700 Michael Smith <msmith@syncword.(none)>
7916 * ext/theora/theoradec.c:
7917 theoradec: fix buffer overrun on 422 decode.
7919 2009-04-27 21:39:01 -0700 Michael Smith <msmith@syncword.(none)>
7921 * ext/theora/theoradec.c:
7922 theoradec: 444 support.
7924 2009-04-27 21:30:04 -0700 Michael Smith <msmith@syncword.(none)>
7926 * ext/theora/theoradec.c:
7927 theoradec: handle 422 images (as YUY2).
7929 2009-04-27 21:01:51 -0700 Michael Smith <msmith@syncword.(none)>
7931 * ext/theora/gsttheoradec.h:
7932 * ext/theora/theoradec.c:
7933 theoradec: rearrange code in preparation for 422 and 444 support.
7935 === release 0.10.23 ===
7937 2009-05-10 23:57:01 +0100 Jan Schmidt <thaytan@noraisin.net>
7943 * docs/plugins/gst-plugins-base-plugins.args:
7944 * docs/plugins/gst-plugins-base-plugins.hierarchy:
7945 * docs/plugins/gst-plugins-base-plugins.interfaces:
7946 * docs/plugins/gst-plugins-base-plugins.prerequisites:
7947 * docs/plugins/gst-plugins-base-plugins.signals:
7948 * docs/plugins/inspect/plugin-adder.xml:
7949 * docs/plugins/inspect/plugin-alsa.xml:
7950 * docs/plugins/inspect/plugin-app.xml:
7951 * docs/plugins/inspect/plugin-audioconvert.xml:
7952 * docs/plugins/inspect/plugin-audiorate.xml:
7953 * docs/plugins/inspect/plugin-audioresample.xml:
7954 * docs/plugins/inspect/plugin-audiotestsrc.xml:
7955 * docs/plugins/inspect/plugin-cdparanoia.xml:
7956 * docs/plugins/inspect/plugin-decodebin.xml:
7957 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
7958 * docs/plugins/inspect/plugin-gdp.xml:
7959 * docs/plugins/inspect/plugin-gio.xml:
7960 * docs/plugins/inspect/plugin-gnomevfs.xml:
7961 * docs/plugins/inspect/plugin-libvisual.xml:
7962 * docs/plugins/inspect/plugin-ogg.xml:
7963 * docs/plugins/inspect/plugin-pango.xml:
7964 * docs/plugins/inspect/plugin-playback.xml:
7965 * docs/plugins/inspect/plugin-queue2.xml:
7966 * docs/plugins/inspect/plugin-subparse.xml:
7967 * docs/plugins/inspect/plugin-tcp.xml:
7968 * docs/plugins/inspect/plugin-theora.xml:
7969 * docs/plugins/inspect/plugin-typefindfunctions.xml:
7970 * docs/plugins/inspect/plugin-uridecodebin.xml:
7971 * docs/plugins/inspect/plugin-video4linux.xml:
7972 * docs/plugins/inspect/plugin-videorate.xml:
7973 * docs/plugins/inspect/plugin-videoscale.xml:
7974 * docs/plugins/inspect/plugin-videotestsrc.xml:
7975 * docs/plugins/inspect/plugin-volume.xml:
7976 * docs/plugins/inspect/plugin-vorbis.xml:
7977 * docs/plugins/inspect/plugin-ximagesink.xml:
7978 * docs/plugins/inspect/plugin-xvimagesink.xml:
7979 * gst-plugins-base.doap:
7980 * win32/common/_stdint.h:
7981 * win32/common/config.h:
7984 2009-05-10 23:56:05 +0100 Jan Schmidt <thaytan@noraisin.net>
8017 2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
8049 * win32/common/_stdint.h:
8050 * win32/common/config.h:
8051 0.10.22.6 pre-release
8053 2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8055 * gst/playback/gstplaysink.c:
8056 playbin2: fix resume after pause
8057 Don't ignore the state change of the children, they might be doing an ASYNC
8060 2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
8093 0.10.22.5 pre-release
8095 2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8097 * gst/tcp/gstmultifdsink.c:
8098 * gst/tcp/gsttcp-marshal.list:
8099 multifdsink: fix signature of the add-full signal
8100 The second parameter is a GstSyncMethod enum, not a boolean.
8102 2009-05-07 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8104 * gst/playback/gstplaysink.c:
8105 playsink: initialize variable too
8107 2009-05-07 14:28:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8109 * gst/playback/gstplaysink.c:
8110 playbin2: make playsink go ASYNC to PAUSED
8111 Make playsink go async to the PAUSED state instead of relying on uridecodebin
8112 for async behaviour in playbin. This solves some problems (mainly with DVD)
8113 where the pipeline would go to PLAYING before preroll completed, failing to
8114 select the audiosink clock.
8117 2009-05-06 16:09:52 +0100 Jan Schmidt <thaytan@noraisin.net>
8149 * win32/common/_stdint.h:
8150 * win32/common/config.h:
8151 0.10.22.4 pre-release
8153 2009-05-06 13:19:34 +0100 Zaheer Merali <zaheerabbas@merali.org>
8155 * ext/theora/theoraenc.c:
8156 * ext/vorbis/vorbisenc.c:
8157 vorbisenc, theoraenc: Ensure gp is computed consistently + clip to segment
8158 With vorbisenc, compute the granulepos with running time and clip incoming
8160 With theoraenc, drop out of segment buffers.
8162 2009-05-01 16:47:53 +0100 Jan Schmidt <thaytan@noraisin.net>
8164 * gst/audioresample/gstaudioresample.c:
8165 audioresample: Fix buffer size transformations
8166 When calculating the input/output buffer sizes in the transform_size function,
8167 take the number of channels into account, so we don't end up calculating
8168 a buffer size that only contains a partial number of audio frames.
8169 Also, when going from output size to input size, round down rather than
8170 up, so as to calculate the minimum number of samples that *might* yield
8171 a buffer of the intended destination size.
8172 Fixes: #580470 and #580952
8174 2009-04-29 16:45:27 +0100 Jan Schmidt <thaytan@noraisin.net>
8176 * ext/vorbis/gstvorbisenc.h:
8177 * ext/vorbis/vorbisenc.c:
8178 vorbisenc: Ensure output buffers fall within the segment
8179 Add the start position of the first segment to the running time
8180 used to generate buffer timestamps in vorbisenc. This avoids generating
8181 buffers which fall outside the initial segment. The element segment
8182 handling requires more extensive fixing, but this at least prevents
8183 regressions. Fixes: #580020
8185 2009-04-29 11:18:42 +0200 Andy Wingo <wingo@oblong.net>
8187 * gst-libs/gst/audio/gstbaseaudiosink.c:
8188 Revert "add can-activate-pull property to baseaudiosink"
8189 This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.
8191 2009-04-29 11:18:33 +0200 Andy Wingo <wingo@oblong.net>
8193 * gst-libs/gst/audio/gstbaseaudiosink.c:
8194 Revert "[baseaudiosink] add docs for can-activate-pull"
8195 This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.
8197 2009-04-28 18:48:33 +0200 Andy Wingo <wingo@oblong.net>
8199 [baseaudiosink] add docs for can-activate-pull
8200 * gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
8203 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
8205 add can-activate-pull property to baseaudiosink
8206 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
8209 2009-04-28 11:32:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8211 * gst/videorate/gstvideorate.c:
8212 * gst/videorate/gstvideorate.h:
8213 videorate: clear discont on duplicated buffers
8214 When videorate duplicates a buffer with a DISCONT flag, it copies the discont on
8215 the first pushed buffer but fails to clear it for subsequent buffers. This
8216 causes theoraenc!oggmux and possibly other elements to consider this a discont
8218 Fix videorate to produce discont as the first buffer and after a flushing seek.
8221 2009-04-24 18:13:00 +0100 Jan Schmidt <thaytan@noraisin.net>
8223 * tests/check/Makefile.am:
8224 check: Disable the playbin2 for this release, as it is a bit racy.
8225 Disable the test, as per the discussion in #580120. Needs re-enabling
8226 after the release, when playbin2 is fixed.
8228 2009-04-23 08:41:19 +0200 Edward Hervey <bilboed@bilboed.com>
8230 * gst/playback/gstdecodebin2.c:
8231 decodebin2: Don't reduce max-size-time of exposed groups. Fixes #579912
8232 The 2s limit is way too small for a lot of files (which have an interleave
8233 in time of between 3 and 5s). Instead, leave it to the initial 5s value
8234 and reduce the other limits (allowing us to stay memory-efficient).
8236 2009-04-21 21:06:59 +0100 Jan Schmidt <thaytan@noraisin.net>
8268 * win32/common/_stdint.h:
8269 * win32/common/config.h:
8270 0.10.22.3 pre-release
8272 2009-04-21 20:41:23 +0100 René Stadler <mail@renestadler.de>
8274 * gst/audioresample/gstaudioresample.c:
8275 audioresample: Fix unused variable in compilation with --disable-gst-debug
8278 2009-04-21 22:12:28 +0100 Jan Schmidt <thaytan@noraisin.net>
8281 Automatic update of common submodule
8282 From b3941ea to 6ab11d1
8284 2009-04-21 20:57:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8286 * gst/playback/gstplaybasebin.c:
8287 playbin: only use raw_decoding_mode when it's true
8288 First check the pad caps if they are raw before setting the raw_decoding_mode to
8289 TRUE. Fixes playback of transport streams and other streams that require large
8293 2009-04-19 18:15:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8295 * gst-libs/gst/cdda/gstcddabasesrc.c:
8296 * tests/check/libs/cddabasesrc.c:
8297 cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core
8298 Don't use REPLACE_ALL merge mode when that's not really what we want,
8299 as now that REPLACE_ALL actually does what it's supposed to do in
8300 core, we drop tags we wanted to keep, such as the various disc id
8301 tags. Add unit test for this as well. Fixes #579463.
8303 2009-04-17 10:34:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8305 * gst-libs/gst/rtsp/gstrtspconnection.c:
8306 rtspconnection: don't use GLib-2.16 API, we require only 2.14
8309 2009-04-17 10:55:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8311 * gst-libs/gst/audio/gstbaseaudiosink.c:
8312 baseaudiosink: don't unparent the ringbuffer
8313 when going to NULL, don't unparent the ringbuffer because we don't support going
8314 back to 0 very well yet.
8317 2009-04-17 10:53:10 +0200 Olivier Crete <tester at tester.ca>
8319 * gst-libs/gst/rtp/gstrtcpbuffer.c:
8320 RTCP: don't fail when retrieving invalid PT
8321 We can't meaningfully assert on valid packet types so just return the type as it
8322 is. Update the comments to reflect this.
8325 2009-04-16 12:12:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8327 * docs/libs/gst-plugins-base-libs-sections.txt:
8328 * gst-libs/gst/app/gstappsink.h:
8329 * gst-libs/gst/app/gstappsrc.h:
8330 app: add trivial cast macros
8331 Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
8332 and add the macros to the standard macros in the docs.
8335 2009-04-16 12:09:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8337 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
8338 pkgconfig: add the app/ directory to Libs
8339 Add the appsrc/appsink directory to the Libs in the uninstalled
8340 pkgconfig file so that one can build against it.
8343 2009-04-15 22:59:31 +0100 Jan Schmidt <thaytan@noraisin.net>
8346 0.10.22.2 pre-release
8348 2009-04-15 22:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
8351 ChangeLog: regenerate changelog with the gen-changelog script
8353 2009-04-16 00:41:13 +0100 Jan Schmidt <thaytan@noraisin.net>
8384 po: Update po files from TP
8386 2009-04-16 00:40:59 +0100 Jan Schmidt <thaytan@noraisin.net>
8388 * win32/common/_stdint.h:
8389 * win32/common/config.h:
8390 * win32/common/gstrtsp-enumtypes.c:
8391 * win32/common/interfaces-enumtypes.c:
8392 * win32/common/interfaces-enumtypes.h:
8393 * win32/common/video-enumtypes.c:
8394 win32: Update win32 build files
8396 2009-04-16 00:31:55 +0100 Jan Schmidt <thaytan@noraisin.net>
8398 * tests/check/libs/video.c:
8399 check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.
8401 2009-04-16 00:31:00 +0100 Jan Schmidt <thaytan@noraisin.net>
8403 * tests/check/elements/playbin2.c:
8404 check: Fix the input uri in playbin2 test.
8405 Don't try and use a random file in wim's home directory as a test input
8407 2009-04-15 15:35:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8409 * gst-libs/gst/video/video.h:
8410 video: Fix typo in the docs
8412 2009-04-15 14:53:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8414 * gst-libs/gst/video/video.c:
8415 * gst-libs/gst/video/video.h:
8416 video: Add support for YVYU YUV colorspace
8418 2009-04-15 00:17:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8420 * docs/libs/gst-plugins-base-libs-docs.sgml:
8421 * gst-libs/gst/fft/gstfft.c:
8422 docs: fix hyperlink and move fft attribution to the right place
8424 2009-04-15 00:02:39 +0300 Stefan Kost <ensonic@users.sf.net>
8426 * gst-libs/gst/audio/gstbaseaudiosink.c:
8427 log: use G_GUINT64_FORMAT instead of llu
8429 2009-04-14 18:31:52 +0200 Josep Torra <n770galaxy at gmail.com>
8431 * gst-libs/gst/rtsp/gstrtspdefs.c:
8432 * gst-libs/gst/rtsp/gstrtspdefs.h:
8433 RTSP: add missing headers for WMS RTSP
8434 Add missing headers related to Windows Media RTSP extension.
8437 2009-04-14 18:16:37 +0200 Olivier Crete <tester at tester.ca>
8439 * docs/design/draft-keyframe-force.txt:
8440 * ext/theora/gsttheoraenc.h:
8441 * ext/theora/theoraenc.c:
8442 theoraenc: implement upstream keyframe force
8443 Implement handling of upstream keyframe forcing.
8444 Update the design documents too.
8447 2009-04-14 17:31:31 +0200 Olivier Crete <tester at tester.ca>
8449 * ext/theora/theoraenc.c:
8450 theoraenc: factor out keyframe forcing
8453 2009-04-14 17:01:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8456 * gst-libs/gst/fft/gstfft.c:
8457 Give credit to Mark Borgerding (kissfft author)
8458 and add myself to AUTHORS as well. Fixes #575638.
8460 2009-04-14 17:04:06 +0200 Jan Urbanski <j.urbanski at students.mimuw.edu.pl>
8462 * gst/tcp/gstmultifdsink.c:
8463 * gst/tcp/gstmultifdsink.h:
8464 multifdsink: add property to resend streamheaders
8465 Adds a new property in multifdsink, resend-streamheader.
8466 If this property is false, the multifdsink will not send the streamheader if
8467 there's already one set for a particular client.
8468 There are some formats in which every stream needs to start with a certain
8469 blob, but you can't inject this blob at leisure. If the producer wants to
8470 change the blob in question and sets in as the streamheader on the outgoing
8471 buffers' caps, new clients of multifdsink will get the new streamheader, but
8472 old clients will break, because they'll see the blob in the middle of the
8474 The property is true by default, so existing code will not see any difference.
8477 2009-04-14 16:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8479 * gst/tcp/gstmultifdsink.c:
8480 * gst/tcp/gstmultifdsink.h:
8481 multifdsink: add property to handle client write
8482 Add a property to disable listening to client writes. This property is usefull
8483 when other code will deal with reading from the client socket.
8484 API: GstMultiFdSink::handle-read property
8486 2009-04-14 16:45:20 +0200 Johann Prieur <johann.prieur at gmail.com>
8488 * docs/libs/gst-plugins-base-libs-sections.txt:
8489 * gst-libs/gst/rtp/gstrtcpbuffer.c:
8490 * gst-libs/gst/rtp/gstrtcpbuffer.h:
8491 * win32/common/libgstrtp.def:
8492 RTCP: add beginnings of Feedback messages
8493 Add the beginnings of parsing and constructing Feedback messages.
8496 2009-04-14 13:51:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8498 * gst/playback/gstplaysink.c:
8499 playbin2: clear the target
8500 Clear the target of our ghostpads before we remove the pad from the element.
8501 This to make sure that the internal pad is not left linked to whatever pad we
8502 were ghosted to. This should only be a problem when we leak the ghostpads.
8503 Also release our subpicture pads.
8506 2009-04-14 12:10:30 +0100 Hannes Bistry <hannesb@gmx.net>
8508 * sys/ximage/ximagesink.c:
8509 ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
8512 2009-04-14 13:16:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8514 * gst-libs/gst/audio/gstbaseaudiosrc.c:
8515 baseaudiosrc: adjust the internal timestamp
8516 Adjust the internal timestamp before comparing it against the adjusted clock
8520 2009-04-14 13:12:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8522 * gst-libs/gst/audio/gstbaseaudiosink.c:
8523 baseaudiosink: use new clock time methods
8524 Use the unadjusted internal clock times to calculate the internal/external
8525 offset when calibrating the clock.
8526 When going to NULL, unparent and free the ringbuffer, like we do in the source
8530 2009-04-14 13:08:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8532 * gst-libs/gst/audio/gstaudioclock.c:
8533 * gst-libs/gst/audio/gstaudioclock.h:
8534 * win32/common/libgstaudio.def:
8535 audioclock: add methods for the internal offset
8536 Add two methods for getting the unadjusted time of the clock and one for
8537 adjusting an internal time. We will need these methods for correctly handling
8538 the time after a gst_audio_clock_reset().
8539 Add a debug category and some debug lines to the audio clock.
8540 API: gst_audio_clock_get_time()
8541 API: gst_audio_clock_adjust()
8542 API: GST_AUDIO_CLOCK_CAST()
8544 2009-04-14 11:34:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8546 * gst/playback/gstdecodebin2.c:
8547 decodebin2: fix up the debugs and warnings
8548 Use _OBJECT variants because we can. Go over some log statements and put them in
8552 2009-04-12 22:26:33 +0200 Luca Ognibene <luca.ognibene at gmail.com>
8554 * gst/tcp/gstmultifdsink.c:
8555 multifdsink: fix error in sync-method
8556 Multifdsink did not handle sync-method=latest-keyframe correctly when the
8557 soft-limit is set to -1 (unlimited).
8560 2009-04-10 21:49:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8562 * gst-libs/gst/audio/gstbaseaudiosink.c:
8563 baseaudiosink: use the internal clock time
8564 We can't assume that the internal clock time is the same as the function we
8565 installed on our provided clock because somebody might have changed it.
8567 2009-04-10 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8569 * tests/examples/seek/seek.c:
8570 seek: handle clock-lost messages
8571 When we receive a clock-lost message we need to pause and play to select a new
8574 2009-04-10 13:44:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8576 * tests/check/Makefile.am:
8577 * tests/check/elements/playbin2.c:
8578 check: add a unit test for playbin2
8579 Add unit test for playbin2 and include the refcount test in #577794.
8581 2009-04-10 13:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8583 * gst/playback/gstplaysink.c:
8584 playbin2: fix refcounting of visualisations
8587 2009-04-10 13:27:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8589 * gst/playback/gstplaysink.c:
8590 playsink: fix refcounting of custom elements
8591 Sink the custom sinks, let other elements we create be sunken by the bin we add
8595 2009-04-10 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8597 * tests/check/elements/appsink.c:
8598 check: fix appsink test
8599 Fix the appsink test now that the method signature changed.
8601 2009-04-10 12:26:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8603 * gst/playback/gstplaybin2.c:
8604 playbin2: handle missing input-selector
8605 Gracefully degrade and disable stream selection when input-selector is
8608 2009-04-09 23:46:17 +0200 Martin Samuelsson <martin.samuelsson at axis.com>
8610 * gst-libs/gst/app/gstappsink.c:
8611 * gst-libs/gst/app/gstappsink.h:
8612 appsink: make callbacks return GstFlowReturn
8613 Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
8614 errors can be reported properly.
8617 2009-04-09 18:04:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8619 * gst-libs/gst/audio/gstringbuffer.c:
8620 * gst-libs/gst/audio/gstringbuffer.h:
8621 ringbuffer: allow for custom commit functions
8622 Allow subclasses to override the commit method.
8624 2009-04-08 18:04:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8626 * gst-libs/gst/audio/gstbaseaudiosink.c:
8627 baseaudiosink: fix a small glitch after pause
8628 After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
8629 the amount of output samples we consumed. We can't do this reliably with the
8630 current API when we are doing trick modes but we can do the right thing for
8633 2009-04-08 16:43:27 +0300 Stefan Kost <ensonic@users.sf.net>
8635 * gst/playback/gstplaysink.c:
8636 playbin2: better error message on sink failure
8637 If we could create the sinks, but the don't work, don't send the missing plugin
8638 message and report that the state-changed failed.
8640 2009-04-07 22:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
8642 * gst-libs/gst/audio/gstaudiofilter.c:
8643 audiofilter: don't leak pad-template
8644 gst_element_class_add_pad_template() does not take ownership.
8646 2009-04-04 21:18:38 +0300 Felipe Contreras <felipe.contreras@gmail.com>
8649 Automatic update of common submodule
8650 From d0ea89e to b3941ea
8652 2009-04-04 16:28:14 +0200 Edward Hervey <bilboed@bilboed.com>
8654 * gst-libs/gst/interfaces/navigation.c:
8655 * sys/v4l/v4lsrc_calls.c:
8656 navigation/v4l: Don't use g_return_val_if_fail for computed/used values.
8658 2009-03-22 09:46:37 +0100 Edward Hervey <bilboed@bilboed.com>
8660 * ext/theora/theoradec.c:
8661 theoradec: return GST_CLOCK_TIME_NONE for negative framecounts.
8662 This fixes most seeking issues when used with gnonlin.
8665 2009-04-04 14:53:42 +0200 Edward Hervey <bilboed@bilboed.com>
8668 Automatic update of common submodule
8669 From f8b3d91 to d0ea89e
8671 2009-04-03 10:51:42 -0700 Michael Smith <msmith@songbirdnest.com>
8673 * gst/playback/gstplaybin2.c:
8674 playbin2: don't leak selector when getting current stream numbers.
8676 2009-04-02 22:28:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8678 * gst-libs/gst/rtsp/gstrtspconnection.c:
8679 rtsp: use fully qualified urls when using a proxy
8680 Use a fully qualified url when specifying the url for tunneled requests through
8684 2009-03-31 00:54:30 +0100 Jan Schmidt <thaytan@noraisin.net>
8686 * docs/libs/gst-plugins-base-libs-sections.txt:
8687 * gst-libs/gst/interfaces/navigation.c:
8688 * gst-libs/gst/interfaces/navigation.h:
8689 * tests/check/Makefile.am:
8690 * tests/check/libs/.gitignore:
8691 * tests/check/libs/navigation.c:
8692 * win32/common/libgstinterfaces.def:
8693 navigation: Extend the navigation interface
8694 Add support for a set of standard commands that can be queried and executed to
8695 support applications like DVD. Add query construction and parsing functions.
8696 Add new messages that can be sent on the bus to provide notifications related
8697 to commands, multiangle changes, and button highlight activity.
8698 Add some helper functions to parse the existing GstNavigation events that
8699 elements might receive.
8700 Document it all and add unit tests.
8702 2009-02-04 17:03:07 +0000 Jan Schmidt <thaytan@noraisin.net>
8704 * gst/playback/gstplaybasebin.c:
8705 * gst/playback/gstplaybasebin.h:
8706 playbin: Add simple 'raw decoding mode'.
8707 Raw decoding mode removes almost all buffering in video and audio queues
8708 when a source providing already decoded video/audio is detected, on the
8709 possibly bogus assumption that such a source should provide sufficient
8710 internal queueing. Fixes playback on some DVDs, and improves it
8713 2009-04-02 09:27:07 +0100 Jan Schmidt <thaytan@noraisin.net>
8715 * tests/check/elements/.gitignore:
8716 ignores: Ignore the videoscale check binary
8718 2009-04-02 12:13:57 +0100 Jan Schmidt <thaytan@noraisin.net>
8720 * win32/common/libgstrtsp.def:
8721 win32: Add gst_rtsp_connection_set_proxy to the win32 exports
8723 2009-04-02 10:42:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8725 * ext/alsa/gstalsamixer.c:
8726 alsamixer: don't forget to release locks in a few places
8729 2009-04-02 11:10:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8731 * gst/videoscale/vs_4tap.c:
8732 videoscale: Don't read over line ends when taking the last Cr or Cb
8734 2009-04-02 10:52:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8736 * gst/videoscale/vs_4tap.c:
8737 videoscale: Don't write to few pixels and don't mix Cr and Cb
8740 2009-04-01 15:15:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8742 * gst/audioresample/gstaudioresample.c:
8743 * tests/check/elements/audioresample.c:
8744 audioresample: fix negotiation so that upstream can actually fixate to downstream's rate
8745 If one side has a preference for a particular sample rate or set of sample rates, we
8746 should honour this in the caps we advertise and transform to and from, so that elements
8747 actually know about the other side's sample rate preference and can negotiate to it
8748 if supported. Also add unit test for this.
8750 2009-03-26 19:34:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8752 * gst/playback/gstplaybin2.c:
8753 docs: add a blurb about redirect messages to playbin2 docs
8755 2009-04-01 09:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8757 * gst-libs/gst/rtsp/gstrtspconnection.c:
8758 rtsp: fix little typo in the comments
8760 2009-03-31 17:52:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8762 * gst-libs/gst/rtsp/gstrtspconnection.c:
8763 rtspconnection: make gst_rtsp_watch_queue_message() thread-safe
8764 People might queue messages from a thread other than the thread in which
8765 the main context which this watch is attached is iterated from, so use
8766 a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
8767 over list nodes just freed in the other thread. This just fixes issues
8768 I've had with gst-rtsp-server. We might need more locking in various
8771 2009-03-31 18:13:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8773 * gst-libs/gst/rtsp/gstrtspconnection.c:
8774 * gst-libs/gst/rtsp/gstrtspmessage.c:
8775 rtsp: clear the entire builder structure
8776 And use structure instead of variable with sizeof when
8777 clearing the rtsp message structure, for clarity.
8779 2009-03-31 17:56:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8781 * gst-libs/gst/rtsp/gstrtspmessage.c:
8782 docs: fix typo in gst_rtsp_message_unset() API docs
8784 2009-03-31 19:00:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8786 * gst-libs/gst/rtsp/gstrtspconnection.c:
8787 * gst-libs/gst/rtsp/gstrtspconnection.h:
8788 rtsp: add support for proxies
8789 Add suport for proxy servers. Currently only used for tunneled HTTP
8790 connections without authentication.
8792 2009-03-31 18:57:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8794 * gst-libs/gst/rtsp/gstrtspmessage.c:
8795 Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
8796 This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.
8798 2009-03-26 18:54:56 +0200 Stefan Kost <ensonic@users.sf.net>
8800 * sys/xvimage/xvimagesink.c:
8801 xvimagesink: use xcontext->depth instead of bits in attr.max_value for colorkey
8802 According to the drivers in http://cgit.freedesktop.org/xorg/driver/ we should
8803 format the colorkey depending on xcontext->depth. This is what they will use to
8804 interprete the value. The max_value in turn is usualy a constant regardless of
8807 2009-03-31 12:22:14 +0300 Stefan Kost <ensonic@users.sf.net>
8809 * gst-libs/gst/rtsp/gstrtspmessage.c:
8810 rtsp: reset whole message (was sizeof pointer instead of sizeof type)
8812 2009-03-31 00:56:18 +0100 Jan Schmidt <thaytan@noraisin.net>
8814 * gst-libs/gst/interfaces/mixer.c:
8815 doc: Fix a typo in the GstMixer docs
8817 2009-03-29 12:01:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8819 * gst/videoscale/vs_scanline.c:
8820 videoscale: Fix linear scaling for one byte components
8823 2009-03-29 11:53:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8825 * gst/videoscale/vs_4tap.c:
8826 videoscale: Fix 4tap scaling of YUYV and friends
8828 2009-03-28 16:08:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8830 * gst/videoscale/vs_image.c:
8831 * gst/videoscale/vs_scanline.c:
8832 * gst/videoscale/vs_scanline.h:
8833 videoscale: Rewrite YUYV (and friends) scaling and don't read/write over line ends
8834 Partially fixes bug #577054, there's just one issue left now.
8836 2009-03-28 12:48:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8838 * tests/check/elements/videoscale.c:
8839 videoscale: Add some more unit tests
8841 2009-03-28 11:51:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8843 * gst/videoscale/gstvideoscale.c:
8844 videoscale: Use bilinear instead of 4tap scaling for heights < 4
8845 Partially fixes bug #577054.
8847 2009-03-28 11:45:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8849 * gst/videoscale/vs_scanline.c:
8850 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY/RGB/RGBA
8851 This case is for upscaling a frame with width=1
8852 Partially fixes bug #577054.
8854 2009-03-28 11:27:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8856 * gst/videoscale/vs_scanline.c:
8857 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
8858 Partially fixes bug #577054.
8860 2009-03-28 10:40:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8862 * gst/videotestsrc/gstvideotestsrc.c:
8863 videotestsrc: Initialize buffer memory with zeroes
8864 This prevents valgrind warnings when accessing the "x" parts
8865 of xRGB and friends in other elements that handle (and can handle)
8866 xRGB like ARGB (for example videoscale).
8868 2009-03-28 10:25:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8870 * tests/check/Makefile.am:
8871 * tests/check/elements/videoscale.c:
8872 videoscale: Add a lot of unit tests
8874 2009-03-28 10:06:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8876 * gst/videoscale/gstvideoscale.c:
8877 videocale: Add support for video/x-raw-gray with bpp=depth=8
8879 2009-03-28 10:01:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8881 * gst/videotestsrc/videotestsrc.c:
8882 videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8
8884 2009-03-28 09:43:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8886 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
8887 ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format
8889 2009-03-27 19:12:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8891 * gst/videoscale/vs_4tap.c:
8892 videoscale: Take the next luma value instead of every second next when scaling UYVY and friends
8894 2009-03-27 19:09:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8896 * gst/videoscale/gstvideoscale.c:
8897 videoscale: Add support for v308 YUV colorspace
8899 2009-03-27 13:15:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8901 * gst/videoscale/vs_4tap.c:
8902 videoscale: Add my copyright to the 4tap scalers
8904 2009-03-27 13:14:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8906 * gst/videoscale/gstvideoscale.c:
8907 videoscale: Enable 4-tap scaling for all supported formats
8909 2009-03-27 13:14:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8911 * gst/videoscale/vs_4tap.c:
8912 * gst/videoscale/vs_4tap.h:
8913 videoscale: Implement 4-tap scaling for RGB565 and RGB555
8915 2009-03-27 10:47:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8917 * gst/videoscale/vs_4tap.c:
8918 * gst/videoscale/vs_4tap.h:
8919 videoscale: Implement 4-tap scaling for UYVY
8921 2009-03-27 09:33:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8923 * gst/videoscale/vs_4tap.c:
8924 * gst/videoscale/vs_4tap.h:
8925 videoscale: Implement 4-tap scaling for YUY2 and YVYU
8927 2009-03-26 22:14:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8929 * gst/videoscale/vs_4tap.c:
8930 * gst/videoscale/vs_4tap.h:
8931 videoscale: Implement 4-tap scaling for RGB and BGR
8933 2009-03-26 22:08:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8935 * gst/videoscale/vs_4tap.c:
8936 * gst/videoscale/vs_4tap.h:
8937 videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats
8939 2009-03-26 11:02:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
8941 * ext/pango/gsttextoverlay.c:
8942 textoverlay: Fix drawing of UYVY text borders
8944 2009-03-26 10:36:27 +0100 Zeeshan Ali <zeeshan.ali@nokia.com>
8946 * ext/pango/gsttextoverlay.c:
8947 * ext/pango/gsttextoverlay.h:
8948 textoverlay: Add support for UYVY colorspace
8951 2009-03-25 19:01:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8953 * gst/playback/gstdecodebin2.c:
8954 decodebin2: do some more cleanup
8955 Free the groups when we go to READY.
8956 Allow for NO_PREROLL elements.
8958 2009-03-25 16:37:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8960 * gst-libs/gst/rtsp/gstrtspconnection.c:
8961 rtsp: start CSeq counting from 1 instead of 0
8962 Start counting from 1 instead of 0 as this is what most other clients
8965 2009-03-25 16:35:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8967 * gst-libs/gst/rtsp/gstrtspdefs.c:
8968 * gst-libs/gst/rtsp/gstrtspdefs.h:
8969 rtsp: add ETag and If-Match headers
8970 Add new headers, we need them for RealMedia support.
8972 2009-03-25 14:16:25 +0200 Stefan Kost <ensonic@users.sf.net>
8974 * sys/xvimage/xvimagesink.c:
8975 xvimagesink: scale the colorkey components in case of 16bit visuals
8976 Use a default that won't be scales to 0,0,0
8978 2009-03-25 11:27:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8980 * gst-libs/gst/audio/gstbaseaudiosrc.c:
8981 audiosrc: improve 'Dropped n samples' warning message
8983 2009-03-24 19:41:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8985 * tests/examples/app/appsrc-ra.c:
8986 * tests/examples/app/appsrc-seekable.c:
8987 examples: use new method to set flags
8988 Use the new core method for setting object enum properties by name.
8990 2009-03-24 18:29:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8992 * gst/playback/gstplaysink.c:
8993 * gst/playback/gstplaysink.h:
8994 playbin2: add more support for subpictures
8996 2009-03-24 17:12:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8998 * gst/playback/gstplaybin2.c:
8999 * gst/playback/gstplaysink.c:
9000 * gst/playback/gstplaysink.h:
9001 playbin2: first support for subpictures
9002 Add beginnings of subpicture support.
9004 2009-03-24 15:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9006 * tests/examples/seek/seek.c:
9007 seek: print tags from the different tracks
9009 2009-03-24 12:22:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9011 * gst/playback/gstplaybin2.c:
9012 playbin2: blacklist subpictures for now
9013 Blacklist the subpictures until we add support for them.
9014 Add some small debug info.
9017 2009-03-24 12:19:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9019 * gst/playback/gsturidecodebin.c:
9020 uridecodebin: expose more media types
9021 Expose more media types from a raw source, such as the subpicture and various
9023 Small cleanups and add some more debugging.
9026 2009-03-24 10:42:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9028 * gst/playback/gstplaysink.c:
9029 playbin2: rescan audio sinks for volume/mute
9030 Rescan the audio sinks for the mute and volume properties.
9033 2009-03-23 19:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9035 * gst/playback/gstplaysink.c:
9036 playbin2: fix reuse of the video chains
9037 When reusing playbin with visualisations, reset the async property on the video
9038 sink because some sinks might dynamically recreate their sinks.
9041 2009-03-23 17:37:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9043 * gst/playback/gstplaysink.c:
9044 playbin2: allow dynamic swtiching of subtitles
9045 When we have the textpad configured, enable and disable the subtitles by setting
9046 the silent flag on the overlay element instead of trying to remove elements.
9049 2009-03-23 16:59:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9051 * tests/icles/playbin-text.c:
9052 tests: print some more info in the text example
9053 Print both the position and the running_time when the subtitle becomes available
9056 2009-03-23 16:04:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9058 * gst/playback/gstplaysink.c:
9059 playbin2: fix dynamic switching of visualisations
9060 Fix the switching of visualisations by requesting and releasing the tee request
9064 2009-03-23 16:19:11 +0200 Stefan Kost <ensonic@users.sf.net>
9067 * gst/tcp/gsttcpclientsink.c:
9068 * gst/tcp/gsttcpclientsrc.c:
9069 * gst/tcp/gsttcpserversink.c:
9070 * gst/tcp/gsttcpserversrc.c:
9071 docs: add examples for tcp elements, also use correct section name. Fixes #564139
9072 Updated the examples in the README to actually work. Add them to api docs. Tests
9073 the api-docs and fix the section names to make the docs actualy show up.
9074 The example for "tcpserversrc" needs review (might be an element bug).
9076 2009-03-17 09:14:02 +0200 Stefan Kost <ensonic@users.sf.net>
9078 * gst/videoscale/gstvideoscale.c:
9079 indent: fix damange that gst-indent did some time ago
9081 2009-03-23 15:27:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9083 * gst/playback/gstplaysink.c:
9084 playbin2: fix linking order
9085 Link after doing the state change and unlink before shutting down. Makes the
9086 window for causing races in toggling the visualisations smaller.
9089 2009-03-23 12:26:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9091 * gst/playback/gsturidecodebin.c:
9092 uridecodebin: reset counter
9093 reset the number of pending dynamic operations back to 0 when we reuse
9097 2009-03-23 11:38:53 +0100 Edward Hervey <bilboed@bilboed.com>
9099 * ext/theora/theoradec.c:
9100 theoradec: Use GST_CLOCK_TIME_NONE for invalid positions. Fixes #543591
9101 The problem was that previously we didn't check whether _theora_granule_frame
9102 returned a negative framecount or not, resulting in bogus timestamps.
9104 2009-03-21 09:46:28 +0100 René Stadler <mail@renestadler.de>
9106 * ext/vorbis/vorbisenc.c:
9107 vorbisenc: Set caps on non-header ouput buffers.
9110 2009-03-20 16:13:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9112 * tests/examples/seek/seek.c:
9113 seek: Add some more debug
9114 Add some more info about the selected streams.
9116 2009-03-20 15:47:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9118 * gst/playback/gstdecodebin2.c:
9119 decodebin2: a pad starts out being not drained.
9120 Mark a new pad as not drained until we get EOS on it.
9122 2009-03-20 14:17:19 +0100 LRN <lrn1986 at gmail dot com>
9124 * gst/playback/gstqueue2.c:
9125 win32: fix seeking in large files
9126 Fix Seeking in large files by using the 64-bit seek functions.
9129 2009-03-19 20:31:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9131 * gst/playback/gstdecodebin2.c:
9132 decodebin2: recover from failing to add a pad
9133 When we cannot add a pad to the decodebin2 for some reason, print a warning but
9134 continue adding the remaining pads.
9136 2009-03-19 19:35:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9138 * gst/playback/gstdecodebin2.c:
9139 decodebin2: more cleanups and docs.
9140 Add some more comments and use g_list_prepend().
9142 2009-03-19 19:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9144 * gst/playback/gstdecodebin2.c:
9145 decodebin2: refactoring and race fixes
9146 Refactor some code so that we can take the right locks and in the right order.
9147 Fixes quite a bit of races already.
9149 2009-03-19 19:03:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9151 * gst/playback/gstplaybin2.c:
9152 playbin2: remove the group cond + cleanups
9153 Remove the group GCond that we used for waiting for groups to finish because we
9154 use pad blocking on the selectors and counters instead for waiting for the
9156 remove the obsolete about_to_finish variable set while emiting the
9157 about-to-finish signal and fix some old comments.
9158 We don't need to take the playbin lock when querying the uridecodebin.
9160 2009-03-18 10:45:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9162 * tests/icles/playbin-text.c:
9163 icles: print better error and warning messages
9166 2009-03-17 22:53:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9168 * gst-libs/gst/rtsp/gstrtspbase64.c:
9169 * gst-libs/gst/rtsp/gstrtspbase64.h:
9170 rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode
9171 This also fixes another instance of CVE-2008-4316.
9173 2009-03-17 19:53:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9175 * ext/ogg/gstoggdemux.c:
9176 oggdemux: report -1 for duration in push mode
9177 In push mode we must return TRUE from the duration query with a value of -1
9178 meaning that we know that we don't know the duration.
9180 2009-03-17 19:09:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9182 * gst/playback/gstdecodebin2.c:
9183 decodebin2: add extra dynamic ref for demuxers
9184 When we make a group connected to a demuxer, keep an extra dynamic refcount for
9185 the group which is only decremented when no_more_pads or a multiqueue overrun is
9186 detected. This way we avoid a race between exposing the group while more dynamic
9187 refs are added from new pads.
9190 2009-03-17 15:39:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9192 * gst/playback/gstplaysink.c:
9193 playbin2: sync state of the sink correctly
9194 Sync the state of the newly added chains to the state of the parent sink element
9195 to avoid lost async-start messages. Fixes cdda:// async-done message storm.
9197 2009-03-17 11:54:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9199 * gst/playback/gstplaybin2.c:
9200 playbin2: return NOT_LINKED for unselected streams
9201 When streams are not selected in the selector, return NOT_LINKED so that
9202 upstream elements can skip decoding. Only do this for audio and video pads
9203 because for text streams the overhead is smaller and they could come from
9206 2009-03-17 11:51:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9208 * gst/playback/gstplaysink.c:
9209 playbin: set custom text sink properties
9210 Set the custom sink async=FALSE to not make it participate in preroll because we
9211 are dealing with sparse streams.
9212 Try to set sync=TRUE on the custom text sink.
9214 2009-03-17 11:30:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9216 * tests/icles/playbin-text.c:
9217 example: use appsink instead of fakesink
9218 Use appsink instead of fakesink to get the subtitles.
9219 Make things more pretty.
9221 2009-03-17 11:24:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9223 * tests/icles/.gitignore:
9224 * tests/icles/Makefile.am:
9225 * tests/icles/playbin-text.c:
9226 examples: add example of intercepting subtitles
9227 Add an example of how to install a custom sink for receiving subtitles in
9230 2009-03-17 11:03:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9232 * tests/check/elements/appsink.c:
9233 tests: fix include in the appsink test
9234 Fix dist by doing the right include.
9236 2009-03-16 16:42:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9238 * gst/playback/gstplaybin2.c:
9239 playbin2: don't try to set invalid stream numbers
9240 Fix a problem with setting the stream numbers because we check for the wrong
9244 2009-03-16 16:16:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9246 * gst/playback/gstplaybin2.c:
9247 playbin2: release the shutdown lock
9248 Release the shutdown lock when we wait for other groups to complete or else we
9249 have a deadlock when the other group completes and tries to grab the shutdown
9253 2009-03-16 15:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9255 * tests/examples/app/appsrc-ra.c:
9256 * tests/examples/app/appsrc-seekable.c:
9257 * tests/examples/app/appsrc-stream.c:
9258 * tests/examples/app/appsrc-stream2.c:
9259 examples: fix g_object_set() value type.
9260 Make sure we cast the length value as a gint64 to the vararg g_object_set() just
9261 incase sizeof(gsize) != sizeof(gint64).
9263 2009-03-15 19:57:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9265 * gst/typefind/gsttypefindfunctions.c:
9266 typefinding: make flac typefinder return lower probability for frame headers
9267 The flac frame header typefinder overstates the likelihood of a match, leading
9268 to false positives with e.g. aac streams and PDF files. Reduce probabilty
9269 returned from LIKELY to POSSIBLE for the frame header matchin code.
9272 2009-03-11 12:59:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9274 * gst/typefind/gsttypefindfunctions.c:
9275 typefinding: improve image/bmp typefinder
9276 Detect more variations and also bail out in more cases where the values
9277 don't make sense. Furthermore, add width/height and bpp to the caps,
9280 2009-03-13 15:22:42 +0000 Jan Schmidt <thaytan@noraisin.net>
9282 * tests/check/Makefile.am:
9283 check: Ignore alsamixer in the states test too
9285 2009-03-13 15:22:11 +0000 Jan Schmidt <thaytan@noraisin.net>
9287 * sys/v4l/v4l_calls.c:
9288 v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.
9290 2009-03-13 16:19:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9292 * gst-libs/gst/rtsp/gstrtspconnection.c:
9293 rtsp: fix resolving of hostnames
9294 We were returning a pointer to a stack variable with the resolved hostname,
9296 return a copy of the resolved ip address instead.
9299 2009-03-13 15:29:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9301 * ext/vorbis/vorbisparse.c:
9302 vorbisparse: be smarter when queueing headers
9303 Look at the first buffer byte to see if a buffer is a header instead of counting
9306 2009-03-13 15:27:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9308 * ext/theora/gsttheoraparse.h:
9309 * ext/theora/theoraparse.c:
9310 theoraparse: be smarter when queuing headers
9311 Look at the first byte of the buffer data (if we can) to decide if the packet is
9312 a header packet or not instead of counting packets.
9314 2009-03-13 15:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9316 * ext/ogg/gstoggdemux.c:
9317 oggdemux: add some debug info
9318 Add some debug info to log when the seek worked.
9320 2009-03-13 15:14:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9322 * gst-libs/gst/app/gstappsrc.c:
9323 appsrc: release lock in _eos flushing case
9324 Release the mutex when we are flushing in gst_app_src_end_of_stream()
9327 2009-03-13 11:49:10 +0000 Jan Schmidt <thaytan@noraisin.net>
9329 * ext/vorbis/vorbisdec.c:
9330 vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.
9332 2009-03-13 11:48:28 +0000 Jan Schmidt <thaytan@noraisin.net>
9334 * ext/theora/theoradec.c:
9335 theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.
9337 2009-03-12 18:27:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9339 * gst/playback/gsturidecodebin.c:
9340 playbin2: fix raw elements like cdda://
9341 Fix a fixme with a one liner and make cd playback work again.
9343 2009-03-12 17:47:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9345 * gst/playback/gstplaybin2.c:
9346 * gst/playback/gstplaysink.c:
9347 * gst/playback/gstplaysink.h:
9348 playbin2: improve subtitle handling
9349 Add property to playbin2 to configure a custom sink that receives the raw
9350 subtitle buffers instead of using a textoverlay.
9351 Improve the property finding code to make it more usable.
9352 Use property find code to find async properties in custom sinks that are bins.
9353 Improve text overlay code to gracefully handle missing elements.
9355 2009-02-24 15:58:42 +0000 Jan Schmidt <thaytan@noraisin.net>
9357 * gst-libs/gst/tag/gstvorbistag.c:
9358 vorbistag: Protect memory allocation calculation from overflow.
9359 Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
9361 2009-03-12 11:34:20 +0000 Jan Urbanski <jurbanski@flumotion.com>
9363 * gst-plugins-base.spec.in:
9366 2009-03-11 18:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9368 * gst-libs/gst/rtsp/gstrtspconnection.c:
9369 rtsp: fix parsing of the timeout parameter
9372 2009-03-11 16:20:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9374 * gst-libs/gst/rtsp/gstrtspmessage.c:
9375 rtsp: fix g_return condition
9376 when parsing a data message, we require a data message.
9378 2009-03-11 13:33:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9380 * gst/typefind/gsttypefindfunctions.c:
9381 typefinding: flac typefinder fixes
9382 Use scan context for initial peek as well. Peek 6 bytes in the initial
9383 peek rather than 5 bytes, to match the length of the memcmp we're doing
9384 on that data later. Return immediately when we found caps from looking
9385 at the beginning of the data - no point in continuing to scan the next
9386 64kB for something matching a frame header.
9388 2009-03-11 14:08:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9390 * gst-libs/gst/rtsp/gstrtspmessage.c:
9391 rtsp: free the right string.
9392 Free the key value before we remove the header item from the array. The item we
9393 retrieved from the array is only valid until we remove it from the array.
9395 2009-03-11 14:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9397 * gst-libs/gst/rtsp/gstrtspconnection.c:
9398 rtsp: keep track of amount of decoded bytes
9399 Keep track of the actual amount of decoded bytes, which can be less than 3 when
9400 we decode the last bits of a base64 message.
9402 2009-03-10 21:00:26 +0200 Stefan Kost <ensonic@users.sf.net>
9404 * gst/adder/gstadder.c:
9405 adder: log details in getcaps like in setcaps
9407 2009-03-10 13:11:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9410 win32: update MANIFEST, fixing 'make dist'
9412 2009-03-09 23:12:00 +0000 Jan Schmidt <thaytan@noraisin.net>
9415 Automatic update of common submodule
9416 From 7032163 to f8b3d91
9418 2009-03-09 16:19:40 +0100 Jonathan Matthew <notverysmart at gmail dot com>
9420 * gst/typefind/gsttypefindfunctions.c:
9421 typefind: add photoshop typefind functions
9422 Add photoshop typefind functions.
9425 2009-03-09 15:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9427 * gst/playback/gstdecodebin2.c:
9428 decodebin2: only remove pads that were added
9429 Flag pads that were added so that we can see if we need to remove them later or
9432 2009-03-09 13:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9434 * gst-libs/gst/rtsp/gstrtsptransport.c:
9435 rtsp: only add ports when not using TCP
9436 Only add the port numbers in the transport string when we are using udp or
9439 2009-03-09 13:53:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9441 * gst-libs/gst/rtsp/gstrtspmessage.c:
9442 rtsp: use gstreamer dump mem
9445 2009-03-09 13:51:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9447 * gst-libs/gst/rtsp/gstrtspconnection.c:
9448 rtsp: use glib base64 encoder
9451 2009-03-06 19:28:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9453 * gst/playback/gstdecodebin2.c:
9454 Unblock blocked ghostpads when shutting down. Fixes #574293.
9456 2009-03-09 10:03:13 +0100 Edward Hervey <bilboed@bilboed.com>
9458 * gst-libs/gst/riff/riff-media.c:
9459 Riff: Add mapping for Fraps video codec.
9460 Found through insanity testrun. Confirmed mapping in libavformat.
9462 2009-03-09 09:07:13 +0100 Edward Hervey <bilboed@bilboed.com>
9464 * gst-libs/gst/riff/riff-media.c:
9465 riff: Add the 'DVR ' mapping for mpeg2video.
9466 Found this in 3 files from the insanity suite and mapping is also present
9469 2009-03-09 09:06:40 +0100 Edward Hervey <bilboed@bilboed.com>
9471 * gst/typefind/gsttypefindfunctions.c:
9472 typefind: Use the proper data pointer instead of poking random memory.
9474 2009-03-08 18:17:48 +0100 LRN <lrn1986@gmail.com>
9476 * gst-libs/gst/rtsp/gstrtspconnection.c:
9477 rtsp: fix compilation on windows.
9478 Remove unused variable when building for windows.
9481 2009-03-08 12:03:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9484 Automatic update of common submodule
9485 From ffa738d to 7032163
9487 2009-03-08 11:19:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9490 Automatic update of common submodule
9491 From 3f13e4e to ffa738d
9493 2009-03-07 11:44:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9496 Automatic update of common submodule
9497 From 3c7456b to 3f13e4e
9499 2009-03-07 10:44:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
9502 Automatic update of common submodule
9503 From 57c83f2 to 3c7456b
9505 2009-03-06 19:02:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9507 * ext/theora/theoradec.c:
9508 theoradec: parse and use codec_data in the caps
9509 Parse the codec_data in the caps and use this as the headers.
9512 2009-03-06 18:53:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9514 * gst-libs/gst/riff/riff-media.c:
9515 riff: add theora mapping
9516 Add theora mappings. See #574169.
9518 2009-03-06 16:31:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9520 * gst-libs/gst/rtsp/gstrtspconnection.c:
9521 * gst-libs/gst/rtsp/gstrtspconnection.h:
9522 * win32/common/libgstrtsp.def:
9523 rtsp: Add methods for getting the read/write fds
9524 API:gst_rtsp_connection_get_readfd()
9525 API:gst_rtsp_connection_get_writefd()
9527 2009-03-06 10:35:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9530 * win32/common/audio-enumtypes.c:
9531 win32: indent copied *-enumtypes.c files in make win32-update
9533 2009-03-06 10:35:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9536 win32: update MANIFEST
9538 2009-03-06 10:30:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9541 * win32/common/config.h:
9542 win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define
9544 2009-03-06 10:05:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9546 * win32/common/_stdint.h:
9547 * win32/common/config.h:
9548 * win32/common/gstrtsp-enumtypes.c:
9549 * win32/common/interfaces-enumtypes.c:
9550 * win32/common/multichannel-enumtypes.c:
9551 * win32/common/pbutils-enumtypes.c:
9552 * win32/common/video-enumtypes.c:
9553 * win32/common/video-enumtypes.h:
9554 win32: update windows files via make win32-update
9555 Updates win32 files using the new system/hook, and defines HAVE_PROCESS_H,
9556 which fixes the build of pbutils on windows (#574319).
9558 2009-03-06 10:03:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9561 gitignore: ignore more
9563 2009-03-06 10:37:38 +0100 Julien Moutte <julien@fluendo.com>
9565 * gst-libs/gst/rtsp/gstrtspconnection.c:
9566 Fix build on Mac OS X
9568 2009-03-05 15:42:23 -0800 Michael Smith <msmith@songbirdnest.com>
9570 * gst/playback/gstdecodebin2.c:
9571 decodebin2: don't stay connected to notify::caps after negotiation
9572 Disconnect the notify::caps signal in our callback (it'll be re-added
9573 if we're not, in fact, finished getting complete caps). Ensures that
9574 caps changes mid-stream (e.g. from an mp3 that changes from
9575 stereo->mono mid-file) don't cause us to try to add a new pad.
9577 2009-03-05 13:48:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9579 * gst-libs/gst/rtsp/gstrtsprange.c:
9580 rtsp: fix parsing of 'now-' ranges.
9583 2009-03-05 12:43:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9585 * tests/examples/dynamic/.gitignore:
9586 * tests/examples/dynamic/Makefile.am:
9587 * tests/examples/dynamic/sprinkle.c:
9588 * tests/examples/dynamic/sprinkle2.c:
9589 * tests/examples/dynamic/sprinkle3.c:
9590 examples: add some more sprinkle examples
9591 Add some more sprinle examples and add some more comments.
9594 2009-03-05 11:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9596 * docs/plugins/gst-plugins-base-plugins-sections.txt:
9597 docs: add appsrc symbols to standard section
9600 2009-03-05 12:27:16 +0200 Stefan Kost <ensonic@users.sf.net>
9602 * gst/adder/gstadder.c:
9603 adder: add variants for unsigned to fix warnings for unneeded check
9604 For unsigned int out+in can't be < 0.
9606 2009-03-05 10:58:12 +0200 Stefan Kost <ensonic@users.sf.net>
9608 * gst/subparse/gstsubparse.c:
9609 subparse: use the right variable in debug log, encoding is not yet initialized
9611 2009-03-05 10:51:25 +0200 Stefan Kost <ensonic@users.sf.net>
9613 * sys/v4l/v4l_calls.c:
9614 v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix
9616 2009-03-05 10:39:33 +0200 Stefan Kost <ensonic@users.sf.net>
9618 * gst/audioresample/gstaudioresample.c:
9619 audioresample: add missing break in event handling, remove dead code
9621 2009-03-04 16:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9623 * gst-libs/gst/rtsp/gstrtspconnection.c:
9624 rtsp: do some more cleanup in _close
9625 Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
9626 unconnected state as it was allocated.
9628 2009-03-04 16:11:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9630 * gst-libs/gst/rtsp/gstrtspconnection.c:
9631 * gst-libs/gst/rtsp/gstrtspconnection.h:
9632 rtsp: fix the memory management of the url
9633 Constify the url parameter in _create.
9634 Make a copy of the url stored in the connection.
9635 Free the url when the connection is freed.
9637 2009-03-04 12:21:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9639 * docs/libs/gst-plugins-base-libs-sections.txt:
9640 * gst-libs/gst/rtsp/gstrtspconnection.c:
9641 * gst-libs/gst/rtsp/gstrtspconnection.h:
9642 * win32/common/libgstrtsp.def:
9643 RTSP: Add support for server tunneling
9644 Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
9645 that a server can store and match the id against other tunnel requests.
9646 Fix the URI in the tunnel requests so that they contain the absolute uri and the
9647 query string if any instead of just the hostname.
9648 Transparently base64 decode the input stream when tunneling.
9649 Add method to set the connection ip address so that it can be included in the
9651 Add method to connect the two tunnel requests.
9652 Add two callbacks for the async mode to notify a tunnel start and tunnel
9654 Add method to reset the watch after the connection has been tunneled.
9655 Various little refactoring to make more stuff reusable.
9656 API: RTSP::gst_rtsp_connection_set_ip()
9657 API: RTSP::gst_rtsp_connection_get_tunnelid()
9658 API: RTSP::gst_rtsp_connection_do_tunnel()
9659 API: RTSP::gst_rtsp_watch_reset()
9661 2009-03-04 12:18:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9663 * gst-libs/gst/rtsp/gstrtspdefs.c:
9664 * gst-libs/gst/rtsp/gstrtspdefs.h:
9665 rtsp: add new defines for tunneling
9666 Add two more result codes for tunneling support.
9668 2009-03-04 12:12:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9670 * gst-libs/gst/rtsp/gstrtspmessage.h:
9671 rtsp: remove , from last enum member
9672 Remove , from last enum member to improve compatibility with other compilers.
9674 2009-02-28 15:23:20 -0800 LRN <lrn1986@gmail.com>
9676 * gst/subparse/gstsubparse.c:
9677 subparse: Convert regex code to GRegex code
9678 Fixes: #572993. Patch author prefers to use an alias, contact
9679 ds if you actually need a real name.
9680 Signed-off-by: David Schleef <ds@schleef.org>
9682 2009-03-02 16:13:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9684 * gst-libs/gst/rtsp/gstrtspconnection.c:
9685 rtsp: remove debugging g_message
9688 2009-03-02 16:03:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9690 * docs/libs/gst-plugins-base-libs-sections.txt:
9691 * gst-libs/gst/rtsp/gstrtspconnection.c:
9692 * gst-libs/gst/rtsp/gstrtspconnection.h:
9693 * win32/common/libgstrtsp.def:
9694 RTSP: add support for Quicktime tunneled RTSP
9695 Add support for tunneling RTSP over HTTP.
9696 Fix documentation some more.
9698 API: RTSP:gst_rtsp_connection_is_tunneled()
9699 API: RTSP:gst_rtsp_connection_set_tunneled()
9701 2009-03-02 15:48:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9703 * gst-libs/gst/rtsp/gstrtsptransport.h:
9704 * gst-libs/gst/rtsp/gstrtspurl.c:
9705 RTSP: parse rtsph uris as RTSP tunneled over HTTP
9706 Add transport define for RTSP tunneled over HTTP.
9707 Parse rtsph:// uris as tunneled HTTP over TCP.
9708 API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
9711 2009-03-02 12:48:18 +0100 Edward Hervey <bilboed@bilboed.com>
9713 * win32/common/libgstrtsp.def:
9714 win32: Add gst_rtsp_connection_get_url definition
9715 No, I'm not wim's buildslave, seriously.
9717 2009-03-02 10:58:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9719 * gst-libs/gst/rtsp/gstrtspconnection.c:
9720 * gst-libs/gst/rtsp/gstrtspconnection.h:
9721 rtsp: add _get_url method and separate sockets
9722 Add gst_rtsp_connection_get_url() method.
9723 Reserve space for 2 sockets, one for reading and one for writing. Use socket
9724 pointers to select the read and write sockets. This should allow us to implement
9725 tunneling over HTTP soon.
9726 API: RTSP::gst_rtsp_connection_get_url()
9728 2009-03-01 18:31:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9730 * gst-libs/gst/app/gstapp-marshal.list:
9731 app: force automatic rebuild of gstapp-marshal.[ch] after previous change
9732 The previous change to appsrc/appsink requires people to 'make clean'
9733 to get the marshallers rebuilt (causing a build failure otherwise).
9734 Change some lines in the .list file around to force a rebuild of
9735 these files automatically.
9737 2009-02-28 11:07:04 -0800 David Schleef <ds@schleef.org>
9740 Bump glib requirement to 2.14
9742 2009-02-28 19:37:53 +0100 LRN <lrn1986@gmail.com>
9744 * ext/gio/gstgiobasesink.c:
9745 gio: Use correct format modifier for size_t
9748 2009-02-28 19:35:33 +0100 LRN <lrn1986@gmail.com>
9750 * gst-libs/gst/rtsp/gstrtspconnection.c:
9751 rtspconnection: Use correct types for some functions on Win32
9754 2009-02-28 13:11:59 +0100 Edward Hervey <bilboed@bilboed.com>
9756 * gst-libs/gst/rtsp/gstrtspconnection.c:
9757 rtspconnection: Fix warning about using unitialized value.
9759 2009-02-28 12:41:28 +0100 Edward Hervey <bilboed@bilboed.com>
9761 * gst-libs/gst/riff/riff-ids.h:
9762 * gst-libs/gst/riff/riff-media.c:
9763 riff: Add more codec mappings.
9764 This comes mostly from a review of ffmpeg/libavformat/riff.c
9766 2009-02-27 11:14:25 +0200 Stefan Kost <ensonic@users.sf.net>
9768 * ext/alsa/gstalsa.c:
9769 alsa: release pcminfo after the strdup
9771 2009-02-26 17:38:47 +0200 Stefan Kost <ensonic@users.sf.net>
9773 * gst-libs/gst/rtsp/gstrtsprange.c:
9774 rtsprange: don't leak the range in case of parsing error.
9775 Free the gstRTSPTimeRange if we don't return it. Also simplify
9776 gst_rtsp_range_free() as it is valid to pass NULL to g_free().
9778 2009-02-26 16:47:39 +0200 Stefan Kost <ensonic@users.sf.net>
9780 * ext/alsa/gstalsa.c:
9781 alsa: cleanup name lookup.
9782 We can break, once we have a name to make sure, we won't read it ever twice.
9784 2009-02-26 16:09:03 +0200 Stefan Kost <ensonic@users.sf.net>
9786 * gst/subparse/gstsubparse.c:
9787 subparse: don't leak line, if flushing
9789 2009-02-26 16:03:39 +0200 Stefan Kost <ensonic@users.sf.net>
9791 * ext/gio/gstgiosink.c:
9792 giosink: reflow error handling to not leak uri
9794 2009-02-26 15:53:10 +0200 Stefan Kost <ensonic@users.sf.net>
9796 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
9797 * gst/ffmpegcolorspace/imgconvert.c:
9798 ffmpegcolorspace: remove unused code/variables
9800 2009-02-26 12:10:47 +0200 Stefan Kost <ensonic@users.sf.net>
9802 * sys/ximage/ximagesink.c:
9803 ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps
9805 2009-02-26 16:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9807 * docs/libs/gst-plugins-base-libs-sections.txt:
9808 * gst-libs/gst/app/gstappsink.c:
9809 * gst-libs/gst/app/gstappsrc.c:
9810 * gst-libs/gst/app/gstappsrc.h:
9811 * win32/common/libgstapp.def:
9812 app: add callbacks to appsrc, cleanups
9813 Add a uri handler to appsink.
9814 don't emit signals when we have installed callbacks on appsink.
9815 Add callbacks to appsrc to replace the signals.
9816 Add property to disable callbacks in appsrc, default to TRUE for backwards
9817 compatibility but disable when callbacks are installed.
9818 API: GstAppSrc::emit-signals
9819 API: GstAppSrc::gst_app_src_set_emit_signals()
9820 API: GstAppSrc::gst_app_src_get_emit_signals()
9821 API: GstAppSrc::gst_app_src_set_callbacks()
9823 2009-02-26 11:42:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9825 * docs/libs/gst-plugins-base-libs-sections.txt:
9826 * gst-libs/gst/app/gstappsink.h:
9827 * tests/check/elements/appsink.c:
9828 Appsink: add padding for callbacks + docs
9829 Add some padding to the callbacks structure just to be safe.
9830 Remove the now invisible marshaller methods from the docs.
9831 Fix a comment in the unit test.
9833 2009-02-26 09:52:59 +0100 Edward Hervey <bilboed@bilboed.com>
9835 * win32/common/libgstapp.def:
9836 win32: Add new libgstapp symbol
9838 2009-02-26 10:07:21 +0200 Stefan Kost <ensonic@users.sf.net>
9840 * docs/plugins/gst-plugins-base-plugins-sections.txt:
9841 docs: clean section.txt file.
9842 Add appsrc/sink symbols to private, as they are covered in the libs docs.
9844 2009-02-26 10:06:23 +0200 Stefan Kost <ensonic@users.sf.net>
9846 * gst/playback/gstplaybasebin.c:
9847 docs: fix random text after since: tag. Also fix class name to make the docs actual appear.
9849 2009-02-26 09:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
9851 * docs/plugins/gst-plugins-base-plugins.args:
9852 * docs/plugins/gst-plugins-base-plugins.hierarchy:
9853 * docs/plugins/gst-plugins-base-plugins.interfaces:
9854 * docs/plugins/gst-plugins-base-plugins.prerequisites:
9855 * docs/plugins/inspect/plugin-adder.xml:
9856 * docs/plugins/inspect/plugin-alsa.xml:
9857 * docs/plugins/inspect/plugin-app.xml:
9858 * docs/plugins/inspect/plugin-audioconvert.xml:
9859 * docs/plugins/inspect/plugin-audiorate.xml:
9860 * docs/plugins/inspect/plugin-audioresample.xml:
9861 * docs/plugins/inspect/plugin-audiotestsrc.xml:
9862 * docs/plugins/inspect/plugin-cdparanoia.xml:
9863 * docs/plugins/inspect/plugin-decodebin.xml:
9864 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
9865 * docs/plugins/inspect/plugin-gdp.xml:
9866 * docs/plugins/inspect/plugin-gio.xml:
9867 * docs/plugins/inspect/plugin-gnomevfs.xml:
9868 * docs/plugins/inspect/plugin-libvisual.xml:
9869 * docs/plugins/inspect/plugin-ogg.xml:
9870 * docs/plugins/inspect/plugin-pango.xml:
9871 * docs/plugins/inspect/plugin-playback.xml:
9872 * docs/plugins/inspect/plugin-queue2.xml:
9873 * docs/plugins/inspect/plugin-subparse.xml:
9874 * docs/plugins/inspect/plugin-tcp.xml:
9875 * docs/plugins/inspect/plugin-theora.xml:
9876 * docs/plugins/inspect/plugin-typefindfunctions.xml:
9877 * docs/plugins/inspect/plugin-uridecodebin.xml:
9878 * docs/plugins/inspect/plugin-video4linux.xml:
9879 * docs/plugins/inspect/plugin-videorate.xml:
9880 * docs/plugins/inspect/plugin-videoscale.xml:
9881 * docs/plugins/inspect/plugin-videotestsrc.xml:
9882 * docs/plugins/inspect/plugin-volume.xml:
9883 * docs/plugins/inspect/plugin-vorbis.xml:
9884 * docs/plugins/inspect/plugin-ximagesink.xml:
9885 * docs/plugins/inspect/plugin-xvimagesink.xml:
9886 * gst/playback/gstplaybin2.c:
9887 docs: playbin2 has no stream-info
9889 2009-02-26 09:53:03 +0200 Stefan Kost <ensonic@users.sf.net>
9891 * gst-libs/gst/video/video.h:
9892 docs: fix newly added interlace constants and plug holes in video format docs
9894 2009-02-26 09:35:43 +0200 Stefan Kost <ensonic@users.sf.net>
9896 * gst-libs/gst/app/gstappsink.c:
9897 * gst-libs/gst/app/gstappsrc.c:
9898 * gst-libs/gst/audio/gstaudiofilter.c:
9899 * gst-libs/gst/audio/gstringbuffer.c:
9900 * gst-libs/gst/rtp/gstrtcpbuffer.c:
9901 docs: don't put random stuff in tags.
9902 Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
9903 tag to append text again to the documentation body.
9905 2009-02-06 11:10:15 +0200 Stefan Kost <ensonic@users.sf.net>
9907 * sys/ximage/ximagesink.c:
9908 ximagsink: do not access uninitialized height variable.
9909 Exit like in xvimagesink, if we have partial caps.
9911 2009-02-25 20:26:05 -0800 David Schleef <ds@schleef.org>
9915 * win32/common/config.h.in:
9916 Change how win32/common/config.h is updated
9917 Generate win32/common/config.h-new directly from config.h.in,
9918 using shell variables in configure and some hard-coded information.
9919 Change top-level makefile so that 'make win32-update' copies the
9920 generated file to win32/common/config.h, which we keep in source
9921 control. It's kept in source control so that the git tree is
9923 This change is similar to the one recently applied to GStreamer,
9924 except that it adds a few -base specific defines.
9926 2009-02-25 19:40:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9928 * gst-libs/gst/app/Makefile.am:
9929 * gst-libs/gst/app/gstappsink.c:
9930 * gst-libs/gst/app/gstappsrc.c:
9931 * win32/common/libgstapp.def:
9932 app: add win32 .def file and only export functions we want exported
9933 Add a .def file for win32 builds (and make check-exports).
9934 Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes #573165).
9935 Make sure private marshaller functions aren't exported by prefixing them with __gst;
9936 also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
9937 a comment why we're not using glib-genmarshal for this one.
9939 2009-02-25 17:08:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9941 * tests/examples/dynamic/.gitignore:
9942 * tests/examples/dynamic/Makefile.am:
9943 * tests/examples/dynamic/sprinkle.c:
9944 sprinkle: Add another example app
9945 Add an example app that dynamically adds and removes audiotestsrc elements from
9948 2009-02-25 16:25:33 +0100 Peter Kjellerstedt <pkj@axis.com>
9950 * gst-libs/gst/rtsp/gstrtspconnection.c:
9953 2009-02-25 11:31:02 +0100 Peter Kjellerstedt <pkj@axis.com>
9955 * gst-libs/gst/rtsp/gstrtspconnection.c:
9956 * gst/tcp/gstmultifdsink.c:
9957 rtsp, multifdsink: Unify the use of union gst_sockaddr.
9959 2009-02-25 14:22:35 +0000 Jan Schmidt <thaytan@noraisin.net>
9963 build: Update shave init statement for changes in common. Bump common.
9965 2009-02-25 13:16:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9967 * sys/xvimage/xvimagesink.c:
9968 * sys/xvimage/xvimagesink.h:
9969 xvimageink: protect buffer_alloc from shutdown
9970 Use the pool_lock in the buffer_alloc function to detect shutdown. Avoids
9971 crashes when the sink is shutdown.
9973 2009-02-25 12:43:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9975 * gst/playback/gstplaybin2.c:
9976 playbin: use flushing pads instead of fakesink
9977 Use the flushing pads on playsink to terminate on shutdown instead of plugging
9978 fakesinks. this should be a little cheaper.
9980 2009-02-25 12:42:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
9982 * gst/playback/gstplaysink.c:
9983 * gst/playback/gstplaysink.h:
9984 playsink: Add FLUSHING pad type
9985 Make it possible to request a flushing pad from the playsink. We can eventually
9986 use these flushing pads to quickly terminate the dataflow when we are shutting
9989 2009-02-25 11:31:52 +0000 Jan Schmidt <thaytan@noraisin.net>
9992 Automatic update of common submodule
9993 From 9cf8c9b to a6ce5c6
9995 2009-02-25 09:52:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
9997 * gst-libs/gst/riff/riff-media.c:
9998 riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
10001 2009-02-25 12:07:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10003 * tests/icles/stress-playbin.c:
10004 stress-playbin: print the current uri
10005 Print the current uri so that we can more easily see what uri caused a crash or
10008 2009-02-25 11:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10010 * tests/icles/stress-playbin.c:
10011 Print the errors more clearly
10012 Print some more verbose messages when dealing with errors.
10014 2009-02-25 10:08:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10016 * gst/playback/gstplaybin2.c:
10017 Release the group lock when setting states
10018 Release the group lock while we perform the state changes on the uridecodebins
10019 because that might trigger callbacks that we need to handle with the group lock
10020 taken. Avoids a possible deadly embrace in some id3/flac files.
10023 2009-02-25 10:05:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10025 * gst/playback/gstdecodebin2.c:
10026 Combine finding and creating groups
10027 Combine the search for the current group and optionally creating one into one
10028 function so that we can avoid taking the lock multiple times.
10030 2009-02-25 08:22:00 +0100 Edward Hervey <bilboed@bilboed.com>
10032 * gst/playback/gstplaybin2.c:
10033 Playbin2: Don't leave unused parameters in debug statements.
10034 Fixes build on macosx
10036 2009-02-24 10:33:05 +0100 Edward Hervey <bilboed@bilboed.com>
10038 * gst-libs/gst/riff/riff-media.c:
10039 Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)
10041 2009-02-24 18:43:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10043 * gst/playback/gstplaybin2.c:
10044 Add some G_UNLIKELY because we can
10045 Add a G_UNLIKELY when checking the shutdown variable.
10047 2009-02-24 17:23:58 +0000 Garret D'Amore <garrett.damore@sun.com>
10049 * gst-libs/gst/interfaces/mixer.h:
10050 * gst-libs/gst/interfaces/mixertrack.h:
10051 mixer interface: Add flags to enhance mixer interfaces
10052 This patch adds a few flags to the mixer and mixerctrl interface to
10053 better support OSSv4 (and potentially other backends).
10054 Patch By: Garret D'Amore <garrett.damore@sun.com>
10055 Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>
10056 API: GST_MIXER_FLAG_HAS_WHITELIST, GST_MIXER_FLAG_GROUPING,
10057 API: GST_MIXER_TRACK_NO_RECORD, GST_MIXER_TRACK_NO_MUTE,
10058 API: GST_MIXER_TRACK_WHITELIST
10060 2009-02-24 17:03:08 +0000 Jan Schmidt <thaytan@noraisin.net>
10062 * gst/tcp/gstmultifdsink.c:
10063 multifdsink: Fix strict aliasing error using a union
10065 2009-02-24 16:49:40 +0000 Jan Schmidt <thaytan@noraisin.net>
10067 * gst-libs/gst/rtsp/gstrtspconnection.c:
10068 rtsp: Fix a strict aliasing warning
10069 Fix strict aliasing warnings from casting a sockaddr_storage and
10070 using it as a sockaddr_in6. Use a union instead.
10072 2009-02-24 16:08:49 +0000 Jan Schmidt <thaytan@noraisin.net>
10074 * docs/libs/.gitignore:
10075 * docs/libs/tmpl/.gitignore:
10076 * docs/plugins/.gitignore:
10077 * docs/plugins/tmpl/.gitignore:
10078 Remove .gitignore files from the docs tmpl dirs, that are killed by make clean.
10080 2009-02-24 14:36:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10082 * docs/plugins/Makefile.am:
10083 * ext/vorbis/Makefile.am:
10084 * ext/vorbis/gstvorbisdec.h:
10085 * ext/vorbis/gstvorbisenc.h:
10086 * ext/vorbis/gstvorbisparse.h:
10087 * ext/vorbis/gstvorbistag.h:
10088 * ext/vorbis/vorbis.c:
10089 * ext/vorbis/vorbisdec.c:
10090 * ext/vorbis/vorbisdec.h:
10091 * ext/vorbis/vorbisenc.c:
10092 * ext/vorbis/vorbisenc.h:
10093 * ext/vorbis/vorbisparse.c:
10094 * ext/vorbis/vorbisparse.h:
10095 * ext/vorbis/vorbistag.c:
10096 * ext/vorbis/vorbistag.h:
10097 vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
10099 2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10101 * gst/ffmpegcolorspace/avcodec.h:
10102 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
10103 * gst/ffmpegcolorspace/imgconvert.c:
10104 ffmpegcolorspace: Add conversion from/to YVYU colorspace
10107 2009-02-24 13:42:01 +0100 Jonas Danielsson <jonas.danielsson@axis.com>
10109 * gst/ffmpegcolorspace/imgconvert.c:
10110 ffmpegcolorspace: Add direct UYVY->GRAY8 conversion
10111 The conversion from UYVY to RGB24 and then to GRAY8
10112 is quite slow. Fixes bug #569655.
10114 2009-02-19 17:16:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
10116 * gst/playback/gstplaybin2.c:
10117 playbin2: fix deadlock when shutting down. Fixes #572577.
10119 2009-02-19 17:15:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
10121 * tests/icles/stress-playbin.c:
10122 stress-playbin: make more flexible, e.g. also useful for playbin2
10124 2009-02-24 12:11:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10126 * gst-libs/gst/rtsp/gstrtspconnection.c:
10127 Match WSAStartup and WSACleanup correctly
10128 Don't randomly call WSAStartup and WSACleanup but instead call the startup when
10129 we create a connection and cleanup when we free it again. Because the internal
10130 datastructure is refcounted, this should not cause any refcounting leaks when
10131 the connection is managed correctly.
10134 2009-02-18 11:59:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
10136 * gst/playback/gstplaysink.c:
10137 playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105.
10139 2009-02-23 10:57:42 -0800 David Flynn <davidf@rd.bbc.co.uk>
10141 * pkgconfig/gstreamer-app-uninstalled.pc.in:
10142 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
10143 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
10144 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
10145 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
10146 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
10147 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
10148 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
10149 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
10150 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
10151 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
10152 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
10153 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
10154 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
10155 * pkgconfig/gstreamer-video-uninstalled.pc.in:
10156 Add srcdir to includes for out-of-source builds
10157 When you use gstreamer uninstalled and build outside
10158 the source tree, the includes need to be specified for
10159 both the source tree and the build tree.
10160 Signed-off-by: David Schleef <ds@schleef.org>
10162 2009-02-22 17:23:52 +0000 Jan Schmidt <thaytan@noraisin.net>
10165 * docs/libs/Makefile.am:
10166 * docs/plugins/Makefile.am:
10167 Use shave for the build output
10169 2009-02-23 12:17:07 +0100 Edward Hervey <bilboed@bilboed.com>
10171 * win32/common/libgstrtsp.def:
10172 win32: Add new symbol to libgstrtsp.def
10174 2009-02-23 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10176 * gst-libs/gst/rtsp/gstrtspextension.c:
10177 * gst-libs/gst/rtsp/gstrtspextension.h:
10178 Add method for handling server requests
10179 Add a receive_request so that extensions can react to server requests.
10181 2009-02-22 19:20:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10183 * tests/check/libs/netbuffer.c:
10184 Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)
10186 2009-02-22 19:19:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10188 * ext/theora/theoraparse.c:
10189 theoraparse: Use the correct unref functions
10191 2009-02-22 19:18:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10193 * sys/ximage/ximagesink.c:
10194 * sys/xvimage/xvimagesink.c:
10195 x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()
10197 2009-02-22 19:12:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10199 * gst-libs/gst/tag/gsttagdemux.c:
10200 tagdemux: Unref the actual buffer instead of the memory address of the buffer
10202 2009-02-22 15:47:53 +0000 Jan Schmidt <thaytan@noraisin.net>
10205 Automatic update of common submodule
10206 From 5d7c9cc to 9cf8c9b
10208 2009-02-22 14:49:29 +0100 Edward Hervey <bilboed@bilboed.com>
10210 * win32/common/libgstrtsp.def:
10211 * win32/common/libgstvideo.def:
10212 win32/common: Update .def files for recent API addition
10214 2009-02-22 13:43:35 +0100 Edward Hervey <bilboed@bilboed.com>
10216 * tests/check/libs/rtp.c:
10217 tests: Fix indentation
10219 2009-02-22 13:42:33 +0100 Edward Hervey <bilboed@bilboed.com>
10221 * gst-libs/gst/video/video.c:
10222 libs/video: Fix gst_video_format_new_caps* functions.
10223 Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
10224 don't add anything.
10226 2009-02-21 11:13:36 -0800 David Schleef <ds@schleef.org>
10229 Automatic update of common submodule
10230 From 80c627d to 5d7c9cc
10232 2009-02-20 17:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10234 * gst-libs/gst/rtsp/gstrtspmessage.c:
10235 Improve key/value parsing
10236 Improve header field parsing by keeping a ref to the key/value instead of
10237 copying it into a local variable.
10239 2009-02-20 12:35:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10241 * gst-libs/gst/rtsp/gstrtspconnection.c:
10242 Add trailing \0 to message length
10243 We always put a trailing 0 at the end of the message body. Reflect this fact in
10244 the length of the message.
10246 2009-02-20 09:50:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10248 * gst-libs/gst/rtsp/gstrtspconnection.c:
10249 Don't parse headers for data messages
10250 Don't try to parse the headers on a data message because they don't have
10253 2009-02-19 12:18:29 -0800 Benjamin M. Schwartz <bens@alum.mit.edu>
10255 * ext/theora/gsttheoraenc.h:
10256 * ext/theora/theoraenc.c:
10257 theoraenc: Add property for speed level control
10258 Add property "speed-level" to control the amount of motion searching
10259 the encoder does. This is only available in libtheora >= 1.0 and
10260 will silently fail with earlier libraries. Fixes: #572275.
10261 Signed-off-by: David Schleef <ds@schleef.org>
10263 2009-02-19 17:40:45 +0100 Edward Hervey <bilboed@bilboed.com>
10265 * gst-libs/gst/video/video.c:
10266 * gst-libs/gst/video/video.h:
10267 video: Fix 'Since' tags
10269 2009-01-26 10:30:53 +0100 Edward Hervey <bilboed@bilboed.com>
10271 * docs/libs/gst-plugins-base-libs-sections.txt:
10272 * gst-libs/gst/video/video.c:
10273 * gst-libs/gst/video/video.h:
10274 video: Add flags for interlaced video along with convenience methods for interlaced caps.
10275 These three flags allow all know combinations of interlaced formats. They should
10276 only be used when the caps contain 'interlaced=True'.
10277 Fixes #163577 (yes, it's a 4 year old bug).
10279 2009-02-19 15:51:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10281 * docs/libs/gst-plugins-base-libs-sections.txt:
10282 * gst-libs/gst/rtsp/gstrtspconnection.c:
10283 * gst-libs/gst/rtsp/gstrtspconnection.h:
10284 Make RTSPConnection opaque and rename RTSPChannel
10285 Make the RTSPConnection object opaque so that we can extend it in the future.
10286 Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
10288 2009-01-26 10:31:14 +0100 Edward Hervey <bilboed@bilboed.com>
10290 * gst-libs/gst/riff/riff-media.c:
10291 Add some more mappings for h264 in riff
10293 2009-02-19 10:49:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10295 * win32/common/libgstrtsp.def:
10296 Add new RTSP symbols to def files
10297 Add the new RTSP symbols to the windows def file.
10299 2009-02-19 10:44:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10301 * docs/libs/gst-plugins-base-libs-sections.txt:
10302 * gst-libs/gst/app/gstappsink.c:
10303 * gst-libs/gst/app/gstappsink.h:
10304 * tests/check/Makefile.am:
10305 * tests/check/elements/.gitignore:
10306 * tests/check/elements/appsink.c:
10307 Add method to install callbacks on appsink
10308 Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
10310 Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
10311 performant alternative to connecting to the signals.
10312 Add a unit test for appsink.
10313 Clean up some of the appsink docs.
10314 API: GstAppSink::gst_app_sink_set_callbacks()
10316 2009-02-18 18:46:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10318 * docs/libs/gst-plugins-base-libs-sections.txt:
10319 * gst-libs/gst/rtsp/gstrtspconnection.c:
10320 * gst-libs/gst/rtsp/gstrtspconnection.h:
10321 Add RTSP accept method
10322 Add a method to accept a connection on a socket and create a GstRTSPConnection
10324 API: gst_rtsp_connection_accept()
10326 2009-02-18 17:42:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10328 * docs/libs/gst-plugins-base-libs-sections.txt:
10329 * gst-libs/gst/rtsp/gstrtspconnection.c:
10330 * gst-libs/gst/rtsp/gstrtspconnection.h:
10331 Add RTSP channel object for async io
10332 Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
10333 that the connection can be monitored from a maincontext. This allows us to
10334 operate in ASYNC mode, which is handy when building a server.
10335 Rework the old code to use the async code under the hood.
10336 API: gst_rtsp_channel_new()
10337 API: gst_rtsp_channel_unref()
10338 API: gst_rtsp_channel_attach()
10339 API: gst_rtsp_channel_queue_message()
10341 2009-02-15 07:30:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10343 * gst/audioresample/gstaudioresample.c:
10344 audioresample: Add locking to protect the resampling context
10345 When setting the quality/filter-length while PLAYING the
10346 resampling context will be destroyed and created again in
10347 some cases, which will cause crashes in the transform function
10348 if it's called at that time.
10350 2009-02-13 10:10:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10352 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
10353 * gst/videotestsrc/videotestsrc.c:
10354 ffmpegcolorspace/videotestsrc: Use v308 instead of V308
10356 2009-02-12 19:02:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10358 * gst/ffmpegcolorspace/avcodec.h:
10359 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
10360 * gst/ffmpegcolorspace/imgconvert.c:
10361 * gst/ffmpegcolorspace/imgconvert_template.h:
10362 ffmpegcolorspace: Add support for packed 4:4:4 YUV (format=V308)
10363 Only conversions from/to are implemented, which
10364 gives (indirect) support for all possible conversions.
10365 Partially fixes bug #571147.
10367 2009-02-12 18:17:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10369 * gst/videotestsrc/videotestsrc.c:
10370 videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
10371 Partially fixes bug #571147.
10373 2009-02-12 09:18:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10375 * gst-libs/gst/tag/gsttagdemux.c:
10376 tagdemux: don't abort when downstream pulls a buffer of size 0
10377 Pulling a 0-sized buffer is allowed, and we should handle this correctly instead of
10378 aborting. Fixes #571009 (wma file with ID3v2 tag).
10380 2009-02-11 16:39:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10382 * gst-libs/gst/riff/riff-read.c:
10383 riff: error out on nonsensical chunk sizes instead of aborting
10384 When encountering a nonsensical chunk size such as (guint)-1, error out cleanly instead of
10385 continuing and trying to g_memdup() 4GB of data that doesn't exist, which will either abort
10386 in g_malloc() or crash.
10387 Fixes #553295, crash with fuzzed AVI file.
10389 2009-02-11 16:39:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10392 Make git ignore backup files.
10394 2009-02-10 20:38:58 -0800 Michael Smith <msmith@syncword.(none)>
10396 * gst/playback/gstplaybin2.c:
10397 Revert "Remove pad-removed handlers after setting the decodebins to NULL."
10398 This reverts commit b36d8f3e119f9edc5993c08025614ee32642972e.
10399 This brought back some deadlocks. A small leak is better, for now. Need to
10400 figure out a way to fix the leak properly.
10402 2009-02-10 17:16:07 -0800 Michael Smith <msmith@songbirdnest.com>
10404 * gst/playback/gstplaybin2.c:
10405 playbin2: Fix segfault on notify after group change.
10406 If our group has been switched, then we get a selector active-pad
10407 notification, we don't need to notify.
10409 2009-02-10 17:10:33 -0800 Michael Smith <msmith@songbirdnest.com>
10411 * gst/playback/gstplaysink.c:
10412 playbin2: Look for volume/mute properties recursively in audio element.
10413 Rather than only checking for volume property on the audio sink
10414 directly, recursively look for it on sinks within it (if it's a bin).
10415 Allows use of sink-as-volume-control where the application has supplied
10416 an audio-sink bin that includes a real audio sink internally.
10418 2009-02-10 18:29:22 +0000 Christian Schaller <cschalle@crazyhorse.localdomain>
10420 * gst-plugins-base.spec.in:
10421 Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base
10423 2009-02-10 17:39:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10425 * gst/videotestsrc/videotestsrc.c:
10426 videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
10427 Partially fixes bug #571147.
10429 2009-02-10 17:37:06 +0100 Peter Kjellerstedt <pkj@axis.com>
10431 * gst-libs/gst/rtsp/gstrtspmessage.c:
10432 gstrtspmessage: Minor documentation correction.
10433 Corrected documentation about what needs to be freed after calling
10434 gst_rtsp_message_new(), gst_rtsp_message_new_request(),
10435 gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
10437 2009-02-10 11:00:12 +0100 Antoine Tremblay <hexa00@gmail.com>
10439 * ext/alsa/gstalsamixer.c:
10440 alsamixer: Fix race condition that made alsamixer not working properly
10441 This is due to race conditions between functions that
10442 modified the mixer like set_volume and
10443 snd_mixer_handle_events since the handle_events
10444 can now be called at any time.
10445 Fixed by adding locking around any snd_mixer call
10446 since even read functions can modify the mixer stucture, since
10447 alsa likes to clear it's values before reading new ones.
10448 The favorite race condition seemed to be that set_volume
10449 called read_elem (in alsalib) that reset the volumes to
10450 0 and then read them with read_x_volume. This read looped
10451 on each channel and as the race condition occured the
10452 channels value could be anything , most of the time
10453 it was 0. Thus no value was read or only the value of
10454 one channel was and the volume was reset to 0.
10457 2009-02-09 12:02:21 +0100 Edward Hervey <bilboed@bilboed.com>
10460 Bump revision to use for common submodule.
10462 2009-02-05 15:47:00 +0200 Stefan Kost <ensonic@users.sf.net>
10464 * sys/xvimage/xvimagesink.c:
10465 xvimagesink: do not call _xwindow_clear on ready->paused.
10466 Calling clear at that transition does things like stopping xvideo (which is not
10467 running at that time) and also clearing anything what the application might have drawn.
10468 This breaks handle-expose and autopaint-colorkey features.
10470 2009-02-04 17:03:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10472 * docs/libs/gst-plugins-base-libs-sections.txt:
10473 * gst-libs/gst/rtsp/gstrtsprange.c:
10474 * gst-libs/gst/rtsp/gstrtsprange.h:
10475 RTSPRange: Add method to serialize ranges
10476 Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
10477 be used by a server.
10478 API: GstRTSPRange::gst_rtsp_range_to_string()
10480 2009-02-04 13:16:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10482 * gst-libs/gst/rtsp/gstrtspurl.c:
10483 * gst-libs/gst/rtsp/gstrtspurl.h:
10484 GstRTSPUrl: Add some const to methods
10485 Add const to the methods that do not modify the object.
10487 2009-02-04 13:53:30 +0200 Stefan Kost <ensonic@users.sf.net>
10489 * gst/playback/gstplaysink.c:
10490 playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
10491 The flags where present but actually not been taken into account.
10493 2009-02-04 12:06:38 +0200 Stefan Kost <ensonic@users.sf.net>
10495 * gst/audioresample/gstaudioresample.c:
10496 audioresample: Add a proper deprecation comment and also drop G_PARAM_CONSTRUCT.
10497 The comment will ensure that is is marked properly in the docs and the
10498 GParamSpecflag was causing a duplicated initialisation of the same value.
10500 2009-02-04 11:18:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10502 * gst-libs/gst/rtsp/gstrtspconnection.c:
10503 Add more g_return_if_fail() calls
10504 Check that we have a valid file descriptor before entering certain functions in
10505 order to avoid undesirable situations.
10506 Add some more debugging in the connect method.
10508 2009-02-04 10:31:21 +0200 Stefan Kost <ensonic@users.sf.net>
10511 * gst/audioresample/Makefile.am:
10512 * gst/audioresample/gstaudioresample.c:
10513 audioresample: Only pull in liboil if its actualy used.
10514 Liboil still has quite significant startup overhead especialy on embedded
10515 platforms. In audioresample it was only used for the profiling timer.
10517 2009-02-03 15:26:08 +0200 Stefan Kost <ensonic@users.sf.net>
10519 * gst/typefind/gsttypefindfunctions.c:
10520 typefind: Make the flac check more tight to not mistace some aac files for flac. Fixes #570356.
10521 Add comments about the flac format. Tighten the check to not allow values that
10524 2009-02-03 10:52:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10526 * win32/common/libgstrtsp.def:
10528 Add new methods to the windows def file.
10530 2009-02-02 17:25:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10532 * gst-libs/gst/pbutils/install-plugins.c:
10533 * tests/check/libs/pbutils.c:
10534 pbutils: remove duplicate detail strings when calling the external codec installer
10535 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.
10537 2009-02-02 18:05:42 +0200 Stefan Kost <ensonic@users.sf.net>
10539 * gst-libs/gst/audio/gstaudiosink.c:
10540 * gst-libs/gst/audio/gstaudiosink.h:
10541 Add a FIXME 0.11. Make the log message a bit more detailed and add comments.
10543 2009-02-02 15:43:03 +0200 Stefan Kost <ensonic@users.sf.net>
10546 * gst/audioresample/gstaudioresample.c:
10547 Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark.
10549 2009-02-02 13:30:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10551 * sys/ximage/ximagesink.c:
10552 Fix buffer_alloc in ximagesink
10553 Remove some useless debug info that reported wrong image sizes.
10554 When upstream does not accept out suggested size, fall back to allocating an
10555 image of the requested width/height instead of the currently configured size.
10556 The problem is that an image is reused from the pool because the width/height
10557 match but the caps on the new buffer are the requested caps with possibly
10558 different height/width resulting in errors.
10560 2009-02-02 12:54:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10562 * gst/playback/gstdecodebin2.c:
10563 * gst/playback/gsturidecodebin.c:
10564 Fix documentation for autoplug-select
10565 fix the documentation strings for the autoplug-select signal.
10568 2009-02-02 10:09:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10570 * gst-libs/gst/rtsp/gstrtspmessage.c:
10571 Fix string leak in rtspmessage
10572 when we remove a header field from a message we must free the value associated
10573 with the key to avoid a memory leak.
10575 2009-01-31 18:45:47 +0200 Stefan Kost <ensonic@users.sf.net>
10577 * docs/libs/gst-plugins-base-libs-docs.sgml:
10578 Its "Base Library" and not just "Library".
10580 2009-01-31 18:44:32 +0200 Stefan Kost <ensonic@users.sf.net>
10582 * gst-libs/gst/audio/gstaudiofilter.c:
10583 Link to the class, as we can't link to the members yet.
10585 2009-01-30 17:48:23 -0800 Michael Smith <msmith@songbirdnest.com>
10587 * gst/playback/gstplaybin2.c:
10588 Remove pad-removed handlers after setting the decodebins to NULL.
10589 They do needed cleanup; without this we leak selector requestpads.
10591 2009-01-30 17:47:07 -0800 Michael Smith <msmith@songbirdnest.com>
10593 * gst/playback/gstplaybin2.c:
10594 Unref selector request pad even if we no longer have a selector.
10595 During destruction, we won't have a selector any more, but we still need
10596 to unref the pad to avoid leaking it.
10598 2009-01-30 15:23:23 -0800 Michael Smith <msmith@songbirdnest.com>
10600 * gst/playback/gstplaybin2.c:
10601 Unref source in playbin2's finalize method
10603 2009-01-30 12:04:01 -0800 Michael Smith <msmith@songbirdnest.com>
10605 * gst/playback/gstplaysink.c:
10606 Fix more leaks of pads and elements in gstplaysink.
10607 Don't keep extra references to volume and mute elements; we don't need
10609 Ensure we unref pads that we have references to, and release request
10612 2009-01-30 11:04:37 -0800 Michael Smith <msmith@songbirdnest.com>
10614 * gst/playback/gstplaysink.c:
10615 Avoid leaking all playsinks. Fix some internal leaks.
10616 Playsink was holding references to itself. Don't do that, it's not cool.
10617 Also, free all chains in dispose.
10619 2009-01-30 10:54:12 -0800 Michael Smith <msmith@songbirdnest.com>
10621 * gst/playback/gstplaybin2.c:
10622 Unref peer request pad after releasing it, since we hold a reference.
10624 2009-01-30 10:52:52 -0800 Michael Smith <msmith@songbirdnest.com>
10626 * gst/playback/gstplaybin2.c:
10627 Fix caps leak in playbin2.
10629 2009-01-30 10:51:11 -0800 Michael Smith <msmith@songbirdnest.com>
10631 * gst/playback/gstplaybin2.c:
10632 Unref active pad from selector when finding active stream.
10634 2009-01-30 10:49:55 -0800 Michael Smith <msmith@songbirdnest.com>
10636 * gst/playback/gstplaybin2.c:
10637 Free uris when finalizing playbin2 instance.
10639 2009-01-30 10:38:17 -0800 Michael Smith <msmith@songbirdnest.com>
10641 * gst/playback/gsturidecodebin.c:
10642 Unref pads when iterating over them in analyse_source.
10643 Fixes leak of source's srcpad when using uridecodebin.
10645 2009-01-30 22:22:07 +0200 Stefan Kost <ensonic@users.sf.net>
10647 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
10648 Add releaseinfo with online url.
10650 2009-01-30 17:58:15 +0000 Jan Schmidt <jan.schmidt@sun.com>
10652 * gst/playback/gstplaybasebin.c:
10653 Fix compilation warning on Forte
10655 2009-01-30 17:16:39 +0000 Jan Schmidt <jan.schmidt@sun.com>
10657 * gst/adder/gstadder.c:
10658 Don't do void pointer arithmetic.
10660 2009-01-30 17:25:51 +0000 Jan Schmidt <thaytan@noraisin.net>
10665 2009-01-30 08:50:53 +0100 Edward Hervey <bilboed@bilboed.com>
10669 Use a symbolic link for the pre-commit client-side hook
10671 2009-01-30 08:12:42 +0100 Edward Hervey <bilboed@bilboed.com>
10674 Add more files/directories to ignore
10676 2009-01-29 14:00:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10678 * gst-libs/gst/rtsp/gstrtspdefs.c:
10680 Fix some typos in the doc string of the new
10681 gst_rtsp_options_as_string() method.
10683 2009-01-29 11:55:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10685 * docs/libs/gst-plugins-base-libs-sections.txt:
10686 * gst-libs/gst/rtsp/gstrtspconnection.c:
10687 * gst-libs/gst/rtsp/gstrtspmessage.c:
10688 * gst-libs/gst/rtsp/gstrtspmessage.h:
10689 Add new RTSP message method to set header
10690 Add gst_rtsp_message_take_header() that takes ownership of the passed header
10691 value. This allows us to avoid an allocations and memory copy in some
10693 API: GstRTSPMessage::gst_rtsp_message_take_header()
10695 2009-01-29 11:51:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10697 * docs/libs/gst-plugins-base-libs-sections.txt:
10698 Add new method to docs
10699 Add the new gst_rtsp_options_as_text() method to the docs.
10701 2009-01-28 11:48:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10703 * gst-libs/gst/rtsp/gstrtspdefs.c:
10704 * gst-libs/gst/rtsp/gstrtspdefs.h:
10705 Add method to serialize RTSP options
10706 Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
10708 API: GstRTSP::gst_rtsp_options_as_text()
10710 2009-01-26 17:59:37 -0800 Michael Smith <msmith@songbirdnest.com>
10712 * gst/typefind/gsttypefindfunctions.c:
10713 Ensure we have sufficient data when using data scan contexts.
10714 Fixes crashes typefinding things that look like they might contain AAC
10715 data (but probably aren't actually AAC).
10717 2009-01-26 23:32:09 +0000 Jan Schmidt <thaytan@noraisin.net>
10719 * ext/gio/Makefile.am:
10720 Fix include order for gio plugin
10722 2009-01-23 23:59:48 +0000 Jan Schmidt <thaytan@noraisin.net>
10724 * win32/common/config.h:
10725 Update win32 config.h for 0.10.22.1 dev cycle
10727 2009-01-23 23:16:11 +0000 Jan Schmidt <thaytan@noraisin.net>
10730 * docs/libs/.gitignore:
10731 * gst-libs/gst/audio/.gitignore:
10732 * gst-libs/gst/video/.gitignore:
10734 * tests/examples/dynamic/.gitignore:
10735 Extend and clean up git ignores
10737 2009-01-23 12:31:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
10740 * docs/plugins/Makefile.am:
10741 * docs/plugins/gst-plugins-base-plugins-sections.txt:
10742 * docs/plugins/gst-plugins-base-plugins.args:
10743 * docs/plugins/gst-plugins-base-plugins.hierarchy:
10744 * docs/plugins/gst-plugins-base-plugins.interfaces:
10745 * docs/plugins/gst-plugins-base-plugins.prerequisites:
10746 * docs/plugins/inspect/plugin-adder.xml:
10747 * docs/plugins/inspect/plugin-alsa.xml:
10748 * docs/plugins/inspect/plugin-app.xml:
10749 * docs/plugins/inspect/plugin-audioconvert.xml:
10750 * docs/plugins/inspect/plugin-audiorate.xml:
10751 * docs/plugins/inspect/plugin-audioresample.xml:
10752 * docs/plugins/inspect/plugin-audiotestsrc.xml:
10753 * docs/plugins/inspect/plugin-cdparanoia.xml:
10754 * docs/plugins/inspect/plugin-decodebin.xml:
10755 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
10756 * docs/plugins/inspect/plugin-gdp.xml:
10757 * docs/plugins/inspect/plugin-gio.xml:
10758 * docs/plugins/inspect/plugin-gnomevfs.xml:
10759 * docs/plugins/inspect/plugin-libvisual.xml:
10760 * docs/plugins/inspect/plugin-ogg.xml:
10761 * docs/plugins/inspect/plugin-pango.xml:
10762 * docs/plugins/inspect/plugin-playback.xml:
10763 * docs/plugins/inspect/plugin-queue2.xml:
10764 * docs/plugins/inspect/plugin-subparse.xml:
10765 * docs/plugins/inspect/plugin-tcp.xml:
10766 * docs/plugins/inspect/plugin-theora.xml:
10767 * docs/plugins/inspect/plugin-typefindfunctions.xml:
10768 * docs/plugins/inspect/plugin-uridecodebin.xml:
10769 * docs/plugins/inspect/plugin-video4linux.xml:
10770 * docs/plugins/inspect/plugin-videorate.xml:
10771 * docs/plugins/inspect/plugin-videoscale.xml:
10772 * docs/plugins/inspect/plugin-videotestsrc.xml:
10773 * docs/plugins/inspect/plugin-volume.xml:
10774 * docs/plugins/inspect/plugin-vorbis.xml:
10775 * docs/plugins/inspect/plugin-ximagesink.xml:
10776 * docs/plugins/inspect/plugin-xvimagesink.xml:
10777 * gst/audioresample/Makefile.am:
10778 * gst/audioresample/README:
10779 * gst/audioresample/arch.h:
10780 * gst/audioresample/buffer.c:
10781 * gst/audioresample/buffer.h:
10782 * gst/audioresample/debug.c:
10783 * gst/audioresample/debug.h:
10784 * gst/audioresample/fixed_arm4.h:
10785 * gst/audioresample/fixed_arm5e.h:
10786 * gst/audioresample/fixed_bfin.h:
10787 * gst/audioresample/fixed_debug.h:
10788 * gst/audioresample/fixed_generic.h:
10789 * gst/audioresample/functable.c:
10790 * gst/audioresample/functable.h:
10791 * gst/audioresample/gstaudioresample.c:
10792 * gst/audioresample/gstaudioresample.h:
10793 * gst/audioresample/resample.c:
10794 * gst/audioresample/resample.h:
10795 * gst/audioresample/resample_chunk.c:
10796 * gst/audioresample/resample_functable.c:
10797 * gst/audioresample/resample_ref.c:
10798 * gst/audioresample/resample_sse.h:
10799 * gst/audioresample/speex_resampler.h:
10800 * gst/audioresample/speex_resampler_double.c:
10801 * gst/audioresample/speex_resampler_float.c:
10802 * gst/audioresample/speex_resampler_int.c:
10803 * gst/audioresample/speex_resampler_wrapper.h:
10804 * gst/speexresample/Makefile.am:
10805 * gst/speexresample/README:
10806 * gst/speexresample/arch.h:
10807 * gst/speexresample/fixed_arm4.h:
10808 * gst/speexresample/fixed_arm5e.h:
10809 * gst/speexresample/fixed_bfin.h:
10810 * gst/speexresample/fixed_debug.h:
10811 * gst/speexresample/fixed_generic.h:
10812 * gst/speexresample/gstspeexresample.c:
10813 * gst/speexresample/gstspeexresample.h:
10814 * gst/speexresample/resample.c:
10815 * gst/speexresample/resample_sse.h:
10816 * gst/speexresample/speex_resampler.h:
10817 * gst/speexresample/speex_resampler_double.c:
10818 * gst/speexresample/speex_resampler_float.c:
10819 * gst/speexresample/speex_resampler_int.c:
10820 * gst/speexresample/speex_resampler_wrapper.h:
10821 * gst/typefind/gsttypefindfunctions.c:
10822 * tests/check/Makefile.am:
10823 * tests/check/elements/audioresample.c:
10824 * tests/check/elements/speexresample.c:
10825 Rename files and types from speexresample to audioresample
10826 Rename files and types from speexresample to audioresample
10827 to finish the move and to prevent any confusion.
10829 2009-01-23 11:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10831 * sys/xvimage/xvimagesink.c:
10832 Add some more debugging to the Xv strides
10833 Add some more debugging to the strides as they are received from the server and
10834 the expected strides.
10836 2009-01-23 11:40:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10838 * gst/typefind/gsttypefindfunctions.c:
10839 Add typefind function for gsm
10840 Because core now supports typefindfactories without a typefind function we can
10841 register a factory fo GSM that will --if all else fails-- assume the file is a
10842 GSM file based on the registered extension.
10845 2009-01-23 11:37:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10847 * gst/playback/gsturidecodebin.c:
10848 Use more performant link function
10849 We can use gst_element_link_pads() instead of the more generic
10850 gst_element_link() function because we know the pads. This saves some cycles
10851 because the more generic function needs to search for possible compatible caps
10854 2009-01-23 11:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10856 * gst-libs/gst/riff/riff-ids.h:
10857 * gst-libs/gst/riff/riff-media.c:
10858 Add more codec ids for RIFF formats
10859 Handle codec ID for various other AAC formats.
10860 Sync the list of possible codec ids with that of ffmpeg.
10863 2009-01-23 11:27:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10865 * ext/theora/theoradec.c:
10866 Use rounded values for image strides and sizes
10867 Round up the height before calculating the expected size and
10868 strides of the output image.
10870 2009-01-23 11:23:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10872 * ext/alsa/gstalsasink.c:
10873 Improve debug message
10874 Improve the debug message when alsa returns an error.
10876 2009-01-23 11:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10878 * gst-libs/gst/app/gstappsrc.c:
10879 Reset queued_bytes counter when flushing
10880 Set the amount of queued bytes in the internal queue back to 0 when we clear the
10884 2009-01-23 10:19:27 +0100 Benjamin Gaignard <benjamin@gaignard.net>
10886 * gst/typefind/gsttypefindfunctions.c:
10887 Add typefinder for Mobile XMF. Fixes bug #568707.
10889 2009-01-23 10:00:11 +0100 Brian Cameron <brian.cameron@sun.com>
10892 Fix linking on Solaris. Fixes bug #568482.
10893 Check for nsl and socket libraries and add them to
10894 LIBS if they're found. They're needed for socket()
10895 and gethostbyname() on Solaris.
10897 2009-01-22 22:09:47 +0000 Jan Schmidt <thaytan@noraisin.net>
10899 * gst/playback/gstplaybasebin.c:
10900 Fix use-after-unref problem noticed by Josep Torra Valles, and run gst-indent
10902 2009-01-22 17:46:59 +0200 Stefan Kost <ensonic@users.sf.net>
10905 Update common snapshot.
10907 2009-01-22 13:47:24 +0100 Sebastian Dröge <slomo@circular-chaos.org>
10910 Fix pre-commit hook
10912 2009-01-22 13:12:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10914 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
10916 2009-01-22 10:14:28 +0100 Sebastian Dröge <slomo@circular-chaos.org>
10918 * gst-libs/gst/fft/gstfftf32.c:
10919 * gst-libs/gst/fft/gstfftf64.c:
10920 * gst-libs/gst/fft/gstffts16.c:
10921 * gst-libs/gst/fft/gstffts32.c:
10922 Reduce the number of allocations for creating FFT contexts
10923 Reduce the number of allocations from 2 to 1 for every FFT
10924 context by allocating enough memory for the FFT context
10925 and passing parts of it to the kissfft allocation functions.
10927 2009-01-22 11:32:56 +0000 Jan Schmidt <thaytan@noraisin.net>
10930 Back to devel -> 0.10.22.1
10932 2009-01-22 05:57:53 +0100 Edward Hervey <bilboed@bilboed.com>
10936 Install and use pre-commit indentation hook from common
10938 2009-01-21 13:09:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
10940 * gst-libs/gst/rtp/gstrtpbuffer.c:
10941 * tests/check/libs/rtp.c:
10942 Avoid overflows in the padding checks by doing the check slightly differently. Add a unit test to check for correct behaviour.
10944 2009-01-21 04:31:32 +0100 Edward Hervey <bilboed@bilboed.com>
10947 autogen.sh : Use git submodule
10949 === release 0.10.22 ===
10951 2009-01-19 23:10:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10957 * docs/plugins/gst-plugins-base-plugins.hierarchy:
10958 * docs/plugins/gst-plugins-base-plugins.interfaces:
10959 * docs/plugins/gst-plugins-base-plugins.prerequisites:
10960 * docs/plugins/inspect/plugin-adder.xml:
10961 * docs/plugins/inspect/plugin-alsa.xml:
10962 * docs/plugins/inspect/plugin-app.xml:
10963 * docs/plugins/inspect/plugin-audioconvert.xml:
10964 * docs/plugins/inspect/plugin-audiorate.xml:
10965 * docs/plugins/inspect/plugin-audioresample.xml:
10966 * docs/plugins/inspect/plugin-audiotestsrc.xml:
10967 * docs/plugins/inspect/plugin-cdparanoia.xml:
10968 * docs/plugins/inspect/plugin-decodebin.xml:
10969 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
10970 * docs/plugins/inspect/plugin-gdp.xml:
10971 * docs/plugins/inspect/plugin-gnomevfs.xml:
10972 * docs/plugins/inspect/plugin-libvisual.xml:
10973 * docs/plugins/inspect/plugin-ogg.xml:
10974 * docs/plugins/inspect/plugin-pango.xml:
10975 * docs/plugins/inspect/plugin-playback.xml:
10976 * docs/plugins/inspect/plugin-queue2.xml:
10977 * docs/plugins/inspect/plugin-subparse.xml:
10978 * docs/plugins/inspect/plugin-tcp.xml:
10979 * docs/plugins/inspect/plugin-theora.xml:
10980 * docs/plugins/inspect/plugin-typefindfunctions.xml:
10981 * docs/plugins/inspect/plugin-uridecodebin.xml:
10982 * docs/plugins/inspect/plugin-video4linux.xml:
10983 * docs/plugins/inspect/plugin-videorate.xml:
10984 * docs/plugins/inspect/plugin-videoscale.xml:
10985 * docs/plugins/inspect/plugin-videotestsrc.xml:
10986 * docs/plugins/inspect/plugin-volume.xml:
10987 * docs/plugins/inspect/plugin-vorbis.xml:
10988 * docs/plugins/inspect/plugin-ximagesink.xml:
10989 * docs/plugins/inspect/plugin-xvimagesink.xml:
10990 * gst-plugins-base.doap:
11020 * win32/common/config.h:
11022 Original commit message from CVS:
11025 2009-01-19 22:01:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11057 Original commit message from CVS:
11060 2009-01-16 11:44:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11062 gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
11063 Original commit message from CVS:
11064 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
11065 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
11066 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
11067 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
11068 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
11069 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
11070 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
11071 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
11072 Use correct struct alignment everywhere to prevent unaligned
11073 memory accesses, resulting in SIGBUS on sparc and probably others.
11076 2009-01-16 11:40:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11078 gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
11079 Original commit message from CVS:
11080 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
11081 Forward unknown events upstream to allow latency configuration.
11084 2009-01-13 14:47:19 +0000 Wim Taymans <wim.taymans@gmail.com>
11086 gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
11087 Original commit message from CVS:
11088 * gst/playback/gstplaybin2.c: (groups_set_locked_state):
11089 Provide the right arguments to a debug line.
11091 2009-01-13 06:51:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11093 sys/xvimage/xvimagesink.c: Don't reset the colorkey when element is reused. Fixes #567511.
11094 Original commit message from CVS:
11095 * sys/xvimage/xvimagesink.c:
11096 Don't reset the colorkey when element is reused. Fixes #567511.
11098 2009-01-09 23:42:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11100 configure.ac: 0.10.21.3 pre-release
11101 Original commit message from CVS:
11103 0.10.21.3 pre-release
11105 2009-01-09 23:13:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11107 gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
11108 Original commit message from CVS:
11109 * gst-libs/gst/app/gstappsink.c:
11110 Store the returned signal id in the right slot when
11111 registering the pull-buffer signal.
11113 Spotted by: Thomas Vander Stichele <thomas at apestaart dot org>
11115 2009-01-09 17:17:50 +0000 Tim-Philipp Müller <tim@centricular.net>
11117 gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
11118 Original commit message from CVS:
11119 * gst-libs/gst/interfaces/mixer.c:
11120 Small docs addition to clarify that one really mustn't free
11121 the constant GList returned (#566812).
11123 2009-01-08 17:18:24 +0000 Wim Taymans <wim.taymans@gmail.com>
11125 Add GType for GstRTSPUrl and expose a copy function because we can.
11126 Original commit message from CVS:
11127 * docs/libs/gst-plugins-base-libs-sections.txt:
11128 * gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
11129 (gst_rtsp_url_get_type), (gst_rtsp_url_copy):
11130 * gst-libs/gst/rtsp/gstrtspurl.h:
11131 * win32/common/libgstrtsp.def:
11132 Add GType for GstRTSPUrl and expose a copy function because we can.
11133 API: gst_rtsp_url_copy()
11136 2009-01-07 18:36:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11138 Add plugin dependency for the GIO and GVfs modules.
11139 Original commit message from CVS:
11141 * ext/gio/gstgio.c: (plugin_init):
11142 Add plugin dependency for the GIO and GVfs modules.
11145 2009-01-07 18:32:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11147 Add plugin dependency for the gnomevfs modules.
11148 Original commit message from CVS:
11150 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
11151 Add plugin dependency for the gnomevfs modules.
11154 2009-01-07 18:30:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11156 win32/common/libgstcdda.def: Add new symbol to the list of exported symbols.
11157 Original commit message from CVS:
11158 * win32/common/libgstcdda.def:
11159 Add new symbol to the list of exported symbols.
11161 2009-01-07 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
11163 gst/playback/gstplaybin2.c: Fix some comments and docs.
11164 Original commit message from CVS:
11165 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
11166 (gst_play_bin_set_uri), (gst_play_bin_set_suburi),
11167 (no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
11168 (activate_group), (deactivate_group), (groups_set_locked_state),
11169 (gst_play_bin_change_state):
11170 Fix some comments and docs.
11171 Post an error message when we fail to link the selector to the sink.
11172 Remove pushing of EOS, this seems unneeded.
11173 Lock the state of deactivated groups so that they don't accidentally
11174 reactivate when the playbin2 state changes.
11175 Reuse uridecodebins.
11176 Unlock and relock state of groups when playbin goes to NULL.
11179 * gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
11180 Only do something in the pad removed callback when we are dealing with
11181 our sourcepads because the sinkpads don't have a ghostpad.
11183 2009-01-07 10:50:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11185 gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
11186 Original commit message from CVS:
11187 * gst-libs/gst/cdda/gstcddabasesrc.c:
11188 * gst-libs/gst/cdda/gstcddabasesrc.h:
11189 Make the GType of GstCDDABaseSrcMode public for bindings.
11192 2009-01-06 18:03:51 +0000 Tim-Philipp Müller <tim@centricular.net>
11194 Use new core API to make registry re-scan the plugin whenever visualisations are added or removed (see #350477).
11195 Original commit message from CVS:
11197 * ext/libvisual/visual.c: (plugin_init):
11198 Use new core API to make registry re-scan the plugin
11199 whenever visualisations are added or removed (see #350477).
11201 2009-01-06 17:30:31 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
11203 gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
11204 Original commit message from CVS:
11205 Patch by: José Alburquerque <jaalburqu svn gnome org>
11206 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
11207 * gst-libs/gst/audio/gstaudioclock.h:
11208 Make gst_audio_clock_new use const gchar* to ease the wrapping of
11209 C++ bindings. Fixes #566723.
11211 2009-01-06 12:16:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11213 Add pkg-config files for libgstapp. Fixes bug #566761.
11214 Original commit message from CVS:
11216 * pkgconfig/Makefile.am:
11217 * pkgconfig/gstreamer-app-uninstalled.pc.in:
11218 * pkgconfig/gstreamer-app.pc.in:
11219 Add pkg-config files for libgstapp. Fixes bug #566761.
11221 2009-01-06 11:10:29 +0000 Tim-Philipp Müller <tim@centricular.net>
11223 gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
11224 Original commit message from CVS:
11225 * gst-libs/gst/app/gstappsink.c:
11226 * gst-libs/gst/app/gstappsink.h:
11227 * gst-libs/gst/app/gstappsrc.c:
11228 * gst-libs/gst/app/gstappsrc.h:
11229 Make debug categories static. Use _element_class_set_details_simple().
11231 2009-01-06 10:56:45 +0000 Tim-Philipp Müller <tim@centricular.net>
11233 gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
11234 Original commit message from CVS:
11235 * gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
11236 (gst_app_sink_class_init), (gst_app_sink_init),
11237 (gst_app_sink_dispose), (gst_app_sink_finalize),
11238 (gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
11239 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
11240 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
11241 (gst_app_sink_render), (gst_app_sink_getcaps),
11242 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
11243 (gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
11244 (gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
11245 (gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
11246 (gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
11247 (gst_app_sink_pull_buffer)::
11248 * gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
11249 * gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
11250 (gst_app_src_class_init), (gst_app_src_init),
11251 (gst_app_src_flush_queued), (gst_app_src_dispose),
11252 (gst_app_src_finalize), (gst_app_src_set_property),
11253 (gst_app_src_get_property), (gst_app_src_unlock),
11254 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
11255 (gst_app_src_is_seekable), (gst_app_src_check_get_range),
11256 (gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
11257 (gst_app_src_set_caps), (gst_app_src_get_caps),
11258 (gst_app_src_set_size), (gst_app_src_get_size),
11259 (gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
11260 (gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
11261 (gst_app_src_set_latencies), (gst_app_src_set_latency),
11262 (gst_app_src_get_latency), (gst_app_src_push_buffer_full),
11263 (gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
11264 * gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
11265 Move private data into a private instance struct. Add padding to
11266 instance and class structures exposed in public headers. Add
11267 Since markers to the gtk-doc blurbs (#566750).
11269 2009-01-06 10:50:37 +0000 Wim Taymans <wim.taymans@gmail.com>
11271 tests/examples/app/appsrc_ex.c: Some comments.
11272 Original commit message from CVS:
11273 * tests/examples/app/appsrc_ex.c: (main):
11275 When pulling a buffer we can get NULL when the element is EOS, don't try
11276 to unref this NULL buffer.
11278 2009-01-06 10:16:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11280 gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
11281 Original commit message from CVS:
11282 * gst-libs/gst/video/Makefile.am:
11283 * gst-libs/gst/video/video.h:
11284 Fix up build flags and include statement for the new generated
11285 enumtypes files, to fix dist.
11287 2009-01-05 23:04:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11289 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
11290 Original commit message from CVS:
11292 * docs/libs/Makefile.am:
11293 * docs/libs/gst-plugins-base-libs-docs.sgml:
11294 * docs/libs/gst-plugins-base-libs-sections.txt:
11295 * docs/plugins/Makefile.am:
11296 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
11297 * docs/plugins/gst-plugins-base-plugins-sections.txt:
11298 * docs/plugins/gst-plugins-base-plugins.args:
11299 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11300 * docs/plugins/gst-plugins-base-plugins.interfaces:
11301 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11302 * docs/plugins/gst-plugins-base-plugins.signals:
11303 * docs/plugins/inspect/plugin-app.xml:
11304 * gst-libs/gst/Makefile.am:
11305 * gst-libs/gst/app/gstappsink.c:
11306 * gst-libs/gst/app/gstappsrc.c:
11307 * tests/examples/Makefile.am:
11308 * tests/examples/app/Makefile.am:
11309 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
11311 2009-01-05 17:13:13 +0000 Wim Taymans <wim.taymans@gmail.com>
11313 gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
11314 Original commit message from CVS:
11315 * gst-libs/gst/audio/gstbaseaudiosink.c:
11316 (gst_base_audio_sink_change_state):
11317 Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
11318 take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
11319 this because the async_play method is deprecated and usually not called
11322 2009-01-05 12:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
11324 gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
11325 Original commit message from CVS:
11326 * gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
11327 Disconnect signal handlers before destroying a previous decodebin so
11328 that we don't end up causing deadlocks. Fixes #566586.
11330 2009-01-05 10:59:35 +0000 Wim Taymans <wim.taymans@gmail.com>
11332 gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
11333 Original commit message from CVS:
11334 * gst/audiotestsrc/gstaudiotestsrc.c:
11335 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
11336 (gst_audio_test_src_check_get_range),
11337 (gst_audio_test_src_set_property),
11338 (gst_audio_test_src_get_property):
11339 * gst/audiotestsrc/gstaudiotestsrc.h:
11340 Add property to control pull/push based scheduling.
11342 2009-01-02 15:04:13 +0000 Alessandro Decina <alessandro.d@gmail.com>
11344 Make the seek and colorkey examples depend on gtk+-x11 as they use
11345 Original commit message from CVS:
11347 * tests/examples/seek/Makefile.am:
11348 * tests/icles/Makefile.am:
11349 Make the seek and colorkey examples depend on gtk+-x11 as they use
11351 Fixes the build with gtk+-quartz.
11353 2008-12-31 16:04:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11355 win32/common/: Add new exports to win32 files.
11356 Original commit message from CVS:
11357 * win32/common/libgstaudio.def:
11358 * win32/common/libgsttag.def:
11359 * win32/common/libgstvideo.def:
11360 Add new exports to win32 files.
11362 2008-12-31 13:31:55 +0000 Edward Hervey <bilboed@bilboed.com>
11364 gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
11365 Original commit message from CVS:
11366 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
11367 * gst-libs/gst/tag/gsttagdemux.h:
11368 Add GType for GstTagDemuxResult enum.
11370 2008-12-31 13:01:30 +0000 Edward Hervey <bilboed@bilboed.com>
11372 gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
11373 Original commit message from CVS:
11374 * gst-libs/gst/video/Makefile.am:
11375 * gst-libs/gst/video/video.h:
11376 Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
11377 This will help bindings to use it.
11379 2008-12-31 11:20:26 +0000 Edward Hervey <bilboed@bilboed.com>
11381 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
11382 Original commit message from CVS:
11383 * gst-libs/gst/audio/Makefile.am:
11384 * gst-libs/gst/audio/audio.c:
11385 * gst-libs/gst/audio/multichannel.h:
11386 * gst-libs/gst/audio/testchannels.c:
11388 * win32/common/audio-enumtypes.c:
11389 (gst_audio_channel_position_get_type),
11390 (gst_ring_buffer_state_get_type),
11391 (gst_ring_buffer_seg_state_get_type),
11392 (gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
11393 * win32/common/audio-enumtypes.h:
11394 * win32/common/multichannel-enumtypes.c:
11395 * win32/common/multichannel-enumtypes.h:
11396 * win32/vs6/grammar.dsp:
11397 * win32/vs6/libgstaudio.dsp:
11398 * win32/vs7/libgstaudio.vcproj:
11399 * win32/vs8/libgstaudio.vcproj:
11400 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
11401 audio- in order to wrap all enums declarations of that library.
11402 This modification should not matter since that header file is not a
11403 public header (it will be included by public headers).
11404 Modify win32 crap^Wfiles accordingly.
11406 2008-12-30 17:55:07 +0000 Edward Hervey <bilboed@bilboed.com>
11408 gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
11409 Original commit message from CVS:
11410 * gst-libs/gst/audio/gstbaseaudiosrc.h:
11411 * gst-libs/gst/audio/gstbaseaudiosink.h:
11412 Complete Sebastien's commit from the 13th by exporting the
11413 _slave_method_get_type() methods.
11415 2008-12-29 16:45:20 +0000 Wim Taymans <wim.taymans@gmail.com>
11417 gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
11418 Original commit message from CVS:
11419 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
11420 (gst_app_src_init), (gst_app_src_set_property),
11421 (gst_app_src_get_property), (gst_app_src_query),
11422 (gst_app_src_set_latencies), (gst_app_src_set_latency),
11423 (gst_app_src_get_latency), (gst_app_src_push_buffer_full):
11424 * gst-libs/gst/app/gstappsrc.h:
11425 Add properties and methods to configure and retrieve the min and max
11428 2008-12-20 17:38:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11430 ext/: Implement URI query. Fixes bug #562949.
11431 Original commit message from CVS:
11432 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_query):
11433 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_class_init),
11434 (gst_gio_base_src_query):
11435 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_query):
11436 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
11437 (gst_gnome_vfs_src_query):
11438 Implement URI query. Fixes bug #562949.
11440 2008-12-20 12:48:43 +0000 Wim Taymans <wim.taymans@gmail.com>
11442 gst/playback/gstplaybin2.c: Add some debug info.
11443 Original commit message from CVS:
11444 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
11445 Add some debug info.
11446 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
11447 (gst_play_sink_reconfigure), (gst_play_sink_request_pad),
11448 (gst_play_sink_release_pad):
11449 Add some more debug info.
11450 Reconfigure the audio chain when we switch between raw and encoded audio
11451 in gapless playback.
11453 2008-12-20 12:45:03 +0000 Wim Taymans <wim.taymans@gmail.com>
11455 gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
11456 Original commit message from CVS:
11457 * gst-libs/gst/audio/gstbaseaudiosink.c:
11458 (gst_base_audio_sink_setcaps):
11459 Pause the write thread before deactivating and releasing the ringbuffer
11460 to avoid a deadlock when we do gapless playback with different sample
11461 rates in playbin2. Fixes #564929.
11463 2008-12-19 13:03:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11465 gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
11466 Original commit message from CVS:
11467 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11468 Make GstAudioSrcSlaveMethod get_type() function non-static
11469 as it's public now.
11470 * win32/common/libgstaudio.def:
11471 * win32/common/libgstnetbuffer.def:
11472 Add some missing functions to the list of exported symbols.
11474 2008-12-18 12:37:33 +0000 Andrew Feren <acferen@yahoo.com>
11476 gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
11477 Original commit message from CVS:
11478 Patch by: Andrew Feren <acferen at yahoo dot com>
11479 * gst-libs/gst/netbuffer/gstnetbuffer.c:
11480 (gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
11481 (gst_netaddress_get_address_bytes),
11482 (gst_netaddress_set_address_bytes):
11483 * gst-libs/gst/netbuffer/gstnetbuffer.h:
11484 Make gst_netaddress_get_ip4_address fail for v6 addresses.
11485 Make gst_netaddress_get_ip6_address either fail or return the v4
11486 address as a transitional v6 address.
11487 Add two convenience functions:
11488 API: gst_netaddress_get_address_bytes()
11489 API: gst_netaddress_set_address_bytes()
11492 2008-12-17 13:51:46 +0000 Wim Taymans <wim.taymans@gmail.com>
11494 Add appsrc and appsink documentation.
11495 Original commit message from CVS:
11496 * docs/plugins/Makefile.am:
11497 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
11498 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
11499 * gst-libs/gst/app/gstappsink.c:
11500 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
11501 Add appsrc and appsink documentation.
11503 2008-12-17 08:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11505 gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
11506 Original commit message from CVS:
11507 * gst/adder/Makefile.am:
11508 * gst/adder/gstadder.c:
11509 Cleanup variable names to make the adder-loop easier to understand.
11510 Also try to use liboil to spee it up, but ifdef it out as it does not
11511 make any change for me (Intel pentim M (sse,sse2) please try on other
11514 2008-12-16 20:16:17 +0000 Wim Taymans <wim.taymans@gmail.com>
11516 Add minimal docs to make the remaining tcp elements show up.
11517 Original commit message from CVS:
11518 * docs/plugins/Makefile.am:
11519 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
11520 * docs/plugins/gst-plugins-base-plugins-sections.txt:
11521 * gst/tcp/gsttcpclientsink.c:
11522 * gst/tcp/gsttcpclientsrc.c:
11523 * gst/tcp/gsttcpserversrc.c:
11524 Add minimal docs to make the remaining tcp elements show up.
11527 2008-12-15 12:02:26 +0000 Wim Taymans <wim.taymans@gmail.com>
11529 examples/app/: Fix example to unref after emiting the push-buffer action.
11530 Original commit message from CVS:
11531 * examples/app/appsrc-ra.c: (feed_data):
11532 * examples/app/appsrc-seekable.c: (feed_data):
11533 * examples/app/appsrc-stream.c: (read_data):
11534 * examples/app/appsrc-stream2.c: (feed_data):
11535 Fix example to unref after emiting the push-buffer action.
11536 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
11537 (gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
11538 (gst_app_src_push_buffer_action):
11539 Don't take the ref on the buffer in push-buffer action because it's too
11540 awkward for bindings. Fixes #564482.
11542 2008-12-13 19:32:13 +0000 Tim-Philipp Müller <tim@centricular.net>
11544 win32/common/config.h: Update to CVS version.
11545 Original commit message from CVS:
11546 * win32/common/config.h:
11547 Update to CVS version.
11548 * win32/common/config.h.in:
11549 Hardcode path to plugin install helper exe, just like we hardcode
11550 the paths in core. Removes another source of VCS conflicts for
11551 people hacking gst-plugins-base on systems with autotools.
11553 2008-12-13 16:21:12 +0000 Edward Hervey <bilboed@bilboed.com>
11555 m4/Makefile.am: And a couple more .m4 that don't exist anymore with gettext 0.17
11556 Original commit message from CVS:
11558 And a couple more .m4 that don't exist anymore with gettext 0.17
11560 2008-12-13 12:41:56 +0000 Edward Hervey <bilboed@bilboed.com>
11562 m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we...
11563 Original commit message from CVS:
11565 inttypes.m4 hasn't been available since gettext-0.15, and since we now
11566 require gettext >= 0.17 ... we can remove it from the list of files to
11569 2008-12-13 06:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11571 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
11572 Original commit message from CVS:
11573 * gst-libs/gst/audio/gstbaseaudiosink.c:
11574 (gst_base_audio_sink_slave_method_get_type),
11575 (gst_base_audio_sink_class_init):
11576 * gst-libs/gst/audio/gstbaseaudiosink.h:
11577 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11578 (gst_base_audio_src_slave_method_get_type),
11579 (gst_base_audio_src_class_init):
11580 * gst-libs/gst/audio/gstbaseaudiosrc.h:
11581 API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
11582 public API. This is needed for the C++ bindings to be able
11583 to use this base classes. Fixes bug #564200, #564206.
11585 2008-12-12 19:41:28 +0000 Edward Hervey <bilboed@bilboed.com>
11587 gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
11588 Original commit message from CVS:
11589 * gst-libs/gst/cdda/gstcddabasesrc.c:
11590 (gst_cdda_base_src_handle_event):
11591 Remove erroneous gst_buffer_ref().
11592 * tests/check/libs/rtp.c: (GST_START_TEST):
11593 Don't forget to unref the buffer once you're done with it.
11595 2008-12-12 13:06:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11597 gst/playback/: XRef to GstXOverlay.
11598 Original commit message from CVS:
11599 * gst/playback/gstplaybin.c:
11600 * gst/playback/gstplaybin2.c:
11601 XRef to GstXOverlay.
11603 2008-12-12 10:54:45 +0000 Edward Hervey <bilboed@bilboed.com>
11605 gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
11606 Original commit message from CVS:
11607 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
11608 Free the factory array when finalizing.
11609 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
11610 Use a GstStaticPadTemplate since the src pad caps are fixed.
11612 2008-12-12 07:17:21 +0000 Edward Hervey <bilboed@bilboed.com>
11614 ext/vorbis/vorbisenc.c: Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with pad templates.
11615 Original commit message from CVS:
11616 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_base_init),
11617 (gst_vorbis_enc_init):
11618 Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with
11621 2008-12-12 07:15:22 +0000 Edward Hervey <bilboed@bilboed.com>
11623 gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
11624 Original commit message from CVS:
11625 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
11626 (gst_riff_create_video_template_caps):
11627 Add mapping for VP6 in avi/riff.
11629 2008-12-11 15:49:12 +0000 Edward Hervey <bilboed@bilboed.com>
11631 gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
11632 Original commit message from CVS:
11633 * gst/subparse/samiparse.c: (sami_context_push_state),
11634 (sami_context_pop_state), (start_sami_element), (end_sami_element):
11635 Some versions of libxml seem to be very picky as to strict formatting
11636 of the input and never 'close' the final </body> tag.
11637 In order to fix that bad behaviour, we trigger the flushing of
11638 remaining data on both </body> and </sami>.
11641 2008-12-11 12:32:03 +0000 Guillaume Emont <guillaume@fluendo.com>
11643 gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
11644 Original commit message from CVS:
11645 Patch by: Guillaume Emont <guillaume at fluendo dot com>
11646 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
11647 Add typefinders for MS Word files and OS X .DS_Store files to
11648 prevent them to be recognized as MPEG files. Fixes bug #564098.
11650 2008-12-11 11:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
11652 gst/playback/gstplaysink.c: Add some more debug info.
11653 Original commit message from CVS:
11654 * gst/playback/gstplaysink.c: (gen_audio_chain),
11655 (gst_play_sink_reconfigure):
11656 Add some more debug info.
11657 Fix linking of just an encoded sink.
11658 Handle failure to create a sink chain more gracefully than crashing.
11660 2008-12-11 10:33:48 +0000 Wim Taymans <wim.taymans@gmail.com>
11662 tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
11663 Original commit message from CVS:
11664 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
11665 Pushing 10 buffers is enough to run the test.
11667 2008-12-11 10:28:43 +0000 Wim Taymans <wim.taymans@gmail.com>
11669 tests/examples/seek/seek.c: Hook up the SKIP seek flag.
11670 Original commit message from CVS:
11671 * tests/examples/seek/seek.c: (do_seek), (stop_cb),
11672 (skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
11674 Hook up the SKIP seek flag.
11676 2008-12-10 18:43:32 +0000 Wim Taymans <wim.taymans@gmail.com>
11678 gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
11679 Original commit message from CVS:
11680 * gst/playback/gstplaybin2.c: (pad_added_cb):
11681 Error out with a missing-plugin error when the input-selector was not
11683 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
11686 2008-12-10 17:39:32 +0000 Wim Taymans <wim.taymans@gmail.com>
11688 gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
11689 Original commit message from CVS:
11690 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
11691 (gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
11692 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
11693 (gst_play_sink_send_event), (gst_play_sink_change_state):
11695 Try to set the selected sink to READY before using it. This will allow
11696 for detection of incompatible formats sooner.
11697 Don't cause a fatal error when conversion elements are missing but post
11698 a missing-element message and a warning instead because things might
11699 still link and run fine.
11700 Simplyfy the construction of audio and video sink chains.
11702 2008-12-10 14:55:10 +0000 Wim Taymans <wim.taymans@gmail.com>
11704 ext/ogg/gstoggdemux.c: Use G_DEFINE_TYPE for the OggPad to get some threadsafe type init from glib.
11705 Original commit message from CVS:
11706 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
11707 (gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
11708 Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
11711 2008-12-10 08:19:13 +0000 Luis Menina <liberforce@freeside.fr>
11713 gst/: Include glib.h instead of a specific GLib header. Including single
11714 Original commit message from CVS:
11715 Patch by: Luis Menina <liberforce at freeside dot fr>
11716 * gst-libs/gst/floatcast/floatcast.h:
11717 * gst/typefind/gsttypefindfunctions.c:
11718 Include glib.h instead of a specific GLib header. Including single
11719 GLib headers is deprecated. Fixes bug #563904.
11721 2008-12-09 18:30:10 +0000 Julien Moutte <julien@moutte.net>
11723 gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
11724 Original commit message from CVS:
11725 2008-12-09 Julien Moutte <julien@fluendo.com>
11726 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
11727 Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
11729 2008-12-09 17:21:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11731 gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
11732 Original commit message from CVS:
11733 * gst-libs/gst/riff/riff-read.c:
11734 Fix handling of odd chunks in riff metadata.
11736 2008-12-08 18:44:22 +0000 Wim Taymans <wim.taymans@gmail.com>
11738 gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
11739 Original commit message from CVS:
11740 * gst/volume/gstvolume.c: (gst_volume_class_init),
11741 (volume_before_transform), (volume_transform_ip):
11742 Use new basetransform vmethod to reconfigure the dynamic properties and
11743 any pending volume/mute changes. Fixes #563508.
11745 2008-12-08 18:12:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11747 configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...
11748 Original commit message from CVS:
11750 First check for "theoraenc theoradec" and if that failed check
11751 for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
11752 deprecate the latter. Also linking on Windows fails with just "theora"
11753 and the version check would fail for the release candidates.
11756 2008-12-08 15:25:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11758 gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
11759 Original commit message from CVS:
11760 * gst/playback/gstdecodebin.c:
11761 * gst/playback/gstdecodebin2.c:
11762 Add basic docs to decodebin and link to decodebin from decodebin2.
11764 2008-12-08 12:08:32 +0000 Olivier Crete <tester@tester.ca>
11766 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
11767 Original commit message from CVS:
11768 Patch by: Olivier Crete <tester at tester ca>
11769 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
11770 * gst-libs/gst/rtp/gstrtcpbuffer.h:
11771 Implement gst_rtcp_packet_remove(). Fixes #563174.
11772 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
11773 Add unit test for some RTCP functions.
11775 2008-12-04 20:09:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11777 configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.
11778 Original commit message from CVS:
11780 Apparently AC_CONFIG_MACRO_DIR breaks when using more
11781 than one macro directory, reverting last change.
11783 2008-12-04 19:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11785 configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros.
11786 Original commit message from CVS:
11788 Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
11791 2008-12-03 17:47:44 +0000 Edward Hervey <bilboed@bilboed.com>
11793 sys/: Clear all flags on buffers returned from the image pool.
11794 Original commit message from CVS:
11795 * sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
11796 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
11797 Clear all flags on buffers returned from the image pool.
11800 2008-12-01 19:36:35 +0000 이문형 <iwings@gmail.com>
11802 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...
11803 Original commit message from CVS:
11804 Patch by: 이문형 <iwings at gmail dot com>
11805 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
11806 Don't forget to release the lock again if we bail out because some
11807 pad is flushing or we've reached EOS, otherwise things will lock up
11808 next time _push_buffer() is called (#562802).
11810 2008-11-29 13:31:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11812 Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
11813 Original commit message from CVS:
11814 Patch by: Cygwin Ports maintainer
11815 <yselkowitz at users dot sourceforge dot net>
11818 Require gettext 0.17 because older versions don't mix with libtool
11819 2.2. At build time an older gettext version will still work.
11822 2008-11-28 13:30:36 +0000 Christian Schaller <uraeus@gnome.org>
11825 * gst/speexresample/Makefile.am:
11827 Original commit message from CVS:
11830 2008-11-28 09:44:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11832 Update documentation of speexresample for the new element name.
11833 Original commit message from CVS:
11834 * docs/plugins/gst-plugins-base-plugins.args:
11835 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11836 * docs/plugins/gst-plugins-base-plugins.interfaces:
11837 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11838 * docs/plugins/inspect/plugin-videorate.xml:
11839 * gst/speexresample/gstspeexresample.c:
11840 Update documentation of speexresample for the new element name.
11842 2008-11-28 09:04:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11844 gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
11845 Original commit message from CVS:
11846 * gst/speexresample/README:
11847 Update README with the latest diff between the Speex resampler
11850 2008-11-28 08:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11852 gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
11853 Original commit message from CVS:
11854 * gst/speexresample/gstspeexresample.c: (plugin_init):
11855 Update the debug category from speex_resample to audioresample.
11857 2008-11-27 19:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11859 Remove audioresample files.
11860 Original commit message from CVS:
11861 * gst/audioresample/Makefile.am:
11862 * gst/audioresample/buffer.c:
11863 * gst/audioresample/buffer.h:
11864 * gst/audioresample/debug.c:
11865 * gst/audioresample/debug.h:
11866 * gst/audioresample/functable.c:
11867 * gst/audioresample/functable.h:
11868 * gst/audioresample/gstaudioresample.c:
11869 * gst/audioresample/gstaudioresample.h:
11870 * gst/audioresample/resample.c:
11871 * gst/audioresample/resample.h:
11872 * gst/audioresample/resample_chunk.c:
11873 * gst/audioresample/resample_functable.c:
11874 * gst/audioresample/resample_ref.c:
11875 * tests/check/elements/audioresample.c:
11876 Remove audioresample files.
11878 2008-11-27 17:04:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11880 docs/plugins/inspect/plugin-audioresample.xml: Regenerated for library filename change.
11881 Original commit message from CVS:
11882 * docs/plugins/inspect/plugin-audioresample.xml:
11883 Regenerated for library filename change.
11885 2008-11-27 16:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11887 Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
11888 Original commit message from CVS:
11890 * docs/plugins/Makefile.am:
11891 * docs/plugins/gst-plugins-base-plugins-sections.txt:
11892 * docs/plugins/gst-plugins-base-plugins.args:
11893 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11894 * docs/plugins/gst-plugins-base-plugins.interfaces:
11895 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11896 * docs/plugins/inspect/plugin-adder.xml:
11897 * docs/plugins/inspect/plugin-alsa.xml:
11898 * docs/plugins/inspect/plugin-audioconvert.xml:
11899 * docs/plugins/inspect/plugin-audiorate.xml:
11900 * docs/plugins/inspect/plugin-audioresample.xml:
11901 * docs/plugins/inspect/plugin-audiotestsrc.xml:
11902 * docs/plugins/inspect/plugin-cdparanoia.xml:
11903 * docs/plugins/inspect/plugin-decodebin.xml:
11904 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
11905 * docs/plugins/inspect/plugin-gdp.xml:
11906 * docs/plugins/inspect/plugin-gio.xml:
11907 * docs/plugins/inspect/plugin-gnomevfs.xml:
11908 * docs/plugins/inspect/plugin-libvisual.xml:
11909 * docs/plugins/inspect/plugin-ogg.xml:
11910 * docs/plugins/inspect/plugin-pango.xml:
11911 * docs/plugins/inspect/plugin-playback.xml:
11912 * docs/plugins/inspect/plugin-queue2.xml:
11913 * docs/plugins/inspect/plugin-subparse.xml:
11914 * docs/plugins/inspect/plugin-tcp.xml:
11915 * docs/plugins/inspect/plugin-theora.xml:
11916 * docs/plugins/inspect/plugin-typefindfunctions.xml:
11917 * docs/plugins/inspect/plugin-uridecodebin.xml:
11918 * docs/plugins/inspect/plugin-video4linux.xml:
11919 * docs/plugins/inspect/plugin-videorate.xml:
11920 * docs/plugins/inspect/plugin-videoscale.xml:
11921 * docs/plugins/inspect/plugin-videotestsrc.xml:
11922 * docs/plugins/inspect/plugin-volume.xml:
11923 * docs/plugins/inspect/plugin-vorbis.xml:
11924 * docs/plugins/inspect/plugin-ximagesink.xml:
11925 * docs/plugins/inspect/plugin-xvimagesink.xml:
11926 * gst/speexresample/gstspeexresample.c: (plugin_init):
11927 * gst/speexresample/Makefile.am:
11928 * tests/check/Makefile.am:
11929 * tests/check/elements/speexresample.c: (setup_speexresample),
11930 (GST_START_TEST), (test_pipeline):
11931 Rename the moved speexresample to audioresample, integrate into the
11932 build system and remove the old audioresample from the build system.
11933 Fixes bug #558124, #385061, #346218, #116051.
11935 2008-11-27 16:47:41 +0000 Wim Taymans <wim.taymans@gmail.com>
11937 gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
11938 Original commit message from CVS:
11939 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11940 (gst_base_audio_src_get_offset), (gst_base_audio_src_create):
11941 Avoid nasty int overflows after about 12 hours and 25 minutes when these
11942 code paths are triggered.
11943 A free beer to Håvard Graff for finding this!
11945 2008-11-27 11:16:44 +0000 이문형 <iwings@gmail.com>
11947 gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
11948 Original commit message from CVS:
11949 Patch by: 이문형 <iwings at gmail dot com>
11950 * gst-libs/gst/rtsp/gstrtspconnection.c:
11951 (gst_rtsp_connection_connect):
11952 A successful gst_poll_wait() doesn't always mean successful connect() on
11953 Windows. We should check errors by calling gst_poll_fd_has_error().
11956 2008-11-25 16:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11958 tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.
11959 Original commit message from CVS:
11960 * tests/check/elements/speexresample.c: (test_pipeline):
11961 Make unit test again faster to prevent timeouts with valgrind.
11963 2008-11-25 15:33:30 +0000 Wim Taymans <wim.taymans@gmail.com>
11965 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
11966 Original commit message from CVS:
11967 * gst-libs/gst/rtp/gstrtcpbuffer.c:
11968 Fix typo in the docs.
11970 2008-11-25 15:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
11972 ext/ogg/gstoggdemux.c: If no stream was found before receiving EOS, post an error message.
11973 Original commit message from CVS:
11974 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
11975 If no stream was found before receiving EOS, post an error message.
11978 2008-11-25 15:14:30 +0000 Wim Taymans <wim.taymans@gmail.com>
11980 ext/theora/: Parse segment events.
11981 Original commit message from CVS:
11982 * ext/theora/gsttheoraenc.h:
11983 * ext/theora/theoraenc.c: (gst_theora_enc_init),
11984 (theora_buffer_from_packet), (theora_push_packet),
11985 (theora_enc_sink_event), (theora_enc_is_discontinuous),
11986 (theora_enc_chain):
11987 Parse segment events.
11988 Pass incomming buffer timestamps to outgoing buffers.
11989 Use the running_time to construct the granulepos.
11992 2008-11-25 11:00:55 +0000 Wim Taymans <wim.taymans@gmail.com>
11994 gst/playback/gstplaybin2.c: Fix buffer-duration property.
11995 Original commit message from CVS:
11996 * gst/playback/gstplaybin2.c: (activate_group):
11997 Fix buffer-duration property.
11999 2008-11-25 10:32:49 +0000 Wim Taymans <wim.taymans@gmail.com>
12001 gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
12002 Original commit message from CVS:
12003 * gst-libs/gst/audio/gstbaseaudiosink.c:
12004 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
12005 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
12006 (gst_base_audio_sink_change_state):
12007 Really fix audiosink drain handling by keeping track of the running_time
12008 of the last sample.
12010 2008-11-24 20:25:24 +0000 Michael Smith <msmith@xiph.org>
12012 gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
12013 Original commit message from CVS:
12014 * gst/playback/gstplaybin2.c:
12015 Add notification of current stream. Add ability to configure buffer
12017 * gst/playback/gsturidecodebin.c:
12018 Add ability to configure buffer sizes for streaming mode.
12021 2008-11-24 20:11:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12023 gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
12024 Original commit message from CVS:
12025 * gst-libs/gst/audio/gstbaseaudiosink.c:
12026 Time is already in running_time. Remove base_time handling. Fixes
12027 audiosinks not draining and thus chopping some audio in the end.
12029 2008-11-24 19:18:59 +0000 David Schleef <ds@schleef.org>
12031 ext/ogg/gstoggmux.*: If we're muxing a dirac stream, flush the page after every picture.
12032 Original commit message from CVS:
12033 * ext/ogg/gstoggmux.c:
12034 * ext/ogg/gstoggmux.h:
12035 If we're muxing a dirac stream, flush the page after every picture.
12037 2008-11-24 12:56:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12039 gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
12040 Original commit message from CVS:
12041 * gst-libs/gst/audio/gstbaseaudiosink.c:
12042 Add one log message to check for audio_drained. Sync one log message
12043 with the condition. Send EOS after draining audio in pull mode.
12045 2008-11-24 12:07:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12047 ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr...
