1 2009-07-24 13:39:55 +0100 Jan Schmidt <thaytan@noraisin.net>
36 2009-07-24 13:46:15 +0100 Jan Schmidt <thaytan@noraisin.net>
38 * tests/examples/v4l/.gitignore:
39 ignores: Ignore v4l probing example binary
41 2009-07-24 09:35:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
43 * gst/typefind/gsttypefindfunctions.c:
44 typefind: recognise Kate spu subtitles as well
45 Recognise spu-subtitles, SUB and K-SPU as valid categories for
46 Kate subtitles as well.
48 2009-07-24 00:42:16 +0300 Stefan Kost <ensonic@users.sf.net>
51 Automatic update of common submodule
52 From fedaaee to 94f95e3
54 2009-07-22 14:21:43 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
56 * gst-plugins-base.spec.in:
57 Update spec file with latest changes
59 2009-07-20 17:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
92 * win32/common/_stdint.h:
93 * win32/common/audio-enumtypes.c:
94 * win32/common/config.h:
95 * win32/common/gstrtsp-enumtypes.c:
96 * win32/common/interfaces-enumtypes.c:
97 * win32/common/video-enumtypes.c:
100 2009-07-20 12:51:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
102 * gst/audiotestsrc/gstaudiotestsrc.c:
103 audiotestsrc: call send_event directly
104 We can't call gst_element_send_event() from a streaming thread as it gets the
105 state lock. Instead call the send_event method directly until we have a nice API
109 2009-07-03 04:42:24 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
111 * gst-libs/gst/audio/gstaudiosink.c:
112 audiosink: Add stream-status messages
115 2009-07-03 04:41:05 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
117 * gst-libs/gst/audio/gstaudiosrc.c:
118 audiosrc: Add stream-status messages
121 2009-07-20 10:53:11 +0200 Edward Hervey <bilboed@bilboed.com>
123 * gst/adder/gstadder.c:
124 gstadder: Don't forget to free pending events on flush/dispose.
127 2009-07-12 10:08:12 +0200 Edward Hervey <bilboed@bilboed.com>
129 * tests/check/elements/adder.c:
130 tests/adder: Add stream consistency checking. Fixes #588748
132 2009-07-12 10:07:34 +0200 Edward Hervey <bilboed@bilboed.com>
134 * gst/audiotestsrc/gstaudiotestsrc.c:
135 audiotestsrc: Make sure tags are properly serialized. Fixes #588746
136 We do this by letting the basesrc base class handle the tags.
138 2009-07-13 09:28:54 +0200 Edward Hervey <bilboed@bilboed.com>
140 * gst/adder/gstadder.c:
141 * gst/adder/gstadder.h:
142 adder: Collect incoming tag events and send them after newsegment. Fixes #588747
144 2009-07-16 09:32:46 +0200 Edward Hervey <bilboed@bilboed.com>
146 * ext/vorbis/vorbisdec.c:
147 vorbisdec: Check for empty tag strings. Fixes #588724
149 2009-07-14 17:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
151 * gst/playback/gstqueue2.c:
152 queue2: fix leak and improve buffering
153 Keep track of the max requested position and compare this to the write position
154 in the temp file to get the current amount of buffered data.
155 Fix memleak of all incomming buffers.
158 2009-07-15 17:40:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
160 * gst/playback/Makefile.am:
161 * gst/playback/gstinputselector.c:
162 * gst/playback/gstinputselector.h:
163 * gst/playback/gstplay-marshal.list:
164 * gst/playback/gstplaybin2.c:
165 playbin2: use private copy of input-selector
166 We shouldn't really depend on elements from -bad for stream
167 selection in playbin2, so use a private copy of input-selector
168 until the selector plugin is ready to be moved to -base or -good.
171 2009-07-15 17:26:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
173 * gst/playback/gstinputselector.c:
174 * gst/playback/gstinputselector.h:
175 playback: add private copy of the input-selector from gst-plugins-bad
176 Not hooked up yet though. See #586356.
178 2009-07-14 19:00:36 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
180 * tests/examples/v4l/Makefile.am:
181 examples: fix v4l probe example build
184 2009-07-14 19:00:10 +0100 Jan Schmidt <thaytan@noraisin.net>
218 0.10.23.2 pre-release
220 2009-07-14 16:24:10 +0100 Jan Schmidt <thaytan@noraisin.net>
224 Add Turkish translations
226 2009-07-14 15:31:13 +0100 Jan Schmidt <thaytan@noraisin.net>
228 * tests/check/elements/adder.c:
229 adder: One more attempt to fix the adder test
230 Give up and discard and recreate the alsasrc after checking it can
231 be opened, due to some strange crash inside alsa when we don't.
233 2009-07-14 15:06:41 +0100 Jan Schmidt <thaytan@noraisin.net>
235 * tests/check/elements/adder.c:
236 adder: Perform get_state() in the unit test
237 Wait for the alsasrc to return to NULL after setting it to PAUSED for
238 testing, otherwise it leads to segfaults later on.
240 2009-07-14 14:39:32 +0100 Jan Schmidt <thaytan@noraisin.net>
242 * tests/check/elements/adder.c:
243 adder: Don't fail when alsasrc is unavailable
244 Make the liveadder test succeed silently when it can't be completed
245 either because alsasrc is unavailable, or because the device is
248 2009-07-13 22:51:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
250 * gst-libs/gst/pbutils/descriptions.c:
251 * gst/typefind/gsttypefindfunctions.c:
252 typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest
253 Differentiate subtitle streams and lyrics/cracktastic/complex streams via
254 the category string in the headers. This seems like a useful distinction
255 to make, and also seems more future-proof. See #525743.
257 2009-02-21 13:18:10 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
259 * ext/ogg/gstoggmux.c:
260 oggmux: add Kate caps to the list of accepted types
263 2009-07-13 21:56:46 +0300 Stefan Kost <ensonic@users.sf.net>
265 * gst/playback/gsturidecodebin.c:
266 uridecodebin: treat uri-schemas incasesensitive
267 Treat uri-schemas incasesensitive. This is mandated in rfc2396 section 3.1.
268 Fixes not showing buffering messages e.g. for HTTP://...
270 2009-07-13 21:54:47 +0300 Stefan Kost <ensonic@users.sf.net>
272 * gst-libs/gst/interfaces/navigation.c:
273 navigation: simplify docs
274 Make short-desc short - its used in the toc. Strip uneeded markup.
276 2009-07-13 18:31:15 +0100 Jan Schmidt <thaytan@noraisin.net>
278 * win32/common/libgstnetbuffer.def:
279 * win32/common/libgstvideo.def:
281 Remove methods from video base classes that have moved to -bad.
282 Add gst_netaddress_to_string
284 2009-07-13 17:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
286 * tests/examples/gio/.gitignore:
287 ignores: ignore the giosrc-mounting example binary
289 2009-07-13 17:54:40 +0100 Jan Schmidt <thaytan@noraisin.net>
291 * gst-libs/gst/interfaces/navigation.c:
292 navigation: Add some partial documentation
293 Add a general documentation blurb for the GstNavigation functionality.
294 Still lacks some example code and detail on how to implement it.
296 2009-07-13 17:52:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
298 * gst-libs/gst/pbutils/descriptions.c:
299 pbutils: add description for Siren codec and make two descriptions non-translatable
301 2009-07-13 12:23:20 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
304 Automatic update of common submodule
305 From 5845b63 to fedaaee
307 2009-07-13 18:21:49 +0200 Elliott Sales de Andrade <quantum.analyst at gmail.com>
309 * gst-libs/gst/riff/riff-ids.h:
310 * gst-libs/gst/riff/riff-media.c:
311 riff: add siren to the RIFF parser
312 Add siren7 caps to the RIFF parser.
314 2009-07-13 14:55:59 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
317 * tests/examples/Makefile.am:
318 * tests/examples/v4l/Makefile.am:
319 * tests/examples/v4l/probe.c:
320 v4lsrc: add a simple test case for device probing
322 2009-07-03 11:38:01 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
325 * sys/v4l/Makefile.am:
326 * sys/v4l/gstv4lelement.c:
327 v4lsrc: optional support for device probing with gudev
328 Enumerate v4l devices using gudev if available.
331 2009-07-10 23:24:36 +0100 Stefan Kost <ensonic@users.sf.net>
333 * gst/adder/gstadder.c:
334 adder: add since tags to docs
336 2009-07-10 21:29:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
338 * tests/examples/seek/seek.c:
339 seek: don't automatically start pipeline in DB
340 Keep the pipeline paused when we detect download buffering. The user has to
341 manually start the pipeline for now because we can't estimate when the buffering
342 will finish or when we have underrun.
344 2009-07-10 21:01:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
346 * gst/playback/gstqueue2.c:
347 queue2: flush differently, avoiding deadlocks
348 Don't flush the file by closing and opening it but instead use g_freopen. This
349 avoids a deadlock in shutdown because we emit the temp-location property change
350 with the wrong lock held.
352 2009-07-10 20:25:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
354 * tests/examples/seek/seek.c:
355 seek: add a checkbox for progressive download
357 2009-07-10 20:24:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
359 * gst/playback/gsturidecodebin.c:
360 uridecodebin: Fix template construction
361 Fix the construction of the temporary filename construction as the application
362 name can be NULL and we don't want a separator between the prgname and the
365 2009-07-10 20:04:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
367 * gst/playback/gstplay-enum.c:
368 * gst/playback/gstplay-enum.h:
369 * gst/playback/gstplaybin2.c:
370 playbin2: add support for progressive download
371 Add a new playbin2 flag (initially disabled) to enable progressive download
372 buffering in uridecodebin.
374 2009-07-10 19:59:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
376 * gst/playback/gsturidecodebin.c:
377 uridecodebin: add download property
378 Add a download property that will attempt to configure queue2 into progressive
380 Make sure we only enable download buffering for quicktime and flv formats.
382 2009-07-10 19:49:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
384 * gst/playback/gstqueue2.c:
385 queue2: add temp-template property
386 Add a new temp-template property so that queue2 can securely allocate a
387 temporary filename. Deprecate the temp-location property for setting the
388 location but still use it to notify the allocated temp file.
390 2009-07-10 20:06:28 +0100 Stefan Kost <ensonic@users.sf.net>
392 * gst/adder/gstadder.c:
393 * gst/adder/gstadder.h:
394 adder: add a caps-property to avoid to need to plug a capsfilter afterwards
395 Adder can only handle one common format accross the pads. Thus one needed to add
396 a capsfilter afterwards and manage the caps. Now one can simply set the caps on
399 2009-07-10 18:59:05 +0100 Stefan Kost <ensonic@users.sf.net>
401 * tests/check/elements/adder.c:
402 adder: skip live-seek text if we have no audiosrc, add new test
403 The seek-test needs a real audiosrc. Also add a test that checks that adder is
404 reusable. Finaly handle warnings as warnings to fix a assertion.
406 2009-07-10 19:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
408 * ext/gio/gstgiosink.c:
409 gio: Also post a "not-mounted" message from giosink
411 2009-07-10 17:15:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
413 * tests/examples/gio/giosrc-mounting.c:
414 gio: Remove workaround for playbin2 bug in the sample application
415 The playbin2 bug was #588078.
417 2009-07-10 17:08:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
419 * gst/playback/gstplaybin2.c:
420 playbin2: Make it possible for READY->PAUSED to succeed after it failed the first time
421 If READY->PAUSED failed in the source element we would've swapped
422 the current and next group already. To allow READY->PAUSED to succeed
423 after the first failure we have to swap the current and next group
424 back again. This also ensure that we're again in the same state
425 as before the failed state change and not at the next group.
426 This was especially a problem for playbin2 pipelines that use the
427 new mounting support in giosrc as the source would fail for READY->PAUSED
428 the first time, the application mounts the location and then tries
429 to go READY->PAUSED again (and this time it would succeed).
432 2009-07-10 11:42:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
435 * tests/examples/Makefile.am:
436 * tests/examples/gio/Makefile.am:
437 * tests/examples/gio/giosrc-mounting.c:
438 gio: Add example application that shows how to handle the "not-mounted" message
440 2009-07-10 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
443 gio: Remove the experimental status from the GIO plugin
446 2009-07-10 11:24:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
448 * ext/gio/gstgiosink.c:
449 * ext/gio/gstgiosrc.c:
450 gio: Add documentation for the new "not-mounted" and "file-exists" messages
452 2009-07-09 13:45:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
454 * ext/gio/gstgiobasesrc.c:
455 gio: Make sure that we have the correct stream position when starting
457 2009-07-08 17:24:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
459 * ext/gio/gstgiobasesink.c:
460 gio: Make sure to flush the output stream if it shouldn't be closed
461 Otherwise there might still be unwritten data after the element
464 2009-07-08 17:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
466 * ext/gio/gstgiobasesink.c:
467 * ext/gio/gstgiobasesink.h:
468 * ext/gio/gstgiobasesrc.c:
469 * ext/gio/gstgiobasesrc.h:
470 * ext/gio/gstgiosink.c:
471 * ext/gio/gstgiosrc.c:
472 gio: Don't close the GIO streams for the giostream{src,sink} elements
473 This makes it possible to do something useful with the streams
474 after the element has stopped. Fixes bug #587896.
476 2009-07-08 17:19:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
478 * tests/check/pipelines/gio.c:
479 gio: Try to reuse the pipeline with the same stream objects
481 2009-07-08 17:02:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
483 * ext/gio/gstgiobasesink.c:
484 * ext/gio/gstgiobasesrc.c:
485 gio: Improve the error message if a stream is already closed before usage
487 2009-07-08 16:55:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
489 * ext/gio/gstgiosink.c:
490 gio: Post a custom file-exists message on the bus if the file already exists
491 An application can handle this message, remove the file in question
492 and restart the pipeline again without showing an error.
493 This fixes bug #529300.
495 2009-07-08 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
497 * ext/gio/gstgiosrc.c:
498 gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted
500 2009-07-08 16:50:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
502 * ext/gio/gstgiosink.c:
503 gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink
505 2009-07-08 15:52:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
507 * ext/gio/gstgiosrc.c:
508 gio: Post a custom "not-mounted" message on the bus
509 This allows applications to mount the GFile if possible and restart
510 the pipeline instead of simply giving an error.
512 2009-07-08 15:08:32 +0200 Philip Jägenstedt <philipj@opera.com>
514 * gst/audioconvert/gstchannelmix.c:
515 audioconvert: Fix compilation when debugging is disabled
518 2009-07-07 20:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
520 * ext/gio/gstgiobasesink.c:
521 * ext/gio/gstgiobasesink.h:
522 * ext/gio/gstgiobasesrc.h:
523 * ext/gio/gstgiosink.c:
524 * ext/gio/gstgiosink.h:
525 * ext/gio/gstgiostreamsink.c:
526 * ext/gio/gstgiostreamsink.h:
527 gio: Add vfunc for requesting the stream for the sinks too
529 2009-07-07 20:21:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
531 * ext/gio/gstgiobasesink.c:
532 * ext/gio/gstgiobasesink.h:
533 * ext/gio/gstgiobasesrc.c:
534 * ext/gio/gstgiosink.c:
535 * ext/gio/gstgiosrc.c:
536 * ext/gio/gstgiostreamsink.c:
537 * ext/gio/gstgiostreamsrc.c:
538 gio: Some more random cleanup
540 2009-07-07 20:20:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
543 * ext/gio/gstgiobasesink.c:
544 * ext/gio/gstgiobasesrc.c:
545 * ext/gio/gstgiobasesrc.h:
546 * ext/gio/gstgiosink.c:
547 * ext/gio/gstgiosrc.c:
548 * ext/gio/gstgiosrc.h:
549 * ext/gio/gstgiostreamsink.c:
550 * ext/gio/gstgiostreamsrc.c:
551 * ext/gio/gstgiostreamsrc.h:
552 gio: Update my mail address and copyright
554 2009-07-07 20:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
556 * ext/gio/gstgiobasesrc.c:
557 * ext/gio/gstgiobasesrc.h:
558 * ext/gio/gstgiosrc.c:
559 * ext/gio/gstgiostreamsrc.c:
560 * ext/gio/gstgiostreamsrc.h:
561 gio: General clean up and simplification
562 The GInputStreams are now requested by a vfunc from
563 the subclasses instead of relying that the subclass
564 sets it until it's needed.
565 This might also fix bug #587896.
567 2009-07-06 22:31:12 +0100 Stefan Kost <ensonic@users.sf.net>
569 * gst/adder/gstadder.c:
570 adder: keep sending newsegments after seeking
571 Adder sends with timestamps from 0 upwards. After seeking we need to send
572 new-segments to get correct positions-queries.
574 2009-07-06 20:44:00 +0100 Stefan Kost <ensonic@users.sf.net>
576 * tests/check/elements/adder.c:
577 adder: make test more robust
578 Add audioconverts to the live-seeking test to make it negotiate.
580 2009-06-30 17:19:50 +0300 Stefan Kost <ensonic@users.sf.net>
582 * sys/xvimage/xvimagesink.c:
583 xvimagesink: use core performance log category
585 2009-07-05 21:29:40 +0200 Edward Hervey <bilboed@bilboed.com>
587 * gst/adder/gstadder.c:
588 adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped.
589 This ensures that collectpads' cookie is properly updated so that when the streaming
590 threads will restart and be checking for the flushing status of all pads there will
591 be no inconsistent state.
593 2009-07-05 18:01:38 +0200 Hans-Peter Nilsson <hp@gcc.gnu.org>
595 * ext/pango/gstclockoverlay.c:
596 pango: Call tzset() before localtime_r()
597 POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't
598 required to set the state variables that define the current timezone. Indeed,
599 glibc (at least 2.9) doesn't do this for subsequent calls. The effect is that
600 if the system timezone is changed for a running program between two calls to
601 gst_clock_overlay_render_time, it won't be noticed. For glibc, changing the
602 timezone equals /etc/localtime being modified.
605 2009-07-01 17:33:14 -0700 David Schleef <ds@schleef.org>
608 build: remove spurious schroedinger reference
610 2009-07-01 10:25:43 -0700 David Schleef <ds@schleef.org>
614 * ext/schroedinger/Makefile.am:
615 * ext/schroedinger/gstschro.c:
616 * ext/schroedinger/gstschrodec.c:
617 * ext/schroedinger/gstschroenc.c:
618 * ext/schroedinger/gstschroparse.c:
619 * ext/schroedinger/gstschroutils.c:
620 * ext/schroedinger/gstschroutils.h:
621 * gst-libs/gst/video/Makefile.am:
622 * gst-libs/gst/video/gstbasevideocodec.c:
623 * gst-libs/gst/video/gstbasevideocodec.h:
624 * gst-libs/gst/video/gstbasevideodecoder.c:
625 * gst-libs/gst/video/gstbasevideodecoder.h:
626 * gst-libs/gst/video/gstbasevideoencoder.c:
627 * gst-libs/gst/video/gstbasevideoencoder.h:
628 * gst-libs/gst/video/gstbasevideoparse.c:
629 * gst-libs/gst/video/gstbasevideoparse.h:
630 * gst-libs/gst/video/gstbasevideoutils.c:
631 * gst-libs/gst/video/gstbasevideoutils.h:
632 basevideo: send basevideo back to remedial school
633 Move basevideo classes and schroedinger plugin to -bad.
635 2009-07-01 12:54:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
637 * docs/libs/gst-plugins-base-libs-sections.txt:
638 * gst-libs/gst/netbuffer/gstnetbuffer.h:
639 netaddress: add constant for max len
641 2009-07-01 12:48:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
643 * docs/libs/gst-plugins-base-libs-sections.txt:
644 * gst-libs/gst/netbuffer/gstnetbuffer.c:
645 * gst-libs/gst/netbuffer/gstnetbuffer.h:
646 netbuffer: add gst_netaddress_to_string
647 Add function to serialize a net address to a string.
648 API: GstNetAddress::gst_netaddress_to_string()
650 2009-06-30 18:44:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
652 * gst/playback/gsturidecodebin.c:
653 uridecodebin: make fd:// uri use buffering too
654 fd:// usually operate in push mode only and are thus suitable for buffering.
656 2009-06-30 14:46:38 +0300 Stefan Kost <ensonic@users.sf.net>
658 * gst/playback/gstplaybin2.c:
659 * gst/volume/gstvolume.c:
660 volume: include "1.0=100%" in property description
662 2009-06-30 14:45:51 +0300 Stefan Kost <ensonic@users.sf.net>
664 * gst/playback/gstplaysink.c:
665 playsink: remove unused property defs
667 2009-06-29 17:11:50 +0300 Stefan Kost <ensonic@users.sf.net>
669 * gst-libs/gst/audio/multichannel.c:
670 multichannel: rewrite the new doc comment a bit
671 Its part of the audio lib.
673 2009-06-29 14:34:02 +0100 Jan Schmidt <thaytan@noraisin.net>
675 * gst/playback/gstplaysink.c:
676 playsink: Avoid a segfault when the video sink fails to start
677 Don't attempt to display the subpictures and segfault when the
678 video sink failed to start (and hence the videochain is NULL).
680 2009-06-29 15:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
682 * gst-libs/gst/audio/gstringbuffer.c:
683 * gst-libs/gst/audio/gstringbuffer.h:
684 ringbuffer: add vmethod to clear the ringbuffer
685 Add a vmethod so that subclasses can be notified when they should clear the data
688 2009-06-29 14:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
690 * gst-libs/gst/riff/riff-media.c:
691 riff-media: Fix the fourcc caps property for VC-1/WMVA
692 The caps property for carrying fourccs is 'format', not 'fourcc'
694 2009-06-29 12:20:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
696 * gst-libs/gst/rtsp/gstrtspconnection.c:
697 rtsp: include in.h for FreeBSD compat
700 2009-06-29 12:20:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
702 * win32/common/libgstapp.def:
703 defs: add defs for new appsink buffer-list method
705 2009-06-29 12:14:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
707 * gst-libs/gst/app/gstappsink.c:
708 * gst-libs/gst/app/gstappsink.h:
709 appsink: add docs and signals
710 Add docs for the new callback.
711 Add signals for the new buffer-list support.
713 2009-06-29 10:24:36 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
715 * tests/check/elements/appsink.c:
716 Added unit tests for buffer list support in appsink.
718 2009-06-17 11:12:08 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
720 * gst-libs/gst/app/gstappsink.c:
721 Added buffer list support.
723 2009-06-17 09:23:11 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
725 * gst-libs/gst/app/gstappsink.h:
726 Added buffer list support.
728 2009-06-29 09:36:27 +0200 Peter Kjellerstedt <pkj@axis.com>
730 * gst-libs/gst/sdp/gstsdpmessage.c:
731 sdp: Include winsock2.h after defining WINVER.
732 Similar to bug #587080.
734 2009-06-29 09:31:40 +0200 Peter Kjellerstedt <pkj@axis.com>
736 * gst-libs/gst/rtsp/gstrtspconnection.c:
737 rtsp: Moved a comment.
739 2009-06-27 23:23:02 +0300 Stefan Kost <ensonic@users.sf.net>
741 * gst-libs/gst/audio/audio.c:
742 * gst-libs/gst/audio/multichannel.c:
743 docs: add basic section docs for multichannel and relocate the ones for audio
744 Add section docs for multichannel, so that it has a short desc in the toc too.
745 Move the section docs in adio up, so that the follow the copyright like
748 2009-06-26 21:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
750 * sys/v4l/gstv4lelement.c:
751 * sys/v4l/gstv4lsrc.c:
752 v4l: open/close device in ready.
753 Simillar change like in v4l2src. This allows probing feature in paused, where
754 streaming is noit yet started.
756 2009-06-10 17:05:22 +0300 René Stadler <rene.stadler@nokia.com>
758 * gst/playback/gstplaysink.c:
759 playbin2: fix initial volume handling also when reusing the element
760 This is a follow-up to commit 452988, making it work correctly when the audio
763 2009-06-26 21:48:58 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
765 * gst-libs/gst/rtsp/gstrtspconnection.c:
766 Define WINVER before including any win headers
769 2009-06-27 00:50:54 +0300 René Stadler <mail@renestadler.de>
771 * gst-libs/gst/riff/riff-read.c:
772 riff: prevent crash if rounded up tag size exceeds data size
773 When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
774 and an invalid read past the buffer data follows.
776 2009-06-26 15:17:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
778 * gst-libs/gst/video/gstbasevideocodec.c:
779 basevideocodec: By default don't allow caps changes on the srcpad
780 This fixed playback of Dirac files with schrodec when upstream wants
781 a different width/height, basevideocodec accepts this and then
782 pushes buffers with new caps but content of the old caps.
783 In the best case this will just result in wrong unit size and a
784 failure in basestransform elements.
786 2009-06-26 14:11:21 +0100 Jan Schmidt <thaytan@noraisin.net>
789 autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
790 Check for more automake command variants. Use printf instead of 'echo -n'
793 2009-06-26 13:41:38 +0100 Jan Schmidt <thaytan@noraisin.net>
796 Automatic update of common submodule
797 From f810030 to 5845b63
799 2009-06-26 13:14:02 +0300 Stefan Kost <ensonic@users.sf.net>
801 * gst/playback/gstscreenshot.c:
802 screenshot: don't leak message
804 2009-06-25 12:04:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
806 * gst/typefind/gsttypefindfunctions.c:
807 typefinding: lower the h264 typefinder's probability
808 A NEARLY_CERTAIN is absolutely not warranted given the kind
809 of things it checks for. Even a LIKELY is probably not entirely
812 2009-06-24 15:13:56 +0100 Jan Schmidt <jan.schmidt@sun.com>
815 Automatic update of common submodule
816 From f3bb51b to f810030
818 2009-06-24 09:48:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
820 * gst-libs/gst/pbutils/descriptions.c:
821 pbutils: add description for multipart
822 So we get slightly nicer error messages when multipartdemux is missing.
824 2009-06-23 18:07:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
826 * gst/adder/gstadder.c:
827 adder: only unflush when we flushed before
828 Ass suggested by Stefan Kost:
829 Keep track of when the sinkpad was set to flushing and unflush the pad when an
830 upstream flushing seek failed.
832 2009-06-23 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
834 * gst/playback/gsturidecodebin.c:
835 uridecodebin: fix leak when the source fails to change state
837 2009-06-23 12:40:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
839 * gst/subparse/gstssaparse.c:
840 ssaparse: avoid leaking all buffers
842 2009-06-22 22:18:03 +0300 Stefan Kost <ensonic@users.sf.net>
844 * tests/check/elements/adder.c:
845 adder: test seek handling in adder
846 This tests seeking on an adder that has a normal and a live source connected.
847 Wheter the current behavior is the desired one needs to be discussed still
850 2009-06-22 16:17:10 +0300 Stefan Kost <ensonic@users.sf.net>
852 * sys/ximage/ximagesink.c:
853 * sys/xvimage/xvimagesink.c:
854 x(v)imagesink: pass the xwindow along to not look at the yet unset var.
855 When we call this from xwindow_new, x(v)imagesink->xwindow is not yet set.
857 2009-06-22 11:40:33 +0300 Stefan Kost <ensonic@users.sf.net>
859 * sys/ximage/ximagesink.c:
860 * sys/ximage/ximagesink.h:
861 * sys/xvimage/xvimagesink.c:
862 * sys/xvimage/xvimagesink.h:
863 x(v)imagesink: catch tags and show title in own window
864 Refactor the code that sets the window title. Catch tag-events and use title
865 metadata for the window title.
867 2009-06-21 19:42:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
869 * gst/audiotestsrc/gstaudiotestsrc.c:
870 audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
871 Also make all the function arrays constant.
873 2009-06-21 12:27:37 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
875 * gst/audiotestsrc/gstaudiotestsrc.c:
876 * gst/audiotestsrc/gstaudiotestsrc.h:
877 audiotestsrc: Add support for generating gaussian white noise
878 This patch adds support for stationary white Gaussian noise.
879 The Box-Muller algorithm is used to generate pairs of independent
880 normally-distributed random numbers.
883 2009-06-20 23:46:28 +0100 Jan Schmidt <thaytan@noraisin.net>
885 * gst/ffmpegcolorspace/imgconvert.c:
886 * gst/ffmpegcolorspace/imgconvert_template.h:
887 ffmpegcolorspace: Fix NV12 and NV21 transformations
888 Fix some stride problems, fix the nv12 to nv21 direct transformation,
889 and implement a direct conversion to yuv444 to save CPU.
891 2009-06-20 22:36:21 +0100 Jan Schmidt <thaytan@noraisin.net>
893 * gst/videotestsrc/videotestsrc.c:
894 videotestsrc: Fix NV12 painting for odd strides/heights
896 2009-06-19 22:16:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
898 * ext/cdparanoia/gstcdparanoiasrc.c:
899 cdparanoia: run-time license is LGPL now that we require cdparanoia 0.10.2
900 cdparanoia has an LGPL v2.1 license since 0.10.1 and we now require 0.10.2.
901 Finally fixes #531035.
903 2009-06-19 21:25:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
905 * ext/cdparanoia/gstcdparanoiasrc.c:
906 cdparanoia: try to guess a good cache size if it's set to -1
907 Try to guess from the paranoia-mode setting whether playback or
908 ripping is wanted, and use a smaller cache size if we're likely
909 to be doing playback, to avoid a long startup delay. Since this
910 was the value used in older cdparanoia versions, it should be
911 fine in any case. See #586331.
913 2009-06-19 11:27:40 +1000 Jonathan Matthew <jonathan@d14n.org>
916 * ext/cdparanoia/gstcdparanoiasrc.c:
917 * ext/cdparanoia/gstcdparanoiasrc.h:
918 cdparanoia: expose cache size setting
919 This setting was added in cdparanoia 10.2. The default value is good
920 for audio extraction, but lower values (previous versions of cdparanoia
921 used 150) are better for realtime playback.
924 2009-06-19 17:43:03 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
926 * gst-plugins-base.spec.in:
927 Make build of schro plugin conditional
929 2009-06-19 15:52:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
931 * docs/libs/gst-plugins-base-libs-sections.txt:
932 * gst-libs/gst/rtp/gstbasertppayload.c:
933 * gst-libs/gst/rtp/gstbasertppayload.h:
934 * win32/common/libgstrtp.def:
935 basertppayload: add support for bufferlists
936 Based on patch from Ognyan Tonchev.
939 2009-06-19 15:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
941 * gst-libs/gst/rtp/gstrtpbuffer.c:
942 rtpbuffer: use new convenience functions
943 New core convenience functions makes the list getters and setters trivial.
944 Maybe even too trivial...
946 2009-06-18 19:07:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
948 * win32/common/libgstrtp.def:
949 defs: add new symbol to win32 defs file
950 Based on patches by Ognyan Tonchev.
953 2009-06-18 19:04:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
955 * docs/libs/gst-plugins-base-libs-sections.txt:
956 * gst-libs/gst/rtp/gstrtpbuffer.c:
957 rtp: cleanups, add _list_get_seq() too
958 Clean up the docs a little.
959 Add missing _list_get_seq method.
960 Add new symbols to the docs
962 2009-06-18 18:47:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
964 * gst-libs/gst/rtp/gstrtpbuffer.c:
965 * win32/common/libgstrtp.def:
967 Add Since tags to docs
968 Move some code around
971 2009-06-18 17:46:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
973 * gst-libs/gst/rtp/gstrtpbuffer.c:
974 * gst-libs/gst/rtp/gstrtpbuffer.h:
975 * tests/check/libs/rtp.c:
976 rtp: add bufferlist support
978 2009-06-18 18:03:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
980 * gst-libs/gst/rtp/gstrtpbuffer.c:
981 rtp: pass data to macros instead of GstBuffer
983 2009-06-18 17:42:10 +0100 Jan Schmidt <thaytan@noraisin.net>
985 * win32/common/libgstrtsp.def:
986 win32: Add gst_rtsp_watch_queue_data() to the exports
987 Fix the tests by exporting the new symbol from the win32 dlls
989 2009-06-18 18:13:22 +0300 Stefan Kost <ensonic@users.sf.net>
991 * sys/xvimage/xvimagesink.c:
992 xvimagesink: appname might be NULL
993 Don't set title if appname is unknown.
995 2009-06-18 17:58:06 +0300 Stefan Kost <ensonic@users.sf.net>
997 * sys/xvimage/xvimagesink.c:
998 xvimagesink: set window title from application name
1000 2009-06-09 19:14:00 +0200 Peter Kjellerstedt <pkj@axis.com>
1002 * gst-libs/gst/rtsp/gstrtspurl.c:
1003 rtsp: Made the parsing of the RTSP URL scheme more generic.
1005 2009-06-15 13:58:26 +0200 Peter Kjellerstedt <pkj@axis.com>
1007 * gst-libs/gst/rtsp/gstrtspconnection.c:
1008 * gst-libs/gst/rtsp/gstrtspconnection.h:
1009 rtsp: Added gst_rtsp_watch_queue_data().
1010 gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
1011 but allows for queuing any data block for writing (much like
1012 gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)
1013 API: gst_rtsp_watch_queue_data()
1015 2009-06-09 16:37:09 +0200 Peter Kjellerstedt <pkj@axis.com>
1017 * gst-libs/gst/rtsp/gstrtspconnection.c:
1018 rtsp: Only extract the session ID from RTSP responses.
1020 2009-06-09 19:06:57 +0200 Peter Kjellerstedt <pkj@axis.com>
1022 * gst-libs/gst/rtsp/gstrtspurl.c:
1023 rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
1025 2009-06-09 14:31:18 +0200 Peter Kjellerstedt <pkj@axis.com>
1027 * gst-libs/gst/rtsp/gstrtspconnection.c:
1028 rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
1030 2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
1032 * gst-libs/gst/rtsp/gstrtspconnection.c:
1033 rtsp: Improved base64 decoding in fill_bytes().
1034 The base64 decoding in fill_bytes() expected the size of the read data to
1035 be evenly divisible by four (which is true for the base64 encoded data
1036 itself). This did not, however, take whitespace (especially line breaks)
1037 into account and would fail the decoding if any whitespace was present.
1039 2009-06-17 14:00:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1041 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1042 audiosrc: fix get_offset
1043 When we need to jump to the most recently captured sample, jump to where the
1044 next sample will be written instead of to some old data.
1047 2009-06-17 13:18:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1049 * gst-libs/gst/audio/gstbaseaudiosink.c:
1050 audiosink: free the ringbuffer when going to NULL
1051 Unparent and free the ringbuffer when going to NULL, like we do with the
1052 audiosrc element. We can do this now because we correctly manage the time
1055 2009-06-17 13:17:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1057 * gst-libs/gst/audio/gstaudiosink.c:
1058 * gst-libs/gst/audio/gstaudiosrc.c:
1059 audio: correctly handle short read/writes
1061 2009-05-05 15:37:54 +0300 René Stadler <rene.stadler@nokia.com>
1063 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1064 baseaudiosrc: add some extra logging for buffer timestamps
1066 2009-06-17 11:22:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1068 * gst/adder/gstadder.c:
1069 adder: more seeking fixes.
1070 When a seek failed upstream, make sure the adder sinkpad is set unflushing again
1071 so that streaming can continue.
1072 We only have a pending segment when we flushed.
1073 Set the flush_stop_pending flag inside the appropriate locks and before we
1074 attempt to perform the upstream seek.
1075 Add some more comments.
1076 Use the right lock to protect the flags in flush_stop.
1079 2009-06-17 07:24:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1081 * gst/playback/gstdecodebin2.c:
1082 decodebin2: Free iterator after removing all groups
1084 2009-06-16 19:38:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1086 * gst-libs/gst/video/gstvideofilter.c:
1087 videofilter: Add a default get_unit_size function
1088 This returns the correct values for all formats that are handled by
1089 GstVideoFormat and makes all the custom get_unit_size functions in
1090 many elements unnecessary.
1092 2009-06-16 18:57:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1094 * gst-libs/gst/rtsp/gstrtspdefs.c:
1095 * gst-libs/gst/rtsp/gstrtspdefs.h:
1096 rtsp: add Timestamp header field
1099 2009-06-16 18:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1101 * gst/playback/gstplaybin2.c:
1102 playbin2: set smarter target state on uridecodebin
1103 Set the target state of the newly added uridecodebins to somthing else that
1104 PAUSED so that we keep their state in sync with the playsink state.
1107 2009-06-16 18:13:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1109 * gst/playback/gstplaysink.c:
1110 playsink: set the sink flag on the element
1112 2009-06-16 18:09:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1114 * gst/playback/gsturidecodebin.c:
1115 uridecodebin: add debug message
1117 2009-06-16 14:05:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1119 * gst-libs/gst/audio/gstaudiosink.c:
1120 * gst-libs/gst/audio/gstaudiosrc.c:
1121 audiosink, audiosrc: do the class_ref()s in the right class_init functions
1122 Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
1124 2009-06-15 15:39:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1126 * gst-libs/gst/audio/gstaudiosink.c:
1127 * gst-libs/gst/audio/gstaudiosrc.c:
1128 audiosink,audiosrc: ref the audio ring buffer class and type in class_init
1129 Hack around thread-safety issues in GObject and our racy _get_type()
1130 functions (we could easily fix the _get_type() functions, but we still
1131 need to hack around the GObject class races until we require a newer
1132 GLib version, I think).
1134 2009-06-15 12:57:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1136 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1137 audiosrc: return FALSE when receiving a SEEK event
1138 When receiving a seek event, return FALSE as we don't implement seeking.
1140 2009-06-15 11:06:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1142 * tests/examples/seek/seek.c:
1143 Don't use deprecated GTK API
1146 2009-06-15 11:40:00 +0300 Stefan Kost <ensonic@users.sf.net>
1148 * gst/adder/gstadder.c:
1149 adder: send flush_stop when seeking failed
1150 At least do the fix to sent the flush_stop when seeking failed to ensure we
1151 keep no pads flushing. before it was send when the seeking worked which is just
1152 plain wrong and was not the intention.
1154 2009-06-12 15:17:14 +0200 Peter Kjellerstedt <pkj@axis.com>
1156 * gst-libs/gst/rtsp/gstrtspconnection.c:
1157 rtsp: Use a more consistent naming of GstRTSPRec variables.
1159 2009-06-12 15:11:05 +0200 Peter Kjellerstedt <pkj@axis.com>
1161 * gst-libs/gst/rtsp/gstrtspconnection.c:
1162 * gst-libs/gst/rtsp/gstrtspconnection.h:
1163 rtsp: Call message_sent() callback for all sent messages.
1164 Previously the messages_sent() callback was only called for messages
1165 which had a CSeq, which excluded all data messages. Instead of using the
1166 CSeq as ID, use a simple index counter.
1168 2009-06-14 22:13:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1170 * ext/ogg/gstoggdemux.c:
1171 * ext/theora/theoradec.c:
1172 * ext/vorbis/vorbisdec.c:
1173 oggdemux: post/send tags with the container-format tag
1174 For this to work properly, theoradec and vorbisdec need to put
1175 tag events received from upstream into the pending_events list
1176 so they get pushed out after any newsegment event, not before.
1178 2009-06-14 20:30:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1180 * tests/examples/seek/scrubby.c:
1181 * tests/examples/seek/seek.c:
1182 * tests/old/examples/seek/cdplayer.c:
1183 Don't use deprecated GTK API
1186 2009-06-12 16:31:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1188 * gst/adder/gstadder.c:
1189 adder: send flush-stop earlier
1190 When no flush-stop has been sent by upstream, we have to send one ourselves to
1191 continue playback. Do this as soon as the collect function is called instead of
1192 after we possibly pushed segment events (that got then flushed out)
1194 2009-06-12 13:55:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1196 * tests/examples/seek/seek.c:
1197 seek: add shuttle controls
1199 2009-06-12 13:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1201 * tests/examples/seek/stepping2.c:
1202 example: fix compile
1204 2009-06-12 13:52:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1206 * tests/examples/seek/Makefile.am:
1207 examples: build the stepping2 example
1209 2009-06-12 13:52:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1211 * gst/playback/gstplaysink.c:
1212 playsink: update for new step API
1214 2009-06-12 13:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1216 * ext/ogg/gstoggdemux.c:
1217 oggdemux: do reverse seeks more accurate
1218 For reverse seeking with the accurate flag set, try to be more precise by
1219 seeking a little bit after the requested position.
1221 2009-06-11 22:32:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1223 * ext/ogg/gstogmparse.c:
1224 * gst/subparse/gstssaparse.c:
1225 * gst/subparse/gstssaparse.h:
1226 * gst/subparse/gstsubparse.c:
1227 * gst/subparse/gstsubparse.h:
1228 subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC
1229 Make subtitle parsers post a taglist with codec tags, so the application
1230 knows what kind of subtitle a subtitle stream is. Fixes #576552.
1232 2009-06-11 19:12:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1234 * gst-libs/gst/audio/gstringbuffer.c:
1235 ringbuffer: handle border cases in resampler
1237 2009-06-11 13:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
1240 * docs/libs/Makefile.am:
1241 * docs/plugins/Makefile.am:
1242 docs: Update common. Use upload-doc.mak instead of upload.mak
1244 2009-06-11 12:39:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1246 * gst-libs/gst/rtp/gstbasertppayload.c:
1249 2009-06-11 12:17:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1251 * gst-libs/gst/audio/gstbaseaudiosink.c:
1252 baseaudiosink: reset accum when dropping samples
1253 When we are resampling and we drop samples because we paused, reset the accum
1254 counter because it's now invalid.
1256 2009-06-11 11:16:15 +0100 Jan Schmidt <thaytan@noraisin.net>
1258 * docs/libs/gst-plugins-base-libs-sections.txt:
1259 * gst-libs/gst/interfaces/mixer.h:
1260 * gst-libs/gst/video/gstbasevideodecoder.h:
1261 docs: Fix a couple of warnings from the docs build.
1263 2009-06-10 21:36:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1265 * gst-libs/gst/audio/testchannels.c:
1266 Don't include config.h multiple times when build audio testchannel app.
1267 Fixes build problem on win32 (#585075).
1269 2009-06-10 16:56:51 +0100 Jan Schmidt <thaytan@noraisin.net>
1271 * gst/playback/gstplaybin2.c:
1272 * gst/playback/gsturidecodebin.c:
1273 playbin2/uridecodebin: Fix connection-speed propagation
1274 uridecodebin expects the passed connection-speed value in kbps, so we
1275 need to divide the value stored in bps by 1000. Also, lower the upper
1276 limit on the properties to the value that we can actually store in our
1277 internal guint (which is plenty high enough)
1279 2009-06-10 14:37:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1281 * gst/subparse/gstsubparse.c:
1282 * tests/check/elements/subparse.c:
1283 subparse: recognise more subrip timestamp variants
1284 Be even less restrictive in what we accept for .srt timestamps when
1285 typefinding and parsing subrip subtitles and add a unit test for
1286 the 'new' format. Fixes #585197.
1288 2009-06-09 22:00:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1290 * gst-libs/gst/rtsp/gstrtsptransport.h:
1291 rtsp: add some more docs
1293 2009-06-09 18:24:55 +0200 Peter Kjellerstedt <pkj@axis.com>
1295 * gst-libs/gst/rtsp/gstrtspmessage.c:
1296 rtsp: Avoid a compiler warning.
1298 2009-06-09 18:23:28 +0200 Peter Kjellerstedt <pkj@axis.com>
1300 * gst-libs/gst/rtsp/gstrtspdefs.h:
1301 rtsp: Updated documentation for GstRTSPResult.
1302 Moved GST_RTSP_ELAST to be last in the documentation to match the actual
1305 2009-05-20 17:30:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1308 autogen: remove -Wno-portability from here
1309 as it is in configure.ac now.
1311 2009-06-09 16:28:20 +0200 Peter Kjellerstedt <pkj@axis.com>
1313 * gst-libs/gst/rtsp/gstrtspconnection.c:
1314 rtsp: Plug a memory leak.
1315 Free memory related to any partially read and/or written RTSP messages.
1317 2009-06-09 12:09:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1319 * gst-libs/gst/audio/gstbaseaudiosink.c:
1320 baseaudiosink: no need to cause discont when clipping
1321 Remove the discont-when-clipping hack now that basesink provides us with
1322 correctly clipped samples when stepping.
1324 2009-06-08 17:26:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1326 * gst-libs/gst/audio/gstbaseaudiosink.c:
1327 audiosink: don't align when we clip
1328 Don't align samples when they were clipped. Not entirely correct but better than
1331 2009-06-08 16:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1333 * tests/examples/seek/.gitignore:
1334 * tests/examples/seek/stepping2.c:
1335 examples: add stepping example in PLAYING
1336 Add stepping example in PLAYING, audio is a bit distorted because basesink does
1337 not provide good clipping info yet.
1339 2009-06-08 10:25:00 +0200 Edward Hervey <bilboed@bilboed.com>
1341 * gst-libs/gst/pbutils/descriptions.c:
1342 pbutils: Add description for hdv/aux-* formats.
1344 2009-06-07 22:20:33 +0400 LRN <lrn1986@gmail.com>
1346 * ext/schroedinger/Makefile.am:
1347 Added libgstbase to schro's LIBADD
1350 2009-06-06 02:15:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1352 * gst-libs/gst/tag/gstid3tag.c:
1353 libgsttag: don't extract genres from empty ID3v1 tags
1354 If we don't have any other info, don't try to interpret the
1355 genre field. In particular we don't want to interpret a genre
1356 of 0 as 'Blues' if no other fields are set and the entire tag
1359 2009-06-05 18:13:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1361 * gst/playback/gstdecodebin2.c:
1362 decodebin2: make sure varargs are of right type
1363 Explicitly cast the variables to g_object_set to their right types.
1365 2009-06-05 16:49:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1367 * gst/playback/gstdecodebin2.c:
1368 decodebin2: increase stream probing queues
1369 When we are probing for streams, we want to set the queue size in such a way
1370 that we can scan a maximum amount of data without consuming too much memory.
1371 Therefore, remove the time limit on the queue and only stop scanning after 2MB
1375 2009-06-05 14:06:17 +0200 Peter Kjellerstedt <pkj@axis.com>
1377 * gst-libs/gst/rtsp/gstrtspconnection.c:
1380 2009-06-05 14:05:54 +0200 Peter Kjellerstedt <pkj@axis.com>
1382 * gst-libs/gst/rtsp/gstrtspconnection.c:
1383 rtsp: Remove an unused variable.
1385 2009-06-05 13:59:14 +0200 Peter Kjellerstedt <pkj@axis.com>
1387 * gst-libs/gst/rtsp/gstrtspconnection.c:
1388 rtsp: Removed duplicate initialization of conn->writefd.
1390 2009-06-05 13:55:08 +0200 Peter Kjellerstedt <pkj@axis.com>
1392 * gst-libs/gst/rtsp/gstrtspconnection.c:
1393 rtsp: Use #defined status codes.
1395 2009-06-05 13:53:29 +0200 Peter Kjellerstedt <pkj@axis.com>
1397 * gst-libs/gst/rtsp/gstrtspconnection.c:
1398 rtsp: Correct gen_tunnel_reply().
1399 Prevent gen_tunnel_reply() from generating an incomplete response
1400 in case an error response code is given.
1402 2009-06-05 10:57:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1405 * win32/common/_stdint.h:
1406 * win32/common/config.h:
1407 * win32/common/video-enumtypes.c:
1408 configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
1409 See #584835. Also update win32 files while we're at it.
1411 2009-06-04 08:57:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1413 * gst/playback/gstplaybin2.c:
1414 playbin2: API: Add {audio,video,text}-tags-changed signals
1417 2009-06-03 20:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1419 * ext/vorbis/vorbisdec.c:
1420 vorbisdec: don't put invalid bitrate values into the taglist
1421 Bitrates are stored as 32-bit signed integers in the vorbis
1422 identification headers, but seem to be read incorrectly,
1423 namely as unsigned 32-bit integers, into the vorbis structure
1424 members which are of type long, which makes our check for
1425 values <= 0 fail with files that put -1 in there for unset
1428 2009-06-03 15:52:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1430 * tests/examples/seek/.gitignore:
1431 ignore: add new stepping app to ignore
1433 2009-06-03 15:31:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1435 * tests/examples/seek/Makefile.am:
1436 * tests/examples/seek/stepping.c:
1437 examples: add stepping example.
1438 Add an example of using playbin2 and frame stepping to simulate variable rate
1439 playback based on a sine wave.
1441 2009-06-03 12:45:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1443 * gst/playback/gstplaybin2.c:
1444 * gst/playback/gstplaysink.h:
1445 playbin2: also set custom text and subp sinks
1446 Set the custom subpicture and text sinks along with the custom audio and video
1448 Fix a little docs blurb too.
1450 2009-06-02 12:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1452 * gst-libs/gst/rtsp/gstrtspconnection.c:
1453 * gst-libs/gst/rtsp/gstrtspconnection.h:
1454 rtsp: add G_LIKELY because we can
1456 2009-06-02 09:53:05 +0200 Edward Hervey <bilboed@bilboed.com>
1458 * gst/typefind/gsttypefindfunctions.c:
1459 typefindfunctions: Fix caps for ogg typefinder.
1461 2009-05-29 11:10:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1463 * docs/libs/gst-plugins-base-libs-sections.txt:
1464 docs: remove some cruft from -sections.txt file
1466 2009-06-01 11:31:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1468 * gst/playback/gstplaysink.c:
1469 * tests/examples/seek/seek.c:
1470 add framestepping to playbin2 and seek
1472 2009-06-01 09:59:22 +0200 Peter Kjellerstedt <pkj@axis.com>
1474 * gst-libs/gst/rtsp/gstrtspconnection.c:
1475 rtsp: Avoid compiler warnings with -Wextra.
1477 2009-06-01 09:58:27 +0200 Peter Kjellerstedt <pkj@axis.com>
1479 * gst-libs/gst/rtsp/gstrtspconnection.h:
1480 rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
1482 2009-06-01 09:43:04 +0200 Peter Kjellerstedt <pkj@axis.com>
1484 * gst-libs/gst/sdp/gstsdpmessage.c:
1485 sdp: Remove an unused variable.
1487 2009-05-30 14:17:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1489 * gst/ffmpegcolorspace/imgconvert.c:
1490 * gst/ffmpegcolorspace/imgconvert_template.h:
1491 ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale
1493 2009-05-29 00:09:15 +0100 Jan Schmidt <thaytan@noraisin.net>
1495 * gst/playback/gstplaybin2.c:
1496 playbin2: Have playbin recognise PGS subpicture streams
1497 Recognise PGS subpicture streams and connect them to the SPU pad
1498 in playsink. Unfortunately this fails badly with negotiation errors
1499 if the SPU is not recent enough to support the stream. I'm not sure
1500 how to add format negotiation in yet.
1502 2009-05-21 23:11:29 +0100 Jan Schmidt <thaytan@noraisin.net>
1504 * gst/playback/gstdecodebin2.c:
1505 * gst/playback/gsturidecodebin.c:
1506 decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.
1508 2009-05-28 20:37:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1510 * gst/playback/gstplaysink.c:
1511 playbin2: fix volume handling for audio sinks without "volume" property
1512 When using an audio sink without a "volume" property, volume control
1513 would only work for the first song. For the next song, we'd try to
1514 re-use the existing audio chain, but inadvertently set chain->volume
1515 to NULL instead of to the existing volume element.
1517 2009-05-28 17:05:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1519 * gst/playback/gstplaysink.c:
1520 playbin2: cosmetic change to avoid unnecessary line breaks
1521 Looks nicer and works around gst-indent silliness.
1523 2009-05-28 17:21:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1525 * gst/playback/gstplaysink.c:
1526 playbin2: don't lose the ref to the volume element
1527 Only release the ref to the volume element when it is controled by a sink. For
1528 software volume we never have to fear that it will change.
1530 2009-05-28 15:21:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1532 * gst/playback/gstplaybin2.c:
1533 * gst/playback/gstplaysink.c:
1534 playbin2: actually use configured audio/video sinks
1535 playbin2 inadvertently used autoaudiosink and autovideosink up to now,
1536 since it would overwrite the sinks configured via the "audio-sink"
1537 and "video-sink" properties with the stream-specific group sinks when
1538 configuring the outputs. Those are usually NULL however, so that would
1539 overwrite the configured sinks with NULL which makes playbin2 then
1540 default to the auto sinks. Fix this by keeping a reference to each
1541 configured sink in playbin2 and setting up the right sinks depending
1542 on whether there is a stream-specific sink or not.
1545 2009-05-27 17:37:38 +0300 Stefan Kost <ensonic@users.sf.net>
1547 * tests/examples/seek/seek.c:
1548 seek: add volume label and sync with sink volume
1549 Look at the volume and have the pulsemixer open at same time. Unfortunately
1550 playbin2 does not emit notify on volume right, so this polls for now.
1552 2009-05-27 18:12:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1554 * gst/playback/gstdecodebin2.c:
1555 decodebin2: remove leftover elements
1556 Remove all of the elements inside decodebin2 when goint to READY and NULL.
1557 Makes decodebin2 reusable.
1560 2009-05-27 15:36:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1562 * gst/playback/gstplaysink.c:
1563 playbin2; release refs to volume/mute properties
1564 Release the refs to the volume and mute property elemens before setting the
1565 child elements to READY or NULL.
1568 2009-05-27 12:10:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1570 * gst/gdp/gstgdppay.c:
1571 gdppay: set caps on outgoing buffers
1572 Set caps on outgoing buffers because NULL caps confuse basetransform.
1575 2009-05-27 11:08:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1577 * gst-libs/gst/netbuffer/gstnetbuffer.c:
1578 netbuffer: also note the order of IP4 addresses
1579 IP4 addresses are also stored in network byte order. Make a note of this in the
1582 2009-05-26 22:43:34 +0200 Alessandro Decina <alessandro.d@gmail.com>
1584 * ext/theora/theoraparse.c:
1585 theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.
1587 2009-05-26 11:13:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1589 * gst-libs/gst/rtsp/gstrtspconnection.c:
1590 Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14"
1591 This reverts commit 418760cf740332c12c3fd9cf3244af134fa9534b.
1592 We now require GLib 2.16.
1594 2009-05-26 15:18:09 +0100 Jan Schmidt <thaytan@noraisin.net>
1599 2009-05-26 15:37:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1601 * gst-libs/gst/netbuffer/gstnetbuffer.c:
1602 netbuffer: document that the port is network order
1603 Document the fact that we store the port number in network order in
1604 GstNetAddress and that the caller should byteswap appropriately.
1606 2009-05-26 15:23:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1608 * gst/videoscale/gstvideoscale.c:
1609 * gst/videoscale/vs_4tap.c:
1610 * gst/videoscale/vs_4tap.h:
1611 * gst/videoscale/vs_image.c:
1612 * gst/videoscale/vs_image.h:
1613 * gst/videoscale/vs_scanline.c:
1614 * gst/videoscale/vs_scanline.h:
1615 videoscale: Add support for 16 bit grayscale in native endianness
1617 2009-05-26 14:58:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1619 * gst/ffmpegcolorspace/avcodec.h:
1620 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
1621 * gst/ffmpegcolorspace/imgconvert.c:
1622 ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian
1624 2009-05-26 14:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1626 * gst/videotestsrc/videotestsrc.c:
1627 * gst/videotestsrc/videotestsrc.h:
1628 videotestsrc: Add support for 16 bit grayscale in native endianness
1630 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
1632 add can-activate-pull property to baseaudiosink
1633 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
1636 2009-05-26 13:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1638 * ext/ogg/gstoggdemux.c:
1639 oggdemux: fix boundary case for seeking.
1640 When we have exactly 0 bytes left to search, make sure we stop instead of going
1641 into an infinite loop.
1643 2009-05-26 11:11:03 +0200 Bastien Nocera <hadess at hadess.net>
1645 * gst-libs/gst/cdda/Makefile.am:
1646 * gst-libs/gst/cdda/gstcddabasesrc.c:
1647 * gst-libs/gst/cdda/sha1.c:
1648 * gst-libs/gst/cdda/sha1.h:
1649 cddabasesrc: Remove copy of sha1 digest
1650 Remove our copy of sha1 digest now that we depend on glib 2.16.
1653 2009-05-25 17:54:01 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
1655 * gst-plugins-base.spec.in:
1658 2009-05-23 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1660 * gst-libs/gst/video/gstbasevideodecoder.c:
1661 * gst-libs/gst/video/gstbasevideoparse.c:
1662 * gst-libs/gst/video/gstbasevideoutils.c:
1663 * gst-libs/gst/video/gstbasevideoutils.h:
1664 * win32/common/libgstvideo.def:
1665 video: don't expose internal gst_adapter_get_buffer() helper function
1666 If it's really needed it should go into GstAdapter in core.
1668 2009-05-22 21:29:51 -0700 David Schleef <ds@schleef.org>
1670 * gst-libs/gst/video/gstbasevideodecoder.c:
1671 basevideo: Fix memleak
1673 2009-05-22 21:27:58 -0700 David Schleef <ds@schleef.org>
1675 * ext/schroedinger/gstschrodec.c:
1676 * ext/schroedinger/gstschroparse.c:
1677 schro: Fix usage of adapter_masked_scan_uint32
1678 Because *somebody* changed the API without telling me.
1680 2009-05-22 21:25:06 -0700 David Schleef <ds@schleef.org>
1682 * ext/schroedinger/gstschro.c:
1683 schro: Change package name to GST_PACKAGE_NAME
1685 2009-05-22 17:34:10 -0700 David Schleef <ds@schleef.org>
1687 * gst-libs/gst/video/gstbasevideoencoder.c:
1688 basevideo: Add preset interface to encoder
1690 2009-05-22 17:31:14 -0700 David Schleef <ds@schleef.org>
1692 * gst/audioresample/gstaudioresample.c:
1693 Run liboil benchmark multiple times
1694 The statistics function requires multiple runs, otherwise
1695 it causes a divide by zero error.
1697 2009-05-22 19:36:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1699 * m4/gst-fionread.m4:
1700 m4: fix 'suspicious cache value' warning for gst-fionread.m4
1701 .. here as well (should really be moved to common, but I'm too lazy).
1703 2009-05-22 17:41:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1705 * ext/vorbis/vorbisdec.c:
1706 vorbisdec: detect and report errors better
1707 Check the return values of a couple more libvorbis functions and post an error
1708 when something is wrong instead of continuing and crashing.
1710 2009-05-22 15:49:14 +0300 Stefan Kost <ensonic@users.sf.net>
1712 * gst/playback/gstplaysink.c:
1713 playbin2: fix initial volume and mute handling
1714 Use two flags to remember volume/mute changes at times when we don't have the
1715 audiochain yet (e.g. construction). Only set values when they were actualy
1716 changed. This makes pulseaudio's stream restore functional.
1718 2009-05-22 10:19:51 +0100 Jan Schmidt <thaytan@noraisin.net>
1721 Automatic update of common submodule
1722 From d3a8fab to 888e0a2
1724 2009-05-22 09:03:22 +0100 Jan Schmidt <thaytan@noraisin.net>
1726 * win32/common/libgstvideo.def:
1727 win32: Remove gst_adapter_masked_scan_uint32 from the exports
1729 2009-05-21 10:48:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1731 * gst-libs/gst/audio/gstbaseaudiosink.c:
1732 audiosink: improve debug message
1734 2009-05-19 18:10:55 -0700 Michael Smith <msmith@songbirdnest.com>
1736 * gst-libs/gst/tag/gstid3tag.c:
1737 gstid3tag: Don't extract a track number unless present.
1738 In ID3v1, a track number is present only if byte 125 is null AND
1739 byte 126 is non-null. If the track number is not present, don't add
1740 a track number tag with value 0.
1742 2009-05-20 00:48:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1744 * gst-libs/gst/video/gstbasevideoutils.c:
1745 * gst-libs/gst/video/gstbasevideoutils.h:
1746 videoutils: remove adapter methods
1747 Remove adapter methods now that they are in core.
1749 2009-05-20 00:42:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1751 * win32/common/libgstvideo.def:
1752 defs: add new symbols
1754 2009-05-19 17:47:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1757 autogen: pass -Wno-portability to automake to suppress warnings
1760 2009-05-19 02:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1762 * docs/libs/.gitignore:
1763 gitignore: remove bogus *.sgml wildcard - these files are tracked in git
1765 2009-05-19 18:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1767 * gst/tcp/gsttcpclientsrc.c:
1768 tcpclientsrc: this is not a live source
1769 Don't mark us as a live source because we are not.
1771 2009-05-19 18:41:02 +0300 Stefan Kost <ensonic@users.sf.net>
1773 * gst/adder/gstadder.c:
1774 adder: only send flush_stop when seek failed
1775 This is still not the ultimate fix. Added some comment to explain the troubles.
1777 2009-05-19 17:17:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1779 * gst-libs/gst/audio/gstbaseaudiosink.c:
1780 audiosink: return the return value of wait_preroll
1781 Return the value that _wait_preroll() returned instead of always WRONG_STATE.
1783 2009-05-19 16:45:56 +0300 Stefan Kost <ensonic@users.sf.net>
1785 * gst/adder/gstadder.c:
1786 * gst/adder/gstadder.h:
1787 adder: send flush_stop to match flush_start
1788 Adder was relying that something else sends a flush stop. When using adder with
1789 a livesource it was not getting a flush_stop and thus all pads downstream where
1790 keept flushing. Mark a pending flush_stop and send it when we are working on
1791 the new segment back in the streaming thread.
1793 2009-05-19 16:02:44 +0300 Stefan Kost <ensonic@users.sf.net>
1795 * tests/examples/seek/seek.c:
1796 seek: ui improvements
1797 Repaint the window black on expose, as this looks nicer when resizing or using
1798 the expander. Also show time after slider, as this saves a whole line (nice on
1801 2009-04-29 18:36:17 +0300 Stefan Kost <ensonic@users.sf.net>
1803 * gst/playback/gstdecodebin.c:
1804 decodebin: use iterators instead of list
1805 The list api is deprecated. Use threadsafe iterators instead.
1807 2009-05-19 15:35:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1809 * gst/playback/gsturidecodebin.c:
1810 uridecodebin: configure caps on decodebin2
1811 Implement the caps property by setting the configured caps on new decodebin2
1815 2009-05-19 15:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1817 * gst/playback/gstdecodebin2.c:
1818 decodebin2: avoid some _caps_ref in some cases
1819 Only mess with the caps refcount when we configure different caps.
1821 2009-05-19 15:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1823 * gst/playback/gsturidecodebin.c:
1824 uridecodebin: fix potential caps leak
1825 Free the user-configured caps in finalize.
1827 2009-05-19 15:20:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1829 * gst/playback/gsturidecodebin.c:
1830 uridecodebin: add queue after cdda://
1831 Add a queue2 after the raw output pads of certain sources such as those for uris
1833 No tuning of the queue is done yet as the defaults seem to work fine for me.
1836 2009-05-19 12:45:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1838 * ext/ogg/gstoggdemux.c:
1839 oggdemux: don't loop when at EOS
1840 When we try to read the last page, don't try to read past the upper boundary, as
1841 this might cause endless loops.
1844 2009-05-19 11:20:19 +0200 Edward Hervey <bilboed@bilboed.com>
1846 * gst/audioresample/gstaudioresample.c:
1847 audioresample: Don't drain remaining buffers after a flush.
1848 If we were resetted (due to a flush), we can not drain the remaining
1849 buffers since they would be pushed before a valid new newsegment event.
1851 2009-05-18 22:29:07 -0700 Michael Smith <msmith@syncword.(none)>
1853 * ext/theora/theoradec.c:
1854 theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.
1856 2009-05-19 01:13:34 +0300 Stefan Kost <ensonic@users.sf.net>
1858 * gst/adder/gstadder.c:
1859 adder: add more logging and return value checking
1861 2009-05-19 01:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
1863 * gst/adder/gstadder.c:
1864 adder: handle the return value from iterator_fold
1866 2009-05-19 01:03:44 +0300 Stefan Kost <ensonic@users.sf.net>
1868 * gst/adder/gstadder.c:
1869 adder: use the pad in logging as objects
1870 Helps to differenciate between source and sinks pads.
1872 2009-04-21 22:54:19 +0300 Stefan Kost <ensonic@users.sf.net>
1874 * tests/examples/seek/seek.c:
1875 seek: use parser for mp3 and rename variable
1877 2009-05-18 11:08:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1879 * tests/examples/seek/seek.c:
1880 seek: add playbin2 options in expander
1881 Add the playbin2 stream selection options inside an expander to preserve some
1884 2009-02-10 15:29:10 -0800 David Schleef <ds@schleef.org>
1886 * gst/videotestsrc/videotestsrc.c:
1887 videotestsrc: Add support for v210 and v216 formats
1889 2009-05-15 16:21:15 -0700 David Schleef <ds@schleef.org>
1891 * gst-libs/gst/video/gstbasevideocodec.c:
1892 * gst-libs/gst/video/gstbasevideodecoder.c:
1893 * gst-libs/gst/video/gstbasevideoencoder.c:
1894 * gst-libs/gst/video/gstbasevideoparse.c:
1895 video: remove // comments
1897 2009-05-15 16:18:18 -0700 David Schleef <ds@schleef.org>
1899 * gst-libs/gst/video/video.c:
1900 * gst-libs/gst/video/video.h:
1901 video: Add Y444, v210, v216 formats
1903 2009-05-15 16:12:37 -0700 David Schleef <ds@schleef.org>
1907 * ext/schroedinger/Makefile.am:
1908 * ext/schroedinger/gstschro.c:
1909 * ext/schroedinger/gstschrodec.c:
1910 * ext/schroedinger/gstschroenc.c:
1911 * ext/schroedinger/gstschroparse.c:
1912 * ext/schroedinger/gstschroutils.c:
1913 * ext/schroedinger/gstschroutils.h:
1914 schro: Move schro plugin from Schroedinger
1915 Previous history is in Schroedinger. Depends on, and is an example
1916 of using, GstBaseVideo* base classes.
1917 Code was reindented, and an #ifdef HAVE_ENCODER removed.
1919 2009-05-15 10:23:08 -0700 David Schleef <ds@schleef.org>
1921 * gst-libs/gst/video/Makefile.am:
1922 * gst-libs/gst/video/gstbasevideocodec.c:
1923 * gst-libs/gst/video/gstbasevideocodec.h:
1924 * gst-libs/gst/video/gstbasevideodecoder.c:
1925 * gst-libs/gst/video/gstbasevideodecoder.h:
1926 * gst-libs/gst/video/gstbasevideoencoder.c:
1927 * gst-libs/gst/video/gstbasevideoencoder.h:
1928 * gst-libs/gst/video/gstbasevideoparse.c:
1929 * gst-libs/gst/video/gstbasevideoparse.h:
1930 * gst-libs/gst/video/gstbasevideoutils.c:
1931 * gst-libs/gst/video/gstbasevideoutils.h:
1932 video: Copy BaseVideo classes from Schroedinger
1934 2009-05-15 23:05:45 +0200 Arnout Vandecappelle <arnout@mind.be>
1936 * gst/tcp/gstmultifdsink.c:
1937 multifdsink: add num-fds property
1938 multifdsink::num-fds
1940 2009-05-15 20:36:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1942 * gst-libs/gst/pbutils/descriptions.c:
1943 pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
1945 2009-05-14 11:44:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1947 * ext/vorbis/vorbisenc.c:
1948 vorbisenc: Implement Preset interface
1950 2009-05-14 11:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1952 * ext/theora/theoraenc.c:
1953 theoraenc: Implement Preset interface
1955 2009-05-14 11:41:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1957 * ext/ogg/gstoggmux.c:
1958 oggmux: Implement Preset interface
1960 2009-05-14 21:37:22 +0100 Jan Schmidt <thaytan@noraisin.net>
1962 * gst/playback/gstplaysink.c:
1963 playbin2: Fix cdda:// playback
1964 Don't send async-start when the playsink has already been configured
1965 before changing state.
1967 2009-05-14 01:31:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1970 configure: require core CVS for gst_adapter_prev_timestamp()
1971 which is used in the libvisual plugin.
1973 2009-04-22 18:34:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1976 AUTHORS: fix my email
1978 2009-04-22 18:35:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1980 * gst-libs/gst/audio/gstaudioclock.c:
1981 audioclock: make our internal time monotonic
1982 Make the internal time increase monotonically.
1984 2009-05-13 19:27:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1986 * ext/libvisual/visual.c:
1987 visual: remove next_ts variable
1988 We can remove the next_ts variable as we don't use it anymore.
1990 2009-05-13 19:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1992 * ext/libvisual/visual.c:
1993 visual: use new adapter timestamp code
1994 Use the new adapter timestamp tracking code to make things easier and produce
1995 vastly better output timestamps.
1997 2009-05-13 01:35:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2029 po: avoid conflicts of local *.po files with files in git
2030 Make it so that filenames and line numbers are only stored in the *.pot file
2031 (which is not in git), but not in the individual *.po files. This information
2032 is hardly useful for translators in our case, and it should avoid the constant
2033 conflicts of local *.po files with the ones in git which are caused by the
2034 source files changing and the line numbers being updated. This commit might
2035 cause one last merge conflict for you, which you can work around with
2036 "git checkout po/*.po" before merging or pulling. After that there should
2037 (hopefully) not be any more local modifications of these files (unless
2038 someone committed additions or changes to translated strings and the
2039 *.po files haven't been updated yet, that is).
2041 2009-05-12 23:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2043 * tests/check/elements/.gitignore:
2044 * tests/check/elements/audioresample.c:
2045 tests: fix audioresample unit test on big endian architectures
2046 Don't hardcode endianness=1234 in the filtercaps, it will cause
2047 pad link failures which will result in the test timing out.
2049 2009-05-12 17:18:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2051 * gst/audiotestsrc/gstaudiotestsrc.c:
2052 audiotestsrc: fix broken enum nick - it should have a hyphen
2053 The enum nick should be 'sine-table', not 'sine table'. Technically this is
2054 an API/ABI change I guess, but anyone who was using this and didn't report
2057 2009-05-01 01:04:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2059 * gst/audiotestsrc/gstaudiotestsrc.c:
2060 audiotestsrc: seek to the requested byte offset, not the expected byte offset
2062 2009-05-01 01:03:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2064 * gst/audiotestsrc/gstaudiotestsrc.c:
2065 * gst/audiotestsrc/gstaudiotestsrc.h:
2066 audiotestsrc: support more than just one channel
2068 2009-05-12 15:52:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2070 * gst-libs/gst/interfaces/propertyprobe.h:
2071 propertyprobe: Fix typo in the docs
2073 2009-05-12 12:17:55 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
2075 * ext/ogg/gstoggmux.c:
2076 * ext/theora/theora.c:
2077 * ext/vorbis/vorbis.c:
2078 Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder
2080 2009-04-30 16:37:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2082 * gst/videorate/gstvideorate.c:
2083 * gst/videorate/gstvideorate.h:
2084 videorate: handle invalid timestamps better
2085 Handle buffers with -1 timestamps better by keeping track of the en time of the
2086 previous buffer and assuming the -1 timestamp buffer goes right after the
2088 when we have two buffers that are equally good, output the oldest buffer once to
2090 don't try to calculate latency when the input framerate is unknown.
2092 2009-04-28 11:37:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2094 * ext/ogg/gstoggmux.c:
2095 oggmux: small debug statement in DISCONT
2097 2009-04-28 11:24:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2099 * ext/ogg/gstoggdemux.c:
2100 * ext/ogg/gstoggdemux.h:
2101 oggdemux: fix abuse of ogg API, handle broken oggs
2102 When we feed the ogg sync layer, we need to feed it contiguous data even if the
2103 sync layer did not consume all of it yet. This makes sure that it always finds
2104 the next page even for more corrupted files. Use a different read_offset for
2105 this purpose. since we now keep track of the sync layer, we don't have to reset
2106 after finding a start of a page.
2107 Add some more debug info for the error paths.
2108 Only reset the sync layer when we perform a seek operation.
2109 Avoid failure when the next chain has no bos pages but instead simply ignore it.
2110 when we receive unknown page serial numbers mid stream, don't fail but post a
2111 warning and hope that we get back on track later.
2114 2009-04-30 16:41:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2116 * gst/playback/gstdecodebin2.c:
2117 decodebin2: make subpictures a raw output format
2118 Subpictures are a raw format, we want those pads exposed so that playbin2 can do
2119 the subpicture mixing.
2121 2009-04-27 10:15:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2123 * gst-libs/gst/rtp/gstbasertppayload.c:
2124 * gst-libs/gst/rtp/gstbasertppayload.h:
2125 rtpdepay: add some more comments
2127 2009-04-17 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2129 * gst-libs/gst/audio/gstaudioclock.c:
2130 audioclock: make sure values are ever increasing
2132 2009-05-05 17:17:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2134 * gst/playback/gstplaysink.c:
2135 playbin2: make fallback identity silent
2136 Set the signal-handoffs to FALSE and silent to TRUE for the fallback identity
2137 element so that it consumes less CPU.
2139 2009-04-17 10:57:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2141 * gst/playback/gstplaybin2.c:
2142 * gst/playback/gstplaysink.c:
2143 playbin2: handle custom audiosinks differently
2144 Keep track of the autoplugged custom sinks and configure them in the playsink
2145 element when we have collected all streams.
2146 Also make sure that we only select one custom sink.
2147 When unreffing the internal sink, we don't need to change the state to NULL.
2149 2009-05-12 10:36:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2151 * gst/playback/gstplaybin2.c:
2152 * gst/playback/gstplaysink.c:
2153 * gst/playback/gstplaysink.h:
2154 playbin2: unify custom sink get/set functions
2155 Use one function to set/get all of the different sink types.
2156 cleanup up the subpicture chain too.
2157 Allow setting a custom subpicture sink.
2159 2009-05-11 18:29:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2161 * gst-libs/gst/interfaces/tunernorm.h:
2162 interfaces: Seperate some more struct definitions from typedefs
2164 2009-05-11 15:48:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2166 * gst-libs/gst/interfaces/navigation.h:
2167 * gst-libs/gst/interfaces/videoorientation.h:
2168 * gst-libs/gst/interfaces/xoverlay.h:
2169 interfaces: Seperate some more struct definitions from typedefs
2171 2009-05-10 17:28:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2173 * win32/common/libgstinterfaces.def:
2174 Add new functions to win32 exports
2176 2009-05-10 17:28:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2178 * docs/libs/gst-plugins-base-libs-sections.txt:
2179 Add new functions to the docs
2181 2009-05-10 17:25:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2183 * gst-libs/gst/interfaces/mixer.c:
2184 * gst-libs/gst/interfaces/mixer.h:
2185 interfaces: API: Add gst_mixer_get_mixer_type()
2186 This is a convenience function that returns the mixer_type
2187 of the interface struct.
2189 2009-05-10 17:25:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2191 * gst-libs/gst/interfaces/colorbalance.c:
2192 interfaces: Add docs for gst_color_balance_get_balance_type()
2194 2009-05-10 11:17:19 +0200 Marc-Andre Lureau <marcandre.lureau@gmail.com>
2197 Run libtoolize before aclocal
2198 This unbreaks the build in some cases. Fixes bug #582021
2200 2009-05-07 17:38:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2202 * ext/pango/gsttextrender.c:
2203 textrender: Correctly initialize the background for ARGB too
2205 2009-05-07 16:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2207 * ext/pango/gsttextrender.c:
2208 * ext/pango/gsttextrender.h:
2209 textrender: Use libgstvideo functions to create caps
2210 Also check if downstream wants ARGB always when we get
2213 2009-05-07 16:52:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2215 * ext/pango/gsttextrender.c:
2216 textrender: Don't always use ARGB if downstream supports it but take it's preference
2218 2009-05-07 16:48:08 +0200 Kapil Agrawal <kapil@mediamagictechnologies.com>
2220 * ext/pango/gsttextrender.c:
2221 * ext/pango/gsttextrender.h:
2222 textrender: Add support for ARGB and alignment properties
2225 2009-05-07 16:42:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2227 * ext/pango/gsttextrender.c:
2228 textrender: Add ; after GST_BOILERPLATE to fix indention
2230 2009-05-07 15:10:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2232 * gst-libs/gst/tag/gstvorbistag.c:
2233 vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
2235 2009-05-07 14:59:36 +0200 Arnout Vandecappelle <arnout@mind.be>
2237 * gst/typefind/gsttypefindfunctions.c:
2238 typefindfunctions: made mp3_type_find less aggressive
2239 mp3_type_find could suggest already when only a single valid header
2240 was found, if it ran out of data before the end of the next frame.