12048 Original commit message from CVS:
12049 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
12050 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
12051 Use gst_buffer_try_new_and_alloc() and fail properly if the
12052 allocation failed. This prevents abort() if downstream elements
12053 request an insane amount of memory.
12055 2008-11-24 12:03:11 +0000 Jon Trowbridge <trow@ximian.com>
12057 gst/volume/gstvolume.*: Cleanup volume, define and use default values.
12058 Original commit message from CVS:
12059 * gst/volume/gstvolume.c: (volume_choose_func),
12060 (volume_update_volume), (gst_volume_set_volume),
12061 (gst_volume_get_volume), (gst_volume_set_mute),
12062 (gst_volume_class_init), (gst_volume_init),
12063 (volume_process_double), (volume_process_float),
12064 (volume_process_int32), (volume_process_int32_clamp),
12065 (volume_process_int24), (volume_process_int24_clamp),
12066 (volume_process_int16), (volume_process_int16_clamp),
12067 (volume_process_int8), (volume_process_int8_clamp), (volume_setup),
12068 (volume_transform_ip), (volume_set_property),
12069 (volume_get_property):
12070 * gst/volume/gstvolume.h:
12071 Cleanup volume, define and use default values.
12072 Recalculate new volume and mute setup before processing. Fixes #561789.
12073 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
12074 Add controller unit test. Patch by: Jonathan Matthew
12075 Fix bogus test that messed with basetransform's internal state.
12077 2008-11-22 15:02:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12079 tests/check/elements/speexresample.c: Make the unit test a bit faster to prevent timeouts, especially with valgrind.
12080 Original commit message from CVS:
12081 * tests/check/elements/speexresample.c: (GST_START_TEST):
12082 Make the unit test a bit faster to prevent timeouts, especially
12085 2008-11-22 14:44:26 +0000 Wim Taymans <wim.taymans@gmail.com>
12087 gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
12088 Original commit message from CVS:
12089 * gst/videorate/gstvideorate.c:
12090 Add jpeg and png image media types to the caps. Fixes #561436.
12092 2008-11-22 14:31:43 +0000 Wim Taymans <wim.taymans@gmail.com>
12094 gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ...
12095 Original commit message from CVS:
12096 * gst/playback/gstplaysink.c: (gen_audio_chain):
12097 Don't post an error when we can't configure the volume but post a
12098 warning instead. Fixes #561780.
12100 2008-11-21 20:32:56 +0000 Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
12102 gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video...
12103 Original commit message from CVS:
12104 Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
12105 * gst/videotestsrc/gstvideotestsrc.c:
12106 * gst/videotestsrc/gstvideotestsrc.h:
12107 * gst/videotestsrc/videotestsrc.c:
12108 * gst/videotestsrc/videotestsrc.h:
12109 Add a zone plate pattern generator based on BBC R&D Report
12110 1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate
12111 kx2=20 ky2=20 kt=1'.
12113 2008-11-21 15:45:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12115 gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
12116 Original commit message from CVS:
12117 * gst/speexresample/gstspeexresample.c:
12118 (gst_speex_resample_class_init), (gst_speex_resample_set_property),
12119 (gst_speex_resample_get_property):
12120 Add a "filter-length" property that maps to the quality values
12121 for compatibilty with audioresample.
12123 2008-11-21 00:04:48 +0000 Michael Smith <msmith@xiph.org>
12125 gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
12126 Original commit message from CVS:
12127 * gst/playback/gstdecodebin2.c:
12128 Fix random fat-fingering making this not compile.
12130 2008-11-20 22:11:38 +0000 Michael Smith <msmith@xiph.org>
12132 gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh...
12133 Original commit message from CVS:
12134 * gst/playback/gstdecodebin2.c:
12135 If the top-level type of the stream is plain text, don't try to decode
12136 it, matching behaviour of decodebin.
12137 * gst/playback/gstplaysink.c:
12138 If we fail to generate a text chain (e.g. due to missing optional
12139 plugins), don't crash.
12141 2008-11-20 22:06:05 +0000 Michael Smith <msmith@xiph.org>
12143 gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
12144 Original commit message from CVS:
12145 * gst-libs/gst/rtsp/gstrtspdefs.c:
12146 Fix win32 build. Oops.
12148 2008-11-20 21:40:49 +0000 Michael Smith <msmith@xiph.org>
12150 gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
12151 Original commit message from CVS:
12152 * gst-libs/gst/rtsp/gstrtspdefs.c:
12153 Use WSAGetLastError() rather than errno/h_errno on win32.
12155 2008-11-20 21:20:27 +0000 Michael Smith <msmith@xiph.org>
12157 gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
12158 Original commit message from CVS:
12159 * gst-libs/gst/riff/riff-media.c:
12160 Support WMA Lossless properly.
12162 2008-11-19 00:24:44 +0000 David Schleef <ds@schleef.org>
12164 gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect...
12165 Original commit message from CVS:
12166 * gst/videotestsrc/gstvideotestsrc.c:
12167 * gst/videotestsrc/gstvideotestsrc.h:
12168 * gst/videotestsrc/videotestsrc.c:
12169 * gst/videotestsrc/videotestsrc.h:
12170 Add "colorspec" property, specifying whether to generate BT.601
12171 or BT.709 video. This only affects YCbCr values, not RGB, since
12172 if you're generating a 709 test pattern, presumably you want
12173 709 RGB primaries, not 601. Also add "smpte75" pattern, which
12174 uses 75% colors instead of 100%, since this is often more useful
12175 for testing (and also follows the SMPTE EG-1 guideline).
12177 2008-11-18 18:08:42 +0000 Alessandro Decina <alessandro.d@gmail.com>
12179 gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2.
12180 Original commit message from CVS:
12181 * gst/playback/gstdecodebin.c:
12182 Add a "sink-caps" property to decodebin like it's done for decodebin2.
12185 2008-11-14 21:44:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12187 gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
12188 Original commit message from CVS:
12189 * gst/audioresample/gstaudioresample.c:
12190 Guard against a NULL dereference I somehow encountered -
12191 with a FLUSH_STOP arriving either before basetransform _start(),
12193 * gst/typefind/gsttypefindfunctions.c:
12194 Make sure we never jump backwards when typefinding corrupt mov files.
12196 2008-11-14 21:39:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12198 gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
12199 Original commit message from CVS:
12200 * gst-libs/gst/interfaces/propertyprobe.c:
12201 Fix random type causing a docs warning.
12203 2008-11-14 15:40:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12205 sys/v4l/gstv4l.c: Give it a minimal rank for autovideosrc.
12206 Original commit message from CVS:
12207 * sys/v4l/gstv4l.c:
12208 Give it a minimal rank for autovideosrc.
12210 2008-11-13 21:11:13 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
12212 gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types.
12213 Original commit message from CVS:
12214 * gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
12216 Improve typefinding of ISO JPEG2000 mime types.
12218 2008-11-13 18:18:32 +0000 Wim Taymans <wim.taymans@gmail.com>
12220 sys/xvimage/xvimagesink.*: Avoid typechecking when we do trivial casts.
12221 Original commit message from CVS:
12222 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
12223 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps),
12224 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
12225 * sys/xvimage/xvimagesink.h:
12226 Avoid typechecking when we do trivial casts.
12227 Move error handling out of the main program flow.
12228 Sneak in the display-region caps property, not completely correct yet.
12229 Cache the width/height in buffer_alloc instead of parsing it from the
12232 2008-11-13 17:27:37 +0000 Wim Taymans <wim.taymans@gmail.com>
12234 gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
12235 Original commit message from CVS:
12236 * gst/playback/gstplaybin2.c: (deactivate_group):
12237 don't try to unlink the selector sinkpad when we don't have it yet. This
12238 can happen if an error occured before the group was complete.
12240 2008-11-13 15:37:40 +0000 Wim Taymans <wim.taymans@gmail.com>
12242 gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
12243 Original commit message from CVS:
12244 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
12245 (gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
12246 (gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
12247 (gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
12248 (gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
12249 (gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
12250 (gst_rtp_buffer_get_extension_data),
12251 (gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
12252 (gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
12253 (gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
12254 (gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
12255 (gst_rtp_buffer_get_payload_type),
12256 (gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
12257 (gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
12258 (gst_rtp_buffer_set_timestamp),
12259 (gst_rtp_buffer_get_payload_subbuffer),
12260 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
12261 Avoid expensive type checks we already did as part of the
12262 _validate() function that should be called first.
12264 2008-11-11 16:40:50 +0000 Wim Taymans <wim.taymans@gmail.com>
12266 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some cases where a newsegment event was not sent.
12267 Original commit message from CVS:
12268 * gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
12269 (gst_base_rtp_depayload_push_full),
12270 (gst_base_rtp_depayload_set_gst_timestamp):
12271 Fix some cases where a newsegment event was not sent.
12273 2008-11-11 15:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
12275 gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co...
12276 Original commit message from CVS:
12277 * gst/playback/gstplaybin2.c: (activate_group):
12278 Catch state change errors and stop from the uridecodebin elements
12279 instead of trying to continue in vain.
12281 2008-11-10 14:53:45 +0000 Edward Hervey <bilboed@bilboed.com>
12283 gst/: Wim, you're a bad boy. You don't want people to contact you or what?
12284 Original commit message from CVS:
12285 * gst-libs/gst/app/gstappsink.c:
12286 * gst-libs/gst/app/gstappsrc.c:
12287 * gst/h264parse/gsth264parse.c:
12288 Wim, you're a bad boy. You don't want people to contact you or what?
12290 2008-11-10 14:22:09 +0000 Wim Taymans <wim.taymans@gmail.com>
12292 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
12293 Original commit message from CVS:
12294 * gst-libs/gst/audio/gstbaseaudiosink.c:
12295 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
12296 (gst_base_audio_sink_callback):
12297 Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
12298 for the latency to expire, fixes #559567.
12300 2008-11-10 13:55:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
12302 gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
12303 Original commit message from CVS:
12304 * gst/adder/gstadder.c:
12305 Change author string after seeing output of gst-inspector.
12307 2008-11-10 10:33:26 +0000 Wim Taymans <wim.taymans@gmail.com>
12309 gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559...
12310 Original commit message from CVS:
12311 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
12312 Don't try to do crazy things when we only have a text pad without a
12313 video pad. Fixes #559478.
12315 2008-11-07 17:35:46 +0000 Wim Taymans <wim.taymans@gmail.com>
12317 gst-libs/gst/app/gstappsrc.*: Add is-live property.
12318 Original commit message from CVS:
12319 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
12320 (gst_app_src_init), (gst_app_src_set_property),
12321 (gst_app_src_get_property), (gst_app_src_push_buffer):
12322 * gst-libs/gst/app/gstappsrc.h:
12323 Add is-live property.
12324 Add some more docs.
12326 2008-11-06 12:14:51 +0000 Wim Taymans <wim.taymans@gmail.com>
12328 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...
12329 Original commit message from CVS:
12330 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
12331 Fix case where we don't have a range for the rates or channels as is the
12332 case with truespeech.
12334 2008-11-05 19:18:25 +0000 Wim Taymans <wim.taymans@gmail.com>
12336 gst/volume/gstvolume.*: Keep negotiated state in a separate variable.
12337 Original commit message from CVS:
12338 * gst/volume/gstvolume.c: (volume_update_real_volume),
12339 (gst_volume_set_volume), (gst_volume_get_volume),
12340 (gst_volume_set_mute), (gst_volume_init), (volume_setup),
12341 (volume_transform_ip), (volume_update_mute),
12342 (volume_update_volume), (volume_get_property):
12343 * gst/volume/gstvolume.h:
12344 Keep negotiated state in a separate variable.
12345 Protect the volume and mute properties with the object lock.
12346 Protect modifying the transform with the transform lock.
12348 2008-11-05 12:20:21 +0000 Wim Taymans <wim.taymans@gmail.com>
12350 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Only convert caps to string when debug is enabled.
12351 Original commit message from CVS:
12352 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
12353 (gst_ffmpeg_pixfmt_to_caps):
12354 Only convert caps to string when debug is enabled.
12356 2008-11-04 18:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
12358 ext/theora/: Copy seqnum.
12359 Original commit message from CVS:
12360 * ext/theora/gsttheoradec.h:
12361 * ext/theora/theoradec.c: (gst_theora_dec_init),
12362 (gst_theora_dec_reset), (theora_dec_src_event),
12363 (theora_dec_sink_event), (theora_handle_type_packet):
12365 Keep events in a pending list, like vorbisdec, instead of trying
12366 to construct a segment event ourselves.
12367 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
12368 (vorbis_dec_src_event), (vorbis_dec_sink_event):
12369 * ext/vorbis/vorbisdec.h:
12372 2008-11-04 17:24:35 +0000 Wim Taymans <wim.taymans@gmail.com>
12374 ext/ogg/gstoggdemux.*: Copy seqnums around to track playback segments and messages.
12375 Original commit message from CVS:
12376 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
12377 (gst_ogg_demux_deactivate_current_chain),
12378 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page),
12379 (gst_ogg_demux_loop):
12380 * ext/ogg/gstoggdemux.h:
12381 Copy seqnums around to track playback segments and messages.
12383 2008-11-04 12:42:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12385 Don't install static libs for plugins. Fixes #550851 for -bad.
12386 Original commit message from CVS:
12387 * ext/alsaspdif/Makefile.am:
12388 * ext/amrwb/Makefile.am:
12389 * ext/apexsink/Makefile.am:
12390 * ext/arts/Makefile.am:
12391 * ext/artsd/Makefile.am:
12392 * ext/audiofile/Makefile.am:
12393 * ext/audioresample/Makefile.am:
12394 * ext/bz2/Makefile.am:
12395 * ext/cdaudio/Makefile.am:
12396 * ext/celt/Makefile.am:
12397 * ext/dc1394/Makefile.am:
12398 * ext/dirac/Makefile.am:
12399 * ext/directfb/Makefile.am:
12400 * ext/divx/Makefile.am:
12401 * ext/dts/Makefile.am:
12402 * ext/faac/Makefile.am:
12403 * ext/faad/Makefile.am:
12404 * ext/gsm/Makefile.am:
12405 * ext/hermes/Makefile.am:
12406 * ext/ivorbis/Makefile.am:
12407 * ext/jack/Makefile.am:
12408 * ext/jp2k/Makefile.am:
12409 * ext/ladspa/Makefile.am:
12410 * ext/lcs/Makefile.am:
12411 * ext/libfame/Makefile.am:
12412 * ext/libmms/Makefile.am:
12413 * ext/metadata/Makefile.am:
12414 * ext/mpeg2enc/Makefile.am:
12415 * ext/mplex/Makefile.am:
12416 * ext/musepack/Makefile.am:
12417 * ext/musicbrainz/Makefile.am:
12418 * ext/mythtv/Makefile.am:
12419 * ext/nas/Makefile.am:
12420 * ext/neon/Makefile.am:
12421 * ext/ofa/Makefile.am:
12422 * ext/polyp/Makefile.am:
12423 * ext/resindvd/Makefile.am:
12424 * ext/sdl/Makefile.am:
12425 * ext/shout/Makefile.am:
12426 * ext/snapshot/Makefile.am:
12427 * ext/sndfile/Makefile.am:
12428 * ext/soundtouch/Makefile.am:
12429 * ext/spc/Makefile.am:
12430 * ext/swfdec/Makefile.am:
12431 * ext/tarkin/Makefile.am:
12432 * ext/theora/Makefile.am:
12433 * ext/timidity/Makefile.am:
12434 * ext/twolame/Makefile.am:
12435 * ext/x264/Makefile.am:
12436 * ext/xine/Makefile.am:
12437 * ext/xvid/Makefile.am:
12438 * gst-libs/gst/app/Makefile.am:
12439 * gst-libs/gst/dshow/Makefile.am:
12440 * gst/aiffparse/Makefile.am:
12441 * gst/app/Makefile.am:
12442 * gst/audiobuffer/Makefile.am:
12443 * gst/bayer/Makefile.am:
12444 * gst/cdxaparse/Makefile.am:
12445 * gst/chart/Makefile.am:
12446 * gst/colorspace/Makefile.am:
12447 * gst/dccp/Makefile.am:
12448 * gst/deinterlace/Makefile.am:
12449 * gst/deinterlace2/Makefile.am:
12450 * gst/dvdspu/Makefile.am:
12451 * gst/festival/Makefile.am:
12452 * gst/filter/Makefile.am:
12453 * gst/flacparse/Makefile.am:
12454 * gst/flv/Makefile.am:
12455 * gst/games/Makefile.am:
12456 * gst/h264parse/Makefile.am:
12457 * gst/librfb/Makefile.am:
12458 * gst/mixmatrix/Makefile.am:
12459 * gst/modplug/Makefile.am:
12460 * gst/mpeg1sys/Makefile.am:
12461 * gst/mpeg4videoparse/Makefile.am:
12462 * gst/mpegdemux/Makefile.am:
12463 * gst/mpegtsmux/Makefile.am:
12464 * gst/mpegvideoparse/Makefile.am:
12465 * gst/mve/Makefile.am:
12466 * gst/nsf/Makefile.am:
12467 * gst/nuvdemux/Makefile.am:
12468 * gst/overlay/Makefile.am:
12469 * gst/passthrough/Makefile.am:
12470 * gst/pcapparse/Makefile.am:
12471 * gst/playondemand/Makefile.am:
12472 * gst/rawparse/Makefile.am:
12473 * gst/real/Makefile.am:
12474 * gst/rtjpeg/Makefile.am:
12475 * gst/rtpmanager/Makefile.am:
12476 * gst/scaletempo/Makefile.am:
12477 * gst/sdp/Makefile.am:
12478 * gst/selector/Makefile.am:
12479 * gst/smooth/Makefile.am:
12480 * gst/smoothwave/Makefile.am:
12481 * gst/speed/Makefile.am:
12482 * gst/speexresample/Makefile.am:
12483 * gst/stereo/Makefile.am:
12484 * gst/subenc/Makefile.am:
12485 * gst/tta/Makefile.am:
12486 * gst/vbidec/Makefile.am:
12487 * gst/videodrop/Makefile.am:
12488 * gst/videosignal/Makefile.am:
12489 * gst/virtualdub/Makefile.am:
12490 * gst/vmnc/Makefile.am:
12491 * gst/y4m/Makefile.am:
12492 * sys/acmenc/Makefile.am:
12493 * sys/cdrom/Makefile.am:
12494 * sys/dshowdecwrapper/Makefile.am:
12495 * sys/dshowsrcwrapper/Makefile.am:
12496 * sys/dvb/Makefile.am:
12497 * sys/dxr3/Makefile.am:
12498 * sys/fbdev/Makefile.am:
12499 * sys/oss4/Makefile.am:
12500 * sys/qcam/Makefile.am:
12501 * sys/qtwrapper/Makefile.am:
12502 * sys/vcd/Makefile.am:
12503 * sys/wininet/Makefile.am:
12504 * win32/common/config.h:
12505 Don't install static libs for plugins. Fixes #550851 for -bad.
12507 2008-11-03 15:30:14 +0000 Matthias Kretz <kretz@kde.org>
12509 ext/alsa/gstalsasink.c: Make all access non-blocking so that we can better handle unplugging of usb devices. Fixes #5...
12510 Original commit message from CVS:
12511 Based on patch by: Matthias Kretz <kretz at kde dot org>
12512 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
12513 (gst_alsasink_prepare), (gst_alsasink_unprepare),
12514 (gst_alsasink_write):
12515 Make all access non-blocking so that we can better handle unplugging
12516 of usb devices. Fixes #559111
12518 2008-11-03 10:49:24 +0000 Damien Lespiau <damien.lespiau@gmail.com>
12520 gst-libs/gst/rtsp/gstrtspconnection.c: Make the next call to poll not depend on previous calls to poll with or withou...
12521 Original commit message from CVS:
12522 Patch by: Damien Lespiau <damien.lespiau gmail com>
12523 * gst-libs/gst/rtsp/gstrtspconnection.c:
12524 (gst_rtsp_connection_write):
12525 Make the next call to poll not depend on previous calls to poll with or
12526 without reading from the active descriptor. Fixes #544293.
12528 2008-11-03 08:55:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12530 gst/speexresample/gstspeexresample.c: Add TODO at the top of the file for enabling SSE/ARM specific optimizations and...
12531 Original commit message from CVS:
12532 * gst/speexresample/gstspeexresample.c:
12533 (gst_speex_resample_convert_buffer):
12534 Add TODO at the top of the file for enabling SSE/ARM specific
12535 optimizations and choosing the fastest implementation at runtime.
12536 Add g_assert_not_reached() at two places that should really never
12539 2008-11-02 09:19:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12541 gst/speexresample/gstspeexresample.c: Fix format string and arguments.
12542 Original commit message from CVS:
12543 * gst/speexresample/gstspeexresample.c:
12544 (gst_speex_resample_check_discont):
12545 Fix format string and arguments.
12546 * gst/speexresample/resample_sse.h:
12549 2008-11-01 19:38:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12551 gst/speexresample/: Add missing headers to Makefile.am.
12552 Original commit message from CVS:
12553 * gst/speexresample/Makefile.am:
12554 * gst/speexresample/gstspeexresample.c:
12555 (gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
12556 (gst_speex_resample_convert_buffer), (_benchmark_int_float),
12557 (_benchmark_int_int), (_benchmark_integer_resampling),
12559 * gst/speexresample/gstspeexresample.h:
12560 * gst/speexresample/resample.c:
12561 * gst/speexresample/speex_resampler_double.c:
12562 * gst/speexresample/speex_resampler_float.c:
12563 * gst/speexresample/speex_resampler_int.c:
12564 * gst/speexresample/speex_resampler_wrapper.h:
12565 Add missing headers to Makefile.am.
12566 Update copyright, years and my mail address.
12567 Benchmark the integer resampling implementation against the
12568 float implementation and use the faster one for 8/16 bit integer
12569 input. On most recent systems the floating point version is faster.
12571 2008-10-31 09:49:57 +0000 Nick Haddad <nick@haddads.net>
12573 gst-libs/gst/riff/: Add support for other fourcc codes that are commonly used for 'uncompressed RGB', including 'RGB ...
12574 Original commit message from CVS:
12575 Patch by: Nick Haddad <nick at haddads dot net>
12576 * gst-libs/gst/riff/riff-ids.h:
12577 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
12578 Add support for other fourcc codes that are commonly used for
12579 'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
12582 2008-10-30 14:55:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12584 gst/speexresample/gstspeexresample.c: The length for the buffer conversion function is the number of audio frames, i....
12585 Original commit message from CVS:
12586 * gst/speexresample/gstspeexresample.c:
12587 (gst_speex_resample_convert_buffer):
12588 The length for the buffer conversion function is the number of
12589 audio frames, i.e. we need to multiply it by the number of channels
12590 to get the number of values. Also spotted by the unit test after
12591 running in valgrind.
12593 2008-10-30 14:46:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12595 tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
12596 Original commit message from CVS:
12597 * tests/check/elements/speexresample.c: (element_message_cb),
12598 (eos_message_cb), (test_pipeline), (GST_START_TEST),
12599 (speexresample_suite):
12600 Add pipeline unit tests for testing all supported formats with
12601 up/downsampling and different in/outrates.
12602 * gst/speexresample/gstspeexresample.c:
12603 (gst_speex_resample_push_drain), (gst_speex_resample_process):
12604 * gst/speexresample/speex_resampler_wrapper.h:
12605 Fix bugs identified by the testsuite.
12607 2008-10-30 13:44:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12609 gst/speexresample/: Add support for int8, int24 and int32 input by converting internally to/from int16 or double.
12610 Original commit message from CVS:
12611 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
12612 (gst_speex_resample_get_funcs),
12613 (gst_speex_resample_transform_size),
12614 (gst_speex_resample_convert_buffer),
12615 (gst_speex_resample_push_drain), (gst_speex_resample_process):
12616 * gst/speexresample/gstspeexresample.h:
12617 * gst/speexresample/speex_resampler_wrapper.h:
12618 Add support for int8, int24 and int32 input by converting internally
12619 to/from int16 or double.
12621 2008-10-30 12:43:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12623 Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
12624 Original commit message from CVS:
12625 * gst/speexresample/Makefile.am:
12626 * gst/speexresample/arch.h:
12627 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
12628 (gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
12629 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
12630 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
12631 (_gcd), (gst_speex_resample_transform_size),
12632 (gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
12633 (gst_speex_resample_process), (gst_speex_resample_transform),
12634 (gst_speex_resample_query), (gst_speex_resample_set_property):
12635 * gst/speexresample/gstspeexresample.h:
12636 * gst/speexresample/resample.c:
12637 * gst/speexresample/speex_resampler.h:
12638 * gst/speexresample/speex_resampler_double.c:
12639 * gst/speexresample/speex_resampler_wrapper.h:
12640 * tests/check/elements/speexresample.c: (setup_speexresample),
12641 (test_perfect_stream_instance), (GST_START_TEST),
12642 (test_discont_stream_instance):
12643 Add support for double samples as input and refactor the usage
12644 of the different compilation flavors of the speex resampler.
12646 2008-10-30 11:43:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12648 gst/audioresample/gstaudioresample.c: Return the result of parent_class->event().
12649 Original commit message from CVS:
12650 * gst/audioresample/gstaudioresample.c:
12651 Return the result of parent_class->event().
12653 2008-10-29 17:02:55 +0000 Wim Taymans <wim.taymans@gmail.com>
12655 gst-libs/gst/app/gstappsink.c: Fix the docs.
12656 Original commit message from CVS:
12657 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
12660 2008-10-29 12:11:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12662 gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
12663 Original commit message from CVS:
12664 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
12665 (gst_speex_resample_get_unit_size),
12666 (gst_speex_resample_push_drain), (gst_speex_resample_event),
12667 (gst_speex_resample_check_discont), (gst_speex_resample_process),
12668 (gst_speex_resample_transform):
12669 * gst/speexresample/gstspeexresample.h:
12670 Rewrite timestamp tracking to make it more robust and guarantee
12671 a continous stream.
12672 * tests/check/Makefile.am:
12673 * tests/check/elements/speexresample.c: (setup_speexresample),
12674 (cleanup_speexresample), (fail_unless_perfect_stream),
12675 (test_perfect_stream_instance), (GST_START_TEST),
12676 (test_discont_stream_instance), (live_switch_alloc_only_48000),
12677 (live_switch_get_sink_caps), (live_switch_push),
12678 (speexresample_suite):
12679 Add unit tests for speexresample based on the audioresample unit tests.
12681 2008-10-28 19:30:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12683 gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
12684 Original commit message from CVS:
12685 * gst/speexresample/gstspeexresample.c:
12686 (gst_speex_resample_get_unit_size),
12687 (gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
12688 (gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
12689 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
12690 (gst_speex_resample_push_drain), (gst_speex_resample_event),
12691 (gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
12692 (gst_speex_resample_process), (gst_speex_resample_transform),
12693 (gst_speex_resample_query), (gst_speex_resample_set_property):
12694 * gst/speexresample/gstspeexresample.h:
12695 Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
12696 instead of GST_DEBUG, ...
12698 2008-10-28 16:28:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12700 gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
12701 Original commit message from CVS:
12702 * gst/speexresample/gstspeexresample.c:
12703 (gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
12704 (gst_speex_resample_process):
12705 Fixate to the nearest supported rate instead of the first one.
12707 2008-10-28 16:25:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12709 gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
12710 Original commit message from CVS:
12711 * gst/audioresample/gstaudioresample.c:
12712 (gst_audioresample_class_init), (audioresample_fixate_caps):
12713 Fixate the rate to the nearest supported rate instead of
12714 the first one. Fixes bug #549510.
12716 2008-10-28 11:46:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12718 gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
12719 Original commit message from CVS:
12720 * gst/speexresample/README:
12721 * gst/speexresample/arch.h:
12722 * gst/speexresample/fixed_arm4.h:
12723 * gst/speexresample/fixed_arm5e.h:
12724 * gst/speexresample/fixed_bfin.h:
12725 * gst/speexresample/fixed_debug.h:
12726 * gst/speexresample/fixed_generic.h:
12727 * gst/speexresample/resample.c: (compute_func), (main), (sinc),
12728 (cubic_coef), (resampler_basic_direct_single),
12729 (resampler_basic_direct_double),
12730 (resampler_basic_interpolate_single),
12731 (resampler_basic_interpolate_double), (update_filter),
12732 (speex_resampler_init_frac), (speex_resampler_process_native),
12733 (speex_resampler_magic), (speex_resampler_process_float),
12734 (speex_resampler_process_int),
12735 (speex_resampler_process_interleaved_float),
12736 (speex_resampler_process_interleaved_int),
12737 (speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
12738 (speex_resampler_reset_mem):
12739 * gst/speexresample/speex_resampler.h:
12740 Update Speex resampler with latest version from Speex GIT.
12742 2008-10-27 14:57:34 +0000 Wim Taymans <wim.taymans@gmail.com>
12744 win32/common/libgstaudio.def: Add new symbols.
12745 Original commit message from CVS:
12746 * win32/common/libgstaudio.def:
12749 2008-10-23 09:57:06 +0000 Wim Taymans <wim.taymans@gmail.com>
12751 ext/vorbis/vorbisdec.c: Attempt to make obfuscated code clearer.
12752 Original commit message from CVS:
12753 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
12754 Attempt to make obfuscated code clearer.
12756 2008-10-23 07:11:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12758 Move float endianness conversion macros to core. Second part of bug ##555196.
12759 Original commit message from CVS:
12760 * docs/libs/gst-plugins-base-libs-sections.txt:
12761 * gst-libs/gst/floatcast/floatcast.h:
12762 Move float endianness conversion macros to core. Second part of
12765 2008-10-22 12:29:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12767 sys/: Don't mark as gtk-doc docs as they aren't public.
12768 Original commit message from CVS:
12769 * sys/ximage/ximagesink.h:
12770 * sys/xvimage/xvimagesink.h:
12771 Don't mark as gtk-doc docs as they aren't public.
12773 2008-10-22 12:25:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12775 Allow setting colorkey if possible. Implement property probe interface for optional X features (autopaint-colorkey, d...
12776 Original commit message from CVS:
12777 * sys/xvimage/xvimagesink.c:
12778 * sys/xvimage/xvimagesink.h:
12779 * tests/icles/Makefile.am:
12780 * tests/icles/test-colorkey.c:
12781 Allow setting colorkey if possible. Implement property probe interface
12782 for optional X features (autopaint-colorkey, double-buffer and
12783 colorkey). Fixes #554533
12785 2008-10-22 12:01:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12787 gst-libs/gst/tag/tags.c: Remove useless buffer size assignment. It already has this value.
12788 Original commit message from CVS:
12789 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
12790 Remove useless buffer size assignment. It already has this value.
12792 2008-10-20 15:35:37 +0000 Wim Taymans <wim.taymans@gmail.com>
12794 gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
12795 Original commit message from CVS:
12796 * gst-libs/gst/audio/gstaudiosink.c:
12797 (gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
12798 (gst_audioringbuffer_activate), (gst_audioringbuffer_release),
12799 (gst_audioringbuffer_stop):
12800 Implement a separate activate functions to start monitoring the segments
12801 or, in pull mode, pulling in data.
12802 * gst-libs/gst/audio/gstbaseaudiosink.c:
12803 (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
12804 (gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
12805 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
12806 (gst_base_audio_sink_activate_pull),
12807 (gst_base_audio_sink_async_play),
12808 (gst_base_audio_sink_change_state):
12809 Implement pad and element convert query function.
12810 Activate the ringbuffer.
12811 Use the segment last_stop value as the offset to pull.
12812 Use new basesink _do_preroll() method to preroll in the pulling thread.
12813 Take appropriate locking in the pulling thread.
12814 * gst-libs/gst/audio/gstringbuffer.h:
12817 2008-10-20 14:08:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12819 gst/typefind/gsttypefindfunctions.c: Improve MXF typefinding a bit by searching for a header partition pack instead o...
12820 Original commit message from CVS:
12821 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
12822 Improve MXF typefinding a bit by searching for a header partition
12823 pack instead of just a general partition pack and checking more
12824 bytes for valid values.
12826 2008-10-20 13:45:55 +0000 Wim Taymans <wim.taymans@gmail.com>
12828 tests/icles/.cvsignore: update ignore file.
12829 Original commit message from CVS:
12830 * tests/icles/.cvsignore:
12831 update ignore file.
12832 * tests/icles/Makefile.am:
12833 * tests/icles/test-box.c: (make_pipeline), (main):
12834 Add another interactive command line experimentation suite for
12835 dynamically boxing/cropping/saling an input video.
12837 2008-10-17 13:19:05 +0000 Wim Taymans <wim.taymans@gmail.com>
12839 Add methods to more accuratly control the pulling thread of a ringbuffer.
12840 Original commit message from CVS:
12841 * docs/libs/gst-plugins-base-libs-sections.txt:
12842 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
12843 (gst_ring_buffer_activate), (gst_ring_buffer_is_active):
12844 * gst-libs/gst/audio/gstringbuffer.h:
12845 Add methods to more accuratly control the pulling thread of a
12847 Add format conversion helper code to the ringbuffer.
12848 API: GstRingBuffer:gst_ring_buffer_activate()
12849 API: GstRingBuffer:gst_ring_buffer_is_active()
12850 API: GstRingBuffer:gst_ring_buffer_convert()
12852 2008-10-16 15:44:37 +0000 Wim Taymans <wim.taymans@gmail.com>
12854 gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
12855 Original commit message from CVS:
12856 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
12857 (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
12858 (gst_audioringbuffer_stop):
12859 Signal thread startup earlier so that we can immediatly go into pull
12860 mode when we have to and block on preroll.
12862 2008-10-16 15:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
12864 gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
12865 Original commit message from CVS:
12866 * gst-libs/gst/audio/gstringbuffer.c:
12867 (gst_ring_buffer_prepare_read):
12868 In pull mode we want the callback to prepull a buffer we can preroll on
12869 even when we are not yet playing.
12871 2008-10-16 15:07:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12873 Don't install static libs for plugins. Fixes #550851 for base.
12874 Original commit message from CVS:
12875 * ext/alsa/Makefile.am:
12876 * ext/cdparanoia/Makefile.am:
12877 * ext/gio/Makefile.am:
12878 * ext/gnomevfs/Makefile.am:
12879 * ext/libvisual/Makefile.am:
12880 * ext/ogg/Makefile.am:
12881 * ext/pango/Makefile.am:
12882 * ext/theora/Makefile.am:
12883 * ext/vorbis/Makefile.am:
12884 * gst/adder/Makefile.am:
12885 * gst/audioconvert/Makefile.am:
12886 * gst/audiorate/Makefile.am:
12887 * gst/audioresample/Makefile.am:
12888 * gst/audiotestsrc/Makefile.am:
12889 * gst/ffmpegcolorspace/Makefile.am:
12890 * gst/gdp/Makefile.am:
12891 * gst/playback/Makefile.am:
12892 * gst/subparse/Makefile.am:
12893 * gst/tcp/Makefile.am:
12894 * gst/typefind/Makefile.am:
12895 * gst/videorate/Makefile.am:
12896 * gst/videoscale/Makefile.am:
12897 * gst/videotestsrc/Makefile.am:
12898 * gst/volume/Makefile.am:
12899 * sys/v4l/Makefile.am:
12900 * sys/ximage/Makefile.am:
12901 * sys/xvimage/Makefile.am:
12902 Don't install static libs for plugins. Fixes #550851 for base.
12904 2008-10-16 13:50:00 +0000 Wim Taymans <wim.taymans@gmail.com>
12906 gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe...
12907 Original commit message from CVS:
12908 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
12909 Set the default blocksize to -1 because we will then use the configured
12910 samplesperbuffer to create our output buffer.
12912 2008-10-15 15:28:41 +0000 Edward Hervey <bilboed@bilboed.com>
12914 gst-libs/gst/riff/riff-media.c: Add mappping for the KMVC (Karl Morton's Video) Codec.
12915 Original commit message from CVS:
12916 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
12917 (gst_riff_create_video_template_caps):
12918 Add mappping for the KMVC (Karl Morton's Video) Codec.
12920 2008-10-15 14:25:50 +0000 Edward Hervey <bilboed@bilboed.com>
12922 gst/typefind/gsttypefindfunctions.c: Don't forget to advance the offset of what we're matching against, else we end u...
12923 Original commit message from CVS:
12924 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
12925 Don't forget to advance the offset of what we're matching against, else
12926 we end up in a forever loop.
12928 2008-10-15 11:25:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12930 gst/subparse/gstsubparse.c: Improve typefinding a bit. If we don't have a Unicode charset try GST_SUBTITLE_ENCODING a...
12931 Original commit message from CVS:
12932 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
12933 Improve typefinding a bit. If we don't have a Unicode charset
12934 try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
12936 2008-10-14 11:13:59 +0000 Edward Hervey <bilboed@bilboed.com>
12938 ext/theora/theoradec.c: Fix build on macosx.
12939 Original commit message from CVS:
12940 * ext/theora/theoradec.c: (theora_dec_decode_buffer):
12941 Fix build on macosx.
12943 2008-10-13 11:36:13 +0000 Robin Stocker <robin@nibor.org>
12945 ext/theora/: Parse input caps and make the PAR override the encoded PAR when specified by a container. Fixes #555699.
12946 Original commit message from CVS:
12947 Based on patch by: Robin Stocker <robin at nibor dot org>
12948 * ext/theora/gsttheoradec.h:
12949 * ext/theora/theoradec.c: (gst_theora_dec_init),
12950 (theora_dec_setcaps), (theora_handle_type_packet),
12951 (theora_dec_decode_buffer), (theora_dec_change_state):
12952 Parse input caps and make the PAR override the encoded PAR when
12953 specified by a container. Fixes #555699.
12955 2008-10-13 09:16:59 +0000 Wim Taymans <wim.taymans@gmail.com>
12957 gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY
12958 Original commit message from CVS:
12959 * gst-libs/gst/rtp/gstbasertpdepayload.c:
12960 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
12961 (gst_base_rtp_depayload_set_gst_timestamp),
12962 (gst_base_rtp_depayload_change_state):
12963 * gst-libs/gst/rtp/gstbasertpdepayload.h:
12964 Add some more G_LIKELY
12965 Fail when the setcaps function was not called.
12966 * gst-libs/gst/rtp/gstbasertppayload.c:
12967 (gst_basertppayload_set_outcaps):
12968 Propagate return value of setcaps.
12970 2008-10-13 08:58:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12972 gst/subparse/: Add support for UTF16/UTF32 subtitles as long as the first bytes of the first buffer contain the BOM. ...
12973 Original commit message from CVS:
12974 * gst/subparse/Makefile.am:
12975 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
12976 (gst_sub_parse_class_init), (gst_sub_parse_init),
12977 (gst_convert_to_utf8), (detect_encoding), (convert_encoding),
12978 (get_next_line), (gst_sub_parse_data_format_autodetect),
12979 (feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
12980 (gst_subparse_type_find):
12981 * gst/subparse/gstsubparse.h:
12982 Add support for UTF16/UTF32 subtitles as long as the first bytes of
12983 the first buffer contain the BOM. This also adds support for other
12984 encodings that allow NUL bytes via the encoding property.
12985 Fixes bugs #552237 and #456788.
12987 2008-10-13 08:15:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12989 gst-libs/gst/tag/tags.c: Don't drop the last byte of image tags if they're not an URI list.
12990 Original commit message from CVS:
12991 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
12992 Don't drop the last byte of image tags if they're not an URI list.
12995 2008-10-13 08:00:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12997 gst/typefind/gsttypefindfunctions.c: For looking at the 4th byte we have to get 4 bytes of course and not 3.
12998 Original commit message from CVS:
12999 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
13000 For looking at the 4th byte we have to get 4 bytes of course
13003 2008-10-13 07:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13005 gst/typefind/gsttypefindfunctions.c: Improve FLAC-without-headers typefinding by looking at most of the frame header ...
13006 Original commit message from CVS:
13007 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
13008 Improve FLAC-without-headers typefinding by looking at most of the
13009 frame header and checking if invalid values are used. Should prevent
13010 quite some false positives compared to the old version which only
13011 check if the first 14 bits are set.
13013 2008-10-11 16:27:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13015 sys/xvimage/xvimagesink.c: Don't assert on caps==NULL.
13016 Original commit message from CVS:
13017 * sys/xvimage/xvimagesink.c:
13018 Don't assert on caps==NULL.
13020 2008-10-10 17:13:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13022 Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
13023 Original commit message from CVS:
13024 * gst/subparse/gstsubparse.c:
13025 (gst_sub_parse_data_format_autodetect), (handle_buffer),
13026 (gst_sub_parse_change_state):
13027 * gst/subparse/gstsubparse.h:
13028 * tests/check/elements/subparse.c: (GST_START_TEST):
13029 Add support for subtitle files with UTF-8 BOM at the beginning
13030 by simple stripping it from the first line before passing it
13031 to any parsing code. Fixes bug #555257 and playback of files
13032 created by Gnome Subtitles.
13034 2008-10-10 15:45:15 +0000 Wim Taymans <wim.taymans@gmail.com>
13036 gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place.
13037 Original commit message from CVS:
13038 * gst/audiotestsrc/gstaudiotestsrc.c:
13039 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
13040 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
13041 (gst_audio_test_src_start), (gst_audio_test_src_stop),
13042 (gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
13043 (gst_audio_test_src_create):
13044 * gst/audiotestsrc/gstaudiotestsrc.h:
13045 Define the default property values in the usual place.
13046 Implement start/stop to reset values correctly.
13047 Calculate the sample size only once when we negotiate.
13048 Rename some values to make more sense.
13049 Keep track of our byte range.
13050 Add support for pull based scheduling. Disabled for now until we have
13051 the whole stack working.
13052 Set the BUFFER_OFFSET correctly.
13054 2008-10-10 15:32:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13056 Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
13057 Original commit message from CVS:
13058 Based on a patch by: xavierb at gmail dot com
13059 * gst/subparse/gstsubparse.c:
13060 (gst_sub_parse_data_format_autodetect):
13061 * tests/check/elements/subparse.c: (GST_START_TEST):
13062 Make the detection of the used subtitle a bit less strict
13063 for srt subtitles. Fixes bug #555607.
13065 2008-10-10 15:21:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13067 ext/vorbis/vorbisenc.c: Fix discontinuity detection which was broken by last commit.
13068 Original commit message from CVS:
13069 * ext/vorbis/vorbisenc.c:
13070 (gst_vorbis_enc_buffer_check_discontinuous):
13071 Fix discontinuity detection which was broken by last commit.
13073 2008-10-09 11:18:09 +0000 Tim-Philipp Müller <tim@centricular.net>
13075 configure.ac: Require core CVS for ghostpad API additions used by decodebin2.
13076 Original commit message from CVS:
13078 Require core CVS for ghostpad API additions used by decodebin2.
13080 2008-10-08 15:30:33 +0000 Edward Hervey <bilboed@bilboed.com>
13082 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
13083 Original commit message from CVS:
13084 * gst-libs/gst/audio/gstbaseaudiosrc.c:
13085 (gst_base_audio_src_create):
13086 Fix debug statements (space between '%' and actual format).
13088 2008-10-08 14:44:04 +0000 Wim Taymans <wim.taymans@gmail.com>
13090 gst/playback/gstdecodebin2.c: Remove bogus assert, the decodepad could have been created inside an already existing g...
13091 Original commit message from CVS:
13092 * gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
13093 Remove bogus assert, the decodepad could have been created inside an
13094 already existing group.
13096 2008-10-08 14:01:42 +0000 Andy Wingo <wingo@pobox.com>
13100 Original commit message from CVS:
13103 2008-10-08 14:00:07 +0000 Andy Wingo <wingo@pobox.com>
13105 gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
13106 Original commit message from CVS:
13107 2008-10-08 Andy Wingo <wingo@pobox.com>
13108 * gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
13109 target instead of setting it.
13110 (gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
13111 API for a decode pad. The bugfix is that we set the group in
13112 activate(), not when the pad was created because it might be NULL
13114 (gst_decode_group_control_source_pad, gst_decode_group_expose):
13115 Update to use the API.
13117 2008-10-08 12:49:40 +0000 Andy Wingo <wingo@pobox.com>
13119 gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
13120 Original commit message from CVS:
13121 2008-10-08 Andy Wingo <wingo@pobox.com>
13122 * gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
13123 be a subclass of GstGhostPad.
13124 (analyze_new_pad): So, when emitting the signals that determine
13125 how we do autoplugging, already create the ghost pad and use it as
13126 the pad in the signal arguments. This allows applications to make
13127 a connection between the pad passed in e.g. autoplug-continue, and
13128 the pad passed in new-decoded-pad.
13129 (connect_pad, expose_pad): Update to receive the ghosted decode
13130 pad in the args, retargetting it as necessary if we have to plug
13131 the target pad through a multiqueue.
13132 (gst_decode_group_control_source_pad): Adapt to receive an
13133 already-ghosted pad that just needs activation, blocking, and
13134 drain notification.
13135 (sort_end_pads): Adapt for decode pads actually being pads.
13136 (gst_decode_group_expose): Adapt for decode pads actually being
13137 pads. Rewrite the decode pad names so they appear in order. Adds a
13138 new error case if we couldn't set the name.
13139 (gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
13141 (gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
13142 New API for the decode pad, needed because we shouldn't do these
13143 things inside gst_decode_pad_new(), but after.
13144 (gst_decode_pad_new): Change to actually make the real pad, and
13145 delay the blocking/drainage bits.
13147 2008-10-08 12:12:01 +0000 Daniel Drake <dsd@laptop.org>
13149 ext/ogg/gstoggmux.c: Unref all buffers when clearing collectpads. Fixes bug #546955.
13150 Original commit message from CVS:
13151 Patch by: Daniel Drake <dsd at laptop dot org>
13152 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
13153 Unref all buffers when clearing collectpads. Fixes bug #546955.
13155 2008-10-08 12:08:01 +0000 Klaas <klaas@rivercrew.net>
13157 ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
13158 Original commit message from CVS:
13159 Based on a patch by: Klaas <klaas at rivercrew dot net>
13160 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
13161 (gst_vorbis_enc_buffer_check_discontinuous),
13162 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
13163 * ext/vorbis/vorbisenc.h:
13164 Keep track of the upstream segments and use the running time on that
13165 segment instead of the buffer timestamp everywhere. Fixes bug #525807.
13167 2008-10-08 11:50:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13169 gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
13170 Original commit message from CVS:
13171 * gst/audioconvert/audioconvert.c: (audio_convert_convert):
13172 Prevent overflows with big buffer when calculating the size of
13173 the intermediate buffer by using gst_util_uint64_scale() instead of
13174 plain arithmetics. Fixes bug #552801.
13176 2008-10-08 10:49:15 +0000 Pavel Zeldin <pzeldin@gmail.com>
13178 ext/pango/gstclockoverlay.*: API: Add ability to specify format for date/time display by adding a "time-format" prope...
13179 Original commit message from CVS:
13180 Patch by: Pavel Zeldin <pzeldin at gmail dot com>
13181 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
13182 (gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
13183 (gst_clock_overlay_init), (gst_clock_overlay_set_property),
13184 (gst_clock_overlay_get_property):
13185 * ext/pango/gstclockoverlay.h:
13186 API: Add ability to specify format for date/time display by
13187 adding a "time-format" property.
13190 2008-10-08 09:22:26 +0000 Jan Gerber <j@oil21.org>
13192 gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
13193 Original commit message from CVS:
13194 Patch by: Jan Gerber <j at oil21 dot org>
13195 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
13196 (gst_riff_create_video_template_caps):
13197 Add FFV1 fourcc to support playback of FFMPEG lossless video
13198 in AVI. Fixes bug #555319.
13200 2008-10-08 09:12:36 +0000 Håvard Graff <havard.graff@tandberg.com>
13202 gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
13203 Original commit message from CVS:
13204 Patch by: Håvard Graff <havard dot graff at tandberg dot com>
13205 * gst-libs/gst/audio/gstbaseaudiosrc.c:
13206 (gst_base_audio_src_create):
13207 Implement skew clock slaving. Fixes #552559.
13209 2008-10-08 09:10:23 +0000 Wim Taymans <wim.taymans@gmail.com>
13211 gst-libs/gst/audio/: Fix include of config.h
13212 Original commit message from CVS:
13213 * gst-libs/gst/audio/multichannel.c:
13214 * gst-libs/gst/audio/testchannels.c:
13215 Fix include of config.h
13217 2008-10-06 16:36:20 +0000 Tero Saarni <tero.saarni@gmail.com>
13219 gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
13220 Original commit message from CVS:
13221 Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
13222 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
13223 (print_media), (gst_sdp_message_dump):
13224 Fix parsing of the c= field containing multicast addresses.
13226 Add the connection info to the session or streams.
13227 Fix parsing of the bandwidth.
13228 Add debugging for the connections and bandwidths for a media.
13229 Add debugging for the bandwidth of the session.
13231 2008-10-06 16:31:27 +0000 Wim Taymans <wim.taymans@gmail.com>
13233 gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
13234 Original commit message from CVS:
13235 * gst-libs/gst/rtp/gstbasertppayload.c:
13236 (gst_basertppayload_change_state):
13237 Configure the next seqnum and timestamp in the state change so that they
13238 can be queried soon after.
13240 2008-10-06 16:29:33 +0000 Wim Taymans <wim.taymans@gmail.com>
13242 gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
13243 Original commit message from CVS:
13244 * gst-libs/gst/rtp/gstbasertpdepayload.c:
13245 (gst_base_rtp_depayload_chain):
13246 Improve debugging of the rtptime.
13248 2008-10-05 11:33:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13250 configure.ac: Back to development -> 0.10.21.1
13251 Original commit message from CVS:
13253 Back to development -> 0.10.21.1
13255 2008-10-05 08:18:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13259 Original commit message from CVS:
13262 2008-10-05 08:11:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13264 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
13265 Original commit message from CVS:
13266 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
13268 Add typefinder for MXF.
13270 2008-10-05 08:10:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13272 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
13273 Original commit message from CVS:
13274 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
13276 Add typefinder for MXF.
13278 2008-10-03 15:19:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13280 tests/icles/Makefile.am: Only build test-colorkey if GTK+ is available.
13281 Original commit message from CVS:
13282 * tests/icles/Makefile.am:
13283 Only build test-colorkey if GTK+ is available.
13285 === release 0.10.21 ===
13287 2008-10-03 00:03:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13293 * docs/plugins/gst-plugins-base-plugins.args:
13294 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13295 * docs/plugins/gst-plugins-base-plugins.interfaces:
13296 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13297 * docs/plugins/inspect/plugin-adder.xml:
13298 * docs/plugins/inspect/plugin-alsa.xml:
13299 * docs/plugins/inspect/plugin-audioconvert.xml:
13300 * docs/plugins/inspect/plugin-audiorate.xml:
13301 * docs/plugins/inspect/plugin-audioresample.xml:
13302 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13303 * docs/plugins/inspect/plugin-cdparanoia.xml:
13304 * docs/plugins/inspect/plugin-decodebin.xml:
13305 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13306 * docs/plugins/inspect/plugin-gdp.xml:
13307 * docs/plugins/inspect/plugin-gio.xml:
13308 * docs/plugins/inspect/plugin-gnomevfs.xml:
13309 * docs/plugins/inspect/plugin-libvisual.xml:
13310 * docs/plugins/inspect/plugin-ogg.xml:
13311 * docs/plugins/inspect/plugin-pango.xml:
13312 * docs/plugins/inspect/plugin-playback.xml:
13313 * docs/plugins/inspect/plugin-queue2.xml:
13314 * docs/plugins/inspect/plugin-subparse.xml:
13315 * docs/plugins/inspect/plugin-tcp.xml:
13316 * docs/plugins/inspect/plugin-theora.xml:
13317 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13318 * docs/plugins/inspect/plugin-uridecodebin.xml:
13319 * docs/plugins/inspect/plugin-video4linux.xml:
13320 * docs/plugins/inspect/plugin-videorate.xml:
13321 * docs/plugins/inspect/plugin-videoscale.xml:
13322 * docs/plugins/inspect/plugin-videotestsrc.xml:
13323 * docs/plugins/inspect/plugin-volume.xml:
13324 * docs/plugins/inspect/plugin-vorbis.xml:
13325 * docs/plugins/inspect/plugin-ximagesink.xml:
13326 * docs/plugins/inspect/plugin-xvimagesink.xml:
13327 * gst-plugins-base.doap:
13328 * win32/common/config.h:
13330 Original commit message from CVS:
13333 2008-10-02 23:44:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13364 Original commit message from CVS:
13367 2008-09-28 22:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13369 configure.ac: 0.10.20.4 pre-release
13370 Original commit message from CVS:
13372 0.10.20.4 pre-release
13374 2008-09-25 10:46:00 +0000 ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>
13376 ext/theora/theoraparse.c: Set the BOS flag on the BOS packet. Fixes #553244.
13377 Original commit message from CVS:
13378 Patch by: ogg.k.ogg.k <ogg dot k dot ogg dot k at googlemail dot com>
13379 * ext/theora/theoraparse.c: (theora_parse_set_streamheader):
13380 Set the BOS flag on the BOS packet. Fixes #553244.
13382 2008-09-23 17:48:14 +0000 Wim Taymans <wim.taymans@gmail.com>
13384 gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
13385 Original commit message from CVS:
13386 * gst-libs/gst/rtsp/gstrtspmessage.c:
13387 (gst_rtsp_message_parse_request),
13388 (gst_rtsp_message_parse_response):
13389 Fix the g_return_val_if_fail() statements.
13391 2008-09-22 17:44:14 +0000 Michael Smith <msmith@xiph.org>
13393 gst-libs/gst/tag/gsttagdemux.c: Fail to activate if there's insufficient data in the file to be usable, preventing an...
13394 Original commit message from CVS:
13395 * gst-libs/gst/tag/gsttagdemux.c:
13396 Fail to activate if there's insufficient data in the file to be usable,
13397 preventing an assertion fail later. Fixes #552960
13399 2008-09-16 15:36:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13401 Commit stuff that should have gone in last week when I made the pre-releases:
13402 Original commit message from CVS:
13403 Commit stuff that should have gone in last week when I made the pre-releases:
13404 2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
13406 0.10.20.2 pre-release
13412 2008-09-15 15:11:18 +0000 Tim-Philipp Müller <tim@centricular.net>
13414 gst/: Recognise Kate subtitle streams (#550582).
13415 Original commit message from CVS:
13416 * gst-libs/gst/pbutils/descriptions.c:
13417 * gst/typefind/gsttypefindfunctions.c:
13418 Recognise Kate subtitle streams (#550582).
13420 2008-09-13 11:04:02 +0000 Tim-Philipp Müller <tim@centricular.net>
13422 gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
13423 Original commit message from CVS:
13424 * gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
13425 Remove trailing comma from enum list, which causes problems
13426 with -pendantic (#550729).
13428 2008-09-05 19:04:47 +0000 Tim-Philipp Müller <tim@centricular.net>
13430 gst-libs/gst/interfaces/propertyprobe.c: More sanity checks for our second-favourite interface.
13431 Original commit message from CVS:
13432 * gst-libs/gst/interfaces/propertyprobe.c:
13433 (gst_property_probe_get_properties),
13434 (gst_property_probe_get_property),
13435 (gst_property_probe_probe_property),
13436 (gst_property_probe_probe_property_name),
13437 (gst_property_probe_needs_probe),
13438 (gst_property_probe_needs_probe_name),
13439 (gst_property_probe_get_values),
13440 (gst_property_probe_get_values_name),
13441 (gst_property_probe_probe_and_get_values),
13442 (gst_property_probe_probe_and_get_values_name):
13443 More sanity checks for our second-favourite interface.
13445 2008-09-05 14:12:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13447 gst-libs/gst/interfaces/propertyprobe.c: Check for NULL pointer, in the hope that this fixes #532864.
13448 Original commit message from CVS:
13449 * gst-libs/gst/interfaces/propertyprobe.c:
13450 Check for NULL pointer, in the hope that this fixes #532864.
13452 2008-09-05 10:24:05 +0000 Tim-Philipp Müller <tim@centricular.net>
13454 sys/xvimage/xvimagesink.c: No really, the next release is 0.10.21 (fix Since: tags in docs).
13455 Original commit message from CVS:
13456 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
13457 No really, the next release is 0.10.21 (fix Since: tags in docs).
13459 2008-09-04 16:25:06 +0000 Wim Taymans <wim.taymans@gmail.com>
13461 gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
13462 Original commit message from CVS:
13463 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
13464 Disable a code path that is now called but causes a deadlock for some
13465 reason and is unneeded.
13467 2008-09-04 13:46:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13469 sys/xvimage/xvimagesink.*: Add a "draw-border" property that can be set to false to disable drawing borders.
13470 Original commit message from CVS:
13471 * sys/xvimage/xvimagesink.c:
13472 * sys/xvimage/xvimagesink.h:
13473 Add a "draw-border" property that can be set to false to disable
13475 * tests/icles/test-colorkey.c:
13476 * tests/icles/Makefile.am:
13477 Add new test application for the colorkey handling.
13479 2008-09-03 14:00:06 +0000 Edward Hervey <bilboed@bilboed.com>
13481 gst-libs/gst/riff/riff-media.c: Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
13482 Original commit message from CVS:
13483 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
13484 Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
13485 This will also be fixed for upcoming gst-ffmpeg release so that once
13486 this release of -base is out, it will work with the latest gst-ffmpeg
13489 2008-09-03 13:27:20 +0000 Edward Hervey <bilboed@bilboed.com>
13491 gst-libs/gst/riff/riff-media.c: Add Truespeech mapping for RIFF formats (AVI/WAV).
13492 Original commit message from CVS:
13493 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
13494 (gst_riff_create_audio_template_caps):
13495 Add Truespeech mapping for RIFF formats (AVI/WAV).
13498 2008-09-03 12:23:44 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
13500 gst/typefind/gsttypefindfunctions.c: Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
13501 Original commit message from CVS:
13502 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
13503 Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
13506 2008-09-03 10:12:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13508 Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
13509 Original commit message from CVS:
13511 * gst/subparse/Makefile.am:
13512 * gst/subparse/gstsubparse.c:
13513 * gst/subparse/samiparse.c:
13514 * tests/check/elements/subparse.c:
13515 Rework last change, so that we build subparse, but just disable the
13516 sami parse functionality, if we're configured to not use xml. In the
13517 tests only the sami test is disabled now.
13519 2008-09-02 15:07:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13521 configure.ac: Disable subparse when xml is disabled. It woundn't work anyway. Fixes test runs.
13522 Original commit message from CVS:
13524 Disable subparse when xml is disabled. It woundn't work anyway. Fixes
13527 2008-09-02 09:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
13529 po/POTFILES.in: Add some more files with strings for translation.
13530 Original commit message from CVS:
13532 Add some more files with strings for translation.
13534 2008-09-02 06:37:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13536 Use new geo location tags from core. Fixes #481169
13537 Original commit message from CVS:
13538 * gst-libs/gst/tag/gstvorbistag.c:
13539 * tests/check/libs/tag.c:
13540 Use new geo location tags from core. Fixes #481169
13542 2008-09-01 16:05:45 +0000 Edward Hervey <bilboed@bilboed.com>
13544 tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
13545 Original commit message from CVS:
13546 * tests/check/elements/audioresample.c: (setup_audioresample),
13547 (fail_unless_perfect_stream), (test_perfect_stream_instance),
13548 (test_discont_stream_instance):
13549 Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
13550 Add debugging for coherence.
13552 2008-08-30 15:55:06 +0000 Jonathan Matthew <notverysmart@gmail.com>
13554 gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
13555 Original commit message from CVS:
13556 Patch by: Jonathan Matthew <notverysmart gmail com>
13557 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
13558 Add typefinder for PDF documents (which is nice to have, since it's a
13559 common format, but also helps prevent false positives). Fixes #549814.
13561 2008-08-27 15:30:16 +0000 Wim Taymans <wim.taymans@gmail.com>
13563 gst/playback/gstplaybin2.c: Fix nasty race where multiple decodebins could start pushing data before we manage to con...
13564 Original commit message from CVS:
13565 * gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
13567 Fix nasty race where multiple decodebins could start pushing data before
13568 we manage to configure the sinks, resulting in not-linked errors in
13569 typical RTSP streaming cases.
13571 2008-08-26 17:24:31 +0000 Wim Taymans <wim.taymans@gmail.com>
13573 gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
13574 Original commit message from CVS:
13575 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
13576 Since we now call stop, we trigger this code path that causes a deadlock
13577 is apparently not needed.
13579 2008-08-26 15:45:36 +0000 Wim Taymans <wim.taymans@gmail.com>
13581 gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
13582 Original commit message from CVS:
13583 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
13584 (gst_ring_buffer_stop):
13585 Also allow the case where the ringbuffer was paused when we try to stop
13586 it so that the basesrc stop function is still called.
13588 2008-08-23 15:25:44 +0000 Mike Ruprecht <cmaiku@gmail.com>
13590 sys/v4l/gstv4lelement.c: Reprobe devices again instead of taking a cached list as new devices could've been plugged i...
13591 Original commit message from CVS:
13592 Patch by: Mike Ruprecht <cmaiku at gmail dot com>
13593 * sys/v4l/gstv4lelement.c: (gst_v4l_class_probe_devices):
13594 Reprobe devices again instead of taking a cached list as new
13595 devices could've been plugged in. Fixes bug #549062.
13597 2008-08-23 15:19:59 +0000 Alessandro Dessina <alessandro@nnva.org>
13599 ext/ogg/gstoggdemux.c: Don't add pads and activate them for skeleton streams. These are already handled inside oggdem...
13600 Original commit message from CVS:
13601 Patch by: Alessandro Dessina <alessandro nnva org>
13602 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
13603 (gst_ogg_demux_activate_chain):
13604 Don't add pads and activate them for skeleton streams. These are already
13605 handled inside oggdemux. Fixes bug #537599.
13607 2008-08-22 15:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
13609 ext/vorbis/vorbisdec.c: Reset variable so that query and convert fail after going back to
13610 Original commit message from CVS:
13611 * ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
13612 Reset variable so that query and convert fail after going back to
13613 READY. Fixes #548898.
13615 2008-08-22 07:24:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13617 ext/vorbis/vorbisenc.c: If a buffer arrives with a timestamp before the timestamp+duration of the previous buffer cli...
13618 Original commit message from CVS:
13619 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
13620 If a buffer arrives with a timestamp before the timestamp+duration
13621 of the previous buffer clip it instead of dropping it completely.
13622 Slight improvement for the unfixable bug #548913.
13624 2008-08-21 14:19:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13626 ext/vorbis/vorbisdec.c: Take the current timestamp instead of timestamp+duration for the offset.
13627 Original commit message from CVS:
13628 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
13629 Take the current timestamp instead of timestamp+duration for the offset.
13630 This offset will later be used for calculating the timestamp and
13631 otherwise vorbisdec will interpolate timestamps wrong if upstream
13632 only sends timestamps and no granulepos.
13634 2008-08-21 11:20:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13636 tests/examples/seek/seek.c: Don't crash when having no visualisations.
13637 Original commit message from CVS:
13638 * tests/examples/seek/seek.c:
13639 Don't crash when having no visualisations.
13641 2008-08-16 20:57:27 +0000 David Schleef <ds@schleef.org>
13643 gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
13644 Original commit message from CVS:
13645 * gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
13646 check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
13649 2008-08-15 07:24:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13651 gst-libs/gst/pbutils/missing-plugins.c: When cleaning up the caps fields also remove "depth" for the same reason we r...
13652 Original commit message from CVS:
13653 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
13654 When cleaning up the caps fields also remove "depth" for the same
13655 reason we remove "width".
13657 2008-08-14 17:14:53 +0000 Tim-Philipp Müller <tim@centricular.net>
13659 gst-libs/gst/pbutils/descriptions.c: Add Lead H.264 here as well.
13660 Original commit message from CVS:
13661 * gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
13662 Add Lead H.264 here as well.
13664 2008-08-14 15:17:31 +0000 Julien Moutte <julien@moutte.net>
13666 gst-libs/gst/riff/riff-media.c: Add Lead H.264 variant.
13667 Original commit message from CVS:
13668 2008-08-14 Julien Moutte <julien@fluendo.com>
13669 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
13670 (gst_riff_create_video_template_caps): Add Lead H.264 variant.
13672 2008-08-13 09:17:38 +0000 Wim Taymans <wim.taymans@gmail.com>
13674 gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
13675 Original commit message from CVS:
13676 * gst-libs/gst/audio/gstbaseaudiosrc.c:
13677 (gst_base_audio_src_create):
13678 When not slaved to another clock also subtract the base_time from our
13679 internal clock time to get the running time.
13681 2008-08-13 00:59:07 +0000 David Schleef <ds@schleef.org>
13683 ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate, since it has no basis in libtheora.
13684 Original commit message from CVS:
13685 * ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
13686 since it has no basis in libtheora.
13688 2008-08-12 06:31:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13690 gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
13691 Original commit message from CVS:
13692 * gst-libs/gst/interfaces/propertyprobe.h:
13693 Remove double "interface" from doc-string.
13694 * gst-libs/gst/interfaces/xoverlay.h:
13695 Document interface.
13696 * gst-libs/gst/riff/riff.c:
13697 Add basic doc blobs.
13699 2008-08-11 15:05:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13701 gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
13702 Original commit message from CVS:
13703 * gst-libs/gst/audio/Makefile.am:
13704 Don't try to build that example anymore.
13706 2008-08-11 14:51:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13708 gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
13709 Original commit message from CVS:
13710 * gst-libs/gst/audio/.cvsignore:
13711 * gst-libs/gst/audio/Makefile.am:
13712 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
13713 * gst-libs/gst/audio/make_filter:
13714 Move audiofiltertemplate to gst-template.
13716 2008-08-11 09:20:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13718 More docs and shuffling. What can we do with the hundreds of #defines.
13719 Original commit message from CVS:
13720 * docs/libs/gst-plugins-base-libs-sections.txt:
13721 * gst-libs/gst/audio/gstaudiosrc.h:
13722 More docs and shuffling. What can we do with the hundreds of #defines.
13724 2008-08-11 08:34:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13726 gst-libs/gst/: Reducing number of dundocumented symbols.
13727 Original commit message from CVS:
13728 * gst-libs/gst/audio/audio.h:
13729 * gst-libs/gst/audio/gstaudiofilter.h:
13730 * gst-libs/gst/audio/gstringbuffer.h:
13731 * gst-libs/gst/interfaces/propertyprobe.h:
13732 * gst-libs/gst/tag/gsttagdemux.h:
13733 Reducing number of dundocumented symbols.
13735 2008-08-11 07:16:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13737 gst-libs/gst/audio/audio.c: Fix doc comment syntax.
13738 Original commit message from CVS:
13739 * gst-libs/gst/audio/audio.c:
13740 Fix doc comment syntax.
13741 * gst-libs/gst/interfaces/propertyprobe.c:
13742 Add more doc-comments and a FIXME: for the signal.
13744 2008-08-07 16:11:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13746 ext/ogg/gstoggmux.*: Don't pretend to support NEWSEGMENT events, instead override the
13747 Original commit message from CVS:
13748 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sink_event),
13749 (gst_ogg_mux_request_new_pad):
13750 * ext/ogg/gstoggmux.h:
13751 Don't pretend to support NEWSEGMENT events, instead override the
13752 GstCollectPads event function to return FALSE on NEWSEGMENT events
13753 and do the normal work for other events.
13754 This prevents elements like flacenc to seek to the start and rewrite
13755 some data which then results in a broken Ogg packet.
13757 2008-08-07 15:58:58 +0000 Frederic Crozat <fcrozat@mandriva.org>
13759 Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
13760 Original commit message from CVS:
13761 Patch by: Frederic Crozat <fcrozat@mandriva.org>
13762 * ext/alsa/gstalsaplugin.c: (plugin_init):
13763 * ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
13764 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
13765 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
13766 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
13767 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
13768 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
13769 * gst/playback/gstdecodebin.c: (plugin_init):
13770 * gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
13771 * gst/playback/gstplayback.c: (plugin_init):
13772 * gst/playback/gstqueue2.c: (plugin_init):
13773 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
13774 * sys/v4l/gstv4l.c: (plugin_init):
13775 Make sure gettext returns translations in UTF-8 encoding rather
13776 than in the current locale encoding (#546822).
13778 2008-08-06 13:12:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13780 gst-libs/gst/pbutils/descriptions.c: Add audio/x-qdm for qtdemux.
13781 Original commit message from CVS:
13782 * gst-libs/gst/pbutils/descriptions.c:
13783 Add audio/x-qdm for qtdemux.
13785 2008-08-05 15:38:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13787 ext/vorbis/vorbisdec.c: Do not leak old taglist.
13788 Original commit message from CVS:
13789 * ext/vorbis/vorbisdec.c:
13790 Do not leak old taglist.
13792 2008-08-04 12:35:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13794 tests/icles/test-scale.c: Include <stdlib.h> for atoi().
13795 Original commit message from CVS:
13796 * tests/icles/test-scale.c:
13797 Include <stdlib.h> for atoi().
13799 2008-08-04 09:11:08 +0000 Andy Wingo <wingo@pobox.com>
13801 gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
13802 Original commit message from CVS:
13803 2008-08-04 Andy Wingo <wingo@pobox.com>
13804 * gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
13807 2008-08-01 13:06:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13809 gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
13810 Original commit message from CVS:
13811 * gst/adder/gstadder.c:
13812 Cleanup lots of empty lines that came from gst-indent going havoc
13813 before I added the INDENT_ON/OFF marker some time agao.
13815 2008-08-01 11:55:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13817 Bump requirement to latest core and use new tag for riff formats.
13818 Original commit message from CVS:
13820 * gst-libs/gst/riff/riff-read.c:
13821 Bump requirement to latest core and use new tag for riff formats.
13822 Needed for #520694.
13824 2008-08-01 11:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
13826 tests/examples/dynamic/: Add example app that dynamically switches between 3 'encoders'.
13827 Original commit message from CVS:
13828 * tests/examples/dynamic/Makefile.am:
13829 * tests/examples/dynamic/codec-select.c: (make_encoder),
13830 (make_pipeline), (do_switch), (my_bus_callback), (main):
13831 Add example app that dynamically switches between 3 'encoders'.
13833 2008-07-31 13:06:13 +0000 Wim Taymans <wim.taymans@gmail.com>
13835 gst/playback/gstplaysink.c: Add some more comments.
13836 Original commit message from CVS:
13837 * gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
13838 Add some more comments.
13840 2008-07-31 12:58:44 +0000 Wim Taymans <wim.taymans@gmail.com>
13842 gst/videotestsrc/gstvideotestsrc.c: Discard buffers of the wrong size after renegotiation, this is perfectly possible...
13843 Original commit message from CVS:
13844 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
13845 (gst_video_test_src_create):
13846 Discard buffers of the wrong size after renegotiation, this is perfectly
13847 possible with things like capsfilter that could suggest caps changes
13848 upstream without knowing the size of the buffer.
13850 2008-07-31 11:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
13852 tests/icles/: Add dynamic rescaling tests for the new basetransform.
13853 Original commit message from CVS:
13854 * tests/icles/.cvsignore:
13855 * tests/icles/Makefile.am:
13856 * tests/icles/test-scale.c: (make_pipeline), (main):
13857 Add dynamic rescaling tests for the new basetransform.
13859 2008-07-30 19:51:36 +0000 Tim-Philipp Müller <tim@centricular.net>
13861 gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
13862 Original commit message from CVS:
13863 * gst/audioconvert/Makefile.am:
13864 Dist recently-added gstfastrandom.h.
13866 2008-07-30 15:29:44 +0000 Edward Hervey <bilboed@bilboed.com>
13868 sys/xvimage/xvimagesink.c: Fix a "may be used uninitialized in this function" which weirdly only appears on macosx (?).
13869 Original commit message from CVS:
13870 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
13871 Fix a "may be used uninitialized in this function" which weirdly only
13872 appears on macosx (?).
13874 2008-07-30 09:02:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13876 gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
13877 Original commit message from CVS:
13878 * gst-libs/gst/riff/riff-ids.h:
13879 Adding acid chunk for tempo and loop information.
13881 2008-07-29 13:01:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13883 sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
13884 Original commit message from CVS:
13885 * sys/xvimage/Makefile.am:
13886 floor() needs linking to $(LIBM).
13888 2008-07-29 12:35:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13890 ext/gnomevfs/gstgnomevfssrc.c: Aggregate short reads and add some comments and debug logging.
13891 Original commit message from CVS:
13892 * ext/gnomevfs/gstgnomevfssrc.c:
13893 Aggregate short reads and add some comments and debug logging.
13896 2008-07-29 10:26:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13898 gst/playback/gstplaybasebin.c: Fix property doc markup (its not a signal).
13899 Original commit message from CVS:
13900 * gst/playback/gstplaybasebin.c:
13901 Fix property doc markup (its not a signal).
13902 * sys/xvimage/xvimagesink.c:
13903 Add since tag for new proeprties (also add sice tags fro the last two
13906 2008-07-29 08:59:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13908 sys/xvimage/xvimagesink.*: Add autofill/colorkey properties. Fixes #538656.
13909 Original commit message from CVS:
13910 * sys/xvimage/xvimagesink.c:
13911 * sys/xvimage/xvimagesink.h:
13912 Add autofill/colorkey properties. Fixes #538656.
13914 2008-07-29 01:58:05 +0000 David Schleef <ds@schleef.org>
13916 sys/xvimage/xvimagesink.c: Fix rounding errors when converting colorbalance values between hardware and object proper...
13917 Original commit message from CVS:
13918 * sys/xvimage/xvimagesink.c:
13919 Fix rounding errors when converting colorbalance values
13920 between hardware and object property ranges. Partial
13921 fix for #537889, however, there still seems to be a small
13922 drift problem that could be totem's fault.
13924 2008-07-28 15:34:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13926 ext/ogg/gstoggdemux.c: Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
13927 Original commit message from CVS:
13928 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
13929 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
13930 Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
13931 This fixes a critical warning.
13933 2008-07-28 13:12:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13935 ext/ogg/gstoggmux.c: Allow muxing of CELT into Ogg streams.
13936 Original commit message from CVS:
13937 * ext/ogg/gstoggmux.c:
13938 Allow muxing of CELT into Ogg streams.
13940 2008-07-28 12:47:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13942 gst/typefind/gsttypefindfunctions.c: Add simple typefinder for the CELT codec (www.celt-codec.org).
13943 Original commit message from CVS:
13944 * gst/typefind/gsttypefindfunctions.c: (celt_type_find),
13946 Add simple typefinder for the CELT codec (www.celt-codec.org).
13948 2008-07-27 11:12:41 +0000 Jan Gerber <j@oil21.org>
13950 ext/ogg/gstoggdemux.c: Fix calculation of the start time from skeleton streams.
13951 Original commit message from CVS:
13952 Patch by: Jan Gerber <j at oil21 dot org>
13953 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
13954 Fix calculation of the start time from skeleton streams.
13957 2008-07-24 13:19:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13959 tests/examples/seek/seek.c: Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
13960 Original commit message from CVS:
13961 * tests/examples/seek/seek.c:
13962 Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
13964 2008-07-23 18:34:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13966 gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
13967 Original commit message from CVS:
13968 * gst/audioconvert/audioconvert.h:
13969 * gst/audioconvert/gstaudioquantize.c:
13970 (gst_audio_quantize_setup_dither),
13971 (gst_audio_quantize_free_dither):
13972 * gst/audioconvert/gstfastrandom.h:
13973 Implement a linear congruential generator as pseudo random number
13974 generator for the dither noise. This is about 2 times faster than
13975 using GLib's mersenne twister. Also this uses only integer math for
13976 generating integers while GLib internally uses floating point math.
13978 2008-07-23 18:27:15 +0000 Michael Smith <msmith@xiph.org>
13980 configure.ac: Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
13981 Original commit message from CVS:
13983 Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
13985 2008-07-23 13:17:31 +0000 Damien Lespiau <damien.lespiau@gmail.com>
13987 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
13988 Original commit message from CVS:
13989 Patch by: Damien Lespiau <damien.lespiau gmail com>
13990 * gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
13991 Use GST_STR_NULL to avoid crashes with libcs that don't
13992 like NULL strings in printf args (such as the win32 one).
13995 2008-07-17 14:21:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13997 sys/xvimage/xvimagesink.c: Oops - set the size of the image used for probing back to 1x1, for consistency with ximage...
13998 Original commit message from CVS:
13999 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
14000 Oops - set the size of the image used for probing back to 1x1, for
14001 consistency with ximagesink
14003 2008-07-17 13:57:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14005 sys/: it's not legal to ask the
14006 Original commit message from CVS:
14007 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
14008 (gst_ximagesink_ximage_new):
14009 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
14010 (gst_xvimagesink_xvimage_new):
14011 Apparently on Solaris and OS/X (at least), it's not legal to ask the
14012 X server to attach to a shared memory segment after we've deleted it,
14013 with the result that MIT-SHM is disabled. Instead, remove it only after
14014 X succeeds in attaching too.
14016 2008-07-17 02:30:24 +0000 David Schleef <ds@schleef.org>
14018 gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second.
14019 Original commit message from CVS:
14020 * gst/audiotestsrc/gstaudiotestsrc.c:
14021 * gst/audiotestsrc/gstaudiotestsrc.h:
14022 Add 'ticks', a 1/30 second sine wave pulse every second.
14024 2008-07-15 22:43:16 +0000 David Schleef <ds@schleef.org>
14026 gst-libs/gst/video/video.c: Revert ABI change.
14027 Original commit message from CVS:
14028 * gst-libs/gst/video/video.c: Revert ABI change.
14030 2008-07-15 13:05:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14032 gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
14033 Original commit message from CVS:
14034 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
14035 Make it impossible to have NULL caps at the point where we set
14036 framerate and other things. Also don't return immediately for "3ivd"
14037 video and let framerate, etc be set. Might fix bug #542508.
14039 2008-07-14 17:06:26 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
14041 gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
14042 Original commit message from CVS:
14043 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
14044 Video format can also be conveniently determined from (many)
14047 2008-07-14 08:18:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14049 gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
14050 Original commit message from CVS:
14051 * gst/playback/gstplaybasebin.c:
14052 * gst/playback/gstplaybasebin.h:
14053 * gst/playback/gstplaybin.c:
14054 * gst/playback/gststreamselector.c:
14055 First stab at integrating DVD subpicture overlay into
14056 playbin. Successfully plugs and plays, but the queues need
14057 shrinking - 3 seconds of video is too much buffering.
14059 2008-07-11 18:06:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14061 gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs.
14062 Original commit message from CVS:
14063 * gst/audioconvert/gstaudioconvert.c:
14064 Remove now obsolete note in the docs.
14066 2008-07-11 06:10:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14068 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
14069 Original commit message from CVS:
14070 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
14071 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
14072 * docs/plugins/gst-plugins-base-plugins-sections.txt:
14073 * docs/plugins/gst-plugins-base-plugins.args:
14074 * docs/plugins/gst-plugins-base-plugins.hierarchy:
14075 * docs/plugins/gst-plugins-base-plugins.interfaces:
14076 * docs/plugins/gst-plugins-base-plugins.prerequisites:
14077 * docs/plugins/gst-plugins-base-plugins.signals:
14078 * docs/plugins/inspect/plugin-adder.xml:
14079 * docs/plugins/inspect/plugin-alsa.xml:
14080 * docs/plugins/inspect/plugin-audioconvert.xml:
14081 * docs/plugins/inspect/plugin-audiorate.xml:
14082 * docs/plugins/inspect/plugin-audioresample.xml:
14083 * docs/plugins/inspect/plugin-audiotestsrc.xml:
14084 * docs/plugins/inspect/plugin-cdparanoia.xml:
14085 * docs/plugins/inspect/plugin-decodebin.xml:
14086 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
14087 * docs/plugins/inspect/plugin-gdp.xml:
14088 * docs/plugins/inspect/plugin-gnomevfs.xml:
14089 * docs/plugins/inspect/plugin-libvisual.xml:
14090 * docs/plugins/inspect/plugin-ogg.xml:
14091 * docs/plugins/inspect/plugin-pango.xml:
14092 * docs/plugins/inspect/plugin-playback.xml:
14093 * docs/plugins/inspect/plugin-queue2.xml:
14094 * docs/plugins/inspect/plugin-subparse.xml:
14095 * docs/plugins/inspect/plugin-tcp.xml:
14096 * docs/plugins/inspect/plugin-theora.xml:
14097 * docs/plugins/inspect/plugin-typefindfunctions.xml:
14098 * docs/plugins/inspect/plugin-uridecodebin.xml:
14099 * docs/plugins/inspect/plugin-video4linux.xml:
14100 * docs/plugins/inspect/plugin-videorate.xml:
14101 * docs/plugins/inspect/plugin-videoscale.xml:
14102 * docs/plugins/inspect/plugin-videotestsrc.xml:
14103 * docs/plugins/inspect/plugin-volume.xml:
14104 * docs/plugins/inspect/plugin-vorbis.xml:
14105 * docs/plugins/inspect/plugin-ximagesink.xml:
14106 * docs/plugins/inspect/plugin-xvimagesink.xml:
14107 * ext/alsa/gstalsamixer.c:
14108 * ext/alsa/gstalsasink.c:
14109 * ext/alsa/gstalsasrc.c:
14110 * ext/gio/gstgiosink.c:
14111 * ext/gio/gstgiosrc.c:
14112 * ext/gio/gstgiostreamsink.c:
14113 * ext/gio/gstgiostreamsrc.c:
14114 * ext/gnomevfs/gstgnomevfssink.c:
14115 * ext/gnomevfs/gstgnomevfssrc.c:
14116 * ext/ogg/gstoggdemux.c:
14117 * ext/ogg/gstoggmux.c:
14118 * ext/pango/gstclockoverlay.c:
14119 * ext/pango/gsttextoverlay.c:
14120 * ext/pango/gsttextrender.c:
14121 * ext/pango/gsttimeoverlay.c:
14122 * ext/theora/theoradec.c:
14123 * ext/theora/theoraenc.c:
14124 * ext/theora/theoraparse.c:
14125 * ext/vorbis/vorbisdec.c:
14126 * ext/vorbis/vorbisenc.c:
14127 * ext/vorbis/vorbisparse.c:
14128 * ext/vorbis/vorbistag.c:
14129 * gst/adder/gstadder.c:
14130 * gst/audioconvert/gstaudioconvert.c:
14131 * gst/audioresample/gstaudioresample.c:
14132 * gst/audiotestsrc/gstaudiotestsrc.c:
14133 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
14134 * gst/gdp/gstgdpdepay.c:
14135 * gst/gdp/gstgdppay.c:
14136 * gst/playback/gstdecodebin2.c:
14137 * gst/playback/gstplaybin.c:
14138 * gst/playback/gstplaybin2.c:
14139 * gst/playback/gstqueue2.c:
14140 * gst/playback/gsturidecodebin.c:
14141 * gst/tcp/gstmultifdsink.c:
14142 * gst/tcp/gsttcpserversink.c:
14143 * gst/videorate/gstvideorate.c:
14144 * gst/videoscale/gstvideoscale.c:
14145 * gst/videotestsrc/gstvideotestsrc.c:
14146 * gst/volume/gstvolume.c:
14147 * sys/ximage/ximagesink.c:
14148 * sys/xvimage/xvimagesink.c:
14149 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
14150 titles. Drop mentining that all our example pipelines are "simple"
14153 2008-07-10 21:06:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14155 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
14156 Original commit message from CVS:
14157 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
14158 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
14159 * docs/plugins/gst-plugins-base-plugins-sections.txt:
14160 * docs/plugins/gst-plugins-base-plugins.args:
14161 * docs/plugins/gst-plugins-base-plugins.hierarchy:
14162 * docs/plugins/gst-plugins-base-plugins.interfaces:
14163 * docs/plugins/gst-plugins-base-plugins.prerequisites:
14164 * docs/plugins/gst-plugins-base-plugins.signals:
14165 * docs/plugins/inspect/plugin-adder.xml:
14166 * docs/plugins/inspect/plugin-alsa.xml:
14167 * docs/plugins/inspect/plugin-audioconvert.xml:
14168 * docs/plugins/inspect/plugin-audiorate.xml:
14169 * docs/plugins/inspect/plugin-audioresample.xml:
14170 * docs/plugins/inspect/plugin-audiotestsrc.xml:
14171 * docs/plugins/inspect/plugin-cdparanoia.xml:
14172 * docs/plugins/inspect/plugin-decodebin.xml:
14173 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
14174 * docs/plugins/inspect/plugin-gdp.xml:
14175 * docs/plugins/inspect/plugin-gnomevfs.xml:
14176 * docs/plugins/inspect/plugin-libvisual.xml:
14177 * docs/plugins/inspect/plugin-ogg.xml:
14178 * docs/plugins/inspect/plugin-pango.xml:
14179 * docs/plugins/inspect/plugin-playback.xml:
14180 * docs/plugins/inspect/plugin-queue2.xml:
14181 * docs/plugins/inspect/plugin-subparse.xml:
14182 * docs/plugins/inspect/plugin-tcp.xml:
14183 * docs/plugins/inspect/plugin-theora.xml:
14184 * docs/plugins/inspect/plugin-typefindfunctions.xml:
14185 * docs/plugins/inspect/plugin-uridecodebin.xml:
14186 * docs/plugins/inspect/plugin-video4linux.xml:
14187 * docs/plugins/inspect/plugin-videorate.xml:
14188 * docs/plugins/inspect/plugin-videoscale.xml:
14189 * docs/plugins/inspect/plugin-videotestsrc.xml:
14190 * docs/plugins/inspect/plugin-volume.xml:
14191 * docs/plugins/inspect/plugin-vorbis.xml:
14192 * docs/plugins/inspect/plugin-ximagesink.xml:
14193 * docs/plugins/inspect/plugin-xvimagesink.xml:
14194 * ext/alsa/gstalsamixer.c:
14195 * ext/alsa/gstalsasink.c:
14196 * ext/alsa/gstalsasrc.c:
14197 * ext/gio/gstgiosink.c:
14198 * ext/gio/gstgiosrc.c:
14199 * ext/gio/gstgiostreamsink.c:
14200 * ext/gio/gstgiostreamsrc.c:
14201 * ext/gnomevfs/gstgnomevfssink.c:
14202 * ext/gnomevfs/gstgnomevfssrc.c:
14203 * ext/ogg/gstoggdemux.c:
14204 * ext/ogg/gstoggmux.c:
14205 * ext/pango/gstclockoverlay.c:
14206 * ext/pango/gsttextoverlay.c:
14207 * ext/pango/gsttextrender.c:
14208 * ext/pango/gsttimeoverlay.c:
14209 * ext/theora/theoradec.c:
14210 * ext/theora/theoraenc.c:
14211 * ext/theora/theoraparse.c:
14212 * ext/vorbis/vorbisdec.c:
14213 * ext/vorbis/vorbisenc.c:
14214 * ext/vorbis/vorbisparse.c:
14215 * ext/vorbis/vorbistag.c:
14216 * gst/adder/gstadder.c:
14217 * gst/audioconvert/gstaudioconvert.c:
14218 * gst/audioresample/gstaudioresample.c:
14219 * gst/audiotestsrc/gstaudiotestsrc.c:
14220 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
14221 * gst/gdp/gstgdpdepay.c:
14222 * gst/gdp/gstgdppay.c:
14223 * gst/playback/gstdecodebin2.c:
14224 * gst/playback/gstplaybin.c:
14225 * gst/playback/gstplaybin2.c:
14226 * gst/playback/gstqueue2.c:
14227 * gst/playback/gsturidecodebin.c:
14228 * gst/tcp/gstmultifdsink.c:
14229 * gst/tcp/gsttcpserversink.c:
14230 * gst/videorate/gstvideorate.c:
14231 * gst/videoscale/gstvideoscale.c:
14232 * gst/videotestsrc/gstvideotestsrc.c:
14233 * gst/volume/gstvolume.c:
14234 * sys/ximage/ximagesink.c:
14235 * sys/xvimage/xvimagesink.c:
14236 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
14237 titles. Drop mentining that all our example pipelines are "simple"
14240 2008-07-07 17:25:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14242 tests/examples/seek/Makefile.am: Fix out of tree build by adding all required CFLAGS.
14243 Original commit message from CVS:
14244 * tests/examples/seek/Makefile.am:
14245 Fix out of tree build by adding all required CFLAGS.
14247 2008-07-07 09:55:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14249 gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
14250 Original commit message from CVS:
14251 * gst/playback/gstdecodebin.c: (add_raw_queue):
14252 And ref the pad before returning it again when linking to the queue
14253 failed. Otherwise we will unref the pad twice later and things break.
14255 2008-07-07 09:48:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14257 gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
14258 Original commit message from CVS:
14259 * gst/playback/gstdecodebin.c: (add_raw_queue):
14260 If linking the raw pad with a queue fails, try it without a queue
14261 instead of failing completely. This should never happen.
14263 2008-07-06 23:22:12 +0000 Evgeniy Stepanov <eugeni.stepanov@gmail.com>
14265 gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
14266 Original commit message from CVS:
14267 Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
14268 * gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
14269 Add a queue after a demuxer if the demuxer outputs raw data. This was
14270 done before only for non-raw data but is required in this case too.
14272 decodebin2 doesn't have this issue because all streams of a group
14273 go through multiqueue.
14275 2008-07-03 09:12:49 +0000 Damien Lespiau <damien.lespiau@gmail.com>
14277 gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
14278 Original commit message from CVS:
14279 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
14280 * gst-libs/gst/sdp/gstsdpmessage.c:
14281 Makes libgstsdp compile with mingw32 by defining the right WINVER so
14282 that getaddrinfo() can be used. Fixes #541358.
14284 2008-07-01 13:22:49 +0000 Wim Taymans <wim.taymans@gmail.com>
14286 gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
14287 Original commit message from CVS:
14288 * gst/videotestsrc/gstvideotestsrc.c:
14289 (gst_video_test_src_class_init), (gst_video_test_src_init),
14290 (gst_video_test_src_set_property),
14291 (gst_video_test_src_get_property), (gst_video_test_src_create):
14292 * gst/videotestsrc/gstvideotestsrc.h:
14293 Cleanups, use default property values as defines.
14294 Add property to enable/disable peer buffer allocation.
14296 2008-06-30 09:46:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14298 tests/check/: Enable unit tests on PPC again as the bugs are now fixed.
14299 Original commit message from CVS:
14300 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
14301 * tests/check/pipelines/streamheader.c: (streamheader_suite):
14302 Enable unit tests on PPC again as the bugs are now fixed.
14304 2008-06-30 09:20:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14306 gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
14307 Original commit message from CVS:
14308 * gst-libs/gst/riff/riff-ids.h:
14309 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
14310 (gst_riff_create_audio_template_caps):
14311 Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
14314 2008-06-30 08:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14316 gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
14317 Original commit message from CVS:
14318 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
14319 (gst_ffmpeg_pixfmt_to_caps):
14320 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
14321 (gst_ffmpegcsp_get_unit_size):
14322 Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
14323 it on other formats. Also adjust the unit size only for that format
14324 to not include the palette. Fixes bug #540497.
14326 2008-06-29 13:45:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14328 gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
14329 Original commit message from CVS:
14330 * gst/adder/gstadder.c:
14331 Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
14333 2008-06-27 07:55:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14335 ChangeLog: ChangeLog surgery.
14336 Original commit message from CVS:
14339 * tests/examples/seek/seek.c:
14340 Move variable into ifdef too.
14342 2008-06-27 07:42:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14344 tests/examples/seek/seek.c: Include config.h and check if we have X. Fixes: #540334.
14345 Original commit message from CVS:
14346 * tests/examples/seek/seek.c:
14347 Include config.h and check if we have X. Fixes: #540334.
14349 2008-06-26 06:03:38 +0000 Sam Morris <sam@robots.org.to.uk>
14351 gst-libs/gst/interfaces/mixertrack.c: API: Add "index" property to GstMixerTrack to differantiate between multiple mi...
14352 Original commit message from CVS:
14353 Patch by: Sam Morris <sam at robots dot org to uk>
14354 * gst-libs/gst/interfaces/mixertrack.c:
14355 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
14356 (gst_mixer_track_set_property):
14357 API: Add "index" property to GstMixerTrack to differantiate between
14358 multiple mixer tracks with the same label.
14359 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
14360 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
14361 Set the "index" property of GstMixerTrack to the index given by ALSA.
14364 2008-06-25 13:15:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14366 tests/examples/seek/: Remove libgstvideo usage. Use gtk_get_option_group instead of gtk_init().
14367 Original commit message from CVS:
14368 * tests/examples/seek/Makefile.am:
14369 * tests/examples/seek/seek.c:
14370 Remove libgstvideo usage. Use gtk_get_option_group instead of
14373 2008-06-24 16:27:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14375 tests/check/Makefile.am: Name the test registry format neutral.
14376 Original commit message from CVS:
14377 * tests/check/Makefile.am:
14378 Name the test registry format neutral.
14380 2008-06-24 16:22:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14382 gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
14383 Original commit message from CVS:
14384 * gst/playback/gstqueue2.c:
14385 Do not double notify. Remove the unsued return value.
14387 2008-06-24 16:15:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14389 ext/alsa/gstalsamixer.c: Also consider "speaker" as a name for master volume. If that doesn't help look for the first...
14390 Original commit message from CVS:
14391 * ext/alsa/gstalsamixer.c:
14392 Also consider "speaker" as a name for master volume. If that doesn't
14393 help look for the first non-mono volume control that also has a
14396 2008-06-24 16:10:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14398 ChangeLog: Forgot to save the ChangeLog :/
14399 Original commit message from CVS:
14401 Forgot to save the ChangeLog :/
14403 2008-06-24 16:05:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14405 tests/examples/seek/: Embedd the xwindow.
14406 Original commit message from CVS:
14407 * tests/examples/seek/Makefile.am:
14408 * tests/examples/seek/seek.c:
14409 Embedd the xwindow.
14411 2008-06-24 01:14:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14413 sys/ximage/ximagesink.h: When the caps change, make sure to re-draw borders in force-aspect-ratio=true mode.
14414 Original commit message from CVS:
14415 * sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
14416 (gst_ximagesink_setcaps):
14417 * sys/ximage/ximagesink.h:
14418 When the caps change, make sure to re-draw borders in
14419 force-aspect-ratio=true mode.
14420 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
14421 Don't clear the border_draw flag until we actually draw the border.
14422 * tests/check/Makefile.am:
14423 Ignore alsasink/src during the states test too, so it doesn't fail
14424 when running without access to the sound device.
14426 2008-06-22 18:35:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14428 tests/examples/seek/seek.c: Fix crasher when playing a parse-launch line the 2nd time.
14429 Original commit message from CVS:
14430 * tests/examples/seek/seek.c:
14431 Fix crasher when playing a parse-launch line the 2nd time.
14433 2008-06-21 18:56:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
14435 tests/check/pipelines/oggmux.c: Properly ifdef tests to fix compilation.
14436 Original commit message from CVS:
14437 * tests/check/pipelines/oggmux.c:
14438 Properly ifdef tests to fix compilation.
14440 2008-06-21 10:25:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
14444 Original commit message from CVS:
14447 2008-06-20 18:24:24 +0000 Michael Smith <msmith@xiph.org>
14449 gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
14450 Original commit message from CVS:
14451 * gst/playback/gstplay-marshal.list:
14452 * gst/playback/gstplaybin2.c:
14453 Add get-video-pad, get-audio-pad, get-text-pad action signals to
14454 playbin2. This allows the user to get to the selector's sinkpads, and
14455 thus inspect a range of things - caps, tags, etc.
14457 2008-06-20 17:27:03 +0000 Michael Smith <msmith@xiph.org>
14459 gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
14460 Original commit message from CVS:
14461 * gst/playback/gstplaybin2.c:
14462 Use a different constant for the convert-frame signal id.
14465 2008-06-20 17:18:55 +0000 Michael Smith <msmith@xiph.org>
14467 gst/playback/: Fix a whole bunch of typos in comments and log statements.
14468 Original commit message from CVS:
14469 * gst/playback/gstplaybin2.c:
14470 * gst/playback/gstplaysink.c:
14471 Fix a whole bunch of typos in comments and log statements.
14473 2008-06-20 17:02:48 +0000 Michael Smith <msmith@xiph.org>
14475 sys/xvimage/xvimagesink.c: Don't set colour balance values on the Xv port if the user hasn't changed them (via proper...
14476 Original commit message from CVS:
14477 * sys/xvimage/xvimagesink.c:
14478 Don't set colour balance values on the Xv port if the user hasn't
14479 changed them (via properties or the interface). Avoids accumulating
14480 rounding errors for the common case.
14481 Partial fix for bug #537889.
14483 2008-06-20 16:56:18 +0000 Michael Smith <msmith@xiph.org>
14485 gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
14486 Original commit message from CVS:
14487 * gst/playback/gstdecodebin2.c:
14488 Ensure decodebin2 emits 'drained' signal once, and only once, when all
14491 2008-06-20 16:12:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
14494 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...
14495 Original commit message from CVS:
14496 apparently it's an error to specify nc -l -p 3000 - though the short usage
14497 does not make it very clear that you can drop the host arg with -l
14499 2008-06-20 09:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
14501 ext/vorbis/vorbisenc.c: Report the encoder latency. Fixes #538232.
14502 Original commit message from CVS:
14503 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
14504 (gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
14505 Report the encoder latency. Fixes #538232.
14507 2008-06-20 09:19:59 +0000 Wim Taymans <wim.taymans@gmail.com>
14509 gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
14510 Original commit message from CVS:
14511 * gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
14512 (notify_source), (activate_group):
14513 Implement the source property, emit notify when it changes in the
14514 underlying uridecodebin.
14516 2008-06-20 09:14:26 +0000 Wim Taymans <wim.taymans@gmail.com>
14518 tests/examples/seek/seek.c: Free and clear the seek element list so that we don't use invalid references when seeking...
14519 Original commit message from CVS:
14520 * tests/examples/seek/seek.c: (stop_cb):
14521 Free and clear the seek element list so that we don't use invalid
14522 references when seeking after recreating a gst-launch line.
14524 2008-06-20 09:09:37 +0000 Wim Taymans <wim.taymans@gmail.com>
14526 gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
14527 Original commit message from CVS:
14528 * gst-libs/gst/audio/gstbaseaudiosink.c:
14529 (gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
14530 (gst_base_audio_sink_render):
14531 Report latency even if we are not live instead of hiding it.
14532 Take ts-offset and render-delay of the basesink into account when
14533 scheduling samples.
14534 Rework the clipping code so that we can take the various offsets into
14535 account and still do correct clipping.
14537 2008-06-20 08:52:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14539 configure.ac: Bump verion back to devel -> 0.10.20.1
14540 Original commit message from CVS:
14542 Bump verion back to devel -> 0.10.20.1
14544 2008-06-20 08:47:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14546 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...
14547 Original commit message from CVS:
14548 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
14549 Don't increase the size of non-string image buffers by one as this
14550 might in theory confuse decoders. Still increase it by one for string
14551 image buffers to append '\0'.
14553 2008-06-20 08:45:13 +0000 Antoine Tremblay <hexa00@gmail.com>
14555 gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
14556 Original commit message from CVS:
14557 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
14558 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
14559 Fix a buffer memleak and remove a confusing and wrong debug output.
14562 2008-06-19 11:25:37 +0000 Wim Taymans <wim.taymans@gmail.com>
14564 examples/app/appsink-src.c: Don't use a buffer after unreffing it.
14565 Original commit message from CVS:
14566 * examples/app/appsink-src.c: (on_new_buffer_from_source):
14567 Don't use a buffer after unreffing it.
14569 === release 0.10.20 ===
14571 2008-06-18 14:36:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14577 * docs/plugins/gst-plugins-base-plugins.args:
14578 * docs/plugins/gst-plugins-base-plugins.hierarchy:
14579 * docs/plugins/gst-plugins-base-plugins.interfaces:
14580 * docs/plugins/gst-plugins-base-plugins.prerequisites:
14581 * docs/plugins/inspect/plugin-adder.xml:
14582 * docs/plugins/inspect/plugin-alsa.xml:
14583 * docs/plugins/inspect/plugin-audioconvert.xml:
14584 * docs/plugins/inspect/plugin-audiorate.xml:
14585 * docs/plugins/inspect/plugin-audioresample.xml:
14586 * docs/plugins/inspect/plugin-audiotestsrc.xml:
14587 * docs/plugins/inspect/plugin-cdparanoia.xml:
14588 * docs/plugins/inspect/plugin-decodebin.xml:
14589 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
14590 * docs/plugins/inspect/plugin-gdp.xml:
14591 * docs/plugins/inspect/plugin-gnomevfs.xml:
14592 * docs/plugins/inspect/plugin-libvisual.xml:
14593 * docs/plugins/inspect/plugin-ogg.xml:
14594 * docs/plugins/inspect/plugin-pango.xml:
14595 * docs/plugins/inspect/plugin-playback.xml:
14596 * docs/plugins/inspect/plugin-queue2.xml:
14597 * docs/plugins/inspect/plugin-subparse.xml:
14598 * docs/plugins/inspect/plugin-tcp.xml:
14599 * docs/plugins/inspect/plugin-theora.xml:
14600 * docs/plugins/inspect/plugin-typefindfunctions.xml:
14601 * docs/plugins/inspect/plugin-uridecodebin.xml:
14602 * docs/plugins/inspect/plugin-video4linux.xml:
14603 * docs/plugins/inspect/plugin-videorate.xml:
14604 * docs/plugins/inspect/plugin-videoscale.xml:
14605 * docs/plugins/inspect/plugin-videotestsrc.xml:
14606 * docs/plugins/inspect/plugin-volume.xml:
14607 * docs/plugins/inspect/plugin-vorbis.xml:
14608 * docs/plugins/inspect/plugin-ximagesink.xml:
14609 * docs/plugins/inspect/plugin-xvimagesink.xml:
14610 * gst-plugins-base.doap:
14612 * win32/common/config.h:
14614 Original commit message from CVS:
14617 2008-06-18 14:32:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14646 Original commit message from CVS:
14649 2008-06-18 06:31:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14651 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
14652 Original commit message from CVS:
14653 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
14654 * examples/app/appsrc-ra.c:
14655 * examples/app/appsrc-seekable.c:
14656 * examples/app/appsrc-stream.c:
14657 * examples/app/appsrc-stream2.c:
14658 * ext/directfb/dfbvideosink.h:
14659 * ext/metadata/gstbasemetadata.c:
14660 * ext/metadata/gstbasemetadata.h:
14661 * ext/metadata/metadata.c:
14662 * ext/metadata/metadataexif.c:
14663 * ext/theora/theoradec.h:
14664 * gst/deinterlace2/gstdeinterlace2.h:
14665 * gst/deinterlace2/tvtime/speedy.c:
14666 * gst/deinterlace2/tvtime/speedy.h:
14667 * gst/deinterlace2/tvtime/vfir.c:
14668 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
14671 2008-06-16 14:11:36 +0000 Andy Wingo <wingo@pobox.com>
14673 * gst-libs/gst/app/gstappsrc.c:
14674 gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
14675 Original commit message from CVS:
14676 2008-06-16 Andy Wingo <wingo@pobox.com>
14677 * gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
14678 (gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
14679 G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
14681 2008-06-16 07:30:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14683 Final round of doc updates.
14684 Original commit message from CVS:
14685 * gst/rtpmanager/gstrtpjitterbuffer.c:
14686 * gst/speed/gstspeed.c:
14687 * gst/speexresample/gstspeexresample.c:
14688 * gst/videosignal/gstvideoanalyse.c:
14689 * gst/videosignal/gstvideodetect.c:
14690 * gst/videosignal/gstvideomark.c:
14691 * sys/dvb/gstdvbsrc.c:
14692 * sys/oss4/oss4-mixer.c:
14693 * sys/oss4/oss4-sink.c:
14694 * sys/oss4/oss4-source.c:
14695 * sys/wininet/gstwininetsrc.c:
14696 Final round of doc updates.
14698 2008-06-13 11:59:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14700 docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
14701 Original commit message from CVS:
14702 * docs/plugins/Makefile.am:
14703 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
14704 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
14705 * docs/plugins/gst-plugins-bad-plugins.args:
14706 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
14707 * docs/plugins/gst-plugins-bad-plugins.interfaces:
14708 * docs/plugins/gst-plugins-bad-plugins.prerequisites:
14709 * docs/plugins/gst-plugins-bad-plugins.signals:
14710 * docs/plugins/inspect/plugin-alsaspdif.xml:
14711 * docs/plugins/inspect/plugin-amrwb.xml:
14712 * docs/plugins/inspect/plugin-app.xml:
14713 * docs/plugins/inspect/plugin-bayer.xml:
14714 * docs/plugins/inspect/plugin-bz2.xml:
14715 * docs/plugins/inspect/plugin-cdaudio.xml:
14716 * docs/plugins/inspect/plugin-cdxaparse.xml:
14717 * docs/plugins/inspect/plugin-dtsdec.xml:
14718 * docs/plugins/inspect/plugin-dvb.xml:
14719 * docs/plugins/inspect/plugin-dvdspu.xml:
14720 * docs/plugins/inspect/plugin-faac.xml:
14721 * docs/plugins/inspect/plugin-faad.xml:
14722 * docs/plugins/inspect/plugin-fbdevsink.xml:
14723 * docs/plugins/inspect/plugin-festival.xml:
14724 * docs/plugins/inspect/plugin-filter.xml:
14725 * docs/plugins/inspect/plugin-flvdemux.xml:
14726 * docs/plugins/inspect/plugin-freeze.xml:
14727 * docs/plugins/inspect/plugin-gsm.xml:
14728 * docs/plugins/inspect/plugin-gstinterlace.xml:
14729 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
14730 * docs/plugins/inspect/plugin-h264parse.xml:
14731 * docs/plugins/inspect/plugin-interleave.xml:
14732 * docs/plugins/inspect/plugin-jack.xml:
14733 * docs/plugins/inspect/plugin-ladspa.xml:
14734 * docs/plugins/inspect/plugin-metadata.xml:
14735 * docs/plugins/inspect/plugin-mms.xml:
14736 * docs/plugins/inspect/plugin-modplug.xml:
14737 * docs/plugins/inspect/plugin-mpeg2enc.xml:
14738 * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
14739 * docs/plugins/inspect/plugin-mpegtsparse.xml:
14740 * docs/plugins/inspect/plugin-mpegvideoparse.xml:
14741 * docs/plugins/inspect/plugin-musepack.xml:
14742 * docs/plugins/inspect/plugin-musicbrainz.xml:
14743 * docs/plugins/inspect/plugin-mve.xml:
14744 * docs/plugins/inspect/plugin-mythtv.xml
14745 * docs/plugins/inspect/plugin-nas.xml:
14746 * docs/plugins/inspect/plugin-neon.xml:
14747 * docs/plugins/inspect/plugin-nsfdec.xml:
14748 * docs/plugins/inspect/plugin-nuvdemux.xml:
14749 * docs/plugins/inspect/plugin-oss4.xml
14750 * docs/plugins/inspect/plugin-rawparse.xml:
14751 * docs/plugins/inspect/plugin-real.xml:
14752 * docs/plugins/inspect/plugin-replaygain.xml:
14753 * docs/plugins/inspect/plugin-rfbsrc.xml:
14754 * docs/plugins/inspect/plugin-sdl.xml:
14755 * docs/plugins/inspect/plugin-sdp.xml:
14756 * docs/plugins/inspect/plugin-selector.xml:
14757 * docs/plugins/inspect/plugin-sndfile.xml:
14758 * docs/plugins/inspect/plugin-soundtouch.xml:
14759 * docs/plugins/inspect/plugin-spcdec.xml:
14760 * docs/plugins/inspect/plugin-speed.xml:
14761 * docs/plugins/inspect/plugin-speexresample.xml:
14762 * docs/plugins/inspect/plugin-stereo.xml:
14763 * docs/plugins/inspect/plugin-subenc.xml
14764 * docs/plugins/inspect/plugin-timidity.xml:
14765 * docs/plugins/inspect/plugin-tta.xml:
14766 * docs/plugins/inspect/plugin-vcdsrc.xml:
14767 * docs/plugins/inspect/plugin-videosignal.xml:
14768 * docs/plugins/inspect/plugin-vmnc.xml:
14769 * docs/plugins/inspect/plugin-wildmidi.xml:
14770 * docs/plugins/inspect/plugin-x264.xml:
14771 * docs/plugins/inspect/plugin-xvid.xml:
14772 * docs/plugins/inspect/plugin-y4menc.xml:
14773 * ext/amrwb/gstamrwbdec.c:
14774 * ext/amrwb/gstamrwbenc.c:
14775 * ext/amrwb/gstamrwbparse.c:
14776 * ext/dc1394/gstdc1394.c:
14777 * ext/directfb/dfbvideosink.c:
14778 * ext/ivorbis/vorbisdec.c:
14779 * ext/jack/gstjackaudiosink.c:
14780 * ext/mpeg2enc/gstmpeg2enc.cc:
14781 * ext/mplex/gstmplex.cc:
14782 * ext/musicbrainz/gsttrm.c:
14783 * ext/mythtv/gstmythtvsrc.c:
14784 * ext/theora/theoradec.c:
14785 * ext/timidity/gsttimidity.c:
14786 * ext/timidity/gstwildmidi.c:
14787 * gst-libs/gst/app/gstappsink.c:
14788 * gst/deinterlace/gstdeinterlace.c:
14789 * gst/dvdspu/gstdvdspu.c:
14790 * gst/festival/gstfestival.c:
14791 * gst/freeze/gstfreeze.c:
14792 * gst/interleave/deinterleave.c:
14793 * gst/interleave/interleave.c:
14794 * gst/modplug/gstmodplug.cc:
14795 * gst/nuvdemux/gstnuvdemux.c:
14796 Add missing elements to docs. Fix doc-markup: use convinience syntax
14797 for examples (produces valid docbook), add several refsec2 when we
14798 have several titles. Fix some types.
14800 2008-06-12 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
14802 examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
14803 Original commit message from CVS:
14804 * examples/app/.cvsignore:
14805 * examples/app/Makefile.am:
14806 * examples/app/appsink-src.c: (on_new_buffer_from_source),
14807 (on_source_message), (on_sink_message), (main):
14808 Add beefed up example app from bug #413418. It now also uses appsink
14809 instead of fakesink for more ultimate coolness.
14810 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
14811 (gst_app_src_init), (gst_app_src_set_property),
14812 (gst_app_src_get_property), (gst_app_src_unlock),
14813 (gst_app_src_unlock_stop), (gst_app_src_create),
14814 (gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
14815 (gst_app_src_end_of_stream):
14816 * gst-libs/gst/app/gstappsrc.h:
14817 Add block property to allow push based implementation to block when we
14818 fill up the appsrc queues.
14819 Emit the enough-data signal while releasing our lock.
14821 2008-06-12 14:50:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14823 examples/app/.cvsignore: Ignore more.
14824 Original commit message from CVS:
14825 * examples/app/.cvsignore:
14828 2008-06-12 14:49:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14830 Do not use short_description in section docs for elements. We extract them from element details and there will be war...
14831 Original commit message from CVS:
14832 * ext/dc1394/gstdc1394.c:
14833 * ext/ivorbis/vorbisdec.c:
14834 * ext/jack/gstjackaudiosink.c:
14835 * ext/metadata/gstmetadatademux.c:
14836 * ext/mythtv/gstmythtvsrc.c:
14837 * ext/theora/theoradec.c:
14838 * gst-libs/gst/app/gstappsink.c:
14839 * gst/bayer/gstbayer2rgb.c:
14840 * gst/deinterlace/gstdeinterlace.c:
14841 * gst/rawparse/gstaudioparse.c:
14842 * gst/rawparse/gstvideoparse.c:
14843 * gst/rtpmanager/gstrtpbin.c:
14844 * gst/rtpmanager/gstrtpclient.c:
14845 * gst/rtpmanager/gstrtpjitterbuffer.c:
14846 * gst/rtpmanager/gstrtpptdemux.c:
14847 * gst/rtpmanager/gstrtpsession.c:
14848 * gst/rtpmanager/gstrtpssrcdemux.c:
14849 * gst/selector/gstinputselector.c:
14850 * gst/selector/gstoutputselector.c:
14851 * gst/videosignal/gstvideoanalyse.c:
14852 * gst/videosignal/gstvideodetect.c:
14853 * gst/videosignal/gstvideomark.c:
14854 * sys/oss4/oss4-mixer.c:
14855 * sys/oss4/oss4-sink.c:
14856 * sys/oss4/oss4-source.c:
14857 Do not use short_description in section docs for elements. We extract
14858 them from element details and there will be warnings if they differ.
14859 Also fixing up the ChangeLog order.
14861 2008-06-11 21:17:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14863 configure.ac: 0.10.19.3 pre-release
14864 Original commit message from CVS:
14866 0.10.19.3 pre-release
14868 2008-06-11 20:13:00 +0000 David Schleef <ds@schleef.org>
14870 gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
14871 Original commit message from CVS:
14872 * gst-libs/gst/rtsp/gstrtspconnection.c:
14873 Fix build on win32.
14874 Patch By: David Schleef <ds@schleef.org>
14877 2008-06-11 09:35:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14879 ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
14880 Original commit message from CVS:
14881 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
14882 (gst_gio_base_src_create):
14883 * ext/gio/gstgiobasesrc.h:
14884 Try to read the requested number of bytes, even if the first
14885 read returns less than requested, until nothing is read anymore
14886 or we have the requested amount of bytes. This fixes playback of
14887 files via Samba as Samba only allows to read 64k at once.
14888 Implement a caching algorithm that makes sure that we read at
14889 least 4k of data every time. Some elements will try to read a few
14890 bytes, then seek, read again a few bytes and so on and this is
14891 painfully slow as every operation has to go over DBus if GVfs is
14893 Fixes bug #536849 and #536848.
14894 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
14895 (gst_gio_src_check_get_range):
14896 Override check_get_range() to blacklist http/https URIs
14897 and whitelist file URIs. More to be added on demand.
14899 2008-06-06 16:50:51 +0000 Wim Taymans <wim.taymans@gmail.com>
14901 examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
14902 Original commit message from CVS:
14903 * examples/app/Makefile.am:
14904 * examples/app/appsrc-ra.c: (feed_data), (seek_data),
14905 (found_source), (bus_message), (main):
14906 * examples/app/appsrc-seekable.c: (feed_data), (seek_data),
14907 (found_source), (bus_message), (main):
14908 * examples/app/appsrc-stream2.c: (feed_data), (found_source),
14909 (bus_message), (main):
14910 Added 3 more example application for using appsrc in random-access mode,
14911 pull-mode streaming and pull mode seekable.
14912 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
14913 (gst_app_src_start), (gst_app_src_do_get_size),
14914 (gst_app_src_create):
14915 * gst-libs/gst/app/gstappsrc.h:
14916 Make stream-type property writable.
14917 Unset flushing when starting so that we reuse appsrc.
14918 Inform basesrc about the configured size.
14919 Emit seek-data signal when we are going to a different offset in
14920 random-access mode.
14922 2008-06-06 14:19:54 +0000 Wim Taymans <wim.taymans@gmail.com>
14924 examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
14925 Original commit message from CVS:
14926 * examples/app/appsrc-stream.c: (found_source), (main):
14927 Use deep-notify until we can depend on a playbin2 with support for the
14930 2008-06-05 16:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
14932 examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
14933 Original commit message from CVS:
14934 * examples/app/.cvsignore:
14935 * examples/app/Makefile.am:
14936 * examples/app/appsrc-stream.c: (read_data), (start_feed),
14937 (stop_feed), (found_source), (bus_message), (main):
14938 Added an example on how to use appsrc in playbin in streaming mode from
14940 * examples/app/appsrc_ex.c: (main):
14941 Set pipeline to NULL to free queued buffers.
14942 * gst-libs/gst/app/gstapp-marshal.list:
14943 * gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
14944 (gst_app_src_class_init), (gst_app_src_init),
14945 (gst_app_src_flush_queued), (gst_app_src_dispose),
14946 (gst_app_src_set_property), (gst_app_src_get_property),
14947 (gst_app_src_unlock), (gst_app_src_unlock_stop),
14948 (gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
14949 (gst_app_src_check_get_range), (gst_app_src_do_seek),
14950 (gst_app_src_create), (gst_app_src_set_stream_type),
14951 (gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
14952 (gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
14953 (gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
14954 (gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
14955 (gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
14956 * gst-libs/gst/app/gstappsrc.h:
14957 Measure max queue size in bytes instead.
14958 Add support for 3 modes of operation, streaming, seekable and
14959 random-access, making basesrc handle the scheduling modes for each.
14960 Add appsrc:// uri handler so that automatic plugging can be done from
14961 playbin2 or uridecodebin, for example.
14962 Added support for custom segment formats.
14963 Add support for push and pull based operations from the application.
14964 Expand the methods so that errors can be detected.
14965 Flush the queued buffers on seeks and when shutting down.
14966 Add signals to inform the app that a seek must happen.
14968 2008-06-05 09:47:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14970 configure.ac: 0.10.19.2 pre-release
14971 Original commit message from CVS:
14973 0.10.19.2 pre-release
14975 2008-06-04 21:48:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14977 win32/common/: Add new API functions to the dll exports
14978 Original commit message from CVS:
14979 * win32/common/libgstrtsp.def:
14980 * win32/common/libgsttag.def:
14981 Add new API functions to the dll exports
14983 2008-06-04 17:42:38 +0000 Michael Smith <msmith@xiph.org>
14985 gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
14986 Original commit message from CVS:
14987 * gst/playback/gstplaybasebin.c:
14988 Disconnect signals from decodebins we created before we remove it from
14989 playbin, to avoid crashes if the decodebin is eventually disposed after
14990 the playbin itself (possible if the app takes a reference on the
14994 2008-06-04 17:12:40 +0000 Tim-Philipp Müller <tim@centricular.net>
14996 gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
14997 Original commit message from CVS:
14998 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
14999 (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
15000 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
15001 (h264_video_type_find), (mpeg_video_stream_type_find),
15002 (dv_type_find), (mmsh_type_find):
15003 Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
15004 copy caps for no good reason (this may be desirable to make it easier
15005 to detect leaks, but then it should probably be done for all caps
15006 in the typefinder somewhere).
15008 2008-06-04 16:06:49 +0000 Peter Kjellerstedt <pkj@axis.com>
15010 tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
15011 Original commit message from CVS:
15012 * tests/check/Makefile.am:
15013 Do not try to run the check tests for subparse unless it has been
15016 2008-06-04 16:00:26 +0000 Peter Kjellerstedt <pkj@axis.com>
15018 tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
15019 Original commit message from CVS:
15020 * tests/check/pipelines/streamheader.c: (buffer_probe_cb),
15021 (test_multifdsink_gdp_vorbisenc), (streamheader_suite):
15022 Do not try to run a test which requires vorbisenc unless we have
15025 2008-06-04 11:53:53 +0000 Peter Kjellerstedt <pkj@axis.com>
15027 gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
15028 Original commit message from CVS:
15029 * gst-libs/gst/rtsp/gstrtspconnection.c:
15030 (gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
15031 (gst_rtsp_connection_clear_auth_params),
15032 (gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
15033 * gst-libs/gst/rtsp/gstrtspconnection.h:
15034 Add a couple of missing argument guards.
15035 Add a way of setting the DSCP for an RTSP connection.
15036 Add an accessor method for the ip member of GstRTSPConnection as all
15037 members are supposed to be private.
15039 2008-06-04 11:33:23 +0000 Peter Kjellerstedt <pkj@axis.com>
15041 gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
15042 Original commit message from CVS:
15043 * gst/tcp/gstmultifdsink.c: (setup_dscp_client):
15044 Fixed accidental use of IPv4 options for all IPv6 addresses.
15046 2008-06-04 10:18:42 +0000 Tim-Philipp Müller <tim@centricular.net>
15048 gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
15049 Original commit message from CVS:
15050 * gst-libs/gst/interfaces/mixertrack.h:
15051 Document mixer track flags.
15053 2008-06-04 05:58:38 +0000 Antoine Tremblay <hexa00@gmail.com>
15055 gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
15056 Original commit message from CVS:
15057 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
15058 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
15059 Don't set caps on the buffers that contain a copy of the buffer
15060 including the caps of them resulting in an always increasing refcount
15061 of the caps and insanely large caps. Instead include a buffer without
15062 caps in the new caps. Fixes bug #536475.
15064 2008-06-04 05:44:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15066 gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
15067 Original commit message from CVS:
15068 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
15069 Transform a given PAR to a range on the struct with the generic
15070 height/width instead of the struct with the possibly restricted
15073 2008-06-04 04:24:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15075 gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
15076 Original commit message from CVS:
15077 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
15078 Prefer the given format if it contains something stricter than [1,MAX]
15079 for height or width and only put a structure that requires rescaling
15080 as second. This makes it possible to use videoscale in pipelines where
15081 the source can actually produce the wanted height/width but usually
15082 selects a different one from the requested.
15084 2008-06-03 20:01:58 +0000 John Millikin <jmillikin@gmail.com>
15086 gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
15087 Original commit message from CVS:
15088 Based on patch by: John Millikin <jmillikin gmail com>
15089 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
15090 (gst_vorbis_tag_add_coverart):
15091 Retrieve COVERART tags from vorbis comments (#512333)
15093 2008-06-03 19:44:48 +0000 Tim-Philipp Müller <tim@centricular.net>
15095 gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
15096 Original commit message from CVS:
15097 * gst-libs/gst/tag/tag.h:
15098 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
15099 Don't forget to add new enum value here too (should probably use
15100 glib-mkenums here...).
15102 2008-06-03 19:29:06 +0000 Tim-Philipp Müller <tim@centricular.net>
15104 gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
15105 Original commit message from CVS:
15106 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
15107 * gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
15108 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
15109 (gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
15110 (gst_tag_image_data_to_image_buffer):
15111 Add two utility functions to avoid code duplication (#512333):
15112 API: add gst_tag_image_data_to_image_buffer()
15113 API: add gst_tag_list_add_id3_image()
15115 2008-06-03 08:54:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15117 win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
15118 Original commit message from CVS:
15119 * win32/common/libgstaudio.def:
15120 Add gst_audio_check_channel_positions() to the exported symbols.
15122 2008-06-03 08:48:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15124 API: Make gst_audio_check_channel_positions() public.
15125 Original commit message from CVS:
15126 * docs/libs/gst-plugins-base-libs-sections.txt:
15127 * gst-libs/gst/audio/multichannel.c:
15128 (gst_audio_check_channel_positions):
15129 * gst-libs/gst/audio/multichannel.h:
15130 API: Make gst_audio_check_channel_positions() public.
15131 * tests/check/libs/audio.c: (GST_START_TEST):
15132 Add some simple checks for gst_audio_check_channel_positions().
15134 2008-06-02 20:09:14 +0000 Tim-Philipp Müller <tim@centricular.net>
15136 sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.
15137 Original commit message from CVS:
15138 * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
15139 minrange and maxrange are scaled according to the frequency
15142 2008-06-02 18:37:02 +0000 Tim-Philipp Müller <tim@centricular.net>
15144 ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
15145 Original commit message from CVS:
15146 * ext/pango/Makefile.am:
15147 * ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
15148 (gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
15149 Use gstvideo functions to calculate strides and plane offsets. Fixes
15150 rendering issue ('ghost' images of the text on the chroma planes)
15151 with widths or heights that are not multiples of 8 (#506659 and
15152 probably also #485729).
15153 * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
15155 Test with odd height/width too.
15157 2008-06-02 12:20:35 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15159 gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
15160 Original commit message from CVS:
15161 * gst/adder/gstadder.c: (gst_adder_query_duration),
15162 (gst_adder_query_latency):
15163 When using gst_element_iterate_pads() one has to unref every pad
15166 2008-05-31 19:57:57 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
15168 gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
15169 Original commit message from CVS:
15170 * gst-libs/gst/audio/gstbaseaudiosrc.c:
15171 (gst_base_audio_src_class_init):
15172 Add a gtk-doc chunk for the new properties to have a Since: indication.
15174 2008-05-31 19:50:59 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
15177 ChangeLog surgery, mark API change
15178 Original commit message from CVS:
15179 ChangeLog surgery, mark API change
15181 2008-05-31 18:10:47 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
15183 gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
15184 Original commit message from CVS:
15185 * gst-libs/gst/audio/gstbaseaudiosrc.c:
15186 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
15187 (gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
15188 (gst_base_audio_src_change_state):
15189 Provide readable actual-buffer-time and actual-latency-time properties
15190 that reflect the configured ringbuffer values. Fixes #524724.
15192 2008-05-30 15:29:20 +0000 Wim Taymans <wim.taymans@gmail.com>
15194 gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
15195 Original commit message from CVS:
15196 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
15197 (gst_basertppayload_change_state):
15198 Simply converting the running time into an RTP timestamp by scaling it
15199 based on the clock-rate is good enough for making an RTP timestamp. This
15200 has the added benefit that we can later on expose a property with the
15201 RTP timestamp of running time 0, as is needed for RTSP servers to
15202 generate the response of the PLAY request.
15204 2008-05-30 08:42:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15206 gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
15207 Original commit message from CVS:
15208 * gst/audioconvert/gstaudioconvert.c:
15209 (structure_has_fixed_channel_positions),
15210 (gst_audio_convert_transform_caps):
15211 Allow up to 11 positioned channels now that audioconvert can handle
15212 this but add no default positions for > 8 channels.
15213 * tests/check/elements/audioconvert.c: (GST_START_TEST):
15214 Add some unit tests for the above change: Test conversion of
15215 11 positioned channels to stereo and the other way around, test
15216 conversion of 15 unpositioned channels in different ways.
15218 2008-05-29 19:45:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15220 win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
15221 Original commit message from CVS:
15222 * win32/common/libgstaudio.def:
15223 Add gst_audio_clock_reset to the list of exported symbols.
15225 2008-05-29 19:37:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15227 tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
15228 Original commit message from CVS:
15229 * tests/check/elements/vorbisdec.c: (vorbisdec_suite):
15230 Remove wrong_channels_identification_header unit test as we now
15231 support 7 (and more channels).
15233 2008-05-29 12:17:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15235 gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
15236 Original commit message from CVS:
15237 * gst/audioconvert/gstchannelmix.c:
15238 (gst_channel_mix_fill_one_other):
15239 If mixing left or right to center (or the other way around) only take
15240 the complete value if we don't already have the original position in
15243 2008-05-29 11:34:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15245 gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
15246 Original commit message from CVS:
15247 * gst-libs/gst/audio/multichannel.c:
15248 (gst_audio_check_channel_positions),
15249 (gst_audio_set_structure_channel_positions_list),
15250 (gst_audio_fixate_channel_positions):
15251 Allow rear center together with rear left/right and other previously
15252 conflicting channel positions. The reason why they weren't allowed
15253 was the channel mixing implementation in audioconvert.
15254 Also take this into account when fixing channel layouts.
15255 Allow setting channel positions for 1/2 channels when using
15256 gst_audio_set_structure_channel_position().
15257 * gst/audioconvert/gstchannelmix.c:
15258 (gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
15259 (gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
15260 (gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
15261 Major rewrite of the channel mixing.
15262 We now allow previously conflicting channel positions to appear
15263 together (rear center and rear left/right for example).
15265 Rework the way channels are mixed together to take more possible
15266 channel positions into account, properly mix from/to side channels
15267 and don't assume that either center, left&right or nothing of a
15268 specific position is available anymore.
15269 * tests/check/elements/audioconvert.c: (GST_START_TEST):
15270 Adjust unit tests with non-standard 1/2 channel layouts to the more
15271 correct new behaviour.
15272 Add a unit test for 5.1->Stereo downmixing.
15274 2008-05-29 07:02:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15276 ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
15277 Original commit message from CVS:
15278 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
15279 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
15280 Add sane defaults for the 7 and 8 channel layouts as those are
15281 undefined in the Vorbis spec. Use NONE channel layouts when decoding
15282 more than 8 channels instead of erroring out. Fixes bug #535356.
15284 2008-05-28 16:10:20 +0000 Wim Taymans <wim.taymans@gmail.com>
15286 Add theoraparse to the docs and fix some docs.
15287 Original commit message from CVS:
15288 * docs/plugins/Makefile.am:
15289 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
15290 * docs/plugins/gst-plugins-base-plugins-sections.txt:
15291 * ext/theora/theoraparse.c:
15292 Add theoraparse to the docs and fix some docs.
15294 2008-05-28 15:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
15296 gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
15297 Original commit message from CVS:
15298 * gst-libs/gst/cdda/gstcddabasesrc.c:
15299 (gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
15300 Fix EOS condition and track addition check, the track.end sector is
15301 included in the track. Fixes #533265.
15303 2008-05-28 14:49:24 +0000 Mark Nauwelaerts <manauw@skynet.be>
15305 gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
15306 Original commit message from CVS:
15307 Patch by: Mark Nauwelaerts <manauw at skynet be>
15308 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
15309 (gst_video_rate_flush_prev), (gst_video_rate_event),
15310 (gst_video_rate_chain):
15311 * gst/videorate/gstvideorate.h:
15312 React (more) to NEWSEGMENT
15313 Small adjustment in timestamp calculation to prevent mismatches
15316 2008-05-28 11:31:44 +0000 Tim-Philipp Müller <tim@centricular.net>
15318 tests/examples/seek/seek.c: Initialise error to NULL as we should.
15319 Original commit message from CVS:
15320 * tests/examples/seek/seek.c: (make_parselaunch_pipeline):
15321 Initialise error to NULL as we should.
15323 2008-05-28 08:14:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15325 gst/adder/gstadder.c: Implement latency query.
15326 Original commit message from CVS:
15327 * gst/adder/gstadder.c: (gst_adder_query_duration),
15328 (gst_adder_query_latency), (gst_adder_query):
15329 Implement latency query.
15331 2008-05-27 18:10:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15333 gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
15334 Original commit message from CVS:
15335 * gst/adder/gstadder.c: (gst_adder_query_duration):
15336 Correctly resync the iterator if gst_iterator_next() returns
15337 GST_ITERATOR_RESYNC.
15339 2008-05-27 17:14:07 +0000 Tim-Philipp Müller <tim@centricular.net>
15341 win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
15342 Original commit message from CVS:
15343 * win32/vs6/libgstpbutils.dsp:
15344 Add pbutils-enumtypes.c to sources (#518037).
15346 2008-05-27 16:20:17 +0000 Wim Taymans <wim.taymans@gmail.com>
15348 gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
15349 Original commit message from CVS:
15350 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
15351 (gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
15352 * gst-libs/gst/audio/gstaudioclock.h:
15353 Add method to inform the clock that the time starts from 0 again. We use
15354 this info to calculate a clock offset so that the time we report in
15355 internal_time is monotonically increasing, as required by the clock base
15356 class. Fixes #521761.
15357 API: GstAudioClock::gst_audio_clock_reset()
15358 * gst-libs/gst/audio/gstbaseaudiosink.c:
15359 (gst_base_audio_sink_skew_slaving),
15360 (gst_base_audio_sink_change_state):
15361 * gst-libs/gst/audio/gstbaseaudiosrc.c:
15362 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
15363 Reset reported time when we (re)create the ringbuffer.
15365 2008-05-27 16:11:32 +0000 Tim-Philipp Müller <tim@centricular.net>
15367 ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
15368 Original commit message from CVS:
15369 * ext/alsa/gstalsamixertrack.c:
15370 (gst_alsa_mixer_track_update_alsa_capabilities):
15371 Make sure playback volumes aren't accidentally overwritten by
15372 capture volumes if an alsa mixer track has both playback and
15373 capture capabilities: we create two GstMixerTracks in that
15374 case, so make sure we query only the alsa capabilities that
15375 refer to the type of GstMixerTrack we created from the dual
15376 capability alsa element. Should fix issues with Audigy2 sound
15379 2008-05-27 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.net>
15381 tests/check/pipelines/oggmux.c: Don't use deprecated function.
15382 Original commit message from CVS:
15383 * tests/check/pipelines/oggmux.c: (test_pipeline):
15384 Don't use deprecated function.
15386 2008-05-27 10:35:55 +0000 Wim Taymans <wim.taymans@gmail.com>
15388 gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
15389 Original commit message from CVS:
15390 * gst/playback/gstdecodebin2.c:
15391 (gst_decode_group_control_source_pad), (gst_decode_group_expose):
15392 Check for NULL cases and log them, creating ghostpads can, for example,
15393 fail when the pad returns wrong caps.
15394 * gst/playback/gstplaybin2.c: (perform_eos):
15395 When pushing out the EOS event, collect the return value and warn when
15398 2008-05-26 17:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
15400 gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
15401 Original commit message from CVS:
15402 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
15403 (gst_riff_create_video_template_caps):
15404 Add support for DVCPRO.
15406 2008-05-26 10:29:20 +0000 Tim-Philipp Müller <tim@centricular.net>
15408 gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
15409 Original commit message from CVS:
15410 * gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
15411 Change default scaling method from nearest-neighbour to bilinear.
15413 2008-05-26 10:26:00 +0000 Tim-Philipp Müller <tim@centricular.net>
15415 tests/check/libs/video.c: More checks.
15416 Original commit message from CVS:
15417 * tests/check/libs/video.c:
15420 2008-05-25 20:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
15422 Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
15423 Original commit message from CVS:
15424 * gst/subparse/gstsubparse.c: (parser_state_init),
15425 (gst_sub_parse_format_autodetect), (handle_buffer):
15426 * gst/subparse/gstsubparse.h:
15427 * tests/check/elements/subparse.c: (test_tmplayer_style3b):
15428 Limit duration to a maximum of five seconds for tmplayer format where
15429 we can guess the duration only from the timestamp of the next line of
15430 text. We don't want to show a text for eternities just because nothing
15431 else is being said for a while.
15433 2008-05-23 14:14:28 +0000 Wim Taymans <wim.taymans@gmail.com>
15435 gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
15436 Original commit message from CVS:
15437 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15438 (gst_base_rtp_depayload_chain),
15439 (gst_base_rtp_depayload_handle_sink_event),
15440 (gst_base_rtp_depayload_push_full),
15441 (gst_base_rtp_depayload_change_state):
15442 Check sequence numbers, mark input buffers with a discont flag for the
15443 subclass when we detected a gap, drop duplicate buffers. We do this
15444 because one can use the element without a jitterbuffer in front and we
15445 don't want to feed the subclasses invalid or reordered data.
15446 Do an error when the subclass did not provide a process function instead
15448 Some other small cleanups.
15450 2008-05-22 22:35:40 +0000 Tim-Philipp Müller <tim@centricular.net>
15452 gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
15453 Original commit message from CVS:
15454 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
15455 May just as well use the precalculated uvstride here.
15457 2008-05-22 22:09:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15459 Add some documentation comments, and some new headers to be scanned.
15460 Original commit message from CVS:
15461 * docs/plugins/Makefile.am:
15462 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
15463 * docs/plugins/gst-plugins-base-plugins-sections.txt:
15464 * docs/plugins/gst-plugins-base-plugins.args:
15465 * docs/plugins/gst-plugins-base-plugins.hierarchy:
15466 * docs/plugins/gst-plugins-base-plugins.interfaces:
15467 * docs/plugins/gst-plugins-base-plugins.prerequisites:
15468 * docs/plugins/inspect/plugin-adder.xml:
15469 * docs/plugins/inspect/plugin-alsa.xml:
15470 * docs/plugins/inspect/plugin-audioconvert.xml:
15471 * docs/plugins/inspect/plugin-audiorate.xml:
15472 * docs/plugins/inspect/plugin-audioresample.xml:
15473 * docs/plugins/inspect/plugin-audiotestsrc.xml:
15474 * docs/plugins/inspect/plugin-cdparanoia.xml:
15475 * docs/plugins/inspect/plugin-decodebin.xml:
15476 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
15477 * docs/plugins/inspect/plugin-gdp.xml:
15478 * docs/plugins/inspect/plugin-gio.xml:
15479 * docs/plugins/inspect/plugin-gnomevfs.xml:
15480 * docs/plugins/inspect/plugin-libvisual.xml:
15481 * docs/plugins/inspect/plugin-ogg.xml:
15482 * docs/plugins/inspect/plugin-pango.xml:
15483 * docs/plugins/inspect/plugin-playback.xml:
15484 * docs/plugins/inspect/plugin-queue2.xml:
15485 * docs/plugins/inspect/plugin-subparse.xml:
15486 * docs/plugins/inspect/plugin-tcp.xml:
15487 * docs/plugins/inspect/plugin-theora.xml:
15488 * docs/plugins/inspect/plugin-typefindfunctions.xml:
15489 * docs/plugins/inspect/plugin-uridecodebin.xml:
15490 * docs/plugins/inspect/plugin-video4linux.xml:
15491 * docs/plugins/inspect/plugin-videorate.xml:
15492 * docs/plugins/inspect/plugin-videoscale.xml:
15493 * docs/plugins/inspect/plugin-videotestsrc.xml:
15494 * docs/plugins/inspect/plugin-volume.xml:
15495 * docs/plugins/inspect/plugin-vorbis.xml:
15496 * docs/plugins/inspect/plugin-ximagesink.xml:
15497 * docs/plugins/inspect/plugin-xvimagesink.xml:
15498 * ext/cdparanoia/gstcdparanoiasrc.c:
15499 * ext/ogg/gstoggdemux.c:
15500 * ext/ogg/gstoggdemux.h:
15501 * ext/ogg/gstoggmux.c:
15502 * ext/ogg/gstoggmux.h:
15503 * gst/audioconvert/audioconvert.c:
15504 * gst/audioconvert/audioconvert.h:
15505 * gst/audioconvert/gstaudioconvert.h:
15506 * gst/gdp/gstgdpdepay.h:
15507 * gst/gdp/gstgdppay.h:
15508 * gst/playback/gstdecodebin.c:
15509 * gst/playback/gstdecodebin2.c:
15510 * gst/playback/gstplaybin.c:
15511 * gst/playback/gstplaybin2.c:
15512 * gst/playback/gsturidecodebin.c:
15513 * gst/tcp/gstmultifdsink.c:
15514 * gst/tcp/gstmultifdsink.h:
15515 * gst/tcp/gsttcp.h:
15516 Add some documentation comments, and some new headers to be scanned.
15517 Rename some internal enum declarations (audioconvert's DitherType and
15518 NoiseShapingType, GstUnitType from the TCP elements) to match the
15519 documented GObject type names so that the docs pick them up.
15520 Name the playbin2 docs markups properly so they get picked up. They'll
15521 need renaming back when/if playbin2 becomes playbin.
15522 100% symbol coverage for the plugin docs, booya.
15524 2008-05-22 18:30:15 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
15526 gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
15527 Original commit message from CVS:
15528 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
15529 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
15530 Fix generation of NV12/NV21 frames. Fixes bug #532454.
15532 2008-05-22 11:59:33 +0000 Sjoerd Simons <sjoerd@luon.net>
15534 gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
15535 Original commit message from CVS:
15536 Patch by: Sjoerd Simons <sjoerd at luon dot net>
15537 * gst/playback/gstdecodebin.c: (remove_fakesink):
15538 Lock the fakesink before setting the state to NULL and removing it from
15539 the bin so that a concurrent state change cannot interfere.
15542 2008-05-21 17:09:42 +0000 Felipe Contreras <felipe.contreras@nokia.com>
15544 docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
15545 Original commit message from CVS:
15546 * docs/Makefile.am:
15547 Fix installing plugin documentation when gtk-doc is disabled.
15549 2008-05-21 17:01:16 +0000 Felipe Contreras <felipe.contreras@nokia.com>
15551 gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
15552 Original commit message from CVS:
15553 * gst-libs/gst/rtsp/Makefile.am:
15554 Distribute, don't install md5.h
15556 2008-05-21 16:47:58 +0000 Julien Moutte <julien@moutte.net>
15558 gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
15559 Original commit message from CVS:
15560 2008-05-21 Julien Moutte <julien@fluendo.com>
15561 * gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
15562 instead of SOL_IP, works on more platforms.
15563 * gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
15566 2008-05-21 16:44:15 +0000 Wim Taymans <wim.taymans@gmail.com>
15568 Some debug and comment fixes.
15569 Original commit message from CVS:
15570 * ext/vorbis/vorbisdec.c:
15571 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
15572 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
15573 Some debug and comment fixes.
15574 * tests/examples/dynamic/addstream.c: (main):
15577 2008-05-21 16:36:50 +0000 Wim Taymans <wim.taymans@gmail.com>
15579 Don't use bad gst_element_get_pad().
15580 Original commit message from CVS:
15581 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
15582 * gst/playback/decodetest.c: (new_decoded_pad_cb):
15583 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
15584 (try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
15585 (cleanup_decodebin):
15586 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
15587 (connect_element), (gst_decode_group_control_demuxer_pad):
15588 * gst/playback/gstplaybasebin.c: (queue_remove_probe),
15589 (queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
15591 * gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
15592 (gst_play_bin_set_property), (handoff), (gen_video_element),
15593 (gen_text_element), (gen_audio_element), (gen_vis_element),
15594 (remove_sinks), (add_sink), (setup_sinks):
15595 * gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
15596 * gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
15597 (gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
15598 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
15599 (gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
15600 (gen_video_chain), (gen_text_chain), (gen_audio_chain),
15601 (gen_vis_chain), (gst_play_sink_reconfigure),
15602 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
15603 (gst_play_sink_request_pad):
15604 * gst/playback/gsturidecodebin.c: (type_found), (setup_source):
15605 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
15607 * gst/playback/test6.c: (new_decoded_pad_cb):
15608 * tests/check/elements/audioconvert.c: (GST_START_TEST):
15609 * tests/check/elements/audiorate.c: (test_injector_chain),
15610 (do_perfect_stream_test):
15611 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
15612 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
15613 * tests/check/elements/gnomevfssink.c:
15614 * tests/check/elements/textoverlay.c:
15615 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
15616 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
15617 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
15618 * tests/check/pipelines/oggmux.c: (test_pipeline):
15619 * tests/check/pipelines/streamheader.c: (GST_START_TEST):
15620 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
15621 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
15622 * tests/examples/seek/scrubby.c: (make_wav_pipeline):
15623 * tests/examples/seek/seek.c: (make_mod_pipeline),
15624 (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
15625 (make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
15626 (make_theora_pipeline), (make_vorbis_theora_pipeline),
15627 (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
15628 (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
15629 (update_fill), (msg_buffering):
15630 Don't use bad gst_element_get_pad().
15632 2008-05-21 14:35:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15634 gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
15635 Original commit message from CVS:
15636 * gst-libs/gst/riff/riff-media.c:
15637 Fix wrong method name in docs. Fix calculation of strf fields for
15639 * gst-libs/gst/riff/riff-read.c:
15640 Whitespace fix and removing double ';'.
15642 2008-05-21 11:52:30 +0000 Wim Taymans <wim.taymans@gmail.com>
15644 docs/design/part-playbin2.txt: Add some leftover doc.
15645 Original commit message from CVS:
15646 * docs/design/part-playbin2.txt:
15647 Add some leftover doc.
15649 2008-05-21 11:36:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15651 gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
15652 Original commit message from CVS:
15653 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
15654 Fix copy & paste error in last commit.
15656 2008-05-21 11:30:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15658 gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
15659 Original commit message from CVS:
15660 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
15661 Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
15662 other channel positions when source has SIDE channels and dest doesn't
15663 or the other way around.
15665 2008-05-21 11:29:25 +0000 Henrik Eriksson <henriken@axis.com>
15667 gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
15668 Original commit message from CVS:
15669 Patch by: Henrik Eriksson <henriken at axis dot com>
15670 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
15671 (gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
15672 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
15673 (gst_multi_fd_sink_get_property):
15674 * gst/tcp/gstmultifdsink.h:
15675 Add support for DSCP QOS. Fixes #469933.
15677 2008-05-21 07:46:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15679 tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
15680 Original commit message from CVS:
15681 * tests/check/elements/audioconvert.c: (GST_START_TEST):
15682 Add another test that checks if conversion between standard 1 and 2
15683 channel layouts with and without positions set is working.
15685 2008-05-21 07:39:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15687 gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
15688 Original commit message from CVS:
15689 * gst-libs/gst/audio/multichannel.c:
15690 (gst_audio_check_channel_positions):
15691 Allow non-standard 2 channel layouts.
15692 * tests/check/elements/audioconvert.c: (GST_START_TEST):
15693 Add some tests for converting and remapping non-standard 1 and 2
15696 2008-05-21 07:28:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15698 gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
15699 Original commit message from CVS:
15700 * gst/audioconvert/gstchannelmix.c:
15701 (gst_channel_mix_fill_normalize):
15702 Prevent division by zero if the channel mix matrix contains only
15705 2008-05-21 06:45:22 +0000 Antoine Tremblay <hexa00@gmail.com>
15707 gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
15708 Original commit message from CVS:
15709 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
15710 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
15711 Close a buffer memory leak. Fixes bug #534071.
15713 2008-05-21 06:39:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15715 gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
15716 Original commit message from CVS:
15717 * gst-libs/gst/rtsp/gstrtsptransport.h:
15718 Make the GstRTSPTransport struct members public as there are no
15719 setters/getters and it's supposed to be changed directly.
15722 2008-05-21 05:48:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15724 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...
15725 Original commit message from CVS:
15726 * gst/adder/gstadder.c:
15727 Adder also doesn't support audio/x-raw-int with width!=depth so don't
15728 claim this on the pad template caps.
15730 2008-05-20 16:26:53 +0000 Wim Taymans <wim.taymans@gmail.com>
15732 gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
15733 Original commit message from CVS:
15734 * gst-libs/gst/audio/gstbaseaudiosink.c:
15735 (gst_base_audio_sink_sync_latency):
15736 We can only use our optimal calibration if we prerolled before the
15739 2008-05-20 14:35:42 +0000 Tim-Philipp Müller <tim@centricular.net>
15741 configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
15742 Original commit message from CVS:
15744 Require core CVS for GstBaseSrc buffer caps setting magic.
15746 2008-05-20 12:26:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15748 gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
15749 Original commit message from CVS:
15750 * gst/audioconvert/gstaudioconvert.c:
15751 (gst_audio_convert_fixate_channels):
15752 Fix logic in last commit.
15754 2008-05-20 12:15:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15756 gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
15757 Original commit message from CVS:
15758 * gst/audioconvert/gstaudioconvert.c:
15759 (gst_audio_convert_fixate_channels):
15760 Passthrough the channel positions if the number of output channels is
15761 the same as the number of input channels, the input had a channel
15762 layout and downstream requests no special one. We did this already for
15763 > 2 channels but now it's also done for 1 channel. Fixes bug #533617.
15765 2008-05-20 11:13:27 +0000 Wim Taymans <wim.taymans@gmail.com>
15767 ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
15768 Original commit message from CVS:
15769 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
15770 (gst_gnome_vfs_src_finalize),
15771 (gst_gnome_vfs_src_received_headers_callback),
15772 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
15773 * ext/gnomevfs/gstgnomevfssrc.h:
15774 Set the ICY caps on the srcpad from where they get picked up by the base
15775 class now and set on the outgoing buffers.
15776 * gst-libs/gst/audio/gstbaseaudiosrc.c:
15777 (gst_base_audio_src_create):
15778 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
15779 BaseSrc now sets the caps on outgoing buffers automatically.
15781 2008-05-20 11:09:06 +0000 Wim Taymans <wim.taymans@gmail.com>
15783 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
15784 Original commit message from CVS:
15785 * gst-libs/gst/audio/gstbaseaudiosink.c:
15786 (gst_base_audio_sink_resample_slaving),
15787 (gst_base_audio_sink_skew_slaving),
15788 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
15789 (gst_base_audio_sink_async_play),
15790 (gst_base_audio_sink_change_state):
15791 Change the way in which the ringbuffer is started when dealing with a
15792 slaved clock and latency. We now sync to the clock until we reach
15793 upstream latency before starting the ringbuffer. This has the effect
15794 that we can accurately align the master and slave clocks and let the
15795 rate correction code take care of the initial drift or rounding errors
15796 instead of leaving them uncorrected with the old approach.
15798 2008-05-20 08:12:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15800 gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
15801 Original commit message from CVS:
15802 * gst/audioconvert/gstaudioconvert.c:
15803 (gst_audio_convert_fixate_channels):
15804 Correctly set the default channel positions when converting to 8
15807 2008-05-19 16:13:25 +0000 Tim-Philipp Müller <tim@centricular.net>
15809 configure.ac: Error out if we don't have the required version of core.
15810 Original commit message from CVS:
15812 Error out if we don't have the required version of core.
15814 2008-05-19 15:59:40 +0000 Tim-Philipp Müller <tim@centricular.net>
15816 gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
15817 Original commit message from CVS:
15818 * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
15819 Use data scan helper in aac typefinder and stop scanning
15820 for headers when we've found a type. Also fix potential invalid
15821 memory access when calculating the frame length.
15823 2008-05-19 14:09:08 +0000 Tim-Philipp Müller <tim@centricular.net>
15825 gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
15826 Original commit message from CVS:
15827 * gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
15828 (mpeg_sys_is_valid_pack):
15829 Don't modify scan context when we return FALSE in ensure_data, so
15830 it's possible to continue scanning, and we don't end up with a NULL
15831 data pointer and a positive size, which might bite us the next time
15832 we're called. Small constification.
15834 2008-05-16 21:12:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15836 gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
15837 Original commit message from CVS:
15838 * gst/adder/gstadder.c:
15839 Adder doesn't support 24 bit samples so don't claim it supports them
15840 in the pad template caps.
15842 2008-05-14 20:28:02 +0000 Wim Taymans <wim.taymans@gmail.com>
15844 gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
15845 Original commit message from CVS:
15846 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15847 (gst_base_rtp_depayload_chain):
15848 Validate the RTP packet before further processing it. It's just too
15849 dangerous to accept random packets and people are not forced to use a
15850 jitterbuffer or session manager to filter out the bad packets.
15851 * gst-libs/gst/rtp/gstrtpbuffer.c:
15852 (gst_rtp_buffer_set_extension_data),
15853 (gst_rtp_buffer_get_payload_subbuffer):
15855 When setting extension data in a buffer that is too small, we fail and
15856 we should not set the extension bit.
15857 Change GST_WARNINGS into g_warning because they really are
15858 programming errors.
15859 * tests/check/libs/rtp.c: (GST_START_TEST):
15860 Catch the g_warnings now in the unit tests and that fact that failing to
15861 set extension data left the extension bit untouched.
15863 2008-05-14 13:57:41 +0000 Tim-Philipp Müller <tim@centricular.net>
15865 gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
15866 Original commit message from CVS:
15867 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
15868 Revert previous change which made basetransform handle buffer_alloc
15869 and which breaks things badly in the non-passthrough case since it
15870 returned buffers with a different (ie. sometimes smaller) size than
15871 the size requested.
15873 2008-05-14 13:43:12 +0000 Bernard B <b-gnome@largestprime.net>
15875 gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
15876 Original commit message from CVS:
15877 Patch by: Bernard B <b-gnome at largestprime dot net>
15878 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
15879 Fix seqnum compare function for bordercase values and fix the docs
15880 again. Fixes #533075.
15881 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
15882 Add a testcase for seqnum compare function.
15884 2008-05-14 10:58:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15886 gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
15887 Original commit message from CVS:
15888 * gst/adder/gstadder.c: (gst_adder_setcaps),
15889 (gst_adder_class_init):
15890 Correctly declare the supported endianness on the pad templates
15891 and check for correct endianness in the set caps function. Adder
15892 only supports native endianness.
15893 Also use gst_element_class_set_details_simple().
15895 2008-05-14 09:12:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15897 sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.
15898 Original commit message from CVS:
15899 * sys/xvimage/xvimagesink.c:
15900 Better debug logging in port value handling. Merging separate port
15901 value loops into one.
15903 2008-05-13 16:02:19 +0000 Hannes Bistry <hannesb@gmx.de>
15905 gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
15906 Original commit message from CVS:
15907 Patch by: Hannes Bistry <hannesb at gmx dot de>
15908 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
15909 * gst/tcp/gsttcpserversink.c:
15910 (gst_tcp_server_sink_handle_server_read),
15911 (gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
15912 Fix regression in clientsrc because we did not add the fd to the poll
15913 set anymore. Fixes #532364.
15914 Do some cleanups here and there.
15916 2008-05-13 13:04:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15918 gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
15919 Original commit message from CVS:
15920 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
15921 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
15922 * gst/playback/gstplay-marshal.list:
15923 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
15924 Use correct marshallers. GstCaps are a boxed type and no GObject
15927 2008-05-13 11:37:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15929 win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.
15930 Original commit message from CVS:
15931 * win32/common/libgstrtsp.def:
15932 Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
15935 2008-05-13 10:59:49 +0000 Sjoerd Simons <sjoerd@luon.net>
15937 tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
15938 Original commit message from CVS:
15939 Patch by: Sjoerd Simons <sjoerd at luon dot net>
15940 * tests/check/elements/audioresample.c:
15941 (live_switch_alloc_only_48000), (live_switch_get_sink_caps),
15942 (live_switch_push), (GST_START_TEST):
15943 Add unit test for the latest basetransform negotiation changes.
15946 2008-05-13 09:14:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15948 gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
15949 Original commit message from CVS:
15950 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
15951 Fix nv12<->nv21 conversion if stride is larger than width.
15953 2008-05-13 07:28:21 +0000 j^ <j@oil21.org>
15955 ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...
15956 Original commit message from CVS:
15957 Patch by: j^ <j at oil21 dot org>
15958 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
15959 (gst_ogg_pad_parse_skeleton_fisbone):
15960 * ext/ogg/gstoggdemux.h:
15961 Parse presentation time from skeleton streams and use it as offset
15962 for the timestamps. Fixes bug #530068.
15964 2008-05-12 08:45:11 +0000 Wim Taymans <wim.taymans@gmail.com>
15966 gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
15967 Original commit message from CVS:
15968 * gst-libs/gst/audio/gstbaseaudiosink.c:
15969 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
15970 Revert previous patch that attempted to more accurately calculate the
15971 initial offset between master and slave clock. The best thing we can do
15972 in general is take the time of both clocks as the diff since we don't
15973 know when the actual preroll happened.
15975 2008-05-11 19:52:59 +0000 Tim-Philipp Müller <tim@centricular.net>
15977 gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
15978 Original commit message from CVS:
15979 * gst-libs/gst/pbutils/install-plugins.c:
15980 Fix docs: type and missing word.
15982 2008-05-10 20:16:21 +0000 Tim-Philipp Müller <tim@centricular.net>
15984 gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
15985 Original commit message from CVS:
15986 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
15987 Don't do lots of 4-byte peeks, but use the 'new' data scan helper
15988 for this instead; don't check if we've found enough markers after
15989 each and every step, it's enough to do that only if we've actually
15990 found a new marker.
15991 Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
15993 2008-05-10 18:19:17 +0000 Tim-Philipp Müller <tim@centricular.net>
15995 gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
15996 Original commit message from CVS:
15997 * gst/typefind/gsttypefindfunctions.c:
15998 (DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
15999 (data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
16000 (mpeg_video_stream_type_find):
16001 Move scan helper thingy to the beginning of the file so we can use
16002 it in other typefind functions. Rename it to something more
16003 generic. Also improve handling of things towards the end of the
16004 typefind data: peek as much as we can if we know the size of the
16005 data, rather than just min_size.
16007 2008-05-09 21:42:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16009 Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
16010 Original commit message from CVS:
16011 * docs/libs/gst-plugins-base-libs-sections.txt:
16012 * gst-libs/gst/interfaces/colorbalance.c:
16013 * gst-libs/gst/interfaces/colorbalance.h:
16014 * gst-libs/gst/interfaces/colorbalancechannel.c:
16015 * gst-libs/gst/interfaces/colorbalancechannel.h:
16016 * gst-libs/gst/interfaces/tuner.c:
16017 * gst-libs/gst/interfaces/tunerchannel.c:
16018 * gst-libs/gst/interfaces/tunerchannel.h:
16019 * gst-libs/gst/interfaces/tunernorm.c:
16020 * gst-libs/gst/interfaces/tunernorm.h:
16021 * gst-libs/gst/video/video.c:
16022 * gst-libs/gst/video/video.h:
16023 Document the GstTuner and GstColorBalance interfaces, and some
16024 other random API functions that needed it. 70% symbol coverage, woo.
16026 2008-05-09 16:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
16028 gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
16029 Original commit message from CVS:
16030 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
16031 Choose to allocate one less segment but require one additional segment
16033 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
16034 No need to increment the number of segments in the source.
16035 * gst-libs/gst/audio/gstbaseaudiosink.c:
16036 (gst_base_audio_sink_get_time), (clock_convert_external),
16037 (gst_base_audio_sink_resample_slaving),
16038 (gst_base_audio_sink_skew_slaving),
16039 (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
16040 (gst_base_audio_sink_async_play):
16041 Remove adding latency when returning the internal time while subtracting
16042 it again when we use the value a little later.
16043 When calculating the end timestamp, we are making a rounding error
16044 with the current algorithm. Ensure that we don't accumulate these
16045 rounding errors when aligning samples by not resampling at all if we
16046 don't need to. Fixes #419351.
16047 Make the initial calibration of the clock slaving a little more
16048 predictable and accurate. Also handle the case where we don't do
16051 2008-05-09 08:34:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16053 gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
16054 Original commit message from CVS:
16055 Based on a patch by:
16056 Björn Benderius <bjoern dot benderius at axis dot com>
16057 * gst/ffmpegcolorspace/avcodec.h:
16058 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
16059 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
16060 (gst_ffmpegcsp_avpicture_fill):
16061 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
16062 * gst/ffmpegcolorspace/imgconvert_template.h:
16063 Add conversions from/to NV12 and NV21 and conversions between those
16064 two formats. Fixes bug #532166.
16066 2008-05-08 17:35:44 +0000 Edward Hervey <bilboed@bilboed.com>
16068 gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
16069 Original commit message from CVS:
16070 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
16071 Abort the h264 typefinding as soon as _peek() doesn't return anything,
16072 which happens for example with files smaller than 128kb.
16074 2008-05-08 14:46:27 +0000 Wouter Cloetens <zombie@e2big.org>
16076 gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
16077 Original commit message from CVS:
16078 Patch by: Wouter Cloetens <zombie at e2big dot org>
16079 * gst-libs/gst/rtsp/Makefile.am:
16080 * gst-libs/gst/rtsp/gstrtspconnection.c:
16081 (gst_rtsp_connection_create), (md5_digest_to_hex_string),
16082 (auth_digest_compute_hex_urp), (auth_digest_compute_response),
16083 (add_auth_header), (gst_rtsp_connection_free),
16084 (gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
16085 (gst_rtsp_connection_set_auth_param),
16086 (gst_rtsp_connection_clear_auth_params):
16087 * gst-libs/gst/rtsp/gstrtspconnection.h:
16088 Add Digest authorization support for RTSP connections. See #532065.
16089 * gst-libs/gst/rtsp/md5.c:
16090 * gst-libs/gst/rtsp/md5.h:
16091 Yeap, another md5 implementation until we can depend on a glib that has
16094 2008-05-08 06:20:42 +0000 Sjoerd Simons <sjoerd@luon.net>
16096 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
16097 Original commit message from CVS:
16098 Patch by: Sjoerd Simons <sjoerd at luon dot net>
16099 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
16100 Let audioresample use the buffer allocation of basetransform instead
16102 * tests/check/elements/audioresample.c: (alloc_only_48000),
16103 (GST_START_TEST), (audioresample_suite):
16104 Add unit test for the recent basetransform bugfix, where upstream
16105 changes caps to something that can't be passed through anymore.
16107 2008-05-07 19:50:27 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
16109 win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
16110 Original commit message from CVS:
16111 * win32/common/config.h.in:
16112 Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
16113 use the real thing than having "???" unconditionally.
16115 2008-05-07 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
16117 gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
16118 Original commit message from CVS:
16119 * gst-libs/gst/audio/gstbaseaudiosink.c:
16120 (gst_base_audio_sink_query):
16121 Report the latency with the new seglatency parameter.
16122 * gst-libs/gst/audio/gstringbuffer.c:
16123 (gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
16124 (gst_ring_buffer_acquire):
16125 * gst-libs/gst/audio/gstringbuffer.h:
16126 Add new field to the ringbufferspec to specify the expected latency
16127 between the underlying device read/write pointer, this is needed
16128 when writing sinks that sit a little closer to the hardware.
16129 Add some more docs for other fields.
16131 2008-05-07 10:38:23 +0000 Wim Taymans <wim.taymans@gmail.com>
16133 gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
16134 Original commit message from CVS:
16135 * gst-libs/gst/app/.cvsignore:
16136 * gst-libs/gst/app/Makefile.am:
16137 * gst-libs/gst/app/gstapp-marshal.list:
16138 Add marshal.list, make it compile and add to cvsignore.
16139 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
16140 (gst_app_sink_stop):
16142 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
16143 (gst_app_src_init), (gst_app_src_set_property),
16144 (gst_app_src_get_property), (gst_app_src_unlock),
16145 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
16146 (gst_app_src_create), (gst_app_src_set_caps),
16147 (gst_app_src_get_caps), (gst_app_src_set_size),
16148 (gst_app_src_get_size), (gst_app_src_set_seekable),
16149 (gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
16150 (gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
16151 (gst_app_src_end_of_stream):
16152 * gst-libs/gst/app/gstappsrc.h:
16153 Beat appsrc in shape, add signals and actions.
16155 Add properties for caps, size, seekability and max-buffers.
16156 Fix unlock/stop code.
16158 2008-05-06 12:35:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16160 gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
16161 Original commit message from CVS:
16162 * gst/volume/gstvolume.c: (volume_transform_ip):
16163 Return NOT_NEGOTIATED if we didn't set a process function yet for some
16164 reason instead of crashing later. Might fix bug #509125.
16166 2008-05-06 12:12:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
16168 gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
16169 Original commit message from CVS:
16170 Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
16171 * gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
16172 * gst/audioconvert/audioconvert.h:
16173 * gst/audioconvert/gstaudioconvert.c:
16174 (gst_audio_convert_parse_caps),
16175 (structure_has_fixed_channel_positions),
16176 (gst_audio_convert_transform_caps):
16177 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
16178 Add support for more than 8 channels and NONE channel layouts. For
16179 more than 8 channels no channel conversion is supported yet, only
16180 format conversions are supported. Fixes bug #398033.
16181 * tests/check/elements/audioconvert.c: (verify_convert),
16182 (GST_START_TEST), (audioconvert_suite):
16183 Add some unit tests by Tim for checking the NONE channel layouts
16184 and more than 8 channels and add some more unit tests for channel
16187 2008-05-06 10:16:49 +0000 Wim Taymans <wim.taymans@gmail.com>
16189 gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
16190 Original commit message from CVS:
16191 * gst/playback/gstdecodebin2.c: (connect_pad):
16192 When autoplugging fails, set the element back to NULL before
16195 2008-05-06 09:59:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16197 win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols.
16198 Original commit message from CVS:
16199 * win32/common/libgstaudio.def:
16200 Add gst_base_audio_src_[sg]et_slave_method() to the exported
16203 2008-05-05 12:33:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16205 gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
16206 Original commit message from CVS:
16207 * gst/subparse/samiparse.c: (handle_start_sync),
16208 (end_sami_element), (characters_sami):
16209 Remove trailing, leading and double whitespaces.
16210 Correctly timestamp buffers and output the last buffer too.
16211 * tests/check/elements/subparse.c: (GST_START_TEST),
16213 Add a simple unit test for SAMI parsing.
16215 2008-05-05 11:14:48 +0000 Young-Ho Cha <ganadist@chollian.net>
16217 gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
16218 Original commit message from CVS:
16219 Patch by: Young-Ho Cha <ganadist at chollian dot net>
16220 * gst/subparse/samiparse.c: (handle_start_sync),
16221 (start_sami_element), (end_sami_element), (characters_sami),
16222 (sami_context_reset):
16223 Only output characters inside the "sync" elements. There could be
16224 other elements like "style" that have some content but should
16225 not be printed. Fixes bug #467911.
16227 2008-05-05 10:27:45 +0000 Wim Taymans <wim.taymans@gmail.com>
16229 gst-libs/gst/app/gstappsink.*: Start some docs.
16230 Original commit message from CVS:
16231 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
16232 (gst_app_sink_init), (gst_app_sink_set_property),
16233 (gst_app_sink_get_property), (gst_app_sink_unlock_start),
16234 (gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
16235 (gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
16236 (gst_app_sink_preroll), (gst_app_sink_render),
16237 (gst_app_sink_set_caps), (gst_app_sink_set_drop),
16238 (gst_app_sink_get_drop):
16239 * gst-libs/gst/app/gstappsink.h:
16241 Add property to drop buffers when the queue is filled
16242 Fix unlocking and flushing when the queues are filled.
16244 2008-05-05 10:03:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16246 gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
16247 Original commit message from CVS:
16248 * gst/playback/gstplaybasebin.c: (set_audio_mute),
16249 (set_active_source):
16250 * gst/playback/gstplaybasebin.h:
16251 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
16252 (playbin_set_audio_mute):
16253 Allow setting -1 as current-audio to mute the current audio stream,
16254 similar to what is done for subtitles. Fixes bug #342294.
16256 2008-05-05 07:41:03 +0000 Edward Hervey <bilboed@bilboed.com>
16258 gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
16259 Original commit message from CVS:
16260 * gst-libs/gst/pbutils/descriptions.c: (formats):
16261 It's SorensOn and not SorensEn.
16263 2008-05-04 15:23:36 +0000 Tim-Philipp Müller <tim@centricular.net>
16265 gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
16266 Original commit message from CVS:
16267 * gst-libs/gst/pbutils/descriptions.c: (formats):
16268 Fix description of video/x-flash-video.
16270 2008-05-04 15:02:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16272 Remove some unused code.
16273 Original commit message from CVS:
16274 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
16275 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
16276 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
16277 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
16278 Remove some unused code.
16279 * gst/audioconvert/gstaudioquantize.c:
16280 (gst_audio_quantize_free_noise_shaping):
16281 Don't return before freeing the noise shaping history.
16283 2008-05-03 16:00:04 +0000 Tim-Philipp Müller <tim@centricular.net>
16285 tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
16286 Original commit message from CVS:
16287 * tests/check/elements/subparse.c: (do_test),
16288 (test_tmplayer_style3b), (subparse_suite):
16289 Add unit test for the tmplayer variant from bug #530962.
16291 2008-05-03 15:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
16293 gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
16294 Original commit message from CVS:
16295 * gst/subparse/gstsubparse.c: (handle_buffer),
16296 (gst_sub_parse_sink_event):
16297 * gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
16298 (tmplayer_parse_line):
16299 Fix parsing of tmplayer subtitle variant where every single line contains
16300 text and there isn't an empty line after each line to determine the
16301 duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
16302 making sure that we push out the last line of text without a duration if
16303 there's still text left in the buffer at the end.
16305 2008-05-03 15:39:04 +0000 Tim-Philipp Müller <tim@centricular.net>
16307 gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
16308 Original commit message from CVS:
16309 * gst/subparse/gstsubparse.c: (feed_textbuf):
16310 Fix detection of discontinuities based on the buffer offset (doesn't work
16311 so well if no buffer offset is set) and also check for the DISCONT buffer
16312 flag. This keeps the parser state from being reset after each buffer in
16315 2008-05-03 12:09:16 +0000 Tim-Philipp Müller <tim@centricular.net>
16317 gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
16318 Original commit message from CVS:
16319 * gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
16320 Further fine-tuning: don't absolutely require sequence or GOP headers
16321 (as introduced in the previous commit), but adjust the typefind
16322 probabilities returned accordingly if we don't see them. Also make sure
16323 picture header and first slice are somewhat close to each other (which
16324 is not perfect but still better than requiring a fixed offset or having
16327 2008-05-02 12:13:08 +0000 Wim Taymans <wim.taymans@gmail.com>
16329 gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
16330 Original commit message from CVS:
16331 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
16332 (gst_basertppayload_sink_setcaps),
16333 (gst_basertppayload_sink_getcaps):
16334 Rename the setcaps/getcaps function internally to make it clear that
16335 they are called for the sink pad.
16337 2008-05-02 12:11:07 +0000 Wim Taymans <wim.taymans@gmail.com>
16339 gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
16340 Original commit message from CVS:
16341 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16342 (gst_base_rtp_depayload_class_init),
16343 (gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
16344 (gst_base_rtp_depayload_packet_lost),
16345 (gst_base_rtp_depayload_set_gst_timestamp):
16346 * gst-libs/gst/rtp/gstbasertpdepayload.h:
16347 Catch packet-lost events from the jitterbuffer and convert them into a
16348 vmethod call (lost-packet) so that depayloaders can do something smart.
16349 Also add a default packet-lost function that sends out a segment update
16352 2008-05-02 11:13:05 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16354 gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
16355 Original commit message from CVS:
16356 * gst/playback/test4.c:
16357 * gst/playback/test5.c:
16358 * gst/playback/test6.c:
16359 * gst/playback/test7.c:
16360 Also include config.h when relying on defines from it. Fixes the
16361 build. Its been a please to serve :)
16363 2008-05-02 10:54:51 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
16366 * gst/videotestsrc/videotestsrc.c:
16367 Add support for NV12 and NV21 in videotestsrc
16368 Original commit message from CVS:
16369 * gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
16370 (paint_setup_NV21), (paint_hline_NV12_NV21):
16371 Add support for NV12 and NV21 in videotestsrc
16373 2008-05-02 10:02:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16375 gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
16376 Original commit message from CVS:
16377 * gst/videoscale/gstvideoscale.c:
16378 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
16379 * gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
16380 (vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
16381 (vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
16382 (vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
16383 (vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
16384 (vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
16385 (vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
16386 (vs_image_scale_linear_RGB555):
16387 Support 1x1 images as input and output as for example the BBC HQ new
16388 streams have 1x1 GIFs in the playlists for some reason.
16390 2008-05-01 19:11:42 +0000 Tim-Philipp Müller <tim@centricular.net>
16392 gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
16393 Original commit message from CVS:
16394 * gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
16396 If we can't activate one of the decoders we plugged in (such as,
16397 say, musepackdec) for some reason (it might not support push mode,
16398 for example), remove any pad probes that close_pad_link() might
16399 have set up. This makes sure we later don't try to remove a probe
16400 for a pad that doesn't exist any longer, and avoids nast warnings
16401 and probably other things too.
16403 2008-04-30 20:54:56 +0000 Tim-Philipp Müller <tim@centricular.net>
16405 gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
16406 Original commit message from CVS:
16407 * gst/typefind/gsttypefindfunctions.c:
16408 (mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
16410 Rework mpeg video stream typefinding a bit more: make sure sequence,
16411 GOP, picture and slice headers appear in the order they should and
16412 that we've in fact at least had one of each; fix picture header
16413 detection; decouple picture and slice header check - don't assume
16414 they're at a fixed offset, there may be extra data in between. Also,
16415 announce varying degrees of probability depending on what we found
16416 exactly (multiple pictures, at least one picture, just sequence and
16417 GOP headers). Finally, in _ensure_data(), take into account that we
16418 might be typefinding smaller amounts of data, such as the first
16419 buffer of a stream, so fall back to the minimum size needed as long
16420 as that's available, instead of erroring out if there's less than
16421 2kB of data. Fixes #526173. Conveniently also doesn't recognise the
16422 fuzzed file from #399342 as valid.
16424 2008-04-30 17:06:45 +0000 Michael Smith <msmith@xiph.org>
16426 ext/theora/theoradec.c: Cool kids don't divide by zero.
16427 Original commit message from CVS:
16428 * ext/theora/theoradec.c:
16429 Cool kids don't divide by zero.
16430 Treat PAR of x:0 as 1:1.
16433 2008-04-30 14:37:52 +0000 Tim-Philipp Müller <tim@centricular.net>
16435 gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
16436 Original commit message from CVS:
16437 * gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
16438 (mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
16439 (mpeg_video_stream_type_find):
16440 Refactor a bit: use context structure to track parsing offset and size of
16441 available data and make the code a bit clearer. Fixes bad memory access
16444 2008-04-28 22:18:49 +0000 Michael Smith <msmith@xiph.org>
16446 gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
16447 Original commit message from CVS:
16448 * gst/playback/test4.c:
16449 * gst/playback/test5.c:
16450 * gst/playback/test6.c:
16451 * gst/tcp/gstmultifdsink.c:
16452 Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
16455 2008-04-28 08:51:38 +0000 Wim Taymans <wim.taymans@gmail.com>
16457 gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
16458 Original commit message from CVS:
16459 * gst-libs/gst/audio/gstbaseaudiosink.h:
16461 * gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
16462 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
16463 (gst_base_audio_src_set_slave_method),
16464 (gst_base_audio_src_get_slave_method),
16465 (gst_base_audio_src_set_property),
16466 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
16467 * gst-libs/gst/audio/gstbaseaudiosrc.h:
16468 Add property and methods for selecting the clock slave method in the
16469 source, like in the sink.
16470 We only implement "none" and "re-timestamp" for now.
16471 API: gst_base_audio_src_set_slave_method()
16472 API: gst_base_audio_src_get_slave_method()
16474 2008-04-25 18:18:47 +0000 Wim Taymans <wim.taymans@gmail.com>
16476 gst-libs/gst/app/gstappsink.*: Add more docs.
16477 Original commit message from CVS:
16478 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
16479 (gst_app_sink_init), (gst_app_sink_set_property),
16480 (gst_app_sink_get_property), (gst_app_sink_event),
16481 (gst_app_sink_preroll), (gst_app_sink_render),
16482 (gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
16483 (gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
16484 (gst_app_sink_pull_buffer):
16485 * gst-libs/gst/app/gstappsink.h:
16487 Add signals for when preroll and render buffers are available.
16488 Add property to control signal emission.
16489 Add property to control the max queue size.
16491 2008-04-25 07:37:09 +0000 Wim Taymans <wim.taymans@gmail.com>
16493 gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
16494 Original commit message from CVS:
16495 * gst-libs/gst/rtp/gstrtpbuffer.c:
16496 Fix the docs about the seqnum compare function, it returns a difference.
16498 2008-04-24 09:27:35 +0000 Edward Hervey <bilboed@bilboed.com>
16500 ext/alsa/gstalsadeviceprobe.c: Don't return before freeing up the allocated structures.
16501 Original commit message from CVS:
16502 * ext/alsa/gstalsadeviceprobe.c:
16503 (gst_alsa_get_device_list): Don't return before freeing up
16504 the allocated structures.
16506 2008-04-24 08:19:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16508 gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
16509 Original commit message from CVS:
16510 * gst/playback/gstplaybin.c:
16511 Remove obsolete streaminfo code and fix a leak. Fixes #529546
16513 2008-04-23 13:50:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16515 ext/ogg/gstoggdemux.c: Revert the event part, that should not go in.
16516 Original commit message from CVS:
16517 * ext/ogg/gstoggdemux.c:
16518 Revert the event part, that should not go in.
16520 2008-04-23 13:45:29 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16522 ext/ogg/gstoggdemux.c: Don't leak GstPluginFeatures when filtering.
16523 Original commit message from CVS:
16524 * ext/ogg/gstoggdemux.c:
16525 Don't leak GstPluginFeatures when filtering.
16527 2008-04-23 08:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16529 sys/xvimage/xvimagesink.c: Add some logging for cases when grabbing the xv failed.
16530 Original commit message from CVS:
16531 * sys/xvimage/xvimagesink.c:
16532 Add some logging for cases when grabbing the xv failed.
16534 2008-04-22 06:18:04 +0000 David Schleef <ds@schleef.org>
16536 ext/ogg/gstoggmux.c: Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos packet. Should conform to what we cu...
16537 Original commit message from CVS:
16538 * ext/ogg/gstoggmux.c:
16539 Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos
16540 packet. Should conform to what we currently think is the
16541 final Ogg/Dirac muxing spec.
16543 2008-04-22 06:13:43 +0000 David Schleef <ds@schleef.org>
16545 sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display. Dark g...
16546 Original commit message from CVS:
16547 * sys/xvimage/xvimagesink.c:
16548 Fix typo that causes the overlay keying color to bright green
16549 on a 16-bit display. Dark grey good. Bright green bad.
16551 2008-04-21 13:47:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16553 ext/gnomevfs/gstgnomevfsuri.c: Add FIXME comment about using uri-list for source and sink.
16554 Original commit message from CVS:
16555 * ext/gnomevfs/gstgnomevfsuri.c:
16556 Add FIXME comment about using uri-list for source and sink.
16558 2008-04-20 11:42:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16560 ext/ogg/gstogmparse.c: GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to vaargs functions to gin...
16561 Original commit message from CVS:
16562 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
16563 GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to
16564 vaargs functions to gint. Otherwise the fractions will get 0 set
16565 instead of the correct value on big endian systems. Fixes bug #529018.
16567 2008-04-20 10:17:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16569 ext/gnomevfs/: Get the list of supported URI schemes in a threadsafe way and use the same list for the source and sink.
16570 Original commit message from CVS:
16571 * ext/gnomevfs/gstgnomevfssink.c:
16572 (gst_gnome_vfs_sink_uri_get_protocols):
16573 * ext/gnomevfs/gstgnomevfssrc.c:
16574 (gst_gnome_vfs_src_uri_get_protocols):
16575 * ext/gnomevfs/gstgnomevfsuri.c: (_internal_get_supported_uris),
16576 (gst_gnomevfs_get_supported_uris):
16577 Get the list of supported URI schemes in a threadsafe way and use the
16578 same list for the source and sink.
16580 2008-04-20 10:11:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16582 ext/gio/gstgio.c: Don't generate a new supported protocols list on each call but cache it. It's supposed to be static...
16583 Original commit message from CVS:
16584 * ext/gio/gstgio.c: (_internal_get_supported_protocols),
16585 (gst_gio_get_supported_protocols):
16586 Don't generate a new supported protocols list on each call but cache
16587 it. It's supposed to be static anyway, this way we only leak it once
16589 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
16590 (gst_gio_sink_class_init), (gst_gio_sink_finalize),
16591 (gst_gio_sink_set_property), (gst_gio_sink_get_property),
16592 (gst_gio_sink_start):
16593 * ext/gio/gstgiosink.h:
16594 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
16595 (gst_gio_src_class_init), (gst_gio_src_finalize),
16596 (gst_gio_src_set_property), (gst_gio_src_get_property),
16597 (gst_gio_src_start):
16598 * ext/gio/gstgiosrc.h:
16599 API: Add "file" properties where one can set a GFile as source/destination.
16600 Add locking to the properties and use gst_element_class_set_details_simple()
16601 instead of a static GstElementDetails struct.
16603 2008-04-19 20:06:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16605 gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
16606 Original commit message from CVS:
16607 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
16609 Add "mpp" and "mp+" as possible extensions for MusePack files.
16610 Add typefinding for MusePack StreamVersion 8 files and include the
16611 stream version in the caps.
16613 2008-04-19 16:33:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16615 gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
16616 Original commit message from CVS:
16617 * gst-libs/gst/rtp/gstrtppayloads.c:
16618 (gst_rtp_payload_info_for_name):
16619 Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
16621 2008-04-18 17:10:43 +0000 Tim-Philipp Müller <tim@centricular.net>
16623 configure.ac: Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level (NB: this only affects compilation of some...
16624 Original commit message from CVS:
16626 Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level
16627 (NB: this only affects compilation of some of the examples).
16628 Remove some configure.ac cruft that's not needed any longer.
16630 2008-04-18 14:54:01 +0000 Edward Hervey <bilboed@bilboed.com>
16632 gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
16633 Original commit message from CVS:
16634 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
16635 Don't validate the payload if there isn't any.
16638 2008-04-17 07:33:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16640 gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
16641 Original commit message from CVS:
16642 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
16643 Use g_atomic_int_set() instead of gst_atomic_int_set().
16645 2008-04-17 07:29:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16647 ext/gio/gstgio.c: Return NULL instead of a gchar * array with one NULL element if we don't get any supported URI sche...
16648 Original commit message from CVS:
16649 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
16650 Return NULL instead of a gchar * array with one NULL element if we
16651 don't get any supported URI schemes from GIO.
16653 2008-04-15 19:06:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16655 gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
16656 Original commit message from CVS:
16657 * gst/audiotestsrc/gstaudiotestsrc.c:
16658 Remove cpp style commented old code.
16660 2008-04-15 19:02:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16662 gst/playback/gstdecodebin2.c: Fix signal docs.
16663 Original commit message from CVS:
16664 * gst/playback/gstdecodebin2.c:
16667 2008-04-14 17:58:19 +0000 Tim-Philipp Müller <tim@centricular.net>
16669 ext/pango/gsttextoverlay.c: Fix textoverlay unit test again by making the supposed default value for the wait-text pr...
16670 Original commit message from CVS:
16671 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
16672 (gst_text_overlay_init):
16673 Fix textoverlay unit test again by making the supposed default
16674 value for the wait-text property the actual default value.
16675 Also fix Since: tag for new property.
16677 2008-04-11 17:13:52 +0000 Tim-Philipp Müller <tim@centricular.net>
16679 gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
16680 Original commit message from CVS:
16681 * gst-libs/gst/video/video.c: (gst_video_format_new_caps),
16682 (gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
16683 (gst_video_format_get_pixel_stride),
16684 (gst_video_format_get_component_width),
16685 (gst_video_format_get_component_height),
16686 (gst_video_format_get_component_offset), (gst_video_format_get_size),
16687 (gst_video_format_convert):
16688 Add guards to these functions to ensure sane input values.
16689 * tests/check/libs/video.c:
16690 Fix unit test not to create caps with width=0 and height=0.
16692 2008-04-11 01:25:01 +0000 Wim Taymans <wim.taymans@gmail.com>
16694 docs/design/draft-keyframe-force.txt: Fix typo.
16695 Original commit message from CVS:
16696 * docs/design/draft-keyframe-force.txt:
16698 * gst/playback/gstqueue2.c: (update_buffering),
16699 (gst_queue_handle_src_query):
16700 Set buffering mode in the messages.
16701 Set buffering percent in the query.
16702 * tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
16703 (do_stream_buffering), (do_download_buffering), (msg_buffering):
16704 Do some more fancy things based on the buffering method in use.
16706 2008-04-09 21:42:24 +0000 Wim Taymans <wim.taymans@gmail.com>
16708 tests/examples/seek/seek.c: Add basic download reports to seek using the new buffering API.
16709 Original commit message from CVS:
16710 * tests/examples/seek/seek.c: (update_fill), (set_update_fill),
16711 (play_cb), (pause_cb), (stop_cb), (msg_state_changed),
16712 (msg_buffering), (main):
16713 Add basic download reports to seek using the new buffering API.
16715 2008-04-09 21:40:17 +0000 Wim Taymans <wim.taymans@gmail.com>
16717 gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
16718 Original commit message from CVS:
16719 * gst/playback/gstqueue2.c: (update_buffering),
16720 (gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
16721 (gst_queue_src_checkgetrange_function):
16722 Include extra buffering stats in the buffering message.
16723 Implement BUFFERING query.
16724 * gst/playback/gsturidecodebin.c: (do_async_start),
16725 (do_async_done), (type_found), (setup_streaming), (setup_source),
16726 (gst_uri_decode_bin_change_state):
16727 Only add decodebin2 when the type is found in streaming mode.
16728 Make uridecodebin async to PAUSED even when we don't have decodebin2
16731 2008-04-09 08:38:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16733 ext/gio/gstgio.c: Filter cdda from the supported URI schemes. We can't support musicbrainz tags and everything else o...
16734 Original commit message from CVS:
16735 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
16736 Filter cdda from the supported URI schemes. We can't support
16737 musicbrainz tags and everything else one expects from a cdda source
16738 with GIO. Fixes bug #526794.
16740 2008-04-07 22:37:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16742 * sys/xvimage/xvimagesink.c:
16743 Fix calculation of 'expected size' for YV12 buffers.
16744 Original commit message from CVS:
16745 2008-04-07 Jan Schmidt <jan.schmidt@sun.com>
16746 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
16747 (gst_xvimagesink_buffer_alloc):
16748 Fix calculation of 'expected size' for YV12 buffers.
16749 Be a little more verbose in the debug output for buffer-alloc'ed
16750 buffers which turn out to have the wrong size.
16752 2008-04-07 22:26:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16755 Fix calculation of 'expected size' for YV12 buffers.
16756 Original commit message from CVS:
16757 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
16758 (gst_xvimagesink_buffer_alloc):
16759 Fix calculation of 'expected size' for YV12 buffers.
16760 Be a little more verbose in the debug output for buffer-alloc'ed
16761 buffers which turn out to have the wrong size.
16763 2008-04-07 10:50:11 +0000 Tim-Philipp Müller <tim@centricular.net>
16765 Merge other changes from 0.10.19 release branch.
16766 Original commit message from CVS:
16769 * gst-plugins-base.doap:
16770 Merge other changes from 0.10.19 release branch.
16772 2008-04-06 20:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
16774 gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
16775 Original commit message from CVS:
16776 * gst-libs/gst/audio/gstbaseaudiosink.c:
16777 (gst_base_audio_sink_class_init):
16778 * gst-libs/gst/audio/gstbaseaudiosrc.c:
16779 (gst_base_audio_src_class_init):
16780 * gst/playback/gstplayback.c: (plugin_init):
16781 * gst/volume/gstvolume.c: (plugin_init):
16782 Work around missing bits of thread-safety on older GLibs some
16783 more to avoid assertions when starting up multiple playbin
16784 objects concurrently (see #512382).
16786 2008-04-06 17:19:39 +0000 Tim-Philipp Müller <tim@centricular.net>
16788 gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
16789 Original commit message from CVS:
16790 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
16791 Remove some more fields.
16793 2008-04-06 08:56:07 +0000 Damien Lespiau <damien.lespiau@gmail.com>
16795 configure.ac: Actually build dlls when cross-compiling with mingw32.
16796 Original commit message from CVS:
16797 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
16799 Actually build dlls when cross-compiling with mingw32.
16802 2008-04-03 23:01:11 +0000 Tim-Philipp Müller <tim@centricular.net>
16804 configure.ac: Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
16805 Original commit message from CVS:
16807 Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
16809 2008-04-03 16:10:53 +0000 Wim Taymans <wim.taymans@gmail.com>
16811 tests/examples/seek/seek.c: Add statusbar.
16812 Original commit message from CVS:
16813 * tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
16814 (msg_buffering), (connect_bus_signals), (main):
16816 Add buffering support with feedback in the statusbar.
16818 2008-04-03 15:58:37 +0000 Tim-Philipp Müller <tim@centricular.net>
16820 ext/ogg/gstoggmux.c: Fix sample pipeline description.
16821 Original commit message from CVS:
16822 * ext/ogg/gstoggmux.c:
16823 Fix sample pipeline description.
16825 2008-04-03 14:58:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16827 docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
16828 Original commit message from CVS:
16829 * docs/plugins/Makefile.am:
16830 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
16831 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
16832 * docs/plugins/gst-plugins-base-plugins-sections.txt:
16833 Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
16834 * docs/plugins/gst-plugins-base-plugins.args:
16835 * docs/plugins/gst-plugins-base-plugins.hierarchy:
16836 * docs/plugins/gst-plugins-base-plugins.interfaces:
16837 * docs/plugins/gst-plugins-base-plugins.prerequisites:
16838 * docs/plugins/inspect/plugin-adder.xml:
16839 * docs/plugins/inspect/plugin-alsa.xml:
16840 * docs/plugins/inspect/plugin-audioconvert.xml:
16841 * docs/plugins/inspect/plugin-audiorate.xml:
16842 * docs/plugins/inspect/plugin-audioresample.xml:
16843 * docs/plugins/inspect/plugin-audiotestsrc.xml:
16844 * docs/plugins/inspect/plugin-cdparanoia.xml:
16845 * docs/plugins/inspect/plugin-decodebin.xml:
16846 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
16847 * docs/plugins/inspect/plugin-gdp.xml:
16848 * docs/plugins/inspect/plugin-gnomevfs.xml:
16849 * docs/plugins/inspect/plugin-libvisual.xml:
16850 * docs/plugins/inspect/plugin-ogg.xml:
16851 * docs/plugins/inspect/plugin-pango.xml:
16852 * docs/plugins/inspect/plugin-playback.xml:
16853 * docs/plugins/inspect/plugin-queue2.xml:
16854 * docs/plugins/inspect/plugin-subparse.xml:
16855 * docs/plugins/inspect/plugin-tcp.xml:
16856 * docs/plugins/inspect/plugin-theora.xml:
16857 * docs/plugins/inspect/plugin-typefindfunctions.xml:
16858 * docs/plugins/inspect/plugin-uridecodebin.xml:
16859 * docs/plugins/inspect/plugin-video4linux.xml:
16860 * docs/plugins/inspect/plugin-videorate.xml:
16861 * docs/plugins/inspect/plugin-videoscale.xml:
16862 * docs/plugins/inspect/plugin-videotestsrc.xml:
16863 * docs/plugins/inspect/plugin-volume.xml:
16864 * docs/plugins/inspect/plugin-vorbis.xml:
16865 * docs/plugins/inspect/plugin-ximagesink.xml:
16866 * docs/plugins/inspect/plugin-xvimagesink.xml:
16867 Update introspection data.
16868 * ext/ogg/gstoggmux.c:
16870 * gst/playback/gstdecodebin2.c:
16871 Don't use gtk-doc style comment start for private stuff, but make it
16872 formatted like this for consistency.
16874 2008-04-03 12:16:04 +0000 Wim Taymans <wim.taymans@gmail.com>
16876 gst/playback/gstdecodebin2.c: Remove fakesink hack, we can now implement this more elegantly.
16877 Original commit message from CVS:
16878 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
16879 (gst_decode_bin_init), (gst_decode_bin_dispose),
16880 (gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
16881 (gst_decode_bin_set_property), (gst_decode_bin_get_property),
16882 (analyze_new_pad), (connect_pad), (expose_pad),
16883 (gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
16884 (gst_decode_group_expose), (gst_decode_group_free),
16885 (do_async_start), (do_async_done), (gst_decode_bin_change_state):
16886 Remove fakesink hack, we can now implement this more elegantly.
16887 Added property to bypass typefinding.
16888 Removed underrun callback and demuxer pad probe, we now use the srcpad
16889 probe to expose groups.
16890 API::sink-caps property
16891 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
16892 Guard against multiple emissions of the no_more_pads signal, which
16893 happens when we are dealing with chained oggs.
16894 * gst/playback/gsturidecodebin.c: (remove_decoders),
16895 (make_decoder), (type_found), (setup_streaming), (source_new_pad),
16897 For streams, use our own typefind element and plug our queue after it.
16898 We will need this to determine the type of buffering to use for the
16901 2008-04-03 10:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
16903 gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
16904 Original commit message from CVS:
16905 * gst-libs/gst/audio/gstbaseaudiosink.c:
16906 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
16907 Guard against over and underflows because of clock slaving.
16908 When we are using our own clock, still compensate for any calibrations
16909 that we might have done to our clock.
16911 2008-04-03 10:22:33 +0000 Wim Taymans <wim.taymans@gmail.com>
16913 ext/theora/theoradec.c: Don't try to do anything fancy with the return code from pushing an event, it does not have e...
16914 Original commit message from CVS:
16915 * ext/theora/theoradec.c: (theora_handle_type_packet),
16916 (theora_dec_chain):
16917 Don't try to do anything fancy with the return code from pushing an
16918 event, it does not have enough information to turn it into a
16921 2008-04-03 10:19:43 +0000 Wim Taymans <wim.taymans@gmail.com>
16923 ext/ogg/gstoggdemux.c: Add small debug line.
16924 Original commit message from CVS:
16925 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_reset),
16926 (gst_ogg_demux_chain_elem_pad):
16927 Add small debug line.
16928 Pass return code from the internal decoder instead of the too generic
16931 2008-04-03 06:39:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16933 gst-libs/gst/cdda/: Use GLib's base64 implementation instead of our own.
16934 Original commit message from CVS:
16935 * gst-libs/gst/cdda/Makefile.am:
16936 * gst-libs/gst/cdda/base64.c:
16937 * gst-libs/gst/cdda/base64.h:
16938 * gst-libs/gst/cdda/gstcddabasesrc.c:
16939 (gst_cddabasesrc_calculate_musicbrainz_discid):
16940 Use GLib's base64 implementation instead of our own.
16942 2008-04-02 15:41:50 +0000 Wim Taymans <wim.taymans@gmail.com>
16944 ext/ogg/gstoggdemux.c: Refix oggdemux, we only have a problem if we failed to find a chain and we are not EOF.
16945 Original commit message from CVS:
16946 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
16947 (gst_ogg_demux_read_chain):
16948 Refix oggdemux, we only have a problem if we failed to find a chain and
16951 2008-04-02 15:07:01 +0000 Victor STINNER <victor.stinner@haypocalc.com>
16953 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...
16954 Original commit message from CVS:
16955 Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
16956 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
16957 (gst_ogg_demux_read_chain):
16958 When we fail to find a BOS page and we and up with no chain, error out
16959 properly instead of segfaulting. Fixes #525665.
16961 2008-04-02 14:58:05 +0000 Wim Taymans <wim.taymans@gmail.com>
16963 ext/ogg/gstoggdemux.c: The new-pad-group sequence is add-pads, no-more-pads, add-pads, no-more-pads...
16964 Original commit message from CVS:
16965 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
16966 (gst_ogg_demux_read_chain), (gst_ogg_demux_handle_page):
16967 The new-pad-group sequence is add-pads, no-more-pads, add-pads,
16970 2008-04-02 11:08:05 +0000 Wim Taymans <wim.taymans@gmail.com>
16972 gst/playback/gstqueue2.c: Update the estimated input data when we push out a buffer.
16973 Original commit message from CVS:
16974 * gst/playback/gstqueue2.c: (update_out_rates),
16975 (gst_queue_open_temp_location_file),
16976 (gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
16977 (gst_queue_handle_src_query), (gst_queue_set_property):
16978 Update the estimated input data when we push out a buffer.
16979 Add some debug info about the temp file.
16980 Only forward src events when we are not using a temp file.
16981 Don't block the duration query, we need to find something better.
16982 Don't leak the temp filename.
16984 2008-04-01 14:01:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16986 configure.ac: Require GLib 2.12 and liboil 0.3.14.
16987 Original commit message from CVS:
16989 Require GLib 2.12 and liboil 0.3.14.
16990 * gst/volume/gstvolume.c: (volume_process_double):
16991 Unconditionally use liboil 0.3.14 function.
16993 2008-03-31 16:08:45 +0000 Wim Taymans <wim.taymans@gmail.com>
16995 gst-libs/gst/riff/riff-media.c: ms-gsm can have arbitrarty sample rates. See #481354.
16996 Original commit message from CVS:
16997 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
16998 ms-gsm can have arbitrarty sample rates. See #481354.
17000 2008-03-28 16:22:35 +0000 Wim Taymans <wim.taymans@gmail.com>
17002 gst-libs/gst/riff/riff-media.c: MP4S is generic MPEG-4, not a microsoft variant.
17003 Original commit message from CVS:
17004 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
17005 MP4S is generic MPEG-4, not a microsoft variant.
17007 2008-03-27 15:26:38 +0000 Michael Smith <msmith@xiph.org>
17009 gst/gdp/gstgdpdepay.c: Check the body CRC (if set) when depayloading.
17010 Original commit message from CVS:
17011 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
17012 Check the body CRC (if set) when depayloading.
17015 2008-03-24 17:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
17017 ext/pango/gsttextoverlay.c: Fix Since: version for new property.
17018 Original commit message from CVS:
17019 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
17020 Fix Since: version for new property.
17022 2008-03-24 16:40:08 +0000 Wim Taymans <wim.taymans@gmail.com>
17024 gst-libs/gst/rtsp/gstrtspconnection.c: Don't error when poll_wait returns EAGAIN.
17025 Original commit message from CVS:
17026 * gst-libs/gst/rtsp/gstrtspconnection.c:
17027 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
17028 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
17029 Don't error when poll_wait returns EAGAIN.
17031 2008-03-24 14:08:22 +0000 Wim Taymans <wim.taymans@gmail.com>
17033 gst/playback/gstqueue2.c: The queue is never filled when there are no buffers in the queue at all.
17034 Original commit message from CVS:
17035 * gst/playback/gstqueue2.c: (gst_queue_is_filled):
17036 The queue is never filled when there are no buffers in the queue at all.
17039 2008-03-24 12:26:30 +0000 Wim Taymans <wim.taymans@gmail.com>
17041 gst/playback/gstplaybin2.c: Update some docs.
17042 Original commit message from CVS:
17043 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
17044 (init_group), (free_group), (gst_play_bin_init),
17045 (gst_play_bin_finalize), (gst_play_bin_set_uri),
17046 (gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
17047 (gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
17048 (gst_play_bin_set_current_video_stream),
17049 (gst_play_bin_set_current_audio_stream),
17050 (gst_play_bin_set_current_text_stream),
17051 (gst_play_bin_set_encoding), (gst_play_bin_set_property),
17052 (gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
17053 (no_more_pads_cb), (perform_eos), (autoplug_select_cb),
17054 (activate_group), (deactivate_group), (setup_next_source),
17055 (save_current_group), (gst_play_bin_change_state):
17057 Add new locks and conds to protect pipeline creation and group
17059 Implement the sub-uri property.
17060 Keep track of pending uridecodebin creation and configure the output
17061 pipeline after all streams are configured.
17062 Propagate subtitle encoding to the uridecodebins.
17063 Implement getting the video/audio/visualisation elements.
17064 Use input-selector for stream switching.
17065 If we are asked to do visualisation, prefer to autoplug raw sinks
17066 instead of sinks that accept encoded data.
17068 2008-03-24 12:15:26 +0000 Wim Taymans <wim.taymans@gmail.com>
17070 gst/playback/gstplaysink.*: Add methods to get audio/video/vis elements.
17071 Original commit message from CVS:
17072 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
17073 (gst_play_sink_init), (gst_play_sink_dispose),
17074 (gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
17075 (gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
17076 (gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
17077 (gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
17078 (gst_play_sink_set_volume), (gst_play_sink_get_volume),
17079 (gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
17080 (gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
17081 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
17082 (gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
17083 * gst/playback/gstplaysink.h:
17084 Add methods to get audio/video/vis elements.
17085 Add methods to set the font description for the overlay.
17086 Remove properties, we're using this element with its methods only.
17087 Add support for subtitles.
17088 Rearrange the locking a bit to not use the object lock for protecting
17089 the pipeline construction.
17090 Try to use the volume and mute property on the sink when its available.
17091 Implement the mute option with volume when the sink does not have a mute
17093 Only add volume element when the sink has no volume property.
17094 Only do visualisations with raw audio pads.
17096 2008-03-24 12:03:02 +0000 Wim Taymans <wim.taymans@gmail.com>
17098 ext/pango/gsttextoverlay.*: Add property to configure waiting for text on the textpad or not, with the default behavi...
17099 Original commit message from CVS:
17100 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
17101 (gst_text_overlay_init), (gst_text_overlay_set_property),
17102 (gst_text_overlay_get_property), (gst_text_overlay_src_event),
17103 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
17104 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
17105 (gst_text_overlay_change_state):
17106 * ext/pango/gsttextoverlay.h:
17107 Add property to configure waiting for text on the textpad or not, with
17108 the default behaviour being the old one (always wait for text before
17109 rendering the video). This default behaviour is usually not the best one
17110 because the text stream can very sparse and could require queueing a lot
17112 Fix the flushing and EOS handing so that we don't mix up their meaning.
17114 2008-03-24 11:54:02 +0000 Wim Taymans <wim.taymans@gmail.com>
17116 gst/playback/gsturidecodebin.c: Add a readonly source property and notify.
17117 Original commit message from CVS:
17118 * gst/playback/gsturidecodebin.c:
17119 (gst_uri_decode_bin_autoplug_factories),
17120 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
17121 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
17122 (gst_uri_decode_bin_set_property),
17123 (gst_uri_decode_bin_get_property), (no_more_pads_full),
17124 (new_decoded_pad_cb), (gen_source_element), (remove_decoders),
17125 (proxy_autoplug_factories_signal), (make_decoder),
17126 (source_new_pad), (setup_source):
17127 Add a readonly source property and notify.
17128 Add new lock for protecting the construction of the pipeline.
17129 Keep track of the decodebins we plugged.
17130 Correctly proxy the autoplug signal so that it actually continues.
17131 Proxy subtitle-encoding to the decodebins.
17133 2008-03-24 11:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
17135 tests/examples/seek/seek.c: Rearrange some buttons in playbin2 and make some other boxes insensitive when needed.
17136 Original commit message from CVS:
17137 * tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
17138 (text_toggle_cb), (update_streams), (main):
17139 Rearrange some buttons in playbin2 and make some other boxes insensitive
17141 Add language codes to subtitle selection boxes when we gind the right
17142 tags for the streams.
17144 2008-03-24 11:36:08 +0000 Wim Taymans <wim.taymans@gmail.com>
17146 gst/playback/gstdecodebin2.c: Protect caps property with the object lock.
17147 Original commit message from CVS:
17148 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
17149 (gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
17150 (gst_decode_bin_set_subs_encoding),
17151 (gst_decode_bin_get_subs_encoding),
17152 (gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
17153 (deactivate_free_recursive):
17154 Protect caps property with the object lock.
17155 Protect encoding property with the object lock.
17156 Keep list of elements we added that have the subtitle-encoding property.
17157 Distribute the subtitle-encoding to all of the elements when it
17160 2008-03-24 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
17162 gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
17163 Original commit message from CVS:
17164 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
17165 Small debug improvement.
17166 * gst-libs/gst/audio/gstbaseaudiosink.c:
17167 (gst_base_audio_sink_render):
17168 Fix bug in determining the sample start/stop position, we want to base
17169 this decision on the fact that we are going forwards or backwards, not
17170 slower or faster. This fixes some ugly resync warnings when playing at
17173 2008-03-23 13:41:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17175 ext/gio/gstgio.c: Correctly set the supported URI schemes and don't leave some schemes in the middle or at the start ...
17176 Original commit message from CVS:
17177 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
17178 Correctly set the supported URI schemes and don't leave
17179 some schemes in the middle or at the start at NULL.
17181 2008-03-23 13:12:41 +0000 Tim-Philipp Müller <tim@centricular.net>
17183 tests/check/elements/gdpdepay.c: Make test compile without unused function/variable warnings on PPC.
17184 Original commit message from CVS:
17185 * tests/check/elements/gdpdepay.c:
17186 Make test compile without unused function/variable warnings on PPC.
17188 2008-03-22 15:00:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17190 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
17191 Original commit message from CVS:
17193 * ext/alsa/gstalsamixerelement.c:
17194 (gst_alsa_mixer_element_class_init):
17195 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
17196 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
17197 * ext/cdparanoia/gstcdparanoiasrc.c:
17198 (gst_cd_paranoia_src_class_init):
17199 * ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
17200 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
17201 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
17202 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
17203 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
17204 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
17205 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
17206 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
17207 * ext/pango/gsttextrender.c: (gst_text_render_class_init):
17208 * ext/theora/theoradec.c: (gst_theora_dec_class_init):
17209 * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
17210 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
17211 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
17212 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
17213 (gst_audio_filter_template_class_init):
17214 * gst-libs/gst/audio/gstbaseaudiosink.c:
17215 (gst_base_audio_sink_class_init):
17216 * gst-libs/gst/audio/gstbaseaudiosrc.c:
17217 (gst_base_audio_src_class_init):
17218 * gst-libs/gst/cdda/gstcddabasesrc.c:
17219 (gst_cdda_base_src_class_init):
17220 * gst-libs/gst/interfaces/mixertrack.c:
17221 (gst_mixer_track_class_init):
17222 * gst-libs/gst/rtp/gstbasertpdepayload.c:
17223 (gst_base_rtp_depayload_class_init):
17224 * gst-libs/gst/rtp/gstbasertppayload.c:
17225 (gst_basertppayload_class_init):
17226 * gst/audioconvert/gstaudioconvert.c:
17227 (gst_audio_convert_class_init):
17228 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
17229 * gst/audioresample/gstaudioresample.c:
17230 (gst_audioresample_class_init):
17231 * gst/audiotestsrc/gstaudiotestsrc.c:
17232 (gst_audio_test_src_class_init):
17233 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
17234 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
17235 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
17236 (preroll_unlinked):
17237 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
17238 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
17239 * gst/playback/gstplaysink.c: (gst_play_sink_class_init):
17240 * gst/playback/gstqueue2.c: (gst_queue_class_init):
17241 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
17242 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
17243 (gst_stream_selector_class_init):
17244 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
17245 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
17246 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
17247 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
17248 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
17249 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
17250 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
17251 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
17252 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
17253 * gst/videotestsrc/gstvideotestsrc.c:
17254 (gst_video_test_src_class_init):
17255 * gst/volume/gstvolume.c: (gst_volume_class_init):
17256 * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
17257 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
17258 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
17259 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
17260 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
17261 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
17262 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
17263 static strings (i.e. all). This gives us less memory usage,
17264 fewer allocations and thus less memory defragmentation. Depend
17265 on core CVS for this. Fixes bug #523806.
17267 2008-03-22 14:13:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17269 ext/gio/gstgio.c: Filter http and https protocols. GIO/GVfs handles them but it's impossible to implement iradio/icec...
17270 Original commit message from CVS:
17271 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
17272 Filter http and https protocols. GIO/GVfs handles them but it's
17273 impossible to implement iradio/icecast with it. Better use
17274 souphttpsrc or something else for this.
17275 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
17276 If getting the file informations by a query fails try it with the
17277 seek-to-end trick too.
17279 2008-03-21 16:46:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17281 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...
17282 Original commit message from CVS:
17283 * gst/volume/gstvolume.c: (gst_volume_interface_supported),
17284 (gst_volume_base_init), (gst_volume_class_init),
17285 (volume_process_double), (volume_process_float),
17286 (volume_transform_ip), (plugin_init):
17287 memset buffers to zero if we get a GAP buffer. We usually see a
17288 buffer as one unit so let's handle it as one and don't care about
17289 volume changes while processing one buffer.
17290 Also clean up some stuff a bit.
17292 2008-03-21 15:58:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17294 gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
17295 Original commit message from CVS:
17296 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
17297 (gst_audio_convert_create_silence_buffer),
17298 (gst_audio_convert_transform):
17299 Make audioconvert GAP-aware by outputting silence buffers when the
17300 input has the GAP flag set. This is up to 8x faster.
17301 Based on a patch by Stefan Kost. Fixes bug #517813.
17303 2008-03-21 15:54:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17305 gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
17306 Original commit message from CVS:
17307 * gst/volume/gstvolume.c: (volume_process_double):
17308 Use oil_scalarmultiply_f64_ns() for double processing when it's
17309 available at compile time.
17311 2008-03-21 13:27:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17313 configure.ac: Fix lrint/lrintf checks to actually work. These functions are in libm on Linux at least so try to link ...
17314 Original commit message from CVS:
17316 Fix lrint/lrintf checks to actually work. These functions are
17317 in libm on Linux at least so try to link to it.
17319 2008-03-21 00:36:20 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17321 configure.ac: Back to development - 0.10.18.1
17322 Original commit message from CVS:
17324 Back to development - 0.10.18.1
17326 === release 0.10.18 ===
17328 2008-03-21 00:26:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17334 * docs/plugins/gst-plugins-base-plugins.args:
17335 * docs/plugins/gst-plugins-base-plugins.hierarchy:
17336 * docs/plugins/gst-plugins-base-plugins.interfaces:
17337 * docs/plugins/gst-plugins-base-plugins.prerequisites:
17338 * docs/plugins/gst-plugins-base-plugins.signals:
17339 * docs/plugins/inspect/plugin-adder.xml:
17340 * docs/plugins/inspect/plugin-alsa.xml:
17341 * docs/plugins/inspect/plugin-audioconvert.xml:
17342 * docs/plugins/inspect/plugin-audiorate.xml:
17343 * docs/plugins/inspect/plugin-audioresample.xml:
17344 * docs/plugins/inspect/plugin-audiotestsrc.xml:
17345 * docs/plugins/inspect/plugin-cdparanoia.xml:
17346 * docs/plugins/inspect/plugin-decodebin.xml:
17347 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
17348 * docs/plugins/inspect/plugin-gdp.xml:
17349 * docs/plugins/inspect/plugin-gnomevfs.xml:
17350 * docs/plugins/inspect/plugin-libvisual.xml:
17351 * docs/plugins/inspect/plugin-ogg.xml:
17352 * docs/plugins/inspect/plugin-pango.xml:
17353 * docs/plugins/inspect/plugin-playback.xml:
17354 * docs/plugins/inspect/plugin-queue2.xml:
17355 * docs/plugins/inspect/plugin-subparse.xml:
17356 * docs/plugins/inspect/plugin-tcp.xml:
17357 * docs/plugins/inspect/plugin-theora.xml:
17358 * docs/plugins/inspect/plugin-typefindfunctions.xml:
17359 * docs/plugins/inspect/plugin-uridecodebin.xml:
17360 * docs/plugins/inspect/plugin-video4linux.xml:
17361 * docs/plugins/inspect/plugin-videorate.xml:
17362 * docs/plugins/inspect/plugin-videoscale.xml:
17363 * docs/plugins/inspect/plugin-videotestsrc.xml:
17364 * docs/plugins/inspect/plugin-volume.xml:
17365 * docs/plugins/inspect/plugin-vorbis.xml:
17366 * docs/plugins/inspect/plugin-ximagesink.xml:
17367 * docs/plugins/inspect/plugin-xvimagesink.xml:
17368 * gst-plugins-base.doap:
17370 * win32/common/config.h:
17372 Original commit message from CVS:
17375 2008-03-21 00:16:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17402 Original commit message from CVS:
17405 2008-03-18 12:19:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17407 0.10.17.4 pre-release
17408 Original commit message from CVS:
17410 * win32/common/config.h:
17411 0.10.17.4 pre-release
17413 2008-03-18 11:20:05 +0000 Wim Taymans <wim.taymans@gmail.com>
17415 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL when trying to print strings that could be NULL because this might...
17416 Original commit message from CVS:
17417 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
17418 Use GST_STR_NULL when trying to print strings that could be NULL because
17419 this might crash on some platforms. See #520808.
17421 2008-03-18 11:10:12 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
17423 gst-libs/gst/rtsp/gstrtspconnection.c: Generic Windows fixes that makes libgstrtsp work on Windows when coupled with ...
17424 Original commit message from CVS:
17425 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
17426 * gst-libs/gst/rtsp/gstrtspconnection.c:
17427 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
17428 (read_line), (gst_rtsp_connection_read_internal):
17429 Generic Windows fixes that makes libgstrtsp work on Windows when
17430 coupled with the new GstPoll API. See #520808.
17432 2008-03-17 22:06:56 +0000 Milosz Derezynski <internalerror@gmail.com>
17434 ext/gio/gstgiobasesrc.c: If seeking to a new position succeeds don't simply return from create() without creating a b...
17435 Original commit message from CVS:
17436 Patch by: Milosz Derezynski <internalerror at gmail dot com>
17437 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
17438 If seeking to a new position succeeds don't simply return from
17439 create() without creating a buffer. Do this only in the case
17440 seeking to the new position fails. Fixes bug #523054.
17442 2008-03-17 10:32:28 +0000 Tim-Philipp Müller <tim@centricular.net>
17444 gst-libs/gst/video/video.c: Fix gst_video_format_parse_caps() for RGB caps with alpha channel (#522635).
17445 Original commit message from CVS:
17446 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
17447 (gst_video_format_from_rgba32_masks):
17448 Fix gst_video_format_parse_caps() for RGB caps with alpha channel
17450 * tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
17451 Add unit test for the RGB caps parsing and creation, checking for
17452 internal consistency of the new API and consistency of the API with
17453 the old GST_VIDEO_CAPS_* defines.
17455 2008-03-14 18:42:35 +0000 David Schleef <ds@schleef.org>
17457 gst/videotestsrc/videotestsrc.c: Oops, revert last change because -base is in freeze.
17458 Original commit message from CVS:
17459 * gst/videotestsrc/videotestsrc.c: Oops, revert last change
17460 because -base is in freeze.
17462 2008-03-14 17:33:09 +0000 William M. Brack <wbrack@mmm.hk>
17464 gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
17465 Original commit message from CVS:
17466 Patch by: William M. Brack
17467 * gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
17469 2008-03-14 09:54:44 +0000 Wim Taymans <wim.taymans@gmail.com>
17471 gst/playback/gststreamselector.*: Revert change that caused regression until a real fix is found.
17472 Original commit message from CVS:
17473 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
17474 (gst_selector_pad_chain):
17475 * gst/playback/gststreamselector.h:
17476 Revert change that caused regression until a real fix is found.
17479 2008-03-12 12:39:13 +0000 Michael Smith <msmith@xiph.org>
17481 gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
17482 Original commit message from CVS:
17483 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
17484 * gst-libs/gst/audio/gstringbuffer.h:
17485 Rename recently added buffer types to make more sense.
17486 * ext/alsa/gstalsasink.c: (alsasink_parse_spec),
17487 (gst_alsasink_write):
17488 Adapt for above API changes.
17491 2008-03-11 13:23:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17493 win32/common/libgstnetbuffer.def: Add new symbol gst_netaddress_equal. Fixes bug #521743.
17494 Original commit message from CVS:
17495 * win32/common/libgstnetbuffer.def:
17496 Add new symbol gst_netaddress_equal. Fixes bug #521743.
17498 2008-03-11 00:25:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17500 0.10.17.3 pre-release
17501 Original commit message from CVS:
17503 * win32/common/config.h:
17504 0.10.17.3 pre-release
17506 2008-03-10 17:19:56 +0000 Wim Taymans <wim.taymans@gmail.com>
17508 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
17509 Original commit message from CVS:
17510 * gst-libs/gst/audio/gstbaseaudiosrc.c:
17511 (gst_base_audio_src_create):
17512 Fix duration when no clock was provided. Fixes #520300.
17514 2008-03-07 18:17:44 +0000 Olivier Crete <tester@tester.ca>
17516 Add trivial function to compare GstNetAddress. See #520626.
17517 Original commit message from CVS:
17518 Patch by: Olivier Crete <tester at tester ca>
17519 * docs/libs/gst-plugins-base-libs-sections.txt:
17520 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
17521 * gst-libs/gst/netbuffer/gstnetbuffer.h:
17522 Add trivial function to compare GstNetAddress. See #520626.
17523 API: GstNetBuffer::gst_netaddress_equal
17525 2008-03-07 16:10:51 +0000 Wim Taymans <wim.taymans@gmail.com>
17527 gst/tcp/gstmultifdsink.c: Update mode property docs, it's deprecated now.
17528 Original commit message from CVS:
17529 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
17530 Update mode property docs, it's deprecated now.
17532 2008-03-07 15:48:51 +0000 Wim Taymans <wim.taymans@gmail.com>
17534 gst/: Remove GstPollMode from gstpoll constructor.
17535 Original commit message from CVS:
17536 * gst-libs/gst/rtsp/gstrtspconnection.c:
17537 (gst_rtsp_connection_create):
17538 * gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
17539 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
17540 * gst/tcp/gstmultifdsink.h:
17541 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
17542 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
17543 Remove GstPollMode from gstpoll constructor.
17545 2008-03-04 00:26:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17547 0.10.17.2 pre-release
17548 Original commit message from CVS:
17550 * win32/common/config.h:
17551 0.10.17.2 pre-release
17553 2008-03-03 23:59:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17555 gst/Makefile.am: GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean them twice
17556 Original commit message from CVS:
17558 GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
17560 * win32/common/libgstinterfaces.def:
17561 * win32/common/libgstrtp.def:
17562 Add new API to the defs
17564 2008-03-03 16:11:50 +0000 Mersad Jelacic <mersad@axis.com>
17566 gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
17567 Original commit message from CVS:
17568 Patch by: Mersad Jelacic <mersad at axis dot com>
17569 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
17570 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
17571 API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
17572 possible to specify the sample size in bits. (#509637)
17574 2008-03-03 13:59:19 +0000 Tim-Philipp Müller <tim@centricular.net>
17576 tests/check/libs/mixer.c: Add a few simple checks for the new message types.
17577 Original commit message from CVS:
17578 * tests/check/libs/mixer.c:
17579 Add a few simple checks for the new message types.
17581 2008-03-03 13:56:38 +0000 Tim-Philipp Müller <tim@centricular.net>
17583 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
17584 Original commit message from CVS:
17585 * docs/libs/gst-plugins-base-libs-sections.txt:
17586 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
17587 (gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
17588 (gst_mixer_message_get_type),
17589 (gst_mixer_message_parse_option_changed),
17590 (gst_mixer_message_parse_options_list_changed):
17591 * gst-libs/gst/interfaces/mixer.h: (GstMixerType),
17592 (GST_MIXER_MESSAGE_OPTION_CHANGED),
17593 (GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
17594 (GST_MIXER_MESSAGE_MIXER_CHANGED):
17595 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
17596 and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
17598 2008-03-03 13:50:18 +0000 Tim-Philipp Müller <tim@centricular.net>
17600 gst-libs/gst/interfaces/mixeroptions.*: API: add GstMixerOptions::get_values vfunc (#519906)
17601 Original commit message from CVS:
17602 * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
17603 (gst_mixer_options_get_values):
17604 * gst-libs/gst/interfaces/mixeroptions.h:
17605 (GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
17606 (_GstMixerOptions), (_GstMixerOptionsClass):
17607 API: add GstMixerOptions::get_values vfunc (#519906)
17609 2008-03-03 12:01:15 +0000 Peter Kjellerstedt <pkj@axis.com>
17611 configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#4...
17612 Original commit message from CVS:
17614 Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
17615 plug-ins are included/excluded. (#498222)
17617 2008-03-03 06:22:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17619 gst/typefind/gsttypefindfunctions.c: Add typefinder for IMelody files, using audio/x-imelody.
17620 Original commit message from CVS:
17621 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
17622 Add typefinder for IMelody files, using audio/x-imelody.
17625 2008-03-03 06:04:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17627 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
17628 Original commit message from CVS:
17629 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
17630 * ext/alsa/gstalsasink.c: (set_hwparams):
17631 * ext/alsa/gstalsasrc.c: (set_hwparams):
17632 * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
17633 * ext/ogg/gstoggmux.h:
17634 * ext/ogg/gstogmparse.c:
17635 * gst-libs/gst/audio/audio.c:
17636 * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
17637 * gst-libs/gst/pbutils/missing-plugins.c:
17638 (gst_missing_uri_sink_message_new),
17639 (gst_missing_element_message_new),
17640 (gst_missing_decoder_message_new),
17641 (gst_missing_encoder_message_new):
17642 * gst-libs/gst/rtp/gstbasertppayload.c:
17643 * gst-libs/gst/rtp/gstrtcpbuffer.c:
17644 (gst_rtcp_packet_bye_get_reason):
17645 * gst/audioconvert/gstaudioconvert.c:
17646 * gst/audioresample/gstaudioresample.c:
17647 * gst/ffmpegcolorspace/imgconvert.c:
17648 * gst/playback/test.c: (gen_video_element), (gen_audio_element):
17649 * gst/typefind/gsttypefindfunctions.c:
17650 * gst/videoscale/vs_4tap.c:
17651 * gst/videoscale/vs_4tap.h:
17652 * sys/v4l/gstv4lelement.c:
17653 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
17654 * sys/v4l/v4l_calls.c:
17655 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
17656 (gst_v4lsrc_try_capture):
17657 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
17658 (gst_ximagesink_ximage_new):
17659 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
17660 (gst_xvimagesink_xvimage_new):
17661 * tests/check/elements/audioconvert.c:
17662 * tests/check/elements/audioresample.c:
17663 (fail_unless_perfect_stream):
17664 * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
17665 * tests/check/elements/decodebin.c:
17666 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
17667 (setup_gdpdepay_streamheader):
17668 * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
17669 (setup_gdppay_streamheader):
17670 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
17671 * tests/check/elements/multifdsink.c: (setup_multifdsink):
17672 * tests/check/elements/textoverlay.c:
17673 * tests/check/elements/videorate.c: (setup_videorate):
17674 * tests/check/elements/videotestsrc.c: (setup_videotestsrc):
17675 * tests/check/elements/volume.c: (setup_volume):
17676 * tests/check/elements/vorbisdec.c: (setup_vorbisdec):
17677 * tests/check/elements/vorbistag.c:
17678 * tests/check/generic/clock-selection.c:
17679 * tests/check/generic/states.c: (setup), (teardown):
17680 * tests/check/libs/cddabasesrc.c:
17681 * tests/check/libs/video.c:
17682 * tests/check/pipelines/gio.c:
17683 * tests/check/pipelines/oggmux.c:
17684 * tests/check/pipelines/simple-launch-lines.c:
17685 (simple_launch_lines_suite):
17686 * tests/check/pipelines/streamheader.c:
17687 * tests/check/pipelines/theoraenc.c:
17688 * tests/check/pipelines/vorbisdec.c:
17689 * tests/check/pipelines/vorbisenc.c:
17690 * tests/examples/seek/scrubby.c:
17691 * tests/examples/seek/seek.c: (query_positions_elems),
17692 (query_positions_pads):
17693 * tests/icles/stress-xoverlay.c: (myclock):
17694 Correct all relevant warnings found by the sparse semantic code
17695 analyzer. This include marking several symbols static, using
17696 NULL instead of 0 for pointers and using "foo (void)" instead
17697 of "foo ()" for declarations.
17698 * win32/common/libgstrtp.def:
17699 Add gst_rtp_buffer_set_extension_data to the symbol definition file.
17701 2008-03-02 18:43:15 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
17703 gst/playback/gstplaybin2.c: Make the function signature of the _get_*_tags() functions match the signature of the vfu...
17704 Original commit message from CVS:
17705 Patch by: José Alburquerque <jaalburqu svn gnome org>
17706 * gst/playback/gstplaybin2.c:
17707 Make the function signature of the _get_*_tags() functions match
17708 the signature of the vfuncs they implement, ie. return a
17709 GstTagList rather than a GstStructure, which is more correct,
17710 even if one is typedef'ed to the other (#518940).
17712 2008-03-02 18:32:36 +0000 Tim-Philipp Müller <tim@centricular.net>
17714 gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
17715 Original commit message from CVS:
17716 * gst-libs/gst/rtsp/gstrtspconnection.c:
17717 Don't include unix headers unconditionally (fixes #518037).
17719 2008-03-02 18:24:37 +0000 Tim-Philipp Müller <tim@centricular.net>
17721 tests/check/libs/video.c: Add unit test that makes sure that the strides, offsets and sizes returned for the various ...
17722 Original commit message from CVS:
17723 * tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
17724 (fourcc_list_struct), (fourcc_list), (fourcc_get_size),
17725 (paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
17726 (paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
17727 (paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
17728 (paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
17729 (gst_video_format_is_packed), (video_format_is_packed):
17730 Add unit test that makes sure that the strides, offsets and
17731 sizes returned for the various YUV formats by the new video API
17732 match the old reference implementation in videotestsrc.
17734 2008-03-02 18:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
17736 gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
17737 Original commit message from CVS:
17738 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
17739 (gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
17740 (gst_video_format_is_rgb), (gst_video_format_is_yuv),
17741 (gst_video_format_has_alpha), (gst_video_format_get_row_stride),
17742 (gst_video_format_get_pixel_stride),
17743 (gst_video_format_get_component_width),
17744 (gst_video_format_get_component_height),
17745 (gst_video_format_get_component_offset), (gst_video_format_get_size):
17746 * gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
17747 (GST_VIDEO_FORMAT_Y42B):
17748 API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
17750 2008-03-02 18:07:10 +0000 Tim-Philipp Müller <tim@centricular.net>
17752 gst-libs/gst/video/video.c: YV12 is I420 with swapped components 1 and 2, so the offset of component 1 for I420 shoul...
17753 Original commit message from CVS:
17754 * gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
17755 YV12 is I420 with swapped components 1 and 2, so the offset of
17756 component 1 for I420 should be the offset for component 2 for YV12
17759 2008-02-29 21:48:00 +0000 Rene Stadler <mail@renestadler.de>
17761 sys/v4l/gstv4lelement.c: Add missing semicolon to fix indentation.
17762 Original commit message from CVS:
17763 * sys/v4l/gstv4lelement.c:
17764 Add missing semicolon to fix indentation.
17766 2008-02-29 18:44:36 +0000 Julien Moutte <julien@moutte.net>
17768 ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
17769 Original commit message from CVS:
17770 2008-02-29 Julien Moutte <julien@fluendo.com>
17771 * ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
17772 (gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
17774 if we can do SPDIF output.
17775 * ext/alsa/gstalsa.h:
17776 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
17777 (gst_alsasink_prepare), (gst_alsasink_close),
17778 (gst_alsasink_write):
17779 * ext/alsa/gstalsasink.h: Initial support for SPDIF.
17780 * gst-libs/gst/audio/gstringbuffer.c:
17781 (gst_ring_buffer_parse_caps):
17782 * gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
17784 to support AC3, EC3 and IEC958 buffers.
17786 2008-02-29 17:59:16 +0000 Tim-Philipp Müller <tim@centricular.net>
17788 gst-libs/gst/interfaces/mixer.c: De-cruft and fix message type assertions (NULL is not a really valid mixer message t...
17789 Original commit message from CVS:
17790 * gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
17791 (gst_mixer_message_parse_mute_toggled),
17792 (gst_mixer_message_parse_record_toggled),
17793 (gst_mixer_message_parse_volume_changed),
17794 (gst_mixer_message_parse_option_changed):
17795 De-cruft and fix message type assertions (NULL is not a really
17796 valid mixer message type string).
17798 2008-02-29 14:52:02 +0000 Wim Taymans <wim.taymans@gmail.com>
17800 ext/libvisual/visual.c: When negotiating, actually start from a format that we can support instead of from the too ge...
17801 Original commit message from CVS:
17802 * ext/libvisual/visual.c: (gst_vis_src_negotiate):
17803 When negotiating, actually start from a format that we can support
17804 instead of from the too generic template.
17806 2008-02-29 12:26:48 +0000 Wim Taymans <wim.taymans@gmail.com>
17808 gst/playback/gstplaybin2.c: Enable vis setting.
17809 Original commit message from CVS:
17810 * gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
17811 Enable vis setting.
17812 * gst/playback/gstplaysink.c: (gst_play_sink_init),
17813 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
17814 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
17816 Implement vis switching while playing.
17818 2008-02-29 00:04:57 +0000 David Schleef <ds@schleef.org>
17820 gst-libs/gst/riff/riff-media.c: Add Dirac mapping
17821 Original commit message from CVS:
17822 * gst-libs/gst/riff/riff-media.c: Add Dirac mapping
17824 2008-02-28 10:54:14 +0000 Peter Kjellerstedt <pkj@axis.com>
17826 gst/tcp/: Removed fdset and stress test, they are now known as GstPoll in core.
17827 Original commit message from CVS:
17828 Patch by: Peter Kjellerstedt <pkj at axis com>
17829 * gst/tcp/Makefile.am:
17830 * gst/tcp/fdsetstress.c:
17831 * gst/tcp/gstfdset.c:
17832 * gst/tcp/gstfdset.h:
17833 Removed fdset and stress test, they are now known as GstPoll in
17835 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
17836 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
17837 (gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
17838 (gst_multi_fd_sink_handle_client_write),
17839 (gst_multi_fd_sink_queue_buffer),
17840 (gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
17841 (gst_multi_fd_sink_stop):
17842 * gst/tcp/gstmultifdsink.h:
17843 * gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
17844 (gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
17845 (gst_tcp_gdp_read_caps):
17846 * gst/tcp/gsttcp.h:
17847 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
17848 (gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
17849 (gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
17850 * gst/tcp/gsttcpclientsink.h:
17851 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
17852 (gst_tcp_client_src_create), (gst_tcp_client_src_start),
17853 (gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
17854 * gst/tcp/gsttcpclientsrc.h:
17855 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
17856 (gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
17857 * gst/tcp/gsttcpserversink.h:
17858 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
17859 (gst_tcp_server_src_create), (gst_tcp_server_src_start),
17860 (gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
17861 * gst/tcp/gsttcpserversrc.h:
17862 Port to GstPoll. See #505417.
17864 2008-02-28 09:54:14 +0000 Wim Taymans <wim.taymans@gmail.com>
17867 Patch Changelog a bit to give credit and refer to the relevant bug.
17868 Original commit message from CVS:
17869 Patch Changelog a bit to give credit and refer to the
17872 2008-02-28 09:50:52 +0000 Wim Taymans <wim.taymans@gmail.com>
17874 gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection.
17875 Original commit message from CVS:
17876 * gst-libs/gst/rtsp/gstrtspconnection.c:
17877 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
17878 (gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
17879 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
17880 (gst_rtsp_connection_free), (gst_rtsp_connection_poll),
17881 (gst_rtsp_connection_flush):
17882 * gst-libs/gst/rtsp/gstrtspconnection.h:
17883 Use GstPoll for the rtsp connection.
17885 2008-02-27 12:19:31 +0000 Wim Taymans <wim.taymans@gmail.com>
17887 tests/examples/seek/seek.c: Add combo box for visualisations, populate it with a factory list of all visualisation pl...
17888 Original commit message from CVS:
17889 * tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
17890 (init_visualization_features), (vis_combo_cb), (shot_cb), (main):
17891 Add combo box for visualisations, populate it with a factory list
17892 of all visualisation plugins, configure vis plugin instance in
17895 2008-02-27 10:55:03 +0000 Wim Taymans <wim.taymans@gmail.com>
17897 tests/check/libs/rtp.c: Add check for RTP buffer defaults, padding and marker bit API.
17898 Original commit message from CVS:
17899 * tests/check/libs/rtp.c: (GST_START_TEST):
17900 Add check for RTP buffer defaults, padding and marker bit API.
17902 2008-02-27 10:42:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17904 gst-libs/gst/cdda/sha1.c: Use memcpy() instead of upcasting a byte array to long *. This fixes an unaligned memory ac...
17905 Original commit message from CVS:
17906 * gst-libs/gst/cdda/sha1.c: (sha_transform):
17907 Use memcpy() instead of upcasting a byte array to long *. This
17908 fixes an unaligned memory access, resulting in SIGBUS on IA64.
17909 This should be ported to GCheckSum once we can use GLib 2.16.
17910 Partially fixes bug #500833.
17912 2008-02-27 10:23:27 +0000 Tim-Philipp Müller <tim@centricular.net>
17914 gst-libs/gst/tag/gsttagdemux.c: Push tag event after the newsegment event. Log the pointer of the buffer we're actual...
17915 Original commit message from CVS:
17916 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
17917 Push tag event after the newsegment event. Log the pointer of
17918 the buffer we're actually going to push rather than the buffer
17919 we're feeding to _make_metadata_writable().
17921 2008-02-25 07:21:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17923 gst/typefind/gsttypefindfunctions.c: Comment smoke typefinder for now. The smokedec plugin needs one frame per buffer...
17924 Original commit message from CVS:
17925 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
17926 Comment smoke typefinder for now. The smokedec plugin needs one
17927 frame per buffer but we have no parser yet, thus it simply crashes
17928 in most situations.
17930 2008-02-25 06:48:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17932 gst/typefind/gsttypefindfunctions.c: Add typefinder for the smoke video codec. Copied from the jpeg plugin.
17933 Original commit message from CVS:
17934 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
17935 Add typefinder for the smoke video codec. Copied from the jpeg plugin.
17937 2008-02-25 06:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17939 gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
17940 Original commit message from CVS:
17941 * gst/typefind/gsttypefindfunctions.c: (mid_type_find),
17943 Add midi typefinder, copied from the timidity plugin.
17945 2008-02-23 09:51:26 +0000 Tomasz Sałaciński <tsalacinski@gmail.com>
17947 Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
17948 Original commit message from CVS:
17949 Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
17950 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
17951 * tests/check/elements/subparse.c: (test_microdvd_with_italics),
17953 Forward slashes at the beginning and end of a line also signify
17954 italics (Fixes: #518162).
17956 2008-02-22 06:38:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17958 tests/check/gst-plugins-base.supp: Add a suppression for a cached value in GIO that wasn't moved while moving gio fro...
17959 Original commit message from CVS:
17960 * tests/check/gst-plugins-base.supp:
17961 Add a suppression for a cached value in GIO that wasn't moved
17962 while moving gio from -bad to -base.
17964 2008-02-22 05:27:24 +0000 Brian Cameron <brian.cameron@sun.com>
17966 configure.ac: Don't hardcode -Wall and -Werror for configure checks, this fails with non-GCC compilers. Fixes bug #51...
17967 Original commit message from CVS:
17968 Patch by: Brian Cameron <brian dot cameron at sun dot com>
17970 Don't hardcode -Wall and -Werror for configure checks, this fails
17971 with non-GCC compilers. Fixes bug #517991.
17973 2008-02-21 08:05:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17975 gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
17976 Original commit message from CVS:
17977 * gst/audiotestsrc/gstaudiotestsrc.c:
17978 Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
17980 2008-02-20 15:37:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17982 ext/gnomevfs/gstgnomevfssink.c: Return FALSE when seeking for a new segment fails instead of silently ignoring the fa...
17983 Original commit message from CVS:
17984 * ext/gnomevfs/gstgnomevfssink.c:
17985 (gst_gnome_vfs_sink_handle_event):
17986 Return FALSE when seeking for a new segment fails instead
17987 of silently ignoring the failure and appending every buffer
17988 that comes for the new segment.
17990 2008-02-20 11:52:28 +0000 Wim Taymans <wim.taymans@gmail.com>
17992 gst/playback/gstplaysink.c: Recursively search the sink element for a last-frame property so that we can also find th...
17993 Original commit message from CVS:
17994 * gst/playback/gstplaysink.c: (find_property),
17995 (gst_play_sink_find_property), (gen_video_chain),
17996 (gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
17997 Recursively search the sink element for a last-frame property so that we
17998 can also find the property in autovideosink and friends that don't
17999 always proxy the internal sink properties.
18001 2008-02-19 20:42:09 +0000 Tim-Philipp Müller <tim@centricular.net>
18003 gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
18004 Original commit message from CVS:
18005 * gst-libs/gst/audio/multichannel.c:
18006 (GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
18007 (gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
18008 (gst_audio_set_structure_channel_positions_list),
18009 (add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
18010 (gst_audio_fixate_channel_positions):
18011 Fix confusing terminology in docs and code: structure fields are
18012 'fields' and not 'properties'.
18014 2008-02-19 20:36:58 +0000 Tim-Philipp Müller <tim@centricular.net>
18016 gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
18017 Original commit message from CVS:
18018 * gst-libs/gst/audio/multichannel.c:
18019 (gst_audio_check_channel_positions), (add_list_to_struct):
18020 Give more useful warning messages if one of the channel
18021 layout enums passed to us is invalid and if the "channels"
18022 field in the caps has a GType we don't expect.
18024 2008-02-19 20:22:09 +0000 Tim-Philipp Müller <tim@centricular.net>
18026 gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
18027 Original commit message from CVS:
18028 * gst-libs/gst/audio/multichannel.c:
18029 Fix typo in docs blurb.
18031 2008-02-19 16:16:55 +0000 Josep Torra Valles <josep@fluendo.com>
18033 gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS typefind lookup to fix typefinding on HD clips.
18034 Original commit message from CVS:
18035 2008-02-19 Julien Moutte <julien@fluendo.com>
18036 Patch by: Josep Torra Valles <josep@fluendo.com>
18037 * gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
18038 typefind lookup to fix typefinding on HD clips.
18040 2008-02-19 15:50:37 +0000 Tim-Philipp Müller <tim@centricular.net>
18042 gst/playback/gstscreenshot.*: Fix up copyright (I rewrote the GStreamer-0.10 code for this from scratch back in the d...
18043 Original commit message from CVS:
18044 * gst/playback/gstscreenshot.c:
18045 * gst/playback/gstscreenshot.h:
18046 Fix up copyright (I rewrote the GStreamer-0.10 code for
18047 this from scratch back in the days).
18049 2008-02-19 15:02:33 +0000 Wim Taymans <wim.taymans@gmail.com>
18051 gst/playback/: Add screenshot conversion code from totem.
18052 Original commit message from CVS:
18053 * gst/playback/Makefile.am:
18054 * gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
18055 (create_element), (gst_play_frame_conv_convert):
18056 * gst/playback/gstscreenshot.h:
18057 Add screenshot conversion code from totem.
18058 * gst/playback/gstplay-marshal.list:
18059 * gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
18060 (gst_play_bin_class_init), (gst_play_bin_convert_frame),
18061 (gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
18062 Implement frame property to get a color-unconverted snapshot.
18063 Implement convert-frame action signal to get a converted snapshot image.
18064 Configure connection speed in uridecodebin.
18065 Document some more properties.
18066 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
18067 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
18068 (gst_play_sink_get_last_frame):
18069 * gst/playback/gstplaysink.h:
18070 Use last-buffer property of the video sink to get a video snapshot.
18071 * tests/examples/seek/seek.c: (shot_cb), (main):
18072 Add snapshot button for playbin2 and use the frame property to save the
18073 frame as a png in the current directory.
18075 2008-02-19 11:45:56 +0000 Josep Torra Valles <josep@fluendo.com>
18077 gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
18078 Original commit message from CVS:
18079 Patch by: Josep Torra Valles <josep at fluendo dot com>
18080 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
18082 Add typefinding support for h264 elementary streams.
18085 2008-02-18 13:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18087 configure.ac: Require CVS of core for new API in collectpads.
18088 Original commit message from CVS:
18090 Require CVS of core for new API in collectpads.
18091 * gst/adder/gstadder.c:
18092 Use new API to make adder sparse stream aware.
18094 2008-02-18 11:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
18096 gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
18097 Original commit message from CVS:
18098 * gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
18100 Get the object data correct so that we can remove our channels
18102 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
18103 (gen_vis_chain), (gst_play_sink_reconfigure),
18104 (gst_play_sink_request_pad):
18105 Add option to disable async behaviour in the sinks when possible. This
18106 makes it possible to avoid an audio queue when dealing with
18108 Add option to add a queue for the audio path.
18109 * tests/examples/seek/seek.c: (clear_streams), (update_streams),
18111 Disable the vis checkbox to match the defaults of playbin2.
18112 Only get the stream info when we need to.
18114 2008-02-17 05:15:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18116 ext/gio/: Don't use async operations as they require a running main loop.
18117 Original commit message from CVS:
18118 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
18119 (gst_gio_base_sink_set_stream):
18120 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
18121 (gst_gio_base_src_set_stream):
18122 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
18123 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
18124 Don't use async operations as they require a running main loop.
18125 This makes us block again when closing streams and unable
18126 to mount the enclosing volume of an URI if it isn't yet.
18128 2008-02-15 18:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
18130 gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
18131 Original commit message from CVS:
18132 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
18133 (gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
18134 (gen_vis_chain), (gst_play_sink_reconfigure),
18135 (gst_play_sink_request_pad):
18136 Move tee in front of the audio and vis pipelines.
18137 Add queue for audio for now.
18138 Add visualisation support.
18139 * tests/examples/seek/seek.c: (main):
18140 Visualisation is by default disabled.
18142 2008-02-15 11:58:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18144 ext/gio/: Improve debugging a bit.
18145 Original commit message from CVS:
18146 * ext/gio/gstgiobasesink.c: (close_stream_cb):
18147 * ext/gio/gstgiobasesrc.c: (close_stream_cb):
18148 Improve debugging a bit.
18149 * ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
18150 * ext/gio/gstgiosink.h:
18151 * ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
18152 * ext/gio/gstgiosrc.h:
18153 Try to mount the enclosing volume of a GFile if it isn't mounted
18154 yet. This requires us to wait for an async operation to finish, done
18155 with an nested GMainLoop. Authentication is not supported yet, will
18158 2008-02-14 18:24:42 +0000 Wim Taymans <wim.taymans@gmail.com>
18160 gst/playback/: Add mute property.
18161 Original commit message from CVS:
18162 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
18163 (gst_play_bin_set_property), (gst_play_bin_get_property),
18164 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
18165 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
18166 (gst_play_sink_get_mute), (gen_audio_chain):
18167 * gst/playback/gstplaysink.h:
18169 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
18170 (gst_selector_pad_chain):
18171 * gst/playback/gststreamselector.h:
18172 Make sure we forward the event only once.
18173 * tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
18174 Add and implement the mute button for playbin2.
18176 2008-02-13 14:34:55 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
18178 ext/alsa/gstalsasink.c: Add some more debug info.
18179 Original commit message from CVS:
18180 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
18181 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
18182 Add some more debug info.
18183 Make sure we never return a negative delay. Fixes #516246.
18185 2008-02-12 20:09:07 +0000 Tim-Philipp Müller <tim@centricular.net>
18187 ext/alsa/gstalsasink.c: Revert patch that makes the sink hold the object lock when calling snd_pcm_delay(), since it ...
18188 Original commit message from CVS:
18189 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
18190 Revert patch that makes the sink hold the object lock when
18191 calling snd_pcm_delay(), since it breaks playback for me.
18193 2008-02-12 19:50:36 +0000 Julien Moutte <julien@moutte.net>
18195 tests/examples/seek/seek.c: Add some seek flags when changing rate.
18196 Original commit message from CVS:
18197 2008-02-12 Julien Moutte <julien@fluendo.com>
18198 * tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
18199 some seek flags when changing rate.
18201 2008-02-12 14:51:26 +0000 Wim Taymans <wim.taymans@gmail.com>
18203 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
18204 Original commit message from CVS:
18205 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18206 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
18207 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
18208 Fix potential leaks.
18209 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
18210 Fix leak when there is no function configured.
18212 2008-02-12 11:36:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18214 sys/v4l/v4lsrc_calls.c: Correctly chain up the finalize method.
18215 Original commit message from CVS:
18216 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
18217 (gst_v4lsrc_buffer_finalize):
18218 Correctly chain up the finalize method.
18220 2008-02-12 09:24:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18222 ext/gio/: Add documentation and example code for giostreamsink/giostreamsrc.
18223 Original commit message from CVS:
18224 * ext/gio/gstgiostreamsink.c:
18225 * ext/gio/gstgiostreamsrc.c:
18226 Add documentation and example code for giostreamsink/giostreamsrc.
18227 * tests/check/pipelines/gio.c: (GST_START_TEST):
18228 Ask the GMemoryOutputStream for the data instead of assuming that
18229 the pointer to the data stayed the same. It could've been realloc'ed.
18231 2008-02-12 08:55:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18233 ext/gio/: Make the documentation of giosink/giosrc complete, large parts are based on the gnomevfssink/gnomevfssrc docs.
18234 Original commit message from CVS:
18235 * ext/gio/gstgiosink.c:
18236 * ext/gio/gstgiosrc.c:
18237 Make the documentation of giosink/giosrc complete, large parts
18238 are based on the gnomevfssink/gnomevfssrc docs.
18240 2008-02-12 08:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18242 docs/plugins/: Add the GIO documentation again and while at that run make update.
18243 Original commit message from CVS:
18244 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
18245 * docs/plugins/gst-plugins-base-plugins-sections.txt:
18246 * docs/plugins/gst-plugins-base-plugins.args:
18247 * docs/plugins/gst-plugins-base-plugins.hierarchy:
18248 * docs/plugins/gst-plugins-base-plugins.interfaces:
18249 * docs/plugins/gst-plugins-base-plugins.prerequisites:
18250 * docs/plugins/gst-plugins-base-plugins.signals:
18251 * docs/plugins/inspect/plugin-adder.xml:
18252 * docs/plugins/inspect/plugin-audioconvert.xml:
18253 * docs/plugins/inspect/plugin-audiorate.xml:
18254 * docs/plugins/inspect/plugin-audioresample.xml:
18255 * docs/plugins/inspect/plugin-decodebin.xml:
18256 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
18257 * docs/plugins/inspect/plugin-gdp.xml:
18258 * docs/plugins/inspect/plugin-gio.xml:
18259 * docs/plugins/inspect/plugin-gnomevfs.xml:
18260 * docs/plugins/inspect/plugin-libvisual.xml:
18261 * docs/plugins/inspect/plugin-ogg.xml:
18262 * docs/plugins/inspect/plugin-pango.xml:
18263 * docs/plugins/inspect/plugin-playback.xml:
18264 * docs/plugins/inspect/plugin-queue2.xml:
18265 * docs/plugins/inspect/plugin-subparse.xml:
18266 * docs/plugins/inspect/plugin-theora.xml:
18267 * docs/plugins/inspect/plugin-uridecodebin.xml:
18268 * docs/plugins/inspect/plugin-videorate.xml:
18269 * docs/plugins/inspect/plugin-videoscale.xml:
18270 * docs/plugins/inspect/plugin-volume.xml:
18271 * docs/plugins/inspect/plugin-vorbis.xml:
18272 Add the GIO documentation again and while at that run make update.
18274 2008-02-11 20:23:44 +0000 Tim-Philipp Müller <tim@centricular.net>
18276 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...
18277 Original commit message from CVS:
18278 * ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
18279 * ext/alsa/gstalsasink.c: (set_swparams):
18280 * ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
18281 Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
18282 against libasound >= 1.0.16, since it's been deprecated in
18283 0.10.16, and alignment is always 1 then, apparently. (#512899)
18285 2008-02-11 18:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
18287 gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
18288 Original commit message from CVS:
18289 * gst/playback/gstplaybin.c: (gen_audio_element):
18290 * gst/playback/gstplaysink.c: (gen_audio_chain):
18291 Handle case where we can't create the volume element a bit
18294 2008-02-11 18:02:13 +0000 Tim-Philipp Müller <tim@centricular.net>
18296 ext/gnomevfs/: Add support for https protocol. Fixes #510229.
18297 Original commit message from CVS:
18298 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
18299 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
18300 Add support for https protocol. Fixes #510229.
18302 2008-02-11 17:03:18 +0000 Alan Peevers <peeves@pacbell.net>
18304 ext/alsa/gstalsasink.c: Take appropriate lock when calling alsa methods.
18305 Original commit message from CVS:
18306 2008-02-11 Julien Moutte <julien@fluendo.com>
18307 Patch by: Alan Peevers <peeves@pacbell.net>
18308 * ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
18309 lock when calling alsa methods.
18311 2008-02-11 13:03:13 +0000 Tim-Philipp Müller <tim@centricular.net>
18313 gst/typefind/gsttypefindfunctions.c: Bump rank of jpeg and png typefinders, which will return maximum probability in ...
18314 Original commit message from CVS:
18315 * gst/typefind/gsttypefindfunctions.c:
18316 Bump rank of jpeg and png typefinders, which will return maximum
18317 probability in the most common cases (thus short-circuiting more
18318 expensive typefinders like the mp3 one for these two quite common
18321 2008-02-11 09:48:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18323 ext/theora/theoraparse.c: Fix long description of the theora parser to be more verbose than just the type name.
18324 Original commit message from CVS:
18325 * ext/theora/theoraparse.c:
18326 Fix long description of the theora parser to be more verbose than just
18329 2008-02-11 06:47:50 +0000 Branko Čibej <brane@xbc.nu>
18331 sys/xvimage/xvimagesink.c: Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
18332 Original commit message from CVS:
18333 Patch by: Branko Čibej <brane at xbc dot nu>
18334 * sys/xvimage/xvimagesink.c:
18335 Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
18338 2008-02-09 10:41:36 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
18340 gst/playback/gstplaybasebin.c: Set is_dynamic as True if there are elements with both request and sometimes src pad t...
18341 Original commit message from CVS:
18342 * gst/playback/gstplaybasebin.c:
18343 Set is_dynamic as True if there are elements with both request
18344 and sometimes src pad templates instead of breaking out when it
18345 finds the first pad template that is a src.
18347 2008-02-08 18:17:51 +0000 Wim Taymans <wim.taymans@gmail.com>
18349 tests/examples/seek/seek.c: Add some stream switching and volume gui for playbin2.
18350 Original commit message from CVS:
18351 * tests/examples/seek/seek.c: (stop_cb), (clear_streams),
18352 (update_streams), (video_combo_cb), (audio_combo_cb),
18353 (text_combo_cb), (volume_spinbutton_changed_cb), (main):
18354 Add some stream switching and volume gui for playbin2.
18356 2008-02-08 17:47:37 +0000 Wim Taymans <wim.taymans@gmail.com>
18358 gst/playback/gstplay-marshal.list: Added marshal for streamselector Tags.
18359 Original commit message from CVS:
18360 * gst/playback/gstplay-marshal.list:
18361 Added marshal for streamselector Tags.
18362 * gst/playback/gstplaybasebin.c: (set_active_source):
18363 Streamselector now selects pads based on the pad object instead of its
18365 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
18366 (init_group), (gst_play_bin_init), (get_group), (get_tags),
18367 (gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
18368 (gst_play_bin_get_text_tags),
18369 (gst_play_bin_set_current_video_stream),
18370 (gst_play_bin_set_current_audio_stream),
18371 (gst_play_bin_set_current_text_stream),
18372 (gst_play_bin_set_property), (gst_play_bin_get_property),
18373 (pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
18374 Remove option to mute streams with the current-a/v/t property, we have
18375 this functionality in the flags.
18376 Add signals to notify when the number of A/V/T channels changed.
18377 Add action signals to get tags for the A/V/T streams.
18378 Implement setting the current A/V/T stream.
18379 Rearrange some things to simplify stream selection.
18381 * gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
18382 (gst_play_sink_get_volume), (gst_play_sink_set_property),
18383 (gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
18384 (activate_vis), (gst_play_sink_reconfigure):
18385 * gst/playback/gstplaysink.h:
18386 Add and implement volume setting methods.
18387 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
18388 (gst_selector_pad_finalize), (gst_selector_pad_get_property),
18389 (gst_selector_pad_event), (gst_stream_selector_class_init),
18390 (gst_stream_selector_init), (gst_stream_selector_finalize),
18391 (gst_stream_selector_set_property),
18392 (gst_stream_selector_get_property),
18393 (gst_stream_selector_get_linked_pad),
18394 (gst_stream_selector_request_new_pad):
18395 * gst/playback/gststreamselector.h:
18396 Add pad properties for tags and status of pads.
18398 Make active pad selection based on pad object instead of name.
18400 2008-02-08 16:10:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18402 configure.ac: Revert last change as we now check in gtk-doc.m4 for sed.
18403 Original commit message from CVS:
18405 Revert last change as we now check in gtk-doc.m4 for sed.
18407 2008-02-08 14:54:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18409 configure.ac: Find and subst SED when building the docs.
18410 Original commit message from CVS:
18412 Find and subst SED when building the docs.
18414 2008-02-08 14:34:41 +0000 Julien Moutte <julien@moutte.net>
18416 tests/examples/seek/seek.c: Make sure bus signals are reconnected when pressing STOP and then PLAY again for a parse ...
18417 Original commit message from CVS:
18418 2008-02-08 Julien Moutte <julien@fluendo.com>
18419 * tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
18420 (main): Make sure bus signals are reconnected when pressing STOP
18421 and then PLAY again for a parse launch pipeline. Fix a ref leak
18423 * win32/common/config.h: Updated.
18425 2008-02-08 00:57:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18427 configure.ac: Make DISABLE_DEPRECATED defined *only* during CVS, not during pre-releases or releases.
18428 Original commit message from CVS:
18430 Make DISABLE_DEPRECATED defined *only* during CVS, not during
18431 pre-releases or releases.
18433 2008-02-08 00:45:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18435 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is reporting
18436 Original commit message from CVS:
18438 * ext/gio/Makefile.am:
18439 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
18442 2008-02-07 23:40:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18444 docs/plugins/Makefile.am: Add the headers which need scanning for the GIO plugin. The rest of the docs still need mig...
18445 Original commit message from CVS:
18446 * docs/plugins/Makefile.am:
18447 Add the headers which need scanning for the GIO plugin. The rest of
18448 the docs still need migrating.
18450 2008-02-07 23:22:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18452 Add gio in a few more places.
18453 Original commit message from CVS:
18455 * tests/check/Makefile.am:
18456 * tests/check/pipelines/.cvsignore:
18457 Add gio in a few more places.
18459 2008-02-07 23:18:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18461 Move gio plugin from -bad and mark as experimental.
18462 Original commit message from CVS:
18465 * tests/check/Makefile.am:
18466 Move gio plugin from -bad and mark as experimental.
18468 2008-02-07 22:39:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18470 gst-libs/gst/interfaces/: Comment out a couple of other things which break the build when
18471 Original commit message from CVS:
18472 * gst-libs/gst/interfaces/mixeroptions.c:
18473 * gst-libs/gst/interfaces/mixertrack.c:
18474 Comment out a couple of other things which break the build when
18475 GST_DISABLE_DEPRECATED isn't on but -Werror is.
18477 2008-02-07 18:28:29 +0000 Tim-Philipp Müller <tim@centricular.net>
18479 docs/libs/gst-plugins-base-libs-sections.txt: Fix pbutils header.
18480 Original commit message from CVS:
18481 * docs/libs/gst-plugins-base-libs-sections.txt:
18482 Fix pbutils header.
18484 2008-02-07 18:07:41 +0000 Christian Schaller <uraeus@gnome.org>
18486 * gst-plugins-base.spec.in:
18487 commit spec file update which includes all the split .pc files
18488 Original commit message from CVS:
18489 commit spec file update which includes all the split .pc files
18491 2008-02-07 12:17:49 +0000 Wim Taymans <wim.taymans@gmail.com>
18493 gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
18494 Original commit message from CVS:
18495 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
18496 Fix compiler warning.
18498 2008-02-07 11:00:45 +0000 Peter Kjellerstedt <pkj@axis.com>
18500 gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
18501 Original commit message from CVS:
18502 Patch by: Peter Kjellerstedt <pkj at axis com>
18503 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
18504 Clear the addrinfo struct using memset. Fixes #514937.
18506 2008-02-06 15:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
18508 gst/tcp/gstfdset.h: Remove unused field to same some memory.
18509 Original commit message from CVS:
18510 * gst/tcp/gstfdset.h:
18511 Remove unused field to same some memory.
18512 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
18513 Mark action signals as such.
18515 2008-02-06 13:35:58 +0000 Michael Smith <msmith@xiph.org>
18517 ext/theora/theoradec.c: Increment granulepos for new-bitstream versions appropriately.
18518 Original commit message from CVS:
18519 * ext/theora/theoradec.c: (_theora_granule_frame),
18521 Increment granulepos for new-bitstream versions appropriately.
18524 2008-02-04 11:51:31 +0000 Wim Taymans <wim.taymans@gmail.com>
18526 tests/examples/seek/seek.c: Remove obsolete stream_time reset after flushing seek, core does that automatically now.
18527 Original commit message from CVS:
18528 * tests/examples/seek/seek.c: (do_seek),
18529 (rate_spinbutton_changed_cb), (update_streams), (main):
18530 Remove obsolete stream_time reset after flushing seek, core does that
18532 Improve accuracy of speed spinbutton.
18533 Only do playbin2 stuff when we actually use it.
18535 2008-02-02 17:29:32 +0000 Tim-Philipp Müller <tim@centricular.net>
18537 tests/check/Makefile.am: Revert previous change of the test environment's GST_PLUGIN_PATH.
18538 Original commit message from CVS:
18539 * tests/check/Makefile.am:
18540 Revert previous change of the test environment's GST_PLUGIN_PATH.
18541 The problem is not with the plugins, but with element factories
18542 and only occurs if elements are split out from existing plugins
18543 or if plugins change name (see #512740).
18545 2008-02-02 15:32:23 +0000 Tim-Philipp Müller <tim@centricular.net>
18547 tests/check/Makefile.am: Fix the tests environment's GST_PLUGIN_PATH: we want the directory with the core's plugins f...
18548 Original commit message from CVS:
18549 * tests/check/Makefile.am:
18550 Fix the tests environment's GST_PLUGIN_PATH: we want the directory
18551 with the core's plugins first and our local build directories last,
18552 since we might be building against an installed core, and that
18553 core's plugin directory may contain older or other versions of
18554 our own -base plugins, but we really do want to test our local
18555 ones (if there are multiple plugins or element factories with the
18556 same name, those inspected last will trump those read in earlier).
18557 Fixes #512740 for the most part.
18559 2008-02-02 07:13:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18561 Use gmtime_r if available as gmtime is not MT-safe.
18562 Original commit message from CVS:
18564 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
18565 Use gmtime_r if available as gmtime is not MT-safe.
18568 2008-02-02 06:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18570 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
18571 Original commit message from CVS:
18572 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
18573 Cast glong to time_t as time_t might have a different type on
18574 other platforms, like FreeBSD, and we get a compiler warning
18575 otherwise. Fixes bug #511825.
18577 2008-02-01 16:44:21 +0000 Wim Taymans <wim.taymans@gmail.com>
18579 gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
18580 Original commit message from CVS:
18581 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
18582 (get_group), (get_n_pads), (gst_play_bin_get_property),
18583 (pad_added_cb), (no_more_pads_cb), (perform_eos),
18584 (autoplug_select_cb), (deactivate_group):
18585 Remove stream-info, we going for something easier.
18586 Refactor getting the current group.
18587 Implement getting the number of audio/video/text streams.
18588 * gst/playback/gststreamselector.c:
18589 (gst_stream_selector_class_init), (gst_stream_selector_init),
18590 (gst_stream_selector_get_property),
18591 (gst_stream_selector_request_new_pad),
18592 (gst_stream_selector_release_pad):
18593 * gst/playback/gststreamselector.h:
18594 Add property for number of pads.
18595 * tests/examples/seek/seek.c: (set_scale), (update_flag),
18596 (vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
18597 (text_toggle_cb), (update_streams), (msg_async_done),
18598 (msg_state_changed), (main):
18599 Block slider callback when updating the slider position.
18600 Add gui elements for controlling playbin2.
18601 Add callback for async_done that updates position/duration.
18603 2008-02-01 12:56:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18605 docs/plugins/: First round of plugin docs cleansups.
18606 Original commit message from CVS:
18607 * docs/plugins/Makefile.am:
18608 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
18609 * docs/plugins/gst-plugins-base-plugins-sections.txt:
18610 * docs/plugins/gst-plugins-base-plugins.hierarchy:
18611 * docs/plugins/gst-plugins-base-plugins.interfaces:
18612 * docs/plugins/gst-plugins-base-plugins.prerequisites:
18613 First round of plugin docs cleansups.
18614 * docs/plugins/inspect/plugin-adder.xml:
18615 * docs/plugins/inspect/plugin-alsa.xml:
18616 * docs/plugins/inspect/plugin-audioconvert.xml:
18617 * docs/plugins/inspect/plugin-audiorate.xml:
18618 * docs/plugins/inspect/plugin-audioresample.xml:
18619 * docs/plugins/inspect/plugin-audiotestsrc.xml:
18620 * docs/plugins/inspect/plugin-cdparanoia.xml:
18621 * docs/plugins/inspect/plugin-decodebin.xml:
18622 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
18623 * docs/plugins/inspect/plugin-gdp.xml:
18624 * docs/plugins/inspect/plugin-gnomevfs.xml:
18625 * docs/plugins/inspect/plugin-libvisual.xml:
18626 * docs/plugins/inspect/plugin-ogg.xml:
18627 * docs/plugins/inspect/plugin-pango.xml:
18628 * docs/plugins/inspect/plugin-subparse.xml:
18629 * docs/plugins/inspect/plugin-tcp.xml:
18630 * docs/plugins/inspect/plugin-theora.xml:
18631 * docs/plugins/inspect/plugin-typefindfunctions.xml:
18632 * docs/plugins/inspect/plugin-video4linux.xml:
18633 * docs/plugins/inspect/plugin-videorate.xml:
18634 * docs/plugins/inspect/plugin-videoscale.xml:
18635 * docs/plugins/inspect/plugin-videotestsrc.xml:
18636 * docs/plugins/inspect/plugin-volume.xml:
18637 * docs/plugins/inspect/plugin-vorbis.xml:
18638 * docs/plugins/inspect/plugin-ximagesink.xml:
18639 * docs/plugins/inspect/plugin-xvimagesink.xml:
18641 * ext/ogg/Makefile.am:
18642 * ext/ogg/gstoggmux.c:
18643 * ext/ogg/gstoggmux.h:
18644 Add header for oggmux. the c-file needs a doc blob still.
18646 2008-02-01 11:09:16 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
18648 Add gst_rtp_buffer_set_extension_data()
18649 Original commit message from CVS:
18650 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
18651 * gst-libs/gst/rtp/gstrtpbuffer.c:
18652 (gst_rtp_buffer_set_extension_data):
18653 * gst-libs/gst/rtp/gstrtpbuffer.h:
18654 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
18655 Add gst_rtp_buffer_set_extension_data()
18656 Add a unit test for this addition. Fixes #511478.
18657 API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
18659 2008-01-31 17:18:46 +0000 Wim Taymans <wim.taymans@gmail.com>
18661 gst-libs/gst/app/gstappsink.c: Really clean up the queue instead of just unreffing all buffers in it.
18662 Original commit message from CVS:
18663 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
18664 Really clean up the queue instead of just unreffing all buffers
18666 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
18667 (gst_app_src_class_init), (gst_app_src_init),
18668 (gst_app_src_dispose), (gst_app_src_finalize):
18669 Fix dispose/finalize.
18671 2008-01-30 15:34:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18673 ext/gio/: Use async variants of the close stream functions to prevent blocking for a long time there and add some mor...
18674 Original commit message from CVS:
18675 * ext/gio/gstgiobasesink.c: (close_stream_cb),
18676 (gst_gio_base_sink_stop), (gst_gio_base_sink_event),
18677 (gst_gio_base_sink_render), (gst_gio_base_sink_set_stream):
18678 * ext/gio/gstgiobasesrc.c: (close_stream_cb),
18679 (gst_gio_base_src_stop), (gst_gio_base_src_create),
18680 (gst_gio_base_src_set_stream):
18681 Use async variants of the close stream functions to prevent blocking
18682 for a long time there and add some more sanity checks for a correct
18685 2008-01-30 14:42:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18687 configure.ac: Back to CVS
18688 Original commit message from CVS:
18692 === release 0.10.17 ===
18694 2008-01-30 14:19:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18700 * docs/plugins/gst-plugins-base-plugins.hierarchy:
18701 * docs/plugins/inspect/plugin-adder.xml:
18702 * docs/plugins/inspect/plugin-alsa.xml:
18703 * docs/plugins/inspect/plugin-audioconvert.xml:
18704 * docs/plugins/inspect/plugin-audiorate.xml:
18705 * docs/plugins/inspect/plugin-audioresample.xml:
18706 * docs/plugins/inspect/plugin-audiotestsrc.xml:
18707 * docs/plugins/inspect/plugin-cdparanoia.xml:
18708 * docs/plugins/inspect/plugin-decodebin.xml:
18709 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
18710 * docs/plugins/inspect/plugin-gdp.xml:
18711 * docs/plugins/inspect/plugin-gnomevfs.xml:
18712 * docs/plugins/inspect/plugin-libvisual.xml:
18713 * docs/plugins/inspect/plugin-ogg.xml:
18714 * docs/plugins/inspect/plugin-pango.xml:
18715 * docs/plugins/inspect/plugin-subparse.xml:
18716 * docs/plugins/inspect/plugin-tcp.xml:
18717 * docs/plugins/inspect/plugin-theora.xml:
18718 * docs/plugins/inspect/plugin-typefindfunctions.xml:
18719 * docs/plugins/inspect/plugin-video4linux.xml:
18720 * docs/plugins/inspect/plugin-videorate.xml:
18721 * docs/plugins/inspect/plugin-videoscale.xml:
18722 * docs/plugins/inspect/plugin-videotestsrc.xml:
18723 * docs/plugins/inspect/plugin-volume.xml:
18724 * docs/plugins/inspect/plugin-vorbis.xml:
18725 * docs/plugins/inspect/plugin-ximagesink.xml:
18726 * docs/plugins/inspect/plugin-xvimagesink.xml:
18727 * gst-plugins-base.doap:
18728 * win32/common/config.h:
18730 Original commit message from CVS:
18733 2008-01-30 13:45:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18735 gst-libs/gst/interfaces/: Also remove the conditional registration of the signals that disappeared with the ABI chang...
18736 Original commit message from CVS:
18737 * gst-libs/gst/interfaces/mixeroptions.c:
18738 * gst-libs/gst/interfaces/mixertrack.c:
18739 Also remove the conditional registration of the signals
18740 that disappeared with the ABI change in 0.10.14
18742 2008-01-30 12:28:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18744 gst-libs/gst/rtsp/gstrtspconnection.c: Revert patch to gstrtspconnection.c for brown paper bag release of -base. Re-o...
18745 Original commit message from CVS:
18746 * gst-libs/gst/rtsp/gstrtspconnection.c:
18747 Revert patch to gstrtspconnection.c for brown paper bag
18748 release of -base. Re-opens: #511825
18750 2008-01-30 12:20:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18752 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
18753 Original commit message from CVS:
18754 * gst-libs/gst/interfaces/mixeroptions.h:
18755 * gst-libs/gst/interfaces/mixertrack.h:
18756 Change the way these deprecated function pointers are removed
18757 so that the compiled ABI is unconditionally smaller. This
18758 sets in stone an ABI break that actually occurred when the
18759 things were deprecated in 0.10.14, which seems to be the best
18760 fix as the only known users are oss-mixer and sunaudio-mixer in
18764 2008-01-30 12:19:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18766 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
18767 Original commit message from CVS:
18768 * gst-libs/gst/interfaces/mixeroptions.h:
18769 * gst-libs/gst/interfaces/mixertrack.h:
18770 Change the way these deprecated function pointers are removed
18771 so that the compiled ABI is unconditionally smaller. This
18772 sets in stone an ABI break that actually occurred when the
18773 things were deprecated in 0.10.14, which seems to be the best
18774 fix as the only known users are oss-mixer and sunaudio-mixer in
18777 2008-01-30 11:43:53 +0000 Tim-Philipp Müller <tim@centricular.net>
18779 win32/common/libgstpbutils.def: Export the two new _get_type() functions which are needed by the python bindings.
18780 Original commit message from CVS:
18781 * win32/common/libgstpbutils.def:
18782 Export the two new _get_type() functions which are needed
18783 by the python bindings.
18785 2008-01-29 09:59:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18787 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
18788 Original commit message from CVS:
18789 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
18790 Cast glong to time_t as time_t might have a different type on
18791 other platforms, like FreeBSD, and we get a compiler warning
18792 otherwise. Fixes bug #511825.
18794 2008-01-29 09:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18796 gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
18797 Original commit message from CVS:
18798 * gst-libs/gst/audio/gstaudiofilter.c:
18799 (gst_audio_filter_class_init):
18800 Initialize the GstRingerBuffer class to get it's debug category
18801 initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
18802 category and otherwise we get some g_critical(). Fixes bug #512334.
18804 2008-01-28 23:35:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18806 configure.ac: Back to CVS
18807 Original commit message from CVS:
18811 === release 0.10.16 ===
18813 2008-01-28 23:31:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18819 * docs/plugins/gst-plugins-base-plugins.args:
18820 * docs/plugins/gst-plugins-base-plugins.hierarchy:
18821 * docs/plugins/gst-plugins-base-plugins.interfaces:
18822 * docs/plugins/gst-plugins-base-plugins.prerequisites:
18823 * docs/plugins/gst-plugins-base-plugins.signals:
18824 * docs/plugins/inspect/plugin-adder.xml:
18825 * docs/plugins/inspect/plugin-alsa.xml:
18826 * docs/plugins/inspect/plugin-audioconvert.xml:
18827 * docs/plugins/inspect/plugin-audiorate.xml:
18828 * docs/plugins/inspect/plugin-audioresample.xml:
18829 * docs/plugins/inspect/plugin-audiotestsrc.xml:
18830 * docs/plugins/inspect/plugin-cdparanoia.xml:
18831 * docs/plugins/inspect/plugin-decodebin.xml:
18832 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
18833 * docs/plugins/inspect/plugin-gdp.xml:
18834 * docs/plugins/inspect/plugin-gnomevfs.xml:
18835 * docs/plugins/inspect/plugin-libvisual.xml:
18836 * docs/plugins/inspect/plugin-ogg.xml:
18837 * docs/plugins/inspect/plugin-pango.xml:
18838 * docs/plugins/inspect/plugin-subparse.xml:
18839 * docs/plugins/inspect/plugin-tcp.xml:
18840 * docs/plugins/inspect/plugin-theora.xml:
18841 * docs/plugins/inspect/plugin-typefindfunctions.xml:
18842 * docs/plugins/inspect/plugin-video4linux.xml:
18843 * docs/plugins/inspect/plugin-videorate.xml:
18844 * docs/plugins/inspect/plugin-videoscale.xml:
18845 * docs/plugins/inspect/plugin-videotestsrc.xml:
18846 * docs/plugins/inspect/plugin-volume.xml:
18847 * docs/plugins/inspect/plugin-vorbis.xml:
18848 * docs/plugins/inspect/plugin-ximagesink.xml:
18849 * docs/plugins/inspect/plugin-xvimagesink.xml:
18850 * gst-plugins-base.doap:
18851 * win32/common/config.h:
18853 Original commit message from CVS:
18856 2008-01-28 22:15:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18882 Original commit message from CVS:
18885 2008-01-22 15:37:49 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
18887 gst-libs/gst/rtp/gstrtpbuffer.c: Fix typos and wrong extension check. Fixes #511274.
18888 Original commit message from CVS:
18889 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
18890 * gst-libs/gst/rtp/gstrtpbuffer.c:
18891 (gst_rtp_buffer_get_extension_data):
18892 Fix typos and wrong extension check. Fixes #511274.
18894 2008-01-18 00:03:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18896 po/sk.po: Oops - add new sk.po mentioned in the LINGUAS I just committed
18897 Original commit message from CVS:
18899 Oops - add new sk.po mentioned in the LINGUAS I just committed
18901 2008-01-17 22:31:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18903 po/LINGUAS: Add ca translation to the disted list.
18904 Original commit message from CVS:
18906 Add ca translation to the disted list.
18907 * win32/vs6/libgstsdp.dsp:
18908 Convert line endings to CRLF
18910 2008-01-17 21:58:53 +0000 Sébastien Moutte <sebastien@moutte.net>
18912 win32/MANIFEST: Add win32/vs6/libgstrtsp.dsp to MANIFEST
18913 Original commit message from CVS:
18915 Add win32/vs6/libgstrtsp.dsp to MANIFEST
18917 2008-01-16 05:40:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18919 Update for API changes in GIO and require GIO 2.15.2 for this.
18920 Original commit message from CVS:
18922 * tests/check/pipelines/gio.c: (GST_START_TEST):
18923 Update for API changes in GIO and require GIO 2.15.2 for this.
18925 2008-01-14 22:20:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18927 win32/common/: Add new API declarations
18928 Original commit message from CVS:
18929 * win32/common/libgstsdp.def:
18930 * win32/common/libgstvideo.def:
18931 Add new API declarations
18933 2008-01-14 17:00:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18935 ext/theora/: Take a 2nd stab at handling libtheora granulepos changes in the decoder and parser by inspecting the bit...
18936 Original commit message from CVS:
18937 * ext/theora/gsttheoradec.h:
18938 * ext/theora/gsttheoraparse.h:
18939 * ext/theora/theoradec.c:
18940 * ext/theora/theoraparse.c:
18941 Take a 2nd stab at handling libtheora granulepos changes in the decoder
18942 and parser by inspecting the bitstream version of the incoming data.
18944 2008-01-14 13:11:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18946 Provide one pkg-config file for every gst-plugins-base library.
18947 Original commit message from CVS:
18949 * pkgconfig/Makefile.am:
18950 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
18951 * pkgconfig/gstreamer-audio.pc.in:
18952 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
18953 * pkgconfig/gstreamer-cdda.pc.in:
18954 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
18955 * pkgconfig/gstreamer-fft.pc.in:
18956 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
18957 * pkgconfig/gstreamer-floatcast.pc.in:
18958 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
18959 * pkgconfig/gstreamer-interfaces.pc.in:
18960 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
18961 * pkgconfig/gstreamer-netbuffer.pc.in:
18962 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
18963 * pkgconfig/gstreamer-pbutils.pc.in:
18964 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
18965 * pkgconfig/gstreamer-riff.pc.in:
18966 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
18967 * pkgconfig/gstreamer-rtp.pc.in:
18968 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
18969 * pkgconfig/gstreamer-rtsp.pc.in:
18970 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
18971 * pkgconfig/gstreamer-sdp.pc.in:
18972 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
18973 * pkgconfig/gstreamer-tag.pc.in:
18974 * pkgconfig/gstreamer-video-uninstalled.pc.in:
18975 * pkgconfig/gstreamer-video.pc.in:
18976 Provide one pkg-config file for every gst-plugins-base library.
18977 This makes linking to those libraries much more intuitive and
18978 provides standard pkg-config behaviour for them. Fixes bug #499697.
18980 2008-01-14 01:19:34 +0000 David Schleef <ds@schleef.org>
18982 gst/videoscale/vs_4tap.c: Fix valgrind error on 4tap scaling method.
18983 Original commit message from CVS:
18984 * gst/videoscale/vs_4tap.c:
18985 Fix valgrind error on 4tap scaling method.
18987 2008-01-13 21:40:45 +0000 Sébastien Moutte <sebastien@moutte.net>
18989 gst-libs/gst/sdp/gstsdpmessage.c: Include Winsock2.h for VS6 and use a different way initialize hints structure so it...
18990 Original commit message from CVS:
18991 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
18992 Include Winsock2.h for VS6 and use a different way initialize
18993 hints structure so it can build with VS6.
18995 * win32/vs6/libgstsdp.dsp:
18996 * win32/common/libgstsdp.def:
18997 Add new files for libgstsdp.
18998 * win32/vs6/grammar.dsp:
18999 Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
19000 * win32/vs6/gst_plugins_base.dsw:
19001 * win32/vs6/libgstdecodebin.dsp:
19002 * win32/vs6/libgstdecodebin2.dsp:
19003 * win32/vs6/libgstplaybin.dsp:
19004 * win32/vs6/libgstvolume.dsp:
19005 Add new dependencies to the link list.
19007 2008-01-13 17:24:49 +0000 Julien Moutte <julien@moutte.net>
19009 win32/common/: Update/Add generated files in the win32 build directory.
19010 Original commit message from CVS:
19011 2008-01-13 Julien Moutte <julien@fluendo.com>
19012 * win32/common/config.h:
19013 * win32/common/gstrtsp-enumtypes.c: (gst_rtsp_result_get_type),
19014 (gst_rtsp_event_get_type), (gst_rtsp_family_get_type),
19015 (gst_rtsp_state_get_type), (gst_rtsp_version_get_type),
19016 (gst_rtsp_method_get_type), (gst_rtsp_auth_method_get_type),
19017 (gst_rtsp_header_field_get_type),
19018 (gst_rtsp_status_code_get_type):
19019 * win32/common/interfaces-enumtypes.c:
19020 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
19021 (gst_mixer_message_type_get_type), (gst_mixer_flags_get_type),
19022 (gst_mixer_track_flags_get_type),
19023 (gst_tuner_channel_flags_get_type):
19024 * win32/common/multichannel-enumtypes.c:
19025 (gst_audio_channel_position_get_type):
19026 * win32/common/pbutils-enumtypes.c:
19027 (gst_install_plugins_return_get_type):
19028 * win32/common/pbutils-enumtypes.h: Update/Add generated files
19029 in the win32 build directory.
19031 2008-01-12 23:24:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19033 tests/check/Makefile.am: Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
19034 Original commit message from CVS:
19035 * tests/check/Makefile.am:
19036 Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
19037 * tests/check/elements/audiorate.c: (do_perfect_stream_test):
19038 * tests/check/elements/playbin.c:
19039 * tests/check/libs/mixer.c: (test_element_interface_supported),
19040 (gst_implements_interface_init):
19041 * tests/check/libs/rtp.c: (GST_START_TEST):
19042 Fix various assignment type mismatches.
19044 2008-01-12 23:08:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19046 Add test to see if hstrerror is available or if we need libresolv (Solaris) for it, then use it in libgstrtsp.
19047 Original commit message from CVS:
19049 * gst-libs/gst/rtsp/Makefile.am:
19050 Add test to see if hstrerror is available or if we need libresolv
19051 (Solaris) for it, then use it in libgstrtsp.
19053 2008-01-12 14:54:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19055 gst-libs/gst/tag/Makefile.am: Fix include path order
19056 Original commit message from CVS:
19057 * gst-libs/gst/tag/Makefile.am:
19058 Fix include path order
19060 2008-01-11 17:15:23 +0000 Tim-Philipp Müller <tim@centricular.net>
19062 * gst-libs/gst/pbutils/.gitignore:
19063 Ignore more and make buildbot happy
19064 Original commit message from CVS:
19065 Ignore more and make buildbot happy
19067 2008-01-11 16:18:10 +0000 Edward Hervey <bilboed@bilboed.com>
19069 gst-libs/gst/pbutils/install-plugins.*: Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping for bi...
19070 Original commit message from CVS:
19071 * gst-libs/gst/pbutils/install-plugins.c:
19072 (gst_install_plugins_context_copy),
19073 (gst_install_plugins_context_get_type):
19074 * gst-libs/gst/pbutils/install-plugins.h:
19075 Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
19078 2008-01-11 15:48:11 +0000 Michael Smith <msmith@xiph.org>
19080 ext/theora/theoradec.c: Adapt for post-alpha meaning of granulepos, when we have a newer version of libtheora.
19081 Original commit message from CVS:
19082 * ext/theora/theoradec.c: (gst_theora_dec_class_init),
19083 (_theora_granule_frame), (_theora_granule_start_time),
19084 (theora_dec_sink_convert), (theora_dec_decode_buffer):
19085 Adapt for post-alpha meaning of granulepos, when we
19086 have a newer version of libtheora.
19087 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
19088 (theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
19089 (theora_enc_is_discontinuous), (theora_enc_chain):
19091 * tests/check/Makefile.am:
19092 Link libtheora into theoraenc test so we can check which version of
19093 libtheora we're testing against.
19094 * tests/check/pipelines/theoraenc.c: (check_libtheora),
19095 (check_buffer_granulepos),
19096 (check_buffer_granulepos_from_starttime), (GST_START_TEST),
19098 Adapt tests to check the values that are now defined for theora; make
19099 the tests backwards-adapt the passed values if we're running against an
19103 2008-01-10 17:55:53 +0000 Tim-Philipp Müller <tim@centricular.net>
19105 gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
19106 Original commit message from CVS:
19107 * gst-libs/gst/audio/gstbaseaudiosink.c:
19108 (gst_base_audio_sink_class_init):
19109 * gst-libs/gst/audio/gstbaseaudiosrc.c:
19110 (gst_base_audio_src_class_init):
19111 Ref audio clock class from a thread-safe context to make sure
19112 we're not bit by GObjects lack of thread-safety here (#349410),
19113 however unlikely that may be in practice.
19115 2008-01-10 12:22:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19117 autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We...
19118 Original commit message from CVS:
19120 Add -Wno-portability to the automake parameters to stop warnings
19121 about GNU make extensions being used. We require GNU make in almost
19122 every Makefile anyway.
19124 Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
19125 at the same time is required for per target flags.
19127 2008-01-08 21:10:02 +0000 Tim-Philipp Müller <tim@centricular.net>
19129 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...
19130 Original commit message from CVS:
19131 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
19132 Post an error message if we can't pull as many bytes as we need
19133 for the tag. This makes sure the user gets to see a proper error
19134 message if a file with a partial ID3 tag is fed to decodebin, and
19135 not a 'no ID3 tag demuxer' error, which would be confusing
19138 2008-01-08 20:59:20 +0000 Tim-Philipp Müller <tim@centricular.net>
19140 gst-libs/gst/pbutils/descriptions.c: Add description strings for ID3, APE, and ICY tags.
19141 Original commit message from CVS:
19142 * gst-libs/gst/pbutils/descriptions.c: (formats):
19143 Add description strings for ID3, APE, and ICY tags.
19145 2008-01-08 20:48:00 +0000 Tim-Philipp Müller <tim@centricular.net>
19147 gst/playback/gstdecodebin.c: Make sure we error out correctly if we can't activate one of the elements we've added. ...
19148 Original commit message from CVS:
19149 * gst/playback/gstdecodebin.c: (try_to_link_1):
19150 Make sure we error out correctly if we can't activate one of
19151 the elements we've added. Fixes #508138.
19153 2008-01-07 13:59:43 +0000 Bastien Nocera <hadess@hadess.net>
19155 ext/alsa/gstalsamixer.c: Use snd_mixer_selem_set_{playback|capture}_volume_all() if the volume is the same for all ch...
19156 Original commit message from CVS:
19157 Patch by: Bastien Nocera <hadess at hadess net>
19158 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
19159 (check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
19160 Use snd_mixer_selem_set_{playback|capture}_volume_all() if
19161 the volume is the same for all channels. This works around
19162 some problem in alsa that leaves us with inconsistent state
19163 for some reason (#486840).
19165 2008-01-07 13:19:50 +0000 Jerone Young <jerone@gmail.com>
19167 ext/alsa/gstalsamixer.c: If there's no mixer track by the name of 'Master' or 'Front', check if there's one called 'P...
19168 Original commit message from CVS:
19169 Patch by: Jerone Young <jerone at gmail com>
19170 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
19171 If there's no mixer track by the name of 'Master' or 'Front',
19172 check if there's one called 'PCM' before trying the generic
19173 fallback logic (fixes #506928, where we pick 'Mic' as master
19174 track for the AD1984 card in a Thinkpad T61/X61 laptop).
19176 2008-01-07 11:40:04 +0000 Wim Taymans <wim.taymans@gmail.com>
19178 gst/playback/gstplay-enum.*: Add enums for configuration flags.
19179 Original commit message from CVS:
19180 * gst/playback/gstplay-enum.c:
19181 (register_gst_autoplug_select_result),
19182 (gst_autoplug_select_result_get_type), (register_gst_play_flags),
19183 (gst_play_flags_get_type):
19184 * gst/playback/gstplay-enum.h:
19185 Add enums for configuration flags.
19186 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
19187 (init_group), (gst_play_bin_init), (gst_play_bin_set_property),
19188 (gst_play_bin_get_property), (no_more_pads_cb),
19189 (autoplug_select_cb), (gst_play_bin_change_state):
19190 Merge mode with flags.
19191 Add more property getters/setters, defaults and docs.
19192 Add properties to get number of audio/video/text streams.
19193 Create sink object in _init so that we can always rely on it being
19195 * gst/playback/gstplaysink.c: (gst_play_sink_init),
19196 (gen_video_chain), (gen_audio_chain), (gen_vis_chain),
19197 (activate_vis), (gst_play_sink_reconfigure),
19198 (gst_play_sink_set_flags), (gst_play_sink_get_flags),
19199 (gst_play_sink_change_state):
19200 * gst/playback/gstplaysink.h:
19201 Use flags to configure the sink pipelines.
19202 Add tee before audio pipeline so that we can use it for visualisations.
19203 Start working on integrating visualisations.
19204 Remove mode, we can do everything with the flags now.
19205 Add method to configue the sink pipeline.
19207 2008-01-06 16:36:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19209 Update to GMemoryInputStream API changes in GLib SVN and require gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
19210 Original commit message from CVS:
19212 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
19213 * tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
19214 Update to GMemoryInputStream API changes in GLib SVN and require
19215 gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
19216 We can also report the duration for every GSeekable, not only
19217 GFileInputStream and GMemoryInputStream.
19219 2008-01-06 14:39:19 +0000 Tim-Philipp Müller <tim@centricular.net>
19221 tests/check/pipelines/theoraenc.c: Turn these functions into macros so we can see right away where the failure occured.
19222 Original commit message from CVS:
19223 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
19224 (check_buffer_timestamp), (check_buffer_duration):
19225 Turn these functions into macros so we can see right away
19226 where the failure occured.
19228 2008-01-05 22:25:05 +0000 Julien Moutte <julien@moutte.net>
19230 sys/xvimage/xvimagesink.c: Add debugging information to understand how X calculates the stride for XvImages.
19231 Original commit message from CVS:
19232 2008-01-05 Julien Moutte <julien@fluendo.com>
19233 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
19234 debugging information to understand how X calculates the stride
19237 2008-01-03 20:33:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19239 gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
19240 Original commit message from CVS:
19241 * gst/volume/Makefile.am:
19242 * gst/volume/gstvolume.c: (volume_choose_func),
19243 (gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
19245 * gst/volume/gstvolume.h:
19246 Use GstAudioFilter as base class for the volume element instead of
19247 plain GstBaseTransform.
19249 2008-01-03 07:17:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19251 gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
19252 Original commit message from CVS:
19253 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
19254 Don't set element details for the abstract GstAudioFilter class.
19256 2008-01-02 12:09:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19258 gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
19259 Original commit message from CVS:
19260 * gst-libs/gst/audio/gstaudiofilter.c:
19261 (gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
19262 Implement get_unit_size() vmethod of GstBaseTransform.
19264 2008-01-01 12:53:48 +0000 Edward Hervey <bilboed@bilboed.com>
19266 gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
19267 Original commit message from CVS:
19268 * gst-libs/gst/pbutils/Makefile.am:
19269 * gst-libs/gst/pbutils/pbutils.h:
19270 Use glib-enum generator to have a proper enum GType for
19271 GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
19273 2008-01-01 01:21:47 +0000 David Schleef <ds@schleef.org>
19275 tests/check/: Reenable theoraenc test, which fails on the buildbot but not locally.
19276 Original commit message from CVS:
19277 * tests/check/Makefile.am:
19278 * tests/check/pipelines/theoraenc.c:
19279 Reenable theoraenc test, which fails on the buildbot but
19282 2007-12-31 21:31:01 +0000 David Schleef <ds@schleef.org>
19284 docs/: Add *-undeclared.txt to fix buildbot.
19285 Original commit message from CVS:
19286 * docs/libs/.cvsignore:
19287 * docs/plugins/.cvsignore:
19288 Add *-undeclared.txt to fix buildbot.
19290 2007-12-31 20:45:28 +0000 David Schleef <ds@schleef.org>
19292 tests/check/Makefile.am: Second attempt at disabling theoraenc test long enough to get buildbot to compile -base.
19293 Original commit message from CVS:
19294 * tests/check/Makefile.am:
19295 Second attempt at disabling theoraenc test long enough to
19296 get buildbot to compile -base.
19298 2007-12-31 20:21:20 +0000 David Schleef <ds@schleef.org>
19300 tests/check/pipelines/theoraenc.c: Disable theoraenc test long enough to get the buildbot to compile a recent -base.
19301 Original commit message from CVS:
19302 * tests/check/pipelines/theoraenc.c:
19303 Disable theoraenc test long enough to get the buildbot to
19304 compile a recent -base.
19306 2007-12-31 13:17:29 +0000 Wim Taymans <wim.taymans@gmail.com>
19308 tests/examples/seek/seek.c: Make sure we reset the slider value to 0.0 without racing against a possible g_idle that ...
19309 Original commit message from CVS:
19310 * tests/examples/seek/seek.c: (stop_cb):
19311 Make sure we reset the slider value to 0.0 without racing against a
19312 possible g_idle that sets it to something else.
19314 2007-12-31 00:32:53 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
19316 sys/ximage/ximagesink.c: fix typo
19317 Original commit message from CVS:
19318 * sys/ximage/ximagesink.c:
19321 2007-12-30 19:21:16 +0000 Wim Taymans <wim.taymans@gmail.com>
19323 gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
19324 Original commit message from CVS:
19325 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
19326 * gst-libs/gst/rtsp/gstrtspdefs.h:
19327 Add Location header so that we can start implementing redirects.
19330 2007-12-29 20:55:39 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
19332 gst/subparse/gstssaparse.c: combine if's
19333 Original commit message from CVS:
19334 * gst/subparse/gstssaparse.c:
19337 2007-12-29 19:23:59 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
19339 gst/subparse/gstssaparse.c: remove duplicate log message
19340 Original commit message from CVS:
19341 * gst/subparse/gstssaparse.c:
19342 remove duplicate log message
19344 2007-12-29 17:29:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19346 Update to latest API changes in GLib/GIO and require at least gio-2.0 2.15.0 for this.
19347 Original commit message from CVS:
19349 * ext/gio/gstgio.c:
19350 * ext/gio/gstgio.h:
19351 * ext/gio/gstgiobasesink.h:
19352 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
19353 * ext/gio/gstgiobasesrc.h:
19354 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
19355 * ext/gio/gstgiosink.h:
19356 * ext/gio/gstgiosrc.h:
19357 * ext/gio/gstgiostreamsink.h:
19358 * ext/gio/gstgiostreamsrc.h:
19359 * tests/check/pipelines/gio.c:
19360 Update to latest API changes in GLib/GIO and require at least
19361 gio-2.0 2.15.0 for this.
19362 * ext/gio/Makefile.am:
19363 Add GST_PLUGIN_LDFLAGS to LDFLAGS.
19365 2007-12-29 16:23:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19367 ext/libvisual/visual.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached()...
19368 Original commit message from CVS:
19369 * ext/libvisual/visual.c: (gst_visual_chain):
19370 Fix 'xyz may be used uninitialized' compiler warnings caused
19371 by broken g_assert_not_reached() macro in GLib-2.15.x and don't
19372 abort() in any case but properly report the error.
19374 2007-12-28 09:00:27 +0000 Wim Taymans <wim.taymans@gmail.com>
19376 gst/playback/gstplaybin2.c: Code cleanups.
19377 Original commit message from CVS:
19378 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
19379 (gst_play_bin_finalize), (gst_play_bin_set_uri),
19380 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
19381 (gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
19382 (autoplug_select_cb), (activate_group), (deactivate_group),
19383 (setup_next_source), (save_current_group),
19384 (gst_play_bin_change_state):
19386 Remove next-uri, we can use the uri property just fine.
19388 Unref uridecodebin when switching.
19389 Fix going to READY.
19390 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
19391 (gst_play_sink_init), (gst_play_sink_dispose),
19392 (gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
19393 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
19394 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
19395 (gst_play_sink_set_property), (gst_play_sink_get_property),
19396 (gen_video_chain), (gen_text_element), (gen_audio_chain),
19397 (gen_vis_element), (gst_play_sink_get_mode),
19398 (gst_play_sink_set_mode), (gst_play_sink_set_flags),
19399 (gst_play_sink_get_flags), (gst_play_sink_request_pad),
19400 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
19401 (gst_play_sink_change_state):
19402 * gst/playback/gstplaysink.h:
19403 Add some locking to make things threadsafe.
19404 * gst/playback/test7.c: (about_to_finish_cb):
19407 2007-12-22 12:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
19409 gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
19410 Original commit message from CVS:
19411 * gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
19412 (gst_video_scale_get_property), (gst_video_scale_transform_caps),
19413 (gst_video_scale_transform):
19414 Don't claim to be able to handle/transform caps that can't really
19415 be handled by the currently selected scaling method (here: RGB or
19416 packed YUV with 4-tap method). Also add locking to method property.
19417 * tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
19418 (test_basetransform_based):
19419 Some test pipelines for the above (not entirely valgrind clean yet
19422 2007-12-22 05:19:00 +0000 David Schleef <ds@schleef.org>
19424 gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
19425 Original commit message from CVS:
19426 * gst-libs/gst/video/video.c:
19427 * gst-libs/gst/video/video.h:
19428 Add additional RGBA and RGB-24 video formats.
19430 2007-12-21 22:46:56 +0000 Tim-Philipp Müller <tim@centricular.net>
19432 tests/check/: Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be deprecated in the future (see #498924).
19433 Original commit message from CVS:
19434 * tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
19435 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
19436 (test_suburi_error_wrongproto), (test_missing_primary_decoder):
19437 * tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
19438 (cddabasesrc_suite):
19439 Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
19440 deprecated in the future (see #498924).
19442 2007-12-21 22:26:47 +0000 Tim-Philipp Müller <tim@centricular.net>
19444 gst/playback/gststreamselector.c: Don't leak event.
19445 Original commit message from CVS:
19446 * gst/playback/gststreamselector.c: (gst_selector_pad_event):
19449 2007-12-20 19:43:25 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
19451 gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
19452 Original commit message from CVS:
19453 * gst-libs/gst/riff/riff-read.c:
19454 Use GST_ROUND_UP_2 macro
19456 2007-12-20 17:13:37 +0000 Tim-Philipp Müller <tim@centricular.net>
19458 gst/playback/.cvsignore: Ignore more.
19459 Original commit message from CVS:
19460 * gst/playback/.cvsignore:
19463 2007-12-20 10:41:29 +0000 Tim-Philipp Müller <tim@centricular.net>
19465 Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
19466 Original commit message from CVS:
19467 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
19468 * gst/playback/gstplaybasebin.c: (set_subtitles_visible),
19469 (set_active_source):
19470 * gst/playback/gstplaybasebin.h:
19471 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
19472 (setup_sinks), (playbin_set_subtitles_visible):
19473 Make switching off of subtitles work. To avoid all kind of
19474 problems with unlinking of the subtitle input, we just keep
19475 the subtitle inputs linked as they are and tell textoverlay
19476 not to render them. Fixes #373011.
19477 Other subtitle switching issues (esp. when there are both
19478 external and in-stream subtitles) remain. They'll be solved
19481 2007-12-18 16:21:35 +0000 Wim Taymans <wim.taymans@gmail.com>
19483 gst/playback/gststreamselector.c: Init the pad segment too.