2241 Therefore, ignore the last found frame if it was incomplete.
2244 2009-05-07 14:48:29 +0200 John Millikin <jmillikin@gmail.com>
2246 * gst-libs/gst/tag/gstvorbistag.c:
2247 vorbistag: Store cover art in vorbiscomments
2250 2009-05-07 06:14:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2252 * gst-libs/gst/interfaces/colorbalance.c:
2253 * gst-libs/gst/interfaces/colorbalance.h:
2254 interfaces: API: Add gst_color_balance_get_balance_type()
2255 This is a convenience function that returns the balance_type
2256 of the interface struct.
2258 2009-05-06 17:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2260 * gst-libs/gst/interfaces/colorbalance.h:
2261 * gst-libs/gst/interfaces/colorbalancechannel.h:
2262 * gst-libs/gst/interfaces/tuner.h:
2263 * gst-libs/gst/interfaces/tunerchannel.h:
2264 interfaces: Separate struct definitions from typedefs
2266 2009-05-06 14:03:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2268 * pkgconfig/gstreamer-app-uninstalled.pc.in:
2269 Fix libdir for uninstalled gstreamer-app library
2271 2009-05-12 01:59:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2273 * gst-libs/gst/pbutils/descriptions.c:
2274 pbutils: add description for APE tag caps
2276 2009-05-12 01:35:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2279 configure: bump core requirement to last release
2280 as that's more likely to be true than that we need
2283 2009-05-12 01:21:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2287 configure: rename CVS -> git in a couple of places
2289 2009-05-12 01:17:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2292 configure: bump GLib requirement to GLib >= 2.16
2293 as per the New Regime (see wiki).
2295 2009-05-01 00:09:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2297 * gst-libs/gst/tag/gsttagdemux.c:
2298 tagdemux: cache events from upstream and re-send them once we have a source pad
2299 Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
2302 2009-05-07 14:07:44 -0700 Michael Smith <msmith@songbirdnest.com>
2304 * gst-libs/gst/riff/riff-media.c:
2305 riff: support UYVY raw 4:2:2 in riff.
2307 2009-05-11 21:20:07 +0100 Jan Schmidt <thaytan@noraisin.net>
2310 Back to development -> 0.10.23.1
2312 2009-04-27 22:42:55 -0700 Michael Smith <msmith@syncword.(none)>
2314 * ext/theora/theoradec.c:
2315 theoradec: fix buffer overrun on 422 decode.
2317 2009-04-27 21:39:01 -0700 Michael Smith <msmith@syncword.(none)>
2319 * ext/theora/theoradec.c:
2320 theoradec: 444 support.
2322 2009-04-27 21:30:04 -0700 Michael Smith <msmith@syncword.(none)>
2324 * ext/theora/theoradec.c:
2325 theoradec: handle 422 images (as YUY2).
2327 2009-04-27 21:01:51 -0700 Michael Smith <msmith@syncword.(none)>
2329 * ext/theora/gsttheoradec.h:
2330 * ext/theora/theoradec.c:
2331 theoradec: rearrange code in preparation for 422 and 444 support.
2333 === release 0.10.23 ===
2335 2009-05-10 23:57:01 +0100 Jan Schmidt <thaytan@noraisin.net>
2341 * docs/plugins/gst-plugins-base-plugins.args:
2342 * docs/plugins/gst-plugins-base-plugins.hierarchy:
2343 * docs/plugins/gst-plugins-base-plugins.interfaces:
2344 * docs/plugins/gst-plugins-base-plugins.prerequisites:
2345 * docs/plugins/gst-plugins-base-plugins.signals:
2346 * docs/plugins/inspect/plugin-adder.xml:
2347 * docs/plugins/inspect/plugin-alsa.xml:
2348 * docs/plugins/inspect/plugin-app.xml:
2349 * docs/plugins/inspect/plugin-audioconvert.xml:
2350 * docs/plugins/inspect/plugin-audiorate.xml:
2351 * docs/plugins/inspect/plugin-audioresample.xml:
2352 * docs/plugins/inspect/plugin-audiotestsrc.xml:
2353 * docs/plugins/inspect/plugin-cdparanoia.xml:
2354 * docs/plugins/inspect/plugin-decodebin.xml:
2355 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
2356 * docs/plugins/inspect/plugin-gdp.xml:
2357 * docs/plugins/inspect/plugin-gio.xml:
2358 * docs/plugins/inspect/plugin-gnomevfs.xml:
2359 * docs/plugins/inspect/plugin-libvisual.xml:
2360 * docs/plugins/inspect/plugin-ogg.xml:
2361 * docs/plugins/inspect/plugin-pango.xml:
2362 * docs/plugins/inspect/plugin-playback.xml:
2363 * docs/plugins/inspect/plugin-queue2.xml:
2364 * docs/plugins/inspect/plugin-subparse.xml:
2365 * docs/plugins/inspect/plugin-tcp.xml:
2366 * docs/plugins/inspect/plugin-theora.xml:
2367 * docs/plugins/inspect/plugin-typefindfunctions.xml:
2368 * docs/plugins/inspect/plugin-uridecodebin.xml:
2369 * docs/plugins/inspect/plugin-video4linux.xml:
2370 * docs/plugins/inspect/plugin-videorate.xml:
2371 * docs/plugins/inspect/plugin-videoscale.xml:
2372 * docs/plugins/inspect/plugin-videotestsrc.xml:
2373 * docs/plugins/inspect/plugin-volume.xml:
2374 * docs/plugins/inspect/plugin-vorbis.xml:
2375 * docs/plugins/inspect/plugin-ximagesink.xml:
2376 * docs/plugins/inspect/plugin-xvimagesink.xml:
2377 * gst-plugins-base.doap:
2378 * win32/common/_stdint.h:
2379 * win32/common/config.h:
2382 2009-05-10 23:56:05 +0100 Jan Schmidt <thaytan@noraisin.net>
2415 2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
2447 * win32/common/_stdint.h:
2448 * win32/common/config.h:
2449 0.10.22.6 pre-release
2451 2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2453 * gst/playback/gstplaysink.c:
2454 playbin2: fix resume after pause
2455 Don't ignore the state change of the children, they might be doing an ASYNC
2458 2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
2491 0.10.22.5 pre-release
2493 2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2495 * gst/tcp/gstmultifdsink.c:
2496 * gst/tcp/gsttcp-marshal.list:
2497 multifdsink: fix signature of the add-full signal
2498 The second parameter is a GstSyncMethod enum, not a boolean.
2500 2009-05-07 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2502 * gst/playback/gstplaysink.c:
2503 playsink: initialize variable too
2505 2009-05-07 14:28:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2507 * gst/playback/gstplaysink.c:
2508 playbin2: make playsink go ASYNC to PAUSED
2509 Make playsink go async to the PAUSED state instead of relying on uridecodebin
2510 for async behaviour in playbin. This solves some problems (mainly with DVD)
2511 where the pipeline would go to PLAYING before preroll completed, failing to
2512 select the audiosink clock.
2515 2009-05-06 16:09:52 +0100 Jan Schmidt <thaytan@noraisin.net>
2547 * win32/common/_stdint.h:
2548 * win32/common/config.h:
2549 0.10.22.4 pre-release
2551 2009-05-06 13:19:34 +0100 Zaheer Merali <zaheerabbas@merali.org>
2553 * ext/theora/theoraenc.c:
2554 * ext/vorbis/vorbisenc.c:
2555 vorbisenc, theoraenc: Ensure gp is computed consistently + clip to segment
2556 With vorbisenc, compute the granulepos with running time and clip incoming
2558 With theoraenc, drop out of segment buffers.
2560 2009-05-01 16:47:53 +0100 Jan Schmidt <thaytan@noraisin.net>
2562 * gst/audioresample/gstaudioresample.c:
2563 audioresample: Fix buffer size transformations
2564 When calculating the input/output buffer sizes in the transform_size function,
2565 take the number of channels into account, so we don't end up calculating
2566 a buffer size that only contains a partial number of audio frames.
2567 Also, when going from output size to input size, round down rather than
2568 up, so as to calculate the minimum number of samples that *might* yield
2569 a buffer of the intended destination size.
2570 Fixes: #580470 and #580952
2572 2009-04-29 16:45:27 +0100 Jan Schmidt <thaytan@noraisin.net>
2574 * ext/vorbis/gstvorbisenc.h:
2575 * ext/vorbis/vorbisenc.c:
2576 vorbisenc: Ensure output buffers fall within the segment
2577 Add the start position of the first segment to the running time
2578 used to generate buffer timestamps in vorbisenc. This avoids generating
2579 buffers which fall outside the initial segment. The element segment
2580 handling requires more extensive fixing, but this at least prevents
2581 regressions. Fixes: #580020
2583 2009-04-29 11:18:42 +0200 Andy Wingo <wingo@oblong.net>
2585 * gst-libs/gst/audio/gstbaseaudiosink.c:
2586 Revert "add can-activate-pull property to baseaudiosink"
2587 This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.
2589 2009-04-29 11:18:33 +0200 Andy Wingo <wingo@oblong.net>
2591 * gst-libs/gst/audio/gstbaseaudiosink.c:
2592 Revert "[baseaudiosink] add docs for can-activate-pull"
2593 This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.
2595 2009-04-28 18:48:33 +0200 Andy Wingo <wingo@oblong.net>
2597 [baseaudiosink] add docs for can-activate-pull
2598 * gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
2601 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
2603 add can-activate-pull property to baseaudiosink
2604 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
2607 2009-04-28 11:32:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2609 * gst/videorate/gstvideorate.c:
2610 * gst/videorate/gstvideorate.h:
2611 videorate: clear discont on duplicated buffers
2612 When videorate duplicates a buffer with a DISCONT flag, it copies the discont on
2613 the first pushed buffer but fails to clear it for subsequent buffers. This
2614 causes theoraenc!oggmux and possibly other elements to consider this a discont
2616 Fix videorate to produce discont as the first buffer and after a flushing seek.
2619 2009-04-24 18:13:00 +0100 Jan Schmidt <thaytan@noraisin.net>
2621 * tests/check/Makefile.am:
2622 check: Disable the playbin2 for this release, as it is a bit racy.
2623 Disable the test, as per the discussion in #580120. Needs re-enabling
2624 after the release, when playbin2 is fixed.
2626 2009-04-23 08:41:19 +0200 Edward Hervey <bilboed@bilboed.com>
2628 * gst/playback/gstdecodebin2.c:
2629 decodebin2: Don't reduce max-size-time of exposed groups. Fixes #579912
2630 The 2s limit is way too small for a lot of files (which have an interleave
2631 in time of between 3 and 5s). Instead, leave it to the initial 5s value
2632 and reduce the other limits (allowing us to stay memory-efficient).
2634 2009-04-21 21:06:59 +0100 Jan Schmidt <thaytan@noraisin.net>
2666 * win32/common/_stdint.h:
2667 * win32/common/config.h:
2668 0.10.22.3 pre-release
2670 2009-04-21 20:41:23 +0100 René Stadler <mail@renestadler.de>
2672 * gst/audioresample/gstaudioresample.c:
2673 audioresample: Fix unused variable in compilation with --disable-gst-debug
2676 2009-04-21 22:12:28 +0100 Jan Schmidt <thaytan@noraisin.net>
2679 Automatic update of common submodule
2680 From b3941ea to 6ab11d1
2682 2009-04-21 20:57:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2684 * gst/playback/gstplaybasebin.c:
2685 playbin: only use raw_decoding_mode when it's true
2686 First check the pad caps if they are raw before setting the raw_decoding_mode to
2687 TRUE. Fixes playback of transport streams and other streams that require large
2691 2009-04-19 18:15:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2693 * gst-libs/gst/cdda/gstcddabasesrc.c:
2694 * tests/check/libs/cddabasesrc.c:
2695 cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core
2696 Don't use REPLACE_ALL merge mode when that's not really what we want,
2697 as now that REPLACE_ALL actually does what it's supposed to do in
2698 core, we drop tags we wanted to keep, such as the various disc id
2699 tags. Add unit test for this as well. Fixes #579463.
2701 2009-04-17 10:34:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2703 * gst-libs/gst/rtsp/gstrtspconnection.c:
2704 rtspconnection: don't use GLib-2.16 API, we require only 2.14
2707 2009-04-17 10:55:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2709 * gst-libs/gst/audio/gstbaseaudiosink.c:
2710 baseaudiosink: don't unparent the ringbuffer
2711 when going to NULL, don't unparent the ringbuffer because we don't support going
2712 back to 0 very well yet.
2715 2009-04-17 10:53:10 +0200 Olivier Crete <tester at tester.ca>
2717 * gst-libs/gst/rtp/gstrtcpbuffer.c:
2718 RTCP: don't fail when retrieving invalid PT
2719 We can't meaningfully assert on valid packet types so just return the type as it
2720 is. Update the comments to reflect this.
2723 2009-04-16 12:12:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2725 * docs/libs/gst-plugins-base-libs-sections.txt:
2726 * gst-libs/gst/app/gstappsink.h:
2727 * gst-libs/gst/app/gstappsrc.h:
2728 app: add trivial cast macros
2729 Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
2730 and add the macros to the standard macros in the docs.
2733 2009-04-16 12:09:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2735 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
2736 pkgconfig: add the app/ directory to Libs
2737 Add the appsrc/appsink directory to the Libs in the uninstalled
2738 pkgconfig file so that one can build against it.
2741 2009-04-15 22:59:31 +0100 Jan Schmidt <thaytan@noraisin.net>
2744 0.10.22.2 pre-release
2746 2009-04-15 22:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
2749 ChangeLog: regenerate changelog with the gen-changelog script
2751 2009-04-16 00:41:13 +0100 Jan Schmidt <thaytan@noraisin.net>
2782 po: Update po files from TP
2784 2009-04-16 00:40:59 +0100 Jan Schmidt <thaytan@noraisin.net>
2786 * win32/common/_stdint.h:
2787 * win32/common/config.h:
2788 * win32/common/gstrtsp-enumtypes.c:
2789 * win32/common/interfaces-enumtypes.c:
2790 * win32/common/interfaces-enumtypes.h:
2791 * win32/common/video-enumtypes.c:
2792 win32: Update win32 build files
2794 2009-04-16 00:31:55 +0100 Jan Schmidt <thaytan@noraisin.net>
2796 * tests/check/libs/video.c:
2797 check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.
2799 2009-04-16 00:31:00 +0100 Jan Schmidt <thaytan@noraisin.net>
2801 * tests/check/elements/playbin2.c:
2802 check: Fix the input uri in playbin2 test.
2803 Don't try and use a random file in wim's home directory as a test input
2805 2009-04-15 15:35:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2807 * gst-libs/gst/video/video.h:
2808 video: Fix typo in the docs
2810 2009-04-15 14:53:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2812 * gst-libs/gst/video/video.c:
2813 * gst-libs/gst/video/video.h:
2814 video: Add support for YVYU YUV colorspace
2816 2009-04-15 00:17:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2818 * docs/libs/gst-plugins-base-libs-docs.sgml:
2819 * gst-libs/gst/fft/gstfft.c:
2820 docs: fix hyperlink and move fft attribution to the right place
2822 2009-04-15 00:02:39 +0300 Stefan Kost <ensonic@users.sf.net>
2824 * gst-libs/gst/audio/gstbaseaudiosink.c:
2825 log: use G_GUINT64_FORMAT instead of llu
2827 2009-04-14 18:31:52 +0200 Josep Torra <n770galaxy at gmail.com>
2829 * gst-libs/gst/rtsp/gstrtspdefs.c:
2830 * gst-libs/gst/rtsp/gstrtspdefs.h:
2831 RTSP: add missing headers for WMS RTSP
2832 Add missing headers related to Windows Media RTSP extension.
2835 2009-04-14 18:16:37 +0200 Olivier Crete <tester at tester.ca>
2837 * docs/design/draft-keyframe-force.txt:
2838 * ext/theora/gsttheoraenc.h:
2839 * ext/theora/theoraenc.c:
2840 theoraenc: implement upstream keyframe force
2841 Implement handling of upstream keyframe forcing.
2842 Update the design documents too.
2845 2009-04-14 17:31:31 +0200 Olivier Crete <tester at tester.ca>
2847 * ext/theora/theoraenc.c:
2848 theoraenc: factor out keyframe forcing
2851 2009-04-14 17:01:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2854 * gst-libs/gst/fft/gstfft.c:
2855 Give credit to Mark Borgerding (kissfft author)
2856 and add myself to AUTHORS as well. Fixes #575638.
2858 2009-04-14 17:04:06 +0200 Jan Urbanski <j.urbanski at students.mimuw.edu.pl>
2860 * gst/tcp/gstmultifdsink.c:
2861 * gst/tcp/gstmultifdsink.h:
2862 multifdsink: add property to resend streamheaders
2863 Adds a new property in multifdsink, resend-streamheader.
2864 If this property is false, the multifdsink will not send the streamheader if
2865 there's already one set for a particular client.
2866 There are some formats in which every stream needs to start with a certain
2867 blob, but you can't inject this blob at leisure. If the producer wants to
2868 change the blob in question and sets in as the streamheader on the outgoing
2869 buffers' caps, new clients of multifdsink will get the new streamheader, but
2870 old clients will break, because they'll see the blob in the middle of the
2872 The property is true by default, so existing code will not see any difference.
2875 2009-04-14 16:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2877 * gst/tcp/gstmultifdsink.c:
2878 * gst/tcp/gstmultifdsink.h:
2879 multifdsink: add property to handle client write
2880 Add a property to disable listening to client writes. This property is usefull
2881 when other code will deal with reading from the client socket.
2882 API: GstMultiFdSink::handle-read property
2884 2009-04-14 16:45:20 +0200 Johann Prieur <johann.prieur at gmail.com>
2886 * docs/libs/gst-plugins-base-libs-sections.txt:
2887 * gst-libs/gst/rtp/gstrtcpbuffer.c:
2888 * gst-libs/gst/rtp/gstrtcpbuffer.h:
2889 * win32/common/libgstrtp.def:
2890 RTCP: add beginnings of Feedback messages
2891 Add the beginnings of parsing and constructing Feedback messages.
2894 2009-04-14 13:51:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2896 * gst/playback/gstplaysink.c:
2897 playbin2: clear the target
2898 Clear the target of our ghostpads before we remove the pad from the element.
2899 This to make sure that the internal pad is not left linked to whatever pad we
2900 were ghosted to. This should only be a problem when we leak the ghostpads.
2901 Also release our subpicture pads.
2904 2009-04-14 12:10:30 +0100 Hannes Bistry <hannesb@gmx.net>
2906 * sys/ximage/ximagesink.c:
2907 ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
2910 2009-04-14 13:16:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2912 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2913 baseaudiosrc: adjust the internal timestamp
2914 Adjust the internal timestamp before comparing it against the adjusted clock
2918 2009-04-14 13:12:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2920 * gst-libs/gst/audio/gstbaseaudiosink.c:
2921 baseaudiosink: use new clock time methods
2922 Use the unadjusted internal clock times to calculate the internal/external
2923 offset when calibrating the clock.
2924 When going to NULL, unparent and free the ringbuffer, like we do in the source
2928 2009-04-14 13:08:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2930 * gst-libs/gst/audio/gstaudioclock.c:
2931 * gst-libs/gst/audio/gstaudioclock.h:
2932 * win32/common/libgstaudio.def:
2933 audioclock: add methods for the internal offset
2934 Add two methods for getting the unadjusted time of the clock and one for
2935 adjusting an internal time. We will need these methods for correctly handling
2936 the time after a gst_audio_clock_reset().
2937 Add a debug category and some debug lines to the audio clock.
2938 API: gst_audio_clock_get_time()
2939 API: gst_audio_clock_adjust()
2940 API: GST_AUDIO_CLOCK_CAST()
2942 2009-04-14 11:34:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2944 * gst/playback/gstdecodebin2.c:
2945 decodebin2: fix up the debugs and warnings
2946 Use _OBJECT variants because we can. Go over some log statements and put them in
2950 2009-04-12 22:26:33 +0200 Luca Ognibene <luca.ognibene at gmail.com>
2952 * gst/tcp/gstmultifdsink.c:
2953 multifdsink: fix error in sync-method
2954 Multifdsink did not handle sync-method=latest-keyframe correctly when the
2955 soft-limit is set to -1 (unlimited).
2958 2009-04-10 21:49:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2960 * gst-libs/gst/audio/gstbaseaudiosink.c:
2961 baseaudiosink: use the internal clock time
2962 We can't assume that the internal clock time is the same as the function we
2963 installed on our provided clock because somebody might have changed it.
2965 2009-04-10 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2967 * tests/examples/seek/seek.c:
2968 seek: handle clock-lost messages
2969 When we receive a clock-lost message we need to pause and play to select a new
2972 2009-04-10 13:44:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2974 * tests/check/Makefile.am:
2975 * tests/check/elements/playbin2.c:
2976 check: add a unit test for playbin2
2977 Add unit test for playbin2 and include the refcount test in #577794.
2979 2009-04-10 13:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2981 * gst/playback/gstplaysink.c:
2982 playbin2: fix refcounting of visualisations
2985 2009-04-10 13:27:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2987 * gst/playback/gstplaysink.c:
2988 playsink: fix refcounting of custom elements
2989 Sink the custom sinks, let other elements we create be sunken by the bin we add
2993 2009-04-10 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2995 * tests/check/elements/appsink.c:
2996 check: fix appsink test
2997 Fix the appsink test now that the method signature changed.
2999 2009-04-10 12:26:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3001 * gst/playback/gstplaybin2.c:
3002 playbin2: handle missing input-selector
3003 Gracefully degrade and disable stream selection when input-selector is
3006 2009-04-09 23:46:17 +0200 Martin Samuelsson <martin.samuelsson at axis.com>
3008 * gst-libs/gst/app/gstappsink.c:
3009 * gst-libs/gst/app/gstappsink.h:
3010 appsink: make callbacks return GstFlowReturn
3011 Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
3012 errors can be reported properly.
3015 2009-04-09 18:04:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3017 * gst-libs/gst/audio/gstringbuffer.c:
3018 * gst-libs/gst/audio/gstringbuffer.h:
3019 ringbuffer: allow for custom commit functions
3020 Allow subclasses to override the commit method.
3022 2009-04-08 18:04:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3024 * gst-libs/gst/audio/gstbaseaudiosink.c:
3025 baseaudiosink: fix a small glitch after pause
3026 After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
3027 the amount of output samples we consumed. We can't do this reliably with the
3028 current API when we are doing trick modes but we can do the right thing for
3031 2009-04-08 16:43:27 +0300 Stefan Kost <ensonic@users.sf.net>
3033 * gst/playback/gstplaysink.c:
3034 playbin2: better error message on sink failure
3035 If we could create the sinks, but the don't work, don't send the missing plugin
3036 message and report that the state-changed failed.
3038 2009-04-07 22:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
3040 * gst-libs/gst/audio/gstaudiofilter.c:
3041 audiofilter: don't leak pad-template
3042 gst_element_class_add_pad_template() does not take ownership.
3044 2009-04-04 21:18:38 +0300 Felipe Contreras <felipe.contreras@gmail.com>
3047 Automatic update of common submodule
3048 From d0ea89e to b3941ea
3050 2009-04-04 16:28:14 +0200 Edward Hervey <bilboed@bilboed.com>
3052 * gst-libs/gst/interfaces/navigation.c:
3053 * sys/v4l/v4lsrc_calls.c:
3054 navigation/v4l: Don't use g_return_val_if_fail for computed/used values.
3056 2009-03-22 09:46:37 +0100 Edward Hervey <bilboed@bilboed.com>
3058 * ext/theora/theoradec.c:
3059 theoradec: return GST_CLOCK_TIME_NONE for negative framecounts.
3060 This fixes most seeking issues when used with gnonlin.
3063 2009-04-04 14:53:42 +0200 Edward Hervey <bilboed@bilboed.com>
3066 Automatic update of common submodule
3067 From f8b3d91 to d0ea89e
3069 2009-04-03 10:51:42 -0700 Michael Smith <msmith@songbirdnest.com>
3071 * gst/playback/gstplaybin2.c:
3072 playbin2: don't leak selector when getting current stream numbers.
3074 2009-04-02 22:28:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3076 * gst-libs/gst/rtsp/gstrtspconnection.c:
3077 rtsp: use fully qualified urls when using a proxy
3078 Use a fully qualified url when specifying the url for tunneled requests through
3082 2009-03-31 00:54:30 +0100 Jan Schmidt <thaytan@noraisin.net>
3084 * docs/libs/gst-plugins-base-libs-sections.txt:
3085 * gst-libs/gst/interfaces/navigation.c:
3086 * gst-libs/gst/interfaces/navigation.h:
3087 * tests/check/Makefile.am:
3088 * tests/check/libs/.gitignore:
3089 * tests/check/libs/navigation.c:
3090 * win32/common/libgstinterfaces.def:
3091 navigation: Extend the navigation interface
3092 Add support for a set of standard commands that can be queried and executed to
3093 support applications like DVD. Add query construction and parsing functions.
3094 Add new messages that can be sent on the bus to provide notifications related
3095 to commands, multiangle changes, and button highlight activity.
3096 Add some helper functions to parse the existing GstNavigation events that
3097 elements might receive.
3098 Document it all and add unit tests.
3100 2009-02-04 17:03:07 +0000 Jan Schmidt <thaytan@noraisin.net>
3102 * gst/playback/gstplaybasebin.c:
3103 * gst/playback/gstplaybasebin.h:
3104 playbin: Add simple 'raw decoding mode'.
3105 Raw decoding mode removes almost all buffering in video and audio queues
3106 when a source providing already decoded video/audio is detected, on the
3107 possibly bogus assumption that such a source should provide sufficient
3108 internal queueing. Fixes playback on some DVDs, and improves it
3111 2009-04-02 09:27:07 +0100 Jan Schmidt <thaytan@noraisin.net>
3113 * tests/check/elements/.gitignore:
3114 ignores: Ignore the videoscale check binary
3116 2009-04-02 12:13:57 +0100 Jan Schmidt <thaytan@noraisin.net>
3118 * win32/common/libgstrtsp.def:
3119 win32: Add gst_rtsp_connection_set_proxy to the win32 exports
3121 2009-04-02 10:42:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3123 * ext/alsa/gstalsamixer.c:
3124 alsamixer: don't forget to release locks in a few places
3127 2009-04-02 11:10:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3129 * gst/videoscale/vs_4tap.c:
3130 videoscale: Don't read over line ends when taking the last Cr or Cb
3132 2009-04-02 10:52:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3134 * gst/videoscale/vs_4tap.c:
3135 videoscale: Don't write to few pixels and don't mix Cr and Cb
3138 2009-04-01 15:15:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3140 * gst/audioresample/gstaudioresample.c:
3141 * tests/check/elements/audioresample.c:
3142 audioresample: fix negotiation so that upstream can actually fixate to downstream's rate
3143 If one side has a preference for a particular sample rate or set of sample rates, we
3144 should honour this in the caps we advertise and transform to and from, so that elements
3145 actually know about the other side's sample rate preference and can negotiate to it
3146 if supported. Also add unit test for this.
3148 2009-03-26 19:34:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3150 * gst/playback/gstplaybin2.c:
3151 docs: add a blurb about redirect messages to playbin2 docs
3153 2009-04-01 09:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3155 * gst-libs/gst/rtsp/gstrtspconnection.c:
3156 rtsp: fix little typo in the comments
3158 2009-03-31 17:52:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3160 * gst-libs/gst/rtsp/gstrtspconnection.c:
3161 rtspconnection: make gst_rtsp_watch_queue_message() thread-safe
3162 People might queue messages from a thread other than the thread in which
3163 the main context which this watch is attached is iterated from, so use
3164 a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
3165 over list nodes just freed in the other thread. This just fixes issues
3166 I've had with gst-rtsp-server. We might need more locking in various
3169 2009-03-31 18:13:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3171 * gst-libs/gst/rtsp/gstrtspconnection.c:
3172 * gst-libs/gst/rtsp/gstrtspmessage.c:
3173 rtsp: clear the entire builder structure
3174 And use structure instead of variable with sizeof when
3175 clearing the rtsp message structure, for clarity.
3177 2009-03-31 17:56:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3179 * gst-libs/gst/rtsp/gstrtspmessage.c:
3180 docs: fix typo in gst_rtsp_message_unset() API docs
3182 2009-03-31 19:00:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3184 * gst-libs/gst/rtsp/gstrtspconnection.c:
3185 * gst-libs/gst/rtsp/gstrtspconnection.h:
3186 rtsp: add support for proxies
3187 Add suport for proxy servers. Currently only used for tunneled HTTP
3188 connections without authentication.
3190 2009-03-31 18:57:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3192 * gst-libs/gst/rtsp/gstrtspmessage.c:
3193 Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
3194 This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.
3196 2009-03-26 18:54:56 +0200 Stefan Kost <ensonic@users.sf.net>
3198 * sys/xvimage/xvimagesink.c:
3199 xvimagesink: use xcontext->depth instead of bits in attr.max_value for colorkey
3200 According to the drivers in http://cgit.freedesktop.org/xorg/driver/ we should
3201 format the colorkey depending on xcontext->depth. This is what they will use to
3202 interprete the value. The max_value in turn is usualy a constant regardless of
3205 2009-03-31 12:22:14 +0300 Stefan Kost <ensonic@users.sf.net>
3207 * gst-libs/gst/rtsp/gstrtspmessage.c:
3208 rtsp: reset whole message (was sizeof pointer instead of sizeof type)
3210 2009-03-31 00:56:18 +0100 Jan Schmidt <thaytan@noraisin.net>
3212 * gst-libs/gst/interfaces/mixer.c:
3213 doc: Fix a typo in the GstMixer docs
3215 2009-03-29 12:01:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3217 * gst/videoscale/vs_scanline.c:
3218 videoscale: Fix linear scaling for one byte components
3221 2009-03-29 11:53:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3223 * gst/videoscale/vs_4tap.c:
3224 videoscale: Fix 4tap scaling of YUYV and friends
3226 2009-03-28 16:08:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3228 * gst/videoscale/vs_image.c:
3229 * gst/videoscale/vs_scanline.c:
3230 * gst/videoscale/vs_scanline.h:
3231 videoscale: Rewrite YUYV (and friends) scaling and don't read/write over line ends
3232 Partially fixes bug #577054, there's just one issue left now.
3234 2009-03-28 12:48:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3236 * tests/check/elements/videoscale.c:
3237 videoscale: Add some more unit tests
3239 2009-03-28 11:51:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3241 * gst/videoscale/gstvideoscale.c:
3242 videoscale: Use bilinear instead of 4tap scaling for heights < 4
3243 Partially fixes bug #577054.
3245 2009-03-28 11:45:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3247 * gst/videoscale/vs_scanline.c:
3248 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY/RGB/RGBA
3249 This case is for upscaling a frame with width=1
3250 Partially fixes bug #577054.
3252 2009-03-28 11:27:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3254 * gst/videoscale/vs_scanline.c:
3255 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
3256 Partially fixes bug #577054.
3258 2009-03-28 10:40:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3260 * gst/videotestsrc/gstvideotestsrc.c:
3261 videotestsrc: Initialize buffer memory with zeroes
3262 This prevents valgrind warnings when accessing the "x" parts
3263 of xRGB and friends in other elements that handle (and can handle)
3264 xRGB like ARGB (for example videoscale).
3266 2009-03-28 10:25:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3268 * tests/check/Makefile.am:
3269 * tests/check/elements/videoscale.c:
3270 videoscale: Add a lot of unit tests
3272 2009-03-28 10:06:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3274 * gst/videoscale/gstvideoscale.c:
3275 videocale: Add support for video/x-raw-gray with bpp=depth=8
3277 2009-03-28 10:01:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3279 * gst/videotestsrc/videotestsrc.c:
3280 videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8
3282 2009-03-28 09:43:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3284 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
3285 ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format
3287 2009-03-27 19:12:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3289 * gst/videoscale/vs_4tap.c:
3290 videoscale: Take the next luma value instead of every second next when scaling UYVY and friends
3292 2009-03-27 19:09:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3294 * gst/videoscale/gstvideoscale.c:
3295 videoscale: Add support for v308 YUV colorspace
3297 2009-03-27 13:15:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3299 * gst/videoscale/vs_4tap.c:
3300 videoscale: Add my copyright to the 4tap scalers
3302 2009-03-27 13:14:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3304 * gst/videoscale/gstvideoscale.c:
3305 videoscale: Enable 4-tap scaling for all supported formats
3307 2009-03-27 13:14:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3309 * gst/videoscale/vs_4tap.c:
3310 * gst/videoscale/vs_4tap.h:
3311 videoscale: Implement 4-tap scaling for RGB565 and RGB555
3313 2009-03-27 10:47:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3315 * gst/videoscale/vs_4tap.c:
3316 * gst/videoscale/vs_4tap.h:
3317 videoscale: Implement 4-tap scaling for UYVY
3319 2009-03-27 09:33:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3321 * gst/videoscale/vs_4tap.c:
3322 * gst/videoscale/vs_4tap.h:
3323 videoscale: Implement 4-tap scaling for YUY2 and YVYU
3325 2009-03-26 22:14:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3327 * gst/videoscale/vs_4tap.c:
3328 * gst/videoscale/vs_4tap.h:
3329 videoscale: Implement 4-tap scaling for RGB and BGR
3331 2009-03-26 22:08:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3333 * gst/videoscale/vs_4tap.c:
3334 * gst/videoscale/vs_4tap.h:
3335 videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats
3337 2009-03-26 11:02:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3339 * ext/pango/gsttextoverlay.c:
3340 textoverlay: Fix drawing of UYVY text borders
3342 2009-03-26 10:36:27 +0100 Zeeshan Ali <zeeshan.ali@nokia.com>
3344 * ext/pango/gsttextoverlay.c:
3345 * ext/pango/gsttextoverlay.h:
3346 textoverlay: Add support for UYVY colorspace
3349 2009-03-25 19:01:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3351 * gst/playback/gstdecodebin2.c:
3352 decodebin2: do some more cleanup
3353 Free the groups when we go to READY.
3354 Allow for NO_PREROLL elements.
3356 2009-03-25 16:37:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3358 * gst-libs/gst/rtsp/gstrtspconnection.c:
3359 rtsp: start CSeq counting from 1 instead of 0
3360 Start counting from 1 instead of 0 as this is what most other clients
3363 2009-03-25 16:35:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3365 * gst-libs/gst/rtsp/gstrtspdefs.c:
3366 * gst-libs/gst/rtsp/gstrtspdefs.h:
3367 rtsp: add ETag and If-Match headers
3368 Add new headers, we need them for RealMedia support.
3370 2009-03-25 14:16:25 +0200 Stefan Kost <ensonic@users.sf.net>
3372 * sys/xvimage/xvimagesink.c:
3373 xvimagesink: scale the colorkey components in case of 16bit visuals
3374 Use a default that won't be scales to 0,0,0
3376 2009-03-25 11:27:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3378 * gst-libs/gst/audio/gstbaseaudiosrc.c:
3379 audiosrc: improve 'Dropped n samples' warning message
3381 2009-03-24 19:41:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3383 * tests/examples/app/appsrc-ra.c:
3384 * tests/examples/app/appsrc-seekable.c:
3385 examples: use new method to set flags
3386 Use the new core method for setting object enum properties by name.
3388 2009-03-24 18:29:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3390 * gst/playback/gstplaysink.c:
3391 * gst/playback/gstplaysink.h:
3392 playbin2: add more support for subpictures
3394 2009-03-24 17:12:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3396 * gst/playback/gstplaybin2.c:
3397 * gst/playback/gstplaysink.c:
3398 * gst/playback/gstplaysink.h:
3399 playbin2: first support for subpictures
3400 Add beginnings of subpicture support.
3402 2009-03-24 15:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3404 * tests/examples/seek/seek.c:
3405 seek: print tags from the different tracks
3407 2009-03-24 12:22:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3409 * gst/playback/gstplaybin2.c:
3410 playbin2: blacklist subpictures for now
3411 Blacklist the subpictures until we add support for them.
3412 Add some small debug info.
3415 2009-03-24 12:19:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3417 * gst/playback/gsturidecodebin.c:
3418 uridecodebin: expose more media types
3419 Expose more media types from a raw source, such as the subpicture and various
3421 Small cleanups and add some more debugging.
3424 2009-03-24 10:42:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3426 * gst/playback/gstplaysink.c:
3427 playbin2: rescan audio sinks for volume/mute
3428 Rescan the audio sinks for the mute and volume properties.
3431 2009-03-23 19:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3433 * gst/playback/gstplaysink.c:
3434 playbin2: fix reuse of the video chains
3435 When reusing playbin with visualisations, reset the async property on the video
3436 sink because some sinks might dynamically recreate their sinks.
3439 2009-03-23 17:37:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3441 * gst/playback/gstplaysink.c:
3442 playbin2: allow dynamic swtiching of subtitles
3443 When we have the textpad configured, enable and disable the subtitles by setting
3444 the silent flag on the overlay element instead of trying to remove elements.
3447 2009-03-23 16:59:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3449 * tests/icles/playbin-text.c:
3450 tests: print some more info in the text example
3451 Print both the position and the running_time when the subtitle becomes available
3454 2009-03-23 16:04:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3456 * gst/playback/gstplaysink.c:
3457 playbin2: fix dynamic switching of visualisations
3458 Fix the switching of visualisations by requesting and releasing the tee request
3462 2009-03-23 16:19:11 +0200 Stefan Kost <ensonic@users.sf.net>
3465 * gst/tcp/gsttcpclientsink.c:
3466 * gst/tcp/gsttcpclientsrc.c:
3467 * gst/tcp/gsttcpserversink.c:
3468 * gst/tcp/gsttcpserversrc.c:
3469 docs: add examples for tcp elements, also use correct section name. Fixes #564139
3470 Updated the examples in the README to actually work. Add them to api docs. Tests
3471 the api-docs and fix the section names to make the docs actualy show up.
3472 The example for "tcpserversrc" needs review (might be an element bug).
3474 2009-03-17 09:14:02 +0200 Stefan Kost <ensonic@users.sf.net>
3476 * gst/videoscale/gstvideoscale.c:
3477 indent: fix damange that gst-indent did some time ago
3479 2009-03-23 15:27:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3481 * gst/playback/gstplaysink.c:
3482 playbin2: fix linking order
3483 Link after doing the state change and unlink before shutting down. Makes the
3484 window for causing races in toggling the visualisations smaller.
3487 2009-03-23 12:26:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3489 * gst/playback/gsturidecodebin.c:
3490 uridecodebin: reset counter
3491 reset the number of pending dynamic operations back to 0 when we reuse
3495 2009-03-23 11:38:53 +0100 Edward Hervey <bilboed@bilboed.com>
3497 * ext/theora/theoradec.c:
3498 theoradec: Use GST_CLOCK_TIME_NONE for invalid positions. Fixes #543591
3499 The problem was that previously we didn't check whether _theora_granule_frame
3500 returned a negative framecount or not, resulting in bogus timestamps.
3502 2009-03-21 09:46:28 +0100 René Stadler <mail@renestadler.de>
3504 * ext/vorbis/vorbisenc.c:
3505 vorbisenc: Set caps on non-header ouput buffers.
3508 2009-03-20 16:13:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3510 * tests/examples/seek/seek.c:
3511 seek: Add some more debug
3512 Add some more info about the selected streams.
3514 2009-03-20 15:47:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3516 * gst/playback/gstdecodebin2.c:
3517 decodebin2: a pad starts out being not drained.
3518 Mark a new pad as not drained until we get EOS on it.
3520 2009-03-20 14:17:19 +0100 LRN <lrn1986 at gmail dot com>
3522 * gst/playback/gstqueue2.c:
3523 win32: fix seeking in large files
3524 Fix Seeking in large files by using the 64-bit seek functions.
3527 2009-03-19 20:31:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3529 * gst/playback/gstdecodebin2.c:
3530 decodebin2: recover from failing to add a pad
3531 When we cannot add a pad to the decodebin2 for some reason, print a warning but
3532 continue adding the remaining pads.
3534 2009-03-19 19:35:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3536 * gst/playback/gstdecodebin2.c:
3537 decodebin2: more cleanups and docs.
3538 Add some more comments and use g_list_prepend().
3540 2009-03-19 19:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3542 * gst/playback/gstdecodebin2.c:
3543 decodebin2: refactoring and race fixes
3544 Refactor some code so that we can take the right locks and in the right order.
3545 Fixes quite a bit of races already.
3547 2009-03-19 19:03:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3549 * gst/playback/gstplaybin2.c:
3550 playbin2: remove the group cond + cleanups
3551 Remove the group GCond that we used for waiting for groups to finish because we
3552 use pad blocking on the selectors and counters instead for waiting for the
3554 remove the obsolete about_to_finish variable set while emiting the
3555 about-to-finish signal and fix some old comments.
3556 We don't need to take the playbin lock when querying the uridecodebin.
3558 2009-03-18 10:45:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3560 * tests/icles/playbin-text.c:
3561 icles: print better error and warning messages
3564 2009-03-17 22:53:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3566 * gst-libs/gst/rtsp/gstrtspbase64.c:
3567 * gst-libs/gst/rtsp/gstrtspbase64.h:
3568 rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode
3569 This also fixes another instance of CVE-2008-4316.
3571 2009-03-17 19:53:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3573 * ext/ogg/gstoggdemux.c:
3574 oggdemux: report -1 for duration in push mode
3575 In push mode we must return TRUE from the duration query with a value of -1
3576 meaning that we know that we don't know the duration.
3578 2009-03-17 19:09:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3580 * gst/playback/gstdecodebin2.c:
3581 decodebin2: add extra dynamic ref for demuxers
3582 When we make a group connected to a demuxer, keep an extra dynamic refcount for
3583 the group which is only decremented when no_more_pads or a multiqueue overrun is
3584 detected. This way we avoid a race between exposing the group while more dynamic
3585 refs are added from new pads.
3588 2009-03-17 15:39:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3590 * gst/playback/gstplaysink.c:
3591 playbin2: sync state of the sink correctly
3592 Sync the state of the newly added chains to the state of the parent sink element
3593 to avoid lost async-start messages. Fixes cdda:// async-done message storm.
3595 2009-03-17 11:54:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3597 * gst/playback/gstplaybin2.c:
3598 playbin2: return NOT_LINKED for unselected streams
3599 When streams are not selected in the selector, return NOT_LINKED so that
3600 upstream elements can skip decoding. Only do this for audio and video pads
3601 because for text streams the overhead is smaller and they could come from
3604 2009-03-17 11:51:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3606 * gst/playback/gstplaysink.c:
3607 playbin: set custom text sink properties
3608 Set the custom sink async=FALSE to not make it participate in preroll because we
3609 are dealing with sparse streams.
3610 Try to set sync=TRUE on the custom text sink.
3612 2009-03-17 11:30:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3614 * tests/icles/playbin-text.c:
3615 example: use appsink instead of fakesink
3616 Use appsink instead of fakesink to get the subtitles.
3617 Make things more pretty.
3619 2009-03-17 11:24:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3621 * tests/icles/.gitignore:
3622 * tests/icles/Makefile.am:
3623 * tests/icles/playbin-text.c:
3624 examples: add example of intercepting subtitles
3625 Add an example of how to install a custom sink for receiving subtitles in
3628 2009-03-17 11:03:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3630 * tests/check/elements/appsink.c:
3631 tests: fix include in the appsink test
3632 Fix dist by doing the right include.
3634 2009-03-16 16:42:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3636 * gst/playback/gstplaybin2.c:
3637 playbin2: don't try to set invalid stream numbers
3638 Fix a problem with setting the stream numbers because we check for the wrong
3642 2009-03-16 16:16:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3644 * gst/playback/gstplaybin2.c:
3645 playbin2: release the shutdown lock
3646 Release the shutdown lock when we wait for other groups to complete or else we
3647 have a deadlock when the other group completes and tries to grab the shutdown
3651 2009-03-16 15:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3653 * tests/examples/app/appsrc-ra.c:
3654 * tests/examples/app/appsrc-seekable.c:
3655 * tests/examples/app/appsrc-stream.c:
3656 * tests/examples/app/appsrc-stream2.c:
3657 examples: fix g_object_set() value type.
3658 Make sure we cast the length value as a gint64 to the vararg g_object_set() just
3659 incase sizeof(gsize) != sizeof(gint64).
3661 2009-03-15 19:57:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3663 * gst/typefind/gsttypefindfunctions.c:
3664 typefinding: make flac typefinder return lower probability for frame headers
3665 The flac frame header typefinder overstates the likelihood of a match, leading
3666 to false positives with e.g. aac streams and PDF files. Reduce probabilty
3667 returned from LIKELY to POSSIBLE for the frame header matchin code.
3670 2009-03-11 12:59:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3672 * gst/typefind/gsttypefindfunctions.c:
3673 typefinding: improve image/bmp typefinder
3674 Detect more variations and also bail out in more cases where the values
3675 don't make sense. Furthermore, add width/height and bpp to the caps,
3678 2009-03-13 15:22:42 +0000 Jan Schmidt <thaytan@noraisin.net>
3680 * tests/check/Makefile.am:
3681 check: Ignore alsamixer in the states test too
3683 2009-03-13 15:22:11 +0000 Jan Schmidt <thaytan@noraisin.net>
3685 * sys/v4l/v4l_calls.c:
3686 v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.
3688 2009-03-13 16:19:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3690 * gst-libs/gst/rtsp/gstrtspconnection.c:
3691 rtsp: fix resolving of hostnames
3692 We were returning a pointer to a stack variable with the resolved hostname,
3694 return a copy of the resolved ip address instead.
3697 2009-03-13 15:29:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3699 * ext/vorbis/vorbisparse.c:
3700 vorbisparse: be smarter when queueing headers
3701 Look at the first buffer byte to see if a buffer is a header instead of counting
3704 2009-03-13 15:27:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3706 * ext/theora/gsttheoraparse.h:
3707 * ext/theora/theoraparse.c:
3708 theoraparse: be smarter when queuing headers
3709 Look at the first byte of the buffer data (if we can) to decide if the packet is
3710 a header packet or not instead of counting packets.
3712 2009-03-13 15:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3714 * ext/ogg/gstoggdemux.c:
3715 oggdemux: add some debug info
3716 Add some debug info to log when the seek worked.
3718 2009-03-13 15:14:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3720 * gst-libs/gst/app/gstappsrc.c:
3721 appsrc: release lock in _eos flushing case
3722 Release the mutex when we are flushing in gst_app_src_end_of_stream()
3725 2009-03-13 11:49:10 +0000 Jan Schmidt <thaytan@noraisin.net>
3727 * ext/vorbis/vorbisdec.c:
3728 vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.
3730 2009-03-13 11:48:28 +0000 Jan Schmidt <thaytan@noraisin.net>
3732 * ext/theora/theoradec.c:
3733 theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.
3735 2009-03-12 18:27:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3737 * gst/playback/gsturidecodebin.c:
3738 playbin2: fix raw elements like cdda://
3739 Fix a fixme with a one liner and make cd playback work again.
3741 2009-03-12 17:47:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3743 * gst/playback/gstplaybin2.c:
3744 * gst/playback/gstplaysink.c:
3745 * gst/playback/gstplaysink.h:
3746 playbin2: improve subtitle handling
3747 Add property to playbin2 to configure a custom sink that receives the raw
3748 subtitle buffers instead of using a textoverlay.
3749 Improve the property finding code to make it more usable.
3750 Use property find code to find async properties in custom sinks that are bins.
3751 Improve text overlay code to gracefully handle missing elements.
3753 2009-02-24 15:58:42 +0000 Jan Schmidt <thaytan@noraisin.net>
3755 * gst-libs/gst/tag/gstvorbistag.c:
3756 vorbistag: Protect memory allocation calculation from overflow.
3757 Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
3759 2009-03-12 11:34:20 +0000 Jan Urbanski <jurbanski@flumotion.com>
3761 * gst-plugins-base.spec.in:
3764 2009-03-11 18:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3766 * gst-libs/gst/rtsp/gstrtspconnection.c:
3767 rtsp: fix parsing of the timeout parameter
3770 2009-03-11 16:20:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3772 * gst-libs/gst/rtsp/gstrtspmessage.c:
3773 rtsp: fix g_return condition
3774 when parsing a data message, we require a data message.
3776 2009-03-11 13:33:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3778 * gst/typefind/gsttypefindfunctions.c:
3779 typefinding: flac typefinder fixes
3780 Use scan context for initial peek as well. Peek 6 bytes in the initial
3781 peek rather than 5 bytes, to match the length of the memcmp we're doing
3782 on that data later. Return immediately when we found caps from looking
3783 at the beginning of the data - no point in continuing to scan the next
3784 64kB for something matching a frame header.
3786 2009-03-11 14:08:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3788 * gst-libs/gst/rtsp/gstrtspmessage.c:
3789 rtsp: free the right string.
3790 Free the key value before we remove the header item from the array. The item we
3791 retrieved from the array is only valid until we remove it from the array.
3793 2009-03-11 14:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3795 * gst-libs/gst/rtsp/gstrtspconnection.c:
3796 rtsp: keep track of amount of decoded bytes
3797 Keep track of the actual amount of decoded bytes, which can be less than 3 when
3798 we decode the last bits of a base64 message.
3800 2009-03-10 21:00:26 +0200 Stefan Kost <ensonic@users.sf.net>
3802 * gst/adder/gstadder.c:
3803 adder: log details in getcaps like in setcaps
3805 2009-03-10 13:11:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3808 win32: update MANIFEST, fixing 'make dist'
3810 2009-03-09 23:12:00 +0000 Jan Schmidt <thaytan@noraisin.net>
3813 Automatic update of common submodule
3814 From 7032163 to f8b3d91
3816 2009-03-09 16:19:40 +0100 Jonathan Matthew <notverysmart at gmail dot com>
3818 * gst/typefind/gsttypefindfunctions.c:
3819 typefind: add photoshop typefind functions
3820 Add photoshop typefind functions.
3823 2009-03-09 15:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3825 * gst/playback/gstdecodebin2.c:
3826 decodebin2: only remove pads that were added
3827 Flag pads that were added so that we can see if we need to remove them later or
3830 2009-03-09 13:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3832 * gst-libs/gst/rtsp/gstrtsptransport.c:
3833 rtsp: only add ports when not using TCP
3834 Only add the port numbers in the transport string when we are using udp or
3837 2009-03-09 13:53:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3839 * gst-libs/gst/rtsp/gstrtspmessage.c:
3840 rtsp: use gstreamer dump mem
3843 2009-03-09 13:51:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3845 * gst-libs/gst/rtsp/gstrtspconnection.c:
3846 rtsp: use glib base64 encoder
3849 2009-03-06 19:28:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
3851 * gst/playback/gstdecodebin2.c:
3852 Unblock blocked ghostpads when shutting down. Fixes #574293.
3854 2009-03-09 10:03:13 +0100 Edward Hervey <bilboed@bilboed.com>
3856 * gst-libs/gst/riff/riff-media.c:
3857 Riff: Add mapping for Fraps video codec.
3858 Found through insanity testrun. Confirmed mapping in libavformat.
3860 2009-03-09 09:07:13 +0100 Edward Hervey <bilboed@bilboed.com>
3862 * gst-libs/gst/riff/riff-media.c:
3863 riff: Add the 'DVR ' mapping for mpeg2video.
3864 Found this in 3 files from the insanity suite and mapping is also present
3867 2009-03-09 09:06:40 +0100 Edward Hervey <bilboed@bilboed.com>
3869 * gst/typefind/gsttypefindfunctions.c:
3870 typefind: Use the proper data pointer instead of poking random memory.
3872 2009-03-08 18:17:48 +0100 LRN <lrn1986@gmail.com>
3874 * gst-libs/gst/rtsp/gstrtspconnection.c:
3875 rtsp: fix compilation on windows.
3876 Remove unused variable when building for windows.
3879 2009-03-08 12:03:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3882 Automatic update of common submodule
3883 From ffa738d to 7032163
3885 2009-03-08 11:19:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3888 Automatic update of common submodule
3889 From 3f13e4e to ffa738d
3891 2009-03-07 11:44:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3894 Automatic update of common submodule
3895 From 3c7456b to 3f13e4e
3897 2009-03-07 10:44:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3900 Automatic update of common submodule
3901 From 57c83f2 to 3c7456b
3903 2009-03-06 19:02:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3905 * ext/theora/theoradec.c:
3906 theoradec: parse and use codec_data in the caps
3907 Parse the codec_data in the caps and use this as the headers.
3910 2009-03-06 18:53:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3912 * gst-libs/gst/riff/riff-media.c:
3913 riff: add theora mapping
3914 Add theora mappings. See #574169.
3916 2009-03-06 16:31:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3918 * gst-libs/gst/rtsp/gstrtspconnection.c:
3919 * gst-libs/gst/rtsp/gstrtspconnection.h:
3920 * win32/common/libgstrtsp.def:
3921 rtsp: Add methods for getting the read/write fds
3922 API:gst_rtsp_connection_get_readfd()
3923 API:gst_rtsp_connection_get_writefd()
3925 2009-03-06 10:35:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3928 * win32/common/audio-enumtypes.c:
3929 win32: indent copied *-enumtypes.c files in make win32-update
3931 2009-03-06 10:35:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3934 win32: update MANIFEST
3936 2009-03-06 10:30:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3939 * win32/common/config.h:
3940 win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define
3942 2009-03-06 10:05:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3944 * win32/common/_stdint.h:
3945 * win32/common/config.h:
3946 * win32/common/gstrtsp-enumtypes.c:
3947 * win32/common/interfaces-enumtypes.c:
3948 * win32/common/multichannel-enumtypes.c:
3949 * win32/common/pbutils-enumtypes.c:
3950 * win32/common/video-enumtypes.c:
3951 * win32/common/video-enumtypes.h:
3952 win32: update windows files via make win32-update
3953 Updates win32 files using the new system/hook, and defines HAVE_PROCESS_H,
3954 which fixes the build of pbutils on windows (#574319).
3956 2009-03-06 10:03:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3959 gitignore: ignore more
3961 2009-03-06 10:37:38 +0100 Julien Moutte <julien@fluendo.com>
3963 * gst-libs/gst/rtsp/gstrtspconnection.c:
3964 Fix build on Mac OS X
3966 2009-03-05 15:42:23 -0800 Michael Smith <msmith@songbirdnest.com>
3968 * gst/playback/gstdecodebin2.c:
3969 decodebin2: don't stay connected to notify::caps after negotiation
3970 Disconnect the notify::caps signal in our callback (it'll be re-added
3971 if we're not, in fact, finished getting complete caps). Ensures that
3972 caps changes mid-stream (e.g. from an mp3 that changes from
3973 stereo->mono mid-file) don't cause us to try to add a new pad.
3975 2009-03-05 13:48:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3977 * gst-libs/gst/rtsp/gstrtsprange.c:
3978 rtsp: fix parsing of 'now-' ranges.
3981 2009-03-05 12:43:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3983 * tests/examples/dynamic/.gitignore:
3984 * tests/examples/dynamic/Makefile.am:
3985 * tests/examples/dynamic/sprinkle.c:
3986 * tests/examples/dynamic/sprinkle2.c:
3987 * tests/examples/dynamic/sprinkle3.c:
3988 examples: add some more sprinkle examples
3989 Add some more sprinle examples and add some more comments.
3992 2009-03-05 11:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3994 * docs/plugins/gst-plugins-base-plugins-sections.txt:
3995 docs: add appsrc symbols to standard section
3998 2009-03-05 12:27:16 +0200 Stefan Kost <ensonic@users.sf.net>
4000 * gst/adder/gstadder.c:
4001 adder: add variants for unsigned to fix warnings for unneeded check
4002 For unsigned int out+in can't be < 0.
4004 2009-03-05 10:58:12 +0200 Stefan Kost <ensonic@users.sf.net>
4006 * gst/subparse/gstsubparse.c:
4007 subparse: use the right variable in debug log, encoding is not yet initialized
4009 2009-03-05 10:51:25 +0200 Stefan Kost <ensonic@users.sf.net>
4011 * sys/v4l/v4l_calls.c:
4012 v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix
4014 2009-03-05 10:39:33 +0200 Stefan Kost <ensonic@users.sf.net>
4016 * gst/audioresample/gstaudioresample.c:
4017 audioresample: add missing break in event handling, remove dead code
4019 2009-03-04 16:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4021 * gst-libs/gst/rtsp/gstrtspconnection.c:
4022 rtsp: do some more cleanup in _close
4023 Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
4024 unconnected state as it was allocated.
4026 2009-03-04 16:11:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4028 * gst-libs/gst/rtsp/gstrtspconnection.c:
4029 * gst-libs/gst/rtsp/gstrtspconnection.h:
4030 rtsp: fix the memory management of the url
4031 Constify the url parameter in _create.
4032 Make a copy of the url stored in the connection.
4033 Free the url when the connection is freed.
4035 2009-03-04 12:21:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4037 * docs/libs/gst-plugins-base-libs-sections.txt:
4038 * gst-libs/gst/rtsp/gstrtspconnection.c:
4039 * gst-libs/gst/rtsp/gstrtspconnection.h:
4040 * win32/common/libgstrtsp.def:
4041 RTSP: Add support for server tunneling
4042 Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
4043 that a server can store and match the id against other tunnel requests.
4044 Fix the URI in the tunnel requests so that they contain the absolute uri and the
4045 query string if any instead of just the hostname.
4046 Transparently base64 decode the input stream when tunneling.
4047 Add method to set the connection ip address so that it can be included in the
4049 Add method to connect the two tunnel requests.
4050 Add two callbacks for the async mode to notify a tunnel start and tunnel
4052 Add method to reset the watch after the connection has been tunneled.
4053 Various little refactoring to make more stuff reusable.
4054 API: RTSP::gst_rtsp_connection_set_ip()
4055 API: RTSP::gst_rtsp_connection_get_tunnelid()
4056 API: RTSP::gst_rtsp_connection_do_tunnel()
4057 API: RTSP::gst_rtsp_watch_reset()
4059 2009-03-04 12:18:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4061 * gst-libs/gst/rtsp/gstrtspdefs.c:
4062 * gst-libs/gst/rtsp/gstrtspdefs.h:
4063 rtsp: add new defines for tunneling
4064 Add two more result codes for tunneling support.
4066 2009-03-04 12:12:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4068 * gst-libs/gst/rtsp/gstrtspmessage.h:
4069 rtsp: remove , from last enum member
4070 Remove , from last enum member to improve compatibility with other compilers.
4072 2009-02-28 15:23:20 -0800 LRN <lrn1986@gmail.com>
4074 * gst/subparse/gstsubparse.c:
4075 subparse: Convert regex code to GRegex code
4076 Fixes: #572993. Patch author prefers to use an alias, contact
4077 ds if you actually need a real name.
4078 Signed-off-by: David Schleef <ds@schleef.org>
4080 2009-03-02 16:13:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4082 * gst-libs/gst/rtsp/gstrtspconnection.c:
4083 rtsp: remove debugging g_message
4086 2009-03-02 16:03:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4088 * docs/libs/gst-plugins-base-libs-sections.txt:
4089 * gst-libs/gst/rtsp/gstrtspconnection.c:
4090 * gst-libs/gst/rtsp/gstrtspconnection.h:
4091 * win32/common/libgstrtsp.def:
4092 RTSP: add support for Quicktime tunneled RTSP
4093 Add support for tunneling RTSP over HTTP.
4094 Fix documentation some more.
4096 API: RTSP:gst_rtsp_connection_is_tunneled()
4097 API: RTSP:gst_rtsp_connection_set_tunneled()
4099 2009-03-02 15:48:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4101 * gst-libs/gst/rtsp/gstrtsptransport.h:
4102 * gst-libs/gst/rtsp/gstrtspurl.c:
4103 RTSP: parse rtsph uris as RTSP tunneled over HTTP
4104 Add transport define for RTSP tunneled over HTTP.
4105 Parse rtsph:// uris as tunneled HTTP over TCP.
4106 API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
4109 2009-03-02 12:48:18 +0100 Edward Hervey <bilboed@bilboed.com>
4111 * win32/common/libgstrtsp.def:
4112 win32: Add gst_rtsp_connection_get_url definition
4113 No, I'm not wim's buildslave, seriously.
4115 2009-03-02 10:58:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4117 * gst-libs/gst/rtsp/gstrtspconnection.c:
4118 * gst-libs/gst/rtsp/gstrtspconnection.h:
4119 rtsp: add _get_url method and separate sockets
4120 Add gst_rtsp_connection_get_url() method.
4121 Reserve space for 2 sockets, one for reading and one for writing. Use socket
4122 pointers to select the read and write sockets. This should allow us to implement
4123 tunneling over HTTP soon.
4124 API: RTSP::gst_rtsp_connection_get_url()
4126 2009-03-01 18:31:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4128 * gst-libs/gst/app/gstapp-marshal.list:
4129 app: force automatic rebuild of gstapp-marshal.[ch] after previous change
4130 The previous change to appsrc/appsink requires people to 'make clean'
4131 to get the marshallers rebuilt (causing a build failure otherwise).
4132 Change some lines in the .list file around to force a rebuild of
4133 these files automatically.
4135 2009-02-28 11:07:04 -0800 David Schleef <ds@schleef.org>
4138 Bump glib requirement to 2.14
4140 2009-02-28 19:37:53 +0100 LRN <lrn1986@gmail.com>
4142 * ext/gio/gstgiobasesink.c:
4143 gio: Use correct format modifier for size_t
4146 2009-02-28 19:35:33 +0100 LRN <lrn1986@gmail.com>
4148 * gst-libs/gst/rtsp/gstrtspconnection.c:
4149 rtspconnection: Use correct types for some functions on Win32
4152 2009-02-28 13:11:59 +0100 Edward Hervey <bilboed@bilboed.com>
4154 * gst-libs/gst/rtsp/gstrtspconnection.c:
4155 rtspconnection: Fix warning about using unitialized value.
4157 2009-02-28 12:41:28 +0100 Edward Hervey <bilboed@bilboed.com>
4159 * gst-libs/gst/riff/riff-ids.h:
4160 * gst-libs/gst/riff/riff-media.c:
4161 riff: Add more codec mappings.
4162 This comes mostly from a review of ffmpeg/libavformat/riff.c
4164 2009-02-27 11:14:25 +0200 Stefan Kost <ensonic@users.sf.net>
4166 * ext/alsa/gstalsa.c:
4167 alsa: release pcminfo after the strdup
4169 2009-02-26 17:38:47 +0200 Stefan Kost <ensonic@users.sf.net>
4171 * gst-libs/gst/rtsp/gstrtsprange.c:
4172 rtsprange: don't leak the range in case of parsing error.
4173 Free the gstRTSPTimeRange if we don't return it. Also simplify
4174 gst_rtsp_range_free() as it is valid to pass NULL to g_free().
4176 2009-02-26 16:47:39 +0200 Stefan Kost <ensonic@users.sf.net>
4178 * ext/alsa/gstalsa.c:
4179 alsa: cleanup name lookup.
4180 We can break, once we have a name to make sure, we won't read it ever twice.
4182 2009-02-26 16:09:03 +0200 Stefan Kost <ensonic@users.sf.net>
4184 * gst/subparse/gstsubparse.c:
4185 subparse: don't leak line, if flushing
4187 2009-02-26 16:03:39 +0200 Stefan Kost <ensonic@users.sf.net>
4189 * ext/gio/gstgiosink.c:
4190 giosink: reflow error handling to not leak uri
4192 2009-02-26 15:53:10 +0200 Stefan Kost <ensonic@users.sf.net>
4194 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
4195 * gst/ffmpegcolorspace/imgconvert.c:
4196 ffmpegcolorspace: remove unused code/variables
4198 2009-02-26 12:10:47 +0200 Stefan Kost <ensonic@users.sf.net>
4200 * sys/ximage/ximagesink.c:
4201 ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps
4203 2009-02-26 16:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4205 * docs/libs/gst-plugins-base-libs-sections.txt:
4206 * gst-libs/gst/app/gstappsink.c:
4207 * gst-libs/gst/app/gstappsrc.c:
4208 * gst-libs/gst/app/gstappsrc.h:
4209 * win32/common/libgstapp.def:
4210 app: add callbacks to appsrc, cleanups
4211 Add a uri handler to appsink.
4212 don't emit signals when we have installed callbacks on appsink.
4213 Add callbacks to appsrc to replace the signals.
4214 Add property to disable callbacks in appsrc, default to TRUE for backwards
4215 compatibility but disable when callbacks are installed.
4216 API: GstAppSrc::emit-signals
4217 API: GstAppSrc::gst_app_src_set_emit_signals()
4218 API: GstAppSrc::gst_app_src_get_emit_signals()
4219 API: GstAppSrc::gst_app_src_set_callbacks()
4221 2009-02-26 11:42:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4223 * docs/libs/gst-plugins-base-libs-sections.txt:
4224 * gst-libs/gst/app/gstappsink.h:
4225 * tests/check/elements/appsink.c:
4226 Appsink: add padding for callbacks + docs
4227 Add some padding to the callbacks structure just to be safe.
4228 Remove the now invisible marshaller methods from the docs.
4229 Fix a comment in the unit test.
4231 2009-02-26 09:52:59 +0100 Edward Hervey <bilboed@bilboed.com>
4233 * win32/common/libgstapp.def:
4234 win32: Add new libgstapp symbol
4236 2009-02-26 10:07:21 +0200 Stefan Kost <ensonic@users.sf.net>
4238 * docs/plugins/gst-plugins-base-plugins-sections.txt:
4239 docs: clean section.txt file.
4240 Add appsrc/sink symbols to private, as they are covered in the libs docs.
4242 2009-02-26 10:06:23 +0200 Stefan Kost <ensonic@users.sf.net>
4244 * gst/playback/gstplaybasebin.c:
4245 docs: fix random text after since: tag. Also fix class name to make the docs actual appear.
4247 2009-02-26 09:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
4249 * docs/plugins/gst-plugins-base-plugins.args:
4250 * docs/plugins/gst-plugins-base-plugins.hierarchy:
4251 * docs/plugins/gst-plugins-base-plugins.interfaces:
4252 * docs/plugins/gst-plugins-base-plugins.prerequisites:
4253 * docs/plugins/inspect/plugin-adder.xml:
4254 * docs/plugins/inspect/plugin-alsa.xml:
4255 * docs/plugins/inspect/plugin-app.xml:
4256 * docs/plugins/inspect/plugin-audioconvert.xml:
4257 * docs/plugins/inspect/plugin-audiorate.xml:
4258 * docs/plugins/inspect/plugin-audioresample.xml:
4259 * docs/plugins/inspect/plugin-audiotestsrc.xml:
4260 * docs/plugins/inspect/plugin-cdparanoia.xml:
4261 * docs/plugins/inspect/plugin-decodebin.xml:
4262 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
4263 * docs/plugins/inspect/plugin-gdp.xml:
4264 * docs/plugins/inspect/plugin-gio.xml:
4265 * docs/plugins/inspect/plugin-gnomevfs.xml:
4266 * docs/plugins/inspect/plugin-libvisual.xml:
4267 * docs/plugins/inspect/plugin-ogg.xml:
4268 * docs/plugins/inspect/plugin-pango.xml:
4269 * docs/plugins/inspect/plugin-playback.xml:
4270 * docs/plugins/inspect/plugin-queue2.xml:
4271 * docs/plugins/inspect/plugin-subparse.xml:
4272 * docs/plugins/inspect/plugin-tcp.xml:
4273 * docs/plugins/inspect/plugin-theora.xml:
4274 * docs/plugins/inspect/plugin-typefindfunctions.xml:
4275 * docs/plugins/inspect/plugin-uridecodebin.xml:
4276 * docs/plugins/inspect/plugin-video4linux.xml:
4277 * docs/plugins/inspect/plugin-videorate.xml:
4278 * docs/plugins/inspect/plugin-videoscale.xml:
4279 * docs/plugins/inspect/plugin-videotestsrc.xml:
4280 * docs/plugins/inspect/plugin-volume.xml:
4281 * docs/plugins/inspect/plugin-vorbis.xml:
4282 * docs/plugins/inspect/plugin-ximagesink.xml:
4283 * docs/plugins/inspect/plugin-xvimagesink.xml:
4284 * gst/playback/gstplaybin2.c:
4285 docs: playbin2 has no stream-info
4287 2009-02-26 09:53:03 +0200 Stefan Kost <ensonic@users.sf.net>
4289 * gst-libs/gst/video/video.h:
4290 docs: fix newly added interlace constants and plug holes in video format docs
4292 2009-02-26 09:35:43 +0200 Stefan Kost <ensonic@users.sf.net>
4294 * gst-libs/gst/app/gstappsink.c:
4295 * gst-libs/gst/app/gstappsrc.c:
4296 * gst-libs/gst/audio/gstaudiofilter.c:
4297 * gst-libs/gst/audio/gstringbuffer.c:
4298 * gst-libs/gst/rtp/gstrtcpbuffer.c:
4299 docs: don't put random stuff in tags.
4300 Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
4301 tag to append text again to the documentation body.
4303 2009-02-06 11:10:15 +0200 Stefan Kost <ensonic@users.sf.net>
4305 * sys/ximage/ximagesink.c:
4306 ximagsink: do not access uninitialized height variable.
4307 Exit like in xvimagesink, if we have partial caps.
4309 2009-02-25 20:26:05 -0800 David Schleef <ds@schleef.org>
4313 * win32/common/config.h.in:
4314 Change how win32/common/config.h is updated
4315 Generate win32/common/config.h-new directly from config.h.in,
4316 using shell variables in configure and some hard-coded information.
4317 Change top-level makefile so that 'make win32-update' copies the
4318 generated file to win32/common/config.h, which we keep in source
4319 control. It's kept in source control so that the git tree is
4321 This change is similar to the one recently applied to GStreamer,
4322 except that it adds a few -base specific defines.
4324 2009-02-25 19:40:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4326 * gst-libs/gst/app/Makefile.am:
4327 * gst-libs/gst/app/gstappsink.c:
4328 * gst-libs/gst/app/gstappsrc.c:
4329 * win32/common/libgstapp.def:
4330 app: add win32 .def file and only export functions we want exported
4331 Add a .def file for win32 builds (and make check-exports).
4332 Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes #573165).
4333 Make sure private marshaller functions aren't exported by prefixing them with __gst;
4334 also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
4335 a comment why we're not using glib-genmarshal for this one.
4337 2009-02-25 17:08:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4339 * tests/examples/dynamic/.gitignore:
4340 * tests/examples/dynamic/Makefile.am:
4341 * tests/examples/dynamic/sprinkle.c:
4342 sprinkle: Add another example app
4343 Add an example app that dynamically adds and removes audiotestsrc elements from
4346 2009-02-25 16:25:33 +0100 Peter Kjellerstedt <pkj@axis.com>
4348 * gst-libs/gst/rtsp/gstrtspconnection.c:
4351 2009-02-25 11:31:02 +0100 Peter Kjellerstedt <pkj@axis.com>
4353 * gst-libs/gst/rtsp/gstrtspconnection.c:
4354 * gst/tcp/gstmultifdsink.c:
4355 rtsp, multifdsink: Unify the use of union gst_sockaddr.
4357 2009-02-25 14:22:35 +0000 Jan Schmidt <thaytan@noraisin.net>
4361 build: Update shave init statement for changes in common. Bump common.
4363 2009-02-25 13:16:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4365 * sys/xvimage/xvimagesink.c:
4366 * sys/xvimage/xvimagesink.h:
4367 xvimageink: protect buffer_alloc from shutdown
4368 Use the pool_lock in the buffer_alloc function to detect shutdown. Avoids
4369 crashes when the sink is shutdown.
4371 2009-02-25 12:43:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4373 * gst/playback/gstplaybin2.c:
4374 playbin: use flushing pads instead of fakesink
4375 Use the flushing pads on playsink to terminate on shutdown instead of plugging
4376 fakesinks. this should be a little cheaper.
4378 2009-02-25 12:42:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4380 * gst/playback/gstplaysink.c:
4381 * gst/playback/gstplaysink.h:
4382 playsink: Add FLUSHING pad type
4383 Make it possible to request a flushing pad from the playsink. We can eventually
4384 use these flushing pads to quickly terminate the dataflow when we are shutting
4387 2009-02-25 11:31:52 +0000 Jan Schmidt <thaytan@noraisin.net>
4390 Automatic update of common submodule
4391 From 9cf8c9b to a6ce5c6
4393 2009-02-25 09:52:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4395 * gst-libs/gst/riff/riff-media.c:
4396 riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
4399 2009-02-25 12:07:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4401 * tests/icles/stress-playbin.c:
4402 stress-playbin: print the current uri
4403 Print the current uri so that we can more easily see what uri caused a crash or
4406 2009-02-25 11:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4408 * tests/icles/stress-playbin.c:
4409 Print the errors more clearly
4410 Print some more verbose messages when dealing with errors.
4412 2009-02-25 10:08:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4414 * gst/playback/gstplaybin2.c:
4415 Release the group lock when setting states
4416 Release the group lock while we perform the state changes on the uridecodebins
4417 because that might trigger callbacks that we need to handle with the group lock
4418 taken. Avoids a possible deadly embrace in some id3/flac files.
4421 2009-02-25 10:05:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4423 * gst/playback/gstdecodebin2.c:
4424 Combine finding and creating groups
4425 Combine the search for the current group and optionally creating one into one
4426 function so that we can avoid taking the lock multiple times.
4428 2009-02-25 08:22:00 +0100 Edward Hervey <bilboed@bilboed.com>
4430 * gst/playback/gstplaybin2.c:
4431 Playbin2: Don't leave unused parameters in debug statements.
4432 Fixes build on macosx
4434 2009-02-24 10:33:05 +0100 Edward Hervey <bilboed@bilboed.com>
4436 * gst-libs/gst/riff/riff-media.c:
4437 Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)
4439 2009-02-24 18:43:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4441 * gst/playback/gstplaybin2.c:
4442 Add some G_UNLIKELY because we can
4443 Add a G_UNLIKELY when checking the shutdown variable.
4445 2009-02-24 17:23:58 +0000 Garret D'Amore <garrett.damore@sun.com>
4447 * gst-libs/gst/interfaces/mixer.h:
4448 * gst-libs/gst/interfaces/mixertrack.h:
4449 mixer interface: Add flags to enhance mixer interfaces
4450 This patch adds a few flags to the mixer and mixerctrl interface to
4451 better support OSSv4 (and potentially other backends).