19484 Original commit message from CVS:
19485 * gst/playback/gststreamselector.c: (gst_selector_pad_init):
19486 Init the pad segment too.
19488 2007-12-18 15:56:51 +0000 Wim Taymans <wim.taymans@gmail.com>
19490 gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
19491 Original commit message from CVS:
19492 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
19493 (gst_audioringbuffer_open_device),
19494 (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
19495 (gst_audioringbuffer_release), (gst_audioringbuffer_start),
19496 (gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
19497 (gst_audio_sink_create_ringbuffer):
19498 Improve debug output.
19499 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
19500 (gst_ring_buffer_pause), (gst_ring_buffer_delay):
19501 Prevent some functions from doing things and failing when the
19502 ringbuffer is not yet acquired.
19504 2007-12-18 15:32:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19506 gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
19507 Original commit message from CVS:
19508 * gst-libs/gst/interfaces/interfaces.h:
19509 Also remove interfaces.h from CVS as it is not needed anymore.
19511 2007-12-18 15:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19513 gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
19514 Original commit message from CVS:
19515 * gst-libs/gst/interfaces/Makefile.am:
19516 interfaces.h is not used anymore so remove it from the build
19519 2007-12-18 01:01:23 +0000 David Schleef <ds@schleef.org>
19521 gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
19522 Original commit message from CVS:
19523 * gst/videotestsrc/gstvideotestsrc.c:
19524 * gst/videotestsrc/gstvideotestsrc.h:
19525 Add a "blink" pattern. Turn on the pain. Apologies. It's useful
19526 for testing vertical refresh synchronization.
19528 2007-12-18 00:13:26 +0000 David Schleef <ds@schleef.org>
19530 Add new GstVideFormat enum and write a bunch of helper functions based around it.
19531 Original commit message from CVS:
19532 * docs/libs/gst-plugins-base-libs-sections.txt:
19533 * gst-libs/gst/video/video.c:
19534 * gst-libs/gst/video/video.h:
19535 Add new GstVideFormat enum and write a bunch of helper functions
19538 2007-12-17 23:41:14 +0000 Tim-Philipp Müller <tim@centricular.net>
19540 Makefile.am: Use new common/win32.mak.
19541 Original commit message from CVS:
19543 Use new common/win32.mak.
19545 2007-12-17 16:44:51 +0000 Wim Taymans <wim.taymans@gmail.com>
19547 gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
19548 Original commit message from CVS:
19549 * gst-libs/gst/audio/gstbaseaudiosrc.c:
19550 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
19552 When going from PLAYING to PAUSED, pause the ringbuffer before calling
19553 the parent state change function, just like the audiosink, because the
19554 parent waits for the element to finish its processing before completing
19555 the state change. This makes going to PAUSED a lot snappier.
19556 When going from READY to PAUSED, don't allow the ringbuffer to start
19559 2007-12-17 00:01:00 +0000 Edward Hervey <bilboed@bilboed.com>
19561 gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
19562 Original commit message from CVS:
19563 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19564 Yet another fix for broken software that produce files with an empty
19565 blockalign field. Instead of completely failing, make a second attempt
19566 at guessing the width/depth by looking at strf->size.
19568 2007-12-16 23:52:58 +0000 Tim-Philipp Müller <tim@centricular.net>
19570 gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
19571 Original commit message from CVS:
19572 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
19573 (gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
19574 * gst-libs/gst/pbutils/install-plugins.c:
19575 (gst_install_plugins_spawn_child), (gst_install_plugins_supported):
19576 * gst-libs/gst/pbutils/missing-plugins.c:
19577 (gst_missing_plugin_message_get_installer_detail),
19578 (gst_missing_encoder_installer_detail_new):
19579 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
19580 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
19581 Turn a few g_assert_not_reached() into g_return_val_if_reached() to
19582 avoid compiler warnings (#503930).
19584 2007-12-16 23:46:16 +0000 Edward Hervey <bilboed@bilboed.com>
19586 gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
19587 Original commit message from CVS:
19588 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
19589 Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
19590 for jpeg video streams.
19591 Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
19592 for the above modification.
19594 2007-12-15 17:27:48 +0000 Tim-Philipp Müller <tim@centricular.net>
19596 gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
19597 Original commit message from CVS:
19598 * gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
19599 (gst_x_overlay_handle_events):
19600 More guards (we don't want klass to end up being NULL).
19602 2007-12-15 03:40:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19604 Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
19605 Original commit message from CVS:
19607 * gst/volume/gstvolume.c: (gst_volume_init):
19608 Use new gst_base_transform_set_gap_aware() function as volume
19609 correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
19612 2007-12-14 19:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
19614 tests/examples/seek/seek.c: Don't go to READY on EOS as this avoids testing of seeking and restarting after EOS, use ...
19615 Original commit message from CVS:
19616 * tests/examples/seek/seek.c: (msg_segment_done), (main):
19617 Don't go to READY on EOS as this avoids testing of seeking and
19618 restarting after EOS, use the stop button when you want to READY.
19619 Don't try to do a flushing seek in segment-done, it does not make
19620 sense to use this for gapless playback and is not needed.
19622 2007-12-14 18:46:12 +0000 Wim Taymans <wim.taymans@gmail.com>
19624 gst/playback/gstqueue2.c: Use separate timers for input and output rates.
19625 Original commit message from CVS:
19626 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
19627 (reset_rate_timer), (update_in_rates), (update_out_rates),
19628 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
19629 (gst_queue_chain), (gst_queue_loop):
19630 Use separate timers for input and output rates.
19631 Pause measuring the output rate when we block for more data.
19634 2007-12-14 16:23:06 +0000 Christian Schaller <uraeus@gnome.org>
19636 * gst/speexresample/Makefile.am:
19637 update spec file and add two missing files for disting
19638 Original commit message from CVS:
19639 update spec file and add two missing files for disting
19641 2007-12-14 09:24:55 +0000 Wim Taymans <wim.taymans@gmail.com>
19643 gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
19644 Original commit message from CVS:
19645 * gst/playback/gstqueue2.c: (gst_queue_chain):
19646 Pause the timer to measure the input rate when we block because the
19647 queue is filled. See #503262.
19649 2007-12-13 15:54:00 +0000 Peter Kjellerstedt <pkj@axis.com>
19651 gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
19652 Original commit message from CVS:
19653 Patch by: Peter Kjellerstedt <pkj at axis com>
19654 * gst-libs/gst/rtsp/gstrtspconnection.c:
19655 (gst_rtsp_connection_free):
19656 Close control sockets. Fixes #503440.
19658 2007-12-13 12:31:38 +0000 Wim Taymans <wim.taymans@gmail.com>
19660 gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
19661 Original commit message from CVS:
19662 * gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
19663 Expose the right pad in the right place with the right element.
19665 2007-12-13 11:40:10 +0000 Tim-Philipp Müller <tim@centricular.net>
19667 gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
19668 Original commit message from CVS:
19669 * gst-libs/gst/pbutils/descriptions.c: (formats):
19670 Add description for 'private' dts caps (who come up with that name?).
19672 2007-12-13 10:10:35 +0000 Tim-Philipp Müller <tim@centricular.net>
19674 Makefile.am: Add check-exports target and run it with 'make check'.
19675 Original commit message from CVS:
19677 Add check-exports target and run it with 'make check'.
19679 Be stricter about what we export in our libraries: change regexp so that
19680 we only export _gst_foo(), but not __gst_foo().
19681 * gst-libs/gst/cdda/base64.h: (rfc822_binary):
19682 * gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
19683 Change internal functions to __gst_foo so they dont' get exported.
19684 * win32/common/libgstaudio.def:
19685 Add missing symbols.
19687 2007-12-11 21:18:57 +0000 David Schleef <ds@schleef.org>
19690 ChangeLog: remove conflict markers
19691 Original commit message from CVS:
19692 ChangeLog: remove conflict markers
19694 2007-12-11 17:14:13 +0000 Tim-Philipp Müller <tim@centricular.net>
19696 ext/gnomevfs/: Use gst_tag_freeform_string_to_utf8() here, which also takes into account any character sets specified...
19697 Original commit message from CVS:
19698 * ext/gnomevfs/Makefile.am:
19699 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
19700 Use gst_tag_freeform_string_to_utf8() here, which also takes
19701 into account any character sets specified by the user via
19702 environment variables.
19704 2007-12-10 15:21:41 +0000 Wim Taymans <wim.taymans@gmail.com>
19706 gst/audioconvert/Makefile.am: Also link to libm.
19707 Original commit message from CVS:
19708 * gst/audioconvert/Makefile.am:
19711 2007-12-10 15:13:55 +0000 Wim Taymans <wim.taymans@gmail.com>
19713 gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
19714 Original commit message from CVS:
19715 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19716 No need for floating point operations here. avoids having to link
19717 against the math library too.
19719 2007-12-10 11:16:25 +0000 Tim-Philipp Müller <tim@centricular.net>
19721 Add one or two missing formats. Generate ADPCM description dynamically depending on layout/format.
19722 Original commit message from CVS:
19723 * gst-libs/gst/pbutils/descriptions.c: (formats),
19724 (format_info_get_desc):
19725 * tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
19727 Add one or two missing formats. Generate ADPCM description
19728 dynamically depending on layout/format.
19730 2007-12-09 04:28:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19732 configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
19733 Original commit message from CVS:
19735 Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
19737 2007-12-08 18:38:39 +0000 Robin Stocker <robin.stocker@gmx.ch>
19739 gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
19740 Original commit message from CVS:
19741 Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
19742 * gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
19743 Some .srt files start with chunk number 0 and not chunk number 1,
19744 recognise and accept those as well (fixes #502497).
19745 * tests/check/elements/subparse.c: (srt_input), (srt_input0),
19747 Add unit test for the above.
19749 2007-12-06 12:08:21 +0000 Wim Taymans <wim.taymans@gmail.com>
19751 gst/playback/gstplay-enum.*: Add missing files.
19752 Original commit message from CVS:
19753 * gst/playback/gstplay-enum.c:
19754 (register_gst_autoplug_select_result),
19755 (gst_autoplug_select_result_get_type):
19756 * gst/playback/gstplay-enum.h:
19759 2007-12-05 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
19761 gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
19762 Original commit message from CVS:
19763 * gst/playback/Makefile.am:
19764 Group decodebin2 and uridecodebin into the same plugin so that they
19765 can share the GEnumType.
19766 * gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
19767 (_gst_select_accumulator), (gst_decode_bin_class_init),
19768 (gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
19769 (gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
19770 (analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
19771 Add signal to sort factories instead of the more awkward autoplug-select
19773 Modify autoplug_select so that we can try, skip or expose the
19774 autopluggin of an element on a pad.
19775 * gst/playback/gstfactorylists.c: (compare_ranks),
19776 (decoders_filter), (sinks_filter), (gst_factory_list_is_type),
19777 (element_filter), (gst_factory_list_get_elements),
19778 (gst_factory_list_debug), (gst_factory_list_filter):
19779 * gst/playback/gstfactorylists.h:
19780 Simplify the API, allow getting elements based on mask.
19781 * gst/playback/gstplay-marshal.list:
19782 Add some more marshallers.
19783 * gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
19784 (gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
19785 (autoplug_select_cb), (activate_group):
19786 Add support for managing non-raw sinks by providing a custom element and
19787 sink list to decodebin2.
19788 Try to plug non-raw sinks when decodebin2 using autoplug-select of
19790 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
19791 (gst_play_sink_set_mode), (gst_play_sink_request_pad):
19792 * gst/playback/gstplaysink.h:
19793 Add support for raw and non-raw sinks.
19794 Add support to force sinks selected by playbin2.
19795 Don't plug raw converters for non-raw sinks.
19796 * gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
19797 (_gst_select_accumulator), (gst_uri_decode_bin_class_init),
19798 (proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
19800 Use right accumulators.
19803 2007-12-03 13:47:00 +0000 Wim Taymans <wim.taymans@gmail.com>
19805 gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
19806 Original commit message from CVS:
19807 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
19808 Use runnning time as the base time instead of the timestamp.
19809 Spotted by Saur on IRC.
19811 2007-12-03 11:32:30 +0000 Edward Hervey <bilboed@bilboed.com>
19813 gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
19814 Original commit message from CVS:
19815 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
19816 Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
19818 2007-12-03 10:58:14 +0000 Wim Taymans <wim.taymans@gmail.com>
19820 ext/ogg/gstoggdemux.c: If we find a new serial number but it does not contain a BOS page, make sure we initialize the...
19821 Original commit message from CVS:
19822 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
19823 (gst_ogg_demux_read_chain):
19824 If we find a new serial number but it does not contain a BOS page, make
19825 sure we initialize the chain to NULL because else we will try to scan it
19826 and crash. Fixes #500763
19828 2007-11-30 17:47:15 +0000 Wim Taymans <wim.taymans@gmail.com>
19830 gst/playback/: Refactor some common code to filter factories and check caps compat.
19831 Original commit message from CVS:
19832 * gst/playback/Makefile.am:
19833 * gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
19834 (get_feature_array), (decoders_filter), (sinks_filter),
19835 (gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
19836 (gst_factory_list_filter):
19837 * gst/playback/gstfactorylists.h:
19838 Refactor some common code to filter factories and check caps compat.
19839 * gst/playback/gstdecodebin.c:
19840 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
19841 (gst_decode_bin_init), (gst_decode_bin_dispose),
19842 (gst_decode_bin_autoplug_continue),
19843 (gst_decode_bin_autoplug_factories),
19844 (gst_decode_bin_autoplug_select), (analyze_new_pad),
19845 (find_compatibles):
19846 * gst/playback/gstplaybin.c:
19847 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
19848 (gst_play_bin_init), (gst_play_bin_finalize),
19849 (autoplug_factories_cb), (activate_group):
19850 * gst/playback/gstqueue2.c:
19851 * gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
19852 (proxy_autoplug_continue_signal),
19853 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
19854 (proxy_drained_signal):
19855 Add some more debug info and use factor filtering code.
19857 2007-11-26 13:19:46 +0000 Julien Moutte <julien@moutte.net>
19859 configure.ac: Add QuickTime Wrapper plug-in.
19860 Original commit message from CVS:
19861 2007-11-26 Julien Moutte <julien@fluendo.com>
19862 * configure.ac: Add QuickTime Wrapper plug-in.
19863 * gst/speexresample/gstspeexresample.c:
19864 (gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
19865 build on Mac OS X Leopard. Incorrect printf format arguments.
19867 * sys/qtwrapper/Makefile.am:
19868 * sys/qtwrapper/audiodecoders.c:
19869 (qtwrapper_audio_decoder_base_init),
19870 (qtwrapper_audio_decoder_class_init),
19871 (qtwrapper_audio_decoder_init),
19872 (clear_AudioStreamBasicDescription), (fill_indesc_mp3),
19873 (fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
19874 (make_samr_magic_cookie), (open_decoder),
19875 (qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
19876 (qtwrapper_audio_decoder_chain),
19877 (qtwrapper_audio_decoder_sink_event),
19878 (qtwrapper_audio_decoders_register):
19879 * sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
19881 * sys/qtwrapper/codecmapping.h:
19882 * sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
19883 (image_description_for_mp4v), (image_description_from_stsd_buffer),
19884 (image_description_from_codec_data):
19885 * sys/qtwrapper/imagedescription.h:
19886 * sys/qtwrapper/qtutils.c: (get_name_info_from_component),
19887 (get_output_info_from_component), (dump_avcc_atom),
19888 (dump_image_description), (dump_codec_decompress_params),
19889 (addSInt32ToDictionary), (dump_cvpixel_buffer),
19890 (DestroyAudioBufferList), (AllocateAudioBufferList):
19891 * sys/qtwrapper/qtutils.h:
19892 * sys/qtwrapper/qtwrapper.c: (plugin_init):
19893 * sys/qtwrapper/qtwrapper.h:
19894 * sys/qtwrapper/videodecoders.c:
19895 (qtwrapper_video_decoder_base_init),
19896 (qtwrapper_video_decoder_class_init),
19897 (qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
19898 (fill_image_description), (new_image_description), (close_decoder),
19899 (open_decoder), (qtwrapper_video_decoder_sink_setcaps),
19900 (decompressCb), (qtwrapper_video_decoder_chain),
19901 (qtwrapper_video_decoder_sink_event),
19902 (qtwrapper_video_decoders_register): Initial import of QuickTime
19903 wrapper jointly developped by Songbird authors (Pioneers of the
19904 Inevitable) and Fluendo.
19906 2007-11-26 12:25:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19908 gst/: Add GAP-flag support.
19909 Original commit message from CVS:
19910 * gst/audiotestsrc/gstaudiotestsrc.c:
19911 * gst/volume/gstvolume.c:
19912 * gst/volume/gstvolume.h:
19913 Add GAP-flag support.
19915 2007-11-26 08:43:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19917 gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
19918 Original commit message from CVS:
19919 * gst/speexresample/README:
19920 * gst/speexresample/arch.h:
19921 * gst/speexresample/resample.c: (resampler_basic_direct_single),
19922 (resampler_basic_direct_double),
19923 (resampler_basic_interpolate_single),
19924 (resampler_basic_interpolate_double),
19925 (speex_resampler_process_native), (speex_resampler_process_float),
19926 (speex_resampler_process_int),
19927 (speex_resampler_process_interleaved_float),
19928 (speex_resampler_process_interleaved_int),
19929 (speex_resampler_get_input_latency),
19930 (speex_resampler_get_output_latency):
19931 * gst/speexresample/speex_resampler.h:
19932 Update speex resampler to latest SVN. We're now down to only the
19933 changes noted in README again.
19934 * gst/speexresample/speex_resampler_wrapper.h:
19935 * gst/speexresample/gstspeexresample.c:
19936 (gst_speex_resample_push_drain), (gst_speex_resample_query):
19937 Adjust to API changes.
19939 2007-11-24 15:02:01 +0000 Julien Moutte <julien@moutte.net>
19941 tests/examples/seek/seek.c: Increase the range of the rate selector as I would like to test QOS behavior at higher fo...
19942 Original commit message from CVS:
19943 2007-11-24 Julien MOUTTE <julien@moutte.net>
19944 * tests/examples/seek/seek.c: (main): Increase the range of the
19945 rate selector as I would like to test QOS behavior at higher
19946 forward and reverse playback speed like say 64x.
19948 2007-11-23 10:21:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19950 gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
19951 Original commit message from CVS:
19952 * gst/speexresample/gstspeexresample.c:
19953 (gst_speex_resample_update_state):
19954 Only post the latency message if we have a resampler state already.
19956 2007-11-23 10:21:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19958 gst/audioresample/gstaudioresample.c: Implement latency query.
19959 Original commit message from CVS:
19960 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
19961 (audioresample_query), (audioresample_query_type),
19962 (gst_audioresample_set_property):
19963 Implement latency query.
19965 2007-11-23 10:01:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19967 gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
19968 Original commit message from CVS:
19969 * gst/speexresample/gstspeexresample.c:
19970 (gst_speex_resample_update_state):
19971 Also post GST_MESSAGE_LATENCY if the latency changes.
19973 2007-11-23 08:48:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19975 gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
19976 Original commit message from CVS:
19977 * gst/speexresample/resample.c: (speex_resampler_get_latency),
19978 (speex_resampler_drain_float), (speex_resampler_drain_int),
19979 (speex_resampler_drain_interleaved_float),
19980 (speex_resampler_drain_interleaved_int):
19981 * gst/speexresample/speex_resampler.h:
19982 * gst/speexresample/speex_resampler_wrapper.h:
19983 Add functions to push the remaining samples and to get the latency
19984 of the resampler. These will get added to Speex SVN in this or a
19985 slightly changed form at some point too and should get merged then
19987 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
19988 (gst_speex_resample_init_state),
19989 (gst_speex_resample_transform_size),
19990 (gst_speex_resample_push_drain), (gst_speex_resample_event),
19991 (gst_speex_fix_output_buffer), (gst_speex_resample_process),
19992 (gst_speex_resample_query), (gst_speex_resample_query_type):
19993 Drop the prepending zeroes and output the remaining samples on EOS.
19994 Also properly implement the latency query for this. speexresample
19995 should be completely ready for production use now.
19997 2007-11-21 18:02:21 +0000 Wim Taymans <wim.taymans@gmail.com>
19999 gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
20000 Original commit message from CVS:
20001 * gst-libs/gst/audio/gstbaseaudiosink.c:
20002 (gst_base_audio_sink_drain):
20003 Our EOS time contains the base_time, _wait_eos() expects a running_time
20004 so we have to subtract the base_time again before calling the function.
20005 This fixes an EOS regression where the base_time was added twice and EOS
20006 took longer and longer in certain situations.
20009 2007-11-21 13:04:17 +0000 Wim Taymans <wim.taymans@gmail.com>
20011 Expose methods for some object properties so that subclasses can more easily configure them.
20012 Original commit message from CVS:
20013 * docs/libs/gst-plugins-base-libs-sections.txt:
20014 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
20015 (gst_base_audio_sink_set_provide_clock),
20016 (gst_base_audio_sink_get_provide_clock),
20017 (gst_base_audio_sink_set_slave_method),
20018 (gst_base_audio_sink_get_slave_method),
20019 (gst_base_audio_sink_set_property),
20020 (gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
20021 (gst_base_audio_sink_none_slaving),
20022 (gst_base_audio_sink_handle_slaving):
20023 * gst-libs/gst/audio/gstbaseaudiosink.h:
20024 Expose methods for some object properties so that subclasses can more
20025 easily configure them.
20026 Added slave method none, that completely disables slaving to the
20028 API: gst_base_audio_sink_set_provide_clock()
20029 API: gst_base_audio_sink_get_provide_clock()
20030 API: gst_base_audio_sink_set_slave_method()
20031 API: gst_base_audio_sink_get_slave_method()
20032 * gst-libs/gst/audio/gstbaseaudiosrc.c:
20033 (gst_base_audio_src_set_provide_clock),
20034 (gst_base_audio_src_get_provide_clock),
20035 (gst_base_audio_src_set_property),
20036 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
20037 * gst-libs/gst/audio/gstbaseaudiosrc.h:
20038 Expose methods for some object properties so that subclasses can more
20039 easily configure them.
20040 API: gst_base_audio_src_set_provide_clock()
20041 API: gst_base_audio_src_get_provide_clock()
20043 2007-11-21 10:18:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20045 gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
20046 Original commit message from CVS:
20047 * gst/speexresample/README:
20048 Add README explaining where the resampling code was taken from
20049 and which changes were done.
20050 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
20052 Use g_malloc() and friends instead of malloc() to achieve higher
20053 portability and define the functions inline.
20054 * gst/speexresample/speex_resampler.h:
20055 Add back some useless preprocessor stuff to keep the diff between
20056 our version and the one from the Speex SVN repository lower.
20058 2007-11-20 20:23:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20060 gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
20061 Original commit message from CVS:
20062 * gst/speexresample/gstspeexresample.c:
20063 (gst_speex_fix_output_buffer), (gst_speex_resample_transform):
20064 Some small cleanup and addition of a TODO item.
20066 2007-11-20 12:56:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20068 gst/speexresample/Makefile.am: Add missing file.
20069 Original commit message from CVS:
20070 * gst/speexresample/Makefile.am:
20073 2007-11-20 07:53:56 +0000 Joe Peterson <lavajoe@gentoo.org>
20075 gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
20076 Original commit message from CVS:
20077 Patch by: Joe Peterson <lavajoe at gentoo dot org>
20078 * gst-libs/gst/sdp/gstsdpmessage.c:
20079 Fix compilation on FreeBSD (Gentoo). Fixes #498228.
20081 2007-11-20 07:47:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20083 Add speexresample to the docs and while at that do a make update.
20084 Original commit message from CVS:
20085 * docs/plugins/Makefile.am:
20086 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
20087 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
20088 * docs/plugins/gst-plugins-bad-plugins.args:
20089 * docs/plugins/gst-plugins-bad-plugins.signals:
20090 * docs/plugins/inspect/plugin-bz2.xml:
20091 * docs/plugins/inspect/plugin-cdxaparse.xml:
20092 * docs/plugins/inspect/plugin-dtsdec.xml:
20093 * docs/plugins/inspect/plugin-equalizer.xml:
20094 * docs/plugins/inspect/plugin-faac.xml:
20095 * docs/plugins/inspect/plugin-faad.xml:
20096 * docs/plugins/inspect/plugin-filter.xml:
20097 * docs/plugins/inspect/plugin-freeze.xml:
20098 * docs/plugins/inspect/plugin-gio.xml:
20099 * docs/plugins/inspect/plugin-gsm.xml:
20100 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
20101 * docs/plugins/inspect/plugin-h264parse.xml:
20102 * docs/plugins/inspect/plugin-modplug.xml:
20103 * docs/plugins/inspect/plugin-mpeg2enc.xml:
20104 * docs/plugins/inspect/plugin-musepack.xml:
20105 * docs/plugins/inspect/plugin-musicbrainz.xml:
20106 * docs/plugins/inspect/plugin-nsfdec.xml:
20107 * docs/plugins/inspect/plugin-replaygain.xml:
20108 * docs/plugins/inspect/plugin-soundtouch.xml:
20109 * docs/plugins/inspect/plugin-spcdec.xml:
20110 * docs/plugins/inspect/plugin-spectrum.xml:
20111 * docs/plugins/inspect/plugin-speed.xml:
20112 * docs/plugins/inspect/plugin-tta.xml:
20113 * docs/plugins/inspect/plugin-videosignal.xml:
20114 * docs/plugins/inspect/plugin-xingheader.xml:
20115 * docs/plugins/inspect/plugin-xvid.xml:
20116 * gst/speexresample/gstspeexresample.h:
20117 Add speexresample to the docs and while at that do a make update.
20119 2007-11-20 07:30:30 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20121 gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
20122 Original commit message from CVS:
20123 * gst/speexresample/gstspeexresample.c:
20124 (gst_speex_fix_output_buffer), (gst_speex_resample_process):
20125 If the resampler gives less output samples than expected
20126 adjust the output buffer and print a warning.
20128 2007-11-20 07:02:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20130 Add resample element based on the Speex resampling algorithm.
20131 Original commit message from CVS:
20133 * gst/speexresample/arch.h:
20134 * gst/speexresample/fixed_generic.h:
20135 * gst/speexresample/gstspeexresample.c:
20136 (gst_speex_resample_base_init), (gst_speex_resample_class_init),
20137 (gst_speex_resample_init), (gst_speex_resample_start),
20138 (gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
20139 (gst_speex_resample_transform_caps),
20140 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
20141 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
20142 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
20143 (gst_speex_resample_event), (gst_speex_resample_check_discont),
20144 (gst_speex_resample_process), (gst_speex_resample_transform),
20145 (gst_speex_resample_set_property),
20146 (gst_speex_resample_get_property), (plugin_init):
20147 * gst/speexresample/gstspeexresample.h:
20148 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
20149 (speex_free), (compute_func), (main), (sinc), (cubic_coef),
20150 (resampler_basic_direct_single), (resampler_basic_direct_double),
20151 (resampler_basic_interpolate_single),
20152 (resampler_basic_interpolate_double), (update_filter),
20153 (speex_resampler_init), (speex_resampler_init_frac),
20154 (speex_resampler_destroy), (speex_resampler_process_native),
20155 (speex_resampler_process_float), (speex_resampler_process_int),
20156 (speex_resampler_process_interleaved_float),
20157 (speex_resampler_process_interleaved_int),
20158 (speex_resampler_set_rate), (speex_resampler_get_rate),
20159 (speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
20160 (speex_resampler_set_quality), (speex_resampler_get_quality),
20161 (speex_resampler_set_input_stride),
20162 (speex_resampler_get_input_stride),
20163 (speex_resampler_set_output_stride),
20164 (speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
20165 (speex_resampler_reset_mem), (speex_resampler_strerror):
20166 * gst/speexresample/speex_resampler.h:
20167 * gst/speexresample/speex_resampler_float.c:
20168 * gst/speexresample/speex_resampler_int.c:
20169 * gst/speexresample/speex_resampler_wrapper.h:
20170 Add resample element based on the Speex resampling algorithm.
20172 2007-11-19 12:30:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20174 tests/check/libs/fft.c: Fix scaling to really have dB instead of something else.
20175 Original commit message from CVS:
20176 * tests/check/libs/fft.c: (GST_START_TEST):
20177 Fix scaling to really have dB instead of something else.
20179 2007-11-19 12:08:16 +0000 Julien Moutte <julien@moutte.net>
20181 tests/examples/seek/seek.c: There's a nice macro to check
20182 Original commit message from CVS:
20183 2007-11-19 Julien MOUTTE <julien@moutte.net>
20184 * tests/examples/seek/seek.c: (main): There's a nice macro to
20186 GTK version, use it.
20188 2007-11-19 11:59:20 +0000 Julien Moutte <julien@moutte.net>
20190 tests/examples/seek/seek.c: Try to support stable version of GTK.
20191 Original commit message from CVS:
20192 2007-11-19 Julien MOUTTE <julien@moutte.net>
20193 * tests/examples/seek/seek.c: (main): Try to support stable version
20196 2007-11-17 15:25:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20198 gst/playback/: Fix the build + little README update.
20199 Original commit message from CVS:
20200 * gst/playback/README:
20201 * gst/playback/test7.c:
20202 Fix the build + little README update.
20204 2007-11-16 16:02:45 +0000 Wim Taymans <wim.taymans@gmail.com>
20206 tests/examples/seek/seek.c: Add playbin2 seek pipeline.
20207 Original commit message from CVS:
20208 * tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
20209 Add playbin2 seek pipeline.
20211 2007-11-16 15:44:48 +0000 Wim Taymans <wim.taymans@gmail.com>
20213 gst/playback/: Add playbin2.
20214 Original commit message from CVS:
20215 * gst/playback/Makefile.am:
20216 * gst/playback/gstplayback.c: (plugin_init):
20217 * gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
20218 (eos_cb), (about_to_finish_cb), (main):
20220 Added gapless playback example.
20221 * gst/playback/gstplaybasebin.c:
20222 * gst/playback/gstplaybasebin.h:
20223 * gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
20224 * gst/playback/gstqueue2.c:
20225 * gst/playback/test.c:
20226 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
20228 * gst/playback/gststreaminfo.h:
20230 * gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
20231 (gst_play_bin_class_init), (init_group), (gst_play_bin_init),
20232 (gst_play_bin_dispose), (gst_play_bin_set_uri),
20233 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
20234 (gst_play_bin_get_property), (gst_play_bin_handle_message),
20235 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
20236 (drained_cb), (unlink_group), (activate_group),
20237 (setup_next_source), (gst_play_bin_change_state),
20238 (gst_play_bin2_plugin_init):
20239 Added raw first version of playbin2. Does chained oggs and gapless
20240 playback fine. No support for raw sinks yet. No visualisations or
20242 * gst/playback/gstplaysink.c: (gst_play_sink_get_type),
20243 (gst_play_sink_class_init), (gst_play_sink_init),
20244 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
20245 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
20246 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
20247 (gst_play_sink_set_property), (gst_play_sink_get_property),
20248 (post_missing_element_message), (free_chain), (add_chain),
20249 (activate_chain), (gen_video_chain), (gen_text_element),
20250 (gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
20251 (gst_play_sink_set_mode), (gst_play_sink_request_pad),
20252 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
20253 (gst_play_sink_send_event), (gst_play_sink_change_state):
20254 * gst/playback/gstplaysink.h:
20255 Added Element that abstracts the sinks and their pipelines for playbin2.
20257 2007-11-16 15:05:07 +0000 Wim Taymans <wim.taymans@gmail.com>
20259 gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
20260 Original commit message from CVS:
20261 * gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
20262 (gst_selector_pad_class_init), (gst_selector_pad_init),
20263 (gst_selector_pad_finalize), (gst_selector_pad_reset),
20264 (gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
20265 (gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
20266 (gst_selector_pad_chain), (gst_stream_selector_get_type),
20267 (gst_stream_selector_base_init), (gst_stream_selector_class_init),
20268 (gst_stream_selector_init), (gst_stream_selector_set_property),
20269 (gst_stream_selector_get_linked_pad),
20270 (gst_stream_selector_getcaps),
20271 (gst_stream_selector_is_active_sinkpad),
20272 (gst_stream_selector_activate_sinkpad),
20273 (gst_stream_selector_get_linked_pads),
20274 (gst_stream_selector_request_new_pad),
20275 (gst_stream_selector_release_pad):
20276 * gst/playback/gststreamselector.h:
20277 Improve streamselector, make it select and unselect the current pad more
20279 Subclass GstPad for the sinkpads of the selector.
20280 Handle segments more correctly.
20281 Fix caps negotiation.
20282 Implement release_pad.
20284 2007-11-16 12:51:44 +0000 Wim Taymans <wim.taymans@gmail.com>
20286 gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
20287 Original commit message from CVS:
20288 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
20289 (gst_decode_group_check_if_drained), (source_pad_event_probe),
20291 Add drained signal fired when decodebin finishes decoding the data.
20292 Remove deprecated STATE_DIRTY message.
20293 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
20294 (unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
20295 (analyse_source), (proxy_drained_signal), (make_decoder),
20296 (source_new_pad), (value_list_append_structure_list),
20297 (handle_redirect_message), (handle_message):
20298 Proxy the new drained signal.
20299 Handle pad removed from decodebin.
20300 Handle redirect messages by sorting multiple redirections based on the
20303 2007-11-16 11:22:09 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
20305 gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
20306 Original commit message from CVS:
20307 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
20308 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
20309 Fix leaking headers. Fixes #496761.
20311 2007-11-16 11:16:58 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
20313 sys/: Don't leak the PAR on errors. Fixes #496731.
20314 Original commit message from CVS:
20315 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
20316 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
20317 (gst_ximagesink_change_state):
20318 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
20319 Don't leak the PAR on errors. Fixes #496731.
20321 2007-11-16 10:14:34 +0000 Tim-Philipp Müller <tim@centricular.net>
20323 gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
20324 Original commit message from CVS:
20325 * gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
20326 (gst_tag_from_id3_user_tag):
20327 Add mapping for audio cd discid tags, so we can extract
20328 them from tags as well (see #347848). Also compare identifiers
20329 in ID3v2 TXXX frames in a case-insensitive way to increase
20330 compatibility when reading tags (discid vs. DiscID vs. DiscId).
20332 2007-11-16 01:21:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20334 gst-plugins-base.doap: Oops, fix the release name.
20335 Original commit message from CVS:
20336 * gst-plugins-base.doap:
20337 Oops, fix the release name.
20339 2007-11-16 00:44:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20341 gst-plugins-base.doap: Add 0.10.15 release
20342 Original commit message from CVS:
20343 * gst-plugins-base.doap:
20344 Add 0.10.15 release
20346 2007-11-16 00:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20348 configure.ac: Back to CVS
20349 Original commit message from CVS:
20353 === release 0.10.15 ===
20355 2007-11-16 00:14:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20357 configure.ac: releasing 0.10.15, "No need to argue"
20358 Original commit message from CVS:
20359 === release 0.10.15 ===
20360 2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
20362 releasing 0.10.15, "No need to argue"
20364 2007-11-16 00:04:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20389 Original commit message from CVS:
20392 2007-11-15 21:40:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20394 win32/vs6/libgstfft.dsp: Convert line endings to DOS.
20395 Original commit message from CVS:
20396 * win32/vs6/libgstfft.dsp:
20397 Convert line endings to DOS.
20399 2007-11-15 21:14:04 +0000 Sébastien Moutte <sebastien@moutte.net>
20401 win32/: Add a project file for fft plugin and remove socket based plugin which don't build from the workspace.* win32...
20402 Original commit message from CVS:
20403 * win32/vs6/gst_plugins_base.dsw:
20404 * win32/vs6/libgstfft.dsp:
20406 Add a project file for fft plugin and remove socket
20407 based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
20408 * win32/vs6/libgstrtp.dsp:
20409 * win32/vs6/libgsttag.dsp:
20410 Convert line endings back to DOS.
20413 2007-11-14 12:27:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20415 win32/vs6/: Convert line endings back to DOS
20416 Original commit message from CVS:
20417 * win32/vs6/libgstinterfaces.dsp:
20418 * win32/vs6/libgstrtsp.dsp:
20419 Convert line endings back to DOS
20421 2007-11-14 11:08:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20423 gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
20424 Original commit message from CVS:
20425 * gst-libs/gst/fft/kiss_fft_f32.h:
20426 * gst-libs/gst/fft/kiss_fft_f64.h:
20427 * gst-libs/gst/fft/kiss_fft_s16.h:
20428 * gst-libs/gst/fft/kiss_fft_s32.h:
20429 Don't include malloc.h which doesn't exist on Mac OSX.
20430 Instead, pull in glib.h and use g_malloc/g_free for
20431 consistency. Fixes: #496548
20433 2007-11-09 15:54:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20435 gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
20436 Original commit message from CVS:
20437 * gst/playback/gstdecodebin2.c:
20438 Dont leak ghostpad. Fixes #475451.
20440 2007-11-09 12:21:52 +0000 Wim Taymans <wim.taymans@gmail.com>
20442 Update some more docs and comments.
20443 Original commit message from CVS:
20444 * docs/design/design-decodebin.txt:
20445 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
20446 Update some more docs and comments.
20448 2007-11-07 16:47:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20450 Require GIO >= 0.1.2 and adjust unit test for an API change.
20451 Original commit message from CVS:
20453 * tests/check/pipelines/gio.c: (GST_START_TEST):
20454 Require GIO >= 0.1.2 and adjust unit test for an API change.
20456 2007-11-07 15:18:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20458 ext/gio/gstgio.h: Add macro to check if a stream supports seeking.
20459 Original commit message from CVS:
20460 * ext/gio/gstgio.h:
20461 Add macro to check if a stream supports seeking.
20462 * ext/gio/Makefile.am:
20463 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
20464 (gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
20465 (gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
20466 (gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
20467 (gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
20468 (gst_gio_base_sink_render), (gst_gio_base_sink_query),
20469 (gst_gio_base_sink_set_stream):
20470 * ext/gio/gstgiobasesink.h:
20471 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
20472 (gst_gio_base_src_class_init), (gst_gio_base_src_init),
20473 (gst_gio_base_src_finalize), (gst_gio_base_src_start),
20474 (gst_gio_base_src_stop), (gst_gio_base_src_get_size),
20475 (gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
20476 (gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
20477 (gst_gio_base_src_create), (gst_gio_base_src_set_stream):
20478 * ext/gio/gstgiobasesrc.h:
20479 Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
20480 base classes that only require a GInputStream or GOutputStream to
20482 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
20483 (gst_gio_sink_class_init), (gst_gio_sink_init),
20484 (gst_gio_sink_finalize), (gst_gio_sink_start):
20485 * ext/gio/gstgiosink.h:
20486 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
20487 (gst_gio_src_class_init), (gst_gio_src_init),
20488 (gst_gio_src_finalize), (gst_gio_src_start):
20489 * ext/gio/gstgiosrc.h:
20490 Use the newly created base classes here.
20491 * ext/gio/gstgio.c: (plugin_init):
20492 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
20493 (gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
20494 (gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
20495 (gst_gio_stream_sink_get_property):
20496 * ext/gio/gstgiostreamsink.h:
20497 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
20498 (gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
20499 (gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
20500 (gst_gio_stream_src_get_property):
20501 * ext/gio/gstgiostreamsrc.h:
20502 Implement GstGioStreamSink and GstGioStreamSrc that have a property
20503 to set the GInputStream/GOutputStream that should be used.
20504 * tests/check/Makefile.am:
20505 * tests/check/pipelines/.cvsignore:
20506 * tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
20507 (gio_testsuite), (main):
20508 Add unit test for giostreamsrc and giostreamsink.
20510 2007-11-07 11:48:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20512 ext/gio/gstgio.c: Remove nowadays unnecessary workaround for a crash.
20513 Original commit message from CVS:
20514 * ext/gio/gstgio.c: (plugin_init):
20515 Remove nowadays unnecessary workaround for a crash.
20516 * ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
20517 (gst_gio_sink_start), (gst_gio_sink_stop),
20518 (gst_gio_sink_unlock_stop):
20519 * ext/gio/gstgiosink.h:
20520 * ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
20521 (gst_gio_src_stop), (gst_gio_src_unlock_stop):
20522 * ext/gio/gstgiosrc.h:
20523 Make the finalize function safer, clean up everything that could stay
20525 Reset the cancellable instead of creating a new one after cancelling
20527 Don't store the GFile in the element, it's only necessary for creating
20530 2007-11-06 23:35:39 +0000 Sebastien Moutte <sebastien@moutte.net>
20532 gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
20533 Original commit message from CVS:
20534 Patch by: Sebastien Moutte <sebastien moutte net>
20535 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
20536 (gst_rtcp_unix_to_ntp):
20537 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
20538 Fix some C99-isms and and a missing function that some versions of
20539 MSVC don't like too much (#494346).
20540 * win32/vs6/gst_plugins_base.dsw:
20541 * win32/vs6/libgstaudio.dsp:
20542 * win32/vs6/libgstrtp.dsp:
20543 * win32/vs6/libgsttag.dsp:
20544 Update vs6 projects files (#494346).
20546 2007-11-06 16:38:49 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
20548 win32/common/: More missing symbols to export (fixes #493986).
20549 Original commit message from CVS:
20550 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
20551 * win32/common/libgstaudio.def:
20552 * win32/common/libgstcdda.def:
20553 * win32/common/libgstinterfaces.def:
20554 * win32/common/libgstnetbuffer.def:
20555 * win32/common/libgstpbutils.def:
20556 * win32/common/libgstrtp.def:
20557 * win32/common/libgstrtsp.def:
20558 * win32/common/libgsttag.def:
20559 * win32/common/libgstvideo.def:
20560 More missing symbols to export (fixes #493986).
20562 2007-11-06 11:58:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20564 Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
20565 Original commit message from CVS:
20566 * docs/libs/gst-plugins-base-libs-sections.txt:
20567 * gst-libs/gst/fft/gstfftf32.c:
20568 * gst-libs/gst/fft/gstfftf32.h:
20569 * gst-libs/gst/fft/gstfftf64.c:
20570 * gst-libs/gst/fft/gstfftf64.h:
20571 * gst-libs/gst/fft/gstffts16.c:
20572 * gst-libs/gst/fft/gstffts16.h:
20573 * gst-libs/gst/fft/gstffts32.c:
20574 * gst-libs/gst/fft/gstffts32.h:
20575 * tests/check/libs/fft.c: (GST_START_TEST):
20576 Remove the magnitude and phase calculation functions as these have
20577 very special use cases and can't even be used for the spectrum
20578 element. Also adjust the docs to mention some properties of the used
20579 FFT implemention, i.e. how the values are scaled. Fixes #492098.
20581 2007-11-06 11:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
20583 gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
20584 Original commit message from CVS:
20585 * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
20587 Avoid crash when there are external subtitles (fixes #491722).
20589 2007-11-03 10:39:21 +0000 Tim-Philipp Müller <tim@centricular.net>
20591 ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
20592 Original commit message from CVS:
20593 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
20594 * ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
20595 'Could not open resource for writing' is not an acceptable
20596 error message when we can't open the audio device (see #492334),
20597 even less so when we're trying to open it to record something.
20599 2007-11-02 21:03:01 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
20601 win32/common/libgstrtp.def: Add some more missing symbols (#492813).
20602 Original commit message from CVS:
20603 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
20604 * win32/common/libgstrtp.def:
20605 Add some more missing symbols (#492813).
20607 2007-11-02 14:59:06 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
20609 tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
20610 Original commit message from CVS:
20611 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
20612 * tests/check/elements/audioconvert.c: (verify_convert):
20613 Add check to make sure that the out caps have a channel layout
20614 set on them where they should have one.
20616 2007-11-01 13:28:59 +0000 Vincent Torri <vtorri@univ-evry.fr>
20618 gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
20619 Original commit message from CVS:
20620 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
20621 * gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
20622 * gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
20623 Include our own _stdint.h instead of sys/types.h, makes MingW happy
20625 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
20626 Use _pipe directly, GLib doesn't have a pipe() macro any longer
20627 (it disappeared in GLib 2.14.0) (#492306).
20628 * gst-libs/gst/sdp/Makefile.am:
20629 * gst-libs/gst/sdp/gstsdpmessage.c:
20630 Fix includes and LIBS for win32/Mingw (#492306).
20631 * tests/examples/dynamic/addstream.c (pause_play_stream):
20632 Use more portable g_usleep() instead of sleep() (#492306).
20634 2007-11-01 12:51:57 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
20636 gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
20637 Original commit message from CVS:
20638 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
20639 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
20640 (gst_ring_buffer_parse_caps):
20641 Return NULL instead of an enum that happens to be 0, fixes warning
20643 * gst-libs/gst/audio/gstringbuffer.h:
20644 No trailing commas in enum list (for gcc-2.9x).
20645 * gst/videotestsrc/videotestsrc.c: (random_char):
20646 Make information loss explicit instead of implicitly truncating to
20647 eight bits via the return value. Fixes runtime error on MSVC when
20648 using the debug CRT (#492114).
20649 * win32/common/config.h.in:
20650 Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
20651 * win32/common/libgstinterfaces.def:
20652 * win32/common/libgstrtp.def:
20653 Export a few more symbols (#492114).
20655 2007-11-01 08:06:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20657 gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
20658 Original commit message from CVS:
20659 * gst-libs/gst/audio/audio.c:
20660 * gst-libs/gst/audio/audio.h:
20661 Readd the deprecation guards, but preserve compilability.
20663 2007-10-31 17:54:48 +0000 Tim-Philipp Müller <tim@centricular.net>
20665 gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
20666 Original commit message from CVS:
20667 * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
20668 (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
20669 Preserve channel layout when fixating the number of channels in the
20670 output caps, or make sure there's a suitable channel position layout
20671 set on the caps if required. Fixes #430677.
20673 2007-10-31 17:32:22 +0000 Tim-Philipp Müller <tim@centricular.net>
20675 tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
20676 Original commit message from CVS:
20677 * tests/check/elements/decodebin.c: (test_text_plain_streams):
20678 Make sure the pipeline really operates in push mode as it should
20681 2007-10-31 15:30:15 +0000 Tim-Philipp Müller <tim@centricular.net>
20683 gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
20684 Original commit message from CVS:
20685 * gst-libs/gst/audio/audio.h:
20686 Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
20687 compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
20688 (ie. normal cvs builds) will fail.
20690 2007-10-31 12:47:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20692 tell gtk-doc about the deprecation guard. Apply more doc fixes.
20693 Original commit message from CVS:
20694 * docs/libs/Makefile.am:
20695 * gst-libs/gst/audio/audio.c:
20696 * gst-libs/gst/audio/audio.h:
20697 * gst-libs/gst/interfaces/mixer.c:
20698 tell gtk-doc about the deprecation guard. Apply more doc fixes.
20700 2007-10-31 12:30:28 +0000 Tim-Philipp Müller <tim@centricular.net>
20702 tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
20703 Original commit message from CVS:
20704 * tests/check/libs/audio.c: (init_value_to_channel_layout),
20705 (test_channel_layout_value_intersect), (audio_suite):
20706 Add simple unit test to make sure GstValue intersection
20707 of channel layouts works the way I think it does.
20709 2007-10-30 20:32:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20711 Fix the docs according to what gtk-doc complained about.
20712 Original commit message from CVS:
20713 * docs/libs/gst-plugins-base-libs-sections.txt:
20714 * gst-libs/gst/audio/gstaudiofilter.h:
20715 * gst-libs/gst/interfaces/mixer.h:
20716 * gst-libs/gst/rtp/gstbasertpdepayload.c:
20717 * gst-libs/gst/rtp/gstbasertpdepayload.h:
20718 * gst-libs/gst/sdp/gstsdpmessage.c:
20719 Fix the docs according to what gtk-doc complained about.
20721 2007-10-30 19:46:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20723 tests/icles/stress-playbin.c: Fix the build.
20724 Original commit message from CVS:
20725 * tests/icles/stress-playbin.c:
20728 2007-10-30 15:54:46 +0000 Tim-Philipp Müller <tim@centricular.net>
20730 gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
20731 Original commit message from CVS:
20732 * gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
20733 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
20734 Post nice/more useful error message if we don't have a decoder for
20737 2007-10-30 15:07:58 +0000 Wim Taymans <wim.taymans@gmail.com>
20739 gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
20740 Original commit message from CVS:
20741 * gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
20742 Be a bit more useful, unblock the pads after we fired the no-more-pads
20743 signal so that we can use the signal to inspect and connect all pads
20744 without having to keep extra state outside of decodebin.
20746 2007-10-30 15:00:06 +0000 Wim Taymans <wim.taymans@gmail.com>
20748 gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
20749 Original commit message from CVS:
20750 * gst/playback/gsturidecodebin.c:
20751 (gst_uri_decode_bin_autoplug_continue),
20752 (gst_uri_decode_bin_class_init), (no_more_pads_full):
20753 Implement default signal handler so that we return TRUE when nothing is
20756 2007-10-28 11:53:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20758 gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
20759 Original commit message from CVS:
20760 * gst-libs/gst/riff/riff-media.c:
20761 (gst_riff_wavext_add_channel_layout),
20762 (gst_riff_wave_add_default_channel_layout),
20763 (gst_riff_wavext_get_default_channel_mask),
20764 (gst_riff_create_audio_caps):
20765 Use the ALSA channel layout as default for wav files without channel
20766 layout information. This fixes playback of chan-id.wav on 5.1 systems
20767 for example. Also refactor the channel layout setting a bit and add
20768 more default channel orders. Fixes #489010.
20770 2007-10-28 11:46:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20773 Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
20774 Original commit message from CVS:
20775 (gst_riff_wavext_add_channel_layout),
20776 (gst_riff_wave_add_default_channel_layout),
20777 (gst_riff_wavext_get_default_channel_mask),
20778 (gst_riff_create_audio_caps):
20779 Use the ALSA channel layout as default for wav files without channel
20780 layout information. This fixes playback of chan-id.wav on 5.1 systems
20781 for example. Also refactor the channel layout setting a bit and add
20782 more default channel orders. Fixes #489010.
20784 2007-10-26 18:57:33 +0000 Tim-Philipp Müller <tim@centricular.net>
20786 tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
20787 Original commit message from CVS:
20788 * tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
20789 GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
20790 -DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
20793 2007-10-26 12:07:14 +0000 Christian Schaller <uraeus@gnome.org>
20795 * gst-plugins-base.spec.in:
20797 Original commit message from CVS:
20800 2007-10-25 17:36:49 +0000 Wim Taymans <wim.taymans@gmail.com>
20802 gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
20803 Original commit message from CVS:
20804 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
20805 (gst_decode_bin_dispose), (gst_decode_bin_set_caps),
20806 (gst_decode_bin_set_subs_encoding),
20807 (gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
20808 (gst_decode_bin_get_property), (analyze_new_pad):
20809 Move subtitle encoding property to decodebin2 so that it can set the
20810 property value on all elements that it autoplugs and that require it.
20811 Make caps refcounting more consistent in get/set.
20812 * gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
20813 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
20814 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
20815 (gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
20816 (proxy_autoplug_continue_signal),
20817 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
20819 Proxy properties and relevant signals from the internal decodebin.
20820 Make properties MT safe.
20822 2007-10-25 15:10:59 +0000 Tim-Philipp Müller <tim@centricular.net>
20824 gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
20825 Original commit message from CVS:
20826 * gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
20827 * gst-libs/gst/tag/tags.c:
20828 Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
20829 GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
20830 * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
20831 Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
20832 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
20833 (gst_tag_to_vorbis_comments):
20834 Map new SORTNAME tags (these tags aren't even semi-official, so I'm
20835 just mapping everything I found in the wild) (#414539).
20837 2007-10-24 11:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
20839 gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
20840 Original commit message from CVS:
20841 Inspired by patch of: René Stadler <mail at renestadler dot de>
20842 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
20843 (gst_decode_bin_autoplug_continue),
20844 (gst_decode_bin_autoplug_factories),
20845 (gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
20846 (find_compatibles):
20847 * gst/playback/gstplay-marshal.list:
20848 Remove the autoplug-sort signal and replace it with a binding friendly
20849 autoplug-select signal.
20850 Add an autoplug-factories signal that can be used to generate a list of
20851 factories to try to autoplug.
20852 Add the GstPad to the autoplugging signal args as it might be needed to
20853 make a good factory selection.
20854 Fix up the marshallers for this. Fixes #407282.
20856 2007-10-23 14:23:14 +0000 Tim-Philipp Müller <tim@centricular.net>
20858 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...
20859 Original commit message from CVS:
20860 * gst-libs/gst/tag/gsttagdemux.c:
20861 Don't abort with an assertion if we receive a seek event with
20862 a start type of NONE (see launchpad bug #155878).
20864 2007-10-22 10:21:46 +0000 Wim Taymans <wim.taymans@gmail.com>
20866 sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
20867 Original commit message from CVS:
20868 * sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
20869 (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
20870 (gst_ximagesink_change_state), (gst_ximagesink_reset):
20871 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
20872 (gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
20873 (gst_xvimagesink_change_state), (gst_xvimagesink_reset):
20874 Make sure that before we clean up the X resources, we shutdown and join
20876 Also make sure the event thread does not shut down immediatly after
20877 startup because the running variable is not yet correctly set.
20880 2007-10-16 16:48:38 +0000 Wim Taymans <wim.taymans@gmail.com>
20882 gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
20883 Original commit message from CVS:
20884 * gst/playback/gstdecodebin.c: (new_pad), (type_found):
20885 Make the window for a race in typefind and shutting down smaller until
20886 we figure out the right locking here. Avoids #485753 usually.
20887 * gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
20888 Remove unneeded lock causing a race in typefind and shutting down.
20890 * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
20891 Also remove sinks when going to NULL because we might not complete the
20892 state change to PAUSED, causing the PAUSED->READY state change not to
20895 2007-10-16 15:33:31 +0000 Wim Taymans <wim.taymans@gmail.com>
20897 gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
20898 Original commit message from CVS:
20899 * gst-libs/gst/audio/gstbaseaudiosink.c:
20900 (gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
20901 Also explicitly release the ringbuffer when going to NULL because it
20902 is required in the setcaps function, before the state change to PAUSED
20905 2007-10-16 14:58:53 +0000 Tim-Philipp Müller <tim@centricular.net>
20907 tests/icles/: Does what it says on the tin.
20908 Original commit message from CVS:
20909 * tests/icles/.cvsignore:
20910 * tests/icles/Makefile.am:
20911 * tests/icles/stress-playbin.c:
20912 Does what it says on the tin.
20914 2007-10-15 11:38:39 +0000 Wim Taymans <wim.taymans@gmail.com>
20916 gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
20917 Original commit message from CVS:
20918 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
20919 Fix queue negotiation. See #486758.
20921 2007-10-12 10:52:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20923 Actual code change to go along with:
20924 Original commit message from CVS:
20925 Actual code change to go along with:
20926 2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
20927 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
20928 (gst_xvimagesink_xwindow_new),
20929 (gst_xvimagesink_update_colorbalance),
20930 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
20931 Fix handling of some of the X atoms. If the last parameter is True,
20932 XInternAtom won't create the atom if it doesn't exist, and therefore
20933 might return None. This causes X errors on Xv implementations that
20934 don't provide the colour balance attributes.
20936 2007-10-12 10:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20939 Remove stray character from the changelog.
20940 Original commit message from CVS:
20941 Remove stray character from the changelog.
20943 2007-10-12 10:33:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20946 I'm too lazy to comment this
20947 Original commit message from CVS:
20948 *** empty log message ***
20950 2007-10-11 18:24:09 +0000 Tim-Philipp Müller <tim@centricular.net>
20952 Extract vorbis comment LICENSE tags correctly.
20953 Original commit message from CVS:
20954 * gst-libs/gst/tag/gstvorbistag.c:
20955 * tests/check/libs/tag.c:
20956 Extract vorbis comment LICENSE tags correctly.
20958 2007-10-11 16:12:21 +0000 Jason Kivlighn <jkivlighn@gmail.com>
20960 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
20961 Original commit message from CVS:
20962 Patch by: Jason Kivlighn <jkivlighn gmail com>
20963 * gst-libs/gst/tag/gstid3tag.c:
20964 * tests/check/libs/tag.c:
20965 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
20967 2007-10-10 17:01:51 +0000 Tim-Philipp Müller <tim@centricular.net>
20969 gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
20970 Original commit message from CVS:
20971 * gst-libs/gst/tag/gsttagdemux.c:
20972 Don't error out when a buggy downstream element doesn't
20973 handle the newsegment event we send properly (especially
20974 not without posting a meaningful error message on the
20975 bus). See bug #471370 and launchpad bug #136264.
20977 2007-10-10 15:36:56 +0000 Wim Taymans <wim.taymans@gmail.com>
20979 gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
20980 Original commit message from CVS:
20981 * gst-libs/gst/audio/gstbaseaudiosink.c:
20982 (gst_base_audio_sink_drain):
20983 Use new basesink method to make our EOS drain interruptable.
20985 2007-10-10 09:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20987 gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
20988 Original commit message from CVS:
20989 * gst-libs/gst/rtp/gstrtppayloads.c:
20990 Fix silly search-replace oversight.
20992 2007-10-09 09:57:17 +0000 Laurent Glayal <spglegle@yahoo.fr>
20994 gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
20995 Original commit message from CVS:
20996 Patch by: Laurent Glayal <spglegle at yahoo dot fr>
20997 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
20998 (gst_basertppayload_set_outcaps):
20999 Fix caps memleak. Fixes #484989.
21001 2007-10-08 18:04:34 +0000 Wim Taymans <wim.taymans@gmail.com>
21003 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
21004 Original commit message from CVS:
21005 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21006 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
21009 2007-10-08 18:02:53 +0000 Wim Taymans <wim.taymans@gmail.com>
21011 gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
21012 Original commit message from CVS:
21013 * gst-libs/gst/audio/gstbaseaudiosrc.c:
21014 (gst_base_audio_src_create):
21015 Also handle the case where there is no clock set on the audio source,
21016 like in the unit tests.
21018 2007-10-08 17:40:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21020 gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
21021 Original commit message from CVS:
21022 * gst-libs/gst/rtp/gstrtppayloads.c:
21023 Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
21024 to avoid compiler warnings
21026 2007-10-08 17:12:32 +0000 Wim Taymans <wim.taymans@gmail.com>
21028 gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
21029 Original commit message from CVS:
21030 * gst/playback/gstdecodebin.c: (type_found),
21031 (gst_decode_bin_change_state):
21032 * gst/playback/gstdecodebin2.c: (type_found),
21033 (gst_decode_bin_change_state):
21034 Don't disconnect the have_type signal because we never reconnect it
21035 later on. Instead keep a variable to see if we already detected a type.
21037 2007-10-08 10:47:26 +0000 Wim Taymans <wim.taymans@gmail.com>
21039 gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
21040 Original commit message from CVS:
21041 * gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
21042 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
21044 Unlink the signal handler when we found the type, we're not going to do
21045 anything sensible with more type_found signals anyway.
21047 2007-10-08 06:07:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21049 ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
21050 Original commit message from CVS:
21051 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
21052 Use GIO function to get a list of supported URI schemes instead of
21053 hard coding something.
21055 2007-10-06 16:49:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21057 gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
21058 Original commit message from CVS:
21059 * gst-libs/gst/tag/gsttagdemux.c:
21062 2007-10-06 15:04:53 +0000 Tim-Philipp Müller <tim@centricular.net>
21064 gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
21065 Original commit message from CVS:
21066 * gst-libs/gst/tag/Makefile.am:
21067 * gst-libs/gst/tag/gsttagdemux.c:
21068 * gst-libs/gst/tag/gsttagdemux.h:
21069 API: add GstTagDemux base class for simple tag demuxers.
21070 * docs/libs/gst-plugins-base-libs-docs.sgml:
21071 * docs/libs/gst-plugins-base-libs-sections.txt:
21072 Add GstTagDemux to docs.
21074 2007-10-05 07:49:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21076 gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
21077 Original commit message from CVS:
21078 * gst-libs/gst/rtp/gstrtpbuffer.c:
21079 (gst_rtp_buffer_get_payload_subbuffer):
21080 Fix bug introduced with last commit which inverted the logic and
21081 caused all buffers to be dropped. Fixes #483620.
21082 Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
21084 2007-10-04 06:50:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21086 gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
21087 Original commit message from CVS:
21088 * gst-libs/gst/rtp/gstrtpbuffer.c:
21089 Replace g_return_if_val (as it could be disabled), with regular return
21092 2007-10-03 14:51:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21094 tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
21095 Original commit message from CVS:
21096 * tests/check/pipelines/simple-launch-lines.c:
21097 Print message name and not just number.
21099 2007-10-02 11:11:13 +0000 Wim Taymans <wim.taymans@gmail.com>
21101 gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
21102 Original commit message from CVS:
21103 * gst-libs/gst/audio/gstbaseaudiosink.c:
21104 (gst_base_audio_sink_async_play):
21105 When slaved to the clock, don't try to align a sample with the previous
21106 one when going to PLAYING again.
21108 2007-10-02 09:04:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21110 tests/examples/snapshot/snapshot.c: Fix the build.
21111 Original commit message from CVS:
21112 * tests/examples/snapshot/snapshot.c:
21115 2007-10-02 07:43:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21117 ext/gio/gstgiosink.c: Update to API changes in GIO.
21118 Original commit message from CVS:
21119 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
21120 Update to API changes in GIO.
21122 2007-10-01 16:33:00 +0000 Wim Taymans <wim.taymans@gmail.com>
21124 gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
21125 Original commit message from CVS:
21126 * gst-libs/gst/sdp/gstsdpmessage.h:
21127 Add RFC 3556 bandwidth modifiers.
21129 2007-10-01 13:37:31 +0000 Wim Taymans <wim.taymans@gmail.com>
21131 Update documentation.
21132 Original commit message from CVS:
21133 * docs/libs/gst-plugins-base-libs-docs.sgml:
21134 * docs/libs/gst-plugins-base-libs-sections.txt:
21135 * gst-libs/gst/rtp/gstrtppayloads.c:
21136 Update documentation.
21138 2007-10-01 13:22:14 +0000 Wim Taymans <wim.taymans@gmail.com>
21140 gst-libs/gst/rtp/: Added new file and header to deal with payload info.
21141 Original commit message from CVS:
21142 * gst-libs/gst/rtp/Makefile.am:
21143 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
21144 (gst_rtp_payload_info_for_name):
21145 * gst-libs/gst/rtp/gstrtppayloads.h:
21146 Added new file and header to deal with payload info.
21147 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
21148 (gst_rtp_buffer_default_clock_rate):
21149 * gst-libs/gst/rtp/gstrtpbuffer.h:
21150 Payload specific stuff is move to new headers.
21151 Implement _default_clock rate using the new payload function.
21152 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
21153 (gst_sdp_parse_line):
21154 * gst-libs/gst/sdp/gstsdpmessage.h:
21155 Add some more comments.
21157 2007-10-01 10:22:46 +0000 Wim Taymans <wim.taymans@gmail.com>
21159 gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
21160 Original commit message from CVS:
21161 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
21162 (sdp_check_header), (sdp_type_find), (plugin_init):
21163 Add typefind function for application/sdp.
21164 Remove some old dirac typefind code that was ifdeffed out.
21166 2007-09-29 12:04:02 +0000 Sébastien Moutte <sebastien@moutte.net>
21168 win32/common/libgstaudio.def: Add new exported functions.
21169 Original commit message from CVS:
21170 * win32/common/libgstaudio.def:
21171 Add new exported functions.
21172 * win32/vs6/grammar.dsp:
21173 Add autogeneration and copy of some autegenerated files from win32/common
21175 * win32/vs6/libgstaudioconvert.dsp:
21176 Add gstaudioquantize.c to the build.
21177 * win32/vs6/libgstinterfaces.dsp:
21178 Add videoorientation.c to the build.
21179 * win32/vs6/libgstriff.dsp:
21180 Add libgsttag to the link libraries list.
21181 * win32/vs6/libgstvolume.dsp:
21182 Add liboil to the link.
21183 * win32/vs6/gst_plugins_base.dsw:
21184 * win32/vs6/libgstrtsp.dsp:
21185 * win32/common/libgstrtsp.def:
21186 Add files to build libgstrtsp library.
21188 2007-09-29 07:01:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21190 ext/gio/: Some minor cleanup and allow setting the location only when the element is not playing or paused.
21191 Original commit message from CVS:
21192 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
21193 (gst_gio_sink_set_property), (gst_gio_sink_render):
21194 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
21195 (gst_gio_src_set_property):
21196 Some minor cleanup and allow setting the location only when the
21197 element is not playing or paused.
21199 2007-09-26 15:14:37 +0000 Wim Taymans <wim.taymans@gmail.com>
21201 tests/examples/snapshot/snapshot.c: Print error when pipeline failed to construct.
21202 Original commit message from CVS:
21203 * tests/examples/snapshot/snapshot.c: (main):
21204 Print error when pipeline failed to construct.
21206 2007-09-25 19:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
21208 Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
21209 Original commit message from CVS:
21211 * gst-libs/gst/tag/gstid3tag.c:
21212 * gst-libs/gst/tag/gstvorbistag.c:
21213 Add mappings for the new GST_TAG_COMPOSER for vorbis comments
21216 2007-09-25 11:54:09 +0000 Tim-Philipp Müller <tim@centricular.net>
21218 gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
21219 Original commit message from CVS:
21220 * gst-libs/gst/floatcast/floatcast.h:
21221 Don't include config.h in an installed public header, this
21222 might break compilation of applications that don't have such
21223 a header and doesn't necessarily do what it's supposed to do
21224 anyway (ie. check for the lrint/lrintf defines) (#442065).
21225 Add docs for the various macros and document how this header
21226 has to be used (link against libm, etc.); add a few FIXMEs;
21227 include math.h for non-c99 code path. Based on patch by
21230 2007-09-25 07:50:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21232 configure.ac: Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead of duplicating these macros in confi...
21233 Original commit message from CVS:
21235 Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
21236 of duplicating these macros in configure.ac.
21238 2007-09-22 17:58:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21240 po/: Updated translations to 0.10.14
21241 Original commit message from CVS:
21245 Updated translations to 0.10.14
21247 2007-09-22 17:57:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21251 Original commit message from CVS:
21254 2007-09-22 17:56:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21256 po/pl.po: Added Polish translation.
21257 Original commit message from CVS:
21258 translated by: Jakub Bogusz <qboosh@pld-linux.org>
21260 Added Polish translation.
21262 2007-09-22 17:55:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21264 po/fi.po: Added Finnish translation.
21265 Original commit message from CVS:
21266 translated by: Ilkka Tuohela <hile@iki.fi>
21268 Added Finnish translation.
21270 2007-09-22 17:54:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21272 po/es.po: Added Spanish translation.
21273 Original commit message from CVS:
21274 translated by: Jorge González González <aloriel@gmail.com>
21276 Added Spanish translation.
21278 2007-09-22 17:53:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21280 po/da.po: Added Danish translation.
21281 Original commit message from CVS:
21282 translated by: Mogens Jaeger <mogens@jaeger.tf>
21284 Added Danish translation.
21286 2007-09-22 17:52:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21288 po/zh_CN.po: Added Chinese (simplified) translation.
21289 Original commit message from CVS:
21290 translated by: Funda Wang <fundawang@linux.net.cn>
21292 Added Chinese (simplified) translation.
21294 2007-09-22 17:51:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21296 po/bg.po: Added Bulgarian translation.
21297 Original commit message from CVS:
21298 translated by: Alexander Shopov <ash@contact.bg>
21300 Added Bulgarian translation.
21302 2007-09-21 18:00:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21304 docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy.
21305 Original commit message from CVS:
21306 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
21308 * ext/gio/gstgiosink.h:
21309 * ext/gio/gstgiosrc.h:
21310 Mark private fields of the instance structs private.
21312 2007-09-21 17:31:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21314 docs/plugins/: Add the GIO plugin to the docs and do a make update while doing that.
21315 Original commit message from CVS:
21316 * docs/plugins/Makefile.am:
21317 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
21318 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
21319 * docs/plugins/gst-plugins-bad-plugins.args:
21320 * docs/plugins/gst-plugins-bad-plugins.signals:
21321 * docs/plugins/inspect/plugin-bz2.xml:
21322 * docs/plugins/inspect/plugin-cdxaparse.xml:
21323 * docs/plugins/inspect/plugin-dfbvideosink.xml:
21324 * docs/plugins/inspect/plugin-dtsdec.xml:
21325 * docs/plugins/inspect/plugin-equalizer.xml:
21326 * docs/plugins/inspect/plugin-faac.xml:
21327 * docs/plugins/inspect/plugin-faad.xml:
21328 * docs/plugins/inspect/plugin-filter.xml:
21329 * docs/plugins/inspect/plugin-freeze.xml:
21330 * docs/plugins/inspect/plugin-gio.xml:
21331 * docs/plugins/inspect/plugin-gsm.xml:
21332 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
21333 * docs/plugins/inspect/plugin-h264parse.xml:
21334 * docs/plugins/inspect/plugin-modplug.xml:
21335 * docs/plugins/inspect/plugin-mpeg2enc.xml:
21336 * docs/plugins/inspect/plugin-musepack.xml:
21337 * docs/plugins/inspect/plugin-musicbrainz.xml:
21338 * docs/plugins/inspect/plugin-nsfdec.xml:
21339 * docs/plugins/inspect/plugin-replaygain.xml:
21340 * docs/plugins/inspect/plugin-soundtouch.xml:
21341 * docs/plugins/inspect/plugin-spcdec.xml:
21342 * docs/plugins/inspect/plugin-spectrum.xml:
21343 * docs/plugins/inspect/plugin-speed.xml:
21344 * docs/plugins/inspect/plugin-tta.xml:
21345 * docs/plugins/inspect/plugin-videosignal.xml:
21346 * docs/plugins/inspect/plugin-xingheader.xml:
21347 * docs/plugins/inspect/plugin-xvid.xml:
21348 Add the GIO plugin to the docs and do a make update
21350 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
21351 Fix a small memleak.
21353 2007-09-21 17:07:56 +0000 René Stadler <mail@renestadler.de>
21355 Add a GIO/GVFS plugin with source and sink elements. This will only be enabled when --enable-experimental is given to...
21356 Original commit message from CVS:
21357 Patch by: René Stadler <mail at renestadler dot de>
21360 * ext/gio/Makefile.am:
21361 * ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek),
21362 (gst_gio_get_supported_protocols),
21363 (gst_gio_uri_handler_get_type_sink),
21364 (gst_gio_uri_handler_get_type_src),
21365 (gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri),
21366 (gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init),
21367 (gst_gio_uri_handler_do_init), (plugin_init):
21368 * ext/gio/gstgio.h:
21369 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
21370 (gst_gio_sink_class_init), (gst_gio_sink_init),
21371 (gst_gio_sink_finalize), (gst_gio_sink_set_property),
21372 (gst_gio_sink_get_property), (gst_gio_sink_start),
21373 (gst_gio_sink_stop), (gst_gio_sink_unlock),
21374 (gst_gio_sink_unlock_stop), (gst_gio_sink_event),
21375 (gst_gio_sink_render), (gst_gio_sink_query):
21376 * ext/gio/gstgiosink.h:
21377 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
21378 (gst_gio_src_class_init), (gst_gio_src_init),
21379 (gst_gio_src_finalize), (gst_gio_src_set_property),
21380 (gst_gio_src_get_property), (gst_gio_src_start),
21381 (gst_gio_src_stop), (gst_gio_src_get_size),
21382 (gst_gio_src_is_seekable), (gst_gio_src_unlock),
21383 (gst_gio_src_unlock_stop), (gst_gio_src_check_get_range),
21384 (gst_gio_src_create):
21385 * ext/gio/gstgiosrc.h:
21386 Add a GIO/GVFS plugin with source and sink elements. This will
21387 only be enabled when --enable-experimental is given to configure
21388 for now as the GIO API is not stable yet. Fixes #476916.
21390 2007-09-21 14:37:26 +0000 Wim Taymans <wim.taymans@gmail.com>
21392 gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
21393 Original commit message from CVS:
21394 * gst/playback/gstqueue2.c: (gst_queue_push_one):
21395 Fix compilation wrt printf arguments.
21397 2007-09-20 17:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
21399 examples/app/appsrc_ex.c: Fix compilation after changing the name of a method.
21400 Original commit message from CVS:
21401 * examples/app/appsrc_ex.c: (main):
21402 Fix compilation after changing the name of a method.
21404 2007-09-20 14:09:24 +0000 Wim Taymans <wim.taymans@gmail.com>
21406 Add simple snapshot example program using appsink.
21407 Original commit message from CVS:
21409 * tests/examples/Makefile.am:
21410 * tests/examples/snapshot/.cvsignore:
21411 * tests/examples/snapshot/Makefile.am:
21412 * tests/examples/snapshot/snapshot.c: (main):
21413 Add simple snapshot example program using appsink.
21415 2007-09-20 13:59:50 +0000 Wim Taymans <wim.taymans@gmail.com>
21417 gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
21418 Original commit message from CVS:
21419 * gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
21420 (gst_app_sink_class_init), (gst_app_sink_init),
21421 (gst_app_sink_dispose), (gst_app_sink_finalize),
21422 (gst_app_sink_set_property), (gst_app_sink_get_property),
21423 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
21424 (gst_app_sink_event), (gst_app_sink_getcaps),
21425 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
21426 (gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
21427 (gst_app_sink_pull_buffer):
21428 * gst-libs/gst/app/gstappsink.h:
21429 Add properties, signals and actions to access the element even without
21430 linking to the library.
21431 Fix some method names and signatures.
21433 2007-09-20 10:37:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21435 tests/check/generic/states.c: Improved state change unit test.
21436 Original commit message from CVS:
21437 * tests/check/generic/states.c:
21438 Improved state change unit test.
21440 2007-09-19 18:16:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21442 Ignore registries in any format.
21443 Original commit message from CVS:
21444 * docs/plugins/.cvsignore:
21445 * tests/check/.cvsignore:
21446 Ignore registries in any format.
21448 2007-09-19 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
21450 gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
21451 Original commit message from CVS:
21452 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21453 (gst_base_rtp_depayload_chain),
21454 (gst_base_rtp_depayload_set_gst_timestamp):
21455 Only copy timestamp on outgoing packets if the depayloader did not set
21457 Also copy duration on outgoing packets.
21459 2007-09-19 15:55:08 +0000 Wim Taymans <wim.taymans@gmail.com>
21461 gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
21462 Original commit message from CVS:
21463 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
21464 (gst_basertppayload_set_outcaps):
21465 Fix compilation because of missing %d in printf.
21466 When fixating caps, fixate what we can and throw away all remaining
21467 unfixed caps, subclasses should do something smart if they need to.
21469 2007-09-19 12:04:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21471 ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
21472 Original commit message from CVS:
21473 * ext/gnomevfs/gstgnomevfssrc.c:
21474 Improve debug logs a bit and be more verbose if things go wrong.
21476 2007-09-17 17:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21478 Fix a bunch of compile warnings shown with Forte.
21479 Original commit message from CVS:
21480 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
21481 (gst_text_overlay_set_property):
21482 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
21483 * gst-libs/gst/audio/gstbaseaudiosink.c:
21484 (gst_base_audio_sink_render):
21485 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
21486 (gst_rtcp_unix_to_ntp):
21487 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
21488 * gst/playback/gstqueue2.c:
21489 * tests/examples/seek/seek.c: (set_scale):
21490 Fix a bunch of compile warnings shown with Forte.
21491 * gst/audiorate/gstaudiorate.c:
21492 Always pull in config.h before including any system headers.
21494 2007-09-17 16:22:17 +0000 Wim Taymans <wim.taymans@gmail.com>
21496 gst/playback/gstqueue2.c: Also fix #476514 for queue2.
21497 Original commit message from CVS:
21498 * gst/playback/gstqueue2.c: (update_buffering),
21499 (gst_queue_locked_flush), (gst_queue_locked_enqueue),
21500 (gst_queue_handle_sink_event), (gst_queue_chain),
21501 (gst_queue_push_one), (gst_queue_sink_activate_push),
21502 (gst_queue_src_activate_push), (gst_queue_src_activate_pull):
21503 Also fix #476514 for queue2.
21505 2007-09-16 19:31:06 +0000 Wim Taymans <wim.taymans@gmail.com>
21507 gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
21508 Original commit message from CVS:
21509 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21510 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
21511 (gst_base_rtp_depayload_chain),
21512 (gst_base_rtp_depayload_handle_sink_event),
21513 (gst_base_rtp_depayload_push_full),
21514 (gst_base_rtp_depayload_set_gst_timestamp),
21515 (gst_base_rtp_depayload_change_state):
21516 Remove code to deal with RTP to GST time conversion, we now just copy
21517 the GST timestamp we receive to the outgoing buffers.
21518 Handle segment and flushes correctly.
21519 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
21520 When we have no valid input timestamp, use the previous rtp timestamp on
21521 the outgoing RTP packet instead of the RTP base time.
21523 2007-09-16 01:56:21 +0000 David Schleef <ds@schleef.org>
21525 ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
21526 Original commit message from CVS:
21527 * ext/alsa/gstalsa.c:
21528 * ext/alsa/gstalsadeviceprobe.c:
21529 * ext/alsa/gstalsamixer.c:
21530 * ext/alsa/gstalsasink.c:
21531 * ext/alsa/gstalsasrc.c:
21532 Change alsa alloca's to malloc to fix warnings on gcc-4.2.
21534 2007-09-15 18:41:27 +0000 Wim Taymans <wim.taymans@gmail.com>
21536 gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
21537 Original commit message from CVS:
21538 * gst-libs/gst/rtp/gstbasertppayload.c:
21539 (gst_basertppayload_set_outcaps), (gst_basertppayload_push):
21540 Add some debug info when negotiating caps.
21542 2007-09-15 00:29:11 +0000 Wim Taymans <wim.taymans@gmail.com>
21544 gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
21545 Original commit message from CVS:
21546 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
21547 A buffer with an empty payload is also a valid buffer.
21549 2007-09-14 20:52:00 +0000 Wim Taymans <wim.taymans@gmail.com>
21551 gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
21552 Original commit message from CVS:
21553 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
21554 (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
21555 (gst_basertppayload_change_state):
21556 Make sure we start our RTP timestamp from the random base RTP
21557 timestamp even if the buffer timestamp starts from some random value.
21559 2007-09-14 16:56:16 +0000 Wim Taymans <wim.taymans@gmail.com>
21561 Add simple exmple app to demonstrate starting and pausing live and non-live bins in a PLAYING pipeline.
21562 Original commit message from CVS:
21564 * tests/examples/Makefile.am:
21565 * tests/examples/dynamic/.cvsignore:
21566 * tests/examples/dynamic/Makefile.am:
21567 * tests/examples/dynamic/addstream.c: (create_stream),
21568 (pause_play_stream), (message_received), (eos_message_received),
21569 (perform_step), (main):
21570 Add simple exmple app to demonstrate starting and pausing live and
21571 non-live bins in a PLAYING pipeline.
21573 2007-09-14 10:42:00 +0000 Julien Moutte <julien@moutte.net>
21575 gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
21576 Original commit message from CVS:
21577 2007-09-14 Julien MOUTTE <julien@moutte.net>
21578 * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
21579 typefind for QCP files (RFC #3625)
21581 2007-09-13 22:52:09 +0000 Wim Taymans <wim.taymans@gmail.com>
21583 gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
21584 Original commit message from CVS:
21585 * gst-libs/gst/audio/gstbaseaudiosink.c:
21586 (gst_base_audio_sink_init):
21587 Disable pull mode scheduling, we're not ready for it yet and it subtly
21588 breaks a lot of things.
21590 2007-09-12 17:35:52 +0000 Tim-Philipp Müller <tim@centricular.net>
21592 tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
21593 Original commit message from CVS:
21594 * tests/check/elements/libvisual.c:
21595 Test all libvisual plugins, not just the first one; this reproduces
21596 bug #450336 quite easily. Looks like a problem with the 'jess'
21599 2007-09-12 17:15:12 +0000 Tim-Philipp Müller <tim@centricular.net>
21601 tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
21602 Original commit message from CVS:
21603 * tests/check/Makefile.am:
21604 * tests/check/elements/.cvsignore:
21605 * tests/check/elements/libvisual.c:
21606 Add basic libvisual test case in an attempt to reproduce bug #450336.
21607 Doesn't reproduce that bug, but some other crasher instead (invalid
21608 free), at least with make elements/libvisual.forever and the bumscope
21609 plugin on x86-64/gutsy. Leaving test disabled for now.
21611 2007-09-12 08:38:21 +0000 Peter Kjellerstedt <pkj@axis.com>
21613 gst/: Printf format fixes (#476128).
21614 Original commit message from CVS:
21615 Patch by: Peter Kjellerstedt <pkj at axis com>
21616 * gst-libs/gst/app/gstappsink.c:
21617 * gst/flv/gstflvdemux.c:
21618 * gst/flv/gstflvparse.c:
21619 * gst/interleave/deinterleave.c:
21620 * gst/switch/gstswitch.c:
21621 Printf format fixes (#476128).
21623 2007-09-11 19:07:57 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
21625 gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
21626 Original commit message from CVS:
21627 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
21628 * gst-libs/gst/rtsp/gstrtspconnection.c:
21629 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
21630 (read_body), (gst_rtsp_connection_receive):
21631 Make sure we can not cancel in the middle of receiving a message.
21634 2007-09-11 11:29:12 +0000 Josep Torra Valles <josep@fluendo.com>
21636 gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
21637 Original commit message from CVS:
21638 Patch by: Josep Torra Valles <josep@fluendo.com>
21639 * gst/playback/gstplaybasebin.c:
21640 Increase upper limit for audio queue a bit; fixes preroll problem
21641 with playbin and decodebin2 when playing a quicktime trailer with
21642 multichannel audio via http (#464666).
21644 2007-09-10 22:10:54 +0000 Wim Taymans <wim.taymans@gmail.com>
21646 gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
21647 Original commit message from CVS:
21648 * gst-libs/gst/audio/gstbaseaudiosrc.c:
21649 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
21650 (gst_base_audio_src_provide_clock),
21651 (gst_base_audio_src_set_property),
21652 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
21653 * gst-libs/gst/audio/gstbaseaudiosrc.h:
21654 Allow othe clocks than the internal clock to be used for the pipeline.
21655 Add property to disable clock provide.
21656 API: GstBaseAudioSrc::provide-clock
21658 2007-09-10 12:05:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21660 gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
21661 Original commit message from CVS:
21662 * gst/playback/gstdecodebin2.c:
21663 Don't leak request pads. Fixes #475395.
21665 2007-09-09 10:25:43 +0000 René Stadler <mail@renestadler.de>
21667 sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
21668 Original commit message from CVS:
21669 Patch by: René Stadler <mail at renestadler dot de>
21670 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
21671 (gst_ximage_buffer_class_init):
21672 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
21673 (gst_xvimage_buffer_class_init):
21674 Correctly chain up finalize with the parent class to prevent
21675 memory leaks. Fixes #474880.
21677 2007-09-09 04:08:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21679 Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
21680 Original commit message from CVS:
21681 * gst/volume/gstvolume.c: (volume_choose_func):
21682 * tests/check/elements/volume.c: (GST_START_TEST):
21683 Revert the latest change: floating point samples are allowed to
21684 have any value, not only values in the range [-1,1]. Thanks to Andy
21685 Wingo for noticing.
21686 Also fix processing of int32 samples with volumes > 4 by making the
21687 unity value smaller which prevents overflows.
21689 2007-09-07 17:37:03 +0000 Tim-Philipp Müller <tim@centricular.net>
21691 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
21692 Original commit message from CVS:
21693 * gst-libs/gst/rtp/gstrtpbuffer.c:
21694 * tests/check/libs/rtp.c:
21695 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
21697 2007-09-07 16:46:05 +0000 Haakon Sporsheim <haakon.sporsheim@tandberg.com>
21699 gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
21700 Original commit message from CVS:
21701 Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
21702 * gst-libs/gst/rtp/gstrtpbuffer.c:
21703 Fix up GstRTPHeader helper struct so that compilers will not under
21704 any circumstances add padding in between our fields, as currently
21705 happens with MSVC on win32, because that would lead to us sending
21706 out RTP payloads with broken RTP headers (#471194).
21707 Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
21708 * tests/check/Makefile.am:
21709 * tests/check/libs/.cvsignore:
21710 * tests/check/libs/rtp.c:
21711 Add some simple unit tests for GstRTPBuffer. Some are disabled
21712 because the code tested still needs fixing (set_csrc() does not work).
21714 2007-09-07 15:05:24 +0000 Christian Schaller <uraeus@gnome.org>
21716 * gst-plugins-base.spec.in:
21717 update spec file to include latest RTSP libraries and headers and more
21718 Original commit message from CVS:
21719 update spec file to include latest RTSP libraries and headers and more
21721 2007-09-07 12:41:01 +0000 Tim-Philipp Müller <tim@centricular.net>
21723 win32/: Add rtsp enumtypes (#474384) and update others.
21724 Original commit message from CVS:
21726 * win32/common/gstrtsp-enumtypes.c:
21727 * win32/common/gstrtsp-enumtypes.h:
21728 * win32/common/interfaces-enumtypes.c:
21729 * win32/common/interfaces-enumtypes.h:
21730 * win32/common/multichannel-enumtypes.c:
21731 Add rtsp enumtypes (#474384) and update others.
21733 2007-09-06 20:31:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21735 configure.ac: Fix configure check for HAVE_LIBXML_HTML.
21736 Original commit message from CVS:
21738 Fix configure check for HAVE_LIBXML_HTML.
21740 2007-09-06 12:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
21742 tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
21743 Original commit message from CVS:
21744 * tests/check/libs/.cvsignore:
21745 Ignore more, in case the build bots work again one day.
21747 2007-09-06 07:00:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21749 Add libgstfft, a FFT library based on Kiss FFT which is
21750 Original commit message from CVS:
21751 Reviewed by: Stefan Kost <ensonic@users.sf.net>
21753 * gst-libs/gst/Makefile.am:
21754 * gst-libs/gst/fft/Makefile.am:
21755 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
21756 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
21757 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
21758 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
21759 * gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
21760 * gst-libs/gst/fft/gstfft.h:
21761 * gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
21762 (gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
21763 (gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
21764 * gst-libs/gst/fft/gstfftf32.h:
21765 * gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
21766 (gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
21767 (gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
21768 * gst-libs/gst/fft/gstfftf64.h:
21769 * gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
21770 (gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
21771 (gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
21772 * gst-libs/gst/fft/gstffts16.h:
21773 * gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
21774 (gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
21775 (gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
21776 * gst-libs/gst/fft/gstffts32.h:
21777 * gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
21778 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
21779 (kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
21780 (kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
21781 * gst-libs/gst/fft/kiss_fft_f32.h:
21782 * gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
21783 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
21784 (kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
21785 (kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
21786 * gst-libs/gst/fft/kiss_fft_f64.h:
21787 * gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
21788 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
21789 (kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
21790 (kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
21791 * gst-libs/gst/fft/kiss_fft_s16.h:
21792 * gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
21793 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
21794 (kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
21795 (kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
21796 * gst-libs/gst/fft/kiss_fft_s32.h:
21797 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
21798 (kiss_fftr_f32), (kiss_fftri_f32):
21799 * gst-libs/gst/fft/kiss_fftr_f32.h:
21800 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
21801 (kiss_fftr_f64), (kiss_fftri_f64):
21802 * gst-libs/gst/fft/kiss_fftr_f64.h:
21803 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
21804 (kiss_fftr_s16), (kiss_fftri_s16):
21805 * gst-libs/gst/fft/kiss_fftr_s16.h:
21806 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
21807 (kiss_fftr_s32), (kiss_fftri_s32):
21808 * gst-libs/gst/fft/kiss_fftr_s32.h:
21809 * gst-libs/gst/fft/kiss_version:
21810 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
21811 * pkgconfig/gstreamer-plugins-base.pc.in:
21812 Add libgstfft, a FFT library based on Kiss FFT which is
21813 BSD licensed. Supported sample formats are int16, int32,
21814 float and double. For those formats a real FFT and IFFT
21815 can be done, different windowing functions can be applied
21816 and functions for extracting the magnitude and phase exist.
21818 * docs/libs/Makefile.am:
21819 * docs/libs/gst-plugins-base-libs-docs.sgml:
21820 * docs/libs/gst-plugins-base-libs-sections.txt:
21821 Integrate libgstfft into the docs.
21822 * tests/check/Makefile.am:
21823 * tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
21824 Add unit tests for libgstfft, currently only testing the FFT.
21825 Unit tests for IFFT will follow soon.
21827 2007-09-05 23:07:40 +0000 Peter Kjellerstedt <pkj@axis.com>
21829 gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
21830 Original commit message from CVS:
21831 Patch by: Peter Kjellerstedt <pkj at axis com>
21832 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
21833 (gst_sdp_message_init), (gst_sdp_message_uninit),
21834 (is_multicast_address), (gst_sdp_message_as_text),
21835 (gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
21836 (gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
21837 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
21838 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
21839 (gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
21840 (gst_sdp_media_init), (gst_sdp_media_uninit),
21841 (gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
21842 (gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
21843 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
21844 (gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
21845 (gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
21846 * gst-libs/gst/sdp/gstsdpmessage.h:
21847 Separate INIT_ARRAY() and related macros into two versions, one for
21848 structures and one for pointers (e.g., INIT_ARRAY() and
21849 INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
21850 lists of emails and phone numbers.
21851 Add missing const as appropriate.
21852 Change all gint to guint since they all actually represent unsigned
21854 Do not use time as a variable name as it shadows the global time().
21855 Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
21856 Actually implement gst_sdp_message_add_time().
21857 Make gst_sdp_message_add_time() take repeat times as an argument.
21858 Store repeat times in GstSDPTime as a GArray rather than as gchar**.
21859 Corrected the definition of gst_sdp_media_get_bandwidth() (was
21860 misspelled as badwidth).
21861 gst-indented and a little clean up. Fixes #471067.
21863 2007-09-05 21:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21865 gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
21866 Original commit message from CVS:
21867 * gst/volume/gstvolume.c: (volume_choose_func),
21868 (volume_process_double), (volume_process_double_clamp),
21869 (volume_process_float_clamp):
21870 Correctly clamp float/double samples in the [-1.0,1.0] range to
21871 prevent weird effects.
21872 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
21873 Add unit tests for all samples types that had none before.
21875 2007-09-05 14:09:15 +0000 Tim-Philipp Müller <tim@centricular.net>
21877 gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
21878 Original commit message from CVS:
21879 * gst-libs/gst/rtp/gstrtpbuffer.c:
21880 Need to include stdlib.h for abs() here too.
21882 2007-09-05 14:01:25 +0000 Tim-Philipp Müller <tim@centricular.net>
21884 gst/playback/gststreaminfo.c: Fix build.
21885 Original commit message from CVS:
21886 * gst/playback/gststreaminfo.c:
21889 2007-09-05 10:32:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21891 gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
21892 Original commit message from CVS:
21893 * gst/playback/gststreaminfo.c:
21894 Clean up some half-disabled code and comment.
21896 2007-09-04 16:18:48 +0000 Wim Taymans <wim.taymans@gmail.com>
21898 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
21899 Original commit message from CVS:
21900 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
21901 (gst_base_rtp_payload_audio_handle_event):
21902 Return FALSE from the event handler to let the parent class handle the
21904 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21905 (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
21906 Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
21907 * gst-libs/gst/rtp/gstbasertppayload.c:
21908 Bump the MTU to 1400.
21910 2007-09-04 01:50:55 +0000 Johan Dahlin <johan@gnome.org>
21912 gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
21913 Original commit message from CVS:
21914 2007-09-03 Johan Dahlin <jdahlin@async.com.br>
21915 * gst/typefind/gsttypefindfunctions.c (plugin_init):
21916 Add an audio/x-nsf typefind function for the nsfdec element.
21918 2007-09-03 20:46:38 +0000 Renato Filho <renato.filho@indt.org.br>
21920 gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
21921 Original commit message from CVS:
21922 * gst/playback/gstplaybasebin.c:
21923 Included "myth://" on stream_uris list for enable buffering to mythtv files
21925 2007-09-03 19:31:11 +0000 Wim Taymans <wim.taymans@gmail.com>
21927 Fix parsing of RB blocks.
21928 Original commit message from CVS:
21929 * docs/libs/gst-plugins-base-libs-sections.txt:
21930 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
21931 (gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
21932 (gst_rtcp_unix_to_ntp):
21933 * gst-libs/gst/rtp/gstrtcpbuffer.h:
21934 Fix parsing of RB blocks.
21936 Added helper functions to convert to/from UNIX and NTP time.
21937 API: gst_rtcp_ntp_to_unix()
21938 API: gst_rtcp_unix_to_ntp()
21939 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
21940 (gst_rtp_buffer_get_header_len),
21941 (gst_rtp_buffer_get_extension_data),
21942 (gst_rtp_buffer_get_payload_subbuffer),
21943 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
21944 (gst_rtp_buffer_ext_timestamp):
21945 * gst-libs/gst/rtp/gstrtpbuffer.h:
21946 Fix some more docs.
21947 Implement handling of packets with extensions.
21948 Fix padding check in _validate().
21949 Added function to get extension data.
21950 API: gst_rtp_buffer_get_header_len()
21951 API: gst_rtp_buffer_get_extension_data()
21953 2007-09-03 19:19:35 +0000 Wim Taymans <wim.taymans@gmail.com>
21955 gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
21956 Original commit message from CVS:
21957 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21958 (gst_base_rtp_depayload_class_init),
21959 (gst_base_rtp_depayload_set_gst_timestamp):
21960 Add some more docs for the queue-delay property and fix a typo in a
21962 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
21965 2007-09-03 19:17:33 +0000 Wim Taymans <wim.taymans@gmail.com>
21967 gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
21968 Original commit message from CVS:
21969 * gst-libs/gst/audio/gstbaseaudiosink.c:
21970 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
21971 (gst_base_audio_sink_change_state):
21972 When skew slaving, try to hover around the middle of a segment so that
21973 we at most drift by half a segment.
21974 If we are aligning in the oposite direction of the clock skew, we don't
21977 2007-08-31 21:07:20 +0000 Wim Taymans <wim.taymans@gmail.com>
21979 gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
21980 Original commit message from CVS:
21981 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21982 (gst_base_rtp_depayload_setcaps),
21983 (gst_base_rtp_depayload_set_gst_timestamp):
21984 Be less silly with the segment start, just apply the clock-base to the
21987 2007-08-31 15:58:30 +0000 Wim Taymans <wim.taymans@gmail.com>
21989 gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
21990 Original commit message from CVS:
21991 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21992 (gst_base_rtp_depayload_class_init),
21993 (gst_base_rtp_depayload_finalize),
21994 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
21995 (gst_base_rtp_depayload_handle_sink_event),
21996 (gst_base_rtp_depayload_set_gst_timestamp),
21997 (gst_base_rtp_depayload_change_state):
21998 * gst-libs/gst/rtp/gstbasertpdepayload.h:
21999 Deprecate the queue handling thread thing and remove the code.
22000 Use new method to calculate the extended timestamp.
22002 2007-08-31 15:21:13 +0000 Wim Taymans <wim.taymans@gmail.com>
22004 gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
22005 Original commit message from CVS:
22006 * gst-libs/gst/rtp/gstrtcpbuffer.c:
22007 (gst_rtcp_packet_sdes_copy_entry):
22008 Use g_strndup which does exactly what we want.
22009 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
22010 (gst_rtp_buffer_ext_timestamp):
22011 * gst-libs/gst/rtp/gstrtpbuffer.h:
22012 Add helper function to compare seqnums.
22013 Add helper function to calculate extended timestamps.
22014 API: gst_rtp_buffer_compare_seqnum()
22015 API: gst_rtp_buffer_ext_timestamp()
22017 2007-08-30 21:59:23 +0000 Wim Taymans <wim.taymans@gmail.com>
22019 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
22020 Original commit message from CVS:
22021 * gst-libs/gst/rtp/gstrtcpbuffer.c:
22022 (gst_rtcp_packet_sdes_get_entry),
22023 (gst_rtcp_packet_sdes_copy_entry):
22024 * gst-libs/gst/rtp/gstrtcpbuffer.h:
22025 Fix and document SDES item data function.
22026 Add new function that makes a proper copy of SDES item data.
22027 API: gst_rtcp_packet_sdes_copy_entry()
22029 2007-08-30 07:29:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22031 The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
22032 Original commit message from CVS:
22035 The tcp and subparse plugins are under gst, but not totaly free of
22036 dependencies. Handle selection inconfigure.ac, so that they show up
22037 on the final list of what is build and what is not. Maybe they should
22038 better be moved to ext.
22040 2007-08-30 06:58:46 +0000 Daniel Díaz <yosoy@danieldiaz.org>
22042 Check if libxml provides HTML parser which subparse needs.
22043 Original commit message from CVS:
22044 Patch by: Daniel Díaz <yosoy@danieldiaz.org>
22047 Check if libxml provides HTML parser which subparse needs.
22050 2007-08-29 14:22:04 +0000 Tim-Philipp Müller <tim@centricular.net>
22052 ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
22053 Original commit message from CVS:
22054 * ext/alsa/gstalsa.c:
22055 Fix typo and compilation on big endian systems.
22057 2007-08-29 12:16:46 +0000 Tim-Philipp Müller <tim@centricular.net>
22059 gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
22060 Original commit message from CVS:
22061 * gst/subparse/gstssaparse.c:
22062 Convert SSA newline codes into actual newline characters (#470766).
22064 2007-08-28 14:58:17 +0000 Tim-Philipp Müller <tim@centricular.net>
22066 API: also add gst_install_plugins_supported() while we're at it (see #470456).
22067 Original commit message from CVS:
22068 * docs/libs/gst-plugins-base-libs-sections.txt:
22069 * gst-libs/gst/pbutils/install-plugins.c:
22070 * gst-libs/gst/pbutils/install-plugins.h:
22071 * tests/check/libs/pbutils.c:
22072 API: also add gst_install_plugins_supported() while we're at it
22075 2007-08-28 14:23:55 +0000 Tim-Philipp Müller <tim@centricular.net>
22077 API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
22078 Original commit message from CVS:
22079 * docs/libs/gst-plugins-base-libs-sections.txt:
22080 * gst-libs/gst/pbutils/missing-plugins.c:
22081 * gst-libs/gst/pbutils/missing-plugins.h:
22082 * tests/check/libs/pbutils.c:
22083 API: add gst_missing_*_installer_detail_new() convenience API so
22084 that applications that know exactly what they're missing can request
22085 installer detail strings for those items directly instead of having
22086 to first create a dummy missing-plugin message and then get the
22087 installer detail string from that. Fixes #470456.
22089 2007-08-27 11:59:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22091 gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
22092 Original commit message from CVS:
22093 * gst/playback/gstdecodebin.c: (close_pad_link):
22094 We need to set up delayed-linking whenever the caps are non-fixed,
22095 not just when there are multiple types - use gst_pad_is_fixed()
22098 2007-08-26 14:14:33 +0000 Tim-Philipp Müller <tim@centricular.net>
22100 gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
22101 Original commit message from CVS:
22102 * gst-libs/gst/pbutils/missing-plugins.c:
22103 (gst_missing_plugin_message_get_installer_detail):
22104 Add missing separator in PID fallback case.
22106 2007-08-24 15:28:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22108 ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
22109 Original commit message from CVS:
22110 * ext/alsa/Makefile.am:
22111 There is no GST_PLUGINS_BASE_LIBS defined.
22112 * ext/alsa/gstalsa.c:
22113 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
22114 * ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
22115 Add support for ALSA 24-bit formats.
22116 snd_pcm_delay can return an error code, especially
22117 during XRUNS. In that case, the best we can do is assume
22119 * gst/audioconvert/Makefile.am:
22120 Add flags from -base before any more-remote dependencies.
22122 2007-08-23 20:45:45 +0000 Davyd <davyd@madeley.id.au>
22124 gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
22125 Original commit message from CVS:
22126 Based on a patch by: Davyd <davyd at madeley dot id dot au>
22127 * gst/volume/gstvolume.c: (volume_choose_func),
22128 (volume_update_real_volume), (gst_volume_set_volume),
22129 (gst_volume_init), (volume_process_int32),
22130 (volume_process_int32_clamp), (volume_process_int24),
22131 (volume_process_int24_clamp), (volume_process_int16),
22132 (volume_process_int16_clamp), (volume_process_int8),
22133 (volume_process_int8_clamp), (volume_update_volume), (plugin_init):
22134 * gst/volume/gstvolume.h:
22135 Add support for int32, int24 and int8 to the volume element.
22138 2007-08-23 12:37:42 +0000 Tim-Philipp Müller <tim@centricular.net>
22140 tests/examples/Makefile.am: Fix even more.
22141 Original commit message from CVS:
22142 * tests/examples/Makefile.am:
22145 2007-08-23 10:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22147 Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
22148 Original commit message from CVS:
22150 * docs/libs/Makefile.am:
22151 * docs/libs/gst-plugins-base-libs-docs.sgml:
22152 * docs/libs/gst-plugins-base-libs-sections.txt:
22153 * ext/gnomevfs/gstgnomevfssrc.c:
22154 * ext/gnomevfs/gstgnomevfssrc.h:
22155 * gst-libs/gst/Makefile.am:
22156 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
22157 * pkgconfig/gstreamer-plugins-base.pc.in:
22158 * sys/v4l/v4lsrc_calls.c:
22159 * tests/examples/Makefile.am:
22160 * win32/common/config.h:
22161 Revert unwanted commit. many thanks to moap. I want a fix for
22162 https://thomas.apestaart.org/moap/trac/ticket/239
22164 2007-08-23 08:33:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22168 * docs/libs/Makefile.am:
22169 * docs/libs/gst-plugins-base-libs-docs.sgml:
22170 * docs/libs/gst-plugins-base-libs-sections.txt:
22171 * ext/gnomevfs/gstgnomevfssrc.c:
22172 * ext/gnomevfs/gstgnomevfssrc.h:
22173 * gst-libs/gst/Makefile.am:
22174 * gst-libs/gst/audio/gstaudiofilter.h:
22175 * gst/typefind/gsttypefindfunctions.c:
22176 * gst/volume/gstvolume.c:
22177 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
22178 * pkgconfig/gstreamer-plugins-base.pc.in:
22179 * sys/v4l/v4lsrc_calls.c:
22180 * tests/examples/Makefile.am:
22181 * win32/common/config.h:
22182 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:
22184 2007-08-22 15:29:04 +0000 Wim Taymans <wim.taymans@gmail.com>
22186 gst-libs/gst/audio/audio.c: Clarify the docs a little.
22187 Original commit message from CVS:
22188 * gst-libs/gst/audio/audio.c:
22189 Clarify the docs a little.
22191 2007-08-22 11:20:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22193 gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
22194 Original commit message from CVS:
22195 * gst/volume/gstvolume.c:
22196 Enable liboil for float and add more details about problems with
22199 2007-08-21 15:43:24 +0000 Wim Taymans <wim.taymans@gmail.com>
22201 sys/v4l/gstv4lsrc.c: Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
22202 Original commit message from CVS:
22203 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
22204 Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
22206 2007-08-21 12:08:43 +0000 Wim Taymans <wim.taymans@gmail.com>
22208 ext/vorbis/vorbisdec.c: When calculating the first timestamp of the buffers, don't go below 0 and clip the samples be...
22209 Original commit message from CVS:
22210 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
22211 When calculating the first timestamp of the buffers, don't go below 0
22212 and clip the samples because the offset was on the eos page.
22215 2007-08-21 11:42:39 +0000 Wim Taymans <wim.taymans@gmail.com>
22217 ext/ogg/gstoggdemux.c: Also submit the eos page when trying to find the first timestamp.
22218 Original commit message from CVS:
22219 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
22220 (gst_ogg_demux_collect_chain_info):
22221 Also submit the eos page when trying to find the first timestamp.
22224 2007-08-17 15:24:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
22226 gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
22227 Original commit message from CVS:
22228 * gst-libs/gst/audio/audio.h:
22229 Use gst_util_uint64_scale() instead of doing the math
22230 with double for GST_FRAMES_TO_CLOCK_TIME() and
22231 GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
22232 prevents rounding errors. Fixes #467667.
22234 2007-08-17 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
22236 gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
22237 Original commit message from CVS:
22238 * gst-libs/gst/rtsp/gstrtspconnection.c:
22239 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
22240 (gst_rtsp_connection_read), (gst_rtsp_connection_poll):
22241 * gst-libs/gst/rtsp/gstrtspconnection.h:
22243 On shutdown, don't read the control socket yet.
22244 Set timeout value correctly in all cases.
22245 Add function to check if the server accepts reads or writes.
22246 API: gst_rtsp_connection_poll()
22247 * gst-libs/gst/rtsp/gstrtspdefs.h:
22248 Fix compilation with -pedantic.
22249 Add enum for _poll.
22251 2007-08-16 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
22253 gst-libs/gst/app/gstappsink.c: Override the preroll vmethod instead of overriding the render method twice.
22254 Original commit message from CVS:
22255 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
22256 Override the preroll vmethod instead of overriding the render method
22259 2007-08-16 16:06:21 +0000 Olivier Crete <tester@tester.ca>
22261 gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
22262 Original commit message from CVS:
22263 Patch by: Olivier Crete <tester at tester ca>
22264 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
22265 (gst_basertppayload_getcaps):
22266 * gst-libs/gst/rtp/gstbasertppayload.h:
22267 Add getcaps vfunc to basertppayload. See #465146.
22269 2007-08-16 11:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
22271 gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
22272 Original commit message from CVS:
22273 * gst/playback/gstplaybasebin.c: (queue_threshold_reached):
22274 Only post buffering messages when we are a stream.
22276 2007-08-15 17:05:45 +0000 Tim-Philipp Müller <tim@centricular.net>
22278 gst-libs/gst/pbutils/: Small docs fix and addition.
22279 Original commit message from CVS:
22280 * gst-libs/gst/pbutils/install-plugins.c:
22281 * gst-libs/gst/pbutils/missing-plugins.c:
22282 Small docs fix and addition.
22284 2007-08-14 17:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
22286 gst-libs/gst/app/gstappsink.c: Don't use new API.
22287 Original commit message from CVS:
22288 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
22291 2007-08-14 17:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
22293 gst-libs/gst/app/gstappsink.*: Make love to appsink.
22294 Original commit message from CVS:
22295 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
22296 (gst_app_sink_class_init), (gst_app_sink_dispose),
22297 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
22298 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
22299 (gst_app_sink_render), (gst_app_sink_get_caps),
22300 (gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
22301 (gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
22302 * gst-libs/gst/app/gstappsink.h:
22303 Make love to appsink.
22304 Make it support pulling of the preroll buffer.
22305 Add docs and debug statements.
22306 Fix some races wrt to EOS handling and stopping.
22308 Implement FLUSHING.
22309 API: gst_app_sink_pull_preroll()
22311 2007-08-13 15:37:29 +0000 Tim-Philipp Müller <tim@centricular.net>
22313 tests/icles/: Add a dumb little test for textoverlay alignments.
22314 Original commit message from CVS:
22315 * tests/icles/.cvsignore:
22316 * tests/icles/Makefile.am:
22317 * tests/icles/test-textoverlay.c:
22318 Add a dumb little test for textoverlay alignments.
22320 2007-08-13 15:26:54 +0000 Dan Williams <dcbw@redhat.com>
22322 ext/pango/gsttextoverlay.*: API: add "line-alignment" property (#459334). Add gtk-doc blurb for "silent" property so ...
22323 Original commit message from CVS:
22324 Patch by: Dan Williams <dcbw redhat com>
22325 * ext/pango/gsttextoverlay.c:
22326 * ext/pango/gsttextoverlay.h:
22327 API: add "line-alignment" property (#459334). Add gtk-doc blurb for
22328 "silent" property so there's a Since tag in the API reference.
22330 2007-08-13 11:21:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
22334 Original commit message from CVS:
22337 2007-08-12 16:30:36 +0000 Wim Taymans <wim.taymans@gmail.com>
22339 gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
22340 Original commit message from CVS:
22341 * gst-libs/gst/rtp/gstbasertppayload.c:
22342 (gst_basertppayload_set_outcaps):
22343 * gst-libs/gst/rtp/gstbasertppayload.h:
22344 Improve caps negotiation so that downstream elements can confiure
22345 certain RTP properties by fixing them on the caps. See #465146.
22348 2007-08-11 12:39:51 +0000 Tim-Philipp Müller <tim@centricular.net>
22350 Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
22351 Original commit message from CVS:
22352 * docs/libs/gst-plugins-base-libs-sections.txt:
22353 * gst-libs/gst/rtp/gstbasertpdepayload.c:
22354 * gst-libs/gst/rtp/gstbasertpdepayload.h:
22355 Mark as deprecated some macros which were presumably meant to be
22356 private API and accidentally exposed in the public header file.
22357 Also actually _init() lock (only works at the moment because the
22358 struct is zeroed out when created and the initial values in the
22359 mutex struct are zeroes too). (#459585)
22361 2007-08-10 17:35:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22363 docs/libs/Makefile.am: Remove cruft and do some cleanups.
22364 Original commit message from CVS:
22365 * docs/libs/Makefile.am:
22366 Remove cruft and do some cleanups.
22367 * docs/libs/gst-plugins-base-libs-docs.sgml:
22368 Prepare for comming gtkdoc features (rebase against online docs).
22370 2007-08-10 13:55:44 +0000 Michael Smith <msmith@xiph.org>
22372 gst/audiorate/gstaudiorate.c: Debug output fixes.
22373 Original commit message from CVS:
22374 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
22375 Debug output fixes.
22376 * tests/check/elements/audiorate.c: (do_perfect_stream_test),
22378 Change the number of buffers used; 500 is too many and leads to
22381 2007-08-10 10:08:05 +0000 Tim-Philipp Müller <tim@centricular.net>
22383 gst/: Printf format fixes (#465028).
22384 Original commit message from CVS:
22385 * gst/playback/gstqueue2.c:
22386 * gst/videorate/gstvideorate.c:
22387 Printf format fixes (#465028).
22389 2007-08-09 15:44:02 +0000 Michael Smith <msmith@xiph.org>
22391 gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
22392 Original commit message from CVS:
22393 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
22394 If we have a large (> 1 second) discontinuity, push a series of
22395 smaller buffers rather than a single very large buffer. Avoids
22396 unreasonably large single buffer allocations when encountering a
22398 * tests/check/elements/audiorate.c: (GST_START_TEST),
22400 Add a test for this.
22402 2007-08-09 12:06:43 +0000 Josep Torra Valles <josep@fluendo.com>
22404 gst/playback/gstplaybasebin.c: Fixes: #465015
22405 Original commit message from CVS:
22406 * gst/playback/gstplaybasebin.c: (group_commit),
22407 (queue_remove_probe), (queue_threshold_reached):
22408 Patch by: Josep Torra Valles <josep@fluendo.com>
22410 Make sure we remove the check_queues buffer probe from the
22411 correct queue to avoid racily going back to "buffering 99%" when
22412 buffering is actually complete.
22413 Also, fix the spelling of Josep's surname in the ChangeLog.
22415 2007-08-09 11:37:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22417 ext/ogg/gstoggmux.c: Do not leak oggmux instance.
22418 Original commit message from CVS:
22419 * ext/ogg/gstoggmux.c:
22420 Do not leak oggmux instance.
22421 * ext/vorbis/vorbisenc.c:
22424 2007-08-09 10:51:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
22426 po/: Updated translations.
22427 Original commit message from CVS:
22433 Updated translations.
22435 2007-08-08 16:07:21 +0000 Yang Hong <hongyang@redflag-linux.com>
22437 ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
22438 Original commit message from CVS:
22439 patch by: Yang Hong <hongyang@redflag-linux.com>
22440 * ext/pango/gsttextoverlay.c:
22441 * ext/pango/gsttextoverlay.h:
22442 Add 'silent' property to GstTimeOverlay. Fixes #462979
22444 2007-08-08 15:05:22 +0000 Josep Torre Valles <josep@fluendo.com>
22446 Add connection-speed property. Fixes #464690.
22447 Original commit message from CVS:
22448 Patch by: Josep Torre Valles <josep@fluendo.com>
22449 * docs/plugins/gst-plugins-base-plugins.args:
22450 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
22451 (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
22452 (gst_uri_decode_bin_get_property), (gen_source_element):
22453 Add connection-speed property. Fixes #464690.
22455 2007-08-07 15:13:46 +0000 Damien Lespiau <damien.lespiau@gmail.com>
22457 Fix compilation on windows. Fixes #464320.
22458 Original commit message from CVS:
22459 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
22461 * gst-libs/gst/rtsp/Makefile.am:
22462 * gst-libs/gst/rtsp/gstrtspconnection.c:
22463 (gst_rtsp_connection_connect):
22464 Fix compilation on windows. Fixes #464320.
22466 2007-08-07 14:14:54 +0000 Josep Torre Valles <josep@fluendo.com>
22468 gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
22469 Original commit message from CVS:
22470 Patch by: Josep Torre Valles <josep@fluendo.com>
22471 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
22472 (gst_play_base_bin_init), (queue_threshold_reached),
22473 (gen_source_element), (setup_substreams),
22474 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
22475 (gst_play_base_bin_get_streaminfo_value_array):
22476 * gst/playback/gstplaybasebin.h:
22477 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
22478 (gst_play_bin_set_property), (gst_play_bin_get_property),
22479 (gst_play_bin_handle_redirect_message):
22480 Move connection-speed property from playbin to playbasebin so that we
22481 can also configure it in source elements that have the connection-speed
22482 property. Fixes #464028.
22483 Add some debug info here and there.
22485 2007-08-06 16:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
22487 gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
22488 Original commit message from CVS:
22489 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
22490 Properly respond to conversion queries. Fixes #464079.
22492 2007-08-03 19:53:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
22494 gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
22495 Original commit message from CVS:
22496 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
22497 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
22498 (gst_audio_test_src_init_sine_table),
22499 (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
22500 * gst/audiotestsrc/gstaudiotestsrc.h:
22501 Add float/double and int32 support to audiotestsrc. Fixes #460422.
22502 Also set the default volume to the default value specified in the
22505 2007-08-03 19:40:14 +0000 Jens Granseuer <jensgr@gmx.net>
22507 gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
22508 Original commit message from CVS:
22509 Patch by: Jens Granseuer <jensgr at gmx dot net>
22510 * gst/audioconvert/gstaudioquantize.c:
22511 Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
22513 2007-08-03 15:44:01 +0000 Wim Taymans <wim.taymans@gmail.com>
22515 gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
22516 Original commit message from CVS:
22517 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
22518 Add rdt manager for rdt transport.
22519 Fix parsing of RDT transport.
22521 2007-08-03 14:43:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22523 configure.ac: Back to CVS
22524 Original commit message from CVS:
22528 === release 0.10.14 ===
22530 2007-08-03 14:41:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22536 * docs/plugins/gst-plugins-base-plugins.args:
22537 * docs/plugins/inspect/plugin-adder.xml:
22538 * docs/plugins/inspect/plugin-alsa.xml:
22539 * docs/plugins/inspect/plugin-audioconvert.xml:
22540 * docs/plugins/inspect/plugin-audiorate.xml:
22541 * docs/plugins/inspect/plugin-audioresample.xml:
22542 * docs/plugins/inspect/plugin-audiotestsrc.xml:
22543 * docs/plugins/inspect/plugin-cdparanoia.xml:
22544 * docs/plugins/inspect/plugin-decodebin.xml:
22545 * docs/plugins/inspect/plugin-decodebin2.xml:
22546 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
22547 * docs/plugins/inspect/plugin-gdp.xml:
22548 * docs/plugins/inspect/plugin-gnomevfs.xml:
22549 * docs/plugins/inspect/plugin-libvisual.xml:
22550 * docs/plugins/inspect/plugin-ogg.xml:
22551 * docs/plugins/inspect/plugin-pango.xml:
22552 * docs/plugins/inspect/plugin-playbin.xml:
22553 * docs/plugins/inspect/plugin-subparse.xml:
22554 * docs/plugins/inspect/plugin-tcp.xml:
22555 * docs/plugins/inspect/plugin-theora.xml:
22556 * docs/plugins/inspect/plugin-typefindfunctions.xml:
22557 * docs/plugins/inspect/plugin-video4linux.xml:
22558 * docs/plugins/inspect/plugin-videorate.xml:
22559 * docs/plugins/inspect/plugin-videoscale.xml:
22560 * docs/plugins/inspect/plugin-videotestsrc.xml:
22561 * docs/plugins/inspect/plugin-volume.xml:
22562 * docs/plugins/inspect/plugin-vorbis.xml:
22563 * docs/plugins/inspect/plugin-ximagesink.xml:
22564 * docs/plugins/inspect/plugin-xvimagesink.xml:
22565 * gst-plugins-base.doap:
22566 * win32/common/config.h:
22568 Original commit message from CVS:
22571 2007-08-03 14:24:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22589 Original commit message from CVS:
22592 2007-07-27 17:37:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22594 tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
22595 Original commit message from CVS:
22596 * tests/check/libs/audio.c: (GST_START_TEST):
22597 Fix the test to reflect the behaviour of gst_audio_clip_buffer.
22599 2007-07-27 17:10:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22601 gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
22602 Original commit message from CVS:
22603 * gst-libs/gst/audio/audio.c:
22604 When clipping a buffer with no timestamp, assume it is
22605 within the segment without warnings.
22608 2007-07-27 11:16:23 +0000 Wim Taymans <wim.taymans@gmail.com>
22610 gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
22611 Original commit message from CVS:
22612 * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
22613 Fire the signal on the object, not the interface.
22615 2007-07-27 09:17:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22617 gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
22618 Original commit message from CVS:
22619 * gst-libs/gst/rtsp/.cvsignore:
22620 Ber. Don't include the full path, idiot.
22622 2007-07-27 08:29:29 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22624 gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
22625 Original commit message from CVS:
22626 * gst-libs/gst/rtsp/.cvsignore:
22627 Ignore generated files.
22629 2007-07-26 19:57:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22631 gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
22632 Original commit message from CVS:
22633 * gst-libs/gst/interfaces/Makefile.am:
22634 * gst-libs/gst/interfaces/interfaces-marshal.list:
22635 * gst-libs/gst/interfaces/rtspextension.c:
22636 * gst-libs/gst/interfaces/rtspextension.h:
22637 * gst-libs/gst/rtsp/Makefile.am:
22638 * gst-libs/gst/rtsp/gstrtsp.h:
22639 * gst-libs/gst/rtsp/gstrtspextension.c:
22640 (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
22641 (gst_rtsp_extension_detect_server),
22642 (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
22643 (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
22644 (gst_rtsp_extension_configure_stream),
22645 (gst_rtsp_extension_get_transports),
22646 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
22647 * gst-libs/gst/rtsp/gstrtspextension.h:
22648 * gst-libs/gst/rtsp/rtsp-marshal.list:
22649 Move the rtspextension.h interface into gstrtspextension.h
22650 as part of libgstrtsp instead of libgstinterfaces, because it's
22651 only for use within plugins, not applications.
22652 Add stuff to do the enum & marshal generation needed in libgstrtsp now.
22653 Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
22654 signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
22657 2007-07-26 15:48:01 +0000 Wim Taymans <wim.taymans@gmail.com>
22659 gst-libs/gst/interfaces/: Fix marshaller for the send signal.
22660 Original commit message from CVS:
22661 * gst-libs/gst/interfaces/Makefile.am:
22662 * gst-libs/gst/interfaces/interfaces-marshal.list:
22663 * gst-libs/gst/interfaces/rtspextension.c:
22664 (gst_rtsp_extension_iface_init),
22665 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
22666 * gst-libs/gst/interfaces/rtspextension.h:
22667 Fix marshaller for the send signal.
22668 Add URL to stream selection interface method.
22670 2007-07-26 15:35:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22672 gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
22673 Original commit message from CVS:
22674 * gst-libs/gst/riff/Makefile.am:
22675 Pull in our dependencies from -base before those from outside.
22677 2007-07-26 14:33:01 +0000 Wim Taymans <wim.taymans@gmail.com>
22679 API: gst_rtsp_base64_decode_ip()
22680 Original commit message from CVS:
22681 * docs/libs/gst-plugins-base-libs-sections.txt:
22682 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
22683 * gst-libs/gst/rtsp/gstrtspbase64.h:
22684 API: gst_rtsp_base64_decode_ip()
22685 Added function to decode Base64 in-place.
22687 2007-07-26 14:08:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22689 tests/check/libs/.cvsignore: Ignore the mixer test binary.
22690 Original commit message from CVS:
22691 * tests/check/libs/.cvsignore:
22692 Ignore the mixer test binary.
22694 2007-07-26 10:00:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22696 ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
22697 Original commit message from CVS:
22698 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
22699 Gratuitous comment change to trigger a rebuild on the buildbots.
22701 2007-07-25 18:20:36 +0000 Wim Taymans <wim.taymans@gmail.com>
22703 gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
22704 Original commit message from CVS:
22705 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
22706 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
22707 (gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
22708 (gst_sdp_media_get_format), (gst_sdp_media_get_information),
22709 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
22710 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
22711 (gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
22712 (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
22713 (gst_sdp_media_get_attribute_val):
22714 * gst-libs/gst/sdp/gstsdpmessage.h:
22715 Constify args where we can.
22717 2007-07-25 18:18:49 +0000 Wim Taymans <wim.taymans@gmail.com>
22719 gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
22720 Original commit message from CVS:
22721 * gst-libs/gst/interfaces/Makefile.am:
22722 * gst-libs/gst/interfaces/rtspextension.c:
22723 (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
22724 (gst_rtsp_extension_detect_server),
22725 (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
22726 (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
22727 (gst_rtsp_extension_configure_stream),
22728 (gst_rtsp_extension_get_transports),
22729 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
22730 * gst-libs/gst/interfaces/rtspextension.h:
22731 Move interface for RTSP extensions from -good to here.
22732 Added helper methods to invoke interface methods.
22734 2007-07-25 11:22:30 +0000 Wim Taymans <wim.taymans@gmail.com>
22736 Fix some more RTSP docs.
22737 Original commit message from CVS:
22738 * docs/libs/gst-plugins-base-libs-sections.txt:
22739 * gst-libs/gst/rtsp/gstrtspdefs.h:
22740 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
22741 (gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
22742 (gst_rtsp_message_init_response),
22743 (gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
22744 (gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
22745 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
22746 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
22747 (gst_rtsp_message_get_body), (dump_key_value):
22748 * gst-libs/gst/rtsp/gstrtspmessage.h:
22749 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
22750 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
22751 (gst_rtsp_range_parse):
22752 * gst-libs/gst/rtsp/gstrtsprange.h:
22753 * gst-libs/gst/rtsp/gstrtsptransport.c:
22754 * gst-libs/gst/rtsp/gstrtspurl.c:
22755 Fix some more RTSP docs.
22756 Add some missing methods for dealing with messages.
22758 2007-07-24 19:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
22760 Added beginnings of RTSP documentation.
22761 Original commit message from CVS:
22762 * docs/libs/gst-plugins-base-libs-docs.sgml:
22763 * docs/libs/gst-plugins-base-libs-sections.txt:
22764 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
22765 * gst-libs/gst/rtsp/gstrtspbase64.h:
22766 * gst-libs/gst/rtsp/gstrtspconnection.c:
22767 (gst_rtsp_connection_connect), (add_auth_header),
22768 (gst_rtsp_connection_write), (gst_rtsp_connection_send),
22769 (read_body), (gst_rtsp_connection_receive),
22770 (gst_rtsp_connection_next_timeout),
22771 (gst_rtsp_connection_reset_timeout),
22772 (gst_rtsp_connection_set_auth):
22773 * gst-libs/gst/rtsp/gstrtspconnection.h:
22774 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
22775 * gst-libs/gst/rtsp/gstrtspdefs.h:
22776 * gst-libs/gst/rtsp/gstrtspmessage.h:
22777 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
22778 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
22779 (gst_rtsp_range_parse):
22780 * gst-libs/gst/rtsp/gstrtspurl.h:
22781 Added beginnings of RTSP documentation.
22783 2007-07-24 17:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
22785 Document the SDP library.
22786 Original commit message from CVS:
22787 * docs/libs/Makefile.am:
22788 * docs/libs/gst-plugins-base-libs-docs.sgml:
22789 * docs/libs/gst-plugins-base-libs-sections.txt:
22790 * gst-libs/gst/sdp/gstsdp.h:
22791 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
22792 (gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
22793 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
22794 (gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
22795 (gst_sdp_message_get_attribute_val),
22796 (gst_sdp_message_add_attribute), (gst_sdp_media_new),
22797 (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
22798 (gst_sdp_media_get_media), (gst_sdp_media_set_media),
22799 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
22800 (gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
22801 (gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
22802 (gst_sdp_media_get_format), (gst_sdp_media_add_format),
22803 (gst_sdp_media_get_information), (gst_sdp_media_set_information),
22804 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
22805 (gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
22806 (gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
22807 (gst_sdp_media_set_key), (gst_sdp_media_get_key),
22808 (gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
22809 (gst_sdp_media_get_attribute_val_n),
22810 (gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
22811 (print_media), (gst_sdp_message_dump):
22812 * gst-libs/gst/sdp/gstsdpmessage.h:
22813 Document the SDP library.
22814 Add some of the missing SDPMedia methods.
22816 2007-07-24 11:52:56 +0000 Wim Taymans <wim.taymans@gmail.com>
22818 Move SDP and RTSP from helper objects in -good to a reusable library.
22819 Original commit message from CVS:
22821 * gst-libs/gst/Makefile.am:
22822 * gst-libs/gst/rtsp/Makefile.am:
22823 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
22824 * gst-libs/gst/rtsp/gstrtspbase64.h:
22825 * gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
22826 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
22827 (add_auth_header), (add_date_header), (gst_rtsp_connection_write),
22828 (gst_rtsp_connection_send), (read_line), (read_string), (read_key),
22829 (parse_response_status), (parse_request_line), (parse_line),
22830 (gst_rtsp_connection_read), (read_body),
22831 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
22832 (gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
22833 (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
22834 (gst_rtsp_connection_set_auth):
22835 * gst-libs/gst/rtsp/gstrtspconnection.h:
22836 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
22837 (gst_rtsp_strresult), (gst_rtsp_method_as_text),
22838 (gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
22839 (gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
22840 (gst_rtsp_find_method):
22841 * gst-libs/gst/rtsp/gstrtspdefs.h:
22842 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
22843 (gst_rtsp_message_new), (gst_rtsp_message_init),
22844 (gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
22845 (gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
22846 (gst_rtsp_message_init_data), (gst_rtsp_message_unset),
22847 (gst_rtsp_message_free), (gst_rtsp_message_add_header),
22848 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
22849 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
22850 (gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
22851 (gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
22852 (gst_rtsp_message_dump):
22853 * gst-libs/gst/rtsp/gstrtspmessage.h:
22854 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
22855 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
22856 (gst_rtsp_range_parse), (gst_rtsp_range_free):
22857 * gst-libs/gst/rtsp/gstrtsprange.h:
22858 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
22859 (gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
22860 (gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
22861 (range_as_text), (rtsp_transport_mode_as_text),
22862 (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
22863 (gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
22864 (gst_rtsp_transport_free):
22865 * gst-libs/gst/rtsp/gstrtsptransport.h:
22866 * gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
22867 (gst_rtsp_url_free), (gst_rtsp_url_set_port),
22868 (gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
22869 * gst-libs/gst/rtsp/gstrtspurl.h:
22870 * gst-libs/gst/sdp/Makefile.am:
22871 * gst-libs/gst/sdp/gstsdp.h:
22872 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
22873 (gst_sdp_connection_init), (gst_sdp_bandwidth_init),
22874 (gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
22875 (gst_sdp_attribute_init), (gst_sdp_message_new),
22876 (gst_sdp_message_init), (gst_sdp_message_uninit),
22877 (gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
22878 (gst_sdp_media_uninit), (gst_sdp_media_free),
22879 (gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
22880 (gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
22881 (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
22882 (gst_sdp_message_add_zone), (gst_sdp_message_set_key),
22883 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
22884 (gst_sdp_message_get_attribute_val),
22885 (gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
22886 (gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
22887 (gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
22888 (gst_sdp_media_get_attribute_val_n),
22889 (gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
22890 (read_string), (read_string_del), (gst_sdp_parse_line),
22891 (gst_sdp_message_parse_buffer), (print_media),
22892 (gst_sdp_message_dump):
22893 * gst-libs/gst/sdp/gstsdpmessage.h:
22894 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
22895 Move SDP and RTSP from helper objects in -good to a reusable library.
22896 Use a proper gst_ namespace.
22898 2007-07-23 18:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
22900 ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
22901 Original commit message from CVS:
22902 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
22903 (vorbis_dec_flush_decode):
22904 Use the new buffer clipping function from gstaudio here.
22906 2007-07-23 18:26:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
22908 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
22909 Original commit message from CVS:
22910 * docs/libs/gst-plugins-base-libs-sections.txt:
22911 * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
22912 * gst-libs/gst/audio/audio.h:
22913 * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
22914 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
22915 Also add deprecation guards for gst_audio_structure_set_int() to the
22918 2007-07-23 14:45:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22920 docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
22921 Original commit message from CVS:
22922 * docs/libs/gst-plugins-base-libs-sections.txt:
22925 2007-07-23 11:18:35 +0000 Dan Williams <dcbw@redhat.com>
22927 gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
22928 Original commit message from CVS:
22929 Patch by: Dan Williams <dcbw at redhat dot com>
22930 * gst/playback/gstplaybasebin.c:
22931 (gst_play_base_bin_get_streaminfo_value_array):
22932 Don't return NULL when querying the stream info value array but instead
22933 return an empty array. Fixes #459204.
22935 2007-07-23 10:41:18 +0000 Tim-Philipp Müller <tim@centricular.net>
22937 gst/playback/gsturidecodebin.c: Init debug category before using it.
22938 Original commit message from CVS:
22939 * gst/playback/gsturidecodebin.c:
22940 Init debug category before using it.
22942 2007-07-21 09:56:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22944 gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
22945 Original commit message from CVS:
22946 * gst-libs/gst/interfaces/mixer.h:
22947 Add padding vars in place of the signal pointers
22948 when building with DISABLE_DEPRECATED so that the
22949 interface structure doesn't change size.
22951 2007-07-21 09:21:12 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
22954 Original commit message from CVS:
22955 * docs/libs/gst-plugins-base-libs-sections.txt:
22956 * ext/alsa/gstalsamixer.c:
22957 * ext/alsa/gstalsamixer.h:
22958 * ext/alsa/gstalsamixerelement.c:
22959 * ext/alsa/gstalsamixertrack.c:
22960 * gst-libs/gst/interfaces/mixer.c:
22961 * gst-libs/gst/interfaces/mixer.h:
22962 * gst-libs/gst/interfaces/mixeroptions.c:
22963 * gst-libs/gst/interfaces/mixeroptions.h:
22964 * gst-libs/gst/interfaces/mixertrack.c:
22965 * gst-libs/gst/interfaces/mixertrack.h:
22966 * tests/check/Makefile.am:
22967 * tests/check/libs/mixer.c:
22968 Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
22970 Add support for notifying mixer changes on the message bus, and
22971 implement it in alsamixer.
22972 API: gst_mixer_get_mixer_flags
22973 API: gst_mixer_message_parse_mute_toggled
22974 API: gst_mixer_message_parse_record_toggled
22975 API: gst_mixer_message_parse_volume_changed
22976 API: gst_mixer_message_parse_option_changed
22977 API: GstMixerMessageType
22980 2007-07-20 16:09:03 +0000 Michael Smith <msmith@xiph.org>
22982 sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
22983 Original commit message from CVS:
22984 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
22985 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
22986 xcontext->im_format is only for testing XShm support (as the header
22987 file comments document). Use xvimage->im_format for everything else.
22988 Avoids spurious warnings on buffer allocation before setcaps.
22990 2007-07-20 07:22:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22992 tests/: We should use $(LIBM).
22993 Original commit message from CVS:
22994 * tests/examples/volume/Makefile.am:
22995 * tests/icles/Makefile.am:
22996 We should use $(LIBM).
22998 2007-07-20 06:13:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23000 tests/icles/Makefile.am: This needs -lm.
23001 Original commit message from CVS:
23002 * tests/icles/Makefile.am:
23005 2007-07-18 07:35:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23007 Add stdlib include (free, atoi, exit).
23008 Original commit message from CVS:
23009 * examples/app/appsrc_ex.c:
23010 * examples/switch/switcher.c:
23011 * ext/neon/gstneonhttpsrc.c:
23012 * ext/timidity/gstwildmidi.c:
23013 * ext/x264/gstx264enc.c:
23014 * gst/mve/mveaudioenc.c: (mve_compress_audio):
23015 * gst/rtpmanager/gstrtpclient.c:
23016 * gst/rtpmanager/gstrtpjitterbuffer.c:
23017 * gst/spectrum/demo-audiotest.c:
23018 * gst/spectrum/demo-osssrc.c:
23019 * sys/dvb/gstdvbsrc.c:
23020 Add stdlib include (free, atoi, exit).
23022 2007-07-16 10:10:28 +0000 Wim Taymans <wim.taymans@gmail.com>
23024 gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
23025 Original commit message from CVS:
23026 * gst-libs/gst/rtp/gstbasertppayload.c:
23027 (gst_basertppayload_class_init), (gst_basertppayload_init),
23028 (gst_basertppayload_set_property),
23029 (gst_basertppayload_get_property):
23030 Don't break ABI, restore previous ranges. Keep the default random
23031 selection of timestamp and seqnum offset but as soon as the app sets a
23032 specific value, use that one.
23034 2007-07-14 18:33:15 +0000 Bastien Nocera <hadess@hadess.net>
23036 sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
23037 Original commit message from CVS:
23038 Patch by: Bastien Nocera <hadess at hadess dot net>
23039 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
23040 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
23041 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
23042 * sys/xvimage/xvimagesink.h:
23043 Add option to turn off double-buffering for debugging purposes.
23046 2007-07-14 18:20:41 +0000 Jorn Baayen <jorn@openedhand.com>
23048 sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
23049 Original commit message from CVS:
23050 Patch by: Jorn Baayen <jorn at openedhand dot com>
23051 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
23052 (gst_ximagesink_set_property), (gst_ximagesink_get_property),
23053 (gst_ximagesink_init), (gst_ximagesink_class_init):
23054 * sys/ximage/ximagesink.h:
23055 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
23056 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
23057 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
23058 * sys/xvimage/xvimagesink.h:
23059 add 'handle-expose' property. Useful for video widgets which may want to
23060 be in control of Expose behaviour. Fixes #380625
23062 2007-07-14 17:23:42 +0000 Wim Taymans <wim.taymans@gmail.com>
23064 gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
23065 Original commit message from CVS:
23066 * gst-libs/gst/rtp/gstbasertppayload.c:
23067 (gst_basertppayload_class_init), (gst_basertppayload_init),
23068 (gst_basertppayload_event), (gst_basertppayload_push),
23069 (gst_basertppayload_set_property),
23070 (gst_basertppayload_get_property),
23071 (gst_basertppayload_change_state):
23072 * gst-libs/gst/rtp/gstbasertppayload.h:
23073 Fix ranges of rtp payloader properties so that the full range can be
23074 used in addition to -1 (random).
23075 Fix wrong seqnum reporting in caps.
23078 2007-07-13 18:12:19 +0000 Wim Taymans <wim.taymans@gmail.com>
23080 gst/videorate/gstvideorate.c: Use boilerplate.
23081 Original commit message from CVS:
23082 * gst/videorate/gstvideorate.c: (gst_video_rate_init),
23083 (gst_video_rate_query):
23085 Add latency query, might not be perfect yet but already works a lot
23086 better. Fixes #442557.
23088 2007-07-13 16:05:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23090 sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
23091 Original commit message from CVS:
23092 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
23093 (gst_xvimagesink_setcaps):
23094 * sys/xvimage/xvimagesink.h:
23095 After a caps change, redraw our borders to avoid garbage left there
23096 when the image format changes to a smaller size, like 16:9 -> 4:3
23097 Also, hold the flow_lock a bit longer in the set_caps while we're
23098 fiddling with the xcontext.
23100 2007-07-13 16:02:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23102 Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
23103 Original commit message from CVS:
23106 * tests/Makefile.am:
23107 Remove bogus check for libcheck, since we check for
23108 gstreamer-check and it pulls in the required info from there, and we
23109 weren't actually _using_ the information for libcheck ourselves
23112 2007-07-13 15:52:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23114 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
23115 Original commit message from CVS:
23116 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
23117 (gst_ffmpeg_caps_to_pixfmt):
23118 Fix the r_mask test for RGBA32 on little-endian.
23119 Fix a stupid typo that would have obviously broken
23120 compilation on big-endian, if anyone was testing.
23122 2007-07-12 15:02:43 +0000 Wim Taymans <wim.taymans@gmail.com>
23124 gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
23125 Original commit message from CVS:
23126 * gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
23127 (paint_hline_str4):
23128 * gst/videotestsrc/videotestsrc.h:
23129 Add alpha to the color struct.
23130 Use a default alpha value of 255 instead of 128.
23132 2007-07-12 12:01:20 +0000 Wim Taymans <wim.taymans@gmail.com>
23134 gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
23135 Original commit message from CVS:
23136 * gst/playback/gstplaybasebin.c: (no_more_pads_full),
23138 Clear the dynamic pads counter when starting a new uri. This makes
23139 reusing playbin work again.
23142 2007-07-12 11:13:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23144 configure.ac: Use pkg-config to locate check.
23145 Original commit message from CVS:
23147 Use pkg-config to locate check.
23149 2007-07-11 23:12:12 +0000 Tim-Philipp Müller <tim@centricular.net>
23151 Fix 'make check' build against core CVS.
23152 Original commit message from CVS:
23154 * tests/check/elements/volume.c: (GST_START_TEST):
23155 Fix 'make check' build against core CVS.
23157 2007-07-10 20:46:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23159 gst-libs/gst/: Make gtk-doc happy.
23160 Original commit message from CVS:
23161 * gst-libs/gst/interfaces/propertyprobe.c:
23162 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23163 * gst-libs/gst/tag/gstvorbistag.c:
23164 Make gtk-doc happy.
23166 2007-07-08 13:07:38 +0000 Tim-Philipp Müller <tim@centricular.net>
23168 gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
23169 Original commit message from CVS:
23170 * gst-libs/gst/audio/gstbaseaudiosink.c:
23171 (gst_base_audio_sink_callback):
23172 Quick hack to make audiosinks stop at EOS when operating in
23173 pull-mode; needs to be fixed properly some day.
23175 2007-07-06 18:19:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23177 docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
23178 Original commit message from CVS:
23179 * docs/libs/gst-plugins-base-libs-sections.txt:
23180 Fix location of includes in the docs.
23182 2007-07-06 11:40:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23184 gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
23185 Original commit message from CVS:
23186 * gst/ffmpegcolorspace/avcodec.h:
23187 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
23188 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
23189 (gst_ffmpegcsp_avpicture_fill):
23190 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
23191 (img_get_alpha_info):
23192 Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
23193 of the existing BGRA32 and RGBA32 formats with the alpha at the other
23194 end of the word. Partially fixes #451908
23196 2007-07-05 08:43:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23198 docs/: Simplify --extra-dir as gtkdoc scans recursively.
23199 Original commit message from CVS:
23200 * docs/libs/Makefile.am:
23201 * docs/plugins/Makefile.am:
23202 Simplify --extra-dir as gtkdoc scans recursively.
23204 2007-07-03 11:52:47 +0000 Wim Taymans <wim.taymans@gmail.com>
23206 gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
23207 Original commit message from CVS:
23208 * gst/adder/gstadder.c: (gst_adder_sink_getcaps),
23209 (gst_adder_request_new_pad):
23210 Make getcaps more robust by not using the proxycaps function. This makes
23211 sure that we don't end up recursively calling getcaps upstream.
23214 2007-06-29 17:21:18 +0000 Wim Taymans <wim.taymans@gmail.com>
23216 gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
23217 Original commit message from CVS:
23218 * gst/audioconvert/audioconvert.c:
23219 Include math.h to fix compilation.
23221 2007-06-29 14:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23223 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
23224 Original commit message from CVS:
23225 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
23226 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
23227 Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
23228 format, as produced by some dc1394 cameras like the iSight.
23229 See http://www.fourcc.org/yuv.php#IYU1
23231 2007-06-28 20:37:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
23233 gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
23234 Original commit message from CVS:
23235 * gst/audioconvert/Makefile.am:
23236 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
23237 (check_default), (audio_convert_prepare_context),
23238 (audio_convert_clean_context), (audio_convert_convert):
23239 * gst/audioconvert/audioconvert.h:
23240 * gst/audioconvert/gstaudioconvert.c:
23241 (gst_audio_convert_dithering_get_type),
23242 (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
23243 (gst_audio_convert_init), (gst_audio_convert_set_caps),
23244 (gst_audio_convert_set_property), (gst_audio_convert_get_property):
23245 * gst/audioconvert/gstaudioconvert.h:
23246 * gst/audioconvert/gstaudioquantize.c:
23247 (gst_audio_quantize_setup_noise_shaping),
23248 (gst_audio_quantize_free_noise_shaping),
23249 (gst_audio_quantize_setup_dither),
23250 (gst_audio_quantize_free_dither),
23251 (gst_audio_quantize_setup_quantize_func),
23252 (gst_audio_quantize_setup), (gst_audio_quantize_free):
23253 * gst/audioconvert/gstaudioquantize.h:
23254 Implement dithering and noise shaping in audioconvert. By default now
23255 TPDF dithering (and no noise shaping) will be used when converting
23256 from a higher bit depth to 20 bit depth or smaller, otherwise
23257 everything will be as it is now.
23258 For the last audioconvert in a pipeline it would make sense to
23259 use some kind of noise shaping, enabling it by default for all
23260 conversions would give undesired results though. Fixes #360246.
23261 * tests/check/elements/audioconvert.c: (setup_audioconvert),
23263 Adjust unit test for the new audioconvert.
23265 2007-06-28 11:06:56 +0000 Wim Taymans <wim.taymans@gmail.com>
23267 gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
23268 Original commit message from CVS:
23269 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
23270 Use other metrics as well when estimating the buffer level.
23272 2007-06-28 10:21:19 +0000 Wim Taymans <wim.taymans@gmail.com>
23274 gst/playback/gstplaybasebin.c: Small debug improvement.
23275 Original commit message from CVS:
23276 * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
23277 Small debug improvement.
23278 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
23280 Tweak the rate estimation period.
23281 When calculating the buffer filledness in rate estimation mode, don't
23282 mix it with other metrics.
23284 2007-06-28 09:46:11 +0000 Wim Taymans <wim.taymans@gmail.com>
23286 gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
23287 Original commit message from CVS:
23288 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
23289 (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
23290 When creating the groups, allow for a 5 second, unlimited buffers
23291 preroll phase after which we expose the group.
23292 When the group is exposed, use a small number of buffers up to a 2
23293 second limit. Also disconnect the overrun signal from multiqueue when we
23294 exposed the group because it is not needed anymore.
23296 2007-06-27 22:30:19 +0000 Tim-Philipp Müller <tim@centricular.net>
23298 gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
23299 Original commit message from CVS:
23300 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
23301 Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
23302 to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
23303 (#451707); also, output some debugging info when dealing with
23305 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
23306 Add unit test for the above.
23308 2007-06-27 12:55:20 +0000 Tim-Philipp Müller <tim@centricular.net>
23310 gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
23311 Original commit message from CVS:
23312 * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
23313 Add description for Windows Media RTP caps.
23314 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
23315 Remove RTP fields that don't define the format from caps.
23317 2007-06-27 10:14:03 +0000 Tim-Philipp Müller <tim@centricular.net>
23319 ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
23320 Original commit message from CVS:
23321 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
23322 Skip empty buffers, but not empty header buffers. That way the original
23323 vorbisdec unit test still passes (#451145); also, take into account
23324 that those empty packets might carry a granulepos.
23325 * tests/check/Makefile.am:
23326 * tests/check/elements/vorbisdec.c:
23327 (_create_codebook_header_buffer), (_create_audio_buffer),
23328 (GST_START_TEST), (vorbisdec_suite):
23329 Add unit test that sends an empty packet.
23331 2007-06-27 09:49:51 +0000 Wim Taymans <wim.taymans@gmail.com>
23333 ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
23334 Original commit message from CVS:
23335 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
23336 Don't error out on 0-sized packets, just emit a warning because this is
23337 not a fatal error. Fixes #451145.
23339 2007-06-25 12:43:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23341 docs/plugins/: Update docs with caps info.
23342 Original commit message from CVS:
23343 * docs/plugins/gst-plugins-base-plugins.args:
23344 * docs/plugins/gst-plugins-base-plugins.signals:
23345 * docs/plugins/inspect/plugin-adder.xml:
23346 * docs/plugins/inspect/plugin-alsa.xml:
23347 * docs/plugins/inspect/plugin-audioconvert.xml:
23348 * docs/plugins/inspect/plugin-audiorate.xml:
23349 * docs/plugins/inspect/plugin-audioresample.xml:
23350 * docs/plugins/inspect/plugin-audiotestsrc.xml:
23351 * docs/plugins/inspect/plugin-cdparanoia.xml:
23352 * docs/plugins/inspect/plugin-decodebin.xml:
23353 * docs/plugins/inspect/plugin-decodebin2.xml:
23354 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
23355 * docs/plugins/inspect/plugin-gdp.xml:
23356 * docs/plugins/inspect/plugin-gnomevfs.xml:
23357 * docs/plugins/inspect/plugin-libvisual.xml:
23358 * docs/plugins/inspect/plugin-ogg.xml:
23359 * docs/plugins/inspect/plugin-pango.xml:
23360 * docs/plugins/inspect/plugin-playbin.xml:
23361 * docs/plugins/inspect/plugin-subparse.xml:
23362 * docs/plugins/inspect/plugin-tcp.xml:
23363 * docs/plugins/inspect/plugin-theora.xml:
23364 * docs/plugins/inspect/plugin-typefindfunctions.xml:
23365 * docs/plugins/inspect/plugin-video4linux.xml:
23366 * docs/plugins/inspect/plugin-videorate.xml:
23367 * docs/plugins/inspect/plugin-videoscale.xml:
23368 * docs/plugins/inspect/plugin-videotestsrc.xml:
23369 * docs/plugins/inspect/plugin-volume.xml:
23370 * docs/plugins/inspect/plugin-vorbis.xml:
23371 * docs/plugins/inspect/plugin-ximagesink.xml:
23372 * docs/plugins/inspect/plugin-xvimagesink.xml:
23373 Update docs with caps info.
23375 2007-06-25 12:04:15 +0000 Tim-Philipp Müller <tim@centricular.net>
23377 po/POTFILES.in: Add more files with translatable strings (#450875).