4452 Patch By: Garret D'Amore <garrett.damore@sun.com>
4453 Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>
4454 API: GST_MIXER_FLAG_HAS_WHITELIST, GST_MIXER_FLAG_GROUPING,
4455 API: GST_MIXER_TRACK_NO_RECORD, GST_MIXER_TRACK_NO_MUTE,
4456 API: GST_MIXER_TRACK_WHITELIST
4458 2009-02-24 17:03:08 +0000 Jan Schmidt <thaytan@noraisin.net>
4460 * gst/tcp/gstmultifdsink.c:
4461 multifdsink: Fix strict aliasing error using a union
4463 2009-02-24 16:49:40 +0000 Jan Schmidt <thaytan@noraisin.net>
4465 * gst-libs/gst/rtsp/gstrtspconnection.c:
4466 rtsp: Fix a strict aliasing warning
4467 Fix strict aliasing warnings from casting a sockaddr_storage and
4468 using it as a sockaddr_in6. Use a union instead.
4470 2009-02-24 16:08:49 +0000 Jan Schmidt <thaytan@noraisin.net>
4472 * docs/libs/.gitignore:
4473 * docs/libs/tmpl/.gitignore:
4474 * docs/plugins/.gitignore:
4475 * docs/plugins/tmpl/.gitignore:
4476 Remove .gitignore files from the docs tmpl dirs, that are killed by make clean.
4478 2009-02-24 14:36:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4480 * docs/plugins/Makefile.am:
4481 * ext/vorbis/Makefile.am:
4482 * ext/vorbis/gstvorbisdec.h:
4483 * ext/vorbis/gstvorbisenc.h:
4484 * ext/vorbis/gstvorbisparse.h:
4485 * ext/vorbis/gstvorbistag.h:
4486 * ext/vorbis/vorbis.c:
4487 * ext/vorbis/vorbisdec.c:
4488 * ext/vorbis/vorbisdec.h:
4489 * ext/vorbis/vorbisenc.c:
4490 * ext/vorbis/vorbisenc.h:
4491 * ext/vorbis/vorbisparse.c:
4492 * ext/vorbis/vorbisparse.h:
4493 * ext/vorbis/vorbistag.c:
4494 * ext/vorbis/vorbistag.h:
4495 vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
4497 2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4499 * gst/ffmpegcolorspace/avcodec.h:
4500 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
4501 * gst/ffmpegcolorspace/imgconvert.c:
4502 ffmpegcolorspace: Add conversion from/to YVYU colorspace
4505 2009-02-24 13:42:01 +0100 Jonas Danielsson <jonas.danielsson@axis.com>
4507 * gst/ffmpegcolorspace/imgconvert.c:
4508 ffmpegcolorspace: Add direct UYVY->GRAY8 conversion
4509 The conversion from UYVY to RGB24 and then to GRAY8
4510 is quite slow. Fixes bug #569655.
4512 2009-02-19 17:16:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4514 * gst/playback/gstplaybin2.c:
4515 playbin2: fix deadlock when shutting down. Fixes #572577.
4517 2009-02-19 17:15:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4519 * tests/icles/stress-playbin.c:
4520 stress-playbin: make more flexible, e.g. also useful for playbin2
4522 2009-02-24 12:11:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4524 * gst-libs/gst/rtsp/gstrtspconnection.c:
4525 Match WSAStartup and WSACleanup correctly
4526 Don't randomly call WSAStartup and WSACleanup but instead call the startup when
4527 we create a connection and cleanup when we free it again. Because the internal
4528 datastructure is refcounted, this should not cause any refcounting leaks when
4529 the connection is managed correctly.
4532 2009-02-18 11:59:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4534 * gst/playback/gstplaysink.c:
4535 playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105.
4537 2009-02-23 10:57:42 -0800 David Flynn <davidf@rd.bbc.co.uk>
4539 * pkgconfig/gstreamer-app-uninstalled.pc.in:
4540 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
4541 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
4542 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
4543 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
4544 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
4545 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
4546 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
4547 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
4548 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
4549 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
4550 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
4551 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
4552 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
4553 * pkgconfig/gstreamer-video-uninstalled.pc.in:
4554 Add srcdir to includes for out-of-source builds
4555 When you use gstreamer uninstalled and build outside
4556 the source tree, the includes need to be specified for
4557 both the source tree and the build tree.
4558 Signed-off-by: David Schleef <ds@schleef.org>
4560 2009-02-22 17:23:52 +0000 Jan Schmidt <thaytan@noraisin.net>
4563 * docs/libs/Makefile.am:
4564 * docs/plugins/Makefile.am:
4565 Use shave for the build output
4567 2009-02-23 12:17:07 +0100 Edward Hervey <bilboed@bilboed.com>
4569 * win32/common/libgstrtsp.def:
4570 win32: Add new symbol to libgstrtsp.def
4572 2009-02-23 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4574 * gst-libs/gst/rtsp/gstrtspextension.c:
4575 * gst-libs/gst/rtsp/gstrtspextension.h:
4576 Add method for handling server requests
4577 Add a receive_request so that extensions can react to server requests.
4579 2009-02-22 19:20:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4581 * tests/check/libs/netbuffer.c:
4582 Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)
4584 2009-02-22 19:19:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4586 * ext/theora/theoraparse.c:
4587 theoraparse: Use the correct unref functions
4589 2009-02-22 19:18:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4591 * sys/ximage/ximagesink.c:
4592 * sys/xvimage/xvimagesink.c:
4593 x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()
4595 2009-02-22 19:12:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4597 * gst-libs/gst/tag/gsttagdemux.c:
4598 tagdemux: Unref the actual buffer instead of the memory address of the buffer
4600 2009-02-22 15:47:53 +0000 Jan Schmidt <thaytan@noraisin.net>
4603 Automatic update of common submodule
4604 From 5d7c9cc to 9cf8c9b
4606 2009-02-22 14:49:29 +0100 Edward Hervey <bilboed@bilboed.com>
4608 * win32/common/libgstrtsp.def:
4609 * win32/common/libgstvideo.def:
4610 win32/common: Update .def files for recent API addition
4612 2009-02-22 13:43:35 +0100 Edward Hervey <bilboed@bilboed.com>
4614 * tests/check/libs/rtp.c:
4615 tests: Fix indentation
4617 2009-02-22 13:42:33 +0100 Edward Hervey <bilboed@bilboed.com>
4619 * gst-libs/gst/video/video.c:
4620 libs/video: Fix gst_video_format_new_caps* functions.
4621 Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
4624 2009-02-21 11:13:36 -0800 David Schleef <ds@schleef.org>
4627 Automatic update of common submodule
4628 From 80c627d to 5d7c9cc
4630 2009-02-20 17:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4632 * gst-libs/gst/rtsp/gstrtspmessage.c:
4633 Improve key/value parsing
4634 Improve header field parsing by keeping a ref to the key/value instead of
4635 copying it into a local variable.
4637 2009-02-20 12:35:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4639 * gst-libs/gst/rtsp/gstrtspconnection.c:
4640 Add trailing \0 to message length
4641 We always put a trailing 0 at the end of the message body. Reflect this fact in
4642 the length of the message.
4644 2009-02-20 09:50:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4646 * gst-libs/gst/rtsp/gstrtspconnection.c:
4647 Don't parse headers for data messages
4648 Don't try to parse the headers on a data message because they don't have
4651 2009-02-19 12:18:29 -0800 Benjamin M. Schwartz <bens@alum.mit.edu>
4653 * ext/theora/gsttheoraenc.h:
4654 * ext/theora/theoraenc.c:
4655 theoraenc: Add property for speed level control
4656 Add property "speed-level" to control the amount of motion searching
4657 the encoder does. This is only available in libtheora >= 1.0 and
4658 will silently fail with earlier libraries. Fixes: #572275.
4659 Signed-off-by: David Schleef <ds@schleef.org>
4661 2009-02-19 17:40:45 +0100 Edward Hervey <bilboed@bilboed.com>
4663 * gst-libs/gst/video/video.c:
4664 * gst-libs/gst/video/video.h:
4665 video: Fix 'Since' tags
4667 2009-01-26 10:30:53 +0100 Edward Hervey <bilboed@bilboed.com>
4669 * docs/libs/gst-plugins-base-libs-sections.txt:
4670 * gst-libs/gst/video/video.c:
4671 * gst-libs/gst/video/video.h:
4672 video: Add flags for interlaced video along with convenience methods for interlaced caps.
4673 These three flags allow all know combinations of interlaced formats. They should
4674 only be used when the caps contain 'interlaced=True'.
4675 Fixes #163577 (yes, it's a 4 year old bug).
4677 2009-02-19 15:51:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4679 * docs/libs/gst-plugins-base-libs-sections.txt:
4680 * gst-libs/gst/rtsp/gstrtspconnection.c:
4681 * gst-libs/gst/rtsp/gstrtspconnection.h:
4682 Make RTSPConnection opaque and rename RTSPChannel
4683 Make the RTSPConnection object opaque so that we can extend it in the future.
4684 Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
4686 2009-01-26 10:31:14 +0100 Edward Hervey <bilboed@bilboed.com>
4688 * gst-libs/gst/riff/riff-media.c:
4689 Add some more mappings for h264 in riff
4691 2009-02-19 10:49:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4693 * win32/common/libgstrtsp.def:
4694 Add new RTSP symbols to def files
4695 Add the new RTSP symbols to the windows def file.
4697 2009-02-19 10:44:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4699 * docs/libs/gst-plugins-base-libs-sections.txt:
4700 * gst-libs/gst/app/gstappsink.c:
4701 * gst-libs/gst/app/gstappsink.h:
4702 * tests/check/Makefile.am:
4703 * tests/check/elements/.gitignore:
4704 * tests/check/elements/appsink.c:
4705 Add method to install callbacks on appsink
4706 Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
4708 Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
4709 performant alternative to connecting to the signals.
4710 Add a unit test for appsink.
4711 Clean up some of the appsink docs.
4712 API: GstAppSink::gst_app_sink_set_callbacks()
4714 2009-02-18 18:46:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4716 * docs/libs/gst-plugins-base-libs-sections.txt:
4717 * gst-libs/gst/rtsp/gstrtspconnection.c:
4718 * gst-libs/gst/rtsp/gstrtspconnection.h:
4719 Add RTSP accept method
4720 Add a method to accept a connection on a socket and create a GstRTSPConnection
4722 API: gst_rtsp_connection_accept()
4724 2009-02-18 17:42:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4726 * docs/libs/gst-plugins-base-libs-sections.txt:
4727 * gst-libs/gst/rtsp/gstrtspconnection.c:
4728 * gst-libs/gst/rtsp/gstrtspconnection.h:
4729 Add RTSP channel object for async io
4730 Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
4731 that the connection can be monitored from a maincontext. This allows us to
4732 operate in ASYNC mode, which is handy when building a server.
4733 Rework the old code to use the async code under the hood.
4734 API: gst_rtsp_channel_new()
4735 API: gst_rtsp_channel_unref()
4736 API: gst_rtsp_channel_attach()
4737 API: gst_rtsp_channel_queue_message()
4739 2009-02-15 07:30:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4741 * gst/audioresample/gstaudioresample.c:
4742 audioresample: Add locking to protect the resampling context
4743 When setting the quality/filter-length while PLAYING the
4744 resampling context will be destroyed and created again in
4745 some cases, which will cause crashes in the transform function
4746 if it's called at that time.
4748 2009-02-13 10:10:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4750 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
4751 * gst/videotestsrc/videotestsrc.c:
4752 ffmpegcolorspace/videotestsrc: Use v308 instead of V308
4754 2009-02-12 19:02:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4756 * gst/ffmpegcolorspace/avcodec.h:
4757 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
4758 * gst/ffmpegcolorspace/imgconvert.c:
4759 * gst/ffmpegcolorspace/imgconvert_template.h:
4760 ffmpegcolorspace: Add support for packed 4:4:4 YUV (format=V308)
4761 Only conversions from/to are implemented, which
4762 gives (indirect) support for all possible conversions.
4763 Partially fixes bug #571147.
4765 2009-02-12 18:17:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4767 * gst/videotestsrc/videotestsrc.c:
4768 videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
4769 Partially fixes bug #571147.
4771 2009-02-12 09:18:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4773 * gst-libs/gst/tag/gsttagdemux.c:
4774 tagdemux: don't abort when downstream pulls a buffer of size 0
4775 Pulling a 0-sized buffer is allowed, and we should handle this correctly instead of
4776 aborting. Fixes #571009 (wma file with ID3v2 tag).
4778 2009-02-11 16:39:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4780 * gst-libs/gst/riff/riff-read.c:
4781 riff: error out on nonsensical chunk sizes instead of aborting
4782 When encountering a nonsensical chunk size such as (guint)-1, error out cleanly instead of
4783 continuing and trying to g_memdup() 4GB of data that doesn't exist, which will either abort
4784 in g_malloc() or crash.
4785 Fixes #553295, crash with fuzzed AVI file.
4787 2009-02-11 16:39:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4790 Make git ignore backup files.
4792 2009-02-10 20:38:58 -0800 Michael Smith <msmith@syncword.(none)>
4794 * gst/playback/gstplaybin2.c:
4795 Revert "Remove pad-removed handlers after setting the decodebins to NULL."
4796 This reverts commit b36d8f3e119f9edc5993c08025614ee32642972e.
4797 This brought back some deadlocks. A small leak is better, for now. Need to
4798 figure out a way to fix the leak properly.
4800 2009-02-10 17:16:07 -0800 Michael Smith <msmith@songbirdnest.com>
4802 * gst/playback/gstplaybin2.c:
4803 playbin2: Fix segfault on notify after group change.
4804 If our group has been switched, then we get a selector active-pad
4805 notification, we don't need to notify.
4807 2009-02-10 17:10:33 -0800 Michael Smith <msmith@songbirdnest.com>
4809 * gst/playback/gstplaysink.c:
4810 playbin2: Look for volume/mute properties recursively in audio element.
4811 Rather than only checking for volume property on the audio sink
4812 directly, recursively look for it on sinks within it (if it's a bin).
4813 Allows use of sink-as-volume-control where the application has supplied
4814 an audio-sink bin that includes a real audio sink internally.
4816 2009-02-10 18:29:22 +0000 Christian Schaller <cschalle@crazyhorse.localdomain>
4818 * gst-plugins-base.spec.in:
4819 Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base
4821 2009-02-10 17:39:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4823 * gst/videotestsrc/videotestsrc.c:
4824 videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
4825 Partially fixes bug #571147.
4827 2009-02-10 17:37:06 +0100 Peter Kjellerstedt <pkj@axis.com>
4829 * gst-libs/gst/rtsp/gstrtspmessage.c:
4830 gstrtspmessage: Minor documentation correction.
4831 Corrected documentation about what needs to be freed after calling
4832 gst_rtsp_message_new(), gst_rtsp_message_new_request(),
4833 gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
4835 2009-02-10 11:00:12 +0100 Antoine Tremblay <hexa00@gmail.com>
4837 * ext/alsa/gstalsamixer.c:
4838 alsamixer: Fix race condition that made alsamixer not working properly
4839 This is due to race conditions between functions that
4840 modified the mixer like set_volume and
4841 snd_mixer_handle_events since the handle_events
4842 can now be called at any time.
4843 Fixed by adding locking around any snd_mixer call
4844 since even read functions can modify the mixer stucture, since
4845 alsa likes to clear it's values before reading new ones.
4846 The favorite race condition seemed to be that set_volume
4847 called read_elem (in alsalib) that reset the volumes to
4848 0 and then read them with read_x_volume. This read looped
4849 on each channel and as the race condition occured the
4850 channels value could be anything , most of the time
4851 it was 0. Thus no value was read or only the value of
4852 one channel was and the volume was reset to 0.
4855 2009-02-09 12:02:21 +0100 Edward Hervey <bilboed@bilboed.com>
4858 Bump revision to use for common submodule.
4860 2009-02-05 15:47:00 +0200 Stefan Kost <ensonic@users.sf.net>
4862 * sys/xvimage/xvimagesink.c:
4863 xvimagesink: do not call _xwindow_clear on ready->paused.
4864 Calling clear at that transition does things like stopping xvideo (which is not
4865 running at that time) and also clearing anything what the application might have drawn.
4866 This breaks handle-expose and autopaint-colorkey features.
4868 2009-02-04 17:03:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4870 * docs/libs/gst-plugins-base-libs-sections.txt:
4871 * gst-libs/gst/rtsp/gstrtsprange.c:
4872 * gst-libs/gst/rtsp/gstrtsprange.h:
4873 RTSPRange: Add method to serialize ranges
4874 Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
4875 be used by a server.
4876 API: GstRTSPRange::gst_rtsp_range_to_string()
4878 2009-02-04 13:16:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4880 * gst-libs/gst/rtsp/gstrtspurl.c:
4881 * gst-libs/gst/rtsp/gstrtspurl.h:
4882 GstRTSPUrl: Add some const to methods
4883 Add const to the methods that do not modify the object.
4885 2009-02-04 13:53:30 +0200 Stefan Kost <ensonic@users.sf.net>
4887 * gst/playback/gstplaysink.c:
4888 playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
4889 The flags where present but actually not been taken into account.
4891 2009-02-04 12:06:38 +0200 Stefan Kost <ensonic@users.sf.net>
4893 * gst/audioresample/gstaudioresample.c:
4894 audioresample: Add a proper deprecation comment and also drop G_PARAM_CONSTRUCT.
4895 The comment will ensure that is is marked properly in the docs and the
4896 GParamSpecflag was causing a duplicated initialisation of the same value.
4898 2009-02-04 11:18:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4900 * gst-libs/gst/rtsp/gstrtspconnection.c:
4901 Add more g_return_if_fail() calls
4902 Check that we have a valid file descriptor before entering certain functions in
4903 order to avoid undesirable situations.
4904 Add some more debugging in the connect method.
4906 2009-02-04 10:31:21 +0200 Stefan Kost <ensonic@users.sf.net>
4909 * gst/audioresample/Makefile.am:
4910 * gst/audioresample/gstaudioresample.c:
4911 audioresample: Only pull in liboil if its actualy used.
4912 Liboil still has quite significant startup overhead especialy on embedded
4913 platforms. In audioresample it was only used for the profiling timer.
4915 2009-02-03 15:26:08 +0200 Stefan Kost <ensonic@users.sf.net>
4917 * gst/typefind/gsttypefindfunctions.c:
4918 typefind: Make the flac check more tight to not mistace some aac files for flac. Fixes #570356.
4919 Add comments about the flac format. Tighten the check to not allow values that
4922 2009-02-03 10:52:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4924 * win32/common/libgstrtsp.def:
4926 Add new methods to the windows def file.
4928 2009-02-02 17:25:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4930 * gst-libs/gst/pbutils/install-plugins.c:
4931 * tests/check/libs/pbutils.c:
4932 pbutils: remove duplicate detail strings when calling the external codec installer
4933 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.
4935 2009-02-02 18:05:42 +0200 Stefan Kost <ensonic@users.sf.net>
4937 * gst-libs/gst/audio/gstaudiosink.c:
4938 * gst-libs/gst/audio/gstaudiosink.h:
4939 Add a FIXME 0.11. Make the log message a bit more detailed and add comments.
4941 2009-02-02 15:43:03 +0200 Stefan Kost <ensonic@users.sf.net>
4944 * gst/audioresample/gstaudioresample.c:
4945 Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark.
4947 2009-02-02 13:30:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4949 * sys/ximage/ximagesink.c:
4950 Fix buffer_alloc in ximagesink
4951 Remove some useless debug info that reported wrong image sizes.
4952 When upstream does not accept out suggested size, fall back to allocating an
4953 image of the requested width/height instead of the currently configured size.
4954 The problem is that an image is reused from the pool because the width/height
4955 match but the caps on the new buffer are the requested caps with possibly
4956 different height/width resulting in errors.
4958 2009-02-02 12:54:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4960 * gst/playback/gstdecodebin2.c:
4961 * gst/playback/gsturidecodebin.c:
4962 Fix documentation for autoplug-select
4963 fix the documentation strings for the autoplug-select signal.
4966 2009-02-02 10:09:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4968 * gst-libs/gst/rtsp/gstrtspmessage.c:
4969 Fix string leak in rtspmessage
4970 when we remove a header field from a message we must free the value associated
4971 with the key to avoid a memory leak.
4973 2009-01-31 18:45:47 +0200 Stefan Kost <ensonic@users.sf.net>
4975 * docs/libs/gst-plugins-base-libs-docs.sgml:
4976 Its "Base Library" and not just "Library".
4978 2009-01-31 18:44:32 +0200 Stefan Kost <ensonic@users.sf.net>
4980 * gst-libs/gst/audio/gstaudiofilter.c:
4981 Link to the class, as we can't link to the members yet.
4983 2009-01-30 17:48:23 -0800 Michael Smith <msmith@songbirdnest.com>
4985 * gst/playback/gstplaybin2.c:
4986 Remove pad-removed handlers after setting the decodebins to NULL.
4987 They do needed cleanup; without this we leak selector requestpads.
4989 2009-01-30 17:47:07 -0800 Michael Smith <msmith@songbirdnest.com>
4991 * gst/playback/gstplaybin2.c:
4992 Unref selector request pad even if we no longer have a selector.
4993 During destruction, we won't have a selector any more, but we still need
4994 to unref the pad to avoid leaking it.
4996 2009-01-30 15:23:23 -0800 Michael Smith <msmith@songbirdnest.com>
4998 * gst/playback/gstplaybin2.c:
4999 Unref source in playbin2's finalize method
5001 2009-01-30 12:04:01 -0800 Michael Smith <msmith@songbirdnest.com>
5003 * gst/playback/gstplaysink.c:
5004 Fix more leaks of pads and elements in gstplaysink.
5005 Don't keep extra references to volume and mute elements; we don't need
5007 Ensure we unref pads that we have references to, and release request
5010 2009-01-30 11:04:37 -0800 Michael Smith <msmith@songbirdnest.com>
5012 * gst/playback/gstplaysink.c:
5013 Avoid leaking all playsinks. Fix some internal leaks.
5014 Playsink was holding references to itself. Don't do that, it's not cool.
5015 Also, free all chains in dispose.
5017 2009-01-30 10:54:12 -0800 Michael Smith <msmith@songbirdnest.com>
5019 * gst/playback/gstplaybin2.c:
5020 Unref peer request pad after releasing it, since we hold a reference.
5022 2009-01-30 10:52:52 -0800 Michael Smith <msmith@songbirdnest.com>
5024 * gst/playback/gstplaybin2.c:
5025 Fix caps leak in playbin2.
5027 2009-01-30 10:51:11 -0800 Michael Smith <msmith@songbirdnest.com>
5029 * gst/playback/gstplaybin2.c:
5030 Unref active pad from selector when finding active stream.
5032 2009-01-30 10:49:55 -0800 Michael Smith <msmith@songbirdnest.com>
5034 * gst/playback/gstplaybin2.c:
5035 Free uris when finalizing playbin2 instance.
5037 2009-01-30 10:38:17 -0800 Michael Smith <msmith@songbirdnest.com>
5039 * gst/playback/gsturidecodebin.c:
5040 Unref pads when iterating over them in analyse_source.
5041 Fixes leak of source's srcpad when using uridecodebin.
5043 2009-01-30 22:22:07 +0200 Stefan Kost <ensonic@users.sf.net>
5045 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
5046 Add releaseinfo with online url.
5048 2009-01-30 17:58:15 +0000 Jan Schmidt <jan.schmidt@sun.com>
5050 * gst/playback/gstplaybasebin.c:
5051 Fix compilation warning on Forte
5053 2009-01-30 17:16:39 +0000 Jan Schmidt <jan.schmidt@sun.com>
5055 * gst/adder/gstadder.c:
5056 Don't do void pointer arithmetic.
5058 2009-01-30 17:25:51 +0000 Jan Schmidt <thaytan@noraisin.net>
5063 2009-01-30 08:50:53 +0100 Edward Hervey <bilboed@bilboed.com>
5067 Use a symbolic link for the pre-commit client-side hook
5069 2009-01-30 08:12:42 +0100 Edward Hervey <bilboed@bilboed.com>
5072 Add more files/directories to ignore
5074 2009-01-29 14:00:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5076 * gst-libs/gst/rtsp/gstrtspdefs.c:
5078 Fix some typos in the doc string of the new
5079 gst_rtsp_options_as_string() method.
5081 2009-01-29 11:55:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5083 * docs/libs/gst-plugins-base-libs-sections.txt:
5084 * gst-libs/gst/rtsp/gstrtspconnection.c:
5085 * gst-libs/gst/rtsp/gstrtspmessage.c:
5086 * gst-libs/gst/rtsp/gstrtspmessage.h:
5087 Add new RTSP message method to set header
5088 Add gst_rtsp_message_take_header() that takes ownership of the passed header
5089 value. This allows us to avoid an allocations and memory copy in some
5091 API: GstRTSPMessage::gst_rtsp_message_take_header()
5093 2009-01-29 11:51:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5095 * docs/libs/gst-plugins-base-libs-sections.txt:
5096 Add new method to docs
5097 Add the new gst_rtsp_options_as_text() method to the docs.
5099 2009-01-28 11:48:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5101 * gst-libs/gst/rtsp/gstrtspdefs.c:
5102 * gst-libs/gst/rtsp/gstrtspdefs.h:
5103 Add method to serialize RTSP options
5104 Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
5106 API: GstRTSP::gst_rtsp_options_as_text()
5108 2009-01-26 17:59:37 -0800 Michael Smith <msmith@songbirdnest.com>
5110 * gst/typefind/gsttypefindfunctions.c:
5111 Ensure we have sufficient data when using data scan contexts.
5112 Fixes crashes typefinding things that look like they might contain AAC
5113 data (but probably aren't actually AAC).
5115 2009-01-26 23:32:09 +0000 Jan Schmidt <thaytan@noraisin.net>
5117 * ext/gio/Makefile.am:
5118 Fix include order for gio plugin
5120 2009-01-23 23:59:48 +0000 Jan Schmidt <thaytan@noraisin.net>
5122 * win32/common/config.h:
5123 Update win32 config.h for 0.10.22.1 dev cycle
5125 2009-01-23 23:16:11 +0000 Jan Schmidt <thaytan@noraisin.net>
5128 * docs/libs/.gitignore:
5129 * gst-libs/gst/audio/.gitignore:
5130 * gst-libs/gst/video/.gitignore:
5132 * tests/examples/dynamic/.gitignore:
5133 Extend and clean up git ignores
5135 2009-01-23 12:31:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5138 * docs/plugins/Makefile.am:
5139 * docs/plugins/gst-plugins-base-plugins-sections.txt:
5140 * docs/plugins/gst-plugins-base-plugins.args:
5141 * docs/plugins/gst-plugins-base-plugins.hierarchy:
5142 * docs/plugins/gst-plugins-base-plugins.interfaces:
5143 * docs/plugins/gst-plugins-base-plugins.prerequisites:
5144 * docs/plugins/inspect/plugin-adder.xml:
5145 * docs/plugins/inspect/plugin-alsa.xml:
5146 * docs/plugins/inspect/plugin-app.xml:
5147 * docs/plugins/inspect/plugin-audioconvert.xml:
5148 * docs/plugins/inspect/plugin-audiorate.xml:
5149 * docs/plugins/inspect/plugin-audioresample.xml:
5150 * docs/plugins/inspect/plugin-audiotestsrc.xml:
5151 * docs/plugins/inspect/plugin-cdparanoia.xml:
5152 * docs/plugins/inspect/plugin-decodebin.xml:
5153 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
5154 * docs/plugins/inspect/plugin-gdp.xml:
5155 * docs/plugins/inspect/plugin-gio.xml:
5156 * docs/plugins/inspect/plugin-gnomevfs.xml:
5157 * docs/plugins/inspect/plugin-libvisual.xml:
5158 * docs/plugins/inspect/plugin-ogg.xml:
5159 * docs/plugins/inspect/plugin-pango.xml:
5160 * docs/plugins/inspect/plugin-playback.xml:
5161 * docs/plugins/inspect/plugin-queue2.xml:
5162 * docs/plugins/inspect/plugin-subparse.xml:
5163 * docs/plugins/inspect/plugin-tcp.xml:
5164 * docs/plugins/inspect/plugin-theora.xml:
5165 * docs/plugins/inspect/plugin-typefindfunctions.xml:
5166 * docs/plugins/inspect/plugin-uridecodebin.xml:
5167 * docs/plugins/inspect/plugin-video4linux.xml:
5168 * docs/plugins/inspect/plugin-videorate.xml:
5169 * docs/plugins/inspect/plugin-videoscale.xml:
5170 * docs/plugins/inspect/plugin-videotestsrc.xml:
5171 * docs/plugins/inspect/plugin-volume.xml:
5172 * docs/plugins/inspect/plugin-vorbis.xml:
5173 * docs/plugins/inspect/plugin-ximagesink.xml:
5174 * docs/plugins/inspect/plugin-xvimagesink.xml:
5175 * gst/audioresample/Makefile.am:
5176 * gst/audioresample/README:
5177 * gst/audioresample/arch.h:
5178 * gst/audioresample/buffer.c:
5179 * gst/audioresample/buffer.h:
5180 * gst/audioresample/debug.c:
5181 * gst/audioresample/debug.h:
5182 * gst/audioresample/fixed_arm4.h:
5183 * gst/audioresample/fixed_arm5e.h:
5184 * gst/audioresample/fixed_bfin.h:
5185 * gst/audioresample/fixed_debug.h:
5186 * gst/audioresample/fixed_generic.h:
5187 * gst/audioresample/functable.c:
5188 * gst/audioresample/functable.h:
5189 * gst/audioresample/gstaudioresample.c:
5190 * gst/audioresample/gstaudioresample.h:
5191 * gst/audioresample/resample.c:
5192 * gst/audioresample/resample.h:
5193 * gst/audioresample/resample_chunk.c:
5194 * gst/audioresample/resample_functable.c:
5195 * gst/audioresample/resample_ref.c:
5196 * gst/audioresample/resample_sse.h:
5197 * gst/audioresample/speex_resampler.h:
5198 * gst/audioresample/speex_resampler_double.c:
5199 * gst/audioresample/speex_resampler_float.c:
5200 * gst/audioresample/speex_resampler_int.c:
5201 * gst/audioresample/speex_resampler_wrapper.h:
5202 * gst/speexresample/Makefile.am:
5203 * gst/speexresample/README:
5204 * gst/speexresample/arch.h:
5205 * gst/speexresample/fixed_arm4.h:
5206 * gst/speexresample/fixed_arm5e.h:
5207 * gst/speexresample/fixed_bfin.h:
5208 * gst/speexresample/fixed_debug.h:
5209 * gst/speexresample/fixed_generic.h:
5210 * gst/speexresample/gstspeexresample.c:
5211 * gst/speexresample/gstspeexresample.h:
5212 * gst/speexresample/resample.c:
5213 * gst/speexresample/resample_sse.h:
5214 * gst/speexresample/speex_resampler.h:
5215 * gst/speexresample/speex_resampler_double.c:
5216 * gst/speexresample/speex_resampler_float.c:
5217 * gst/speexresample/speex_resampler_int.c:
5218 * gst/speexresample/speex_resampler_wrapper.h:
5219 * gst/typefind/gsttypefindfunctions.c:
5220 * tests/check/Makefile.am:
5221 * tests/check/elements/audioresample.c:
5222 * tests/check/elements/speexresample.c:
5223 Rename files and types from speexresample to audioresample
5224 Rename files and types from speexresample to audioresample
5225 to finish the move and to prevent any confusion.
5227 2009-01-23 11:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5229 * sys/xvimage/xvimagesink.c:
5230 Add some more debugging to the Xv strides
5231 Add some more debugging to the strides as they are received from the server and
5232 the expected strides.
5234 2009-01-23 11:40:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5236 * gst/typefind/gsttypefindfunctions.c:
5237 Add typefind function for gsm
5238 Because core now supports typefindfactories without a typefind function we can
5239 register a factory fo GSM that will --if all else fails-- assume the file is a
5240 GSM file based on the registered extension.
5243 2009-01-23 11:37:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5245 * gst/playback/gsturidecodebin.c:
5246 Use more performant link function
5247 We can use gst_element_link_pads() instead of the more generic
5248 gst_element_link() function because we know the pads. This saves some cycles
5249 because the more generic function needs to search for possible compatible caps
5252 2009-01-23 11:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5254 * gst-libs/gst/riff/riff-ids.h:
5255 * gst-libs/gst/riff/riff-media.c:
5256 Add more codec ids for RIFF formats
5257 Handle codec ID for various other AAC formats.
5258 Sync the list of possible codec ids with that of ffmpeg.
5261 2009-01-23 11:27:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5263 * ext/theora/theoradec.c:
5264 Use rounded values for image strides and sizes
5265 Round up the height before calculating the expected size and
5266 strides of the output image.
5268 2009-01-23 11:23:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5270 * ext/alsa/gstalsasink.c:
5271 Improve debug message
5272 Improve the debug message when alsa returns an error.
5274 2009-01-23 11:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5276 * gst-libs/gst/app/gstappsrc.c:
5277 Reset queued_bytes counter when flushing
5278 Set the amount of queued bytes in the internal queue back to 0 when we clear the
5282 2009-01-23 10:19:27 +0100 Benjamin Gaignard <benjamin@gaignard.net>
5284 * gst/typefind/gsttypefindfunctions.c:
5285 Add typefinder for Mobile XMF. Fixes bug #568707.
5287 2009-01-23 10:00:11 +0100 Brian Cameron <brian.cameron@sun.com>
5290 Fix linking on Solaris. Fixes bug #568482.
5291 Check for nsl and socket libraries and add them to
5292 LIBS if they're found. They're needed for socket()
5293 and gethostbyname() on Solaris.
5295 2009-01-22 22:09:47 +0000 Jan Schmidt <thaytan@noraisin.net>
5297 * gst/playback/gstplaybasebin.c:
5298 Fix use-after-unref problem noticed by Josep Torra Valles, and run
5301 2009-01-22 17:46:59 +0200 Stefan Kost <ensonic@users.sf.net>
5304 Update common snapshot.
5306 2009-01-22 13:47:24 +0100 Sebastian Dröge <slomo@circular-chaos.org>
5311 2009-01-22 13:12:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5313 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
5315 2009-01-22 10:14:28 +0100 Sebastian Dröge <slomo@circular-chaos.org>
5317 * gst-libs/gst/fft/gstfftf32.c:
5318 * gst-libs/gst/fft/gstfftf64.c:
5319 * gst-libs/gst/fft/gstffts16.c:
5320 * gst-libs/gst/fft/gstffts32.c:
5321 Reduce the number of allocations for creating FFT contexts
5322 Reduce the number of allocations from 2 to 1 for every FFT
5323 context by allocating enough memory for the FFT context
5324 and passing parts of it to the kissfft allocation functions.
5326 2009-01-22 11:32:56 +0000 Jan Schmidt <thaytan@noraisin.net>
5329 Back to devel -> 0.10.22.1
5331 2009-01-22 05:57:53 +0100 Edward Hervey <bilboed@bilboed.com>
5335 Install and use pre-commit indentation hook from common
5337 2009-01-21 13:09:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5339 * gst-libs/gst/rtp/gstrtpbuffer.c:
5340 * tests/check/libs/rtp.c:
5341 Avoid overflows in the padding checks by doing the check slightly
5343 Add a unit test to check for correct behaviour.
5345 2009-01-21 04:31:32 +0100 Edward Hervey <bilboed@bilboed.com>
5348 autogen.sh : Use git submodule
5350 === release 0.10.22 ===
5352 2009-01-19 23:10:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5358 * docs/plugins/gst-plugins-base-plugins.hierarchy:
5359 * docs/plugins/gst-plugins-base-plugins.interfaces:
5360 * docs/plugins/gst-plugins-base-plugins.prerequisites:
5361 * docs/plugins/inspect/plugin-adder.xml:
5362 * docs/plugins/inspect/plugin-alsa.xml:
5363 * docs/plugins/inspect/plugin-app.xml:
5364 * docs/plugins/inspect/plugin-audioconvert.xml:
5365 * docs/plugins/inspect/plugin-audiorate.xml:
5366 * docs/plugins/inspect/plugin-audioresample.xml:
5367 * docs/plugins/inspect/plugin-audiotestsrc.xml:
5368 * docs/plugins/inspect/plugin-cdparanoia.xml:
5369 * docs/plugins/inspect/plugin-decodebin.xml:
5370 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
5371 * docs/plugins/inspect/plugin-gdp.xml:
5372 * docs/plugins/inspect/plugin-gnomevfs.xml:
5373 * docs/plugins/inspect/plugin-libvisual.xml:
5374 * docs/plugins/inspect/plugin-ogg.xml:
5375 * docs/plugins/inspect/plugin-pango.xml:
5376 * docs/plugins/inspect/plugin-playback.xml:
5377 * docs/plugins/inspect/plugin-queue2.xml:
5378 * docs/plugins/inspect/plugin-subparse.xml:
5379 * docs/plugins/inspect/plugin-tcp.xml:
5380 * docs/plugins/inspect/plugin-theora.xml:
5381 * docs/plugins/inspect/plugin-typefindfunctions.xml:
5382 * docs/plugins/inspect/plugin-uridecodebin.xml:
5383 * docs/plugins/inspect/plugin-video4linux.xml:
5384 * docs/plugins/inspect/plugin-videorate.xml:
5385 * docs/plugins/inspect/plugin-videoscale.xml:
5386 * docs/plugins/inspect/plugin-videotestsrc.xml:
5387 * docs/plugins/inspect/plugin-volume.xml:
5388 * docs/plugins/inspect/plugin-vorbis.xml:
5389 * docs/plugins/inspect/plugin-ximagesink.xml:
5390 * docs/plugins/inspect/plugin-xvimagesink.xml:
5391 * gst-plugins-base.doap:
5421 * win32/common/config.h:
5423 Original commit message from CVS:
5426 2009-01-19 22:01:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5458 Original commit message from CVS:
5461 2009-01-16 11:44:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5463 gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
5464 Original commit message from CVS:
5465 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
5466 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
5467 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
5468 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
5469 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
5470 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
5471 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
5472 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
5473 Use correct struct alignment everywhere to prevent unaligned
5474 memory accesses, resulting in SIGBUS on sparc and probably others.
5477 2009-01-16 11:40:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5479 gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
5480 Original commit message from CVS:
5481 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
5482 Forward unknown events upstream to allow latency configuration.
5485 2009-01-13 14:47:19 +0000 Wim Taymans <wim.taymans@gmail.com>
5487 gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
5488 Original commit message from CVS:
5489 * gst/playback/gstplaybin2.c: (groups_set_locked_state):
5490 Provide the right arguments to a debug line.
5492 2009-01-13 06:51:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
5494 sys/xvimage/xvimagesink.c: Don't reset the colorkey when element is reused. Fixes #567511.
5495 Original commit message from CVS:
5496 * sys/xvimage/xvimagesink.c:
5497 Don't reset the colorkey when element is reused. Fixes #567511.
5499 2009-01-09 23:42:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5501 configure.ac: 0.10.21.3 pre-release
5502 Original commit message from CVS:
5504 0.10.21.3 pre-release
5506 2009-01-09 23:13:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5508 gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
5509 Original commit message from CVS:
5510 * gst-libs/gst/app/gstappsink.c:
5511 Store the returned signal id in the right slot when
5512 registering the pull-buffer signal.
5514 Spotted by: Thomas Vander Stichele <thomas at apestaart dot org>
5516 2009-01-09 17:17:50 +0000 Tim-Philipp Müller <tim@centricular.net>
5518 gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
5519 Original commit message from CVS:
5520 * gst-libs/gst/interfaces/mixer.c:
5521 Small docs addition to clarify that one really mustn't free
5522 the constant GList returned (#566812).
5524 2009-01-08 17:18:24 +0000 Wim Taymans <wim.taymans@gmail.com>
5526 Add GType for GstRTSPUrl and expose a copy function because we can.
5527 Original commit message from CVS:
5528 * docs/libs/gst-plugins-base-libs-sections.txt:
5529 * gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
5530 (gst_rtsp_url_get_type), (gst_rtsp_url_copy):
5531 * gst-libs/gst/rtsp/gstrtspurl.h:
5532 * win32/common/libgstrtsp.def:
5533 Add GType for GstRTSPUrl and expose a copy function because we can.
5534 API: gst_rtsp_url_copy()
5537 2009-01-07 18:36:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5539 Add plugin dependency for the GIO and GVfs modules.
5540 Original commit message from CVS:
5542 * ext/gio/gstgio.c: (plugin_init):
5543 Add plugin dependency for the GIO and GVfs modules.
5546 2009-01-07 18:32:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5548 Add plugin dependency for the gnomevfs modules.
5549 Original commit message from CVS:
5551 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
5552 Add plugin dependency for the gnomevfs modules.
5555 2009-01-07 18:30:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5557 win32/common/libgstcdda.def: Add new symbol to the list of exported symbols.
5558 Original commit message from CVS:
5559 * win32/common/libgstcdda.def:
5560 Add new symbol to the list of exported symbols.
5562 2009-01-07 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
5564 gst/playback/gstplaybin2.c: Fix some comments and docs.
5565 Original commit message from CVS:
5566 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
5567 (gst_play_bin_set_uri), (gst_play_bin_set_suburi),
5568 (no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
5569 (activate_group), (deactivate_group), (groups_set_locked_state),
5570 (gst_play_bin_change_state):
5571 Fix some comments and docs.
5572 Post an error message when we fail to link the selector to the sink.
5573 Remove pushing of EOS, this seems unneeded.
5574 Lock the state of deactivated groups so that they don't accidentally
5575 reactivate when the playbin2 state changes.
5576 Reuse uridecodebins.
5577 Unlock and relock state of groups when playbin goes to NULL.
5580 * gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
5581 Only do something in the pad removed callback when we are dealing with
5582 our sourcepads because the sinkpads don't have a ghostpad.
5584 2009-01-07 10:50:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5586 gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
5587 Original commit message from CVS:
5588 * gst-libs/gst/cdda/gstcddabasesrc.c:
5589 * gst-libs/gst/cdda/gstcddabasesrc.h:
5590 Make the GType of GstCDDABaseSrcMode public for bindings.
5593 2009-01-06 18:03:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5595 Use new core API to make registry re-scan the plugin whenever visualisations are added or removed (see #350477).
5596 Original commit message from CVS:
5598 * ext/libvisual/visual.c: (plugin_init):
5599 Use new core API to make registry re-scan the plugin
5600 whenever visualisations are added or removed (see #350477).
5602 2009-01-06 17:30:31 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
5604 gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
5605 Original commit message from CVS:
5606 Patch by: José Alburquerque <jaalburqu svn gnome org>
5607 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
5608 * gst-libs/gst/audio/gstaudioclock.h:
5609 Make gst_audio_clock_new use const gchar* to ease the wrapping of
5610 C++ bindings. Fixes #566723.
5612 2009-01-06 12:16:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5614 Add pkg-config files for libgstapp. Fixes bug #566761.
5615 Original commit message from CVS:
5617 * pkgconfig/Makefile.am:
5618 * pkgconfig/gstreamer-app-uninstalled.pc.in:
5619 * pkgconfig/gstreamer-app.pc.in:
5620 Add pkg-config files for libgstapp. Fixes bug #566761.
5622 2009-01-06 11:10:29 +0000 Tim-Philipp Müller <tim@centricular.net>
5624 gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
5625 Original commit message from CVS:
5626 * gst-libs/gst/app/gstappsink.c:
5627 * gst-libs/gst/app/gstappsink.h:
5628 * gst-libs/gst/app/gstappsrc.c:
5629 * gst-libs/gst/app/gstappsrc.h:
5630 Make debug categories static. Use _element_class_set_details_simple().
5632 2009-01-06 10:56:45 +0000 Tim-Philipp Müller <tim@centricular.net>
5634 gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
5635 Original commit message from CVS:
5636 * gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
5637 (gst_app_sink_class_init), (gst_app_sink_init),
5638 (gst_app_sink_dispose), (gst_app_sink_finalize),
5639 (gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
5640 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
5641 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
5642 (gst_app_sink_render), (gst_app_sink_getcaps),
5643 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
5644 (gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
5645 (gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
5646 (gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
5647 (gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
5648 (gst_app_sink_pull_buffer)::
5649 * gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
5650 * gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
5651 (gst_app_src_class_init), (gst_app_src_init),
5652 (gst_app_src_flush_queued), (gst_app_src_dispose),
5653 (gst_app_src_finalize), (gst_app_src_set_property),
5654 (gst_app_src_get_property), (gst_app_src_unlock),
5655 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
5656 (gst_app_src_is_seekable), (gst_app_src_check_get_range),
5657 (gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
5658 (gst_app_src_set_caps), (gst_app_src_get_caps),
5659 (gst_app_src_set_size), (gst_app_src_get_size),
5660 (gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
5661 (gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
5662 (gst_app_src_set_latencies), (gst_app_src_set_latency),
5663 (gst_app_src_get_latency), (gst_app_src_push_buffer_full),
5664 (gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
5665 * gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
5666 Move private data into a private instance struct. Add padding to
5667 instance and class structures exposed in public headers. Add
5668 Since markers to the gtk-doc blurbs (#566750).
5670 2009-01-06 10:50:37 +0000 Wim Taymans <wim.taymans@gmail.com>
5672 tests/examples/app/appsrc_ex.c: Some comments.
5673 Original commit message from CVS:
5674 * tests/examples/app/appsrc_ex.c: (main):
5676 When pulling a buffer we can get NULL when the element is EOS, don't try
5677 to unref this NULL buffer.
5679 2009-01-06 10:16:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5681 gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
5682 Original commit message from CVS:
5683 * gst-libs/gst/video/Makefile.am:
5684 * gst-libs/gst/video/video.h:
5685 Fix up build flags and include statement for the new generated
5686 enumtypes files, to fix dist.
5688 2009-01-05 23:04:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5690 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
5691 Original commit message from CVS:
5693 * docs/libs/Makefile.am:
5694 * docs/libs/gst-plugins-base-libs-docs.sgml:
5695 * docs/libs/gst-plugins-base-libs-sections.txt:
5696 * docs/plugins/Makefile.am:
5697 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
5698 * docs/plugins/gst-plugins-base-plugins-sections.txt:
5699 * docs/plugins/gst-plugins-base-plugins.args:
5700 * docs/plugins/gst-plugins-base-plugins.hierarchy:
5701 * docs/plugins/gst-plugins-base-plugins.interfaces:
5702 * docs/plugins/gst-plugins-base-plugins.prerequisites:
5703 * docs/plugins/gst-plugins-base-plugins.signals:
5704 * docs/plugins/inspect/plugin-app.xml:
5705 * gst-libs/gst/Makefile.am:
5706 * gst-libs/gst/app/gstappsink.c:
5707 * gst-libs/gst/app/gstappsrc.c:
5708 * tests/examples/Makefile.am:
5709 * tests/examples/app/Makefile.am:
5710 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
5712 2009-01-05 17:13:13 +0000 Wim Taymans <wim.taymans@gmail.com>
5714 gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
5715 Original commit message from CVS:
5716 * gst-libs/gst/audio/gstbaseaudiosink.c:
5717 (gst_base_audio_sink_change_state):
5718 Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
5719 take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
5720 this because the async_play method is deprecated and usually not called
5723 2009-01-05 12:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
5725 gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
5726 Original commit message from CVS:
5727 * gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
5728 Disconnect signal handlers before destroying a previous decodebin so
5729 that we don't end up causing deadlocks. Fixes #566586.
5731 2009-01-05 10:59:35 +0000 Wim Taymans <wim.taymans@gmail.com>
5733 gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
5734 Original commit message from CVS:
5735 * gst/audiotestsrc/gstaudiotestsrc.c:
5736 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
5737 (gst_audio_test_src_check_get_range),
5738 (gst_audio_test_src_set_property),
5739 (gst_audio_test_src_get_property):
5740 * gst/audiotestsrc/gstaudiotestsrc.h:
5741 Add property to control pull/push based scheduling.
5743 2009-01-02 15:04:13 +0000 Alessandro Decina <alessandro.d@gmail.com>
5745 Make the seek and colorkey examples depend on gtk+-x11 as they use
5746 Original commit message from CVS:
5748 * tests/examples/seek/Makefile.am:
5749 * tests/icles/Makefile.am:
5750 Make the seek and colorkey examples depend on gtk+-x11 as they use
5752 Fixes the build with gtk+-quartz.
5754 2008-12-31 16:04:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5756 win32/common/: Add new exports to win32 files.
5757 Original commit message from CVS:
5758 * win32/common/libgstaudio.def:
5759 * win32/common/libgsttag.def:
5760 * win32/common/libgstvideo.def:
5761 Add new exports to win32 files.
5763 2008-12-31 13:31:55 +0000 Edward Hervey <bilboed@bilboed.com>
5765 gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
5766 Original commit message from CVS:
5767 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
5768 * gst-libs/gst/tag/gsttagdemux.h:
5769 Add GType for GstTagDemuxResult enum.
5771 2008-12-31 13:01:30 +0000 Edward Hervey <bilboed@bilboed.com>
5773 gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
5774 Original commit message from CVS:
5775 * gst-libs/gst/video/Makefile.am:
5776 * gst-libs/gst/video/video.h:
5777 Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
5778 This will help bindings to use it.
5780 2008-12-31 11:20:26 +0000 Edward Hervey <bilboed@bilboed.com>
5782 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
5783 Original commit message from CVS:
5784 * gst-libs/gst/audio/Makefile.am:
5785 * gst-libs/gst/audio/audio.c:
5786 * gst-libs/gst/audio/multichannel.h:
5787 * gst-libs/gst/audio/testchannels.c:
5789 * win32/common/audio-enumtypes.c:
5790 (gst_audio_channel_position_get_type),
5791 (gst_ring_buffer_state_get_type),
5792 (gst_ring_buffer_seg_state_get_type),
5793 (gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
5794 * win32/common/audio-enumtypes.h:
5795 * win32/common/multichannel-enumtypes.c:
5796 * win32/common/multichannel-enumtypes.h:
5797 * win32/vs6/grammar.dsp:
5798 * win32/vs6/libgstaudio.dsp:
5799 * win32/vs7/libgstaudio.vcproj:
5800 * win32/vs8/libgstaudio.vcproj:
5801 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
5802 audio- in order to wrap all enums declarations of that library.
5803 This modification should not matter since that header file is not a
5804 public header (it will be included by public headers).
5805 Modify win32 crap^Wfiles accordingly.
5807 2008-12-30 17:55:07 +0000 Edward Hervey <bilboed@bilboed.com>
5809 gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
5810 Original commit message from CVS:
5811 * gst-libs/gst/audio/gstbaseaudiosrc.h:
5812 * gst-libs/gst/audio/gstbaseaudiosink.h:
5813 Complete Sebastien's commit from the 13th by exporting the
5814 _slave_method_get_type() methods.
5816 2008-12-29 16:45:20 +0000 Wim Taymans <wim.taymans@gmail.com>
5818 gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
5819 Original commit message from CVS:
5820 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
5821 (gst_app_src_init), (gst_app_src_set_property),
5822 (gst_app_src_get_property), (gst_app_src_query),
5823 (gst_app_src_set_latencies), (gst_app_src_set_latency),
5824 (gst_app_src_get_latency), (gst_app_src_push_buffer_full):
5825 * gst-libs/gst/app/gstappsrc.h:
5826 Add properties and methods to configure and retrieve the min and max
5829 2008-12-20 17:38:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5831 ext/: Implement URI query. Fixes bug #562949.
5832 Original commit message from CVS:
5833 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_query):
5834 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_class_init),
5835 (gst_gio_base_src_query):
5836 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_query):
5837 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
5838 (gst_gnome_vfs_src_query):
5839 Implement URI query. Fixes bug #562949.
5841 2008-12-20 12:48:43 +0000 Wim Taymans <wim.taymans@gmail.com>
5843 gst/playback/gstplaybin2.c: Add some debug info.
5844 Original commit message from CVS:
5845 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
5846 Add some debug info.
5847 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
5848 (gst_play_sink_reconfigure), (gst_play_sink_request_pad),
5849 (gst_play_sink_release_pad):
5850 Add some more debug info.
5851 Reconfigure the audio chain when we switch between raw and encoded audio
5852 in gapless playback.
5854 2008-12-20 12:45:03 +0000 Wim Taymans <wim.taymans@gmail.com>
5856 gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
5857 Original commit message from CVS:
5858 * gst-libs/gst/audio/gstbaseaudiosink.c:
5859 (gst_base_audio_sink_setcaps):
5860 Pause the write thread before deactivating and releasing the ringbuffer
5861 to avoid a deadlock when we do gapless playback with different sample
5862 rates in playbin2. Fixes #564929.
5864 2008-12-19 13:03:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5866 gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
5867 Original commit message from CVS:
5868 * gst-libs/gst/audio/gstbaseaudiosrc.c:
5869 Make GstAudioSrcSlaveMethod get_type() function non-static
5871 * win32/common/libgstaudio.def:
5872 * win32/common/libgstnetbuffer.def:
5873 Add some missing functions to the list of exported symbols.
5875 2008-12-18 12:37:33 +0000 Andrew Feren <acferen@yahoo.com>
5877 gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
5878 Original commit message from CVS:
5879 Patch by: Andrew Feren <acferen at yahoo dot com>
5880 * gst-libs/gst/netbuffer/gstnetbuffer.c:
5881 (gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
5882 (gst_netaddress_get_address_bytes),
5883 (gst_netaddress_set_address_bytes):
5884 * gst-libs/gst/netbuffer/gstnetbuffer.h:
5885 Make gst_netaddress_get_ip4_address fail for v6 addresses.
5886 Make gst_netaddress_get_ip6_address either fail or return the v4
5887 address as a transitional v6 address.
5888 Add two convenience functions:
5889 API: gst_netaddress_get_address_bytes()
5890 API: gst_netaddress_set_address_bytes()
5893 2008-12-17 13:51:46 +0000 Wim Taymans <wim.taymans@gmail.com>
5895 Add appsrc and appsink documentation.
5896 Original commit message from CVS:
5897 * docs/plugins/Makefile.am:
5898 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
5899 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
5900 * gst-libs/gst/app/gstappsink.c:
5901 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
5902 Add appsrc and appsink documentation.
5904 2008-12-17 08:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
5906 gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
5907 Original commit message from CVS:
5908 * gst/adder/Makefile.am:
5909 * gst/adder/gstadder.c:
5910 Cleanup variable names to make the adder-loop easier to understand.
5911 Also try to use liboil to spee it up, but ifdef it out as it does not
5912 make any change for me (Intel pentim M (sse,sse2) please try on other
5915 2008-12-16 20:16:17 +0000 Wim Taymans <wim.taymans@gmail.com>
5917 Add minimal docs to make the remaining tcp elements show up.
5918 Original commit message from CVS:
5919 * docs/plugins/Makefile.am:
5920 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
5921 * docs/plugins/gst-plugins-base-plugins-sections.txt:
5922 * gst/tcp/gsttcpclientsink.c:
5923 * gst/tcp/gsttcpclientsrc.c:
5924 * gst/tcp/gsttcpserversrc.c:
5925 Add minimal docs to make the remaining tcp elements show up.
5928 2008-12-15 12:02:26 +0000 Wim Taymans <wim.taymans@gmail.com>
5930 examples/app/: Fix example to unref after emiting the push-buffer action.
5931 Original commit message from CVS:
5932 * examples/app/appsrc-ra.c: (feed_data):
5933 * examples/app/appsrc-seekable.c: (feed_data):
5934 * examples/app/appsrc-stream.c: (read_data):
5935 * examples/app/appsrc-stream2.c: (feed_data):
5936 Fix example to unref after emiting the push-buffer action.
5937 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
5938 (gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
5939 (gst_app_src_push_buffer_action):
5940 Don't take the ref on the buffer in push-buffer action because it's too
5941 awkward for bindings. Fixes #564482.
5943 2008-12-13 19:32:13 +0000 Tim-Philipp Müller <tim@centricular.net>
5945 win32/common/config.h: Update to CVS version.
5946 Original commit message from CVS:
5947 * win32/common/config.h:
5948 Update to CVS version.
5949 * win32/common/config.h.in:
5950 Hardcode path to plugin install helper exe, just like we hardcode
5951 the paths in core. Removes another source of VCS conflicts for
5952 people hacking gst-plugins-base on systems with autotools.
5954 2008-12-13 16:21:12 +0000 Edward Hervey <bilboed@bilboed.com>
5956 m4/Makefile.am: And a couple more .m4 that don't exist anymore with gettext 0.17
5957 Original commit message from CVS:
5959 And a couple more .m4 that don't exist anymore with gettext 0.17
5961 2008-12-13 12:41:56 +0000 Edward Hervey <bilboed@bilboed.com>
5963 m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we...
5964 Original commit message from CVS:
5966 inttypes.m4 hasn't been available since gettext-0.15, and since we now
5967 require gettext >= 0.17 ... we can remove it from the list of files to
5970 2008-12-13 06:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5972 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
5973 Original commit message from CVS:
5974 * gst-libs/gst/audio/gstbaseaudiosink.c:
5975 (gst_base_audio_sink_slave_method_get_type),
5976 (gst_base_audio_sink_class_init):
5977 * gst-libs/gst/audio/gstbaseaudiosink.h:
5978 * gst-libs/gst/audio/gstbaseaudiosrc.c:
5979 (gst_base_audio_src_slave_method_get_type),
5980 (gst_base_audio_src_class_init):
5981 * gst-libs/gst/audio/gstbaseaudiosrc.h:
5982 API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
5983 public API. This is needed for the C++ bindings to be able
5984 to use this base classes. Fixes bug #564200, #564206.
5986 2008-12-12 19:41:28 +0000 Edward Hervey <bilboed@bilboed.com>
5988 gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
5989 Original commit message from CVS:
5990 * gst-libs/gst/cdda/gstcddabasesrc.c:
5991 (gst_cdda_base_src_handle_event):
5992 Remove erroneous gst_buffer_ref().
5993 * tests/check/libs/rtp.c: (GST_START_TEST):
5994 Don't forget to unref the buffer once you're done with it.
5996 2008-12-12 13:06:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
5998 gst/playback/: XRef to GstXOverlay.
5999 Original commit message from CVS:
6000 * gst/playback/gstplaybin.c:
6001 * gst/playback/gstplaybin2.c:
6002 XRef to GstXOverlay.
6004 2008-12-12 10:54:45 +0000 Edward Hervey <bilboed@bilboed.com>
6006 gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
6007 Original commit message from CVS:
6008 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
6009 Free the factory array when finalizing.
6010 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
6011 Use a GstStaticPadTemplate since the src pad caps are fixed.
6013 2008-12-12 07:17:21 +0000 Edward Hervey <bilboed@bilboed.com>
6015 ext/vorbis/vorbisenc.c: Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with pad templates.
6016 Original commit message from CVS:
6017 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_base_init),
6018 (gst_vorbis_enc_init):
6019 Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with
6022 2008-12-12 07:15:22 +0000 Edward Hervey <bilboed@bilboed.com>
6024 gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
6025 Original commit message from CVS:
6026 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
6027 (gst_riff_create_video_template_caps):
6028 Add mapping for VP6 in avi/riff.
6030 2008-12-11 15:49:12 +0000 Edward Hervey <bilboed@bilboed.com>
6032 gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
6033 Original commit message from CVS:
6034 * gst/subparse/samiparse.c: (sami_context_push_state),
6035 (sami_context_pop_state), (start_sami_element), (end_sami_element):
6036 Some versions of libxml seem to be very picky as to strict formatting
6037 of the input and never 'close' the final </body> tag.
6038 In order to fix that bad behaviour, we trigger the flushing of
6039 remaining data on both </body> and </sami>.
6042 2008-12-11 12:32:03 +0000 Guillaume Emont <guillaume@fluendo.com>
6044 gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
6045 Original commit message from CVS:
6046 Patch by: Guillaume Emont <guillaume at fluendo dot com>
6047 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
6048 Add typefinders for MS Word files and OS X .DS_Store files to
6049 prevent them to be recognized as MPEG files. Fixes bug #564098.
6051 2008-12-11 11:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
6053 gst/playback/gstplaysink.c: Add some more debug info.
6054 Original commit message from CVS:
6055 * gst/playback/gstplaysink.c: (gen_audio_chain),
6056 (gst_play_sink_reconfigure):
6057 Add some more debug info.
6058 Fix linking of just an encoded sink.
6059 Handle failure to create a sink chain more gracefully than crashing.
6061 2008-12-11 10:33:48 +0000 Wim Taymans <wim.taymans@gmail.com>
6063 tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
6064 Original commit message from CVS:
6065 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
6066 Pushing 10 buffers is enough to run the test.
6068 2008-12-11 10:28:43 +0000 Wim Taymans <wim.taymans@gmail.com>
6070 tests/examples/seek/seek.c: Hook up the SKIP seek flag.
6071 Original commit message from CVS:
6072 * tests/examples/seek/seek.c: (do_seek), (stop_cb),
6073 (skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
6075 Hook up the SKIP seek flag.
6077 2008-12-10 18:43:32 +0000 Wim Taymans <wim.taymans@gmail.com>
6079 gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
6080 Original commit message from CVS:
6081 * gst/playback/gstplaybin2.c: (pad_added_cb):
6082 Error out with a missing-plugin error when the input-selector was not
6084 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
6087 2008-12-10 17:39:32 +0000 Wim Taymans <wim.taymans@gmail.com>
6089 gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
6090 Original commit message from CVS:
6091 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
6092 (gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
6093 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
6094 (gst_play_sink_send_event), (gst_play_sink_change_state):
6096 Try to set the selected sink to READY before using it. This will allow
6097 for detection of incompatible formats sooner.
6098 Don't cause a fatal error when conversion elements are missing but post
6099 a missing-element message and a warning instead because things might
6100 still link and run fine.
6101 Simplyfy the construction of audio and video sink chains.
6103 2008-12-10 14:55:10 +0000 Wim Taymans <wim.taymans@gmail.com>
6105 ext/ogg/gstoggdemux.c: Use G_DEFINE_TYPE for the OggPad to get some threadsafe type init from glib.
6106 Original commit message from CVS:
6107 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
6108 (gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
6109 Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
6112 2008-12-10 08:19:13 +0000 Luis Menina <liberforce@freeside.fr>
6114 gst/: Include glib.h instead of a specific GLib header. Including single
6115 Original commit message from CVS:
6116 Patch by: Luis Menina <liberforce at freeside dot fr>
6117 * gst-libs/gst/floatcast/floatcast.h:
6118 * gst/typefind/gsttypefindfunctions.c:
6119 Include glib.h instead of a specific GLib header. Including single
6120 GLib headers is deprecated. Fixes bug #563904.
6122 2008-12-09 18:30:10 +0000 Julien Moutte <julien@moutte.net>
6124 gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
6125 Original commit message from CVS:
6126 2008-12-09 Julien Moutte <julien@fluendo.com>
6127 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
6128 Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
6130 2008-12-09 17:21:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6132 gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
6133 Original commit message from CVS:
6134 * gst-libs/gst/riff/riff-read.c:
6135 Fix handling of odd chunks in riff metadata.
6137 2008-12-08 18:44:22 +0000 Wim Taymans <wim.taymans@gmail.com>
6139 gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
6140 Original commit message from CVS:
6141 * gst/volume/gstvolume.c: (gst_volume_class_init),
6142 (volume_before_transform), (volume_transform_ip):
6143 Use new basetransform vmethod to reconfigure the dynamic properties and
6144 any pending volume/mute changes. Fixes #563508.
6146 2008-12-08 18:12:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6148 configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...
6149 Original commit message from CVS:
6151 First check for "theoraenc theoradec" and if that failed check
6152 for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
6153 deprecate the latter. Also linking on Windows fails with just "theora"
6154 and the version check would fail for the release candidates.
6157 2008-12-08 15:25:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6159 gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
6160 Original commit message from CVS:
6161 * gst/playback/gstdecodebin.c:
6162 * gst/playback/gstdecodebin2.c:
6163 Add basic docs to decodebin and link to decodebin from decodebin2.
6165 2008-12-08 12:08:32 +0000 Olivier Crete <tester@tester.ca>
6167 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
6168 Original commit message from CVS:
6169 Patch by: Olivier Crete <tester at tester ca>
6170 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
6171 * gst-libs/gst/rtp/gstrtcpbuffer.h:
6172 Implement gst_rtcp_packet_remove(). Fixes #563174.
6173 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
6174 Add unit test for some RTCP functions.
6176 2008-12-04 20:09:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6178 configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.
6179 Original commit message from CVS:
6181 Apparently AC_CONFIG_MACRO_DIR breaks when using more
6182 than one macro directory, reverting last change.
6184 2008-12-04 19:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6186 configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros.
6187 Original commit message from CVS:
6189 Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
6192 2008-12-03 17:47:44 +0000 Edward Hervey <bilboed@bilboed.com>
6194 sys/: Clear all flags on buffers returned from the image pool.
6195 Original commit message from CVS:
6196 * sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
6197 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
6198 Clear all flags on buffers returned from the image pool.
6201 2008-12-01 19:36:35 +0000 이문형 <iwings@gmail.com>
6203 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...
6204 Original commit message from CVS:
6205 Patch by: 이문형 <iwings at gmail dot com>
6206 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
6207 Don't forget to release the lock again if we bail out because some
6208 pad is flushing or we've reached EOS, otherwise things will lock up
6209 next time _push_buffer() is called (#562802).
6211 2008-11-29 13:31:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6213 Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
6214 Original commit message from CVS:
6215 Patch by: Cygwin Ports maintainer
6216 <yselkowitz at users dot sourceforge dot net>
6219 Require gettext 0.17 because older versions don't mix with libtool
6220 2.2. At build time an older gettext version will still work.
6223 2008-11-28 13:30:36 +0000 Christian Schaller <uraeus@gnome.org>
6226 * gst/speexresample/Makefile.am:
6228 Original commit message from CVS:
6231 2008-11-28 09:44:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6233 Update documentation of speexresample for the new element name.
6234 Original commit message from CVS:
6235 * docs/plugins/gst-plugins-base-plugins.args:
6236 * docs/plugins/gst-plugins-base-plugins.hierarchy:
6237 * docs/plugins/gst-plugins-base-plugins.interfaces:
6238 * docs/plugins/gst-plugins-base-plugins.prerequisites:
6239 * docs/plugins/inspect/plugin-videorate.xml:
6240 * gst/speexresample/gstspeexresample.c:
6241 Update documentation of speexresample for the new element name.
6243 2008-11-28 09:04:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6245 gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
6246 Original commit message from CVS:
6247 * gst/speexresample/README:
6248 Update README with the latest diff between the Speex resampler
6251 2008-11-28 08:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6253 gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
6254 Original commit message from CVS:
6255 * gst/speexresample/gstspeexresample.c: (plugin_init):
6256 Update the debug category from speex_resample to audioresample.
6258 2008-11-27 19:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6260 Remove audioresample files.
6261 Original commit message from CVS:
6262 * gst/audioresample/Makefile.am:
6263 * gst/audioresample/buffer.c:
6264 * gst/audioresample/buffer.h:
6265 * gst/audioresample/debug.c:
6266 * gst/audioresample/debug.h:
6267 * gst/audioresample/functable.c:
6268 * gst/audioresample/functable.h:
6269 * gst/audioresample/gstaudioresample.c:
6270 * gst/audioresample/gstaudioresample.h:
6271 * gst/audioresample/resample.c:
6272 * gst/audioresample/resample.h:
6273 * gst/audioresample/resample_chunk.c:
6274 * gst/audioresample/resample_functable.c:
6275 * gst/audioresample/resample_ref.c:
6276 * tests/check/elements/audioresample.c:
6277 Remove audioresample files.
6279 2008-11-27 17:04:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6281 docs/plugins/inspect/plugin-audioresample.xml: Regenerated for library filename change.
6282 Original commit message from CVS:
6283 * docs/plugins/inspect/plugin-audioresample.xml:
6284 Regenerated for library filename change.
6286 2008-11-27 16:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6288 Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
6289 Original commit message from CVS:
6291 * docs/plugins/Makefile.am:
6292 * docs/plugins/gst-plugins-base-plugins-sections.txt:
6293 * docs/plugins/gst-plugins-base-plugins.args:
6294 * docs/plugins/gst-plugins-base-plugins.hierarchy:
6295 * docs/plugins/gst-plugins-base-plugins.interfaces:
6296 * docs/plugins/gst-plugins-base-plugins.prerequisites:
6297 * docs/plugins/inspect/plugin-adder.xml:
6298 * docs/plugins/inspect/plugin-alsa.xml:
6299 * docs/plugins/inspect/plugin-audioconvert.xml:
6300 * docs/plugins/inspect/plugin-audiorate.xml:
6301 * docs/plugins/inspect/plugin-audioresample.xml:
6302 * docs/plugins/inspect/plugin-audiotestsrc.xml:
6303 * docs/plugins/inspect/plugin-cdparanoia.xml:
6304 * docs/plugins/inspect/plugin-decodebin.xml:
6305 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
6306 * docs/plugins/inspect/plugin-gdp.xml:
6307 * docs/plugins/inspect/plugin-gio.xml:
6308 * docs/plugins/inspect/plugin-gnomevfs.xml:
6309 * docs/plugins/inspect/plugin-libvisual.xml:
6310 * docs/plugins/inspect/plugin-ogg.xml:
6311 * docs/plugins/inspect/plugin-pango.xml:
6312 * docs/plugins/inspect/plugin-playback.xml:
6313 * docs/plugins/inspect/plugin-queue2.xml:
6314 * docs/plugins/inspect/plugin-subparse.xml:
6315 * docs/plugins/inspect/plugin-tcp.xml:
6316 * docs/plugins/inspect/plugin-theora.xml:
6317 * docs/plugins/inspect/plugin-typefindfunctions.xml:
6318 * docs/plugins/inspect/plugin-uridecodebin.xml:
6319 * docs/plugins/inspect/plugin-video4linux.xml:
6320 * docs/plugins/inspect/plugin-videorate.xml:
6321 * docs/plugins/inspect/plugin-videoscale.xml:
6322 * docs/plugins/inspect/plugin-videotestsrc.xml:
6323 * docs/plugins/inspect/plugin-volume.xml:
6324 * docs/plugins/inspect/plugin-vorbis.xml:
6325 * docs/plugins/inspect/plugin-ximagesink.xml:
6326 * docs/plugins/inspect/plugin-xvimagesink.xml:
6327 * gst/speexresample/gstspeexresample.c: (plugin_init):
6328 * gst/speexresample/Makefile.am:
6329 * tests/check/Makefile.am:
6330 * tests/check/elements/speexresample.c: (setup_speexresample),
6331 (GST_START_TEST), (test_pipeline):
6332 Rename the moved speexresample to audioresample, integrate into the
6333 build system and remove the old audioresample from the build system.
6334 Fixes bug #558124, #385061, #346218, #116051.
6336 2008-11-27 16:47:41 +0000 Wim Taymans <wim.taymans@gmail.com>
6338 gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
6339 Original commit message from CVS:
6340 * gst-libs/gst/audio/gstbaseaudiosrc.c:
6341 (gst_base_audio_src_get_offset), (gst_base_audio_src_create):
6342 Avoid nasty int overflows after about 12 hours and 25 minutes when these
6343 code paths are triggered.
6344 A free beer to Håvard Graff for finding this!
6346 2008-11-27 11:16:44 +0000 이문형 <iwings@gmail.com>
6348 gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
6349 Original commit message from CVS:
6350 Patch by: 이문형 <iwings at gmail dot com>
6351 * gst-libs/gst/rtsp/gstrtspconnection.c:
6352 (gst_rtsp_connection_connect):
6353 A successful gst_poll_wait() doesn't always mean successful connect() on
6354 Windows. We should check errors by calling gst_poll_fd_has_error().
6357 2008-11-25 16:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6359 tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.
6360 Original commit message from CVS:
6361 * tests/check/elements/speexresample.c: (test_pipeline):
6362 Make unit test again faster to prevent timeouts with valgrind.
6364 2008-11-25 15:33:30 +0000 Wim Taymans <wim.taymans@gmail.com>
6366 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
6367 Original commit message from CVS:
6368 * gst-libs/gst/rtp/gstrtcpbuffer.c:
6369 Fix typo in the docs.
6371 2008-11-25 15:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
6373 ext/ogg/gstoggdemux.c: If no stream was found before receiving EOS, post an error message.
6374 Original commit message from CVS:
6375 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
6376 If no stream was found before receiving EOS, post an error message.
6379 2008-11-25 15:14:30 +0000 Wim Taymans <wim.taymans@gmail.com>
6381 ext/theora/: Parse segment events.
6382 Original commit message from CVS:
6383 * ext/theora/gsttheoraenc.h:
6384 * ext/theora/theoraenc.c: (gst_theora_enc_init),
6385 (theora_buffer_from_packet), (theora_push_packet),
6386 (theora_enc_sink_event), (theora_enc_is_discontinuous),
6388 Parse segment events.
6389 Pass incomming buffer timestamps to outgoing buffers.
6390 Use the running_time to construct the granulepos.
6393 2008-11-25 11:00:55 +0000 Wim Taymans <wim.taymans@gmail.com>
6395 gst/playback/gstplaybin2.c: Fix buffer-duration property.
6396 Original commit message from CVS:
6397 * gst/playback/gstplaybin2.c: (activate_group):
6398 Fix buffer-duration property.
6400 2008-11-25 10:32:49 +0000 Wim Taymans <wim.taymans@gmail.com>
6402 gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
6403 Original commit message from CVS:
6404 * gst-libs/gst/audio/gstbaseaudiosink.c:
6405 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
6406 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
6407 (gst_base_audio_sink_change_state):
6408 Really fix audiosink drain handling by keeping track of the running_time
6411 2008-11-24 20:25:24 +0000 Michael Smith <msmith@xiph.org>
6413 gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
6414 Original commit message from CVS:
6415 * gst/playback/gstplaybin2.c:
6416 Add notification of current stream. Add ability to configure buffer
6418 * gst/playback/gsturidecodebin.c:
6419 Add ability to configure buffer sizes for streaming mode.
6422 2008-11-24 20:11:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6424 gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
6425 Original commit message from CVS:
6426 * gst-libs/gst/audio/gstbaseaudiosink.c:
6427 Time is already in running_time. Remove base_time handling. Fixes
6428 audiosinks not draining and thus chopping some audio in the end.
6430 2008-11-24 19:18:59 +0000 David Schleef <ds@schleef.org>
6432 ext/ogg/gstoggmux.*: If we're muxing a dirac stream, flush the page after every picture.
6433 Original commit message from CVS:
6434 * ext/ogg/gstoggmux.c:
6435 * ext/ogg/gstoggmux.h:
6436 If we're muxing a dirac stream, flush the page after every picture.
6438 2008-11-24 12:56:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6440 gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
6441 Original commit message from CVS:
6442 * gst-libs/gst/audio/gstbaseaudiosink.c:
6443 Add one log message to check for audio_drained. Sync one log message
6444 with the condition. Send EOS after draining audio in pull mode.
6446 2008-11-24 12:07:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6448 ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr...
6449 Original commit message from CVS:
6450 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
6451 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
6452 Use gst_buffer_try_new_and_alloc() and fail properly if the
6453 allocation failed. This prevents abort() if downstream elements
6454 request an insane amount of memory.
6456 2008-11-24 12:03:11 +0000 Jon Trowbridge <trow@ximian.com>
6458 gst/volume/gstvolume.*: Cleanup volume, define and use default values.
6459 Original commit message from CVS:
6460 * gst/volume/gstvolume.c: (volume_choose_func),
6461 (volume_update_volume), (gst_volume_set_volume),
6462 (gst_volume_get_volume), (gst_volume_set_mute),
6463 (gst_volume_class_init), (gst_volume_init),
6464 (volume_process_double), (volume_process_float),
6465 (volume_process_int32), (volume_process_int32_clamp),
6466 (volume_process_int24), (volume_process_int24_clamp),
6467 (volume_process_int16), (volume_process_int16_clamp),
6468 (volume_process_int8), (volume_process_int8_clamp), (volume_setup),
6469 (volume_transform_ip), (volume_set_property),
6470 (volume_get_property):
6471 * gst/volume/gstvolume.h:
6472 Cleanup volume, define and use default values.
6473 Recalculate new volume and mute setup before processing. Fixes #561789.
6474 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
6475 Add controller unit test. Patch by: Jonathan Matthew
6476 Fix bogus test that messed with basetransform's internal state.
6478 2008-11-22 15:02:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6480 tests/check/elements/speexresample.c: Make the unit test a bit faster to prevent timeouts, especially with valgrind.
6481 Original commit message from CVS:
6482 * tests/check/elements/speexresample.c: (GST_START_TEST):
6483 Make the unit test a bit faster to prevent timeouts, especially
6486 2008-11-22 14:44:26 +0000 Wim Taymans <wim.taymans@gmail.com>
6488 gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
6489 Original commit message from CVS:
6490 * gst/videorate/gstvideorate.c:
6491 Add jpeg and png image media types to the caps. Fixes #561436.
6493 2008-11-22 14:31:43 +0000 Wim Taymans <wim.taymans@gmail.com>
6495 gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ...
6496 Original commit message from CVS:
6497 * gst/playback/gstplaysink.c: (gen_audio_chain):
6498 Don't post an error when we can't configure the volume but post a
6499 warning instead. Fixes #561780.
6501 2008-11-21 20:32:56 +0000 Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
6503 gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video...
6504 Original commit message from CVS:
6505 Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
6506 * gst/videotestsrc/gstvideotestsrc.c:
6507 * gst/videotestsrc/gstvideotestsrc.h:
6508 * gst/videotestsrc/videotestsrc.c:
6509 * gst/videotestsrc/videotestsrc.h:
6510 Add a zone plate pattern generator based on BBC R&D Report
6511 1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate
6512 kx2=20 ky2=20 kt=1'.
6514 2008-11-21 15:45:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6516 gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
6517 Original commit message from CVS:
6518 * gst/speexresample/gstspeexresample.c:
6519 (gst_speex_resample_class_init), (gst_speex_resample_set_property),
6520 (gst_speex_resample_get_property):
6521 Add a "filter-length" property that maps to the quality values
6522 for compatibilty with audioresample.
6524 2008-11-21 00:04:48 +0000 Michael Smith <msmith@xiph.org>
6526 gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
6527 Original commit message from CVS:
6528 * gst/playback/gstdecodebin2.c:
6529 Fix random fat-fingering making this not compile.
6531 2008-11-20 22:11:38 +0000 Michael Smith <msmith@xiph.org>
6533 gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh...
6534 Original commit message from CVS:
6535 * gst/playback/gstdecodebin2.c:
6536 If the top-level type of the stream is plain text, don't try to decode
6537 it, matching behaviour of decodebin.
6538 * gst/playback/gstplaysink.c:
6539 If we fail to generate a text chain (e.g. due to missing optional
6540 plugins), don't crash.
6542 2008-11-20 22:06:05 +0000 Michael Smith <msmith@xiph.org>
6544 gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
6545 Original commit message from CVS:
6546 * gst-libs/gst/rtsp/gstrtspdefs.c:
6547 Fix win32 build. Oops.
6549 2008-11-20 21:40:49 +0000 Michael Smith <msmith@xiph.org>
6551 gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
6552 Original commit message from CVS:
6553 * gst-libs/gst/rtsp/gstrtspdefs.c:
6554 Use WSAGetLastError() rather than errno/h_errno on win32.
6556 2008-11-20 21:20:27 +0000 Michael Smith <msmith@xiph.org>
6558 gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
6559 Original commit message from CVS:
6560 * gst-libs/gst/riff/riff-media.c:
6561 Support WMA Lossless properly.
6563 2008-11-19 00:24:44 +0000 David Schleef <ds@schleef.org>
6565 gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect...
6566 Original commit message from CVS:
6567 * gst/videotestsrc/gstvideotestsrc.c:
6568 * gst/videotestsrc/gstvideotestsrc.h:
6569 * gst/videotestsrc/videotestsrc.c:
6570 * gst/videotestsrc/videotestsrc.h:
6571 Add "colorspec" property, specifying whether to generate BT.601
6572 or BT.709 video. This only affects YCbCr values, not RGB, since
6573 if you're generating a 709 test pattern, presumably you want
6574 709 RGB primaries, not 601. Also add "smpte75" pattern, which
6575 uses 75% colors instead of 100%, since this is often more useful
6576 for testing (and also follows the SMPTE EG-1 guideline).
6578 2008-11-18 18:08:42 +0000 Alessandro Decina <alessandro.d@gmail.com>
6580 gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2.
6581 Original commit message from CVS:
6582 * gst/playback/gstdecodebin.c:
6583 Add a "sink-caps" property to decodebin like it's done for decodebin2.
6586 2008-11-14 21:44:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
6588 gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
6589 Original commit message from CVS:
6590 * gst/audioresample/gstaudioresample.c:
6591 Guard against a NULL dereference I somehow encountered -
6592 with a FLUSH_STOP arriving either before basetransform _start(),
6594 * gst/typefind/gsttypefindfunctions.c:
6595 Make sure we never jump backwards when typefinding corrupt mov files.
6597 2008-11-14 21:39:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
6599 gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
6600 Original commit message from CVS:
6601 * gst-libs/gst/interfaces/propertyprobe.c:
6602 Fix random type causing a docs warning.
6604 2008-11-14 15:40:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6606 sys/v4l/gstv4l.c: Give it a minimal rank for autovideosrc.
6607 Original commit message from CVS:
6609 Give it a minimal rank for autovideosrc.
6611 2008-11-13 21:11:13 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6613 gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types.
6614 Original commit message from CVS:
6615 * gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
6617 Improve typefinding of ISO JPEG2000 mime types.
6619 2008-11-13 18:18:32 +0000 Wim Taymans <wim.taymans@gmail.com>
6621 sys/xvimage/xvimagesink.*: Avoid typechecking when we do trivial casts.
6622 Original commit message from CVS:
6623 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
6624 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps),
6625 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
6626 * sys/xvimage/xvimagesink.h:
6627 Avoid typechecking when we do trivial casts.
6628 Move error handling out of the main program flow.
6629 Sneak in the display-region caps property, not completely correct yet.
6630 Cache the width/height in buffer_alloc instead of parsing it from the
6633 2008-11-13 17:27:37 +0000 Wim Taymans <wim.taymans@gmail.com>
6635 gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
6636 Original commit message from CVS:
6637 * gst/playback/gstplaybin2.c: (deactivate_group):
6638 don't try to unlink the selector sinkpad when we don't have it yet. This
6639 can happen if an error occured before the group was complete.
6641 2008-11-13 15:37:40 +0000 Wim Taymans <wim.taymans@gmail.com>
6643 gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
6644 Original commit message from CVS:
6645 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
6646 (gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
6647 (gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
6648 (gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
6649 (gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
6650 (gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
6651 (gst_rtp_buffer_get_extension_data),
6652 (gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
6653 (gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
6654 (gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
6655 (gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
6656 (gst_rtp_buffer_get_payload_type),
6657 (gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
6658 (gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
6659 (gst_rtp_buffer_set_timestamp),
6660 (gst_rtp_buffer_get_payload_subbuffer),
6661 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
6662 Avoid expensive type checks we already did as part of the
6663 _validate() function that should be called first.
6665 2008-11-11 16:40:50 +0000 Wim Taymans <wim.taymans@gmail.com>
6667 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some cases where a newsegment event was not sent.
6668 Original commit message from CVS:
6669 * gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
6670 (gst_base_rtp_depayload_push_full),
6671 (gst_base_rtp_depayload_set_gst_timestamp):
6672 Fix some cases where a newsegment event was not sent.
6674 2008-11-11 15:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
6676 gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co...
6677 Original commit message from CVS:
6678 * gst/playback/gstplaybin2.c: (activate_group):
6679 Catch state change errors and stop from the uridecodebin elements
6680 instead of trying to continue in vain.
6682 2008-11-10 14:53:45 +0000 Edward Hervey <bilboed@bilboed.com>
6684 gst/: Wim, you're a bad boy. You don't want people to contact you or what?
6685 Original commit message from CVS:
6686 * gst-libs/gst/app/gstappsink.c:
6687 * gst-libs/gst/app/gstappsrc.c:
6688 * gst/h264parse/gsth264parse.c:
6689 Wim, you're a bad boy. You don't want people to contact you or what?
6691 2008-11-10 14:22:09 +0000 Wim Taymans <wim.taymans@gmail.com>
6693 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
6694 Original commit message from CVS:
6695 * gst-libs/gst/audio/gstbaseaudiosink.c:
6696 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
6697 (gst_base_audio_sink_callback):
6698 Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
6699 for the latency to expire, fixes #559567.
6701 2008-11-10 13:55:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
6703 gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
6704 Original commit message from CVS:
6705 * gst/adder/gstadder.c:
6706 Change author string after seeing output of gst-inspector.
6708 2008-11-10 10:33:26 +0000 Wim Taymans <wim.taymans@gmail.com>
6710 gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559...
6711 Original commit message from CVS:
6712 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
6713 Don't try to do crazy things when we only have a text pad without a
6714 video pad. Fixes #559478.
6716 2008-11-07 17:35:46 +0000 Wim Taymans <wim.taymans@gmail.com>
6718 gst-libs/gst/app/gstappsrc.*: Add is-live property.
6719 Original commit message from CVS:
6720 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
6721 (gst_app_src_init), (gst_app_src_set_property),
6722 (gst_app_src_get_property), (gst_app_src_push_buffer):
6723 * gst-libs/gst/app/gstappsrc.h:
6724 Add is-live property.
6727 2008-11-06 12:14:51 +0000 Wim Taymans <wim.taymans@gmail.com>
6729 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...
6730 Original commit message from CVS:
6731 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
6732 Fix case where we don't have a range for the rates or channels as is the
6733 case with truespeech.
6735 2008-11-05 19:18:25 +0000 Wim Taymans <wim.taymans@gmail.com>
6737 gst/volume/gstvolume.*: Keep negotiated state in a separate variable.
6738 Original commit message from CVS:
6739 * gst/volume/gstvolume.c: (volume_update_real_volume),
6740 (gst_volume_set_volume), (gst_volume_get_volume),
6741 (gst_volume_set_mute), (gst_volume_init), (volume_setup),
6742 (volume_transform_ip), (volume_update_mute),
6743 (volume_update_volume), (volume_get_property):
6744 * gst/volume/gstvolume.h:
6745 Keep negotiated state in a separate variable.
6746 Protect the volume and mute properties with the object lock.
6747 Protect modifying the transform with the transform lock.
6749 2008-11-05 12:20:21 +0000 Wim Taymans <wim.taymans@gmail.com>
6751 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Only convert caps to string when debug is enabled.
6752 Original commit message from CVS:
6753 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
6754 (gst_ffmpeg_pixfmt_to_caps):
6755 Only convert caps to string when debug is enabled.
6757 2008-11-04 18:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
6759 ext/theora/: Copy seqnum.
6760 Original commit message from CVS:
6761 * ext/theora/gsttheoradec.h:
6762 * ext/theora/theoradec.c: (gst_theora_dec_init),
6763 (gst_theora_dec_reset), (theora_dec_src_event),
6764 (theora_dec_sink_event), (theora_handle_type_packet):
6766 Keep events in a pending list, like vorbisdec, instead of trying
6767 to construct a segment event ourselves.
6768 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
6769 (vorbis_dec_src_event), (vorbis_dec_sink_event):
6770 * ext/vorbis/vorbisdec.h:
6773 2008-11-04 17:24:35 +0000 Wim Taymans <wim.taymans@gmail.com>
6775 ext/ogg/gstoggdemux.*: Copy seqnums around to track playback segments and messages.
6776 Original commit message from CVS:
6777 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
6778 (gst_ogg_demux_deactivate_current_chain),
6779 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page),
6780 (gst_ogg_demux_loop):
6781 * ext/ogg/gstoggdemux.h:
6782 Copy seqnums around to track playback segments and messages.
6784 2008-11-04 12:42:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6786 Don't install static libs for plugins. Fixes #550851 for -bad.
6787 Original commit message from CVS:
6788 * ext/alsaspdif/Makefile.am:
6789 * ext/amrwb/Makefile.am:
6790 * ext/apexsink/Makefile.am:
6791 * ext/arts/Makefile.am:
6792 * ext/artsd/Makefile.am:
6793 * ext/audiofile/Makefile.am:
6794 * ext/audioresample/Makefile.am:
6795 * ext/bz2/Makefile.am:
6796 * ext/cdaudio/Makefile.am:
6797 * ext/celt/Makefile.am:
6798 * ext/dc1394/Makefile.am:
6799 * ext/dirac/Makefile.am:
6800 * ext/directfb/Makefile.am:
6801 * ext/divx/Makefile.am:
6802 * ext/dts/Makefile.am:
6803 * ext/faac/Makefile.am:
6804 * ext/faad/Makefile.am:
6805 * ext/gsm/Makefile.am:
6806 * ext/hermes/Makefile.am:
6807 * ext/ivorbis/Makefile.am:
6808 * ext/jack/Makefile.am:
6809 * ext/jp2k/Makefile.am:
6810 * ext/ladspa/Makefile.am:
6811 * ext/lcs/Makefile.am:
6812 * ext/libfame/Makefile.am:
6813 * ext/libmms/Makefile.am:
6814 * ext/metadata/Makefile.am:
6815 * ext/mpeg2enc/Makefile.am:
6816 * ext/mplex/Makefile.am:
6817 * ext/musepack/Makefile.am:
6818 * ext/musicbrainz/Makefile.am:
6819 * ext/mythtv/Makefile.am:
6820 * ext/nas/Makefile.am:
6821 * ext/neon/Makefile.am:
6822 * ext/ofa/Makefile.am:
6823 * ext/polyp/Makefile.am:
6824 * ext/resindvd/Makefile.am:
6825 * ext/sdl/Makefile.am:
6826 * ext/shout/Makefile.am:
6827 * ext/snapshot/Makefile.am:
6828 * ext/sndfile/Makefile.am:
6829 * ext/soundtouch/Makefile.am:
6830 * ext/spc/Makefile.am:
6831 * ext/swfdec/Makefile.am:
6832 * ext/tarkin/Makefile.am:
6833 * ext/theora/Makefile.am:
6834 * ext/timidity/Makefile.am:
6835 * ext/twolame/Makefile.am:
6836 * ext/x264/Makefile.am:
6837 * ext/xine/Makefile.am:
6838 * ext/xvid/Makefile.am:
6839 * gst-libs/gst/app/Makefile.am:
6840 * gst-libs/gst/dshow/Makefile.am:
6841 * gst/aiffparse/Makefile.am:
6842 * gst/app/Makefile.am:
6843 * gst/audiobuffer/Makefile.am:
6844 * gst/bayer/Makefile.am:
6845 * gst/cdxaparse/Makefile.am:
6846 * gst/chart/Makefile.am:
6847 * gst/colorspace/Makefile.am:
6848 * gst/dccp/Makefile.am:
6849 * gst/deinterlace/Makefile.am:
6850 * gst/deinterlace2/Makefile.am:
6851 * gst/dvdspu/Makefile.am:
6852 * gst/festival/Makefile.am:
6853 * gst/filter/Makefile.am:
6854 * gst/flacparse/Makefile.am:
6855 * gst/flv/Makefile.am:
6856 * gst/games/Makefile.am:
6857 * gst/h264parse/Makefile.am:
6858 * gst/librfb/Makefile.am:
6859 * gst/mixmatrix/Makefile.am:
6860 * gst/modplug/Makefile.am:
6861 * gst/mpeg1sys/Makefile.am:
6862 * gst/mpeg4videoparse/Makefile.am:
6863 * gst/mpegdemux/Makefile.am:
6864 * gst/mpegtsmux/Makefile.am:
6865 * gst/mpegvideoparse/Makefile.am:
6866 * gst/mve/Makefile.am:
6867 * gst/nsf/Makefile.am:
6868 * gst/nuvdemux/Makefile.am:
6869 * gst/overlay/Makefile.am:
6870 * gst/passthrough/Makefile.am:
6871 * gst/pcapparse/Makefile.am:
6872 * gst/playondemand/Makefile.am:
6873 * gst/rawparse/Makefile.am:
6874 * gst/real/Makefile.am:
6875 * gst/rtjpeg/Makefile.am:
6876 * gst/rtpmanager/Makefile.am:
6877 * gst/scaletempo/Makefile.am:
6878 * gst/sdp/Makefile.am:
6879 * gst/selector/Makefile.am:
6880 * gst/smooth/Makefile.am:
6881 * gst/smoothwave/Makefile.am:
6882 * gst/speed/Makefile.am:
6883 * gst/speexresample/Makefile.am:
6884 * gst/stereo/Makefile.am:
6885 * gst/subenc/Makefile.am:
6886 * gst/tta/Makefile.am:
6887 * gst/vbidec/Makefile.am:
6888 * gst/videodrop/Makefile.am:
6889 * gst/videosignal/Makefile.am:
6890 * gst/virtualdub/Makefile.am:
6891 * gst/vmnc/Makefile.am:
6892 * gst/y4m/Makefile.am:
6893 * sys/acmenc/Makefile.am:
6894 * sys/cdrom/Makefile.am:
6895 * sys/dshowdecwrapper/Makefile.am:
6896 * sys/dshowsrcwrapper/Makefile.am:
6897 * sys/dvb/Makefile.am:
6898 * sys/dxr3/Makefile.am:
6899 * sys/fbdev/Makefile.am:
6900 * sys/oss4/Makefile.am:
6901 * sys/qcam/Makefile.am:
6902 * sys/qtwrapper/Makefile.am:
6903 * sys/vcd/Makefile.am:
6904 * sys/wininet/Makefile.am:
6905 * win32/common/config.h:
6906 Don't install static libs for plugins. Fixes #550851 for -bad.
6908 2008-11-03 15:30:14 +0000 Matthias Kretz <kretz@kde.org>
6910 ext/alsa/gstalsasink.c: Make all access non-blocking so that we can better handle unplugging of usb devices. Fixes #5...
6911 Original commit message from CVS:
6912 Based on patch by: Matthias Kretz <kretz at kde dot org>
6913 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
6914 (gst_alsasink_prepare), (gst_alsasink_unprepare),
6915 (gst_alsasink_write):
6916 Make all access non-blocking so that we can better handle unplugging
6917 of usb devices. Fixes #559111
6919 2008-11-03 10:49:24 +0000 Damien Lespiau <damien.lespiau@gmail.com>
6921 gst-libs/gst/rtsp/gstrtspconnection.c: Make the next call to poll not depend on previous calls to poll with or withou...
6922 Original commit message from CVS:
6923 Patch by: Damien Lespiau <damien.lespiau gmail com>
6924 * gst-libs/gst/rtsp/gstrtspconnection.c:
6925 (gst_rtsp_connection_write):
6926 Make the next call to poll not depend on previous calls to poll with or
6927 without reading from the active descriptor. Fixes #544293.
6929 2008-11-03 08:55:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6931 gst/speexresample/gstspeexresample.c: Add TODO at the top of the file for enabling SSE/ARM specific optimizations and...
6932 Original commit message from CVS:
6933 * gst/speexresample/gstspeexresample.c:
6934 (gst_speex_resample_convert_buffer):
6935 Add TODO at the top of the file for enabling SSE/ARM specific
6936 optimizations and choosing the fastest implementation at runtime.
6937 Add g_assert_not_reached() at two places that should really never
6940 2008-11-02 09:19:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6942 gst/speexresample/gstspeexresample.c: Fix format string and arguments.
6943 Original commit message from CVS:
6944 * gst/speexresample/gstspeexresample.c:
6945 (gst_speex_resample_check_discont):
6946 Fix format string and arguments.
6947 * gst/speexresample/resample_sse.h:
6950 2008-11-01 19:38:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6952 gst/speexresample/: Add missing headers to Makefile.am.
6953 Original commit message from CVS:
6954 * gst/speexresample/Makefile.am:
6955 * gst/speexresample/gstspeexresample.c:
6956 (gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
6957 (gst_speex_resample_convert_buffer), (_benchmark_int_float),
6958 (_benchmark_int_int), (_benchmark_integer_resampling),
6960 * gst/speexresample/gstspeexresample.h:
6961 * gst/speexresample/resample.c:
6962 * gst/speexresample/speex_resampler_double.c:
6963 * gst/speexresample/speex_resampler_float.c:
6964 * gst/speexresample/speex_resampler_int.c:
6965 * gst/speexresample/speex_resampler_wrapper.h:
6966 Add missing headers to Makefile.am.
6967 Update copyright, years and my mail address.
6968 Benchmark the integer resampling implementation against the
6969 float implementation and use the faster one for 8/16 bit integer
6970 input. On most recent systems the floating point version is faster.
6972 2008-10-31 09:49:57 +0000 Nick Haddad <nick@haddads.net>
6974 gst-libs/gst/riff/: Add support for other fourcc codes that are commonly used for 'uncompressed RGB', including 'RGB ...
6975 Original commit message from CVS:
6976 Patch by: Nick Haddad <nick at haddads dot net>
6977 * gst-libs/gst/riff/riff-ids.h:
6978 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
6979 Add support for other fourcc codes that are commonly used for
6980 'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
6983 2008-10-30 14:55:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6985 gst/speexresample/gstspeexresample.c: The length for the buffer conversion function is the number of audio frames, i....
6986 Original commit message from CVS:
6987 * gst/speexresample/gstspeexresample.c:
6988 (gst_speex_resample_convert_buffer):
6989 The length for the buffer conversion function is the number of
6990 audio frames, i.e. we need to multiply it by the number of channels
6991 to get the number of values. Also spotted by the unit test after
6992 running in valgrind.
6994 2008-10-30 14:46:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6996 tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
6997 Original commit message from CVS:
6998 * tests/check/elements/speexresample.c: (element_message_cb),
6999 (eos_message_cb), (test_pipeline), (GST_START_TEST),
7000 (speexresample_suite):
7001 Add pipeline unit tests for testing all supported formats with
7002 up/downsampling and different in/outrates.
7003 * gst/speexresample/gstspeexresample.c:
7004 (gst_speex_resample_push_drain), (gst_speex_resample_process):
7005 * gst/speexresample/speex_resampler_wrapper.h:
7006 Fix bugs identified by the testsuite.
7008 2008-10-30 13:44:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7010 gst/speexresample/: Add support for int8, int24 and int32 input by converting internally to/from int16 or double.
7011 Original commit message from CVS:
7012 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
7013 (gst_speex_resample_get_funcs),
7014 (gst_speex_resample_transform_size),
7015 (gst_speex_resample_convert_buffer),
7016 (gst_speex_resample_push_drain), (gst_speex_resample_process):
7017 * gst/speexresample/gstspeexresample.h:
7018 * gst/speexresample/speex_resampler_wrapper.h:
7019 Add support for int8, int24 and int32 input by converting internally
7020 to/from int16 or double.
7022 2008-10-30 12:43:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7024 Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
7025 Original commit message from CVS:
7026 * gst/speexresample/Makefile.am:
7027 * gst/speexresample/arch.h:
7028 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
7029 (gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
7030 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
7031 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
7032 (_gcd), (gst_speex_resample_transform_size),
7033 (gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
7034 (gst_speex_resample_process), (gst_speex_resample_transform),
7035 (gst_speex_resample_query), (gst_speex_resample_set_property):
7036 * gst/speexresample/gstspeexresample.h:
7037 * gst/speexresample/resample.c:
7038 * gst/speexresample/speex_resampler.h:
7039 * gst/speexresample/speex_resampler_double.c:
7040 * gst/speexresample/speex_resampler_wrapper.h:
7041 * tests/check/elements/speexresample.c: (setup_speexresample),
7042 (test_perfect_stream_instance), (GST_START_TEST),
7043 (test_discont_stream_instance):
7044 Add support for double samples as input and refactor the usage
7045 of the different compilation flavors of the speex resampler.
7047 2008-10-30 11:43:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7049 gst/audioresample/gstaudioresample.c: Return the result of parent_class->event().
7050 Original commit message from CVS:
7051 * gst/audioresample/gstaudioresample.c:
7052 Return the result of parent_class->event().
7054 2008-10-29 17:02:55 +0000 Wim Taymans <wim.taymans@gmail.com>
7056 gst-libs/gst/app/gstappsink.c: Fix the docs.
7057 Original commit message from CVS:
7058 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
7061 2008-10-29 12:11:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7063 gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
7064 Original commit message from CVS:
7065 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
7066 (gst_speex_resample_get_unit_size),
7067 (gst_speex_resample_push_drain), (gst_speex_resample_event),
7068 (gst_speex_resample_check_discont), (gst_speex_resample_process),
7069 (gst_speex_resample_transform):
7070 * gst/speexresample/gstspeexresample.h:
7071 Rewrite timestamp tracking to make it more robust and guarantee
7073 * tests/check/Makefile.am:
7074 * tests/check/elements/speexresample.c: (setup_speexresample),
7075 (cleanup_speexresample), (fail_unless_perfect_stream),
7076 (test_perfect_stream_instance), (GST_START_TEST),
7077 (test_discont_stream_instance), (live_switch_alloc_only_48000),
7078 (live_switch_get_sink_caps), (live_switch_push),
7079 (speexresample_suite):
7080 Add unit tests for speexresample based on the audioresample unit tests.
7082 2008-10-28 19:30:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7084 gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
7085 Original commit message from CVS:
7086 * gst/speexresample/gstspeexresample.c:
7087 (gst_speex_resample_get_unit_size),
7088 (gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
7089 (gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
7090 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
7091 (gst_speex_resample_push_drain), (gst_speex_resample_event),
7092 (gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
7093 (gst_speex_resample_process), (gst_speex_resample_transform),
7094 (gst_speex_resample_query), (gst_speex_resample_set_property):
7095 * gst/speexresample/gstspeexresample.h:
7096 Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
7097 instead of GST_DEBUG, ...
7099 2008-10-28 16:28:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7101 gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
7102 Original commit message from CVS:
7103 * gst/speexresample/gstspeexresample.c:
7104 (gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
7105 (gst_speex_resample_process):
7106 Fixate to the nearest supported rate instead of the first one.
7108 2008-10-28 16:25:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7110 gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
7111 Original commit message from CVS:
7112 * gst/audioresample/gstaudioresample.c:
7113 (gst_audioresample_class_init), (audioresample_fixate_caps):
7114 Fixate the rate to the nearest supported rate instead of
7115 the first one. Fixes bug #549510.
7117 2008-10-28 11:46:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7119 gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
7120 Original commit message from CVS:
7121 * gst/speexresample/README:
7122 * gst/speexresample/arch.h:
7123 * gst/speexresample/fixed_arm4.h:
7124 * gst/speexresample/fixed_arm5e.h:
7125 * gst/speexresample/fixed_bfin.h:
7126 * gst/speexresample/fixed_debug.h:
7127 * gst/speexresample/fixed_generic.h:
7128 * gst/speexresample/resample.c: (compute_func), (main), (sinc),
7129 (cubic_coef), (resampler_basic_direct_single),
7130 (resampler_basic_direct_double),
7131 (resampler_basic_interpolate_single),
7132 (resampler_basic_interpolate_double), (update_filter),
7133 (speex_resampler_init_frac), (speex_resampler_process_native),
7134 (speex_resampler_magic), (speex_resampler_process_float),
7135 (speex_resampler_process_int),
7136 (speex_resampler_process_interleaved_float),
7137 (speex_resampler_process_interleaved_int),
7138 (speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
7139 (speex_resampler_reset_mem):
7140 * gst/speexresample/speex_resampler.h:
7141 Update Speex resampler with latest version from Speex GIT.
7143 2008-10-27 14:57:34 +0000 Wim Taymans <wim.taymans@gmail.com>
7145 win32/common/libgstaudio.def: Add new symbols.
7146 Original commit message from CVS:
7147 * win32/common/libgstaudio.def:
7150 2008-10-23 09:57:06 +0000 Wim Taymans <wim.taymans@gmail.com>
7152 ext/vorbis/vorbisdec.c: Attempt to make obfuscated code clearer.
7153 Original commit message from CVS:
7154 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
7155 Attempt to make obfuscated code clearer.
7157 2008-10-23 07:11:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7159 Move float endianness conversion macros to core. Second part of bug ##555196.
7160 Original commit message from CVS:
7161 * docs/libs/gst-plugins-base-libs-sections.txt:
7162 * gst-libs/gst/floatcast/floatcast.h:
7163 Move float endianness conversion macros to core. Second part of
7166 2008-10-22 12:29:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7168 sys/: Don't mark as gtk-doc docs as they aren't public.
7169 Original commit message from CVS:
7170 * sys/ximage/ximagesink.h:
7171 * sys/xvimage/xvimagesink.h:
7172 Don't mark as gtk-doc docs as they aren't public.
7174 2008-10-22 12:25:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7176 Allow setting colorkey if possible. Implement property probe interface for optional X features (autopaint-colorkey, d...
7177 Original commit message from CVS:
7178 * sys/xvimage/xvimagesink.c:
7179 * sys/xvimage/xvimagesink.h:
7180 * tests/icles/Makefile.am:
7181 * tests/icles/test-colorkey.c:
7182 Allow setting colorkey if possible. Implement property probe interface
7183 for optional X features (autopaint-colorkey, double-buffer and
7184 colorkey). Fixes #554533
7186 2008-10-22 12:01:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7188 gst-libs/gst/tag/tags.c: Remove useless buffer size assignment. It already has this value.
7189 Original commit message from CVS:
7190 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
7191 Remove useless buffer size assignment. It already has this value.
7193 2008-10-20 15:35:37 +0000 Wim Taymans <wim.taymans@gmail.com>
7195 gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
7196 Original commit message from CVS:
7197 * gst-libs/gst/audio/gstaudiosink.c:
7198 (gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
7199 (gst_audioringbuffer_activate), (gst_audioringbuffer_release),
7200 (gst_audioringbuffer_stop):
7201 Implement a separate activate functions to start monitoring the segments
7202 or, in pull mode, pulling in data.
7203 * gst-libs/gst/audio/gstbaseaudiosink.c:
7204 (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
7205 (gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
7206 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
7207 (gst_base_audio_sink_activate_pull),
7208 (gst_base_audio_sink_async_play),
7209 (gst_base_audio_sink_change_state):
7210 Implement pad and element convert query function.
7211 Activate the ringbuffer.
7212 Use the segment last_stop value as the offset to pull.
7213 Use new basesink _do_preroll() method to preroll in the pulling thread.
7214 Take appropriate locking in the pulling thread.
7215 * gst-libs/gst/audio/gstringbuffer.h:
7218 2008-10-20 14:08:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7220 gst/typefind/gsttypefindfunctions.c: Improve MXF typefinding a bit by searching for a header partition pack instead o...
7221 Original commit message from CVS:
7222 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
7223 Improve MXF typefinding a bit by searching for a header partition
7224 pack instead of just a general partition pack and checking more
7225 bytes for valid values.
7227 2008-10-20 13:45:55 +0000 Wim Taymans <wim.taymans@gmail.com>
7229 tests/icles/.cvsignore: update ignore file.
7230 Original commit message from CVS:
7231 * tests/icles/.cvsignore:
7233 * tests/icles/Makefile.am:
7234 * tests/icles/test-box.c: (make_pipeline), (main):
7235 Add another interactive command line experimentation suite for
7236 dynamically boxing/cropping/saling an input video.
7238 2008-10-17 13:19:05 +0000 Wim Taymans <wim.taymans@gmail.com>
7240 Add methods to more accuratly control the pulling thread of a ringbuffer.
7241 Original commit message from CVS:
7242 * docs/libs/gst-plugins-base-libs-sections.txt:
7243 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
7244 (gst_ring_buffer_activate), (gst_ring_buffer_is_active):
7245 * gst-libs/gst/audio/gstringbuffer.h:
7246 Add methods to more accuratly control the pulling thread of a
7248 Add format conversion helper code to the ringbuffer.
7249 API: GstRingBuffer:gst_ring_buffer_activate()
7250 API: GstRingBuffer:gst_ring_buffer_is_active()
7251 API: GstRingBuffer:gst_ring_buffer_convert()
7253 2008-10-16 15:44:37 +0000 Wim Taymans <wim.taymans@gmail.com>
7255 gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
7256 Original commit message from CVS:
7257 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
7258 (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
7259 (gst_audioringbuffer_stop):
7260 Signal thread startup earlier so that we can immediatly go into pull
7261 mode when we have to and block on preroll.
7263 2008-10-16 15:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
7265 gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
7266 Original commit message from CVS:
7267 * gst-libs/gst/audio/gstringbuffer.c:
7268 (gst_ring_buffer_prepare_read):
7269 In pull mode we want the callback to prepull a buffer we can preroll on
7270 even when we are not yet playing.
7272 2008-10-16 15:07:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7274 Don't install static libs for plugins. Fixes #550851 for base.
7275 Original commit message from CVS:
7276 * ext/alsa/Makefile.am:
7277 * ext/cdparanoia/Makefile.am:
7278 * ext/gio/Makefile.am:
7279 * ext/gnomevfs/Makefile.am:
7280 * ext/libvisual/Makefile.am:
7281 * ext/ogg/Makefile.am:
7282 * ext/pango/Makefile.am:
7283 * ext/theora/Makefile.am:
7284 * ext/vorbis/Makefile.am:
7285 * gst/adder/Makefile.am:
7286 * gst/audioconvert/Makefile.am:
7287 * gst/audiorate/Makefile.am:
7288 * gst/audioresample/Makefile.am:
7289 * gst/audiotestsrc/Makefile.am:
7290 * gst/ffmpegcolorspace/Makefile.am:
7291 * gst/gdp/Makefile.am:
7292 * gst/playback/Makefile.am:
7293 * gst/subparse/Makefile.am:
7294 * gst/tcp/Makefile.am:
7295 * gst/typefind/Makefile.am:
7296 * gst/videorate/Makefile.am:
7297 * gst/videoscale/Makefile.am:
7298 * gst/videotestsrc/Makefile.am:
7299 * gst/volume/Makefile.am:
7300 * sys/v4l/Makefile.am:
7301 * sys/ximage/Makefile.am:
7302 * sys/xvimage/Makefile.am:
7303 Don't install static libs for plugins. Fixes #550851 for base.
7305 2008-10-16 13:50:00 +0000 Wim Taymans <wim.taymans@gmail.com>
7307 gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe...
7308 Original commit message from CVS:
7309 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
7310 Set the default blocksize to -1 because we will then use the configured
7311 samplesperbuffer to create our output buffer.
7313 2008-10-15 15:28:41 +0000 Edward Hervey <bilboed@bilboed.com>
7315 gst-libs/gst/riff/riff-media.c: Add mappping for the KMVC (Karl Morton's Video) Codec.
7316 Original commit message from CVS:
7317 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
7318 (gst_riff_create_video_template_caps):
7319 Add mappping for the KMVC (Karl Morton's Video) Codec.
7321 2008-10-15 14:25:50 +0000 Edward Hervey <bilboed@bilboed.com>
7323 gst/typefind/gsttypefindfunctions.c: Don't forget to advance the offset of what we're matching against, else we end u...
7324 Original commit message from CVS:
7325 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
7326 Don't forget to advance the offset of what we're matching against, else
7327 we end up in a forever loop.
7329 2008-10-15 11:25:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7331 gst/subparse/gstsubparse.c: Improve typefinding a bit. If we don't have a Unicode charset try GST_SUBTITLE_ENCODING a...
7332 Original commit message from CVS:
7333 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
7334 Improve typefinding a bit. If we don't have a Unicode charset
7335 try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
7337 2008-10-14 11:13:59 +0000 Edward Hervey <bilboed@bilboed.com>
7339 ext/theora/theoradec.c: Fix build on macosx.
7340 Original commit message from CVS:
7341 * ext/theora/theoradec.c: (theora_dec_decode_buffer):
7342 Fix build on macosx.
7344 2008-10-13 11:36:13 +0000 Robin Stocker <robin@nibor.org>
7346 ext/theora/: Parse input caps and make the PAR override the encoded PAR when specified by a container. Fixes #555699.
7347 Original commit message from CVS:
7348 Based on patch by: Robin Stocker <robin at nibor dot org>
7349 * ext/theora/gsttheoradec.h:
7350 * ext/theora/theoradec.c: (gst_theora_dec_init),
7351 (theora_dec_setcaps), (theora_handle_type_packet),
7352 (theora_dec_decode_buffer), (theora_dec_change_state):
7353 Parse input caps and make the PAR override the encoded PAR when
7354 specified by a container. Fixes #555699.
7356 2008-10-13 09:16:59 +0000 Wim Taymans <wim.taymans@gmail.com>
7358 gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY
7359 Original commit message from CVS:
7360 * gst-libs/gst/rtp/gstbasertpdepayload.c:
7361 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
7362 (gst_base_rtp_depayload_set_gst_timestamp),
7363 (gst_base_rtp_depayload_change_state):
7364 * gst-libs/gst/rtp/gstbasertpdepayload.h:
7365 Add some more G_LIKELY
7366 Fail when the setcaps function was not called.
7367 * gst-libs/gst/rtp/gstbasertppayload.c:
7368 (gst_basertppayload_set_outcaps):
7369 Propagate return value of setcaps.
7371 2008-10-13 08:58:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7373 gst/subparse/: Add support for UTF16/UTF32 subtitles as long as the first bytes of the first buffer contain the BOM. ...
7374 Original commit message from CVS:
7375 * gst/subparse/Makefile.am:
7376 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
7377 (gst_sub_parse_class_init), (gst_sub_parse_init),
7378 (gst_convert_to_utf8), (detect_encoding), (convert_encoding),
7379 (get_next_line), (gst_sub_parse_data_format_autodetect),
7380 (feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
7381 (gst_subparse_type_find):
7382 * gst/subparse/gstsubparse.h:
7383 Add support for UTF16/UTF32 subtitles as long as the first bytes of
7384 the first buffer contain the BOM. This also adds support for other
7385 encodings that allow NUL bytes via the encoding property.
7386 Fixes bugs #552237 and #456788.
7388 2008-10-13 08:15:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7390 gst-libs/gst/tag/tags.c: Don't drop the last byte of image tags if they're not an URI list.
7391 Original commit message from CVS:
7392 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
7393 Don't drop the last byte of image tags if they're not an URI list.
7396 2008-10-13 08:00:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7398 gst/typefind/gsttypefindfunctions.c: For looking at the 4th byte we have to get 4 bytes of course and not 3.
7399 Original commit message from CVS:
7400 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
7401 For looking at the 4th byte we have to get 4 bytes of course
7404 2008-10-13 07:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7406 gst/typefind/gsttypefindfunctions.c: Improve FLAC-without-headers typefinding by looking at most of the frame header ...
7407 Original commit message from CVS:
7408 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
7409 Improve FLAC-without-headers typefinding by looking at most of the
7410 frame header and checking if invalid values are used. Should prevent
7411 quite some false positives compared to the old version which only
7412 check if the first 14 bits are set.
7414 2008-10-11 16:27:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7416 sys/xvimage/xvimagesink.c: Don't assert on caps==NULL.
7417 Original commit message from CVS:
7418 * sys/xvimage/xvimagesink.c:
7419 Don't assert on caps==NULL.
7421 2008-10-10 17:13:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7423 Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
7424 Original commit message from CVS:
7425 * gst/subparse/gstsubparse.c:
7426 (gst_sub_parse_data_format_autodetect), (handle_buffer),
7427 (gst_sub_parse_change_state):
7428 * gst/subparse/gstsubparse.h:
7429 * tests/check/elements/subparse.c: (GST_START_TEST):
7430 Add support for subtitle files with UTF-8 BOM at the beginning
7431 by simple stripping it from the first line before passing it
7432 to any parsing code. Fixes bug #555257 and playback of files
7433 created by Gnome Subtitles.
7435 2008-10-10 15:45:15 +0000 Wim Taymans <wim.taymans@gmail.com>
7437 gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place.
7438 Original commit message from CVS:
7439 * gst/audiotestsrc/gstaudiotestsrc.c:
7440 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
7441 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
7442 (gst_audio_test_src_start), (gst_audio_test_src_stop),
7443 (gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
7444 (gst_audio_test_src_create):
7445 * gst/audiotestsrc/gstaudiotestsrc.h:
7446 Define the default property values in the usual place.
7447 Implement start/stop to reset values correctly.
7448 Calculate the sample size only once when we negotiate.
7449 Rename some values to make more sense.
7450 Keep track of our byte range.
7451 Add support for pull based scheduling. Disabled for now until we have
7452 the whole stack working.
7453 Set the BUFFER_OFFSET correctly.
7455 2008-10-10 15:32:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7457 Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
7458 Original commit message from CVS:
7459 Based on a patch by: xavierb at gmail dot com
7460 * gst/subparse/gstsubparse.c:
7461 (gst_sub_parse_data_format_autodetect):
7462 * tests/check/elements/subparse.c: (GST_START_TEST):
7463 Make the detection of the used subtitle a bit less strict
7464 for srt subtitles. Fixes bug #555607.
7466 2008-10-10 15:21:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7468 ext/vorbis/vorbisenc.c: Fix discontinuity detection which was broken by last commit.
7469 Original commit message from CVS:
7470 * ext/vorbis/vorbisenc.c:
7471 (gst_vorbis_enc_buffer_check_discontinuous):
7472 Fix discontinuity detection which was broken by last commit.
7474 2008-10-09 11:18:09 +0000 Tim-Philipp Müller <tim@centricular.net>
7476 configure.ac: Require core CVS for ghostpad API additions used by decodebin2.
7477 Original commit message from CVS:
7479 Require core CVS for ghostpad API additions used by decodebin2.
7481 2008-10-08 15:30:33 +0000 Edward Hervey <bilboed@bilboed.com>
7483 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
7484 Original commit message from CVS:
7485 * gst-libs/gst/audio/gstbaseaudiosrc.c:
7486 (gst_base_audio_src_create):
7487 Fix debug statements (space between '%' and actual format).
7489 2008-10-08 14:44:04 +0000 Wim Taymans <wim.taymans@gmail.com>
7491 gst/playback/gstdecodebin2.c: Remove bogus assert, the decodepad could have been created inside an already existing g...
7492 Original commit message from CVS:
7493 * gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
7494 Remove bogus assert, the decodepad could have been created inside an
7495 already existing group.
7497 2008-10-08 14:01:42 +0000 Andy Wingo <wingo@pobox.com>
7501 Original commit message from CVS:
7504 2008-10-08 14:00:07 +0000 Andy Wingo <wingo@pobox.com>
7506 gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
7507 Original commit message from CVS:
7508 2008-10-08 Andy Wingo <wingo@pobox.com>
7509 * gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
7510 target instead of setting it.
7511 (gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
7512 API for a decode pad. The bugfix is that we set the group in
7513 activate(), not when the pad was created because it might be NULL
7515 (gst_decode_group_control_source_pad, gst_decode_group_expose):
7516 Update to use the API.
7518 2008-10-08 12:49:40 +0000 Andy Wingo <wingo@pobox.com>
7520 gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
7521 Original commit message from CVS:
7522 2008-10-08 Andy Wingo <wingo@pobox.com>
7523 * gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
7524 be a subclass of GstGhostPad.
7525 (analyze_new_pad): So, when emitting the signals that determine
7526 how we do autoplugging, already create the ghost pad and use it as
7527 the pad in the signal arguments. This allows applications to make
7528 a connection between the pad passed in e.g. autoplug-continue, and
7529 the pad passed in new-decoded-pad.
7530 (connect_pad, expose_pad): Update to receive the ghosted decode
7531 pad in the args, retargetting it as necessary if we have to plug
7532 the target pad through a multiqueue.
7533 (gst_decode_group_control_source_pad): Adapt to receive an
7534 already-ghosted pad that just needs activation, blocking, and
7536 (sort_end_pads): Adapt for decode pads actually being pads.
7537 (gst_decode_group_expose): Adapt for decode pads actually being
7538 pads. Rewrite the decode pad names so they appear in order. Adds a
7539 new error case if we couldn't set the name.
7540 (gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
7542 (gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
7543 New API for the decode pad, needed because we shouldn't do these
7544 things inside gst_decode_pad_new(), but after.
7545 (gst_decode_pad_new): Change to actually make the real pad, and
7546 delay the blocking/drainage bits.
7548 2008-10-08 12:12:01 +0000 Daniel Drake <dsd@laptop.org>
7550 ext/ogg/gstoggmux.c: Unref all buffers when clearing collectpads. Fixes bug #546955.
7551 Original commit message from CVS:
7552 Patch by: Daniel Drake <dsd at laptop dot org>
7553 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
7554 Unref all buffers when clearing collectpads. Fixes bug #546955.
7556 2008-10-08 12:08:01 +0000 Klaas <klaas@rivercrew.net>
7558 ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
7559 Original commit message from CVS:
7560 Based on a patch by: Klaas <klaas at rivercrew dot net>
7561 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
7562 (gst_vorbis_enc_buffer_check_discontinuous),
7563 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
7564 * ext/vorbis/vorbisenc.h:
7565 Keep track of the upstream segments and use the running time on that
7566 segment instead of the buffer timestamp everywhere. Fixes bug #525807.
7568 2008-10-08 11:50:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7570 gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
7571 Original commit message from CVS:
7572 * gst/audioconvert/audioconvert.c: (audio_convert_convert):
7573 Prevent overflows with big buffer when calculating the size of
7574 the intermediate buffer by using gst_util_uint64_scale() instead of
7575 plain arithmetics. Fixes bug #552801.
7577 2008-10-08 10:49:15 +0000 Pavel Zeldin <pzeldin@gmail.com>
7579 ext/pango/gstclockoverlay.*: API: Add ability to specify format for date/time display by adding a "time-format" prope...
7580 Original commit message from CVS:
7581 Patch by: Pavel Zeldin <pzeldin at gmail dot com>
7582 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
7583 (gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
7584 (gst_clock_overlay_init), (gst_clock_overlay_set_property),
7585 (gst_clock_overlay_get_property):
7586 * ext/pango/gstclockoverlay.h:
7587 API: Add ability to specify format for date/time display by
7588 adding a "time-format" property.
7591 2008-10-08 09:22:26 +0000 Jan Gerber <j@oil21.org>
7593 gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
7594 Original commit message from CVS:
7595 Patch by: Jan Gerber <j at oil21 dot org>
7596 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
7597 (gst_riff_create_video_template_caps):
7598 Add FFV1 fourcc to support playback of FFMPEG lossless video
7599 in AVI. Fixes bug #555319.
7601 2008-10-08 09:12:36 +0000 Håvard Graff <havard.graff@tandberg.com>
7603 gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
7604 Original commit message from CVS:
7605 Patch by: Håvard Graff <havard dot graff at tandberg dot com>
7606 * gst-libs/gst/audio/gstbaseaudiosrc.c:
7607 (gst_base_audio_src_create):
7608 Implement skew clock slaving. Fixes #552559.
7610 2008-10-08 09:10:23 +0000 Wim Taymans <wim.taymans@gmail.com>
7612 gst-libs/gst/audio/: Fix include of config.h
7613 Original commit message from CVS:
7614 * gst-libs/gst/audio/multichannel.c:
7615 * gst-libs/gst/audio/testchannels.c:
7616 Fix include of config.h
7618 2008-10-06 16:36:20 +0000 Tero Saarni <tero.saarni@gmail.com>
7620 gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
7621 Original commit message from CVS:
7622 Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
7623 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
7624 (print_media), (gst_sdp_message_dump):
7625 Fix parsing of the c= field containing multicast addresses.
7627 Add the connection info to the session or streams.
7628 Fix parsing of the bandwidth.
7629 Add debugging for the connections and bandwidths for a media.
7630 Add debugging for the bandwidth of the session.
7632 2008-10-06 16:31:27 +0000 Wim Taymans <wim.taymans@gmail.com>
7634 gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
7635 Original commit message from CVS:
7636 * gst-libs/gst/rtp/gstbasertppayload.c:
7637 (gst_basertppayload_change_state):
7638 Configure the next seqnum and timestamp in the state change so that they
7639 can be queried soon after.
7641 2008-10-06 16:29:33 +0000 Wim Taymans <wim.taymans@gmail.com>
7643 gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
7644 Original commit message from CVS:
7645 * gst-libs/gst/rtp/gstbasertpdepayload.c:
7646 (gst_base_rtp_depayload_chain):
7647 Improve debugging of the rtptime.
7649 2008-10-05 11:33:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7651 configure.ac: Back to development -> 0.10.21.1
7652 Original commit message from CVS:
7654 Back to development -> 0.10.21.1
7656 2008-10-05 08:18:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7660 Original commit message from CVS:
7663 2008-10-05 08:11:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7665 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
7666 Original commit message from CVS:
7667 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
7669 Add typefinder for MXF.
7671 2008-10-05 08:10:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7673 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
7674 Original commit message from CVS:
7675 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
7677 Add typefinder for MXF.
7679 2008-10-03 15:19:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7681 tests/icles/Makefile.am: Only build test-colorkey if GTK+ is available.
7682 Original commit message from CVS:
7683 * tests/icles/Makefile.am:
7684 Only build test-colorkey if GTK+ is available.
7686 === release 0.10.21 ===
7688 2008-10-03 00:03:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7694 * docs/plugins/gst-plugins-base-plugins.args:
7695 * docs/plugins/gst-plugins-base-plugins.hierarchy:
7696 * docs/plugins/gst-plugins-base-plugins.interfaces:
7697 * docs/plugins/gst-plugins-base-plugins.prerequisites:
7698 * docs/plugins/inspect/plugin-adder.xml:
7699 * docs/plugins/inspect/plugin-alsa.xml:
7700 * docs/plugins/inspect/plugin-audioconvert.xml:
7701 * docs/plugins/inspect/plugin-audiorate.xml:
7702 * docs/plugins/inspect/plugin-audioresample.xml:
7703 * docs/plugins/inspect/plugin-audiotestsrc.xml:
7704 * docs/plugins/inspect/plugin-cdparanoia.xml:
7705 * docs/plugins/inspect/plugin-decodebin.xml:
7706 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
7707 * docs/plugins/inspect/plugin-gdp.xml:
7708 * docs/plugins/inspect/plugin-gio.xml:
7709 * docs/plugins/inspect/plugin-gnomevfs.xml:
7710 * docs/plugins/inspect/plugin-libvisual.xml:
7711 * docs/plugins/inspect/plugin-ogg.xml:
7712 * docs/plugins/inspect/plugin-pango.xml:
7713 * docs/plugins/inspect/plugin-playback.xml:
7714 * docs/plugins/inspect/plugin-queue2.xml:
7715 * docs/plugins/inspect/plugin-subparse.xml:
7716 * docs/plugins/inspect/plugin-tcp.xml:
7717 * docs/plugins/inspect/plugin-theora.xml:
7718 * docs/plugins/inspect/plugin-typefindfunctions.xml:
7719 * docs/plugins/inspect/plugin-uridecodebin.xml:
7720 * docs/plugins/inspect/plugin-video4linux.xml:
7721 * docs/plugins/inspect/plugin-videorate.xml:
7722 * docs/plugins/inspect/plugin-videoscale.xml:
7723 * docs/plugins/inspect/plugin-videotestsrc.xml:
7724 * docs/plugins/inspect/plugin-volume.xml:
7725 * docs/plugins/inspect/plugin-vorbis.xml:
7726 * docs/plugins/inspect/plugin-ximagesink.xml:
7727 * docs/plugins/inspect/plugin-xvimagesink.xml:
7728 * gst-plugins-base.doap:
7729 * win32/common/config.h:
7731 Original commit message from CVS:
7734 2008-10-02 23:44:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7765 Original commit message from CVS:
7768 2008-09-28 22:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7770 configure.ac: 0.10.20.4 pre-release
7771 Original commit message from CVS:
7773 0.10.20.4 pre-release
7775 2008-09-25 10:46:00 +0000 ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>
7777 ext/theora/theoraparse.c: Set the BOS flag on the BOS packet. Fixes #553244.
7778 Original commit message from CVS:
7779 Patch by: ogg.k.ogg.k <ogg dot k dot ogg dot k at googlemail dot com>
7780 * ext/theora/theoraparse.c: (theora_parse_set_streamheader):
7781 Set the BOS flag on the BOS packet. Fixes #553244.
7783 2008-09-23 17:48:14 +0000 Wim Taymans <wim.taymans@gmail.com>
7785 gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
7786 Original commit message from CVS:
7787 * gst-libs/gst/rtsp/gstrtspmessage.c:
7788 (gst_rtsp_message_parse_request),
7789 (gst_rtsp_message_parse_response):
7790 Fix the g_return_val_if_fail() statements.
7792 2008-09-22 17:44:14 +0000 Michael Smith <msmith@xiph.org>
7794 gst-libs/gst/tag/gsttagdemux.c: Fail to activate if there's insufficient data in the file to be usable, preventing an...
7795 Original commit message from CVS:
7796 * gst-libs/gst/tag/gsttagdemux.c:
7797 Fail to activate if there's insufficient data in the file to be usable,
7798 preventing an assertion fail later. Fixes #552960
7800 2008-09-16 15:36:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7802 Commit stuff that should have gone in last week when I made the pre-releases:
7803 Original commit message from CVS:
7804 Commit stuff that should have gone in last week when I made the pre-releases:
7805 2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
7807 0.10.20.2 pre-release
7813 2008-09-15 15:11:18 +0000 Tim-Philipp Müller <tim@centricular.net>
7815 gst/: Recognise Kate subtitle streams (#550582).
7816 Original commit message from CVS:
7817 * gst-libs/gst/pbutils/descriptions.c:
7818 * gst/typefind/gsttypefindfunctions.c:
7819 Recognise Kate subtitle streams (#550582).
7821 2008-09-13 11:04:02 +0000 Tim-Philipp Müller <tim@centricular.net>
7823 gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
7824 Original commit message from CVS:
7825 * gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
7826 Remove trailing comma from enum list, which causes problems
7827 with -pendantic (#550729).
7829 2008-09-05 19:04:47 +0000 Tim-Philipp Müller <tim@centricular.net>
7831 gst-libs/gst/interfaces/propertyprobe.c: More sanity checks for our second-favourite interface.
7832 Original commit message from CVS:
7833 * gst-libs/gst/interfaces/propertyprobe.c:
7834 (gst_property_probe_get_properties),
7835 (gst_property_probe_get_property),
7836 (gst_property_probe_probe_property),
7837 (gst_property_probe_probe_property_name),
7838 (gst_property_probe_needs_probe),
7839 (gst_property_probe_needs_probe_name),
7840 (gst_property_probe_get_values),
7841 (gst_property_probe_get_values_name),
7842 (gst_property_probe_probe_and_get_values),
7843 (gst_property_probe_probe_and_get_values_name):
7844 More sanity checks for our second-favourite interface.
7846 2008-09-05 14:12:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7848 gst-libs/gst/interfaces/propertyprobe.c: Check for NULL pointer, in the hope that this fixes #532864.
7849 Original commit message from CVS:
7850 * gst-libs/gst/interfaces/propertyprobe.c:
7851 Check for NULL pointer, in the hope that this fixes #532864.
7853 2008-09-05 10:24:05 +0000 Tim-Philipp Müller <tim@centricular.net>
7855 sys/xvimage/xvimagesink.c: No really, the next release is 0.10.21 (fix Since: tags in docs).
7856 Original commit message from CVS:
7857 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
7858 No really, the next release is 0.10.21 (fix Since: tags in docs).
7860 2008-09-04 16:25:06 +0000 Wim Taymans <wim.taymans@gmail.com>
7862 gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
7863 Original commit message from CVS:
7864 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
7865 Disable a code path that is now called but causes a deadlock for some
7866 reason and is unneeded.
7868 2008-09-04 13:46:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7870 sys/xvimage/xvimagesink.*: Add a "draw-border" property that can be set to false to disable drawing borders.
7871 Original commit message from CVS:
7872 * sys/xvimage/xvimagesink.c:
7873 * sys/xvimage/xvimagesink.h:
7874 Add a "draw-border" property that can be set to false to disable
7876 * tests/icles/test-colorkey.c:
7877 * tests/icles/Makefile.am:
7878 Add new test application for the colorkey handling.
7880 2008-09-03 14:00:06 +0000 Edward Hervey <bilboed@bilboed.com>
7882 gst-libs/gst/riff/riff-media.c: Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
7883 Original commit message from CVS:
7884 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
7885 Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
7886 This will also be fixed for upcoming gst-ffmpeg release so that once
7887 this release of -base is out, it will work with the latest gst-ffmpeg
7890 2008-09-03 13:27:20 +0000 Edward Hervey <bilboed@bilboed.com>
7892 gst-libs/gst/riff/riff-media.c: Add Truespeech mapping for RIFF formats (AVI/WAV).
7893 Original commit message from CVS:
7894 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
7895 (gst_riff_create_audio_template_caps):
7896 Add Truespeech mapping for RIFF formats (AVI/WAV).
7899 2008-09-03 12:23:44 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
7901 gst/typefind/gsttypefindfunctions.c: Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
7902 Original commit message from CVS:
7903 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
7904 Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
7907 2008-09-03 10:12:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7909 Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
7910 Original commit message from CVS:
7912 * gst/subparse/Makefile.am:
7913 * gst/subparse/gstsubparse.c:
7914 * gst/subparse/samiparse.c:
7915 * tests/check/elements/subparse.c:
7916 Rework last change, so that we build subparse, but just disable the
7917 sami parse functionality, if we're configured to not use xml. In the
7918 tests only the sami test is disabled now.
7920 2008-09-02 15:07:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7922 configure.ac: Disable subparse when xml is disabled. It woundn't work anyway. Fixes test runs.
7923 Original commit message from CVS:
7925 Disable subparse when xml is disabled. It woundn't work anyway. Fixes
7928 2008-09-02 09:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
7930 po/POTFILES.in: Add some more files with strings for translation.
7931 Original commit message from CVS:
7933 Add some more files with strings for translation.
7935 2008-09-02 06:37:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7937 Use new geo location tags from core. Fixes #481169
7938 Original commit message from CVS:
7939 * gst-libs/gst/tag/gstvorbistag.c:
7940 * tests/check/libs/tag.c:
7941 Use new geo location tags from core. Fixes #481169
7943 2008-09-01 16:05:45 +0000 Edward Hervey <bilboed@bilboed.com>
7945 tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
7946 Original commit message from CVS:
7947 * tests/check/elements/audioresample.c: (setup_audioresample),
7948 (fail_unless_perfect_stream), (test_perfect_stream_instance),
7949 (test_discont_stream_instance):
7950 Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
7951 Add debugging for coherence.
7953 2008-08-30 15:55:06 +0000 Jonathan Matthew <notverysmart@gmail.com>
7955 gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
7956 Original commit message from CVS:
7957 Patch by: Jonathan Matthew <notverysmart gmail com>
7958 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
7959 Add typefinder for PDF documents (which is nice to have, since it's a
7960 common format, but also helps prevent false positives). Fixes #549814.
7962 2008-08-27 15:30:16 +0000 Wim Taymans <wim.taymans@gmail.com>
7964 gst/playback/gstplaybin2.c: Fix nasty race where multiple decodebins could start pushing data before we manage to con...
7965 Original commit message from CVS:
7966 * gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
7968 Fix nasty race where multiple decodebins could start pushing data before
7969 we manage to configure the sinks, resulting in not-linked errors in
7970 typical RTSP streaming cases.
7972 2008-08-26 17:24:31 +0000 Wim Taymans <wim.taymans@gmail.com>
7974 gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
7975 Original commit message from CVS:
7976 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
7977 Since we now call stop, we trigger this code path that causes a deadlock
7978 is apparently not needed.
7980 2008-08-26 15:45:36 +0000 Wim Taymans <wim.taymans@gmail.com>
7982 gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
7983 Original commit message from CVS:
7984 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
7985 (gst_ring_buffer_stop):
7986 Also allow the case where the ringbuffer was paused when we try to stop
7987 it so that the basesrc stop function is still called.
7989 2008-08-23 15:25:44 +0000 Mike Ruprecht <cmaiku@gmail.com>
7991 sys/v4l/gstv4lelement.c: Reprobe devices again instead of taking a cached list as new devices could've been plugged i...
7992 Original commit message from CVS:
7993 Patch by: Mike Ruprecht <cmaiku at gmail dot com>
7994 * sys/v4l/gstv4lelement.c: (gst_v4l_class_probe_devices):
7995 Reprobe devices again instead of taking a cached list as new
7996 devices could've been plugged in. Fixes bug #549062.
7998 2008-08-23 15:19:59 +0000 Alessandro Dessina <alessandro@nnva.org>
8000 ext/ogg/gstoggdemux.c: Don't add pads and activate them for skeleton streams. These are already handled inside oggdem...
8001 Original commit message from CVS:
8002 Patch by: Alessandro Dessina <alessandro nnva org>
8003 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
8004 (gst_ogg_demux_activate_chain):
8005 Don't add pads and activate them for skeleton streams. These are already
8006 handled inside oggdemux. Fixes bug #537599.
8008 2008-08-22 15:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
8010 ext/vorbis/vorbisdec.c: Reset variable so that query and convert fail after going back to
8011 Original commit message from CVS:
8012 * ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
8013 Reset variable so that query and convert fail after going back to
8014 READY. Fixes #548898.
8016 2008-08-22 07:24:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8018 ext/vorbis/vorbisenc.c: If a buffer arrives with a timestamp before the timestamp+duration of the previous buffer cli...
8019 Original commit message from CVS:
8020 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
8021 If a buffer arrives with a timestamp before the timestamp+duration
8022 of the previous buffer clip it instead of dropping it completely.
8023 Slight improvement for the unfixable bug #548913.
8025 2008-08-21 14:19:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8027 ext/vorbis/vorbisdec.c: Take the current timestamp instead of timestamp+duration for the offset.
8028 Original commit message from CVS:
8029 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
8030 Take the current timestamp instead of timestamp+duration for the offset.
8031 This offset will later be used for calculating the timestamp and
8032 otherwise vorbisdec will interpolate timestamps wrong if upstream
8033 only sends timestamps and no granulepos.
8035 2008-08-21 11:20:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8037 tests/examples/seek/seek.c: Don't crash when having no visualisations.
8038 Original commit message from CVS:
8039 * tests/examples/seek/seek.c:
8040 Don't crash when having no visualisations.
8042 2008-08-16 20:57:27 +0000 David Schleef <ds@schleef.org>
8044 gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
8045 Original commit message from CVS:
8046 * gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
8047 check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
8050 2008-08-15 07:24:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8052 gst-libs/gst/pbutils/missing-plugins.c: When cleaning up the caps fields also remove "depth" for the same reason we r...
8053 Original commit message from CVS:
8054 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
8055 When cleaning up the caps fields also remove "depth" for the same
8056 reason we remove "width".
8058 2008-08-14 17:14:53 +0000 Tim-Philipp Müller <tim@centricular.net>
8060 gst-libs/gst/pbutils/descriptions.c: Add Lead H.264 here as well.
8061 Original commit message from CVS:
8062 * gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
8063 Add Lead H.264 here as well.
8065 2008-08-14 15:17:31 +0000 Julien Moutte <julien@moutte.net>
8067 gst-libs/gst/riff/riff-media.c: Add Lead H.264 variant.
8068 Original commit message from CVS:
8069 2008-08-14 Julien Moutte <julien@fluendo.com>
8070 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
8071 (gst_riff_create_video_template_caps): Add Lead H.264 variant.
8073 2008-08-13 09:17:38 +0000 Wim Taymans <wim.taymans@gmail.com>
8075 gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
8076 Original commit message from CVS:
8077 * gst-libs/gst/audio/gstbaseaudiosrc.c:
8078 (gst_base_audio_src_create):
8079 When not slaved to another clock also subtract the base_time from our
8080 internal clock time to get the running time.
8082 2008-08-13 00:59:07 +0000 David Schleef <ds@schleef.org>
8084 ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate, since it has no basis in libtheora.
8085 Original commit message from CVS:
8086 * ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
8087 since it has no basis in libtheora.
8089 2008-08-12 06:31:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8091 gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
8092 Original commit message from CVS:
8093 * gst-libs/gst/interfaces/propertyprobe.h:
8094 Remove double "interface" from doc-string.
8095 * gst-libs/gst/interfaces/xoverlay.h:
8097 * gst-libs/gst/riff/riff.c:
8098 Add basic doc blobs.
8100 2008-08-11 15:05:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8102 gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
8103 Original commit message from CVS:
8104 * gst-libs/gst/audio/Makefile.am:
8105 Don't try to build that example anymore.
8107 2008-08-11 14:51:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8109 gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
8110 Original commit message from CVS:
8111 * gst-libs/gst/audio/.cvsignore:
8112 * gst-libs/gst/audio/Makefile.am:
8113 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
8114 * gst-libs/gst/audio/make_filter:
8115 Move audiofiltertemplate to gst-template.
8117 2008-08-11 09:20:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8119 More docs and shuffling. What can we do with the hundreds of #defines.
8120 Original commit message from CVS:
8121 * docs/libs/gst-plugins-base-libs-sections.txt:
8122 * gst-libs/gst/audio/gstaudiosrc.h:
8123 More docs and shuffling. What can we do with the hundreds of #defines.
8125 2008-08-11 08:34:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8127 gst-libs/gst/: Reducing number of dundocumented symbols.
8128 Original commit message from CVS:
8129 * gst-libs/gst/audio/audio.h:
8130 * gst-libs/gst/audio/gstaudiofilter.h:
8131 * gst-libs/gst/audio/gstringbuffer.h:
8132 * gst-libs/gst/interfaces/propertyprobe.h:
8133 * gst-libs/gst/tag/gsttagdemux.h:
8134 Reducing number of dundocumented symbols.
8136 2008-08-11 07:16:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8138 gst-libs/gst/audio/audio.c: Fix doc comment syntax.
8139 Original commit message from CVS:
8140 * gst-libs/gst/audio/audio.c:
8141 Fix doc comment syntax.
8142 * gst-libs/gst/interfaces/propertyprobe.c:
8143 Add more doc-comments and a FIXME: for the signal.
8145 2008-08-07 16:11:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8147 ext/ogg/gstoggmux.*: Don't pretend to support NEWSEGMENT events, instead override the
8148 Original commit message from CVS:
8149 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sink_event),
8150 (gst_ogg_mux_request_new_pad):
8151 * ext/ogg/gstoggmux.h:
8152 Don't pretend to support NEWSEGMENT events, instead override the
8153 GstCollectPads event function to return FALSE on NEWSEGMENT events
8154 and do the normal work for other events.
8155 This prevents elements like flacenc to seek to the start and rewrite
8156 some data which then results in a broken Ogg packet.
8158 2008-08-07 15:58:58 +0000 Frederic Crozat <fcrozat@mandriva.org>
8160 Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
8161 Original commit message from CVS:
8162 Patch by: Frederic Crozat <fcrozat@mandriva.org>
8163 * ext/alsa/gstalsaplugin.c: (plugin_init):
8164 * ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
8165 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
8166 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
8167 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
8168 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
8169 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
8170 * gst/playback/gstdecodebin.c: (plugin_init):
8171 * gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
8172 * gst/playback/gstplayback.c: (plugin_init):
8173 * gst/playback/gstqueue2.c: (plugin_init):
8174 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
8175 * sys/v4l/gstv4l.c: (plugin_init):
8176 Make sure gettext returns translations in UTF-8 encoding rather
8177 than in the current locale encoding (#546822).
8179 2008-08-06 13:12:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8181 gst-libs/gst/pbutils/descriptions.c: Add audio/x-qdm for qtdemux.
8182 Original commit message from CVS:
8183 * gst-libs/gst/pbutils/descriptions.c:
8184 Add audio/x-qdm for qtdemux.
8186 2008-08-05 15:38:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8188 ext/vorbis/vorbisdec.c: Do not leak old taglist.
8189 Original commit message from CVS:
8190 * ext/vorbis/vorbisdec.c:
8191 Do not leak old taglist.
8193 2008-08-04 12:35:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8195 tests/icles/test-scale.c: Include <stdlib.h> for atoi().
8196 Original commit message from CVS:
8197 * tests/icles/test-scale.c:
8198 Include <stdlib.h> for atoi().
8200 2008-08-04 09:11:08 +0000 Andy Wingo <wingo@pobox.com>
8202 gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
8203 Original commit message from CVS:
8204 2008-08-04 Andy Wingo <wingo@pobox.com>
8205 * gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
8208 2008-08-01 13:06:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8210 gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
8211 Original commit message from CVS:
8212 * gst/adder/gstadder.c:
8213 Cleanup lots of empty lines that came from gst-indent going havoc
8214 before I added the INDENT_ON/OFF marker some time agao.
8216 2008-08-01 11:55:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8218 Bump requirement to latest core and use new tag for riff formats.
8219 Original commit message from CVS:
8221 * gst-libs/gst/riff/riff-read.c:
8222 Bump requirement to latest core and use new tag for riff formats.
8225 2008-08-01 11:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
8227 tests/examples/dynamic/: Add example app that dynamically switches between 3 'encoders'.
8228 Original commit message from CVS:
8229 * tests/examples/dynamic/Makefile.am:
8230 * tests/examples/dynamic/codec-select.c: (make_encoder),
8231 (make_pipeline), (do_switch), (my_bus_callback), (main):
8232 Add example app that dynamically switches between 3 'encoders'.
8234 2008-07-31 13:06:13 +0000 Wim Taymans <wim.taymans@gmail.com>
8236 gst/playback/gstplaysink.c: Add some more comments.
8237 Original commit message from CVS:
8238 * gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
8239 Add some more comments.
8241 2008-07-31 12:58:44 +0000 Wim Taymans <wim.taymans@gmail.com>
8243 gst/videotestsrc/gstvideotestsrc.c: Discard buffers of the wrong size after renegotiation, this is perfectly possible...
8244 Original commit message from CVS:
8245 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
8246 (gst_video_test_src_create):
8247 Discard buffers of the wrong size after renegotiation, this is perfectly
8248 possible with things like capsfilter that could suggest caps changes
8249 upstream without knowing the size of the buffer.
8251 2008-07-31 11:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
8253 tests/icles/: Add dynamic rescaling tests for the new basetransform.
8254 Original commit message from CVS:
8255 * tests/icles/.cvsignore:
8256 * tests/icles/Makefile.am:
8257 * tests/icles/test-scale.c: (make_pipeline), (main):
8258 Add dynamic rescaling tests for the new basetransform.
8260 2008-07-30 19:51:36 +0000 Tim-Philipp Müller <tim@centricular.net>
8262 gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
8263 Original commit message from CVS:
8264 * gst/audioconvert/Makefile.am:
8265 Dist recently-added gstfastrandom.h.
8267 2008-07-30 15:29:44 +0000 Edward Hervey <bilboed@bilboed.com>
8269 sys/xvimage/xvimagesink.c: Fix a "may be used uninitialized in this function" which weirdly only appears on macosx (?).
8270 Original commit message from CVS:
8271 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
8272 Fix a "may be used uninitialized in this function" which weirdly only
8273 appears on macosx (?).
8275 2008-07-30 09:02:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8277 gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
8278 Original commit message from CVS:
8279 * gst-libs/gst/riff/riff-ids.h:
8280 Adding acid chunk for tempo and loop information.
8282 2008-07-29 13:01:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8284 sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
8285 Original commit message from CVS:
8286 * sys/xvimage/Makefile.am:
8287 floor() needs linking to $(LIBM).
8289 2008-07-29 12:35:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8291 ext/gnomevfs/gstgnomevfssrc.c: Aggregate short reads and add some comments and debug logging.
8292 Original commit message from CVS:
8293 * ext/gnomevfs/gstgnomevfssrc.c:
8294 Aggregate short reads and add some comments and debug logging.
8297 2008-07-29 10:26:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8299 gst/playback/gstplaybasebin.c: Fix property doc markup (its not a signal).
8300 Original commit message from CVS:
8301 * gst/playback/gstplaybasebin.c:
8302 Fix property doc markup (its not a signal).
8303 * sys/xvimage/xvimagesink.c:
8304 Add since tag for new proeprties (also add sice tags fro the last two
8307 2008-07-29 08:59:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8309 sys/xvimage/xvimagesink.*: Add autofill/colorkey properties. Fixes #538656.
8310 Original commit message from CVS:
8311 * sys/xvimage/xvimagesink.c:
8312 * sys/xvimage/xvimagesink.h:
8313 Add autofill/colorkey properties. Fixes #538656.
8315 2008-07-29 01:58:05 +0000 David Schleef <ds@schleef.org>
8317 sys/xvimage/xvimagesink.c: Fix rounding errors when converting colorbalance values between hardware and object proper...
8318 Original commit message from CVS:
8319 * sys/xvimage/xvimagesink.c:
8320 Fix rounding errors when converting colorbalance values
8321 between hardware and object property ranges. Partial
8322 fix for #537889, however, there still seems to be a small
8323 drift problem that could be totem's fault.
8325 2008-07-28 15:34:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8327 ext/ogg/gstoggdemux.c: Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
8328 Original commit message from CVS:
8329 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
8330 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
8331 Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
8332 This fixes a critical warning.
8334 2008-07-28 13:12:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8336 ext/ogg/gstoggmux.c: Allow muxing of CELT into Ogg streams.
8337 Original commit message from CVS:
8338 * ext/ogg/gstoggmux.c:
8339 Allow muxing of CELT into Ogg streams.
8341 2008-07-28 12:47:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8343 gst/typefind/gsttypefindfunctions.c: Add simple typefinder for the CELT codec (www.celt-codec.org).
8344 Original commit message from CVS:
8345 * gst/typefind/gsttypefindfunctions.c: (celt_type_find),
8347 Add simple typefinder for the CELT codec (www.celt-codec.org).
8349 2008-07-27 11:12:41 +0000 Jan Gerber <j@oil21.org>
8351 ext/ogg/gstoggdemux.c: Fix calculation of the start time from skeleton streams.
8352 Original commit message from CVS:
8353 Patch by: Jan Gerber <j at oil21 dot org>
8354 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
8355 Fix calculation of the start time from skeleton streams.
8358 2008-07-24 13:19:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8360 tests/examples/seek/seek.c: Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
8361 Original commit message from CVS:
8362 * tests/examples/seek/seek.c:
8363 Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
8365 2008-07-23 18:34:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8367 gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
8368 Original commit message from CVS:
8369 * gst/audioconvert/audioconvert.h:
8370 * gst/audioconvert/gstaudioquantize.c:
8371 (gst_audio_quantize_setup_dither),
8372 (gst_audio_quantize_free_dither):
8373 * gst/audioconvert/gstfastrandom.h:
8374 Implement a linear congruential generator as pseudo random number
8375 generator for the dither noise. This is about 2 times faster than
8376 using GLib's mersenne twister. Also this uses only integer math for
8377 generating integers while GLib internally uses floating point math.
8379 2008-07-23 18:27:15 +0000 Michael Smith <msmith@xiph.org>
8381 configure.ac: Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
8382 Original commit message from CVS:
8384 Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
8386 2008-07-23 13:17:31 +0000 Damien Lespiau <damien.lespiau@gmail.com>
8388 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
8389 Original commit message from CVS:
8390 Patch by: Damien Lespiau <damien.lespiau gmail com>
8391 * gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
8392 Use GST_STR_NULL to avoid crashes with libcs that don't
8393 like NULL strings in printf args (such as the win32 one).
8396 2008-07-17 14:21:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8398 sys/xvimage/xvimagesink.c: Oops - set the size of the image used for probing back to 1x1, for consistency with ximage...