23378 Original commit message from CVS:
23380 Add more files with translatable strings (#450875).
23382 2007-06-23 14:44:07 +0000 Edward Hervey <bilboed@bilboed.com>
23384 ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
23385 Original commit message from CVS:
23386 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
23387 The chain should be freed if we error out here, else it will leak.
23388 * gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
23389 (cleanup_decodebin):
23390 Don't forget to *properly* remove the signals, else it will leak.
23392 2007-06-22 14:25:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23394 MAINTAINERS: Updating all the maintainers files
23395 Original commit message from CVS:
23397 Updating all the maintainers files
23399 2007-06-21 08:34:46 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23401 tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
23402 Original commit message from CVS:
23403 * tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
23405 Destroy and recreate parse-launch based pipeline after stop to be able
23406 to play again. Reorder some code and add more comments.
23408 2007-06-20 11:09:03 +0000 Wim Taymans <wim@fluendo.com>
23410 gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
23411 Original commit message from CVS:
23412 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
23413 When handling a delayed-caps notification case, mark
23414 the group as dynamic so that the nbdynamic count is
23415 incremented and decremented correctly. Fixes: #449156
23416 Patch by: Wim Taymans <wim@fluendo.com>
23418 2007-06-19 19:13:04 +0000 Andy Wingo <wingo@pobox.com>
23421 * gst-libs/gst/audio/gstbaseaudiosink.c:
23422 * win32/common/config.h:
23423 gst-libs/gst/audio/gstbaseaudiosink.c
23424 Original commit message from CVS:
23425 2007-06-19 Andy Wingo <wingo@pobox.com>
23426 * gst-libs/gst/audio/gstbaseaudiosink.c
23427 (gst_base_audio_sink_init): Enable pull-mode operation.
23429 2007-06-19 09:34:35 +0000 Michael Smith <msmith@xiph.org>
23431 gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
23432 Original commit message from CVS:
23433 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
23434 Change minimum rate back to 1000 to allow low-sample-rate wav files
23437 2007-06-17 17:27:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23439 po/vi.po: Update translations.
23440 Original commit message from CVS:
23442 Update translations.
23444 2007-06-16 03:42:14 +0000 David Schleef <ds@schleef.org>
23446 gst/playback/gstqueue2.c: Fix compile error from ignored return value.
23447 Original commit message from CVS:
23448 * gst/playback/gstqueue2.c:
23449 Fix compile error from ignored return value.
23451 2007-06-15 15:23:36 +0000 Michael Smith <msmith@xiph.org>
23453 gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
23454 Original commit message from CVS:
23455 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
23456 Update tmpbuf for all neccesary rows, not just one, as is required
23460 2007-06-15 11:15:28 +0000 Michael Smith <msmith@xiph.org>
23462 tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
23463 Original commit message from CVS:
23464 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
23465 (eos_buffer_probe):
23466 Add a test that ensures we set DELTA_UNIT on all non-header,
23467 non-video buffers, if we have a video stream.
23468 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
23469 (gst_ogg_mux_process_best_pad):
23470 Move setting delta_pad to earlier, where we inspect all pads, so
23471 that leading audio pages don't get DELTA_UNIT unset if they come
23472 before the first DELTA_UNIT from video pages. Fixes the newly-added
23473 test. Fixes #385527.
23475 2007-06-14 19:53:27 +0000 Tim-Philipp Müller <tim@centricular.net>
23477 tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
23478 Original commit message from CVS:
23479 * tests/check/pipelines/streamheader.c: (streamheader_suite):
23480 Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
23481 fails on the p5-ppc64 build bot and the failure looks like it is due
23482 to the same issue as #348114, ie. a compiler bug.
23484 2007-06-13 18:20:57 +0000 Edward Hervey <bilboed@bilboed.com>
23486 gst/playback/gstqueue2.c: Fix build on MacOSX.
23487 Original commit message from CVS:
23488 * gst/playback/gstqueue2.c: (gst_queue_create_read):
23489 Fix build on MacOSX.
23491 2007-06-13 09:01:32 +0000 Wim Taymans <wim.taymans@gmail.com>
23493 ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
23494 Original commit message from CVS:
23495 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
23496 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
23497 Fix compilation on mingw. Fixes #446972.
23499 2007-06-12 08:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
23501 gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
23502 Original commit message from CVS:
23503 Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
23504 * gst/playback/gstqueue2.c: (update_buffering),
23505 (gst_queue_locked_enqueue):
23506 Fix a division by zero when the max percent is <= 0. Fixes #446572.
23507 also update the buffering status when receiving events. Fixes #446551.
23509 2007-06-11 11:32:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
23511 gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
23512 Original commit message from CVS:
23513 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
23514 * gst/playback/gstqueue2.c: (gst_queue_peer_query),
23515 (gst_queue_handle_src_query):
23516 Wait for preroll before attempting to forward a duration query upstream.
23519 2007-06-07 21:08:38 +0000 Sébastien Moutte <sebastien@moutte.net>
23521 gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
23522 Original commit message from CVS:
23523 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23524 (gst_base_rtp_depayload_set_gst_timestamp):
23525 Use G_GINT64_CONSTANT macro for int64 constant.
23526 * win32/common/libgstinterfaces.def:
23527 * win32/common/libgsttag.def:
23528 Add new exported functions.
23530 2007-06-07 14:25:32 +0000 Tim-Philipp Müller <tim@centricular.net>
23532 ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
23533 Original commit message from CVS:
23534 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
23535 The BOS page of the first Dirac video stream needs to come before
23536 the BOS page of any Vorbis streams or other audio streams, just like
23539 2007-06-07 09:11:27 +0000 Wim Taymans <wim.taymans@gmail.com>
23541 gst/playback/gstqueue2.c: Fix compilation.
23542 Original commit message from CVS:
23543 * gst/playback/gstqueue2.c: (gst_queue_get_range):
23546 2007-06-06 13:36:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
23548 gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
23549 Original commit message from CVS:
23550 Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
23551 * gst/playback/gstqueue2.c: (gst_queue_init),
23552 (gst_queue_handle_sink_event), (gst_queue_chain),
23553 (gst_queue_get_range), (gst_queue_src_checkgetrange_function),
23554 (gst_queue_sink_activate_push), (gst_queue_src_activate_push),
23555 (gst_queue_src_activate_pull):
23556 Add pull based scheduling and fix some deadlocks. Fixes #444523.
23557 Does not yet completely work because duration queries upstream won't
23560 2007-06-06 09:08:50 +0000 Wim Taymans <wim.taymans@gmail.com>
23562 Some more fseeko checks.
23563 Original commit message from CVS:
23565 * gst/playback/gstqueue2.c: (gst_queue_create_read):
23566 Some more fseeko checks.
23568 2007-06-06 08:01:42 +0000 Wim Taymans <wim.taymans@gmail.com>
23570 configure.ac: check for large file support.
23571 Original commit message from CVS:
23573 check for large file support.
23575 2007-06-05 21:36:11 +0000 Sven Arvidsson <sa@whiz.se>
23577 gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
23578 Original commit message from CVS:
23579 Based on a patch by Sven Arvidsson <sa at whiz dot se>:
23580 * gst/subparse/gstsubparse.c: (parse_subrip),
23581 (subviewer_unescape_newlines), (parse_subviewer),
23582 (gst_sub_parse_data_format_autodetect),
23583 (gst_sub_parse_format_autodetect), (gst_subparse_type_find):
23584 * gst/subparse/gstsubparse.h:
23585 Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
23586 * tests/check/elements/subparse.c: (GST_START_TEST),
23588 Add a unit test for both SubViewer formats.
23590 2007-06-05 17:08:04 +0000 Michael Smith <msmith@xiph.org>
23592 gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
23593 Original commit message from CVS:
23594 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
23595 Don't overflow intermediate values when seeking to large time values
23598 2007-06-05 17:02:13 +0000 Wim Taymans <wim.taymans@gmail.com>
23600 gst/playback/gstqueue2.c: Include stdio to define fseeko.
23601 Original commit message from CVS:
23602 * gst/playback/gstqueue2.c: (gst_queue_have_data),
23603 (gst_queue_create_read), (gst_queue_read_item_from_file),
23604 (gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
23605 Include stdio to define fseeko.
23607 2007-06-05 16:37:09 +0000 Edward Hervey <edward@fluendo.com>
23609 sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
23610 Original commit message from CVS:
23611 Patch by: Edward Hervey <edward@fluendo.com>
23612 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
23613 (gst_v4lsrc_query):
23614 Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
23616 2007-06-05 16:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
23618 gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
23619 Original commit message from CVS:
23620 * gst-libs/gst/riff/Makefile.am:
23621 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
23622 Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
23623 our own implementation.
23625 2007-06-05 16:19:30 +0000 Wim Taymans <wim.taymans@gmail.com>
23627 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
23628 Original commit message from CVS:
23629 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23630 (gst_base_rtp_depayload_setcaps),
23631 (gst_base_rtp_depayload_set_gst_timestamp),
23632 (gst_base_rtp_depayload_change_state):
23633 Handle timestamp wraparound.
23635 2007-06-05 16:17:30 +0000 Wim Taymans <wim.taymans@gmail.com>
23637 gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
23638 Original commit message from CVS:
23639 * gst/playback/gsturidecodebin.c: (no_more_pads_full),
23640 (new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
23641 (gst_uri_decode_bin_change_state):
23642 Make sure we name srcpads uniquely even when using different internal
23644 Signal no-more-pads when no more dynamic elements exist.
23645 Remove pads on cleanup.
23647 2007-06-05 16:14:23 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
23649 gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
23650 Original commit message from CVS:
23651 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
23652 * gst/playback/gstqueue2.c: (gst_queue_class_init),
23653 (gst_queue_init), (gst_queue_finalize),
23654 (gst_queue_write_buffer_to_file), (gst_queue_have_data),
23655 (gst_queue_create_read), (gst_queue_read_item_from_file),
23656 (gst_queue_open_temp_location_file),
23657 (gst_queue_close_temp_location_file), (gst_queue_locked_flush),
23658 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
23659 (gst_queue_is_empty), (gst_queue_is_filled),
23660 (gst_queue_change_state), (gst_queue_set_temp_location),
23661 (gst_queue_set_property):
23662 Add support for filebased buffering. Fixes #441264.
23664 2007-06-05 16:05:19 +0000 Wim Taymans <wim.taymans@gmail.com>
23666 gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
23667 Original commit message from CVS:
23668 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
23669 (analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
23670 (caps_notify_group_cb), (gst_decode_group_new),
23671 (gst_decode_group_free):
23672 Add support for delayed caps fixation when autoplugging.
23673 Optimize cases where a multiqueue is not needed/wanted, like right after
23674 anything that is not a demuxer.
23676 2007-06-05 16:02:57 +0000 Wim Taymans <wim.taymans@gmail.com>
23678 ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
23679 Original commit message from CVS:
23680 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
23681 (gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
23682 (gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
23683 consideratly speedup ogg chain detection by not trying to find a base
23684 timestamp for skeleton streams.
23686 2007-06-05 16:00:33 +0000 Wim Taymans <wim.taymans@gmail.com>
23688 gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
23689 Original commit message from CVS:
23690 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
23691 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
23692 (gst_multi_fd_sink_remove_flush),
23693 (gst_multi_fd_sink_remove_client_link),
23694 (gst_multi_fd_sink_handle_client_write),
23695 (gst_multi_fd_sink_handle_clients):
23696 * gst/tcp/gstmultifdsink.h:
23697 Add support for remuve_flush.
23699 2007-06-05 15:59:00 +0000 Wim Taymans <wim.taymans@gmail.com>
23701 Add draft design for forcing keyframes in encoders and implement in theoraenc.
23702 Original commit message from CVS:
23703 * docs/design/draft-keyframe-force.txt:
23704 * ext/theora/theoraenc.c: (theora_enc_sink_event),
23705 (theora_enc_chain):
23706 Add draft design for forcing keyframes in encoders and implement in
23709 2007-06-05 13:22:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23711 configure.ac: Back to CVS
23712 Original commit message from CVS:
23716 === release 0.10.13 ===
23718 2007-06-05 12:50:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23724 * docs/plugins/gst-plugins-base-plugins.args:
23725 * docs/plugins/inspect/plugin-adder.xml:
23726 * docs/plugins/inspect/plugin-alsa.xml:
23727 * docs/plugins/inspect/plugin-audioconvert.xml:
23728 * docs/plugins/inspect/plugin-audiorate.xml:
23729 * docs/plugins/inspect/plugin-audioresample.xml:
23730 * docs/plugins/inspect/plugin-audiotestsrc.xml:
23731 * docs/plugins/inspect/plugin-cdparanoia.xml:
23732 * docs/plugins/inspect/plugin-decodebin.xml:
23733 * docs/plugins/inspect/plugin-decodebin2.xml:
23734 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
23735 * docs/plugins/inspect/plugin-gdp.xml:
23736 * docs/plugins/inspect/plugin-gnomevfs.xml:
23737 * docs/plugins/inspect/plugin-libvisual.xml:
23738 * docs/plugins/inspect/plugin-ogg.xml:
23739 * docs/plugins/inspect/plugin-pango.xml:
23740 * docs/plugins/inspect/plugin-playbin.xml:
23741 * docs/plugins/inspect/plugin-subparse.xml:
23742 * docs/plugins/inspect/plugin-tcp.xml:
23743 * docs/plugins/inspect/plugin-theora.xml:
23744 * docs/plugins/inspect/plugin-typefindfunctions.xml:
23745 * docs/plugins/inspect/plugin-video4linux.xml:
23746 * docs/plugins/inspect/plugin-videorate.xml:
23747 * docs/plugins/inspect/plugin-videoscale.xml:
23748 * docs/plugins/inspect/plugin-videotestsrc.xml:
23749 * docs/plugins/inspect/plugin-volume.xml:
23750 * docs/plugins/inspect/plugin-vorbis.xml:
23751 * docs/plugins/inspect/plugin-ximagesink.xml:
23752 * docs/plugins/inspect/plugin-xvimagesink.xml:
23753 * gst-plugins-base.doap:
23754 * win32/common/config.h:
23755 * win32/vs6/grammar.dsp:
23756 * win32/vs6/gst_plugins_base.dsw:
23757 * win32/vs6/libgstadder.dsp:
23758 * win32/vs6/libgstaudio.dsp:
23759 * win32/vs6/libgstaudioconvert.dsp:
23760 * win32/vs6/libgstaudiorate.dsp:
23761 * win32/vs6/libgstaudioresample.dsp:
23762 * win32/vs6/libgstaudioscale.dsp:
23763 * win32/vs6/libgstaudiotestsrc.dsp:
23764 * win32/vs6/libgstcdda.dsp:
23765 * win32/vs6/libgstdecodebin.dsp:
23766 * win32/vs6/libgstdecodebin2.dsp:
23767 * win32/vs6/libgstdirectsound.dsp:
23768 * win32/vs6/libgstffmpegcolorspace.dsp:
23769 * win32/vs6/libgstgdp.dsp:
23770 * win32/vs6/libgstinterfaces.dsp:
23771 * win32/vs6/libgstnetbuffer.dsp:
23772 * win32/vs6/libgstogg.dsp:
23773 * win32/vs6/libgstpbutils.dsp:
23774 * win32/vs6/libgstplaybin.dsp:
23775 * win32/vs6/libgstriff.dsp:
23776 * win32/vs6/libgstrtp.dsp:
23777 * win32/vs6/libgstsinesrc.dsp:
23778 * win32/vs6/libgstsubparse.dsp:
23779 * win32/vs6/libgsttag.dsp:
23780 * win32/vs6/libgsttheora.dsp:
23781 * win32/vs6/libgsttypefindfunctions.dsp:
23782 * win32/vs6/libgstutils.dsp:
23783 * win32/vs6/libgstvideo.dsp:
23784 * win32/vs6/libgstvideorate.dsp:
23785 * win32/vs6/libgstvideoscale.dsp:
23786 * win32/vs6/libgstvideotestsrc.dsp:
23787 * win32/vs6/libgstvolume.dsp:
23788 * win32/vs6/libgstvorbis.dsp:
23789 Release 0.10.13 "What's going on?"
23790 Original commit message from CVS:
23791 Release 0.10.13 "What's going on?"
23793 2007-06-05 12:32:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23811 Original commit message from CVS:
23814 2007-05-31 17:08:58 +0000 Wim Taymans <wim@fluendo.com>
23816 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...
23817 Original commit message from CVS:
23818 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
23819 In riff, the depth is stored in the size field but it just means that
23820 the least significant bits are cleared. We can therefore just play
23821 the sample as if it had a depth == width. Fixes: #440997
23822 Patch by: Wim Taymans <wim@fluendo.com>
23823 Patch by: Sebastian Dröge <slomo@circular-chaos.org>
23825 2007-05-31 16:36:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23827 gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
23828 Original commit message from CVS:
23829 * gst-libs/gst/floatcast/floatcast.h:
23830 Define inline when needed on win32 builds. Fixes: #441295
23832 2007-05-29 13:38:35 +0000 Wim Taymans <wim.taymans@gmail.com>
23834 gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
23835 Original commit message from CVS:
23836 * gst/playback/gstplaybasebin.c: (queue_overrun),
23837 (no_more_pads_full):
23838 Stop buffering when the group is commited because the queues filled up.
23841 2007-05-25 10:07:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23843 Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
23844 Original commit message from CVS:
23845 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
23846 (gst_alsa_mixer_free), (gst_alsa_mixer_update),
23847 (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
23848 (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
23849 (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
23850 * ext/alsa/gstalsamixer.h:
23851 * ext/alsa/gstalsamixerelement.c:
23852 (gst_alsa_mixer_element_interface_supported),
23853 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
23854 (gst_alsa_mixer_element_set_property),
23855 (gst_alsa_mixer_element_get_property),
23856 (gst_alsa_mixer_element_change_state):
23857 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
23858 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
23859 (gst_mixer_option_changed):
23860 * gst-libs/gst/interfaces/mixer.h:
23861 Revert commits towards #152864 made so far. We'll pick it up again
23862 after the 0.10.13 release.
23864 2007-05-24 16:22:23 +0000 Wim Taymans <wim.taymans@gmail.com>
23866 gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
23867 Original commit message from CVS:
23868 * gst-libs/gst/audio/gstbaseaudiosink.c:
23869 (gst_base_audio_sink_render):
23870 After an interrupt (PAUSED/flush) assume that the next sample should not
23871 be aligned to the previous sample. Fixes #417992.
23873 2007-05-24 15:16:59 +0000 Tim-Philipp Müller <tim@centricular.net>
23875 gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
23876 Original commit message from CVS:
23877 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
23878 Don't add channels and rate fields to the template caps for
23879 audio/x-dts, as wavparse might not always be able to set them,
23880 which would then lead to 'caps are not a real subset of the
23881 template caps' warnings.
23883 2007-05-24 11:15:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23885 gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
23886 Original commit message from CVS:
23887 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
23888 Handle unknown or invalid pads without crashing, as might occur if
23889 a media file like an mp3 is specified as a subtitle file.
23892 2007-05-24 10:19:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23894 gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
23895 Original commit message from CVS:
23896 * gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
23898 Block the subtitle bin output queue before ghosting it and linking,
23899 then unblock after. This avoids spurious not-linked errors caused
23900 by the queue starting up (because it gets linked when it is ghosted).
23903 2007-05-23 15:54:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23905 tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
23906 Original commit message from CVS:
23907 * tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
23908 Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
23909 file. Avoids flukes where the input gets typefound to some valid but
23912 2007-05-22 15:45:19 +0000 Tim-Philipp Müller <tim@centricular.net>
23914 tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
23915 Original commit message from CVS:
23916 * tests/check/Makefile.am:
23917 * tests/check/elements/.cvsignore:
23918 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
23919 (cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
23920 Add unit test for gnomevfssink seeking and position reporting for
23923 2007-05-22 15:30:26 +0000 Mark Nauwelaerts <manauw@skynet.be>
23925 ext/gnomevfs/gstgnomevfssink.*: see #412648.
23926 Original commit message from CVS:
23927 Patch by: Mark Nauwelaerts <manauw at skynet be>
23928 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
23929 (gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
23930 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
23931 * ext/gnomevfs/gstgnomevfssink.h:
23932 Fix position reporting, especially after a seek (from upstream),
23935 2007-05-22 15:04:41 +0000 Tim-Philipp Müller <tim@centricular.net>
23937 ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
23938 Original commit message from CVS:
23939 * ext/cdparanoia/gstcdparanoiasrc.c:
23942 2007-05-22 11:40:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23944 gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
23945 Original commit message from CVS:
23946 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
23947 Specify the full valid range for MP3 samplerates. Fixes a regression
23948 caused by extra header checks since the last release.
23950 2007-05-21 15:32:42 +0000 Mike Smith <msmith@xiph.org>
23952 sys/: Fix a locking-order bug I introduced with my changes the other day.
23953 Original commit message from CVS:
23954 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
23955 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
23956 Fix a locking-order bug I introduced with my changes the other day.
23957 Patch by Mike Smith.
23959 2007-05-21 15:24:21 +0000 Michael Smith <msmith@xiph.org>
23961 ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
23962 Original commit message from CVS:
23963 * ext/theora/theoradec.c: (theora_handle_data_packet):
23964 Don't look inside 0-length packets (which indicate duplicated
23967 2007-05-21 10:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
23970 Original commit message from CVS:
23971 * ext/cdparanoia/gstcdparanoiasrc.c:
23972 (gst_cd_paranoia_src_read_sector):
23973 * gst-libs/gst/audio/gstbaseaudiosrc.c:
23974 (gst_base_audio_src_create):
23976 * ext/theora/theoradec.c: (theora_dec_sink_event):
23978 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23979 (gst_base_rtp_depayload_set_gst_timestamp):
23981 * gst/playback/gstdecodebin.c: (queue_underrun_cb):
23982 And some debug info when a FIXME path is hit.
23984 2007-05-21 09:45:28 +0000 Wim Taymans <wim.taymans@gmail.com>
23986 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
23987 Original commit message from CVS:
23988 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23989 (gst_base_rtp_audio_payload_class_init),
23990 (gst_base_rtp_audio_payload_init),
23991 (gst_base_rtp_audio_payload_finalize),
23992 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
23993 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
23994 (gst_base_rtp_payload_audio_handle_event):
23995 Some cleanups, remove minptime property as it is now in the parent
23997 Override parent class event function.
23998 * gst-libs/gst/rtp/gstbasertppayload.c:
23999 (gst_basertppayload_class_init), (gst_basertppayload_init),
24000 (gst_basertppayload_event), (gst_basertppayload_set_property),
24001 (gst_basertppayload_get_property):
24002 * gst-libs/gst/rtp/gstbasertppayload.h:
24003 Add min-ptime property.
24004 Add handle-event vmethod. Fixes #415001.
24006 2007-05-18 17:10:03 +0000 Christian Schaller <uraeus@gnome.org>
24008 * gst-plugins-base.spec.in:
24010 Original commit message from CVS:
24013 2007-05-18 15:23:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24015 gst-libs/gst/audio/gstbaseaudiosink.c
24016 Original commit message from CVS:
24017 * gst-libs/gst/audio/gstbaseaudiosink.c
24018 (gst_base_audio_sink_change_state):
24019 Fix typo in comment.
24020 * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
24021 free_dynamics, pad_probe, close_pad_link, try_to_link_1,
24022 get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
24024 * gst/playback/gstplaybin.c (gst_play_bin_set_property,
24025 gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
24026 Remove trailing whitespaces in comments.
24027 * gst/volume/Makefile.am:
24030 2007-05-18 15:10:08 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
24033 * gst-libs/gst/interfaces/mixer.h:
24034 gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
24035 Original commit message from CVS:
24036 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
24037 * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
24038 set_option, get_option, _gst_reserved):
24039 Revert reordering functions (keep ABI).
24041 2007-05-17 17:35:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24043 sys/: When we create our own window, indicate that we handle the
24044 Original commit message from CVS:
24045 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
24046 (gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
24047 (gst_ximagesink_show_frame):
24048 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
24049 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
24050 (gst_xvimagesink_show_frame):
24051 When we create our own window, indicate that we handle the
24052 WM_DELETE client message from the window manager, so that it won't
24053 kill our window (and our app) along with it. Handle ClientMessage,
24054 post an error on the bus, and close the window. Further buffers
24055 arriving will result in a FlowError because the window has been
24058 Clean up the X event handling loop and make them the same for
24059 both xvimagesink and ximagesink while I'm at it.
24061 2007-05-17 16:27:32 +0000 Wim Taymans <wim.taymans@gmail.com>
24063 gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
24064 Original commit message from CVS:
24065 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
24066 Make decodebin2 autoplug depayloaders too.
24067 * gst/playback/gsturidecodebin.c: (source_new_pad):
24068 Set the newly created decoder in a usable state when autoplugging a
24069 dynamic source such as RTSP.
24071 2007-05-17 16:11:03 +0000 Tim-Philipp Müller <tim@centricular.net>
24073 gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
24074 Original commit message from CVS:
24075 * gst/playback/gststreaminfo.c: (cb_probe):
24076 Ignore video-codec tag for audio streams and ignore audio-codec tags
24077 for video streams. Should make codec name collection a bit more
24078 robust against sloppy demuxers that send tag events containing both
24079 tags down each pad.
24081 2007-05-17 15:22:44 +0000 Wim Taymans <wim.taymans@gmail.com>
24083 gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
24084 Original commit message from CVS:
24085 * gst/playback/gstqueue2.c: (update_rates):
24086 Tweak the buffering thresholds a little.
24087 Update the buffer size with the previously calculate rate instead of
24088 only when we calculate a new rate so that we get smoother buffering
24090 * gst/playback/Makefile.am:
24091 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
24092 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
24093 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
24094 (gst_uri_decode_bin_get_property), (unknown_type),
24095 (add_element_stream), (no_more_pads_full), (no_more_pads),
24096 (source_no_more_pads), (new_decoded_pad), (array_has_value),
24097 (gen_source_element), (has_all_raw_caps), (analyse_source),
24098 (remove_decoders), (make_decoder), (remove_source),
24099 (source_new_pad), (setup_source), (decoder_query_init),
24100 (decoder_query_duration_fold), (decoder_query_duration_done),
24101 (decoder_query_position_fold), (decoder_query_position_done),
24102 (decoder_query_latency_fold), (decoder_query_latency_done),
24103 (decoder_query_seeking_fold), (decoder_query_seeking_done),
24104 (decoder_query_generic_fold), (gst_uri_decode_bin_query),
24105 (gst_uri_decode_bin_change_state), (plugin_init):
24106 New element that intergrates a source, optional buffering element and
24109 2007-05-17 14:17:17 +0000 Tim-Philipp Müller <tim@centricular.net>
24111 configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
24112 Original commit message from CVS:
24114 Bump libtheora requirement to 1.0alpha5 for the pixformat check
24115 (also has a .pc file, so we don't need the fallback check any
24116 longer). Fixes #438840.
24118 2007-05-17 13:36:11 +0000 Wim Taymans <wim.taymans@gmail.com>
24120 gst/playback/gstqueue2.c: fix build.
24121 Original commit message from CVS:
24122 * gst/playback/gstqueue2.c: (gst_queue_get_type),
24123 (gst_queue_class_init), (gst_queue_finalize), (update_time_level),
24124 (apply_segment), (apply_buffer), (update_buffering),
24125 (reset_rate_timer), (update_rates), (gst_queue_locked_flush),
24126 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
24127 (gst_queue_handle_sink_event), (gst_queue_is_filled),
24128 (gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
24132 2007-05-17 11:57:44 +0000 Wim Taymans <wim.taymans@gmail.com>
24134 gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
24135 Original commit message from CVS:
24136 * gst/playback/Makefile.am:
24137 * gst/playback/gstqueue2.c: (gst_queue_get_type),
24138 (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
24139 (gst_queue_getcaps), (gst_queue_bufferalloc),
24140 (gst_queue_acceptcaps), (update_time_level), (apply_segment),
24141 (apply_buffer), (update_buffering), (reset_rate_timer),
24142 (update_rates), (gst_queue_locked_flush),
24143 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
24144 (gst_queue_handle_sink_event), (gst_queue_is_empty),
24145 (gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
24146 (gst_queue_loop), (gst_queue_handle_src_event),
24147 (gst_queue_handle_src_query), (gst_queue_sink_activate_push),
24148 (gst_queue_src_activate_push), (gst_queue_change_state),
24149 (gst_queue_set_property), (gst_queue_get_property), (plugin_init):
24150 On our way to playbin2 this is the new network queue that does buffering
24151 all by itself using high and low watermarks. It can also measure up and
24152 downstream bandwidth to optimally size the queue.
24154 2007-05-17 11:16:14 +0000 Michael Smith <msmith@xiph.org>
24156 gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
24157 Original commit message from CVS:
24158 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
24159 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
24160 Use the segment->last_stop value to calculate the next timestamp to
24161 generate after a seek; not the segment->start value.
24163 2007-05-15 20:14:06 +0000 David Schleef <ds@schleef.org>
24165 docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
24166 Original commit message from CVS:
24167 * docs/Makefile.am: Install docs even when --disable-gtk-doc
24168 is disabled. This matches the behavior of gtk+. Fixes #349099.
24170 2007-05-15 17:11:09 +0000 Wim Taymans <wim.taymans@gmail.com>
24172 ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
24173 Original commit message from CVS:
24174 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
24175 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
24176 Some more chained streaming ogg timestamp fixes.
24178 2007-05-15 16:46:10 +0000 Wim Taymans <wim.taymans@gmail.com>
24180 ext/ogg/gstoggdemux.c: Add some FIXMEs.
24181 Original commit message from CVS:
24182 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
24183 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
24184 (gst_ogg_demux_handle_page):
24186 Fix chain start/stop segment handling based on patch by
24187 <ahalda at cs dot mcgill dot ca> see #320984.
24189 2007-05-15 15:33:54 +0000 Michael Smith <msmith@xiph.org>
24191 configure.ac: We don't require a C++ compiler. So don't require one.
24192 Original commit message from CVS:
24194 We don't require a C++ compiler. So don't require one.
24196 2007-05-15 15:29:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24199 * ext/alsa/gstalsamixer.c:
24200 ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
24201 Original commit message from CVS:
24202 * ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
24203 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
24204 gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
24205 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
24206 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
24207 gst_alsa_mixer_update_track):
24208 Apply some of the cleanup Tim suggested in #152864 afterwards.
24210 2007-05-15 14:01:26 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
24212 ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
24213 Original commit message from CVS:
24214 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
24215 * ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
24216 _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
24217 gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
24218 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
24219 gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
24220 gst_alsa_mixer_handle_source_callback,
24221 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
24222 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
24223 gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
24224 gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
24225 gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
24226 gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
24227 * ext/alsa/gstalsamixer.h (handle_source, interface, dir):
24228 * ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
24229 gst_alsa_mixer_element_interface_supported,
24230 gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
24231 gst_alsa_mixer_element_set_property,
24232 gst_alsa_mixer_element_get_property,
24233 gst_alsa_mixer_element_change_state):
24234 * ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
24235 * gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
24236 gst_mixer_option_changed):
24237 * gst-libs/gst/interfaces/mixer.h (set_option, get_option,
24238 volume_changed, option_changed, _gst_reserved):
24239 Implement notification for alsamixer. Fixes #152864
24241 2007-05-15 03:53:11 +0000 David Schleef <ds@schleef.org>
24243 gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
24244 Original commit message from CVS:
24245 * gst/videotestsrc/videotestsrc.c:
24246 * gst/videotestsrc/videotestsrc.h:
24247 Add support for video/x-raw-bayer.
24249 2007-05-13 01:06:19 +0000 David Schleef <ds@schleef.org>
24251 sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
24252 Original commit message from CVS:
24253 * sys/xvimage/xvimagesink.c:
24254 Add some sanity checking for the XVImage size returned by X.
24255 Related to #377400.
24257 2007-05-12 16:18:39 +0000 Wim Taymans <wim.taymans@gmail.com>
24259 gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
24260 Original commit message from CVS:
24261 * gst-libs/gst/rtp/gstbasertpdepayload.c:
24262 (gst_base_rtp_depayload_setcaps),
24263 (gst_base_rtp_depayload_set_gst_timestamp):
24264 Parse and use additional caps fields as described in updated
24265 application/x-rtp caps spec.
24267 2007-05-12 16:16:22 +0000 Wim Taymans <wim.taymans@gmail.com>
24269 ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
24270 Original commit message from CVS:
24271 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
24272 (gst_ogg_demux_collect_chain_info):
24273 If there is a stream in a chain without any data packets, ignore the
24274 stream in the total length calculations. Might be related to #436820.
24276 2007-05-11 17:33:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24278 gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
24279 Original commit message from CVS:
24280 * gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
24281 (mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
24282 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
24283 (mpeg_video_type_find), (mpeg_video_stream_type_find),
24285 Consolidate and re-work our mpeg system stream detection to probe
24286 more packets and produce a higher confidence result. Fixes a
24287 regression caused by lowering the typefind probability last year
24288 - related to bug #397810. Remove the redundant MPEG-1 specific
24289 typefind function, as the new one detects both MPEG-1 & MPEG-2
24291 Also cleanup the MPEG elementary and MPEG-TS detection functions a
24293 Tested against my media test directory, with some improvements and
24296 2007-05-10 15:28:13 +0000 Wim Taymans <wim.taymans@gmail.com>
24298 gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
24299 Original commit message from CVS:
24300 * gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
24301 (queue_out_of_data):
24302 Connect to the new queue "pushing" signal instead of the broken
24305 2007-05-09 21:17:40 +0000 Sébastien Moutte <sebastien@moutte.net>
24307 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
24308 Original commit message from CVS:
24309 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
24310 (gst_base_rtp_audio_payload_handle_frame_based_buffer):
24311 Move variable declaration before the first instruction.
24312 * gst/videotestsrc/videotestsrc.c:
24313 Define M_PI if it's not defined yet.
24314 * win32/common/libgstrtp.def:
24315 Add new exported functions.
24317 2007-05-09 11:54:32 +0000 Michael Smith <msmith@xiph.org>
24319 ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
24320 Original commit message from CVS:
24321 * ext/theora/theoradec.c: (theora_handle_type_packet):
24322 gst_pad_push_event() does not return a GstFlowReturn!
24324 2007-05-09 11:25:34 +0000 Wim Taymans <wim.taymans@gmail.com>
24326 tests/examples/seek/: Some small cosmetic changes.
24327 Original commit message from CVS:
24328 * tests/examples/seek/scrubby.c: (stop_cb), (main):
24329 * tests/examples/seek/seek.c: (do_seek):
24330 Some small cosmetic changes.
24332 2007-05-08 19:24:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24335 * gst/adder/gstadder.c:
24336 * gst/adder/gstadder.h:
24337 gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
24338 Original commit message from CVS:
24339 * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
24340 gst_adder_change_state):
24341 * gst/adder/gstadder.h (bps, offset, collect_event, segment,
24342 segment_pending, segment_position, segment_rate):
24343 Handle playback-rate on adder.
24345 2007-05-07 11:43:31 +0000 Michael Smith <msmith@xiph.org>
24347 ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
24348 Original commit message from CVS:
24349 * ext/theora/gsttheoradec.h:
24350 * ext/theora/theoradec.c: (gst_theora_dec_reset),
24351 (theora_dec_sink_event), (theora_handle_comment_packet),
24352 (theora_handle_type_packet), (theora_dec_change_state):
24353 Don't push events (newsegment, tags) before initialising the
24355 This is neccesary for seeking to work correctly in gnonlin.
24357 2007-05-04 13:10:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24359 gst/: gst/audiotestsrc/gstaudiotestsrc.c
24360 Original commit message from CVS:
24361 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
24362 * gst/adder/gstadder.c:
24363 * gst/audiotestsrc/gstaudiotestsrc.c
24364 (gst_audio_test_src_create_white_noise):
24365 * gst/videotestsrc/gstvideotestsrc.c:
24366 * gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
24367 VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
24368 volume_sink_template, volume_src_template, gst_volume_init,
24369 volume_process_double, volume_process_int16,
24370 volume_process_int16_clamp):
24371 Doc fixes and formatting.
24373 2007-05-04 12:41:21 +0000 Tim-Philipp Müller <tim@centricular.net>
24375 tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
24376 Original commit message from CVS:
24377 * tests/check/Makefile.am:
24378 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
24379 Minimal check for volume's GstController usability; also another
24382 2007-05-04 09:06:38 +0000 Tim-Philipp Müller <tim@centricular.net>
24384 gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
24385 Original commit message from CVS:
24386 * gst-libs/gst/cdda/gstcddabasesrc.c:
24387 (gst_cdda_base_src_add_track):
24388 Fix it so that it (a) makes sense and (b) doesn't break
24389 everything cdda-related including the unit test.
24391 2007-05-04 08:46:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24393 gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
24394 Original commit message from CVS:
24395 * gst-libs/gst/cdda/gstcddabasesrc.c:
24396 (gst_cdda_base_src_add_track):
24397 Fix build when disabling asserts.
24399 2007-05-03 16:29:10 +0000 Tim-Philipp Müller <tim@centricular.net>
24401 sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
24402 Original commit message from CVS:
24403 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
24404 When XShm is not available, we might get row strides that are not
24405 rounded up to multiples of four; this is bad, because virtually
24406 every RGB-processing element in GStreamer assumes rowstrides are
24407 rounded up to multiples of four, so let's allocate at least enough
24408 memory to avoid crashes in this case. The image will still be
24409 displayed distorted though if this happens, so that still needs
24410 fixing (maybe by allocating a bigger image with an 'even' width
24411 and then clipping it appropriately when rendering - something for
24412 Xlib aficionados in any case).
24414 2007-05-03 13:16:21 +0000 Michael Smith <msmith@xiph.org>
24416 gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
24417 Original commit message from CVS:
24418 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
24419 If a buffer doesn't have a timestamp, assume it's contiguous with
24420 the previous buffer, and synthesise timestamps appropriately.
24422 2007-05-03 11:24:00 +0000 Edward Hervey <bilboed@bilboed.com>
24424 tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
24425 Original commit message from CVS:
24426 * tests/check/elements/videorate.c: (GST_START_TEST):
24427 Set buffer timestamp to a valid value in order to test the buffer
24428 really does stay in videorate.
24430 2007-05-03 10:47:22 +0000 Edward Hervey <bilboed@bilboed.com>
24432 gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
24433 Original commit message from CVS:
24434 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
24435 There is no sensible way to handle incoming buffers which don't have a
24436 valid timestamp. We therefore discard them and wait for the next one.
24438 2007-05-01 18:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
24440 gst/playback/: Better error message for text files.
24441 Original commit message from CVS:
24442 * gst/playback/gstdecodebin.c: (type_found), (plugin_init):
24443 * gst/playback/gstdecodebin2.c: (plugin_init):
24444 Better error message for text files.
24446 2007-04-29 14:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
24448 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
24449 Original commit message from CVS:
24450 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
24451 Fix offset bug in generation RR packets.
24453 2007-04-27 15:33:46 +0000 Julien Moutte <julien@moutte.net>
24455 ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
24456 Original commit message from CVS:
24457 2007-04-27 Julien MOUTTE <julien@moutte.net>
24458 * ext/theora/theoradec.c: (_theora_granule_time),
24459 (theora_dec_push_forward), (theora_handle_data_packet),
24460 (theora_dec_decode_buffer): Calculate buffer duration correctly
24461 to generate a perfect stream (#433888).
24462 * gst/audioresample/gstaudioresample.c:
24463 (audioresample_check_discont): Glib provides ABS.
24465 2007-04-27 15:01:40 +0000 Wim Taymans <wim.taymans@gmail.com>
24467 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
24468 Original commit message from CVS:
24469 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
24470 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
24471 (gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
24472 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
24473 (gst_rtcp_packet_bye_set_reason):
24474 * gst-libs/gst/rtp/gstrtcpbuffer.h:
24475 Fix RB block parsing and writing.
24476 Add support for constructing BYE packets.
24478 2007-04-25 08:54:34 +0000 Tim-Philipp Müller <tim@centricular.net>
24480 When posting a warning message because samples were dropped, post something more intelligible than he default error m...
24481 Original commit message from CVS:
24482 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
24483 (gst_base_audio_src_create):
24485 When posting a warning message because samples were dropped, post
24486 something more intelligible than he default error message for clock
24487 errors which is just confusing in this context (#432984).
24489 2007-04-25 08:10:26 +0000 Wim Taymans <wim.taymans@gmail.com>
24491 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
24492 Original commit message from CVS:
24493 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
24494 (gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
24495 (read_packet_header), (gst_rtcp_packet_move_to_next),
24496 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
24497 (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
24498 (gst_rtcp_packet_sdes_get_item_count),
24499 (gst_rtcp_packet_sdes_first_item),
24500 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
24501 (gst_rtcp_packet_sdes_first_entry),
24502 (gst_rtcp_packet_sdes_next_entry),
24503 (gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
24504 (gst_rtcp_packet_sdes_add_entry):
24505 * gst-libs/gst/rtp/gstrtcpbuffer.h:
24506 Implement code to write SR, RR and SDES packets.
24508 2007-04-24 20:45:24 +0000 Christian Kirbach <Christian.Kirbach@googlemail.com>
24510 sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
24511 Original commit message from CVS:
24512 Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
24513 * sys/ximage/ximagesink.c:
24514 Fix build if XShm is not available (#432362).
24516 2007-04-24 18:58:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
24518 gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
24519 Original commit message from CVS:
24520 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
24521 Initalize the AudioConvertCtx with zeroes, otherwise it will contain
24522 pointers to random memory which are passed to g_free() when
24523 audio_convert_prepare_context() is called the first time.
24525 2007-04-24 15:00:07 +0000 Dan Williams <dcbw@redhat.com>
24527 gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
24528 Original commit message from CVS:
24529 Patch by: Dan Williams <dcbw redhat com>
24530 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
24531 Don't leak incoming buffer if gst_pad_push() returns a
24532 non-OK flow. Fixes #432755.
24533 * tests/check/elements/videorate.c: (GST_START_TEST),
24535 Unit test for the above by Yours Truly.
24537 2007-04-23 20:04:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24539 gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
24540 Original commit message from CVS:
24541 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
24542 (gst_adder_sink_event), (gst_adder_collected):
24543 Fix non-flushing segmented seeks, Fixes #340060 for me
24545 2007-04-21 15:29:27 +0000 Tim-Philipp Müller <tim@centricular.net>
24548 ChangeLog surgery: add API keyword
24549 Original commit message from CVS:
24550 ChangeLog surgery: add API keyword
24552 2007-04-21 15:25:22 +0000 Olivier Crete <tester@tester.ca>
24554 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
24555 Original commit message from CVS:
24556 Patch by: Olivier Crete <tester at tester ca>
24557 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
24558 (gst_base_rtp_audio_payload_class_init),
24559 (gst_base_rtp_audio_payload_init),
24560 (gst_base_rtp_audio_payload_dispose):
24561 Chain up to parent class in dispose function; get rid of
24562 unnecessary 'diposed' flag in private structure (#415001).
24564 2007-04-21 15:10:25 +0000 Tim-Philipp Müller <tim@centricular.net>
24566 Some minor docs fixes and additions; also add missing 'Since' bits.
24567 Original commit message from CVS:
24568 * docs/libs/gst-plugins-base-libs.types:
24569 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
24570 (gst_base_rtp_audio_payload_class_init):
24571 * gst-libs/gst/rtp/gstbasertpdepayload.c:
24572 * gst-libs/gst/rtp/gstbasertppayload.c:
24573 Some minor docs fixes and additions; also add missing 'Since' bits.
24575 2007-04-21 14:40:45 +0000 Zeeshan Ali <zeenix@gmail.com>
24577 gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
24578 Original commit message from CVS:
24579 Patch by: Zeeshan Ali <zeenix gmail com>
24580 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
24581 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
24582 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
24583 (gst_base_rtp_audio_payload_push):
24584 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
24585 The recently-added gst_base_rtp_audio_payload_push() should take an
24586 object of type GstBaseRTPAudioPayload as first argument (#431672).
24588 2007-04-21 14:14:24 +0000 Tim-Philipp Müller <tim@centricular.net>
24590 gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
24591 Original commit message from CVS:
24592 * gst/audioresample/gstaudioresample.c:
24593 Make more functions static, just because we can.
24595 2007-04-21 13:54:39 +0000 Tim-Philipp Müller <tim@centricular.net>
24597 tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
24598 Original commit message from CVS:
24599 * tests/check/elements/audioresample.c:
24600 Add unit test for audioresample shutdown crasher (#420106).
24602 2007-04-20 10:42:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24604 gst/subparse/: Use GST_DISABLE_XML here
24605 Original commit message from CVS:
24606 * gst/subparse/gstsubparse.c:
24607 * gst/subparse/samiparse.c:
24608 Use GST_DISABLE_XML here
24609 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
24610 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
24611 (gst_xvimagesink_buffer_alloc),
24612 (gst_xvimagesink_navigation_send_event):
24613 * sys/xvimage/xvimagesink.h:
24614 Include stdlib.h when using atoi.
24615 * tests/check/elements/playbin.c: (playbin_suite):
24616 Use GST_DISABLE_REGISTRY here
24618 2007-04-19 16:58:53 +0000 Michael Smith <msmith@xiph.org>
24620 ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
24621 Original commit message from CVS:
24622 * ext/theora/gsttheoraenc.h:
24623 * ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
24624 (theora_enc_sink_event), (theora_enc_change_state):
24625 Track initialisation state; don't try to use encoder state if we're
24626 not initialised (it'll segfault).
24628 2007-04-18 11:06:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24630 tests/check/pipelines/.cvsignore: Fix build.
24631 Original commit message from CVS:
24632 * tests/check/pipelines/.cvsignore:
24635 2007-04-17 10:56:37 +0000 Tim-Philipp Müller <tim@centricular.net>
24637 gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
24638 Original commit message from CVS:
24639 * gst/app/Makefile.am:
24640 Fix CFLAGS and hopefully #430594.
24642 2007-04-17 02:53:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
24644 gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
24645 Original commit message from CVS:
24646 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24647 Allow random depths between 1 and 32 instead of only multiplies of 8.
24649 2007-04-17 02:04:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
24651 gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
24652 Original commit message from CVS:
24653 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24654 Set the maximum number of channels for PCM and float in the correct
24655 place to have it also used when creating the template caps.
24657 2007-04-17 01:56:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
24659 gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
24660 Original commit message from CVS:
24661 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24662 Correctly support 4, 6 and 8 channels with normal PCM and float
24664 Fix the depth and signedness calculation in extensible wav files and
24665 also handle 1, 2, 4, 6, 8 channels here when a file without channel
24667 Add support for float, alaw and mulaw in extensible wav files.
24668 This allows correct playback of all but 5 files from
24669 http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
24670 (gst_riff_create_audio_template_caps):
24671 Add voxware and float formats to the template caps.
24673 2007-04-16 22:20:03 +0000 Vincent Torri <vtorri@univ-evry.fr>
24675 ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
24676 Original commit message from CVS:
24677 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
24678 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
24679 Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
24680 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24681 * gst/audioresample/gstaudioresample.c: (audioresample_do_output):
24682 Use the correct format strings for integer formats.
24684 2007-04-16 21:44:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24686 * gst-plugins-base.doap:
24688 Original commit message from CVS:
24691 2007-04-16 21:42:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24693 * gst-plugins-base.doap:
24695 Original commit message from CVS:
24698 2007-04-15 14:35:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24700 ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
24701 Original commit message from CVS:
24702 * ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
24703 Don't use pad_alloc_buffer_and_set_caps to create a small header
24704 packet, or, worse, to create a big temporary video buffer using the
24707 2007-04-14 12:34:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24709 gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
24710 Original commit message from CVS:
24711 * gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
24712 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
24713 GST_START_TEST, buffer_probe_cb, GST_START_TEST):
24714 Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
24716 2007-04-13 22:10:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24718 * gst/tcp/gstmultifdsink.c:
24720 Original commit message from CVS:
24723 2007-04-13 21:55:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24726 * tests/check/pipelines/streamheader.c:
24727 tests/check/pipelines/streamheader.c (tag_event_probe_cb,
24728 Original commit message from CVS:
24729 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
24730 GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
24731 streamheader_suite):
24732 Add another test set up for failure
24734 2007-04-13 21:09:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24736 * ext/ogg/gstoggmux.c:
24737 * gst/gdp/gstgdpdepay.c:
24739 Original commit message from CVS:
24742 2007-04-13 21:08:11 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24744 tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
24745 Original commit message from CVS:
24746 * tests/check/Makefile.am:
24747 * tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
24748 GST_START_TEST, streamheader_suite, main):
24749 Add a test for the streamheader bug Wim fixed.
24751 2007-04-13 11:42:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24753 ext/theora/theoradec.c: Fix misleading comment.
24754 Original commit message from CVS:
24755 * ext/theora/theoradec.c: (theora_dec_sink_event):
24756 Fix misleading comment.
24758 2007-04-13 06:17:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24760 gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
24761 Original commit message from CVS:
24762 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24763 More sanity checks for the header fields.
24765 2007-04-12 16:36:36 +0000 Tim-Philipp Müller <tim@centricular.net>
24767 gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
24768 Original commit message from CVS:
24769 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
24770 Try encodings from all environment variables, not just those in the
24771 first environment variable that is set.
24773 2007-04-12 15:00:03 +0000 Wim Taymans <wim.taymans@gmail.com>
24775 gst/videorate/gstvideorate.c: Add some debug.
24776 Original commit message from CVS:
24777 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
24778 (gst_video_rate_chain):
24780 * tests/check/elements/videorate.c: (GST_START_TEST),
24782 Added check for videorate changing caps handling. Closes #421834.
24784 2007-04-12 12:57:33 +0000 Michael Smith <msmith@xiph.org>
24786 ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
24787 Original commit message from CVS:
24788 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
24789 Use scale functions to avoid overflow when calculating duration of
24792 2007-04-12 12:19:20 +0000 Tim-Philipp Müller <tim@centricular.net>
24794 API: add gst_tag_freeform_string_to_utf8() (#405072).
24795 Original commit message from CVS:
24796 * docs/libs/gst-plugins-base-libs-sections.txt:
24797 * gst-libs/gst/tag/tag.h:
24798 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
24799 API: add gst_tag_freeform_string_to_utf8() (#405072).
24800 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
24801 Use gst_tag_freeform_string_to_utf8() here.
24803 2007-04-12 10:38:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24805 * gst/tcp/gstmultifdsink.c:
24807 Original commit message from CVS:
24810 2007-04-12 10:03:22 +0000 Wim Taymans <wim.taymans@gmail.com>
24812 gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
24813 Original commit message from CVS:
24814 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
24815 (gst_gdp_pay_sink_event):
24816 Make sure we set the IN_CAPS flag correctly.
24817 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
24818 Get the IN_CAPS flag before we call functions that mess with the flags.
24820 2007-04-10 20:37:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24823 * gst/gdp/gstgdppay.c:
24824 gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
24825 Original commit message from CVS:
24826 * gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
24827 gst_gdp_pay_chain, gst_gdp_pay_sink_event):
24828 Only stamp buffers with offset/offset_end right before they get
24829 pushed. This ensures offset continuity, which was not the case
24831 gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
24833 2007-04-10 20:25:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24835 * gst/gdp/gstgdpdepay.c:
24836 * gst/gdp/gstgdppay.c:
24838 Original commit message from CVS:
24841 2007-04-10 11:23:18 +0000 Christian Schaller <uraeus@gnome.org>
24844 * gst-plugins-base.spec.in:
24845 update spec file for RTP changes
24846 Original commit message from CVS:
24847 update spec file for RTP changes
24849 2007-04-06 12:58:06 +0000 Wim Taymans <wim.taymans@gmail.com>
24851 gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
24852 Original commit message from CVS:
24853 * gst/playback/gstplaybin.c: (add_sink),
24854 (gst_play_bin_change_state):
24855 Activate sync in playbin, we are ready to handle it for live streams.
24857 2007-04-06 09:56:18 +0000 Tim-Philipp Müller <tim@centricular.net>
24859 tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
24860 Original commit message from CVS:
24861 * tests/check/elements/playbin.c:
24862 (test_sink_usage_video_only_stream), (playbin_suite):
24863 Add small test for stream-info-value-array code paths.
24865 2007-04-05 15:44:40 +0000 Wim Taymans <wim.taymans@gmail.com>
24867 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
24868 Original commit message from CVS:
24869 * gst-libs/gst/audio/gstbaseaudiosink.c:
24870 (gst_base_audio_sink_skew_slaving):
24871 Don't try to create invalid calibration parameters by making the
24872 internal time go backwards, instead make external time go forward.
24874 2007-04-05 10:27:06 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
24876 gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
24877 Original commit message from CVS:
24878 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
24879 * gst/playback/gstplaybasebin.c: (add_stream):
24880 Fix leak in add_stream(), when g_value_set_object() increases the
24881 refcount of streaminfo object. Fixes #426250.
24883 2007-04-04 02:45:03 +0000 David Schleef <ds@schleef.org>
24885 gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
24886 Original commit message from CVS:
24887 * gst/videotestsrc/gstvideotestsrc.c:
24888 * gst/videotestsrc/gstvideotestsrc.h:
24889 * gst/videotestsrc/videotestsrc.c:
24890 * gst/videotestsrc/videotestsrc.h:
24891 Add a test pattern called "circular", which has concentric
24892 rings with varying radial frequency. The main purpose of this
24893 pattern is to test fidelity loss in a filter or scaler element.
24894 Notably, this pattern is scale invariant, and is optimally viewed
24895 with a width (and height) of 400.
24897 2007-04-03 11:10:52 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
24899 gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
24900 Original commit message from CVS:
24901 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
24902 * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
24903 (deactivate_free_recursive):
24904 Decodebin2 doesn't unref pads it obtains in some occasions:
24905 - multiqueue src pads, when either connecting further or exposing
24906 - sink pads of new autoplugged elements
24907 - peer pads when recursively freeing elements
24910 2007-03-30 17:05:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
24912 gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
24913 Original commit message from CVS:
24914 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24915 Add audio/x-raw-float support, now that audioconvert support
24916 non-native endianness floats.
24918 2007-03-30 15:00:49 +0000 Tim-Philipp Müller <tim@centricular.net>
24920 docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
24921 Original commit message from CVS:
24922 * docs/libs/gst-plugins-base-libs-docs.sgml:
24923 gstreamer-plugins-base.pc doesn't exist, it's
24924 gstreamer-plugins-base-0.10.pc.
24926 2007-03-29 18:42:34 +0000 René Stadler <mail@renestadler.de>
24928 with some minor changes
24929 Original commit message from CVS:
24930 Patch by: René Stadler <mail at renestadler dot de>
24931 with some minor changes
24932 * gst-libs/gst/floatcast/floatcast.h:
24933 Use more efficient float endianness conversion functions that don't
24934 involve 2 function calls per value.
24935 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
24936 (check_default), (audio_convert_prepare_context):
24937 * gst/audioconvert/gstaudioconvert.c:
24938 (gst_audio_convert_parse_caps), (make_lossless_changes):
24939 Support non-native endianness floats as input and output.
24941 * tests/check/elements/audioconvert.c: (verify_convert),
24943 Add unit tests for the non-native endianness float conversions.
24945 2007-03-29 16:23:53 +0000 Wim Taymans <wim.taymans@gmail.com>
24947 gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
24948 Original commit message from CVS:
24949 * gst-libs/gst/rtp/gstbasertpdepayload.c:
24950 (gst_base_rtp_depayload_base_init),
24951 (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
24952 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
24953 (gst_base_rtp_depayload_set_gst_timestamp),
24954 (gst_base_rtp_depayload_change_state),
24955 (gst_base_rtp_depayload_set_property),
24956 (gst_base_rtp_depayload_get_property):
24957 * gst-libs/gst/rtp/gstbasertpdepayload.h:
24958 Add Private structure.
24959 Bring element code to 2007.
24960 Parse clock-base caps param and use it when generating the
24962 Reset variables before going to PAUSED.
24965 2007-03-29 16:20:31 +0000 Wim Taymans <wim.taymans@gmail.com>
24968 Original commit message from CVS:
24969 * docs/libs/gst-plugins-base-libs-docs.sgml:
24970 * docs/libs/gst-plugins-base-libs-sections.txt:
24971 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
24972 (gst_base_rtp_audio_payload_get_adapter):
24974 Fix some more docs.
24975 * gst-libs/gst/rtp/Makefile.am:
24976 * gst-libs/gst/rtp/gstrtcpbuffer.c:
24977 (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
24978 (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
24979 (gst_rtcp_buffer_get_packet_count), (read_packet_header),
24980 (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
24981 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
24982 (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
24983 (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
24984 (gst_rtcp_packet_sr_get_sender_info),
24985 (gst_rtcp_packet_sr_set_sender_info),
24986 (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
24987 (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
24988 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
24989 (gst_rtcp_packet_sdes_get_chunk_count),
24990 (gst_rtcp_packet_sdes_first_chunk),
24991 (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
24992 (gst_rtcp_packet_sdes_first_item),
24993 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
24994 (gst_rtcp_packet_bye_get_ssrc_count),
24995 (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
24996 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
24997 (gst_rtcp_packet_bye_get_reason_len),
24998 (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
24999 * gst-libs/gst/rtp/gstrtcpbuffer.h:
25000 Add new helper object for parsing and creating RTCP messages.
25002 2007-03-29 12:07:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
25004 gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
25005 Original commit message from CVS:
25006 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
25007 PCM samples with width=8 must be always unsigned, no matter what
25010 2007-03-29 11:24:47 +0000 Andy Wingo <wingo@pobox.com>
25012 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
25013 Original commit message from CVS:
25014 2007-03-29 Andy Wingo <wingo@pobox.com>
25015 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
25016 perfect offsets also, not just timestamps.
25017 * tests/check/elements/videorate.c (test_more): Test that given
25018 any incoming offsets, that videorate produces perfect offsets.
25020 2007-03-29 10:19:45 +0000 Wim Taymans <wim.taymans@gmail.com>
25022 gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
25023 Original commit message from CVS:
25024 * gst-libs/gst/riff/riff-ids.h:
25025 Add some more RIFF formats.
25027 2007-03-29 10:17:52 +0000 Wim Taymans <wim.taymans@gmail.com>
25029 gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
25030 Original commit message from CVS:
25031 * gst-libs/gst/rtp/gstrtpbuffer.c:
25032 (gst_rtp_buffer_default_clock_rate):
25033 * gst-libs/gst/rtp/gstrtpbuffer.h:
25034 Fix fixed payload names and docs.
25035 Added method to get the default clock rates of fixed payload types.
25036 API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
25038 2007-03-28 15:24:40 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
25040 tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
25041 Original commit message from CVS:
25042 * tests/check/pipelines/.cvsignore:
25043 Add new vorbisdec test to cvsignore.
25045 2007-03-28 14:50:47 +0000 Wim Taymans <wim.taymans@gmail.com>
25047 gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
25048 Original commit message from CVS:
25049 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
25050 (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
25051 (gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
25052 (gst_base_audio_sink_set_property),
25053 (gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
25054 (clock_convert_external), (gst_base_audio_sink_resample_slaving),
25055 (gst_base_audio_sink_skew_slaving),
25056 (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
25057 (gst_base_audio_sink_async_play):
25058 * gst-libs/gst/audio/gstbaseaudiosink.h:
25059 Store private stuff in GstBaseAudioSinkPrivate.
25060 Add configurable clock slaving modes property.
25061 API:: GstBaseAudioSink::slave-method property
25062 Some more latency reporting tweaks.
25063 Added skew based clock slaving correction and make it the default until
25064 the resampling method is more robust.
25066 2007-03-27 12:44:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
25068 gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
25069 Original commit message from CVS:
25070 * gst/audioconvert/audioconvert.c:
25071 Add docs to the integer pack functions and implement proper
25072 rounding. Before we had rounding towards negative infinity, i.e.
25073 always the smaller number was taken. Now we use natural rounding,
25074 i.e. rounding to the nearest integer and to the one with the largest
25075 absolute value for X.5. The old rounding introduced some minor
25076 distortions. Fixes #420079
25077 * tests/check/elements/audioconvert.c: (GST_START_TEST):
25078 Fix one unit test that assumed the old rounding and added unit tests
25079 for checking signed/unsigned int16 <-> signed/unsigned int16 with
25080 depth 8, one for signed int16 <-> unsigned int16 and one for the new
25081 rounding from signed int32 to signed/unsigned int16.
25083 2007-03-27 11:31:17 +0000 Michael Smith <msmith@xiph.org>
25085 gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
25086 Original commit message from CVS:
25087 * gst/audioconvert/gstaudioconvert.c: (strip_width_64),
25088 (gst_audio_convert_transform_caps):
25089 Fix typo in debug line introduced recently, as pointed out on irc.
25091 2007-03-27 10:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
25093 Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
25094 Original commit message from CVS:
25095 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
25096 * tests/check/libs/tag.c: (GST_START_TEST):
25097 Make sure we parse floating-point numbers in vorbis comments
25098 correctly with either '.' or ',' as separator, no matter what
25099 the current locale is. Add unit test for this too.
25101 2007-03-27 09:37:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25103 * tests/check/pipelines/vorbisdec.c:
25105 Original commit message from CVS:
25108 2007-03-26 22:38:19 +0000 René Stadler <mail@renestadler.de>
25110 gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
25111 Original commit message from CVS:
25112 Patch by: René Stadler <mail at renestadler de>
25113 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
25114 When writing out floating-point numbers to vorbis comment tags, always
25115 use the same character as separator no matter what the current locale is
25117 * tests/check/libs/tag.c: (GST_START_TEST):
25118 Add unit tests for replaygain tags in vorbis comments (closes #423055).
25120 2007-03-26 20:56:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25122 ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
25123 Original commit message from CVS:
25124 * ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
25125 vorbis_handle_data_packet):
25126 Correctly set DURATION to generate a timestamp-continuous stream.
25127 One bug left at the end; see
25128 ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
25129 * tests/check/Makefile.am:
25130 * tests/check/pipelines/vorbisenc.c (GST_START_TEST):
25131 Add a test to check this. Without the above patch this test fails.
25133 2007-03-26 11:44:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25135 gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
25136 Original commit message from CVS:
25137 * gst-libs/gst/rtp/Makefile.am:
25138 The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
25140 2007-03-23 15:43:24 +0000 Christian Schaller <uraeus@gnome.org>
25142 * gst-plugins-base.spec.in:
25144 Original commit message from CVS:
25147 2007-03-23 12:32:33 +0000 Michael Smith <msmith@xiph.org>
25149 gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
25150 Original commit message from CVS:
25151 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
25152 (gst_video_rate_reset), (gst_video_rate_chain):
25153 If videorate changes caps, we can no longer use the old buffer
25154 (which may have a different size, incompatible with our caps).
25155 So don't do that; just duplicate the new frame more times.
25157 2007-03-22 17:43:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25159 gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
25160 Original commit message from CVS:
25161 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
25162 Remove playbin's override of the set_clock vmethod. It's irrelevant
25163 after Wim's commit on the 19th.
25165 2007-03-22 14:37:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25167 gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
25168 Original commit message from CVS:
25169 * gst-libs/gst/app/Makefile.am:
25170 Use GST_ALL_LDFLAGS, which actually exists, but maybe David
25171 can confirm that was what he wanted.
25173 2007-03-22 09:26:02 +0000 Wim Taymans <wim.taymans@gmail.com>
25175 ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
25176 Original commit message from CVS:
25177 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
25178 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
25179 * ext/gnomevfs/gstgnomevfssrc.h:
25180 Don't cache file sizes. Fixes #341078.
25182 2007-03-21 11:03:23 +0000 Tim-Philipp Müller <tim@centricular.net>
25184 gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
25185 Original commit message from CVS:
25186 * gst/playback/gstplaybin.c: (add_sink):
25187 Use GST_PTR_FORMAT to log caps.
25189 2007-03-21 10:23:11 +0000 Young-Ho Cha <ganadist@chollian.net>
25191 gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
25192 Original commit message from CVS:
25193 Patch by: Young-Ho Cha <ganadist at chollian net>
25194 * gst/subparse/samiparse.c: (handle_start_font):
25195 Special-case some more colour names that pango doesn't handle by
25196 default. Fixes #420578.
25198 2007-03-20 11:49:55 +0000 Michael Smith <msmith@xiph.org>
25200 ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
25201 Original commit message from CVS:
25202 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
25203 If we get a zero-sized input buffer, don't pass it to libvorbis, as
25204 that marks EOS internally. After that, libvorbis will buffer all
25205 input data, and encode none of it, eventually leading to memory
25208 2007-03-19 10:52:50 +0000 Wim Taymans <wim.taymans@gmail.com>
25210 gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
25211 Original commit message from CVS:
25212 * gst/playback/gstdecodebin.c: (remove_fakesink):
25213 Don't post STATE_DIRTY anymore.
25214 * gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
25215 (gst_play_bin_change_state):
25216 Remove stream_time reset in seek handling, core does that now.
25217 Disable clocking for live pipelines by forcing a NULL clock to the
25218 complete pipeline, core is too smart now for our previous hack.
25219 We can always autoplug in PAUSED now.
25221 2007-03-18 03:14:01 +0000 David Schleef <ds@schleef.org>
25223 REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
25224 Original commit message from CVS:
25225 * REQUIREMENTS: Update this file, change the formatting to make
25226 it more consistent, plus more machine readable.
25228 2007-03-16 17:29:09 +0000 Michael Smith <msmith@xiph.org>
25230 gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
25231 Original commit message from CVS:
25232 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
25233 (strip_width_64), (append_with_other_format):
25234 Previous fix was too simplistic, and broke the tests. Use a better
25235 approach; only strip 64 from widths for integer audio.
25237 2007-03-16 16:42:23 +0000 Michael Smith <msmith@xiph.org>
25239 gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
25240 Original commit message from CVS:
25241 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
25242 (gst_audio_convert_transform_caps):
25243 We don't support 64 bit integer audio, so don't try to claim we can.
25244 Stops us producing caps don't match our template caps.
25247 2007-03-15 10:52:21 +0000 Michael Smith <msmith@xiph.org>
25249 gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
25250 Original commit message from CVS:
25251 * gst/audioresample/gstaudioresample.c:
25252 (audioresample_check_discont), (audioresample_transform):
25253 Don't trigger discontinuities for very small imperfections; a filter
25254 flush will sound bad, and many plugins have rounding errors leading
25257 2007-03-14 21:11:18 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
25259 gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
25260 Original commit message from CVS:
25261 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
25262 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
25263 Add min-ptime property to RTP base audio payloader. Patch by
25264 olivier.crete@collabora.co.uk.
25266 Indentation/whitespace/documentation fixes.
25268 2007-03-14 17:16:30 +0000 Julien Moutte <julien@moutte.net>
25270 gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
25271 Original commit message from CVS:
25272 2007-03-14 Julien MOUTTE <julien@moutte.net>
25273 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
25274 (audioresample_transform_size), (audioresample_do_output),
25275 (audioresample_transform), (audioresample_pushthrough): Handle
25276 discontinuous streams.
25277 * gst/audioresample/gstaudioresample.h:
25278 * tests/check/elements/audioresample.c:
25279 (test_discont_stream_instance), (GST_START_TEST),
25280 (audioresample_suite): Add a test for discontinuous streams.
25281 * win32/common/config.h: Updated.
25283 2007-03-14 15:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25285 po/: Update translations from translation project.
25286 Original commit message from CVS:
25300 Update translations from translation project.
25302 2007-03-14 15:05:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25304 * gst/gdp/gstgdpdepay.c:
25306 Original commit message from CVS:
25309 2007-03-14 14:48:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25311 gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
25312 Original commit message from CVS:
25313 * gst/audioresample/debug.h:
25314 * gst/audioresample/resample.c: (resample_init):
25315 Since I really am not interested in a debug line for each sample
25316 being processed, move the library's debugging to its own category,
25319 2007-03-14 14:09:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25321 * gst/audioresample/gstaudioresample.c:
25322 add debugging and reformat docs
25323 Original commit message from CVS:
25324 add debugging and reformat docs
25326 2007-03-12 23:29:07 +0000 Michael Smith <msmith@xiph.org>
25328 ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
25329 Original commit message from CVS:
25330 * ext/theora/theoradec.c: (theora_handle_type_packet):
25331 Since the plugin doesn't support anything other than 4:2:0 right
25332 now, post an error and fail if we get something else. Won't matter
25333 until libtheora supports the other pixel formats, but hopefully
25336 2007-03-12 15:50:35 +0000 Alex Lancaster <alexlan@fedoraproject.org>
25339 I'm too lazy to comment this
25340 Original commit message from CVS:
25341 Mention Patch by: Alex Lancaster in a recent commit.
25343 2007-03-12 11:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25345 examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.
25346 Original commit message from CVS:
25347 * examples/app/.cvsignore:
25348 The buildbot demands .cvsignore files, and I comply.
25350 2007-03-11 00:48:26 +0000 David Schleef <ds@schleef.org>
25352 Add appsrc/appsink example.
25353 Original commit message from CVS:
25355 * examples/Makefile.am:
25356 * examples/app/Makefile.am:
25357 * examples/app/appsrc_ex.c:
25358 Add appsrc/appsink example.
25359 * gst-libs/gst/app/Makefile.am:
25360 * gst-libs/gst/app/gstapp.c:
25361 * gst-libs/gst/app/gstappsink.c:
25362 * gst-libs/gst/app/gstappsink.h:
25363 * gst/app/gstapp.c:
25366 2007-03-10 15:59:33 +0000 Sébastien Moutte <sebastien@moutte.net>
25368 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
25369 Original commit message from CVS:
25370 * gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
25371 Use gst_guint64_to_gdouble for conversion.
25373 Add new files to the win32 MANIFEST.
25374 * win32/common/libgstaudio.def:
25375 * win32/common/libgstpbutils.def:
25376 Add new exported functions.
25377 * win32/vs6/gst_plugins_base.dsw:
25378 * win32/vs6/libgstdecodebin.dsp:
25379 * win32/vs6/libgstplaybin.dsp:
25380 Change the link to libgstpbutils.lib.
25381 * win32/vs6/libgstdecodebin2.dsp:
25382 Add a new project for decodebin2.
25383 * win32/vs6/libgstpbutils.dsp:
25384 Add a new project for pbutils.
25386 2007-03-10 12:18:58 +0000 Tim-Philipp Müller <tim@centricular.net>
25388 gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
25389 Original commit message from CVS:
25390 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
25391 Also accept partial dates with only year and month,
25392 like 1999-12-00 (fixes #410396 even more).
25393 * tests/check/libs/tag.c: (GST_START_TEST):
25394 Add unit test for the above.
25396 2007-03-10 11:21:08 +0000 Tim-Philipp Müller <tim@centricular.net>
25398 tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
25399 Original commit message from CVS:
25400 * tests/check/elements/subparse.c: (GST_START_TEST),
25402 Add unit test for MPL2 subtitle format (#413799).
25404 2007-03-10 11:17:52 +0000 Kamil Pawlowski <kamilpe@gmail.com>
25406 gst/subparse/: Add support for MPL2 subtitle format (#413799).
25407 Original commit message from CVS:
25408 Patch by: Kamil Pawlowski <kamilpe gmail com>
25409 * gst/subparse/Makefile.am:
25410 * gst/subparse/gstsubparse.c:
25411 (gst_sub_parse_data_format_autodetect),
25412 (gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
25413 (gst_subparse_type_find):
25414 * gst/subparse/gstsubparse.h:
25415 * gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
25416 * gst/subparse/mpl2parse.h:
25417 Add support for MPL2 subtitle format (#413799).
25419 2007-03-09 17:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
25421 configure.ac: We require core CVS for the new buffer metadata copy functions.
25422 Original commit message from CVS:
25424 We require core CVS for the new buffer metadata copy functions.
25426 2007-03-09 16:51:13 +0000 Wim Taymans <wim.taymans@gmail.com>
25428 gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
25429 Original commit message from CVS:
25430 * gst-libs/gst/tag/gstid3tag.c:
25431 Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
25434 2007-03-09 16:46:35 +0000 Wim Taymans <wim.taymans@gmail.com>
25436 ext/libvisual/visual.c: Improve adapter usage and comments.
25437 Original commit message from CVS:
25438 * ext/libvisual/visual.c: (gst_visual_sink_setcaps),
25439 (gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
25440 Improve adapter usage and comments.
25442 2007-03-09 16:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
25444 Use new metadata copy function.
25445 Original commit message from CVS:
25446 * ext/pango/gsttextrender.c: (gst_text_render_chain):
25447 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
25448 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
25449 Use new metadata copy function.
25450 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
25451 (gst_ffmpegcsp_transform):
25452 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
25453 Basetransform copied the metadata for us.
25455 2007-03-09 16:28:04 +0000 Tim-Philipp Müller <tim@centricular.net>
25457 ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
25458 Original commit message from CVS:
25459 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
25460 (gst_text_overlay_video_event):
25461 Some more logging. Only accept newsegment events in TIME format and
25462 send a WARNING message if they are not in TIME format.
25463 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
25464 (gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
25465 (gst_sub_parse_chain), (gst_sub_parse_sink_event):
25466 * gst/subparse/gstsubparse.h:
25467 No need to allocate GstSegment structure dynamically, just put it
25468 into the instance structure; ignore newsegment events in BYTE
25469 format and in particular don't let it overwrite our saved TIME
25470 segment from the last seek.
25472 2007-03-09 13:05:04 +0000 Michael Smith <msmith@xiph.org>
25474 gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
25475 Original commit message from CVS:
25476 * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
25477 Replace AC3 typefinder with one that isn't terrible, and actually
25480 2007-03-09 12:22:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25482 gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
25483 Original commit message from CVS:
25484 * gst/audioconvert/gstaudioconvert.c:
25485 (gst_audio_convert_transform):
25486 fix error category and translatable string
25488 2007-03-09 11:23:32 +0000 Tim-Philipp Müller <tim@centricular.net>
25490 pkgconfig/: Fix up utils => pbutils here too.
25491 Original commit message from CVS:
25492 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
25493 * pkgconfig/gstreamer-plugins-base.pc.in:
25494 Fix up utils => pbutils here too.
25496 2007-03-09 10:49:53 +0000 Tim-Philipp Müller <tim@centricular.net>
25498 gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
25499 Original commit message from CVS:
25500 * gst/subparse/gstsubparse.c: (handle_buffer):
25501 Break out of loop in chain function as soon as possible if we get
25502 a non-OK flow return.
25504 2007-03-08 18:26:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25506 tests/check/elements/alsa.c: Unref the mixer if the state change fails too (if the alsa devices are inaccessible, for...
25507 Original commit message from CVS:
25508 * tests/check/elements/alsa.c: (GST_START_TEST):
25509 Unref the mixer if the state change fails too (if the
25510 alsa devices are inaccessible, for example)
25512 2007-03-08 17:49:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25514 tests/check/Makefile.am: Don't test libvisual elements in the states check, because libvisual seems to leak internally.
25515 Original commit message from CVS:
25516 * tests/check/Makefile.am:
25517 Don't test libvisual elements in the states check, because libvisual
25518 seems to leak internally.
25519 Re-enable the alsa and states tests now that there's new suppressions
25521 * tests/check/elements/alsa.c: (GST_START_TEST):
25522 Don't leak the alsamixer we instantiated.
25524 2007-03-08 15:22:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25526 sys/: Move some cleanup stuff from the state change handler into a _reset() function that can be called from _finaliz...
25527 Original commit message from CVS:
25528 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
25529 (gst_ximagesink_change_state), (gst_ximagesink_reset),
25530 (gst_ximagesink_finalize):
25531 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
25532 (gst_xvimagesink_reset), (gst_xvimagesink_finalize):
25533 Move some cleanup stuff from the state change handler into a _reset()
25534 function that can be called from _finalize(). This ensures that things
25535 get freed even if (for some reason) the NULL->READY state transition
25536 fails in the parent class.
25537 Even if a parent state change fails, process our downward state change
25538 logic instead of bailing out early.
25539 Free the correct xcontext pointer in ximagesink's xcontext_clear.
25541 2007-03-08 12:53:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25543 ext/alsa/gstalsasink.c: Extra log line.
25544 Original commit message from CVS:
25545 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
25547 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
25548 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
25549 Use pango_font_description_set_family_static instead of
25550 pango_font_description_set_family to save a string copy (it was
25551 leaking due to the strdup anyway)
25552 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
25553 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
25554 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
25555 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
25556 Chain up in finalize.
25558 2007-03-07 18:50:10 +0000 Tim-Philipp Müller <tim@centricular.net>
25560 gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
25561 Original commit message from CVS:
25562 * gst-libs/gst/interfaces/mixertrack.c:
25563 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
25564 (gst_mixer_track_set_property):
25565 API: add "untranslated-label" property which should be set by
25566 implementations at construct time (#414645).
25567 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
25568 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
25569 Set "untranslated-label" when constructing mixer track objects.
25570 * tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
25571 Unit test to check the above.
25573 2007-03-07 17:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
25575 ext/ogg/gstoggdemux.c: Fix confusing debug message.
25576 Original commit message from CVS:
25577 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
25578 Fix confusing debug message.
25580 2007-03-07 17:12:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25582 gst-plugins-base.doap: update doap file with new version
25583 Original commit message from CVS:
25584 * gst-plugins-base.doap:
25585 update doap file with new version
25587 2007-03-07 17:05:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25589 * gst/tcp/gstmultifdsink.c:
25591 Original commit message from CVS:
25594 2007-03-07 16:56:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25596 configure.ac: Back to CVS
25597 Original commit message from CVS:
25601 === release 0.10.12 ===
25603 2007-03-07 16:46:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25609 * docs/plugins/gst-plugins-base-plugins.args:
25610 * docs/plugins/inspect/plugin-adder.xml:
25611 * docs/plugins/inspect/plugin-alsa.xml:
25612 * docs/plugins/inspect/plugin-audioconvert.xml:
25613 * docs/plugins/inspect/plugin-audiorate.xml:
25614 * docs/plugins/inspect/plugin-audioresample.xml:
25615 * docs/plugins/inspect/plugin-audiotestsrc.xml:
25616 * docs/plugins/inspect/plugin-cdparanoia.xml:
25617 * docs/plugins/inspect/plugin-decodebin.xml:
25618 * docs/plugins/inspect/plugin-decodebin2.xml:
25619 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
25620 * docs/plugins/inspect/plugin-gdp.xml:
25621 * docs/plugins/inspect/plugin-gnomevfs.xml:
25622 * docs/plugins/inspect/plugin-libvisual.xml:
25623 * docs/plugins/inspect/plugin-ogg.xml:
25624 * docs/plugins/inspect/plugin-pango.xml:
25625 * docs/plugins/inspect/plugin-playbin.xml:
25626 * docs/plugins/inspect/plugin-subparse.xml:
25627 * docs/plugins/inspect/plugin-tcp.xml:
25628 * docs/plugins/inspect/plugin-theora.xml:
25629 * docs/plugins/inspect/plugin-typefindfunctions.xml:
25630 * docs/plugins/inspect/plugin-video4linux.xml:
25631 * docs/plugins/inspect/plugin-videorate.xml:
25632 * docs/plugins/inspect/plugin-videoscale.xml:
25633 * docs/plugins/inspect/plugin-videotestsrc.xml:
25634 * docs/plugins/inspect/plugin-volume.xml:
25635 * docs/plugins/inspect/plugin-vorbis.xml:
25636 * docs/plugins/inspect/plugin-ximagesink.xml:
25637 * docs/plugins/inspect/plugin-xvimagesink.xml:
25638 * win32/common/config.h:
25640 Original commit message from CVS:
25643 2007-03-07 15:35:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25662 Original commit message from CVS:
25665 2007-03-06 12:31:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25667 configure.ac: Bump version to 0.10.11.4 pre-release
25668 Original commit message from CVS:
25670 Bump version to 0.10.11.4 pre-release
25672 2007-03-06 12:10:08 +0000 Wim Taymans <wim.taymans@gmail.com>
25674 gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
25675 Original commit message from CVS:
25676 * gst-libs/gst/audio/gstbaseaudiosink.c:
25677 (gst_base_audio_sink_async_play):
25678 Fix regression that made GStreamer skip the first samples of audio.
25681 2007-03-05 11:21:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25683 configure.ac: Bump version to 0.10.11.3 pre-release
25684 Original commit message from CVS:
25686 Bump version to 0.10.11.3 pre-release
25688 2007-03-05 09:35:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
25690 po/POTFILES.in: Update paths for the rename from utils to pbutils to fix the build.
25691 Original commit message from CVS:
25693 Update paths for the rename from utils to pbutils to fix the build.
25695 2007-03-05 09:27:55 +0000 Tim-Philipp Müller <tim@centricular.net>
25697 gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
25698 Original commit message from CVS:
25699 * gst-libs/gst/pbutils/Makefile.am:
25700 Change directory to install headers in from gst/utils to gst/pbutils
25703 2007-03-04 23:41:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25705 * tests/check/libs/.gitignore:
25707 Original commit message from CVS:
25710 2007-03-04 23:41:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25712 * win32/common/config.h:
25713 * win32/common/libgstutils.def:
25715 Original commit message from CVS:
25718 2007-03-04 23:39:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25720 rename utils to pbutils
25721 Original commit message from CVS:
25723 * docs/libs/gst-plugins-base-libs-docs.sgml:
25724 * docs/libs/gst-plugins-base-libs-sections.txt:
25725 * gst-libs/gst/Makefile.am:
25726 * gst-libs/gst/interfaces/mixer.c:
25727 * gst-libs/gst/pbutils/Makefile.am:
25728 * gst-libs/gst/pbutils/descriptions.c:
25729 (gst_pb_utils_get_source_description),
25730 (gst_pb_utils_get_sink_description),
25731 (gst_pb_utils_get_decoder_description),
25732 (gst_pb_utils_get_encoder_description),
25733 (gst_pb_utils_get_element_description),
25734 (gst_pb_utils_add_codec_description_to_tag_list),
25735 (gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
25736 * gst-libs/gst/pbutils/descriptions.h:
25737 * gst-libs/gst/pbutils/install-plugins.c:
25738 * gst-libs/gst/pbutils/install-plugins.h:
25739 * gst-libs/gst/pbutils/missing-plugins.c:
25740 (gst_missing_uri_source_message_new),
25741 (gst_missing_uri_sink_message_new),
25742 (gst_missing_element_message_new),
25743 (gst_missing_decoder_message_new),
25744 (gst_missing_encoder_message_new),
25745 (gst_missing_plugin_message_get_description):
25746 * gst-libs/gst/pbutils/missing-plugins.h:
25747 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
25748 * gst-libs/gst/pbutils/pbutils.h:
25749 * gst-libs/gst/utils/Makefile.am:
25750 * gst-libs/gst/utils/base-utils.c:
25751 * gst-libs/gst/utils/base-utils.h:
25752 * gst-libs/gst/utils/descriptions.c:
25753 * gst-libs/gst/utils/descriptions.h:
25754 * gst-libs/gst/utils/install-plugins.c:
25755 * gst-libs/gst/utils/install-plugins.h:
25756 * gst-libs/gst/utils/missing-plugins.c:
25757 * gst-libs/gst/utils/missing-plugins.h:
25758 * gst-plugins-base.spec.in:
25759 * gst/playback/Makefile.am:
25760 * gst/playback/gstdecodebin.c:
25761 * gst/playback/gstdecodebin2.c:
25762 * gst/playback/gstplaybasebin.c: (setup_subtitle),
25763 (gen_source_element):
25764 * gst/playback/gstplaybin.c: (plugin_init):
25765 * tests/check/Makefile.am:
25766 * tests/check/libs/pbutils.c: (GST_START_TEST),
25767 (test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
25768 * tests/check/libs/utils.c:
25769 rename utils to pbutils
25771 2007-03-03 10:23:03 +0000 David Schleef <ds@schleef.org>
25773 gst-libs/gst/app/Makefile.am: Install the headers.
25774 Original commit message from CVS:
25775 * gst-libs/gst/app/Makefile.am:
25776 Install the headers.
25778 2007-03-03 10:10:30 +0000 David Schleef <ds@schleef.org>
25780 gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.
25781 Original commit message from CVS:
25782 * gst-libs/gst/app/Makefile.am:
25783 * gst-libs/gst/app/gstappbuffer.c:
25784 * gst-libs/gst/app/gstappbuffer.h:
25785 * gst-libs/gst/app/gstappsrc.c:
25786 Add GstAppBuffer that includes a callback and closure for
25787 proper handling of data chunks.
25789 2007-03-03 09:06:06 +0000 David Schleef <ds@schleef.org>
25791 gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.
25792 Original commit message from CVS:
25793 * gst-libs/gst/app/gstappsrc.c:
25794 * gst-libs/gst/app/gstappsrc.h:
25795 Hacking to address issues in 413418.
25797 2007-03-03 08:16:57 +0000 David Schleef <ds@schleef.org>
25799 Move the app library to gst-libs/gst/app (duh!)
25800 Original commit message from CVS:
25804 * gst-libs/gst/Makefile.am:
25805 * gst-libs/gst/app/Makefile.am:
25806 * gst-libs/gst/app/gstapp.c:
25807 * gst-libs/gst/app/gstappsrc.c:
25808 * gst-libs/gst/app/gstappsrc.h:
25809 * gst/app/Makefile.am:
25810 * gst/app/gstapp.c:
25811 * gst/app/gstappsrc.c:
25812 * gst/app/gstappsrc.h:
25813 Move the app library to gst-libs/gst/app (duh!)
25815 2007-03-02 12:59:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25817 Add documentation for decodebin2 that indicates that the API is still unstable.
25818 Original commit message from CVS:
25819 * docs/plugins/Makefile.am:
25820 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
25821 * docs/plugins/gst-plugins-base-plugins-sections.txt:
25822 * docs/plugins/inspect/plugin-decodebin2.xml:
25823 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
25824 Add documentation for decodebin2 that indicates that the API
25827 2007-03-01 18:50:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25829 configure.ac: Update to 0.10.11.2 (0.10.12 pre-release)
25830 Original commit message from CVS:
25832 Update to 0.10.11.2 (0.10.12 pre-release)
25834 2007-03-01 17:29:55 +0000 Wim Taymans <wim.taymans@gmail.com>
25836 gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
25837 Original commit message from CVS:
25838 * gst-libs/gst/audio/gstbaseaudiosink.c:
25839 (gst_base_audio_sink_async_play):
25840 base time is irrelevant here.
25842 2007-03-01 17:01:43 +0000 Wim Taymans <wim.taymans@gmail.com>
25844 gst-libs/gst/audio/: Improve debugging.
25845 Original commit message from CVS:
25846 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
25847 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
25849 * gst-libs/gst/audio/gstbaseaudiosink.c:
25850 (gst_base_audio_sink_query), (gst_base_audio_sink_event),
25851 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
25852 Improve latency and clock slaving calculations.
25853 Improve slave clock calibration.
25854 * gst-libs/gst/audio/gstringbuffer.c:
25855 (gst_ring_buffer_commit_full):
25856 When we are asked to render N sample to 0 bytes, return N.
25858 2007-03-01 16:48:45 +0000 Wim Taymans <wim.taymans@gmail.com>
25860 ext/alsa/gstalsasink.*: Remove unused dispose function.
25861 Original commit message from CVS:
25862 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
25863 (gst_alsasink_write), (gst_alsasink_reset):
25864 * ext/alsa/gstalsasink.h:
25865 Remove unused dispose function.
25866 Rename lock to not interfere with alsasrc lock.
25867 * ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
25868 (gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
25869 (gst_alsasrc_read), (gst_alsasrc_reset):
25870 * ext/alsa/gstalsasrc.h:
25871 Implement finalize function.
25872 Use lock to protect alsa access.
25874 Fine tune sw params.
25876 2007-03-01 10:20:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25881 Original commit message from CVS:
25884 2007-02-28 19:27:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25886 configure.ac: Convert to new AG_GST style.
25887 Original commit message from CVS:
25889 Convert to new AG_GST style.
25891 2007-02-28 15:17:20 +0000 Ed Catmur <ed@catmur.co.uk>
25893 gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
25894 Original commit message from CVS:
25895 Patch by: Ed Catmur <ed at catmur dot co dot uk>
25896 * gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
25897 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
25898 Fix race condition when rapidly switching visualisations in playbin.
25901 2007-02-28 15:11:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25903 tests/check/Makefile.am: Include local stuff before system installed things in LDFLAGS and
25904 Original commit message from CVS:
25905 * tests/check/Makefile.am:
25906 Include local stuff before system installed things in LDFLAGS and
25909 2007-02-28 15:10:06 +0000 Wim Taymans <wim.taymans@gmail.com>
25911 ext/ogg/gstoggdemux.c: Improve debugging.
25912 Original commit message from CVS:
25913 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
25916 2007-02-28 15:05:03 +0000 Wim Taymans <wim.taymans@gmail.com>
25918 sys/v4l/: Fix duration and timestamping, taking latency into account.
25919 Original commit message from CVS:
25920 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
25921 (gst_v4lsrc_fixate), (gst_v4lsrc_query):
25922 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
25923 Fix duration and timestamping, taking latency into account.
25924 Implement latency query.
25926 2007-02-28 15:02:25 +0000 Wim Taymans <wim.taymans@gmail.com>
25928 gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
25929 Original commit message from CVS:
25930 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
25931 (gst_audio_clock_new):
25933 * gst-libs/gst/audio/gstbaseaudiosink.c:
25934 (gst_base_audio_sink_init), (gst_base_audio_sink_query):
25935 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
25936 (gst_base_audio_src_query), (gst_base_audio_src_get_offset),
25937 (gst_base_audio_src_create):
25938 Improve latency query code.
25939 Use proper clock names.
25941 2007-02-28 12:57:46 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25943 * tests/check/generic/states.c:
25945 Original commit message from CVS:
25948 2007-02-28 12:44:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25950 tests/check/generic/states.c: Copy the states.c test from core again
25951 Original commit message from CVS:
25952 * tests/check/generic/states.c: (GST_START_TEST):
25953 Copy the states.c test from core again
25954 * tests/check/Makefile.am:
25955 ignore cdio and cdparanoiasrc
25957 2007-02-28 12:08:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25959 gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
25960 Original commit message from CVS:
25961 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
25962 (double_hq), (audio_convert_get_func_index), (check_default),
25963 (audio_convert_prepare_context), (audio_convert_convert):
25964 Also make valgrind happy and avoid copying data in some cases.
25966 2007-02-28 11:58:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25968 * tests/check/generic/states.c:
25970 Original commit message from CVS:
25973 2007-02-28 11:47:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25975 Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
25976 Original commit message from CVS:
25977 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
25978 (double_hq), (audio_convert_get_func_index),
25979 (audio_convert_prepare_context), (audio_convert_convert):
25980 * gst/audioconvert/gstaudioconvert.c:
25981 (gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
25982 (gst_audio_convert_transform_caps):
25983 * tests/check/elements/audioconvert.c: (GST_START_TEST),
25984 (audioconvert_suite):
25985 Don't run inplace if that overwrites source data as we go. Add more
25986 tests. Fixes #339837 even more.
25988 2007-02-27 18:45:37 +0000 Julien Moutte <julien@moutte.net>
25990 tests/examples/seek/seek.c: Fix various seeking bugs (Slider was not updating when doing a non flushing seek, Reverse...
25991 Original commit message from CVS:
25992 2007-02-27 Julien MOUTTE <julien@moutte.net>
25993 * tests/examples/seek/seek.c: (do_seek), (set_update_scale),
25994 (msg_segment_done): Fix various seeking bugs (Slider was not
25995 updating when doing a non flushing seek, Reverse playback
25996 on segment seek was wrong).
25998 2007-02-26 21:01:03 +0000 David Schleef <ds@schleef.org>
26000 Add a new plugin/library to make it easy for apps to shove data into a pipeline.
26001 Original commit message from CVS:
26003 * gst/app/Makefile.am:
26004 * gst/app/gstapp.c:
26005 * gst/app/gstappsrc.c:
26006 * gst/app/gstappsrc.h:
26007 Add a new plugin/library to make it easy for apps to shove
26008 data into a pipeline.
26010 2007-02-26 11:48:49 +0000 Wim Taymans <wim.taymans@gmail.com>
26012 tests/examples/seek/seek.c: When we stop scrubbing, don't leave the pipeline PLAYING when we requested a PAUSED state.
26013 Original commit message from CVS:
26014 * tests/examples/seek/seek.c: (stop_seek):
26015 When we stop scrubbing, don't leave the pipeline PLAYING when we
26016 requested a PAUSED state.
26018 2007-02-25 23:51:03 +0000 René Stadler <mail@renestadler.de>
26020 gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
26021 Original commit message from CVS:
26022 Patch by: René Stadler <mail at renestadler de>
26023 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
26024 Parse date strings in vorbis comments that have an invalid (zero)
26025 month or day (#410396).
26026 * tests/check/libs/tag.c: (GST_START_TEST):
26027 Test case for the above.
26029 2007-02-24 20:12:49 +0000 Loïc Minier <lool+gnome@via.ecp.fr>
26031 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
26032 Original commit message from CVS:
26033 Patch by: Loïc Minier <lool+gnome at via ecp fr>
26035 * ext/alsa/Makefile.am:
26036 * gst/audiotestsrc/Makefile.am:
26037 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
26039 2007-02-23 18:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
26041 gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
26042 Original commit message from CVS:
26043 * gst/playback/gstplaybin.c:
26044 Improve docs: point out that the application needs to assist playbin
26047 2007-02-23 13:10:50 +0000 Tim-Philipp Müller <tim@centricular.net>
26049 Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
26050 Original commit message from CVS:
26051 * gst-libs/gst/utils/install-plugins.c:
26052 * gst-libs/gst/utils/missing-plugins.c:
26053 * tests/check/libs/utils.c: (missing_msg_check_getters):
26054 Change GStreamer marker prefix in detail string from 'gstreamer.net'
26055 to just 'gstreamer'. Document the caps string component of the
26056 decoder/encoder detail a bit better, since not everyone will be
26057 familiar with the GStreamer media type/caps system (but they better
26058 enjoy nested itemized lists).
26060 2007-02-22 12:57:47 +0000 Tim-Philipp Müller <tim@centricular.net>
26062 gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
26063 Original commit message from CVS:
26064 * gst-libs/gst/netbuffer/gstnetbuffer.c:
26065 (notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
26066 Fix copying of GstNetBuffer (would crash before, or at least lead to
26067 invalid memory access, #410772), for now by copying the GstBuffer copy
26068 code from the core over here so we can copy the GstBuffer fields on a
26069 provided buffer instance (of type GstNetBuffer in this case). Would be
26070 better to fix this with some support by the core though (and in the long
26071 run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
26072 * tests/check/Makefile.am:
26073 Enable unit test for GstNetBuffer.
26075 2007-02-22 11:04:10 +0000 Andy Wingo <wingo@pobox.com>
26078 * gst-libs/gst/audio/gstbaseaudiosink.c:
26079 gst-libs/gst/audio/gstbaseaudiosink.c
26080 Original commit message from CVS:
26081 2007-02-22 Andy Wingo <wingo@pobox.com>
26082 * gst-libs/gst/audio/gstbaseaudiosink.c
26083 (gst_base_audio_sink_init): Disable pull-mode activation until we
26084 figure out how to make audio sinks go to PLAYING.
26086 2007-02-22 09:04:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26088 Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
26089 Original commit message from CVS:
26090 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
26091 (double_hq), (audio_convert_get_func_index),
26092 (audio_convert_prepare_context), (audio_convert_convert):
26093 * gst/audioconvert/audioconvert.h:
26094 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
26095 (gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
26096 * gst/audioconvert/gstchannelmix.h:
26097 * tests/check/elements/audioconvert.c: (GST_START_TEST):
26098 Add float as an intermediate format, as well as float mixing. Enable
26099 test that was failing before. Fixes #339837
26101 2007-02-21 16:12:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26103 tests/examples/seek/seek.c: Undo the previous commit: -1 as a stop time implies that the stop time is the end of file...
26104 Original commit message from CVS:
26105 * tests/examples/seek/seek.c: (do_seek):
26106 Undo the previous commit: -1 as a stop time implies that the stop
26107 time is the end of file, clearing any previously configured segment.
26109 2007-02-21 15:36:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26111 tests/examples/seek/seek.c: Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
26112 Original commit message from CVS:
26113 * tests/examples/seek/seek.c: (do_seek):
26114 Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
26116 2007-02-21 13:55:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26118 gst/volume/gstvolume.c: Unbreak volume, value remains gint.
26119 Original commit message from CVS:
26120 * gst/volume/gstvolume.c: (volume_process_int16),
26121 (volume_process_int16_clamp), (volume_set_caps):
26122 Unbreak volume, value remains gint.
26124 2007-02-21 13:08:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26126 gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
26127 Original commit message from CVS:
26128 * gst/volume/gstvolume.c: (volume_choose_func),
26129 (volume_update_real_volume), (gst_volume_set_volume),
26130 (gst_volume_init), (volume_process_double), (volume_process_float),
26131 (volume_process_int16), (volume_process_int16_clamp),
26132 (volume_set_caps), (volume_transform_ip), (volume_update_volume):
26133 * gst/volume/gstvolume.h:
26134 Extend float audio support (double) and some int->uint cleanups.
26136 2007-02-20 15:44:32 +0000 Edward Hervey <bilboed@bilboed.com>
26138 gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
26139 Original commit message from CVS:
26140 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
26141 (multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
26142 (sort_end_pads), (gst_decode_group_expose),
26143 (gst_decode_group_hide):
26144 Don't free groups from the streaming threads. Just put them aside and
26145 free them in dispose.
26147 2007-02-20 11:20:52 +0000 Edward Hervey <bilboed@bilboed.com>
26149 gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
26150 Original commit message from CVS:
26151 * gst/playback/gstdecodebin2.c: (connect_element),
26152 (pad_added_group_cb), (gst_decode_group_check_if_blocked),
26153 (sort_end_pads), (gst_decode_group_expose):
26154 Handle dynamic pads within groups.
26155 Sort pads before exposing them in order to make playbin happy.
26156 There still is a race with the multiqueue filling up. This should be
26160 2007-02-18 21:02:36 +0000 Tim-Philipp Müller <tim@centricular.net>
26162 gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
26163 Original commit message from CVS:
26164 * gst-libs/gst/utils/base-utils.c:
26165 * gst-libs/gst/utils/descriptions.c:
26166 * gst-libs/gst/utils/install-plugins.c:
26167 * gst-libs/gst/utils/missing-plugins.c:
26168 Some more docs (and descriptions for two subtitle formats).
26170 2007-02-16 10:19:45 +0000 Tim-Philipp Müller <tim@centricular.net>
26172 gst-libs/gst/audio/audio.c: Fix documentation.
26173 Original commit message from CVS:
26174 * gst-libs/gst/audio/audio.c:
26177 2007-02-16 10:15:46 +0000 Yves Lefebvre <ivanohe@abacom.com>
26179 gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
26180 Original commit message from CVS:
26181 Patch by: Yves Lefebvre <ivanohe abacom com>
26182 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
26183 Don't leak caps. Fixes #408278.
26185 2007-02-15 15:17:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26187 More docs coverage and some ChangeLog surgery (add missing names)
26188 Original commit message from CVS:
26189 * ext/cdparanoia/gstcdparanoiasrc.h:
26190 * ext/ogg/gstoggdemux.h:
26191 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
26192 (gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
26193 (gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
26194 * gst-libs/gst/audio/audio.h:
26195 * gst-libs/gst/audio/gstaudiofilter.h:
26196 * gst-libs/gst/interfaces/videoorientation.h:
26197 * gst/adder/gstadder.h:
26198 More docs coverage and some ChangeLog surgery (add missing names)
26200 2007-02-15 12:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
26202 sys/: Small constifications.
26203 Original commit message from CVS:
26204 * sys/ximage/ximagesink.c:
26205 (gst_ximagesink_calculate_pixel_aspect_ratio):
26206 * sys/xvimage/xvimagesink.c:
26207 (gst_xvimagesink_calculate_pixel_aspect_ratio):
26208 Small constifications.
26210 2007-02-15 12:06:25 +0000 Wim Taymans <wim.taymans@gmail.com>
26212 gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
26213 Original commit message from CVS:
26214 * gst-libs/gst/audio/gstbaseaudiosink.c:
26215 (gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
26216 (gst_base_audio_sink_render), (gst_base_audio_sink_callback),
26217 (gst_base_audio_sink_async_play),
26218 (gst_base_audio_sink_change_state):
26219 Answer latency query.
26220 Use configured latency when syncing.
26222 * gst-libs/gst/audio/gstbaseaudiosrc.c:
26223 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
26224 (gst_base_audio_src_query), (gst_base_audio_src_change_state):
26225 Fix possible memleak.
26226 Implement latency query.
26229 2007-02-15 11:59:41 +0000 Wim Taymans <wim.taymans@gmail.com>
26231 ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
26232 Original commit message from CVS:
26233 * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
26234 Ignore errors in reset, these are not fatal. They also grab the element
26235 lock which is already taking when this function is called. Fixes
26238 2007-02-13 13:50:56 +0000 Christian Schaller <uraeus@gnome.org>
26240 * gst-plugins-base.spec.in:
26241 add header file for easy codec install
26242 Original commit message from CVS:
26243 add header file for easy codec install
26245 2007-02-13 10:24:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26247 configure.ac: Remove 'tests/examples/xerror/Makefile' from output files again.
26248 Original commit message from CVS:
26250 Remove 'tests/examples/xerror/Makefile' from output files again.
26252 2007-02-13 09:12:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26254 Also crossref against gst-plugins-base-libs.
26255 Original commit message from CVS:
26257 * docs/plugins/Makefile.am:
26258 Also crossref against gst-plugins-base-libs.
26260 2007-02-12 20:42:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26262 Add crossreferences to glib/gobject/gstream docs.
26263 Original commit message from CVS:
26265 * docs/libs/Makefile.am:
26266 * docs/plugins/Makefile.am:
26267 Add crossreferences to glib/gobject/gstream docs.
26268 * gst-libs/gst/audio/audio.h:
26270 * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
26271 Add own debug category.
26273 2007-02-12 11:01:04 +0000 René Stadler <mail@renestadler.de>
26275 gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
26276 Original commit message from CVS:
26277 Patch by: René Stadler <mail at renestadler de>
26278 * gst-libs/gst/tag/gstvorbistag.c:
26279 Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
26282 2007-02-12 10:33:40 +0000 Tim-Philipp Müller <tim@centricular.net>
26284 gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
26285 Original commit message from CVS:
26286 * gst/playback/gstplaybasebin.c: (setup_source):
26287 When we have external subtitles and wait for the subtitle decodebin
26288 to get up and running, we set up a (sync) bus handler for the
26289 subtitle decodebin, so we can stop waiting when it posts an error
26290 message. However, we should do that before we set the subtitle
26291 decodebin's state to playing, otherwise things are racy and we might
26292 miss error messages posted before we had a chance to set up the bus.
26293 This should finally fix totem hanging on .txt pseudo-subtitle files.
26295 2007-02-10 19:27:48 +0000 Sébastien Moutte <sebastien@moutte.net>
26297 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
26298 Original commit message from CVS:
26299 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
26300 Use gst_gdouble_to_guint64 for conversions.
26301 * win32/common/config.h.in:
26302 Add a define for GST_INSTALL_PLUGINS_HELPER
26303 * win32/common/libgstaudio.def:
26304 * win32/common/libgstcdda.def:
26305 * win32/common/libgstnetbuffer.def:
26306 * win32/common/libgstrtp.def:
26307 * win32/common/libgutils.def:
26308 Add new exported functions.
26309 * win32/vs6/gst_plugins_base.dsw:
26310 * win32/vs6/libgstdecodebin.dsp:
26311 * win32/vs6/libgstnetbuffer.dsp:
26312 * win32/vs6/libgstplaybin.dsp:
26313 * win32/vs6/libgstrtp.dsp:
26314 * win32/vs6/libgstvorbis.dsp:
26315 * win32/vs6/libgstcdda.dsp:
26316 * win32/vs6/libgstgdp.dsp:
26317 * win32/vs6/libgstutils.dsp:
26318 Update and add new project files.
26320 2007-02-10 18:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
26322 gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
26323 Original commit message from CVS:
26324 * gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
26325 (subrip_remove_unhandled_tags), (parse_subrip):
26326 For SubRip (.srt) subtitles, ignore all markup tags we don't
26327 handle (like font tags, for example).
26328 * tests/check/elements/subparse.c:
26331 2007-02-09 13:28:01 +0000 Tim-Philipp Müller <tim@centricular.net>
26335 Original commit message from CVS:
26338 2007-02-09 13:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
26340 gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
26341 Original commit message from CVS:
26342 * gst/playback/gstdecodebin.c: (add_fakesink),
26343 (gst_decode_bin_change_state):
26344 * gst/playback/gstdecodebin2.c: (add_fakesink),
26345 (gst_decode_bin_change_state):
26346 Don't error out if there is no fakesink in the READY to NULL state
26347 change, since when decodebin is re-used, we're only adding the
26348 fakesink element in READY to PAUSED.
26349 * tests/check/elements/decodebin.c:
26350 (new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
26352 Minimal unit test to make sure we can use the same decodebin
26353 instance twice (at least with audiotestsrc input).
26355 2007-02-09 09:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
26357 ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
26358 Original commit message from CVS:
26359 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
26360 Try to get devic-name from device string first, and from handle only
26361 as fallback (seems to yield better results and is more robust
26362 against buggy probing code on the application side).
26364 2007-02-08 15:43:26 +0000 Julien Puydt <julien.puydt@laposte.net>
26366 ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
26367 Original commit message from CVS:
26368 Based on patch by: Julien Puydt <julien.puydt at laposte net>
26369 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
26370 (gst_alsa_find_device_name):
26371 * ext/alsa/gstalsa.h:
26372 * ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
26373 * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
26374 Improve device-name detection a bit, especially in the case where
26375 the device is not actually open (#405020, #405024). Move common code
26376 into gstalsa.c instead of duplicating it.
26378 2007-02-07 13:05:01 +0000 Tim-Philipp Müller <tim@centricular.net>
26380 gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
26381 Original commit message from CVS:
26382 * gst/audioconvert/gstaudioconvert.c:
26383 Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
26385 2007-02-06 17:47:32 +0000 Julien Moutte <julien@moutte.net>
26387 sys/xvimage/xvimagesink.*: Implement PropertyProbe Interface for XVAdaptors so that one can choose the adaptor to use...
26388 Original commit message from CVS:
26389 2007-02-06 Julien MOUTTE <julien@moutte.net>
26390 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
26391 (gst_xvimagesink_get_xv_support),
26392 (gst_xvimagesink_xcontext_clear),
26393 (gst_xvimagesink_interface_supported),
26394 (gst_xvimagesink_probe_get_properties),
26395 (gst_xvimagesink_probe_probe_property),
26396 (gst_xvimagesink_probe_needs_probe),
26397 (gst_xvimagesink_probe_get_values),
26398 (gst_xvimagesink_property_probe_interface_init),
26399 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
26400 (gst_xvimagesink_init), (gst_xvimagesink_class_init),
26401 (gst_xvimagesink_get_type):
26402 * sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
26403 for XVAdaptors so that one can choose the adaptor to use with
26404 gstreamer-properties.
26406 2007-02-06 14:00:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26408 gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
26409 Original commit message from CVS:
26410 * gst/audioconvert/gstaudioconvert.c:
26411 Also mention that a conversion from double to float is suboptimal still.
26413 2007-02-06 09:42:05 +0000 Tim-Philipp Müller <tim@centricular.net>
26415 gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
26416 Original commit message from CVS:
26417 * gst-libs/gst/audio/gstaudiofilter.c:
26418 (gst_audio_filter_class_init), (gst_audio_filter_change_state):
26419 Clear our formats structure and free the caps contained in it when
26422 2007-02-05 18:39:51 +0000 Andy Wingo <wingo@pobox.com>
26425 * gst-libs/gst/audio/gstbaseaudiosink.c:
26426 gst-libs/gst/audio/gstbaseaudiosink.c
26427 Original commit message from CVS:
26428 2007-02-05 Andy Wingo <wingo@pobox.com>
26429 * gst-libs/gst/audio/gstbaseaudiosink.c
26430 (gst_base_audio_sink_callback): Update basesink->offset so that we
26431 pull monotonically increasing offsets instead of, um, seeking back
26432 to 0 each time. Fixes alsasrc ! alsasink!
26434 2007-02-05 11:44:52 +0000 Tim-Philipp Müller <tim@centricular.net>
26436 gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
26437 Original commit message from CVS:
26438 * gst/videoscale/gstvideoscale.c:
26439 A width and height of 1 makes us crash, so increase minimum size to
26440 2x2 pixels until someone feels like fixing this (#404512).
26442 2007-02-04 16:23:37 +0000 Tim-Philipp Müller <tim@centricular.net>
26444 tests/check/pipelines/oggmux.c: Add small test to make sure request pads are cleaned up properly even if oggmux never...
26445 Original commit message from CVS:
26446 * tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
26447 Add small test to make sure request pads are cleaned up properly
26448 even if oggmux never changes state out of NULL.
26450 2007-02-04 14:11:51 +0000 Tim-Philipp Müller <tim@centricular.net>
26452 tests/check/libs/utils.c: Fix unit test. Turns out things work much better when you
26453 Original commit message from CVS:
26454 * tests/check/libs/utils.c: (GST_START_TEST):
26455 Fix unit test. Turns out things work much better when you
26456 NULL-terminate string arrays. Should make p5 build bot happy again.
26458 2007-02-03 23:28:45 +0000 Tim-Philipp Müller <tim@centricular.net>
26460 gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
26461 Original commit message from CVS:
26462 * gst-libs/gst/audio/Makefile.am:
26463 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
26464 (gst_audio_filter_template_base_init),
26465 (gst_audio_filter_template_class_init),
26466 (gst_audio_filter_template_init),
26467 (gst_audio_filter_template_set_property),
26468 (gst_audio_filter_template_get_property),
26469 (gst_audio_filter_template_setup),
26470 (gst_audio_filter_template_filter),
26471 (gst_audio_filter_template_filter_inplace), (plugin_init):
26472 Oops, forgot to commit fixed-up example.
26474 2007-02-03 20:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
26476 Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
26477 Original commit message from CVS:
26478 * docs/libs/gst-plugins-base-libs-sections.txt:
26479 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
26480 (gst_audio_filter_class_init), (gst_audio_filter_init),
26481 (gst_audio_filter_set_caps),
26482 (gst_audio_filter_class_add_pad_templates):
26483 * gst-libs/gst/audio/gstaudiofilter.h:
26484 Port GstAudioFilter to 0.10. This change technically breaks
26485 API and ABI (and thus also every library developer's heart),
26486 but seems justifiable on the grounds that the base class was
26487 completely unusable before (ie. would crash immediately when
26488 actually used). Fixes #403963 (and eventually also #403572).
26489 Also document all of this a bit.
26491 2007-02-03 14:26:54 +0000 Tim-Philipp Müller <tim@centricular.net>
26493 Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
26494 Original commit message from CVS:
26495 * gst-libs/gst/utils/install-plugins.c:
26496 (gst_install_plugins_spawn_child):
26497 * tests/check/libs/utils.c:
26498 (test_base_utils_install_plugins_do_callout):
26499 Lowering log level to see why things fail on the p5 build bot;
26500 fix some typos in unit test messages.
26502 2007-02-03 13:59:27 +0000 Tim-Philipp Müller <tim@centricular.net>
26504 tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
26505 Original commit message from CVS:
26506 * tests/check/libs/utils.c:
26507 (test_base_utils_install_plugins_do_callout):
26508 Don't hard-code temp directory for test helper; use GLib functions
26509 to write out file and do error checking etc.
26511 2007-02-02 20:42:08 +0000 Tim-Philipp Müller <tim@centricular.net>
26513 gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
26514 Original commit message from CVS:
26515 * gst-libs/gst/utils/Makefile.am:
26516 * gst-libs/gst/utils/base-utils.h:
26517 * gst-libs/gst/utils/install-plugins.c:
26518 (gst_install_plugins_context_set_xid),
26519 (gst_install_plugins_context_new),
26520 (gst_install_plugins_context_free),
26521 (gst_install_plugins_get_helper),
26522 (gst_install_plugins_spawn_child),
26523 (gst_install_plugins_return_from_status),
26524 (gst_install_plugins_installer_exited),
26525 (gst_install_plugins_async), (gst_install_plugins_sync),
26526 (gst_install_plugins_return_get_name),
26527 (gst_install_plugins_installation_in_progress):
26528 * gst-libs/gst/utils/install-plugins.h:
26529 API: add API for applications to initiate installation of missing
26530 plugins, ie. gst_install_plugins_async() primarily.
26531 Based on libgimme-codec by Ryan Lortie.
26533 Add --with-install-plugins-helper configure option so distros can specify
26534 the path of the helper script or program to call when plugin installation
26535 is requested (distros: please do any argument munging in this helper
26536 script instead of patching GStreamer to pass arguments differently
26537 to another program directly).
26538 * docs/libs/gst-plugins-base-libs-docs.sgml:
26539 * docs/libs/gst-plugins-base-libs-sections.txt:
26540 Build and document new API.
26541 * tests/check/libs/utils.c: (result_cb),
26542 (test_base_utils_install_plugins_do_callout), (GST_START_TEST),
26543 (libgstbaseutils_suite):
26544 Some simple checks for the new API.
26546 2007-02-02 14:44:29 +0000 Tim-Philipp Müller <tim@centricular.net>
26548 tests/check/elements/audioconvert.c: Add small test for 32bit float <=> 64bit float conversion (works only one way so...
26549 Original commit message from CVS:
26550 * tests/check/elements/audioconvert.c: (test_float_conversion):
26551 Add small test for 32bit float <=> 64bit float conversion (works
26552 only one way so far, 32=>64 produces structured noise).
26554 2007-02-02 11:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
26556 gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
26557 Original commit message from CVS:
26558 * gst/audioconvert/gstaudioconvert.c:
26559 (set_structure_widths_32_and_64), (make_lossless_changes):
26560 We don't support floats with a width of 40, 48 or 56 bits.
26562 2007-02-02 09:48:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26564 gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
26565 Original commit message from CVS:
26566 * gst/audioconvert/audioconvert.c: (float), (double),
26567 (audio_convert_get_func_index):
26568 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
26569 (make_lossless_changes):
26570 Support for 64-bit float audio in audioconvert (#339837)
26572 2007-02-01 18:50:08 +0000 Holger Wansing <linux@wansing-online.de>
26574 po/: Add German translation (#352069).
26575 Original commit message from CVS:
26576 Patch by: Holger Wansing <linux wansing-online de>
26579 Add German translation (#352069).
26581 2007-02-01 17:52:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
26583 ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
26584 Original commit message from CVS:
26585 reviewed by: Wim Taymans <wim@fluendo.com>
26586 * ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
26587 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
26588 Use newly added GstCollectPads API to free the allocated resources in
26589 the GstOggPad structures (#402393).
26591 2007-01-31 15:58:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26593 gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
26594 Original commit message from CVS:
26595 * gst/playback/gstplaybin.c: (gen_vis_element):
26596 Add audioresample+audioconvert in front of the visualisation
26597 element, so that elements like libvisual 0.4 that don't support all
26598 samplerates can work.
26601 2007-01-30 19:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
26603 gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
26604 Original commit message from CVS:
26605 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
26606 (gst_play_base_bin_get_streaminfo_value_array):
26607 Take some locks and make a copy of the streaminfo value array we
26608 maintain while holding the lock, so that the application can
26609 retrieve the stream-info as a value array in a thread-safe way.
26611 2007-01-30 11:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
26613 gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
26614 Original commit message from CVS:
26615 * gst/audioconvert/gstaudioconvert.c:
26616 Don't fail on 0 sized buffers. Fixes #396835.
26618 2007-01-29 21:13:07 +0000 David Schleef <ds@schleef.org>
26620 gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
26621 Original commit message from CVS:
26622 * gst/typefind/gsttypefindfunctions.c:
26623 Detect BBCD as video/x-dirac, so we can play raw dirac
26626 2007-01-29 18:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
26628 ext/theora/theoraenc.c: Check return value of theora_encode_header(), or we might try to allocate a random number of ...
26629 Original commit message from CVS:
26630 * ext/theora/theoraenc.c: (theora_enc_chain):
26631 Check return value of theora_encode_header(), or we might try to
26632 allocate a random number of bytes. theora_encode_header() can fail
26633 if libtheora has been compiled with encoding support disabled.
26636 2007-01-29 10:53:06 +0000 Wim Taymans <wim.taymans@gmail.com>
26638 tests/check/gst/.cvsignore: Do as buildbot says.
26639 Original commit message from CVS:
26640 * tests/check/gst/.cvsignore:
26641 Do as buildbot says.
26643 2007-01-29 10:25:11 +0000 Wim Taymans <wim.taymans@gmail.com>
26645 ext/libvisual/visual.c: Fix strides in libvisual. Gst uses X strides.
26646 Original commit message from CVS:
26647 * ext/libvisual/visual.c: (gst_visual_src_setcaps):
26648 Fix strides in libvisual. Gst uses X strides.
26649 Inspired by: <ed at catmur dot co dot uk> and
26650 <tim at centricular dot net>
26653 2007-01-27 13:32:24 +0000 Wim Taymans <wim.taymans@gmail.com>
26655 ext/ogg/gstoggdemux.*: Properly propagate streaming errors when we are scanning the file for chains so that we don't ...
26656 Original commit message from CVS:
26657 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
26658 (gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
26659 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
26660 (gst_ogg_demux_perform_seek),
26661 (gst_ogg_demux_bisect_forward_serialno),
26662 (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
26663 (gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
26664 (gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
26665 (gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
26666 * ext/ogg/gstoggdemux.h:
26667 Properly propagate streaming errors when we are scanning the file for
26668 chains so that we don't crash when shut down. Might fix some crashers
26669 when quickly switching oggs in RB such as #332503 and #378436.
26671 2007-01-26 12:44:46 +0000 Tim-Philipp Müller <tim@centricular.net>
26673 ext/gnomevfs/gstgnomevfssrc.c: Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND error code as well.
26674 Original commit message from CVS:
26675 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
26676 Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
26677 error code as well.
26679 2007-01-25 16:02:41 +0000 Wim Taymans <wim.taymans@gmail.com>
26681 gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
26682 Original commit message from CVS:
26683 * gst/playback/gstplaybasebin.c: (remove_source):
26684 Don't try to disconnect a signal from a finalized object.
26686 2007-01-25 14:29:21 +0000 Tim-Philipp Müller <tim@centricular.net>
26688 gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
26689 Original commit message from CVS:
26690 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
26691 Cast lock macro parameters to make sure we're actually accessing the
26692 lock member at the right class level. Free list itself in _dispose()
26693 as well and NULL it in case dispose gets called multiple times.
26695 2007-01-25 14:02:37 +0000 Edward Hervey <bilboed@bilboed.com>
26697 gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
26698 Original commit message from CVS:
26699 * gst/playback/gstdecodebin2.c:
26700 (gst_decode_bin_dispose),(gst_decode_bin_finalize):
26701 Free GstDecodeGroups no longer used.
26702 (gst_decode_group_expose):
26703 Don't unlock too many times !
26704 (deactivate_free_recursive):
26705 Free iterator once we're done with it.
26706 Fix for recursively deactivating elements (stop at ghostpads).
26708 2007-01-25 12:24:18 +0000 Tim-Philipp Müller <tim@centricular.net>
26710 gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
26711 Original commit message from CVS:
26712 * gst/playback/gstplaybin.c: (handoff):
26713 Fix up caps on the frame buffer before we save it and potentially
26714 make it accessible to other threads via g_object_get; also use
26715 gst_buffer_replace() instead of gst_mini_object_replace().
26717 2007-01-25 12:06:59 +0000 Tim-Philipp Müller <tim@centricular.net>
26719 gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
26720 Original commit message from CVS:
26721 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
26722 Make getting the current frame thread-safe.
26724 2007-01-25 11:48:10 +0000 Edward Hervey <bilboed@bilboed.com>
26726 gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
26727 Original commit message from CVS:
26728 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
26729 (gst_decode_group_new), (gst_decode_group_free):
26730 Set queues to bigger sizes to cope with HD contents.
26731 Fix some mutex freeing and add comment about MT safe methods.
26733 2007-01-24 12:51:20 +0000 Tim-Philipp Müller <tim@centricular.net>
26735 ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
26736 Original commit message from CVS:
26737 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
26738 (gst_text_overlay_text_event):
26739 Don't unnecessarily ref (and then leak) upstream events if the text
26740 pad is not linked. Fixes #399948.
26741 * tests/check/gst-plugins-base.supp:
26742 Add suppression for pango on edgy/x86 for textoverlay test.
26744 2007-01-24 12:10:56 +0000 Wim Taymans <wim.taymans@gmail.com>
26746 gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
26747 Original commit message from CVS:
26748 * gst-libs/gst/rtp/gstrtpbuffer.h:
26749 Add some more fixed payloads.
26751 2007-01-23 18:39:45 +0000 Tim-Philipp Müller <tim@centricular.net>
26753 ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
26754 Original commit message from CVS:
26755 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
26756 Error out properly if we get an error from libogg while reading the
26757 BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).
26759 2007-01-23 17:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
26761 gst/playback/gstdecodebin2.c: Don't leak mutex.
26762 Original commit message from CVS:
26763 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
26765 * tests/check/elements/playbin.c:
26766 (test_sink_usage_video_only_stream),
26767 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
26768 (test_suburi_error_wrongproto), (test_missing_urisource_handler),
26769 (test_missing_suburisource_handler),
26770 (test_missing_primary_decoder), (playbin_suite):
26771 Run all tests once with decodebin and once with decodebin2.
26772 One test does not pass yet with decodebin2.
26774 2007-01-23 14:30:28 +0000 Edward Hervey <bilboed@bilboed.com>
26776 ext/ogg/gstoggmux.c: Fix the cases where oggmux doesn't properly figure out that all sinkpads have gone EOS, and ther...
26777 Original commit message from CVS:
26778 * ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
26779 Fix the cases where oggmux doesn't properly figure out that all
26780 sinkpads have gone EOS, and therefore doesn't push out the remaining
26781 buffers and the final EOS event.
26784 2007-01-23 13:19:19 +0000 Julien Moutte <julien@moutte.net>
26786 sys/: Don't lock on navigation event push, just on keysym to string.
26787 Original commit message from CVS:
26788 2007-01-23 Julien MOUTTE <julien@moutte.net>
26789 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
26790 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
26791 Don't lock on navigation event push, just on keysym to string.
26792 Fixes #397673 again.
26794 2007-01-22 17:37:38 +0000 Edward Hervey <bilboed@bilboed.com>
26796 gst/playback/gstdecodebin2.c: Cleanups.
26797 Original commit message from CVS:
26798 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
26799 (get_current_group), (group_demuxer_event_probe),
26800 (gst_decode_group_expose), (deactivate_free_recursive),
26801 (gst_decode_group_free):
26803 Don't forget to emit 'no-more-pads' once a group is exposed.
26804 Cleanup elements from a DecodeGroup once we remove it.
26805 Protect call to gst_decode_group_expose() with the decodebin lock.
26807 2007-01-22 13:16:42 +0000 Julien Moutte <julien@moutte.net>
26809 sys/: Looking at Xorg code i can't figure out if that XKeysymToString function is thread sensible or not. Lock it jus...
26810 Original commit message from CVS:
26811 2007-01-22 Julien MOUTTE <julien@moutte.net>
26812 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
26813 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
26814 Looking at Xorg code i can't figure out if that XKeysymToString
26815 function is thread sensible or not. Lock it just in case as
26816 recommended by Radek Doulik <rodo at ximian dot com>.
26818 2007-01-22 13:10:13 +0000 Julien Moutte <julien@moutte.net>
26820 sys/: Lock that X Call as well. Fixes #397673.
26821 Original commit message from CVS:
26822 2007-01-22 Julien MOUTTE <julien@moutte.net>
26823 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
26824 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
26825 Lock that X Call as well. Fixes #397673.
26827 2007-01-22 12:03:27 +0000 Tim-Philipp Müller <tim@centricular.net>
26829 gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
26830 Original commit message from CVS:
26831 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
26832 Don't go into an endless loop if the file starts with 00 00 01 2X,
26833 like quicktime redirect files might. Fixes #396042.
26834 * tests/check/Makefile.am:
26835 * tests/check/gst/.cvsignore:
26836 * tests/check/gst/typefindfunctions.c: (GST_START_TEST),
26837 (typefindfunctions_suite):
26838 Add unit test for the above.
26840 2007-01-22 10:27:26 +0000 Tim-Philipp Müller <tim@centricular.net>
26842 gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
26843 Original commit message from CVS:
26844 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
26845 On second thought, use "depth" field rather than "bpp" field.
26847 2007-01-22 09:23:01 +0000 Tim-Philipp Müller <tim@centricular.net>
26849 gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
26850 Original commit message from CVS:
26851 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
26852 Camtasia caps apparently need a bpp field (#398875).
26854 2007-01-19 19:09:05 +0000 Tim-Philipp Müller <tim@centricular.net>
26856 gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
26857 Original commit message from CVS:
26858 * gst/playback/gstplaybasebin.c: (setup_subtitle),
26859 (gen_source_element), (gst_play_base_bin_change_state):
26860 Attempt at a better error message in case we don't have the required
26861 URI handler installed; post missing-plugin message also when we're
26862 missing an URI handler for the subtitle URI; clean up properly also
26863 when an error occurs and we never made it to PAUSED state.
26864 * tests/check/elements/playbin.c: (GST_START_TEST),
26866 Check that we're also getting a missing-plugin messsage for a
26867 missing subtitle URI handler (and clean up properly).
26869 2007-01-19 18:47:30 +0000 Tim-Philipp Müller <tim@centricular.net>
26871 gst/playback/gstplaybasebin.c: Plug a few reference leaks.
26872 Original commit message from CVS:
26873 * gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
26874 Plug a few reference leaks.
26876 2007-01-19 12:23:06 +0000 Tim-Philipp Müller <tim@centricular.net>
26878 gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
26879 Original commit message from CVS:
26880 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
26881 Lower probability a bit if the marker isn't right at the start,
26882 to decrease the chance of false positives.
26884 2007-01-19 11:31:50 +0000 Tim-Philipp Müller <tim@centricular.net>
26886 gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
26887 Original commit message from CVS:
26888 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
26889 Small mpeg2 system stream typefinding improvement: make typefinder
26890 probe a bit into the stream instead of just looking for a marker
26891 at the beginning. Fixes #397810.
26893 2007-01-18 16:23:35 +0000 Tim-Philipp Müller <tim@centricular.net>
26895 gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
26896 Original commit message from CVS:
26897 * gst/audioconvert/gstchannelmix.c:
26898 Remove compatibility cruft for prehistoric GLib versions.
26900 2007-01-17 16:11:14 +0000 Tim-Philipp Müller <tim@centricular.net>
26902 gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
26903 Original commit message from CVS:
26904 * gst/playback/Makefile.am:
26905 * gst/playback/gstdecodebin.c: (close_pad_link):
26906 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
26907 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
26908 (gst_play_base_bin_handle_message_func), (unknown_type):
26909 Let decodebin be the element to post missing-plugin messages for
26910 missing decoders (rather than playbin); make playbin implement
26911 GstBin::handle_message so we can suppress missing-plugin messages
26912 for types we're not handling on purpose (don't want to bring up an
26913 installer in those cases).
26915 2007-01-16 19:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
26917 gst/: Fix potentially unaligned access (#397207).
26918 Original commit message from CVS:
26919 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
26920 * gst-libs/gst/tag/gstvorbistag.c:
26921 (gst_tag_list_to_vorbiscomment_buffer):
26922 * gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
26923 Fix potentially unaligned access (#397207).
26925 2007-01-16 12:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26927 tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
26928 Original commit message from CVS:
26929 * tests/examples/seek/seek.c: (set_scale), (update_scale),
26930 (do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
26931 (rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
26933 Allow to toggle looping while it plays. Fix callback prototype. Clean
26934 up code a bit more. Add copyright header.
26936 2007-01-16 11:41:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26938 sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
26939 Original commit message from CVS:
26940 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
26941 Red and blue mask was swapped (spotted by Dan Williams).
26943 2007-01-15 13:58:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26945 gst-libs/gst/tag/: Use new beats-per-minute tag from core.
26946 Original commit message from CVS:
26947 * gst-libs/gst/tag/gstid3tag.c:
26948 * gst-libs/gst/tag/gstvorbistag.c:
26949 Use new beats-per-minute tag from core.
26951 2007-01-15 11:30:53 +0000 Tim-Philipp Müller <tim@centricular.net>
26953 po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
26954 Original commit message from CVS:
26956 Add new files with translatable strings, so they actually make it
26957 into the template file one day.
26959 2007-01-12 21:19:35 +0000 Andy Wingo <wingo@pobox.com>
26962 * gst-libs/gst/audio/gstbaseaudiosink.c:
26963 * gst-libs/gst/audio/gstbaseaudiosrc.c:
26964 gst-libs/gst/audio/gstbaseaudiosink.c
26965 Original commit message from CVS:
26966 2007-01-12 Andy Wingo <wingo@pobox.com>
26967 * gst-libs/gst/audio/gstbaseaudiosink.c
26968 (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
26969 (gst_base_audio_sink_activate_pull): Remove the handwavey nego
26970 stuff, as the base class handles this now. Actually tell the ring
26972 (gst_base_audio_sink_callback): Cast the ring buffer correctly.
26973 How did this work before? Maybe I'm not as awesome a programmer as
26975 * gst-libs/gst/audio/gstbaseaudiosrc.c
26976 (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
26979 2007-01-12 18:08:23 +0000 Tim-Philipp Müller <tim@centricular.net>
26981 gst-libs/gst/utils/missing-plugins.c: Remove more fields so that the application can better blacklist formats that ha...
26982 Original commit message from CVS:
26983 * gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
26984 Remove more fields so that the application can better blacklist
26985 formats that have been tried before.
26987 2007-01-12 17:43:40 +0000 Christian Schaller <uraeus@gnome.org>
26989 * gst-plugins-base.spec.in:
26991 Original commit message from CVS:
26994 2007-01-12 12:47:29 +0000 Tim-Philipp Müller <tim@centricular.net>
26996 gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
26997 Original commit message from CVS:
26998 * gst-libs/gst/audio/mixerutils.h:
26999 Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
27000 used when compiling with c++ compilers as well.
27002 2007-01-12 09:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
27004 gst/typefind/gsttypefindfunctions.c: Fix comment.
27005 Original commit message from CVS:
27006 * gst/typefind/gsttypefindfunctions.c:
27009 2007-01-11 13:12:17 +0000 Tim-Philipp Müller <tim@centricular.net>
27011 gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
27012 Original commit message from CVS:
27013 * gst/playback/gstplaybin.c: (post_missing_element_message),
27014 (gen_video_element), (gen_text_element), (gen_audio_element),
27016 Post missing-plugin messages also when we error out because
27017 converters, textoverlay or auto*sinks are missing (#161922).
27019 2007-01-10 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
27021 gst/playback/: Fix the case where we try to ref a NULL element when we delay a link because of unfixed caps.
27022 Original commit message from CVS:
27023 * gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
27024 (is_demuxer_element), (new_caps):
27025 * gst/playback/gstplaybasebin.c: (source_new_pad):
27026 Fix the case where we try to ref a NULL element when we delay a link
27027 because of unfixed caps.
27028 Set the state of autoplugged decodebins to PAUSED.
27029 RTSP now works in playbin, we can remove it from the blacklist.
27031 2007-01-09 14:33:24 +0000 Tim-Philipp Müller <tim@centricular.net>
27033 gst/playback/: Post missing-plugin messages on the bus for missing sources and missing decoders/demuxers/depayloaders...
27034 Original commit message from CVS:
27035 * gst/playback/Makefile.am:
27036 * gst/playback/gstplaybasebin.c: (string_arr_has_str),
27037 (unknown_type), (setup_subtitle), (gen_source_element):
27038 * gst/playback/gstplaybin.c: (plugin_init):
27039 Post missing-plugin messages on the bus for missing sources and
27040 missing decoders/demuxers/depayloaders; fix error code used when
27041 we're missing an URI handler source; for media types that we are not
27042 handling on purpose at the moment, don't print "don't know how to
27043 handle xyz" messages to the terminal or post missing-plugin
27044 messages on the bus.
27045 * tests/check/elements/playbin.c: (create_playbin),
27046 (GST_START_TEST), (gst_codec_src_uri_get_type),
27047 (gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
27048 (gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
27049 (gst_codec_src_init_type), (gst_codec_src_base_init),
27050 (gst_codec_src_create), (gst_codec_src_class_init),
27051 (gst_codec_src_init), (plugin_init), (playbin_suite):
27052 Add some tests for the missing-plugin stuff.
27054 2007-01-09 14:20:08 +0000 Tim-Philipp Müller <tim@centricular.net>
27056 API: add new libgstbaseutils library with functions
27057 Original commit message from CVS:
27059 * gst-libs/gst/Makefile.am:
27060 * gst-libs/gst/utils/Makefile.am:
27061 * gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
27062 * gst-libs/gst/utils/base-utils.h:
27063 * gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
27064 (find_format_info), (caps_are_rtp_caps),
27065 (gst_base_utils_get_source_description),
27066 (gst_base_utils_get_sink_description),
27067 (gst_base_utils_get_decoder_description),
27068 (gst_base_utils_get_encoder_description),
27069 (gst_base_utils_get_element_description),
27070 (gst_base_utils_add_codec_description_to_tag_list),
27071 (gst_base_utils_get_codec_description), (gst_base_utils_list_all):
27072 * gst-libs/gst/utils/descriptions.h:
27073 * gst-libs/gst/utils/missing-plugins.c:
27074 (missing_structure_get_type), (copy_and_clean_caps),
27075 (gst_missing_uri_source_message_new),
27076 (gst_missing_uri_sink_message_new),
27077 (gst_missing_element_message_new),
27078 (gst_missing_decoder_message_new),
27079 (gst_missing_encoder_message_new),
27080 (missing_structure_get_string_detail),
27081 (missing_structure_get_caps_detail),
27082 (gst_missing_plugin_message_get_installer_detail),
27083 (gst_missing_plugin_message_get_description),
27084 (gst_is_missing_plugin_message):
27085 * gst-libs/gst/utils/missing-plugins.h:
27086 API: add new libgstbaseutils library with functions
27087 - to create and parse missing-plugins messages
27088 - that provide (translated) descriptions for caps/decoders/sources/etc.
27090 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
27091 * pkgconfig/gstreamer-plugins-base.pc.in:
27093 * docs/libs/gst-plugins-base-libs-docs.sgml:
27094 * docs/libs/gst-plugins-base-libs-sections.txt:
27095 Generate docs for new lib and API.
27096 * tests/check/Makefile.am:
27097 * tests/check/libs/.cvsignore:
27098 * tests/check/libs/utils.c: (missing_msg_check_getters),
27099 (GST_START_TEST), (libgstbaseutils_suite):
27100 Add some basic unit tests.
27102 2007-01-09 13:35:08 +0000 Tim-Philipp Müller <tim@centricular.net>
27104 ext/ogg/Makefile.am: Dist gstoggdemux.h to fix 'make distcheck'.
27105 Original commit message from CVS:
27106 * ext/ogg/Makefile.am:
27107 Dist gstoggdemux.h to fix 'make distcheck'.
27108 * sys/v4l/Makefile.am:
27109 Fix 'make distcheck' even more.
27111 2007-01-09 12:30:46 +0000 Wim Taymans <wim.taymans@gmail.com>
27114 Original commit message from CVS:
27115 * docs/plugins/Makefile.am:
27116 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
27117 * docs/plugins/gst-plugins-base-plugins-sections.txt:
27118 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
27119 (gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
27120 (gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
27121 (gst_ogg_demux_perform_seek):
27122 * ext/ogg/gstoggdemux.h:
27124 Add some more comments.
27127 2007-01-09 11:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
27129 Small documentation updates/fixes
27130 Original commit message from CVS:
27131 * ext/theora/theoradec.c:
27132 * ext/vorbis/vorbisdec.c:
27133 * gst-libs/gst/audio/gstringbuffer.c:
27134 (gst_ring_buffer_commit_full):
27135 * gst-libs/gst/audio/gstringbuffer.h:
27136 * gst-libs/gst/rtp/gstrtpbuffer.c:
27137 * gst-libs/gst/tag/gstvorbistag.c:
27138 Small documentation updates/fixes
27140 2007-01-09 10:37:01 +0000 Tim-Philipp Müller <tim@centricular.net>
27142 configure.ac: Require core CVS HEAD for Andy's basesrc/sink API additions.
27143 Original commit message from CVS:
27145 Require core CVS HEAD for Andy's basesrc/sink API additions.
27147 2007-01-08 14:01:23 +0000 Günter Thelen <daedalus.inc@gmx.net>
27149 gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
27150 Original commit message from CVS:
27151 Patch by: Günter Thelen <daedalus dot inc at gmx net>
27152 * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
27154 Add typefinder for flac-in-ogg in conformance with the ogg-mapping
27155 on flac.sf.net (there appear to be other versions of the first
27156 ogg page in the wild) (#391365).
27158 2007-01-08 13:32:32 +0000 Tim-Philipp Müller <tim@centricular.net>
27160 configure.ac: Check if localtime_r() is available.
27161 Original commit message from CVS:
27163 Check if localtime_r() is available.
27164 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
27165 If localtime_r() is not available, fall back to localtime(). Should
27166 fix build on MingW (#393310).
27168 2007-01-08 12:30:03 +0000 Tim-Philipp Müller <tim@centricular.net>
27170 gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
27171 Original commit message from CVS:
27172 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
27173 * gst/subparse/gstsubparse.h:
27174 Remove spurious 1000 subtrahend when calculating the timestamp from
27175 the frame number and the frame rate . Also, use the frames/second
27176 value specified in the first line of the file, if one is specified
27177 there. Should fix #357503.
27178 * tests/check/elements/subparse.c: (do_test),
27179 (test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
27181 Add some basic unit tests for the microdvd subtitle format.
27183 2007-01-07 21:53:38 +0000 Young-Ho Cha <ganadist@chollian.net>
27185 sys/xvimage/xvimagesink.c: Fixes : #390076.
27186 Original commit message from CVS:
27187 2007-01-07 Julien MOUTTE <julien@moutte.net>
27188 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
27189 (gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
27190 (gst_xvimagesink_xvimage_put),
27191 (gst_lookup_xv_port_from_adaptor),
27192 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
27193 (gst_xvimagesink_set_xwindow_id),
27194 (gst_xvimagesink_set_event_handling),
27195 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
27196 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
27197 Patch by : Young-Ho Cha <ganadist at chollian dot net>
27199 Add an adaptor property to select a specific XV adaptor.
27200 * sys/xvimage/xvimagesink.h:
27202 2007-01-07 18:50:13 +0000 Julien Moutte <julien@moutte.net>
27204 sys/: Use flow_lock much more to protect every access to xwindow.
27205 Original commit message from CVS:
27206 2007-01-07 Julien MOUTTE <julien@moutte.net>
27207 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
27208 (gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
27209 (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
27210 (gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
27211 (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
27212 (gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
27213 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
27214 (gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
27215 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
27216 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
27217 (gst_xvimagesink_change_state),
27218 (gst_xvimagesink_set_xwindow_id),
27219 (gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
27220 Use flow_lock much more to protect every access to xwindow.
27221 Try to catch erros while creating images in case some drivers
27223 just generating an XError when the requested image is too big.
27224 Should fix : #354698, #384008, #384060.
27225 * tests/icles/stress-xoverlay.c: (cycle_window),
27227 Implement some stress testing of setting window xid.
27229 2007-01-07 10:33:55 +0000 Sébastien Moutte <sebastien@moutte.net>
27231 win32/common/libgsaudio.def: Add new exported function.
27232 Original commit message from CVS:
27233 * win32/common/libgsaudio.def:
27234 Add new exported function.
27235 * win32/common/libgstogg.dsp:
27236 Add gstoggaviparse.c to the build.
27237 * win32/common/libgstvideoscale.dsp:
27238 Add vs_4tap.c to the build.
27239 * win32/common/libgstvorbis.dsp:
27240 Add vorbistag.c to the build.
27242 2007-01-06 17:28:40 +0000 Andy Wingo <wingo@pobox.com>
27245 * gst-libs/gst/audio/gstbaseaudiosink.c:
27246 gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
27247 Original commit message from CVS:
27248 2007-01-06 Andy Wingo <wingo@pobox.com>
27249 * gst-libs/gst/audio/gstbaseaudiosink.c
27250 (gst_base_audio_sink_class_init)
27251 (gst_base_audio_sink_init):
27252 (gst_base_audio_sink_activate_pull): Add an activate_pull function
27253 to baseaudiosink, and tell basesink that we can work in pull mode.
27254 This way the ring buffer thread drives the pipeline directly, if
27255 pull mode is possible. There is some lingering nastiness regarding
27257 (gst_base_audio_sink_callback): Implement the callback to pull
27258 data. This interface is a bit light, though -- it should get a
27259 GstFlowReturn return value at least.
27261 2007-01-05 19:43:55 +0000 Tim-Philipp Müller <tim@centricular.net>
27263 Printf format and missing argument fixes.
27264 Original commit message from CVS:
27265 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
27266 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
27267 * gst/playback/gstdecodebin2.c:
27268 (gst_decode_group_check_if_blocked):
27269 Printf format and missing argument fixes.
27271 2007-01-05 18:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27273 ext/ogg/gstogmparse.c: Activate pads before adding them to the element.
27274 Original commit message from CVS:
27275 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
27276 (gst_ogm_parse_change_state):
27277 Activate pads before adding them to the element.
27279 2007-01-05 16:02:50 +0000 Tim-Philipp Müller <tim@centricular.net>
27281 tests/examples/seek/: Call g_thread_init() first thing in main() (see #391278).
27282 Original commit message from CVS:
27283 * tests/examples/seek/scrubby.c: (main):
27284 * tests/examples/seek/seek.c: (main):
27285 Call g_thread_init() first thing in main() (see #391278).
27287 2007-01-05 12:19:34 +0000 Tim-Philipp Müller <tim@centricular.net>
27289 tests/check/: Add test for GstNetBuffer + gst_buffer_copy(). Disabled for the time being, since it's broken, see #393...
27290 Original commit message from CVS:
27291 * tests/check/Makefile.am:
27292 * tests/check/libs/.cvsignore:
27293 * tests/check/libs/netbuffer.c: (GST_START_TEST),
27295 Add test for GstNetBuffer + gst_buffer_copy(). Disabled
27296 for the time being, since it's broken, see #393099.
27298 2007-01-05 12:13:24 +0000 Tim-Philipp Müller <tim@centricular.net>
27300 tests/check/Makefile.am: Update to use GST_PLUGINS_BASE_CFLAGS as well.
27301 Original commit message from CVS:
27302 * tests/check/Makefile.am:
27303 Update to use GST_PLUGINS_BASE_CFLAGS as well.
27305 2007-01-04 12:49:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27307 configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
27308 Original commit message from CVS:
27310 split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
27311 so that GST_BASE_CFLAGS can go inbetween them, making sure
27312 we use uninstalled gst-libs headers
27313 * docs/libs/Makefile.am:
27314 * ext/alsa/Makefile.am:
27315 * ext/cdparanoia/Makefile.am:
27316 * ext/gnomevfs/Makefile.am:
27317 * ext/libvisual/Makefile.am:
27318 * ext/ogg/Makefile.am:
27319 * ext/theora/Makefile.am:
27320 * ext/vorbis/Makefile.am:
27321 * gst-libs/gst/audio/Makefile.am:
27322 * gst-libs/gst/cdda/Makefile.am:
27323 * gst-libs/gst/interfaces/Makefile.am:
27324 * gst-libs/gst/riff/Makefile.am:
27325 * gst-libs/gst/rtp/Makefile.am:
27326 * gst-libs/gst/tag/Makefile.am:
27327 * gst/adder/Makefile.am:
27328 * gst/audioconvert/Makefile.am:
27329 * gst/audiorate/Makefile.am:
27330 * gst/audioresample/Makefile.am:
27331 * gst/playback/Makefile.am:
27332 * gst/tcp/Makefile.am:
27333 * gst/videoscale/Makefile.am:
27334 * gst/volume/Makefile.am:
27335 * sys/ximage/Makefile.am:
27336 * sys/xvimage/Makefile.am:
27337 * tests/icles/Makefile.am:
27340 2007-01-04 11:30:53 +0000 Julien Moutte <julien@moutte.net>
27342 Add a method to the XOverlay interface to allow disabling of event handling in x[v]imagesink elements. This will let ...
27343 Original commit message from CVS:
27344 2007-01-04 Julien MOUTTE <julien@moutte.net>
27345 * gst-libs/gst/interfaces/xoverlay.c:
27346 (gst_x_overlay_handle_events):
27347 * gst-libs/gst/interfaces/xoverlay.h:
27348 * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
27349 (gst_ximagesink_set_xwindow_id),
27350 (gst_ximagesink_set_event_handling),
27351 (gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
27352 (gst_ximagesink_get_property), (gst_ximagesink_init),
27353 (gst_ximagesink_class_init):
27354 * sys/ximage/ximagesink.h:
27355 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
27356 (gst_xvimagesink_set_xwindow_id),
27357 (gst_xvimagesink_set_event_handling),
27358 (gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
27359 (gst_xvimagesink_get_property), (gst_xvimagesink_init),
27360 (gst_xvimagesink_class_init):
27361 * sys/xvimage/xvimagesink.h:
27362 * tests/icles/stress-xoverlay.c: (toggle_events),
27364 Add a method to the XOverlay interface to allow disabling of
27365 event handling in x[v]imagesink elements. This will let X events
27366 propagate to parent windows which can be usefull in some cases.
27367 Be carefull that the application is then responsible of pushing
27368 navigation events and expose events to the video sink.
27371 2007-01-03 15:45:06 +0000 Tim-Philipp Müller <tim@centricular.net>
27373 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION (fixes #392070).
27374 Original commit message from CVS:
27375 * gst-libs/gst/tag/gstvorbistag.c:
27376 * tests/check/libs/tag.c: (GST_START_TEST):
27377 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
27380 2007-01-01 18:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
27383 Original commit message from CVS:
27385 * docs/Makefile.am:
27386 * docs/design/Makefile.am:
27389 2006-12-27 17:15:35 +0000 Julien Moutte <julien@moutte.net>
27391 docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation typo. Fixes: #390063.
27392 Original commit message from CVS:
27393 2006-12-27 Julien MOUTTE <julien@moutte.net>
27394 * docs/libs/gst-plugins-base-libs-sections.txt: Fix a
27396 typo. Fixes: #390063.
27398 2006-12-27 12:08:13 +0000 Julien Moutte <julien@moutte.net>
27400 sys/: Plug a caps leak.
27401 Original commit message from CVS:
27402 2006-12-27 Julien MOUTTE <julien@moutte.net>
27403 * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
27404 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
27406 * win32/common/config.h: Updated.