8399 Original commit message from CVS:
8400 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
8401 Oops - set the size of the image used for probing back to 1x1, for
8402 consistency with ximagesink
8404 2008-07-17 13:57:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8406 sys/: it's not legal to ask the
8407 Original commit message from CVS:
8408 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
8409 (gst_ximagesink_ximage_new):
8410 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
8411 (gst_xvimagesink_xvimage_new):
8412 Apparently on Solaris and OS/X (at least), it's not legal to ask the
8413 X server to attach to a shared memory segment after we've deleted it,
8414 with the result that MIT-SHM is disabled. Instead, remove it only after
8415 X succeeds in attaching too.
8417 2008-07-17 02:30:24 +0000 David Schleef <ds@schleef.org>
8419 gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second.
8420 Original commit message from CVS:
8421 * gst/audiotestsrc/gstaudiotestsrc.c:
8422 * gst/audiotestsrc/gstaudiotestsrc.h:
8423 Add 'ticks', a 1/30 second sine wave pulse every second.
8425 2008-07-15 22:43:16 +0000 David Schleef <ds@schleef.org>
8427 gst-libs/gst/video/video.c: Revert ABI change.
8428 Original commit message from CVS:
8429 * gst-libs/gst/video/video.c: Revert ABI change.
8431 2008-07-15 13:05:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8433 gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
8434 Original commit message from CVS:
8435 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
8436 Make it impossible to have NULL caps at the point where we set
8437 framerate and other things. Also don't return immediately for "3ivd"
8438 video and let framerate, etc be set. Might fix bug #542508.
8440 2008-07-14 17:06:26 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
8442 gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
8443 Original commit message from CVS:
8444 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
8445 Video format can also be conveniently determined from (many)
8448 2008-07-14 08:18:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8450 gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
8451 Original commit message from CVS:
8452 * gst/playback/gstplaybasebin.c:
8453 * gst/playback/gstplaybasebin.h:
8454 * gst/playback/gstplaybin.c:
8455 * gst/playback/gststreamselector.c:
8456 First stab at integrating DVD subpicture overlay into
8457 playbin. Successfully plugs and plays, but the queues need
8458 shrinking - 3 seconds of video is too much buffering.
8460 2008-07-11 18:06:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8462 gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs.
8463 Original commit message from CVS:
8464 * gst/audioconvert/gstaudioconvert.c:
8465 Remove now obsolete note in the docs.
8467 2008-07-11 06:10:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8469 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
8470 Original commit message from CVS:
8471 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
8472 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
8473 * docs/plugins/gst-plugins-base-plugins-sections.txt:
8474 * docs/plugins/gst-plugins-base-plugins.args:
8475 * docs/plugins/gst-plugins-base-plugins.hierarchy:
8476 * docs/plugins/gst-plugins-base-plugins.interfaces:
8477 * docs/plugins/gst-plugins-base-plugins.prerequisites:
8478 * docs/plugins/gst-plugins-base-plugins.signals:
8479 * docs/plugins/inspect/plugin-adder.xml:
8480 * docs/plugins/inspect/plugin-alsa.xml:
8481 * docs/plugins/inspect/plugin-audioconvert.xml:
8482 * docs/plugins/inspect/plugin-audiorate.xml:
8483 * docs/plugins/inspect/plugin-audioresample.xml:
8484 * docs/plugins/inspect/plugin-audiotestsrc.xml:
8485 * docs/plugins/inspect/plugin-cdparanoia.xml:
8486 * docs/plugins/inspect/plugin-decodebin.xml:
8487 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
8488 * docs/plugins/inspect/plugin-gdp.xml:
8489 * docs/plugins/inspect/plugin-gnomevfs.xml:
8490 * docs/plugins/inspect/plugin-libvisual.xml:
8491 * docs/plugins/inspect/plugin-ogg.xml:
8492 * docs/plugins/inspect/plugin-pango.xml:
8493 * docs/plugins/inspect/plugin-playback.xml:
8494 * docs/plugins/inspect/plugin-queue2.xml:
8495 * docs/plugins/inspect/plugin-subparse.xml:
8496 * docs/plugins/inspect/plugin-tcp.xml:
8497 * docs/plugins/inspect/plugin-theora.xml:
8498 * docs/plugins/inspect/plugin-typefindfunctions.xml:
8499 * docs/plugins/inspect/plugin-uridecodebin.xml:
8500 * docs/plugins/inspect/plugin-video4linux.xml:
8501 * docs/plugins/inspect/plugin-videorate.xml:
8502 * docs/plugins/inspect/plugin-videoscale.xml:
8503 * docs/plugins/inspect/plugin-videotestsrc.xml:
8504 * docs/plugins/inspect/plugin-volume.xml:
8505 * docs/plugins/inspect/plugin-vorbis.xml:
8506 * docs/plugins/inspect/plugin-ximagesink.xml:
8507 * docs/plugins/inspect/plugin-xvimagesink.xml:
8508 * ext/alsa/gstalsamixer.c:
8509 * ext/alsa/gstalsasink.c:
8510 * ext/alsa/gstalsasrc.c:
8511 * ext/gio/gstgiosink.c:
8512 * ext/gio/gstgiosrc.c:
8513 * ext/gio/gstgiostreamsink.c:
8514 * ext/gio/gstgiostreamsrc.c:
8515 * ext/gnomevfs/gstgnomevfssink.c:
8516 * ext/gnomevfs/gstgnomevfssrc.c:
8517 * ext/ogg/gstoggdemux.c:
8518 * ext/ogg/gstoggmux.c:
8519 * ext/pango/gstclockoverlay.c:
8520 * ext/pango/gsttextoverlay.c:
8521 * ext/pango/gsttextrender.c:
8522 * ext/pango/gsttimeoverlay.c:
8523 * ext/theora/theoradec.c:
8524 * ext/theora/theoraenc.c:
8525 * ext/theora/theoraparse.c:
8526 * ext/vorbis/vorbisdec.c:
8527 * ext/vorbis/vorbisenc.c:
8528 * ext/vorbis/vorbisparse.c:
8529 * ext/vorbis/vorbistag.c:
8530 * gst/adder/gstadder.c:
8531 * gst/audioconvert/gstaudioconvert.c:
8532 * gst/audioresample/gstaudioresample.c:
8533 * gst/audiotestsrc/gstaudiotestsrc.c:
8534 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
8535 * gst/gdp/gstgdpdepay.c:
8536 * gst/gdp/gstgdppay.c:
8537 * gst/playback/gstdecodebin2.c:
8538 * gst/playback/gstplaybin.c:
8539 * gst/playback/gstplaybin2.c:
8540 * gst/playback/gstqueue2.c:
8541 * gst/playback/gsturidecodebin.c:
8542 * gst/tcp/gstmultifdsink.c:
8543 * gst/tcp/gsttcpserversink.c:
8544 * gst/videorate/gstvideorate.c:
8545 * gst/videoscale/gstvideoscale.c:
8546 * gst/videotestsrc/gstvideotestsrc.c:
8547 * gst/volume/gstvolume.c:
8548 * sys/ximage/ximagesink.c:
8549 * sys/xvimage/xvimagesink.c:
8550 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
8551 titles. Drop mentining that all our example pipelines are "simple"
8554 2008-07-10 21:06:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8556 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
8557 Original commit message from CVS:
8558 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
8559 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
8560 * docs/plugins/gst-plugins-base-plugins-sections.txt:
8561 * docs/plugins/gst-plugins-base-plugins.args:
8562 * docs/plugins/gst-plugins-base-plugins.hierarchy:
8563 * docs/plugins/gst-plugins-base-plugins.interfaces:
8564 * docs/plugins/gst-plugins-base-plugins.prerequisites:
8565 * docs/plugins/gst-plugins-base-plugins.signals:
8566 * docs/plugins/inspect/plugin-adder.xml:
8567 * docs/plugins/inspect/plugin-alsa.xml:
8568 * docs/plugins/inspect/plugin-audioconvert.xml:
8569 * docs/plugins/inspect/plugin-audiorate.xml:
8570 * docs/plugins/inspect/plugin-audioresample.xml:
8571 * docs/plugins/inspect/plugin-audiotestsrc.xml:
8572 * docs/plugins/inspect/plugin-cdparanoia.xml:
8573 * docs/plugins/inspect/plugin-decodebin.xml:
8574 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
8575 * docs/plugins/inspect/plugin-gdp.xml:
8576 * docs/plugins/inspect/plugin-gnomevfs.xml:
8577 * docs/plugins/inspect/plugin-libvisual.xml:
8578 * docs/plugins/inspect/plugin-ogg.xml:
8579 * docs/plugins/inspect/plugin-pango.xml:
8580 * docs/plugins/inspect/plugin-playback.xml:
8581 * docs/plugins/inspect/plugin-queue2.xml:
8582 * docs/plugins/inspect/plugin-subparse.xml:
8583 * docs/plugins/inspect/plugin-tcp.xml:
8584 * docs/plugins/inspect/plugin-theora.xml:
8585 * docs/plugins/inspect/plugin-typefindfunctions.xml:
8586 * docs/plugins/inspect/plugin-uridecodebin.xml:
8587 * docs/plugins/inspect/plugin-video4linux.xml:
8588 * docs/plugins/inspect/plugin-videorate.xml:
8589 * docs/plugins/inspect/plugin-videoscale.xml:
8590 * docs/plugins/inspect/plugin-videotestsrc.xml:
8591 * docs/plugins/inspect/plugin-volume.xml:
8592 * docs/plugins/inspect/plugin-vorbis.xml:
8593 * docs/plugins/inspect/plugin-ximagesink.xml:
8594 * docs/plugins/inspect/plugin-xvimagesink.xml:
8595 * ext/alsa/gstalsamixer.c:
8596 * ext/alsa/gstalsasink.c:
8597 * ext/alsa/gstalsasrc.c:
8598 * ext/gio/gstgiosink.c:
8599 * ext/gio/gstgiosrc.c:
8600 * ext/gio/gstgiostreamsink.c:
8601 * ext/gio/gstgiostreamsrc.c:
8602 * ext/gnomevfs/gstgnomevfssink.c:
8603 * ext/gnomevfs/gstgnomevfssrc.c:
8604 * ext/ogg/gstoggdemux.c:
8605 * ext/ogg/gstoggmux.c:
8606 * ext/pango/gstclockoverlay.c:
8607 * ext/pango/gsttextoverlay.c:
8608 * ext/pango/gsttextrender.c:
8609 * ext/pango/gsttimeoverlay.c:
8610 * ext/theora/theoradec.c:
8611 * ext/theora/theoraenc.c:
8612 * ext/theora/theoraparse.c:
8613 * ext/vorbis/vorbisdec.c:
8614 * ext/vorbis/vorbisenc.c:
8615 * ext/vorbis/vorbisparse.c:
8616 * ext/vorbis/vorbistag.c:
8617 * gst/adder/gstadder.c:
8618 * gst/audioconvert/gstaudioconvert.c:
8619 * gst/audioresample/gstaudioresample.c:
8620 * gst/audiotestsrc/gstaudiotestsrc.c:
8621 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
8622 * gst/gdp/gstgdpdepay.c:
8623 * gst/gdp/gstgdppay.c:
8624 * gst/playback/gstdecodebin2.c:
8625 * gst/playback/gstplaybin.c:
8626 * gst/playback/gstplaybin2.c:
8627 * gst/playback/gstqueue2.c:
8628 * gst/playback/gsturidecodebin.c:
8629 * gst/tcp/gstmultifdsink.c:
8630 * gst/tcp/gsttcpserversink.c:
8631 * gst/videorate/gstvideorate.c:
8632 * gst/videoscale/gstvideoscale.c:
8633 * gst/videotestsrc/gstvideotestsrc.c:
8634 * gst/volume/gstvolume.c:
8635 * sys/ximage/ximagesink.c:
8636 * sys/xvimage/xvimagesink.c:
8637 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
8638 titles. Drop mentining that all our example pipelines are "simple"
8641 2008-07-07 17:25:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8643 tests/examples/seek/Makefile.am: Fix out of tree build by adding all required CFLAGS.
8644 Original commit message from CVS:
8645 * tests/examples/seek/Makefile.am:
8646 Fix out of tree build by adding all required CFLAGS.
8648 2008-07-07 09:55:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8650 gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
8651 Original commit message from CVS:
8652 * gst/playback/gstdecodebin.c: (add_raw_queue):
8653 And ref the pad before returning it again when linking to the queue
8654 failed. Otherwise we will unref the pad twice later and things break.
8656 2008-07-07 09:48:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8658 gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
8659 Original commit message from CVS:
8660 * gst/playback/gstdecodebin.c: (add_raw_queue):
8661 If linking the raw pad with a queue fails, try it without a queue
8662 instead of failing completely. This should never happen.
8664 2008-07-06 23:22:12 +0000 Evgeniy Stepanov <eugeni.stepanov@gmail.com>
8666 gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
8667 Original commit message from CVS:
8668 Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
8669 * gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
8670 Add a queue after a demuxer if the demuxer outputs raw data. This was
8671 done before only for non-raw data but is required in this case too.
8673 decodebin2 doesn't have this issue because all streams of a group
8674 go through multiqueue.
8676 2008-07-03 09:12:49 +0000 Damien Lespiau <damien.lespiau@gmail.com>
8678 gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
8679 Original commit message from CVS:
8680 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
8681 * gst-libs/gst/sdp/gstsdpmessage.c:
8682 Makes libgstsdp compile with mingw32 by defining the right WINVER so
8683 that getaddrinfo() can be used. Fixes #541358.
8685 2008-07-01 13:22:49 +0000 Wim Taymans <wim.taymans@gmail.com>
8687 gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
8688 Original commit message from CVS:
8689 * gst/videotestsrc/gstvideotestsrc.c:
8690 (gst_video_test_src_class_init), (gst_video_test_src_init),
8691 (gst_video_test_src_set_property),
8692 (gst_video_test_src_get_property), (gst_video_test_src_create):
8693 * gst/videotestsrc/gstvideotestsrc.h:
8694 Cleanups, use default property values as defines.
8695 Add property to enable/disable peer buffer allocation.
8697 2008-06-30 09:46:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8699 tests/check/: Enable unit tests on PPC again as the bugs are now fixed.
8700 Original commit message from CVS:
8701 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
8702 * tests/check/pipelines/streamheader.c: (streamheader_suite):
8703 Enable unit tests on PPC again as the bugs are now fixed.
8705 2008-06-30 09:20:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8707 gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
8708 Original commit message from CVS:
8709 * gst-libs/gst/riff/riff-ids.h:
8710 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
8711 (gst_riff_create_audio_template_caps):
8712 Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
8715 2008-06-30 08:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8717 gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
8718 Original commit message from CVS:
8719 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
8720 (gst_ffmpeg_pixfmt_to_caps):
8721 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
8722 (gst_ffmpegcsp_get_unit_size):
8723 Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
8724 it on other formats. Also adjust the unit size only for that format
8725 to not include the palette. Fixes bug #540497.
8727 2008-06-29 13:45:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8729 gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
8730 Original commit message from CVS:
8731 * gst/adder/gstadder.c:
8732 Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
8734 2008-06-27 07:55:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8736 ChangeLog: ChangeLog surgery.
8737 Original commit message from CVS:
8740 * tests/examples/seek/seek.c:
8741 Move variable into ifdef too.
8743 2008-06-27 07:42:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8745 tests/examples/seek/seek.c: Include config.h and check if we have X. Fixes: #540334.
8746 Original commit message from CVS:
8747 * tests/examples/seek/seek.c:
8748 Include config.h and check if we have X. Fixes: #540334.
8750 2008-06-26 06:03:38 +0000 Sam Morris <sam@robots.org.to.uk>
8752 gst-libs/gst/interfaces/mixertrack.c: API: Add "index" property to GstMixerTrack to differantiate between multiple mi...
8753 Original commit message from CVS:
8754 Patch by: Sam Morris <sam at robots dot org to uk>
8755 * gst-libs/gst/interfaces/mixertrack.c:
8756 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
8757 (gst_mixer_track_set_property):
8758 API: Add "index" property to GstMixerTrack to differantiate between
8759 multiple mixer tracks with the same label.
8760 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
8761 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
8762 Set the "index" property of GstMixerTrack to the index given by ALSA.
8765 2008-06-25 13:15:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8767 tests/examples/seek/: Remove libgstvideo usage. Use gtk_get_option_group instead of gtk_init().
8768 Original commit message from CVS:
8769 * tests/examples/seek/Makefile.am:
8770 * tests/examples/seek/seek.c:
8771 Remove libgstvideo usage. Use gtk_get_option_group instead of
8774 2008-06-24 16:27:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8776 tests/check/Makefile.am: Name the test registry format neutral.
8777 Original commit message from CVS:
8778 * tests/check/Makefile.am:
8779 Name the test registry format neutral.
8781 2008-06-24 16:22:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8783 gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
8784 Original commit message from CVS:
8785 * gst/playback/gstqueue2.c:
8786 Do not double notify. Remove the unsued return value.
8788 2008-06-24 16:15:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8790 ext/alsa/gstalsamixer.c: Also consider "speaker" as a name for master volume. If that doesn't help look for the first...
8791 Original commit message from CVS:
8792 * ext/alsa/gstalsamixer.c:
8793 Also consider "speaker" as a name for master volume. If that doesn't
8794 help look for the first non-mono volume control that also has a
8797 2008-06-24 16:10:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8799 ChangeLog: Forgot to save the ChangeLog :/
8800 Original commit message from CVS:
8802 Forgot to save the ChangeLog :/
8804 2008-06-24 16:05:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8806 tests/examples/seek/: Embedd the xwindow.
8807 Original commit message from CVS:
8808 * tests/examples/seek/Makefile.am:
8809 * tests/examples/seek/seek.c:
8812 2008-06-24 01:14:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8814 sys/ximage/ximagesink.h: When the caps change, make sure to re-draw borders in force-aspect-ratio=true mode.
8815 Original commit message from CVS:
8816 * sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
8817 (gst_ximagesink_setcaps):
8818 * sys/ximage/ximagesink.h:
8819 When the caps change, make sure to re-draw borders in
8820 force-aspect-ratio=true mode.
8821 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
8822 Don't clear the border_draw flag until we actually draw the border.
8823 * tests/check/Makefile.am:
8824 Ignore alsasink/src during the states test too, so it doesn't fail
8825 when running without access to the sound device.
8827 2008-06-22 18:35:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8829 tests/examples/seek/seek.c: Fix crasher when playing a parse-launch line the 2nd time.
8830 Original commit message from CVS:
8831 * tests/examples/seek/seek.c:
8832 Fix crasher when playing a parse-launch line the 2nd time.
8834 2008-06-21 18:56:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
8836 tests/check/pipelines/oggmux.c: Properly ifdef tests to fix compilation.
8837 Original commit message from CVS:
8838 * tests/check/pipelines/oggmux.c:
8839 Properly ifdef tests to fix compilation.
8841 2008-06-21 10:25:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
8845 Original commit message from CVS:
8848 2008-06-20 18:24:24 +0000 Michael Smith <msmith@xiph.org>
8850 gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
8851 Original commit message from CVS:
8852 * gst/playback/gstplay-marshal.list:
8853 * gst/playback/gstplaybin2.c:
8854 Add get-video-pad, get-audio-pad, get-text-pad action signals to
8855 playbin2. This allows the user to get to the selector's sinkpads, and
8856 thus inspect a range of things - caps, tags, etc.
8858 2008-06-20 17:27:03 +0000 Michael Smith <msmith@xiph.org>
8860 gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
8861 Original commit message from CVS:
8862 * gst/playback/gstplaybin2.c:
8863 Use a different constant for the convert-frame signal id.
8866 2008-06-20 17:18:55 +0000 Michael Smith <msmith@xiph.org>
8868 gst/playback/: Fix a whole bunch of typos in comments and log statements.
8869 Original commit message from CVS:
8870 * gst/playback/gstplaybin2.c:
8871 * gst/playback/gstplaysink.c:
8872 Fix a whole bunch of typos in comments and log statements.
8874 2008-06-20 17:02:48 +0000 Michael Smith <msmith@xiph.org>
8876 sys/xvimage/xvimagesink.c: Don't set colour balance values on the Xv port if the user hasn't changed them (via proper...
8877 Original commit message from CVS:
8878 * sys/xvimage/xvimagesink.c:
8879 Don't set colour balance values on the Xv port if the user hasn't
8880 changed them (via properties or the interface). Avoids accumulating
8881 rounding errors for the common case.
8882 Partial fix for bug #537889.
8884 2008-06-20 16:56:18 +0000 Michael Smith <msmith@xiph.org>
8886 gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
8887 Original commit message from CVS:
8888 * gst/playback/gstdecodebin2.c:
8889 Ensure decodebin2 emits 'drained' signal once, and only once, when all
8892 2008-06-20 16:12:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
8895 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...
8896 Original commit message from CVS:
8897 apparently it's an error to specify nc -l -p 3000 - though the short usage
8898 does not make it very clear that you can drop the host arg with -l
8900 2008-06-20 09:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
8902 ext/vorbis/vorbisenc.c: Report the encoder latency. Fixes #538232.
8903 Original commit message from CVS:
8904 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
8905 (gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
8906 Report the encoder latency. Fixes #538232.
8908 2008-06-20 09:19:59 +0000 Wim Taymans <wim.taymans@gmail.com>
8910 gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
8911 Original commit message from CVS:
8912 * gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
8913 (notify_source), (activate_group):
8914 Implement the source property, emit notify when it changes in the
8915 underlying uridecodebin.
8917 2008-06-20 09:14:26 +0000 Wim Taymans <wim.taymans@gmail.com>
8919 tests/examples/seek/seek.c: Free and clear the seek element list so that we don't use invalid references when seeking...
8920 Original commit message from CVS:
8921 * tests/examples/seek/seek.c: (stop_cb):
8922 Free and clear the seek element list so that we don't use invalid
8923 references when seeking after recreating a gst-launch line.
8925 2008-06-20 09:09:37 +0000 Wim Taymans <wim.taymans@gmail.com>
8927 gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
8928 Original commit message from CVS:
8929 * gst-libs/gst/audio/gstbaseaudiosink.c:
8930 (gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
8931 (gst_base_audio_sink_render):
8932 Report latency even if we are not live instead of hiding it.
8933 Take ts-offset and render-delay of the basesink into account when
8935 Rework the clipping code so that we can take the various offsets into
8936 account and still do correct clipping.
8938 2008-06-20 08:52:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8940 configure.ac: Bump verion back to devel -> 0.10.20.1
8941 Original commit message from CVS:
8943 Bump verion back to devel -> 0.10.20.1
8945 2008-06-20 08:47:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8947 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...
8948 Original commit message from CVS:
8949 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
8950 Don't increase the size of non-string image buffers by one as this
8951 might in theory confuse decoders. Still increase it by one for string
8952 image buffers to append '\0'.
8954 2008-06-20 08:45:13 +0000 Antoine Tremblay <hexa00@gmail.com>
8956 gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
8957 Original commit message from CVS:
8958 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
8959 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
8960 Fix a buffer memleak and remove a confusing and wrong debug output.
8963 2008-06-19 11:25:37 +0000 Wim Taymans <wim.taymans@gmail.com>
8965 examples/app/appsink-src.c: Don't use a buffer after unreffing it.
8966 Original commit message from CVS:
8967 * examples/app/appsink-src.c: (on_new_buffer_from_source):
8968 Don't use a buffer after unreffing it.
8970 === release 0.10.20 ===
8972 2008-06-18 14:36:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8978 * docs/plugins/gst-plugins-base-plugins.args:
8979 * docs/plugins/gst-plugins-base-plugins.hierarchy:
8980 * docs/plugins/gst-plugins-base-plugins.interfaces:
8981 * docs/plugins/gst-plugins-base-plugins.prerequisites:
8982 * docs/plugins/inspect/plugin-adder.xml:
8983 * docs/plugins/inspect/plugin-alsa.xml:
8984 * docs/plugins/inspect/plugin-audioconvert.xml:
8985 * docs/plugins/inspect/plugin-audiorate.xml:
8986 * docs/plugins/inspect/plugin-audioresample.xml:
8987 * docs/plugins/inspect/plugin-audiotestsrc.xml:
8988 * docs/plugins/inspect/plugin-cdparanoia.xml:
8989 * docs/plugins/inspect/plugin-decodebin.xml:
8990 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
8991 * docs/plugins/inspect/plugin-gdp.xml:
8992 * docs/plugins/inspect/plugin-gnomevfs.xml:
8993 * docs/plugins/inspect/plugin-libvisual.xml:
8994 * docs/plugins/inspect/plugin-ogg.xml:
8995 * docs/plugins/inspect/plugin-pango.xml:
8996 * docs/plugins/inspect/plugin-playback.xml:
8997 * docs/plugins/inspect/plugin-queue2.xml:
8998 * docs/plugins/inspect/plugin-subparse.xml:
8999 * docs/plugins/inspect/plugin-tcp.xml:
9000 * docs/plugins/inspect/plugin-theora.xml:
9001 * docs/plugins/inspect/plugin-typefindfunctions.xml:
9002 * docs/plugins/inspect/plugin-uridecodebin.xml:
9003 * docs/plugins/inspect/plugin-video4linux.xml:
9004 * docs/plugins/inspect/plugin-videorate.xml:
9005 * docs/plugins/inspect/plugin-videoscale.xml:
9006 * docs/plugins/inspect/plugin-videotestsrc.xml:
9007 * docs/plugins/inspect/plugin-volume.xml:
9008 * docs/plugins/inspect/plugin-vorbis.xml:
9009 * docs/plugins/inspect/plugin-ximagesink.xml:
9010 * docs/plugins/inspect/plugin-xvimagesink.xml:
9011 * gst-plugins-base.doap:
9013 * win32/common/config.h:
9015 Original commit message from CVS:
9018 2008-06-18 14:32:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9047 Original commit message from CVS:
9050 2008-06-18 06:31:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9052 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
9053 Original commit message from CVS:
9054 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
9055 * examples/app/appsrc-ra.c:
9056 * examples/app/appsrc-seekable.c:
9057 * examples/app/appsrc-stream.c:
9058 * examples/app/appsrc-stream2.c:
9059 * ext/directfb/dfbvideosink.h:
9060 * ext/metadata/gstbasemetadata.c:
9061 * ext/metadata/gstbasemetadata.h:
9062 * ext/metadata/metadata.c:
9063 * ext/metadata/metadataexif.c:
9064 * ext/theora/theoradec.h:
9065 * gst/deinterlace2/gstdeinterlace2.h:
9066 * gst/deinterlace2/tvtime/speedy.c:
9067 * gst/deinterlace2/tvtime/speedy.h:
9068 * gst/deinterlace2/tvtime/vfir.c:
9069 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
9072 2008-06-16 14:11:36 +0000 Andy Wingo <wingo@pobox.com>
9074 * gst-libs/gst/app/gstappsrc.c:
9075 gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
9076 Original commit message from CVS:
9077 2008-06-16 Andy Wingo <wingo@pobox.com>
9078 * gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
9079 (gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
9080 G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
9082 2008-06-16 07:30:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9084 Final round of doc updates.
9085 Original commit message from CVS:
9086 * gst/rtpmanager/gstrtpjitterbuffer.c:
9087 * gst/speed/gstspeed.c:
9088 * gst/speexresample/gstspeexresample.c:
9089 * gst/videosignal/gstvideoanalyse.c:
9090 * gst/videosignal/gstvideodetect.c:
9091 * gst/videosignal/gstvideomark.c:
9092 * sys/dvb/gstdvbsrc.c:
9093 * sys/oss4/oss4-mixer.c:
9094 * sys/oss4/oss4-sink.c:
9095 * sys/oss4/oss4-source.c:
9096 * sys/wininet/gstwininetsrc.c:
9097 Final round of doc updates.
9099 2008-06-13 11:59:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9101 docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
9102 Original commit message from CVS:
9103 * docs/plugins/Makefile.am:
9104 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
9105 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
9106 * docs/plugins/gst-plugins-bad-plugins.args:
9107 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
9108 * docs/plugins/gst-plugins-bad-plugins.interfaces:
9109 * docs/plugins/gst-plugins-bad-plugins.prerequisites:
9110 * docs/plugins/gst-plugins-bad-plugins.signals:
9111 * docs/plugins/inspect/plugin-alsaspdif.xml:
9112 * docs/plugins/inspect/plugin-amrwb.xml:
9113 * docs/plugins/inspect/plugin-app.xml:
9114 * docs/plugins/inspect/plugin-bayer.xml:
9115 * docs/plugins/inspect/plugin-bz2.xml:
9116 * docs/plugins/inspect/plugin-cdaudio.xml:
9117 * docs/plugins/inspect/plugin-cdxaparse.xml:
9118 * docs/plugins/inspect/plugin-dtsdec.xml:
9119 * docs/plugins/inspect/plugin-dvb.xml:
9120 * docs/plugins/inspect/plugin-dvdspu.xml:
9121 * docs/plugins/inspect/plugin-faac.xml:
9122 * docs/plugins/inspect/plugin-faad.xml:
9123 * docs/plugins/inspect/plugin-fbdevsink.xml:
9124 * docs/plugins/inspect/plugin-festival.xml:
9125 * docs/plugins/inspect/plugin-filter.xml:
9126 * docs/plugins/inspect/plugin-flvdemux.xml:
9127 * docs/plugins/inspect/plugin-freeze.xml:
9128 * docs/plugins/inspect/plugin-gsm.xml:
9129 * docs/plugins/inspect/plugin-gstinterlace.xml:
9130 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
9131 * docs/plugins/inspect/plugin-h264parse.xml:
9132 * docs/plugins/inspect/plugin-interleave.xml:
9133 * docs/plugins/inspect/plugin-jack.xml:
9134 * docs/plugins/inspect/plugin-ladspa.xml:
9135 * docs/plugins/inspect/plugin-metadata.xml:
9136 * docs/plugins/inspect/plugin-mms.xml:
9137 * docs/plugins/inspect/plugin-modplug.xml:
9138 * docs/plugins/inspect/plugin-mpeg2enc.xml:
9139 * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
9140 * docs/plugins/inspect/plugin-mpegtsparse.xml:
9141 * docs/plugins/inspect/plugin-mpegvideoparse.xml:
9142 * docs/plugins/inspect/plugin-musepack.xml:
9143 * docs/plugins/inspect/plugin-musicbrainz.xml:
9144 * docs/plugins/inspect/plugin-mve.xml:
9145 * docs/plugins/inspect/plugin-mythtv.xml
9146 * docs/plugins/inspect/plugin-nas.xml:
9147 * docs/plugins/inspect/plugin-neon.xml:
9148 * docs/plugins/inspect/plugin-nsfdec.xml:
9149 * docs/plugins/inspect/plugin-nuvdemux.xml:
9150 * docs/plugins/inspect/plugin-oss4.xml
9151 * docs/plugins/inspect/plugin-rawparse.xml:
9152 * docs/plugins/inspect/plugin-real.xml:
9153 * docs/plugins/inspect/plugin-replaygain.xml:
9154 * docs/plugins/inspect/plugin-rfbsrc.xml:
9155 * docs/plugins/inspect/plugin-sdl.xml:
9156 * docs/plugins/inspect/plugin-sdp.xml:
9157 * docs/plugins/inspect/plugin-selector.xml:
9158 * docs/plugins/inspect/plugin-sndfile.xml:
9159 * docs/plugins/inspect/plugin-soundtouch.xml:
9160 * docs/plugins/inspect/plugin-spcdec.xml:
9161 * docs/plugins/inspect/plugin-speed.xml:
9162 * docs/plugins/inspect/plugin-speexresample.xml:
9163 * docs/plugins/inspect/plugin-stereo.xml:
9164 * docs/plugins/inspect/plugin-subenc.xml
9165 * docs/plugins/inspect/plugin-timidity.xml:
9166 * docs/plugins/inspect/plugin-tta.xml:
9167 * docs/plugins/inspect/plugin-vcdsrc.xml:
9168 * docs/plugins/inspect/plugin-videosignal.xml:
9169 * docs/plugins/inspect/plugin-vmnc.xml:
9170 * docs/plugins/inspect/plugin-wildmidi.xml:
9171 * docs/plugins/inspect/plugin-x264.xml:
9172 * docs/plugins/inspect/plugin-xvid.xml:
9173 * docs/plugins/inspect/plugin-y4menc.xml:
9174 * ext/amrwb/gstamrwbdec.c:
9175 * ext/amrwb/gstamrwbenc.c:
9176 * ext/amrwb/gstamrwbparse.c:
9177 * ext/dc1394/gstdc1394.c:
9178 * ext/directfb/dfbvideosink.c:
9179 * ext/ivorbis/vorbisdec.c:
9180 * ext/jack/gstjackaudiosink.c:
9181 * ext/mpeg2enc/gstmpeg2enc.cc:
9182 * ext/mplex/gstmplex.cc:
9183 * ext/musicbrainz/gsttrm.c:
9184 * ext/mythtv/gstmythtvsrc.c:
9185 * ext/theora/theoradec.c:
9186 * ext/timidity/gsttimidity.c:
9187 * ext/timidity/gstwildmidi.c:
9188 * gst-libs/gst/app/gstappsink.c:
9189 * gst/deinterlace/gstdeinterlace.c:
9190 * gst/dvdspu/gstdvdspu.c:
9191 * gst/festival/gstfestival.c:
9192 * gst/freeze/gstfreeze.c:
9193 * gst/interleave/deinterleave.c:
9194 * gst/interleave/interleave.c:
9195 * gst/modplug/gstmodplug.cc:
9196 * gst/nuvdemux/gstnuvdemux.c:
9197 Add missing elements to docs. Fix doc-markup: use convinience syntax
9198 for examples (produces valid docbook), add several refsec2 when we
9199 have several titles. Fix some types.
9201 2008-06-12 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
9203 examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
9204 Original commit message from CVS:
9205 * examples/app/.cvsignore:
9206 * examples/app/Makefile.am:
9207 * examples/app/appsink-src.c: (on_new_buffer_from_source),
9208 (on_source_message), (on_sink_message), (main):
9209 Add beefed up example app from bug #413418. It now also uses appsink
9210 instead of fakesink for more ultimate coolness.
9211 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
9212 (gst_app_src_init), (gst_app_src_set_property),
9213 (gst_app_src_get_property), (gst_app_src_unlock),
9214 (gst_app_src_unlock_stop), (gst_app_src_create),
9215 (gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
9216 (gst_app_src_end_of_stream):
9217 * gst-libs/gst/app/gstappsrc.h:
9218 Add block property to allow push based implementation to block when we
9219 fill up the appsrc queues.
9220 Emit the enough-data signal while releasing our lock.
9222 2008-06-12 14:50:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9224 examples/app/.cvsignore: Ignore more.
9225 Original commit message from CVS:
9226 * examples/app/.cvsignore:
9229 2008-06-12 14:49:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9231 Do not use short_description in section docs for elements. We extract them from element details and there will be war...
9232 Original commit message from CVS:
9233 * ext/dc1394/gstdc1394.c:
9234 * ext/ivorbis/vorbisdec.c:
9235 * ext/jack/gstjackaudiosink.c:
9236 * ext/metadata/gstmetadatademux.c:
9237 * ext/mythtv/gstmythtvsrc.c:
9238 * ext/theora/theoradec.c:
9239 * gst-libs/gst/app/gstappsink.c:
9240 * gst/bayer/gstbayer2rgb.c:
9241 * gst/deinterlace/gstdeinterlace.c:
9242 * gst/rawparse/gstaudioparse.c:
9243 * gst/rawparse/gstvideoparse.c:
9244 * gst/rtpmanager/gstrtpbin.c:
9245 * gst/rtpmanager/gstrtpclient.c:
9246 * gst/rtpmanager/gstrtpjitterbuffer.c:
9247 * gst/rtpmanager/gstrtpptdemux.c:
9248 * gst/rtpmanager/gstrtpsession.c:
9249 * gst/rtpmanager/gstrtpssrcdemux.c:
9250 * gst/selector/gstinputselector.c:
9251 * gst/selector/gstoutputselector.c:
9252 * gst/videosignal/gstvideoanalyse.c:
9253 * gst/videosignal/gstvideodetect.c:
9254 * gst/videosignal/gstvideomark.c:
9255 * sys/oss4/oss4-mixer.c:
9256 * sys/oss4/oss4-sink.c:
9257 * sys/oss4/oss4-source.c:
9258 Do not use short_description in section docs for elements. We extract
9259 them from element details and there will be warnings if they differ.
9260 Also fixing up the ChangeLog order.
9262 2008-06-11 21:17:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9264 configure.ac: 0.10.19.3 pre-release
9265 Original commit message from CVS:
9267 0.10.19.3 pre-release
9269 2008-06-11 20:13:00 +0000 David Schleef <ds@schleef.org>
9271 gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
9272 Original commit message from CVS:
9273 * gst-libs/gst/rtsp/gstrtspconnection.c:
9275 Patch By: David Schleef <ds@schleef.org>
9278 2008-06-11 09:35:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9280 ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
9281 Original commit message from CVS:
9282 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
9283 (gst_gio_base_src_create):
9284 * ext/gio/gstgiobasesrc.h:
9285 Try to read the requested number of bytes, even if the first
9286 read returns less than requested, until nothing is read anymore
9287 or we have the requested amount of bytes. This fixes playback of
9288 files via Samba as Samba only allows to read 64k at once.
9289 Implement a caching algorithm that makes sure that we read at
9290 least 4k of data every time. Some elements will try to read a few
9291 bytes, then seek, read again a few bytes and so on and this is
9292 painfully slow as every operation has to go over DBus if GVfs is
9294 Fixes bug #536849 and #536848.
9295 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
9296 (gst_gio_src_check_get_range):
9297 Override check_get_range() to blacklist http/https URIs
9298 and whitelist file URIs. More to be added on demand.
9300 2008-06-06 16:50:51 +0000 Wim Taymans <wim.taymans@gmail.com>
9302 examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
9303 Original commit message from CVS:
9304 * examples/app/Makefile.am:
9305 * examples/app/appsrc-ra.c: (feed_data), (seek_data),
9306 (found_source), (bus_message), (main):
9307 * examples/app/appsrc-seekable.c: (feed_data), (seek_data),
9308 (found_source), (bus_message), (main):
9309 * examples/app/appsrc-stream2.c: (feed_data), (found_source),
9310 (bus_message), (main):
9311 Added 3 more example application for using appsrc in random-access mode,
9312 pull-mode streaming and pull mode seekable.
9313 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
9314 (gst_app_src_start), (gst_app_src_do_get_size),
9315 (gst_app_src_create):
9316 * gst-libs/gst/app/gstappsrc.h:
9317 Make stream-type property writable.
9318 Unset flushing when starting so that we reuse appsrc.
9319 Inform basesrc about the configured size.
9320 Emit seek-data signal when we are going to a different offset in
9323 2008-06-06 14:19:54 +0000 Wim Taymans <wim.taymans@gmail.com>
9325 examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
9326 Original commit message from CVS:
9327 * examples/app/appsrc-stream.c: (found_source), (main):
9328 Use deep-notify until we can depend on a playbin2 with support for the
9331 2008-06-05 16:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
9333 examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
9334 Original commit message from CVS:
9335 * examples/app/.cvsignore:
9336 * examples/app/Makefile.am:
9337 * examples/app/appsrc-stream.c: (read_data), (start_feed),
9338 (stop_feed), (found_source), (bus_message), (main):
9339 Added an example on how to use appsrc in playbin in streaming mode from
9341 * examples/app/appsrc_ex.c: (main):
9342 Set pipeline to NULL to free queued buffers.
9343 * gst-libs/gst/app/gstapp-marshal.list:
9344 * gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
9345 (gst_app_src_class_init), (gst_app_src_init),
9346 (gst_app_src_flush_queued), (gst_app_src_dispose),
9347 (gst_app_src_set_property), (gst_app_src_get_property),
9348 (gst_app_src_unlock), (gst_app_src_unlock_stop),
9349 (gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
9350 (gst_app_src_check_get_range), (gst_app_src_do_seek),
9351 (gst_app_src_create), (gst_app_src_set_stream_type),
9352 (gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
9353 (gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
9354 (gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
9355 (gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
9356 (gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
9357 * gst-libs/gst/app/gstappsrc.h:
9358 Measure max queue size in bytes instead.
9359 Add support for 3 modes of operation, streaming, seekable and
9360 random-access, making basesrc handle the scheduling modes for each.
9361 Add appsrc:// uri handler so that automatic plugging can be done from
9362 playbin2 or uridecodebin, for example.
9363 Added support for custom segment formats.
9364 Add support for push and pull based operations from the application.
9365 Expand the methods so that errors can be detected.
9366 Flush the queued buffers on seeks and when shutting down.
9367 Add signals to inform the app that a seek must happen.
9369 2008-06-05 09:47:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9371 configure.ac: 0.10.19.2 pre-release
9372 Original commit message from CVS:
9374 0.10.19.2 pre-release
9376 2008-06-04 21:48:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9378 win32/common/: Add new API functions to the dll exports
9379 Original commit message from CVS:
9380 * win32/common/libgstrtsp.def:
9381 * win32/common/libgsttag.def:
9382 Add new API functions to the dll exports
9384 2008-06-04 17:42:38 +0000 Michael Smith <msmith@xiph.org>
9386 gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
9387 Original commit message from CVS:
9388 * gst/playback/gstplaybasebin.c:
9389 Disconnect signals from decodebins we created before we remove it from
9390 playbin, to avoid crashes if the decodebin is eventually disposed after
9391 the playbin itself (possible if the app takes a reference on the
9395 2008-06-04 17:12:40 +0000 Tim-Philipp Müller <tim@centricular.net>
9397 gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
9398 Original commit message from CVS:
9399 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
9400 (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
9401 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
9402 (h264_video_type_find), (mpeg_video_stream_type_find),
9403 (dv_type_find), (mmsh_type_find):
9404 Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
9405 copy caps for no good reason (this may be desirable to make it easier
9406 to detect leaks, but then it should probably be done for all caps
9407 in the typefinder somewhere).
9409 2008-06-04 16:06:49 +0000 Peter Kjellerstedt <pkj@axis.com>
9411 tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
9412 Original commit message from CVS:
9413 * tests/check/Makefile.am:
9414 Do not try to run the check tests for subparse unless it has been
9417 2008-06-04 16:00:26 +0000 Peter Kjellerstedt <pkj@axis.com>
9419 tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
9420 Original commit message from CVS:
9421 * tests/check/pipelines/streamheader.c: (buffer_probe_cb),
9422 (test_multifdsink_gdp_vorbisenc), (streamheader_suite):
9423 Do not try to run a test which requires vorbisenc unless we have
9426 2008-06-04 11:53:53 +0000 Peter Kjellerstedt <pkj@axis.com>
9428 gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
9429 Original commit message from CVS:
9430 * gst-libs/gst/rtsp/gstrtspconnection.c:
9431 (gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
9432 (gst_rtsp_connection_clear_auth_params),
9433 (gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
9434 * gst-libs/gst/rtsp/gstrtspconnection.h:
9435 Add a couple of missing argument guards.
9436 Add a way of setting the DSCP for an RTSP connection.
9437 Add an accessor method for the ip member of GstRTSPConnection as all
9438 members are supposed to be private.
9440 2008-06-04 11:33:23 +0000 Peter Kjellerstedt <pkj@axis.com>
9442 gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
9443 Original commit message from CVS:
9444 * gst/tcp/gstmultifdsink.c: (setup_dscp_client):
9445 Fixed accidental use of IPv4 options for all IPv6 addresses.
9447 2008-06-04 10:18:42 +0000 Tim-Philipp Müller <tim@centricular.net>
9449 gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
9450 Original commit message from CVS:
9451 * gst-libs/gst/interfaces/mixertrack.h:
9452 Document mixer track flags.
9454 2008-06-04 05:58:38 +0000 Antoine Tremblay <hexa00@gmail.com>
9456 gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
9457 Original commit message from CVS:
9458 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
9459 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
9460 Don't set caps on the buffers that contain a copy of the buffer
9461 including the caps of them resulting in an always increasing refcount
9462 of the caps and insanely large caps. Instead include a buffer without
9463 caps in the new caps. Fixes bug #536475.
9465 2008-06-04 05:44:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9467 gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
9468 Original commit message from CVS:
9469 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
9470 Transform a given PAR to a range on the struct with the generic
9471 height/width instead of the struct with the possibly restricted
9474 2008-06-04 04:24:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9476 gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
9477 Original commit message from CVS:
9478 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
9479 Prefer the given format if it contains something stricter than [1,MAX]
9480 for height or width and only put a structure that requires rescaling
9481 as second. This makes it possible to use videoscale in pipelines where
9482 the source can actually produce the wanted height/width but usually
9483 selects a different one from the requested.
9485 2008-06-03 20:01:58 +0000 John Millikin <jmillikin@gmail.com>
9487 gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
9488 Original commit message from CVS:
9489 Based on patch by: John Millikin <jmillikin gmail com>
9490 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
9491 (gst_vorbis_tag_add_coverart):
9492 Retrieve COVERART tags from vorbis comments (#512333)
9494 2008-06-03 19:44:48 +0000 Tim-Philipp Müller <tim@centricular.net>
9496 gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
9497 Original commit message from CVS:
9498 * gst-libs/gst/tag/tag.h:
9499 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
9500 Don't forget to add new enum value here too (should probably use
9501 glib-mkenums here...).
9503 2008-06-03 19:29:06 +0000 Tim-Philipp Müller <tim@centricular.net>
9505 gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
9506 Original commit message from CVS:
9507 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
9508 * gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
9509 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
9510 (gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
9511 (gst_tag_image_data_to_image_buffer):
9512 Add two utility functions to avoid code duplication (#512333):
9513 API: add gst_tag_image_data_to_image_buffer()
9514 API: add gst_tag_list_add_id3_image()
9516 2008-06-03 08:54:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9518 win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
9519 Original commit message from CVS:
9520 * win32/common/libgstaudio.def:
9521 Add gst_audio_check_channel_positions() to the exported symbols.
9523 2008-06-03 08:48:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9525 API: Make gst_audio_check_channel_positions() public.
9526 Original commit message from CVS:
9527 * docs/libs/gst-plugins-base-libs-sections.txt:
9528 * gst-libs/gst/audio/multichannel.c:
9529 (gst_audio_check_channel_positions):
9530 * gst-libs/gst/audio/multichannel.h:
9531 API: Make gst_audio_check_channel_positions() public.
9532 * tests/check/libs/audio.c: (GST_START_TEST):
9533 Add some simple checks for gst_audio_check_channel_positions().
9535 2008-06-02 20:09:14 +0000 Tim-Philipp Müller <tim@centricular.net>
9537 sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.
9538 Original commit message from CVS:
9539 * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
9540 minrange and maxrange are scaled according to the frequency
9543 2008-06-02 18:37:02 +0000 Tim-Philipp Müller <tim@centricular.net>
9545 ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
9546 Original commit message from CVS:
9547 * ext/pango/Makefile.am:
9548 * ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
9549 (gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
9550 Use gstvideo functions to calculate strides and plane offsets. Fixes
9551 rendering issue ('ghost' images of the text on the chroma planes)
9552 with widths or heights that are not multiples of 8 (#506659 and
9553 probably also #485729).
9554 * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
9556 Test with odd height/width too.
9558 2008-06-02 12:20:35 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9560 gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
9561 Original commit message from CVS:
9562 * gst/adder/gstadder.c: (gst_adder_query_duration),
9563 (gst_adder_query_latency):
9564 When using gst_element_iterate_pads() one has to unref every pad
9567 2008-05-31 19:57:57 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9569 gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
9570 Original commit message from CVS:
9571 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9572 (gst_base_audio_src_class_init):
9573 Add a gtk-doc chunk for the new properties to have a Since: indication.
9575 2008-05-31 19:50:59 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9578 ChangeLog surgery, mark API change
9579 Original commit message from CVS:
9580 ChangeLog surgery, mark API change
9582 2008-05-31 18:10:47 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9584 gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
9585 Original commit message from CVS:
9586 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9587 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
9588 (gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
9589 (gst_base_audio_src_change_state):
9590 Provide readable actual-buffer-time and actual-latency-time properties
9591 that reflect the configured ringbuffer values. Fixes #524724.
9593 2008-05-30 15:29:20 +0000 Wim Taymans <wim.taymans@gmail.com>
9595 gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
9596 Original commit message from CVS:
9597 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
9598 (gst_basertppayload_change_state):
9599 Simply converting the running time into an RTP timestamp by scaling it
9600 based on the clock-rate is good enough for making an RTP timestamp. This
9601 has the added benefit that we can later on expose a property with the
9602 RTP timestamp of running time 0, as is needed for RTSP servers to
9603 generate the response of the PLAY request.
9605 2008-05-30 08:42:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9607 gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
9608 Original commit message from CVS:
9609 * gst/audioconvert/gstaudioconvert.c:
9610 (structure_has_fixed_channel_positions),
9611 (gst_audio_convert_transform_caps):
9612 Allow up to 11 positioned channels now that audioconvert can handle
9613 this but add no default positions for > 8 channels.
9614 * tests/check/elements/audioconvert.c: (GST_START_TEST):
9615 Add some unit tests for the above change: Test conversion of
9616 11 positioned channels to stereo and the other way around, test
9617 conversion of 15 unpositioned channels in different ways.
9619 2008-05-29 19:45:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9621 win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
9622 Original commit message from CVS:
9623 * win32/common/libgstaudio.def:
9624 Add gst_audio_clock_reset to the list of exported symbols.
9626 2008-05-29 19:37:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9628 tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
9629 Original commit message from CVS:
9630 * tests/check/elements/vorbisdec.c: (vorbisdec_suite):
9631 Remove wrong_channels_identification_header unit test as we now
9632 support 7 (and more channels).
9634 2008-05-29 12:17:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9636 gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
9637 Original commit message from CVS:
9638 * gst/audioconvert/gstchannelmix.c:
9639 (gst_channel_mix_fill_one_other):
9640 If mixing left or right to center (or the other way around) only take
9641 the complete value if we don't already have the original position in
9644 2008-05-29 11:34:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9646 gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
9647 Original commit message from CVS:
9648 * gst-libs/gst/audio/multichannel.c:
9649 (gst_audio_check_channel_positions),
9650 (gst_audio_set_structure_channel_positions_list),
9651 (gst_audio_fixate_channel_positions):
9652 Allow rear center together with rear left/right and other previously
9653 conflicting channel positions. The reason why they weren't allowed
9654 was the channel mixing implementation in audioconvert.
9655 Also take this into account when fixing channel layouts.
9656 Allow setting channel positions for 1/2 channels when using
9657 gst_audio_set_structure_channel_position().
9658 * gst/audioconvert/gstchannelmix.c:
9659 (gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
9660 (gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
9661 (gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
9662 Major rewrite of the channel mixing.
9663 We now allow previously conflicting channel positions to appear
9664 together (rear center and rear left/right for example).
9666 Rework the way channels are mixed together to take more possible
9667 channel positions into account, properly mix from/to side channels
9668 and don't assume that either center, left&right or nothing of a
9669 specific position is available anymore.
9670 * tests/check/elements/audioconvert.c: (GST_START_TEST):
9671 Adjust unit tests with non-standard 1/2 channel layouts to the more
9672 correct new behaviour.
9673 Add a unit test for 5.1->Stereo downmixing.
9675 2008-05-29 07:02:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9677 ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
9678 Original commit message from CVS:
9679 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
9680 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
9681 Add sane defaults for the 7 and 8 channel layouts as those are
9682 undefined in the Vorbis spec. Use NONE channel layouts when decoding
9683 more than 8 channels instead of erroring out. Fixes bug #535356.
9685 2008-05-28 16:10:20 +0000 Wim Taymans <wim.taymans@gmail.com>
9687 Add theoraparse to the docs and fix some docs.
9688 Original commit message from CVS:
9689 * docs/plugins/Makefile.am:
9690 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
9691 * docs/plugins/gst-plugins-base-plugins-sections.txt:
9692 * ext/theora/theoraparse.c:
9693 Add theoraparse to the docs and fix some docs.
9695 2008-05-28 15:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
9697 gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
9698 Original commit message from CVS:
9699 * gst-libs/gst/cdda/gstcddabasesrc.c:
9700 (gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
9701 Fix EOS condition and track addition check, the track.end sector is
9702 included in the track. Fixes #533265.
9704 2008-05-28 14:49:24 +0000 Mark Nauwelaerts <manauw@skynet.be>
9706 gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
9707 Original commit message from CVS:
9708 Patch by: Mark Nauwelaerts <manauw at skynet be>
9709 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
9710 (gst_video_rate_flush_prev), (gst_video_rate_event),
9711 (gst_video_rate_chain):
9712 * gst/videorate/gstvideorate.h:
9713 React (more) to NEWSEGMENT
9714 Small adjustment in timestamp calculation to prevent mismatches
9717 2008-05-28 11:31:44 +0000 Tim-Philipp Müller <tim@centricular.net>
9719 tests/examples/seek/seek.c: Initialise error to NULL as we should.
9720 Original commit message from CVS:
9721 * tests/examples/seek/seek.c: (make_parselaunch_pipeline):
9722 Initialise error to NULL as we should.
9724 2008-05-28 08:14:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9726 gst/adder/gstadder.c: Implement latency query.
9727 Original commit message from CVS:
9728 * gst/adder/gstadder.c: (gst_adder_query_duration),
9729 (gst_adder_query_latency), (gst_adder_query):
9730 Implement latency query.
9732 2008-05-27 18:10:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9734 gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
9735 Original commit message from CVS:
9736 * gst/adder/gstadder.c: (gst_adder_query_duration):
9737 Correctly resync the iterator if gst_iterator_next() returns
9738 GST_ITERATOR_RESYNC.
9740 2008-05-27 17:14:07 +0000 Tim-Philipp Müller <tim@centricular.net>
9742 win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
9743 Original commit message from CVS:
9744 * win32/vs6/libgstpbutils.dsp:
9745 Add pbutils-enumtypes.c to sources (#518037).
9747 2008-05-27 16:20:17 +0000 Wim Taymans <wim.taymans@gmail.com>
9749 gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
9750 Original commit message from CVS:
9751 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
9752 (gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
9753 * gst-libs/gst/audio/gstaudioclock.h:
9754 Add method to inform the clock that the time starts from 0 again. We use
9755 this info to calculate a clock offset so that the time we report in
9756 internal_time is monotonically increasing, as required by the clock base
9757 class. Fixes #521761.
9758 API: GstAudioClock::gst_audio_clock_reset()
9759 * gst-libs/gst/audio/gstbaseaudiosink.c:
9760 (gst_base_audio_sink_skew_slaving),
9761 (gst_base_audio_sink_change_state):
9762 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9763 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
9764 Reset reported time when we (re)create the ringbuffer.
9766 2008-05-27 16:11:32 +0000 Tim-Philipp Müller <tim@centricular.net>
9768 ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
9769 Original commit message from CVS:
9770 * ext/alsa/gstalsamixertrack.c:
9771 (gst_alsa_mixer_track_update_alsa_capabilities):
9772 Make sure playback volumes aren't accidentally overwritten by
9773 capture volumes if an alsa mixer track has both playback and
9774 capture capabilities: we create two GstMixerTracks in that
9775 case, so make sure we query only the alsa capabilities that
9776 refer to the type of GstMixerTrack we created from the dual
9777 capability alsa element. Should fix issues with Audigy2 sound
9780 2008-05-27 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.net>
9782 tests/check/pipelines/oggmux.c: Don't use deprecated function.
9783 Original commit message from CVS:
9784 * tests/check/pipelines/oggmux.c: (test_pipeline):
9785 Don't use deprecated function.
9787 2008-05-27 10:35:55 +0000 Wim Taymans <wim.taymans@gmail.com>
9789 gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
9790 Original commit message from CVS:
9791 * gst/playback/gstdecodebin2.c:
9792 (gst_decode_group_control_source_pad), (gst_decode_group_expose):
9793 Check for NULL cases and log them, creating ghostpads can, for example,
9794 fail when the pad returns wrong caps.
9795 * gst/playback/gstplaybin2.c: (perform_eos):
9796 When pushing out the EOS event, collect the return value and warn when
9799 2008-05-26 17:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
9801 gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
9802 Original commit message from CVS:
9803 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
9804 (gst_riff_create_video_template_caps):
9805 Add support for DVCPRO.
9807 2008-05-26 10:29:20 +0000 Tim-Philipp Müller <tim@centricular.net>
9809 gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
9810 Original commit message from CVS:
9811 * gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
9812 Change default scaling method from nearest-neighbour to bilinear.
9814 2008-05-26 10:26:00 +0000 Tim-Philipp Müller <tim@centricular.net>
9816 tests/check/libs/video.c: More checks.
9817 Original commit message from CVS:
9818 * tests/check/libs/video.c:
9821 2008-05-25 20:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
9823 Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
9824 Original commit message from CVS:
9825 * gst/subparse/gstsubparse.c: (parser_state_init),
9826 (gst_sub_parse_format_autodetect), (handle_buffer):
9827 * gst/subparse/gstsubparse.h:
9828 * tests/check/elements/subparse.c: (test_tmplayer_style3b):
9829 Limit duration to a maximum of five seconds for tmplayer format where
9830 we can guess the duration only from the timestamp of the next line of
9831 text. We don't want to show a text for eternities just because nothing
9832 else is being said for a while.
9834 2008-05-23 14:14:28 +0000 Wim Taymans <wim.taymans@gmail.com>
9836 gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
9837 Original commit message from CVS:
9838 * gst-libs/gst/rtp/gstbasertpdepayload.c:
9839 (gst_base_rtp_depayload_chain),
9840 (gst_base_rtp_depayload_handle_sink_event),
9841 (gst_base_rtp_depayload_push_full),
9842 (gst_base_rtp_depayload_change_state):
9843 Check sequence numbers, mark input buffers with a discont flag for the
9844 subclass when we detected a gap, drop duplicate buffers. We do this
9845 because one can use the element without a jitterbuffer in front and we
9846 don't want to feed the subclasses invalid or reordered data.
9847 Do an error when the subclass did not provide a process function instead
9849 Some other small cleanups.
9851 2008-05-22 22:35:40 +0000 Tim-Philipp Müller <tim@centricular.net>
9853 gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
9854 Original commit message from CVS:
9855 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
9856 May just as well use the precalculated uvstride here.
9858 2008-05-22 22:09:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9860 Add some documentation comments, and some new headers to be scanned.
9861 Original commit message from CVS:
9862 * docs/plugins/Makefile.am:
9863 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
9864 * docs/plugins/gst-plugins-base-plugins-sections.txt:
9865 * docs/plugins/gst-plugins-base-plugins.args:
9866 * docs/plugins/gst-plugins-base-plugins.hierarchy:
9867 * docs/plugins/gst-plugins-base-plugins.interfaces:
9868 * docs/plugins/gst-plugins-base-plugins.prerequisites:
9869 * docs/plugins/inspect/plugin-adder.xml:
9870 * docs/plugins/inspect/plugin-alsa.xml:
9871 * docs/plugins/inspect/plugin-audioconvert.xml:
9872 * docs/plugins/inspect/plugin-audiorate.xml:
9873 * docs/plugins/inspect/plugin-audioresample.xml:
9874 * docs/plugins/inspect/plugin-audiotestsrc.xml:
9875 * docs/plugins/inspect/plugin-cdparanoia.xml:
9876 * docs/plugins/inspect/plugin-decodebin.xml:
9877 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
9878 * docs/plugins/inspect/plugin-gdp.xml:
9879 * docs/plugins/inspect/plugin-gio.xml:
9880 * docs/plugins/inspect/plugin-gnomevfs.xml:
9881 * docs/plugins/inspect/plugin-libvisual.xml:
9882 * docs/plugins/inspect/plugin-ogg.xml:
9883 * docs/plugins/inspect/plugin-pango.xml:
9884 * docs/plugins/inspect/plugin-playback.xml:
9885 * docs/plugins/inspect/plugin-queue2.xml:
9886 * docs/plugins/inspect/plugin-subparse.xml:
9887 * docs/plugins/inspect/plugin-tcp.xml:
9888 * docs/plugins/inspect/plugin-theora.xml:
9889 * docs/plugins/inspect/plugin-typefindfunctions.xml:
9890 * docs/plugins/inspect/plugin-uridecodebin.xml:
9891 * docs/plugins/inspect/plugin-video4linux.xml:
9892 * docs/plugins/inspect/plugin-videorate.xml:
9893 * docs/plugins/inspect/plugin-videoscale.xml:
9894 * docs/plugins/inspect/plugin-videotestsrc.xml:
9895 * docs/plugins/inspect/plugin-volume.xml:
9896 * docs/plugins/inspect/plugin-vorbis.xml:
9897 * docs/plugins/inspect/plugin-ximagesink.xml:
9898 * docs/plugins/inspect/plugin-xvimagesink.xml:
9899 * ext/cdparanoia/gstcdparanoiasrc.c:
9900 * ext/ogg/gstoggdemux.c:
9901 * ext/ogg/gstoggdemux.h:
9902 * ext/ogg/gstoggmux.c:
9903 * ext/ogg/gstoggmux.h:
9904 * gst/audioconvert/audioconvert.c:
9905 * gst/audioconvert/audioconvert.h:
9906 * gst/audioconvert/gstaudioconvert.h:
9907 * gst/gdp/gstgdpdepay.h:
9908 * gst/gdp/gstgdppay.h:
9909 * gst/playback/gstdecodebin.c:
9910 * gst/playback/gstdecodebin2.c:
9911 * gst/playback/gstplaybin.c:
9912 * gst/playback/gstplaybin2.c:
9913 * gst/playback/gsturidecodebin.c:
9914 * gst/tcp/gstmultifdsink.c:
9915 * gst/tcp/gstmultifdsink.h:
9917 Add some documentation comments, and some new headers to be scanned.
9918 Rename some internal enum declarations (audioconvert's DitherType and
9919 NoiseShapingType, GstUnitType from the TCP elements) to match the
9920 documented GObject type names so that the docs pick them up.
9921 Name the playbin2 docs markups properly so they get picked up. They'll
9922 need renaming back when/if playbin2 becomes playbin.
9923 100% symbol coverage for the plugin docs, booya.
9925 2008-05-22 18:30:15 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
9927 gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
9928 Original commit message from CVS:
9929 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
9930 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
9931 Fix generation of NV12/NV21 frames. Fixes bug #532454.
9933 2008-05-22 11:59:33 +0000 Sjoerd Simons <sjoerd@luon.net>
9935 gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
9936 Original commit message from CVS:
9937 Patch by: Sjoerd Simons <sjoerd at luon dot net>
9938 * gst/playback/gstdecodebin.c: (remove_fakesink):
9939 Lock the fakesink before setting the state to NULL and removing it from
9940 the bin so that a concurrent state change cannot interfere.
9943 2008-05-21 17:09:42 +0000 Felipe Contreras <felipe.contreras@nokia.com>
9945 docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
9946 Original commit message from CVS:
9948 Fix installing plugin documentation when gtk-doc is disabled.
9950 2008-05-21 17:01:16 +0000 Felipe Contreras <felipe.contreras@nokia.com>
9952 gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
9953 Original commit message from CVS:
9954 * gst-libs/gst/rtsp/Makefile.am:
9955 Distribute, don't install md5.h
9957 2008-05-21 16:47:58 +0000 Julien Moutte <julien@moutte.net>
9959 gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
9960 Original commit message from CVS:
9961 2008-05-21 Julien Moutte <julien@fluendo.com>
9962 * gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
9963 instead of SOL_IP, works on more platforms.
9964 * gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
9967 2008-05-21 16:44:15 +0000 Wim Taymans <wim.taymans@gmail.com>
9969 Some debug and comment fixes.
9970 Original commit message from CVS:
9971 * ext/vorbis/vorbisdec.c:
9972 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
9973 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
9974 Some debug and comment fixes.
9975 * tests/examples/dynamic/addstream.c: (main):
9978 2008-05-21 16:36:50 +0000 Wim Taymans <wim.taymans@gmail.com>
9980 Don't use bad gst_element_get_pad().
9981 Original commit message from CVS:
9982 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
9983 * gst/playback/decodetest.c: (new_decoded_pad_cb):
9984 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
9985 (try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
9986 (cleanup_decodebin):
9987 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
9988 (connect_element), (gst_decode_group_control_demuxer_pad):
9989 * gst/playback/gstplaybasebin.c: (queue_remove_probe),
9990 (queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
9992 * gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
9993 (gst_play_bin_set_property), (handoff), (gen_video_element),
9994 (gen_text_element), (gen_audio_element), (gen_vis_element),
9995 (remove_sinks), (add_sink), (setup_sinks):
9996 * gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
9997 * gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
9998 (gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
9999 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
10000 (gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
10001 (gen_video_chain), (gen_text_chain), (gen_audio_chain),
10002 (gen_vis_chain), (gst_play_sink_reconfigure),
10003 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
10004 (gst_play_sink_request_pad):
10005 * gst/playback/gsturidecodebin.c: (type_found), (setup_source):
10006 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
10008 * gst/playback/test6.c: (new_decoded_pad_cb):
10009 * tests/check/elements/audioconvert.c: (GST_START_TEST):
10010 * tests/check/elements/audiorate.c: (test_injector_chain),
10011 (do_perfect_stream_test):
10012 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
10013 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
10014 * tests/check/elements/gnomevfssink.c:
10015 * tests/check/elements/textoverlay.c:
10016 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
10017 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
10018 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
10019 * tests/check/pipelines/oggmux.c: (test_pipeline):
10020 * tests/check/pipelines/streamheader.c: (GST_START_TEST):
10021 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
10022 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
10023 * tests/examples/seek/scrubby.c: (make_wav_pipeline):
10024 * tests/examples/seek/seek.c: (make_mod_pipeline),
10025 (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
10026 (make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
10027 (make_theora_pipeline), (make_vorbis_theora_pipeline),
10028 (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
10029 (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
10030 (update_fill), (msg_buffering):
10031 Don't use bad gst_element_get_pad().
10033 2008-05-21 14:35:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10035 gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
10036 Original commit message from CVS:
10037 * gst-libs/gst/riff/riff-media.c:
10038 Fix wrong method name in docs. Fix calculation of strf fields for
10040 * gst-libs/gst/riff/riff-read.c:
10041 Whitespace fix and removing double ';'.
10043 2008-05-21 11:52:30 +0000 Wim Taymans <wim.taymans@gmail.com>
10045 docs/design/part-playbin2.txt: Add some leftover doc.
10046 Original commit message from CVS:
10047 * docs/design/part-playbin2.txt:
10048 Add some leftover doc.
10050 2008-05-21 11:36:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10052 gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
10053 Original commit message from CVS:
10054 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
10055 Fix copy & paste error in last commit.
10057 2008-05-21 11:30:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10059 gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
10060 Original commit message from CVS:
10061 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
10062 Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
10063 other channel positions when source has SIDE channels and dest doesn't
10064 or the other way around.
10066 2008-05-21 11:29:25 +0000 Henrik Eriksson <henriken@axis.com>
10068 gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
10069 Original commit message from CVS:
10070 Patch by: Henrik Eriksson <henriken at axis dot com>
10071 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
10072 (gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
10073 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
10074 (gst_multi_fd_sink_get_property):
10075 * gst/tcp/gstmultifdsink.h:
10076 Add support for DSCP QOS. Fixes #469933.
10078 2008-05-21 07:46:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10080 tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
10081 Original commit message from CVS:
10082 * tests/check/elements/audioconvert.c: (GST_START_TEST):
10083 Add another test that checks if conversion between standard 1 and 2
10084 channel layouts with and without positions set is working.
10086 2008-05-21 07:39:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10088 gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
10089 Original commit message from CVS:
10090 * gst-libs/gst/audio/multichannel.c:
10091 (gst_audio_check_channel_positions):
10092 Allow non-standard 2 channel layouts.
10093 * tests/check/elements/audioconvert.c: (GST_START_TEST):
10094 Add some tests for converting and remapping non-standard 1 and 2
10097 2008-05-21 07:28:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10099 gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
10100 Original commit message from CVS:
10101 * gst/audioconvert/gstchannelmix.c:
10102 (gst_channel_mix_fill_normalize):
10103 Prevent division by zero if the channel mix matrix contains only
10106 2008-05-21 06:45:22 +0000 Antoine Tremblay <hexa00@gmail.com>
10108 gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
10109 Original commit message from CVS:
10110 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
10111 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
10112 Close a buffer memory leak. Fixes bug #534071.
10114 2008-05-21 06:39:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10116 gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
10117 Original commit message from CVS:
10118 * gst-libs/gst/rtsp/gstrtsptransport.h:
10119 Make the GstRTSPTransport struct members public as there are no
10120 setters/getters and it's supposed to be changed directly.
10123 2008-05-21 05:48:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10125 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...
10126 Original commit message from CVS:
10127 * gst/adder/gstadder.c:
10128 Adder also doesn't support audio/x-raw-int with width!=depth so don't
10129 claim this on the pad template caps.
10131 2008-05-20 16:26:53 +0000 Wim Taymans <wim.taymans@gmail.com>
10133 gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
10134 Original commit message from CVS:
10135 * gst-libs/gst/audio/gstbaseaudiosink.c:
10136 (gst_base_audio_sink_sync_latency):
10137 We can only use our optimal calibration if we prerolled before the
10140 2008-05-20 14:35:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10142 configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
10143 Original commit message from CVS:
10145 Require core CVS for GstBaseSrc buffer caps setting magic.
10147 2008-05-20 12:26:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10149 gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
10150 Original commit message from CVS:
10151 * gst/audioconvert/gstaudioconvert.c:
10152 (gst_audio_convert_fixate_channels):
10153 Fix logic in last commit.
10155 2008-05-20 12:15:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10157 gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
10158 Original commit message from CVS:
10159 * gst/audioconvert/gstaudioconvert.c:
10160 (gst_audio_convert_fixate_channels):
10161 Passthrough the channel positions if the number of output channels is
10162 the same as the number of input channels, the input had a channel
10163 layout and downstream requests no special one. We did this already for
10164 > 2 channels but now it's also done for 1 channel. Fixes bug #533617.
10166 2008-05-20 11:13:27 +0000 Wim Taymans <wim.taymans@gmail.com>
10168 ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
10169 Original commit message from CVS:
10170 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
10171 (gst_gnome_vfs_src_finalize),
10172 (gst_gnome_vfs_src_received_headers_callback),
10173 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
10174 * ext/gnomevfs/gstgnomevfssrc.h:
10175 Set the ICY caps on the srcpad from where they get picked up by the base
10176 class now and set on the outgoing buffers.
10177 * gst-libs/gst/audio/gstbaseaudiosrc.c:
10178 (gst_base_audio_src_create):
10179 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
10180 BaseSrc now sets the caps on outgoing buffers automatically.
10182 2008-05-20 11:09:06 +0000 Wim Taymans <wim.taymans@gmail.com>
10184 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
10185 Original commit message from CVS:
10186 * gst-libs/gst/audio/gstbaseaudiosink.c:
10187 (gst_base_audio_sink_resample_slaving),
10188 (gst_base_audio_sink_skew_slaving),
10189 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
10190 (gst_base_audio_sink_async_play),
10191 (gst_base_audio_sink_change_state):
10192 Change the way in which the ringbuffer is started when dealing with a
10193 slaved clock and latency. We now sync to the clock until we reach
10194 upstream latency before starting the ringbuffer. This has the effect
10195 that we can accurately align the master and slave clocks and let the
10196 rate correction code take care of the initial drift or rounding errors
10197 instead of leaving them uncorrected with the old approach.
10199 2008-05-20 08:12:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10201 gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
10202 Original commit message from CVS:
10203 * gst/audioconvert/gstaudioconvert.c:
10204 (gst_audio_convert_fixate_channels):
10205 Correctly set the default channel positions when converting to 8
10208 2008-05-19 16:13:25 +0000 Tim-Philipp Müller <tim@centricular.net>
10210 configure.ac: Error out if we don't have the required version of core.
10211 Original commit message from CVS:
10213 Error out if we don't have the required version of core.
10215 2008-05-19 15:59:40 +0000 Tim-Philipp Müller <tim@centricular.net>
10217 gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
10218 Original commit message from CVS:
10219 * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
10220 Use data scan helper in aac typefinder and stop scanning
10221 for headers when we've found a type. Also fix potential invalid
10222 memory access when calculating the frame length.
10224 2008-05-19 14:09:08 +0000 Tim-Philipp Müller <tim@centricular.net>
10226 gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
10227 Original commit message from CVS:
10228 * gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
10229 (mpeg_sys_is_valid_pack):
10230 Don't modify scan context when we return FALSE in ensure_data, so
10231 it's possible to continue scanning, and we don't end up with a NULL
10232 data pointer and a positive size, which might bite us the next time
10233 we're called. Small constification.
10235 2008-05-16 21:12:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10237 gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
10238 Original commit message from CVS:
10239 * gst/adder/gstadder.c:
10240 Adder doesn't support 24 bit samples so don't claim it supports them
10241 in the pad template caps.
10243 2008-05-14 20:28:02 +0000 Wim Taymans <wim.taymans@gmail.com>
10245 gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
10246 Original commit message from CVS:
10247 * gst-libs/gst/rtp/gstbasertpdepayload.c:
10248 (gst_base_rtp_depayload_chain):
10249 Validate the RTP packet before further processing it. It's just too
10250 dangerous to accept random packets and people are not forced to use a
10251 jitterbuffer or session manager to filter out the bad packets.
10252 * gst-libs/gst/rtp/gstrtpbuffer.c:
10253 (gst_rtp_buffer_set_extension_data),
10254 (gst_rtp_buffer_get_payload_subbuffer):
10256 When setting extension data in a buffer that is too small, we fail and
10257 we should not set the extension bit.
10258 Change GST_WARNINGS into g_warning because they really are
10259 programming errors.
10260 * tests/check/libs/rtp.c: (GST_START_TEST):
10261 Catch the g_warnings now in the unit tests and that fact that failing to
10262 set extension data left the extension bit untouched.
10264 2008-05-14 13:57:41 +0000 Tim-Philipp Müller <tim@centricular.net>
10266 gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
10267 Original commit message from CVS:
10268 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
10269 Revert previous change which made basetransform handle buffer_alloc
10270 and which breaks things badly in the non-passthrough case since it
10271 returned buffers with a different (ie. sometimes smaller) size than
10272 the size requested.
10274 2008-05-14 13:43:12 +0000 Bernard B <b-gnome@largestprime.net>
10276 gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
10277 Original commit message from CVS:
10278 Patch by: Bernard B <b-gnome at largestprime dot net>
10279 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
10280 Fix seqnum compare function for bordercase values and fix the docs
10281 again. Fixes #533075.
10282 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
10283 Add a testcase for seqnum compare function.
10285 2008-05-14 10:58:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10287 gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
10288 Original commit message from CVS:
10289 * gst/adder/gstadder.c: (gst_adder_setcaps),
10290 (gst_adder_class_init):
10291 Correctly declare the supported endianness on the pad templates
10292 and check for correct endianness in the set caps function. Adder
10293 only supports native endianness.
10294 Also use gst_element_class_set_details_simple().
10296 2008-05-14 09:12:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10298 sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.
10299 Original commit message from CVS:
10300 * sys/xvimage/xvimagesink.c:
10301 Better debug logging in port value handling. Merging separate port
10302 value loops into one.
10304 2008-05-13 16:02:19 +0000 Hannes Bistry <hannesb@gmx.de>
10306 gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
10307 Original commit message from CVS:
10308 Patch by: Hannes Bistry <hannesb at gmx dot de>
10309 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
10310 * gst/tcp/gsttcpserversink.c:
10311 (gst_tcp_server_sink_handle_server_read),
10312 (gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
10313 Fix regression in clientsrc because we did not add the fd to the poll
10314 set anymore. Fixes #532364.
10315 Do some cleanups here and there.
10317 2008-05-13 13:04:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10319 gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
10320 Original commit message from CVS:
10321 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
10322 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
10323 * gst/playback/gstplay-marshal.list:
10324 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
10325 Use correct marshallers. GstCaps are a boxed type and no GObject
10328 2008-05-13 11:37:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10330 win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.
10331 Original commit message from CVS:
10332 * win32/common/libgstrtsp.def:
10333 Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
10336 2008-05-13 10:59:49 +0000 Sjoerd Simons <sjoerd@luon.net>
10338 tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
10339 Original commit message from CVS:
10340 Patch by: Sjoerd Simons <sjoerd at luon dot net>
10341 * tests/check/elements/audioresample.c:
10342 (live_switch_alloc_only_48000), (live_switch_get_sink_caps),
10343 (live_switch_push), (GST_START_TEST):
10344 Add unit test for the latest basetransform negotiation changes.
10347 2008-05-13 09:14:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10349 gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
10350 Original commit message from CVS:
10351 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
10352 Fix nv12<->nv21 conversion if stride is larger than width.
10354 2008-05-13 07:28:21 +0000 j^ <j@oil21.org>
10356 ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...
10357 Original commit message from CVS:
10358 Patch by: j^ <j at oil21 dot org>
10359 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
10360 (gst_ogg_pad_parse_skeleton_fisbone):
10361 * ext/ogg/gstoggdemux.h:
10362 Parse presentation time from skeleton streams and use it as offset
10363 for the timestamps. Fixes bug #530068.
10365 2008-05-12 08:45:11 +0000 Wim Taymans <wim.taymans@gmail.com>
10367 gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
10368 Original commit message from CVS:
10369 * gst-libs/gst/audio/gstbaseaudiosink.c:
10370 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
10371 Revert previous patch that attempted to more accurately calculate the
10372 initial offset between master and slave clock. The best thing we can do
10373 in general is take the time of both clocks as the diff since we don't
10374 know when the actual preroll happened.
10376 2008-05-11 19:52:59 +0000 Tim-Philipp Müller <tim@centricular.net>
10378 gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
10379 Original commit message from CVS:
10380 * gst-libs/gst/pbutils/install-plugins.c:
10381 Fix docs: type and missing word.
10383 2008-05-10 20:16:21 +0000 Tim-Philipp Müller <tim@centricular.net>
10385 gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
10386 Original commit message from CVS:
10387 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
10388 Don't do lots of 4-byte peeks, but use the 'new' data scan helper
10389 for this instead; don't check if we've found enough markers after
10390 each and every step, it's enough to do that only if we've actually
10391 found a new marker.
10392 Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
10394 2008-05-10 18:19:17 +0000 Tim-Philipp Müller <tim@centricular.net>
10396 gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
10397 Original commit message from CVS:
10398 * gst/typefind/gsttypefindfunctions.c:
10399 (DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
10400 (data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
10401 (mpeg_video_stream_type_find):
10402 Move scan helper thingy to the beginning of the file so we can use
10403 it in other typefind functions. Rename it to something more
10404 generic. Also improve handling of things towards the end of the
10405 typefind data: peek as much as we can if we know the size of the
10406 data, rather than just min_size.
10408 2008-05-09 21:42:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10410 Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
10411 Original commit message from CVS:
10412 * docs/libs/gst-plugins-base-libs-sections.txt:
10413 * gst-libs/gst/interfaces/colorbalance.c:
10414 * gst-libs/gst/interfaces/colorbalance.h:
10415 * gst-libs/gst/interfaces/colorbalancechannel.c:
10416 * gst-libs/gst/interfaces/colorbalancechannel.h:
10417 * gst-libs/gst/interfaces/tuner.c:
10418 * gst-libs/gst/interfaces/tunerchannel.c:
10419 * gst-libs/gst/interfaces/tunerchannel.h:
10420 * gst-libs/gst/interfaces/tunernorm.c:
10421 * gst-libs/gst/interfaces/tunernorm.h:
10422 * gst-libs/gst/video/video.c:
10423 * gst-libs/gst/video/video.h:
10424 Document the GstTuner and GstColorBalance interfaces, and some
10425 other random API functions that needed it. 70% symbol coverage, woo.
10427 2008-05-09 16:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
10429 gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
10430 Original commit message from CVS:
10431 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
10432 Choose to allocate one less segment but require one additional segment
10434 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
10435 No need to increment the number of segments in the source.
10436 * gst-libs/gst/audio/gstbaseaudiosink.c:
10437 (gst_base_audio_sink_get_time), (clock_convert_external),
10438 (gst_base_audio_sink_resample_slaving),
10439 (gst_base_audio_sink_skew_slaving),
10440 (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
10441 (gst_base_audio_sink_async_play):
10442 Remove adding latency when returning the internal time while subtracting
10443 it again when we use the value a little later.
10444 When calculating the end timestamp, we are making a rounding error
10445 with the current algorithm. Ensure that we don't accumulate these
10446 rounding errors when aligning samples by not resampling at all if we
10447 don't need to. Fixes #419351.
10448 Make the initial calibration of the clock slaving a little more
10449 predictable and accurate. Also handle the case where we don't do
10452 2008-05-09 08:34:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10454 gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
10455 Original commit message from CVS:
10456 Based on a patch by:
10457 Björn Benderius <bjoern dot benderius at axis dot com>
10458 * gst/ffmpegcolorspace/avcodec.h:
10459 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
10460 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
10461 (gst_ffmpegcsp_avpicture_fill):
10462 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
10463 * gst/ffmpegcolorspace/imgconvert_template.h:
10464 Add conversions from/to NV12 and NV21 and conversions between those
10465 two formats. Fixes bug #532166.
10467 2008-05-08 17:35:44 +0000 Edward Hervey <bilboed@bilboed.com>
10469 gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
10470 Original commit message from CVS:
10471 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
10472 Abort the h264 typefinding as soon as _peek() doesn't return anything,
10473 which happens for example with files smaller than 128kb.
10475 2008-05-08 14:46:27 +0000 Wouter Cloetens <zombie@e2big.org>
10477 gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
10478 Original commit message from CVS:
10479 Patch by: Wouter Cloetens <zombie at e2big dot org>
10480 * gst-libs/gst/rtsp/Makefile.am:
10481 * gst-libs/gst/rtsp/gstrtspconnection.c:
10482 (gst_rtsp_connection_create), (md5_digest_to_hex_string),
10483 (auth_digest_compute_hex_urp), (auth_digest_compute_response),
10484 (add_auth_header), (gst_rtsp_connection_free),
10485 (gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
10486 (gst_rtsp_connection_set_auth_param),
10487 (gst_rtsp_connection_clear_auth_params):
10488 * gst-libs/gst/rtsp/gstrtspconnection.h:
10489 Add Digest authorization support for RTSP connections. See #532065.
10490 * gst-libs/gst/rtsp/md5.c:
10491 * gst-libs/gst/rtsp/md5.h:
10492 Yeap, another md5 implementation until we can depend on a glib that has
10495 2008-05-08 06:20:42 +0000 Sjoerd Simons <sjoerd@luon.net>
10497 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
10498 Original commit message from CVS:
10499 Patch by: Sjoerd Simons <sjoerd at luon dot net>
10500 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
10501 Let audioresample use the buffer allocation of basetransform instead
10503 * tests/check/elements/audioresample.c: (alloc_only_48000),
10504 (GST_START_TEST), (audioresample_suite):
10505 Add unit test for the recent basetransform bugfix, where upstream
10506 changes caps to something that can't be passed through anymore.
10508 2008-05-07 19:50:27 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
10510 win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
10511 Original commit message from CVS:
10512 * win32/common/config.h.in:
10513 Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
10514 use the real thing than having "???" unconditionally.
10516 2008-05-07 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
10518 gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
10519 Original commit message from CVS:
10520 * gst-libs/gst/audio/gstbaseaudiosink.c:
10521 (gst_base_audio_sink_query):
10522 Report the latency with the new seglatency parameter.
10523 * gst-libs/gst/audio/gstringbuffer.c:
10524 (gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
10525 (gst_ring_buffer_acquire):
10526 * gst-libs/gst/audio/gstringbuffer.h:
10527 Add new field to the ringbufferspec to specify the expected latency
10528 between the underlying device read/write pointer, this is needed
10529 when writing sinks that sit a little closer to the hardware.
10530 Add some more docs for other fields.
10532 2008-05-07 10:38:23 +0000 Wim Taymans <wim.taymans@gmail.com>
10534 gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
10535 Original commit message from CVS:
10536 * gst-libs/gst/app/.cvsignore:
10537 * gst-libs/gst/app/Makefile.am:
10538 * gst-libs/gst/app/gstapp-marshal.list:
10539 Add marshal.list, make it compile and add to cvsignore.
10540 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
10541 (gst_app_sink_stop):
10543 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
10544 (gst_app_src_init), (gst_app_src_set_property),
10545 (gst_app_src_get_property), (gst_app_src_unlock),
10546 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
10547 (gst_app_src_create), (gst_app_src_set_caps),
10548 (gst_app_src_get_caps), (gst_app_src_set_size),
10549 (gst_app_src_get_size), (gst_app_src_set_seekable),
10550 (gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
10551 (gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
10552 (gst_app_src_end_of_stream):
10553 * gst-libs/gst/app/gstappsrc.h:
10554 Beat appsrc in shape, add signals and actions.
10556 Add properties for caps, size, seekability and max-buffers.
10557 Fix unlock/stop code.
10559 2008-05-06 12:35:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10561 gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
10562 Original commit message from CVS:
10563 * gst/volume/gstvolume.c: (volume_transform_ip):
10564 Return NOT_NEGOTIATED if we didn't set a process function yet for some
10565 reason instead of crashing later. Might fix bug #509125.
10567 2008-05-06 12:12:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10569 gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
10570 Original commit message from CVS:
10571 Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
10572 * gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
10573 * gst/audioconvert/audioconvert.h:
10574 * gst/audioconvert/gstaudioconvert.c:
10575 (gst_audio_convert_parse_caps),
10576 (structure_has_fixed_channel_positions),
10577 (gst_audio_convert_transform_caps):
10578 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
10579 Add support for more than 8 channels and NONE channel layouts. For
10580 more than 8 channels no channel conversion is supported yet, only
10581 format conversions are supported. Fixes bug #398033.
10582 * tests/check/elements/audioconvert.c: (verify_convert),
10583 (GST_START_TEST), (audioconvert_suite):
10584 Add some unit tests by Tim for checking the NONE channel layouts
10585 and more than 8 channels and add some more unit tests for channel
10588 2008-05-06 10:16:49 +0000 Wim Taymans <wim.taymans@gmail.com>
10590 gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
10591 Original commit message from CVS:
10592 * gst/playback/gstdecodebin2.c: (connect_pad):
10593 When autoplugging fails, set the element back to NULL before
10596 2008-05-06 09:59:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10598 win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols.
10599 Original commit message from CVS:
10600 * win32/common/libgstaudio.def:
10601 Add gst_base_audio_src_[sg]et_slave_method() to the exported
10604 2008-05-05 12:33:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10606 gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
10607 Original commit message from CVS:
10608 * gst/subparse/samiparse.c: (handle_start_sync),
10609 (end_sami_element), (characters_sami):
10610 Remove trailing, leading and double whitespaces.
10611 Correctly timestamp buffers and output the last buffer too.
10612 * tests/check/elements/subparse.c: (GST_START_TEST),
10614 Add a simple unit test for SAMI parsing.
10616 2008-05-05 11:14:48 +0000 Young-Ho Cha <ganadist@chollian.net>
10618 gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
10619 Original commit message from CVS:
10620 Patch by: Young-Ho Cha <ganadist at chollian dot net>
10621 * gst/subparse/samiparse.c: (handle_start_sync),
10622 (start_sami_element), (end_sami_element), (characters_sami),
10623 (sami_context_reset):
10624 Only output characters inside the "sync" elements. There could be
10625 other elements like "style" that have some content but should
10626 not be printed. Fixes bug #467911.
10628 2008-05-05 10:27:45 +0000 Wim Taymans <wim.taymans@gmail.com>
10630 gst-libs/gst/app/gstappsink.*: Start some docs.
10631 Original commit message from CVS:
10632 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
10633 (gst_app_sink_init), (gst_app_sink_set_property),
10634 (gst_app_sink_get_property), (gst_app_sink_unlock_start),
10635 (gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
10636 (gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
10637 (gst_app_sink_preroll), (gst_app_sink_render),
10638 (gst_app_sink_set_caps), (gst_app_sink_set_drop),
10639 (gst_app_sink_get_drop):
10640 * gst-libs/gst/app/gstappsink.h:
10642 Add property to drop buffers when the queue is filled
10643 Fix unlocking and flushing when the queues are filled.
10645 2008-05-05 10:03:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10647 gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
10648 Original commit message from CVS:
10649 * gst/playback/gstplaybasebin.c: (set_audio_mute),
10650 (set_active_source):
10651 * gst/playback/gstplaybasebin.h:
10652 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
10653 (playbin_set_audio_mute):
10654 Allow setting -1 as current-audio to mute the current audio stream,
10655 similar to what is done for subtitles. Fixes bug #342294.
10657 2008-05-05 07:41:03 +0000 Edward Hervey <bilboed@bilboed.com>
10659 gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
10660 Original commit message from CVS:
10661 * gst-libs/gst/pbutils/descriptions.c: (formats):
10662 It's SorensOn and not SorensEn.
10664 2008-05-04 15:23:36 +0000 Tim-Philipp Müller <tim@centricular.net>
10666 gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
10667 Original commit message from CVS:
10668 * gst-libs/gst/pbutils/descriptions.c: (formats):
10669 Fix description of video/x-flash-video.
10671 2008-05-04 15:02:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10673 Remove some unused code.
10674 Original commit message from CVS:
10675 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
10676 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
10677 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
10678 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
10679 Remove some unused code.
10680 * gst/audioconvert/gstaudioquantize.c:
10681 (gst_audio_quantize_free_noise_shaping):
10682 Don't return before freeing the noise shaping history.
10684 2008-05-03 16:00:04 +0000 Tim-Philipp Müller <tim@centricular.net>
10686 tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
10687 Original commit message from CVS:
10688 * tests/check/elements/subparse.c: (do_test),
10689 (test_tmplayer_style3b), (subparse_suite):
10690 Add unit test for the tmplayer variant from bug #530962.
10692 2008-05-03 15:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
10694 gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
10695 Original commit message from CVS:
10696 * gst/subparse/gstsubparse.c: (handle_buffer),
10697 (gst_sub_parse_sink_event):
10698 * gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
10699 (tmplayer_parse_line):
10700 Fix parsing of tmplayer subtitle variant where every single line contains
10701 text and there isn't an empty line after each line to determine the
10702 duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
10703 making sure that we push out the last line of text without a duration if
10704 there's still text left in the buffer at the end.
10706 2008-05-03 15:39:04 +0000 Tim-Philipp Müller <tim@centricular.net>
10708 gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
10709 Original commit message from CVS:
10710 * gst/subparse/gstsubparse.c: (feed_textbuf):
10711 Fix detection of discontinuities based on the buffer offset (doesn't work
10712 so well if no buffer offset is set) and also check for the DISCONT buffer
10713 flag. This keeps the parser state from being reset after each buffer in
10716 2008-05-03 12:09:16 +0000 Tim-Philipp Müller <tim@centricular.net>
10718 gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
10719 Original commit message from CVS:
10720 * gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
10721 Further fine-tuning: don't absolutely require sequence or GOP headers
10722 (as introduced in the previous commit), but adjust the typefind
10723 probabilities returned accordingly if we don't see them. Also make sure
10724 picture header and first slice are somewhat close to each other (which
10725 is not perfect but still better than requiring a fixed offset or having
10728 2008-05-02 12:13:08 +0000 Wim Taymans <wim.taymans@gmail.com>
10730 gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
10731 Original commit message from CVS:
10732 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
10733 (gst_basertppayload_sink_setcaps),
10734 (gst_basertppayload_sink_getcaps):
10735 Rename the setcaps/getcaps function internally to make it clear that
10736 they are called for the sink pad.
10738 2008-05-02 12:11:07 +0000 Wim Taymans <wim.taymans@gmail.com>
10740 gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
10741 Original commit message from CVS:
10742 * gst-libs/gst/rtp/gstbasertpdepayload.c:
10743 (gst_base_rtp_depayload_class_init),
10744 (gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
10745 (gst_base_rtp_depayload_packet_lost),
10746 (gst_base_rtp_depayload_set_gst_timestamp):
10747 * gst-libs/gst/rtp/gstbasertpdepayload.h:
10748 Catch packet-lost events from the jitterbuffer and convert them into a
10749 vmethod call (lost-packet) so that depayloaders can do something smart.
10750 Also add a default packet-lost function that sends out a segment update
10753 2008-05-02 11:13:05 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10755 gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
10756 Original commit message from CVS:
10757 * gst/playback/test4.c:
10758 * gst/playback/test5.c:
10759 * gst/playback/test6.c:
10760 * gst/playback/test7.c:
10761 Also include config.h when relying on defines from it. Fixes the
10762 build. Its been a please to serve :)
10764 2008-05-02 10:54:51 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
10767 * gst/videotestsrc/videotestsrc.c:
10768 Add support for NV12 and NV21 in videotestsrc
10769 Original commit message from CVS:
10770 * gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
10771 (paint_setup_NV21), (paint_hline_NV12_NV21):
10772 Add support for NV12 and NV21 in videotestsrc
10774 2008-05-02 10:02:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10776 gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
10777 Original commit message from CVS:
10778 * gst/videoscale/gstvideoscale.c:
10779 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
10780 * gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
10781 (vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
10782 (vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
10783 (vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
10784 (vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
10785 (vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
10786 (vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
10787 (vs_image_scale_linear_RGB555):
10788 Support 1x1 images as input and output as for example the BBC HQ new
10789 streams have 1x1 GIFs in the playlists for some reason.
10791 2008-05-01 19:11:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10793 gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
10794 Original commit message from CVS:
10795 * gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
10797 If we can't activate one of the decoders we plugged in (such as,
10798 say, musepackdec) for some reason (it might not support push mode,
10799 for example), remove any pad probes that close_pad_link() might
10800 have set up. This makes sure we later don't try to remove a probe
10801 for a pad that doesn't exist any longer, and avoids nast warnings
10802 and probably other things too.
10804 2008-04-30 20:54:56 +0000 Tim-Philipp Müller <tim@centricular.net>
10806 gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
10807 Original commit message from CVS:
10808 * gst/typefind/gsttypefindfunctions.c:
10809 (mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
10811 Rework mpeg video stream typefinding a bit more: make sure sequence,
10812 GOP, picture and slice headers appear in the order they should and
10813 that we've in fact at least had one of each; fix picture header
10814 detection; decouple picture and slice header check - don't assume
10815 they're at a fixed offset, there may be extra data in between. Also,
10816 announce varying degrees of probability depending on what we found
10817 exactly (multiple pictures, at least one picture, just sequence and
10818 GOP headers). Finally, in _ensure_data(), take into account that we
10819 might be typefinding smaller amounts of data, such as the first
10820 buffer of a stream, so fall back to the minimum size needed as long
10821 as that's available, instead of erroring out if there's less than
10822 2kB of data. Fixes #526173. Conveniently also doesn't recognise the
10823 fuzzed file from #399342 as valid.
10825 2008-04-30 17:06:45 +0000 Michael Smith <msmith@xiph.org>
10827 ext/theora/theoradec.c: Cool kids don't divide by zero.
10828 Original commit message from CVS:
10829 * ext/theora/theoradec.c:
10830 Cool kids don't divide by zero.
10831 Treat PAR of x:0 as 1:1.
10834 2008-04-30 14:37:52 +0000 Tim-Philipp Müller <tim@centricular.net>
10836 gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
10837 Original commit message from CVS:
10838 * gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
10839 (mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
10840 (mpeg_video_stream_type_find):
10841 Refactor a bit: use context structure to track parsing offset and size of
10842 available data and make the code a bit clearer. Fixes bad memory access
10845 2008-04-28 22:18:49 +0000 Michael Smith <msmith@xiph.org>
10847 gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
10848 Original commit message from CVS:
10849 * gst/playback/test4.c:
10850 * gst/playback/test5.c:
10851 * gst/playback/test6.c:
10852 * gst/tcp/gstmultifdsink.c:
10853 Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
10856 2008-04-28 08:51:38 +0000 Wim Taymans <wim.taymans@gmail.com>
10858 gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
10859 Original commit message from CVS:
10860 * gst-libs/gst/audio/gstbaseaudiosink.h:
10862 * gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
10863 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
10864 (gst_base_audio_src_set_slave_method),
10865 (gst_base_audio_src_get_slave_method),
10866 (gst_base_audio_src_set_property),
10867 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
10868 * gst-libs/gst/audio/gstbaseaudiosrc.h:
10869 Add property and methods for selecting the clock slave method in the
10870 source, like in the sink.
10871 We only implement "none" and "re-timestamp" for now.
10872 API: gst_base_audio_src_set_slave_method()
10873 API: gst_base_audio_src_get_slave_method()
10875 2008-04-25 18:18:47 +0000 Wim Taymans <wim.taymans@gmail.com>
10877 gst-libs/gst/app/gstappsink.*: Add more docs.
10878 Original commit message from CVS:
10879 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
10880 (gst_app_sink_init), (gst_app_sink_set_property),
10881 (gst_app_sink_get_property), (gst_app_sink_event),
10882 (gst_app_sink_preroll), (gst_app_sink_render),
10883 (gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
10884 (gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
10885 (gst_app_sink_pull_buffer):
10886 * gst-libs/gst/app/gstappsink.h:
10888 Add signals for when preroll and render buffers are available.
10889 Add property to control signal emission.
10890 Add property to control the max queue size.
10892 2008-04-25 07:37:09 +0000 Wim Taymans <wim.taymans@gmail.com>
10894 gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
10895 Original commit message from CVS:
10896 * gst-libs/gst/rtp/gstrtpbuffer.c:
10897 Fix the docs about the seqnum compare function, it returns a difference.
10899 2008-04-24 09:27:35 +0000 Edward Hervey <bilboed@bilboed.com>
10901 ext/alsa/gstalsadeviceprobe.c: Don't return before freeing up the allocated structures.
10902 Original commit message from CVS:
10903 * ext/alsa/gstalsadeviceprobe.c:
10904 (gst_alsa_get_device_list): Don't return before freeing up
10905 the allocated structures.
10907 2008-04-24 08:19:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10909 gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
10910 Original commit message from CVS:
10911 * gst/playback/gstplaybin.c:
10912 Remove obsolete streaminfo code and fix a leak. Fixes #529546
10914 2008-04-23 13:50:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10916 ext/ogg/gstoggdemux.c: Revert the event part, that should not go in.
10917 Original commit message from CVS:
10918 * ext/ogg/gstoggdemux.c:
10919 Revert the event part, that should not go in.
10921 2008-04-23 13:45:29 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10923 ext/ogg/gstoggdemux.c: Don't leak GstPluginFeatures when filtering.
10924 Original commit message from CVS:
10925 * ext/ogg/gstoggdemux.c:
10926 Don't leak GstPluginFeatures when filtering.
10928 2008-04-23 08:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10930 sys/xvimage/xvimagesink.c: Add some logging for cases when grabbing the xv failed.
10931 Original commit message from CVS:
10932 * sys/xvimage/xvimagesink.c:
10933 Add some logging for cases when grabbing the xv failed.
10935 2008-04-22 06:18:04 +0000 David Schleef <ds@schleef.org>
10937 ext/ogg/gstoggmux.c: Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos packet. Should conform to what we cu...
10938 Original commit message from CVS:
10939 * ext/ogg/gstoggmux.c:
10940 Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos
10941 packet. Should conform to what we currently think is the
10942 final Ogg/Dirac muxing spec.
10944 2008-04-22 06:13:43 +0000 David Schleef <ds@schleef.org>
10946 sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display. Dark g...
10947 Original commit message from CVS:
10948 * sys/xvimage/xvimagesink.c:
10949 Fix typo that causes the overlay keying color to bright green
10950 on a 16-bit display. Dark grey good. Bright green bad.
10952 2008-04-21 13:47:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10954 ext/gnomevfs/gstgnomevfsuri.c: Add FIXME comment about using uri-list for source and sink.
10955 Original commit message from CVS:
10956 * ext/gnomevfs/gstgnomevfsuri.c:
10957 Add FIXME comment about using uri-list for source and sink.
10959 2008-04-20 11:42:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10961 ext/ogg/gstogmparse.c: GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to vaargs functions to gin...
10962 Original commit message from CVS:
10963 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
10964 GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to
10965 vaargs functions to gint. Otherwise the fractions will get 0 set
10966 instead of the correct value on big endian systems. Fixes bug #529018.
10968 2008-04-20 10:17:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10970 ext/gnomevfs/: Get the list of supported URI schemes in a threadsafe way and use the same list for the source and sink.
10971 Original commit message from CVS:
10972 * ext/gnomevfs/gstgnomevfssink.c:
10973 (gst_gnome_vfs_sink_uri_get_protocols):
10974 * ext/gnomevfs/gstgnomevfssrc.c:
10975 (gst_gnome_vfs_src_uri_get_protocols):
10976 * ext/gnomevfs/gstgnomevfsuri.c: (_internal_get_supported_uris),
10977 (gst_gnomevfs_get_supported_uris):
10978 Get the list of supported URI schemes in a threadsafe way and use the
10979 same list for the source and sink.
10981 2008-04-20 10:11:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10983 ext/gio/gstgio.c: Don't generate a new supported protocols list on each call but cache it. It's supposed to be static...
10984 Original commit message from CVS:
10985 * ext/gio/gstgio.c: (_internal_get_supported_protocols),
10986 (gst_gio_get_supported_protocols):
10987 Don't generate a new supported protocols list on each call but cache
10988 it. It's supposed to be static anyway, this way we only leak it once
10990 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
10991 (gst_gio_sink_class_init), (gst_gio_sink_finalize),
10992 (gst_gio_sink_set_property), (gst_gio_sink_get_property),
10993 (gst_gio_sink_start):
10994 * ext/gio/gstgiosink.h:
10995 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
10996 (gst_gio_src_class_init), (gst_gio_src_finalize),
10997 (gst_gio_src_set_property), (gst_gio_src_get_property),
10998 (gst_gio_src_start):
10999 * ext/gio/gstgiosrc.h:
11000 API: Add "file" properties where one can set a GFile as source/destination.
11001 Add locking to the properties and use gst_element_class_set_details_simple()
11002 instead of a static GstElementDetails struct.
11004 2008-04-19 20:06:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11006 gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
11007 Original commit message from CVS:
11008 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
11010 Add "mpp" and "mp+" as possible extensions for MusePack files.
11011 Add typefinding for MusePack StreamVersion 8 files and include the
11012 stream version in the caps.
11014 2008-04-19 16:33:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11016 gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
11017 Original commit message from CVS:
11018 * gst-libs/gst/rtp/gstrtppayloads.c:
11019 (gst_rtp_payload_info_for_name):
11020 Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
11022 2008-04-18 17:10:43 +0000 Tim-Philipp Müller <tim@centricular.net>
11024 configure.ac: Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level (NB: this only affects compilation of some...
11025 Original commit message from CVS:
11027 Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level
11028 (NB: this only affects compilation of some of the examples).
11029 Remove some configure.ac cruft that's not needed any longer.
11031 2008-04-18 14:54:01 +0000 Edward Hervey <bilboed@bilboed.com>
11033 gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
11034 Original commit message from CVS:
11035 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
11036 Don't validate the payload if there isn't any.
11039 2008-04-17 07:33:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11041 gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
11042 Original commit message from CVS:
11043 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
11044 Use g_atomic_int_set() instead of gst_atomic_int_set().
11046 2008-04-17 07:29:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11048 ext/gio/gstgio.c: Return NULL instead of a gchar * array with one NULL element if we don't get any supported URI sche...
11049 Original commit message from CVS:
11050 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11051 Return NULL instead of a gchar * array with one NULL element if we
11052 don't get any supported URI schemes from GIO.
11054 2008-04-15 19:06:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11056 gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
11057 Original commit message from CVS:
11058 * gst/audiotestsrc/gstaudiotestsrc.c:
11059 Remove cpp style commented old code.
11061 2008-04-15 19:02:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11063 gst/playback/gstdecodebin2.c: Fix signal docs.
11064 Original commit message from CVS:
11065 * gst/playback/gstdecodebin2.c:
11068 2008-04-14 17:58:19 +0000 Tim-Philipp Müller <tim@centricular.net>
11070 ext/pango/gsttextoverlay.c: Fix textoverlay unit test again by making the supposed default value for the wait-text pr...
11071 Original commit message from CVS:
11072 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
11073 (gst_text_overlay_init):
11074 Fix textoverlay unit test again by making the supposed default
11075 value for the wait-text property the actual default value.
11076 Also fix Since: tag for new property.
11078 2008-04-11 17:13:52 +0000 Tim-Philipp Müller <tim@centricular.net>
11080 gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
11081 Original commit message from CVS:
11082 * gst-libs/gst/video/video.c: (gst_video_format_new_caps),
11083 (gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
11084 (gst_video_format_get_pixel_stride),
11085 (gst_video_format_get_component_width),
11086 (gst_video_format_get_component_height),
11087 (gst_video_format_get_component_offset), (gst_video_format_get_size),
11088 (gst_video_format_convert):
11089 Add guards to these functions to ensure sane input values.
11090 * tests/check/libs/video.c:
11091 Fix unit test not to create caps with width=0 and height=0.
11093 2008-04-11 01:25:01 +0000 Wim Taymans <wim.taymans@gmail.com>
11095 docs/design/draft-keyframe-force.txt: Fix typo.
11096 Original commit message from CVS:
11097 * docs/design/draft-keyframe-force.txt:
11099 * gst/playback/gstqueue2.c: (update_buffering),
11100 (gst_queue_handle_src_query):
11101 Set buffering mode in the messages.
11102 Set buffering percent in the query.
11103 * tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
11104 (do_stream_buffering), (do_download_buffering), (msg_buffering):
11105 Do some more fancy things based on the buffering method in use.
11107 2008-04-09 21:42:24 +0000 Wim Taymans <wim.taymans@gmail.com>
11109 tests/examples/seek/seek.c: Add basic download reports to seek using the new buffering API.
11110 Original commit message from CVS:
11111 * tests/examples/seek/seek.c: (update_fill), (set_update_fill),
11112 (play_cb), (pause_cb), (stop_cb), (msg_state_changed),
11113 (msg_buffering), (main):
11114 Add basic download reports to seek using the new buffering API.
11116 2008-04-09 21:40:17 +0000 Wim Taymans <wim.taymans@gmail.com>
11118 gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
11119 Original commit message from CVS:
11120 * gst/playback/gstqueue2.c: (update_buffering),
11121 (gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
11122 (gst_queue_src_checkgetrange_function):
11123 Include extra buffering stats in the buffering message.
11124 Implement BUFFERING query.
11125 * gst/playback/gsturidecodebin.c: (do_async_start),
11126 (do_async_done), (type_found), (setup_streaming), (setup_source),
11127 (gst_uri_decode_bin_change_state):
11128 Only add decodebin2 when the type is found in streaming mode.
11129 Make uridecodebin async to PAUSED even when we don't have decodebin2
11132 2008-04-09 08:38:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11134 ext/gio/gstgio.c: Filter cdda from the supported URI schemes. We can't support musicbrainz tags and everything else o...
11135 Original commit message from CVS:
11136 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11137 Filter cdda from the supported URI schemes. We can't support
11138 musicbrainz tags and everything else one expects from a cdda source
11139 with GIO. Fixes bug #526794.
11141 2008-04-07 22:37:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11143 * sys/xvimage/xvimagesink.c:
11144 Fix calculation of 'expected size' for YV12 buffers.
11145 Original commit message from CVS:
11146 2008-04-07 Jan Schmidt <jan.schmidt@sun.com>
11147 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
11148 (gst_xvimagesink_buffer_alloc):
11149 Fix calculation of 'expected size' for YV12 buffers.
11150 Be a little more verbose in the debug output for buffer-alloc'ed
11151 buffers which turn out to have the wrong size.
11153 2008-04-07 22:26:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11156 Fix calculation of 'expected size' for YV12 buffers.
11157 Original commit message from CVS:
11158 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
11159 (gst_xvimagesink_buffer_alloc):
11160 Fix calculation of 'expected size' for YV12 buffers.
11161 Be a little more verbose in the debug output for buffer-alloc'ed
11162 buffers which turn out to have the wrong size.
11164 2008-04-07 10:50:11 +0000 Tim-Philipp Müller <tim@centricular.net>
11166 Merge other changes from 0.10.19 release branch.
11167 Original commit message from CVS:
11170 * gst-plugins-base.doap:
11171 Merge other changes from 0.10.19 release branch.
11173 2008-04-06 20:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
11175 gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
11176 Original commit message from CVS:
11177 * gst-libs/gst/audio/gstbaseaudiosink.c:
11178 (gst_base_audio_sink_class_init):
11179 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11180 (gst_base_audio_src_class_init):
11181 * gst/playback/gstplayback.c: (plugin_init):
11182 * gst/volume/gstvolume.c: (plugin_init):
11183 Work around missing bits of thread-safety on older GLibs some
11184 more to avoid assertions when starting up multiple playbin
11185 objects concurrently (see #512382).
11187 2008-04-06 17:19:39 +0000 Tim-Philipp Müller <tim@centricular.net>
11189 gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
11190 Original commit message from CVS:
11191 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
11192 Remove some more fields.
11194 2008-04-06 08:56:07 +0000 Damien Lespiau <damien.lespiau@gmail.com>
11196 configure.ac: Actually build dlls when cross-compiling with mingw32.
11197 Original commit message from CVS:
11198 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
11200 Actually build dlls when cross-compiling with mingw32.
11203 2008-04-03 23:01:11 +0000 Tim-Philipp Müller <tim@centricular.net>
11205 configure.ac: Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
11206 Original commit message from CVS:
11208 Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
11210 2008-04-03 16:10:53 +0000 Wim Taymans <wim.taymans@gmail.com>
11212 tests/examples/seek/seek.c: Add statusbar.
11213 Original commit message from CVS:
11214 * tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
11215 (msg_buffering), (connect_bus_signals), (main):
11217 Add buffering support with feedback in the statusbar.
11219 2008-04-03 15:58:37 +0000 Tim-Philipp Müller <tim@centricular.net>
11221 ext/ogg/gstoggmux.c: Fix sample pipeline description.
11222 Original commit message from CVS:
11223 * ext/ogg/gstoggmux.c:
11224 Fix sample pipeline description.
11226 2008-04-03 14:58:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11228 docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
11229 Original commit message from CVS:
11230 * docs/plugins/Makefile.am:
11231 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
11232 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
11233 * docs/plugins/gst-plugins-base-plugins-sections.txt:
11234 Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
11235 * docs/plugins/gst-plugins-base-plugins.args:
11236 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11237 * docs/plugins/gst-plugins-base-plugins.interfaces:
11238 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11239 * docs/plugins/inspect/plugin-adder.xml:
11240 * docs/plugins/inspect/plugin-alsa.xml:
11241 * docs/plugins/inspect/plugin-audioconvert.xml:
11242 * docs/plugins/inspect/plugin-audiorate.xml:
11243 * docs/plugins/inspect/plugin-audioresample.xml:
11244 * docs/plugins/inspect/plugin-audiotestsrc.xml:
11245 * docs/plugins/inspect/plugin-cdparanoia.xml:
11246 * docs/plugins/inspect/plugin-decodebin.xml:
11247 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
11248 * docs/plugins/inspect/plugin-gdp.xml:
11249 * docs/plugins/inspect/plugin-gnomevfs.xml:
11250 * docs/plugins/inspect/plugin-libvisual.xml:
11251 * docs/plugins/inspect/plugin-ogg.xml:
11252 * docs/plugins/inspect/plugin-pango.xml:
11253 * docs/plugins/inspect/plugin-playback.xml:
11254 * docs/plugins/inspect/plugin-queue2.xml:
11255 * docs/plugins/inspect/plugin-subparse.xml:
11256 * docs/plugins/inspect/plugin-tcp.xml:
11257 * docs/plugins/inspect/plugin-theora.xml:
11258 * docs/plugins/inspect/plugin-typefindfunctions.xml:
11259 * docs/plugins/inspect/plugin-uridecodebin.xml:
11260 * docs/plugins/inspect/plugin-video4linux.xml:
11261 * docs/plugins/inspect/plugin-videorate.xml:
11262 * docs/plugins/inspect/plugin-videoscale.xml:
11263 * docs/plugins/inspect/plugin-videotestsrc.xml:
11264 * docs/plugins/inspect/plugin-volume.xml:
11265 * docs/plugins/inspect/plugin-vorbis.xml:
11266 * docs/plugins/inspect/plugin-ximagesink.xml:
11267 * docs/plugins/inspect/plugin-xvimagesink.xml:
11268 Update introspection data.
11269 * ext/ogg/gstoggmux.c:
11271 * gst/playback/gstdecodebin2.c:
11272 Don't use gtk-doc style comment start for private stuff, but make it
11273 formatted like this for consistency.
11275 2008-04-03 12:16:04 +0000 Wim Taymans <wim.taymans@gmail.com>
11277 gst/playback/gstdecodebin2.c: Remove fakesink hack, we can now implement this more elegantly.
11278 Original commit message from CVS:
11279 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
11280 (gst_decode_bin_init), (gst_decode_bin_dispose),
11281 (gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
11282 (gst_decode_bin_set_property), (gst_decode_bin_get_property),
11283 (analyze_new_pad), (connect_pad), (expose_pad),
11284 (gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
11285 (gst_decode_group_expose), (gst_decode_group_free),
11286 (do_async_start), (do_async_done), (gst_decode_bin_change_state):
11287 Remove fakesink hack, we can now implement this more elegantly.
11288 Added property to bypass typefinding.
11289 Removed underrun callback and demuxer pad probe, we now use the srcpad
11290 probe to expose groups.
11291 API::sink-caps property
11292 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
11293 Guard against multiple emissions of the no_more_pads signal, which
11294 happens when we are dealing with chained oggs.
11295 * gst/playback/gsturidecodebin.c: (remove_decoders),
11296 (make_decoder), (type_found), (setup_streaming), (source_new_pad),
11298 For streams, use our own typefind element and plug our queue after it.
11299 We will need this to determine the type of buffering to use for the
11302 2008-04-03 10:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
11304 gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
11305 Original commit message from CVS:
11306 * gst-libs/gst/audio/gstbaseaudiosink.c:
11307 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
11308 Guard against over and underflows because of clock slaving.
11309 When we are using our own clock, still compensate for any calibrations
11310 that we might have done to our clock.
11312 2008-04-03 10:22:33 +0000 Wim Taymans <wim.taymans@gmail.com>
11314 ext/theora/theoradec.c: Don't try to do anything fancy with the return code from pushing an event, it does not have e...
11315 Original commit message from CVS:
11316 * ext/theora/theoradec.c: (theora_handle_type_packet),
11317 (theora_dec_chain):
11318 Don't try to do anything fancy with the return code from pushing an
11319 event, it does not have enough information to turn it into a
11322 2008-04-03 10:19:43 +0000 Wim Taymans <wim.taymans@gmail.com>
11324 ext/ogg/gstoggdemux.c: Add small debug line.
11325 Original commit message from CVS:
11326 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_reset),
11327 (gst_ogg_demux_chain_elem_pad):
11328 Add small debug line.
11329 Pass return code from the internal decoder instead of the too generic
11332 2008-04-03 06:39:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11334 gst-libs/gst/cdda/: Use GLib's base64 implementation instead of our own.
11335 Original commit message from CVS:
11336 * gst-libs/gst/cdda/Makefile.am:
11337 * gst-libs/gst/cdda/base64.c:
11338 * gst-libs/gst/cdda/base64.h:
11339 * gst-libs/gst/cdda/gstcddabasesrc.c:
11340 (gst_cddabasesrc_calculate_musicbrainz_discid):
11341 Use GLib's base64 implementation instead of our own.
11343 2008-04-02 15:41:50 +0000 Wim Taymans <wim.taymans@gmail.com>
11345 ext/ogg/gstoggdemux.c: Refix oggdemux, we only have a problem if we failed to find a chain and we are not EOF.
11346 Original commit message from CVS:
11347 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
11348 (gst_ogg_demux_read_chain):
11349 Refix oggdemux, we only have a problem if we failed to find a chain and
11352 2008-04-02 15:07:01 +0000 Victor STINNER <victor.stinner@haypocalc.com>
11354 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...
11355 Original commit message from CVS:
11356 Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
11357 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
11358 (gst_ogg_demux_read_chain):
11359 When we fail to find a BOS page and we and up with no chain, error out
11360 properly instead of segfaulting. Fixes #525665.
11362 2008-04-02 14:58:05 +0000 Wim Taymans <wim.taymans@gmail.com>
11364 ext/ogg/gstoggdemux.c: The new-pad-group sequence is add-pads, no-more-pads, add-pads, no-more-pads...
11365 Original commit message from CVS:
11366 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
11367 (gst_ogg_demux_read_chain), (gst_ogg_demux_handle_page):
11368 The new-pad-group sequence is add-pads, no-more-pads, add-pads,
11371 2008-04-02 11:08:05 +0000 Wim Taymans <wim.taymans@gmail.com>
11373 gst/playback/gstqueue2.c: Update the estimated input data when we push out a buffer.
11374 Original commit message from CVS:
11375 * gst/playback/gstqueue2.c: (update_out_rates),
11376 (gst_queue_open_temp_location_file),
11377 (gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
11378 (gst_queue_handle_src_query), (gst_queue_set_property):
11379 Update the estimated input data when we push out a buffer.
11380 Add some debug info about the temp file.
11381 Only forward src events when we are not using a temp file.
11382 Don't block the duration query, we need to find something better.
11383 Don't leak the temp filename.
11385 2008-04-01 14:01:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11387 configure.ac: Require GLib 2.12 and liboil 0.3.14.
11388 Original commit message from CVS:
11390 Require GLib 2.12 and liboil 0.3.14.
11391 * gst/volume/gstvolume.c: (volume_process_double):
11392 Unconditionally use liboil 0.3.14 function.
11394 2008-03-31 16:08:45 +0000 Wim Taymans <wim.taymans@gmail.com>
11396 gst-libs/gst/riff/riff-media.c: ms-gsm can have arbitrarty sample rates. See #481354.
11397 Original commit message from CVS:
11398 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
11399 ms-gsm can have arbitrarty sample rates. See #481354.
11401 2008-03-28 16:22:35 +0000 Wim Taymans <wim.taymans@gmail.com>
11403 gst-libs/gst/riff/riff-media.c: MP4S is generic MPEG-4, not a microsoft variant.
11404 Original commit message from CVS:
11405 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
11406 MP4S is generic MPEG-4, not a microsoft variant.
11408 2008-03-27 15:26:38 +0000 Michael Smith <msmith@xiph.org>
11410 gst/gdp/gstgdpdepay.c: Check the body CRC (if set) when depayloading.
11411 Original commit message from CVS:
11412 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
11413 Check the body CRC (if set) when depayloading.
11416 2008-03-24 17:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
11418 ext/pango/gsttextoverlay.c: Fix Since: version for new property.
11419 Original commit message from CVS:
11420 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
11421 Fix Since: version for new property.
11423 2008-03-24 16:40:08 +0000 Wim Taymans <wim.taymans@gmail.com>
11425 gst-libs/gst/rtsp/gstrtspconnection.c: Don't error when poll_wait returns EAGAIN.
11426 Original commit message from CVS:
11427 * gst-libs/gst/rtsp/gstrtspconnection.c:
11428 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
11429 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
11430 Don't error when poll_wait returns EAGAIN.
11432 2008-03-24 14:08:22 +0000 Wim Taymans <wim.taymans@gmail.com>
11434 gst/playback/gstqueue2.c: The queue is never filled when there are no buffers in the queue at all.
11435 Original commit message from CVS:
11436 * gst/playback/gstqueue2.c: (gst_queue_is_filled):
11437 The queue is never filled when there are no buffers in the queue at all.
11440 2008-03-24 12:26:30 +0000 Wim Taymans <wim.taymans@gmail.com>
11442 gst/playback/gstplaybin2.c: Update some docs.
11443 Original commit message from CVS:
11444 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
11445 (init_group), (free_group), (gst_play_bin_init),
11446 (gst_play_bin_finalize), (gst_play_bin_set_uri),
11447 (gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
11448 (gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
11449 (gst_play_bin_set_current_video_stream),
11450 (gst_play_bin_set_current_audio_stream),
11451 (gst_play_bin_set_current_text_stream),
11452 (gst_play_bin_set_encoding), (gst_play_bin_set_property),
11453 (gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
11454 (no_more_pads_cb), (perform_eos), (autoplug_select_cb),
11455 (activate_group), (deactivate_group), (setup_next_source),
11456 (save_current_group), (gst_play_bin_change_state):
11458 Add new locks and conds to protect pipeline creation and group
11460 Implement the sub-uri property.
11461 Keep track of pending uridecodebin creation and configure the output
11462 pipeline after all streams are configured.
11463 Propagate subtitle encoding to the uridecodebins.
11464 Implement getting the video/audio/visualisation elements.
11465 Use input-selector for stream switching.
11466 If we are asked to do visualisation, prefer to autoplug raw sinks
11467 instead of sinks that accept encoded data.
11469 2008-03-24 12:15:26 +0000 Wim Taymans <wim.taymans@gmail.com>
11471 gst/playback/gstplaysink.*: Add methods to get audio/video/vis elements.
11472 Original commit message from CVS:
11473 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
11474 (gst_play_sink_init), (gst_play_sink_dispose),
11475 (gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
11476 (gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
11477 (gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
11478 (gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
11479 (gst_play_sink_set_volume), (gst_play_sink_get_volume),
11480 (gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
11481 (gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
11482 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
11483 (gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
11484 * gst/playback/gstplaysink.h:
11485 Add methods to get audio/video/vis elements.
11486 Add methods to set the font description for the overlay.
11487 Remove properties, we're using this element with its methods only.
11488 Add support for subtitles.
11489 Rearrange the locking a bit to not use the object lock for protecting
11490 the pipeline construction.
11491 Try to use the volume and mute property on the sink when its available.
11492 Implement the mute option with volume when the sink does not have a mute
11494 Only add volume element when the sink has no volume property.
11495 Only do visualisations with raw audio pads.
11497 2008-03-24 12:03:02 +0000 Wim Taymans <wim.taymans@gmail.com>
11499 ext/pango/gsttextoverlay.*: Add property to configure waiting for text on the textpad or not, with the default behavi...
11500 Original commit message from CVS:
11501 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
11502 (gst_text_overlay_init), (gst_text_overlay_set_property),
11503 (gst_text_overlay_get_property), (gst_text_overlay_src_event),
11504 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
11505 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
11506 (gst_text_overlay_change_state):
11507 * ext/pango/gsttextoverlay.h:
11508 Add property to configure waiting for text on the textpad or not, with
11509 the default behaviour being the old one (always wait for text before
11510 rendering the video). This default behaviour is usually not the best one
11511 because the text stream can very sparse and could require queueing a lot
11513 Fix the flushing and EOS handing so that we don't mix up their meaning.
11515 2008-03-24 11:54:02 +0000 Wim Taymans <wim.taymans@gmail.com>
11517 gst/playback/gsturidecodebin.c: Add a readonly source property and notify.
11518 Original commit message from CVS:
11519 * gst/playback/gsturidecodebin.c:
11520 (gst_uri_decode_bin_autoplug_factories),
11521 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
11522 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
11523 (gst_uri_decode_bin_set_property),
11524 (gst_uri_decode_bin_get_property), (no_more_pads_full),
11525 (new_decoded_pad_cb), (gen_source_element), (remove_decoders),
11526 (proxy_autoplug_factories_signal), (make_decoder),
11527 (source_new_pad), (setup_source):
11528 Add a readonly source property and notify.
11529 Add new lock for protecting the construction of the pipeline.
11530 Keep track of the decodebins we plugged.
11531 Correctly proxy the autoplug signal so that it actually continues.
11532 Proxy subtitle-encoding to the decodebins.
11534 2008-03-24 11:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
11536 tests/examples/seek/seek.c: Rearrange some buttons in playbin2 and make some other boxes insensitive when needed.
11537 Original commit message from CVS:
11538 * tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
11539 (text_toggle_cb), (update_streams), (main):
11540 Rearrange some buttons in playbin2 and make some other boxes insensitive
11542 Add language codes to subtitle selection boxes when we gind the right
11543 tags for the streams.
11545 2008-03-24 11:36:08 +0000 Wim Taymans <wim.taymans@gmail.com>
11547 gst/playback/gstdecodebin2.c: Protect caps property with the object lock.
11548 Original commit message from CVS:
11549 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
11550 (gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
11551 (gst_decode_bin_set_subs_encoding),
11552 (gst_decode_bin_get_subs_encoding),
11553 (gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
11554 (deactivate_free_recursive):
11555 Protect caps property with the object lock.
11556 Protect encoding property with the object lock.
11557 Keep list of elements we added that have the subtitle-encoding property.
11558 Distribute the subtitle-encoding to all of the elements when it
11561 2008-03-24 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
11563 gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
11564 Original commit message from CVS:
11565 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
11566 Small debug improvement.
11567 * gst-libs/gst/audio/gstbaseaudiosink.c:
11568 (gst_base_audio_sink_render):
11569 Fix bug in determining the sample start/stop position, we want to base
11570 this decision on the fact that we are going forwards or backwards, not
11571 slower or faster. This fixes some ugly resync warnings when playing at
11574 2008-03-23 13:41:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11576 ext/gio/gstgio.c: Correctly set the supported URI schemes and don't leave some schemes in the middle or at the start ...
11577 Original commit message from CVS:
11578 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11579 Correctly set the supported URI schemes and don't leave
11580 some schemes in the middle or at the start at NULL.
11582 2008-03-23 13:12:41 +0000 Tim-Philipp Müller <tim@centricular.net>
11584 tests/check/elements/gdpdepay.c: Make test compile without unused function/variable warnings on PPC.
11585 Original commit message from CVS:
11586 * tests/check/elements/gdpdepay.c:
11587 Make test compile without unused function/variable warnings on PPC.
11589 2008-03-22 15:00:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11591 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
11592 Original commit message from CVS:
11594 * ext/alsa/gstalsamixerelement.c:
11595 (gst_alsa_mixer_element_class_init):
11596 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
11597 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
11598 * ext/cdparanoia/gstcdparanoiasrc.c:
11599 (gst_cd_paranoia_src_class_init):
11600 * ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
11601 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
11602 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
11603 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
11604 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
11605 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
11606 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
11607 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
11608 * ext/pango/gsttextrender.c: (gst_text_render_class_init):
11609 * ext/theora/theoradec.c: (gst_theora_dec_class_init):
11610 * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
11611 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
11612 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
11613 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
11614 (gst_audio_filter_template_class_init):
11615 * gst-libs/gst/audio/gstbaseaudiosink.c:
11616 (gst_base_audio_sink_class_init):
11617 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11618 (gst_base_audio_src_class_init):
11619 * gst-libs/gst/cdda/gstcddabasesrc.c:
11620 (gst_cdda_base_src_class_init):
11621 * gst-libs/gst/interfaces/mixertrack.c:
11622 (gst_mixer_track_class_init):
11623 * gst-libs/gst/rtp/gstbasertpdepayload.c:
11624 (gst_base_rtp_depayload_class_init):
11625 * gst-libs/gst/rtp/gstbasertppayload.c:
11626 (gst_basertppayload_class_init):
11627 * gst/audioconvert/gstaudioconvert.c:
11628 (gst_audio_convert_class_init):
11629 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
11630 * gst/audioresample/gstaudioresample.c:
11631 (gst_audioresample_class_init):
11632 * gst/audiotestsrc/gstaudiotestsrc.c:
11633 (gst_audio_test_src_class_init):
11634 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
11635 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
11636 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
11637 (preroll_unlinked):
11638 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
11639 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
11640 * gst/playback/gstplaysink.c: (gst_play_sink_class_init):
11641 * gst/playback/gstqueue2.c: (gst_queue_class_init):
11642 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
11643 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
11644 (gst_stream_selector_class_init):
11645 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
11646 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
11647 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
11648 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
11649 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
11650 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
11651 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
11652 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
11653 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
11654 * gst/videotestsrc/gstvideotestsrc.c:
11655 (gst_video_test_src_class_init):
11656 * gst/volume/gstvolume.c: (gst_volume_class_init):
11657 * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
11658 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
11659 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
11660 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
11661 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
11662 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
11663 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
11664 static strings (i.e. all). This gives us less memory usage,
11665 fewer allocations and thus less memory defragmentation. Depend
11666 on core CVS for this. Fixes bug #523806.
11668 2008-03-22 14:13:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11670 ext/gio/gstgio.c: Filter http and https protocols. GIO/GVfs handles them but it's impossible to implement iradio/icec...
11671 Original commit message from CVS:
11672 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11673 Filter http and https protocols. GIO/GVfs handles them but it's
11674 impossible to implement iradio/icecast with it. Better use
11675 souphttpsrc or something else for this.
11676 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
11677 If getting the file informations by a query fails try it with the
11678 seek-to-end trick too.
11680 2008-03-21 16:46:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11682 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...
11683 Original commit message from CVS:
11684 * gst/volume/gstvolume.c: (gst_volume_interface_supported),
11685 (gst_volume_base_init), (gst_volume_class_init),
11686 (volume_process_double), (volume_process_float),
11687 (volume_transform_ip), (plugin_init):
11688 memset buffers to zero if we get a GAP buffer. We usually see a
11689 buffer as one unit so let's handle it as one and don't care about
11690 volume changes while processing one buffer.
11691 Also clean up some stuff a bit.
11693 2008-03-21 15:58:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11695 gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
11696 Original commit message from CVS:
11697 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
11698 (gst_audio_convert_create_silence_buffer),
11699 (gst_audio_convert_transform):
11700 Make audioconvert GAP-aware by outputting silence buffers when the
11701 input has the GAP flag set. This is up to 8x faster.
11702 Based on a patch by Stefan Kost. Fixes bug #517813.
11704 2008-03-21 15:54:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11706 gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
11707 Original commit message from CVS:
11708 * gst/volume/gstvolume.c: (volume_process_double):
11709 Use oil_scalarmultiply_f64_ns() for double processing when it's
11710 available at compile time.
11712 2008-03-21 13:27:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11714 configure.ac: Fix lrint/lrintf checks to actually work. These functions are in libm on Linux at least so try to link ...
11715 Original commit message from CVS:
11717 Fix lrint/lrintf checks to actually work. These functions are
11718 in libm on Linux at least so try to link to it.
11720 2008-03-21 00:36:20 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11722 configure.ac: Back to development - 0.10.18.1
11723 Original commit message from CVS:
11725 Back to development - 0.10.18.1
11727 === release 0.10.18 ===
11729 2008-03-21 00:26:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11735 * docs/plugins/gst-plugins-base-plugins.args:
11736 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11737 * docs/plugins/gst-plugins-base-plugins.interfaces:
11738 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11739 * docs/plugins/gst-plugins-base-plugins.signals:
11740 * docs/plugins/inspect/plugin-adder.xml:
11741 * docs/plugins/inspect/plugin-alsa.xml:
11742 * docs/plugins/inspect/plugin-audioconvert.xml:
11743 * docs/plugins/inspect/plugin-audiorate.xml:
11744 * docs/plugins/inspect/plugin-audioresample.xml:
11745 * docs/plugins/inspect/plugin-audiotestsrc.xml:
11746 * docs/plugins/inspect/plugin-cdparanoia.xml:
11747 * docs/plugins/inspect/plugin-decodebin.xml:
11748 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
11749 * docs/plugins/inspect/plugin-gdp.xml:
11750 * docs/plugins/inspect/plugin-gnomevfs.xml:
11751 * docs/plugins/inspect/plugin-libvisual.xml:
11752 * docs/plugins/inspect/plugin-ogg.xml:
11753 * docs/plugins/inspect/plugin-pango.xml:
11754 * docs/plugins/inspect/plugin-playback.xml:
11755 * docs/plugins/inspect/plugin-queue2.xml:
11756 * docs/plugins/inspect/plugin-subparse.xml:
11757 * docs/plugins/inspect/plugin-tcp.xml:
11758 * docs/plugins/inspect/plugin-theora.xml:
11759 * docs/plugins/inspect/plugin-typefindfunctions.xml:
11760 * docs/plugins/inspect/plugin-uridecodebin.xml:
11761 * docs/plugins/inspect/plugin-video4linux.xml:
11762 * docs/plugins/inspect/plugin-videorate.xml:
11763 * docs/plugins/inspect/plugin-videoscale.xml:
11764 * docs/plugins/inspect/plugin-videotestsrc.xml:
11765 * docs/plugins/inspect/plugin-volume.xml:
11766 * docs/plugins/inspect/plugin-vorbis.xml:
11767 * docs/plugins/inspect/plugin-ximagesink.xml:
11768 * docs/plugins/inspect/plugin-xvimagesink.xml:
11769 * gst-plugins-base.doap:
11771 * win32/common/config.h:
11773 Original commit message from CVS:
11776 2008-03-21 00:16:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11803 Original commit message from CVS:
11806 2008-03-18 12:19:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11808 0.10.17.4 pre-release
11809 Original commit message from CVS:
11811 * win32/common/config.h:
11812 0.10.17.4 pre-release
11814 2008-03-18 11:20:05 +0000 Wim Taymans <wim.taymans@gmail.com>
11816 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL when trying to print strings that could be NULL because this might...
11817 Original commit message from CVS:
11818 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
11819 Use GST_STR_NULL when trying to print strings that could be NULL because
11820 this might crash on some platforms. See #520808.
11822 2008-03-18 11:10:12 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
11824 gst-libs/gst/rtsp/gstrtspconnection.c: Generic Windows fixes that makes libgstrtsp work on Windows when coupled with ...
11825 Original commit message from CVS:
11826 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
11827 * gst-libs/gst/rtsp/gstrtspconnection.c:
11828 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
11829 (read_line), (gst_rtsp_connection_read_internal):
11830 Generic Windows fixes that makes libgstrtsp work on Windows when
11831 coupled with the new GstPoll API. See #520808.
11833 2008-03-17 22:06:56 +0000 Milosz Derezynski <internalerror@gmail.com>
11835 ext/gio/gstgiobasesrc.c: If seeking to a new position succeeds don't simply return from create() without creating a b...
11836 Original commit message from CVS:
11837 Patch by: Milosz Derezynski <internalerror at gmail dot com>
11838 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
11839 If seeking to a new position succeeds don't simply return from
11840 create() without creating a buffer. Do this only in the case
11841 seeking to the new position fails. Fixes bug #523054.
11843 2008-03-17 10:32:28 +0000 Tim-Philipp Müller <tim@centricular.net>
11845 gst-libs/gst/video/video.c: Fix gst_video_format_parse_caps() for RGB caps with alpha channel (#522635).
11846 Original commit message from CVS:
11847 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
11848 (gst_video_format_from_rgba32_masks):
11849 Fix gst_video_format_parse_caps() for RGB caps with alpha channel
11851 * tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
11852 Add unit test for the RGB caps parsing and creation, checking for
11853 internal consistency of the new API and consistency of the API with
11854 the old GST_VIDEO_CAPS_* defines.
11856 2008-03-14 18:42:35 +0000 David Schleef <ds@schleef.org>
11858 gst/videotestsrc/videotestsrc.c: Oops, revert last change because -base is in freeze.
11859 Original commit message from CVS:
11860 * gst/videotestsrc/videotestsrc.c: Oops, revert last change
11861 because -base is in freeze.
11863 2008-03-14 17:33:09 +0000 William M. Brack <wbrack@mmm.hk>
11865 gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
11866 Original commit message from CVS:
11867 Patch by: William M. Brack
11868 * gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
11870 2008-03-14 09:54:44 +0000 Wim Taymans <wim.taymans@gmail.com>
11872 gst/playback/gststreamselector.*: Revert change that caused regression until a real fix is found.
11873 Original commit message from CVS:
11874 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
11875 (gst_selector_pad_chain):
11876 * gst/playback/gststreamselector.h:
11877 Revert change that caused regression until a real fix is found.
11880 2008-03-12 12:39:13 +0000 Michael Smith <msmith@xiph.org>
11882 gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
11883 Original commit message from CVS:
11884 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
11885 * gst-libs/gst/audio/gstringbuffer.h:
11886 Rename recently added buffer types to make more sense.
11887 * ext/alsa/gstalsasink.c: (alsasink_parse_spec),
11888 (gst_alsasink_write):
11889 Adapt for above API changes.
11892 2008-03-11 13:23:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11894 win32/common/libgstnetbuffer.def: Add new symbol gst_netaddress_equal. Fixes bug #521743.
11895 Original commit message from CVS:
11896 * win32/common/libgstnetbuffer.def:
11897 Add new symbol gst_netaddress_equal. Fixes bug #521743.
11899 2008-03-11 00:25:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11901 0.10.17.3 pre-release
11902 Original commit message from CVS:
11904 * win32/common/config.h:
11905 0.10.17.3 pre-release
11907 2008-03-10 17:19:56 +0000 Wim Taymans <wim.taymans@gmail.com>
11909 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
11910 Original commit message from CVS:
11911 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11912 (gst_base_audio_src_create):
11913 Fix duration when no clock was provided. Fixes #520300.
11915 2008-03-07 18:17:44 +0000 Olivier Crete <tester@tester.ca>
11917 Add trivial function to compare GstNetAddress. See #520626.
11918 Original commit message from CVS:
11919 Patch by: Olivier Crete <tester at tester ca>
11920 * docs/libs/gst-plugins-base-libs-sections.txt:
11921 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
11922 * gst-libs/gst/netbuffer/gstnetbuffer.h:
11923 Add trivial function to compare GstNetAddress. See #520626.
11924 API: GstNetBuffer::gst_netaddress_equal
11926 2008-03-07 16:10:51 +0000 Wim Taymans <wim.taymans@gmail.com>
11928 gst/tcp/gstmultifdsink.c: Update mode property docs, it's deprecated now.
11929 Original commit message from CVS:
11930 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
11931 Update mode property docs, it's deprecated now.
11933 2008-03-07 15:48:51 +0000 Wim Taymans <wim.taymans@gmail.com>
11935 gst/: Remove GstPollMode from gstpoll constructor.
11936 Original commit message from CVS:
11937 * gst-libs/gst/rtsp/gstrtspconnection.c:
11938 (gst_rtsp_connection_create):
11939 * gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
11940 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
11941 * gst/tcp/gstmultifdsink.h:
11942 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
11943 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
11944 Remove GstPollMode from gstpoll constructor.
11946 2008-03-04 00:26:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11948 0.10.17.2 pre-release
11949 Original commit message from CVS:
11951 * win32/common/config.h:
11952 0.10.17.2 pre-release
11954 2008-03-03 23:59:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11956 gst/Makefile.am: GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean them twice
11957 Original commit message from CVS:
11959 GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
11961 * win32/common/libgstinterfaces.def:
11962 * win32/common/libgstrtp.def:
11963 Add new API to the defs
11965 2008-03-03 16:11:50 +0000 Mersad Jelacic <mersad@axis.com>
11967 gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
11968 Original commit message from CVS:
11969 Patch by: Mersad Jelacic <mersad at axis dot com>
11970 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
11971 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
11972 API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
11973 possible to specify the sample size in bits. (#509637)
11975 2008-03-03 13:59:19 +0000 Tim-Philipp Müller <tim@centricular.net>
11977 tests/check/libs/mixer.c: Add a few simple checks for the new message types.
11978 Original commit message from CVS:
11979 * tests/check/libs/mixer.c:
11980 Add a few simple checks for the new message types.
11982 2008-03-03 13:56:38 +0000 Tim-Philipp Müller <tim@centricular.net>
11984 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
11985 Original commit message from CVS:
11986 * docs/libs/gst-plugins-base-libs-sections.txt:
11987 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
11988 (gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
11989 (gst_mixer_message_get_type),
11990 (gst_mixer_message_parse_option_changed),
11991 (gst_mixer_message_parse_options_list_changed):
11992 * gst-libs/gst/interfaces/mixer.h: (GstMixerType),
11993 (GST_MIXER_MESSAGE_OPTION_CHANGED),
11994 (GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
11995 (GST_MIXER_MESSAGE_MIXER_CHANGED):
11996 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
11997 and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
11999 2008-03-03 13:50:18 +0000 Tim-Philipp Müller <tim@centricular.net>
12001 gst-libs/gst/interfaces/mixeroptions.*: API: add GstMixerOptions::get_values vfunc (#519906)
12002 Original commit message from CVS:
12003 * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
12004 (gst_mixer_options_get_values):
12005 * gst-libs/gst/interfaces/mixeroptions.h:
12006 (GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
12007 (_GstMixerOptions), (_GstMixerOptionsClass):
12008 API: add GstMixerOptions::get_values vfunc (#519906)
12010 2008-03-03 12:01:15 +0000 Peter Kjellerstedt <pkj@axis.com>
12012 configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#4...
12013 Original commit message from CVS:
12015 Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
12016 plug-ins are included/excluded. (#498222)
12018 2008-03-03 06:22:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12020 gst/typefind/gsttypefindfunctions.c: Add typefinder for IMelody files, using audio/x-imelody.
12021 Original commit message from CVS:
12022 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
12023 Add typefinder for IMelody files, using audio/x-imelody.
12026 2008-03-03 06:04:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12028 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
12029 Original commit message from CVS:
12030 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
12031 * ext/alsa/gstalsasink.c: (set_hwparams):
12032 * ext/alsa/gstalsasrc.c: (set_hwparams):
12033 * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
12034 * ext/ogg/gstoggmux.h:
12035 * ext/ogg/gstogmparse.c:
12036 * gst-libs/gst/audio/audio.c:
12037 * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
12038 * gst-libs/gst/pbutils/missing-plugins.c:
12039 (gst_missing_uri_sink_message_new),
12040 (gst_missing_element_message_new),
12041 (gst_missing_decoder_message_new),
12042 (gst_missing_encoder_message_new):
12043 * gst-libs/gst/rtp/gstbasertppayload.c:
12044 * gst-libs/gst/rtp/gstrtcpbuffer.c:
12045 (gst_rtcp_packet_bye_get_reason):
12046 * gst/audioconvert/gstaudioconvert.c:
12047 * gst/audioresample/gstaudioresample.c:
12048 * gst/ffmpegcolorspace/imgconvert.c:
12049 * gst/playback/test.c: (gen_video_element), (gen_audio_element):
12050 * gst/typefind/gsttypefindfunctions.c:
12051 * gst/videoscale/vs_4tap.c:
12052 * gst/videoscale/vs_4tap.h:
12053 * sys/v4l/gstv4lelement.c:
12054 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
12055 * sys/v4l/v4l_calls.c:
12056 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
12057 (gst_v4lsrc_try_capture):
12058 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
12059 (gst_ximagesink_ximage_new):
12060 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
12061 (gst_xvimagesink_xvimage_new):
12062 * tests/check/elements/audioconvert.c:
12063 * tests/check/elements/audioresample.c:
12064 (fail_unless_perfect_stream):
12065 * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
12066 * tests/check/elements/decodebin.c:
12067 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
12068 (setup_gdpdepay_streamheader):
12069 * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
12070 (setup_gdppay_streamheader):
12071 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
12072 * tests/check/elements/multifdsink.c: (setup_multifdsink):
12073 * tests/check/elements/textoverlay.c:
12074 * tests/check/elements/videorate.c: (setup_videorate):
12075 * tests/check/elements/videotestsrc.c: (setup_videotestsrc):
12076 * tests/check/elements/volume.c: (setup_volume):
12077 * tests/check/elements/vorbisdec.c: (setup_vorbisdec):
12078 * tests/check/elements/vorbistag.c:
12079 * tests/check/generic/clock-selection.c:
12080 * tests/check/generic/states.c: (setup), (teardown):
12081 * tests/check/libs/cddabasesrc.c:
12082 * tests/check/libs/video.c:
12083 * tests/check/pipelines/gio.c:
12084 * tests/check/pipelines/oggmux.c:
12085 * tests/check/pipelines/simple-launch-lines.c:
12086 (simple_launch_lines_suite):
12087 * tests/check/pipelines/streamheader.c:
12088 * tests/check/pipelines/theoraenc.c:
12089 * tests/check/pipelines/vorbisdec.c:
12090 * tests/check/pipelines/vorbisenc.c:
12091 * tests/examples/seek/scrubby.c:
12092 * tests/examples/seek/seek.c: (query_positions_elems),
12093 (query_positions_pads):
12094 * tests/icles/stress-xoverlay.c: (myclock):
12095 Correct all relevant warnings found by the sparse semantic code
12096 analyzer. This include marking several symbols static, using
12097 NULL instead of 0 for pointers and using "foo (void)" instead
12098 of "foo ()" for declarations.
12099 * win32/common/libgstrtp.def:
12100 Add gst_rtp_buffer_set_extension_data to the symbol definition file.
12102 2008-03-02 18:43:15 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
12104 gst/playback/gstplaybin2.c: Make the function signature of the _get_*_tags() functions match the signature of the vfu...
12105 Original commit message from CVS:
12106 Patch by: José Alburquerque <jaalburqu svn gnome org>
12107 * gst/playback/gstplaybin2.c:
12108 Make the function signature of the _get_*_tags() functions match
12109 the signature of the vfuncs they implement, ie. return a
12110 GstTagList rather than a GstStructure, which is more correct,
12111 even if one is typedef'ed to the other (#518940).
12113 2008-03-02 18:32:36 +0000 Tim-Philipp Müller <tim@centricular.net>
12115 gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
12116 Original commit message from CVS:
12117 * gst-libs/gst/rtsp/gstrtspconnection.c:
12118 Don't include unix headers unconditionally (fixes #518037).
12120 2008-03-02 18:24:37 +0000 Tim-Philipp Müller <tim@centricular.net>
12122 tests/check/libs/video.c: Add unit test that makes sure that the strides, offsets and sizes returned for the various ...
12123 Original commit message from CVS:
12124 * tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
12125 (fourcc_list_struct), (fourcc_list), (fourcc_get_size),
12126 (paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
12127 (paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
12128 (paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
12129 (paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
12130 (gst_video_format_is_packed), (video_format_is_packed):
12131 Add unit test that makes sure that the strides, offsets and
12132 sizes returned for the various YUV formats by the new video API
12133 match the old reference implementation in videotestsrc.
12135 2008-03-02 18:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
12137 gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
12138 Original commit message from CVS:
12139 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
12140 (gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
12141 (gst_video_format_is_rgb), (gst_video_format_is_yuv),
12142 (gst_video_format_has_alpha), (gst_video_format_get_row_stride),
12143 (gst_video_format_get_pixel_stride),
12144 (gst_video_format_get_component_width),
12145 (gst_video_format_get_component_height),
12146 (gst_video_format_get_component_offset), (gst_video_format_get_size):
12147 * gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
12148 (GST_VIDEO_FORMAT_Y42B):
12149 API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
12151 2008-03-02 18:07:10 +0000 Tim-Philipp Müller <tim@centricular.net>
12153 gst-libs/gst/video/video.c: YV12 is I420 with swapped components 1 and 2, so the offset of component 1 for I420 shoul...
12154 Original commit message from CVS:
12155 * gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
12156 YV12 is I420 with swapped components 1 and 2, so the offset of
12157 component 1 for I420 should be the offset for component 2 for YV12
12160 2008-02-29 21:48:00 +0000 Rene Stadler <mail@renestadler.de>
12162 sys/v4l/gstv4lelement.c: Add missing semicolon to fix indentation.
12163 Original commit message from CVS:
12164 * sys/v4l/gstv4lelement.c:
12165 Add missing semicolon to fix indentation.
12167 2008-02-29 18:44:36 +0000 Julien Moutte <julien@moutte.net>
12169 ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
12170 Original commit message from CVS:
12171 2008-02-29 Julien Moutte <julien@fluendo.com>
12172 * ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
12173 (gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
12175 if we can do SPDIF output.
12176 * ext/alsa/gstalsa.h:
12177 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
12178 (gst_alsasink_prepare), (gst_alsasink_close),
12179 (gst_alsasink_write):
12180 * ext/alsa/gstalsasink.h: Initial support for SPDIF.
12181 * gst-libs/gst/audio/gstringbuffer.c:
12182 (gst_ring_buffer_parse_caps):
12183 * gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
12185 to support AC3, EC3 and IEC958 buffers.
12187 2008-02-29 17:59:16 +0000 Tim-Philipp Müller <tim@centricular.net>
12189 gst-libs/gst/interfaces/mixer.c: De-cruft and fix message type assertions (NULL is not a really valid mixer message t...
12190 Original commit message from CVS:
12191 * gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
12192 (gst_mixer_message_parse_mute_toggled),
12193 (gst_mixer_message_parse_record_toggled),
12194 (gst_mixer_message_parse_volume_changed),
12195 (gst_mixer_message_parse_option_changed):
12196 De-cruft and fix message type assertions (NULL is not a really
12197 valid mixer message type string).
12199 2008-02-29 14:52:02 +0000 Wim Taymans <wim.taymans@gmail.com>
12201 ext/libvisual/visual.c: When negotiating, actually start from a format that we can support instead of from the too ge...
12202 Original commit message from CVS:
12203 * ext/libvisual/visual.c: (gst_vis_src_negotiate):
12204 When negotiating, actually start from a format that we can support
12205 instead of from the too generic template.
12207 2008-02-29 12:26:48 +0000 Wim Taymans <wim.taymans@gmail.com>
12209 gst/playback/gstplaybin2.c: Enable vis setting.
12210 Original commit message from CVS:
12211 * gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
12212 Enable vis setting.
12213 * gst/playback/gstplaysink.c: (gst_play_sink_init),
12214 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
12215 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
12217 Implement vis switching while playing.
12219 2008-02-29 00:04:57 +0000 David Schleef <ds@schleef.org>
12221 gst-libs/gst/riff/riff-media.c: Add Dirac mapping
12222 Original commit message from CVS:
12223 * gst-libs/gst/riff/riff-media.c: Add Dirac mapping
12225 2008-02-28 10:54:14 +0000 Peter Kjellerstedt <pkj@axis.com>
12227 gst/tcp/: Removed fdset and stress test, they are now known as GstPoll in core.
12228 Original commit message from CVS:
12229 Patch by: Peter Kjellerstedt <pkj at axis com>
12230 * gst/tcp/Makefile.am:
12231 * gst/tcp/fdsetstress.c:
12232 * gst/tcp/gstfdset.c:
12233 * gst/tcp/gstfdset.h:
12234 Removed fdset and stress test, they are now known as GstPoll in
12236 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
12237 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
12238 (gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
12239 (gst_multi_fd_sink_handle_client_write),
12240 (gst_multi_fd_sink_queue_buffer),
12241 (gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
12242 (gst_multi_fd_sink_stop):
12243 * gst/tcp/gstmultifdsink.h:
12244 * gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
12245 (gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
12246 (gst_tcp_gdp_read_caps):
12247 * gst/tcp/gsttcp.h:
12248 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
12249 (gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
12250 (gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
12251 * gst/tcp/gsttcpclientsink.h:
12252 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
12253 (gst_tcp_client_src_create), (gst_tcp_client_src_start),
12254 (gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
12255 * gst/tcp/gsttcpclientsrc.h:
12256 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
12257 (gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
12258 * gst/tcp/gsttcpserversink.h:
12259 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
12260 (gst_tcp_server_src_create), (gst_tcp_server_src_start),
12261 (gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
12262 * gst/tcp/gsttcpserversrc.h:
12263 Port to GstPoll. See #505417.
12265 2008-02-28 09:54:14 +0000 Wim Taymans <wim.taymans@gmail.com>
12268 Patch Changelog a bit to give credit and refer to the relevant bug.
12269 Original commit message from CVS:
12270 Patch Changelog a bit to give credit and refer to the
12273 2008-02-28 09:50:52 +0000 Wim Taymans <wim.taymans@gmail.com>
12275 gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection.