27408 2006-12-22 12:10:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27410 tests/check/elements/: Fix the dp tests, but activating the pads for the streamheader tests too and cleaning up condi...
27411 Original commit message from CVS:
27412 * tests/check/elements/gdpdepay.c: (cleanup_gdpdepay),
27413 (setup_gdpdepay_streamheader):
27414 * tests/check/elements/gdppay.c: (cleanup_gdppay),
27415 (setup_gdppay_streamheader):
27416 Fix the dp tests, but activating the pads for the streamheader tests
27417 too and cleaning up conditionaly
27419 2006-12-22 11:09:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27421 gst/ffmpegcolorspace/: Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the other end of the wo...
27422 Original commit message from CVS:
27423 * gst/ffmpegcolorspace/avcodec.h:
27424 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
27425 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
27426 (gst_ffmpegcsp_avpicture_fill):
27427 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
27428 (img_get_alpha_info):
27429 Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
27430 other end of the word. Fixes: #387073.
27431 Add some inconsequential branch hints in a couple of places.
27433 2006-12-21 12:30:11 +0000 Tim-Philipp Müller <tim@centricular.net>
27435 gst/ffmpegcolorspace/gstffmpegcodecmap.c: The "signed" field in raw audio caps is of boolean type, trying to extract ...
27436 Original commit message from CVS:
27437 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
27438 (gst_ffmpeg_caps_to_smpfmt):
27439 The "signed" field in raw audio caps is of boolean type, trying to
27440 extract the value with _get_int() will fail (fix to keep in sync with
27441 the copy in gst-ffmpeg)
27443 2006-12-21 08:12:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27445 tests/check/elements/: consistent pad (de)activation
27446 Original commit message from CVS:
27447 * tests/check/elements/audioresample.c: (cleanup_audioresample):
27448 * tests/check/elements/audiotestsrc.c: (cleanup_audiotestsrc):
27449 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
27450 (cleanup_gdpdepay):
27451 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay):
27452 * tests/check/elements/subparse.c: (teardown_subparse):
27453 * tests/check/elements/textoverlay.c: (cleanup_textoverlay):
27454 * tests/check/elements/videorate.c: (cleanup_videorate):
27455 * tests/check/elements/videotestsrc.c: (cleanup_videotestsrc):
27456 * tests/check/elements/volume.c: (cleanup_volume):
27457 * tests/check/elements/vorbisdec.c: (setup_vorbisdec),
27458 (cleanup_vorbisdec):
27459 * tests/check/elements/vorbistag.c: (setup_vorbistag),
27460 (cleanup_vorbistag):
27461 consistent pad (de)activation
27463 2006-12-20 10:29:58 +0000 Tim-Philipp Müller <tim@centricular.net>
27465 gst/typefind/gsttypefindfunctions.c: Forgot to register the extensions.
27466 Original commit message from CVS:
27467 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
27468 Forgot to register the extensions.
27470 2006-12-20 09:25:55 +0000 Tim-Philipp Müller <tim@centricular.net>
27472 gst/typefind/gsttypefindfunctions.c: Add typefinder for VIVO files (my christmas present to the 90s).
27473 Original commit message from CVS:
27474 * gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
27476 Add typefinder for VIVO files (my christmas present to the 90s).
27478 2006-12-16 13:59:09 +0000 Tim-Philipp Müller <tim@centricular.net>
27480 gst/playback/gstdecodebin.c: Special-case the text/plain media type: we only want to recognise it as a 'raw' decoded ...
27481 Original commit message from CVS:
27482 * gst/playback/gstdecodebin.c: (type_found):
27483 Special-case the text/plain media type: we only want to recognise it
27484 as a 'raw' decoded media type if it comes from a demuxer or subtitle
27485 parser, but not if the entire stream is of text/plain type. If the
27486 entire stream is text/plain, we should just error out.
27487 This fixes playback of audio files with lyrics in totem. Totem can't
27488 distinguish between text files and subtitle files and passes any
27489 .txt file with the same basename as the main file to playbin as
27490 suburi, and playbin will then throw a 'subtitle found, but no video
27491 stream' error, which isn't entirely helpful. See #380342.
27492 Also, with this change we'll show a slightly more correct error
27493 message in case totem passes a playlist file to us (although a
27494 custom error message wording instead of the default text would
27495 probably not be a bad idea either).
27496 Same problem also needs to be fixed for playbin+decodebin2.
27497 * tests/check/Makefile.am:
27498 * tests/check/elements/decodebin.c: (src_handoff_cb),
27499 (decodebin_new_decoded_pad_cb), (GST_START_TEST),
27501 Add simple unit test for decodebin for the above.
27503 2006-12-16 12:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
27505 gst/playback/: Refuse to change state to READY when we failed to create any of the required elements in our instance ...
27506 Original commit message from CVS:
27507 * gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
27508 * gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
27509 Refuse to change state to READY when we failed to create any of the
27510 required elements in our instance init function.
27512 2006-12-15 10:52:23 +0000 Tim-Philipp Müller <tim@centricular.net>
27514 docs/libs/gst-plugins-base-libs-sections.txt: Small docs fixes/updates.
27515 Original commit message from CVS:
27516 * docs/libs/gst-plugins-base-libs-sections.txt:
27517 Small docs fixes/updates.
27518 * gst-libs/gst/video/gstvideosink.h:
27519 Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
27520 from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
27521 removed from the base sink API between 0.9.6 and 0.9.7).
27522 API: add GST_VIDEO_SINK_CAST and use it for the height/width
27523 accessor macros, so we don't do a runtime GObject type check every
27526 2006-12-15 00:20:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27529 Original commit message from CVS:
27531 * gst-plugins-base.doap:
27532 * gst-plugins-base.spec.in:
27535 2006-12-09 15:12:38 +0000 Jens Granseuer <jensgr@gmx.net>
27537 Declare variables at the beginning of a block. Fixes #383195.
27538 Original commit message from CVS:
27539 Patch by: Jens Granseuer <jensgr at gmx net>
27540 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
27541 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
27542 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
27543 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
27544 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
27545 Declare variables at the beginning of a block. Fixes #383195.
27547 2006-12-07 02:38:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27549 configure.ac: Bump version nano - back to CVS.
27550 Original commit message from CVS:
27552 Bump version nano - back to CVS.
27554 === release 0.10.11 ===
27556 2006-12-07 02:30:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27558 configure.ac: releasing 0.10.11, "Dumb things"
27559 Original commit message from CVS:
27560 === release 0.10.11 ===
27561 2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
27563 releasing 0.10.11, "Dumb things"
27565 2006-12-05 12:44:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27567 gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
27568 Original commit message from CVS:
27569 * gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
27570 (close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
27571 Handle the case where an element has multiple pads with
27572 unfixed caps as well as still possibly producing more dynamic
27573 pads by storing each case as a distinct entry in the dynamic list.
27574 Fixes #38223 again.
27576 2006-12-04 13:02:37 +0000 Wim Taymans <wim.taymans@gmail.com>
27578 gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
27579 Original commit message from CVS:
27580 * gst/playback/gstdecodebin.c: (close_pad_link):
27581 Fix #382223, add more dynamic caps handling.
27583 2006-12-01 11:35:57 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
27586 Ignore all pot files
27587 Original commit message from CVS:
27588 Ignore all pot files
27590 2006-12-01 10:36:50 +0000 Michael Smith <msmith@xiph.org>
27592 gst/audiorate/gstaudiorate.c: Delete bad debug code.
27593 Original commit message from CVS:
27594 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
27595 Delete bad debug code.
27598 2006-12-01 10:27:54 +0000 Sergey Scobich <sergey.scobich@gmail.com>
27600 Fix compilation on win32 under VS8
27601 Original commit message from CVS:
27602 * gst/videoscale/vs_4tap.c:
27604 * win32/common/config.h:
27605 * win32/vs8/libgstvideoscale.vcproj:
27606 Fix compilation on win32 under VS8
27607 Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
27608 Partially fixes #381175
27610 2006-11-30 23:46:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27627 Original commit message from CVS:
27630 2006-11-30 12:50:42 +0000 Michael Smith <msmith@xiph.org>
27632 tests/check/pipelines/theoraenc.c: It would be very bad if, after a discont buffer, we thought every single following...
27633 Original commit message from CVS:
27634 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
27636 It would be very bad if, after a discont buffer, we thought every
27637 single following buffer was also discont. So, add to the test to
27638 ensure that this isn't the case.
27639 * ext/theora/theoraenc.c: (theora_enc_is_discontinuous):
27640 ... it was the case. So fix it.
27642 2006-11-28 16:43:18 +0000 Wim Taymans <wim.taymans@gmail.com>
27644 gst/playback/gstplaybasebin.c: Improve debug.
27645 Original commit message from CVS:
27646 * gst/playback/gstplaybasebin.c: (check_queue_event):
27648 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
27649 Fix width and height range from 16 - 4096 to 1 - MAXINT, just like the
27650 padtemplate caps. Refixes #357577.
27652 2006-11-28 16:21:27 +0000 Wim Taymans <wim.taymans@gmail.com>
27654 gst/playback/gstplaybasebin.c: Add event probe to see when EOS is in a queue and we can disable the underrun signals....
27655 Original commit message from CVS:
27656 * gst/playback/gstplaybasebin.c: (check_queue_event),
27657 (queue_threshold_reached), (queue_out_of_data),
27658 (gen_preroll_element):
27659 Add event probe to see when EOS is in a queue and we can disable the
27660 underrun signals. Fixes #357577.
27662 2006-11-28 14:40:39 +0000 Edward Hervey <bilboed@bilboed.com>
27664 gst/playback/: New decodebin2 element.
27665 Original commit message from CVS:
27666 * gst/playback/Makefile.am:
27667 * gst/playback/gstdecodebin2.c: (gst_decode_bin_get_type),
27668 (_gst_boolean_accumulator), (gst_decode_bin_class_init),
27669 (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
27670 (gst_decode_bin_init), (gst_decode_bin_dispose),
27671 (gst_decode_bin_finalize), (gst_decode_bin_set_property),
27672 (gst_decode_bin_get_property), (gst_decode_bin_set_caps),
27673 (gst_decode_bin_get_caps), (gst_decode_bin_autoplug_continue),
27674 (gst_decode_bin_autoplug_sort), (analyze_new_pad), (connect_pad),
27675 (connect_element), (expose_pad), (type_found),
27676 (pad_added_group_cb), (pad_removed_group_cb),
27677 (no_more_pads_group_cb), (pad_added_cb), (pad_removed_cb),
27678 (no_more_pads_cb), (find_compatibles), (is_demuxer_element),
27679 (are_raw_caps), (multi_queue_overrun_cb),
27680 (multi_queue_underrun_cb), (gst_decode_group_new),
27681 (get_current_group), (group_demuxer_event_probe),
27682 (gst_decode_group_control_demuxer_pad),
27683 (gst_decode_group_control_source_pad),
27684 (gst_decode_group_check_if_blocked),
27685 (gst_decode_group_check_if_drained), (gst_decode_group_expose),
27686 (gst_decode_group_hide), (gst_decode_group_free),
27687 (gst_decode_group_set_complete), (source_pad_blocked_cb),
27688 (source_pad_event_probe), (gst_decode_pad_new), (add_fakesink),
27689 (remove_fakesink), (find_sink_pad), (gst_decode_bin_change_state),
27691 New decodebin2 element.
27693 * gst/playback/gstplay-marshal.list:
27694 Added marshallers for new signals in decodebin2
27695 * gst/playback/gstplaybasebin.c: (setup_subtitle), (make_decoder):
27696 Use decodebin2 if *and only if* the USE_DECODEBIN2 environment variable
27699 2006-11-28 10:45:40 +0000 Wim Taymans <wim.taymans@gmail.com>
27701 gst/playback/gstplaybasebin.c: Disable rtsp:// uris for the release, it's not good enough yet.
27702 Original commit message from CVS:
27703 * gst/playback/gstplaybasebin.c: (setup_source),
27704 (gst_play_base_bin_change_state):
27705 Disable rtsp:// uris for the release, it's not good enough yet.
27708 2006-11-26 16:39:41 +0000 Wim Taymans <wim.taymans@gmail.com>
27710 ext/theora/theoradec.c: Implement reverse playback.
27711 Original commit message from CVS:
27712 * ext/theora/theoradec.c: (gst_theora_dec_reset),
27713 (theora_dec_push_forward), (theora_dec_push_reverse),
27714 (theora_handle_data_packet), (theora_dec_decode_buffer),
27715 (theora_dec_flush_decode), (theora_dec_chain_reverse),
27716 (theora_dec_chain_forward), (theora_dec_chain):
27717 Implement reverse playback.
27718 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
27719 (vorbis_dec_decode_buffer), (vorbis_dec_flush_decode),
27720 (vorbis_dec_chain_forward):
27721 Clear buffers used for reverse playback in _reset.
27722 No need to set the eos flag, we clip samples using the segment.
27724 2006-11-24 15:40:58 +0000 Wim Taymans <wim.taymans@gmail.com>
27726 ext/ogg/gstoggdemux.c: Some cleanups.
27727 Original commit message from CVS:
27728 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
27729 (gst_ogg_pad_init), (gst_ogg_pad_dispose), (gst_ogg_pad_reset),
27730 (gst_ogg_pad_stream_out), (gst_ogg_pad_submit_page),
27731 (gst_ogg_chain_reset), (gst_ogg_demux_perform_seek):
27733 Handle continued pages in reverse mode.
27735 2006-11-24 15:39:03 +0000 Wim Taymans <wim.taymans@gmail.com>
27737 ext/vorbis/vorbisdec.c: Small cleanups.
27738 Original commit message from CVS:
27739 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
27740 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
27741 (vorbis_dec_flush_decode):
27743 Don't try to add invalid timestamps.
27744 Clipping will unref the buffer.
27746 2006-11-24 08:56:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27748 gst/: remove obsolete _factory_init protos
27749 Original commit message from CVS:
27750 * gst/adder/gstadder.h:
27751 * gst/audiotestsrc/gstaudiotestsrc.h:
27752 remove obsolete _factory_init protos
27754 2006-11-24 08:35:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27756 sys/xvimage/xvimagesink.c: Fix spacing in debug message.
27757 Original commit message from CVS:
27758 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
27759 Fix spacing in debug message.
27761 2006-11-23 11:07:23 +0000 Wim Taymans <wim.taymans@gmail.com>
27763 ext/ogg/gstoggdemux.c: Don't just ignore return values from _pad_push().
27764 Original commit message from CVS:
27765 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
27766 (gst_ogg_demux_chain):
27767 Don't just ignore return values from _pad_push().
27768 Small debug improvements.
27770 2006-11-23 11:02:11 +0000 Michael Smith <msmith@xiph.org>
27772 ext/ogg/gstoggmux.c: If our incoming buffer is marked as DISCONT, then increment the page number (so that the discont...
27773 Original commit message from CVS:
27774 * ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
27775 If our incoming buffer is marked as DISCONT, then increment the page
27776 number (so that the discontinuity is marked in the final ogg
27777 bitstream) and flush the previous page.
27779 2006-11-22 14:34:03 +0000 Michael Smith <msmith@xiph.org>
27781 ext/theora/: Mark discontinuities of > 3/4 of a frame, reinit encoder.
27782 Original commit message from CVS:
27783 * ext/theora/gsttheoraenc.h:
27784 * ext/theora/theoraenc.c: (gst_theora_enc_init),
27785 (theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
27786 (theora_buffer_from_packet), (theora_enc_is_discontinuous),
27787 (theora_enc_chain), (theora_enc_change_state):
27788 Mark discontinuities of > 3/4 of a frame, reinit encoder.
27789 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
27790 (GST_START_TEST), (theoraenc_suite):
27791 Enable discontinuity test, fix it.
27793 2006-11-21 18:39:34 +0000 Tim-Philipp Müller <tim@centricular.net>
27795 ext/pango/gsttextoverlay.*: Some textoverlay fixes: for one, in the video chain function, actually wait for a text bu...
27796 Original commit message from CVS:
27797 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
27798 (gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
27799 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
27800 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
27801 (gst_text_overlay_change_state):
27802 * ext/pango/gsttextoverlay.h:
27803 Some textoverlay fixes: for one, in the video chain function,
27804 actually wait for a text buffer to come in if there is none at the
27805 moment and there should be one; also, deal more gracefully with
27806 incoming buffers that do not have a timestamp or duration; discard
27807 text buffer when not needed any longer. Fixes #341681.
27808 * tests/check/Makefile.am:
27809 * tests/check/elements/.cvsignore:
27810 * tests/check/elements/textoverlay.c:
27811 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
27812 (setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
27813 (create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
27814 (test_video_waits_for_text_send_text_newsegment_thread),
27815 (test_video_waits_for_text_shutdown_element),
27816 (test_render_continuity_push_video_buffers_thread),
27817 (textoverlay_suite):
27818 Add some unit tests for textoverlay.
27820 2006-11-21 09:29:56 +0000 Tim-Philipp Müller <tim@centricular.net>
27822 gst/typefind/gsttypefindfunctions.c: Avoid integer underflow when the found probability for mp3 is smaller than the '...
27823 Original commit message from CVS:
27824 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
27825 Avoid integer underflow when the found probability for mp3 is
27826 smaller than the 'penalty' we subtract if there's not a clean
27827 mp3 header sync at offset 0.
27829 2006-11-21 08:17:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27831 docs/libs/gst-plugins-base-libs-sections.txt: Add some new symbols to the docs
27832 Original commit message from CVS:
27833 * docs/libs/gst-plugins-base-libs-sections.txt:
27834 Add some new symbols to the docs
27836 2006-11-20 16:44:28 +0000 Tim-Philipp Müller <tim@centricular.net>
27838 tests/check/: Enable ffmpegcolorspace test now that the RGBA32 issue is fixed (for now not for valgrinding though, si...
27839 Original commit message from CVS:
27840 * tests/check/Makefile.am:
27841 * tests/check/elements/ffmpegcolorspace.c:
27842 (ffmpegcolorspace_suite):
27843 Enable ffmpegcolorspace test now that the RGBA32 issue is fixed
27844 (for now not for valgrinding though, since it takes too long).
27846 2006-11-20 15:01:09 +0000 Wim Taymans <wim.taymans@gmail.com>
27848 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix RGBA32 caps. Fixes #357038.
27849 Original commit message from CVS:
27850 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
27851 (gst_ffmpeg_pixfmt_to_caps):
27852 Fix RGBA32 caps. Fixes #357038.
27854 2006-11-20 12:20:39 +0000 Tim-Philipp Müller <tim@centricular.net>
27856 gst-libs/gst/interfaces/mixertrack.h: Add FIXME so we can add some padding here in 0.11
27857 Original commit message from CVS:
27858 * gst-libs/gst/interfaces/mixertrack.h:
27859 Add FIXME so we can add some padding here in 0.11
27861 2006-11-19 17:07:34 +0000 Tim-Philipp Müller <tim@centricular.net>
27863 gst-libs/gst/rtp/gstbasertpaudiopayload.h: Fix GstBaseRTPAudioPayload structure so the whole GObject inheritance busi...
27864 Original commit message from CVS:
27865 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
27866 Fix GstBaseRTPAudioPayload structure so the whole GObject
27867 inheritance business actually works (parent class instance structure
27868 must always come first in the derived class instance structure).
27870 2006-11-16 14:35:30 +0000 Tim-Philipp Müller <tim@centricular.net>
27872 Make sure our checks and the videotestsrc plugin link against the local uninstalled gst libs and not any installed gs...
27873 Original commit message from CVS:
27874 * gst/videotestsrc/Makefile.am:
27875 * tests/check/Makefile.am:
27876 Make sure our checks and the videotestsrc plugin link against the
27877 local uninstalled gst libs and not any installed gst libs that
27878 might happen to exist as well.
27879 * tests/check/elements/adder.c: (message_received),
27880 (test_event_message_received), (test_play_twice_message_received):
27881 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
27882 Fix compiler warnings when compiling against core with disabled
27885 2006-11-16 12:55:08 +0000 Michael Smith <msmith@xiph.org>
27887 gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
27888 Original commit message from CVS:
27889 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
27890 (gst_audio_rate_sink_event), (gst_audio_rate_chain):
27891 Fix audiorate, so that it accurately sets offsets and timestamps.
27892 Doesn't change the fundamental algorithmic decisions; so should be
27894 * tests/check/Makefile.am:
27895 Enable audiorate test now that it passes.
27897 2006-11-15 10:05:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27899 sys/xvimage/xvimagesink.c: clear xv when going to NULL, remove // commented non-existant proto
27900 Original commit message from CVS:
27901 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
27902 clear xv when going to NULL, remove // commented non-existant proto
27903 * tests/examples/seek/seek.c: (main):
27904 add missing tooltip description for scrub and play_scrub
27906 2006-11-14 23:34:19 +0000 David Schleef <ds@schleef.org>
27908 configure.ac: Bump liboil requirement to 0.3.8.
27909 Original commit message from CVS:
27911 Bump liboil requirement to 0.3.8.
27912 * gst-libs/gst/riff/riff-media.c:
27914 * gst/videoscale/vs_image.h:
27915 * gst/videoscale/vs_scanline.h:
27916 Use liboil's stdint.h.
27917 * gst/videotestsrc/videotestsrc.c:
27918 Remove liboil related ifdef's, since they aren't needed now, and
27919 won't work with future versions.
27921 2006-11-14 23:08:38 +0000 David Schleef <ds@schleef.org>
27923 gst/videoscale/: Add a 4-tap image scaler. Theoretically looks much prettier.
27924 Original commit message from CVS:
27925 * gst/videoscale/Makefile.am:
27926 * gst/videoscale/gstvideoscale.c:
27927 * gst/videoscale/gstvideoscale.h:
27928 * gst/videoscale/vs_4tap.c:
27929 * gst/videoscale/vs_4tap.h:
27930 * gst/videoscale/vs_image.c:
27931 * gst/videoscale/vs_image.h:
27932 * gst/videoscale/vs_scanline.c:
27933 * gst/videoscale/vs_scanline.h:
27934 Add a 4-tap image scaler. Theoretically looks much prettier.
27935 The tap calculation could use some improvement.
27937 2006-11-14 11:54:14 +0000 Jan David Mol <j.j.d.mol@tudelft.nl>
27939 Various gsize and gssize printf fixes. Fixes #372507.
27940 Original commit message from CVS:
27941 Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
27942 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
27943 (gst_riff_parse_strf_iavs):
27944 * gst/subparse/gstsubparse.c: (convert_encoding):
27945 * gst/tcp/gstmultifdsink.c:
27946 (gst_multi_fd_sink_handle_client_write):
27947 * gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
27948 (gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
27949 (gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
27950 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
27951 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
27952 (gst_ximagesink_ximage_new):
27953 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
27954 Various gsize and gssize printf fixes. Fixes #372507.
27956 2006-11-13 18:14:48 +0000 Wim Taymans <wim.taymans@gmail.com>
27958 ext/vorbis/vorbisdec.*: First stab at vorbis reverse playback.
27959 Original commit message from CVS:
27960 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
27961 (vorbis_dec_push_forward), (vorbis_dec_push_reverse),
27962 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
27963 (vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
27964 (vorbis_dec_chain_forward), (vorbis_dec_chain):
27965 * ext/vorbis/vorbisdec.h:
27966 First stab at vorbis reverse playback.
27968 2006-11-13 17:30:17 +0000 Wim Taymans <wim.taymans@gmail.com>
27970 gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
27971 Original commit message from CVS:
27972 * gst-libs/gst/audio/gstbaseaudiosink.c:
27973 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
27974 * gst-libs/gst/audio/gstbaseaudiosink.h:
27975 Make the clock sync code more accurate wrt resampling and playback
27976 at different rates.
27977 * gst-libs/gst/audio/gstringbuffer.c:
27978 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
27979 * gst-libs/gst/audio/gstringbuffer.h:
27980 Use better algorithm to interpolate sample rates.
27982 2006-11-13 15:31:01 +0000 Michael Smith <msmith@xiph.org>
27984 ext/ogg/gstoggdemux.c: Improve a debug line slightly.
27985 Original commit message from CVS:
27986 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
27987 Improve a debug line slightly.
27988 * ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
27989 Call gst_riff_init() in plugin_init, to avoid getting errors from
27990 the debug system (unrelated changes to another plugin made this turn
27993 2006-11-10 19:20:21 +0000 Sergey Scobich <sergery.scobich@gmail.com>
27995 win32/common/libgsttag.def: Add missing symbol (#366492).
27996 Original commit message from CVS:
27997 Patch by: Sergey Scobich <sergery.scobich at gmail com>
27998 * win32/common/libgsttag.def:
27999 Add missing symbol (#366492).
28001 2006-11-10 00:52:55 +0000 Tim-Philipp Müller <tim@centricular.net>
28003 gst/playback/gststreamselector.c: Don't unref a NULL pad.
28004 Original commit message from CVS:
28005 * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
28006 Don't unref a NULL pad.
28008 2006-11-09 00:50:00 +0000 Christian Schaller <uraeus@gnome.org>
28010 ext/ogg/gstoggdemux.c: Implement first stab at reverse playback.
28011 Original commit message from CVS:
28012 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
28013 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_perform_seek),
28014 (gst_ogg_demux_handle_page), (gst_ogg_demux_chain),
28015 (gst_ogg_demux_loop_forward), (gst_ogg_demux_loop_reverse),
28016 (gst_ogg_demux_loop):
28017 Implement first stab at reverse playback.
28019 2006-11-07 07:22:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28021 gst-libs/gst/riff/riff-media.c: add h263/h264 variants to the caps, Fixes #363118
28022 Original commit message from CVS:
28023 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
28024 (gst_riff_create_video_template_caps):
28025 add h263/h264 variants to the caps, Fixes #363118
28027 2006-11-06 18:24:59 +0000 Tim-Philipp Müller <tim@centricular.net>
28029 gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
28030 Original commit message from CVS:
28031 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
28032 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
28033 Use g_strerror instead of strerror so we get UTF-8.
28035 2006-11-04 07:25:58 +0000 David Schleef <ds@schleef.org>
28037 ext/ogg/: Add/remove KW-DIRAC header here, since it is ogg-specific.
28038 Original commit message from CVS:
28039 * ext/ogg/gstoggdemux.c:
28040 * ext/ogg/gstoggmux.c:
28041 Add/remove KW-DIRAC header here, since it is ogg-specific.
28043 2006-11-03 15:44:31 +0000 Michael Smith <msmith@xiph.org>
28045 gst/typefind/gsttypefindfunctions.c: Recognise more mpeg4 elementary video streams.
28046 Original commit message from CVS:
28047 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
28048 Recognise more mpeg4 elementary video streams.
28050 2006-11-02 17:26:03 +0000 Edward Hervey <bilboed@bilboed.com>
28052 gst/typefind/gsttypefindfunctions.c: Lower the probability of mp3 typefinding functions if we don't find a valid mp3 ...
28053 Original commit message from CVS:
28054 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
28055 Lower the probability of mp3 typefinding functions if we don't find a
28056 valid mp3 header at the start of the file.
28059 2006-11-02 15:06:36 +0000 Wim Taymans <wim.taymans@gmail.com>
28061 ext/theora/: Document and partially implement an algorithm for doing reverse playback of theora video.
28062 Original commit message from CVS:
28063 * ext/theora/gsttheoradec.h:
28064 * ext/theora/theoradec.c: (gst_theora_dec_init),
28065 (theora_dec_sink_event), (theora_dec_chain_forward),
28066 (theora_dec_flush_decode), (theora_dec_chain_reverse),
28067 (theora_dec_chain):
28068 Document and partially implement an algorithm for doing reverse playback
28071 2006-11-02 14:18:45 +0000 Sergey Scobich <sergey.scobich@gmail.com>
28073 win32/: Misc. VS8 build fixes: fix syntax in config.h, add missing entries to libgsttag.def; add missing dependencies...
28074 Original commit message from CVS:
28075 Patch by: Sergey Scobich <sergey.scobich at gmail com>
28076 * win32/common/config.h:
28077 * win32/common/interfaces-enumtypes.c:
28078 * win32/common/libgsttag.def:
28079 * win32/vs8/gst-plugins-base.sln:
28080 * win32/vs8/libgstaudioresample.vcproj:
28081 * win32/vs8/libgstinterfaces.vcproj:
28082 * win32/vs8/libgstogg.vcproj:
28083 * win32/vs8/libgstriff.vcproj:
28084 * win32/vs8/libgsttag.vcproj:
28085 * win32/vs8/libgsttheora.vcproj:
28086 * win32/vs8/libgstvideoscale.vcproj:
28087 * win32/vs8/libgstvorbis.vcproj:
28088 Misc. VS8 build fixes: fix syntax in config.h, add missing entries
28089 to libgsttag.def; add missing dependencies for some vs8 projects;
28090 re-arrange placement of .def files in vs8 projects (#366334).
28092 2006-11-01 14:08:31 +0000 Tim-Philipp Müller <tim@centricular.net>
28094 ext/ogg/gstogg.c: Remove unused variable.
28095 Original commit message from CVS:
28096 * ext/ogg/gstogg.c:
28097 Remove unused variable.
28098 * ext/ogg/gstoggdemux.c:
28099 Fix Wim's surname in plugin description.
28101 2006-10-31 15:05:33 +0000 Wim Taymans <wim.taymans@gmail.com>
28103 gst-plugins-base.spec.in: spec new .h file. Fixes #368310.
28104 Original commit message from CVS:
28105 * gst-plugins-base.spec.in:
28106 spec new .h file. Fixes #368310.
28108 2006-10-31 14:19:07 +0000 Michael Smith <msmith@xiph.org>
28110 gst/tcp/gstmultifdsink.*: Make using the remove or clear signals threadsafe.
28111 Original commit message from CVS:
28112 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
28113 (gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
28114 (gst_multi_fd_sink_get_stats),
28115 (gst_multi_fd_sink_remove_client_link),
28116 (gst_multi_fd_sink_queue_buffer),
28117 (gst_multi_fd_sink_handle_clients):
28118 * gst/tcp/gstmultifdsink.h:
28119 Make using the remove or clear signals threadsafe.
28120 Make calling get-stats with an invalid fd not segfault.
28123 2006-10-31 10:49:19 +0000 Wim Taymans <wim.taymans@gmail.com>
28125 gst-libs/gst/rtp/: Fix and activate base audio payloader.
28126 Original commit message from CVS:
28127 * gst-libs/gst/rtp/Makefile.am:
28128 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
28129 (gst_base_rtp_audio_payload_init):
28130 Fix and activate base audio payloader.
28132 2006-10-28 17:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
28134 gst/typefind/gsttypefindfunctions.c: Add typefinder for QuickTime Image Files (see #366156).
28135 Original commit message from CVS:
28136 * gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
28138 Add typefinder for QuickTime Image Files (see #366156).
28140 2006-10-28 16:00:51 +0000 Tim-Philipp Müller <tim@centricular.net>
28142 gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
28143 Original commit message from CVS:
28144 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
28145 Another typo fix (#366212).
28147 2006-10-27 17:13:35 +0000 Wim Taymans <wim.taymans@gmail.com>
28149 gst/volume/gstvolume.c: Use stream time to synchronize volume property instead of rather random timestamps. This is n...
28150 Original commit message from CVS:
28151 * gst/volume/gstvolume.c: (volume_transform_ip):
28152 Use stream time to synchronize volume property instead of rather random
28153 timestamps. This is needed when gnonlin does its time shifting.
28155 2006-10-27 16:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
28158 I'm too lazy to comment this
28159 Original commit message from CVS:
28160 *** empty log message ***
28162 2006-10-27 16:45:30 +0000 Mark Nauwelaerts <manauw@skynet.be>
28164 ext/ogg/gstoggmux.c: Remove the pad from the element in release_pad.
28165 Original commit message from CVS:
28166 Patch by: Mark Nauwelaerts <manauw at skynet dot be>
28167 * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
28168 Remove the pad from the element in release_pad.
28170 2006-10-27 11:57:18 +0000 Tim-Philipp Müller <tim@centricular.net>
28172 sys/: Explicitly create our custom buffer classes at a thread-safe location as well, since g_type_class_ref() doesn't...
28173 Original commit message from CVS:
28174 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
28175 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
28176 Explicitly create our custom buffer classes at a thread-safe
28177 location as well, since g_type_class_ref() doesn't seem to be
28178 entirely thread-safe either (#365501; also see #349410).
28180 2006-10-26 10:49:00 +0000 Tim-Philipp Müller <tim@centricular.net>
28182 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...
28183 Original commit message from CVS:
28184 * gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
28185 (gst_riff_parse_info):
28186 If strings in INFO chunk are not UTF-8, do something similar to
28187 what we do for ID3v1 tags: check a number of environment variables
28188 (GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
28189 character sets to try, otherwise try the current locale and/or fall
28190 back on ISO-8859-1. Fixes #360552.
28192 2006-10-23 12:46:41 +0000 Tim-Philipp Müller <tim@centricular.net>
28194 gst/videotestsrc/: Add a bunch of exciting new checkers patterns.
28195 Original commit message from CVS:
28196 * gst/videotestsrc/gstvideotestsrc.c:
28197 (gst_video_test_src_pattern_get_type),
28198 (gst_video_test_src_set_pattern):
28199 * gst/videotestsrc/gstvideotestsrc.h:
28200 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_checkers1),
28201 (gst_video_test_src_checkers2), (gst_video_test_src_checkers4),
28202 (gst_video_test_src_checkers8):
28203 * gst/videotestsrc/videotestsrc.h:
28204 Add a bunch of exciting new checkers patterns.
28206 2006-10-23 12:06:44 +0000 Tim-Philipp Müller <tim@centricular.net>
28208 gst/subparse/: Add support for TMPlayer-type subtitles (#362845).
28209 Original commit message from CVS:
28210 * gst/subparse/Makefile.am:
28211 * gst/subparse/gstsubparse.c:
28212 (gst_sub_parse_data_format_autodetect),
28213 (gst_sub_parse_format_autodetect), (handle_buffer),
28214 (gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
28215 * gst/subparse/gstsubparse.h:
28216 * gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
28218 * gst/subparse/tmplayerparse.h:
28219 Add support for TMPlayer-type subtitles (#362845).
28220 * tests/check/elements/subparse.c: (test_tmplayer_do_test),
28221 (GST_START_TEST), (subparse_suite):
28222 Add some basic unit tests for the above.
28224 2006-10-23 11:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
28226 tests/check/elements/audiorate.c: More tests for audiorate: inject buffers to check behaviour when buffers overlap.
28227 Original commit message from CVS:
28228 * tests/check/elements/audiorate.c: (test_injector_base_init),
28229 (test_injector_class_init), (test_injector_chain),
28230 (test_injector_init), (probe_cb), (do_perfect_stream_test),
28231 (GST_START_TEST), (audiorate_suite):
28232 More tests for audiorate: inject buffers to check behaviour when
28235 2006-10-21 16:39:54 +0000 Tim-Philipp Müller <tim@centricular.net>
28237 tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...
28238 Original commit message from CVS:
28239 * tests/check/Makefile.am:
28240 * tests/check/elements/.cvsignore:
28241 * tests/check/elements/audiorate.c: (probe_cb), (got_buf),
28242 (do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
28243 Add some basic unit tests for audiorate. Disabled at the moment
28244 since it doesn't pass yet (see bug #363119).
28246 2006-10-20 17:02:19 +0000 Tim-Philipp Müller <tim@centricular.net>
28248 gst/subparse/gstsubparse.c: Add missing closing tags for markup and fix broken markup, otherwise pango won't render a...
28249 Original commit message from CVS:
28250 * gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
28251 (parse_subrip), (handle_buffer):
28252 Add missing closing tags for markup and fix broken markup,
28253 otherwise pango won't render anything (fixes #357531). Also,
28254 make sure the text we send out is always NUL-terminated
28255 (better safe than sorry etc.).
28256 * tests/check/elements/subparse.c: (test_srt_do_test),
28258 Some more tests for .srt incl. tests for the above stuff.
28260 2006-10-20 13:56:55 +0000 Stefan Kost <ensonic@users.sf.net>
28262 sys/: Try to redraw borders only when needed. Apparently this consumes resources on small devices... :-O (#363607)
28263 Original commit message from CVS:
28264 2006-10-20 Julien MOUTTE <julien@moutte.net>
28265 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
28266 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
28267 Patch by: Stefan Kost <ensonic@users.sf.net>
28268 Try to redraw borders only when needed. Apparently this consumes
28269 resources on small devices... :-O (#363607)
28271 2006-10-20 13:54:19 +0000 Michael Smith <msmith@xiph.org>
28273 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...
28274 Original commit message from CVS:
28275 * gst/tcp/gstmultifdsink.c:
28276 (gst_multi_fd_sink_client_queue_buffer):
28277 If caps change, then update the client's idea of the caps so that we
28278 don't end up re-sending streamheaders for every single buffer after
28281 2006-10-20 12:31:02 +0000 Michael Smith <msmith@xiph.org>
28283 ext/ogg/gstoggparse.c: Set caps on pushed buffers; fix up refcounting of caps objects.
28284 Original commit message from CVS:
28285 * ext/ogg/gstoggparse.c: (gst_ogg_parse_dispose),
28286 (gst_ogg_parse_append_header), (gst_ogg_parse_chain):
28287 Set caps on pushed buffers; fix up refcounting of caps objects.
28289 2006-10-19 14:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
28291 gst/typefind/gsttypefindfunctions.c: Typefind mmsh header data packet to application/x-mmsh (#362625).
28292 Original commit message from CVS:
28293 * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
28295 Typefind mmsh header data packet to application/x-mmsh (#362625).
28297 2006-10-19 09:17:48 +0000 Tim-Philipp Müller <tim@centricular.net>
28299 tests/check/: Add very simple unit test for subparse.
28300 Original commit message from CVS:
28301 * tests/check/Makefile.am:
28302 * tests/check/elements/.cvsignore:
28303 * tests/check/elements/subparse.c: (buffer_from_static_string),
28304 (setup_subparse), (teardown_subparse), (test_srt_do_test),
28305 (GST_START_TEST), (subparse_suite):
28306 Add very simple unit test for subparse.
28308 2006-10-19 09:00:21 +0000 Tim-Philipp Müller <tim@centricular.net>
28310 gst/subparse/gstsubparse.c: Strip trailing newlines from subtitle text output.
28311 Original commit message from CVS:
28312 * gst/subparse/gstsubparse.c: (strip_trailing_newlines),
28314 Strip trailing newlines from subtitle text output.
28316 2006-10-18 18:40:12 +0000 Tim-Philipp Müller <tim@centricular.net>
28318 gst/subparse/gstsubparse.c: Fix memleak; clear subparse->textbuf n state change function.
28319 Original commit message from CVS:
28320 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
28321 (gst_sub_parse_change_state):
28322 Fix memleak; clear subparse->textbuf n state change function.
28324 2006-10-18 15:13:59 +0000 Tim-Philipp Müller <tim@centricular.net>
28326 gst/subparse/gstsubparse.c: Don't require subrip (.srt) files to start with a chunk number of 1.
28327 Original commit message from CVS:
28328 * gst/subparse/gstsubparse.c:
28329 (gst_sub_parse_data_format_autodetect):
28330 Don't require subrip (.srt) files to start with a chunk number of 1.
28332 2006-10-18 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
28334 gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
28335 Original commit message from CVS:
28336 * gst-libs/gst/audio/gstbaseaudiosink.c:
28337 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
28338 * gst-libs/gst/audio/gstbaseaudiosink.h:
28339 Extract rate from the NEWSEGMENT event.
28340 Use commit_full to also take rate adjustment into account when writing
28341 samples to the ringbuffer.
28342 * gst-libs/gst/audio/gstringbuffer.c:
28343 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
28344 (gst_ring_buffer_read):
28345 * gst-libs/gst/audio/gstringbuffer.h:
28346 Added _commit_full() to also take rate into account.
28347 Use simple interpolation algorithm to resample audio.
28348 API: gst_ring_buffer_commit_full()
28349 * tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
28350 * tests/examples/seek/seek.c: (segment_done):
28351 Don't try to seek with 0.0 rate, just pause instead.
28352 Remove bogus debug line.
28354 2006-10-18 12:57:54 +0000 Tim-Philipp Müller <tim@centricular.net>
28356 gst/playback/gstplaybasebin.c: Catch async errors when starting up the subtitle bin, so we can stop waiting and conti...
28357 Original commit message from CVS:
28358 * gst/playback/gstplaybasebin.c: (subbin_startup_sync_msg),
28360 Catch async errors when starting up the subtitle bin, so we can
28361 stop waiting and continue with the main film instead of hanging
28362 forever. Fixes #339366.
28363 * tests/check/elements/playbin.c: (playbin_suite):
28364 Enable unit test for the above.
28366 2006-10-18 09:53:03 +0000 Tim-Philipp Müller <tim@centricular.net>
28368 tests/check/: Some small and basic unit tests for playbin; not very useful yet, but at least a start.
28369 Original commit message from CVS:
28370 * tests/check/Makefile.am:
28371 * tests/check/elements/.cvsignore:
28372 * tests/check/elements/playbin.c: (GST_START_TEST),
28373 (gst_red_video_src_uri_get_type),
28374 (gst_red_video_src_uri_get_protocols),
28375 (gst_red_video_src_uri_get_uri), (gst_red_video_src_uri_set_uri),
28376 (gst_red_video_src_uri_handler_init),
28377 (gst_red_video_src_init_type), (gst_red_video_src_base_init),
28378 (gst_red_video_src_create), (gst_red_video_src_class_init),
28379 (gst_red_video_src_init), (plugin_init), (playbin_suite):
28380 Some small and basic unit tests for playbin; not very useful yet,
28381 but at least a start.
28383 2006-10-18 09:46:35 +0000 Tim-Philipp Müller <tim@centricular.net>
28385 gst/playback/gstplaybin.c: The old pad activation spiel.
28386 Original commit message from CVS:
28387 * gst/playback/gstplaybin.c: (setup_sinks):
28388 The old pad activation spiel.
28390 2006-10-18 09:31:49 +0000 Tim-Philipp Müller <tim@centricular.net>
28392 gst/playback/gstplaybasebin.c: Don't hang forever if the subbin already fails to start up in the state change to PAUS...
28393 Original commit message from CVS:
28394 * gst/playback/gstplaybasebin.c: (setup_source):
28395 Don't hang forever if the subbin already fails to start up in
28396 the state change to PAUSED (#339366).
28398 2006-10-17 17:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
28400 gst-libs/gst/interfaces/tuner.c: Fix some function guards, add some more function guards.
28401 Original commit message from CVS:
28402 * gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
28403 (gst_tuner_set_channel), (gst_tuner_get_channel),
28404 (gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
28405 (gst_tuner_set_frequency), (gst_tuner_get_frequency),
28406 (gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
28407 (gst_tuner_find_channel_by_name):
28408 Fix some function guards, add some more function guards.
28410 2006-10-17 11:34:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28412 gst/playback/gstdecodebin.c: Don't return a pad from get_our_ghost_pad unless it is actually the one we want.
28413 Original commit message from CVS:
28414 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
28415 (remove_element_chain):
28416 Don't return a pad from get_our_ghost_pad unless it is actually the
28418 Change a cast in remove_element_chain slightly.
28420 2006-10-13 15:20:29 +0000 Julien Moutte <julien@moutte.net>
28422 tests/examples/seek/seek.c: Segment seeking needs to use the rate and set stop to -1.
28423 Original commit message from CVS:
28424 2006-10-13 Julien MOUTTE <julien@moutte.net>
28425 * tests/examples/seek/seek.c: (do_seek), (start_seek),
28426 (rate_spinbutton_changed_cb), (segment_done),
28427 (msg_state_changed):
28428 Segment seeking needs to use the rate and set stop to -1.
28430 2006-10-13 14:15:42 +0000 Ville Syrjala <ville.syrjala@movial.fi>
28432 gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
28433 Original commit message from CVS:
28434 * gst-libs/gst/audio/gstbaseaudiosink.c:
28435 (gst_base_audio_sink_setcaps):
28436 Don't crash when ringbuffer is not yet created.
28437 Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
28439 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
28440 * gst/playback/gststreamselector.c:
28441 (gst_stream_selector_request_new_pad):
28442 Activate pads befre adding them to running elements.
28444 2006-10-13 11:25:10 +0000 Julien Moutte <julien@moutte.net>
28446 tests/examples/seek/seek.c: Stop the scale updater when we start grabing the slider. Don't wait for the pipeline to b...
28447 Original commit message from CVS:
28448 2006-10-13 Julien MOUTTE <julien@moutte.net>
28449 * tests/examples/seek/seek.c: (do_seek), (start_seek),
28450 (rate_spinbutton_changed_cb), (msg_state_changed): Stop the
28452 updater when we start grabing the slider. Don't wait for the
28453 pipeline to be PAUSED.
28455 2006-10-13 08:57:52 +0000 Tim-Philipp Müller <tim@centricular.net>
28457 gst-libs/gst/interfaces/mixer.c: Guard mixer interface functions against bogus arguments.
28458 Original commit message from CVS:
28459 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
28460 (gst_mixer_set_volume), (gst_mixer_get_volume),
28461 (gst_mixer_set_mute), (gst_mixer_set_option),
28462 (gst_mixer_get_option), (gst_mixer_mute_toggled),
28463 (gst_mixer_record_toggled), (gst_mixer_volume_changed),
28464 (gst_mixer_option_changed):
28465 Guard mixer interface functions against bogus arguments.
28467 2006-10-12 19:39:07 +0000 Julien Moutte <julien@moutte.net>
28469 tests/examples/seek/seek.c: Use state-changed messages to trigger start/stop of scale update timer. Indeed the scale ...
28470 Original commit message from CVS:
28471 2006-10-12 Julien MOUTTE <julien@moutte.net>
28472 * tests/examples/seek/seek.c: (do_seek), (start_seek),
28474 (play_cb), (pause_cb), (stop_cb),
28475 (rate_spinbutton_changed_cb),
28476 (msg_state_changed), (main): Use state-changed messages to
28478 start/stop of scale update timer. Indeed the scale slider was
28479 jumping here and there because the update timer was activated
28480 before seek completed. This fixes instant applying of rate
28482 by pressing the spinbutton like a crazy man !
28484 2006-10-12 19:09:06 +0000 Sebastien Cote <sebas642@yahoo.ca>
28486 gst-libs/gst/rtp/gstbasertppayload.c: Fix two small memory leaks (#361456).
28487 Original commit message from CVS:
28488 Patch by: Sebastien Cote <sebas642 at yahoo.ca>
28489 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
28490 (gst_basertppayload_finalize):
28491 Fix two small memory leaks (#361456).
28493 2006-10-10 18:56:01 +0000 Julien Moutte <julien@moutte.net>
28495 tests/examples/seek/seek.c: When changing spinbutton we try to change the rate on the fly.
28496 Original commit message from CVS:
28497 2006-10-10 Julien MOUTTE <julien@moutte.net>
28498 * tests/examples/seek/seek.c: (do_seek),
28499 (rate_spinbutton_changed_cb): When changing spinbutton we try
28500 to change the rate on the fly.
28502 2006-10-10 16:50:06 +0000 Wim Taymans <wim.taymans@gmail.com>
28504 gst-libs/gst/riff/: Add WMS caps.
28505 Original commit message from CVS:
28506 * gst-libs/gst/riff/riff-ids.h:
28507 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
28508 (gst_riff_create_audio_template_caps):
28511 2006-10-10 12:49:03 +0000 Josep Torre Valles <josep@fluendo.com>
28513 ext/gnomevfs/: Fix URI interface implementation return type.
28514 Original commit message from CVS:
28515 2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
28516 Patch by: Josep Torre Valles <josep@fluendo.com>
28517 * ext/gnomevfs/gstgnomevfssink.c:
28518 * ext/gnomevfs/gstgnomevfssrc.c:
28519 Fix URI interface implementation return type.
28520 * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
28521 Fix what looks like a copy/paste issue when assigning values.
28522 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
28523 (gst_audio_filter_template_get_type):
28524 Cast to prevent Forte warnings.
28525 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
28526 Fix URI interface implementation return type.
28527 gst_pad_query_position requires a signed integer pointer as
28528 3rd parameter, GstClockTime is unsigned.
28529 * gst/audioconvert/audioconvert.c:
28530 Fix integer overflow when treated as signed.
28531 * gst/audioresample/resample.c: (resample_add_input_data):
28532 Cast to prevent warnings on Forte.
28533 * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
28534 Fix integer overflow when treated as signed.
28535 * gst/ffmpegcolorspace/imgconvert_template.h:
28536 Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
28537 * gst/playback/gstdecodebin.c: (queue_filled_cb),
28538 (cleanup_decodebin):
28539 Who initialises a guint to -1!
28540 Cast function pointers to prevent warnings on Forte.
28541 * gst/playback/gstplaybasebin.c: (queue_deadlock_check),
28542 (queue_threshold_reached):
28543 Cast function pointers correctly to prevent warnings on Forte.
28544 * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
28545 Cast function pointers correctly to prevent warnings on Forte.
28546 * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
28547 Obvious change to unsigned, 0xEF > max signed char.
28548 * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
28549 GstClockTime is unsigned, initialise correctly.
28550 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
28551 Cast so pointer arithemetic doesn't cause warnings on Forte.
28552 * gst/videorate/gstvideorate.c:
28553 Use correct return value.
28554 * tests/examples/seek/scrubby.c:
28555 GstClockTime is unsigned, initialise correctly.
28557 2006-10-10 11:20:03 +0000 Ferenc Gerlits <fgerlits@gmail.com>
28559 gst/typefind/gsttypefindfunctions.c: Recognise XML files and XML-like files shorter than 256 bytes as well (fixes #35...
28560 Original commit message from CVS:
28561 Patch by: Ferenc Gerlits <fgerlits at gmail com>
28562 * gst/typefind/gsttypefindfunctions.c:
28563 Recognise XML files and XML-like files shorter than 256 bytes as
28564 well (fixes #359237).
28566 2006-10-09 15:01:30 +0000 Edgard Lima <edgard.lima@indt.org.br>
28570 * gst/typefind/gsttypefindfunctions.c:
28571 Added typefind functions to video/x-nuv media.
28572 Original commit message from CVS:
28573 Added typefind functions to video/x-nuv media.
28575 2006-10-08 16:59:31 +0000 Tim-Philipp Müller <tim@centricular.net>
28577 gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
28578 Original commit message from CVS:
28579 * gst-libs/gst/interfaces/xoverlay.c:
28580 (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
28581 Some more guards against invalid input.
28583 2006-10-07 18:35:39 +0000 Julien Moutte <julien@moutte.net>
28585 ext/pango/gsttextoverlay.c: Useless goto.
28586 Original commit message from CVS:
28587 2006-10-07 Julien MOUTTE <julien@moutte.net>
28588 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
28590 * tests/examples/seek/seek.c: (do_seek),
28591 (rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
28592 seek example to experiment with rates != 1.0 (reverse playback
28595 2006-10-06 19:20:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28597 gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
28598 Original commit message from CVS:
28599 * gst-libs/gst/interfaces/xoverlay.c:
28600 Unref message in doc-example (spotted by Robert McQueen)
28602 2006-10-06 17:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
28604 gst/typefind/gsttypefindfunctions.c: printf fix.
28605 Original commit message from CVS:
28606 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
28607 (mpeg1_parse_header), (mpeg1_sys_type_find):
28610 2006-10-06 14:37:43 +0000 Wim Taymans <wim.taymans@gmail.com>
28612 gst/playback/: Activate dynamic pads before adding them to the element.
28613 Original commit message from CVS:
28614 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
28616 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
28617 Activate dynamic pads before adding them to the element.
28619 2006-10-06 14:04:53 +0000 Michael Smith <msmith@xiph.org>
28621 gst-libs/gst/floatcast/floatcast.h: Fix obviously-bogus macros; use the correct types.
28622 Original commit message from CVS:
28623 * gst-libs/gst/floatcast/floatcast.h:
28624 Fix obviously-bogus macros; use the correct types.
28626 2006-10-06 13:34:46 +0000 Wim Taymans <wim.taymans@gmail.com>
28628 gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
28629 Original commit message from CVS:
28630 * gst-libs/gst/rtp/gstbasertpdepayload.c:
28631 (gst_base_rtp_depayload_change_state):
28632 Also call parent state change function to activate pads.
28633 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
28634 (mpeg1_parse_header), (mpeg1_sys_type_find):
28635 Add some more debug info in mpeg typefinding.
28637 2006-10-06 12:57:10 +0000 Michael Smith <msmith@xiph.org>
28639 ext/theora/theoradec.c: Zero byte theora packets are valid and well-defined; don't warn on them.
28640 Original commit message from CVS:
28641 * ext/theora/theoradec.c: (theora_dec_chain):
28642 Zero byte theora packets are valid and well-defined; don't warn on
28645 2006-10-06 10:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28647 gst/tcp/gstmultifdsink.c: API: add dropped_buffers to the get-stats GValueArray
28648 Original commit message from CVS:
28649 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
28650 (gst_multi_fd_sink_get_stats), (find_limits),
28651 (gst_multi_fd_sink_queue_buffer):
28652 API: add dropped_buffers to the get-stats GValueArray
28654 2006-10-05 15:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
28656 Printf format fixes.
28657 Original commit message from CVS:
28658 * ext/alsa/gstalsadeviceprobe.c:
28659 (gst_alsa_device_property_probe_get_values):
28660 * ext/alsa/gstalsasink.c: (set_hwparams):
28661 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
28662 (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
28663 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
28664 (gst_ogg_mux_process_best_pad):
28665 * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
28666 (gst_ogg_parse_chain):
28667 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
28668 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
28669 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
28670 (gst_vorbis_enc_buffer_check_discontinuous):
28671 * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
28672 * gst-libs/gst/audio/gstbaseaudiosink.c:
28673 (gst_base_audio_sink_render):
28674 * gst-libs/gst/cdda/gstcddabasesrc.c:
28675 (gst_cdda_base_src_handle_track_seek):
28676 * gst-libs/gst/rtp/gstbasertpdepayload.c:
28677 (gst_base_rtp_depayload_push_full):
28678 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
28679 * gst/audioresample/resample.c: (resample_input_pushthrough):
28680 * gst/playback/gstplaybasebin.c: (queue_out_of_data):
28681 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
28682 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
28683 (wavpack_type_find):
28684 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
28685 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
28686 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
28687 * tests/check/elements/volume.c: (GST_START_TEST):
28688 Printf format fixes.
28690 2006-10-04 13:18:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28692 gst/tcp/gsttcp.c: Fix a simple mistake (see the docs)
28693 Original commit message from CVS:
28694 * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
28695 Fix a simple mistake (see the docs)
28698 2006-10-04 13:15:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28700 * win32/common/config.h:
28702 Original commit message from CVS:
28705 2006-10-03 12:11:45 +0000 Tim-Philipp Müller <tim@centricular.net>
28707 docs/plugins/: Add vorbistag element to docs; update version numbers to 0.10.10.1.
28708 Original commit message from CVS:
28709 * docs/plugins/Makefile.am:
28710 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
28711 * docs/plugins/gst-plugins-base-plugins-sections.txt:
28712 * docs/plugins/gst-plugins-base-plugins.args:
28713 * docs/plugins/gst-plugins-base-plugins.hierarchy:
28714 * docs/plugins/inspect/plugin-adder.xml:
28715 * docs/plugins/inspect/plugin-alsa.xml:
28716 * docs/plugins/inspect/plugin-audioconvert.xml:
28717 * docs/plugins/inspect/plugin-audiorate.xml:
28718 * docs/plugins/inspect/plugin-audioresample.xml:
28719 * docs/plugins/inspect/plugin-audiotestsrc.xml:
28720 * docs/plugins/inspect/plugin-cdparanoia.xml:
28721 * docs/plugins/inspect/plugin-decodebin.xml:
28722 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
28723 * docs/plugins/inspect/plugin-gdp.xml:
28724 * docs/plugins/inspect/plugin-gnomevfs.xml:
28725 * docs/plugins/inspect/plugin-libvisual.xml:
28726 * docs/plugins/inspect/plugin-ogg.xml:
28727 * docs/plugins/inspect/plugin-pango.xml:
28728 * docs/plugins/inspect/plugin-playbin.xml:
28729 * docs/plugins/inspect/plugin-subparse.xml:
28730 * docs/plugins/inspect/plugin-tcp.xml:
28731 * docs/plugins/inspect/plugin-theora.xml:
28732 * docs/plugins/inspect/plugin-typefindfunctions.xml:
28733 * docs/plugins/inspect/plugin-video4linux.xml:
28734 * docs/plugins/inspect/plugin-videorate.xml:
28735 * docs/plugins/inspect/plugin-videoscale.xml:
28736 * docs/plugins/inspect/plugin-videotestsrc.xml:
28737 * docs/plugins/inspect/plugin-volume.xml:
28738 * docs/plugins/inspect/plugin-vorbis.xml:
28739 * docs/plugins/inspect/plugin-ximagesink.xml:
28740 * docs/plugins/inspect/plugin-xvimagesink.xml:
28741 Add vorbistag element to docs; update version numbers to 0.10.10.1.
28743 2006-10-03 11:51:48 +0000 James Doc Livingston <doclivingston@gmail.com>
28745 ext/vorbis/: Add new vorbistag element which derives from vorbisparse and is essentially the same as well, only that ...
28746 Original commit message from CVS:
28747 Patch by: James "Doc" Livingston <doclivingston at gmail com>
28748 * ext/vorbis/Makefile.am:
28749 * ext/vorbis/vorbis.c: (plugin_init):
28750 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_class_init),
28751 (vorbis_parse_parse_packet), (vorbis_parse_chain):
28752 * ext/vorbis/vorbisparse.h:
28753 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_base_init),
28754 (gst_vorbis_tag_class_init), (gst_vorbis_tag_init),
28755 (gst_vorbis_tag_parse_packet):
28756 * ext/vorbis/vorbistag.h:
28757 Add new vorbistag element which derives from vorbisparse
28758 and is essentially the same as well, only that it implements
28759 the GstTagSetter interface and can modify the stream's
28760 vorbiscomment on the fly (#335635).
28761 * tests/check/Makefile.am:
28762 * tests/check/elements/.cvsignore:
28763 * tests/check/elements/vorbistag.c: (setup_vorbistag),
28764 (cleanup_vorbistag), (buffer_probe), (start_pipeline),
28765 (get_buffer), (stop_pipeline), (_create_codebook_header_buffer),
28766 (_create_audio_buffer), (GST_START_TEST), (vorbistag_suite):
28767 Add unit test for new vorbistag element.
28769 2006-10-03 10:36:38 +0000 Tim-Philipp Müller <tim@centricular.net>
28771 ext/vorbis/vorbisparse.c: Set BOS flag in packet structure to fix 'jump depends on unitialized value' errors in valgr...
28772 Original commit message from CVS:
28773 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init),
28774 (vorbis_parse_push_headers), (vorbis_parse_chain):
28775 Set BOS flag in packet structure to fix 'jump depends
28776 on unitialized value' errors in valgrind; various minor
28779 2006-09-30 15:30:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28781 gst/playback/gstdecodebin.c: Fix typo in a debug statement.
28782 Original commit message from CVS:
28783 * gst/playback/gstdecodebin.c: (close_pad_link):
28784 Fix typo in a debug statement.
28785 * gst/playback/gstplaybasebin.c: (probe_triggered),
28786 (new_decoded_pad_full), (new_decoded_pad), (subs_new_decoded_pad),
28787 (gen_source_element), (source_new_pad), (analyse_source),
28789 When handling no_more_pads in new_decoded_pad, make sure to treat
28790 subtitle pads correctly. Fixes playback with subtitle files.
28791 Move a recurring message to LOG level.
28792 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
28793 The maximum value for the Xv colorkey on this Radeon is 0xFFFFFFFF,
28794 which ends up as -1 when cast to an int. Make the logic handle the
28795 max value as an unsigned mask and only change the colorkey when it's
28796 a value we recognise.
28798 2006-09-30 00:14:20 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28800 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
28801 Original commit message from CVS:
28802 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
28803 Removed empty * between paragraphs
28805 2006-09-29 23:50:53 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28807 gst-libs/gst/rtp/: Moved some documentation into .c file
28808 Original commit message from CVS:
28809 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
28810 * gst-libs/gst/rtp/README:
28811 Moved some documentation into .c file
28813 2006-09-29 17:35:01 +0000 Wim Taymans <wim.taymans@gmail.com>
28815 gst/playback/gstdecodebin.c: Fix compilation.
28816 Original commit message from CVS:
28817 * gst/playback/gstdecodebin.c: (no_more_pads):
28820 2006-09-29 16:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
28822 gst/playback/gstdecodebin.c: Remove g_print
28823 Original commit message from CVS:
28824 * gst/playback/gstdecodebin.c: (new_caps):
28826 * gst/playback/gstplaybin.c:
28829 2006-09-29 15:16:32 +0000 Tim-Philipp Müller <tim@centricular.net>
28831 tests/check/Makefile.am: Re-enable cddabasesrc test to see if it works again now.
28832 Original commit message from CVS:
28833 * tests/check/Makefile.am:
28834 Re-enable cddabasesrc test to see if it works again
28837 2006-09-29 13:46:45 +0000 Wim Taymans <wim.taymans@gmail.com>
28839 gst/playback/gstplaybasebin.c: Handle invalid URIs a bit more gracefully.
28840 Original commit message from CVS:
28841 * gst/playback/gstplaybasebin.c: (setup_subtitle),
28842 (gen_source_element):
28843 Handle invalid URIs a bit more gracefully.
28845 2006-09-29 12:54:28 +0000 Tim-Philipp Müller <tim@centricular.net>
28847 tests/check/pipelines/oggmux.c: Remove obsolete comment.
28848 Original commit message from CVS:
28849 * tests/check/pipelines/oggmux.c:
28850 Remove obsolete comment.
28852 2006-09-29 10:43:05 +0000 James Doc Livingston <doclivingston@gmail.com>
28854 ext/ogg/gstoggmux.c: Commit patch from James "Doc" Livingston, adds proper EOS handling in oggmux. GStreamer can, for...
28855 Original commit message from CVS:
28856 * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
28857 (gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
28858 (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
28859 (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
28860 (gst_ogg_mux_collected):
28861 Commit patch from James "Doc" Livingston, adds proper EOS handling
28862 in oggmux. GStreamer can, for the first time ever, create a valid
28864 * tests/check/pipelines/oggmux.c: (check_chain_final_state),
28866 Reenable tests now that they pass.
28868 2006-09-29 08:20:03 +0000 Wim Taymans <wim.taymans@gmail.com>
28870 gst/tcp/gstmultifdsink.c: Stop reading commands when EOF (we read 0) as well.
28871 Original commit message from CVS:
28872 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
28873 Stop reading commands when EOF (we read 0) as well.
28875 2006-09-28 15:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
28877 gst/playback/gstdecodebin.c: Implement delayed caps linking needed for element with a lot of different caps on the sr...
28878 Original commit message from CVS:
28879 * gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
28880 (close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
28881 (find_dynamic), (unlinked), (close_link):
28882 Implement delayed caps linking needed for element with a lot of
28883 different caps on the src pads that get fixed at runtime.
28884 Improve management of dynamic elements.
28885 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
28886 (group_destroy), (group_commit), (check_queue), (queue_overrun),
28887 (gen_preroll_element), (remove_groups), (unknown_type),
28888 (add_element_stream), (no_more_pads_full), (no_more_pads),
28889 (sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
28890 (new_decoded_pad), (setup_subtitle), (array_has_value),
28891 (gen_source_element), (source_new_pad), (has_all_raw_caps),
28892 (analyse_source), (remove_decoders), (make_decoder),
28893 (remove_source), (setup_source), (finish_source), (prepare_output),
28894 (gst_play_base_bin_change_state):
28895 * gst/playback/gstplaybasebin.h:
28896 Use more _CAST instead of full type checking casts.
28897 Small cleanups, plug some leaks.
28898 Handle dynamic sources.
28899 Add some helper functions to create lists of strings used for
28900 blacklisting and other stuff.
28901 Refactor some code dealing with analysing the source.
28902 Re-enable sources without pads (like cd:// or other selfcontained
28905 2006-09-28 15:08:15 +0000 Wim Taymans <wim.taymans@gmail.com>
28907 gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
28908 Original commit message from CVS:
28909 * gst-libs/gst/audio/gstbaseaudiosink.c:
28910 (gst_base_audio_sink_render):
28911 When we have a timestamp, we can still perform clipping.
28912 When we have no clock, we must play the sample ASAP.
28914 2006-09-28 11:46:26 +0000 Wim Taymans <wim.taymans@gmail.com>
28916 gst/audiorate/gstaudiorate.c: Set caps on outgoing buffers.
28917 Original commit message from CVS:
28918 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
28919 Set caps on outgoing buffers.
28920 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
28921 (gst_video_rate_event), (gst_video_rate_chain):
28922 * gst/videorate/gstvideorate.h:
28923 Fix videorate some more. Fixes #357977
28925 2006-09-28 11:34:05 +0000 Tim-Philipp Müller <tim@centricular.net>
28927 tests/check/elements/adder.c: Don't set timeout to 6 seconds when we're running in valgrind ... (and how is 6 seconds...
28928 Original commit message from CVS:
28929 * tests/check/elements/adder.c: (adder_suite):
28930 Don't set timeout to 6 seconds when we're running
28931 in valgrind ... (and how is 6 seconds longer than
28932 the default anyway?)
28934 2006-09-28 10:49:56 +0000 Wim Taymans <wim.taymans@gmail.com>
28936 gst/audiorate/gstaudiorate.c: Keep sink and src segment to keep track of time and support more input formats.
28937 Original commit message from CVS:
28938 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
28939 (gst_audio_rate_sink_event), (gst_audio_rate_convert),
28940 (gst_audio_rate_convert_segments), (gst_audio_rate_chain):
28941 Keep sink and src segment to keep track of time and support more
28943 Fix bogus next_offset and run_time calculation, don't understand how
28944 this could have worked before. Fixes #357976.
28945 Remove some unneeded vars.
28947 2006-09-28 09:41:20 +0000 Tim-Philipp Müller <tim@centricular.net>
28949 gst/playback/gstplaybin.c: Only remove visualisation from visbin if there is a visbin (or: don't throw warnings when ...
28950 Original commit message from CVS:
28951 * gst/playback/gstplaybin.c: (remove_sinks):
28952 Only remove visualisation from visbin if there is a visbin (or:
28953 don't throw warnings when closing totem without playing a file).
28955 2006-09-27 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
28957 gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
28958 Original commit message from CVS:
28959 * gst-libs/gst/audio/gstbaseaudiosink.c:
28960 (gst_base_audio_sink_render):
28961 Add some more info in a WARNING.
28962 * gst-libs/gst/audio/gstbaseaudiosrc.c:
28963 (gst_base_audio_src_create):
28964 Handle PAUSE in create function, use new -core addition to
28965 wait for playing. Fixes pausing and resuming capture from an
28967 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
28968 (gst_ring_buffer_read):
28969 Constify some more.
28970 Caller supports interrupted reads now.
28972 2006-09-27 13:29:49 +0000 Christian Schaller <uraeus@gnome.org>
28974 * gst-plugins-base.spec.in:
28975 add new header file to spec
28976 Original commit message from CVS:
28977 add new header file to spec
28979 2006-09-27 12:55:45 +0000 Tim-Philipp Müller <tim@centricular.net>
28981 tests/check/Makefile.am: Another attempt to make the gen64 buildbot happy.
28982 Original commit message from CVS:
28983 * tests/check/Makefile.am:
28984 Another attempt to make the gen64 buildbot happy.
28986 2006-09-27 11:58:17 +0000 Jonathan Matthew <jonathan@kaolin.wh9.net>
28988 ext/libvisual/visual.c: Libvisual plugin was not passing audio data to libvisual 0.4.0 correctly. Fixes #357800
28989 Original commit message from CVS:
28990 Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
28991 * ext/libvisual/visual.c: (gst_visual_clear_actors),
28992 (gst_visual_chain), (gst_visual_change_state):
28993 Libvisual plugin was not passing audio data to libvisual 0.4.0
28994 correctly. Fixes #357800
28996 2006-09-27 11:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
28998 tests/check/pipelines/simple-launch-lines.c: Add timeout to _get_state() so we see which pipeline it is that causes t...
28999 Original commit message from CVS:
29000 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
29001 Add timeout to _get_state() so we see which pipeline it is
29002 that causes trouble on the gen64 build bot.
29004 2006-09-27 11:06:54 +0000 Wim Taymans <wim.taymans@gmail.com>
29006 gst-libs/gst/rtp/gstbasertpdepayload.c: the source pad always uses fixed caps.
29007 Original commit message from CVS:
29008 * gst-libs/gst/rtp/gstbasertpdepayload.c:
29009 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
29010 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
29011 (gst_base_rtp_depayload_set_gst_timestamp):
29012 the source pad always uses fixed caps.
29014 2006-09-27 11:05:08 +0000 Wim Taymans <wim.taymans@gmail.com>
29016 Added docs for the audio libs.
29017 Original commit message from CVS:
29018 * docs/libs/gst-plugins-base-libs-docs.sgml:
29019 * docs/libs/gst-plugins-base-libs-sections.txt:
29020 * gst-libs/gst/audio/gstaudioclock.c:
29021 * gst-libs/gst/audio/gstaudioclock.h:
29022 * gst-libs/gst/audio/gstaudiosink.c:
29023 * gst-libs/gst/audio/gstaudiosink.h:
29024 * gst-libs/gst/audio/gstaudiosrc.c:
29025 * gst-libs/gst/audio/gstbaseaudiosink.c:
29026 (gst_base_audio_sink_render):
29027 * gst-libs/gst/audio/gstbaseaudiosink.h:
29028 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
29029 * gst-libs/gst/audio/gstbaseaudiosrc.h:
29030 * gst-libs/gst/audio/gstringbuffer.h:
29031 Added docs for the audio libs.
29033 2006-09-27 10:59:24 +0000 Tim-Philipp Müller <tim@centricular.net>
29035 tests/check/Makefile.am: Temporarily disable test that fails on the bots for unknown reasons.
29036 Original commit message from CVS:
29037 * tests/check/Makefile.am:
29038 Temporarily disable test that fails on the bots for unknown reasons.
29040 2006-09-27 00:13:29 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
29042 gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
29043 Original commit message from CVS:
29044 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
29045 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
29046 Moved AudioCodecType into priv
29047 Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
29049 2006-09-25 15:47:25 +0000 Wim Taymans <wim.taymans@gmail.com>
29051 gst/playback/gstdecodebin.c: Cleanups and small leak fixes.
29052 Original commit message from CVS:
29053 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
29054 (add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
29055 (is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
29057 Cleanups and small leak fixes.
29058 Added Depayloaders to valid list of autopluggable elements.
29060 2006-09-25 13:24:59 +0000 Wim Taymans <wim.taymans@gmail.com>
29062 gst/playback/gstplaybin.c: Detect NO_PREROLL state change returns and disable clock distribution to the sinks so that...
29063 Original commit message from CVS:
29064 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
29065 (gst_play_bin_vis_blocked), (gst_play_bin_set_property),
29066 (gen_video_element), (gen_text_element), (gen_audio_element),
29067 (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
29068 (gst_play_bin_set_clock_func), (gst_play_bin_change_state):
29069 Detect NO_PREROLL state change returns and disable clock distribution to
29070 the sinks so that sync is disabled.
29071 Avoid some type checking and do simple casts instead.
29072 Small cleanups, fix some FIXMEs.
29073 Be more robust when linking user specified elements, catch an report
29074 errors. Fixes #357404.
29075 Fix some leaks in the error paths.
29077 2006-09-25 12:55:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29080 ChangeLog surgery for missing bug-number
29081 Original commit message from CVS:
29082 ChangeLog surgery for missing bug-number
29084 2006-09-25 11:28:15 +0000 Peter Kjellerstedt <pkj@axis.com>
29086 gst/playback/test.c: Fix compilation with uClibc and -Werror (#357591).
29087 Original commit message from CVS:
29088 Patch by: Peter Kjellerstedt <pkj at axis com>
29089 * gst/playback/test.c:
29090 Fix compilation with uClibc and -Werror (#357591).
29092 2006-09-25 10:21:31 +0000 Tim-Philipp Müller <tim@centricular.net>
29094 gst-libs/gst/tag/gstvorbistag.c: Parse dates that are followed by a time as well (#357532).
29095 Original commit message from CVS:
29096 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
29097 Parse dates that are followed by a time as well (#357532).
29098 * tests/check/libs/tag.c: (test_vorbis_tags):
29099 Add unit test for this.
29101 2006-09-23 15:24:55 +0000 Tim-Philipp Müller <tim@centricular.net>
29103 gst/: A few array const-ifications.
29104 Original commit message from CVS:
29105 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
29106 (gst_audio_convert_transform_caps):
29107 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
29108 * gst/videotestsrc/videotestsrc.h:
29109 A few array const-ifications.
29111 2006-09-23 15:02:51 +0000 Tim-Philipp Müller <tim@centricular.net>
29113 tests/check/Makefile.am: See if this makes the build bots happy.
29114 Original commit message from CVS:
29115 * tests/check/Makefile.am:
29116 See if this makes the build bots happy.
29117 * tests/check/libs/cddabasesrc.c:
29120 2006-09-23 14:30:53 +0000 Young-Ho Cha <ganadist@chollian.net>
29122 gst/subparse/samiparse.c: More case-insensitivity for certain tags; recognise entities with decimal codes as special ...
29123 Original commit message from CVS:
29124 Patch by: Young-Ho Cha <ganadist at chollian dot net>
29125 * gst/subparse/samiparse.c: (handle_start_font),
29126 (fix_invalid_entities):
29127 More case-insensitivity for certain tags; recognise entities with
29128 decimal codes as special entities as well (#357330).
29130 2006-09-23 13:32:07 +0000 Tim-Philipp Müller <tim@centricular.net>
29132 gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
29133 Original commit message from CVS:
29134 * gst-libs/gst/Makefile.am:
29135 Need to build tag directory before cdda.
29137 2006-09-23 13:21:07 +0000 Tim-Philipp Müller <tim@centricular.net>
29139 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
29140 Original commit message from CVS:
29141 * docs/libs/gst-plugins-base-libs-sections.txt:
29142 * gst-libs/gst/cdda/Makefile.am:
29143 * gst-libs/gst/cdda/gstcddabasesrc.c:
29144 (gst_cdda_base_src_base_init):
29145 * gst-libs/gst/cdda/gstcddabasesrc.h:
29146 * gst-libs/gst/tag/tag.h:
29147 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
29148 (gst_tag_register_musicbrainz_tags):
29149 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
29150 depend on libgsttag. This is required so we can extract/read tags like
29151 DISCID without depending on libgstcddabasesrc (which used to register
29153 * gst-libs/gst/tag/gstvorbistag.c:
29154 Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
29155 tags (also see #347848).
29156 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
29157 Log vorbis comments we are actually writing. Const-ify array.
29159 2006-09-23 08:53:30 +0000 Wim Taymans <wim.taymans@gmail.com>
29161 gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
29162 Original commit message from CVS:
29163 * gst/playback/gstplaybasebin.c: (gen_preroll_element):
29164 Improve buffering a bit by avoiding a deadlock because we cannot assume
29165 the underrun is always called.
29167 2006-09-23 08:51:14 +0000 Young-Ho Cha <ganadist@chollian.net>
29169 gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
29170 Original commit message from CVS:
29171 Patch by: Young-Ho Cha <ganadist at chollian dot net>
29172 * gst-libs/gst/riff/riff-ids.h:
29173 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
29174 (gst_riff_create_audio_template_caps):
29175 Added MPEG-4 AAC and id and caps. Fixes #357289
29176 Added WMA9 Lossless id.
29178 2006-09-22 14:50:01 +0000 Tim-Philipp Müller <tim@centricular.net>
29180 ext/gnomevfs/gstgnomevfssrc.c: Fix misleading docs addition.
29181 Original commit message from CVS:
29182 * ext/gnomevfs/gstgnomevfssrc.c:
29183 Fix misleading docs addition.
29184 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
29185 Get rid of compiler warning the right way.
29187 2006-09-22 14:13:34 +0000 Wim Taymans <wim.taymans@gmail.com>
29189 gst-libs/gst/rtp/gstbasertpdepayload.*: Small cleanups.
29190 Original commit message from CVS:
29191 * gst-libs/gst/rtp/gstbasertpdepayload.c:
29192 (gst_base_rtp_depayload_finalize),
29193 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
29194 (gst_base_rtp_depayload_push_full),
29195 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
29196 (gst_base_rtp_depayload_process),
29197 (gst_base_rtp_depayload_set_gst_timestamp),
29198 (gst_base_rtp_depayload_queue_release):
29199 * gst-libs/gst/rtp/gstbasertpdepayload.h:
29202 Refactored the process method and added methods to push from the process
29204 Use _scale functions.
29205 API: gst_base_rtp_depayload_push_ts
29206 API: gst_base_rtp_depayload_push
29207 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
29208 timestamps are uint.
29210 2006-09-22 11:59:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29212 gst-libs/gst/interfaces/xoverlay.c: Remove unused statement from doc example.
29213 Original commit message from CVS:
29214 * gst-libs/gst/interfaces/xoverlay.c:
29215 Remove unused statement from doc example.
29217 2006-09-22 09:52:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29219 * gst/videorate/gstvideorate.c:
29221 Original commit message from CVS:
29224 2006-09-21 13:49:47 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29226 gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
29227 Original commit message from CVS:
29228 * gst-libs/gst/interfaces/videoorientation.c:
29229 (gst_video_orientation_iface_init),
29230 (gst_video_orientation_get_hflip),
29231 (gst_video_orientation_get_vflip),
29232 (gst_video_orientation_get_hcenter),
29233 (gst_video_orientation_get_vcenter),
29234 (gst_video_orientation_set_hflip),
29235 (gst_video_orientation_set_vflip),
29236 (gst_video_orientation_set_hcenter),
29237 (gst_video_orientation_set_vcenter):
29238 Add since tags to new API docs, ChangeLog surgery (forgot API keyword
29241 2006-09-21 09:27:47 +0000 Tim-Philipp Müller <tim@centricular.net>
29243 tests/check/: but disable for now since it doesn't pass (something wrong with
29244 Original commit message from CVS:
29245 * tests/check/Makefile.am:
29246 * tests/check/elements/.cvsignore:
29247 * tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
29248 (create_rgb_conversions), (rgb_conversion_free),
29249 (right_shift_colour), (fix_expected_colour), (check_rgb_buf),
29250 (got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
29251 Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
29252 but disable for now since it doesn't pass (something wrong with
29255 2006-09-21 07:01:48 +0000 Wim Taymans <wim.taymans@gmail.com>
29257 gst/playback/gstplaybasebin.c: Refactor handling of overrun detection.
29258 Original commit message from CVS:
29259 * gst/playback/gstplaybasebin.c: (group_commit),
29260 (queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
29261 (queue_out_of_data), (gen_preroll_element),
29262 (preroll_remove_overrun), (probe_triggered):
29263 Refactor handling of overrun detection.
29264 Separate handling of group completion and deadlock detection when doing
29265 network buffering. This should fix some deadlocks that were not detected
29266 because the group was completed.
29267 Add more comments, improve debugging.
29269 2006-09-21 05:31:00 +0000 Wim Taymans <wim.taymans@gmail.com>
29271 tests/check/: Some more compilation fixes.
29272 Original commit message from CVS:
29273 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
29274 * tests/check/libs/audio.c:
29275 Some more compilation fixes.
29277 2006-09-21 05:12:18 +0000 Wim Taymans <wim.taymans@gmail.com>
29279 gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
29280 Original commit message from CVS:
29281 * gst-libs/gst/audio/gstringbuffer.c:
29282 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
29283 (gst_ring_buffer_read):
29284 Early morning compilation fix.
29286 2006-09-20 18:09:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29290 Original commit message from CVS:
29293 2006-09-20 17:04:57 +0000 Wim Taymans <wim.taymans@gmail.com>
29295 tests/check/: Fix some warnings.
29296 Original commit message from CVS:
29297 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
29298 * tests/check/elements/multifdsink.c: (GST_START_TEST):
29299 * tests/check/elements/videorate.c: (GST_START_TEST):
29300 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
29301 * tests/check/pipelines/oggmux.c: (eos_buffer_probe):
29304 2006-09-20 10:59:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29306 sys/xvimage/xvimagesink.c: change colorkey behaviour back according to #354773 comment 6/7
29307 Original commit message from CVS:
29308 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
29309 (gst_xvimagesink_get_times):
29310 change colorkey behaviour back according to #354773 comment 6/7
29312 2006-09-20 10:42:34 +0000 Tim-Philipp Müller <tim@centricular.net>
29315 ChangeLog surgery: remove junk
29316 Original commit message from CVS:
29317 ChangeLog surgery: remove junk
29319 2006-09-19 11:31:06 +0000 Michael Smith <msmith@xiph.org>
29321 gst/tcp/gstmultifdsink.*: Implement stubbed out properties unit-type, units-soft-max, units-max, to allow specifying ...
29322 Original commit message from CVS:
29323 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
29324 (gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
29325 (gst_multi_fd_sink_recover_client),
29326 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
29327 (gst_multi_fd_sink_get_property):
29328 * gst/tcp/gstmultifdsink.h:
29329 Implement stubbed out properties unit-type, units-soft-max,
29330 units-max, to allow specifying maximum sizes in units other than
29334 2006-09-19 10:23:03 +0000 Wim Taymans <wim.taymans@gmail.com>
29336 gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
29337 Original commit message from CVS:
29338 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
29339 (gst_riff_create_audio_template_caps):
29340 Reorder the audio formats a bit for clarity.
29341 Detect and create caps for MSGSM and MSN (WAV49).
29343 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
29344 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
29345 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
29346 Small cleanups, move error handling out of normal flow for clarity.
29348 2006-09-18 15:59:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29350 Add new interface to control video orientation (fixes #354908)
29351 Original commit message from CVS:
29352 * docs/libs/gst-plugins-base-libs-docs.sgml:
29353 * docs/libs/gst-plugins-base-libs.types:
29354 * gst-libs/gst/interfaces/Makefile.am:
29355 * gst-libs/gst/interfaces/videoorientation.c:
29356 (gst_video_orientation_get_type),
29357 (gst_video_orientation_iface_init),
29358 (gst_video_orientation_get_hflip),
29359 (gst_video_orientation_get_vflip),
29360 (gst_video_orientation_get_hcenter),
29361 (gst_video_orientation_get_vcenter),
29362 (gst_video_orientation_set_hflip),
29363 (gst_video_orientation_set_vflip),
29364 (gst_video_orientation_set_hcenter),
29365 (gst_video_orientation_set_vcenter):
29366 * gst-libs/gst/interfaces/videoorientation.h:
29367 Add new interface to control video orientation (fixes #354908)
29369 2006-09-18 15:48:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29371 gst/videotestsrc/gstvideotestsrc.c: Use G_UNLIKELY in _create and log one more detail.
29372 Original commit message from CVS:
29373 * gst/videotestsrc/gstvideotestsrc.c:
29374 Use G_UNLIKELY in _create and log one more detail.
29375 (gst_video_test_src_get_times), (gst_video_test_src_create):
29376 * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
29377 Use gst_util_uint64_scale_int in _get_times().
29379 2006-09-18 15:00:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29381 sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
29382 Original commit message from CVS:
29383 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
29384 Give better warning message (add object and detail).
29386 2006-09-18 14:42:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29388 sys/xvimage/xvimagesink.c: xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes #354773), use gst_util...
29389 Original commit message from CVS:
29390 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
29391 (gst_xvimagesink_get_times):
29392 xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
29393 #354773), use gst_util_uint64_scale_int in _get_times()
29395 2006-09-18 14:21:45 +0000 Michael Smith <msmith@xiph.org>
29397 ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
29398 Original commit message from CVS:
29399 * ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
29400 Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
29401 always true, leading to dropping all timestamps.
29403 2006-09-18 11:40:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29405 ext/libvisual/visual.c: update to work also with libvisual 0.4 API
29406 Original commit message from CVS:
29407 * ext/libvisual/visual.c: (gst_vis_src_negotiate),
29408 (gst_visual_chain), (gst_visual_change_state):
29409 update to work also with libvisual 0.4 API
29410 * tools/gst-launch-ext.1.in:
29411 * tools/gst-visualise.1.in:
29412 remove references to old man-pages
29413 * tests/examples/seek/seek.c: (main):
29414 add real meadi-buttons, add tool-tips for the seek-options, arrange
29415 seek options in a table
29417 2006-09-18 10:57:28 +0000 Michael Smith <msmith@xiph.org>
29419 ext/ogg/gstoggmux.c: Don't generate out-of-order timestamps from oggmux, instead clamp output timestamps to be >= the...
29420 Original commit message from CVS:
29421 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
29422 (gst_ogg_mux_push_buffer):
29423 Don't generate out-of-order timestamps from oggmux, instead clamp
29424 output timestamps to be >= the previously output ts.
29427 2006-09-18 10:18:22 +0000 Michael Smith <msmith@xiph.org>
29429 gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
29430 Original commit message from CVS:
29431 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
29432 (gst_multi_fd_sink_class_init):
29433 Updates, fixes, and typo corrections for multifdsink. No functional
29436 2006-09-17 21:58:06 +0000 Michael Smith <msmith@xiph.org>
29438 gst/typefind/gsttypefindfunctions.c: Don't crash on truncated files - check that we got an 8 byte buffer before tryin...
29439 Original commit message from CVS:
29440 * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
29441 Don't crash on truncated files - check that we got an 8 byte buffer
29442 before trying to memcmp it.
29444 2006-09-17 20:32:09 +0000 Tim-Philipp Müller <tim@centricular.net>
29446 gst/playback/gstplaybasebin.c: Make stream-switching appear instant to the application (ie. make sure that a g_object...
29447 Original commit message from CVS:
29448 * gst/playback/gstplaybasebin.c: (get_active_source):
29449 Make stream-switching appear instant to the application
29450 (ie. make sure that a g_object_get on 'current-foo' returns
29451 the stream previously set with g_object_set(). Totem needs
29452 this to update stream-related meta-info (like audio-codec)
29453 correctly when switching streams.
29455 2006-09-17 20:14:43 +0000 Tim-Philipp Müller <tim@centricular.net>
29457 ext/alsa/gstalsamixer.c: Try harder to guess which mixer track is the master mixer track (instead of just taking the ...
29458 Original commit message from CVS:
29459 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
29460 (gst_alsa_mixer_ensure_track_list):
29461 Try harder to guess which mixer track is the master mixer
29462 track (instead of just taking the first one that has a pvolume).
29465 2006-09-17 11:24:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29467 gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
29468 Original commit message from CVS:
29469 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
29470 (gst_audio_convert_transform_caps):
29471 Get structure-name just once.
29473 2006-09-16 22:30:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29475 tests/check/: Fix big batch of compiler warnings.
29476 Original commit message from CVS:
29477 * tests/check/elements/audioresample.c: (GST_START_TEST):
29478 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
29479 * tests/check/elements/volume.c: (GST_START_TEST):
29480 * tests/check/elements/vorbisdec.c: (GST_START_TEST):
29481 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
29482 (test_pipeline), (GST_START_TEST):
29483 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
29484 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
29485 Fix big batch of compiler warnings.
29487 2006-09-16 21:54:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29489 ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
29490 Original commit message from CVS:
29491 * ext/gnomevfs/gstgnomevfssrc.c:
29492 Add docs about icydemux usage in connection with gnomevfssrc
29493 * ext/libvisual/visual.c:
29494 * ext/ogg/gstoggaviparse.c:
29495 * ext/ogg/gstoggdemux.c:
29496 * ext/ogg/gstoggmux.c:
29497 * ext/ogg/gstoggparse.c:
29498 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
29499 * gst-libs/gst/audio/gstaudiosink.c:
29500 * gst-libs/gst/audio/gstaudiosrc.c:
29501 * gst/audiorate/gstaudiorate.c:
29502 More G_OBJECT macro fixing.
29503 * gst/audiotestsrc/gstaudiotestsrc.h:
29504 Fix wrong info in header due to copy & paste
29506 2006-09-15 14:53:44 +0000 Wim Taymans <wim.taymans@gmail.com>
29508 gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
29509 Original commit message from CVS:
29510 * gst-libs/gst/audio/gstbaseaudiosink.c:
29511 (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
29512 * gst-libs/gst/audio/gstbaseaudiosrc.c:
29513 (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
29514 (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
29515 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
29516 Do the delay calculation in the source/sink base classes as this is
29517 specific for the capture/playback mode.
29518 Try to fixate a bit better, like round depth up to a multiple of 8
29520 Handle underruns correctly by marking DISCONT on buffers and adjusting
29521 timestamps to handle the gap.
29522 Set offset/offset_end correctly on buffers.
29523 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
29524 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
29525 (gst_ring_buffer_read):
29526 Remove resync and underrun recovery from the ringbuffer.
29527 Fix ringbuffer read code on under/overrun.
29529 2006-09-15 11:17:02 +0000 Wim Taymans <wim.taymans@gmail.com>
29531 gst/playback/gstplaybasebin.*: Don't use a 0 low watermark when buffering, it is catching starvation way too late. In...
29532 Original commit message from CVS:
29533 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
29534 (gst_play_base_bin_init), (fill_buffer), (check_queue),
29535 (queue_threshold_reached), (gst_play_base_bin_set_property),
29536 (gst_play_base_bin_get_property):
29537 * gst/playback/gstplaybasebin.h:
29538 Don't use a 0 low watermark when buffering, it is catching starvation
29539 way too late. Instead, use a 3 second queue with 30 and 95
29540 percent low/high watermarks.
29541 Added queue-min-threshold property to configure low watermark.
29542 Use new _buffering message API.
29543 Make queue_threshold variable big enough to store a uint64 time value.
29544 API: playbin::queue-min-threshold property.
29546 2006-09-15 09:13:50 +0000 Wim Taymans <wim.taymans@gmail.com>
29548 configure.ac: We require 0.10.10.1 now because of _wait_preroll().
29549 Original commit message from CVS:
29551 We require 0.10.10.1 now because of _wait_preroll().
29552 * gst-libs/gst/audio/gstbaseaudiosink.c:
29553 (gst_base_audio_sink_render):
29554 Use gst_base_sink_wait_preroll().
29556 2006-09-15 09:09:00 +0000 Wim Taymans <wim.taymans@gmail.com>
29558 ext/alsa/: Use DEBUG_OBJECT more.
29559 Original commit message from CVS:
29560 * ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
29561 * ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
29562 Use DEBUG_OBJECT more.
29564 === release 0.10.10 ===
29566 2006-09-14 20:09:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29573 * docs/plugins/gst-plugins-base-plugins.args:
29574 * docs/plugins/inspect/plugin-adder.xml:
29575 * docs/plugins/inspect/plugin-alsa.xml:
29576 * docs/plugins/inspect/plugin-audioconvert.xml:
29577 * docs/plugins/inspect/plugin-audiorate.xml:
29578 * docs/plugins/inspect/plugin-audioresample.xml:
29579 * docs/plugins/inspect/plugin-audiotestsrc.xml:
29580 * docs/plugins/inspect/plugin-cdparanoia.xml:
29581 * docs/plugins/inspect/plugin-decodebin.xml:
29582 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
29583 * docs/plugins/inspect/plugin-gdp.xml:
29584 * docs/plugins/inspect/plugin-gnomevfs.xml:
29585 * docs/plugins/inspect/plugin-libvisual.xml:
29586 * docs/plugins/inspect/plugin-ogg.xml:
29587 * docs/plugins/inspect/plugin-pango.xml:
29588 * docs/plugins/inspect/plugin-playbin.xml:
29589 * docs/plugins/inspect/plugin-subparse.xml:
29590 * docs/plugins/inspect/plugin-tcp.xml:
29591 * docs/plugins/inspect/plugin-theora.xml:
29592 * docs/plugins/inspect/plugin-typefindfunctions.xml:
29593 * docs/plugins/inspect/plugin-video4linux.xml:
29594 * docs/plugins/inspect/plugin-videorate.xml:
29595 * docs/plugins/inspect/plugin-videoscale.xml:
29596 * docs/plugins/inspect/plugin-videotestsrc.xml:
29597 * docs/plugins/inspect/plugin-volume.xml:
29598 * docs/plugins/inspect/plugin-vorbis.xml:
29599 * docs/plugins/inspect/plugin-ximagesink.xml:
29600 * docs/plugins/inspect/plugin-xvimagesink.xml:
29601 * ext/theora/theoraparse.c:
29602 * gst-libs/gst/rtp/gstrtpbuffer.c:
29603 * gst/playback/gstplaybin.c:
29604 * tests/check/Makefile.am:
29605 * win32/common/config.h:
29607 Original commit message from CVS:
29610 2006-09-09 16:08:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29613 * win32/common/config.h:
29615 Original commit message from CVS:
29618 2006-09-07 19:01:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29621 update bug in changelog
29622 Original commit message from CVS:
29623 update bug in changelog
29625 2006-09-07 19:00:33 +0000 Michael Smith <msmith@fluendo.com>
29627 Fix implementation of sync-method 'next-keyframe'
29628 Original commit message from CVS:
29629 patch by: Michael Smith <msmith at fluendo dot com>
29630 * gst/tcp/gstmultifdsink.c: (is_sync_frame),
29631 (gst_multi_fd_sink_client_queue_buffer),
29632 (gst_multi_fd_sink_new_client):
29633 * tests/check/elements/multifdsink.c: (GST_START_TEST),
29634 (multifdsink_suite):
29635 Fix implementation of sync-method 'next-keyframe'
29637 2006-09-07 15:00:08 +0000 Wim Taymans <wim@fluendo.com>
29639 ext/gnomevfs/gstgnomevfssrc.c: This patch removes the RANDOM flag that was incorrectly introduced with revision 1.91....
29640 Original commit message from CVS:
29641 patch by: Wim Taymans <wim at fluendo dot com>
29642 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
29643 This patch removes the RANDOM flag that was incorrectly introduced with
29644 revision 1.91. Fixes #354590
29646 2006-09-07 14:56:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29649 * win32/common/config.h:
29651 Original commit message from CVS:
29654 2006-09-07 14:56:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29671 Original commit message from CVS:
29674 2006-09-05 09:12:25 +0000 Tim-Philipp Müller <tim@centricular.net>
29676 tests/check/Makefile.am: Random variation in Makefile line to see if it makes the gen64-base-full bot any happier.
29677 Original commit message from CVS:
29678 * tests/check/Makefile.am:
29679 Random variation in Makefile line to see if it makes the
29680 gen64-base-full bot any happier.
29682 2006-09-04 19:04:35 +0000 Tim-Philipp Müller <tim@centricular.net>
29684 tests/check/pipelines/oggmux.c: Disable test that fails at the moment (killed after timeout).
29685 Original commit message from CVS:
29686 * tests/check/pipelines/oggmux.c: (oggmux_suite):
29687 Disable test that fails at the moment (killed after timeout).
29689 2006-09-04 18:19:06 +0000 James Livingston <doclivingston@gmail.com>
29691 tests/check/: Add simple unit test for oggmux from #337026 with checking for the
29692 Original commit message from CVS:
29693 Patch by: James Livingston <doclivingston at gmail.com>
29694 * tests/check/Makefile.am:
29695 * tests/check/pipelines/.cvsignore:
29696 * tests/check/pipelines/oggmux.c: (get_page_codec),
29697 (check_chain_final_state), (fail_if_audio), (validate_ogg_page),
29698 (eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
29699 (test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
29700 (test_theora_vorbis), (oggmux_suite):
29701 Add simple unit test for oggmux from #337026 with checking for the
29702 EOS flags disabled for the time being.
29704 2006-09-04 09:13:01 +0000 Alessandro Dessina <alessandro@nnva.org>
29706 ext/ogg/gstoggmux.c: Add cmml caps to oggmux. Fixes #353912
29707 Original commit message from CVS:
29708 patch by: Alessandro Dessina <alessandro nnva org>
29709 * ext/ogg/gstoggmux.c:
29710 Add cmml caps to oggmux. Fixes #353912
29712 2006-09-02 13:20:59 +0000 Tim-Philipp Müller <tim@centricular.net>
29714 tests/check/elements/videotestsrc.c: Returning a return value often helps. In this case, we don't need the return val...
29715 Original commit message from CVS:
29716 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
29717 Returning a return value often helps. In this case, we
29718 don't need the return value anyway, so just get rid of it.
29719 Should make build bots much happier.
29721 2006-09-02 12:59:48 +0000 Tim-Philipp Müller <tim@centricular.net>
29723 gst/videotestsrc/videotestsrc.*: Add support for AYUV and the various RGBA formats. Initialise fields of paintinfo st...
29724 Original commit message from CVS:
29725 * gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
29726 (paint_get_structure), (gst_video_test_src_get_size),
29727 (gst_video_test_src_smpte), (gst_video_test_src_snow),
29728 (gst_video_test_src_unicolor), (paint_setup_AYUV),
29729 (paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
29730 (paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
29731 * gst/videotestsrc/videotestsrc.h:
29732 Add support for AYUV and the various RGBA formats. Initialise
29733 fields of paintinfo structs allocated on the stack.
29734 * tests/check/elements/videotestsrc.c: (right_shift_colour),
29735 (fix_expected_colour), (check_rgb_buf), (got_buf_cb),
29736 (GST_START_TEST), (videotestsrc_suite):
29737 Add unit tests for videotestsrc's RGB output.
29739 2006-09-01 16:12:35 +0000 Tim-Philipp Müller <tim@centricular.net>
29741 gst/videotestsrc/: Add more uni-colour patterns ("white", "red", "green", and "blue").
29742 Original commit message from CVS:
29743 * gst/videotestsrc/gstvideotestsrc.c:
29744 (gst_video_test_src_pattern_get_type),
29745 (gst_video_test_src_set_pattern):
29746 * gst/videotestsrc/gstvideotestsrc.h:
29747 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
29748 (gst_video_test_src_black), (gst_video_test_src_white),
29749 (gst_video_test_src_red), (gst_video_test_src_green),
29750 (gst_video_test_src_blue):
29751 * gst/videotestsrc/videotestsrc.h:
29752 Add more uni-colour patterns ("white", "red", "green", and "blue").
29754 2006-09-01 10:07:05 +0000 Tim-Philipp Müller <tim@centricular.net>
29756 gst/videotestsrc/videotestsrc.c: Fix stride for YVYU, should be word-aligned (#353658).
29757 Original commit message from CVS:
29758 * gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
29759 Fix stride for YVYU, should be word-aligned (#353658).
29761 2006-08-31 14:37:33 +0000 Tim-Philipp Müller <tim@centricular.net>
29763 gst/adder/gstadder.c: Fix build.
29764 Original commit message from CVS:
29765 * gst/adder/gstadder.c: (gst_adder_src_event):
29768 2006-08-31 12:39:17 +0000 Edward Hervey <bilboed@bilboed.com>
29770 gst/adder/gstadder.*: Remember the start position asked in the incoming seeks, so we can output GST_EVENT_NEW_SEGMENT...
29771 Original commit message from CVS:
29772 * gst/adder/gstadder.c: (forward_event_func),
29773 (gst_adder_src_event), (gst_adder_collected),
29774 (gst_adder_change_state):
29775 * gst/adder/gstadder.h:
29776 Remember the start position asked in the incoming seeks, so we can
29777 output GST_EVENT_NEW_SEGMENT with a correct position value (instead
29778 of assuming it will always be 0).
29780 2006-08-31 12:31:00 +0000 Edward Hervey <bilboed@bilboed.com>
29782 ext/ogg/gstoggdemux.c: Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
29783 Original commit message from CVS:
29784 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
29785 (gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
29786 (gst_ogg_demux_loop):
29787 Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
29789 2006-08-30 17:22:27 +0000 Tim-Philipp Müller <tim@centricular.net>
29791 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Return FALSE instead of returning a random false unit size when the forma...
29792 Original commit message from CVS:
29793 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
29794 (gst_ffmpegcsp_get_unit_size):
29795 Return FALSE instead of returning a random false unit
29796 size when the format isn't known/supported (even if
29797 this shouldn't happen under normal circumstances).
29799 2006-08-29 15:23:46 +0000 Tim-Philipp Müller <tim@centricular.net>
29801 ext/gnomevfs/gstgnomevfssrc.c: Try harder to get the size from a uri by using _info_uri() when _info_from_handle() do...
29802 Original commit message from CVS:
29803 Patch by: Tim-Philipp Müller <tim at centricular dot net>
29804 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
29805 (gst_gnome_vfs_src_start):
29806 Try harder to get the size from a uri by using _info_uri() when
29807 _info_from_handle() does not give us enough info.
29808 Also follow symlinks when getting the size.
29809 Partially Fixes #332864.
29811 2006-08-29 11:50:51 +0000 Viktor Peters <viktor.peters@gmail.com>
29813 ext/alsa/: Improve and fix mixer track handling, in particular better handling of alsa's pvolume/pswitch/cvolume/cswi...
29814 Original commit message from CVS:
29815 Patch by: Viktor Peters <viktor dot peters at gmail dot com>
29816 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
29817 (gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
29818 (gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
29819 (gst_alsa_mixer_set_record):
29820 * ext/alsa/gstalsamixertrack.c:
29821 (gst_alsa_mixer_track_update_alsa_capabilities),
29822 (alsa_track_has_cap), (gst_alsa_mixer_track_new),
29823 (gst_alsa_mixer_track_update):
29824 * ext/alsa/gstalsamixertrack.h:
29825 Improve and fix mixer track handling, in particular better handling
29826 of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
29827 track objects for tracks that have both capture and playback volume
29828 (and label them differently as well so they're not mistakenly
29829 assumed to be duplicates); classify mixer tracks that only affect
29830 the audible volume of something (rather than the capture volume)
29831 as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
29832 for capture tracks to correspond to alsa-pswitch alsa-cswitch
29833 (following the meaning documented in the mixer interface header
29834 file); add support for alsa's exclusive cswitch groups; update/sync
29835 state/flags better if mixer settings are changed by another
29836 application. Fixes #336075.
29838 2006-08-29 10:58:43 +0000 Tim-Philipp Müller <tim@centricular.net>
29840 gst/playback/gstplaybin.c: Improve docs: add section about BUFFERING messages sent by playbin.
29841 Original commit message from CVS:
29842 * gst/playback/gstplaybin.c:
29843 Improve docs: add section about BUFFERING messages sent by playbin.
29845 2006-08-29 10:51:12 +0000 Michael Smith <msmith@xiph.org>
29847 ext/vorbis/vorbisenc.c: Ignore explicit DISCONT marked on buffers (which is often spurious, particularly when using m...
29848 Original commit message from CVS:
29849 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
29850 (gst_vorbis_enc_buffer_check_discontinuous),
29851 (gst_vorbis_enc_chain):
29852 Ignore explicit DISCONT marked on buffers (which is often spurious,
29853 particularly when using multiple segments), in favour of solely
29854 using the timestamps/durations.
29856 2006-08-29 10:32:34 +0000 Edward Hervey <bilboed@bilboed.com>
29858 gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
29859 Original commit message from CVS:
29860 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
29861 Don't rely on incoming buffers offset anymore, since it is completely
29862 broken when using multiple segments.
29863 Instead convert the incoming buffers timestamp to running time, and
29864 then convert that value to the offsets.
29865 Also inform GstSegment of the last outputted stop position, which is
29866 needed if we received several segments with an unknown stop value.
29868 2006-08-29 08:03:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29870 ext/vorbis/vorbisenc.c: fix buffer unreffing on a header push failure
29871 Original commit message from CVS:
29872 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
29873 fix buffer unreffing on a header push failure
29875 2006-08-28 16:17:13 +0000 Wim Taymans <wim.taymans@gmail.com>
29877 gst/audiorate/gstaudiorate.c: Make the metadata of the buffer writable before changing its flags.
29878 Original commit message from CVS:
29879 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
29880 (gst_audio_rate_chain):
29881 Make the metadata of the buffer writable before changing its
29884 2006-08-28 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
29887 Fix changelog with bugzilla bug it fixed.
29888 Original commit message from CVS:
29889 Fix changelog with bugzilla bug it fixed.
29891 2006-08-28 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
29893 gst/audiorate/gstaudiorate.c: Fix audiorate some more.
29894 Original commit message from CVS:
29895 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
29896 (gst_audio_rate_setcaps), (gst_audio_rate_init),
29897 (gst_audio_rate_sink_event), (gst_audio_rate_src_event),
29898 (gst_audio_rate_chain), (gst_audio_rate_change_state):
29899 Fix audiorate some more.
29900 Reset and resync counters on flush and READY.
29901 Handle the DISCONT flag correctly.
29902 Use GstSegment to track position.
29903 Fail when not negotiated.
29905 2006-08-25 16:48:28 +0000 Michael Smith <msmith@xiph.org>
29907 gst/tcp/gstmultifdsink.c: Fix spelling.
29908 Original commit message from CVS:
29909 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
29911 Remove accidently included debug line.
29913 2006-08-25 16:39:38 +0000 Wim Taymans <wim.taymans@gmail.com>
29915 gst/tcp/gstmultifdsink.c: Small cleanups.
29916 Original commit message from CVS:
29917 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
29919 If a buffer is received with no caps, make the buffer metadata
29920 writable and set the caps, making sure that we don't screw up the
29923 2006-08-25 16:19:55 +0000 Michael Smith <msmith@xiph.org>
29925 gst/gdp/gstgdppay.c: Fix memory leaks and misleading debug messages, add a couple of comments.
29926 Original commit message from CVS:
29927 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
29928 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
29929 Fix memory leaks and misleading debug messages, add a couple of
29931 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
29932 (gst_multi_fd_sink_render):
29933 Do not use gst_buffer_make_writable() in a basesink render method,
29934 as it may incorrectly unref the buffer. Instead, use convoluted
29935 dance to avoid copying the buffer except when we need to.
29937 2006-08-25 09:54:56 +0000 Michael Smith <msmith@xiph.org>
29939 ext/vorbis/vorbisenc.c: Allow very small discontinuities in the timestamps. These we can't do anything useful with an...
29940 Original commit message from CVS:
29941 * ext/vorbis/vorbisenc.c:
29942 (gst_vorbis_enc_buffer_check_discontinuous):
29943 Allow very small discontinuities in the timestamps. These we can't
29944 do anything useful with anyway (because vorbis's timestamps have
29945 only sample granularity), and are commonly produced by elements with
29946 minor bugs. Allow up to 1/2 a sample out.
29949 2006-08-24 11:18:56 +0000 Wim Taymans <wim.taymans@gmail.com>
29951 tests/examples/seek/seek.c: Add a checkbox to enable play scrubbing. Makes it possible to disable normal scrubbing.
29952 Original commit message from CVS:
29953 * tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
29954 (play_scrub_toggle_cb), (main):
29955 Add a checkbox to enable play scrubbing. Makes it possible to disable
29958 2006-08-23 19:37:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29960 tests/check/elements/.cvsignore: make buildbot happy
29961 Original commit message from CVS:
29962 * tests/check/elements/.cvsignore:
29963 make buildbot happy
29965 2006-08-23 16:43:03 +0000 Tim-Philipp Müller <tim@centricular.net>
29967 ext/ogg/gstogmparse.c: Refactor ogm parse, do better input checking, misc. clean-ups.
29968 Original commit message from CVS:
29969 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
29970 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
29971 (gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
29972 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
29973 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
29974 (gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
29975 (gst_ogm_text_parse_strip_trailing_zeroes),
29976 (gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
29977 (gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
29978 Refactor ogm parse, do better input checking, misc. clean-ups.
29979 Cache incoming events and push them once the source pad has
29980 been created. Don't pass unterminated strings to sscanf().
29981 Strip trailing zeroes from subtitle text output, since they
29982 are not valid UTF-8. Don't push vorbiscomment packets on
29983 the subtitle text pad. Output perfect streams if possible.
29985 2006-08-23 15:27:38 +0000 Wim Taymans <wim.taymans@gmail.com>
29987 tests/check/libs/cddabasesrc.c: Waits for tasks to settle down so that we clean up correctly for valgrind.
29988 Original commit message from CVS:
29989 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
29990 Waits for tasks to settle down so that we clean up correctly for
29993 2006-08-23 15:11:56 +0000 Tim-Philipp Müller <tim@centricular.net>
29995 tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
29996 Original commit message from CVS:
29997 * tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
29998 Unit test fixes: \377 is more likely to fit into 8 bits than \777;
29999 actually return return value in taglists_are_equal.
30001 2006-08-23 12:14:20 +0000 Tim-Philipp Müller <tim@centricular.net>
30003 ext/ogg/gstogmparse.c: Fix crash due to broken bitstream parsing on x86-64: can't make any assumptions about sizeof(s...
30004 Original commit message from CVS:
30005 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
30006 Fix crash due to broken bitstream parsing on x86-64: can't make
30007 any assumptions about sizeof(struct) due to alignment/packing
30008 differences on different architectures. Fixes #351790.
30010 2006-08-22 16:31:47 +0000 Wim Taymans <wim.taymans@gmail.com>
30012 gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
30013 Original commit message from CVS:
30014 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
30015 (gst_riff_parse_chunk), (gst_riff_parse_file_header),
30016 (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
30017 (gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
30018 (gst_riff_parse_info):
30019 Protect public functions against bad input.
30023 2006-08-22 15:50:36 +0000 Tim-Philipp Müller <tim@centricular.net>
30025 gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
30026 Original commit message from CVS:
30027 * gst-libs/gst/riff/riff-ids.h:
30028 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
30029 Add voxware audio IDs (even if we can't play it) (#351795).
30031 2006-08-22 15:11:52 +0000 Tim-Philipp Müller <tim@centricular.net>
30033 gst-libs/gst/riff/riff-media.c: Const-ify some arrays and use G_N_ELEMENTS instead of wasting oodles of RAM on termin...
30034 Original commit message from CVS:
30035 * gst-libs/gst/riff/riff-media.c:
30036 (gst_riff_create_video_template_caps),
30037 (gst_riff_create_audio_template_caps),
30038 (gst_riff_create_iavs_template_caps):
30039 Const-ify some arrays and use G_N_ELEMENTS instead
30040 of wasting oodles of RAM on terminator bits.
30042 2006-08-22 08:27:07 +0000 Tim-Philipp Müller <tim@centricular.net>
30044 And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
30045 Original commit message from CVS:
30046 * gst-libs/gst/tag/gstvorbistag.c:
30047 (gst_tag_list_to_vorbiscomment_buffer):
30048 * tests/check/libs/tag.c: (GST_START_TEST):
30049 And the same for _to_vorbiscomment_buffer(): allow
30050 id_data_len == 0 for speex.
30052 2006-08-21 19:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30056 Original commit message from CVS:
30059 2006-08-21 19:01:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30061 Move GDP plugin to -base from -bad. Closes #347783.
30062 Original commit message from CVS:
30064 * docs/plugins/Makefile.am:
30065 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
30066 * docs/plugins/gst-plugins-base-plugins-sections.txt:
30067 * docs/plugins/inspect/plugin-gdp.xml:
30068 * gst/gdp/Makefile.am:
30069 * tests/check/Makefile.am:
30070 Move GDP plugin to -base from -bad. Closes #347783.
30072 2006-08-21 18:34:46 +0000 Tim-Philipp Müller <tim@centricular.net>
30074 gst-libs/gst/tag/gstvorbistag.c: Allow id_data_len == 0 (needed for vorbis comments in Speex files).
30075 Original commit message from CVS:
30076 * gst-libs/gst/tag/gstvorbistag.c:
30077 (gst_tag_list_from_vorbiscomment_buffer):
30078 Allow id_data_len == 0 (needed for vorbis comments in Speex files).
30079 Also add some checks to make sure we don't memcmp() beyond the end of
30080 vorbiscomment buffer if the ID to check for is larger than the buffer.
30081 * tests/check/libs/tag.c: (GST_START_TEST):
30082 Some more tests for gst_tag_list_from_vorbiscomment_buffer().
30084 2006-08-21 16:39:25 +0000 Tim-Philipp Müller <tim@centricular.net>
30086 ext/vorbis/vorbisenc.c: Use vorbis comment utility functions from libgsttag instead of re-inventing the wheel (partia...
30087 Original commit message from CVS:
30088 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
30089 (gst_vorbis_enc_set_metadata):
30090 Use vorbis comment utility functions from libgsttag
30091 instead of re-inventing the wheel (partially fixes #347091).
30093 2006-08-21 11:42:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30095 tests/check/elements/audioconvert.c: Fix leaks. Wait for state transitions that might happen ASYNC, as well as some t...
30096 Original commit message from CVS:
30097 * tests/check/elements/audioconvert.c: (GST_START_TEST):
30098 Fix leaks. Wait for state transitions that might happen ASYNC, as well
30099 as some that won't.
30101 2006-08-21 10:32:51 +0000 Wim Taymans <wim.taymans@gmail.com>
30103 docs/libs/: Don't try to GObject scan the netbuffer as it's not a GObject.
30104 Original commit message from CVS:
30105 * docs/libs/Makefile.am:
30106 * docs/libs/gst-plugins-base-libs-sections.txt:
30107 * docs/libs/gst-plugins-base-libs.types:
30108 Don't try to GObject scan the netbuffer as it's not a GObject.
30110 * gst-libs/gst/netbuffer/gstnetbuffer.c:
30111 * gst-libs/gst/netbuffer/gstnetbuffer.h:
30112 Document GstNetBuffer.
30114 2006-08-21 08:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30116 tests/check/elements/audioconvert.c: Add testcase for caps-size-explosion
30117 Original commit message from CVS:
30118 * tests/check/elements/audioconvert.c: (GST_START_TEST),
30119 (audioconvert_suite):
30120 Add testcase for caps-size-explosion
30122 2006-08-20 13:05:44 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30124 gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size
30125 Original commit message from CVS:
30126 * gst/audioconvert/gstaudioconvert.c:
30127 (gst_audio_convert_get_unit_size), (set_structure_widths):
30128 Lower debug, use g_assert in _get_unit_size
30129 * gst/audioresample/gstaudioresample.c:
30130 (audioresample_get_unit_size):
30131 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30132 (gst_ffmpegcsp_get_unit_size):
30133 * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
30134 use g_assert in _get_unit_size
30136 2006-08-18 21:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
30139 ChangeLog surgery: fix bug number
30140 Original commit message from CVS:
30141 ChangeLog surgery: fix bug number
30143 2006-08-18 16:43:26 +0000 Wim Taymans <wim.taymans@gmail.com>
30145 Document GstRTPBuffer.
30146 Original commit message from CVS:
30147 * docs/libs/gst-plugins-base-libs-sections.txt:
30148 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
30149 (gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
30150 (gst_rtp_buffer_get_payload_buffer):
30151 * gst-libs/gst/rtp/gstrtpbuffer.h:
30152 Document GstRTPBuffer.
30153 Added function to efficiently strip payload headers.
30154 API: gst_rtp_buffer_get_payload_subbuffer()
30156 2006-08-17 16:52:06 +0000 Tim-Philipp Müller <tim@centricular.net>
30158 gst-libs/gst/tag/gstvorbistag.c: Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT tags and deserialise...
30159 Original commit message from CVS:
30160 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
30161 (gst_tag_to_vorbis_comments):
30162 Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
30163 tags and deserialise them properly as well (#351768).
30164 Add some more gtk-doc blurbs and also some g_return_if_fail().
30165 * tests/check/libs/tag.c: (GST_START_TEST),
30166 (back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
30169 2006-08-17 15:43:40 +0000 Wim Taymans <wim.taymans@gmail.com>
30171 ext/ogg/: Added ogg-in-avi parser element. Fixes #140139.
30172 Original commit message from CVS:
30173 * ext/ogg/Makefile.am:
30174 * ext/ogg/gstogg.c: (plugin_init):
30175 * ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
30176 (gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
30177 (gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
30178 (gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
30179 (gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
30180 (gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
30181 Added ogg-in-avi parser element. Fixes #140139.
30182 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
30183 Fixed a bug in oggdemux debug code.
30184 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
30185 (gst_riff_create_audio_template_caps):
30186 Recognise Ogg in the AVI extensible wave format.
30188 2006-08-17 10:00:00 +0000 Tim-Philipp Müller <tim@centricular.net>
30190 gst-libs/gst/cdda/gstcddabasesrc.c: Make buffer durations add up (duration should be next_ts-ts for perfect streams)....
30191 Original commit message from CVS:
30192 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
30193 Make buffer durations add up (duration should be next_ts-ts for
30194 perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
30196 * tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
30197 (test_buffer_timestamps), (cddabasesrc_suite):
30198 Add unit test for the above.
30199 * tests/check/Makefile.am:
30200 Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
30201 to see what happens.
30203 2006-08-16 11:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
30205 ext/alsa/: Avoid setting and using a NULL device name.
30206 Original commit message from CVS:
30207 * ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
30208 (gst_alsasink_open):
30209 * ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
30210 (gst_alsasrc_open):
30211 Avoid setting and using a NULL device name.
30212 Print more info when we fail to open a device.
30214 2006-08-16 11:28:57 +0000 Tim-Philipp Müller <tim@centricular.net>
30216 API: add gst_tag_parse_extended_comment() (#351426).
30217 Original commit message from CVS:
30218 * docs/libs/gst-plugins-base-libs-sections.txt:
30219 * gst-libs/gst/tag/tag.h:
30220 * gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
30221 API: add gst_tag_parse_extended_comment() (#351426).
30222 * tests/check/Makefile.am:
30223 * tests/check/libs/.cvsignore:
30224 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
30225 Add unit test for gst_tag_parse_extended_comment().
30227 2006-08-15 19:20:16 +0000 Tim-Philipp Müller <tim@centricular.net>
30229 sys/: Fix leak (#351502).
30230 Original commit message from CVS:
30231 * sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
30232 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
30233 Fix leak (#351502).
30235 2006-08-15 17:21:33 +0000 Tim-Philipp Müller <tim@centricular.net>
30238 Original commit message from CVS:
30239 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
30240 * docs/plugins/gst-plugins-base-plugins-sections.txt:
30241 * docs/plugins/gst-plugins-base-plugins.args:
30242 * gst/playback/gstplaybin.c:
30244 * docs/plugins/inspect/plugin-adder.xml:
30245 * docs/plugins/inspect/plugin-alsa.xml:
30246 * docs/plugins/inspect/plugin-audioconvert.xml:
30247 * docs/plugins/inspect/plugin-audiorate.xml:
30248 * docs/plugins/inspect/plugin-audioresample.xml:
30249 * docs/plugins/inspect/plugin-audiotestsrc.xml:
30250 * docs/plugins/inspect/plugin-cdparanoia.xml:
30251 * docs/plugins/inspect/plugin-decodebin.xml:
30252 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
30253 * docs/plugins/inspect/plugin-gnomevfs.xml:
30254 * docs/plugins/inspect/plugin-ogg.xml:
30255 * docs/plugins/inspect/plugin-pango.xml:
30256 * docs/plugins/inspect/plugin-playbin.xml:
30257 * docs/plugins/inspect/plugin-subparse.xml:
30258 * docs/plugins/inspect/plugin-tcp.xml:
30259 * docs/plugins/inspect/plugin-theora.xml:
30260 * docs/plugins/inspect/plugin-typefindfunctions.xml:
30261 * docs/plugins/inspect/plugin-video4linux.xml:
30262 * docs/plugins/inspect/plugin-videorate.xml:
30263 * docs/plugins/inspect/plugin-videoscale.xml:
30264 * docs/plugins/inspect/plugin-videotestsrc.xml:
30265 * docs/plugins/inspect/plugin-volume.xml:
30266 * docs/plugins/inspect/plugin-vorbis.xml:
30267 * docs/plugins/inspect/plugin-ximagesink.xml:
30268 * docs/plugins/inspect/plugin-xvimagesink.xml:
30269 Update to CVS version.
30271 2006-08-14 17:54:01 +0000 Tim-Philipp Müller <tim@centricular.net>
30273 gst/playback/gstplaybin.c: Add "connection-speed" property; re-order redirect messages with multiple redirect locatio...
30274 Original commit message from CVS:
30275 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
30276 (gst_play_bin_set_property), (gst_play_bin_get_property),
30277 (value_list_append_structure_list),
30278 (gst_play_bin_handle_redirect_message),
30279 (gst_play_bin_handle_message):
30280 Add "connection-speed" property; re-order redirect messages with
30281 multiple redirect locations depending on the minimum bitrate if
30282 that information is available and a connection speed is set
30285 2006-08-14 11:41:04 +0000 Tim-Philipp Müller <tim@centricular.net>
30287 gst/playback/gstplaybin.c: Update max volume to the same value that the volume element uses.
30288 Original commit message from CVS:
30289 * gst/playback/gstplaybin.c:
30290 Update max volume to the same value that the volume element uses.
30292 2006-08-14 10:50:15 +0000 Wim Taymans <wim.taymans@gmail.com>
30294 ext/alsa/gstalsamixer.c: Less uglyness..
30295 Original commit message from CVS:
30296 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
30299 2006-08-14 10:49:10 +0000 Wim Taymans <wim.taymans@gmail.com>
30301 ext/ogg/gstoggdemux.c: Add some more debug info.
30302 Original commit message from CVS:
30303 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
30304 (gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
30305 (gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
30306 Add some more debug info.
30307 Don't crash when a seek failed.
30308 Actually return the result of the seek instead of TRUE.
30309 Ignore multiple BOS pages with the same serial so that we don't create
30310 the same stream multiple times.
30311 Post an error when we fail to do the initial seek.
30313 2006-08-13 14:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
30315 ext/alsa/gstalsa.c: Small code cleanup.
30316 Original commit message from CVS:
30317 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
30318 (gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
30319 Small code cleanup.
30320 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
30321 (gst_alsa_mixer_new):
30322 Remove hack that always set the device to hw:0*.
30323 Properly find the card name for whatever device was configured.
30324 Do some better debugging.
30326 * ext/alsa/gstalsamixerelement.c:
30327 (gst_alsa_mixer_element_set_property),
30328 (gst_alsa_mixer_element_change_state):
30330 Handle setting of a NULL device name better.
30332 2006-08-11 15:53:43 +0000 Wim Taymans <wim.taymans@gmail.com>
30334 gst/adder/gstadder.c: Don't clip float values. Fixes #350900.
30335 Original commit message from CVS:
30336 * gst/adder/gstadder.c:
30337 Don't clip float values. Fixes #350900.
30339 2006-08-11 15:33:17 +0000 Andy Wingo <wingo@pobox.com>
30341 gst/tcp/gsttcp.c: Really fix the build?
30342 Original commit message from CVS:
30343 2006-08-11 Andy Wingo <wingo@pobox.com>
30344 * gst/tcp/gsttcp.c: Really fix the build?
30346 2006-08-11 15:29:56 +0000 Andy Wingo <wingo@pobox.com>
30348 gst/tcp/gsttcp.h: For now, always disable deprecation here -- fixes the build.
30349 Original commit message from CVS:
30350 2006-08-11 Andy Wingo <wingo@pobox.com>
30351 * gst/tcp/gsttcp.h: For now, always disable deprecation here --
30354 2006-08-10 13:01:31 +0000 Tim-Philipp Müller <tim@centricular.net>
30356 gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field.
30357 Original commit message from CVS:
30358 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
30359 Float caps shouldn't have a "signed" field.
30361 2006-08-10 08:56:22 +0000 Tim-Philipp Müller <tim@centricular.net>
30363 ext/ogg/gstoggdemux.c: Implement SEEKING query in its most basic form, so that we can at least check if we're seekabl...
30364 Original commit message from CVS:
30365 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
30366 Implement SEEKING query in its most basic form, so that we can
30367 at least check if we're seekable or not (#350655).
30369 2006-08-09 14:42:58 +0000 Tim-Philipp Müller <tim@centricular.net>
30371 gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
30372 Original commit message from CVS:
30373 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
30374 The checks here are not even close to anything that would
30375 justify MAXIMUM probability, lowering to POSSIBLE until someone
30376 fixes the checks (case at hand: quicktime redirection files
30377 might start with 00 00 01 XX and pass the checks here just
30378 fine, see #350399).
30380 2006-08-08 13:57:29 +0000 Edward Hervey <bilboed@bilboed.com>
30382 tests/check/elements/gdpdepay.c: I forgot to include the file containing the #define :)
30383 Original commit message from CVS:
30384 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
30385 I forgot to include the file containing the #define :)
30386 Now includes "config.h"
30388 2006-08-08 13:45:44 +0000 Edward Hervey <bilboed@bilboed.com>
30390 tests/check/elements/gdpdepay.c: Ignore test known to fail on PPC64. See #348114.
30391 Original commit message from CVS:
30392 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
30393 Ignore test known to fail on PPC64. See #348114.
30395 2006-08-08 08:41:13 +0000 Sjoerd Simons <sjoerd@luon.net>
30397 gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
30398 Original commit message from CVS:
30399 Patch by: Sjoerd Simons <sjoerd at luon net>
30400 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
30401 Better detection for multipart/x-mixed-replace: accept leading
30402 whitespaces before the boundary marker as well (as our very own
30403 multipartmux used to produce) (#349068).
30405 2006-08-07 08:26:03 +0000 Young-Ho Cha <ganadist@chollian.net>
30407 gst-libs/gst/riff/: Detect DTS audio streams (#350157).
30408 Original commit message from CVS:
30409 Patch by: Young-Ho Cha <ganadist at chollian net>
30410 * gst-libs/gst/riff/riff-ids.h:
30411 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
30412 (gst_riff_create_audio_template_caps):
30413 Detect DTS audio streams (#350157).
30415 2006-08-05 17:08:05 +0000 Andy Wingo <wingo@pobox.com>
30417 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (gst_theora_parse_class_init) (theora_parse_dispose, theora_par...
30418 Original commit message from CVS:
30419 2006-08-05 Andy Wingo <wingo@pobox.com>
30420 * ext/theora/gsttheoraparse.h:
30421 * ext/theora/theoraparse.c (gst_theora_parse_class_init)
30422 (theora_parse_dispose, theora_parse_set_property)
30423 (theora_parse_get_property, theora_parse_munge_granulepos)
30424 (theora_parse_push_buffer, theora_parse_change_state): Add a
30425 property 'synchronization-points' to fix badly synchronized oggs.
30427 2006-08-04 13:20:23 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
30429 gst/gdp/gstgdpdepay.c: Fix event parsing by gdpdepay. Fixes #349916.
30430 Original commit message from CVS:
30431 2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
30432 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
30433 Fix event parsing by gdpdepay. Fixes #349916.
30435 2006-08-03 15:04:42 +0000 Tim-Philipp Müller <tim@centricular.net>
30437 tests/check/: Add a few tests for the channel position stuff in libgstaudio.
30438 Original commit message from CVS:
30439 * tests/check/Makefile.am:
30440 * tests/check/libs/.cvsignore:
30441 * tests/check/libs/audio.c: (structure_contains_channel_positions),
30442 (fixed_caps_have_channel_positions), (GST_START_TEST),
30443 (audio_suite), (main):
30444 Add a few tests for the channel position stuff in libgstaudio.
30446 2006-08-03 14:16:06 +0000 Tim-Philipp Müller <tim@centricular.net>
30448 ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
30449 Original commit message from CVS:
30450 * ext/alsa/gstalsa.c: (caps_add_channel_configuration),
30451 (gst_alsa_detect_channels):
30452 * ext/alsa/gstalsasink.c:
30453 Add support for cards that (only) do more than 8 channels,
30454 like the Delta 44 (#345188).
30455 * gst-libs/gst/audio/multichannel.c:
30456 (gst_audio_check_channel_positions):
30457 * gst-libs/gst/audio/multichannel.h:
30458 API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
30459 unspecified channel position and cannot be combined with any
30460 of the other audio channel positions; adjust position layout
30461 checks accordingly (#345188).
30463 2006-08-03 11:15:53 +0000 Tim-Philipp Müller <tim@centricular.net>
30465 gst/typefind/gsttypefindfunctions.c: Recognise ancient RealAudio files (see #349779).
30466 Original commit message from CVS:
30467 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
30468 Recognise ancient RealAudio files (see #349779).
30470 2006-08-03 09:01:25 +0000 Jens Granseuer <jensgr@gmx.net>
30472 gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
30473 Original commit message from CVS:
30474 Patch by: Jens Granseuer <jensgr at gmx net>
30475 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
30476 Add typefinder for Interplay's MVE format (#348973).
30478 2006-08-02 17:03:29 +0000 Marcel Moreaux <marcelm@luon.net>
30480 gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
30481 Original commit message from CVS:
30482 Patch by: Marcel Moreaux <marcelm at luon dot net>
30483 * gst-libs/gst/rtp/gstbasertpdepayload.c:
30484 (gst_base_rtp_depayload_add_to_queue):
30485 * gst-libs/gst/rtp/gstbasertpdepayload.h:
30486 Handle RTP sequence number rollover.
30487 Disable jitterbuffer by default.
30489 2006-08-02 16:56:19 +0000 Wim Taymans <wim.taymans@gmail.com>
30491 gst/gdp/gstgdpdepay.c: Disable seeking.
30492 Original commit message from CVS:
30493 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
30494 (gst_gdp_depay_finalize), (gst_gdp_depay_sink_event),
30495 (gst_gdp_depay_src_event), (gst_gdp_depay_chain),
30496 (gst_gdp_depay_change_state):
30499 Clear adapter on disconts.
30500 Clear caps when going to READY instead of NULL
30501 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
30502 (gst_gdp_pay_init), (gst_gdp_pay_finalize), (gst_gdp_pay_reset),
30503 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
30504 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
30505 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
30506 (gst_gdp_pay_sink_event), (gst_gdp_pay_src_event),
30507 (gst_gdp_pay_change_state):
30508 * gst/gdp/gstgdppay.h:
30509 Reset payloader when going to READY.
30510 Fix leaked buffers in ->queue on push errors.
30513 Create packetizer in _init, free in _finalize.
30515 2006-07-31 08:48:36 +0000 Wim Taymans <wim.taymans@gmail.com>
30517 gst/gdp/gstgdpdepay.c: Consume all events except EOS because we generate events from the gdp payload instead. Fixes #...
30518 Original commit message from CVS:
30519 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
30520 (gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
30521 Consume all events except EOS because we generate events from
30522 the gdp payload instead. Fixes #349204
30524 2006-07-28 17:17:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30526 gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
30527 Original commit message from CVS:
30528 * gst/audioresample/gstaudioresample.c: (audioresample_stop),
30529 (audioresample_set_caps):
30530 Don't leak references to the incoming caps. Clean them up when
30532 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
30533 (gst_video_scale_finalize):
30534 Don't leak our temporary pixel buffer.
30535 * tests/check/Makefile.am:
30536 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
30537 (GST_START_TEST), (simple_launch_lines_suite):
30538 Fix leaks and re-enable the test for valgrind checking.
30540 2006-07-28 16:39:31 +0000 Sjoerd Simons <sjoerd@luon.net>
30542 gst/typefind/gsttypefindfunctions.c: Add typefind function for multipart/x-mixed-replace (#348916).
30543 Original commit message from CVS:
30544 Patch by: Sjoerd Simons <sjoerd at luon net>
30545 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
30547 Add typefind function for multipart/x-mixed-replace (#348916).
30549 2006-07-28 14:14:58 +0000 Wim Taymans <wim.taymans@gmail.com>
30551 gst/adder/gstadder.c: Fix leak in duration query.
30552 Original commit message from CVS:
30553 * gst/adder/gstadder.c: (gst_adder_setcaps),
30554 (gst_adder_query_duration):
30555 Fix leak in duration query.
30556 Reflow some docs and notes.
30558 2006-07-28 13:42:48 +0000 Michael Smith <msmith@xiph.org>
30560 tests/check/pipelines/vorbisenc.c: Enable Andy's extra vorbisenc test, now that it passes. Also fix one aspect of it.
30561 Original commit message from CVS:
30562 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
30564 Enable Andy's extra vorbisenc test, now that it passes. Also fix one
30567 2006-07-28 12:48:21 +0000 Michael Smith <msmith@xiph.org>
30569 ext/vorbis/vorbisenc.*: Handle discontinuities in the input vorbis stream correctly, so that the output is properly t...
30570 Original commit message from CVS:
30571 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
30572 (gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
30573 (gst_vorbis_enc_push_buffer),
30574 (gst_vorbis_enc_buffer_check_discontinuous),
30575 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
30576 * ext/vorbis/vorbisenc.h:
30577 Handle discontinuities in the input vorbis stream correctly,
30578 so that the output is properly timestamped (and has good granulepos
30579 values). Needs some oggmux fixes too.
30581 2006-07-27 10:52:52 +0000 Kai Vehmanen <kv2004@eca.cx>
30583 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
30584 Original commit message from CVS:
30585 patch by: Kai Vehmanen <kv2004 eca cx>
30586 * gst-libs/gst/rtp/gstbasertpdepayload.c:
30587 (gst_base_rtp_depayload_chain),
30588 (gst_base_rtp_depayload_handle_sink_event),
30589 (gst_base_rtp_depayload_change_state):
30590 Don't send multiple newsegments with different formats.
30593 2006-07-26 15:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
30595 ext/ogg/gstoggdemux.c: Make seeking in ogg more accurate again by doing the more correct granuletime to stream time c...
30596 Original commit message from CVS:
30597 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
30598 (gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
30599 Make seeking in ogg more accurate again by doing the more correct
30600 granuletime to stream time conversion.
30602 2006-07-26 10:59:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30604 gst/tcp/gstmultifdsink.c: debug a little more understandably do not use goto as a substitute for break, especially if...
30605 Original commit message from CVS:
30606 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
30607 (gst_multi_fd_sink_new_client):
30608 debug a little more understandably
30609 do not use goto as a substitute for break, especially if
30610 break is also being used
30612 2006-07-26 10:55:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30614 * gst/tcp/gsttcp.c:
30615 move a recurring normal event to LOG, where it should be
30616 Original commit message from CVS:
30617 move a recurring normal event to LOG, where it should be
30619 2006-07-26 10:54:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30621 * ext/vorbis/vorbisdec.c:
30623 Original commit message from CVS:
30626 2006-07-26 10:52:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30628 gst/gdp/gstgdpdepay.c: proxying get/set caps is the wrong thing to do, since we really do change caps quite fundament...
30629 Original commit message from CVS:
30630 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
30631 proxying get/set caps is the wrong thing to do, since we really
30632 do change caps quite fundamentally
30633 * tests/check/elements/gdpdepay.c:
30634 * tests/check/elements/gdppay.c:
30635 remove declaration of buffers, it's already done in gstcheck.h
30637 2006-07-26 10:31:06 +0000 Tim-Philipp Müller <tim@centricular.net>
30639 gst/playback/: Remove GLib-2.6 compatibility cruft.
30640 Original commit message from CVS:
30641 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
30642 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
30643 Remove GLib-2.6 compatibility cruft.
30645 2006-07-24 16:47:10 +0000 Wim Taymans <wim.taymans@gmail.com>
30647 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
30648 Original commit message from CVS:
30649 * gst-libs/gst/audio/gstbaseaudiosink.c:
30650 (gst_base_audio_sink_render):
30651 Don't try to align a sample to an unknown value.
30653 2006-07-24 15:14:17 +0000 Wim Taymans <wim.taymans@gmail.com>
30655 gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
30656 Original commit message from CVS:
30657 * gst-libs/gst/audio/gstbaseaudiosink.c:
30658 (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
30659 When the audio clock is slaved to another clock, never try to align
30660 samples but trust the rate interpolation algorithm.
30662 2006-07-24 14:34:42 +0000 Wim Taymans <wim.taymans@gmail.com>
30664 ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
30665 Original commit message from CVS:
30666 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
30667 Don't try to calculate silence samples, base class does this much
30669 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
30670 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
30671 (gst_ring_buffer_acquire):
30672 Calculate silence samples correctly.
30673 * gst-libs/gst/audio/gstringbuffer.h:
30676 2006-07-22 17:01:12 +0000 Tim-Philipp Müller <tim@centricular.net>
30678 gst/typefind/gsttypefindfunctions.c: Limit search for the first markup tag to the first few kB of the file. If we don...
30679 Original commit message from CVS:
30680 * gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
30681 Limit search for the first markup tag to the first few kB of
30682 the file. If we don't find one there, it's highly unlikely that
30683 this is an XML(-ish) file.
30685 2006-07-21 17:04:06 +0000 Andy Wingo <wingo@pobox.com>
30687 tests/check/pipelines/theoraenc.c (test_discontinuity): Similar test to the one in vorbisenc. Also commented out.
30688 Original commit message from CVS:
30689 2006-07-21 Andy Wingo <wingo@pobox.com>
30690 * tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
30691 test to the one in vorbisenc. Also commented out.
30693 2006-07-21 16:54:19 +0000 Andy Wingo <wingo@pobox.com>
30695 tests/check/pipelines/vorbisenc.c: New test, commented out until Mike lands some elite vorbisenc patches.
30696 Original commit message from CVS:
30697 2006-07-21 Andy Wingo <wingo@pobox.com>
30698 * tests/check/pipelines/vorbisenc.c:
30699 (test_discontinuity): New test, commented out until Mike lands
30700 some elite vorbisenc patches.
30702 2006-07-21 15:59:24 +0000 Andy Wingo <wingo@pobox.com>
30704 tests/check/pipelines/: Port to bufferstraw.
30705 Original commit message from CVS:
30706 2006-07-21 Andy Wingo <wingo@pobox.com>
30707 * tests/check/pipelines/vorbisenc.c:
30708 * tests/check/pipelines/theoraenc.c: Port to bufferstraw.
30709 Bufferstraw was actually factored out of these tests. Now we share
30712 2006-07-21 11:03:28 +0000 Wim Taymans <wim.taymans@gmail.com>
30714 ext/theora/theoradec.c: Better clipping.
30715 Original commit message from CVS:
30716 * ext/theora/theoradec.c: (clip_buffer):
30719 2006-07-21 10:43:54 +0000 Wim Taymans <wim.taymans@gmail.com>
30721 gst-libs/gst/audio/gstaudiosink.c: Fix leak.
30722 Original commit message from CVS:
30723 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
30724 (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
30725 (gst_audioringbuffer_release), (gst_audioringbuffer_stop):
30727 Avoid type casting when we can.
30728 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
30731 2006-07-20 16:57:29 +0000 Tim-Philipp Müller <tim@centricular.net>
30733 ext/alsa/gstalsamixerelement.c: Make state change fail if the specified device can't be opened for some reason.
30734 Original commit message from CVS:
30735 * ext/alsa/gstalsamixerelement.c:
30736 (gst_alsa_mixer_element_change_state):
30737 Make state change fail if the specified device can't be opened
30740 2006-07-20 10:42:21 +0000 Wim Taymans <wim.taymans@gmail.com>
30742 gst/playback/test.c: Example of a small audio/video player using decodebin.
30743 Original commit message from CVS:
30744 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
30745 (cb_newpad), (main):
30746 Example of a small audio/video player using decodebin.
30748 2006-07-20 05:56:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30750 gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
30751 Original commit message from CVS:
30752 * gst-libs/gst/riff/riff-ids.h:
30753 Add 'fact' chunk id
30755 2006-07-19 18:20:43 +0000 Wim Taymans <wim.taymans@gmail.com>
30757 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't assert when not negotiated but post a meaningfull error message. Fixes ...
30758 Original commit message from CVS:
30759 * gst-libs/gst/rtp/gstbasertpdepayload.c:
30760 (gst_base_rtp_depayload_chain),
30761 (gst_base_rtp_depayload_change_state):
30762 Don't assert when not negotiated but post a meaningfull
30763 error message. Fixes #347918.
30764 * gst-libs/gst/rtp/gstbasertppayload.c:
30765 Add comment about better default MTU size.
30766 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
30767 Small cleanups, start docs.
30769 2006-07-19 14:46:36 +0000 Martin Szulecki <compiz@sukimashita.com>
30771 sys/v4l/gstv4lelement.c: If "device-name" is requested and the device is not open, try to temporarily open it to obta...
30772 Original commit message from CVS:
30773 Patch by: Martin Szulecki
30774 * sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
30775 If "device-name" is requested and the device is not
30776 open, try to temporarily open it to obtain this
30777 information (#342494).
30779 2006-07-19 12:25:00 +0000 Tim-Philipp Müller <tim@centricular.net>
30781 gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
30782 Original commit message from CVS:
30783 * gst-libs/gst/tag/gstid3tag.c:
30784 Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
30785 * gst-libs/gst/tag/gsttageditingprivate.h:
30786 * gst-libs/gst/tag/gstvorbistag.c:
30787 Some more random const-ifications.
30789 2006-07-18 19:48:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30791 gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
30792 Original commit message from CVS:
30793 * gst-libs/gst/riff/riff-ids.h:
30794 * gst-libs/gst/riff/riff-media.c:
30795 (gst_riff_create_video_template_caps):
30796 Add more FOURCCs (sort list to make stuff easier to find),
30797 add comment what those 16 bytes in struct _gst_riff_strh according to
30800 2006-07-17 14:17:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30802 gst/gdp/: remove parent_class setting, BOILERPLATE does this fix typo in comment
30803 Original commit message from CVS:
30804 2006-07-17 Thomas Vander Stichele <thomas at apestaart dot org>
30805 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
30806 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
30807 remove parent_class setting, BOILERPLATE does this
30808 (gst_gdp_pay_reset_streamheader):
30809 fix typo in comment
30811 2006-07-17 13:48:10 +0000 Tim-Philipp Müller <tim@centricular.net>
30813 gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
30814 Original commit message from CVS:
30815 * gst-libs/gst/audio/multichannel.c:
30816 (gst_audio_check_channel_positions),
30817 (gst_audio_fixate_channel_positions):
30818 Const-ify two arrays.
30820 2006-07-17 12:33:42 +0000 Tim-Philipp Müller <tim@centricular.net>
30822 ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
30823 Original commit message from CVS:
30824 * ext/alsa/gstalsa.c: (caps_add_channel_configuration):
30825 Fix typo, so that alsasink also advertises 8 channels
30826 if that's supported (tags: can, worms, open, alsa, ph34r).
30828 2006-07-17 12:01:04 +0000 Wim Taymans <wim.taymans@gmail.com>
30830 ext/ogg/gstoggdemux.c: *sigh*, when is the compiler going to warn when the comments are out-of-sync with the code.. R...
30831 Original commit message from CVS:
30832 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
30833 (gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
30834 *sigh*, when is the compiler going to warn when the comments
30835 are out-of-sync with the code.. Refix case of busted theora
30836 headers with 0 granule pos.
30838 2006-07-14 17:56:59 +0000 Wim Taymans <wim.taymans@gmail.com>
30840 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix 99% cpu load by waiting for absolute times on the clock. Fixes #347300.
30841 Original commit message from CVS:
30842 * gst-libs/gst/rtp/gstbasertpdepayload.c:
30843 (gst_base_rtp_depayload_wait),
30844 (gst_base_rtp_depayload_change_state),
30845 (gst_base_rtp_depayload_set_property),
30846 (gst_base_rtp_depayload_get_property):
30847 Fix 99% cpu load by waiting for absolute times on the
30848 clock. Fixes #347300.
30850 2006-07-14 17:07:08 +0000 Andy Wingo <wingo@pobox.com>
30852 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (theora_parse_drain_event_queue) (theora_parse_push_headers, th...
30853 Original commit message from CVS:
30854 2006-07-14 Andy Wingo <wingo@pobox.com>
30855 * ext/theora/gsttheoraparse.h:
30856 * ext/theora/theoraparse.c (theora_parse_drain_event_queue)
30857 (theora_parse_push_headers, theora_parse_clear_queue)
30858 (theora_parse_drain_queue_prematurely, )
30859 (theora_parse_sink_event, theora_parse_change_state): Queue events
30860 until we initialized our state, like in vorbisparse.
30862 2006-07-14 16:45:17 +0000 Iain * <iaingnome@gmail.com>
30864 ext/vorbis/vorbisparse.h: ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue) (vorbis_parse_push_headers, vorbi...
30865 Original commit message from CVS:
30866 2006-07-14 Andy Wingo <wingo@pobox.com>
30867 * ext/vorbis/vorbisparse.h:
30868 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
30869 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
30870 (vorbis_parse_drain_queue_prematurely, )
30871 (vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
30872 until we have initialized our state. Fixes seeking after an
30874 2006-07-14 Andy Wingo <wingo@pobox.com>
30875 Patch by: Iain * <iaingnome@gmail.com>
30876 * ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.
30878 2006-07-14 15:52:39 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30880 configure.ac: Bump nano back to CVS
30881 Original commit message from CVS:
30883 Bump nano back to CVS
30885 === release 0.10.9 ===
30887 2006-07-14 15:51:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30889 configure.ac: releasing 0.10.9, "I walk the line"
30890 Original commit message from CVS:
30891 2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
30893 releasing 0.10.9, "I walk the line"
30895 2006-07-14 14:12:40 +0000 Michael Smith <msmith@xiph.org>
30897 tests/check/pipelines/vorbisenc.c: Move a g_cond_signal to earlier to avoid sometimes deadlocking (commonly happens w...
30898 Original commit message from CVS:
30899 * tests/check/pipelines/vorbisenc.c: (stop_pipeline):
30900 Move a g_cond_signal to earlier to avoid sometimes deadlocking
30901 (commonly happens when running this test under valgrind) when trying
30902 to remove the buffer probe.
30904 2006-07-14 10:34:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30906 * gst/gdp/Makefile.am:
30907 build as a plugin, not a lib
30908 Original commit message from CVS:
30909 build as a plugin, not a lib
30911 2006-07-13 16:43:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30913 sys/xvimage/xvimagesink.c: Fix missing g_unlock from the previous commit
30914 Original commit message from CVS:
30915 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
30916 Fix missing g_unlock from the previous commit
30918 2006-07-13 16:34:04 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30920 sys/: Implement a locking order to ensure we always take the object lock before the x_lock and never vice-versa.
30921 Original commit message from CVS:
30922 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
30923 (gst_ximagesink_change_state):
30924 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
30925 (gst_xvimagesink_change_state):
30926 Implement a locking order to ensure we always take the object lock
30927 before the x_lock and never vice-versa.
30929 2006-07-13 15:25:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30931 docs/plugins/: add more plugins and elements to docs
30932 Original commit message from CVS:
30933 * docs/plugins/Makefile.am:
30934 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
30935 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
30936 add more plugins and elements to docs
30937 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
30938 fix segfaults due to wrong g_free
30940 * gst/gdp/gstgdppay.c:
30943 2006-07-13 14:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30945 gst/playback/gstdecodebin.c: Fix a caps leak when linking (#347304)
30946 Original commit message from CVS:
30947 * gst/playback/gstdecodebin.c: (find_compatibles):
30948 Fix a caps leak when linking (#347304)
30949 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
30950 (gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
30951 (gst_ximagesink_change_state):
30952 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
30953 (gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
30954 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
30955 (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
30956 Don't leak shared memory resources. Use the object lock to protect
30957 against the xcontext disappearing while returning a buffer from the
30958 pipeline. (#347304)
30960 2006-07-12 14:20:43 +0000 Edward Hervey <bilboed@bilboed.com>
30962 ext/vorbis/vorbisdec.c: gst_tag_list_merge() returns a new object. Take that into account when using it. This avoids ...
30963 Original commit message from CVS:
30964 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
30965 (vorbis_handle_comment_packet):
30966 gst_tag_list_merge() returns a new object. Take that into account when
30967 using it. This avoids memleak.
30968 Revert previous commit which is not needed.
30970 2006-07-12 13:30:20 +0000 Edward Hervey <bilboed@bilboed.com>
30972 ext/vorbis/vorbisdec.c: Reset the decoder in finalize so that all fields get cleared.
30973 Original commit message from CVS:
30974 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
30975 Reset the decoder in finalize so that all fields get cleared.
30977 2006-07-12 13:24:19 +0000 Wim Taymans <wim.taymans@gmail.com>
30979 gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
30980 Original commit message from CVS:
30981 * gst-libs/gst/audio/gstbaseaudiosrc.c:
30982 (gst_base_audio_src_set_clock),
30983 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
30984 Don't try to post an error message when setting the clock fails
30985 as this can happen when adding an element to a bin which will then
30986 deadlock. Fixes #347296.
30988 2006-07-12 13:04:15 +0000 Edward Hervey <bilboed@bilboed.com>
30990 ext/vorbis/vorbisdec.c: Post tag messages on the bus even if we're not initialized.
30991 Original commit message from CVS:
30992 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
30993 (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
30994 (vorbis_handle_type_packet):
30995 Post tag messages on the bus even if we're not initialized.
30996 If we're not initialized, we still postpone the event pushing of tags.
30998 2006-07-12 11:28:37 +0000 Wim Taymans <wim.taymans@gmail.com>
31000 Revert last two changes that broke the freeze.
31001 Original commit message from CVS:
31002 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
31003 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
31004 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
31005 Revert last two changes that broke the freeze.
31007 2006-07-12 10:59:55 +0000 Wim Taymans <wim.taymans@gmail.com>
31009 ext/alsa/gstalsasink.c: basesink calculates silence sample correctly for us.
31010 Original commit message from CVS:
31011 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
31012 basesink calculates silence sample correctly for us.
31014 2006-07-12 10:58:42 +0000 Wim Taymans <wim.taymans@gmail.com>
31016 gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
31017 Original commit message from CVS:
31018 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
31019 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
31020 Calculate correct silence samples so we don't fill our ringbuffer
31023 2006-07-12 10:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
31025 ext/vorbis/vorbisdec.*: Delay sending events (newsegment, tags) until the decoder is properly initialized.
31026 Original commit message from CVS:
31027 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
31028 (gst_vorbis_dec_reset), (vorbis_dec_sink_event),
31029 (vorbis_handle_comment_packet), (vorbis_handle_type_packet):
31030 * ext/vorbis/vorbisdec.h:
31031 Delay sending events (newsegment, tags) until the decoder is properly
31035 2006-07-11 22:40:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31052 Original commit message from CVS:
31055 2006-07-11 21:04:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31057 tests/check/elements/audioconvert.c: Patch from #347221 adding a test for audioconvert channel remappings.
31058 Original commit message from CVS:
31059 * tests/check/elements/audioconvert.c: (get_float_mc_caps),
31060 (get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
31061 Patch from #347221 adding a test for audioconvert
31062 channel remappings.
31064 2006-07-11 12:03:25 +0000 Tim-Philipp Müller <tim@centricular.net>
31066 gst/subparse/gstssaparse.c: Don't include the terminating NUL in the buffer size, it's only there for extra paranoia ...
31067 Original commit message from CVS:
31068 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
31069 (gst_ssa_parse_parse_line):
31070 Don't include the terminating NUL in the buffer size,
31071 it's only there for extra paranoia (would add random
31072 '*' characters at the end of each subtitle since the
31073 terminator itself is not valid UTF-8 technically).
31074 Also fix indenting after boilerplate macro.
31076 2006-07-10 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.net>
31078 gst/playback/gstdecodebin.c: Also emit 'unknown-type' signal (which should really be called unhandled-type) if we fou...
31079 Original commit message from CVS:
31080 * gst/playback/gstdecodebin.c: (close_pad_link):
31081 Also emit 'unknown-type' signal (which should really be
31082 called unhandled-type) if we found potential decoders/demuxers
31083 in the registry but none of them worked in the end (as in the
31084 case where the plugins don't exist any longer but are still
31085 listed in the registry). Fixes #329798.
31087 2006-07-08 13:48:58 +0000 Andy Wingo <wingo@pobox.com>
31090 * ext/theora/theoraparse.c:
31091 theoraparse.c (theora_parse_push_buffer)
31092 Original commit message from CVS:
31093 2006-07-08 Andy Wingo <wingo@pobox.com>
31094 * theoraparse.c (theora_parse_push_buffer)
31095 (theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
31096 Add some more debugging. Fix granulepos reconstruction in the face
31097 of discontinuities.
31099 2006-07-06 15:54:50 +0000 Wim Taymans <wim.taymans@gmail.com>
31101 gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
31102 Original commit message from CVS:
31103 * gst-libs/gst/audio/gstbaseaudiosink.c:
31104 (gst_base_audio_sink_class_init),
31105 (gst_base_audio_sink_provide_clock):
31106 Use gobject_class instead of G_OBJECT_CLASS (klass)
31107 * gst-libs/gst/audio/gstbaseaudiosrc.c:
31108 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
31109 (gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
31110 (gst_base_audio_src_get_time),
31111 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
31112 (gst_base_audio_src_create_ringbuffer):
31113 Fix latency and buffer-time constants and properties ala basesink.
31114 Implement pull based scheduling. Fixes #346527.
31115 Set default blocksize in GstBaseSrc to 0, we default to pushing out
31117 Refuse slaving to another clock instead of silently not working.
31118 Only provide a clock when we are actually able to do so.
31119 Various small cleanups and compiler hints.
31121 2006-07-06 13:23:07 +0000 Lutz Mueller <lutz@topfrose.de>
31123 gst/typefind/gsttypefindfunctions.c: Add typefinding for text/html (#346581).
31124 Original commit message from CVS:
31125 Patch by: Lutz Mueller <lutz at topfrose de>
31126 * gst/typefind/gsttypefindfunctions.c: (html_type_find),
31128 Add typefinding for text/html (#346581).
31130 2006-07-06 13:12:02 +0000 Tim-Philipp Müller <tim@centricular.net>
31132 gst/typefind/gsttypefindfunctions.c: Fix SMIL typefinding, make xml_check_first_element() more useful.
31133 Original commit message from CVS:
31134 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
31135 (xml_check_first_element), (xml_type_find), (smil_type_find):
31136 Fix SMIL typefinding, make xml_check_first_element() more
31139 2006-07-06 13:04:24 +0000 Tim-Philipp Müller <tim@centricular.net>
31141 gst/playback/gstplaybasebin.*: Protect list of elements with a subtitle-encoding property and the subtitle encoding m...
31142 Original commit message from CVS:
31143 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
31144 (gst_play_base_bin_finalize), (decodebin_element_added_cb),
31145 (decodebin_element_removed_cb), (gst_play_base_bin_set_property):
31146 * gst/playback/gstplaybasebin.h:
31147 Protect list of elements with a subtitle-encoding property and
31148 the subtitle encoding member itself with a lock of their own
31149 instead of using the object lock. This prevents a dead-lock in
31150 the element-remove callback in some circumstances when shutting
31153 2006-07-05 20:11:13 +0000 Sébastien Moutte <sebastien@moutte.net>
31155 win32/common/libgsttag.def: Export some new functions.
31156 Original commit message from CVS:
31157 * win32/common/libgsttag.def:
31158 Export some new functions.
31159 * win32/vs6/libgstogg.dsp:
31160 Add a link to libgsttag-0.10.lib.
31162 2006-07-04 16:50:21 +0000 Tim-Philipp Müller <tim@centricular.net>
31164 ext/alsa/gstalsamixertrack.c: Some const-ification.
31165 Original commit message from CVS:
31166 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
31167 Some const-ification.
31169 2006-07-04 14:06:03 +0000 Wim Taymans <wim.taymans@gmail.com>
31171 gst/playback/gstplaybasebin.c: Improve checking if we are dealing with a stream. Added some more uris that need buffe...
31172 Original commit message from CVS:
31173 * gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
31174 Improve checking if we are dealing with a stream. Added some
31175 more uris that need buffering.
31177 2006-07-03 10:43:31 +0000 Edward Hervey <bilboed@bilboed.com>
31179 ext/vorbis/vorbisdec.c: Remove unused variable.
31180 Original commit message from CVS:
31181 * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
31182 Remove unused variable.
31184 2006-07-02 21:48:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31186 Makefile.am: include lcov.mak
31187 Original commit message from CVS:
31191 add GCOV_LIBS to GST_LIBS
31193 2006-07-02 11:08:58 +0000 Michael Sheldon <webmaster@mikeasoft.com>
31195 ext/alsa/gstalsasrc.c: Add 32 bps to template caps and increase channels range from [1,2] to [1,MAX]. See #346326.
31196 Original commit message from CVS:
31197 Patch by: Michael Sheldon <webmaster at mikeasoft com>
31198 * ext/alsa/gstalsasrc.c:
31199 Add 32 bps to template caps and increase channels range
31200 from [1,2] to [1,MAX]. See #346326.
31202 2006-06-30 12:04:51 +0000 Tim-Philipp Müller <tim@centricular.net>
31204 gst-libs/gst/riff/riff-media.c: Recognise 'WMVA' video codec fourcc (#345879).
31205 Original commit message from CVS:
31206 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
31207 Recognise 'WMVA' video codec fourcc (#345879).
31209 2006-06-29 12:21:06 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
31211 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
31212 Original commit message from CVS:
31213 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
31214 Fixed nasty memory leak
31216 2006-06-26 13:19:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31218 gst/tcp/gsttcp.c: fix logging
31219 Original commit message from CVS:
31220 * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
31221 (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
31224 2006-06-23 16:45:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31226 gst/playback/gstdecodebin.c: Protect remove_fakesink using a mutex, so that we don't try and remove the fakesink simu...
31227 Original commit message from CVS:
31228 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
31229 (gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
31230 (remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
31231 Protect remove_fakesink using a mutex, so that we don't try and
31232 remove the fakesink simultaneously from multiple threads.
31233 When going from READY to PAUSED, restore the fakesink, so that
31234 it is there when decodebin gets reused.
31236 2006-06-23 09:53:09 +0000 Tim-Philipp Müller <tim@centricular.net>
31238 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
31239 Original commit message from CVS:
31240 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
31241 * gst-libs/gst/rtp/gstbasertpdepayload.c:
31242 * gst-libs/gst/rtp/gstbasertppayload.c:
31243 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
31244 * gst/tcp/gstmultifdsink.c:
31245 * gst/tcp/gsttcpclientsink.c:
31246 * gst/tcp/gsttcpclientsrc.c:
31247 * gst/tcp/gsttcpserversink.c:
31248 * gst/tcp/gsttcpserversrc.c:
31249 * gst/videorate/gstvideorate.c:
31250 * gst/videotestsrc/gstvideotestsrc.c:
31251 * sys/v4l/gstv4ljpegsrc.c:
31252 * sys/v4l/gstv4lmjpegsink.c:
31253 * sys/v4l/gstv4lsrc.c:
31254 * tests/examples/seek/scrubby.c:
31255 * tests/examples/seek/seek.c:
31256 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
31258 2006-06-23 09:09:44 +0000 Tim-Philipp Müller <tim@centricular.net>
31260 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) and fix one GObject boilerplate macro.
31261 Original commit message from CVS:
31262 * ext/directfb/dfbvideosink.c:
31263 * ext/gsm/gstgsmdec.c:
31264 * ext/gsm/gstgsmenc.c:
31265 * ext/libmms/gstmms.c:
31266 * ext/neon/gstneonhttpsrc.c:
31267 * ext/theora/theoradec.c:
31268 * gst/freeze/gstfreeze.c:
31269 * gst/gdp/gstgdpdepay.c:
31270 * gst/gdp/gstgdppay.c:
31271 * sys/glsink/glimagesink.c:
31272 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
31273 and fix one GObject boilerplate macro.
31275 2006-06-22 12:13:31 +0000 Tim-Philipp Müller <tim@centricular.net>
31277 gst-libs/gst/tag/tags.c: Second field in GEnumValue shouldn't be a description, but a stringified version of the enum...
31278 Original commit message from CVS:
31279 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
31280 Second field in GEnumValue shouldn't be a description,
31281 but a stringified version of the enum value.
31283 2006-06-22 12:03:14 +0000 Wim Taymans <wim.taymans@gmail.com>
31285 sys/ximage/ximagesink.c: Avoid type checking in buffer casts.
31286 Original commit message from CVS:
31287 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
31288 (gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
31289 (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
31290 Avoid type checking in buffer casts.
31291 Avoid caps copy in buffer_alloc when we can.
31292 Use pad_peer_accept.
31294 2006-06-22 11:01:41 +0000 Tim-Philipp Müller <tim@centricular.net>
31296 gst-libs/gst/tag/tag.h: Oops, make that 'Since: 0.10.9'.
31297 Original commit message from CVS:
31298 * gst-libs/gst/tag/tag.h:
31299 Oops, make that 'Since: 0.10.9'.
31301 2006-06-22 10:55:05 +0000 Tim-Philipp Müller <tim@centricular.net>
31303 API: add GstTagImageType enum to describe images contained in image tags (#345641).
31304 Original commit message from CVS:
31305 * docs/libs/gst-plugins-base-libs-sections.txt:
31306 * gst-libs/gst/tag/tag.h:
31307 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
31308 (gst_tag_image_type_get_type):
31309 API: add GstTagImageType enum to describe images contained
31310 in image tags (#345641).
31312 2006-06-22 10:31:22 +0000 Tim-Philipp Müller <tim@centricular.net>
31314 gst/tcp/gstmultifdsink.c: Fix warnings with gst-inspect: "buffers-min" property should be of G_TYPE_INT and not G_TYP...
31315 Original commit message from CVS:
31316 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
31317 Fix warnings with gst-inspect: "buffers-min" property
31318 should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
31319 typo in property description.
31321 2006-06-22 10:10:51 +0000 Cody Russell <bratsche@gnome.org>
31323 gst/: Avoid unnecessary class cast check in class_init functions (#337747).
31324 Original commit message from CVS:
31325 Patch by: Cody Russell <bratsche at gnome org>
31326 * gst/audioresample/gstaudioresample.c:
31327 (gst_audioresample_class_init):
31328 * gst/playback/gststreamselector.c:
31329 (gst_stream_selector_class_init):
31330 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
31331 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
31332 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
31333 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
31334 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
31335 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
31336 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
31337 * gst/videotestsrc/gstvideotestsrc.c:
31338 (gst_video_test_src_class_init):
31339 * gst/volume/gstvolume.c: (gst_volume_class_init):
31340 Avoid unnecessary class cast check in class_init
31341 functions (#337747).
31343 2006-06-21 18:39:07 +0000 Tim-Philipp Müller <tim@centricular.net>
31345 ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
31346 Original commit message from CVS:
31347 * ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
31348 (gst_text_overlay_video_chain):
31349 g_markup_escape_text() REALLY doesn't like non-UTF8 input
31350 and doesn't validate its input either (and neither did
31351 textoverlay it seems). Let's do that then and fix #345206.
31353 2006-06-19 17:12:57 +0000 Wim Taymans <wim.taymans@gmail.com>
31355 gst/tcp/gstmultifdsink.*: Added shiny new burst-on-connect methods.
31356 Original commit message from CVS:
31357 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
31358 (gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
31359 (gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
31360 (gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
31361 (find_syncframe), (find_limits), (assign_value),
31362 (count_burst_unit), (gst_multi_fd_sink_new_client),
31363 (gst_multi_fd_sink_handle_client_write),
31364 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
31365 (gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
31366 (gst_multi_fd_sink_change_state):
31367 * gst/tcp/gstmultifdsink.h:
31368 Added shiny new burst-on-connect methods.
31369 Add properties to control the minimal amount of data queued.
31371 API: bytes-min property
31372 API: time-min property
31373 API: buffers-min property
31374 API: burst-unit property
31375 API: burst-value property
31376 API: add-full signal
31377 * gst/tcp/gsttcp-marshal.list:
31378 Added new marshaller code for the new signal.
31379 * tests/check/elements/multifdsink.c: (GST_START_TEST),
31380 (multifdsink_suite):
31381 Added testcases for new burst methods.
31383 2006-06-19 11:35:47 +0000 Christian Schaller <uraeus@gnome.org>
31385 * gst-plugins-base.spec.in:
31386 update for latest changes
31387 Original commit message from CVS:
31388 update for latest changes
31390 2006-06-19 09:57:50 +0000 Edward Hervey <bilboed@bilboed.com>
31392 ext/theora/theoradec.c: Implement clipping for accurate seeking.
31393 Original commit message from CVS:
31394 * ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
31395 Implement clipping for accurate seeking.
31398 2006-06-19 09:08:05 +0000 Philip Jaegenstedt <philip@lysator.liu.se>
31400 gst/videoscale/gstvideoscale.c: Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
31401 Original commit message from CVS:
31402 Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
31403 * gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
31404 (gst_video_scale_transform):
31405 Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
31407 2006-06-17 14:18:41 +0000 Tim-Philipp Müller <tim@centricular.net>
31411 Original commit message from CVS:
31414 2006-06-17 14:13:03 +0000 Tim-Philipp Müller <tim@centricular.net>
31416 configure.ac: Fix --disable-extern (can't set conditionals conditionally, #343602).
31417 Original commit message from CVS:
31419 Fix --disable-extern (can't set conditionals conditionally,
31422 2006-06-16 15:43:23 +0000 Tim-Philipp Müller <tim@centricular.net>
31424 tests/check/elements/audioresample.c: Add test case for bug #342789 fixed below.
31425 Original commit message from CVS:
31426 * tests/check/elements/audioresample.c: (test_reuse),
31427 (audioresample_suite):
31428 Add test case for bug #342789 fixed below.
31430 2006-06-16 15:17:44 +0000 Tim-Philipp Müller <tim@centricular.net>
31432 gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
31433 Original commit message from CVS:
31434 * gst/audioresample/gstaudioresample.c:
31435 (gst_audioresample_class_init), (gst_audioresample_init),
31436 (audioresample_start), (audioresample_stop),
31437 (gst_audioresample_set_property), (gst_audioresample_get_property):
31438 Implement GstBaseTransform::start and ::stop so that audioresample
31439 can clear its internal state properly and be reused insted of
31440 causing non-negotiated errors with playbin under some circumstances
31442 * tests/check/elements/audioresample.c: (setup_audioresample),
31443 (cleanup_audioresample):
31444 Need to set element state here so that ::start and ::stop are
31447 2006-06-16 13:59:29 +0000 Young-Ho Cha <ganadist@chollian.net>
31449 gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
31450 Original commit message from CVS:
31451 Patch by: Young-Ho Cha <ganadist at chollian dot net>
31452 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
31453 Parse extra data better, apparently it's right behind
31454 the normal strf header size. Fixes #343500.
31456 2006-06-16 11:04:21 +0000 Wim Taymans <wim.taymans@gmail.com>
31458 ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
31459 Original commit message from CVS:
31460 * ext/alsa/gstalsasink.c: (set_hwparams):
31461 If we fail to set the buffer_time and period_time alsa
31462 parameters, post a warning and leave alsa select a
31463 default instead of failing. Fixes #342085
31465 2006-06-16 10:30:25 +0000 Tim-Philipp Müller <tim@centricular.net>
31468 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
31469 Original commit message from CVS:
31470 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
31472 2006-06-16 10:20:10 +0000 Tim-Philipp Müller <tim@centricular.net>
31474 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.
31475 Original commit message from CVS:
31476 * docs/libs/gst-plugins-base-libs-sections.txt:
31477 * gst-libs/gst/cdda/gstcddabasesrc.h:
31478 Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
31479 out in the header file and shouldn't be listed in the docs.
31480 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
31481 Fix it so that it doesn't crash in the debug statement.
31483 2006-06-16 10:02:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31485 docs/libs/: add remaining symbols into correct setions
31486 Original commit message from CVS:
31487 * docs/libs/Makefile.am:
31488 * docs/libs/gst-plugins-base-libs-docs.sgml:
31489 * docs/libs/gst-plugins-base-libs-sections.txt:
31490 * docs/libs/gst-plugins-base-libs.types:
31491 add remaining symbols into correct setions
31492 * gst-libs/gst/audio/gstringbuffer.c:
31493 fix incomplete docs
31494 * gst-libs/gst/audio/gstringbuffer.h:
31495 comment out not yet implemented function
31496 * gst-libs/gst/floatcast/floatcast.h:
31497 * gst-libs/gst/netbuffer/gstnetbuffer.c:
31498 add short descriptions
31499 * gst-libs/gst/interfaces/propertyprobe.c:
31500 fix return value docs
31501 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
31502 simplify debug logging
31503 * gst-libs/gst/riff/riff-read.h:
31504 sync function prototype and docs
31505 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
31506 remove left over symbol
31508 2006-06-16 09:56:41 +0000 Tim-Philipp Müller <tim@centricular.net>
31510 Use GST_PLUGIN_DOCS macro in configure.ac, add
31511 Original commit message from CVS:
31514 * docs/Makefile.am:
31515 Use GST_PLUGIN_DOCS macro in configure.ac, add
31516 --enable-plugin-docs default to autogen.sh and use
31517 ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
31519 2006-06-15 15:27:49 +0000 Wim Taymans <wim.taymans@gmail.com>
31521 ext/ogg/gstoggdemux.c: Combine GstFlowReturn from the source pads to give a meaningfull result to the upstream peer o...
31522 Original commit message from CVS:
31523 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
31524 (gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
31525 (gst_ogg_demux_loop):
31526 Combine GstFlowReturn from the source pads to give a
31527 meaningfull result to the upstream peer or to stop the
31528 processing task in case of errors.
31530 2006-06-14 14:49:33 +0000 Tim-Philipp Müller <tim@centricular.net>
31532 gst/playback/gststreaminfo.c: Try GST_TAG_CODEC as fallback when extracting the codec name; more debug info.
31533 Original commit message from CVS:
31534 * gst/playback/gststreaminfo.c: (cb_probe):
31535 Try GST_TAG_CODEC as fallback when extracting the
31536 codec name; more debug info.
31538 2006-06-14 14:34:28 +0000 Tim-Philipp Müller <tim@centricular.net>
31540 ext/ogg/: Extract language tags from ogm subtitle streams, so that the subtitle menu choices are labelled correctly in
31541 Original commit message from CVS:
31542 * ext/ogg/Makefile.am:
31543 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
31544 Extract language tags from ogm subtitle streams, so that
31545 the subtitle menu choices are labelled correctly in
31546 Totem (fixes #344708).
31548 2006-06-14 09:13:54 +0000 Alessandro Decina <alessandro@nnva.org>
31550 ext/ogg/gstoggmux.c: Fix various leaks. Fixes #343699.
31551 Original commit message from CVS:
31552 Patch by: Alessandro Decina <alessandro at nnva dot org>
31553 * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
31554 (gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
31555 (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
31556 (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
31557 Fix various leaks. Fixes #343699.
31558 Add x-smoke mime type.
31560 2006-06-14 08:17:45 +0000 Tim-Philipp Müller <tim@centricular.net>
31562 gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
31563 Original commit message from CVS:
31564 * gst-libs/gst/riff/riff-ids.h:
31565 Add IDs for 'bext' chunks (see #343837).
31567 2006-06-12 12:44:38 +0000 Young-Ho Cha <ganadist@chollian.net>
31569 gst/subparse/samiparse.c: Honour font face tags in SAMI subtitles (#344503).
31570 Original commit message from CVS:
31571 Patch by: Young-Ho Cha <ganadist at chollian net>
31572 * gst/subparse/samiparse.c: (sami_context_pop_state),
31573 (handle_start_font), (end_sami_element):
31574 Honour font face tags in SAMI subtitles (#344503).
31576 2006-06-11 20:41:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31578 po/POTFILES.in: add missing files containing translatable strings
31579 Original commit message from CVS:
31581 add missing files containing translatable strings
31583 2006-06-11 19:55:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31585 docs/libs/tmpl/.cvsignore: we don't want those *.sgml files in CVS either
31586 Original commit message from CVS:
31587 * docs/libs/tmpl/.cvsignore:
31588 we don't want those *.sgml files in CVS either
31590 2006-06-11 19:44:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31593 Original commit message from CVS:
31594 * docs/libs/.cvsignore:
31595 * tests/check/elements/.cvsignore:
31596 * tests/check/libs/.cvsignore:
31599 2006-06-11 18:33:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31601 docs/libs/Makefile.am: also commiting the changed Makefile.am (added more libs to the doc-build)
31602 Original commit message from CVS:
31603 * docs/libs/Makefile.am:
31604 also commiting the changed Makefile.am (added more libs to the
31607 2006-06-11 17:08:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31609 docs/libs/: first batch of reordering things, add index & hierarchy
31610 Original commit message from CVS:
31611 * docs/libs/gst-plugins-base-libs-docs.sgml:
31612 * docs/libs/gst-plugins-base-libs-sections.txt:
31613 * docs/libs/gst-plugins-base-libs.types:
31614 first batch of reordering things, add index & hierarchy
31616 2006-06-11 14:08:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31619 * ext/alsa/Makefile.am:
31620 * ext/cdparanoia/Makefile.am:
31621 * ext/gnomevfs/Makefile.am:
31622 * ext/libvisual/Makefile.am:
31623 * ext/ogg/Makefile.am:
31624 * ext/pango/Makefile.am:
31625 * ext/theora/Makefile.am:
31626 * ext/vorbis/Makefile.am:
31627 * sys/v4l/Makefile.am:
31628 * sys/ximage/Makefile.am:
31629 * sys/xvimage/Makefile.am:
31630 further clean up build
31631 Original commit message from CVS:
31632 further clean up build
31634 2006-06-11 12:14:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31636 configure.ac: use GST_PKG_CHECK_MODULES, cleans up output
31637 Original commit message from CVS:
31639 use GST_PKG_CHECK_MODULES, cleans up output
31641 2006-06-11 12:10:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31644 * win32/common/config.h:
31646 Original commit message from CVS:
31649 2006-06-10 18:52:03 +0000 Tim-Philipp Müller <tim@centricular.net>
31651 ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
31652 Original commit message from CVS:
31653 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
31654 Add support for burn:// URIs (#343385); const-ify things a bit,
31655 use G_N_ELEMENTS instead of hard-coded array size.
31657 2006-06-10 18:25:07 +0000 Young-Ho Cha <ganadist@chollian.net>
31659 gst/subparse/samiparse.c: Fix up broken entities before passing them to libxml *sigh*. (#343303).
31660 Original commit message from CVS:
31661 Patch by: Young-Ho Cha <ganadist at chollian net>
31662 * gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
31663 Fix up broken entities before passing them to libxml *sigh*.
31666 2006-06-09 18:52:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31671 Original commit message from CVS:
31674 === release 0.10.8 ===
31676 2006-06-09 18:49:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31682 * docs/plugins/gst-plugins-base-plugins.args:
31683 * docs/plugins/inspect/plugin-adder.xml:
31684 * docs/plugins/inspect/plugin-alsa.xml:
31685 * docs/plugins/inspect/plugin-audioconvert.xml:
31686 * docs/plugins/inspect/plugin-audiorate.xml:
31687 * docs/plugins/inspect/plugin-audioresample.xml:
31688 * docs/plugins/inspect/plugin-audiotestsrc.xml:
31689 * docs/plugins/inspect/plugin-cdparanoia.xml:
31690 * docs/plugins/inspect/plugin-decodebin.xml:
31691 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
31692 * docs/plugins/inspect/plugin-gnomevfs.xml:
31693 * docs/plugins/inspect/plugin-libvisual.xml:
31694 * docs/plugins/inspect/plugin-ogg.xml:
31695 * docs/plugins/inspect/plugin-pango.xml:
31696 * docs/plugins/inspect/plugin-playbin.xml:
31697 * docs/plugins/inspect/plugin-subparse.xml:
31698 * docs/plugins/inspect/plugin-tcp.xml:
31699 * docs/plugins/inspect/plugin-theora.xml:
31700 * docs/plugins/inspect/plugin-typefindfunctions.xml:
31701 * docs/plugins/inspect/plugin-video4linux.xml:
31702 * docs/plugins/inspect/plugin-videorate.xml:
31703 * docs/plugins/inspect/plugin-videoscale.xml:
31704 * docs/plugins/inspect/plugin-videotestsrc.xml:
31705 * docs/plugins/inspect/plugin-volume.xml:
31706 * docs/plugins/inspect/plugin-vorbis.xml:
31707 * docs/plugins/inspect/plugin-ximagesink.xml:
31708 * docs/plugins/inspect/plugin-xvimagesink.xml:
31709 * win32/common/config.h:
31711 Original commit message from CVS:
31714 2006-06-07 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31716 0.10.7.2 prerelease
31717 Original commit message from CVS:
31733 * win32/common/config.h:
31734 0.10.7.2 prerelease
31736 2006-06-07 11:03:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31738 move last template doc snippets to source code and delete them
31739 Original commit message from CVS:
31740 * docs/libs/tmpl/gstaudio.sgml:
31741 * docs/libs/tmpl/gstcolorbalance.sgml:
31742 * docs/libs/tmpl/gstmixer.sgml:
31743 * docs/libs/tmpl/gstringbuffer.sgml:
31744 * docs/libs/tmpl/gsttuner.sgml:
31745 * docs/libs/tmpl/gstxoverlay.sgml:
31746 * gst-libs/gst/audio/audio.c:
31747 * gst-libs/gst/audio/gstringbuffer.c:
31748 * gst-libs/gst/interfaces/colorbalance.c:
31749 * gst-libs/gst/interfaces/mixer.c:
31750 * gst-libs/gst/interfaces/tuner.c:
31751 * gst-libs/gst/interfaces/xoverlay.c:
31752 move last template doc snippets to source code and delete them
31754 2006-06-06 16:26:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31756 * gst/gdp/gstgdppay.c:
31758 Original commit message from CVS:
31761 2006-06-06 14:39:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31763 configure.ac: enable building of GDP elements
31764 Original commit message from CVS:
31766 enable building of GDP elements
31767 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
31768 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
31769 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
31770 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
31771 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
31772 (gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
31773 (gst_gdp_pay_change_state):
31774 * gst/gdp/gstgdppay.h:
31777 2006-06-06 11:13:18 +0000 Michael Smith <msmith@xiph.org>
31779 ext/theora/theoraparse.c: Mark DELTA_UNIT on non-keyframes.
31780 Original commit message from CVS:
31781 * ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
31782 (theora_parse_drain_queue):
31783 Mark DELTA_UNIT on non-keyframes.
31785 2006-06-03 21:06:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31787 gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
31788 Original commit message from CVS:
31789 * gst-libs/gst/audio/gstbaseaudiosink.c:
31790 (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
31791 * gst-libs/gst/audio/gstbaseaudiosink.h:
31792 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
31793 (gst_ring_buffer_samples_done):
31794 * gst-libs/gst/audio/gstringbuffer.h:
31795 Document better the fact that latency_time and buffer_time are values
31796 stored in microseconds, and not the usual GStreamer nanoseconds.
31797 Change the variables (compatibly) that store them from GstClockTime
31798 to guint64 to make it more clear that they're not storing clock times.
31799 Also, remove the bogus property description that says the user can
31800 specify -1 to get the default value, since that's never been the case.
31801 When computing the default segment size for the ring buffer, make it
31802 an integer number of samples.
31803 When the sub-class indicates a delay greater than the number of
31804 samples we've written return 0 from the audio sink get_time method.
31806 2006-06-02 17:01:02 +0000 Michael Smith <msmith@xiph.org>
31808 tests/check/: Don't busy-wait in tests; this was causing test timeouts very frequently when running under valgrind.
31809 Original commit message from CVS:
31810 * tests/check/elements/audioconvert.c: (set_channel_positions),
31811 (get_float_mc_caps), (get_int_mc_caps):
31812 * tests/check/elements/audioresample.c:
31813 * tests/check/elements/audiotestsrc.c: (GST_START_TEST):
31814 * tests/check/elements/videorate.c:
31815 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
31816 * tests/check/elements/volume.c:
31817 * tests/check/elements/vorbisdec.c:
31818 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
31819 Don't busy-wait in tests; this was causing test timeouts very
31820 frequently when running under valgrind.
31822 2006-06-02 16:45:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31824 * gst/gdp/gstgdpdepay.c:
31825 * gst/gdp/gstgdppay.h:
31827 Original commit message from CVS:
31830 2006-06-02 16:35:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31832 * tests/check/elements/multifdsink.c:
31833 fail_if_can_read is racy
31834 Original commit message from CVS:
31835 fail_if_can_read is racy
31837 2006-06-02 16:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31839 gst/tcp/: make multifdsink properly deal with streamheader:
31840 Original commit message from CVS:
31842 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
31843 (gst_multi_fd_sink_remove_client_link),
31844 (gst_multi_fd_sink_client_queue_caps),
31845 (gst_multi_fd_sink_client_queue_buffer),
31846 (gst_multi_fd_sink_handle_client_write),
31847 (gst_multi_fd_sink_render):
31848 * gst/tcp/gstmultifdsink.h:
31849 make multifdsink properly deal with streamheader:
31850 - streamheader is taken from caps
31851 - buffers marked with IN_CAPS are not sent
31852 - streamheaders are sent, on connection, from the caps of the
31853 buffer where the client gets positioned to
31854 - further streamheader changes are done every time the client
31855 will receive a buffer with different caps
31856 * tests/check/elements/multifdsink.c: (GST_START_TEST),
31857 (gst_multifdsink_create_streamheader):
31860 2006-06-02 15:06:59 +0000 Michael Smith <msmith@xiph.org>
31862 ext/vorbis/vorbisdec.c: Reinstate limit on channel count. Vorbis does not define the meaning of > 6 channels, so they...
31863 Original commit message from CVS:
31864 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
31865 Reinstate limit on channel count. Vorbis does not define the meaning
31866 of > 6 channels, so they're just independent channels. Gstreamer
31867 currently has no mechanism to represent N independent channels.
31869 2006-06-02 14:23:34 +0000 Michael Smith <msmith@xiph.org>
31871 ext/vorbis/vorbisdec.c: Don't arbitrarily restrict channel counts and rate in vorbis.
31872 Original commit message from CVS:
31873 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
31874 Don't arbitrarily restrict channel counts and rate in vorbis.
31875 In terms of effects likely on real-world files, this fixes 96kHz
31876 playback of vorbis.
31878 2006-06-02 14:19:18 +0000 Michael Smith <msmith@xiph.org>
31880 gst/audioconvert/audioconvert.c: More correct float->int conversion.
31881 Original commit message from CVS:
31882 * gst/audioconvert/audioconvert.c: (float):
31883 More correct float->int conversion.
31885 2006-06-02 14:07:42 +0000 Michael Smith <msmith@xiph.org>
31887 ext/ogg/gstoggdemux.c: Don't accidently send GST_CLOCK_TIME_NONE as a new segment start value. Fixes g-critical on tr...
31888 Original commit message from CVS:
31889 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
31890 Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
31891 value. Fixes g-critical on trying to play back ogg containing
31894 2006-06-02 10:34:12 +0000 Wim Taymans <wim.taymans@gmail.com>
31896 gst/playback/gstplaybasebin.*: Make the subtitle detection work from any thread so we don't deadlock. Fixes #343397.
31897 Original commit message from CVS:
31898 * gst/playback/gstplaybasebin.c: (group_create), (group_commit),
31900 * gst/playback/gstplaybasebin.h:
31901 Make the subtitle detection work from any thread so we don't
31902 deadlock. Fixes #343397.
31904 2006-06-02 10:28:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31906 gst/gdp/gstgdppay.c: add crc-header and crc-payload properties don't error out on some things that are recoverable
31907 Original commit message from CVS:
31908 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
31909 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
31910 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
31911 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
31912 (gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
31913 (gst_gdp_pay_get_property):
31914 add crc-header and crc-payload properties
31915 don't error out on some things that are recoverable
31916 * tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
31919 2006-06-02 09:17:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31921 * gst/tcp/gsttcp.c:
31922 show type number when packet is of the wrong type
31923 Original commit message from CVS:
31924 show type number when packet is of the wrong type
31926 2006-06-01 23:04:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31928 gst/volume/Makefile.am: Seriously, it's not *that* hard to get compilation right. Even a drunk can do it ! Add LIBOI...
31929 Original commit message from CVS:
31930 * gst/volume/Makefile.am:
31931 Seriously, it's not *that* hard to get compilation right. Even
31932 a drunk can do it ! Add LIBOIL CFLAGS and LIBS
31934 2006-06-01 22:00:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31936 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
31937 Original commit message from CVS:
31938 * ext/alsaspdif/alsaspdifsink.h:
31939 * ext/amrwb/gstamrwbdec.h:
31940 * ext/amrwb/gstamrwbenc.h:
31941 * ext/amrwb/gstamrwbparse.h:
31942 * ext/arts/gst_arts.h:
31943 * ext/artsd/gstartsdsink.h:
31944 * ext/audiofile/gstafparse.h:
31945 * ext/audiofile/gstafsink.h:
31946 * ext/audiofile/gstafsrc.h:
31947 * ext/audioresample/gstaudioresample.h:
31948 * ext/bz2/gstbz2dec.h:
31949 * ext/bz2/gstbz2enc.h:
31950 * ext/dirac/gstdiracdec.h:
31951 * ext/directfb/dfbvideosink.h:
31952 * ext/divx/gstdivxdec.h:
31953 * ext/divx/gstdivxenc.h:
31954 * ext/dts/gstdtsdec.h:
31955 * ext/faac/gstfaac.h:
31956 * ext/gsm/gstgsmdec.h:
31957 * ext/gsm/gstgsmenc.h:
31958 * ext/ivorbis/vorbisenc.h:
31959 * ext/libfame/gstlibfame.h:
31960 * ext/nas/nassink.h:
31961 * ext/neon/gstneonhttpsrc.h:
31962 * ext/polyp/polypsink.h:
31963 * ext/sdl/sdlaudiosink.h:
31964 * ext/sdl/sdlvideosink.h:
31965 * ext/shout/gstshout.h:
31966 * ext/snapshot/gstsnapshot.h:
31967 * ext/sndfile/gstsf.h:
31968 * ext/swfdec/gstswfdec.h:
31969 * ext/tarkin/gsttarkindec.h:
31970 * ext/tarkin/gsttarkinenc.h:
31971 * ext/theora/theoradec.h:
31972 * ext/wavpack/gstwavpackdec.h:
31973 * ext/wavpack/gstwavpackparse.h:
31974 * ext/xine/gstxine.h:
31975 * ext/xvid/gstxviddec.h:
31976 * ext/xvid/gstxvidenc.h:
31977 * gst/cdxaparse/gstcdxaparse.h:
31978 * gst/cdxaparse/gstcdxastrip.h:
31979 * gst/colorspace/gstcolorspace.h:
31980 * gst/festival/gstfestival.h:
31981 * gst/freeze/gstfreeze.h:
31982 * gst/gdp/gstgdpdepay.h:
31983 * gst/gdp/gstgdppay.h:
31984 * gst/modplug/gstmodplug.h:
31985 * gst/mpeg1sys/gstmpeg1systemencode.h:
31986 * gst/mpeg1videoparse/gstmp1videoparse.h:
31987 * gst/mpeg2sub/gstmpeg2subt.h:
31988 * gst/mpegaudioparse/gstmpegaudioparse.h:
31989 * gst/multifilesink/gstmultifilesink.h:
31990 * gst/overlay/gstoverlay.h:
31991 * gst/playondemand/gstplayondemand.h:
31992 * gst/qtdemux/qtdemux.h:
31993 * gst/rtjpeg/gstrtjpegdec.h:
31994 * gst/rtjpeg/gstrtjpegenc.h:
31995 * gst/smooth/gstsmooth.h:
31996 * gst/smoothwave/gstsmoothwave.h:
31997 * gst/spectrum/gstspectrum.h:
31998 * gst/speed/gstspeed.h:
31999 * gst/stereo/gststereo.h:
32000 * gst/switch/gstswitch.h:
32001 * gst/tta/gstttadec.h:
32002 * gst/tta/gstttaparse.h:
32003 * gst/videodrop/gstvideodrop.h:
32004 * gst/xingheader/gstxingmux.h:
32005 * sys/directdraw/gstdirectdrawsink.h:
32006 * sys/directsound/gstdirectsoundsink.h:
32007 * sys/dxr3/dxr3audiosink.h:
32008 * sys/dxr3/dxr3spusink.h:
32009 * sys/dxr3/dxr3videosink.h:
32010 * sys/qcam/gstqcamsrc.h:
32011 * sys/vcd/vcdsrc.h:
32012 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
32014 2006-06-01 20:39:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32016 gst/volume/gstvolume.*: rewrite the passthrough check, split _int16 and _int16_clamp, fix another property desc., rem...
32017 Original commit message from CVS:
32018 * gst/volume/gstvolume.c: (volume_choose_func),
32019 (volume_update_real_volume), (gst_volume_class_init),
32020 (gst_volume_init), (volume_process_float), (volume_process_int16),
32021 (volume_process_int16_clamp), (volume_set_caps),
32022 (volume_transform_ip), (plugin_init):
32023 * gst/volume/gstvolume.h:
32024 rewrite the passthrough check, split _int16 and _int16_clamp, fix
32025 another property desc., remove unused param from process function
32026 * tests/check/elements/volume.c: (volume_suite):
32027 reactivate the passthrough test
32029 2006-06-01 19:19:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32031 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
32032 Original commit message from CVS:
32033 * ext/alsa/gstalsamixerelement.h:
32034 * ext/alsa/gstalsamixeroptions.h:
32035 * ext/alsa/gstalsamixertrack.h:
32036 * ext/gnomevfs/gstgnomevfssink.h:
32037 * ext/gnomevfs/gstgnomevfssrc.h:
32038 * ext/theora/gsttheoradec.h:
32039 * ext/theora/gsttheoraenc.h:
32040 * ext/theora/gsttheoraparse.h:
32041 * ext/vorbis/vorbisparse.h:
32042 * gst-libs/gst/audio/gstaudioclock.h:
32043 * gst-libs/gst/audio/gstaudiofilter.h:
32044 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
32045 * gst/audioconvert/gstaudioconvert.h:
32046 * gst/audioresample/gstaudioresample.h:
32047 * gst/audiotestsrc/gstaudiotestsrc.h:
32048 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
32049 * gst/playback/gststreamselector.h:
32050 * gst/tcp/gstmultifdsink.h:
32051 * gst/tcp/gsttcpclientsink.h:
32052 * gst/tcp/gsttcpclientsrc.h:
32053 * gst/tcp/gsttcpserversink.h:
32054 * gst/tcp/gsttcpserversrc.h:
32055 * gst/videorate/gstvideorate.h:
32056 * gst/videoscale/gstvideoscale.h:
32057 * gst/videotestsrc/gstvideotestsrc.h:
32058 * gst/volume/gstvolume.h:
32059 * sys/v4l/gstv4ljpegsrc.h:
32060 * sys/v4l/gstv4lmjpegsink.h:
32061 * sys/v4l/gstv4lmjpegsrc.h:
32062 * sys/v4l/gstv4lsrc.h:
32063 * sys/ximage/ximagesink.h:
32064 * sys/xvimage/xvimagesink.h:
32065 * tests/old/testsuite/alsa/sinesrc.h:
32066 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
32068 2006-05-31 16:56:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32070 * tests/check/elements/multifdsink.c:
32071 remove wrong commit
32072 Original commit message from CVS:
32073 remove wrong commit
32075 2006-05-31 16:21:48 +0000 Wim Taymans <wim.taymans@gmail.com>
32077 ext/libvisual/visual.c: Handle DISCONT.
32078 Original commit message from CVS:
32079 * ext/libvisual/visual.c: (gst_visual_reset),
32080 (gst_visual_sink_setcaps), (gst_visual_sink_event),
32081 (gst_visual_src_event), (get_buffer), (gst_visual_chain):
32083 Use running time before doing QoS.
32086 2006-05-31 14:17:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32088 docs/libs/Makefile.am: set a magic variable to indicate we know the docs are incomplete
32089 Original commit message from CVS:
32090 * docs/libs/Makefile.am:
32091 set a magic variable to indicate we know the docs are incomplete
32093 2006-05-30 20:33:59 +0000 Sébastien Moutte <sebastien@moutte.net>
32095 win32/common/libgstvideo.def: export gst_video_calculate_display_ratio
32096 Original commit message from CVS:
32097 * win32/common/libgstvideo.def:
32098 export gst_video_calculate_display_ratio
32099 * win32/vs6/libgstvideoscale.dsp:
32100 add link to libgstvideo-0.10.lib
32102 2006-05-30 19:00:39 +0000 Tim-Philipp Müller <tim@centricular.net>
32104 gst/playback/gstplaybasebin.c: Throw a more comprehensible error for rtsp:// URIs (rather than erroring out with a ne...
32105 Original commit message from CVS:
32106 * gst/playback/gstplaybasebin.c: (gen_source_element):
32107 Throw a more comprehensible error for rtsp:// URIs (rather
32108 than erroring out with a negotiation error later on) until
32109 we fix playbin to handle rtspsrc etc.
32111 2006-05-30 16:09:36 +0000 Wim Taymans <wim.taymans@gmail.com>
32113 ext/pango/gsttextoverlay.c: Added some FIXMEs.
32114 Original commit message from CVS:
32115 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
32116 (gst_text_overlay_text_event):
32119 2006-05-30 16:07:50 +0000 Wim Taymans <wim.taymans@gmail.com>
32121 gst/adder/gstadder.*: Implement release_request_pad.
32122 Original commit message from CVS:
32123 * gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
32124 (gst_adder_request_new_pad), (gst_adder_release_pad):
32125 * gst/adder/gstadder.h:
32126 Implement release_request_pad.
32127 Make padcounter atomic.
32128 * tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
32129 Added check for release_pad in adder.
32131 2006-05-30 16:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
32133 ext/ogg/gstoggdemux.c: Fix build again.
32134 Original commit message from CVS:
32135 * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
32138 2006-05-30 14:59:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32140 ext/ogg/gstoggdemux.c: add more debugging clean up printf formats for granulepos and serialno
32141 Original commit message from CVS:
32142 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
32143 (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
32144 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
32145 (gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
32146 (gst_ogg_demux_seek), (gst_ogg_demux_get_data),
32147 (gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
32148 (gst_ogg_demux_bisect_forward_serialno),
32149 (gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
32150 (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
32152 clean up printf formats for granulepos and serialno
32154 2006-05-30 14:31:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32156 * tests/check/elements/multifdsink.c:
32157 * tests/check/generic/states.c:
32158 properly fail if we can't make an element
32159 Original commit message from CVS:
32160 properly fail if we can't make an element
32162 2006-05-30 13:22:58 +0000 Michael Smith <msmith@xiph.org>
32164 ext/vorbis/vorbisenc.*: Multi-channel caps negotiation, so we can do proper multichannel vorbis encoding, negotiated ...
32165 Original commit message from CVS:
32166 * ext/vorbis/vorbisenc.c: (raw_caps_factory),
32167 (gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
32168 (gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
32169 (gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
32170 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
32171 * ext/vorbis/vorbisenc.h:
32172 Multi-channel caps negotiation, so we can do proper multichannel
32173 vorbis encoding, negotiated through audioconvert.
32175 2006-05-30 11:45:52 +0000 Wim Taymans <wim.taymans@gmail.com>
32177 tests/check/elements/adder.c: Added check to show that #339935 is fixed with ongoing adder and collectpads fixes.
32178 Original commit message from CVS:
32179 * tests/check/elements/adder.c: (test_event_message_received),
32180 (test_play_twice_message_received), (GST_START_TEST),
32182 Added check to show that #339935 is fixed with ongoing
32183 adder and collectpads fixes.
32185 2006-05-29 17:19:48 +0000 Wim Taymans <wim.taymans@gmail.com>
32187 gst/adder/gstadder.c: Don't leak pad name.
32188 Original commit message from CVS:
32189 * gst/adder/gstadder.c: (gst_adder_request_new_pad):
32190 Don't leak pad name.
32192 2006-05-29 15:49:53 +0000 Wim Taymans <wim.taymans@gmail.com>
32194 gst/adder/gstadder.c: Fix adder seeking.
32195 Original commit message from CVS:
32196 * gst/adder/gstadder.c: (gst_adder_query_duration),
32197 (forward_event_func), (forward_event), (gst_adder_src_event):
32199 Make query/seeking code threadsafe.
32200 * tests/check/Makefile.am:
32201 * tests/check/elements/adder.c: (test_event_message_received),
32202 (GST_START_TEST), (test_play_twice_message_received):
32203 Fix adder test case.
32205 2006-05-29 13:21:00 +0000 Young-Ho Cha <ganadist@chollian.net>
32207 gst/playback/gstplaybasebin.*: Add 'subtitle-encoding' property to playbin, so applications can force a subtitle enco...
32208 Original commit message from CVS:
32209 Patch by: Young-Ho Cha <ganadist at chollian net>
32210 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
32211 (gst_play_base_bin_init), (gst_play_base_bin_dispose),
32212 (set_encoding_element), (decodebin_element_added_cb),
32213 (decodebin_element_removed_cb), (setup_subtitle), (setup_source),
32214 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
32215 * gst/playback/gstplaybasebin.h:
32216 Add 'subtitle-encoding' property to playbin, so applications can
32217 force a subtitle encoding for non-UTF8 subtitles (#342268).
32218 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
32219 (gst_sub_parse_set_property):
32220 Rename recently-added 'encoding' property to 'subtitle-encoding'
32221 (so it can be proxied by playbin/decodebin in a generic way
32222 with less danger of false positives).
32224 2006-05-29 11:04:48 +0000 Michael Smith <msmith@xiph.org>
32226 gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf...
32227 Original commit message from CVS:
32228 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
32229 (append_with_other_format), (set_structure_widths),
32230 (gst_audio_convert_transform_caps):
32231 Patch from #341562: give more specific audio caps in get_caps, so
32232 that basetransform can make better decisions on what caps to
32235 2006-05-28 20:04:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32237 tests/check/elements/volume.c: make it compile again
32238 Original commit message from CVS:
32239 * tests/check/elements/volume.c:
32240 make it compile again
32242 2006-05-28 19:56:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32244 tests/check/elements/volume.c: disable test until #343196 gets resolved
32245 Original commit message from CVS:
32246 * tests/check/elements/volume.c: (volume_suite):
32247 disable test until #343196 gets resolved
32249 2006-05-28 19:42:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32251 gst/adder/gstadder.c: Make it easier to copy&paste
32252 Original commit message from CVS:
32253 * gst/adder/gstadder.c: (gst_adder_get_type):
32254 Make it easier to copy&paste
32255 * gst/volume/Makefile.am:
32256 * gst/volume/gstvolume.c: (volume_update_real_volume),
32257 (gst_volume_set_volume), (gst_volume_set_mute),
32258 (gst_volume_class_init), (volume_process_int16), (volume_set_caps),
32259 (volume_transform_ip), (volume_update_mute),
32260 (volume_update_volume):
32261 * gst/volume/gstvolume.h:
32262 Add own debug category, move duplicate code to helper function, fix
32263 property texts, add more comments and prepare ffor liboil-goodness
32264 * tests/check/Makefile.am:
32265 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
32266 add test for mute and passtrough case, be a bit more verbose to track
32268 * tests/check/generic/states.c: (GST_START_TEST):
32269 catch elements that fail to instantiate
32271 2006-05-28 09:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
32273 tests/check/pipelines/: Comment out tests using parse_launch() if core was built without parsing capabilities.
32274 Original commit message from CVS:
32275 * tests/check/pipelines/simple-launch-lines.c:
32276 * tests/check/pipelines/theoraenc.c:
32277 * tests/check/pipelines/vorbisenc.c:
32278 Comment out tests using parse_launch() if core was built without
32279 parsing capabilities.
32281 2006-05-27 13:34:03 +0000 Edward Hervey <bilboed@bilboed.com>
32283 tests/check/Makefile.am: Extra bonus points for whoever explains to ensonic that you are meant to test unit tests tho...
32284 Original commit message from CVS:
32285 * tests/check/Makefile.am:
32286 Extra bonus points for whoever explains to ensonic that you are meant
32287 to test unit tests thoroughly before commiting them, especially if
32288 you know it's going to break.
32289 De-activated element/adder tests.
32291 2006-05-27 13:09:16 +0000 Edward Hervey <bilboed@bilboed.com>
32293 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Marking caps conversion issues as GST_WARNING is way too verbose,
32294 Original commit message from CVS:
32295 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
32296 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
32297 Marking caps conversion issues as GST_WARNING is way too verbose,
32298 Moving them to GST_LOG.
32300 2006-05-27 11:26:18 +0000 Tim-Philipp Müller <tim@centricular.net>
32302 README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from...
32303 Original commit message from CVS:
32305 Replace current README (containing the release notes from
32306 some 0.9.x version) with a proper README taken from the core.
32308 2006-05-26 15:52:23 +0000 Wim Taymans <wim.taymans@gmail.com>
32310 ext/vorbis/vorbisdec.c: Small cleanups.
32311 Original commit message from CVS:
32312 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
32313 (vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
32314 (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
32315 (vorbis_dec_change_state):
32318 Clip output samples to segment boundaries.
32320 2006-05-26 11:17:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32322 sys/ximage/ximagesink.c: Improve the errors produced on bad output, including some human readable description strings.
32323 Original commit message from CVS:
32324 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
32325 (gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
32326 Improve the errors produced on bad output, including some human
32327 readable description strings.
32328 Handle the (theoretical for ximagesink) case where the XServer
32329 has a different idea about the size required for a particular
32330 frame and gives us too small a memory allocation.
32332 2006-05-26 10:18:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32335 Mention bugs fixed by previous commit
32336 Original commit message from CVS:
32337 Mention bugs fixed by previous commit
32339 2006-05-26 09:40:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32341 sys/xvimage/xvimagesink.c: Improve the errors produced on bad output, including some human readable description strings.
32342 Original commit message from CVS:
32343 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
32344 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
32345 (gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
32346 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
32347 Improve the errors produced on bad output, including some human
32348 readable description strings.
32349 Handle RGB Xv formats properly by transforming them into our
32350 big-endian caps description.
32351 Use gst_caps_truncate to ensure that we never try and choose a
32352 non-fixed caps in buffer_alloc.
32353 Handle the case where the XServer has a different idea about the size
32354 required for a particular frame and gives us too small a memory
32356 Use -1 to indicate 'no image format', because 0 is a valid XServer
32357 image format number.
32358 Put RGB Xv formats at the end of the caps, so that we always prefer
32360 Iterate the available Xv Encodings to determine the maximum width and
32361 height, and then return that in our caps.
32363 2006-05-25 16:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32365 gst/playback/gstdecodebin.c: When there is only one unfinished pad and it receives an event that doesn't match our re...
32366 Original commit message from CVS:
32367 * gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
32368 When there is only one unfinished pad and it receives an event that
32369 doesn't match our requirements, we need to set alldone=FALSE so that
32370 the fakesink is not removed yet.
32372 2006-05-25 09:32:31 +0000 Tim-Philipp Müller <tim@centricular.net>
32374 ext/ogg/gstoggdemux.c: Use gst_type_find_helper_for_buffer() to find the type of stream from the first packet.
32375 Original commit message from CVS:
32376 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
32377 Use gst_type_find_helper_for_buffer() to find the type
32378 of stream from the first packet.
32380 Bump requirements to core CVS (needed for vorbis
32381 typefinding to work).
32383 2006-05-24 08:34:53 +0000 Edward Hervey <bilboed@bilboed.com>
32385 gst/typefind/gsttypefindfunctions.c: Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
32386 Original commit message from CVS:
32387 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
32388 Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
32389 Else they play perfectly fine with qtdemux.
32391 2006-05-23 20:38:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32393 make more debug catagories static
32394 Original commit message from CVS:
32395 * ext/theora/theoradec.c:
32396 * ext/theora/theoraenc.c:
32397 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
32398 * gst/audiorate/gstaudiorate.c:
32399 make more debug catagories static
32400 * tests/check/Makefile.am:
32401 * tests/check/elements/adder.c: (message_received),
32402 (test_event_message_received), (GST_START_TEST),
32403 (test_play_twice_message_received), (adder_suite):
32404 added test case for using element twice, extra bonus points for anyone
32405 who can make these test run reliably
32407 2006-05-23 15:18:40 +0000 Tim-Philipp Müller <tim@centricular.net>
32409 ext/theora/theoradec.c: Make work with time-stamped input buffers that do not have a granulepos in BUFFER_OFFSET_END ...
32410 Original commit message from CVS:
32411 * ext/theora/theoradec.c: (theora_dec_chain):
32412 Make work with time-stamped input buffers that do not
32413 have a granulepos in BUFFER_OFFSET_END (like theora
32414 buffers coming from matroskademux). Fixes #342448.
32416 2006-05-22 15:53:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32418 gst/gdp/: Handle error cases when calling functions do downwards state change after parent's change_state
32419 Original commit message from CVS:
32420 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain),
32421 (gst_gdp_depay_change_state):
32422 * gst/gdp/gstgdpdepay.h:
32423 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader),
32424 (gst_gdp_pay_chain), (gst_gdp_pay_sink_event),
32425 (gst_gdp_pay_change_state):
32426 * gst/gdp/gstgdppay.h:
32427 Handle error cases when calling functions
32428 do downwards state change after parent's change_state
32429 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
32430 * tests/check/elements/gdppay.c: (GST_START_TEST):
32433 2006-05-22 13:25:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32435 adding GDP payloader and depayloader. Build integration will follow later when the GDP issues for core are sorted out.
32436 Original commit message from CVS:
32437 * gst/gdp/Makefile.am:
32438 * gst/gdp/gstgdp.c: (plugin_init):
32439 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_base_init),
32440 (gst_gdp_depay_class_init), (gst_gdp_depay_init),
32441 (gst_gdp_depay_finalize), (gst_gdp_depay_chain),
32442 (gst_gdp_depay_change_state), (gst_gdp_depay_plugin_init):
32443 * gst/gdp/gstgdpdepay.h:
32444 * gst/gdp/gstgdppay.c: (gst_gdp_pay_base_init),
32445 (gst_gdp_pay_class_init), (gst_gdp_pay_init),
32446 (gst_gdp_pay_dispose), (gst_gdp_stamp_buffer),
32447 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
32448 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
32449 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
32450 (gst_gdp_pay_sink_event), (gst_gdp_pay_change_state),
32451 (gst_gdp_pay_plugin_init):
32452 * gst/gdp/gstgdppay.h:
32453 * tests/check/Makefile.am:
32454 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
32455 (cleanup_gdpdepay), (gdpdepay_push_per_byte), (GST_START_TEST),
32456 (setup_gdpdepay_streamheader), (gdpdepay_suite), (main):
32457 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay),
32458 (GST_START_TEST), (setup_gdppay_streamheader), (gdppay_suite),
32460 adding GDP payloader and depayloader. Build integration will
32461 follow later when the GDP issues for core are sorted out.
32463 2006-05-22 11:42:03 +0000 Peter Kjellerstedt <pkj@axis.com>
32465 gst/tcp/Makefile.am: fdstresstest doesn't need Gtk+, fix compilation if gtk is not available (#342566).
32466 Original commit message from CVS:
32467 Patch by: Peter Kjellerstedt <pkj at axis com>
32468 * gst/tcp/Makefile.am:
32469 fdstresstest doesn't need Gtk+, fix compilation if
32470 gtk is not available (#342566).
32472 2006-05-19 17:57:56 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
32474 gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
32475 Original commit message from CVS:
32476 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
32478 Removed redundant floor()
32480 2006-05-19 15:00:43 +0000 Tim-Philipp Müller <tim@centricular.net>
32482 gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
32483 Original commit message from CVS:
32484 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
32485 On second thought, just skip JUNK chunks automatically, so
32486 the caller doesn't have to handle this. Fixes #342345.
32487 Also, return GST_FLOW_UNEXPECTED if we get a short read,
32488 not GST_FLOW_ERROR.
32490 2006-05-19 13:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
32492 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...
32493 Original commit message from CVS:
32494 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
32495 Don't bail out on JUNK chunks with a size of 0 (would try to
32496 pull_range 0 bytes before, which sources don't like too much).
32499 2006-05-19 13:02:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32501 Use the gstutil scaling function to preserve 64 bits while calculating output width and height from the display-aspec...
32502 Original commit message from CVS:
32503 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
32504 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
32505 Use the gstutil scaling function to preserve 64 bits while calculating
32506 output width and height from the display-aspect-ratio. (A continuation
32509 2006-05-19 11:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32511 sys/xvimage/xvimagesink.*: When performing buffer allocations, remember the caps and image format we return so that i...
32512 Original commit message from CVS:
32513 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
32514 (gst_xvimagesink_buffer_alloc):
32515 * sys/xvimage/xvimagesink.h:
32516 When performing buffer allocations, remember the caps and image format
32517 we return so that if the same caps are asked for next time we can
32518 return them immediately without doing any caps intersections.
32520 2006-05-18 23:00:02 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
32522 gst-libs/gst/rtp/README: Some new documentation
32523 Original commit message from CVS:
32524 2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
32525 * gst-libs/gst/rtp/README:
32526 Some new documentation
32527 * gst-libs/gst/rtp/gstrtpbuffer.h:
32528 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
32529 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
32530 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
32531 New RTP audio base payloader class. Supports frame or sample based codecs.
32532 Not enabled in Makefile.am until approved.
32534 2006-05-18 20:30:26 +0000 Tim-Philipp Müller <tim@centricular.net>
32536 tests/check/elements/alsa.c: Fix test case: don't try to free NULL GValueArray when there are no devices.
32537 Original commit message from CVS:
32538 * tests/check/elements/alsa.c: (test_device_property_probe):
32539 Fix test case: don't try to free NULL GValueArray when there
32542 2006-05-18 19:21:53 +0000 Tim-Philipp Müller <tim@centricular.net>
32544 tests/check/: Add simple test that runs a device property probe on alsasrc, alsasink and alsamixer. Disable valgrind ...
32545 Original commit message from CVS:
32546 * tests/check/Makefile.am:
32547 * tests/check/elements/alsa.c: (test_device_property_probe),
32548 (alsa_suite), (main):
32549 Add simple test that runs a device property probe on alsasrc,
32550 alsasink and alsamixer. Disable valgrind check for now (too
32551 many leaks in libasound, and valgrind ignored my suppressions
32554 2006-05-18 17:19:39 +0000 Martin Szulecki <gnomebugzilla@sukimashita.com>
32556 ext/alsa/: Clean up and simplify alsa device probing. Make it actually work for multiple classes. Don't cache results...
32557 Original commit message from CVS:
32558 * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
32559 (gst_alsa_device_property_probe_probe_property),
32560 (gst_alsa_device_property_probe_needs_probe),
32561 (gst_alsa_device_property_probe_get_values),
32562 (gst_alsa_type_add_device_property_probe_interface):
32563 * ext/alsa/gstalsadeviceprobe.h:
32564 * ext/alsa/gstalsamixerelement.c:
32565 (gst_alsa_mixer_element_init_interfaces):
32566 * ext/alsa/gstalsamixerelement.h:
32567 Clean up and simplify alsa device probing. Make it actually work
32568 for multiple classes. Don't cache results any longer.
32569 * ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
32570 (gst_alsasink_init):
32571 * ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
32572 (gst_alsasrc_interface_supported), (gst_implements_interface_init),
32573 (gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
32574 Make alsasink and alsasrc implement the GstPropertyProbe interface
32575 for device probing (#342181).
32576 Patch by: Martin Szulecki <gnomebugzilla at sukimashita com>
32578 2006-05-18 10:05:23 +0000 Tim-Philipp Müller <tim@centricular.net>
32580 gst/subparse/samiparse.c: Don't ignore return value of strtol (++compiler_happiness).
32581 Original commit message from CVS:
32582 * gst/subparse/samiparse.c: (handle_start_font):
32583 Don't ignore return value of strtol (++compiler_happiness).
32585 2006-05-17 17:49:10 +0000 Young-Ho Cha <ganadist@chollian.net>
32587 gst/subparse/gstsubparse.*: Add 'encoding' property (#341681).
32588 Original commit message from CVS:
32589 Patch by: Young-Ho Cha <ganadist chollian net>
32590 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
32591 (gst_sub_parse_class_init), (gst_sub_parse_init),
32592 (gst_sub_parse_set_property), (gst_sub_parse_get_property),
32593 (convert_encoding):
32594 * gst/subparse/gstsubparse.h:
32595 Add 'encoding' property (#341681).
32596 * gst/subparse/samiparse.c: (characters_sami):
32597 Output is pango markup, so we need to escape text
32598 between tags (#342143).
32600 2006-05-16 17:34:14 +0000 Tim-Philipp Müller <tim@centricular.net>
32602 gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
32603 Original commit message from CVS:
32604 * gst-libs/gst/audio/multichannel.c:
32605 (gst_audio_check_channel_positions):
32606 It's okay to have caps with channels=1 and a channel position
32607 different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
32608 (deinterleavers might want to keep the position in the caps,
32609 so that they can be re-interleaved again properly later).
32610 Leave check for unexpected 2-channel layouts intact for now.
32612 2006-05-16 16:28:10 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
32614 gst/tcp/gsttcp.c: Return GST_FLOW_UNEXPECTED when we have an eos on the socket so basesrc can do its job correctly.
32615 Original commit message from CVS:
32616 2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
32617 * gst/tcp/gsttcp.c: (gst_tcp_socket_read):
32618 Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
32619 basesrc can do its job correctly.
32621 2006-05-16 15:52:17 +0000 Tim-Philipp Müller <tim@centricular.net>
32623 ext/alsa/: Refactor and improve caps probing code: probe signedness when we probe the supported formats/widths; set e...
32624 Original commit message from CVS:
32625 * ext/alsa/Makefile.am:
32626 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
32627 (gst_alsa_detect_formats), (get_channel_free_structure),
32628 (caps_add_channel_configuration), (gst_alsa_detect_channels),
32629 (gst_alsa_probe_supported_formats):
32630 * ext/alsa/gstalsa.h:
32631 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
32632 Refactor and improve caps probing code: probe signedness
32633 when we probe the supported formats/widths; set endianness
32634 to the one we actually probed for (ie. cpu endianness).
32635 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
32636 (gst_alsasrc_close):
32637 * ext/alsa/gstalsasrc.h:
32638 Implement caps probing for alsasrc.
32640 2006-05-15 17:42:19 +0000 Wim Taymans <wim.taymans@gmail.com>
32642 ext/theora/theoradec.c: Cleanups, add some G_LIKELY.
32643 Original commit message from CVS:
32644 * ext/theora/theoradec.c: (gst_theora_dec_reset),
32645 (theora_dec_src_query), (theora_dec_src_event),
32646 (theora_dec_sink_event), (theora_handle_comment_packet),
32647 (theora_handle_data_packet), (theora_dec_change_state):
32648 Cleanups, add some G_LIKELY.
32649 Use segment helpers instead of our own wrong code.
32650 Clear queued buffers on seek and READY.
32651 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
32652 (vorbis_dec_convert), (vorbis_dec_src_query),
32653 (vorbis_dec_src_event), (vorbis_dec_sink_event),
32654 (vorbis_handle_comment_packet), (vorbis_dec_push),
32655 (vorbis_handle_data_packet), (vorbis_dec_chain),
32656 (vorbis_dec_change_state):
32657 * ext/vorbis/vorbisdec.h:
32658 Remove old useless packetno variable.
32659 Do position query properly.
32661 Do cleanup of queued buffers in new helper function
32664 2006-05-15 17:17:22 +0000 Tim-Philipp Müller <tim@centricular.net>
32666 ext/alsa/gstalsasink.c: Query supported sample rates. Fixes #341732.
32667 Original commit message from CVS:
32668 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
32669 Query supported sample rates. Fixes #341732.
32671 2006-05-15 17:01:02 +0000 Julien Moutte <julien@moutte.net>
32673 gst/playback/gstdecodebin.c: Make decodebin reusable when going from PAUSE_TO_READY and then back to PAUSED.
32674 Original commit message from CVS:
32675 2006-05-15 Julien MOUTTE <julien@moutte.net>
32676 * gst/playback/gstdecodebin.c: (cleanup_decodebin),
32677 (gst_decode_bin_change_state): Make decodebin reusable
32678 when going from PAUSE_TO_READY and then back to PAUSED.
32681 2006-05-15 16:49:31 +0000 Wim Taymans <wim.taymans@gmail.com>
32683 ext/vorbis/vorbisdec.c: Cleanups. Use refcounting and DEBUG_OBJECT.
32684 Original commit message from CVS:
32685 * ext/vorbis/vorbisdec.c: (vorbis_get_query_types),
32686 (vorbis_dec_convert), (vorbis_dec_src_query),
32687 (vorbis_dec_sink_query), (vorbis_dec_src_event),
32688 (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
32689 (vorbis_dec_clean_queued), (vorbis_dec_push),
32690 (vorbis_handle_data_packet), (vorbis_dec_change_state):
32691 Cleanups. Use refcounting and DEBUG_OBJECT.
32692 Reset segment on flush, use code methods instead of our
32694 Fix potential memleak.
32696 2006-05-15 16:46:44 +0000 Tim-Philipp Müller <tim@centricular.net>
32698 ext/alsa/gstalsasink.*: Don't leak allocated snd_output_t structure if there's more than one alsasink instance at a t...
32699 Original commit message from CVS:
32700 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
32701 (gst_alsasink_init):
32702 * ext/alsa/gstalsasink.h:
32703 Don't leak allocated snd_output_t structure if there's
32704 more than one alsasink instance at a time (#341873).
32705 Also fix GObject macros in header file.
32707 2006-05-15 15:31:30 +0000 Tim-Philipp Müller <tim@centricular.net>
32709 gst/subparse/gstsubparse.c: Don't use libxml functions in the typefinding code.
32710 Original commit message from CVS:
32711 * gst/subparse/gstsubparse.c:
32712 (gst_sub_parse_data_format_autodetect):
32713 Don't use libxml functions in the typefinding code.
32715 2006-05-15 15:01:08 +0000 Wim Taymans <wim.taymans@gmail.com>
32717 ext/ogg/gstoggdemux.c: Fix seeking performance in the case where a non-header packet has a 0 granulepos (busted theor...
32718 Original commit message from CVS:
32719 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
32720 Fix seeking performance in the case where a non-header
32721 packet has a 0 granulepos (busted theora case).
32724 2006-05-15 14:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
32726 gst/subparse/gstsubparse.c: Improve SAMI typefinding: handle case where there are whitespaces or newlines in front of...
32727 Original commit message from CVS:
32728 * gst/subparse/gstsubparse.c:
32729 (gst_sub_parse_data_format_autodetect):
32730 Improve SAMI typefinding: handle case where there are
32731 whitespaces or newlines in front of the first <SAMI>
32734 2006-05-15 12:18:13 +0000 Tim-Philipp Müller <tim@centricular.net>
32736 configure.ac: Build video4linux plugin even if there's no XVIDEO, just without implementing the GstXOverlay interface...
32737 Original commit message from CVS:
32739 Build video4linux plugin even if there's no XVIDEO, just
32740 without implementing the GstXOverlay interface (#334002).
32742 2006-05-15 10:17:04 +0000 Tim-Philipp Müller <tim@centricular.net>
32744 Add tentative support for libvisual-0.4 (#336881).
32745 Original commit message from CVS:
32747 * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
32749 Add tentative support for libvisual-0.4 (#336881).
32751 2006-05-15 09:41:03 +0000 Young-Ho Cha <ganadist@chollian.net>
32753 gst/subparse/samiparse.c: Need to map "silver" colour explicitly (#169936).
32754 Original commit message from CVS:
32755 Patch by: Young-Ho Cha <ganadist at chollian net>
32756 * gst/subparse/samiparse.c: (handle_start_font):
32757 Need to map "silver" colour explicitly (#169936).
32759 2006-05-15 09:14:35 +0000 Young-Ho Cha <ganadist@chollian.net>
32761 gst/subparse/: Add support for SAMI subtitles (#169936).
32762 Original commit message from CVS:
32763 Patch by: Young-Ho Cha <ganadist at chollian net>
32764 * gst/subparse/Makefile.am:
32765 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
32766 (parser_state_dispose), (gst_sub_parse_data_format_autodetect),
32767 (gst_sub_parse_format_autodetect), (feed_textbuf),
32768 (gst_subparse_type_find), (plugin_init):
32769 * gst/subparse/gstsubparse.h:
32770 * gst/subparse/samiparse.c:
32771 * gst/subparse/samiparse.h:
32772 Add support for SAMI subtitles (#169936).
32774 2006-05-14 21:18:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32776 * win32/common/config.h:
32778 Original commit message from CVS:
32781 2006-05-14 21:18:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32784 fix mistakes in README
32785 Original commit message from CVS:
32786 fix mistakes in README
32788 2006-05-14 18:15:17 +0000 Michael Smith <msmith@xiph.org>
32790 gst/audioconvert/gstchannelmix.c: Fix #341696: crash when mixing L+R+C to mono or stereo.
32791 Original commit message from CVS:
32792 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
32793 Fix #341696: crash when mixing L+R+C to mono or stereo.
32794 * tests/check/Makefile.am:
32795 * tests/check/elements/audioconvert.c: (set_channel_positions),
32796 (get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
32797 (audioconvert_suite):
32798 Add test for the above, including some generic framework bits for
32799 testing multichannel things.
32801 2006-05-14 16:05:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32805 Original commit message from CVS:
32808 === release 0.10.7 ===
32810 2006-05-14 16:00:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32812 configure.ac: releasing 0.10.7, "Leave the gun"
32813 Original commit message from CVS:
32814 2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
32816 releasing 0.10.7, "Leave the gun"
32818 2006-05-14 15:55:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32836 Original commit message from CVS:
32839 2006-05-12 22:22:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32842 Original commit message from CVS:
32843 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
32844 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
32847 2006-05-12 21:30:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32849 Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
32850 Original commit message from CVS:
32851 * docs/libs/gst-plugins-base-libs-docs.sgml:
32852 * docs/libs/gst-plugins-base-libs-sections.txt:
32853 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
32854 * gst-libs/gst/video/video.h:
32855 * gst/videoscale/Makefile.am:
32856 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
32857 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
32858 * tests/check/Makefile.am:
32859 * tests/check/libs/video.c: (GST_START_TEST), (video_suite),
32861 Fix integer overflow problem with pixel-aspect-ratio calculations
32862 in videoscale and xvimagesink (#341542)
32864 2006-05-12 16:56:52 +0000 Tim-Philipp Müller <tim@centricular.net>
32866 gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
32867 Original commit message from CVS:
32868 * gst-libs/gst/tag/gstid3tag.c:
32869 Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
32871 2006-05-12 10:39:08 +0000 Sébastien Moutte <sebastien@moutte.net>
32873 win32/MANIFEST: update win32 files listing
32874 Original commit message from CVS:
32876 update win32 files listing
32878 2006-05-11 21:47:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32880 * tests/check/elements/multifdsink.c:
32881 disable failing check on gentoo64
32882 Original commit message from CVS:
32883 disable failing check on gentoo64
32885 2006-05-11 21:35:44 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32887 * tests/check/elements/multifdsink.c:
32888 disable failing check on gentoo64
32889 Original commit message from CVS:
32890 disable failing check on gentoo64
32892 2006-05-11 21:20:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32894 * tests/check/elements/multifdsink.c:
32895 macros show the correct line
32896 Original commit message from CVS:
32897 macros show the correct line
32899 2006-05-11 21:04:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32901 * tests/check/elements/multifdsink.c:
32902 macros show the correct line
32903 Original commit message from CVS:
32904 macros show the correct line
32906 2006-05-11 21:01:05 +0000 Sjoerd Simons <sjoerd@luon.net>
32908 gst/playback/gstplaybasebin.*: API: GstPlayBaseBin::stream-info-value-array property use a more bindings-friendly way...
32909 Original commit message from CVS:
32910 2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
32911 patch by: Sjoerd Simons (sjoerd@luon.net)
32912 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
32913 (group_create), (group_destroy), (add_stream),
32914 (gst_play_base_bin_get_property),
32915 (gst_play_base_bin_get_streaminfo_value_array):
32916 * gst/playback/gstplaybasebin.h:
32917 API: GstPlayBaseBin::stream-info-value-array property
32918 use a more bindings-friendly way of exposing streaminfo
32919 using a GValueArray. Tested in ipython.
32922 2006-05-11 19:44:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32924 * tests/check/elements/multifdsink.c:
32925 fix some type warnings
32926 Original commit message from CVS:
32927 fix some type warnings
32929 2006-05-11 19:38:22 +0000 Wim Taymans <wim.taymans@gmail.com>
32931 gst/playback/gstdecodebin.c: Also catch queue underruns but don't do anything yet.
32932 Original commit message from CVS:
32933 * gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
32934 (queue_underrun_cb), (queue_filled_cb):
32935 Also catch queue underruns but don't do anything yet.
32936 Refactor and comment queue enlarging code a bit.
32937 * gst/playback/gstplaybasebin.c: (queue_overrun),
32938 (queue_threshold_reached), (queue_out_of_data),
32939 (gen_preroll_element):
32940 If a queue over/underruns check that we don't create nasty
32941 deadlocks when the min-threshold is not reached but the
32942 max-bytes is. In those cases disable max-bytes when we
32943 know that the queue is fed timed data.
32946 2006-05-11 18:06:18 +0000 Tim-Philipp Müller <tim@centricular.net>
32948 gst/playback/gstplaybin.c: Make playbin automatically plug an 'audioresample' element before the audio sink as well. ...
32949 Original commit message from CVS:
32950 * gst/playback/gstplaybin.c: (gen_audio_element):
32951 Make playbin automatically plug an 'audioresample'
32952 element before the audio sink as well. This solves
32953 problems with sinks that only accept a very specific
32954 sample rate, like esdsink (e.g. #340379).
32956 2006-05-11 16:04:28 +0000 Tim-Philipp Müller <tim@centricular.net>
32958 gst/playback/gstplaybasebin.c: Make http sources send special headers so that we receive icecast metadata if the http...
32959 Original commit message from CVS:
32960 * gst/playback/gstplaybasebin.c: (gen_source_element):
32961 Make http sources send special headers so that we receive
32962 icecast metadata if the http stream is an icecast stream
32963 (otherwise the server will just ignore them). This also
32964 means that from now on users will need the 'icydemux'
32965 element from gst-plugins-good installed if they want to
32966 listen to icecast radio streams. (#341432, #333657).
32968 2006-05-11 12:34:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32970 * gst/tcp/gstmultifdsink.c:
32972 Original commit message from CVS:
32975 2006-05-11 11:40:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32977 gst/tcp/gstmultifdsink.c: remove stupid example from docs - it should come with a simple
32978 Original commit message from CVS:
32979 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
32980 (gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
32981 remove stupid example from docs - it should come with a simple
32984 * tests/check/elements/multifdsink.c: (wait_bytes_served),
32985 (fail_if_can_read), (GST_START_TEST),
32986 (gst_multifdsink_create_streamheader), (multifdsink_suite):
32987 add a test for changing streamheader which exposes a bug in
32990 2006-05-11 10:33:46 +0000 Michael Smith <msmith@xiph.org>
32992 ext/gnomevfs/gstgnomevfssrc.*: Don't set icy-caps unless we have a sane interval value. Move interval to a local vari...
32993 Original commit message from CVS:
32994 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
32995 (gst_gnome_vfs_src_received_headers_callback):
32996 * ext/gnomevfs/gstgnomevfssrc.h:
32997 Don't set icy-caps unless we have a sane interval value. Move
32998 interval to a local variable; we never use it outside this function.
33000 2006-05-11 10:14:20 +0000 Wim Taymans <wim.taymans@gmail.com>
33002 sys/: Register special buffer types along with the objects so that they are not registered at runtime from N differen...
33003 Original commit message from CVS:
33004 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
33005 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
33006 Register special buffer types along with the objects so
33007 that they are not registered at runtime from N different
33008 streaming threads since they are not threadsafe.
33010 2006-05-10 18:31:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33012 * tests/check/elements/multifdsink.c:
33013 set caps and plug leaks
33014 Original commit message from CVS:
33015 set caps and plug leaks
33017 2006-05-10 18:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33019 tests/check/elements/multifdsink.c: add two more tests, one doing streamheader
33020 Original commit message from CVS:
33021 * tests/check/elements/multifdsink.c: (wait_bytes_served),
33022 (GST_START_TEST), (fail_unless_read), (multifdsink_suite):
33023 add two more tests, one doing streamheader
33025 2006-05-10 16:34:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33027 gst/tcp/gstmultifdsink.c: clean up the bufqueue when shutting down
33028 Original commit message from CVS:
33029 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
33030 clean up the bufqueue when shutting down
33031 * tests/check/Makefile.am:
33032 * tests/check/elements/multifdsink.c: (setup_multifdsink),
33033 (cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
33035 add a test for the leak that was just fixed
33037 2006-05-10 15:16:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33039 * gst/tcp/gstmultifdsink.c:
33041 Original commit message from CVS:
33044 2006-05-10 15:14:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33046 * gst/tcp/gstmultifdsink.c:
33047 * gst/tcp/gstmultifdsink.h:
33049 Original commit message from CVS:
33052 2006-05-10 11:54:36 +0000 Wim Taymans <wim.taymans@gmail.com>
33054 gst/adder/gstadder.*: Updated some docs. Added comments and FIXMEs all over the place.
33055 Original commit message from CVS:
33056 * gst/adder/gstadder.c: (gst_adder_setcaps),
33057 (gst_adder_query_duration), (gst_adder_query), (forward_event),
33058 (gst_adder_src_event), (gst_adder_sink_event),
33059 (gst_adder_class_init), (gst_adder_finalize),
33060 (gst_adder_request_new_pad), (gst_adder_collected):
33061 * gst/adder/gstadder.h:
33062 Updated some docs. Added comments and FIXMEs all over the place.
33063 Improve debugging info.
33064 Fix leak on finalize by not calling the parent.
33065 Implement duration query.
33066 Make event forwarding threadsafe.
33067 Correctly send NEWSEGMENT at start and after flush.
33068 Handle EOS correctly.
33069 Post error when not negotiated.
33070 * tests/check/elements/adder.c: (GST_START_TEST):
33071 Added FIXME in the test.
33073 2006-05-09 19:24:46 +0000 Tim-Philipp Müller <tim@centricular.net>
33075 Const-ify GEnumValue and GFlagsValue arrays. Use
33076 Original commit message from CVS:
33077 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
33078 (gst_text_overlay_halign_get_type),
33079 (gst_text_overlay_wrap_mode_get_type):
33080 * ext/theora/theoradec.c: (theora_handle_type_packet),
33081 (theora_handle_data_packet):
33082 * ext/theora/theoraenc.c: (gst_border_mode_get_type),
33083 (theora_enc_sink_setcaps), (theora_enc_chain):
33084 * gst-libs/gst/cdda/gstcddabasesrc.c:
33085 (gst_cdda_base_src_mode_get_type):
33086 * gst/audiotestsrc/gstaudiotestsrc.c:
33087 (gst_audiostestsrc_wave_get_type):
33088 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
33089 * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
33090 * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
33091 (gst_sync_method_get_type), (gst_unit_type_get_type),
33092 (gst_client_status_get_type):
33093 * gst/videoscale/gstvideoscale.c:
33094 (gst_video_scale_method_get_type):
33095 * gst/videotestsrc/gstvideotestsrc.c:
33096 (gst_video_test_src_pattern_get_type):
33097 * gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
33098 (paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
33099 (paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
33100 (paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
33101 (paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
33102 (paint_setup_RGB565), (paint_setup_xRGB1555):
33103 Const-ify GEnumValue and GFlagsValue arrays. Use
33104 GST_ROUND_UP_* macros instead of home-made ones.
33106 2006-05-09 17:40:41 +0000 Tim-Philipp Müller <tim@centricular.net>
33108 configure.ac: Require core CVS for the new newsegment stuff.
33109 Original commit message from CVS:
33111 Require core CVS for the new newsegment stuff.
33113 2006-05-09 17:30:48 +0000 Sjoerd Simons <sjoerd@luon.net>
33115 gst/tcp/gstmultifdsink.c: Register nick for enum value (#341160).
33116 Original commit message from CVS:
33117 Patch by: Sjoerd Simons <sjoerd at luon net>
33118 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
33119 Register nick for enum value (#341160).
33121 2006-05-09 16:46:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33123 gst/typefind/gsttypefindfunctions.c: backout typefind patch #340375
33124 Original commit message from CVS:
33125 * gst/typefind/gsttypefindfunctions.c: (m4a_type_find),
33127 backout typefind patch #340375
33128 * tests/check/elements/adder.c: (message_received),
33129 (GST_START_TEST), (adder_suite):
33130 redo, signal-handling of test
33132 2006-05-09 16:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
33134 gst/adder/gstadder.*: Remove bogus segment merging and forwarding, we don't care about timestamps anyway and we just ...
33135 Original commit message from CVS:
33136 * gst/adder/gstadder.c: (gst_adder_request_new_pad),
33137 (gst_adder_collected):
33138 * gst/adder/gstadder.h:
33139 Remove bogus segment merging and forwarding, we don't
33140 care about timestamps anyway and we just produce a
33142 Also create a nice NEWSEGMENT event when we start.
33143 Use _scale_int some more.
33145 2006-05-09 11:59:13 +0000 Edward Hervey <bilboed@bilboed.com>
33147 tests/icles/stress-xoverlay.c: Fix if core was built without parsing support.
33148 Original commit message from CVS:
33149 * tests/icles/stress-xoverlay.c:
33150 Fix if core was built without parsing support.
33152 2006-05-09 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
33154 gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
33155 Original commit message from CVS:
33156 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
33157 Add SEDG (Samsung MPEG-4) fourcc.
33159 2006-05-09 11:31:47 +0000 Edward Hervey <bilboed@bilboed.com>
33161 tests/examples/volume/volume.c: Fox if core was built without parsing support.
33162 Original commit message from CVS:
33163 * tests/examples/volume/volume.c:
33164 Fox if core was built without parsing support.
33165 * tests/examples/seek/seek.c:
33166 Disable the parse_launch example if core was built without parsing
33169 2006-05-09 11:21:24 +0000 Edward Hervey <bilboed@bilboed.com>
33171 tests/examples/seek/seek.c: Disable the parse_launch example if core was built without parsing support.
33172 Original commit message from CVS:
33173 * tests/examples/seek/seek.c:
33174 Disable the parse_launch example if core was built without parsing
33177 2006-05-08 15:51:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33179 * docs/libs/tmpl/gstcolorbalance.sgml:
33180 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
33181 * gst/tcp/gstmultifdsink.c:
33182 * gst/videoscale/gstvideoscale.c:
33183 doc reparagraphing and DEBUG_FUNCPTRing
33184 Original commit message from CVS:
33185 doc reparagraphing and DEBUG_FUNCPTRing
33187 2006-05-08 11:51:43 +0000 Edward Hervey <bilboed@bilboed.com>
33189 autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize
33190 Original commit message from CVS:
33191 * autogen.sh: (CONFIGURE_DEF_OPT):
33192 libtoolize on Darwin/MacOSX is called glibtoolize
33194 2006-05-07 17:39:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33196 tests/check/: Disable the adder test, until the build-slaves posses the kindness to either like it or to give valid r...
33197 Original commit message from CVS:
33198 * tests/check/Makefile.am:
33199 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
33200 Disable the adder test, until the build-slaves posses the kindness to
33201 either like it or to give valid reason for not doing so
33203 2006-05-07 17:25:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33205 tests/check/elements/adder.c: Shuffle NULL state change around and raise timeout more
33206 Original commit message from CVS:
33207 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
33209 Shuffle NULL state change around and raise timeout more
33211 2006-05-07 17:07:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33213 gst/typefind/gsttypefindfunctions.c: Add typefind to distinguish between "audio/x-m4a" and new type "video/mp4". Fixe...
33214 Original commit message from CVS:
33215 * gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
33216 (mp4_type_find), (plugin_init):
33217 Add typefind to distinguish between "audio/x-m4a" and new type
33218 "video/mp4". Fixes #340375
33219 * tests/check/elements/adder.c: (adder_suite):
33220 Raise timeout to make buildbot happy
33222 2006-05-07 16:39:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33224 Add sink-event handling to adder. It tries to merge incomming newsegment-events. Added test to check if segment_done ...
33225 Original commit message from CVS:
33226 * gst/adder/gstadder.c: (gst_adder_sink_event),
33227 (gst_adder_request_new_pad), (gst_adder_change_state):
33228 * gst/adder/gstadder.h:
33229 * tests/check/Makefile.am:
33230 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
33231 (adder_suite), (main):
33232 Add sink-event handling to adder. It tries to merge incomming
33233 newsegment-events. Added test to check if segment_done is comming
33236 2006-05-05 16:34:15 +0000 Andy Wingo <wingo@pobox.com>
33239 * ext/theora/theoraparse.c:
33240 * ext/vorbis/vorbisparse.c:
33241 ext/theora/theoraparse.c (gst_theora_parse_init) ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
33242 Original commit message from CVS:
33243 2006-05-05 Andy Wingo <wingo@pobox.com>
33244 * ext/theora/theoraparse.c (gst_theora_parse_init)
33245 (theora_parse_src_convert, theora_parse_src_query):
33246 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
33247 (vorbis_parse_convert, vorbis_parse_src_query): Add convert and
33248 query functions on the source pads of the theora and vorbis parse
33249 elements. Fixes position querying when doing a remux.
33251 2006-05-05 13:46:37 +0000 Michael Smith <msmith@xiph.org>
33253 ext/theora/theoraparse.c: Fix flushing.
33254 Original commit message from CVS:
33255 * ext/theora/theoraparse.c: (parse_granulepos),
33256 (theora_parse_drain_queue_prematurely),
33257 (theora_parse_queue_buffer), (theora_parse_sink_event):
33259 Fix invalid granulepos outputs when starting with a non-keyframe.
33261 2006-05-05 12:37:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33263 gst/typefind/gsttypefindfunctions.c: Rearrange MPEG system stream detection, fixing some memleaks in the process.
33264 Original commit message from CVS:
33265 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
33266 (mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
33267 Rearrange MPEG system stream detection, fixing some memleaks in the
33269 Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
33270 they clean up their data correctly.
33271 Remove unused ogganx caps and move the 'is_annodex' check to inside
33272 the 'is_ogg' if statement.
33274 2006-05-05 11:33:37 +0000 Wim Taymans <wim.taymans@gmail.com>
33276 gst/playback/gstdecodebin.c: Properly remove ghostpads. Fixes #340392
33277 Original commit message from CVS:
33278 * gst/playback/gstdecodebin.c: (cleanup_decodebin):
33279 Properly remove ghostpads. Fixes #340392
33281 2006-05-04 18:43:58 +0000 David Schleef <ds@schleef.org>
33283 gst/typefind/gsttypefindfunctions.c:
33284 Original commit message from CVS:
33285 * gst/typefind/gsttypefindfunctions.c:
33287 2006-05-03 16:32:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33289 gst/typefind/gsttypefindfunctions.c: When typefinding an MP3 in push-based mode, don't penalise the probability down ...
33290 Original commit message from CVS:
33291 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
33292 (mpeg_ts_probe_headers), (mpeg_ts_type_find):
33293 When typefinding an MP3 in push-based mode, don't penalise the
33294 probability down to 74% when we found 5 valid frames just because we
33295 can't peek the end of the file.
33296 Make the probability for detecting MPEG Transport Streams based on the
33297 number of sequential headers we successfully detected.
33299 2006-05-03 15:52:46 +0000 Wim Taymans <wim.taymans@gmail.com>
33301 ext/vorbis/vorbisdec.c: Still produce an error when we receive an empty packet.
33302 Original commit message from CVS:
33303 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
33304 (vorbis_dec_push), (vorbis_dec_chain):
33305 Still produce an error when we receive an empty packet.
33307 2006-05-03 15:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
33309 ext/ogg/gstoggdemux.c: Mark buffers with DISCONT after seek and after activating new chains.
33310 Original commit message from CVS:
33311 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
33312 (gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
33313 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
33314 Mark buffers with DISCONT after seek and after activating new
33316 * ext/theora/gsttheoradec.h:
33317 * ext/theora/theoradec.c: (gst_theora_dec_reset),
33318 (theora_get_query_types), (theora_dec_sink_event),
33319 (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
33320 (theora_dec_change_state):
33322 Detect and mark DISCONT buffers.
33323 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
33324 (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
33325 (vorbis_dec_change_state):
33326 * ext/vorbis/vorbisdec.h:
33328 Detect and mark DISCONT buffers.
33329 Don't crash on 0 sized buffers.
33331 2006-05-03 08:58:13 +0000 Wim Taymans <wim.taymans@gmail.com>
33333 gst/volume/gstvolume.c: Increase "volume" property to 10.0. Fixes #340369.
33334 Original commit message from CVS:
33335 * gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
33336 (volume_transform_ip):
33337 Increase "volume" property to 10.0. Fixes #340369.
33338 Set the process function to NULL when capsnego fails so that
33339 we properly error out.
33341 2006-05-02 18:15:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33343 gst/playback/: free cpas using gst_caps_unref, don't leak caps-strings
33344 Original commit message from CVS:
33345 * gst/playback/gstplaybin.c: (add_sink):
33346 * gst/playback/test.c: (main):
33347 * gst/playback/test5.c: (dump_element_stats):
33348 * gst/playback/test6.c: (main):
33349 free cpas using gst_caps_unref, don't leak caps-strings
33351 2006-05-02 06:33:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33353 * gst-libs/gst/rtp/gstbasertppayload.c:
33355 Original commit message from CVS:
33358 2006-05-01 19:08:40 +0000 Tim-Philipp Müller <tim@centricular.net>
33360 gst/typefind/gsttypefindfunctions.c: Refine musepack typefinding a bit. Return MAXIMUM probability when we detect str...
33361 Original commit message from CVS:
33362 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
33364 Refine musepack typefinding a bit. Return MAXIMUM
33365 probability when we detect stream version 7 to make
33366 sure the mpeg audio typefinder doesn't trump us.
33368 2006-04-29 16:25:58 +0000 Tim-Philipp Müller <tim@centricular.net>
33370 gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
33371 Original commit message from CVS:
33372 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
33373 Protect against unexpected NULL strf_data buffer.
33375 2006-04-29 13:09:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33377 tests/check/elements/audioconvert.c: interpret the out[] buffer in the order the bytes are actually put in, which is ...
33378 Original commit message from CVS:
33379 * tests/check/elements/audioconvert.c: (verify_convert),
33381 interpret the out[] buffer in the order the bytes are actually
33382 put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
33383 Other tests should use BYTE_ORDER since the array is filled in
33386 2006-04-29 12:10:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33388 * tests/check/elements/audioconvert.c:
33389 dump expected data when audioconvert test fails
33390 Original commit message from CVS:
33391 dump expected data when audioconvert test fails
33393 2006-04-29 11:55:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33395 tests/check/elements/audioconvert.c: when a test fails, give an indication of which it is
33396 Original commit message from CVS:
33397 * tests/check/elements/audioconvert.c: (verify_convert),
33399 when a test fails, give an indication of which it is
33401 2006-04-29 09:48:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33403 * ext/ogg/gstoggmux.c:
33404 * ext/theora/theoraenc.c:
33405 add another include
33406 Original commit message from CVS:
33407 add another include
33409 2006-04-29 01:24:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33411 * gst/subparse/gstssaparse.c:
33412 atoi() needs stdlib.h
33413 Original commit message from CVS:
33414 atoi() needs stdlib.h
33416 2006-04-29 01:18:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33418 * gst/playback/test4.c:
33419 * gst/playback/test5.c:
33420 * gst/playback/test6.c:
33421 exit needs stdlib.h
33422 Original commit message from CVS:
33423 exit needs stdlib.h
33425 2006-04-29 01:10:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33427 gst-libs/gst/cdda/gstcddabasesrc.c: compile fix; strtol() needs <stdlib.h>
33428 Original commit message from CVS:
33429 * gst-libs/gst/cdda/gstcddabasesrc.c:
33430 compile fix; strtol() needs <stdlib.h>
33432 2006-04-29 01:04:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33436 * docs/Makefile.am:
33437 * docs/libs/Makefile.am:
33438 * docs/libs/tmpl/gstcolorbalance.sgml:
33439 * docs/plugins/Makefile.am:
33441 use common upload.mak
33442 Original commit message from CVS:
33443 use common upload.mak
33445 2006-04-28 19:46:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33447 make GstElementDetails const
33448 Original commit message from CVS:
33449 * ext/alsa/gstalsamixerelement.c:
33450 * ext/alsa/gstalsasrc.c:
33451 * ext/cdparanoia/gstcdparanoiasrc.c:
33452 * ext/gnomevfs/gstgnomevfssink.c:
33453 * ext/gnomevfs/gstgnomevfssrc.c:
33454 * ext/ogg/gstoggdemux.c:
33455 * ext/ogg/gstoggmux.c:
33456 * ext/ogg/gstoggparse.c:
33457 * ext/ogg/gstogmparse.c:
33458 * ext/pango/gstclockoverlay.c:
33459 * ext/pango/gsttextoverlay.c:
33460 * ext/pango/gsttextrender.c:
33461 * ext/pango/gsttimeoverlay.c:
33462 * ext/theora/theoradec.c:
33463 * ext/theora/theoraenc.c:
33464 * ext/vorbis/vorbisdec.c:
33465 * ext/vorbis/vorbisenc.c:
33466 * gst-libs/gst/audio/gstaudiofilter.c:
33467 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
33468 * gst/audioconvert/gstaudioconvert.c:
33469 * gst/audiorate/gstaudiorate.c:
33470 * gst/audioresample/gstaudioresample.c:
33471 * gst/audiotestsrc/gstaudiotestsrc.c:
33472 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
33473 * gst/playback/gstdecodebin.c:
33474 * gst/playback/gstplaybin.c:
33475 * gst/playback/gststreamselector.c:
33476 * gst/subparse/gstsubparse.c:
33477 * gst/tcp/gstmultifdsink.c:
33478 * gst/tcp/gsttcpclientsink.c:
33479 * gst/tcp/gsttcpclientsrc.c:
33480 * gst/tcp/gsttcpserversink.c:
33481 * gst/tcp/gsttcpserversrc.c:
33482 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
33483 * gst/videorate/gstvideorate.c:
33484 * gst/videoscale/gstvideoscale.c:
33485 * gst/videotestsrc/gstvideotestsrc.c:
33486 * gst/volume/gstvolume.c:
33487 * sys/v4l/gstv4ljpegsrc.c:
33488 * sys/v4l/gstv4lmjpegsink.c:
33489 * sys/v4l/gstv4lmjpegsrc.c:
33490 * sys/v4l/gstv4lsrc.c:
33491 * sys/ximage/ximagesink.c:
33492 * sys/xvimage/xvimagesink.c:
33493 * tests/check/libs/cddabasesrc.c:
33494 make GstElementDetails const
33496 2006-04-28 19:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33498 gst/adder/gstadder.c: send events from src-pad to all sink-pads fixes #338657
33499 Original commit message from CVS:
33500 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
33502 send events from src-pad to all sink-pads fixes #338657
33504 2006-04-28 19:08:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33506 ext/alsa/gstalsasink.c: query witdh capabilities from alsa, fixes #338919
33507 Original commit message from CVS:
33508 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
33509 (alsasink_parse_spec):
33510 query witdh capabilities from alsa, fixes #338919
33512 2006-04-28 15:31:28 +0000 Wim Taymans <wim.taymans@gmail.com>
33514 gst/tcp/gstmultifdsink.*: Fix race condition in multifdsink that can lead to spurious duplicate clients. this patch a...
33515 Original commit message from CVS:
33516 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
33517 (gst_multi_fd_sink_remove_client_link):
33518 * gst/tcp/gstmultifdsink.h:
33519 Fix race condition in multifdsink that can lead to spurious
33520 duplicate clients. this patch adds a new signal that is fired when
33521 multifdsink has removed all references to the fd.
33523 Updated documentation.
33524 API: client-fd-removed signal added
33526 2006-04-28 15:24:00 +0000 Michael Smith <msmith@xiph.org>
33528 gst/tcp/gstmultifdsink.c: When asking g_value_array_new to prealloc elements, we may as well ask for the right number...
33529 Original commit message from CVS:
33530 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
33531 When asking g_value_array_new to prealloc elements, we may as well
33532 ask for the right number of elements.
33534 2006-04-28 15:08:09 +0000 Wim Taymans <wim.taymans@gmail.com>
33536 gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
33537 Original commit message from CVS:
33538 * gst-libs/gst/audio/gstbaseaudiosink.c:
33539 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
33540 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
33541 patch to make timestamp checking more tollerant to rounding
33542 errors given that real discontinuities are to be marked on
33543 buffers. Fixes some asf files and #338778.
33544 Also avoid some crashers when we receive an event in the
33547 2006-04-28 15:01:58 +0000 Michael Smith <msmith@xiph.org>
33549 ext/gnomevfs/gstgnomevfssrc.*: Remove ICY handling (mostly) from gnomevfssrc, in favour of proper shared support with...
33550 Original commit message from CVS:
33551 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
33552 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
33553 (gst_gnome_vfs_src_get_property),
33554 (gst_gnome_vfs_src_send_additional_headers_callback),
33555 (gst_gnome_vfs_src_received_headers_callback),
33556 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
33557 (gst_gnome_vfs_src_stop):
33558 * ext/gnomevfs/gstgnomevfssrc.h:
33559 Remove ICY handling (mostly) from gnomevfssrc, in favour of
33560 proper shared support within icydemux.
33562 2006-04-28 14:49:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33564 gst/videorate/gstvideorate.c: fix up docs fix a leak when no caps negotiated fix counting of input frames
33565 Original commit message from CVS:
33566 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
33567 (gst_video_rate_swap_prev), (gst_video_rate_chain):
33569 fix a leak when no caps negotiated
33570 fix counting of input frames
33571 * tests/check/elements/.cvsignore:
33572 * tests/check/elements/videorate.c: (assert_videorate_stats),
33573 (GST_START_TEST), (videorate_suite):
33574 add tests for these
33576 2006-04-28 14:48:11 +0000 Wim Taymans <wim.taymans@gmail.com>
33578 gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
33579 Original commit message from CVS:
33580 * gst-libs/gst/audio/gstringbuffer.c:
33581 (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
33582 (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
33583 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
33584 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
33585 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
33586 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
33587 (gst_ring_buffer_commit), (gst_ring_buffer_read),
33588 (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
33589 (gst_ring_buffer_clear), (gst_ring_buffer_may_start):
33590 Check arguments passed to public functions instead of
33593 2006-04-28 14:37:46 +0000 Wim Taymans <wim.taymans@gmail.com>
33595 gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
33596 Original commit message from CVS:
33597 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
33598 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
33599 GstBaseAudioSrc must be live or it does not work.
33600 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
33601 Don't set live to TRUE as this is the default in the parentclass.
33603 2006-04-28 14:37:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33605 * win32/common/config.h:
33607 Original commit message from CVS:
33610 2006-04-28 14:33:45 +0000 Wim Taymans <wim.taymans@gmail.com>
33612 gst/videoscale/gstvideoscale.c: Videoscale doesn't pass on pixel-aspect ratio. Handle all fixation cases better. Fixe...
33613 Original commit message from CVS:
33614 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps),
33615 (gst_video_scale_fixate_caps), (gst_video_scale_src_event):
33616 Videoscale doesn't pass on pixel-aspect ratio. Handle all
33617 fixation cases better. Fixes #338991
33619 2006-04-28 14:24:38 +0000 Wim Taymans <wim.taymans@gmail.com>
33621 gst/videotestsrc/gstvideotestsrc.c: Handle 0/1 framerate correctly Fixes #331901.
33622 Original commit message from CVS:
33623 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
33624 Handle 0/1 framerate correctly Fixes #331901.
33626 2006-04-28 14:22:16 +0000 Wim Taymans <wim.taymans@gmail.com>
33628 tests/check/elements/audioconvert.c: Added check for correct clipping when doing float samples in audioconvert.
33629 Original commit message from CVS:
33630 * tests/check/elements/audioconvert.c: (get_float_caps),
33631 (GST_START_TEST), (audioconvert_suite):
33632 Added check for correct clipping when doing float samples
33635 2006-04-28 14:19:49 +0000 Wim Taymans <wim.taymans@gmail.com>
33637 gst/videorate/gstvideorate.c: Print more debugging info.
33638 Original commit message from CVS:
33639 * gst/videorate/gstvideorate.c: (gst_video_rate_event),
33640 (gst_video_rate_chain):
33641 Print more debugging info.
33643 2006-04-28 14:17:00 +0000 Wim Taymans <wim.taymans@gmail.com>
33645 gst/audioresample/gstaudioresample.c: Add support for other formats audioresample can handle such as 32 bits in and f...
33646 Original commit message from CVS:
33647 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
33648 (resample_set_state_from_caps):
33649 Add support for other formats audioresample can handle such as
33650 32 bits in and float and 64 bits float. Fixes #301759
33652 2006-04-28 14:12:28 +0000 Wim Taymans <wim.taymans@gmail.com>
33654 gst/audioconvert/audioconvert.c: correctly clip float samples > 1.0. Fixes #338718
33655 Original commit message from CVS:
33656 * gst/audioconvert/audioconvert.c: (float):
33657 correctly clip float samples > 1.0. Fixes #338718
33659 2006-04-28 13:35:34 +0000 Young-Ho Cha <ganadist@chollian.net>
33661 ext/pango/gsttextoverlay.c: Don't strip newlines from the text. Also, center lines within multi-line paragraphs (#339...
33662 Original commit message from CVS:
33663 Patch by: Young-Ho Cha <ganadist at chollian net>
33664 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
33665 (gst_text_overlay_render_text):
33666 Don't strip newlines from the text. Also, center lines
33667 within multi-line paragraphs (#339405).
33669 2006-04-28 12:15:33 +0000 Tim-Philipp Müller <tim@centricular.net>
33671 gst/typefind/gsttypefindfunctions.c: Fix wavpack typefinding to work in more cases (don't peek for chunks of multiple...
33672 Original commit message from CVS:
33673 * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
33674 Fix wavpack typefinding to work in more cases (don't peek
33675 for chunks of multiple hundred kBs at once, but process
33676 things step-by-step in smaller units). Fixes #339786.
33678 2006-04-28 10:58:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33683 Original commit message from CVS:
33686 === release 0.10.6 ===
33688 2006-04-28 10:53:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33694 * docs/plugins/gst-plugins-base-plugins.signals:
33695 * docs/plugins/inspect/plugin-adder.xml:
33696 * docs/plugins/inspect/plugin-alsa.xml:
33697 * docs/plugins/inspect/plugin-audioconvert.xml:
33698 * docs/plugins/inspect/plugin-audiorate.xml:
33699 * docs/plugins/inspect/plugin-audioresample.xml:
33700 * docs/plugins/inspect/plugin-audiotestsrc.xml:
33701 * docs/plugins/inspect/plugin-cdparanoia.xml:
33702 * docs/plugins/inspect/plugin-decodebin.xml:
33703 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
33704 * docs/plugins/inspect/plugin-gnomevfs.xml:
33705 * docs/plugins/inspect/plugin-libvisual.xml:
33706 * docs/plugins/inspect/plugin-ogg.xml:
33707 * docs/plugins/inspect/plugin-pango.xml:
33708 * docs/plugins/inspect/plugin-playbin.xml:
33709 * docs/plugins/inspect/plugin-subparse.xml:
33710 * docs/plugins/inspect/plugin-tcp.xml:
33711 * docs/plugins/inspect/plugin-theora.xml:
33712 * docs/plugins/inspect/plugin-typefindfunctions.xml:
33713 * docs/plugins/inspect/plugin-video4linux.xml:
33714 * docs/plugins/inspect/plugin-videorate.xml:
33715 * docs/plugins/inspect/plugin-videoscale.xml:
33716 * docs/plugins/inspect/plugin-videotestsrc.xml:
33717 * docs/plugins/inspect/plugin-volume.xml:
33718 * docs/plugins/inspect/plugin-vorbis.xml:
33719 * docs/plugins/inspect/plugin-ximagesink.xml:
33720 * docs/plugins/inspect/plugin-xvimagesink.xml:
33723 Original commit message from CVS:
33726 2006-04-28 10:42:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33729 * win32/common/config.h:
33730 dist more win32 files
33731 Original commit message from CVS:
33732 dist more win32 files
33734 2006-04-28 10:41:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33751 Original commit message from CVS:
33754 2006-04-27 00:19:29 +0000 David Schleef <ds@schleef.org>
33756 gst/videoscale/gstvideoscale.c: Add call to oil_init().
33757 Original commit message from CVS:
33758 * gst/videoscale/gstvideoscale.c: Add call to oil_init().
33761 2006-04-26 17:20:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33765 * win32/common/config.h:
33767 Original commit message from CVS:
33770 2006-04-26 17:17:39 +0000 Wim Taymans <wim.taymans@gmail.com>
33772 ext/ogg/gstoggdemux.c: make sure correct newsegments are sent, so that the decoder and the demuxer agree on timestamp...
33773 Original commit message from CVS:
33774 2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
33775 patch by: Wim Taymans
33776 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
33777 (gst_ogg_demux_perform_seek):
33778 make sure correct newsegments are sent, so that the decoder
33779 and the demuxer agree on timestamps. Fixes playback of a lot
33780 of Ogg files that do not start from 0. Fixes #339833.
33782 2006-04-26 16:44:20 +0000 Edward Hervey <edward@fluendo.com>
33784 Fix an infinite loop if frames are passed in with wrongly ordered timestamps. Fixes #339013.
33785 Original commit message from CVS:
33786 Patch by: Edward Hervey <edward@fluendo.com>
33787 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
33788 * tests/check/Makefile.am:
33789 * tests/check/elements/videorate.c: (assert_videorate_stats),
33790 (setup_videorate), (cleanup_videorate), (GST_START_TEST),
33791 (videorate_suite), (main):
33792 Fix an infinite loop if frames are passed in with wrongly ordered
33793 timestamps. Fixes #339013.
33795 2006-04-26 13:55:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33798 * win32/common/config.h:
33800 Original commit message from CVS:
33803 2006-04-22 21:25:41 +0000 Tim-Philipp Müller <tim@centricular.net>
33805 gst/typefind/gsttypefindfunctions.c: fix typefinding on some ISO files. Fixes #339212.
33806 Original commit message from CVS:
33807 Patch by: Tim-Philipp Müller <tim at centricular dot net>
33808 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
33809 fix typefinding on some ISO files. Fixes #339212.
33811 2006-04-22 21:19:06 +0000 Tim-Philipp Müller <tim@centricular.net>
33813 gst-libs/gst/riff/riff-media.c: add another H264 fourcc. Fixes #339047.
33814 Original commit message from CVS:
33815 Patch by: Tim-Philipp Müller <tim at centricular dot net>
33816 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
33817 add another H264 fourcc. Fixes #339047.
33819 2006-04-22 21:12:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33821 gst/playback/gststreamselector.c: Restore old StreamSelector behaviour.
33822 Original commit message from CVS:
33823 Patch by: Jan Schmidt
33824 * gst/playback/gststreamselector.c:
33825 (gst_stream_selector_bufferalloc):
33826 Restore old StreamSelector behaviour.
33829 2006-04-13 09:26:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33832 * gst-libs/gst/rtp/Makefile.am:
33833 * gst-libs/gst/rtp/gstrtpbuffer.h:
33834 reverting rtp patches to fix freeze break on -base as explained on the list
33835 Original commit message from CVS:
33836 reverting rtp patches to fix freeze break on -base as explained on the list
33838 2006-04-13 03:55:12 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
33840 gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
33841 Original commit message from CVS:
33842 2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
33843 * gst-libs/gst/rtp/gstrtpbuffer.h:
33844 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
33845 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
33846 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
33847 New RTP audio base payloader class. Supports frame or sample based codecs
33849 2006-04-12 11:04:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33867 update libtool versioning
33868 Original commit message from CVS:
33869 update libtool versioning
33871 2006-04-12 10:58:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33875 * win32/common/config.h:
33877 Original commit message from CVS:
33880 2006-04-11 17:31:29 +0000 Antoine Tremblay <hexa00@gmail.com>
33882 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
33883 Original commit message from CVS:
33884 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
33885 * gst-libs/gst/rtp/gstbasertpdepayload.c:
33886 (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
33887 Fix some memory leaks: on finalize, free buffers left in the queue
33888 before destroying the queue; in _push(), unref rtp_buf even if
33889 the process vfunc returned a NULL buffer as output buffer (#337548);
33890 demote some recuring debug messages to LOG level.
33892 2006-04-11 15:01:51 +0000 Christian Schaller <uraeus@gnome.org>
33894 * gst-plugins-base.spec.in:
33895 fix version number macro
33896 Original commit message from CVS:
33897 fix version number macro
33899 2006-04-11 14:42:33 +0000 Wim Taymans <wim.taymans@gmail.com>
33901 ext/ogg/gstoggdemux.c: More cleanups.
33902 Original commit message from CVS:
33903 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
33904 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
33905 (gst_ogg_chain_free), (gst_ogg_demux_sink_event),
33906 (gst_ogg_demux_loop):
33908 Respect segment stop when emiting EOS or SEGMENT_DONE.
33911 2006-04-11 10:45:32 +0000 Tim-Philipp Müller <tim@centricular.net>
33913 gst/playback/gststreamselector.c: Don't leak pad name.
33914 Original commit message from CVS:
33915 * gst/playback/gststreamselector.c:
33916 (gst_stream_selector_get_property):
33917 Don't leak pad name.
33919 2006-04-11 09:42:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33922 Mention bug #336617 closed by recent commit
33923 Original commit message from CVS:
33924 Mention bug #336617 closed by recent commit
33926 2006-04-10 20:32:46 +0000 Michael Smith <msmith@xiph.org>
33928 tests/check/: so that FC4 buildslaves can pass.
33929 Original commit message from CVS:
33930 * tests/check/Makefile.am:
33931 * tests/check/gst-plugins-base.supp:
33932 Suppress an old libtheora bug (fixed in more recent versions), so
33933 that FC4 buildslaves can pass.
33935 2006-04-10 19:13:30 +0000 Wim Taymans <wim.taymans@gmail.com>
33937 ext/ogg/gstoggdemux.c: Don't leak events.
33938 Original commit message from CVS:
33939 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
33940 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
33941 (gst_ogg_demux_init), (gst_ogg_demux_finalize),
33942 (gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
33943 (gst_ogg_demux_loop):
33945 Remember what error we got when finding chains, if we
33946 were shutdown, that would not be an error.
33948 2006-04-10 17:05:46 +0000 Wim Taymans <wim.taymans@gmail.com>
33950 gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
33951 Original commit message from CVS:
33952 * gst-libs/gst/audio/gstbaseaudiosink.c:
33953 (gst_base_audio_sink_event):
33954 Starting the ringbuffer when we did not acquire it can cause
33955 a deadlock, is pointless and causes nasty things for
33957 Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
33959 2006-04-10 15:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
33961 ext/ogg/gstoggdemux.c: Add some more debugging.
33962 Original commit message from CVS:
33963 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
33964 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
33965 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
33966 (gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
33967 (gst_ogg_demux_deactivate_current_chain),
33968 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
33969 (gst_ogg_demux_bisect_forward_serialno),
33970 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
33971 Add some more debugging.
33973 2006-04-10 14:52:10 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33976 * ext/theora/theoraenc.c:
33978 Original commit message from CVS:
33981 2006-04-10 10:29:21 +0000 Wim Taymans <wim.taymans@gmail.com>
33983 ext/theora/theoradec.c: Some more debug info.
33984 Original commit message from CVS:
33985 * ext/theora/theoradec.c: (theora_dec_src_event),
33986 (theora_handle_data_packet):
33987 Some more debug info.
33988 * tests/examples/seek/seek.c: (start_seek), (main):
33989 Print element messages too.
33991 2006-04-09 17:14:22 +0000 Sébastien Moutte <sebastien@moutte.net>
33993 gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer sta...
33994 Original commit message from CVS:
33995 * gst/audioresample/debug.h:
33996 replace debug macros with variable number of parameters
33997 by a simple alias to gstreamer standard debug macros
33998 (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
33999 supported by MSVC 6.0 and 7.1)
34000 * gst/audioresample/resample.h:
34001 define M_PI and rint for WIN32
34002 * win32/common/libgstaudio.def:
34003 * win32/common/libgstriff.def:
34004 * win32/common/libgsttag.def:
34005 * win32/common/libgstvideo.def:
34006 add new exported functions
34008 update project files
34010 2006-04-08 21:02:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
34012 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
34013 Original commit message from CVS:
34014 * ext/alsa/gstalsamixeroptions.c:
34015 (gst_alsa_mixer_options_class_init):
34016 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
34017 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
34018 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
34019 * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
34020 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
34021 * gst-libs/gst/audio/gstaudiofilter.c:
34022 (gst_audio_filter_class_init):
34023 * gst-libs/gst/audio/gstaudiosink.c:
34024 (gst_audioringbuffer_class_init):
34025 * gst-libs/gst/audio/gstaudiosrc.c:
34026 (gst_audioringbuffer_class_init):
34027 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
34028 * gst-libs/gst/interfaces/colorbalancechannel.c:
34029 (gst_color_balance_channel_class_init):
34030 * gst-libs/gst/interfaces/mixeroptions.c:
34031 (gst_mixer_options_class_init):
34032 * gst-libs/gst/interfaces/mixertrack.c:
34033 (gst_mixer_track_class_init):
34034 * gst-libs/gst/interfaces/tunerchannel.c:
34035 (gst_tuner_channel_class_init):
34036 * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
34037 * gst-libs/gst/netbuffer/gstnetbuffer.c:
34038 (gst_netbuffer_class_init):
34039 * gst-libs/gst/rtp/gstbasertppayload.c:
34040 (gst_basertppayload_class_init):
34041 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
34042 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
34043 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
34044 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
34045 * gst/playback/gststreamselector.c:
34046 (gst_stream_selector_class_init):
34047 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
34048 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
34049 * sys/v4l/gstv4lcolorbalance.c:
34050 (gst_v4l_color_balance_channel_class_init):
34051 * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
34052 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
34053 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
34054 * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
34055 (gst_v4l_tuner_norm_class_init):
34056 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
34057 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
34058 * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
34059 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
34061 2006-04-08 18:09:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
34063 Fix broken GObject macros
34064 Original commit message from CVS:
34065 * ext/pango/gsttextrender.h:
34066 * gst-libs/gst/audio/gstaudiosink.h:
34067 * gst-libs/gst/audio/gstaudiosrc.h:
34068 * gst-libs/gst/audio/gstbaseaudiosink.h:
34069 * gst-libs/gst/audio/gstbaseaudiosrc.h:
34070 * gst-libs/gst/audio/gstringbuffer.h:
34071 * gst-libs/gst/rtp/gstbasertpdepayload.h:
34072 * gst-libs/gst/rtp/gstbasertppayload.h:
34073 * gst-libs/gst/video/gstvideofilter.h:
34074 * gst-libs/gst/video/gstvideosink.h:
34075 * gst/playback/gstplaybasebin.h:
34076 * gst/tcp/gstmultifdsink.h:
34077 * sys/v4l/gstv4lelement.h:
34078 Fix broken GObject macros
34080 2006-04-08 16:21:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
34082 ext/alsa/gstalsasink.c: More debug to trace why my USB headset is not working with gst
34083 Original commit message from CVS:
34084 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
34085 More debug to trace why my USB headset is not working with gst
34087 2006-04-07 17:18:11 +0000 Jan Schmidt <thaytan@mad.scientist.com>
34089 gst/playback/gstplaybasebin.c: Clean up our group elements properly in the case where it never got committed - it sti...
34090 Original commit message from CVS:
34091 * gst/playback/gstplaybasebin.c: (group_destroy):
34092 Clean up our group elements properly in the case where it never
34093 got committed - it still got added unconditionally to the bin.
34095 2006-04-07 15:14:32 +0000 Wim Taymans <wim.taymans@gmail.com>
34097 ext/theora/theoradec.c: Unref unhandled events.
34098 Original commit message from CVS:
34099 * ext/theora/theoradec.c: (theora_dec_sink_event),
34100 (theora_handle_data_packet), (theora_dec_chain):
34101 Unref unhandled events.
34102 Protect against empty buffers.
34103 Perform QoS on running time.
34105 2006-04-07 13:24:54 +0000 Michael Smith <msmith@xiph.org>
34107 ext/vorbis/vorbisenc.c: Remove leaks from vorbisenc.
34108 Original commit message from CVS:
34109 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
34110 (gst_vorbis_enc_chain):
34111 Remove leaks from vorbisenc.
34112 Mostly minor changes, the only significant one is that now the
34113 buffers we set as 'streamheader' on the caps are copies of the
34114 original buffers, to avoid circular refcounting problems.
34116 2006-04-07 09:51:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
34118 gst/playback/gstplaybasebin.c: Don't remove our mute-probe if someone else already did so.
34119 Original commit message from CVS:
34120 * gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
34121 Don't remove our mute-probe if someone else already did so.
34122 Don't set a 2nd one if there is already one pending on the pad.
34123 * gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
34125 When a seek fails, ensure that playbin is still set back to playing.
34126 * gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
34127 (mpeg_ts_type_find), (plugin_init):
34128 Add a typefind function for mpeg-ts streams.
34130 2006-04-06 11:40:45 +0000 Andy Wingo <wingo@pobox.com>
34133 * gst/audiotestsrc/gstaudiotestsrc.c:
34134 * gst/videorate/gstvideorate.c:
34135 gst/videorate/gstvideorate.c (gst_video_rate_reset)
34136 Original commit message from CVS:
34137 2006-04-06 Andy Wingo <wingo@pobox.com>
34138 * gst/videorate/gstvideorate.c (gst_video_rate_reset)
34139 (gst_video_rate_init): Caps-related parameters should not be reset
34140 by a flush -- move their inits to the instance init function.
34141 (gst_video_rate_flush_prev): Don't complain if gst_pad_push
34142 is not OK, just return the result.
34143 * gst/audiotestsrc/gstaudiotestsrc.c
34144 (gst_audio_test_src_class_init)
34145 (gst_audio_test_src_get_times): Re-enable is-live=true, as was
34146 broken by Stefan's commit on 24 March.
34148 2006-04-06 10:50:14 +0000 Andy Wingo <wingo@pobox.com>
34150 ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on buffers being pushed out. Fixes oggmux ! multifdsink.
34151 Original commit message from CVS:
34152 2006-04-06 Andy Wingo <wingo@pobox.com>
34153 * ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
34154 buffers being pushed out. Fixes oggmux ! multifdsink.
34156 2006-04-05 13:05:25 +0000 Tim-Philipp Müller <tim@centricular.net>
34158 ext/vorbis/: Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make vorbisenc adhere to the official nomenclature; u...
34159 Original commit message from CVS:
34160 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
34161 (gst_vorbis_dec_init), (vorbis_dec_finalize):
34162 * ext/vorbis/vorbisdec.h:
34163 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_add_interfaces),
34164 (gst_vorbis_enc_base_init), (gst_vorbis_enc_class_init),
34165 (gst_vorbis_enc_sink_setcaps), (gst_vorbis_enc_convert_src),
34166 (gst_vorbis_enc_convert_sink), (gst_vorbis_enc_get_query_types),
34167 (gst_vorbis_enc_src_query), (gst_vorbis_enc_sink_query),
34168 (gst_vorbis_enc_init), (gst_vorbis_enc_get_tag_value),
34169 (gst_vorbis_enc_metadata_set1), (gst_vorbis_enc_set_metadata),
34170 (gst_vorbis_enc_setup), (gst_vorbis_enc_clear),
34171 (gst_vorbis_enc_buffer_from_packet),
34172 (gst_vorbis_enc_buffer_from_header_packet),
34173 (gst_vorbis_enc_push_buffer), (gst_vorbis_enc_push_packet),
34174 (gst_vorbis_enc_set_header_on_caps), (gst_vorbis_enc_sink_event),
34175 (gst_vorbis_enc_chain), (gst_vorbis_enc_output_buffers),
34176 (gst_vorbis_enc_get_property), (gst_vorbis_enc_set_property),
34177 (gst_vorbis_enc_change_state):
34178 * ext/vorbis/vorbisenc.h:
34179 Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make
34180 vorbisenc adhere to the official nomenclature; use boilerplate
34183 2006-04-04 11:20:24 +0000 Andy Wingo <wingo@pobox.com>
34185 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Whoops, fix bug introduced. Bad hacker!
34186 Original commit message from CVS:
34187 2006-04-04 Andy Wingo <wingo@pobox.com>
34188 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
34189 Whoops, fix bug introduced. Bad hacker!
34191 2006-04-04 11:15:00 +0000 Andy Wingo <wingo@pobox.com>
34193 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Properly handle the case where you get EOS before any buffe...
34194 Original commit message from CVS:
34195 2006-04-04 Andy Wingo <wingo@pobox.com>
34196 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
34197 Properly handle the case where you get EOS before any buffers are
34198 received. Use gst_buffer_make_metadata_writable where appropriate.
34200 2006-04-04 10:16:46 +0000 Andy Wingo <wingo@pobox.com>
34202 ext/theora/theoradec.c (theora_handle_data_packet): This value is often negative -- make it signed so as not to wrap ...
34203 Original commit message from CVS:
34204 2006-04-04 Andy Wingo <wingo@pobox.com>
34205 * ext/theora/theoradec.c (theora_handle_data_packet): This value
34206 is often negative -- make it signed so as not to wrap around.
34207 Fixes segfaults introduced on 9 March.
34209 2006-04-03 16:43:10 +0000 Wim Taymans <wim.taymans@gmail.com>
34211 ext/theora/: Don't try to store a gdouble in a gboolean.
34212 Original commit message from CVS:
34213 * ext/theora/gsttheoradec.h:
34214 * ext/theora/theoradec.c: (theora_dec_src_event):
34215 Don't try to store a gdouble in a gboolean.
34218 2006-04-03 12:55:18 +0000 Michael Smith <msmith@xiph.org>
34220 ext/ogg/gstoggmux.c: Oggmux sucks.
34221 Original commit message from CVS:
34222 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
34224 Make it suck slightly less by writing out the final page.
34225 Still can't encode a vorbis-in-ogg file correctly, though.
34227 2006-04-03 08:49:06 +0000 Andy Wingo <wingo@pobox.com>
34229 ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove a g_print.
34230 Original commit message from CVS:
34231 2006-04-03 Andy Wingo <wingo@pobox.com>
34232 * ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
34235 2006-04-03 08:32:21 +0000 Andy Wingo <wingo@pobox.com>
34237 ext/theora/theora.c (plugin_init): Register theoraparse.
34238 Original commit message from CVS:
34239 2006-04-03 Andy Wingo <wingo@pobox.com>
34240 * ext/theora/theora.c (plugin_init): Register theoraparse.
34241 * ext/theora/gsttheoraparse.h:
34242 * ext/theora/theoraparse.c: New files implementing a theora
34243 parser. Now we can properly remux ogg/theora+vorbis, yay.
34245 2006-04-03 08:28:58 +0000 Andy Wingo <wingo@pobox.com>
34247 ext/vorbis/vorbisparse.c: Add some docs and a copyright.
34248 Original commit message from CVS:
34249 2006-04-03 Andy Wingo <wingo@pobox.com>
34250 * ext/vorbis/vorbisparse.c: Add some docs and a copyright.
34252 2006-04-01 15:34:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34256 don't use AS_LIBTOOL_TAGS, it doesn't work
34257 Original commit message from CVS:
34258 don't use AS_LIBTOOL_TAGS, it doesn't work
34260 2006-04-01 11:41:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34263 * ext/pango/gsttextoverlay.c:
34264 * sys/v4l/gstv4lsrc.c:
34265 remove BT8x8 from description, works for more devices
34266 Original commit message from CVS:
34267 remove BT8x8 from description, works for more devices
34269 2006-04-01 11:21:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34271 gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798)
34272 Original commit message from CVS:
34273 * gst/audiotestsrc/gstaudiotestsrc.c:
34274 Fixed the sample pipeline (see #323798)
34276 2006-04-01 09:50:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34278 use AS_VERSION and AS_NANO more cleanups
34279 Original commit message from CVS:
34281 * win32/common/config.h:
34282 * win32/common/config.h.in:
34283 use AS_VERSION and AS_NANO
34286 2006-03-31 17:08:41 +0000 Andy Wingo <wingo@pobox.com>
34288 ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix uninitialized variable return that would happen.
34289 Original commit message from CVS:
34290 2006-03-31 Andy Wingo <wingo@pobox.com>
34291 * ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
34292 uninitialized variable return that would happen.
34294 2006-03-31 16:57:47 +0000 Andy Wingo <wingo@pobox.com>
34296 ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix uninitialized variable return that would never happen.
34297 Original commit message from CVS:
34298 2006-03-31 Andy Wingo <wingo@pobox.com>
34299 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
34300 uninitialized variable return that would never happen.
34302 2006-03-31 16:43:43 +0000 Andy Wingo <wingo@pobox.com>
34304 ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
34305 Original commit message from CVS:
34306 2006-03-31 Andy Wingo <wingo@pobox.com>
34307 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
34308 (vorbis_parse_sink_event): Add an event function to flush our
34309 state on a seek, and to drain buffers on a premature EOS.
34310 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
34311 (vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
34312 (vorbis_parse_chain, vorbis_parse_queue_buffer)
34313 (vorbis_parse_drain_queue): Queue up buffers until we can set
34314 their timestamps and granulepos values.
34315 * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
34316 and keep track of data needed for deriving granulepos and
34317 timestamps for buffers.
34319 2006-03-30 11:05:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34321 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
34322 * pkgconfig/gstreamer-plugins-base.pc.in:
34323 expose pluginsdir so gonlin can use it for tests
34324 Original commit message from CVS:
34325 expose pluginsdir so gonlin can use it for tests
34327 2006-03-30 10:03:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34329 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
34330 * pkgconfig/gstreamer-plugins-base.pc.in:
34331 add ccda to libraries
34332 Original commit message from CVS:
34333 add ccda to libraries
34335 2006-03-29 14:00:08 +0000 j^ <j@bootlab.org>
34337 better/unified long descriptions
34338 Original commit message from CVS:
34339 Patch by: j^ <j at bootlab dot org>
34340 * ext/alsa/gstalsamixerelement.c:
34341 (gst_alsa_mixer_element_class_init):
34342 * ext/alsa/gstalsasink.c:
34343 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
34344 * ext/ogg/gstoggdemux.c:
34345 * ext/ogg/gstoggmux.c:
34346 * ext/ogg/gstoggparse.c:
34347 * ext/pango/gstclockoverlay.c:
34348 * ext/pango/gsttextoverlay.c:
34349 * ext/pango/gsttextrender.c:
34350 * ext/pango/gsttimeoverlay.c:
34351 * ext/theora/theoradec.c:
34352 * ext/theora/theoraenc.c:
34353 * ext/vorbis/vorbisdec.c:
34354 * ext/vorbis/vorbisenc.c:
34355 * gst/audioconvert/gstaudioconvert.c:
34356 * gst/subparse/gstsubparse.c:
34357 * gst/tcp/gstmultifdsink.c:
34358 * gst/tcp/gsttcpclientsink.c:
34359 * gst/tcp/gsttcpclientsrc.c:
34360 * gst/tcp/gsttcpserversink.c:
34361 * gst/tcp/gsttcpserversrc.c:
34362 better/unified long descriptions
34365 2006-03-29 13:54:24 +0000 Wim Taymans <wim.taymans@gmail.com>
34367 tests/examples/seek/seek.c: Don't let double and tripple clicks mess up our state.
34368 Original commit message from CVS:
34369 * tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
34371 Don't let double and tripple clicks mess up our state.
34373 2006-03-28 13:13:43 +0000 Tim-Philipp Müller <tim@centricular.net>
34375 gst/playback/gstplaybin.c: Error out gracefully when we can't create any of the usual conversion elements for some re...
34376 Original commit message from CVS:
34377 * gst/playback/gstplaybin.c: (gen_video_element),
34378 (gen_text_element), (gen_audio_element), (gen_vis_element):
34379 Error out gracefully when we can't create any of the usual
34380 conversion elements for some reason. Also, don't try to
34381 create an audioscale (sic) element that's not used anyway.
34383 2006-03-28 10:21:52 +0000 Tim-Philipp Müller <tim@centricular.net>
34385 gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
34386 Original commit message from CVS:
34387 * gst/playback/gstplaybasebin.c: (setup_source):
34388 Don't post RESOURCE_NOT_FOUND error when we can't find a source
34389 element for a particular protocol, that's confusing for users.
34390 Instead, post a RESOURCE_FAILED error, so that our own error
34391 message is actually shown in totem etc. (#336303).
34393 2006-03-27 16:36:46 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
34395 ext/gnomevfs/gstgnomevfssrc.c: Fix some minor memory leaks (#336194).
34396 Original commit message from CVS:
34397 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
34398 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
34399 (gst_gnome_vfs_src_get_icy_metadata):
34400 Fix some minor memory leaks (#336194).
34402 2006-03-27 16:15:00 +0000 Tim-Philipp Müller <tim@centricular.net>
34404 ext/gnomevfs/: Make gnomevfssink accept filenames as well as URIs for the "location" property, just like gnomevfssrc ...
34405 Original commit message from CVS:
34406 * ext/gnomevfs/gstgnomevfs.c:
34407 (gst_gnome_vfs_location_to_uri_string):
34408 * ext/gnomevfs/gstgnomevfs.h:
34409 * ext/gnomevfs/gstgnomevfssink.c:
34410 (gst_gnome_vfs_sink_set_property):
34411 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
34412 Make gnomevfssink accept filenames as well as URIs for the
34413 "location" property, just like gnomevfssrc does (and
34414 filesrc/filesink do) (#336190).
34416 2006-03-24 20:35:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34418 tests/check/generic/clock-selection.c: set to NULL before unreffing, fixes a valgrind leak.
34419 Original commit message from CVS:
34420 * tests/check/generic/clock-selection.c: (GST_START_TEST):
34421 set to NULL before unreffing, fixes a valgrind leak.
34422 Why was this not triggering the error that an object needs to
34423 be NULL before unreffing ?
34424 * win32/common/config.h:
34427 2006-03-24 17:57:39 +0000 Tim-Philipp Müller <tim@centricular.net>
34429 gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
34430 Original commit message from CVS:
34431 * gst/subparse/gstsubparse.c: (convert_encoding),
34432 (gst_sub_parse_change_state):
34433 * gst/subparse/gstsubparse.h:
34434 Text subtitle files may or may not be UTF-8. If it's not, we
34435 don't really want to see '?' characters in place of non-ASCII
34436 characters like accented characters. So let's assume the input
34437 is UTF-8 until we come across text that is clearly not. If it's
34438 not UTF-8, we don't really know what it is, so try the following:
34439 (a) see whether the GST_SUBTITLE_ENCODING environment variable
34440 is set; if not, check (b) if the current locale encoding is
34441 non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
34442 the current locale encoding is UTF-8 and the environment variable
34443 was not set to any particular encoding. Not perfect, but better
34444 than nothing (and better than before, I think) (fixes #172848).
34446 2006-03-24 17:39:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34448 * docs/plugins/tmpl/.gitignore:
34449 * tests/check/libs/.gitignore:
34450 * tests/check/pipelines/.gitignore:
34451 * tests/examples/volume/.gitignore:
34453 Original commit message from CVS:
34456 2006-03-24 17:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34458 configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
34459 Original commit message from CVS:
34460 2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
34462 update core requirement to 0.10.4.1 because of async_playback
34463 vmethod on GstBaseSink
34465 2006-03-24 17:11:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
34467 use DEBUG_FUNCPTR for collectpads
34468 Original commit message from CVS:
34469 * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
34470 * gst/adder/gstadder.c: (gst_adder_init):
34471 use DEBUG_FUNCPTR for collectpads
34473 2006-03-24 14:11:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34476 don't go through check-torture if no check installed
34477 Original commit message from CVS:
34478 don't go through check-torture if no check installed
34480 2006-03-24 10:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
34482 Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
34483 Original commit message from CVS:
34484 * docs/plugins/Makefile.am:
34485 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
34486 * docs/plugins/gst-plugins-base-plugins-sections.txt:
34487 * ext/cdparanoia/gstcdparanoiasrc.c:
34488 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
34489 (gst_gnome_vfs_sink_class_init):
34490 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
34491 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
34492 * ext/ogg/gstoggmux.c:
34493 * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
34494 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
34495 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
34496 * ext/pango/gsttextoverlay.c:
34497 * ext/pango/gsttextrender.c:
34498 * ext/theora/theoradec.c:
34499 * ext/theora/theoraenc.c:
34500 * ext/vorbis/vorbisdec.c:
34501 * ext/vorbis/vorbisenc.c:
34502 * gst-libs/gst/audio/gstaudiofilter.c:
34503 (gst_audio_filter_base_init):
34504 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
34505 (gst_audio_filter_template_base_init):
34506 * gst/adder/gstadder.c: (gst_adder_get_type):
34507 * gst/adder/gstadder.h:
34508 * gst/audioconvert/gstaudioconvert.c:
34509 * gst/audiotestsrc/gstaudiotestsrc.c:
34510 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
34511 (gst_audio_test_src_create):
34512 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
34513 * gst/playback/gstdecodebin.c:
34514 * gst/playback/gstplaybin.c:
34515 * gst/playback/gststreamselector.c:
34516 (gst_stream_selector_base_init):
34517 * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
34518 * gst/volume/gstvolume.c:
34519 * sys/v4l/gstv4lmjpegsink.c:
34520 * sys/v4l/gstv4lmjpegsrc.c:
34521 * tests/check/libs/cddabasesrc.c:
34522 * tests/old/examples/gob/gst-identity2.gob:
34523 Add docs for adder, use GST_ELEMENT_DETAILS macro,
34524 define GstElementDetails at the top
34526 2006-03-23 21:48:18 +0000 Sébastien Moutte <sebastien@moutte.net>
34528 win32/common/libgstinterfaces.def: Add a lot of export functions for gst-python
34529 Original commit message from CVS:
34530 * win32/common/libgstinterfaces.def:
34531 Add a lot of export functions for gst-python
34532 * win32/common/libgstinterfaces.dsp:
34533 Add a missing include folder in the project configuration
34535 2006-03-23 16:58:03 +0000 Wim Taymans <wim.taymans@gmail.com>
34537 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
34538 Original commit message from CVS:
34539 * gst-libs/gst/audio/gstbaseaudiosrc.c:
34540 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
34541 (gst_base_audio_src_change_state):
34542 Fix audio sources, forgot to make the ringbuffer
34545 2006-03-23 16:29:58 +0000 Wim Taymans <wim.taymans@gmail.com>
34547 gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
34548 Original commit message from CVS:
34549 * gst-libs/gst/audio/gstbaseaudiosrc.c:
34550 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
34551 (gst_base_audio_src_change_state):
34552 unparent instead of unref the ringbuffer.
34554 2006-03-23 16:24:23 +0000 Wim Taymans <wim.taymans@gmail.com>
34556 gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
34557 Original commit message from CVS:
34558 * gst-libs/gst/audio/gstbaseaudiosink.c:
34559 (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
34560 (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
34561 Implement new async_play vmethod to start slaving and allow
34562 playback start in case of async PLAY state changes.
34563 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
34564 Enable QoS with new method in base class.
34566 2006-03-23 11:18:19 +0000 Julien MOUTTE <julien@moutte.net>
34568 gst/videotestsrc/gstvideotestsrc.c: Partially handle 0 framerate, only EOS after the first frame is missing.
34569 Original commit message from CVS:
34570 Patch by: Julien MOUTTE <julien at moutte dot net>
34571 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
34572 (gst_video_test_src_do_seek), (gst_video_test_src_create):
34573 Partially handle 0 framerate, only EOS after the first frame
34576 2006-03-23 09:38:59 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
34578 gst/: Patch for support of YVU9 AVI files (#334822)
34579 Original commit message from CVS:
34580 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
34581 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
34582 (gst_riff_create_video_template_caps):
34583 * gst/ffmpegcolorspace/avcodec.h:
34584 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
34585 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
34586 (gst_ffmpegcsp_avpicture_fill):
34587 * gst/ffmpegcolorspace/imgconvert.c:
34588 Patch for support of YVU9 AVI files (#334822)
34590 2006-03-22 15:29:25 +0000 Edward Hervey <bilboed@bilboed.com>
34592 docs/design/design-decodebin.txt: Added design document for new decodebin text/x-pango-markup is also a default targe...
34593 Original commit message from CVS:
34594 * docs/design/design-decodebin.txt:
34595 Added design document for new decodebin
34596 (Target Caps): text/x-pango-markup is also a default target caps.
34598 2006-03-22 15:11:47 +0000 Edward Hervey <bilboed@bilboed.com>
34600 docs/design/design-decodebin.txt: Added design document for new decodebin
34601 Original commit message from CVS:
34602 * docs/design/design-decodebin.txt:
34603 Added design document for new decodebin
34605 2006-03-22 12:33:09 +0000 Wim Taymans <wim.taymans@gmail.com>
34607 gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
34608 Original commit message from CVS:
34609 * gst-libs/gst/audio/gstbaseaudiosink.c:
34610 (gst_base_audio_sink_dispose):
34611 Since we _parent the ringbuffer, we also need to
34612 _unparent instead of a plain _unref.
34614 2006-03-22 12:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
34616 tests/examples/seek/seek.c: Add scrub checkbox.
34617 Original commit message from CVS:
34618 * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
34619 (stop_seek), (scrub_toggle_cb), (main):
34620 Add scrub checkbox.
34622 2006-03-21 17:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
34624 ext/ogg/gstoggparse.c: Fix very inefficient usage of linked lists (#335365).
34625 Original commit message from CVS:
34626 * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
34627 (gst_ogg_parse_chain):
34628 Fix very inefficient usage of linked lists (#335365).
34630 2006-03-21 14:26:01 +0000 Edward Hervey <bilboed@bilboed.com>
34632 gcc 4.1 unreferenced pointer fixes.
34633 Original commit message from CVS:
34634 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
34635 * gst/playback/gstplaybin.c: (handoff):
34636 * gst/playback/gststreamselector.c:
34637 (gst_stream_selector_set_property):
34638 gcc 4.1 unreferenced pointer fixes.
34639 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
34640 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
34641 gst_buffer_ref() now takes a GstBuffer*.
34643 2006-03-20 18:09:41 +0000 Julien Moutte <julien@moutte.net>
34645 sys/xvimage/xvimagesink.c: Fix a memleak reported by Jan Schmidt.
34646 Original commit message from CVS:
34647 2006-03-20 Julien MOUTTE <julien@moutte.net>
34648 * sys/xvimage/xvimagesink.c:
34649 (gst_xvimagesink_get_format_from_caps): Fix a memleak reported
34652 2006-03-19 11:37:46 +0000 Tim-Philipp Müller <tim@centricular.net>
34654 gst/typefind/gsttypefindfunctions.c: Can't do tag preferences via probability, as tags would then lose against types ...
34655 Original commit message from CVS:
34656 * gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
34657 (id3v1_type_find), (apetag_type_find), (plugin_init):
34658 Can't do tag preferences via probability, as tags would then
34659 lose against types that are recognised with MAXIMUM probability
34660 (like .wav); so let all tag typefinders return MAXIMUM themselves
34661 and order them via the rank. Split ID3v1 and ID3v2 typefinders so
34662 that we can prefer APE to ID3v1 (fixes #335028).
34664 2006-03-17 17:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
34666 gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
34667 Original commit message from CVS:
34668 * gst-libs/gst/audio/gstbaseaudiosink.c:
34669 (gst_base_audio_sink_change_state):
34670 * gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
34671 (gst_ring_buffer_may_start):
34672 * gst-libs/gst/audio/gstringbuffer.h:
34673 Only start playback if we are playing.
34674 should fix #330748.
34676 2006-03-17 13:11:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
34678 Revert accidental commits to these files.
34679 Original commit message from CVS:
34680 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
34681 * win32/common/config.h:
34682 Revert accidental commits to these files.
34684 2006-03-16 20:01:03 +0000 Michal Benes <michal.benes@xeris.cz>
34686 tests/Makefile.am: Don't try to build tests in tests/icles if we don't have X (#323852)
34687 Original commit message from CVS:
34688 Patch by: Michal Benes <michal dot benes at xeris dot cz>
34689 * tests/Makefile.am:
34690 Don't try to build tests in tests/icles if we
34691 don't have X (#323852)
34693 2006-03-16 13:08:01 +0000 Tim-Philipp Müller <tim@centricular.net>
34695 gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
34696 Original commit message from CVS:
34697 * gst-libs/gst/tag/gstid3tag.c:
34698 Add TXXX frame identifiers for replaygain stuff as used
34699 by some taggers (see #323721).
34701 2006-03-16 10:22:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
34703 gst/playback/gststreamselector.c: Preserve the existing buggy streamselector behaviour by performing a fallback buffe...
34704 Original commit message from CVS:
34705 * gst/playback/gststreamselector.c:
34706 (gst_stream_selector_set_property),
34707 (gst_stream_selector_bufferalloc):
34708 Preserve the existing buggy streamselector behaviour by performing
34709 a fallback buffer allocation when downstream isn't linked yet.
34710 This should really be fixed in playbin by blocking pads until it's
34712 Also, use gst_pad_alloc_buffer instead of
34713 gst_pad_alloc_buffer_and_set.
34715 2006-03-15 22:40:08 +0000 Tim-Philipp Müller <tim@centricular.net>
34717 gst-libs/gst/tag/gstid3tag.c: Don't crash on unknown ID3v2 TXXX frames.
34718 Original commit message from CVS:
34719 * gst-libs/gst/tag/gstid3tag.c:
34720 Don't crash on unknown ID3v2 TXXX frames.
34722 2006-03-15 17:59:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
34724 ext/alsa/gstalsasink.c: Chain up to the parent finalize method.
34725 Original commit message from CVS:
34726 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
34727 Chain up to the parent finalize method.
34728 Add 32-bit sample size to the template caps.
34729 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
34730 (gst_riff_create_video_template_caps):
34731 Add the fourcc that the VMWare codec uses.
34732 * gst/playback/gststreamselector.c:
34733 (gst_stream_selector_set_property),
34734 (gst_stream_selector_bufferalloc),
34735 (gst_stream_selector_request_new_pad):
34736 For the active pad, forward buffer-alloc requests, otherwise
34737 return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
34738 having to memcpy every frame when used by playbin.