12276 Original commit message from CVS:
12277 * gst-libs/gst/rtsp/gstrtspconnection.c:
12278 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
12279 (gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
12280 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
12281 (gst_rtsp_connection_free), (gst_rtsp_connection_poll),
12282 (gst_rtsp_connection_flush):
12283 * gst-libs/gst/rtsp/gstrtspconnection.h:
12284 Use GstPoll for the rtsp connection.
12286 2008-02-27 12:19:31 +0000 Wim Taymans <wim.taymans@gmail.com>
12288 tests/examples/seek/seek.c: Add combo box for visualisations, populate it with a factory list of all visualisation pl...
12289 Original commit message from CVS:
12290 * tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
12291 (init_visualization_features), (vis_combo_cb), (shot_cb), (main):
12292 Add combo box for visualisations, populate it with a factory list
12293 of all visualisation plugins, configure vis plugin instance in
12296 2008-02-27 10:55:03 +0000 Wim Taymans <wim.taymans@gmail.com>
12298 tests/check/libs/rtp.c: Add check for RTP buffer defaults, padding and marker bit API.
12299 Original commit message from CVS:
12300 * tests/check/libs/rtp.c: (GST_START_TEST):
12301 Add check for RTP buffer defaults, padding and marker bit API.
12303 2008-02-27 10:42:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12305 gst-libs/gst/cdda/sha1.c: Use memcpy() instead of upcasting a byte array to long *. This fixes an unaligned memory ac...
12306 Original commit message from CVS:
12307 * gst-libs/gst/cdda/sha1.c: (sha_transform):
12308 Use memcpy() instead of upcasting a byte array to long *. This
12309 fixes an unaligned memory access, resulting in SIGBUS on IA64.
12310 This should be ported to GCheckSum once we can use GLib 2.16.
12311 Partially fixes bug #500833.
12313 2008-02-27 10:23:27 +0000 Tim-Philipp Müller <tim@centricular.net>
12315 gst-libs/gst/tag/gsttagdemux.c: Push tag event after the newsegment event. Log the pointer of the buffer we're actual...
12316 Original commit message from CVS:
12317 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
12318 Push tag event after the newsegment event. Log the pointer of
12319 the buffer we're actually going to push rather than the buffer
12320 we're feeding to _make_metadata_writable().
12322 2008-02-25 07:21:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12324 gst/typefind/gsttypefindfunctions.c: Comment smoke typefinder for now. The smokedec plugin needs one frame per buffer...
12325 Original commit message from CVS:
12326 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
12327 Comment smoke typefinder for now. The smokedec plugin needs one
12328 frame per buffer but we have no parser yet, thus it simply crashes
12329 in most situations.
12331 2008-02-25 06:48:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12333 gst/typefind/gsttypefindfunctions.c: Add typefinder for the smoke video codec. Copied from the jpeg plugin.
12334 Original commit message from CVS:
12335 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
12336 Add typefinder for the smoke video codec. Copied from the jpeg plugin.
12338 2008-02-25 06:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12340 gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
12341 Original commit message from CVS:
12342 * gst/typefind/gsttypefindfunctions.c: (mid_type_find),
12344 Add midi typefinder, copied from the timidity plugin.
12346 2008-02-23 09:51:26 +0000 Tomasz Sałaciński <tsalacinski@gmail.com>
12348 Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
12349 Original commit message from CVS:
12350 Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
12351 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
12352 * tests/check/elements/subparse.c: (test_microdvd_with_italics),
12354 Forward slashes at the beginning and end of a line also signify
12355 italics (Fixes: #518162).
12357 2008-02-22 06:38:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12359 tests/check/gst-plugins-base.supp: Add a suppression for a cached value in GIO that wasn't moved while moving gio fro...
12360 Original commit message from CVS:
12361 * tests/check/gst-plugins-base.supp:
12362 Add a suppression for a cached value in GIO that wasn't moved
12363 while moving gio from -bad to -base.
12365 2008-02-22 05:27:24 +0000 Brian Cameron <brian.cameron@sun.com>
12367 configure.ac: Don't hardcode -Wall and -Werror for configure checks, this fails with non-GCC compilers. Fixes bug #51...
12368 Original commit message from CVS:
12369 Patch by: Brian Cameron <brian dot cameron at sun dot com>
12371 Don't hardcode -Wall and -Werror for configure checks, this fails
12372 with non-GCC compilers. Fixes bug #517991.
12374 2008-02-21 08:05:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12376 gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
12377 Original commit message from CVS:
12378 * gst/audiotestsrc/gstaudiotestsrc.c:
12379 Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
12381 2008-02-20 15:37:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12383 ext/gnomevfs/gstgnomevfssink.c: Return FALSE when seeking for a new segment fails instead of silently ignoring the fa...
12384 Original commit message from CVS:
12385 * ext/gnomevfs/gstgnomevfssink.c:
12386 (gst_gnome_vfs_sink_handle_event):
12387 Return FALSE when seeking for a new segment fails instead
12388 of silently ignoring the failure and appending every buffer
12389 that comes for the new segment.
12391 2008-02-20 11:52:28 +0000 Wim Taymans <wim.taymans@gmail.com>
12393 gst/playback/gstplaysink.c: Recursively search the sink element for a last-frame property so that we can also find th...
12394 Original commit message from CVS:
12395 * gst/playback/gstplaysink.c: (find_property),
12396 (gst_play_sink_find_property), (gen_video_chain),
12397 (gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
12398 Recursively search the sink element for a last-frame property so that we
12399 can also find the property in autovideosink and friends that don't
12400 always proxy the internal sink properties.
12402 2008-02-19 20:42:09 +0000 Tim-Philipp Müller <tim@centricular.net>
12404 gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
12405 Original commit message from CVS:
12406 * gst-libs/gst/audio/multichannel.c:
12407 (GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
12408 (gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
12409 (gst_audio_set_structure_channel_positions_list),
12410 (add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
12411 (gst_audio_fixate_channel_positions):
12412 Fix confusing terminology in docs and code: structure fields are
12413 'fields' and not 'properties'.
12415 2008-02-19 20:36:58 +0000 Tim-Philipp Müller <tim@centricular.net>
12417 gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
12418 Original commit message from CVS:
12419 * gst-libs/gst/audio/multichannel.c:
12420 (gst_audio_check_channel_positions), (add_list_to_struct):
12421 Give more useful warning messages if one of the channel
12422 layout enums passed to us is invalid and if the "channels"
12423 field in the caps has a GType we don't expect.
12425 2008-02-19 20:22:09 +0000 Tim-Philipp Müller <tim@centricular.net>
12427 gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
12428 Original commit message from CVS:
12429 * gst-libs/gst/audio/multichannel.c:
12430 Fix typo in docs blurb.
12432 2008-02-19 16:16:55 +0000 Josep Torra Valles <josep@fluendo.com>
12434 gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS typefind lookup to fix typefinding on HD clips.
12435 Original commit message from CVS:
12436 2008-02-19 Julien Moutte <julien@fluendo.com>
12437 Patch by: Josep Torra Valles <josep@fluendo.com>
12438 * gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
12439 typefind lookup to fix typefinding on HD clips.
12441 2008-02-19 15:50:37 +0000 Tim-Philipp Müller <tim@centricular.net>
12443 gst/playback/gstscreenshot.*: Fix up copyright (I rewrote the GStreamer-0.10 code for this from scratch back in the d...
12444 Original commit message from CVS:
12445 * gst/playback/gstscreenshot.c:
12446 * gst/playback/gstscreenshot.h:
12447 Fix up copyright (I rewrote the GStreamer-0.10 code for
12448 this from scratch back in the days).
12450 2008-02-19 15:02:33 +0000 Wim Taymans <wim.taymans@gmail.com>
12452 gst/playback/: Add screenshot conversion code from totem.
12453 Original commit message from CVS:
12454 * gst/playback/Makefile.am:
12455 * gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
12456 (create_element), (gst_play_frame_conv_convert):
12457 * gst/playback/gstscreenshot.h:
12458 Add screenshot conversion code from totem.
12459 * gst/playback/gstplay-marshal.list:
12460 * gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
12461 (gst_play_bin_class_init), (gst_play_bin_convert_frame),
12462 (gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
12463 Implement frame property to get a color-unconverted snapshot.
12464 Implement convert-frame action signal to get a converted snapshot image.
12465 Configure connection speed in uridecodebin.
12466 Document some more properties.
12467 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
12468 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
12469 (gst_play_sink_get_last_frame):
12470 * gst/playback/gstplaysink.h:
12471 Use last-buffer property of the video sink to get a video snapshot.
12472 * tests/examples/seek/seek.c: (shot_cb), (main):
12473 Add snapshot button for playbin2 and use the frame property to save the
12474 frame as a png in the current directory.
12476 2008-02-19 11:45:56 +0000 Josep Torra Valles <josep@fluendo.com>
12478 gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
12479 Original commit message from CVS:
12480 Patch by: Josep Torra Valles <josep at fluendo dot com>
12481 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
12483 Add typefinding support for h264 elementary streams.
12486 2008-02-18 13:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12488 configure.ac: Require CVS of core for new API in collectpads.
12489 Original commit message from CVS:
12491 Require CVS of core for new API in collectpads.
12492 * gst/adder/gstadder.c:
12493 Use new API to make adder sparse stream aware.
12495 2008-02-18 11:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
12497 gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
12498 Original commit message from CVS:
12499 * gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
12501 Get the object data correct so that we can remove our channels
12503 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
12504 (gen_vis_chain), (gst_play_sink_reconfigure),
12505 (gst_play_sink_request_pad):
12506 Add option to disable async behaviour in the sinks when possible. This
12507 makes it possible to avoid an audio queue when dealing with
12509 Add option to add a queue for the audio path.
12510 * tests/examples/seek/seek.c: (clear_streams), (update_streams),
12512 Disable the vis checkbox to match the defaults of playbin2.
12513 Only get the stream info when we need to.
12515 2008-02-17 05:15:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12517 ext/gio/: Don't use async operations as they require a running main loop.
12518 Original commit message from CVS:
12519 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
12520 (gst_gio_base_sink_set_stream):
12521 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
12522 (gst_gio_base_src_set_stream):
12523 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
12524 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
12525 Don't use async operations as they require a running main loop.
12526 This makes us block again when closing streams and unable
12527 to mount the enclosing volume of an URI if it isn't yet.
12529 2008-02-15 18:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
12531 gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
12532 Original commit message from CVS:
12533 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
12534 (gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
12535 (gen_vis_chain), (gst_play_sink_reconfigure),
12536 (gst_play_sink_request_pad):
12537 Move tee in front of the audio and vis pipelines.
12538 Add queue for audio for now.
12539 Add visualisation support.
12540 * tests/examples/seek/seek.c: (main):
12541 Visualisation is by default disabled.
12543 2008-02-15 11:58:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12545 ext/gio/: Improve debugging a bit.
12546 Original commit message from CVS:
12547 * ext/gio/gstgiobasesink.c: (close_stream_cb):
12548 * ext/gio/gstgiobasesrc.c: (close_stream_cb):
12549 Improve debugging a bit.
12550 * ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
12551 * ext/gio/gstgiosink.h:
12552 * ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
12553 * ext/gio/gstgiosrc.h:
12554 Try to mount the enclosing volume of a GFile if it isn't mounted
12555 yet. This requires us to wait for an async operation to finish, done
12556 with an nested GMainLoop. Authentication is not supported yet, will
12559 2008-02-14 18:24:42 +0000 Wim Taymans <wim.taymans@gmail.com>
12561 gst/playback/: Add mute property.
12562 Original commit message from CVS:
12563 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
12564 (gst_play_bin_set_property), (gst_play_bin_get_property),
12565 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
12566 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
12567 (gst_play_sink_get_mute), (gen_audio_chain):
12568 * gst/playback/gstplaysink.h:
12570 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
12571 (gst_selector_pad_chain):
12572 * gst/playback/gststreamselector.h:
12573 Make sure we forward the event only once.
12574 * tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
12575 Add and implement the mute button for playbin2.
12577 2008-02-13 14:34:55 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
12579 ext/alsa/gstalsasink.c: Add some more debug info.
12580 Original commit message from CVS:
12581 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
12582 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
12583 Add some more debug info.
12584 Make sure we never return a negative delay. Fixes #516246.
12586 2008-02-12 20:09:07 +0000 Tim-Philipp Müller <tim@centricular.net>
12588 ext/alsa/gstalsasink.c: Revert patch that makes the sink hold the object lock when calling snd_pcm_delay(), since it ...
12589 Original commit message from CVS:
12590 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
12591 Revert patch that makes the sink hold the object lock when
12592 calling snd_pcm_delay(), since it breaks playback for me.
12594 2008-02-12 19:50:36 +0000 Julien Moutte <julien@moutte.net>
12596 tests/examples/seek/seek.c: Add some seek flags when changing rate.
12597 Original commit message from CVS:
12598 2008-02-12 Julien Moutte <julien@fluendo.com>
12599 * tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
12600 some seek flags when changing rate.
12602 2008-02-12 14:51:26 +0000 Wim Taymans <wim.taymans@gmail.com>
12604 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
12605 Original commit message from CVS:
12606 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
12607 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
12608 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
12609 Fix potential leaks.
12610 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
12611 Fix leak when there is no function configured.
12613 2008-02-12 11:36:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12615 sys/v4l/v4lsrc_calls.c: Correctly chain up the finalize method.
12616 Original commit message from CVS:
12617 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
12618 (gst_v4lsrc_buffer_finalize):
12619 Correctly chain up the finalize method.
12621 2008-02-12 09:24:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12623 ext/gio/: Add documentation and example code for giostreamsink/giostreamsrc.
12624 Original commit message from CVS:
12625 * ext/gio/gstgiostreamsink.c:
12626 * ext/gio/gstgiostreamsrc.c:
12627 Add documentation and example code for giostreamsink/giostreamsrc.
12628 * tests/check/pipelines/gio.c: (GST_START_TEST):
12629 Ask the GMemoryOutputStream for the data instead of assuming that
12630 the pointer to the data stayed the same. It could've been realloc'ed.
12632 2008-02-12 08:55:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12634 ext/gio/: Make the documentation of giosink/giosrc complete, large parts are based on the gnomevfssink/gnomevfssrc docs.
12635 Original commit message from CVS:
12636 * ext/gio/gstgiosink.c:
12637 * ext/gio/gstgiosrc.c:
12638 Make the documentation of giosink/giosrc complete, large parts
12639 are based on the gnomevfssink/gnomevfssrc docs.
12641 2008-02-12 08:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12643 docs/plugins/: Add the GIO documentation again and while at that run make update.
12644 Original commit message from CVS:
12645 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
12646 * docs/plugins/gst-plugins-base-plugins-sections.txt:
12647 * docs/plugins/gst-plugins-base-plugins.args:
12648 * docs/plugins/gst-plugins-base-plugins.hierarchy:
12649 * docs/plugins/gst-plugins-base-plugins.interfaces:
12650 * docs/plugins/gst-plugins-base-plugins.prerequisites:
12651 * docs/plugins/gst-plugins-base-plugins.signals:
12652 * docs/plugins/inspect/plugin-adder.xml:
12653 * docs/plugins/inspect/plugin-audioconvert.xml:
12654 * docs/plugins/inspect/plugin-audiorate.xml:
12655 * docs/plugins/inspect/plugin-audioresample.xml:
12656 * docs/plugins/inspect/plugin-decodebin.xml:
12657 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
12658 * docs/plugins/inspect/plugin-gdp.xml:
12659 * docs/plugins/inspect/plugin-gio.xml:
12660 * docs/plugins/inspect/plugin-gnomevfs.xml:
12661 * docs/plugins/inspect/plugin-libvisual.xml:
12662 * docs/plugins/inspect/plugin-ogg.xml:
12663 * docs/plugins/inspect/plugin-pango.xml:
12664 * docs/plugins/inspect/plugin-playback.xml:
12665 * docs/plugins/inspect/plugin-queue2.xml:
12666 * docs/plugins/inspect/plugin-subparse.xml:
12667 * docs/plugins/inspect/plugin-theora.xml:
12668 * docs/plugins/inspect/plugin-uridecodebin.xml:
12669 * docs/plugins/inspect/plugin-videorate.xml:
12670 * docs/plugins/inspect/plugin-videoscale.xml:
12671 * docs/plugins/inspect/plugin-volume.xml:
12672 * docs/plugins/inspect/plugin-vorbis.xml:
12673 Add the GIO documentation again and while at that run make update.
12675 2008-02-11 20:23:44 +0000 Tim-Philipp Müller <tim@centricular.net>
12677 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...
12678 Original commit message from CVS:
12679 * ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
12680 * ext/alsa/gstalsasink.c: (set_swparams):
12681 * ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
12682 Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
12683 against libasound >= 1.0.16, since it's been deprecated in
12684 0.10.16, and alignment is always 1 then, apparently. (#512899)
12686 2008-02-11 18:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
12688 gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
12689 Original commit message from CVS:
12690 * gst/playback/gstplaybin.c: (gen_audio_element):
12691 * gst/playback/gstplaysink.c: (gen_audio_chain):
12692 Handle case where we can't create the volume element a bit
12695 2008-02-11 18:02:13 +0000 Tim-Philipp Müller <tim@centricular.net>
12697 ext/gnomevfs/: Add support for https protocol. Fixes #510229.
12698 Original commit message from CVS:
12699 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
12700 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
12701 Add support for https protocol. Fixes #510229.
12703 2008-02-11 17:03:18 +0000 Alan Peevers <peeves@pacbell.net>
12705 ext/alsa/gstalsasink.c: Take appropriate lock when calling alsa methods.
12706 Original commit message from CVS:
12707 2008-02-11 Julien Moutte <julien@fluendo.com>
12708 Patch by: Alan Peevers <peeves@pacbell.net>
12709 * ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
12710 lock when calling alsa methods.
12712 2008-02-11 13:03:13 +0000 Tim-Philipp Müller <tim@centricular.net>
12714 gst/typefind/gsttypefindfunctions.c: Bump rank of jpeg and png typefinders, which will return maximum probability in ...
12715 Original commit message from CVS:
12716 * gst/typefind/gsttypefindfunctions.c:
12717 Bump rank of jpeg and png typefinders, which will return maximum
12718 probability in the most common cases (thus short-circuiting more
12719 expensive typefinders like the mp3 one for these two quite common
12722 2008-02-11 09:48:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12724 ext/theora/theoraparse.c: Fix long description of the theora parser to be more verbose than just the type name.
12725 Original commit message from CVS:
12726 * ext/theora/theoraparse.c:
12727 Fix long description of the theora parser to be more verbose than just
12730 2008-02-11 06:47:50 +0000 Branko Čibej <brane@xbc.nu>
12732 sys/xvimage/xvimagesink.c: Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
12733 Original commit message from CVS:
12734 Patch by: Branko Čibej <brane at xbc dot nu>
12735 * sys/xvimage/xvimagesink.c:
12736 Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
12739 2008-02-09 10:41:36 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
12741 gst/playback/gstplaybasebin.c: Set is_dynamic as True if there are elements with both request and sometimes src pad t...
12742 Original commit message from CVS:
12743 * gst/playback/gstplaybasebin.c:
12744 Set is_dynamic as True if there are elements with both request
12745 and sometimes src pad templates instead of breaking out when it
12746 finds the first pad template that is a src.
12748 2008-02-08 18:17:51 +0000 Wim Taymans <wim.taymans@gmail.com>
12750 tests/examples/seek/seek.c: Add some stream switching and volume gui for playbin2.
12751 Original commit message from CVS:
12752 * tests/examples/seek/seek.c: (stop_cb), (clear_streams),
12753 (update_streams), (video_combo_cb), (audio_combo_cb),
12754 (text_combo_cb), (volume_spinbutton_changed_cb), (main):
12755 Add some stream switching and volume gui for playbin2.
12757 2008-02-08 17:47:37 +0000 Wim Taymans <wim.taymans@gmail.com>
12759 gst/playback/gstplay-marshal.list: Added marshal for streamselector Tags.
12760 Original commit message from CVS:
12761 * gst/playback/gstplay-marshal.list:
12762 Added marshal for streamselector Tags.
12763 * gst/playback/gstplaybasebin.c: (set_active_source):
12764 Streamselector now selects pads based on the pad object instead of its
12766 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
12767 (init_group), (gst_play_bin_init), (get_group), (get_tags),
12768 (gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
12769 (gst_play_bin_get_text_tags),
12770 (gst_play_bin_set_current_video_stream),
12771 (gst_play_bin_set_current_audio_stream),
12772 (gst_play_bin_set_current_text_stream),
12773 (gst_play_bin_set_property), (gst_play_bin_get_property),
12774 (pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
12775 Remove option to mute streams with the current-a/v/t property, we have
12776 this functionality in the flags.
12777 Add signals to notify when the number of A/V/T channels changed.
12778 Add action signals to get tags for the A/V/T streams.
12779 Implement setting the current A/V/T stream.
12780 Rearrange some things to simplify stream selection.
12782 * gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
12783 (gst_play_sink_get_volume), (gst_play_sink_set_property),
12784 (gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
12785 (activate_vis), (gst_play_sink_reconfigure):
12786 * gst/playback/gstplaysink.h:
12787 Add and implement volume setting methods.
12788 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
12789 (gst_selector_pad_finalize), (gst_selector_pad_get_property),
12790 (gst_selector_pad_event), (gst_stream_selector_class_init),
12791 (gst_stream_selector_init), (gst_stream_selector_finalize),
12792 (gst_stream_selector_set_property),
12793 (gst_stream_selector_get_property),
12794 (gst_stream_selector_get_linked_pad),
12795 (gst_stream_selector_request_new_pad):
12796 * gst/playback/gststreamselector.h:
12797 Add pad properties for tags and status of pads.
12799 Make active pad selection based on pad object instead of name.
12801 2008-02-08 16:10:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12803 configure.ac: Revert last change as we now check in gtk-doc.m4 for sed.
12804 Original commit message from CVS:
12806 Revert last change as we now check in gtk-doc.m4 for sed.
12808 2008-02-08 14:54:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12810 configure.ac: Find and subst SED when building the docs.
12811 Original commit message from CVS:
12813 Find and subst SED when building the docs.
12815 2008-02-08 14:34:41 +0000 Julien Moutte <julien@moutte.net>
12817 tests/examples/seek/seek.c: Make sure bus signals are reconnected when pressing STOP and then PLAY again for a parse ...
12818 Original commit message from CVS:
12819 2008-02-08 Julien Moutte <julien@fluendo.com>
12820 * tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
12821 (main): Make sure bus signals are reconnected when pressing STOP
12822 and then PLAY again for a parse launch pipeline. Fix a ref leak
12824 * win32/common/config.h: Updated.
12826 2008-02-08 00:57:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12828 configure.ac: Make DISABLE_DEPRECATED defined *only* during CVS, not during pre-releases or releases.
12829 Original commit message from CVS:
12831 Make DISABLE_DEPRECATED defined *only* during CVS, not during
12832 pre-releases or releases.
12834 2008-02-08 00:45:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12836 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is reporting
12837 Original commit message from CVS:
12839 * ext/gio/Makefile.am:
12840 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
12843 2008-02-07 23:40:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12845 docs/plugins/Makefile.am: Add the headers which need scanning for the GIO plugin. The rest of the docs still need mig...
12846 Original commit message from CVS:
12847 * docs/plugins/Makefile.am:
12848 Add the headers which need scanning for the GIO plugin. The rest of
12849 the docs still need migrating.
12851 2008-02-07 23:22:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12853 Add gio in a few more places.
12854 Original commit message from CVS:
12856 * tests/check/Makefile.am:
12857 * tests/check/pipelines/.cvsignore:
12858 Add gio in a few more places.
12860 2008-02-07 23:18:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12862 Move gio plugin from -bad and mark as experimental.
12863 Original commit message from CVS:
12866 * tests/check/Makefile.am:
12867 Move gio plugin from -bad and mark as experimental.
12869 2008-02-07 22:39:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12871 gst-libs/gst/interfaces/: Comment out a couple of other things which break the build when
12872 Original commit message from CVS:
12873 * gst-libs/gst/interfaces/mixeroptions.c:
12874 * gst-libs/gst/interfaces/mixertrack.c:
12875 Comment out a couple of other things which break the build when
12876 GST_DISABLE_DEPRECATED isn't on but -Werror is.
12878 2008-02-07 18:28:29 +0000 Tim-Philipp Müller <tim@centricular.net>
12880 docs/libs/gst-plugins-base-libs-sections.txt: Fix pbutils header.
12881 Original commit message from CVS:
12882 * docs/libs/gst-plugins-base-libs-sections.txt:
12883 Fix pbutils header.
12885 2008-02-07 18:07:41 +0000 Christian Schaller <uraeus@gnome.org>
12887 * gst-plugins-base.spec.in:
12888 commit spec file update which includes all the split .pc files
12889 Original commit message from CVS:
12890 commit spec file update which includes all the split .pc files
12892 2008-02-07 12:17:49 +0000 Wim Taymans <wim.taymans@gmail.com>
12894 gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
12895 Original commit message from CVS:
12896 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
12897 Fix compiler warning.
12899 2008-02-07 11:00:45 +0000 Peter Kjellerstedt <pkj@axis.com>
12901 gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
12902 Original commit message from CVS:
12903 Patch by: Peter Kjellerstedt <pkj at axis com>
12904 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
12905 Clear the addrinfo struct using memset. Fixes #514937.
12907 2008-02-06 15:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
12909 gst/tcp/gstfdset.h: Remove unused field to same some memory.
12910 Original commit message from CVS:
12911 * gst/tcp/gstfdset.h:
12912 Remove unused field to same some memory.
12913 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
12914 Mark action signals as such.
12916 2008-02-06 13:35:58 +0000 Michael Smith <msmith@xiph.org>
12918 ext/theora/theoradec.c: Increment granulepos for new-bitstream versions appropriately.
12919 Original commit message from CVS:
12920 * ext/theora/theoradec.c: (_theora_granule_frame),
12922 Increment granulepos for new-bitstream versions appropriately.
12925 2008-02-04 11:51:31 +0000 Wim Taymans <wim.taymans@gmail.com>
12927 tests/examples/seek/seek.c: Remove obsolete stream_time reset after flushing seek, core does that automatically now.
12928 Original commit message from CVS:
12929 * tests/examples/seek/seek.c: (do_seek),
12930 (rate_spinbutton_changed_cb), (update_streams), (main):
12931 Remove obsolete stream_time reset after flushing seek, core does that
12933 Improve accuracy of speed spinbutton.
12934 Only do playbin2 stuff when we actually use it.
12936 2008-02-02 17:29:32 +0000 Tim-Philipp Müller <tim@centricular.net>
12938 tests/check/Makefile.am: Revert previous change of the test environment's GST_PLUGIN_PATH.
12939 Original commit message from CVS:
12940 * tests/check/Makefile.am:
12941 Revert previous change of the test environment's GST_PLUGIN_PATH.
12942 The problem is not with the plugins, but with element factories
12943 and only occurs if elements are split out from existing plugins
12944 or if plugins change name (see #512740).
12946 2008-02-02 15:32:23 +0000 Tim-Philipp Müller <tim@centricular.net>
12948 tests/check/Makefile.am: Fix the tests environment's GST_PLUGIN_PATH: we want the directory with the core's plugins f...
12949 Original commit message from CVS:
12950 * tests/check/Makefile.am:
12951 Fix the tests environment's GST_PLUGIN_PATH: we want the directory
12952 with the core's plugins first and our local build directories last,
12953 since we might be building against an installed core, and that
12954 core's plugin directory may contain older or other versions of
12955 our own -base plugins, but we really do want to test our local
12956 ones (if there are multiple plugins or element factories with the
12957 same name, those inspected last will trump those read in earlier).
12958 Fixes #512740 for the most part.
12960 2008-02-02 07:13:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12962 Use gmtime_r if available as gmtime is not MT-safe.
12963 Original commit message from CVS:
12965 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
12966 Use gmtime_r if available as gmtime is not MT-safe.
12969 2008-02-02 06:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12971 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
12972 Original commit message from CVS:
12973 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
12974 Cast glong to time_t as time_t might have a different type on
12975 other platforms, like FreeBSD, and we get a compiler warning
12976 otherwise. Fixes bug #511825.
12978 2008-02-01 16:44:21 +0000 Wim Taymans <wim.taymans@gmail.com>
12980 gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
12981 Original commit message from CVS:
12982 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
12983 (get_group), (get_n_pads), (gst_play_bin_get_property),
12984 (pad_added_cb), (no_more_pads_cb), (perform_eos),
12985 (autoplug_select_cb), (deactivate_group):
12986 Remove stream-info, we going for something easier.
12987 Refactor getting the current group.
12988 Implement getting the number of audio/video/text streams.
12989 * gst/playback/gststreamselector.c:
12990 (gst_stream_selector_class_init), (gst_stream_selector_init),
12991 (gst_stream_selector_get_property),
12992 (gst_stream_selector_request_new_pad),
12993 (gst_stream_selector_release_pad):
12994 * gst/playback/gststreamselector.h:
12995 Add property for number of pads.
12996 * tests/examples/seek/seek.c: (set_scale), (update_flag),
12997 (vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
12998 (text_toggle_cb), (update_streams), (msg_async_done),
12999 (msg_state_changed), (main):
13000 Block slider callback when updating the slider position.
13001 Add gui elements for controlling playbin2.
13002 Add callback for async_done that updates position/duration.
13004 2008-02-01 12:56:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13006 docs/plugins/: First round of plugin docs cleansups.
13007 Original commit message from CVS:
13008 * docs/plugins/Makefile.am:
13009 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
13010 * docs/plugins/gst-plugins-base-plugins-sections.txt:
13011 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13012 * docs/plugins/gst-plugins-base-plugins.interfaces:
13013 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13014 First round of plugin docs cleansups.
13015 * docs/plugins/inspect/plugin-adder.xml:
13016 * docs/plugins/inspect/plugin-alsa.xml:
13017 * docs/plugins/inspect/plugin-audioconvert.xml:
13018 * docs/plugins/inspect/plugin-audiorate.xml:
13019 * docs/plugins/inspect/plugin-audioresample.xml:
13020 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13021 * docs/plugins/inspect/plugin-cdparanoia.xml:
13022 * docs/plugins/inspect/plugin-decodebin.xml:
13023 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13024 * docs/plugins/inspect/plugin-gdp.xml:
13025 * docs/plugins/inspect/plugin-gnomevfs.xml:
13026 * docs/plugins/inspect/plugin-libvisual.xml:
13027 * docs/plugins/inspect/plugin-ogg.xml:
13028 * docs/plugins/inspect/plugin-pango.xml:
13029 * docs/plugins/inspect/plugin-subparse.xml:
13030 * docs/plugins/inspect/plugin-tcp.xml:
13031 * docs/plugins/inspect/plugin-theora.xml:
13032 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13033 * docs/plugins/inspect/plugin-video4linux.xml:
13034 * docs/plugins/inspect/plugin-videorate.xml:
13035 * docs/plugins/inspect/plugin-videoscale.xml:
13036 * docs/plugins/inspect/plugin-videotestsrc.xml:
13037 * docs/plugins/inspect/plugin-volume.xml:
13038 * docs/plugins/inspect/plugin-vorbis.xml:
13039 * docs/plugins/inspect/plugin-ximagesink.xml:
13040 * docs/plugins/inspect/plugin-xvimagesink.xml:
13042 * ext/ogg/Makefile.am:
13043 * ext/ogg/gstoggmux.c:
13044 * ext/ogg/gstoggmux.h:
13045 Add header for oggmux. the c-file needs a doc blob still.
13047 2008-02-01 11:09:16 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13049 Add gst_rtp_buffer_set_extension_data()
13050 Original commit message from CVS:
13051 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
13052 * gst-libs/gst/rtp/gstrtpbuffer.c:
13053 (gst_rtp_buffer_set_extension_data):
13054 * gst-libs/gst/rtp/gstrtpbuffer.h:
13055 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
13056 Add gst_rtp_buffer_set_extension_data()
13057 Add a unit test for this addition. Fixes #511478.
13058 API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
13060 2008-01-31 17:18:46 +0000 Wim Taymans <wim.taymans@gmail.com>
13062 gst-libs/gst/app/gstappsink.c: Really clean up the queue instead of just unreffing all buffers in it.
13063 Original commit message from CVS:
13064 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
13065 Really clean up the queue instead of just unreffing all buffers
13067 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
13068 (gst_app_src_class_init), (gst_app_src_init),
13069 (gst_app_src_dispose), (gst_app_src_finalize):
13070 Fix dispose/finalize.
13072 2008-01-30 15:34:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13074 ext/gio/: Use async variants of the close stream functions to prevent blocking for a long time there and add some mor...
13075 Original commit message from CVS:
13076 * ext/gio/gstgiobasesink.c: (close_stream_cb),
13077 (gst_gio_base_sink_stop), (gst_gio_base_sink_event),
13078 (gst_gio_base_sink_render), (gst_gio_base_sink_set_stream):
13079 * ext/gio/gstgiobasesrc.c: (close_stream_cb),
13080 (gst_gio_base_src_stop), (gst_gio_base_src_create),
13081 (gst_gio_base_src_set_stream):
13082 Use async variants of the close stream functions to prevent blocking
13083 for a long time there and add some more sanity checks for a correct
13086 2008-01-30 14:42:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13088 configure.ac: Back to CVS
13089 Original commit message from CVS:
13093 === release 0.10.17 ===
13095 2008-01-30 14:19:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13101 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13102 * docs/plugins/inspect/plugin-adder.xml:
13103 * docs/plugins/inspect/plugin-alsa.xml:
13104 * docs/plugins/inspect/plugin-audioconvert.xml:
13105 * docs/plugins/inspect/plugin-audiorate.xml:
13106 * docs/plugins/inspect/plugin-audioresample.xml:
13107 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13108 * docs/plugins/inspect/plugin-cdparanoia.xml:
13109 * docs/plugins/inspect/plugin-decodebin.xml:
13110 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13111 * docs/plugins/inspect/plugin-gdp.xml:
13112 * docs/plugins/inspect/plugin-gnomevfs.xml:
13113 * docs/plugins/inspect/plugin-libvisual.xml:
13114 * docs/plugins/inspect/plugin-ogg.xml:
13115 * docs/plugins/inspect/plugin-pango.xml:
13116 * docs/plugins/inspect/plugin-subparse.xml:
13117 * docs/plugins/inspect/plugin-tcp.xml:
13118 * docs/plugins/inspect/plugin-theora.xml:
13119 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13120 * docs/plugins/inspect/plugin-video4linux.xml:
13121 * docs/plugins/inspect/plugin-videorate.xml:
13122 * docs/plugins/inspect/plugin-videoscale.xml:
13123 * docs/plugins/inspect/plugin-videotestsrc.xml:
13124 * docs/plugins/inspect/plugin-volume.xml:
13125 * docs/plugins/inspect/plugin-vorbis.xml:
13126 * docs/plugins/inspect/plugin-ximagesink.xml:
13127 * docs/plugins/inspect/plugin-xvimagesink.xml:
13128 * gst-plugins-base.doap:
13129 * win32/common/config.h:
13131 Original commit message from CVS:
13134 2008-01-30 13:45:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13136 gst-libs/gst/interfaces/: Also remove the conditional registration of the signals that disappeared with the ABI chang...
13137 Original commit message from CVS:
13138 * gst-libs/gst/interfaces/mixeroptions.c:
13139 * gst-libs/gst/interfaces/mixertrack.c:
13140 Also remove the conditional registration of the signals
13141 that disappeared with the ABI change in 0.10.14
13143 2008-01-30 12:28:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13145 gst-libs/gst/rtsp/gstrtspconnection.c: Revert patch to gstrtspconnection.c for brown paper bag release of -base. Re-o...
13146 Original commit message from CVS:
13147 * gst-libs/gst/rtsp/gstrtspconnection.c:
13148 Revert patch to gstrtspconnection.c for brown paper bag
13149 release of -base. Re-opens: #511825
13151 2008-01-30 12:20:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13153 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
13154 Original commit message from CVS:
13155 * gst-libs/gst/interfaces/mixeroptions.h:
13156 * gst-libs/gst/interfaces/mixertrack.h:
13157 Change the way these deprecated function pointers are removed
13158 so that the compiled ABI is unconditionally smaller. This
13159 sets in stone an ABI break that actually occurred when the
13160 things were deprecated in 0.10.14, which seems to be the best
13161 fix as the only known users are oss-mixer and sunaudio-mixer in
13165 2008-01-30 12:19:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13167 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
13168 Original commit message from CVS:
13169 * gst-libs/gst/interfaces/mixeroptions.h:
13170 * gst-libs/gst/interfaces/mixertrack.h:
13171 Change the way these deprecated function pointers are removed
13172 so that the compiled ABI is unconditionally smaller. This
13173 sets in stone an ABI break that actually occurred when the
13174 things were deprecated in 0.10.14, which seems to be the best
13175 fix as the only known users are oss-mixer and sunaudio-mixer in
13178 2008-01-30 11:43:53 +0000 Tim-Philipp Müller <tim@centricular.net>
13180 win32/common/libgstpbutils.def: Export the two new _get_type() functions which are needed by the python bindings.
13181 Original commit message from CVS:
13182 * win32/common/libgstpbutils.def:
13183 Export the two new _get_type() functions which are needed
13184 by the python bindings.
13186 2008-01-29 09:59:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13188 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
13189 Original commit message from CVS:
13190 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
13191 Cast glong to time_t as time_t might have a different type on
13192 other platforms, like FreeBSD, and we get a compiler warning
13193 otherwise. Fixes bug #511825.
13195 2008-01-29 09:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13197 gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
13198 Original commit message from CVS:
13199 * gst-libs/gst/audio/gstaudiofilter.c:
13200 (gst_audio_filter_class_init):
13201 Initialize the GstRingerBuffer class to get it's debug category
13202 initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
13203 category and otherwise we get some g_critical(). Fixes bug #512334.
13205 2008-01-28 23:35:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13207 configure.ac: Back to CVS
13208 Original commit message from CVS:
13212 === release 0.10.16 ===
13214 2008-01-28 23:31:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13220 * docs/plugins/gst-plugins-base-plugins.args:
13221 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13222 * docs/plugins/gst-plugins-base-plugins.interfaces:
13223 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13224 * docs/plugins/gst-plugins-base-plugins.signals:
13225 * docs/plugins/inspect/plugin-adder.xml:
13226 * docs/plugins/inspect/plugin-alsa.xml:
13227 * docs/plugins/inspect/plugin-audioconvert.xml:
13228 * docs/plugins/inspect/plugin-audiorate.xml:
13229 * docs/plugins/inspect/plugin-audioresample.xml:
13230 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13231 * docs/plugins/inspect/plugin-cdparanoia.xml:
13232 * docs/plugins/inspect/plugin-decodebin.xml:
13233 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13234 * docs/plugins/inspect/plugin-gdp.xml:
13235 * docs/plugins/inspect/plugin-gnomevfs.xml:
13236 * docs/plugins/inspect/plugin-libvisual.xml:
13237 * docs/plugins/inspect/plugin-ogg.xml:
13238 * docs/plugins/inspect/plugin-pango.xml:
13239 * docs/plugins/inspect/plugin-subparse.xml:
13240 * docs/plugins/inspect/plugin-tcp.xml:
13241 * docs/plugins/inspect/plugin-theora.xml:
13242 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13243 * docs/plugins/inspect/plugin-video4linux.xml:
13244 * docs/plugins/inspect/plugin-videorate.xml:
13245 * docs/plugins/inspect/plugin-videoscale.xml:
13246 * docs/plugins/inspect/plugin-videotestsrc.xml:
13247 * docs/plugins/inspect/plugin-volume.xml:
13248 * docs/plugins/inspect/plugin-vorbis.xml:
13249 * docs/plugins/inspect/plugin-ximagesink.xml:
13250 * docs/plugins/inspect/plugin-xvimagesink.xml:
13251 * gst-plugins-base.doap:
13252 * win32/common/config.h:
13254 Original commit message from CVS:
13257 2008-01-28 22:15:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13283 Original commit message from CVS:
13286 2008-01-22 15:37:49 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13288 gst-libs/gst/rtp/gstrtpbuffer.c: Fix typos and wrong extension check. Fixes #511274.
13289 Original commit message from CVS:
13290 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
13291 * gst-libs/gst/rtp/gstrtpbuffer.c:
13292 (gst_rtp_buffer_get_extension_data):
13293 Fix typos and wrong extension check. Fixes #511274.
13295 2008-01-18 00:03:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13297 po/sk.po: Oops - add new sk.po mentioned in the LINGUAS I just committed
13298 Original commit message from CVS:
13300 Oops - add new sk.po mentioned in the LINGUAS I just committed
13302 2008-01-17 22:31:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13304 po/LINGUAS: Add ca translation to the disted list.
13305 Original commit message from CVS:
13307 Add ca translation to the disted list.
13308 * win32/vs6/libgstsdp.dsp:
13309 Convert line endings to CRLF
13311 2008-01-17 21:58:53 +0000 Sébastien Moutte <sebastien@moutte.net>
13313 win32/MANIFEST: Add win32/vs6/libgstrtsp.dsp to MANIFEST
13314 Original commit message from CVS:
13316 Add win32/vs6/libgstrtsp.dsp to MANIFEST
13318 2008-01-16 05:40:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13320 Update for API changes in GIO and require GIO 2.15.2 for this.
13321 Original commit message from CVS:
13323 * tests/check/pipelines/gio.c: (GST_START_TEST):
13324 Update for API changes in GIO and require GIO 2.15.2 for this.
13326 2008-01-14 22:20:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13328 win32/common/: Add new API declarations
13329 Original commit message from CVS:
13330 * win32/common/libgstsdp.def:
13331 * win32/common/libgstvideo.def:
13332 Add new API declarations
13334 2008-01-14 17:00:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13336 ext/theora/: Take a 2nd stab at handling libtheora granulepos changes in the decoder and parser by inspecting the bit...
13337 Original commit message from CVS:
13338 * ext/theora/gsttheoradec.h:
13339 * ext/theora/gsttheoraparse.h:
13340 * ext/theora/theoradec.c:
13341 * ext/theora/theoraparse.c:
13342 Take a 2nd stab at handling libtheora granulepos changes in the decoder
13343 and parser by inspecting the bitstream version of the incoming data.
13345 2008-01-14 13:11:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13347 Provide one pkg-config file for every gst-plugins-base library.
13348 Original commit message from CVS:
13350 * pkgconfig/Makefile.am:
13351 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
13352 * pkgconfig/gstreamer-audio.pc.in:
13353 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
13354 * pkgconfig/gstreamer-cdda.pc.in:
13355 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
13356 * pkgconfig/gstreamer-fft.pc.in:
13357 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
13358 * pkgconfig/gstreamer-floatcast.pc.in:
13359 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
13360 * pkgconfig/gstreamer-interfaces.pc.in:
13361 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
13362 * pkgconfig/gstreamer-netbuffer.pc.in:
13363 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
13364 * pkgconfig/gstreamer-pbutils.pc.in:
13365 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
13366 * pkgconfig/gstreamer-riff.pc.in:
13367 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
13368 * pkgconfig/gstreamer-rtp.pc.in:
13369 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
13370 * pkgconfig/gstreamer-rtsp.pc.in:
13371 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
13372 * pkgconfig/gstreamer-sdp.pc.in:
13373 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
13374 * pkgconfig/gstreamer-tag.pc.in:
13375 * pkgconfig/gstreamer-video-uninstalled.pc.in:
13376 * pkgconfig/gstreamer-video.pc.in:
13377 Provide one pkg-config file for every gst-plugins-base library.
13378 This makes linking to those libraries much more intuitive and
13379 provides standard pkg-config behaviour for them. Fixes bug #499697.
13381 2008-01-14 01:19:34 +0000 David Schleef <ds@schleef.org>
13383 gst/videoscale/vs_4tap.c: Fix valgrind error on 4tap scaling method.
13384 Original commit message from CVS:
13385 * gst/videoscale/vs_4tap.c:
13386 Fix valgrind error on 4tap scaling method.
13388 2008-01-13 21:40:45 +0000 Sébastien Moutte <sebastien@moutte.net>
13390 gst-libs/gst/sdp/gstsdpmessage.c: Include Winsock2.h for VS6 and use a different way initialize hints structure so it...
13391 Original commit message from CVS:
13392 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
13393 Include Winsock2.h for VS6 and use a different way initialize
13394 hints structure so it can build with VS6.
13396 * win32/vs6/libgstsdp.dsp:
13397 * win32/common/libgstsdp.def:
13398 Add new files for libgstsdp.
13399 * win32/vs6/grammar.dsp:
13400 Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
13401 * win32/vs6/gst_plugins_base.dsw:
13402 * win32/vs6/libgstdecodebin.dsp:
13403 * win32/vs6/libgstdecodebin2.dsp:
13404 * win32/vs6/libgstplaybin.dsp:
13405 * win32/vs6/libgstvolume.dsp:
13406 Add new dependencies to the link list.
13408 2008-01-13 17:24:49 +0000 Julien Moutte <julien@moutte.net>
13410 win32/common/: Update/Add generated files in the win32 build directory.
13411 Original commit message from CVS:
13412 2008-01-13 Julien Moutte <julien@fluendo.com>
13413 * win32/common/config.h:
13414 * win32/common/gstrtsp-enumtypes.c: (gst_rtsp_result_get_type),
13415 (gst_rtsp_event_get_type), (gst_rtsp_family_get_type),
13416 (gst_rtsp_state_get_type), (gst_rtsp_version_get_type),
13417 (gst_rtsp_method_get_type), (gst_rtsp_auth_method_get_type),
13418 (gst_rtsp_header_field_get_type),
13419 (gst_rtsp_status_code_get_type):
13420 * win32/common/interfaces-enumtypes.c:
13421 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
13422 (gst_mixer_message_type_get_type), (gst_mixer_flags_get_type),
13423 (gst_mixer_track_flags_get_type),
13424 (gst_tuner_channel_flags_get_type):
13425 * win32/common/multichannel-enumtypes.c:
13426 (gst_audio_channel_position_get_type):
13427 * win32/common/pbutils-enumtypes.c:
13428 (gst_install_plugins_return_get_type):
13429 * win32/common/pbutils-enumtypes.h: Update/Add generated files
13430 in the win32 build directory.
13432 2008-01-12 23:24:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13434 tests/check/Makefile.am: Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
13435 Original commit message from CVS:
13436 * tests/check/Makefile.am:
13437 Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
13438 * tests/check/elements/audiorate.c: (do_perfect_stream_test):
13439 * tests/check/elements/playbin.c:
13440 * tests/check/libs/mixer.c: (test_element_interface_supported),
13441 (gst_implements_interface_init):
13442 * tests/check/libs/rtp.c: (GST_START_TEST):
13443 Fix various assignment type mismatches.
13445 2008-01-12 23:08:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13447 Add test to see if hstrerror is available or if we need libresolv (Solaris) for it, then use it in libgstrtsp.
13448 Original commit message from CVS:
13450 * gst-libs/gst/rtsp/Makefile.am:
13451 Add test to see if hstrerror is available or if we need libresolv
13452 (Solaris) for it, then use it in libgstrtsp.
13454 2008-01-12 14:54:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13456 gst-libs/gst/tag/Makefile.am: Fix include path order
13457 Original commit message from CVS:
13458 * gst-libs/gst/tag/Makefile.am:
13459 Fix include path order
13461 2008-01-11 17:15:23 +0000 Tim-Philipp Müller <tim@centricular.net>
13463 * gst-libs/gst/pbutils/.gitignore:
13464 Ignore more and make buildbot happy
13465 Original commit message from CVS:
13466 Ignore more and make buildbot happy
13468 2008-01-11 16:18:10 +0000 Edward Hervey <bilboed@bilboed.com>
13470 gst-libs/gst/pbutils/install-plugins.*: Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping for bi...
13471 Original commit message from CVS:
13472 * gst-libs/gst/pbutils/install-plugins.c:
13473 (gst_install_plugins_context_copy),
13474 (gst_install_plugins_context_get_type):
13475 * gst-libs/gst/pbutils/install-plugins.h:
13476 Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
13479 2008-01-11 15:48:11 +0000 Michael Smith <msmith@xiph.org>
13481 ext/theora/theoradec.c: Adapt for post-alpha meaning of granulepos, when we have a newer version of libtheora.
13482 Original commit message from CVS:
13483 * ext/theora/theoradec.c: (gst_theora_dec_class_init),
13484 (_theora_granule_frame), (_theora_granule_start_time),
13485 (theora_dec_sink_convert), (theora_dec_decode_buffer):
13486 Adapt for post-alpha meaning of granulepos, when we
13487 have a newer version of libtheora.
13488 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
13489 (theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
13490 (theora_enc_is_discontinuous), (theora_enc_chain):
13492 * tests/check/Makefile.am:
13493 Link libtheora into theoraenc test so we can check which version of
13494 libtheora we're testing against.
13495 * tests/check/pipelines/theoraenc.c: (check_libtheora),
13496 (check_buffer_granulepos),
13497 (check_buffer_granulepos_from_starttime), (GST_START_TEST),
13499 Adapt tests to check the values that are now defined for theora; make
13500 the tests backwards-adapt the passed values if we're running against an
13504 2008-01-10 17:55:53 +0000 Tim-Philipp Müller <tim@centricular.net>
13506 gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
13507 Original commit message from CVS:
13508 * gst-libs/gst/audio/gstbaseaudiosink.c:
13509 (gst_base_audio_sink_class_init):
13510 * gst-libs/gst/audio/gstbaseaudiosrc.c:
13511 (gst_base_audio_src_class_init):
13512 Ref audio clock class from a thread-safe context to make sure
13513 we're not bit by GObjects lack of thread-safety here (#349410),
13514 however unlikely that may be in practice.
13516 2008-01-10 12:22:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13518 autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We...
13519 Original commit message from CVS:
13521 Add -Wno-portability to the automake parameters to stop warnings
13522 about GNU make extensions being used. We require GNU make in almost
13523 every Makefile anyway.
13525 Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
13526 at the same time is required for per target flags.
13528 2008-01-08 21:10:02 +0000 Tim-Philipp Müller <tim@centricular.net>
13530 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...
13531 Original commit message from CVS:
13532 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
13533 Post an error message if we can't pull as many bytes as we need
13534 for the tag. This makes sure the user gets to see a proper error
13535 message if a file with a partial ID3 tag is fed to decodebin, and
13536 not a 'no ID3 tag demuxer' error, which would be confusing
13539 2008-01-08 20:59:20 +0000 Tim-Philipp Müller <tim@centricular.net>
13541 gst-libs/gst/pbutils/descriptions.c: Add description strings for ID3, APE, and ICY tags.
13542 Original commit message from CVS:
13543 * gst-libs/gst/pbutils/descriptions.c: (formats):
13544 Add description strings for ID3, APE, and ICY tags.
13546 2008-01-08 20:48:00 +0000 Tim-Philipp Müller <tim@centricular.net>
13548 gst/playback/gstdecodebin.c: Make sure we error out correctly if we can't activate one of the elements we've added. ...
13549 Original commit message from CVS:
13550 * gst/playback/gstdecodebin.c: (try_to_link_1):
13551 Make sure we error out correctly if we can't activate one of
13552 the elements we've added. Fixes #508138.
13554 2008-01-07 13:59:43 +0000 Bastien Nocera <hadess@hadess.net>
13556 ext/alsa/gstalsamixer.c: Use snd_mixer_selem_set_{playback|capture}_volume_all() if the volume is the same for all ch...
13557 Original commit message from CVS:
13558 Patch by: Bastien Nocera <hadess at hadess net>
13559 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
13560 (check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
13561 Use snd_mixer_selem_set_{playback|capture}_volume_all() if
13562 the volume is the same for all channels. This works around
13563 some problem in alsa that leaves us with inconsistent state
13564 for some reason (#486840).
13566 2008-01-07 13:19:50 +0000 Jerone Young <jerone@gmail.com>
13568 ext/alsa/gstalsamixer.c: If there's no mixer track by the name of 'Master' or 'Front', check if there's one called 'P...
13569 Original commit message from CVS:
13570 Patch by: Jerone Young <jerone at gmail com>
13571 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
13572 If there's no mixer track by the name of 'Master' or 'Front',
13573 check if there's one called 'PCM' before trying the generic
13574 fallback logic (fixes #506928, where we pick 'Mic' as master
13575 track for the AD1984 card in a Thinkpad T61/X61 laptop).
13577 2008-01-07 11:40:04 +0000 Wim Taymans <wim.taymans@gmail.com>
13579 gst/playback/gstplay-enum.*: Add enums for configuration flags.
13580 Original commit message from CVS:
13581 * gst/playback/gstplay-enum.c:
13582 (register_gst_autoplug_select_result),
13583 (gst_autoplug_select_result_get_type), (register_gst_play_flags),
13584 (gst_play_flags_get_type):
13585 * gst/playback/gstplay-enum.h:
13586 Add enums for configuration flags.
13587 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
13588 (init_group), (gst_play_bin_init), (gst_play_bin_set_property),
13589 (gst_play_bin_get_property), (no_more_pads_cb),
13590 (autoplug_select_cb), (gst_play_bin_change_state):
13591 Merge mode with flags.
13592 Add more property getters/setters, defaults and docs.
13593 Add properties to get number of audio/video/text streams.
13594 Create sink object in _init so that we can always rely on it being
13596 * gst/playback/gstplaysink.c: (gst_play_sink_init),
13597 (gen_video_chain), (gen_audio_chain), (gen_vis_chain),
13598 (activate_vis), (gst_play_sink_reconfigure),
13599 (gst_play_sink_set_flags), (gst_play_sink_get_flags),
13600 (gst_play_sink_change_state):
13601 * gst/playback/gstplaysink.h:
13602 Use flags to configure the sink pipelines.
13603 Add tee before audio pipeline so that we can use it for visualisations.
13604 Start working on integrating visualisations.
13605 Remove mode, we can do everything with the flags now.
13606 Add method to configue the sink pipeline.
13608 2008-01-06 16:36:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13610 Update to GMemoryInputStream API changes in GLib SVN and require gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
13611 Original commit message from CVS:
13613 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
13614 * tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
13615 Update to GMemoryInputStream API changes in GLib SVN and require
13616 gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
13617 We can also report the duration for every GSeekable, not only
13618 GFileInputStream and GMemoryInputStream.
13620 2008-01-06 14:39:19 +0000 Tim-Philipp Müller <tim@centricular.net>
13622 tests/check/pipelines/theoraenc.c: Turn these functions into macros so we can see right away where the failure occured.
13623 Original commit message from CVS:
13624 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
13625 (check_buffer_timestamp), (check_buffer_duration):
13626 Turn these functions into macros so we can see right away
13627 where the failure occured.
13629 2008-01-05 22:25:05 +0000 Julien Moutte <julien@moutte.net>
13631 sys/xvimage/xvimagesink.c: Add debugging information to understand how X calculates the stride for XvImages.
13632 Original commit message from CVS:
13633 2008-01-05 Julien Moutte <julien@fluendo.com>
13634 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
13635 debugging information to understand how X calculates the stride
13638 2008-01-03 20:33:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13640 gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
13641 Original commit message from CVS:
13642 * gst/volume/Makefile.am:
13643 * gst/volume/gstvolume.c: (volume_choose_func),
13644 (gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
13646 * gst/volume/gstvolume.h:
13647 Use GstAudioFilter as base class for the volume element instead of
13648 plain GstBaseTransform.
13650 2008-01-03 07:17:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13652 gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
13653 Original commit message from CVS:
13654 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
13655 Don't set element details for the abstract GstAudioFilter class.
13657 2008-01-02 12:09:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13659 gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
13660 Original commit message from CVS:
13661 * gst-libs/gst/audio/gstaudiofilter.c:
13662 (gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
13663 Implement get_unit_size() vmethod of GstBaseTransform.
13665 2008-01-01 12:53:48 +0000 Edward Hervey <bilboed@bilboed.com>
13667 gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
13668 Original commit message from CVS:
13669 * gst-libs/gst/pbutils/Makefile.am:
13670 * gst-libs/gst/pbutils/pbutils.h:
13671 Use glib-enum generator to have a proper enum GType for
13672 GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
13674 2008-01-01 01:21:47 +0000 David Schleef <ds@schleef.org>
13676 tests/check/: Reenable theoraenc test, which fails on the buildbot but not locally.
13677 Original commit message from CVS:
13678 * tests/check/Makefile.am:
13679 * tests/check/pipelines/theoraenc.c:
13680 Reenable theoraenc test, which fails on the buildbot but
13683 2007-12-31 21:31:01 +0000 David Schleef <ds@schleef.org>
13685 docs/: Add *-undeclared.txt to fix buildbot.
13686 Original commit message from CVS:
13687 * docs/libs/.cvsignore:
13688 * docs/plugins/.cvsignore:
13689 Add *-undeclared.txt to fix buildbot.
13691 2007-12-31 20:45:28 +0000 David Schleef <ds@schleef.org>
13693 tests/check/Makefile.am: Second attempt at disabling theoraenc test long enough to get buildbot to compile -base.
13694 Original commit message from CVS:
13695 * tests/check/Makefile.am:
13696 Second attempt at disabling theoraenc test long enough to
13697 get buildbot to compile -base.
13699 2007-12-31 20:21:20 +0000 David Schleef <ds@schleef.org>
13701 tests/check/pipelines/theoraenc.c: Disable theoraenc test long enough to get the buildbot to compile a recent -base.
13702 Original commit message from CVS:
13703 * tests/check/pipelines/theoraenc.c:
13704 Disable theoraenc test long enough to get the buildbot to
13705 compile a recent -base.
13707 2007-12-31 13:17:29 +0000 Wim Taymans <wim.taymans@gmail.com>
13709 tests/examples/seek/seek.c: Make sure we reset the slider value to 0.0 without racing against a possible g_idle that ...
13710 Original commit message from CVS:
13711 * tests/examples/seek/seek.c: (stop_cb):
13712 Make sure we reset the slider value to 0.0 without racing against a
13713 possible g_idle that sets it to something else.
13715 2007-12-31 00:32:53 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13717 sys/ximage/ximagesink.c: fix typo
13718 Original commit message from CVS:
13719 * sys/ximage/ximagesink.c:
13722 2007-12-30 19:21:16 +0000 Wim Taymans <wim.taymans@gmail.com>
13724 gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
13725 Original commit message from CVS:
13726 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
13727 * gst-libs/gst/rtsp/gstrtspdefs.h:
13728 Add Location header so that we can start implementing redirects.
13731 2007-12-29 20:55:39 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13733 gst/subparse/gstssaparse.c: combine if's
13734 Original commit message from CVS:
13735 * gst/subparse/gstssaparse.c:
13738 2007-12-29 19:23:59 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13740 gst/subparse/gstssaparse.c: remove duplicate log message
13741 Original commit message from CVS:
13742 * gst/subparse/gstssaparse.c:
13743 remove duplicate log message
13745 2007-12-29 17:29:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13747 Update to latest API changes in GLib/GIO and require at least gio-2.0 2.15.0 for this.
13748 Original commit message from CVS:
13750 * ext/gio/gstgio.c:
13751 * ext/gio/gstgio.h:
13752 * ext/gio/gstgiobasesink.h:
13753 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
13754 * ext/gio/gstgiobasesrc.h:
13755 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
13756 * ext/gio/gstgiosink.h:
13757 * ext/gio/gstgiosrc.h:
13758 * ext/gio/gstgiostreamsink.h:
13759 * ext/gio/gstgiostreamsrc.h:
13760 * tests/check/pipelines/gio.c:
13761 Update to latest API changes in GLib/GIO and require at least
13762 gio-2.0 2.15.0 for this.
13763 * ext/gio/Makefile.am:
13764 Add GST_PLUGIN_LDFLAGS to LDFLAGS.
13766 2007-12-29 16:23:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13768 ext/libvisual/visual.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached()...
13769 Original commit message from CVS:
13770 * ext/libvisual/visual.c: (gst_visual_chain):
13771 Fix 'xyz may be used uninitialized' compiler warnings caused
13772 by broken g_assert_not_reached() macro in GLib-2.15.x and don't
13773 abort() in any case but properly report the error.
13775 2007-12-28 09:00:27 +0000 Wim Taymans <wim.taymans@gmail.com>
13777 gst/playback/gstplaybin2.c: Code cleanups.
13778 Original commit message from CVS:
13779 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
13780 (gst_play_bin_finalize), (gst_play_bin_set_uri),
13781 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
13782 (gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
13783 (autoplug_select_cb), (activate_group), (deactivate_group),
13784 (setup_next_source), (save_current_group),
13785 (gst_play_bin_change_state):
13787 Remove next-uri, we can use the uri property just fine.
13789 Unref uridecodebin when switching.
13790 Fix going to READY.
13791 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
13792 (gst_play_sink_init), (gst_play_sink_dispose),
13793 (gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
13794 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
13795 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
13796 (gst_play_sink_set_property), (gst_play_sink_get_property),
13797 (gen_video_chain), (gen_text_element), (gen_audio_chain),
13798 (gen_vis_element), (gst_play_sink_get_mode),
13799 (gst_play_sink_set_mode), (gst_play_sink_set_flags),
13800 (gst_play_sink_get_flags), (gst_play_sink_request_pad),
13801 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
13802 (gst_play_sink_change_state):
13803 * gst/playback/gstplaysink.h:
13804 Add some locking to make things threadsafe.
13805 * gst/playback/test7.c: (about_to_finish_cb):
13808 2007-12-22 12:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
13810 gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
13811 Original commit message from CVS:
13812 * gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
13813 (gst_video_scale_get_property), (gst_video_scale_transform_caps),
13814 (gst_video_scale_transform):
13815 Don't claim to be able to handle/transform caps that can't really
13816 be handled by the currently selected scaling method (here: RGB or
13817 packed YUV with 4-tap method). Also add locking to method property.
13818 * tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
13819 (test_basetransform_based):
13820 Some test pipelines for the above (not entirely valgrind clean yet
13823 2007-12-22 05:19:00 +0000 David Schleef <ds@schleef.org>
13825 gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
13826 Original commit message from CVS:
13827 * gst-libs/gst/video/video.c:
13828 * gst-libs/gst/video/video.h:
13829 Add additional RGBA and RGB-24 video formats.
13831 2007-12-21 22:46:56 +0000 Tim-Philipp Müller <tim@centricular.net>
13833 tests/check/: Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be deprecated in the future (see #498924).
13834 Original commit message from CVS:
13835 * tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
13836 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
13837 (test_suburi_error_wrongproto), (test_missing_primary_decoder):
13838 * tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
13839 (cddabasesrc_suite):
13840 Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
13841 deprecated in the future (see #498924).
13843 2007-12-21 22:26:47 +0000 Tim-Philipp Müller <tim@centricular.net>
13845 gst/playback/gststreamselector.c: Don't leak event.
13846 Original commit message from CVS:
13847 * gst/playback/gststreamselector.c: (gst_selector_pad_event):
13850 2007-12-20 19:43:25 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13852 gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
13853 Original commit message from CVS:
13854 * gst-libs/gst/riff/riff-read.c:
13855 Use GST_ROUND_UP_2 macro
13857 2007-12-20 17:13:37 +0000 Tim-Philipp Müller <tim@centricular.net>
13859 gst/playback/.cvsignore: Ignore more.
13860 Original commit message from CVS:
13861 * gst/playback/.cvsignore:
13864 2007-12-20 10:41:29 +0000 Tim-Philipp Müller <tim@centricular.net>
13866 Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
13867 Original commit message from CVS:
13868 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
13869 * gst/playback/gstplaybasebin.c: (set_subtitles_visible),
13870 (set_active_source):
13871 * gst/playback/gstplaybasebin.h:
13872 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
13873 (setup_sinks), (playbin_set_subtitles_visible):
13874 Make switching off of subtitles work. To avoid all kind of
13875 problems with unlinking of the subtitle input, we just keep
13876 the subtitle inputs linked as they are and tell textoverlay
13877 not to render them. Fixes #373011.
13878 Other subtitle switching issues (esp. when there are both
13879 external and in-stream subtitles) remain. They'll be solved
13882 2007-12-18 16:21:35 +0000 Wim Taymans <wim.taymans@gmail.com>
13884 gst/playback/gststreamselector.c: Init the pad segment too.
13885 Original commit message from CVS:
13886 * gst/playback/gststreamselector.c: (gst_selector_pad_init):
13887 Init the pad segment too.
13889 2007-12-18 15:56:51 +0000 Wim Taymans <wim.taymans@gmail.com>
13891 gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
13892 Original commit message from CVS:
13893 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
13894 (gst_audioringbuffer_open_device),
13895 (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
13896 (gst_audioringbuffer_release), (gst_audioringbuffer_start),
13897 (gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
13898 (gst_audio_sink_create_ringbuffer):
13899 Improve debug output.
13900 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
13901 (gst_ring_buffer_pause), (gst_ring_buffer_delay):
13902 Prevent some functions from doing things and failing when the
13903 ringbuffer is not yet acquired.
13905 2007-12-18 15:32:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13907 gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
13908 Original commit message from CVS:
13909 * gst-libs/gst/interfaces/interfaces.h:
13910 Also remove interfaces.h from CVS as it is not needed anymore.
13912 2007-12-18 15:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13914 gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
13915 Original commit message from CVS:
13916 * gst-libs/gst/interfaces/Makefile.am:
13917 interfaces.h is not used anymore so remove it from the build
13920 2007-12-18 01:01:23 +0000 David Schleef <ds@schleef.org>
13922 gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
13923 Original commit message from CVS:
13924 * gst/videotestsrc/gstvideotestsrc.c:
13925 * gst/videotestsrc/gstvideotestsrc.h:
13926 Add a "blink" pattern. Turn on the pain. Apologies. It's useful
13927 for testing vertical refresh synchronization.
13929 2007-12-18 00:13:26 +0000 David Schleef <ds@schleef.org>
13931 Add new GstVideFormat enum and write a bunch of helper functions based around it.
13932 Original commit message from CVS:
13933 * docs/libs/gst-plugins-base-libs-sections.txt:
13934 * gst-libs/gst/video/video.c:
13935 * gst-libs/gst/video/video.h:
13936 Add new GstVideFormat enum and write a bunch of helper functions
13939 2007-12-17 23:41:14 +0000 Tim-Philipp Müller <tim@centricular.net>
13941 Makefile.am: Use new common/win32.mak.
13942 Original commit message from CVS:
13944 Use new common/win32.mak.
13946 2007-12-17 16:44:51 +0000 Wim Taymans <wim.taymans@gmail.com>
13948 gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
13949 Original commit message from CVS:
13950 * gst-libs/gst/audio/gstbaseaudiosrc.c:
13951 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
13953 When going from PLAYING to PAUSED, pause the ringbuffer before calling
13954 the parent state change function, just like the audiosink, because the
13955 parent waits for the element to finish its processing before completing
13956 the state change. This makes going to PAUSED a lot snappier.
13957 When going from READY to PAUSED, don't allow the ringbuffer to start
13960 2007-12-17 00:01:00 +0000 Edward Hervey <bilboed@bilboed.com>
13962 gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
13963 Original commit message from CVS:
13964 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
13965 Yet another fix for broken software that produce files with an empty
13966 blockalign field. Instead of completely failing, make a second attempt
13967 at guessing the width/depth by looking at strf->size.
13969 2007-12-16 23:52:58 +0000 Tim-Philipp Müller <tim@centricular.net>
13971 gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
13972 Original commit message from CVS:
13973 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
13974 (gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
13975 * gst-libs/gst/pbutils/install-plugins.c:
13976 (gst_install_plugins_spawn_child), (gst_install_plugins_supported):
13977 * gst-libs/gst/pbutils/missing-plugins.c:
13978 (gst_missing_plugin_message_get_installer_detail),
13979 (gst_missing_encoder_installer_detail_new):
13980 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
13981 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
13982 Turn a few g_assert_not_reached() into g_return_val_if_reached() to
13983 avoid compiler warnings (#503930).
13985 2007-12-16 23:46:16 +0000 Edward Hervey <bilboed@bilboed.com>
13987 gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
13988 Original commit message from CVS:
13989 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
13990 Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
13991 for jpeg video streams.
13992 Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
13993 for the above modification.
13995 2007-12-15 17:27:48 +0000 Tim-Philipp Müller <tim@centricular.net>
13997 gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
13998 Original commit message from CVS:
13999 * gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
14000 (gst_x_overlay_handle_events):
14001 More guards (we don't want klass to end up being NULL).
14003 2007-12-15 03:40:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14005 Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
14006 Original commit message from CVS:
14008 * gst/volume/gstvolume.c: (gst_volume_init):
14009 Use new gst_base_transform_set_gap_aware() function as volume
14010 correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
14013 2007-12-14 19:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
14015 tests/examples/seek/seek.c: Don't go to READY on EOS as this avoids testing of seeking and restarting after EOS, use ...
14016 Original commit message from CVS:
14017 * tests/examples/seek/seek.c: (msg_segment_done), (main):
14018 Don't go to READY on EOS as this avoids testing of seeking and
14019 restarting after EOS, use the stop button when you want to READY.
14020 Don't try to do a flushing seek in segment-done, it does not make
14021 sense to use this for gapless playback and is not needed.
14023 2007-12-14 18:46:12 +0000 Wim Taymans <wim.taymans@gmail.com>
14025 gst/playback/gstqueue2.c: Use separate timers for input and output rates.
14026 Original commit message from CVS:
14027 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
14028 (reset_rate_timer), (update_in_rates), (update_out_rates),
14029 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
14030 (gst_queue_chain), (gst_queue_loop):
14031 Use separate timers for input and output rates.
14032 Pause measuring the output rate when we block for more data.
14035 2007-12-14 16:23:06 +0000 Christian Schaller <uraeus@gnome.org>
14037 * gst/speexresample/Makefile.am:
14038 update spec file and add two missing files for disting
14039 Original commit message from CVS:
14040 update spec file and add two missing files for disting
14042 2007-12-14 09:24:55 +0000 Wim Taymans <wim.taymans@gmail.com>
14044 gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
14045 Original commit message from CVS:
14046 * gst/playback/gstqueue2.c: (gst_queue_chain):
14047 Pause the timer to measure the input rate when we block because the
14048 queue is filled. See #503262.
14050 2007-12-13 15:54:00 +0000 Peter Kjellerstedt <pkj@axis.com>
14052 gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
14053 Original commit message from CVS:
14054 Patch by: Peter Kjellerstedt <pkj at axis com>
14055 * gst-libs/gst/rtsp/gstrtspconnection.c:
14056 (gst_rtsp_connection_free):
14057 Close control sockets. Fixes #503440.
14059 2007-12-13 12:31:38 +0000 Wim Taymans <wim.taymans@gmail.com>
14061 gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
14062 Original commit message from CVS:
14063 * gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
14064 Expose the right pad in the right place with the right element.
14066 2007-12-13 11:40:10 +0000 Tim-Philipp Müller <tim@centricular.net>
14068 gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
14069 Original commit message from CVS:
14070 * gst-libs/gst/pbutils/descriptions.c: (formats):
14071 Add description for 'private' dts caps (who come up with that name?).
14073 2007-12-13 10:10:35 +0000 Tim-Philipp Müller <tim@centricular.net>
14075 Makefile.am: Add check-exports target and run it with 'make check'.
14076 Original commit message from CVS:
14078 Add check-exports target and run it with 'make check'.
14080 Be stricter about what we export in our libraries: change regexp so that
14081 we only export _gst_foo(), but not __gst_foo().
14082 * gst-libs/gst/cdda/base64.h: (rfc822_binary):
14083 * gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
14084 Change internal functions to __gst_foo so they dont' get exported.
14085 * win32/common/libgstaudio.def:
14086 Add missing symbols.
14088 2007-12-11 21:18:57 +0000 David Schleef <ds@schleef.org>
14091 ChangeLog: remove conflict markers
14092 Original commit message from CVS:
14093 ChangeLog: remove conflict markers
14095 2007-12-11 17:14:13 +0000 Tim-Philipp Müller <tim@centricular.net>
14097 ext/gnomevfs/: Use gst_tag_freeform_string_to_utf8() here, which also takes into account any character sets specified...
14098 Original commit message from CVS:
14099 * ext/gnomevfs/Makefile.am:
14100 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
14101 Use gst_tag_freeform_string_to_utf8() here, which also takes
14102 into account any character sets specified by the user via
14103 environment variables.
14105 2007-12-10 15:21:41 +0000 Wim Taymans <wim.taymans@gmail.com>
14107 gst/audioconvert/Makefile.am: Also link to libm.
14108 Original commit message from CVS:
14109 * gst/audioconvert/Makefile.am:
14112 2007-12-10 15:13:55 +0000 Wim Taymans <wim.taymans@gmail.com>
14114 gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
14115 Original commit message from CVS:
14116 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
14117 No need for floating point operations here. avoids having to link
14118 against the math library too.
14120 2007-12-10 11:16:25 +0000 Tim-Philipp Müller <tim@centricular.net>
14122 Add one or two missing formats. Generate ADPCM description dynamically depending on layout/format.
14123 Original commit message from CVS:
14124 * gst-libs/gst/pbutils/descriptions.c: (formats),
14125 (format_info_get_desc):
14126 * tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
14128 Add one or two missing formats. Generate ADPCM description
14129 dynamically depending on layout/format.
14131 2007-12-09 04:28:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14133 configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
14134 Original commit message from CVS:
14136 Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
14138 2007-12-08 18:38:39 +0000 Robin Stocker <robin.stocker@gmx.ch>
14140 gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
14141 Original commit message from CVS:
14142 Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
14143 * gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
14144 Some .srt files start with chunk number 0 and not chunk number 1,
14145 recognise and accept those as well (fixes #502497).
14146 * tests/check/elements/subparse.c: (srt_input), (srt_input0),
14148 Add unit test for the above.
14150 2007-12-06 12:08:21 +0000 Wim Taymans <wim.taymans@gmail.com>
14152 gst/playback/gstplay-enum.*: Add missing files.
14153 Original commit message from CVS:
14154 * gst/playback/gstplay-enum.c:
14155 (register_gst_autoplug_select_result),
14156 (gst_autoplug_select_result_get_type):
14157 * gst/playback/gstplay-enum.h:
14160 2007-12-05 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
14162 gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
14163 Original commit message from CVS:
14164 * gst/playback/Makefile.am:
14165 Group decodebin2 and uridecodebin into the same plugin so that they
14166 can share the GEnumType.
14167 * gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
14168 (_gst_select_accumulator), (gst_decode_bin_class_init),
14169 (gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
14170 (gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
14171 (analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
14172 Add signal to sort factories instead of the more awkward autoplug-select
14174 Modify autoplug_select so that we can try, skip or expose the
14175 autopluggin of an element on a pad.