34739 * gst/tcp/gstmultifdsink.c:
34740 (gst_multi_fd_sink_handle_client_write):
34741 Get negotiated caps from the sink pad, rather than the sink
34744 2006-03-15 17:11:34 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
34746 ext/gnomevfs/gstgnomevfssrc.c: Don't forget to set src->callbacks_pushed to FALSE again when popping them, otherwise ...
34747 Original commit message from CVS:
34748 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
34749 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_pop_callbacks):
34750 Don't forget to set src->callbacks_pushed to FALSE again when
34751 popping them, otherwise re-activation in a different mode won't
34754 2006-03-15 11:30:29 +0000 Sebastien Moutte <sebastien@moutte.net>
34756 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
34757 Original commit message from CVS:
34758 Patch by: Sebastien Moutte <sebastien moutte net>
34759 * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
34760 (gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
34761 (gst_ffmpeg_smpfmt_to_caps):
34762 Replace __VA_ARGS__ caps creation macros with varargs functions.
34763 Makes things compile on MSVC (#320765), looks nicer, and we can
34764 tell the compiler to check for the NULL terminator.
34766 2006-03-14 15:13:04 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
34768 gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
34769 Original commit message from CVS:
34770 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
34771 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
34772 Make sure the buffer we copy into is really always big
34773 enough, this time for real (#333488).
34775 2006-03-14 13:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
34777 gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
34778 Original commit message from CVS:
34779 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
34780 Add support for 24bpp DIB (#305279).
34782 2006-03-14 11:11:59 +0000 Wim Taymans <wim.taymans@gmail.com>
34784 gst/: Re-enable QoS after the release.
34785 Original commit message from CVS:
34786 * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
34787 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
34788 * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
34789 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
34790 (gst_video_scale_init), (gst_video_scale_src_event):
34791 Re-enable QoS after the release.
34792 Rework videoscale to use the base class src_event handler.
34794 2006-03-14 09:51:01 +0000 Tim-Philipp Müller <tim@centricular.net>
34796 configure.ac: back to CVS.
34797 Original commit message from CVS:
34801 === release 0.10.5 ===
34803 2006-03-13 19:50:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34809 * docs/plugins/inspect/plugin-adder.xml:
34810 * docs/plugins/inspect/plugin-alsa.xml:
34811 * docs/plugins/inspect/plugin-audioconvert.xml:
34812 * docs/plugins/inspect/plugin-audiorate.xml:
34813 * docs/plugins/inspect/plugin-audioresample.xml:
34814 * docs/plugins/inspect/plugin-audiotestsrc.xml:
34815 * docs/plugins/inspect/plugin-cdparanoia.xml:
34816 * docs/plugins/inspect/plugin-decodebin.xml:
34817 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
34818 * docs/plugins/inspect/plugin-gnomevfs.xml:
34819 * docs/plugins/inspect/plugin-libvisual.xml:
34820 * docs/plugins/inspect/plugin-ogg.xml:
34821 * docs/plugins/inspect/plugin-pango.xml:
34822 * docs/plugins/inspect/plugin-playbin.xml:
34823 * docs/plugins/inspect/plugin-subparse.xml:
34824 * docs/plugins/inspect/plugin-tcp.xml:
34825 * docs/plugins/inspect/plugin-theora.xml:
34826 * docs/plugins/inspect/plugin-typefindfunctions.xml:
34827 * docs/plugins/inspect/plugin-video4linux.xml:
34828 * docs/plugins/inspect/plugin-videorate.xml:
34829 * docs/plugins/inspect/plugin-videoscale.xml:
34830 * docs/plugins/inspect/plugin-videotestsrc.xml:
34831 * docs/plugins/inspect/plugin-volume.xml:
34832 * docs/plugins/inspect/plugin-vorbis.xml:
34833 * docs/plugins/inspect/plugin-ximagesink.xml:
34834 * docs/plugins/inspect/plugin-xvimagesink.xml:
34835 * win32/common/config.h:
34837 Original commit message from CVS:
34840 2006-03-13 17:28:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34857 Original commit message from CVS:
34860 2006-03-13 11:17:19 +0000 Tim-Philipp Müller <tim@centricular.net>
34862 docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
34863 Original commit message from CVS:
34864 * docs/plugins/Makefile.am:
34865 Part of previous cdparanoiasrc docs fixes, forgot to commit.
34867 2006-03-12 14:56:31 +0000 Tim-Philipp Müller <tim@centricular.net>
34869 docs/plugins/: Add cdparanoiasrc to docs.
34870 Original commit message from CVS:
34871 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
34872 * docs/plugins/gst-plugins-base-plugins-sections.txt:
34873 * docs/plugins/gst-plugins-base-plugins.hierarchy:
34874 Add cdparanoiasrc to docs.
34875 * gst-libs/gst/cdda/gstcddabasesrc.c:
34876 More GstCddaBaseSrc docs.
34878 2006-03-12 13:47:22 +0000 Tim-Philipp Müller <tim@centricular.net>
34880 Add new API to libgsttag: gst_tag_from_id3_user_tag().
34881 Original commit message from CVS:
34882 * docs/libs/gst-plugins-base-libs-sections.txt:
34883 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
34884 * gst-libs/gst/tag/tag.h:
34885 Add new API to libgsttag: gst_tag_from_id3_user_tag().
34887 2006-03-11 19:47:16 +0000 Tim-Philipp Müller <tim@centricular.net>
34889 gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
34890 Original commit message from CVS:
34891 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
34892 NULL-terminate array of mpeg4 video file extensions.
34893 Fixes crash on PPC (#334226).
34895 2006-03-11 16:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
34897 ext/gnomevfs/gstgnomevfssrc.c: gnome_vfs_uri_is_local() alone is not a good indicator whether we can operate in pull-...
34898 Original commit message from CVS:
34899 * ext/gnomevfs/gstgnomevfssrc.c:
34900 (gst_gnome_vfs_src_check_get_range):
34901 gnome_vfs_uri_is_local() alone is not a good indicator
34902 whether we can operate in pull-mode with a specific URI,
34903 as it returns FALSE for file:// URIs that point to an
34904 NFS-mounted path. Be more conservative here: whitelist
34905 local files, blacklist http URIs and use the old
34906 mechanism for anything else (fixes #334216).
34908 2006-03-10 19:15:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34910 configure.ac: back to trunk
34911 Original commit message from CVS:
34915 === release 0.10.4 ===
34917 2006-03-10 19:05:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34923 * docs/plugins/gst-plugins-base-plugins.args:
34924 * docs/plugins/inspect/plugin-adder.xml:
34925 * docs/plugins/inspect/plugin-alsa.xml:
34926 * docs/plugins/inspect/plugin-audioconvert.xml:
34927 * docs/plugins/inspect/plugin-audiorate.xml:
34928 * docs/plugins/inspect/plugin-audioresample.xml:
34929 * docs/plugins/inspect/plugin-audiotestsrc.xml:
34930 * docs/plugins/inspect/plugin-cdparanoia.xml:
34931 * docs/plugins/inspect/plugin-decodebin.xml:
34932 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
34933 * docs/plugins/inspect/plugin-gnomevfs.xml:
34934 * docs/plugins/inspect/plugin-libvisual.xml:
34935 * docs/plugins/inspect/plugin-ogg.xml:
34936 * docs/plugins/inspect/plugin-pango.xml:
34937 * docs/plugins/inspect/plugin-playbin.xml:
34938 * docs/plugins/inspect/plugin-subparse.xml:
34939 * docs/plugins/inspect/plugin-tcp.xml:
34940 * docs/plugins/inspect/plugin-theora.xml:
34941 * docs/plugins/inspect/plugin-typefindfunctions.xml:
34942 * docs/plugins/inspect/plugin-video4linux.xml:
34943 * docs/plugins/inspect/plugin-videorate.xml:
34944 * docs/plugins/inspect/plugin-videoscale.xml:
34945 * docs/plugins/inspect/plugin-videotestsrc.xml:
34946 * docs/plugins/inspect/plugin-volume.xml:
34947 * docs/plugins/inspect/plugin-vorbis.xml:
34948 * docs/plugins/inspect/plugin-ximagesink.xml:
34949 * docs/plugins/inspect/plugin-xvimagesink.xml:
34951 * win32/common/config.h:
34953 Original commit message from CVS:
34956 2006-03-10 12:37:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
34958 gst-libs/gst/video/gstvideosink.c: Disable max-lateness by setting it to -1 for now, so that we can bed QoS stuff in ...
34959 Original commit message from CVS:
34960 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
34961 Disable max-lateness by setting it to -1 for now, so that
34962 we can bed QoS stuff in thoroughly between now and the next
34965 2006-03-10 11:09:23 +0000 Fabrizio <fabrizio.ge@tiscali.it>
34967 gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
34968 Original commit message from CVS:
34969 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
34970 Make sure we don't read beyond the palette buffer in case of
34971 broken or manipulated files (#333488, patch by: Fabrizio
34974 2006-03-10 10:44:02 +0000 Edward Hervey <bilboed@bilboed.com>
34976 gst/typefind/gsttypefindfunctions.c: Fix for variable not initialized.
34977 Original commit message from CVS:
34978 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
34979 Fix for variable not initialized.
34981 2006-03-09 19:02:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34984 * docs/libs/tmpl/gstringbuffer.sgml:
34999 * win32/common/config.h:
35001 Original commit message from CVS:
35004 2006-03-09 17:58:00 +0000 Wim Taymans <wim.taymans@gmail.com>
35006 ext/libvisual/visual.c: Small cleanups.
35007 Original commit message from CVS:
35008 * ext/libvisual/visual.c: (gst_visual_get_type),
35009 (gst_visual_src_setcaps), (gst_vis_src_negotiate),
35010 (gst_visual_chain):
35012 * ext/theora/gsttheoradec.h:
35013 * ext/theora/theoradec.c: (gst_theora_dec_init),
35014 (gst_theora_dec_reset), (_theora_granule_time),
35015 (theora_dec_src_convert), (theora_dec_sink_convert),
35016 (theora_dec_src_query), (theora_dec_src_event),
35017 (theora_dec_sink_event), (theora_handle_comment_packet),
35018 (theora_handle_header_packet), (theora_dec_push),
35019 (theora_handle_data_packet), (theora_dec_chain),
35020 (theora_dec_change_state):
35023 2006-03-09 17:50:59 +0000 Wim Taymans <wim.taymans@gmail.com>
35025 ext/gnomevfs/gstgnomevfssrc.c: Some cleanups.
35026 Original commit message from CVS:
35027 * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
35028 (audiocast_register_listener), (gst_gnome_vfs_src_start):
35031 2006-03-09 17:45:39 +0000 Wim Taymans <wim.taymans@gmail.com>
35033 ext/ogg/gstoggdemux.c: Don't try to activate NULL chains.
35034 Original commit message from CVS:
35035 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
35036 Don't try to activate NULL chains.
35038 2006-03-09 16:30:41 +0000 Tim-Philipp Müller <tim@centricular.net>
35040 gst/typefind/gsttypefindfunctions.c: Fix invalid memory access to region before peek'd data (#332964).
35041 Original commit message from CVS:
35042 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
35043 Fix invalid memory access to region before peek'd data (#332964).
35045 2006-03-09 15:05:03 +0000 Christophe Fergeau <teuf@gnome.org>
35048 Original commit message from CVS:
35049 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init):
35050 * ext/pango/gsttextrender.c: (gst_text_render_init):
35051 * gst/adder/gstadder.c: (gst_adder_init):
35052 Don't leak padtemplates, patch by Christophe Fergeau,
35055 2006-03-09 12:56:35 +0000 Tim-Philipp Müller <tim@centricular.net>
35057 gst/subparse/gstsubparse.c: Fix invalid memory access: make sure string passed to regexec() is NUL-termianted.
35058 Original commit message from CVS:
35059 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
35060 Fix invalid memory access: make sure string passed to
35061 regexec() is NUL-termianted.
35063 2006-03-09 12:37:59 +0000 Tim-Philipp Müller <tim@centricular.net>
35065 gst/typefind/gsttypefindfunctions.c: Refactor mpeg/audio typefinding to make it more maintainable and easier to fine-...
35066 Original commit message from CVS:
35067 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
35069 Refactor mpeg/audio typefinding to make it more maintainable
35070 and easier to fine-tune. Make probing into middle of the file
35071 work properly (fixes #333900, also see #152688).
35073 2006-03-09 11:10:03 +0000 Tim-Philipp Müller <tim@centricular.net>
35075 gst/typefind/gsttypefindfunctions.c: Remove part from previous commit that was bogus: g_utf8_validate() does in fact ...
35076 Original commit message from CVS:
35077 * gst/typefind/gsttypefindfunctions.c:
35078 (utf8_type_find_have_valid_utf8_at_offset):
35079 Remove part from previous commit that was bogus:
35080 g_utf8_validate() does in fact not accept embedded
35081 zeroes, so we don't need to check for those (thanks
35082 to Mike for the hint).
35084 2006-03-08 17:11:29 +0000 Tim-Philipp Müller <tim@centricular.net>
35086 gst/typefind/gsttypefindfunctions.c: Make plain/text typefinder more conservative: firstly, check for embedded zeroes...
35087 Original commit message from CVS:
35088 * gst/typefind/gsttypefindfunctions.c:
35089 (utf8_type_find_count_embedded_zeroes),
35090 (utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
35091 Make plain/text typefinder more conservative: firstly, check
35092 for embedded zeroes, which are perfectly valid UTF-8 characters,
35093 but also a fairly good sign that something is not a plain text
35094 file; secondly, probe into the middle of the file if possible.
35095 If we can't probe into the middle, limit the probability value
35096 to be returned to TYPE_FIND_POSSIBLE (see #333900).
35098 2006-03-08 11:34:45 +0000 Michael Smith <msmith@xiph.org>
35100 gst/typefind/gsttypefindfunctions.c: Make typefind function name for mpeg4 video unique.
35101 Original commit message from CVS:
35102 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
35103 Make typefind function name for mpeg4 video unique.
35105 2006-03-08 09:53:31 +0000 Wim Taymans <wim.taymans@gmail.com>
35107 ext/libvisual/visual.c: Cleanups, post nice errors.
35108 Original commit message from CVS:
35109 * ext/libvisual/visual.c: (gst_visual_init),
35110 (gst_visual_clear_actors), (gst_visual_dispose),
35111 (gst_visual_reset), (gst_visual_src_setcaps),
35112 (gst_visual_sink_setcaps), (gst_vis_src_negotiate),
35113 (gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
35114 (gst_visual_chain), (gst_visual_change_state):
35115 Cleanups, post nice errors.
35116 Handle sink and src events.
35117 Implement simple QoS.
35118 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
35119 Use new basesink methods to configure max-lateness.
35121 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
35122 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
35123 Debug statement cleanups.
35124 * gst/volume/gstvolume.c: (gst_volume_class_init):
35127 2006-03-08 09:50:23 +0000 Tim-Philipp Müller <tim@centricular.net>
35129 ext/pango/gsttextoverlay.c: Revert API/ABI break from March 1. Keep 'halign' and 'valign' as string type properties, ...
35130 Original commit message from CVS:
35131 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
35132 (gst_text_overlay_init), (gst_text_overlay_set_property),
35133 (gst_text_overlay_get_property):
35134 Revert API/ABI break from March 1. Keep 'halign' and 'valign'
35135 as string type properties, but mark them deprecated. Add
35136 'halignment' and 'valignment' properties that use enums
35137 instead of strings.
35139 2006-03-08 09:37:12 +0000 Fabrizio <fabrizio.ge@tiscali.it>
35141 gst-libs/gst/riff/riff-media.c: Allow palettes with less than 256 colours in AVI files
35142 Original commit message from CVS:
35143 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
35144 Allow palettes with less than 256 colours in AVI files
35145 (#333488, patch by: Fabrizio Gennari).
35147 2006-03-07 21:56:09 +0000 Julien Moutte <julien@moutte.net>
35149 ext/pango/gsttextoverlay.c: Fix wrong EOS handling on text pad. We were releasing the queued text buffer when we shou...
35150 Original commit message from CVS:
35151 2006-03-07 Julien MOUTTE <julien@moutte.net>
35152 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
35153 (gst_text_overlay_video_event): Fix wrong EOS handling on text
35154 pad. We were releasing the queued text buffer when we should keep
35155 it until video pad gets EOS or discard the text buffer because it's
35156 too old. That was eating the last subtitle buffer. Add some more
35159 2006-03-07 17:28:36 +0000 Tim-Philipp Müller <tim@centricular.net>
35161 ext/pango/gsttextoverlay.c: Fix invalid memory access (we can't access a buffer after it's been pushed downstream wit...
35162 Original commit message from CVS:
35163 * ext/pango/gsttextoverlay.c: (gst_text_overlay_render_text),
35164 (gst_text_overlay_video_chain):
35165 Fix invalid memory access (we can't access a buffer after it's been
35166 pushed downstream without taking a reference); fix memory leak (if
35167 there's no text to render, bail out before allocating stuff).
35169 2006-03-07 15:08:15 +0000 Tim-Philipp Müller <tim@centricular.net>
35171 ext/pango/gsttextoverlay.*: If input is plain text, escape it before passing it to pango_layout_set_markup().
35172 Original commit message from CVS:
35173 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
35174 (gst_text_overlay_setcaps_txt), (gst_text_overlay_video_chain):
35175 * ext/pango/gsttextoverlay.h:
35176 If input is plain text, escape it before passing it to
35177 pango_layout_set_markup().
35179 2006-03-07 13:01:21 +0000 Tim-Philipp Müller <tim@centricular.net>
35181 gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
35182 Original commit message from CVS:
35183 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
35184 Don't ignore flow return from gst_pad_push().
35186 2006-03-07 12:49:03 +0000 Christophe Fergeau <teuf@gnome.org>
35188 Don't leak references returned by gst_pad_get_parent()
35189 Original commit message from CVS:
35190 * ext/libvisual/visual.c: (gst_visual_getcaps),
35191 (gst_visual_src_setcaps), (gst_visual_sink_setcaps):
35192 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
35193 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
35194 (gst_vorbisenc_convert_sink):
35195 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
35196 (gst_audio_duration_from_pad_buffer):
35197 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
35198 (gst_audio_filter_chain):
35199 * gst-libs/gst/rtp/gstbasertpdepayload.c:
35200 (gst_base_rtp_depayload_setcaps):
35201 * gst-libs/gst/video/video.c: (gst_video_frame_rate),
35202 (gst_video_get_size):
35203 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
35204 Don't leak references returned by gst_pad_get_parent()
35205 (#333663, based on patch by: Christophe Fergeau).
35207 2006-03-06 20:52:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
35209 ext/gnomevfs/gstgnomevfssink.c: change location param details
35210 Original commit message from CVS:
35211 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
35212 change location param details
35213 * gst/volume/gstvolume.c: (plugin_init):
35214 correct plugin description
35216 2006-03-06 20:07:55 +0000 Tim-Philipp Müller <tim@centricular.net>
35218 ext/gnomevfs/gstgnomevfssrc.c: Override GstBaseSrc::check_get_range() in order to avoid opening the resource just to ...
35219 Original commit message from CVS:
35220 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
35221 (gst_gnome_vfs_src_check_get_range):
35222 Override GstBaseSrc::check_get_range() in order to avoid opening
35223 the resource just to check whether we can operate in pull-mode or
35224 not - we can predict that pretty well from the URI alone. Should
35225 fix problems with last.fm (#331690). (Requires latest core CVS).
35227 2006-03-06 16:18:51 +0000 Wim Taymans <wim.taymans@gmail.com>
35229 gst-libs/gst/video/gstvideosink.c: Throw away frames that are later than 20 ms.
35230 Original commit message from CVS:
35231 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
35232 (gst_video_sink_class_init):
35233 Throw away frames that are later than 20 ms.
35235 2006-03-06 14:14:47 +0000 Fabrizio <fabrizio.ge@tiscali.it>
35237 gst-libs/gst/riff/riff-media.c:
35238 Original commit message from CVS:
35239 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
35240 Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).
35242 2006-03-05 23:39:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35244 ext/ogg/gstoggmux.c: put Theora BOS pages before others. This hardcodes the Ogg/Theora I profile, but hey.
35245 Original commit message from CVS:
35246 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
35247 (gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
35248 put Theora BOS pages before others. This hardcodes
35249 the Ogg/Theora I profile, but hey.
35251 2006-03-05 23:06:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35253 * ext/ogg/gstoggmux.c:
35254 changed more than 5 lines
35255 Original commit message from CVS:
35256 changed more than 5 lines
35258 2006-03-05 22:57:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35260 ogg muxing of vorbis and theora now has pages ordered correctly again, even with delays.
35261 Original commit message from CVS:
35262 ogg muxing of vorbis and theora now has pages ordered correctly again,
35265 updated with some examples
35266 * ext/theora/theoraenc.c: (granulepos_to_timestamp),
35267 (granulepos_add), (theora_buffer_from_packet):
35268 * ext/vorbis/vorbisenc.c: (granulepos_to_timestamp_offset),
35269 (granulepos_to_timestamp), (gst_vorbisenc_buffer_from_packet),
35270 (gst_vorbisenc_chain):
35271 implement strategy from ext/ogg/README
35272 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
35273 (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
35274 (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_compare_pads),
35275 (gst_ogg_mux_queue_pads), (gst_ogg_mux_collected):
35276 Fix muxer so that oggz-validate is happy with all streams;
35277 except for no eos mark, and the BOS page ordering
35278 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
35279 (check_buffer_granulepos):
35280 * tests/check/pipelines/vorbisenc.c: (check_buffer_granulepos):
35281 update tests to check for OFFSET being set as requested
35282 fixed type of granulepos, it's not a ClockTime
35284 2006-03-05 21:34:23 +0000 Julien Moutte <julien@moutte.net>
35286 sys/xvimage/xvimagesink.c: Check that the xvimage we are creating has a correct size before returning it. (#3...
35287 Original commit message from CVS:
35288 2006-03-05 Julien MOUTTE <julien@moutte.net>
35289 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
35290 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
35291 Check that the xvimage we are creating has a correct size before returning it. (#314897)
35293 2006-03-05 13:44:05 +0000 Tim-Philipp Müller <tim@centricular.net>
35295 gst/typefind/gsttypefindfunctions.c: Give id3 and ape tag typefinders a rank slightly higher than PRIMARY to ensure t...
35296 Original commit message from CVS:
35297 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
35298 Give id3 and ape tag typefinders a rank slightly higher
35299 than PRIMARY to ensure they're always run before any of
35300 the other typefinders (in particular wav and mp3) (#324186).
35302 2006-03-05 13:08:37 +0000 Tim-Philipp Müller <tim@centricular.net>
35304 gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
35305 Original commit message from CVS:
35306 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
35307 Add support for '3IVD' fourcc (#333403).
35309 2006-03-04 14:35:10 +0000 Tim-Philipp Müller <tim@centricular.net>
35311 configure.ac: Bump requirements to GStreamer CVS for the new error enum.
35312 Original commit message from CVS:
35314 Bump requirements to GStreamer CVS for the new error enum.
35315 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
35316 Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
35317 space left on the device (fixes #333352).
35319 2006-03-03 23:53:50 +0000 Sébastien Moutte <sebastien@moutte.net>
35321 win32/vs6: add a project file for libgstvolume update the workspace
35322 Original commit message from CVS:
35324 add a project file for libgstvolume
35325 update the workspace
35327 2006-03-03 15:26:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35330 * ext/ogg/gstoggmux.c:
35332 Original commit message from CVS:
35335 2006-03-03 15:22:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35337 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
35338 Original commit message from CVS:
35339 2006-03-03 Thomas Vander Stichele <thomas at apestaart dot org>
35340 * ext/theora/theoraenc.c: (theora_set_header_on_caps):
35341 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
35343 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
35344 Set IN_CAPS on header buffers
35346 2006-03-02 18:23:55 +0000 Wim Taymans <wim.taymans@gmail.com>
35348 docs/plugins/: Add audioresample to docs.
35349 Original commit message from CVS:
35350 * docs/plugins/Makefile.am:
35351 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
35352 * docs/plugins/gst-plugins-base-plugins-sections.txt:
35353 Add audioresample to docs.
35354 * gst/audioconvert/gstaudioconvert.c:
35356 * gst/audioresample/gstaudioresample.c:
35357 (gst_audioresample_base_init), (gst_audioresample_class_init),
35358 (gst_audioresample_init), (gst_audioresample_dispose),
35359 (audioresample_get_unit_size), (audioresample_transform_caps),
35360 (resample_set_state_from_caps), (audioresample_transform_size),
35361 (audioresample_set_caps), (audioresample_event),
35362 (audioresample_do_output), (audioresample_transform),
35363 (audioresample_pushthrough), (gst_audioresample_set_property),
35364 (gst_audioresample_get_property), (plugin_init):
35365 * gst/audioresample/gstaudioresample.h:
35367 Small code cleanups.
35369 2006-03-02 18:12:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35371 * gst/videorate/Makefile.am:
35373 Original commit message from CVS:
35376 2006-03-02 17:48:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35378 * ext/ogg/gstoggmux.c:
35379 debug using the actual GstPad, that allows us to see the serialno in the padname
35380 Original commit message from CVS:
35381 debug using the actual GstPad, that allows us to see the serialno in the padname
35383 2006-03-02 17:46:36 +0000 Wim Taymans <wim.taymans@gmail.com>
35385 docs/plugins/: Added videoscale to docs.
35386 Original commit message from CVS:
35387 * docs/plugins/Makefile.am:
35388 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
35389 * docs/plugins/gst-plugins-base-plugins-sections.txt:
35390 Added videoscale to docs.
35391 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
35392 (gst_video_rate_swap_prev), (gst_video_rate_event),
35393 (gst_video_rate_chain):
35395 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
35396 (gst_video_scale_init), (gst_video_scale_prepare_size),
35397 (gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
35398 (gst_video_scale_fixate_caps), (gst_video_scale_transform):
35399 * gst/videoscale/gstvideoscale.h:
35400 Added docs, examples.
35401 Some code cleanups.
35402 Post errors instead of g_warning.
35404 2006-03-02 17:30:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35406 * ext/ogg/gstoggmux.c:
35407 clean up debug messages
35408 Original commit message from CVS:
35409 clean up debug messages
35411 2006-03-02 17:15:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35413 * ext/ogg/gstoggmux.c:
35414 extra debugging from older version, makes it easier to compare
35415 Original commit message from CVS:
35416 extra debugging from older version, makes it easier to compare
35418 2006-03-02 17:04:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
35420 * ext/ogg/gstoggmux.c:
35421 some space cleanup and debug fixes
35422 Original commit message from CVS:
35423 some space cleanup and debug fixes
35425 2006-03-02 16:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
35427 docs/: Added some more docs to libs and plugins.
35428 Original commit message from CVS:
35429 * docs/libs/gst-plugins-base-libs-docs.sgml:
35430 * docs/libs/gst-plugins-base-libs-sections.txt:
35431 * docs/libs/gst-plugins-base-libs.types:
35432 * docs/plugins/Makefile.am:
35433 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
35434 * docs/plugins/gst-plugins-base-plugins-sections.txt:
35435 Added some more docs to libs and plugins.
35436 * gst-libs/gst/audio/gstringbuffer.c:
35437 (gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
35438 * gst-libs/gst/audio/gstringbuffer.h:
35439 Document ringbuffer some more.
35440 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
35441 (gst_video_rate_setcaps), (gst_video_rate_reset),
35442 (gst_video_rate_init), (gst_video_rate_flush_prev),
35443 (gst_video_rate_swap_prev), (gst_video_rate_event),
35444 (gst_video_rate_chain), (gst_video_rate_change_state):
35445 * gst/videorate/gstvideorate.h:
35446 Fix videorate to use segments.
35447 Make it work with 0/1 framerates (closes #331903)
35448 Handle EOS correctly.
35451 2006-03-02 13:13:00 +0000 Tim-Philipp Müller <tim@centricular.net>
35453 ext/ogg/gstogmparse.c: In state change function, first chain up to parent class, then handle downwards state change s...
35454 Original commit message from CVS:
35455 * ext/ogg/gstogmparse.c: (gst_ogm_parse_class_init),
35456 (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
35457 (gst_ogm_text_parse_init), (gst_ogm_parse_change_state):
35458 In state change function, first chain up to parent class,
35459 then handle downwards state change stuff. Remove some
35460 commented out cruft from 0.8 code.
35462 2006-03-02 12:35:59 +0000 Tim-Philipp Müller <tim@centricular.net>
35464 ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
35465 Original commit message from CVS:
35466 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
35467 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
35468 (gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
35469 (gst_ogm_parse_chain):
35470 Don't remove/re-add source pad if the new caps are the same as
35471 the old caps anyway (#333042). When removing source pad, don't
35472 unref it afterwards - we didn't ref it when adding. Sprinkle some
35473 GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
35474 after using gst_pad_get_parent(). Return downstream flow return
35475 value in chain function.
35477 2006-03-02 11:28:23 +0000 Wim Taymans <wim.taymans@gmail.com>
35479 docs/plugins/: Fix hierarchy, added some more elements to the docs.
35480 Original commit message from CVS:
35481 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
35482 * docs/plugins/gst-plugins-base-plugins.args:
35483 * docs/plugins/gst-plugins-base-plugins.hierarchy:
35484 * docs/plugins/gst-plugins-base-plugins.interfaces:
35485 * docs/plugins/gst-plugins-base-plugins.signals:
35486 Fix hierarchy, added some more elements to the docs.
35487 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
35488 (gst_ffmpegcsp_get_type):
35489 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
35490 Fix docs for ffmpegcolorspace.
35492 2006-03-01 19:24:44 +0000 Tim-Philipp Müller <tim@centricular.net>
35494 gst/typefind/gsttypefindfunctions.c: Some typefinding fine-tuning:
35495 Original commit message from CVS:
35496 * gst/typefind/gsttypefindfunctions.c: (id3_type_find),
35497 (apetag_type_find), (ape_type_find), (plugin_init):
35498 Some typefinding fine-tuning:
35499 - rank ID3/APE tags in order of preference via probabilities, so that
35500 ID3v2 > APEv2 > APEv1 > ID3v1.
35501 - three or four bytes don't really justify MAXIMUM probability,
35502 change those to 'very likely' (musepack and monkeysaudio).
35504 2006-03-01 18:25:18 +0000 Wim Taymans <wim.taymans@gmail.com>
35507 Original commit message from CVS:
35508 * docs/plugins/Makefile.am:
35509 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
35510 * docs/plugins/gst-plugins-base-plugins-sections.txt:
35511 * ext/alsa/gstalsamixer.c:
35512 * ext/alsa/gstalsamixer.h:
35513 * ext/alsa/gstalsamixerelement.c:
35514 (gst_alsa_mixer_element_class_init), (gst_alsa_mixer_element_init):
35515 * ext/alsa/gstalsamixerelement.h:
35516 * ext/alsa/gstalsasink.c:
35517 * ext/alsa/gstalsasink.h:
35518 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
35519 (gst_alsasrc_init):
35520 * ext/alsa/gstalsasrc.h:
35522 Small code cleanups.
35524 2006-03-01 17:52:45 +0000 Wim Taymans <wim.taymans@gmail.com>
35526 ext/theora/Makefile.am: Dist new header too,
35527 Original commit message from CVS:
35528 * ext/theora/Makefile.am:
35529 Dist new header too,
35531 2006-03-01 17:39:28 +0000 Wim Taymans <wim.taymans@gmail.com>
35533 Fix some more docs.
35534 Original commit message from CVS:
35535 * docs/plugins/Makefile.am:
35536 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
35537 * docs/plugins/gst-plugins-base-plugins-sections.txt:
35538 * ext/gnomevfs/gstgnomevfssink.h:
35539 * ext/gnomevfs/gstgnomevfssrc.h:
35540 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
35541 * ext/vorbis/vorbisdec.h:
35542 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
35543 * ext/vorbis/vorbisenc.h:
35544 * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
35545 (vorbis_parse_chain), (vorbis_parse_change_state):
35546 * ext/vorbis/vorbisparse.h:
35547 * gst/audioconvert/gstaudioconvert.h:
35548 * gst/tcp/gsttcpserversink.h:
35549 * gst/videotestsrc/gstvideotestsrc.c:
35550 * gst/videotestsrc/gstvideotestsrc.h:
35551 * gst/volume/gstvolume.c:
35552 * gst/volume/gstvolume.h:
35553 Fix some more docs.
35554 Added docs for vorbisdec and vorbisparse.
35557 2006-03-01 16:24:37 +0000 Wim Taymans <wim.taymans@gmail.com>
35559 Updated/added documentation.
35560 Original commit message from CVS:
35561 * docs/plugins/Makefile.am:
35562 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
35563 * docs/plugins/gst-plugins-base-plugins-sections.txt:
35564 * ext/pango/gstclockoverlay.h:
35565 * ext/pango/gsttextoverlay.h:
35566 * ext/pango/gsttextrender.h:
35567 * ext/pango/gsttimeoverlay.h:
35568 * ext/theora/gsttheoradec.h:
35569 * ext/theora/gsttheoraenc.h:
35570 * ext/theora/theoradec.c:
35571 * ext/theora/theoraenc.c:
35572 * gst/audioconvert/gstaudioconvert.h:
35573 * gst/audiotestsrc/gstaudiotestsrc.h:
35574 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
35575 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
35576 * gst/tcp/gstmultifdsink.h:
35577 Updated/added documentation.
35578 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
35579 (gst_text_overlay_halign_get_type),
35580 (gst_text_overlay_wrap_mode_get_type),
35581 (gst_text_overlay_base_init), (gst_text_overlay_class_init),
35582 (gst_text_overlay_init), (gst_text_overlay_set_property),
35583 (gst_text_overlay_get_property):
35584 Fix up properties to be enums instead of string to make bindings,
35585 introspection and automatic GUI creation possible.
35586 Add getters for the properties.
35588 2006-02-28 21:21:07 +0000 Sébastien Moutte <sebastien@moutte.net>
35590 gst/audiotestsrc/gstaudiotestsrc.c: added defines of M_PI and M_PI_2
35591 Original commit message from CVS:
35592 * gst/audiotestsrc/gstaudiotestsrc.c:
35593 added defines of M_PI and M_PI_2
35594 * gst/ffmpegcolorspace/avcodec.h:
35595 removed #include "stdint.h" for win32 as _stdint.h is
35596 autogenerated to win32/common
35597 * win32/common/libgstaudio.def:
35598 * win32/common/libgsttag.def:
35601 some project files bugs corrected
35603 project files are reset to the default vs7 configuration
35604 (they link to msvcr71.dll using default optimizations)
35606 2006-02-28 19:08:12 +0000 Wim Taymans <wim.taymans@gmail.com>
35608 ext/gnomevfs/gstgnomevfssink.c: Fix some docs.
35609 Original commit message from CVS:
35610 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
35613 2006-02-28 13:52:04 +0000 Edward Hervey <bilboed@bilboed.com>
35615 ext/alsa/gstalsasrc.c: Set proper class on the ElementDetails:
35616 Original commit message from CVS:
35617 * ext/alsa/gstalsasrc.c:
35618 Set proper class on the ElementDetails:
35619 Source/Audio instead of Src/Audio
35621 2006-02-28 12:19:11 +0000 Edward Hervey <bilboed@bilboed.com>
35623 gst/videoscale/vs_scanline.c: Revert optimization in videoscale. It should go in liboil and have an appropriate liboi...
35624 Original commit message from CVS:
35625 * gst/videoscale/vs_scanline.c:
35626 (vs_scanline_resample_nearest_RGBA):
35627 Revert optimization in videoscale. It should go in liboil and have
35628 an appropriate liboil function.
35630 2006-02-28 11:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
35632 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
35633 Original commit message from CVS:
35634 * gst-libs/gst/audio/gstbaseaudiosink.c:
35635 (gst_base_audio_sink_provide_clock):
35636 Don't try to provide a clock in the NULL state.
35638 2006-02-28 11:04:47 +0000 Wim Taymans <wim.taymans@gmail.com>
35640 ext/ogg/gstoggdemux.c: Use GstSegment infrastructure to remove duplicated code and handle more seek cases correctly.
35641 Original commit message from CVS:
35642 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
35643 (gst_ogg_pad_event), (gst_ogg_pad_internal_chain),
35644 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
35645 (gst_ogg_demux_deactivate_current_chain),
35646 (gst_ogg_demux_activate_chain), (gst_ogg_demux_do_seek),
35647 (gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_info),
35648 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
35649 (gst_ogg_demux_loop), (gst_ogg_demux_change_state):
35650 Use GstSegment infrastructure to remove duplicated code
35651 and handle more seek cases correctly.
35653 2006-02-28 10:39:19 +0000 Wim Taymans <wim.taymans@gmail.com>
35655 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Don't ignore return code from ffmpeg convert function.
35656 Original commit message from CVS:
35657 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
35658 (gst_ffmpegcsp_transform):
35659 Don't ignore return code from ffmpeg convert function.
35660 * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
35661 Split out some long statements to ease debugging.
35663 2006-02-27 12:08:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35665 ext/libvisual/visual.c: Don't use gst_pad_use_fixed_caps, because it prevents downstream from being able to renegotia...
35666 Original commit message from CVS:
35667 * ext/libvisual/visual.c: (gst_visual_init),
35668 (gst_vis_src_negotiate), (get_buffer), (plugin_init):
35669 Don't use gst_pad_use_fixed_caps, because it prevents downstream from
35670 being able to renegotiate the size. Instead, use the negotiation
35671 algorithm from the goom plugin to pick an initial output caps.
35672 Also, allow theoretical libvisual plugins that might support non-GL
35673 output even if they also do GL.
35675 2006-02-26 21:05:46 +0000 Julien Moutte <julien@moutte.net>
35677 ext/libvisual/visual.c: Load only non GL plugins. Fix some memleaks and possible negotiation issues.
35678 Original commit message from CVS:
35679 2006-02-26 Julien MOUTTE <julien@moutte.net>
35680 * ext/libvisual/visual.c: (gst_visual_init),
35681 (gst_visual_src_setcaps), (get_buffer), (gst_visual_chain),
35682 (plugin_init): Load only non GL plugins. Fix some memleaks and
35683 possible negotiation issues.
35685 2006-02-24 23:19:44 +0000 Julien Moutte <julien@moutte.net>
35687 gst-libs/gst/tag/tag.h: Adding Annodex tags here.
35688 Original commit message from CVS:
35689 2006-02-25 Julien MOUTTE <julien@moutte.net>
35690 * gst-libs/gst/tag/tag.h: Adding Annodex tags here.
35692 2006-02-24 18:55:27 +0000 Michael Smith <msmith@xiph.org>
35694 gst/typefind/gsttypefindfunctions.c: Fix CMML type find function to not require a specific minor version of the CMML ...
35695 Original commit message from CVS:
35696 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
35697 (cmml_type_find), (plugin_init):
35698 Fix CMML type find function to not require a specific minor version
35699 of the CMML header.
35700 Add an MPEG4 video elementary stream typefind function.
35702 2006-02-24 17:31:53 +0000 Michael Smith <msmith@xiph.org>
35704 ext/ogg/gstoggdemux.c: Annodex support in ogg demuxer. Doesn't do very much without the other annodex patches (to come).
35705 Original commit message from CVS:
35706 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
35707 (gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
35708 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
35709 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
35710 (gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
35711 (gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
35712 Annodex support in ogg demuxer. Doesn't do very much without the
35713 other annodex patches (to come).
35715 2006-02-24 16:21:34 +0000 Tim-Philipp Müller <tim@centricular.net>
35717 gst-libs/gst/riff/riff-media.c:
35718 Original commit message from CVS:
35719 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
35720 Pick up palette for MS video v1 (#327028, patch by:
35721 Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
35723 2006-02-24 13:54:04 +0000 Tim-Philipp Müller <tim@centricular.net>
35725 gst/ffmpegcolorspace/gstffmpegcolorspace.c: The 'palette_data' field from incoming RGB caps shouldn't be proxied on o...
35726 Original commit message from CVS:
35727 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
35728 (gst_ffmpegcsp_caps_remove_format_info),
35729 (gst_ffmpegcsp_get_unit_size):
35730 The 'palette_data' field from incoming RGB caps shouldn't be
35731 proxied on outgoing YUV caps; also, restrict unit size
35732 adjustment in case of paletted data only to the unit that
35733 actually has a palette. Fixes #330711.
35735 2006-02-24 12:18:14 +0000 Tim-Philipp Müller <tim@centricular.net>
35737 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Plug some memory leaks.
35738 Original commit message from CVS:
35739 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
35740 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
35741 (gst_ffmpegcsp_finalize), (gst_ffmpegcsp_class_init),
35742 (gst_ffmpegcsp_get_unit_size):
35743 Plug some memory leaks.
35745 2006-02-24 10:18:52 +0000 Tim-Philipp Müller <tim@centricular.net>
35747 sys/: Add some _CFLAGS and _LIBS that seem to be missing and/or required for Cygwin (see #317048).
35748 Original commit message from CVS:
35749 * sys/ximage/Makefile.am:
35750 * sys/xvimage/Makefile.am:
35751 Add some _CFLAGS and _LIBS that seem to be missing
35752 and/or required for Cygwin (see #317048).
35754 2006-02-24 00:07:18 +0000 Tim-Philipp Müller <tim@centricular.net>
35757 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
35758 Original commit message from CVS:
35759 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
35761 2006-02-22 18:46:46 +0000 Wim Taymans <wim.taymans@gmail.com>
35763 ext/alsa/gstalsasrc.c: Fix description as pointed out by caugier.
35764 Original commit message from CVS:
35765 * ext/alsa/gstalsasrc.c:
35766 Fix description as pointed out by caugier.
35768 2006-02-22 10:29:22 +0000 Edward Hervey <bilboed@bilboed.com>
35770 gst/typefind/gsttypefindfunctions.c: Better 3gp typefinding.
35771 Original commit message from CVS:
35772 Reviewed by : Edward Hervey <edward@fluendo.com>
35773 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
35775 Better 3gp typefinding.
35777 2006-02-21 12:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
35779 ext/gnomevfs/gstgnomevfssrc.c: Don't send EOS event here, the base class will send one for us.
35780 Original commit message from CVS:
35781 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
35782 Don't send EOS event here, the base class will send one for us.
35783 * gst/playback/gstplaybasebin.c: (prepare_output):
35784 Subpictures without video stream aren't allowed either.
35785 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
35786 Fix debug statement copy'n'paste-o.
35788 2006-02-21 12:05:18 +0000 Tim-Philipp Müller <tim@centricular.net>
35790 ext/alsa/gstalsamixer.c: Fix issues with mixer keeping state when muting/unmuting and when changing the volume whilst...
35791 Original commit message from CVS:
35792 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
35793 Fix issues with mixer keeping state when muting/unmuting
35794 and when changing the volume whilst muted (see #331763
35797 2006-02-20 18:27:06 +0000 Tim-Philipp Müller <tim@centricular.net>
35799 gst/subparse/gstsubparse.c: Set right caps given that we send escaped text. Also, honour <i></i>, <b></b> and <u></u>...
35800 Original commit message from CVS:
35801 * gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
35802 (parse_subrip), (gst_sub_parse_format_autodetect):
35803 Set right caps given that we send escaped text. Also,
35804 honour <i></i>, <b></b> and <u></u> markers that can be found
35805 in .srt files (fixes #310202).
35807 2006-02-20 16:21:14 +0000 Tim-Philipp Müller <tim@centricular.net>
35809 gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
35810 Original commit message from CVS:
35811 * gst-libs/gst/audio/mixerutils.c:
35812 (element_factory_rank_compare_func):
35813 Make order in which elements are tried more determinable.
35815 2006-02-20 15:57:51 +0000 Julien Moutte <julien@moutte.net>
35817 gst/playback/gstdecodebin.c: Make decodebin reusable by fixing remove_element_chain first and then introduce a cleane...
35818 Original commit message from CVS:
35819 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
35820 (remove_element_chain), (cleanup_decodebin),
35821 (gst_decode_bin_change_state): Make decodebin reusable by
35822 fixing remove_element_chain first and then introduce a
35823 cleaner in state change to ->NULL. (Closes #331678)
35824 ------------------------------------------------------
35826 2006-02-19 14:32:35 +0000 Wim Taymans <wim.taymans@gmail.com>
35828 ext/gnomevfs/gstgnomevfssink.c: use 0666 mask when creating files so umask gets applied correctly. Fixes #331295.
35829 Original commit message from CVS:
35830 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
35831 use 0666 mask when creating files so umask gets applied
35832 correctly. Fixes #331295.
35834 2006-02-19 14:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
35836 gst/subparse/: Add very basic parser for SSA subtitle streams (as often found in matroska files).
35837 Original commit message from CVS:
35838 * gst/subparse/Makefile.am:
35839 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
35840 (gst_ssa_parse_dispose), (gst_ssa_parse_init),
35841 (gst_ssa_parse_class_init), (gst_ssa_parse_src_event),
35842 (gst_ssa_parse_sink_event), (gst_ssa_parse_setcaps),
35843 (gst_ssa_parse_remove_override_codes), (gst_ssa_parse_parse_line),
35844 (gst_ssa_parse_chain), (gst_ssa_parse_change_state):
35845 * gst/subparse/gstssaparse.h:
35846 * gst/subparse/gstsubparse.c: (plugin_init):
35847 Add very basic parser for SSA subtitle streams (as often
35848 found in matroska files).
35850 2006-02-19 14:09:40 +0000 Tim-Philipp Müller <tim@centricular.net>
35852 gst/playback/gstdecodebin.c: That should be text/x-pango-markup, not text/x-pango-layout.
35853 Original commit message from CVS:
35854 * gst/playback/gstdecodebin.c: (mimetype_is_raw):
35855 That should be text/x-pango-markup, not text/x-pango-layout.
35857 2006-02-19 12:41:03 +0000 Julien Moutte <julien@moutte.net>
35859 ext/pango/gsttextoverlay.c: Polishing.
35860 Original commit message from CVS:
35861 2006-02-19 Julien MOUTTE <julien@moutte.net>
35862 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize):
35865 2006-02-19 12:05:23 +0000 Julien Moutte <julien@moutte.net>
35867 ext/pango/gsttextoverlay.c: Fix state change deadlock.
35868 Original commit message from CVS:
35869 2006-02-19 Julien MOUTTE <julien@moutte.net>
35870 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
35871 (gst_text_overlay_finalize), (gst_text_overlay_init),
35872 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
35873 (gst_text_overlay_render_text),
35874 (gst_text_overlay_text_pad_link),
35875 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
35876 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
35877 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
35878 Fix state change deadlock.
35880 2006-02-19 11:56:28 +0000 Julien Moutte <julien@moutte.net>
35882 ext/pango/gsttextoverlay.*: Fix seeking both for muxed formats and subtitles files.
35883 Original commit message from CVS:
35884 2006-02-19 Julien MOUTTE <julien@moutte.net>
35885 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
35886 (gst_text_overlay_finalize), (gst_text_overlay_init),
35887 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
35888 (gst_text_overlay_render_text),
35889 (gst_text_overlay_text_pad_link),
35890 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
35891 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
35892 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
35893 * ext/pango/gsttextoverlay.h: Fix seeking both for muxed formats
35894 and subtitles files.
35896 2006-02-19 00:40:38 +0000 Julien Moutte <julien@moutte.net>
35898 gst/playback/gstdecodebin.c: pango layout should be considered as row.
35899 Original commit message from CVS:
35900 2006-02-19 Julien MOUTTE <julien@moutte.net>
35901 * gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
35902 should be considered as row.
35904 2006-02-19 00:25:16 +0000 Julien Moutte <julien@moutte.net>
35906 gst/playback/gststreaminfo.*: Introduce language informations.
35907 Original commit message from CVS:
35908 2006-02-19 Julien MOUTTE <julien@moutte.net>
35909 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
35911 * gst/playback/gststreaminfo.h: Introduce language informations.
35913 2006-02-18 22:41:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
35915 sys/: Set shared memory segments to be deleted as soon as we have attached, that way they get cleaned up automaticall...
35916 Original commit message from CVS:
35917 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
35918 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
35919 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
35920 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
35921 Set shared memory segments to be deleted as soon as we have attached,
35922 that way they get cleaned up automatically if we crash.
35924 2006-02-18 19:53:48 +0000 Julien Moutte <julien@moutte.net>
35926 ext/pango/: Those functions are called with lock held.
35927 Original commit message from CVS:
35928 2006-02-18 Julien MOUTTE <julien@moutte.net>
35929 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_get_text):
35930 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_get_text): Those
35931 functions are called with lock held.
35933 2006-02-18 19:51:47 +0000 Julien Moutte <julien@moutte.net>
35937 Original commit message from CVS:
35940 2006-02-18 19:10:35 +0000 Julien Moutte <julien@moutte.net>
35942 ext/pango/gsttextoverlay.c: Refactoring of textoverlay without collectpads. This now supports sparse subtitles coming...
35943 Original commit message from CVS:
35944 2006-02-18 Julien MOUTTE <julien@moutte.net>
35945 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
35946 (gst_text_overlay_finalize), (gst_text_overlay_init),
35947 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
35948 (gst_text_overlay_render_text),
35949 (gst_text_overlay_text_pad_link),
35950 (gst_text_overlay_text_pad_unlink),
35951 (gst_text_overlay_text_event),
35952 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
35953 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
35954 (gst_text_overlay_change_state): Refactoring of textoverlay
35955 without collectpads. This now supports sparse subtitles coming
35956 from a demuxer instead of a sub file. Seeking is still broken
35957 though. Need to discuss with wtay some more on how to handle
35959 * ext/pango/gsttextoverlay.h:
35960 * gst/playback/gstplaybin.c: (setup_sinks): Support linking with
35961 subtitles coming from the demuxer.
35963 2006-02-17 19:31:12 +0000 Wim Taymans <wim.taymans@gmail.com>
35965 ext/vorbis/vorbisenc.c: Use some more scaling functions.
35966 Original commit message from CVS:
35967 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
35968 (gst_vorbisenc_convert_sink):
35969 Use some more scaling functions.
35971 2006-02-17 16:12:11 +0000 Tim-Philipp Müller <tim@centricular.net>
35973 ext/cdparanoia/gstcdparanoiasrc.*: Add back 'transport-error' and 'uncorrected-error' signals and make them actually ...
35974 Original commit message from CVS:
35975 * ext/cdparanoia/gstcdparanoiasrc.c:
35976 (gst_cd_paranoia_src_class_init), (gst_cd_paranoia_dummy_callback),
35977 (gst_cd_paranoia_paranoia_callback),
35978 (gst_cd_paranoia_src_signal_is_being_watched),
35979 (gst_cd_paranoia_src_read_sector):
35980 * ext/cdparanoia/gstcdparanoiasrc.h:
35981 Add back 'transport-error' and 'uncorrected-error' signals and
35982 make them actually be fired when bad stuff happens (#319340).
35984 2006-02-17 14:07:01 +0000 Wim Taymans <wim.taymans@gmail.com>
35986 gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
35987 Original commit message from CVS:
35988 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
35989 (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
35990 (gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
35991 (gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
35992 (gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
35993 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
35994 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
35995 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
35996 (gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
35997 (gst_ring_buffer_clear):
35999 Added some G_LIKELY.
36001 2006-02-17 10:15:52 +0000 Wim Taymans <wim.taymans@gmail.com>
36003 gst-libs/gst/audio/TODO: Update TODO
36004 Original commit message from CVS:
36005 * gst-libs/gst/audio/TODO:
36007 * gst-libs/gst/audio/gstbaseaudiosink.c:
36008 (gst_base_audio_sink_get_offset):
36009 When trying to play samples ASAP and we don't have a
36010 previous sample, try to play at position 0 instead of
36011 an invalid position.
36013 2006-02-17 09:24:56 +0000 Wim Taymans <wim.taymans@gmail.com>
36015 ext/alsa/gstalsasink.c: Also release lock when we get an error in _reset(); fix an error message.
36016 Original commit message from CVS:
36017 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
36018 (gst_alsasink_reset):
36019 Also release lock when we get an error in _reset();
36020 fix an error message.
36022 2006-02-16 21:01:23 +0000 Tim-Philipp Müller <tim@centricular.net>
36024 ext/alsa/gstalsasink.*: Add support for more than 2 channels (#326720).
36025 Original commit message from CVS:
36026 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
36027 (gst_alsasink_init), (get_channel_free_structure),
36028 (caps_add_channel_configuration), (gst_alsasink_getcaps),
36029 (gst_alsasink_close):
36030 * ext/alsa/gstalsasink.h:
36031 Add support for more than 2 channels (#326720).
36033 2006-02-16 20:19:51 +0000 Tim-Philipp Müller <tim@centricular.net>
36035 gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
36036 Original commit message from CVS:
36037 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
36038 Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
36039 with 4 or 6 channels, assume a default channel layout to make things
36040 work (not sure there's anything else we can do in those cases).
36042 2006-02-16 19:18:46 +0000 Tim-Philipp Müller <tim@centricular.net>
36044 gst-libs/gst/audio/multichannel.c: Minor docs fix.
36045 Original commit message from CVS:
36046 * gst-libs/gst/audio/multichannel.c:
36048 * gst-libs/gst/riff/Makefile.am:
36049 * gst-libs/gst/riff/riff-ids.h:
36050 * gst-libs/gst/riff/riff-media.c:
36051 (gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
36052 Add support for WAVEFORMATEX, eg. PCM audio with more than two
36053 channels and a channel layout map.
36055 2006-02-16 17:06:46 +0000 Edward Hervey <bilboed@bilboed.com>
36057 gst/videoscale/vs_scanline.c: C-level optimization of the RGBA nearest neighbour function.
36058 Original commit message from CVS:
36059 Reviewed by Edward Hervey <edward@fluendo.com>
36060 * gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
36061 C-level optimization of the RGBA nearest neighbour function.
36062 Eventually this might end up in liboil with vectorized versions.
36064 2006-02-16 11:44:43 +0000 Tim-Philipp Müller <tim@centricular.net>
36066 gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
36067 Original commit message from CVS:
36068 * gst-libs/gst/audio/multichannel.c:
36069 (gst_audio_get_channel_positions):
36070 When we have more than 2 channels, but no channel layout is
36071 specified in the caps, return some default channel layout
36072 to the caller and warn about about a possibly buggy element
36073 (could be buggy filtercaps as well of course) (#317038).
36075 2006-02-16 09:29:38 +0000 Tim-Philipp Müller <tim@centricular.net>
36077 pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: Add gst-libs/gst/cdda to list of lib search paths.
36078 Original commit message from CVS:
36079 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
36080 Add gst-libs/gst/cdda to list of lib search paths.
36082 2006-02-15 12:20:47 +0000 Andy Wingo <wingo@pobox.com>
36084 ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating timestamp, update timestamp_end as well. Fixes a bugaboo. ...
36085 Original commit message from CVS:
36086 2006-02-15 Andy Wingo <wingo@pobox.com>
36087 * ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating
36088 timestamp, update timestamp_end as well. Fixes a bugaboo. I hope
36089 to the Lord Jesus that I do not have to touch the ogg muxer ever
36092 2006-02-15 12:07:57 +0000 Edward Hervey <bilboed@bilboed.com>
36094 gst/typefind/gsttypefindfunctions.c: quicktime movie files can also contain 'uuid' atoms.
36095 Original commit message from CVS:
36096 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
36097 quicktime movie files can also contain 'uuid' atoms.
36099 2006-02-14 18:52:52 +0000 Tim-Philipp Müller <tim@centricular.net>
36101 gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun...
36102 Original commit message from CVS:
36103 * gst/audioconvert/plugin.c: (plugin_init):
36104 Register the GstAudioChannelPosition enum type with the type
36105 system in the plugin_init function, so that it is known before
36106 any element actually makes use of multi-channel stuff. This is
36107 required for example if one wants to be able to deserialise/use
36108 a caps string with channel positions before any pipeline has
36109 been setup and started, like with gst-launch.
36111 2006-02-14 13:45:35 +0000 Wim Taymans <wim.taymans@gmail.com>
36113 gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
36114 Original commit message from CVS:
36115 * gst-libs/gst/audio/gstringbuffer.c:
36116 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
36117 (gst_ring_buffer_samples_done), (wait_segment),
36118 (gst_ring_buffer_commit), (gst_ring_buffer_clear):
36119 Add some compiler G_(UN_)LIKELY help.
36120 SIGNAL the ringbuffer waiters when going to PAUSED as well to
36121 make sure they can exit their functions. Should fix #330748
36123 2006-02-13 20:49:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36125 Windows does not have long long; copy the generated _stdint.h
36126 Original commit message from CVS:
36130 * win32/common/_stdint.h:
36131 Windows does not have long long; copy the generated _stdint.h
36132 * win32/common/interfaces-enumtypes.c:
36133 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
36134 (gst_mixer_track_flags_get_type),
36135 (gst_tuner_channel_flags_get_type):
36136 * win32/common/multichannel-enumtypes.c:
36137 (gst_audio_channel_position_get_type):
36140 2006-02-13 18:49:02 +0000 Wim Taymans <wim.taymans@gmail.com>
36142 gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
36143 Original commit message from CVS:
36144 * gst-libs/gst/audio/gstbaseaudiosink.c:
36145 (gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
36146 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
36147 Always sync on first sample we receive when starting.
36149 2006-02-13 15:59:48 +0000 Wim Taymans <wim.taymans@gmail.com>
36151 gst/playback/gstplaybin.c: Update vis bin docs.
36152 Original commit message from CVS:
36153 * gst/playback/gstplaybin.c: (gen_vis_element):
36154 Update vis bin docs.
36155 Move queue after tee so we don't queue video buffers but
36156 audio samples instead. Fixes problems where the video queue
36157 is filled and the audio queue empty.
36159 2006-02-13 15:17:34 +0000 Tim-Philipp Müller <tim@centricular.net>
36161 gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
36162 Original commit message from CVS:
36163 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
36164 No need to push an EOS event here, GstBaseSrc will do that for us
36165 when we return FLOW_UNEXPECTED.
36167 2006-02-12 14:54:55 +0000 Wim Taymans <wim.taymans@gmail.com>
36169 gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
36170 Original commit message from CVS:
36171 * gst-libs/gst/audio/gstbaseaudiosink.c:
36172 (gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
36173 (gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
36174 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
36175 Use scale functions when possible.
36176 Fix error messages.
36177 Free clockid when after waiting for EOS.
36178 Use G_(UN_)LIKLY when it makes sense.
36179 Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
36181 2006-02-12 14:26:55 +0000 Edward Hervey <bilboed@bilboed.com>
36183 gst/playback/gstplaybasebin.c: Remove stray semi-colon (fixes #330888).
36184 Original commit message from CVS:
36185 * gst/playback/gstplaybasebin.c: (prepare_output):
36186 Remove stray semi-colon (fixes #330888).
36188 2006-02-11 23:35:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36190 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...
36191 Original commit message from CVS:
36192 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
36193 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
36194 Fix up the XShm call testing so that we catch errors, and don't
36195 cause new ones by attempting to detach from a segment we failed
36196 to attach to. Fixes #312439.
36198 2006-02-10 11:29:55 +0000 Edward Hervey <bilboed@bilboed.com>
36200 gst/typefind/gsttypefindfunctions.c: Added flv file typefind (video/x-flv).
36201 Original commit message from CVS:
36202 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
36203 Added flv file typefind (video/x-flv).
36205 2006-02-10 10:53:33 +0000 Edward Hervey <bilboed@bilboed.com>
36207 gst-libs/gst/riff/riff-media.c: Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
36208 Original commit message from CVS:
36209 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
36210 (gst_riff_create_video_template_caps):
36211 Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
36212 Also added the caps to the default set of riff video caps.
36214 2006-02-09 19:05:23 +0000 Andy Wingo <wingo@pobox.com>
36216 ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start time and the end time of the last packet in the page.
36217 Original commit message from CVS:
36218 2006-02-09 Andy Wingo <wingo@pobox.com>
36219 * ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start
36220 time and the end time of the last packet in the page.
36221 (gst_ogg_mux_pad_queue_page): In addition to setting the timestamp
36222 on the pages in our queue, set the duration as well. Reflow a
36224 (gst_ogg_mux_collected): Keep track of GstOggPad->timestamp_end.
36225 Fixes bad muxing order.
36227 2006-02-09 17:04:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36229 gst-libs/gst/rtp/gstbasertppayload.c: update seqnum before setting it on the packet; this makes sure that the timesta...
36230 Original commit message from CVS:
36231 * gst-libs/gst/rtp/gstbasertppayload.c:
36232 (gst_basertppayload_setcaps), (gst_basertppayload_push):
36233 update seqnum before setting it on the packet; this makes sure
36234 that the timestamp and seqnum properties match after pushing
36237 2006-02-09 12:16:35 +0000 Andy Wingo <wingo@pobox.com>
36241 Original commit message from CVS:
36244 2006-02-09 11:46:03 +0000 Andy Wingo <wingo@pobox.com>
36246 * gst-libs/gst/audio/gstringbuffer.c:
36247 * win32/common/config.h:
36249 Original commit message from CVS:
36252 2006-02-09 11:36:18 +0000 Andy Wingo <wingo@pobox.com>
36254 gst-libs/gst/audio/gstringbuffer.c
36255 Original commit message from CVS:
36256 2006-02-09 Andy Wingo <wingo@pobox.com>
36257 * gst-libs/gst/audio/gstringbuffer.c
36258 (gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
36259 overflow after 13.5 hours of recording. Kapow!
36260 * ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
36261 the buffer size -- we don't care about underrun/overrun reporting
36262 right now, just need to return a useful value.
36264 2006-02-09 11:21:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36266 configure.ac: Back to CVS
36267 Original commit message from CVS:
36271 === release 0.10.3 ===
36273 2006-02-09 11:18:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36279 * docs/plugins/inspect/plugin-adder.xml:
36280 * docs/plugins/inspect/plugin-alsa.xml:
36281 * docs/plugins/inspect/plugin-audioconvert.xml:
36282 * docs/plugins/inspect/plugin-audiorate.xml:
36283 * docs/plugins/inspect/plugin-audioresample.xml:
36284 * docs/plugins/inspect/plugin-audiotestsrc.xml:
36285 * docs/plugins/inspect/plugin-cdparanoia.xml:
36286 * docs/plugins/inspect/plugin-decodebin.xml:
36287 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
36288 * docs/plugins/inspect/plugin-gnomevfs.xml:
36289 * docs/plugins/inspect/plugin-libvisual.xml:
36290 * docs/plugins/inspect/plugin-ogg.xml:
36291 * docs/plugins/inspect/plugin-pango.xml:
36292 * docs/plugins/inspect/plugin-playbin.xml:
36293 * docs/plugins/inspect/plugin-subparse.xml:
36294 * docs/plugins/inspect/plugin-tcp.xml:
36295 * docs/plugins/inspect/plugin-theora.xml:
36296 * docs/plugins/inspect/plugin-typefindfunctions.xml:
36297 * docs/plugins/inspect/plugin-video4linux.xml:
36298 * docs/plugins/inspect/plugin-videorate.xml:
36299 * docs/plugins/inspect/plugin-videoscale.xml:
36300 * docs/plugins/inspect/plugin-videotestsrc.xml:
36301 * docs/plugins/inspect/plugin-volume.xml:
36302 * docs/plugins/inspect/plugin-vorbis.xml:
36303 * docs/plugins/inspect/plugin-ximagesink.xml:
36304 * docs/plugins/inspect/plugin-xvimagesink.xml:
36305 * win32/common/config.h:
36307 Original commit message from CVS:
36310 2006-02-08 18:37:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36312 configure.ac: Drat. Bump libtool version number for new API.
36313 Original commit message from CVS:
36315 Drat. Bump libtool version number for new API.
36316 Prelease 0.10.2.3 (of 0.10.3)
36318 2006-02-08 15:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36320 0.10.2.2 prerelease (of 0.10.3).
36321 Original commit message from CVS:
36323 * win32/common/config.h:
36324 0.10.2.2 prerelease (of 0.10.3).
36326 2006-02-08 15:50:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36328 gst/tcp/gsttcpclientsrc.c: Revert Andy's newsegment change pending a more correct fix.
36329 Original commit message from CVS:
36330 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
36331 Revert Andy's newsegment change pending a more correct
36334 2006-02-08 12:46:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36351 Original commit message from CVS:
36354 2006-02-08 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36356 * gst/tcp/gstmultifdsink.c:
36358 Original commit message from CVS:
36361 2006-02-08 09:20:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36363 gst/typefind/gsttypefindfunctions.c: detect more files as 3gp group and reorder the iso file formats
36364 Original commit message from CVS:
36366 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
36367 (qt_type_find), (plugin_init):
36368 detect more files as 3gp
36369 group and reorder the iso file formats
36371 2006-02-07 18:32:00 +0000 Tim-Philipp Müller <tim@centricular.net>
36373 ext/vorbis/vorbis.c: Register musicbrainz tags, so apps don't have to.
36374 Original commit message from CVS:
36375 * ext/vorbis/vorbis.c: (plugin_init):
36376 Register musicbrainz tags, so apps don't have to.
36378 2006-02-07 17:44:53 +0000 Tim-Philipp Müller <tim@centricular.net>
36380 gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
36381 Original commit message from CVS:
36382 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
36383 (gst_tag_to_vorbis_tag):
36384 Make sure we called gst_tag_register_musicbrainz_tags()
36385 before possibly mapping a vorbiscomment string from/to a
36388 2006-02-07 16:16:41 +0000 Tim-Philipp Müller <tim@centricular.net>
36390 gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
36391 Original commit message from CVS:
36392 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
36393 In case we can't find the required number of consecutive
36394 mpeg audio frames to positively identify an MPEG audio
36395 stream, check if there's at least a valid mpeg audio
36396 frame right at offset 0 and if so suggest mpeg/audio
36397 caps with a very low probability (#153004).
36399 2006-02-07 15:52:26 +0000 Andy Wingo <wingo@pobox.com>
36401 gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to a TIME segment if we get timestamped buffers. Requir...
36402 Original commit message from CVS:
36403 2006-02-07 Andy Wingo <wingo@pobox.com>
36404 * gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
36405 a TIME segment if we get timestamped buffers. Requires recent
36406 fixes in core to work properly.
36408 2006-02-07 14:57:46 +0000 Tim-Philipp Müller <tim@centricular.net>
36410 gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
36411 Original commit message from CVS:
36412 * gst/playback/gstplaybasebin.c: (prepare_output):
36413 Don't print the URI as part of the error message, it
36414 makes error dialogs look rather ugly, especially if
36415 the URI is very long or has characters in it that
36418 2006-02-07 13:11:31 +0000 Tim-Philipp Müller <tim@centricular.net>
36420 gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
36421 Original commit message from CVS:
36422 * gst/playback/gstplaybasebin.c: (prepare_output):
36423 Error out if we have only text or subtitles, but nothing
36424 else. Also error out if we have subtitles but no video
36427 2006-02-07 11:44:39 +0000 Tim-Philipp Müller <tim@centricular.net>
36429 ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
36430 Original commit message from CVS:
36431 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
36432 Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
36433 Post an error message on the bus when we encounter an
36434 error, which will hopefully be more meaningful than the
36435 'Internal Flow Error' message users get to see if we
36436 just return GST_FLOW_ERROR.
36438 2006-02-07 11:28:04 +0000 Andy Wingo <wingo@pobox.com>
36440 configure.ac (GST_MAJORMINOR): Update core version req to 0.10.2.2, for the collectpads API addition (#330244).
36441 Original commit message from CVS:
36442 2006-02-07 Andy Wingo <wingo@pobox.com>
36443 * configure.ac (GST_MAJORMINOR): Update core version req to
36444 0.10.2.2, for the collectpads API addition (#330244).
36446 2006-02-06 19:09:26 +0000 Tim-Philipp Müller <tim@centricular.net>
36448 ext/gnomevfs/gstgnomevfs.c: Return FALSE from plugin_init() when GnomeVFS can't be initialised for some reason (#3284...
36449 Original commit message from CVS:
36450 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
36451 Return FALSE from plugin_init() when GnomeVFS can't
36452 be initialised for some reason (#328423).
36454 2006-02-06 13:26:54 +0000 Julien Moutte <julien@moutte.net>
36456 ext/pango/gsttextoverlay.c: Stick to seeking theory until i find the bug.
36457 Original commit message from CVS:
36458 2006-02-06 Julien MOUTTE <julien@moutte.net>
36459 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
36460 Stick to seeking theory until i find the bug.
36461 * gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
36463 2006-02-06 12:38:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36465 Make theoraenc and the tests leak free. Like, really.
36466 Original commit message from CVS:
36467 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
36468 (theora_enc_finalize), (theora_enc_sink_setcaps),
36469 (theora_set_header_on_caps), (theora_enc_chain),
36470 (theora_enc_change_state):
36471 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
36472 Make theoraenc and the tests leak free. Like, really.
36474 2006-02-05 23:31:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36476 Add a finalize method to ensure we clean up state even if someone omitted the state change back to NULL.
36477 Original commit message from CVS:
36478 (theora_enc_finalize), (theora_enc_sink_setcaps):
36479 Add a finalize method to ensure we clean up state even if
36480 someone omitted the state change back to NULL.
36481 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
36482 (gst_vorbisenc_chain):
36483 Free some more leaked bits.
36484 * tests/check/pipelines/theoraenc.c: (start_pipeline),
36486 Wait for state changes to happen if they're ASYNC.
36487 This ought to teach those fancy pants buildbots a lesson.
36489 2006-02-05 22:47:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36491 gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
36492 Original commit message from CVS:
36493 * gst-libs/gst/tag/gstid3tag.c:
36494 Add mapping for ID3 International Standard Recording Code
36497 2006-02-05 22:44:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36499 ext/vorbis/vorbisenc.c: Don't leak tag names.
36500 Original commit message from CVS:
36501 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
36502 Don't leak tag names.
36504 2006-02-05 18:22:01 +0000 Tim-Philipp Müller <tim@centricular.net>
36506 Split libgsttag docs into multiple sections.
36507 Original commit message from CVS:
36508 * docs/libs/gst-plugins-base-libs-docs.sgml:
36509 * docs/libs/gst-plugins-base-libs-sections.txt:
36510 * gst-libs/gst/tag/gstid3tag.c:
36511 * gst-libs/gst/tag/gstvorbistag.c:
36512 * gst-libs/gst/tag/tags.c:
36513 Split libgsttag docs into multiple sections.
36515 2006-02-05 18:01:33 +0000 Tim-Philipp Müller <tim@centricular.net>
36517 Add libgsttag to the docs.
36518 Original commit message from CVS:
36519 * docs/libs/Makefile.am:
36520 * docs/libs/gst-plugins-base-libs-docs.sgml:
36521 * docs/libs/gst-plugins-base-libs-sections.txt:
36522 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
36523 * gst-libs/gst/tag/gstvorbistag.c:
36524 * gst-libs/gst/tag/tag.h:
36525 * gst-libs/gst/tag/tags.c:
36526 Add libgsttag to the docs.
36528 2006-02-05 17:21:23 +0000 Julien Moutte <julien@moutte.net>
36530 ext/pango/gsttextoverlay.c: Fix clockoverlay.
36531 Original commit message from CVS:
36532 2006-02-05 Julien MOUTTE <julien@moutte.net>
36533 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
36534 (gst_text_overlay_init), (gst_text_overlay_src_event),
36535 (gst_text_overlay_collected): Fix clockoverlay.
36537 2006-02-05 17:15:17 +0000 Tim-Philipp Müller <tim@centricular.net>
36539 docs/libs/compiling.sgml: Fix typo: it's pkg-config, not pkg-gconfig
36540 Original commit message from CVS:
36541 * docs/libs/compiling.sgml:
36542 Fix typo: it's pkg-config, not pkg-gconfig
36543 * docs/libs/gst-plugins-base-libs-docs.sgml:
36544 * docs/libs/gst-plugins-base-libs-sections.txt:
36545 * docs/libs/tmpl/gstgconf.sgml:
36546 There is no libgstgconf in 0.10, remove it
36549 2006-02-05 16:03:48 +0000 Julien Moutte <julien@moutte.net>
36551 docs/libs/tmpl/gstcolorbalance.sgml: Updated.
36552 Original commit message from CVS:
36553 2006-02-05 Julien MOUTTE <julien@moutte.net>
36554 * docs/libs/tmpl/gstcolorbalance.sgml: Updated.
36555 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
36556 (gst_text_overlay_src_event), (gst_text_overlay_collected):
36557 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
36558 (gst_sub_parse_class_init), (gst_sub_parse_init),
36559 (gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
36560 (parse_mpsub), (parser_state_init), (handle_buffer),
36561 (gst_sub_parse_chain), (gst_sub_parse_sink_event),
36563 * gst/subparse/gstsubparse.h: Introduce seeking code.
36565 2006-02-05 15:14:06 +0000 Tim-Philipp Müller <tim@centricular.net>
36567 gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
36568 Original commit message from CVS:
36569 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
36570 Add comment about LANGUAGE tag inconsistency (we want
36571 ISO-639-1, but extract three-letter identifiers?)
36573 Add two translatable files.
36575 2006-02-05 14:59:28 +0000 Tim-Philipp Müller <tim@centricular.net>
36577 gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
36578 Original commit message from CVS:
36579 * gst-libs/gst/tag/Makefile.am:
36580 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
36581 * gst-libs/gst/tag/tag.h:
36582 * gst-libs/gst/tag/tags.c:
36583 (gst_tag_register_musicbrainz_tags_internal),
36584 (gst_tag_register_musicbrainz_tags):
36585 Forward-port some tags stuff from the 0.8 branch. This is
36586 mostly the addition of musicbrainz tags and their mapping
36587 to vorbistags, and a vorbistag mapping of the language tag.
36589 2006-02-05 12:06:25 +0000 Julien Moutte <julien@moutte.net>
36591 gst/playback/gstplaybin.c: Fix broken code refactoring.
36592 Original commit message from CVS:
36593 2006-02-05 Julien MOUTTE <julien@moutte.net>
36594 * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
36597 2006-02-05 03:05:41 +0000 David Schleef <ds@schleef.org>
36599 Add Dirac typefinding and add dirac format to oggmux.
36600 Original commit message from CVS:
36601 * ext/ogg/gstoggmux.c:
36602 * gst/typefind/gsttypefindfunctions.c:
36603 Add Dirac typefinding and add dirac format to oggmux.
36605 2006-02-04 07:49:03 +0000 Michael Smith <msmith@xiph.org>
36608 Improve error message for liboil missingness.
36609 Original commit message from CVS:
36610 Improve error message for liboil missingness.
36612 2006-02-03 19:23:41 +0000 Tim-Philipp Müller <tim@centricular.net>
36614 gst/playback/gstdecodebin.c: Don't put essential function call into g_return_*() macro, otherwise it'll all be replac...
36615 Original commit message from CVS:
36616 * gst/playback/gstdecodebin.c: (try_to_link_1):
36617 Don't put essential function call into
36618 g_return_*() macro, otherwise it'll all be
36619 replaced by NOOPs when compiling with
36620 G_DISABLE_CHECKS defined.
36622 2006-02-03 17:45:44 +0000 Edgard Lima <edgard.lima@indt.org.br>
36625 * ext/ogg/gstoggdemux.c:
36626 * ext/ogg/gstoggparse.c:
36627 * gst/tcp/gsttcpserversink.c:
36628 * sys/v4l/v4lsrc_calls.c:
36629 * sys/v4l/v4lsrc_calls.h:
36630 Just make it compile with --disable-gst-debug.
36631 Original commit message from CVS:
36632 Just make it compile with --disable-gst-debug.
36634 2006-02-03 12:51:47 +0000 Wim Taymans <wim.taymans@gmail.com>
36636 ext/alsa/gstalsasink.*: Add lock to protect alsa calls.
36637 Original commit message from CVS:
36638 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
36639 (gst_alsasink_class_init), (gst_alsasink_init),
36640 (gst_alsasink_write), (gst_alsasink_reset):
36641 * ext/alsa/gstalsasink.h:
36642 Add lock to protect alsa calls.
36643 Implement reset to flush samples ASAP, does not work
36646 2006-02-02 18:18:31 +0000 Wim Taymans <wim.taymans@gmail.com>
36648 gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
36649 Original commit message from CVS:
36650 * gst-libs/gst/audio/gstbaseaudiosink.c:
36651 (gst_base_audio_sink_provide_clock):
36652 Ugh.. getting late I guess...
36654 2006-02-02 18:13:26 +0000 Wim Taymans <wim.taymans@gmail.com>
36656 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
36657 Original commit message from CVS:
36658 * gst-libs/gst/audio/gstbaseaudiosink.c:
36659 (gst_base_audio_sink_provide_clock),
36660 (gst_base_audio_sink_set_property),
36661 (gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
36662 Don't try to provide a clock when we are not negotiated since
36663 we might not be able to make it run.
36665 2006-02-02 17:51:48 +0000 Tim-Philipp Müller <tim@centricular.net>
36667 gst/playback/gstdecodebin.c: Unlinking two source pads is ... hard.
36668 Original commit message from CVS:
36669 * gst/playback/gstdecodebin.c: (try_to_link_1):
36670 Unlinking two source pads is ... hard.
36672 2006-02-02 12:14:35 +0000 Wim Taymans <wim.taymans@gmail.com>
36674 gst-libs/gst/audio/TODO: Updated.
36675 Original commit message from CVS:
36676 * gst-libs/gst/audio/TODO:
36678 * gst-libs/gst/audio/gstbaseaudiosink.c:
36679 (gst_base_audio_sink_drain), (gst_base_audio_sink_event):
36680 On EOS, wait till the last sample is played before posting EOS.
36682 2006-02-02 08:53:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36684 * tests/check/pipelines/theoraenc.c:
36685 comment on my understanding
36686 Original commit message from CVS:
36687 comment on my understanding
36689 2006-02-02 08:47:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36692 * tests/check/pipelines/theoraenc.c:
36693 reformat to fit 80 chars
36694 Original commit message from CVS:
36695 reformat to fit 80 chars
36697 2006-02-02 00:04:37 +0000 Kai Vehmanen <kv2004@eca.cx>
36699 gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
36700 Original commit message from CVS:
36701 2006-02-01 Philippe Kalaf <burger at speedy dot org>
36702 * gst-libs/gst/rtp/gstbasertpdepayload.c:
36703 Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
36704 setting queue_delay to zero. Also avoid thread being started if
36705 queue_delay is zero.
36707 2006-02-01 14:51:29 +0000 Tim-Philipp Müller <tim@centricular.net>
36709 gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
36710 Original commit message from CVS:
36711 * gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
36712 Make test work again by connecting fakesinks to each decoded pad,
36713 which makes the pipeline wait until each fakesink has a buffer
36714 queued before going to PAUSED state. At that point we know the
36715 decodebin pads are negotiated.
36717 2006-02-01 11:59:47 +0000 Tim-Philipp Müller <tim@centricular.net>
36719 gst/: Pass unhandled queries to the parent class's query function.
36720 Original commit message from CVS:
36721 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
36722 (gst_cdda_base_src_handle_event):
36723 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
36724 Pass unhandled queries to the parent class's query function.
36726 2006-02-01 11:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
36728 Pass unhandled queries upstream instead of just dropping them (#326447). Also, fix supported query types list for som...
36729 Original commit message from CVS:
36730 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
36731 (gst_ogg_pad_src_query):
36732 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
36733 * ext/theora/theoradec.c: (theora_dec_src_query),
36734 (theora_dec_sink_query):
36735 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
36736 (vorbis_dec_sink_query):
36737 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
36738 (gst_vorbisenc_sink_query):
36739 * gst/adder/gstadder.c: (gst_adder_query):
36740 Pass unhandled queries upstream instead of just
36741 dropping them (#326447). Also, fix supported
36742 query types list for some elements.
36744 2006-02-01 09:58:15 +0000 Tim-Philipp Müller <tim@centricular.net>
36746 gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
36747 Original commit message from CVS:
36748 * gst/typefind/gsttypefindfunctions.c: (au_type_find),
36749 (paris_type_find), (ilbc_type_find), (plugin_init):
36750 Fix typefinding for audio/x-au, audio/x-paris and
36751 audio/iLBC-sh. We cannot use the START_WITH macros
36752 here, because there can only be one typefind factory
36753 with the same name (caps), so the second one would
36754 replace the first one and the first one would never
36755 be called when doing typefinding (see #161712).
36757 2006-01-31 19:25:10 +0000 Wim Taymans <wim.taymans@gmail.com>
36759 ext/vorbis/vorbisdec.c: Use scale_int when we can, add some more scaling.
36760 Original commit message from CVS:
36761 * ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
36762 (vorbis_handle_header_packet), (vorbis_dec_push),
36763 (vorbis_handle_data_packet):
36764 Use scale_int when we can, add some more scaling.
36765 Check packettype before parsing it.
36767 2006-01-31 17:44:35 +0000 Wim Taymans <wim.taymans@gmail.com>
36769 ext/theora/theoradec.c: Call right _scale functions.
36770 Original commit message from CVS:
36771 * ext/theora/theoradec.c: (_theora_granule_time),
36772 (theora_dec_src_convert), (theora_dec_sink_convert):
36773 Call right _scale functions.
36774 Use parameter instead of some other random value.
36776 2006-01-31 17:27:00 +0000 Wim Taymans <wim.taymans@gmail.com>
36778 ext/theora/theoradec.c: Use higher precision timestamps calculation.
36779 Original commit message from CVS:
36780 * ext/theora/theoradec.c: (_theora_granule_frame),
36781 (_theora_granule_time), (_inc_granulepos),
36782 (theora_dec_src_convert), (theora_dec_sink_convert),
36783 (theora_handle_type_packet), (theora_handle_data_packet),
36784 (theora_dec_chain):
36785 Use higher precision timestamps calculation.
36786 Convert some other conversions to _scale.
36788 2006-01-31 17:19:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
36790 gst/: initialize gst_controller before using
36791 Original commit message from CVS:
36792 * gst/audiotestsrc/gstaudiotestsrc.c:
36793 (gst_audio_test_src_create_sine_table), (plugin_init):
36794 * gst/volume/gstvolume.c: (plugin_init):
36795 initialize gst_controller before using
36797 2006-01-31 16:26:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
36799 tests/check/pipelines/: Define constant using G_GINT64_CONSTANT to avoid errors when passing it around - otherwise it...
36800 Original commit message from CVS:
36801 * tests/check/pipelines/theoraenc.c:
36802 * tests/check/pipelines/vorbisenc.c:
36803 Define constant using G_GINT64_CONSTANT to avoid errors when
36804 passing it around - otherwise it gets truncated to 32 bits.
36805 Fixes failing tests.
36807 2006-01-31 15:36:13 +0000 Andy Wingo <wingo@pobox.com>
36809 sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the caps being set doesn't have a framerate value. Basic...
36810 Original commit message from CVS:
36811 2006-01-31 Andy Wingo <wingo@pobox.com>
36812 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
36813 caps being set doesn't have a framerate value. Basically a stopgap
36815 * ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
36816 technically correct enough to put into core though.
36817 (gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
36818 DURATION. Fixes theoraenc ! oggmux.
36819 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
36820 fraction, not double.
36822 2006-01-31 12:23:35 +0000 Christian Schaller <uraeus@gnome.org>
36824 * gst-plugins-base.spec.in:
36825 update with latest files
36826 Original commit message from CVS:
36827 update with latest files
36829 2006-01-30 23:42:54 +0000 Sébastien Moutte <sebastien@moutte.net>
36831 win32/vs7: add vs7 project files created by Sergey Scobich
36832 Original commit message from CVS:
36834 add vs7 project files created by Sergey Scobich
36836 2006-01-30 22:18:53 +0000 Sébastien Moutte <sebastien@moutte.net>
36838 win32/vs8: add vs8 project files created by Sergey Scobich
36839 Original commit message from CVS:
36841 add vs8 project files created by Sergey Scobich
36843 2006-01-30 19:22:22 +0000 Andy Wingo <wingo@pobox.com>
36845 ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare timestamp + duration, not just timestamp -- ogg pages should ...
36846 Original commit message from CVS:
36847 2006-01-30 Andy Wingo <wingo@pobox.com>
36848 * ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
36849 timestamp + duration, not just timestamp -- ogg pages should be
36850 ordered by stop time. Necessary fix given the change in vorbis
36853 2006-01-30 19:21:07 +0000 Andy Wingo <wingo@pobox.com>
36856 * ext/theora/gsttheoraenc.h:
36857 * ext/theora/theoraenc.c:
36858 * tests/check/pipelines/theoraenc.c:
36859 ext/theora/theoraenc.c (theora_enc_sink_setcaps)
36860 Original commit message from CVS:
36861 2006-01-30 Andy Wingo <wingo@pobox.com>
36862 * ext/theora/theoraenc.c (theora_enc_sink_setcaps)
36863 (gst_theora_enc_init): Pull the granule shift out of the encoder.
36864 (granulepos_add): New function, handles the messiness of adjusting
36866 (theora_buffer_from_packet):
36867 (theora_enc_chain):
36868 (theora_enc_sink_event): Use granulepos_add, not +.
36869 * tests/check/pipelines/theoraenc.c
36870 (check_buffer_granulepos_from_starttime): Just check the frame
36871 count, not the actual granulepos -- we can't dictate to the
36872 encoder when it should be placing keyframes.
36874 2006-01-30 18:17:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36876 ext/gnomevfs/gstgnomevfssrc.c: SERVICE_NOT_AVAILABLE happens for example when you're trying to play an http:// stream...
36877 Original commit message from CVS:
36878 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
36879 SERVICE_NOT_AVAILABLE happens for example when you're trying to
36880 play an http:// stream from a server that's not serving
36882 2006-01-30 17:08:11 +0000 Andy Wingo <wingo@pobox.com>
36884 tests/check/pipelines/: Totally remove the UINT64_CONSTANT macro, doesn't appear to be needed or available.
36885 Original commit message from CVS:
36886 2006-01-30 Andy Wingo <wingo@pobox.com>
36887 * tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
36888 * tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
36889 remove the UINT64_CONSTANT macro, doesn't appear to be needed or
36892 2006-01-30 17:01:54 +0000 Andy Wingo <wingo@pobox.com>
36894 ext/theora/: Same changes as were done to vorbisenc, although theoraenc was timestamping correctly. Added handling of...
36895 Original commit message from CVS:
36896 2006-01-30 Andy Wingo <wingo@pobox.com>
36897 * ext/theora/gsttheoraenc.h:
36898 * ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
36899 although theoraenc was timestamping correctly. Added handling of
36900 streams that start with nonzero timestamps.
36901 * tests/check/Makefile.am:
36902 * tests/check/pipelines/theoraenc.c: New file, basically does same
36903 tests as vorbisenc.
36904 * tests/check/pipelines/vorbisenc.c: I claim these bugs.
36906 2006-01-30 16:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
36908 gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
36909 Original commit message from CVS:
36910 * gst-libs/gst/audio/gstaudiosink.c:
36911 (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
36912 (gst_audioringbuffer_pause):
36913 Implement pause that does not wait for completion.
36914 * gst-libs/gst/audio/gstbaseaudiosink.c:
36915 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
36916 Don't drop buffers when going to PAUSED but perform preroll on
36917 remaining samples now that core base class supports this.
36918 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
36919 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
36920 (gst_ring_buffer_commit):
36921 Pause should not signal waiters.
36922 Implement return value of _commit correctly.
36924 2006-01-30 15:01:28 +0000 Andy Wingo <wingo@pobox.com>
36926 tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
36927 Original commit message from CVS:
36928 2006-01-30 Andy Wingo <wingo@pobox.com>
36929 * tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
36930 * ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
36931 updated to timestamp from the first sample, not the last.
36932 (gst_vorbisenc_buffer_from_header_packet): New function, takes
36933 special care of granulepos and timestamp for header packets.
36934 (gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
36935 when the first buffer has a nonzero timestamp.
36936 * ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
36937 (GstVorbisEnc.subgranule_offset): New members. Take care of the
36938 case when the first audio buffer we get has a nonzero timestamp.
36939 (GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
36940 properly timestamp vorbis buffers with the time of the first
36941 sample, not the last.
36942 * ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
36943 vorbis_granule_time_copy -- now it takes the granule/subgranule
36944 offset into account.
36945 * tests/check/pipelines/vorbisenc.c: New test for correctness of
36946 timestamps, durations, and granulepos on buffers produced by
36949 2006-01-30 14:42:28 +0000 Eric Jonas <jonas@mit.edu>
36951 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Patch from Eric Jonas to support conversions to/from UYVY (Fixes: #324626)
36952 Original commit message from CVS:
36953 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
36954 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
36955 Patch from Eric Jonas to support conversions to/from UYVY
36958 2006-01-30 08:11:14 +0000 Julien Moutte <julien@moutte.net>
36960 gst/playback/: Implement subtitles.
36961 Original commit message from CVS:
36962 2006-01-30 Julien MOUTTE <julien@moutte.net>
36963 * gst/playback/gstplaybasebin.c: (group_commit),
36965 (setup_subtitle), (setup_source), (set_active_source):
36966 * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
36967 (gen_text_element), (gen_audio_element), (gen_vis_element),
36968 (remove_sinks), (add_sink), (setup_sinks): Implement subtitles.
36970 2006-01-29 19:13:39 +0000 Sébastien Moutte <sebastien@moutte.net>
36972 gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
36973 Original commit message from CVS:
36974 * gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
36975 * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
36976 use of gst_guint64_to_gdouble to be compliant with vs6
36977 * gst/playback/gstdecodebin.c: (try_to_link_1)
36978 * gst/videorate/videorate.c: (gst_video_rate_blank_data)
36979 use of G_GINT64_CONSTANT for int64 constants
36980 * win32/common/libgstinterfaces.def:
36981 export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
36983 update and add new project files
36985 2006-01-29 18:21:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
36987 add a win32-update rule like in core, and copy over enumtypes files
36988 Original commit message from CVS:
36991 * win32/common/interfaces-enumtypes.c:
36992 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
36993 (gst_mixer_track_flags_get_type),
36994 (gst_tuner_channel_flags_get_type):
36995 * win32/common/interfaces-enumtypes.h:
36996 * win32/common/multichannel-enumtypes.c:
36997 (gst_audio_channel_position_get_type):
36998 * win32/common/multichannel-enumtypes.h:
36999 add a win32-update rule like in core, and copy over enumtypes files
37001 2006-01-29 18:07:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37004 generate win32/common/config.h
37005 Original commit message from CVS:
37006 generate win32/common/config.h
37008 2006-01-29 18:05:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37010 win32/: add config files just like in core
37011 Original commit message from CVS:
37013 * win32/common/config.h:
37014 * win32/common/config.h.in:
37015 add config files just like in core
37017 2006-01-28 18:22:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37019 ext/alsa/: Update all error messages. All of them should either use the default translated message, or actually prov...
37020 Original commit message from CVS:
37021 * ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
37022 (set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
37023 (gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
37024 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
37025 (set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
37026 (gst_alsasrc_unprepare), (gst_alsasrc_read):
37027 Update all error messages. All of them should either use
37028 the default translated message, or actually provide a
37029 translatable string.
37030 Make the string for channel count problems meaningful.
37032 2006-01-28 18:19:18 +0000 Tim-Philipp Müller <tim@centricular.net>
37034 gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
37035 Original commit message from CVS:
37036 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
37037 Make gcc-4.1 happy (part of #327357).
37039 2006-01-28 16:35:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37041 sys/v4l/v4l_calls.c: check for and throw RESOURCE_BUSY
37042 Original commit message from CVS:
37043 * sys/v4l/v4l_calls.c: (gst_v4l_open):
37044 check for and throw RESOURCE_BUSY
37046 2006-01-28 02:13:14 +0000 David Schleef <ds@schleef.org>
37048 gst/videoscale/vs_scanline.c: Oops, *that's* why I never checked in this change -- it requires liboil features not in...
37049 Original commit message from CVS:
37050 * gst/videoscale/vs_scanline.c: Oops, *that's* why I never
37051 checked in this change -- it requires liboil features not
37052 in 0.3.6. Revert parts.
37054 2006-01-27 23:40:19 +0000 David Schleef <ds@schleef.org>
37056 update liboil requirement to 0.3.6
37057 Original commit message from CVS:
37059 * configure.ac: update liboil requirement to 0.3.6
37060 * gst/videoscale/Makefile.am:
37061 * gst/videoscale/vs_scanline.c: liboilify
37063 2006-01-27 17:00:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
37065 ext/libvisual/visual.c: When pad_alloc returns a GstFlowReturn other than GST_FLOW_OK, make sure it is passed upstream.
37066 Original commit message from CVS:
37067 * ext/libvisual/visual.c: (get_buffer):
37068 When pad_alloc returns a GstFlowReturn other
37069 than GST_FLOW_OK, make sure it is passed upstream.
37071 2006-01-27 01:36:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
37073 ext/alsa/gstalsasink.c: Free the device name string.
37074 Original commit message from CVS:
37075 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
37076 (gst_alsasink_class_init):
37077 Free the device name string.
37078 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
37079 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
37080 (gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
37081 Don't remove a pad from the collectpads structure until it
37082 is released - it's a request pad, and may receive data again
37083 if the element gets moved back to PLAYING state.
37084 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
37085 Ensure we turn on double buffering on the Xv port, and
37086 set the colour key to something dark and mysterious that
37089 2006-01-27 01:06:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37091 ext/: - a library should not call setlocale. see Libraries node in gettext manual
37092 Original commit message from CVS:
37093 * ext/alsa/gstalsaplugin.c: (plugin_init):
37094 * ext/cdparanoia/gstcdparanoiasrc.c:
37095 (gst_cd_paranoia_src_base_init), (plugin_init):
37096 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
37097 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
37098 - a library should not call setlocale. see Libraries node in
37100 - make sure all plugins that use translation do bindtextdomain
37101 to point to the localedir
37102 * gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
37103 (setup_sinks), (plugin_init):
37104 all this, and check for NULL when creating sinks
37106 2006-01-26 23:21:31 +0000 Julien Moutte <julien@moutte.net>
37108 gst/subparse/gstsubparse.c: Make typefinding of subtitles work again.
37109 Original commit message from CVS:
37110 2006-01-27 Julien MOUTTE <julien@moutte.net>
37111 * gst/subparse/gstsubparse.c: (gst_subparse_type_find),
37112 (plugin_init): Make typefinding of subtitles work again.
37114 2006-01-26 20:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
37116 gst/typefind/gsttypefindfunctions.c: Backport a bunch of typefinding fixes from the 0.8 branch.
37117 Original commit message from CVS:
37118 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
37119 (mp3_type_frame_length_from_header), (mp3_type_find),
37120 (wavpack_type_find), (m4a_type_find), (ircam_type_find),
37122 Backport a bunch of typefinding fixes from the 0.8 branch.
37123 Also, improve wavpack typefinding: if we can't peek the
37124 entire wavpack block, try to parse the bits we can get and
37125 see if we find what we're looking for in those.
37127 2006-01-26 19:17:38 +0000 Julien Moutte <julien@moutte.net>
37129 sys/: Handle some more cases of pixel aspect ratio.
37130 Original commit message from CVS:
37131 2006-01-26 Julien MOUTTE <julien@moutte.net>
37132 * sys/ximage/ximagesink.c:
37133 (gst_ximagesink_calculate_pixel_aspect_ratio):
37134 * sys/xvimage/xvimagesink.c:
37135 (gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
37136 more cases of pixel aspect ratio.
37138 2006-01-26 13:09:24 +0000 Edward Hervey <bilboed@bilboed.com>
37140 gst/playback/gstdecodebin.c: Also consider the flush-start and tag events as unblockers for the pad probes.
37141 Original commit message from CVS:
37142 * gst/playback/gstdecodebin.c: (pad_probe):
37143 Also consider the flush-start and tag events as unblockers
37144 for the pad probes.
37146 2006-01-26 12:32:58 +0000 Julien Moutte <julien@moutte.net>
37148 gst/playback/gstplaybin.c: On the fly visualisation switch, works disabling, enabling as well but it won't be able to...
37149 Original commit message from CVS:
37150 2006-01-26 Julien MOUTTE <julien@moutte.net>
37151 * gst/playback/gstplaybin.c: (gst_play_bin_init),
37152 (gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
37153 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
37154 On the fly visualisation switch, works disabling, enabling as
37155 well but it won't be able to enable vis in a playbin that was
37156 created with no visualisation.
37158 2006-01-25 10:50:32 +0000 Wim Taymans <wim.taymans@gmail.com>
37160 gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
37161 Original commit message from CVS:
37162 * gst-libs/gst/audio/gstbaseaudiosink.c:
37163 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
37164 Undo previous commit, it breaks resume after pause.
37166 2006-01-25 09:27:01 +0000 Wim Taymans <wim.taymans@gmail.com>
37168 gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
37169 Original commit message from CVS:
37170 * gst-libs/gst/audio/gstbaseaudiosink.c:
37171 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
37172 (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
37174 Post error when caps cannot be parsed.
37175 Resync on discontinuity in the stream.
37176 Clip samples to segment boundaries.
37177 return WRONG_STATE sooner when we are flushing.
37178 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
37179 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
37180 Make audiosrc operate in TIME.
37181 Set TIMESTAMP and DURATION on buffers.
37183 2006-01-24 21:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
37185 tests/examples/seek/seek.c: Output tag messages as well.
37186 Original commit message from CVS:
37187 * tests/examples/seek/seek.c: (main):
37188 Output tag messages as well.
37190 2006-01-23 15:05:24 +0000 Edward Hervey <bilboed@bilboed.com>
37192 gst/playback/gstdecodebin.c: Replace GstPadBlockCallback with pad probes that detect first buffer AND eos before remo...
37193 Original commit message from CVS:
37194 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
37195 (free_pad_probes), (remove_fakesink), (pad_probe),
37196 (close_pad_link), (gst_decode_bin_change_state):
37197 Replace GstPadBlockCallback with pad probes that detect
37198 first buffer AND eos before removing fakesink.
37199 Fixes hang with demuxers doing EOS while pre-rolling.
37202 2006-01-23 10:10:36 +0000 Jens Granseuer <jensgr@gmx.net>
37204 GCC 2.95 fixes (#328263).
37205 Original commit message from CVS:
37206 2006-01-23 Andy Wingo <wingo@pobox.com>
37207 * ext/alsa/gstalsasink.c:
37208 * gst-libs/gst/rtp/gstbasertpdepayload.c:
37209 (gst_base_rtp_depayload_setcaps),
37210 (gst_base_rtp_depayload_add_to_queue),
37211 (gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
37212 Patch by: Jens Granseuer <jensgr at gmx dot net>
37214 2006-01-22 17:24:02 +0000 Julien Moutte <julien@moutte.net>
37216 sys/: Playbin keeps some ref to some frames. We might get a frame destroyed after changing state to
37217 Original commit message from CVS:
37218 2006-01-22 Julien MOUTTE <julien@moutte.net>
37219 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
37220 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
37221 (gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
37222 frames. We might get a frame destroyed after changing state to
37223 NULL, adding a safety check on xcontext.
37225 2006-01-22 14:50:53 +0000 Tim-Philipp Müller <tim@centricular.net>
37227 gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
37228 Original commit message from CVS:
37229 * gst-libs/gst/interfaces/xoverlay.c:
37230 Fix prepare-xwindow-id code example in the docs - we need to
37231 ignore all messages that aren't element messages as well.
37233 2006-01-21 22:40:03 +0000 Julien Moutte <julien@moutte.net>
37235 sys/xvimage/xvimagesink.c: I think one day i'll completely undestand how caps negotiation is supposed to work. This r...
37236 Original commit message from CVS:
37237 2006-01-21 Julien MOUTTE <julien@moutte.net>
37238 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
37239 I think one day i'll completely undestand how caps negotiation
37240 is supposed to work. This refactoring handles buffer_alloc
37241 called with caps we can't handle. We definitely don't want a
37242 set_caps with those caps, so we define and allocate a buffer
37243 we would like to receive.
37245 2006-01-20 19:10:26 +0000 Christian Schaller <uraeus@gnome.org>
37249 up automake requirement to 1.7
37250 Original commit message from CVS:
37251 up automake requirement to 1.7
37253 2006-01-19 10:59:51 +0000 Tim-Philipp Müller <tim@centricular.net>
37255 gst/playback/gstplaybasebin.c: Free iterator when done.
37256 Original commit message from CVS:
37257 * gst/playback/gstplaybasebin.c: (setup_source):
37258 Free iterator when done.
37260 2006-01-17 11:43:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
37262 gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
37263 Original commit message from CVS:
37264 * gst-libs/gst/audio/gstbaseaudiosink.c:
37265 (gst_base_audio_sink_render):
37266 Fix playback of non-synchronised streams by assuming a rate
37267 of 1.0 instead of a random one.
37268 Makes this work again:
37269 gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
37270 endianness=(int)4321, signed=(boolean)true, width=(int)16,
37271 depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
37272 audioresample ! alsasink
37274 2006-01-16 21:01:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37278 Original commit message from CVS:
37281 === release 0.10.2 ===
37283 2006-01-16 20:59:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37289 * docs/plugins/gst-plugins-base-plugins.args:
37290 * docs/plugins/inspect/plugin-adder.xml:
37291 * docs/plugins/inspect/plugin-alsa.xml:
37292 * docs/plugins/inspect/plugin-audioconvert.xml:
37293 * docs/plugins/inspect/plugin-audiorate.xml:
37294 * docs/plugins/inspect/plugin-audioresample.xml:
37295 * docs/plugins/inspect/plugin-audiotestsrc.xml:
37296 * docs/plugins/inspect/plugin-cdparanoia.xml:
37297 * docs/plugins/inspect/plugin-decodebin.xml:
37298 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
37299 * docs/plugins/inspect/plugin-gnomevfs.xml:
37300 * docs/plugins/inspect/plugin-libvisual.xml:
37301 * docs/plugins/inspect/plugin-ogg.xml:
37302 * docs/plugins/inspect/plugin-pango.xml:
37303 * docs/plugins/inspect/plugin-playbin.xml:
37304 * docs/plugins/inspect/plugin-subparse.xml:
37305 * docs/plugins/inspect/plugin-tcp.xml:
37306 * docs/plugins/inspect/plugin-theora.xml:
37307 * docs/plugins/inspect/plugin-typefindfunctions.xml:
37308 * docs/plugins/inspect/plugin-video4linux.xml:
37309 * docs/plugins/inspect/plugin-videorate.xml:
37310 * docs/plugins/inspect/plugin-videoscale.xml:
37311 * docs/plugins/inspect/plugin-videotestsrc.xml:
37312 * docs/plugins/inspect/plugin-volume.xml:
37313 * docs/plugins/inspect/plugin-vorbis.xml:
37314 * docs/plugins/inspect/plugin-ximagesink.xml:
37315 * docs/plugins/inspect/plugin-xvimagesink.xml:
37317 Original commit message from CVS:
37320 2006-01-16 16:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
37322 gst/playback/: Comment out broken code that connects to the state-changed signal.
37323 Original commit message from CVS:
37324 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
37325 * gst/playback/gststreamselector.c:
37326 (gst_stream_selector_set_property):
37327 Comment out broken code that connects to the state-changed signal.
37328 At this point, changing current stream selection is broken, but
37329 stuff like gst-launch playbin current-audio=1 works and filters
37330 to the chosen stream.
37332 2006-01-16 15:31:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37334 ext/vorbis/vorbisdec.c: Fix #327216 (null dereference in vorbisdec)
37335 Original commit message from CVS:
37336 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
37337 Fix #327216 (null dereference in vorbisdec)
37339 2006-01-16 15:19:55 +0000 Tim-Philipp Müller <tim@centricular.net>
37341 ext/theora/theoradec.c: Post taglist actually on bus instead of just freeing it (fixes #327114 and totem bug #327080).
37342 Original commit message from CVS:
37343 * ext/theora/theoradec.c: (theora_handle_comment_packet):
37344 Post taglist actually on bus instead of just freeing it
37345 (fixes #327114 and totem bug #327080).
37346 * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
37347 Use gst_element_found_tags_for_pad(), so that the tags
37348 are sent downstream as an event as well.
37350 2006-01-15 10:06:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37352 sys/: move all regularly occurring messages to GST_LOG level add some more object logs
37353 Original commit message from CVS:
37354 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
37355 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
37356 (gst_ximagesink_buffer_alloc):
37357 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
37358 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
37359 (gst_xvimagesink_buffer_alloc):
37360 move all regularly occurring messages to GST_LOG level
37361 add some more object logs
37363 2006-01-14 22:59:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37381 Original commit message from CVS:
37384 2006-01-14 20:46:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37386 ext/ogg/gstoggmux.c: fix a silly segfault
37387 Original commit message from CVS:
37388 2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
37389 * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
37390 fix a silly segfault
37392 2006-01-14 12:52:22 +0000 Tim-Philipp Müller <tim@centricular.net>
37394 Add docs for mixerutils stuff.
37395 Original commit message from CVS:
37396 * docs/libs/gst-plugins-base-libs-docs.sgml:
37397 * docs/libs/gst-plugins-base-libs-sections.txt:
37398 * gst-libs/gst/audio/mixerutils.c:
37399 * gst-libs/gst/audio/mixerutils.h:
37400 Add docs for mixerutils stuff.
37402 2006-01-13 17:17:07 +0000 Tim-Philipp Müller <tim@centricular.net>
37404 gst/playback/gstplaybasebin.c: Fix playback for sources that emit raw audio or raw video streams (e.g.: cd audio sour...
37405 Original commit message from CVS:
37406 * gst/playback/gstplaybasebin.c: (setup_source):
37407 Fix playback for sources that emit raw audio or
37408 raw video streams (e.g.: cd audio sources) (#325984).
37410 2006-01-13 16:45:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37412 gst-libs/gst/audio/mixerutils.c: actually save the element we create
37413 Original commit message from CVS:
37414 * gst-libs/gst/audio/mixerutils.c:
37415 (gst_audio_mixer_filter_do_filter):
37416 actually save the element we create
37418 2006-01-13 16:17:50 +0000 Christian Schaller <uraeus@gnome.org>
37420 * gst-plugins-base.spec.in:
37421 remove version suffix
37422 Original commit message from CVS:
37423 remove version suffix
37425 2006-01-12 14:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
37427 gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
37428 Original commit message from CVS:
37429 * gst-libs/gst/cdda/gstcddabasesrc.c:
37430 (gst_cdda_base_src_handle_track_seek):
37431 No need to post a tag message on the bus when seeking
37432 within the same track, only post it when the current
37435 2006-01-11 18:30:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
37437 gst/playback/: Reenable stream selection. These mechanisms need a complete overhaul in the face of 0.8->0.10 changes ...
37438 Original commit message from CVS:
37439 * gst/playback/gstplaybasebin.c: (group_destroy),
37440 (probe_triggered), (new_decoded_pad), (mute_group_type),
37441 (set_active_source):
37442 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
37443 * gst/playback/gststreamselector.c:
37444 (gst_stream_selector_base_init),
37445 (gst_stream_selector_set_property),
37446 (gst_stream_selector_request_new_pad):
37447 Reenable stream selection. These mechanisms need a complete overhaul
37448 in the face of 0.8->0.10 changes though.
37450 2006-01-11 18:03:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
37452 ext/ogg/gstoggdemux.c: Change the pad template to src_%d to match the pads that are created from it. decodebin needs ...
37453 Original commit message from CVS:
37454 * ext/ogg/gstoggdemux.c:
37455 Change the pad template to src_%d to match the pads that
37456 are created from it. decodebin needs this information in order
37457 to decide that oggdemux is capable of producing multiple pads
37458 (and hence needs queues inserted).
37459 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
37460 (gst_ogg_mux_collected):
37461 Make debug output more useful by using GST_PTR_FORMAT.
37463 2006-01-11 17:38:35 +0000 Christian Schaller <uraeus@gnome.org>
37465 * gst-plugins-base.spec.in:
37466 update spec.in file
37467 Original commit message from CVS:
37468 update spec.in file
37470 2006-01-11 15:11:20 +0000 Tim-Philipp Müller <tim@centricular.net>
37472 gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
37473 Original commit message from CVS:
37474 Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
37475 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
37476 Set depth and width for alaw/mulaw (fixes #326601).
37478 2006-01-10 23:58:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37480 tests/icles/Makefile.am: don't build the tests if we don't have the libs
37481 Original commit message from CVS:
37482 * tests/icles/Makefile.am:
37483 don't build the tests if we don't have the libs
37485 2006-01-10 18:06:56 +0000 Tim-Philipp Müller <tim@centricular.net>
37487 ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
37488 Original commit message from CVS:
37489 * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
37490 (gst_cd_paranoia_paranoia_callback):
37491 Don't try to free NULL pointers.
37493 2006-01-10 15:47:48 +0000 Edward Hervey <bilboed@bilboed.com>
37495 gst/audiorate/gstaudiorate.c: Add debugging category.
37496 Original commit message from CVS:
37497 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
37498 (gst_audio_rate_change_state), (plugin_init):
37499 Add debugging category.
37501 Add case for incoming buffers without valid offset/offset_end.
37503 2006-01-10 12:25:59 +0000 Michael Smith <msmith@xiph.org>
37505 gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
37506 Original commit message from CVS:
37507 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
37508 Don't leak GCond in audio sources.
37510 2006-01-10 11:49:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
37512 gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
37513 Original commit message from CVS:
37514 * gst/playback/gstplaybin.c: (gen_audio_element):
37515 Don't leak an autoaudiosink/alsasink when we generate
37516 a new audio element. (old code, I guess)
37518 2006-01-10 11:04:21 +0000 Michael Smith <msmith@xiph.org>
37520 gst/audiorate/gstaudiorate.c: Support float audio in audiorate.
37521 Original commit message from CVS:
37522 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
37523 Support float audio in audiorate.
37524 Use width rather than depth for selecting sample width.
37526 2006-01-10 10:06:53 +0000 Tim-Philipp Müller <tim@centricular.net>
37528 gst/videotestsrc/videotestsrc.h: Use GLib types here (that way we don't have to include the generated _stdint.h heade...
37529 Original commit message from CVS:
37530 * gst/videotestsrc/videotestsrc.h:
37531 Use GLib types here (that way we don't have to include the
37532 generated _stdint.h header, which makes life easier for win32
37533 folks that don't use autotools for the build) (#325990, patch
37534 by: Sergey Scobich).
37536 2006-01-10 09:38:44 +0000 Tim-Philipp Müller <tim@centricular.net>
37538 gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
37539 Original commit message from CVS:
37540 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
37541 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
37542 (gst_ring_buffer_pause), (wait_segment):
37543 * gst-libs/gst/audio/gstringbuffer.h:
37544 Name (private) union, makes Forte compiler happy (this time
37545 for real) (#324900).
37547 2006-01-09 10:52:33 +0000 Tim-Philipp Müller <tim@centricular.net>
37549 gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
37550 Original commit message from CVS:
37551 * gst-libs/gst/audio/Makefile.am:
37552 Link against libgstinterfaces, needed for mixer
37553 and property probe stuff.
37555 2006-01-09 10:46:52 +0000 Edward Hervey <bilboed@bilboed.com>
37557 gst-libs/gst/Makefile.am:
37558 Original commit message from CVS:
37559 * gst-libs/gst/Makefile.am:
37561 2006-01-09 09:38:34 +0000 Tim-Philipp Müller <tim@centricular.net>
37563 gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
37564 Original commit message from CVS:
37565 * gst-libs/gst/audio/Makefile.am:
37566 * gst-libs/gst/audio/mixerutils.c:
37567 (gst_audio_mixer_filter_do_filter),
37568 (gst_audio_mixer_filter_check_element),
37569 (gst_audio_mixer_filter_probe_feature),
37570 (element_factory_rank_compare_func),
37571 (gst_audio_default_registry_mixer_filter):
37572 * gst-libs/gst/audio/mixerutils.h:
37573 Add gst_audio_default_registry_mixer_filter() utility
37576 2006-01-03 17:33:16 +0000 Michael Smith <msmith@xiph.org>
37578 gst/audioresample/resample.h: As before, but for o_buf
37579 Original commit message from CVS:
37580 * gst/audioresample/resample.h:
37581 As before, but for o_buf
37583 2006-01-03 17:27:13 +0000 Michael Smith <msmith@xiph.org>
37585 gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
37586 Original commit message from CVS:
37587 * gst/audioresample/resample.h:
37588 Declare struct _ResampleState.buffer as unsigned char *, not void *,
37589 since we do arithmetic on it.
37591 2006-01-02 23:37:38 +0000 Tim-Philipp Müller <tim@centricular.net>
37593 gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
37594 Original commit message from CVS:
37595 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
37596 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
37597 (gst_ring_buffer_pause), (wait_segment):
37598 * gst-libs/gst/audio/gstringbuffer.h:
37599 Sun's Forte compiler doesn't seem to like anonymous structs,
37600 so use same setup as in GstBaseSrc (fixes #324900).
37602 2005-12-30 14:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
37604 move old example to tests/examples/volume/volune.c
37605 Original commit message from CVS:
37607 * gst/volume/Makefile.am:
37608 * gst/volume/demo.c:
37609 move old example to tests/examples/volume/volune.c
37610 * tests/examples/Makefile.am:
37611 * tests/examples/seek/seek.c: (main):
37612 change window-close event from "delete-event" to "destroy"
37613 * tests/examples/volume/Makefile.am:
37614 * tests/examples/volume/volume.c: (value_changed_callback),
37615 (setup_gui), (message_received), (eos_message_received), (main):
37616 fix event handling and bus usage
37618 2005-12-29 20:37:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
37620 gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...
37621 Original commit message from CVS:
37622 * gst/audiotestsrc/gstaudiotestsrc.c:
37623 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
37624 (gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
37625 (gst_audio_test_src_query), (gst_audio_test_src_create_sine),
37626 (gst_audio_test_src_create_square),
37627 (gst_audio_test_src_create_saw),
37628 (gst_audio_test_src_create_triangle),
37629 (gst_audio_test_src_create_silence),
37630 (gst_audio_test_src_create_white_noise),
37631 (gst_audio_test_src_create_pink_noise),
37632 (gst_audio_test_src_init_sine_table),
37633 (gst_audio_test_src_create_sine_table),
37634 (gst_audio_test_src_change_wave),
37635 (gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
37636 (gst_audio_test_src_create), (gst_audio_test_src_set_property):
37637 * gst/audiotestsrc/gstaudiotestsrc.h:
37638 update to basesrc changes, implement segmented seeking and eos handling,
37639 add a 'sine-tab' waveform for performance critical playback
37641 2005-12-29 16:17:55 +0000 Tim-Philipp Müller <tim@centricular.net>
37643 po/POTFILES.in: ... and this time the other modified file that I missed last time.
37644 Original commit message from CVS:
37646 ... and this time the other modified file that I missed last time.
37648 2005-12-29 14:31:49 +0000 Michael Smith <msmith@xiph.org>
37650 gst/playback/gstdecodebin.c: Fix non-C89 variable declaration not at the start of a block. Should help some compilers.
37651 Original commit message from CVS:
37652 * gst/playback/gstdecodebin.c: (new_pad):
37653 Fix non-C89 variable declaration not at the start of a block. Should
37654 help some compilers.
37656 2005-12-29 12:43:22 +0000 Tim-Philipp Müller <tim@centricular.net>
37658 tests/check/Makefile.am: And now fix 'make distcheck' (builddir != srcdir)
37659 Original commit message from CVS:
37660 * tests/check/Makefile.am:
37661 And now fix 'make distcheck' (builddir != srcdir)
37663 2005-12-29 12:22:24 +0000 Tim-Philipp Müller <tim@centricular.net>
37665 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia plugin again (there are still fixes required to pla...
37666 Original commit message from CVS:
37668 * ext/cdparanoia/Makefile.am:
37669 * ext/cdparanoia/gstcdparanoia.c:
37670 * ext/cdparanoia/gstcdparanoia.h:
37671 * ext/cdparanoia/gstcdparanoiasrc.c:
37672 (gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
37673 (gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
37674 (gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
37675 (gst_cd_paranoia_paranoia_callback),
37676 (gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
37677 (gst_cd_paranoia_src_set_property),
37678 (gst_cd_paranoia_src_get_property), (plugin_init):
37679 * ext/cdparanoia/gstcdparanoiasrc.h:
37680 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
37681 plugin again (there are still fixes required to playbin to make
37682 cdda:// uris work there).
37684 2005-12-29 12:13:57 +0000 Tim-Philipp Müller <tim@centricular.net>
37686 tests/check/Makefile.am: Fix test case compilation.
37687 Original commit message from CVS:
37688 * tests/check/Makefile.am:
37689 Fix test case compilation.
37691 2005-12-29 11:49:11 +0000 Tim-Philipp Müller <tim@centricular.net>
37693 gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
37694 Original commit message from CVS:
37695 * gst-libs/gst/cdda/gstcddabasesrc.c:
37696 (gst_cdda_base_src_update_duration),
37697 (gst_cdda_base_src_calculate_cddb_id):
37698 An integer is not a string. Fix access to uninitialised variable.
37699 * tests/check/Makefile.am:
37700 Add cddabasesrc unit test; also actually enable the vorbis test.
37701 * tests/check/generic/states.c:
37702 Blacklist new cd audio elements as well.
37703 * tests/check/libs/cddabasesrc.c:
37704 Unit test for GstCddaBaseSrc (discid calculation mostly).
37706 2005-12-28 18:19:25 +0000 Tim-Philipp Müller <tim@centricular.net>
37708 docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
37709 Original commit message from CVS:
37710 * docs/libs/Makefile.am:
37711 * docs/libs/gst-plugins-base-libs-docs.sgml:
37712 * docs/libs/gst-plugins-base-libs-sections.txt:
37713 * docs/libs/gst-plugins-base-libs.types:
37714 Add docs for libgstcdda/GstCddaBaseSrc.
37715 * gst-libs/gst/interfaces/mixertrack.h:
37716 Do one struct member per line with a semicolon at the end, that way
37717 even gtk-doc might parse it without complaining.
37719 2005-12-28 18:06:50 +0000 Tim-Philipp Müller <tim@centricular.net>
37721 Add new libgstcdda with GstCddaBaseSrc class.
37722 Original commit message from CVS:
37724 * gst-libs/gst/Makefile.am:
37725 * gst-libs/gst/cdda/Makefile.am:
37726 * gst-libs/gst/cdda/base64.c:
37727 * gst-libs/gst/cdda/base64.h:
37728 * gst-libs/gst/cdda/gstcddabasesrc.c:
37729 (gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
37730 (gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
37731 (gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
37732 (gst_cdda_base_src_get_property),
37733 (gst_cdda_base_src_get_track_from_sector),
37734 (gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
37735 (gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
37736 (gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
37737 (gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
37738 (gst_cdda_base_src_uri_get_protocols),
37739 (gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
37740 (gst_cdda_base_src_uri_handler_init),
37741 (gst_cdda_base_src_setup_interfaces),
37742 (gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
37743 (cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
37744 (lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
37745 (gst_cdda_base_src_add_tags),
37746 (gst_cdda_base_src_add_index_associations),
37747 (gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
37748 (gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
37749 (gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
37750 (gst_cdda_base_src_create):
37751 * gst-libs/gst/cdda/gstcddabasesrc.h:
37752 * gst-libs/gst/cdda/sha1.c:
37753 * gst-libs/gst/cdda/sha1.h:
37754 Add new libgstcdda with GstCddaBaseSrc class.
37756 2005-12-28 14:59:41 +0000 Tim-Philipp Müller <tim@centricular.net>
37758 ext/gnomevfs/gstgnomevfssink.h: Use GstBaseSinkClass as parent_class member for class struct, not
37759 Original commit message from CVS:
37760 * ext/gnomevfs/gstgnomevfssink.h:
37761 Use GstBaseSinkClass as parent_class member for class struct, not
37764 2005-12-27 22:29:43 +0000 Tim-Philipp Müller <tim@centricular.net>
37766 gst/videotestsrc/gstvideotestsrc.c: Add start method to reset running time and number of frames sent
37767 Original commit message from CVS:
37768 * gst/videotestsrc/gstvideotestsrc.c:
37769 (gst_video_test_src_class_init), (gst_video_test_src_start):
37770 Add start method to reset running time and number of frames sent
37771 when starting up (fixes #324696; patch by: Michal Benes).
37773 2005-12-27 21:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
37775 docs/plugins/: Add docs stuff for gnomevfssrc and gnomevfssink.
37776 Original commit message from CVS:
37777 * docs/plugins/Makefile.am:
37778 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
37779 * docs/plugins/gst-plugins-base-plugins-sections.txt:
37780 * docs/plugins/gst-plugins-base-plugins.args:
37781 * docs/plugins/gst-plugins-base-plugins.hierarchy:
37782 * docs/plugins/gst-plugins-base-plugins.signals:
37783 Add docs stuff for gnomevfssrc and gnomevfssink.
37784 * ext/gnomevfs/gstgnomevfssrc.c:
37785 Fix example pipeline in gtk-doc blurb.
37787 2005-12-27 21:42:23 +0000 Tim-Philipp Müller <tim@centricular.net>
37789 ext/gnomevfs/: Port gnomevfssink; add gtk-doc blurb.
37790 Original commit message from CVS:
37791 * ext/gnomevfs/Makefile.am:
37792 * ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
37793 (gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
37794 (gst_gnome_vfs_handle_get_type), (plugin_init):
37795 * ext/gnomevfs/gstgnomevfs.h:
37796 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
37797 (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
37798 (gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
37799 (gst_gnome_vfs_sink_set_property),
37800 (gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
37801 (gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
37802 (gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
37803 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
37804 (gst_gnome_vfs_sink_uri_get_type),
37805 (gst_gnome_vfs_sink_uri_get_protocols),
37806 (gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
37807 (gst_gnome_vfs_sink_uri_handler_init):
37808 * ext/gnomevfs/gstgnomevfssink.h:
37809 Port gnomevfssink; add gtk-doc blurb.
37810 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
37811 (gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
37812 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
37813 (gst_gnome_vfs_src_uri_get_type),
37814 (gst_gnome_vfs_src_uri_get_protocols),
37815 (gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
37816 (gst_gnome_vfs_src_uri_handler_init),
37817 (gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
37818 (gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
37819 (gst_gnome_vfs_src_send_additional_headers_callback),
37820 (gst_gnome_vfs_src_received_headers_callback),
37821 (gst_gnome_vfs_src_push_callbacks),
37822 (gst_gnome_vfs_src_pop_callbacks),
37823 (gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
37824 (gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
37825 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
37826 * ext/gnomevfs/gstgnomevfssrc.h:
37827 s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
37828 file; add gtk-doc blurb with example pipelines.
37830 2005-12-23 18:16:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37834 Original commit message from CVS:
37837 === release 0.10.1 ===
37839 2005-12-23 18:08:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37845 * docs/libs/tmpl/gstcolorbalance.sgml:
37846 * docs/plugins/gst-plugins-base-plugins.args:
37847 * docs/plugins/gst-plugins-base-plugins.signals:
37848 * docs/plugins/inspect/plugin-adder.xml:
37849 * docs/plugins/inspect/plugin-alsa.xml:
37850 * docs/plugins/inspect/plugin-audioconvert.xml:
37851 * docs/plugins/inspect/plugin-audiorate.xml:
37852 * docs/plugins/inspect/plugin-audioresample.xml:
37853 * docs/plugins/inspect/plugin-audiotestsrc.xml:
37854 * docs/plugins/inspect/plugin-decodebin.xml:
37855 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
37856 * docs/plugins/inspect/plugin-gnomevfs.xml:
37857 * docs/plugins/inspect/plugin-libvisual.xml:
37858 * docs/plugins/inspect/plugin-ogg.xml:
37859 * docs/plugins/inspect/plugin-pango.xml:
37860 * docs/plugins/inspect/plugin-playbin.xml:
37861 * docs/plugins/inspect/plugin-subparse.xml:
37862 * docs/plugins/inspect/plugin-tcp.xml:
37863 * docs/plugins/inspect/plugin-theora.xml:
37864 * docs/plugins/inspect/plugin-typefindfunctions.xml:
37865 * docs/plugins/inspect/plugin-video4linux.xml:
37866 * docs/plugins/inspect/plugin-videorate.xml:
37867 * docs/plugins/inspect/plugin-videoscale.xml:
37868 * docs/plugins/inspect/plugin-videotestsrc.xml:
37869 * docs/plugins/inspect/plugin-volume.xml:
37870 * docs/plugins/inspect/plugin-vorbis.xml:
37871 * docs/plugins/inspect/plugin-ximagesink.xml:
37872 * docs/plugins/inspect/plugin-xvimagesink.xml:
37874 Original commit message from CVS:
37877 2005-12-21 20:59:52 +0000 Edgard Lima <edgard.lima@indt.org.br>
37880 * gst/typefind/gsttypefindfunctions.c:
37881 iLBC30 and iLBC20 added to typefind.
37882 Original commit message from CVS:
37883 iLBC30 and iLBC20 added to typefind.
37885 2005-12-20 15:57:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37889 * docs/libs/tmpl/gstcolorbalance.sgml:
37905 Original commit message from CVS:
37908 2005-12-20 12:24:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37910 * gst-libs/gst/audio/gstbaseaudiosink.c:
37911 * gst-libs/gst/audio/gstbaseaudiosrc.c:
37912 stop making fun of older compilers
37913 Original commit message from CVS:
37914 stop making fun of older compilers
37916 2005-12-20 12:00:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
37918 gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
37919 Original commit message from CVS:
37920 * gst-libs/gst/audio/gstbaseaudiosink.c:
37921 (gst_base_audio_sink_class_init):
37922 * gst-libs/gst/audio/gstbaseaudiosrc.c:
37923 (gst_base_audio_src_class_init):
37924 update strings, values are in microseconds
37925 change the default sink buffer time to something that is smaller
37926 (to help software volume mixing have a slightly lower delay) but
37927 still be acceptable on Wim's laptop
37929 2005-12-20 10:13:05 +0000 Edward Hervey <bilboed@bilboed.com>
37931 gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
37932 Original commit message from CVS:
37933 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
37934 Made a quack, forgot to add DUCK to the riff video template.
37936 2005-12-19 15:00:38 +0000 Edward Hervey <bilboed@bilboed.com>
37938 ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
37939 Original commit message from CVS:
37940 * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
37941 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
37942 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
37943 (gst_ogm_parse_chain):
37944 Make sure pads are initialized correctly.
37945 * gst-libs/gst/riff/riff-ids.h:
37946 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
37947 (gst_riff_create_video_template_caps):
37948 Add a whole bunch of FOURCC <=> MimeType.
37949 Extend the riff video pad template to support the newly added fourcc.
37951 2005-12-18 15:04:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
37953 ext/ogg/gstoggdemux.c: Extra debug output when activating/deactivating chains.
37954 Original commit message from CVS:
37955 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
37956 (gst_ogg_demux_activate_chain):
37957 Extra debug output when activating/deactivating chains.
37958 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
37959 (is_demuxer_element), (try_to_link_1), (remove_element_chain),
37961 Remove a queue from our list when it becomes unlinked.
37962 Don't add queues to elements in class 'Demux' if they
37963 can only produce one pad
37965 2005-12-18 10:54:33 +0000 Julien Moutte <julien@moutte.net>
37967 gst-libs/gst/video/gstvideosink.c: Add a debug category.
37968 Original commit message from CVS:
37969 2005-12-18 Julien MOUTTE <julien@moutte.net>
37970 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
37971 (gst_video_sink_get_type): Add a debug category.
37973 2005-12-18 00:56:07 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
37975 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
37976 Original commit message from CVS:
37977 2005-12-17 Philippe Khalaf <burger@speedy.org>
37978 * gst-libs/gst/rtp/gstbasertpdepayload.c:
37979 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
37980 Handle downstream newsegment by sending our own newsegment before the
37981 next buffer to be released. (#323900)
37983 2005-12-18 00:41:10 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
37985 gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
37986 Original commit message from CVS:
37987 2005-12-17 Philippe Khalaf <burger@speedy.org>
37988 * gst-libs/gst/rtp/gstbasertpdepayload.c:
37989 (gst_base_rtp_depayload_set_gst_timestamp):
37990 add queue delay to new segment as well (as opposed to just the first
37991 buffer). (bug #322347)
37993 2005-12-16 22:00:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
37995 ext/libvisual/visual.c: change some char* into char[]
37996 Original commit message from CVS:
37997 * ext/libvisual/visual.c: (make_valid_name):
37998 change some char* into char[]
37999 * gst/audiotestsrc/gstaudiotestsrc.c:
38000 (gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
38001 (gst_audio_test_src_create):
38002 * gst/audiotestsrc/gstaudiotestsrc.h:
38003 prepare to handle EOS and SEGMENT_DONE
38005 2005-12-16 12:32:37 +0000 Tim-Philipp Müller <tim@centricular.net>
38007 tests/check/generic/states.c: Blacklist cdparanoia element in state test.
38008 Original commit message from CVS:
38009 * tests/check/generic/states.c: (GST_START_TEST):
38010 Blacklist cdparanoia element in state test.
38012 2005-12-16 11:25:51 +0000 Benjamin Pineau <ben.pineau@gmail.com>
38014 gst/tcp/: Add <string.h> includes for memset and FD_ZERO (fixes #323878;
38015 Original commit message from CVS:
38016 * gst/tcp/gsttcp.c:
38017 * gst/tcp/gsttcpclientsink.c:
38018 * gst/tcp/gsttcpserversink.c:
38019 * gst/tcp/gsttcpserversrc.c:
38020 Add <string.h> includes for memset and FD_ZERO (fixes #323878;
38021 patch by: Benjamin Pineau).
38023 2005-12-15 14:43:38 +0000 Michael Smith <msmith@xiph.org>
38025 gst/videorate/gstvideorate.c: Fix timestamping for videorate when the first buffer it sees has a non-zero timestamp. ...
38026 Original commit message from CVS:
38027 * gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
38028 (gst_video_rate_chain):
38029 Fix timestamping for videorate when the first buffer it sees has a
38030 non-zero timestamp. Fix some misleading debug output.
38032 2005-12-15 10:30:14 +0000 Michael Smith <msmith@xiph.org>
38034 gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
38035 Original commit message from CVS:
38036 * gst/audioresample/gstaudioresample.c:
38037 Don't leak all input buffers to audioresample.
38039 2005-12-15 10:15:10 +0000 Tim-Philipp Müller <tim@centricular.net>
38041 ext/pango/gsttextoverlay.c: Don't operate on empty text buffers. Strip newlines and tabs only from the end of the tex...
38042 Original commit message from CVS:
38043 * ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
38044 Don't operate on empty text buffers. Strip newlines and
38045 tabs only from the end of the text, but leave them intact
38046 in the middle. Fix typo in gtk-doc description.
38048 2005-12-15 09:48:19 +0000 Tim-Philipp Müller <tim@centricular.net>
38050 gst/playback/: Make sure the video frame buffer we return to apps via the "frame" property always has caps set on it....
38051 Original commit message from CVS:
38052 * gst/playback/gstplaybasebin.c:
38053 * gst/playback/gstplaybin.c: (handoff):
38054 Make sure the video frame buffer we return to apps via the
38055 "frame" property always has caps set on it. Modify
38056 _gst_gvalue_set_object() macro to handle NULL objects
38059 2005-12-14 20:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
38061 gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
38062 Original commit message from CVS:
38063 * gst/audiotestsrc/gstaudiotestsrc.c:
38064 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
38065 (gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
38066 (gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
38067 (gst_audio_test_src_create):
38068 * gst/audiotestsrc/gstaudiotestsrc.h:
38069 Adjust to some recent api changes and add wtays new cool seeking
38072 2005-12-14 17:58:48 +0000 Tim-Philipp Müller <tim@centricular.net>
38074 ext/alsa/: Helper functions to add device probing via the GstPropertyProbe interface to a class.
38075 Original commit message from CVS:
38076 * ext/alsa/Makefile.am:
38077 * ext/alsa/gstalsadeviceprobe.c:
38078 * ext/alsa/gstalsadeviceprobe.h:
38079 Helper functions to add device probing via the GstPropertyProbe
38080 interface to a class.
38081 * ext/alsa/gstalsamixer.h:
38082 Comment out GST_ALSA_MIXER, it returns a struct that's not
38084 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
38085 Add some debug info.
38086 * ext/alsa/gstalsamixerelement.c:
38087 (gst_alsa_mixer_element_interface_supported),
38088 (gst_implements_interface_init),
38089 (gst_alsa_mixer_element_init_interfaces),
38090 (gst_alsa_mixer_element_class_init),
38091 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
38092 (gst_alsa_mixer_element_set_property),
38093 (gst_alsa_mixer_element_get_property),
38094 (gst_alsa_mixer_element_change_state):
38095 * ext/alsa/gstalsamixerelement.h:
38096 Add 'device' and 'device-name' properties. Add GstPropertyProbe
38097 for device handling (gnome-volume-control will need that).
38099 2005-12-12 20:31:24 +0000 Christian Schaller <uraeus@gnome.org>
38103 * gst-plugins-base.spec.in:
38104 updates to activate cdparanoia plugin
38105 Original commit message from CVS:
38106 updates to activate cdparanoia plugin
38108 2005-12-12 19:13:09 +0000 Michael Smith <msmith@xiph.org>
38110 ext/ogg/gstoggdemux.c: Use the correct function to free list of typefind factories.
38111 Original commit message from CVS:
38112 * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
38113 Use the correct function to free list of typefind factories.
38115 2005-12-12 15:09:55 +0000 Wim Taymans <wim.taymans@gmail.com>
38117 gst/videotestsrc/gstvideotestsrc.*: Implement seeking in videotestsrc.
38118 Original commit message from CVS:
38119 * gst/videotestsrc/gstvideotestsrc.c:
38120 (gst_video_test_src_class_init), (gst_video_test_src_init),
38121 (gst_video_test_src_parse_caps), (gst_video_test_src_query),
38122 (gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
38123 (gst_video_test_src_create):
38124 * gst/videotestsrc/gstvideotestsrc.h:
38125 Implement seeking in videotestsrc.
38128 2005-12-12 15:06:46 +0000 Wim Taymans <wim.taymans@gmail.com>
38130 ext/cdparanoia/: Partially ported cdparanoia now that basesrc can support a plugin like this..
38131 Original commit message from CVS:
38132 * ext/cdparanoia/Makefile.am:
38133 * ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
38134 (gst_paranoia_endian_get_type), (_do_init),
38135 (cdparanoia_class_init), (cdparanoia_init),
38136 (cdparanoia_set_property), (cdparanoia_get_property),
38137 (cdparanoia_do_seek), (cdparanoia_is_seekable),
38138 (cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
38139 (cdparanoia_convert), (cdparanoia_get_query_types),
38140 (cdparanoia_query), (cdparanoia_set_index),
38141 (cdparanoia_uri_set_uri):
38142 * ext/cdparanoia/gstcdparanoia.h:
38143 Partially ported cdparanoia now that basesrc can support a
38146 2005-12-12 13:03:50 +0000 Wim Taymans <wim.taymans@gmail.com>
38148 tests/examples/seek/scrubby.c: Set higher priority for bus events so they don't get reordered with gtk gui events.
38149 Original commit message from CVS:
38150 * tests/examples/seek/scrubby.c: (main):
38151 Set higher priority for bus events so they don't get reordered with
38153 * tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
38154 (flush_toggle_cb), (main):
38155 Added checkbox do disable flushing seeks.
38156 Disable scrubbing when doing non flushing seeks.
38158 2005-12-12 09:52:37 +0000 Tim-Philipp Müller <tim@centricular.net>
38160 gst/subparse/gstsubparse.c: Implement some sort of event handling that doesn't rely on g_return_if_fail; make sure we...
38161 Original commit message from CVS:
38162 * gst/subparse/gstsubparse.c: (gst_sub_parse_init),
38163 (gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
38164 (parser_state_init), (handle_buffer), (gst_sub_parse_chain),
38165 (gst_sub_parse_sink_event), (gst_sub_parse_change_state):
38166 Implement some sort of event handling that doesn't rely on
38167 g_return_if_fail; make sure we always push the last chunk of an
38168 .srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
38169 state change function; remove some old cruft. Seeking is still
38170 rather unlikely to work though.
38171 * tools/.cvsignore:
38174 2005-12-11 22:26:08 +0000 Julien Moutte <julien@moutte.net>
38176 sys/xvimage/xvimagesink.c: Fixed a leak of the current image reference when cleaning up.
38177 Original commit message from CVS:
38178 2005-12-11 Julien MOUTTE <julien@moutte.net>
38179 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
38180 Fixed a leak of the current image reference when cleaning up.
38181 Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
38183 2005-12-09 10:23:42 +0000 Michael Smith <msmith@xiph.org>
38185 tools/: Remove gst-launch-ext. It doesn't work, and is no longer particularly useful.
38186 Original commit message from CVS:
38187 * tools/Makefile.am:
38188 * tools/gst-launch-ext-m.m:
38189 Remove gst-launch-ext. It doesn't work, and is no longer
38190 particularly useful.
38192 2005-12-08 18:53:57 +0000 Luca Ognibene <luogni@tin.it>
38194 ext/ogg/gstogmparse.c: don't pass random values to ogmparse convert function.
38195 Original commit message from CVS:
38196 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
38197 don't pass random values to ogmparse convert function.
38198 Make seeking possible in the exile1.ogm file.
38200 2005-12-07 18:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
38202 gst/playback/: Work around refcount problem with g_value_set_object() that occur if the core has been compiled agains...
38203 Original commit message from CVS:
38204 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
38205 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
38206 Work around refcount problem with g_value_set_object() that occur
38207 if the core has been compiled against GLib-2.6 (g_value_set_object()
38208 will only g_object_ref() the element, but the caller will
38209 gst_object_unref() it and bad things will happen due to the way
38210 GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
38211 totem for people on FC4 using Thomas's 0.10 RPMs.
38213 2005-12-07 11:34:37 +0000 Edward Hervey <bilboed@bilboed.com>
38215 Time to welcome ogm to 0.10 :)
38216 Original commit message from CVS:
38217 Time to welcome ogm to 0.10 :)
38218 * ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
38219 (gst_ogg_pad_typefind):
38220 Oggdemux can now properly typefind elements with dynamic pads.
38221 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
38222 Properly set caps on src pad, and set caps on outgoing buffers.
38224 2005-12-06 19:42:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
38227 * ext/alsa/gstalsamixer.h:
38228 * ext/alsa/gstalsamixerelement.h:
38229 * ext/alsa/gstalsamixeroptions.h:
38230 * ext/alsa/gstalsamixertrack.h:
38231 * ext/alsa/gstalsasink.c:
38232 * ext/alsa/gstalsasink.h:
38233 * ext/alsa/gstalsasrc.c:
38234 * ext/alsa/gstalsasrc.h:
38235 * ext/cdparanoia/gstcdparanoia.h:
38236 * ext/gnomevfs/gstgnomevfsuri.h:
38237 * ext/ogg/gstoggdemux.c:
38238 * ext/ogg/gstoggmux.c:
38239 * ext/pango/gsttextoverlay.h:
38240 * ext/theora/theoradec.c:
38241 * ext/theora/theoraenc.c:
38242 * ext/vorbis/vorbisdec.h:
38243 * ext/vorbis/vorbisenc.c:
38244 * ext/vorbis/vorbisenc.h:
38245 * ext/vorbis/vorbisparse.h:
38246 * gst-libs/gst/audio/gstaudioclock.h:
38247 * gst-libs/gst/audio/gstaudiosink.c:
38248 * gst-libs/gst/audio/gstaudiosink.h:
38249 * gst-libs/gst/audio/gstaudiosrc.c:
38250 * gst-libs/gst/audio/gstaudiosrc.h:
38251 * gst-libs/gst/audio/gstbaseaudiosink.c:
38252 * gst-libs/gst/audio/gstbaseaudiosink.h:
38253 * gst-libs/gst/audio/gstbaseaudiosrc.c:
38254 * gst-libs/gst/audio/gstbaseaudiosrc.h:
38255 * gst-libs/gst/audio/gstringbuffer.h:
38256 * gst-libs/gst/audio/multichannel.h:
38257 * gst-libs/gst/floatcast/floatcast.h:
38258 * gst-libs/gst/interfaces/colorbalance.c:
38259 * gst-libs/gst/interfaces/colorbalance.h:
38260 * gst-libs/gst/interfaces/colorbalancechannel.h:
38261 * gst-libs/gst/interfaces/mixer.h:
38262 * gst-libs/gst/interfaces/mixeroptions.h:
38263 * gst-libs/gst/interfaces/mixertrack.h:
38264 * gst-libs/gst/interfaces/navigation.h:
38265 * gst-libs/gst/interfaces/propertyprobe.h:
38266 * gst-libs/gst/interfaces/tuner.h:
38267 * gst-libs/gst/interfaces/tunerchannel.h:
38268 * gst-libs/gst/interfaces/tunernorm.h:
38269 * gst-libs/gst/interfaces/xoverlay.h:
38270 * gst-libs/gst/netbuffer/gstnetbuffer.h:
38271 * gst-libs/gst/riff/riff-ids.h:
38272 * gst-libs/gst/riff/riff-media.h:
38273 * gst-libs/gst/riff/riff-read.h:
38274 * gst-libs/gst/rtp/gstbasertpdepayload.h:
38275 * gst-libs/gst/rtp/gstbasertppayload.c:
38276 * gst-libs/gst/rtp/gstbasertppayload.h:
38277 * gst-libs/gst/rtp/gstrtpbuffer.c:
38278 * gst-libs/gst/rtp/gstrtpbuffer.h:
38279 * gst-libs/gst/tag/gsttageditingprivate.h:
38280 * gst-libs/gst/tag/gstvorbistag.c:
38281 * gst-libs/gst/tag/tag.h:
38282 * gst-libs/gst/video/video.h:
38283 * gst/adder/gstadder.c:
38284 * gst/adder/gstadder.h:
38285 * gst/audioconvert/audioconvert.c:
38286 * gst/audioconvert/audioconvert.h:
38287 * gst/audioconvert/gstaudioconvert.c:
38288 * gst/audioconvert/gstchannelmix.c:
38289 * gst/audioconvert/gstchannelmix.h:
38290 * gst/audiorate/gstaudiorate.c:
38291 * gst/audioresample/buffer.h:
38292 * gst/audioresample/functable.h:
38293 * gst/audioresample/gstaudioresample.c:
38294 * gst/audioresample/resample.h:
38295 * gst/ffmpegcolorspace/avcodec.h:
38296 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
38297 * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
38298 * gst/ffmpegcolorspace/imgconvert.c:
38299 * gst/ffmpegcolorspace/imgconvert_template.h:
38300 * gst/playback/gstdecodebin.c:
38301 * gst/playback/gstplaybasebin.h:
38302 * gst/playback/gstplaybin.c:
38303 * gst/playback/gststreaminfo.h:
38304 * gst/tcp/gstfdset.c:
38305 * gst/tcp/gstfdset.h:
38306 * gst/tcp/gstmultifdsink.c:
38307 * gst/tcp/gstmultifdsink.h:
38308 * gst/tcp/gsttcp.h:
38309 * gst/tcp/gsttcpclientsrc.c:
38310 * gst/tcp/gsttcpclientsrc.h:
38311 * gst/tcp/gsttcpplugin.h:
38312 * gst/tcp/gsttcpserversink.c:
38313 * gst/tcp/gsttcpserversrc.c:
38314 * gst/typefind/gsttypefindfunctions.c:
38315 * gst/videorate/gstvideorate.c:
38316 * gst/videotestsrc/gstvideotestsrc.h:
38317 * gst/videotestsrc/videotestsrc.h:
38318 * sys/v4l/gstv4lcolorbalance.h:
38319 * sys/v4l/gstv4ltuner.h:
38320 * sys/v4l/gstv4lxoverlay.h:
38321 * sys/v4l/v4l_calls.h:
38322 * sys/v4l/videodev_mjpeg.h:
38323 * tests/check/elements/audioconvert.c:
38324 * tests/check/elements/audioresample.c:
38325 * tests/check/elements/audiotestsrc.c:
38326 * tests/check/elements/videotestsrc.c:
38327 * tests/check/elements/volume.c:
38328 * tests/examples/seek/scrubby.c:
38329 * tests/examples/seek/seek.c:
38331 Original commit message from CVS:
38334 2005-12-06 19:28:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
38336 * docs/libs/tmpl/gstaudio.sgml:
38337 * docs/libs/tmpl/gstcolorbalance.sgml:
38338 * docs/libs/tmpl/gstgconf.sgml:
38339 * docs/libs/tmpl/gstmixer.sgml:
38340 * docs/libs/tmpl/gstringbuffer.sgml:
38341 * docs/libs/tmpl/gsttuner.sgml:
38342 * docs/libs/tmpl/gstxoverlay.sgml:
38343 put back stability level
38344 Original commit message from CVS:
38345 put back stability level
38347 2005-12-05 18:11:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
38351 Original commit message from CVS:
38354 === release 0.10.0 ===
38356 2005-12-05 18:02:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
38362 * docs/libs/tmpl/gstcolorbalance.sgml:
38363 * docs/plugins/inspect/plugin-adder.xml:
38364 * docs/plugins/inspect/plugin-alsa.xml:
38365 * docs/plugins/inspect/plugin-audioconvert.xml:
38366 * docs/plugins/inspect/plugin-audiorate.xml:
38367 * docs/plugins/inspect/plugin-audioresample.xml:
38368 * docs/plugins/inspect/plugin-audiotestsrc.xml:
38369 * docs/plugins/inspect/plugin-decodebin.xml:
38370 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
38371 * docs/plugins/inspect/plugin-gnomevfs.xml:
38372 * docs/plugins/inspect/plugin-libvisual.xml:
38373 * docs/plugins/inspect/plugin-ogg.xml:
38374 * docs/plugins/inspect/plugin-pango.xml:
38375 * docs/plugins/inspect/plugin-playbin.xml:
38376 * docs/plugins/inspect/plugin-subparse.xml:
38377 * docs/plugins/inspect/plugin-tcp.xml:
38378 * docs/plugins/inspect/plugin-theora.xml:
38379 * docs/plugins/inspect/plugin-typefindfunctions.xml:
38380 * docs/plugins/inspect/plugin-video4linux.xml:
38381 * docs/plugins/inspect/plugin-videorate.xml:
38382 * docs/plugins/inspect/plugin-videoscale.xml:
38383 * docs/plugins/inspect/plugin-videotestsrc.xml:
38384 * docs/plugins/inspect/plugin-volume.xml:
38385 * docs/plugins/inspect/plugin-vorbis.xml:
38386 * docs/plugins/inspect/plugin-ximagesink.xml:
38387 * docs/plugins/inspect/plugin-xvimagesink.xml:
38389 Original commit message from CVS:
projects / platform / upstream / gstreamer.git / blob