14176 * gst/playback/gstfactorylists.c: (compare_ranks),
14177 (decoders_filter), (sinks_filter), (gst_factory_list_is_type),
14178 (element_filter), (gst_factory_list_get_elements),
14179 (gst_factory_list_debug), (gst_factory_list_filter):
14180 * gst/playback/gstfactorylists.h:
14181 Simplify the API, allow getting elements based on mask.
14182 * gst/playback/gstplay-marshal.list:
14183 Add some more marshallers.
14184 * gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
14185 (gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
14186 (autoplug_select_cb), (activate_group):
14187 Add support for managing non-raw sinks by providing a custom element and
14188 sink list to decodebin2.
14189 Try to plug non-raw sinks when decodebin2 using autoplug-select of
14191 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
14192 (gst_play_sink_set_mode), (gst_play_sink_request_pad):
14193 * gst/playback/gstplaysink.h:
14194 Add support for raw and non-raw sinks.
14195 Add support to force sinks selected by playbin2.
14196 Don't plug raw converters for non-raw sinks.
14197 * gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
14198 (_gst_select_accumulator), (gst_uri_decode_bin_class_init),
14199 (proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
14201 Use right accumulators.
14204 2007-12-03 13:47:00 +0000 Wim Taymans <wim.taymans@gmail.com>
14206 gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
14207 Original commit message from CVS:
14208 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
14209 Use runnning time as the base time instead of the timestamp.
14210 Spotted by Saur on IRC.
14212 2007-12-03 11:32:30 +0000 Edward Hervey <bilboed@bilboed.com>
14214 gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
14215 Original commit message from CVS:
14216 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
14217 Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
14219 2007-12-03 10:58:14 +0000 Wim Taymans <wim.taymans@gmail.com>
14221 ext/ogg/gstoggdemux.c: If we find a new serial number but it does not contain a BOS page, make sure we initialize the...
14222 Original commit message from CVS:
14223 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
14224 (gst_ogg_demux_read_chain):
14225 If we find a new serial number but it does not contain a BOS page, make
14226 sure we initialize the chain to NULL because else we will try to scan it
14227 and crash. Fixes #500763
14229 2007-11-30 17:47:15 +0000 Wim Taymans <wim.taymans@gmail.com>
14231 gst/playback/: Refactor some common code to filter factories and check caps compat.
14232 Original commit message from CVS:
14233 * gst/playback/Makefile.am:
14234 * gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
14235 (get_feature_array), (decoders_filter), (sinks_filter),
14236 (gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
14237 (gst_factory_list_filter):
14238 * gst/playback/gstfactorylists.h:
14239 Refactor some common code to filter factories and check caps compat.
14240 * gst/playback/gstdecodebin.c:
14241 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
14242 (gst_decode_bin_init), (gst_decode_bin_dispose),
14243 (gst_decode_bin_autoplug_continue),
14244 (gst_decode_bin_autoplug_factories),
14245 (gst_decode_bin_autoplug_select), (analyze_new_pad),
14246 (find_compatibles):
14247 * gst/playback/gstplaybin.c:
14248 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
14249 (gst_play_bin_init), (gst_play_bin_finalize),
14250 (autoplug_factories_cb), (activate_group):
14251 * gst/playback/gstqueue2.c:
14252 * gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
14253 (proxy_autoplug_continue_signal),
14254 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
14255 (proxy_drained_signal):
14256 Add some more debug info and use factor filtering code.
14258 2007-11-26 13:19:46 +0000 Julien Moutte <julien@moutte.net>
14260 configure.ac: Add QuickTime Wrapper plug-in.
14261 Original commit message from CVS:
14262 2007-11-26 Julien Moutte <julien@fluendo.com>
14263 * configure.ac: Add QuickTime Wrapper plug-in.
14264 * gst/speexresample/gstspeexresample.c:
14265 (gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
14266 build on Mac OS X Leopard. Incorrect printf format arguments.
14268 * sys/qtwrapper/Makefile.am:
14269 * sys/qtwrapper/audiodecoders.c:
14270 (qtwrapper_audio_decoder_base_init),
14271 (qtwrapper_audio_decoder_class_init),
14272 (qtwrapper_audio_decoder_init),
14273 (clear_AudioStreamBasicDescription), (fill_indesc_mp3),
14274 (fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
14275 (make_samr_magic_cookie), (open_decoder),
14276 (qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
14277 (qtwrapper_audio_decoder_chain),
14278 (qtwrapper_audio_decoder_sink_event),
14279 (qtwrapper_audio_decoders_register):
14280 * sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
14282 * sys/qtwrapper/codecmapping.h:
14283 * sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
14284 (image_description_for_mp4v), (image_description_from_stsd_buffer),
14285 (image_description_from_codec_data):
14286 * sys/qtwrapper/imagedescription.h:
14287 * sys/qtwrapper/qtutils.c: (get_name_info_from_component),
14288 (get_output_info_from_component), (dump_avcc_atom),
14289 (dump_image_description), (dump_codec_decompress_params),
14290 (addSInt32ToDictionary), (dump_cvpixel_buffer),
14291 (DestroyAudioBufferList), (AllocateAudioBufferList):
14292 * sys/qtwrapper/qtutils.h:
14293 * sys/qtwrapper/qtwrapper.c: (plugin_init):
14294 * sys/qtwrapper/qtwrapper.h:
14295 * sys/qtwrapper/videodecoders.c:
14296 (qtwrapper_video_decoder_base_init),
14297 (qtwrapper_video_decoder_class_init),
14298 (qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
14299 (fill_image_description), (new_image_description), (close_decoder),
14300 (open_decoder), (qtwrapper_video_decoder_sink_setcaps),
14301 (decompressCb), (qtwrapper_video_decoder_chain),
14302 (qtwrapper_video_decoder_sink_event),
14303 (qtwrapper_video_decoders_register): Initial import of QuickTime
14304 wrapper jointly developped by Songbird authors (Pioneers of the
14305 Inevitable) and Fluendo.
14307 2007-11-26 12:25:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14309 gst/: Add GAP-flag support.
14310 Original commit message from CVS:
14311 * gst/audiotestsrc/gstaudiotestsrc.c:
14312 * gst/volume/gstvolume.c:
14313 * gst/volume/gstvolume.h:
14314 Add GAP-flag support.
14316 2007-11-26 08:43:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14318 gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
14319 Original commit message from CVS:
14320 * gst/speexresample/README:
14321 * gst/speexresample/arch.h:
14322 * gst/speexresample/resample.c: (resampler_basic_direct_single),
14323 (resampler_basic_direct_double),
14324 (resampler_basic_interpolate_single),
14325 (resampler_basic_interpolate_double),
14326 (speex_resampler_process_native), (speex_resampler_process_float),
14327 (speex_resampler_process_int),
14328 (speex_resampler_process_interleaved_float),
14329 (speex_resampler_process_interleaved_int),
14330 (speex_resampler_get_input_latency),
14331 (speex_resampler_get_output_latency):
14332 * gst/speexresample/speex_resampler.h:
14333 Update speex resampler to latest SVN. We're now down to only the
14334 changes noted in README again.
14335 * gst/speexresample/speex_resampler_wrapper.h:
14336 * gst/speexresample/gstspeexresample.c:
14337 (gst_speex_resample_push_drain), (gst_speex_resample_query):
14338 Adjust to API changes.
14340 2007-11-24 15:02:01 +0000 Julien Moutte <julien@moutte.net>
14342 tests/examples/seek/seek.c: Increase the range of the rate selector as I would like to test QOS behavior at higher fo...
14343 Original commit message from CVS:
14344 2007-11-24 Julien MOUTTE <julien@moutte.net>
14345 * tests/examples/seek/seek.c: (main): Increase the range of the
14346 rate selector as I would like to test QOS behavior at higher
14347 forward and reverse playback speed like say 64x.
14349 2007-11-23 10:21:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14351 gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
14352 Original commit message from CVS:
14353 * gst/speexresample/gstspeexresample.c:
14354 (gst_speex_resample_update_state):
14355 Only post the latency message if we have a resampler state already.
14357 2007-11-23 10:21:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14359 gst/audioresample/gstaudioresample.c: Implement latency query.
14360 Original commit message from CVS:
14361 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
14362 (audioresample_query), (audioresample_query_type),
14363 (gst_audioresample_set_property):
14364 Implement latency query.
14366 2007-11-23 10:01:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14368 gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
14369 Original commit message from CVS:
14370 * gst/speexresample/gstspeexresample.c:
14371 (gst_speex_resample_update_state):
14372 Also post GST_MESSAGE_LATENCY if the latency changes.
14374 2007-11-23 08:48:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14376 gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
14377 Original commit message from CVS:
14378 * gst/speexresample/resample.c: (speex_resampler_get_latency),
14379 (speex_resampler_drain_float), (speex_resampler_drain_int),
14380 (speex_resampler_drain_interleaved_float),
14381 (speex_resampler_drain_interleaved_int):
14382 * gst/speexresample/speex_resampler.h:
14383 * gst/speexresample/speex_resampler_wrapper.h:
14384 Add functions to push the remaining samples and to get the latency
14385 of the resampler. These will get added to Speex SVN in this or a
14386 slightly changed form at some point too and should get merged then
14388 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
14389 (gst_speex_resample_init_state),
14390 (gst_speex_resample_transform_size),
14391 (gst_speex_resample_push_drain), (gst_speex_resample_event),
14392 (gst_speex_fix_output_buffer), (gst_speex_resample_process),
14393 (gst_speex_resample_query), (gst_speex_resample_query_type):
14394 Drop the prepending zeroes and output the remaining samples on EOS.
14395 Also properly implement the latency query for this. speexresample
14396 should be completely ready for production use now.
14398 2007-11-21 18:02:21 +0000 Wim Taymans <wim.taymans@gmail.com>
14400 gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
14401 Original commit message from CVS:
14402 * gst-libs/gst/audio/gstbaseaudiosink.c:
14403 (gst_base_audio_sink_drain):
14404 Our EOS time contains the base_time, _wait_eos() expects a running_time
14405 so we have to subtract the base_time again before calling the function.
14406 This fixes an EOS regression where the base_time was added twice and EOS
14407 took longer and longer in certain situations.
14410 2007-11-21 13:04:17 +0000 Wim Taymans <wim.taymans@gmail.com>
14412 Expose methods for some object properties so that subclasses can more easily configure them.
14413 Original commit message from CVS:
14414 * docs/libs/gst-plugins-base-libs-sections.txt:
14415 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
14416 (gst_base_audio_sink_set_provide_clock),
14417 (gst_base_audio_sink_get_provide_clock),
14418 (gst_base_audio_sink_set_slave_method),
14419 (gst_base_audio_sink_get_slave_method),
14420 (gst_base_audio_sink_set_property),
14421 (gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
14422 (gst_base_audio_sink_none_slaving),
14423 (gst_base_audio_sink_handle_slaving):
14424 * gst-libs/gst/audio/gstbaseaudiosink.h:
14425 Expose methods for some object properties so that subclasses can more
14426 easily configure them.
14427 Added slave method none, that completely disables slaving to the
14429 API: gst_base_audio_sink_set_provide_clock()
14430 API: gst_base_audio_sink_get_provide_clock()
14431 API: gst_base_audio_sink_set_slave_method()
14432 API: gst_base_audio_sink_get_slave_method()
14433 * gst-libs/gst/audio/gstbaseaudiosrc.c:
14434 (gst_base_audio_src_set_provide_clock),
14435 (gst_base_audio_src_get_provide_clock),
14436 (gst_base_audio_src_set_property),
14437 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
14438 * gst-libs/gst/audio/gstbaseaudiosrc.h:
14439 Expose methods for some object properties so that subclasses can more
14440 easily configure them.
14441 API: gst_base_audio_src_set_provide_clock()
14442 API: gst_base_audio_src_get_provide_clock()
14444 2007-11-21 10:18:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14446 gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
14447 Original commit message from CVS:
14448 * gst/speexresample/README:
14449 Add README explaining where the resampling code was taken from
14450 and which changes were done.
14451 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
14453 Use g_malloc() and friends instead of malloc() to achieve higher
14454 portability and define the functions inline.
14455 * gst/speexresample/speex_resampler.h:
14456 Add back some useless preprocessor stuff to keep the diff between
14457 our version and the one from the Speex SVN repository lower.
14459 2007-11-20 20:23:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14461 gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
14462 Original commit message from CVS:
14463 * gst/speexresample/gstspeexresample.c:
14464 (gst_speex_fix_output_buffer), (gst_speex_resample_transform):
14465 Some small cleanup and addition of a TODO item.
14467 2007-11-20 12:56:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14469 gst/speexresample/Makefile.am: Add missing file.
14470 Original commit message from CVS:
14471 * gst/speexresample/Makefile.am:
14474 2007-11-20 07:53:56 +0000 Joe Peterson <lavajoe@gentoo.org>
14476 gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
14477 Original commit message from CVS:
14478 Patch by: Joe Peterson <lavajoe at gentoo dot org>
14479 * gst-libs/gst/sdp/gstsdpmessage.c:
14480 Fix compilation on FreeBSD (Gentoo). Fixes #498228.
14482 2007-11-20 07:47:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14484 Add speexresample to the docs and while at that do a make update.
14485 Original commit message from CVS:
14486 * docs/plugins/Makefile.am:
14487 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
14488 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
14489 * docs/plugins/gst-plugins-bad-plugins.args:
14490 * docs/plugins/gst-plugins-bad-plugins.signals:
14491 * docs/plugins/inspect/plugin-bz2.xml:
14492 * docs/plugins/inspect/plugin-cdxaparse.xml:
14493 * docs/plugins/inspect/plugin-dtsdec.xml:
14494 * docs/plugins/inspect/plugin-equalizer.xml:
14495 * docs/plugins/inspect/plugin-faac.xml:
14496 * docs/plugins/inspect/plugin-faad.xml:
14497 * docs/plugins/inspect/plugin-filter.xml:
14498 * docs/plugins/inspect/plugin-freeze.xml:
14499 * docs/plugins/inspect/plugin-gio.xml:
14500 * docs/plugins/inspect/plugin-gsm.xml:
14501 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
14502 * docs/plugins/inspect/plugin-h264parse.xml:
14503 * docs/plugins/inspect/plugin-modplug.xml:
14504 * docs/plugins/inspect/plugin-mpeg2enc.xml:
14505 * docs/plugins/inspect/plugin-musepack.xml:
14506 * docs/plugins/inspect/plugin-musicbrainz.xml:
14507 * docs/plugins/inspect/plugin-nsfdec.xml:
14508 * docs/plugins/inspect/plugin-replaygain.xml:
14509 * docs/plugins/inspect/plugin-soundtouch.xml:
14510 * docs/plugins/inspect/plugin-spcdec.xml:
14511 * docs/plugins/inspect/plugin-spectrum.xml:
14512 * docs/plugins/inspect/plugin-speed.xml:
14513 * docs/plugins/inspect/plugin-tta.xml:
14514 * docs/plugins/inspect/plugin-videosignal.xml:
14515 * docs/plugins/inspect/plugin-xingheader.xml:
14516 * docs/plugins/inspect/plugin-xvid.xml:
14517 * gst/speexresample/gstspeexresample.h:
14518 Add speexresample to the docs and while at that do a make update.
14520 2007-11-20 07:30:30 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14522 gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
14523 Original commit message from CVS:
14524 * gst/speexresample/gstspeexresample.c:
14525 (gst_speex_fix_output_buffer), (gst_speex_resample_process):
14526 If the resampler gives less output samples than expected
14527 adjust the output buffer and print a warning.
14529 2007-11-20 07:02:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14531 Add resample element based on the Speex resampling algorithm.
14532 Original commit message from CVS:
14534 * gst/speexresample/arch.h:
14535 * gst/speexresample/fixed_generic.h:
14536 * gst/speexresample/gstspeexresample.c:
14537 (gst_speex_resample_base_init), (gst_speex_resample_class_init),
14538 (gst_speex_resample_init), (gst_speex_resample_start),
14539 (gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
14540 (gst_speex_resample_transform_caps),
14541 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
14542 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
14543 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
14544 (gst_speex_resample_event), (gst_speex_resample_check_discont),
14545 (gst_speex_resample_process), (gst_speex_resample_transform),
14546 (gst_speex_resample_set_property),
14547 (gst_speex_resample_get_property), (plugin_init):
14548 * gst/speexresample/gstspeexresample.h:
14549 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
14550 (speex_free), (compute_func), (main), (sinc), (cubic_coef),
14551 (resampler_basic_direct_single), (resampler_basic_direct_double),
14552 (resampler_basic_interpolate_single),
14553 (resampler_basic_interpolate_double), (update_filter),
14554 (speex_resampler_init), (speex_resampler_init_frac),
14555 (speex_resampler_destroy), (speex_resampler_process_native),
14556 (speex_resampler_process_float), (speex_resampler_process_int),
14557 (speex_resampler_process_interleaved_float),
14558 (speex_resampler_process_interleaved_int),
14559 (speex_resampler_set_rate), (speex_resampler_get_rate),
14560 (speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
14561 (speex_resampler_set_quality), (speex_resampler_get_quality),
14562 (speex_resampler_set_input_stride),
14563 (speex_resampler_get_input_stride),
14564 (speex_resampler_set_output_stride),
14565 (speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
14566 (speex_resampler_reset_mem), (speex_resampler_strerror):
14567 * gst/speexresample/speex_resampler.h:
14568 * gst/speexresample/speex_resampler_float.c:
14569 * gst/speexresample/speex_resampler_int.c:
14570 * gst/speexresample/speex_resampler_wrapper.h:
14571 Add resample element based on the Speex resampling algorithm.
14573 2007-11-19 12:30:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14575 tests/check/libs/fft.c: Fix scaling to really have dB instead of something else.
14576 Original commit message from CVS:
14577 * tests/check/libs/fft.c: (GST_START_TEST):
14578 Fix scaling to really have dB instead of something else.
14580 2007-11-19 12:08:16 +0000 Julien Moutte <julien@moutte.net>
14582 tests/examples/seek/seek.c: There's a nice macro to check
14583 Original commit message from CVS:
14584 2007-11-19 Julien MOUTTE <julien@moutte.net>
14585 * tests/examples/seek/seek.c: (main): There's a nice macro to
14587 GTK version, use it.
14589 2007-11-19 11:59:20 +0000 Julien Moutte <julien@moutte.net>
14591 tests/examples/seek/seek.c: Try to support stable version of GTK.
14592 Original commit message from CVS:
14593 2007-11-19 Julien MOUTTE <julien@moutte.net>
14594 * tests/examples/seek/seek.c: (main): Try to support stable version
14597 2007-11-17 15:25:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14599 gst/playback/: Fix the build + little README update.
14600 Original commit message from CVS:
14601 * gst/playback/README:
14602 * gst/playback/test7.c:
14603 Fix the build + little README update.
14605 2007-11-16 16:02:45 +0000 Wim Taymans <wim.taymans@gmail.com>
14607 tests/examples/seek/seek.c: Add playbin2 seek pipeline.
14608 Original commit message from CVS:
14609 * tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
14610 Add playbin2 seek pipeline.
14612 2007-11-16 15:44:48 +0000 Wim Taymans <wim.taymans@gmail.com>
14614 gst/playback/: Add playbin2.
14615 Original commit message from CVS:
14616 * gst/playback/Makefile.am:
14617 * gst/playback/gstplayback.c: (plugin_init):
14618 * gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
14619 (eos_cb), (about_to_finish_cb), (main):
14621 Added gapless playback example.
14622 * gst/playback/gstplaybasebin.c:
14623 * gst/playback/gstplaybasebin.h:
14624 * gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
14625 * gst/playback/gstqueue2.c:
14626 * gst/playback/test.c:
14627 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
14629 * gst/playback/gststreaminfo.h:
14631 * gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
14632 (gst_play_bin_class_init), (init_group), (gst_play_bin_init),
14633 (gst_play_bin_dispose), (gst_play_bin_set_uri),
14634 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
14635 (gst_play_bin_get_property), (gst_play_bin_handle_message),
14636 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
14637 (drained_cb), (unlink_group), (activate_group),
14638 (setup_next_source), (gst_play_bin_change_state),
14639 (gst_play_bin2_plugin_init):
14640 Added raw first version of playbin2. Does chained oggs and gapless
14641 playback fine. No support for raw sinks yet. No visualisations or
14643 * gst/playback/gstplaysink.c: (gst_play_sink_get_type),
14644 (gst_play_sink_class_init), (gst_play_sink_init),
14645 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
14646 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
14647 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
14648 (gst_play_sink_set_property), (gst_play_sink_get_property),
14649 (post_missing_element_message), (free_chain), (add_chain),
14650 (activate_chain), (gen_video_chain), (gen_text_element),
14651 (gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
14652 (gst_play_sink_set_mode), (gst_play_sink_request_pad),
14653 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
14654 (gst_play_sink_send_event), (gst_play_sink_change_state):
14655 * gst/playback/gstplaysink.h:
14656 Added Element that abstracts the sinks and their pipelines for playbin2.
14658 2007-11-16 15:05:07 +0000 Wim Taymans <wim.taymans@gmail.com>
14660 gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
14661 Original commit message from CVS:
14662 * gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
14663 (gst_selector_pad_class_init), (gst_selector_pad_init),
14664 (gst_selector_pad_finalize), (gst_selector_pad_reset),
14665 (gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
14666 (gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
14667 (gst_selector_pad_chain), (gst_stream_selector_get_type),
14668 (gst_stream_selector_base_init), (gst_stream_selector_class_init),
14669 (gst_stream_selector_init), (gst_stream_selector_set_property),
14670 (gst_stream_selector_get_linked_pad),
14671 (gst_stream_selector_getcaps),
14672 (gst_stream_selector_is_active_sinkpad),
14673 (gst_stream_selector_activate_sinkpad),
14674 (gst_stream_selector_get_linked_pads),
14675 (gst_stream_selector_request_new_pad),
14676 (gst_stream_selector_release_pad):
14677 * gst/playback/gststreamselector.h:
14678 Improve streamselector, make it select and unselect the current pad more
14680 Subclass GstPad for the sinkpads of the selector.
14681 Handle segments more correctly.
14682 Fix caps negotiation.
14683 Implement release_pad.
14685 2007-11-16 12:51:44 +0000 Wim Taymans <wim.taymans@gmail.com>
14687 gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
14688 Original commit message from CVS:
14689 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
14690 (gst_decode_group_check_if_drained), (source_pad_event_probe),
14692 Add drained signal fired when decodebin finishes decoding the data.
14693 Remove deprecated STATE_DIRTY message.
14694 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
14695 (unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
14696 (analyse_source), (proxy_drained_signal), (make_decoder),
14697 (source_new_pad), (value_list_append_structure_list),
14698 (handle_redirect_message), (handle_message):
14699 Proxy the new drained signal.
14700 Handle pad removed from decodebin.
14701 Handle redirect messages by sorting multiple redirections based on the
14704 2007-11-16 11:22:09 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
14706 gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
14707 Original commit message from CVS:
14708 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
14709 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
14710 Fix leaking headers. Fixes #496761.
14712 2007-11-16 11:16:58 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
14714 sys/: Don't leak the PAR on errors. Fixes #496731.
14715 Original commit message from CVS:
14716 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
14717 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
14718 (gst_ximagesink_change_state):
14719 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
14720 Don't leak the PAR on errors. Fixes #496731.
14722 2007-11-16 10:14:34 +0000 Tim-Philipp Müller <tim@centricular.net>
14724 gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
14725 Original commit message from CVS:
14726 * gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
14727 (gst_tag_from_id3_user_tag):
14728 Add mapping for audio cd discid tags, so we can extract
14729 them from tags as well (see #347848). Also compare identifiers
14730 in ID3v2 TXXX frames in a case-insensitive way to increase
14731 compatibility when reading tags (discid vs. DiscID vs. DiscId).
14733 2007-11-16 01:21:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14735 gst-plugins-base.doap: Oops, fix the release name.
14736 Original commit message from CVS:
14737 * gst-plugins-base.doap:
14738 Oops, fix the release name.
14740 2007-11-16 00:44:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14742 gst-plugins-base.doap: Add 0.10.15 release
14743 Original commit message from CVS:
14744 * gst-plugins-base.doap:
14745 Add 0.10.15 release
14747 2007-11-16 00:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14749 configure.ac: Back to CVS
14750 Original commit message from CVS:
14754 === release 0.10.15 ===
14756 2007-11-16 00:14:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14758 configure.ac: releasing 0.10.15, "No need to argue"
14759 Original commit message from CVS:
14760 === release 0.10.15 ===
14761 2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
14763 releasing 0.10.15, "No need to argue"
14765 2007-11-16 00:04:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14790 Original commit message from CVS:
14793 2007-11-15 21:40:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14795 win32/vs6/libgstfft.dsp: Convert line endings to DOS.
14796 Original commit message from CVS:
14797 * win32/vs6/libgstfft.dsp:
14798 Convert line endings to DOS.
14800 2007-11-15 21:14:04 +0000 Sébastien Moutte <sebastien@moutte.net>
14802 win32/: Add a project file for fft plugin and remove socket based plugin which don't build from the workspace.* win32...
14803 Original commit message from CVS:
14804 * win32/vs6/gst_plugins_base.dsw:
14805 * win32/vs6/libgstfft.dsp:
14807 Add a project file for fft plugin and remove socket
14808 based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
14809 * win32/vs6/libgstrtp.dsp:
14810 * win32/vs6/libgsttag.dsp:
14811 Convert line endings back to DOS.
14814 2007-11-14 12:27:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14816 win32/vs6/: Convert line endings back to DOS
14817 Original commit message from CVS:
14818 * win32/vs6/libgstinterfaces.dsp:
14819 * win32/vs6/libgstrtsp.dsp:
14820 Convert line endings back to DOS
14822 2007-11-14 11:08:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14824 gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
14825 Original commit message from CVS:
14826 * gst-libs/gst/fft/kiss_fft_f32.h:
14827 * gst-libs/gst/fft/kiss_fft_f64.h:
14828 * gst-libs/gst/fft/kiss_fft_s16.h:
14829 * gst-libs/gst/fft/kiss_fft_s32.h:
14830 Don't include malloc.h which doesn't exist on Mac OSX.
14831 Instead, pull in glib.h and use g_malloc/g_free for
14832 consistency. Fixes: #496548
14834 2007-11-09 15:54:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14836 gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
14837 Original commit message from CVS:
14838 * gst/playback/gstdecodebin2.c:
14839 Dont leak ghostpad. Fixes #475451.
14841 2007-11-09 12:21:52 +0000 Wim Taymans <wim.taymans@gmail.com>
14843 Update some more docs and comments.
14844 Original commit message from CVS:
14845 * docs/design/design-decodebin.txt:
14846 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
14847 Update some more docs and comments.
14849 2007-11-07 16:47:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14851 Require GIO >= 0.1.2 and adjust unit test for an API change.
14852 Original commit message from CVS:
14854 * tests/check/pipelines/gio.c: (GST_START_TEST):
14855 Require GIO >= 0.1.2 and adjust unit test for an API change.
14857 2007-11-07 15:18:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14859 ext/gio/gstgio.h: Add macro to check if a stream supports seeking.
14860 Original commit message from CVS:
14861 * ext/gio/gstgio.h:
14862 Add macro to check if a stream supports seeking.
14863 * ext/gio/Makefile.am:
14864 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
14865 (gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
14866 (gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
14867 (gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
14868 (gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
14869 (gst_gio_base_sink_render), (gst_gio_base_sink_query),
14870 (gst_gio_base_sink_set_stream):
14871 * ext/gio/gstgiobasesink.h:
14872 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
14873 (gst_gio_base_src_class_init), (gst_gio_base_src_init),
14874 (gst_gio_base_src_finalize), (gst_gio_base_src_start),
14875 (gst_gio_base_src_stop), (gst_gio_base_src_get_size),
14876 (gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
14877 (gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
14878 (gst_gio_base_src_create), (gst_gio_base_src_set_stream):
14879 * ext/gio/gstgiobasesrc.h:
14880 Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
14881 base classes that only require a GInputStream or GOutputStream to
14883 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
14884 (gst_gio_sink_class_init), (gst_gio_sink_init),
14885 (gst_gio_sink_finalize), (gst_gio_sink_start):
14886 * ext/gio/gstgiosink.h:
14887 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
14888 (gst_gio_src_class_init), (gst_gio_src_init),
14889 (gst_gio_src_finalize), (gst_gio_src_start):
14890 * ext/gio/gstgiosrc.h:
14891 Use the newly created base classes here.
14892 * ext/gio/gstgio.c: (plugin_init):
14893 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
14894 (gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
14895 (gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
14896 (gst_gio_stream_sink_get_property):
14897 * ext/gio/gstgiostreamsink.h:
14898 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
14899 (gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
14900 (gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
14901 (gst_gio_stream_src_get_property):
14902 * ext/gio/gstgiostreamsrc.h:
14903 Implement GstGioStreamSink and GstGioStreamSrc that have a property
14904 to set the GInputStream/GOutputStream that should be used.
14905 * tests/check/Makefile.am:
14906 * tests/check/pipelines/.cvsignore:
14907 * tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
14908 (gio_testsuite), (main):
14909 Add unit test for giostreamsrc and giostreamsink.
14911 2007-11-07 11:48:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14913 ext/gio/gstgio.c: Remove nowadays unnecessary workaround for a crash.
14914 Original commit message from CVS:
14915 * ext/gio/gstgio.c: (plugin_init):
14916 Remove nowadays unnecessary workaround for a crash.
14917 * ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
14918 (gst_gio_sink_start), (gst_gio_sink_stop),
14919 (gst_gio_sink_unlock_stop):
14920 * ext/gio/gstgiosink.h:
14921 * ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
14922 (gst_gio_src_stop), (gst_gio_src_unlock_stop):
14923 * ext/gio/gstgiosrc.h:
14924 Make the finalize function safer, clean up everything that could stay
14926 Reset the cancellable instead of creating a new one after cancelling
14928 Don't store the GFile in the element, it's only necessary for creating
14931 2007-11-06 23:35:39 +0000 Sebastien Moutte <sebastien@moutte.net>
14933 gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
14934 Original commit message from CVS:
14935 Patch by: Sebastien Moutte <sebastien moutte net>
14936 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
14937 (gst_rtcp_unix_to_ntp):
14938 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
14939 Fix some C99-isms and and a missing function that some versions of
14940 MSVC don't like too much (#494346).
14941 * win32/vs6/gst_plugins_base.dsw:
14942 * win32/vs6/libgstaudio.dsp:
14943 * win32/vs6/libgstrtp.dsp:
14944 * win32/vs6/libgsttag.dsp:
14945 Update vs6 projects files (#494346).
14947 2007-11-06 16:38:49 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
14949 win32/common/: More missing symbols to export (fixes #493986).
14950 Original commit message from CVS:
14951 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
14952 * win32/common/libgstaudio.def:
14953 * win32/common/libgstcdda.def:
14954 * win32/common/libgstinterfaces.def:
14955 * win32/common/libgstnetbuffer.def:
14956 * win32/common/libgstpbutils.def:
14957 * win32/common/libgstrtp.def:
14958 * win32/common/libgstrtsp.def:
14959 * win32/common/libgsttag.def:
14960 * win32/common/libgstvideo.def:
14961 More missing symbols to export (fixes #493986).
14963 2007-11-06 11:58:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14965 Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
14966 Original commit message from CVS:
14967 * docs/libs/gst-plugins-base-libs-sections.txt:
14968 * gst-libs/gst/fft/gstfftf32.c:
14969 * gst-libs/gst/fft/gstfftf32.h:
14970 * gst-libs/gst/fft/gstfftf64.c:
14971 * gst-libs/gst/fft/gstfftf64.h:
14972 * gst-libs/gst/fft/gstffts16.c:
14973 * gst-libs/gst/fft/gstffts16.h:
14974 * gst-libs/gst/fft/gstffts32.c:
14975 * gst-libs/gst/fft/gstffts32.h:
14976 * tests/check/libs/fft.c: (GST_START_TEST):
14977 Remove the magnitude and phase calculation functions as these have
14978 very special use cases and can't even be used for the spectrum
14979 element. Also adjust the docs to mention some properties of the used
14980 FFT implemention, i.e. how the values are scaled. Fixes #492098.
14982 2007-11-06 11:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
14984 gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
14985 Original commit message from CVS:
14986 * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
14988 Avoid crash when there are external subtitles (fixes #491722).
14990 2007-11-03 10:39:21 +0000 Tim-Philipp Müller <tim@centricular.net>
14992 ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
14993 Original commit message from CVS:
14994 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
14995 * ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
14996 'Could not open resource for writing' is not an acceptable
14997 error message when we can't open the audio device (see #492334),
14998 even less so when we're trying to open it to record something.
15000 2007-11-02 21:03:01 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15002 win32/common/libgstrtp.def: Add some more missing symbols (#492813).
15003 Original commit message from CVS:
15004 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15005 * win32/common/libgstrtp.def:
15006 Add some more missing symbols (#492813).
15008 2007-11-02 14:59:06 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
15010 tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
15011 Original commit message from CVS:
15012 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
15013 * tests/check/elements/audioconvert.c: (verify_convert):
15014 Add check to make sure that the out caps have a channel layout
15015 set on them where they should have one.
15017 2007-11-01 13:28:59 +0000 Vincent Torri <vtorri@univ-evry.fr>
15019 gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
15020 Original commit message from CVS:
15021 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
15022 * gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
15023 * gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
15024 Include our own _stdint.h instead of sys/types.h, makes MingW happy
15026 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
15027 Use _pipe directly, GLib doesn't have a pipe() macro any longer
15028 (it disappeared in GLib 2.14.0) (#492306).
15029 * gst-libs/gst/sdp/Makefile.am:
15030 * gst-libs/gst/sdp/gstsdpmessage.c:
15031 Fix includes and LIBS for win32/Mingw (#492306).
15032 * tests/examples/dynamic/addstream.c (pause_play_stream):
15033 Use more portable g_usleep() instead of sleep() (#492306).
15035 2007-11-01 12:51:57 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15037 gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
15038 Original commit message from CVS:
15039 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15040 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
15041 (gst_ring_buffer_parse_caps):
15042 Return NULL instead of an enum that happens to be 0, fixes warning
15044 * gst-libs/gst/audio/gstringbuffer.h:
15045 No trailing commas in enum list (for gcc-2.9x).
15046 * gst/videotestsrc/videotestsrc.c: (random_char):
15047 Make information loss explicit instead of implicitly truncating to
15048 eight bits via the return value. Fixes runtime error on MSVC when
15049 using the debug CRT (#492114).
15050 * win32/common/config.h.in:
15051 Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
15052 * win32/common/libgstinterfaces.def:
15053 * win32/common/libgstrtp.def:
15054 Export a few more symbols (#492114).
15056 2007-11-01 08:06:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15058 gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
15059 Original commit message from CVS:
15060 * gst-libs/gst/audio/audio.c:
15061 * gst-libs/gst/audio/audio.h:
15062 Readd the deprecation guards, but preserve compilability.
15064 2007-10-31 17:54:48 +0000 Tim-Philipp Müller <tim@centricular.net>
15066 gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
15067 Original commit message from CVS:
15068 * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
15069 (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
15070 Preserve channel layout when fixating the number of channels in the
15071 output caps, or make sure there's a suitable channel position layout
15072 set on the caps if required. Fixes #430677.
15074 2007-10-31 17:32:22 +0000 Tim-Philipp Müller <tim@centricular.net>
15076 tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
15077 Original commit message from CVS:
15078 * tests/check/elements/decodebin.c: (test_text_plain_streams):
15079 Make sure the pipeline really operates in push mode as it should
15082 2007-10-31 15:30:15 +0000 Tim-Philipp Müller <tim@centricular.net>
15084 gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
15085 Original commit message from CVS:
15086 * gst-libs/gst/audio/audio.h:
15087 Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
15088 compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
15089 (ie. normal cvs builds) will fail.
15091 2007-10-31 12:47:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15093 tell gtk-doc about the deprecation guard. Apply more doc fixes.
15094 Original commit message from CVS:
15095 * docs/libs/Makefile.am:
15096 * gst-libs/gst/audio/audio.c:
15097 * gst-libs/gst/audio/audio.h:
15098 * gst-libs/gst/interfaces/mixer.c:
15099 tell gtk-doc about the deprecation guard. Apply more doc fixes.
15101 2007-10-31 12:30:28 +0000 Tim-Philipp Müller <tim@centricular.net>
15103 tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
15104 Original commit message from CVS:
15105 * tests/check/libs/audio.c: (init_value_to_channel_layout),
15106 (test_channel_layout_value_intersect), (audio_suite):
15107 Add simple unit test to make sure GstValue intersection
15108 of channel layouts works the way I think it does.
15110 2007-10-30 20:32:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15112 Fix the docs according to what gtk-doc complained about.
15113 Original commit message from CVS:
15114 * docs/libs/gst-plugins-base-libs-sections.txt:
15115 * gst-libs/gst/audio/gstaudiofilter.h:
15116 * gst-libs/gst/interfaces/mixer.h:
15117 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15118 * gst-libs/gst/rtp/gstbasertpdepayload.h:
15119 * gst-libs/gst/sdp/gstsdpmessage.c:
15120 Fix the docs according to what gtk-doc complained about.
15122 2007-10-30 19:46:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15124 tests/icles/stress-playbin.c: Fix the build.
15125 Original commit message from CVS:
15126 * tests/icles/stress-playbin.c:
15129 2007-10-30 15:54:46 +0000 Tim-Philipp Müller <tim@centricular.net>
15131 gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
15132 Original commit message from CVS:
15133 * gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
15134 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
15135 Post nice/more useful error message if we don't have a decoder for
15138 2007-10-30 15:07:58 +0000 Wim Taymans <wim.taymans@gmail.com>
15140 gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
15141 Original commit message from CVS:
15142 * gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
15143 Be a bit more useful, unblock the pads after we fired the no-more-pads
15144 signal so that we can use the signal to inspect and connect all pads
15145 without having to keep extra state outside of decodebin.
15147 2007-10-30 15:00:06 +0000 Wim Taymans <wim.taymans@gmail.com>
15149 gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
15150 Original commit message from CVS:
15151 * gst/playback/gsturidecodebin.c:
15152 (gst_uri_decode_bin_autoplug_continue),
15153 (gst_uri_decode_bin_class_init), (no_more_pads_full):
15154 Implement default signal handler so that we return TRUE when nothing is
15157 2007-10-28 11:53:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15159 gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
15160 Original commit message from CVS:
15161 * gst-libs/gst/riff/riff-media.c:
15162 (gst_riff_wavext_add_channel_layout),
15163 (gst_riff_wave_add_default_channel_layout),
15164 (gst_riff_wavext_get_default_channel_mask),
15165 (gst_riff_create_audio_caps):
15166 Use the ALSA channel layout as default for wav files without channel
15167 layout information. This fixes playback of chan-id.wav on 5.1 systems
15168 for example. Also refactor the channel layout setting a bit and add
15169 more default channel orders. Fixes #489010.
15171 2007-10-28 11:46:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15174 Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
15175 Original commit message from CVS:
15176 (gst_riff_wavext_add_channel_layout),
15177 (gst_riff_wave_add_default_channel_layout),
15178 (gst_riff_wavext_get_default_channel_mask),
15179 (gst_riff_create_audio_caps):
15180 Use the ALSA channel layout as default for wav files without channel
15181 layout information. This fixes playback of chan-id.wav on 5.1 systems
15182 for example. Also refactor the channel layout setting a bit and add
15183 more default channel orders. Fixes #489010.
15185 2007-10-26 18:57:33 +0000 Tim-Philipp Müller <tim@centricular.net>
15187 tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
15188 Original commit message from CVS:
15189 * tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
15190 GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
15191 -DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
15194 2007-10-26 12:07:14 +0000 Christian Schaller <uraeus@gnome.org>
15196 * gst-plugins-base.spec.in:
15198 Original commit message from CVS:
15201 2007-10-25 17:36:49 +0000 Wim Taymans <wim.taymans@gmail.com>
15203 gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
15204 Original commit message from CVS:
15205 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
15206 (gst_decode_bin_dispose), (gst_decode_bin_set_caps),
15207 (gst_decode_bin_set_subs_encoding),
15208 (gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
15209 (gst_decode_bin_get_property), (analyze_new_pad):
15210 Move subtitle encoding property to decodebin2 so that it can set the
15211 property value on all elements that it autoplugs and that require it.
15212 Make caps refcounting more consistent in get/set.
15213 * gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
15214 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
15215 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
15216 (gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
15217 (proxy_autoplug_continue_signal),
15218 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
15220 Proxy properties and relevant signals from the internal decodebin.
15221 Make properties MT safe.
15223 2007-10-25 15:10:59 +0000 Tim-Philipp Müller <tim@centricular.net>
15225 gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
15226 Original commit message from CVS:
15227 * gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
15228 * gst-libs/gst/tag/tags.c:
15229 Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
15230 GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
15231 * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
15232 Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
15233 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
15234 (gst_tag_to_vorbis_comments):
15235 Map new SORTNAME tags (these tags aren't even semi-official, so I'm
15236 just mapping everything I found in the wild) (#414539).
15238 2007-10-24 11:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
15240 gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
15241 Original commit message from CVS:
15242 Inspired by patch of: René Stadler <mail at renestadler dot de>
15243 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
15244 (gst_decode_bin_autoplug_continue),
15245 (gst_decode_bin_autoplug_factories),
15246 (gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
15247 (find_compatibles):
15248 * gst/playback/gstplay-marshal.list:
15249 Remove the autoplug-sort signal and replace it with a binding friendly
15250 autoplug-select signal.
15251 Add an autoplug-factories signal that can be used to generate a list of
15252 factories to try to autoplug.
15253 Add the GstPad to the autoplugging signal args as it might be needed to
15254 make a good factory selection.
15255 Fix up the marshallers for this. Fixes #407282.
15257 2007-10-23 14:23:14 +0000 Tim-Philipp Müller <tim@centricular.net>
15259 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...
15260 Original commit message from CVS:
15261 * gst-libs/gst/tag/gsttagdemux.c:
15262 Don't abort with an assertion if we receive a seek event with
15263 a start type of NONE (see launchpad bug #155878).
15265 2007-10-22 10:21:46 +0000 Wim Taymans <wim.taymans@gmail.com>
15267 sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
15268 Original commit message from CVS:
15269 * sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
15270 (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
15271 (gst_ximagesink_change_state), (gst_ximagesink_reset):
15272 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
15273 (gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
15274 (gst_xvimagesink_change_state), (gst_xvimagesink_reset):
15275 Make sure that before we clean up the X resources, we shutdown and join
15277 Also make sure the event thread does not shut down immediatly after
15278 startup because the running variable is not yet correctly set.
15281 2007-10-16 16:48:38 +0000 Wim Taymans <wim.taymans@gmail.com>
15283 gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
15284 Original commit message from CVS:
15285 * gst/playback/gstdecodebin.c: (new_pad), (type_found):
15286 Make the window for a race in typefind and shutting down smaller until
15287 we figure out the right locking here. Avoids #485753 usually.
15288 * gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
15289 Remove unneeded lock causing a race in typefind and shutting down.
15291 * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
15292 Also remove sinks when going to NULL because we might not complete the
15293 state change to PAUSED, causing the PAUSED->READY state change not to
15296 2007-10-16 15:33:31 +0000 Wim Taymans <wim.taymans@gmail.com>
15298 gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
15299 Original commit message from CVS:
15300 * gst-libs/gst/audio/gstbaseaudiosink.c:
15301 (gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
15302 Also explicitly release the ringbuffer when going to NULL because it
15303 is required in the setcaps function, before the state change to PAUSED
15306 2007-10-16 14:58:53 +0000 Tim-Philipp Müller <tim@centricular.net>
15308 tests/icles/: Does what it says on the tin.
15309 Original commit message from CVS:
15310 * tests/icles/.cvsignore:
15311 * tests/icles/Makefile.am:
15312 * tests/icles/stress-playbin.c:
15313 Does what it says on the tin.
15315 2007-10-15 11:38:39 +0000 Wim Taymans <wim.taymans@gmail.com>
15317 gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
15318 Original commit message from CVS:
15319 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
15320 Fix queue negotiation. See #486758.
15322 2007-10-12 10:52:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15324 Actual code change to go along with:
15325 Original commit message from CVS:
15326 Actual code change to go along with:
15327 2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
15328 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
15329 (gst_xvimagesink_xwindow_new),
15330 (gst_xvimagesink_update_colorbalance),
15331 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
15332 Fix handling of some of the X atoms. If the last parameter is True,
15333 XInternAtom won't create the atom if it doesn't exist, and therefore
15334 might return None. This causes X errors on Xv implementations that
15335 don't provide the colour balance attributes.
15337 2007-10-12 10:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15340 Remove stray character from the changelog.
15341 Original commit message from CVS:
15342 Remove stray character from the changelog.
15344 2007-10-12 10:33:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15347 I'm too lazy to comment this
15348 Original commit message from CVS:
15349 *** empty log message ***
15351 2007-10-11 18:24:09 +0000 Tim-Philipp Müller <tim@centricular.net>
15353 Extract vorbis comment LICENSE tags correctly.
15354 Original commit message from CVS:
15355 * gst-libs/gst/tag/gstvorbistag.c:
15356 * tests/check/libs/tag.c:
15357 Extract vorbis comment LICENSE tags correctly.
15359 2007-10-11 16:12:21 +0000 Jason Kivlighn <jkivlighn@gmail.com>
15361 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
15362 Original commit message from CVS:
15363 Patch by: Jason Kivlighn <jkivlighn gmail com>
15364 * gst-libs/gst/tag/gstid3tag.c:
15365 * tests/check/libs/tag.c:
15366 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
15368 2007-10-10 17:01:51 +0000 Tim-Philipp Müller <tim@centricular.net>
15370 gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
15371 Original commit message from CVS:
15372 * gst-libs/gst/tag/gsttagdemux.c:
15373 Don't error out when a buggy downstream element doesn't
15374 handle the newsegment event we send properly (especially
15375 not without posting a meaningful error message on the
15376 bus). See bug #471370 and launchpad bug #136264.
15378 2007-10-10 15:36:56 +0000 Wim Taymans <wim.taymans@gmail.com>
15380 gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
15381 Original commit message from CVS:
15382 * gst-libs/gst/audio/gstbaseaudiosink.c:
15383 (gst_base_audio_sink_drain):
15384 Use new basesink method to make our EOS drain interruptable.
15386 2007-10-10 09:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15388 gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
15389 Original commit message from CVS:
15390 * gst-libs/gst/rtp/gstrtppayloads.c:
15391 Fix silly search-replace oversight.
15393 2007-10-09 09:57:17 +0000 Laurent Glayal <spglegle@yahoo.fr>
15395 gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
15396 Original commit message from CVS:
15397 Patch by: Laurent Glayal <spglegle at yahoo dot fr>
15398 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
15399 (gst_basertppayload_set_outcaps):
15400 Fix caps memleak. Fixes #484989.
15402 2007-10-08 18:04:34 +0000 Wim Taymans <wim.taymans@gmail.com>
15404 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
15405 Original commit message from CVS:
15406 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15407 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
15410 2007-10-08 18:02:53 +0000 Wim Taymans <wim.taymans@gmail.com>
15412 gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
15413 Original commit message from CVS:
15414 * gst-libs/gst/audio/gstbaseaudiosrc.c:
15415 (gst_base_audio_src_create):
15416 Also handle the case where there is no clock set on the audio source,
15417 like in the unit tests.
15419 2007-10-08 17:40:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15421 gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
15422 Original commit message from CVS:
15423 * gst-libs/gst/rtp/gstrtppayloads.c:
15424 Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
15425 to avoid compiler warnings
15427 2007-10-08 17:12:32 +0000 Wim Taymans <wim.taymans@gmail.com>
15429 gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
15430 Original commit message from CVS:
15431 * gst/playback/gstdecodebin.c: (type_found),
15432 (gst_decode_bin_change_state):
15433 * gst/playback/gstdecodebin2.c: (type_found),
15434 (gst_decode_bin_change_state):
15435 Don't disconnect the have_type signal because we never reconnect it
15436 later on. Instead keep a variable to see if we already detected a type.
15438 2007-10-08 10:47:26 +0000 Wim Taymans <wim.taymans@gmail.com>
15440 gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
15441 Original commit message from CVS:
15442 * gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
15443 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
15445 Unlink the signal handler when we found the type, we're not going to do
15446 anything sensible with more type_found signals anyway.
15448 2007-10-08 06:07:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15450 ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
15451 Original commit message from CVS:
15452 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
15453 Use GIO function to get a list of supported URI schemes instead of
15454 hard coding something.
15456 2007-10-06 16:49:55 +0000 Tim-Philipp Müller <tim@centricular.net>
15458 gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
15459 Original commit message from CVS:
15460 * gst-libs/gst/tag/gsttagdemux.c:
15463 2007-10-06 15:04:53 +0000 Tim-Philipp Müller <tim@centricular.net>
15465 gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
15466 Original commit message from CVS:
15467 * gst-libs/gst/tag/Makefile.am:
15468 * gst-libs/gst/tag/gsttagdemux.c:
15469 * gst-libs/gst/tag/gsttagdemux.h:
15470 API: add GstTagDemux base class for simple tag demuxers.
15471 * docs/libs/gst-plugins-base-libs-docs.sgml:
15472 * docs/libs/gst-plugins-base-libs-sections.txt:
15473 Add GstTagDemux to docs.
15475 2007-10-05 07:49:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15477 gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
15478 Original commit message from CVS:
15479 * gst-libs/gst/rtp/gstrtpbuffer.c:
15480 (gst_rtp_buffer_get_payload_subbuffer):
15481 Fix bug introduced with last commit which inverted the logic and
15482 caused all buffers to be dropped. Fixes #483620.
15483 Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
15485 2007-10-04 06:50:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15487 gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
15488 Original commit message from CVS:
15489 * gst-libs/gst/rtp/gstrtpbuffer.c:
15490 Replace g_return_if_val (as it could be disabled), with regular return
15493 2007-10-03 14:51:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15495 tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
15496 Original commit message from CVS:
15497 * tests/check/pipelines/simple-launch-lines.c:
15498 Print message name and not just number.
15500 2007-10-02 11:11:13 +0000 Wim Taymans <wim.taymans@gmail.com>
15502 gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
15503 Original commit message from CVS:
15504 * gst-libs/gst/audio/gstbaseaudiosink.c:
15505 (gst_base_audio_sink_async_play):
15506 When slaved to the clock, don't try to align a sample with the previous
15507 one when going to PLAYING again.
15509 2007-10-02 09:04:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15511 tests/examples/snapshot/snapshot.c: Fix the build.
15512 Original commit message from CVS:
15513 * tests/examples/snapshot/snapshot.c:
15516 2007-10-02 07:43:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15518 ext/gio/gstgiosink.c: Update to API changes in GIO.
15519 Original commit message from CVS:
15520 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
15521 Update to API changes in GIO.
15523 2007-10-01 16:33:00 +0000 Wim Taymans <wim.taymans@gmail.com>
15525 gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
15526 Original commit message from CVS:
15527 * gst-libs/gst/sdp/gstsdpmessage.h:
15528 Add RFC 3556 bandwidth modifiers.
15530 2007-10-01 13:37:31 +0000 Wim Taymans <wim.taymans@gmail.com>
15532 Update documentation.
15533 Original commit message from CVS:
15534 * docs/libs/gst-plugins-base-libs-docs.sgml:
15535 * docs/libs/gst-plugins-base-libs-sections.txt:
15536 * gst-libs/gst/rtp/gstrtppayloads.c:
15537 Update documentation.
15539 2007-10-01 13:22:14 +0000 Wim Taymans <wim.taymans@gmail.com>
15541 gst-libs/gst/rtp/: Added new file and header to deal with payload info.
15542 Original commit message from CVS:
15543 * gst-libs/gst/rtp/Makefile.am:
15544 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
15545 (gst_rtp_payload_info_for_name):
15546 * gst-libs/gst/rtp/gstrtppayloads.h:
15547 Added new file and header to deal with payload info.
15548 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
15549 (gst_rtp_buffer_default_clock_rate):
15550 * gst-libs/gst/rtp/gstrtpbuffer.h:
15551 Payload specific stuff is move to new headers.
15552 Implement _default_clock rate using the new payload function.
15553 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
15554 (gst_sdp_parse_line):
15555 * gst-libs/gst/sdp/gstsdpmessage.h:
15556 Add some more comments.
15558 2007-10-01 10:22:46 +0000 Wim Taymans <wim.taymans@gmail.com>
15560 gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
15561 Original commit message from CVS:
15562 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
15563 (sdp_check_header), (sdp_type_find), (plugin_init):
15564 Add typefind function for application/sdp.
15565 Remove some old dirac typefind code that was ifdeffed out.
15567 2007-09-29 12:04:02 +0000 Sébastien Moutte <sebastien@moutte.net>
15569 win32/common/libgstaudio.def: Add new exported functions.
15570 Original commit message from CVS:
15571 * win32/common/libgstaudio.def:
15572 Add new exported functions.
15573 * win32/vs6/grammar.dsp:
15574 Add autogeneration and copy of some autegenerated files from win32/common
15576 * win32/vs6/libgstaudioconvert.dsp:
15577 Add gstaudioquantize.c to the build.
15578 * win32/vs6/libgstinterfaces.dsp:
15579 Add videoorientation.c to the build.
15580 * win32/vs6/libgstriff.dsp:
15581 Add libgsttag to the link libraries list.
15582 * win32/vs6/libgstvolume.dsp:
15583 Add liboil to the link.
15584 * win32/vs6/gst_plugins_base.dsw:
15585 * win32/vs6/libgstrtsp.dsp:
15586 * win32/common/libgstrtsp.def:
15587 Add files to build libgstrtsp library.
15589 2007-09-29 07:01:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15591 ext/gio/: Some minor cleanup and allow setting the location only when the element is not playing or paused.
15592 Original commit message from CVS:
15593 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
15594 (gst_gio_sink_set_property), (gst_gio_sink_render):
15595 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
15596 (gst_gio_src_set_property):
15597 Some minor cleanup and allow setting the location only when the
15598 element is not playing or paused.
15600 2007-09-26 15:14:37 +0000 Wim Taymans <wim.taymans@gmail.com>
15602 tests/examples/snapshot/snapshot.c: Print error when pipeline failed to construct.
15603 Original commit message from CVS:
15604 * tests/examples/snapshot/snapshot.c: (main):
15605 Print error when pipeline failed to construct.
15607 2007-09-25 19:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
15609 Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
15610 Original commit message from CVS:
15612 * gst-libs/gst/tag/gstid3tag.c:
15613 * gst-libs/gst/tag/gstvorbistag.c:
15614 Add mappings for the new GST_TAG_COMPOSER for vorbis comments
15617 2007-09-25 11:54:09 +0000 Tim-Philipp Müller <tim@centricular.net>
15619 gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
15620 Original commit message from CVS:
15621 * gst-libs/gst/floatcast/floatcast.h:
15622 Don't include config.h in an installed public header, this
15623 might break compilation of applications that don't have such
15624 a header and doesn't necessarily do what it's supposed to do
15625 anyway (ie. check for the lrint/lrintf defines) (#442065).
15626 Add docs for the various macros and document how this header
15627 has to be used (link against libm, etc.); add a few FIXMEs;
15628 include math.h for non-c99 code path. Based on patch by
15631 2007-09-25 07:50:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15633 configure.ac: Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead of duplicating these macros in confi...
15634 Original commit message from CVS:
15636 Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
15637 of duplicating these macros in configure.ac.
15639 2007-09-22 17:58:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15641 po/: Updated translations to 0.10.14
15642 Original commit message from CVS:
15646 Updated translations to 0.10.14
15648 2007-09-22 17:57:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15652 Original commit message from CVS:
15655 2007-09-22 17:56:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15657 po/pl.po: Added Polish translation.
15658 Original commit message from CVS:
15659 translated by: Jakub Bogusz <qboosh@pld-linux.org>
15661 Added Polish translation.
15663 2007-09-22 17:55:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15665 po/fi.po: Added Finnish translation.
15666 Original commit message from CVS:
15667 translated by: Ilkka Tuohela <hile@iki.fi>
15669 Added Finnish translation.
15671 2007-09-22 17:54:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15673 po/es.po: Added Spanish translation.
15674 Original commit message from CVS:
15675 translated by: Jorge González González <aloriel@gmail.com>
15677 Added Spanish translation.
15679 2007-09-22 17:53:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15681 po/da.po: Added Danish translation.
15682 Original commit message from CVS:
15683 translated by: Mogens Jaeger <mogens@jaeger.tf>
15685 Added Danish translation.
15687 2007-09-22 17:52:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15689 po/zh_CN.po: Added Chinese (simplified) translation.
15690 Original commit message from CVS:
15691 translated by: Funda Wang <fundawang@linux.net.cn>
15693 Added Chinese (simplified) translation.
15695 2007-09-22 17:51:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15697 po/bg.po: Added Bulgarian translation.
15698 Original commit message from CVS:
15699 translated by: Alexander Shopov <ash@contact.bg>
15701 Added Bulgarian translation.
15703 2007-09-21 18:00:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15705 docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy.
15706 Original commit message from CVS:
15707 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
15709 * ext/gio/gstgiosink.h:
15710 * ext/gio/gstgiosrc.h:
15711 Mark private fields of the instance structs private.
15713 2007-09-21 17:31:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15715 docs/plugins/: Add the GIO plugin to the docs and do a make update while doing that.
15716 Original commit message from CVS:
15717 * docs/plugins/Makefile.am:
15718 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
15719 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
15720 * docs/plugins/gst-plugins-bad-plugins.args:
15721 * docs/plugins/gst-plugins-bad-plugins.signals:
15722 * docs/plugins/inspect/plugin-bz2.xml:
15723 * docs/plugins/inspect/plugin-cdxaparse.xml:
15724 * docs/plugins/inspect/plugin-dfbvideosink.xml:
15725 * docs/plugins/inspect/plugin-dtsdec.xml:
15726 * docs/plugins/inspect/plugin-equalizer.xml:
15727 * docs/plugins/inspect/plugin-faac.xml:
15728 * docs/plugins/inspect/plugin-faad.xml:
15729 * docs/plugins/inspect/plugin-filter.xml:
15730 * docs/plugins/inspect/plugin-freeze.xml:
15731 * docs/plugins/inspect/plugin-gio.xml:
15732 * docs/plugins/inspect/plugin-gsm.xml:
15733 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
15734 * docs/plugins/inspect/plugin-h264parse.xml:
15735 * docs/plugins/inspect/plugin-modplug.xml:
15736 * docs/plugins/inspect/plugin-mpeg2enc.xml:
15737 * docs/plugins/inspect/plugin-musepack.xml:
15738 * docs/plugins/inspect/plugin-musicbrainz.xml:
15739 * docs/plugins/inspect/plugin-nsfdec.xml:
15740 * docs/plugins/inspect/plugin-replaygain.xml:
15741 * docs/plugins/inspect/plugin-soundtouch.xml:
15742 * docs/plugins/inspect/plugin-spcdec.xml:
15743 * docs/plugins/inspect/plugin-spectrum.xml:
15744 * docs/plugins/inspect/plugin-speed.xml:
15745 * docs/plugins/inspect/plugin-tta.xml:
15746 * docs/plugins/inspect/plugin-videosignal.xml:
15747 * docs/plugins/inspect/plugin-xingheader.xml:
15748 * docs/plugins/inspect/plugin-xvid.xml:
15749 Add the GIO plugin to the docs and do a make update
15751 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
15752 Fix a small memleak.
15754 2007-09-21 17:07:56 +0000 René Stadler <mail@renestadler.de>
15756 Add a GIO/GVFS plugin with source and sink elements. This will only be enabled when --enable-experimental is given to...
15757 Original commit message from CVS:
15758 Patch by: René Stadler <mail at renestadler dot de>
15761 * ext/gio/Makefile.am:
15762 * ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek),
15763 (gst_gio_get_supported_protocols),
15764 (gst_gio_uri_handler_get_type_sink),
15765 (gst_gio_uri_handler_get_type_src),
15766 (gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri),
15767 (gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init),
15768 (gst_gio_uri_handler_do_init), (plugin_init):
15769 * ext/gio/gstgio.h:
15770 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
15771 (gst_gio_sink_class_init), (gst_gio_sink_init),
15772 (gst_gio_sink_finalize), (gst_gio_sink_set_property),
15773 (gst_gio_sink_get_property), (gst_gio_sink_start),
15774 (gst_gio_sink_stop), (gst_gio_sink_unlock),
15775 (gst_gio_sink_unlock_stop), (gst_gio_sink_event),
15776 (gst_gio_sink_render), (gst_gio_sink_query):
15777 * ext/gio/gstgiosink.h:
15778 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
15779 (gst_gio_src_class_init), (gst_gio_src_init),
15780 (gst_gio_src_finalize), (gst_gio_src_set_property),
15781 (gst_gio_src_get_property), (gst_gio_src_start),
15782 (gst_gio_src_stop), (gst_gio_src_get_size),
15783 (gst_gio_src_is_seekable), (gst_gio_src_unlock),
15784 (gst_gio_src_unlock_stop), (gst_gio_src_check_get_range),
15785 (gst_gio_src_create):
15786 * ext/gio/gstgiosrc.h:
15787 Add a GIO/GVFS plugin with source and sink elements. This will
15788 only be enabled when --enable-experimental is given to configure
15789 for now as the GIO API is not stable yet. Fixes #476916.
15791 2007-09-21 14:37:26 +0000 Wim Taymans <wim.taymans@gmail.com>
15793 gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
15794 Original commit message from CVS:
15795 * gst/playback/gstqueue2.c: (gst_queue_push_one):
15796 Fix compilation wrt printf arguments.
15798 2007-09-20 17:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
15800 examples/app/appsrc_ex.c: Fix compilation after changing the name of a method.
15801 Original commit message from CVS:
15802 * examples/app/appsrc_ex.c: (main):
15803 Fix compilation after changing the name of a method.
15805 2007-09-20 14:09:24 +0000 Wim Taymans <wim.taymans@gmail.com>
15807 Add simple snapshot example program using appsink.
15808 Original commit message from CVS:
15810 * tests/examples/Makefile.am:
15811 * tests/examples/snapshot/.cvsignore:
15812 * tests/examples/snapshot/Makefile.am:
15813 * tests/examples/snapshot/snapshot.c: (main):
15814 Add simple snapshot example program using appsink.
15816 2007-09-20 13:59:50 +0000 Wim Taymans <wim.taymans@gmail.com>
15818 gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
15819 Original commit message from CVS:
15820 * gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
15821 (gst_app_sink_class_init), (gst_app_sink_init),
15822 (gst_app_sink_dispose), (gst_app_sink_finalize),
15823 (gst_app_sink_set_property), (gst_app_sink_get_property),
15824 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
15825 (gst_app_sink_event), (gst_app_sink_getcaps),
15826 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
15827 (gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
15828 (gst_app_sink_pull_buffer):
15829 * gst-libs/gst/app/gstappsink.h:
15830 Add properties, signals and actions to access the element even without
15831 linking to the library.
15832 Fix some method names and signatures.
15834 2007-09-20 10:37:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15836 tests/check/generic/states.c: Improved state change unit test.
15837 Original commit message from CVS:
15838 * tests/check/generic/states.c:
15839 Improved state change unit test.
15841 2007-09-19 18:16:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15843 Ignore registries in any format.
15844 Original commit message from CVS:
15845 * docs/plugins/.cvsignore:
15846 * tests/check/.cvsignore:
15847 Ignore registries in any format.
15849 2007-09-19 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
15851 gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
15852 Original commit message from CVS:
15853 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15854 (gst_base_rtp_depayload_chain),
15855 (gst_base_rtp_depayload_set_gst_timestamp):
15856 Only copy timestamp on outgoing packets if the depayloader did not set
15858 Also copy duration on outgoing packets.
15860 2007-09-19 15:55:08 +0000 Wim Taymans <wim.taymans@gmail.com>
15862 gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
15863 Original commit message from CVS:
15864 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
15865 (gst_basertppayload_set_outcaps):
15866 Fix compilation because of missing %d in printf.
15867 When fixating caps, fixate what we can and throw away all remaining
15868 unfixed caps, subclasses should do something smart if they need to.
15870 2007-09-19 12:04:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15872 ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
15873 Original commit message from CVS:
15874 * ext/gnomevfs/gstgnomevfssrc.c:
15875 Improve debug logs a bit and be more verbose if things go wrong.
15877 2007-09-17 17:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15879 Fix a bunch of compile warnings shown with Forte.
15880 Original commit message from CVS:
15881 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
15882 (gst_text_overlay_set_property):
15883 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
15884 * gst-libs/gst/audio/gstbaseaudiosink.c:
15885 (gst_base_audio_sink_render):
15886 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
15887 (gst_rtcp_unix_to_ntp):
15888 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
15889 * gst/playback/gstqueue2.c:
15890 * tests/examples/seek/seek.c: (set_scale):
15891 Fix a bunch of compile warnings shown with Forte.
15892 * gst/audiorate/gstaudiorate.c:
15893 Always pull in config.h before including any system headers.
15895 2007-09-17 16:22:17 +0000 Wim Taymans <wim.taymans@gmail.com>
15897 gst/playback/gstqueue2.c: Also fix #476514 for queue2.
15898 Original commit message from CVS:
15899 * gst/playback/gstqueue2.c: (update_buffering),
15900 (gst_queue_locked_flush), (gst_queue_locked_enqueue),
15901 (gst_queue_handle_sink_event), (gst_queue_chain),
15902 (gst_queue_push_one), (gst_queue_sink_activate_push),
15903 (gst_queue_src_activate_push), (gst_queue_src_activate_pull):
15904 Also fix #476514 for queue2.
15906 2007-09-16 19:31:06 +0000 Wim Taymans <wim.taymans@gmail.com>
15908 gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
15909 Original commit message from CVS:
15910 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15911 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
15912 (gst_base_rtp_depayload_chain),
15913 (gst_base_rtp_depayload_handle_sink_event),
15914 (gst_base_rtp_depayload_push_full),
15915 (gst_base_rtp_depayload_set_gst_timestamp),
15916 (gst_base_rtp_depayload_change_state):
15917 Remove code to deal with RTP to GST time conversion, we now just copy
15918 the GST timestamp we receive to the outgoing buffers.
15919 Handle segment and flushes correctly.
15920 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
15921 When we have no valid input timestamp, use the previous rtp timestamp on
15922 the outgoing RTP packet instead of the RTP base time.
15924 2007-09-16 01:56:21 +0000 David Schleef <ds@schleef.org>
15926 ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
15927 Original commit message from CVS:
15928 * ext/alsa/gstalsa.c:
15929 * ext/alsa/gstalsadeviceprobe.c:
15930 * ext/alsa/gstalsamixer.c:
15931 * ext/alsa/gstalsasink.c:
15932 * ext/alsa/gstalsasrc.c:
15933 Change alsa alloca's to malloc to fix warnings on gcc-4.2.
15935 2007-09-15 18:41:27 +0000 Wim Taymans <wim.taymans@gmail.com>
15937 gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
15938 Original commit message from CVS:
15939 * gst-libs/gst/rtp/gstbasertppayload.c:
15940 (gst_basertppayload_set_outcaps), (gst_basertppayload_push):
15941 Add some debug info when negotiating caps.
15943 2007-09-15 00:29:11 +0000 Wim Taymans <wim.taymans@gmail.com>
15945 gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
15946 Original commit message from CVS:
15947 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
15948 A buffer with an empty payload is also a valid buffer.
15950 2007-09-14 20:52:00 +0000 Wim Taymans <wim.taymans@gmail.com>
15952 gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
15953 Original commit message from CVS:
15954 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
15955 (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
15956 (gst_basertppayload_change_state):
15957 Make sure we start our RTP timestamp from the random base RTP
15958 timestamp even if the buffer timestamp starts from some random value.
15960 2007-09-14 16:56:16 +0000 Wim Taymans <wim.taymans@gmail.com>
15962 Add simple exmple app to demonstrate starting and pausing live and non-live bins in a PLAYING pipeline.
15963 Original commit message from CVS:
15965 * tests/examples/Makefile.am:
15966 * tests/examples/dynamic/.cvsignore:
15967 * tests/examples/dynamic/Makefile.am:
15968 * tests/examples/dynamic/addstream.c: (create_stream),
15969 (pause_play_stream), (message_received), (eos_message_received),
15970 (perform_step), (main):
15971 Add simple exmple app to demonstrate starting and pausing live and
15972 non-live bins in a PLAYING pipeline.
15974 2007-09-14 10:42:00 +0000 Julien Moutte <julien@moutte.net>
15976 gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
15977 Original commit message from CVS:
15978 2007-09-14 Julien MOUTTE <julien@moutte.net>
15979 * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
15980 typefind for QCP files (RFC #3625)
15982 2007-09-13 22:52:09 +0000 Wim Taymans <wim.taymans@gmail.com>
15984 gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
15985 Original commit message from CVS:
15986 * gst-libs/gst/audio/gstbaseaudiosink.c:
15987 (gst_base_audio_sink_init):
15988 Disable pull mode scheduling, we're not ready for it yet and it subtly
15989 breaks a lot of things.
15991 2007-09-12 17:35:52 +0000 Tim-Philipp Müller <tim@centricular.net>
15993 tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
15994 Original commit message from CVS:
15995 * tests/check/elements/libvisual.c:
15996 Test all libvisual plugins, not just the first one; this reproduces
15997 bug #450336 quite easily. Looks like a problem with the 'jess'
16000 2007-09-12 17:15:12 +0000 Tim-Philipp Müller <tim@centricular.net>
16002 tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
16003 Original commit message from CVS:
16004 * tests/check/Makefile.am:
16005 * tests/check/elements/.cvsignore:
16006 * tests/check/elements/libvisual.c:
16007 Add basic libvisual test case in an attempt to reproduce bug #450336.
16008 Doesn't reproduce that bug, but some other crasher instead (invalid
16009 free), at least with make elements/libvisual.forever and the bumscope
16010 plugin on x86-64/gutsy. Leaving test disabled for now.
16012 2007-09-12 08:38:21 +0000 Peter Kjellerstedt <pkj@axis.com>
16014 gst/: Printf format fixes (#476128).
16015 Original commit message from CVS:
16016 Patch by: Peter Kjellerstedt <pkj at axis com>
16017 * gst-libs/gst/app/gstappsink.c:
16018 * gst/flv/gstflvdemux.c:
16019 * gst/flv/gstflvparse.c:
16020 * gst/interleave/deinterleave.c:
16021 * gst/switch/gstswitch.c:
16022 Printf format fixes (#476128).
16024 2007-09-11 19:07:57 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
16026 gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
16027 Original commit message from CVS:
16028 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
16029 * gst-libs/gst/rtsp/gstrtspconnection.c:
16030 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
16031 (read_body), (gst_rtsp_connection_receive):
16032 Make sure we can not cancel in the middle of receiving a message.
16035 2007-09-11 11:29:12 +0000 Josep Torra Valles <josep@fluendo.com>
16037 gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
16038 Original commit message from CVS:
16039 Patch by: Josep Torra Valles <josep@fluendo.com>
16040 * gst/playback/gstplaybasebin.c:
16041 Increase upper limit for audio queue a bit; fixes preroll problem
16042 with playbin and decodebin2 when playing a quicktime trailer with
16043 multichannel audio via http (#464666).
16045 2007-09-10 22:10:54 +0000 Wim Taymans <wim.taymans@gmail.com>
16047 gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
16048 Original commit message from CVS:
16049 * gst-libs/gst/audio/gstbaseaudiosrc.c:
16050 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
16051 (gst_base_audio_src_provide_clock),
16052 (gst_base_audio_src_set_property),
16053 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
16054 * gst-libs/gst/audio/gstbaseaudiosrc.h:
16055 Allow othe clocks than the internal clock to be used for the pipeline.
16056 Add property to disable clock provide.
16057 API: GstBaseAudioSrc::provide-clock
16059 2007-09-10 12:05:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16061 gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
16062 Original commit message from CVS:
16063 * gst/playback/gstdecodebin2.c:
16064 Don't leak request pads. Fixes #475395.
16066 2007-09-09 10:25:43 +0000 René Stadler <mail@renestadler.de>
16068 sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
16069 Original commit message from CVS:
16070 Patch by: René Stadler <mail at renestadler dot de>
16071 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
16072 (gst_ximage_buffer_class_init):
16073 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
16074 (gst_xvimage_buffer_class_init):
16075 Correctly chain up finalize with the parent class to prevent
16076 memory leaks. Fixes #474880.
16078 2007-09-09 04:08:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16080 Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
16081 Original commit message from CVS:
16082 * gst/volume/gstvolume.c: (volume_choose_func):
16083 * tests/check/elements/volume.c: (GST_START_TEST):
16084 Revert the latest change: floating point samples are allowed to
16085 have any value, not only values in the range [-1,1]. Thanks to Andy
16086 Wingo for noticing.
16087 Also fix processing of int32 samples with volumes > 4 by making the
16088 unity value smaller which prevents overflows.
16090 2007-09-07 17:37:03 +0000 Tim-Philipp Müller <tim@centricular.net>
16092 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
16093 Original commit message from CVS:
16094 * gst-libs/gst/rtp/gstrtpbuffer.c:
16095 * tests/check/libs/rtp.c:
16096 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
16098 2007-09-07 16:46:05 +0000 Haakon Sporsheim <haakon.sporsheim@tandberg.com>
16100 gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
16101 Original commit message from CVS:
16102 Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
16103 * gst-libs/gst/rtp/gstrtpbuffer.c:
16104 Fix up GstRTPHeader helper struct so that compilers will not under
16105 any circumstances add padding in between our fields, as currently
16106 happens with MSVC on win32, because that would lead to us sending
16107 out RTP payloads with broken RTP headers (#471194).
16108 Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
16109 * tests/check/Makefile.am:
16110 * tests/check/libs/.cvsignore:
16111 * tests/check/libs/rtp.c:
16112 Add some simple unit tests for GstRTPBuffer. Some are disabled
16113 because the code tested still needs fixing (set_csrc() does not work).
16115 2007-09-07 15:05:24 +0000 Christian Schaller <uraeus@gnome.org>
16117 * gst-plugins-base.spec.in:
16118 update spec file to include latest RTSP libraries and headers and more
16119 Original commit message from CVS:
16120 update spec file to include latest RTSP libraries and headers and more
16122 2007-09-07 12:41:01 +0000 Tim-Philipp Müller <tim@centricular.net>
16124 win32/: Add rtsp enumtypes (#474384) and update others.
16125 Original commit message from CVS:
16127 * win32/common/gstrtsp-enumtypes.c:
16128 * win32/common/gstrtsp-enumtypes.h:
16129 * win32/common/interfaces-enumtypes.c:
16130 * win32/common/interfaces-enumtypes.h:
16131 * win32/common/multichannel-enumtypes.c:
16132 Add rtsp enumtypes (#474384) and update others.
16134 2007-09-06 20:31:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16136 configure.ac: Fix configure check for HAVE_LIBXML_HTML.
16137 Original commit message from CVS:
16139 Fix configure check for HAVE_LIBXML_HTML.
16141 2007-09-06 12:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
16143 tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
16144 Original commit message from CVS:
16145 * tests/check/libs/.cvsignore:
16146 Ignore more, in case the build bots work again one day.
16148 2007-09-06 07:00:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16150 Add libgstfft, a FFT library based on Kiss FFT which is
16151 Original commit message from CVS:
16152 Reviewed by: Stefan Kost <ensonic@users.sf.net>
16154 * gst-libs/gst/Makefile.am:
16155 * gst-libs/gst/fft/Makefile.am:
16156 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
16157 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
16158 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
16159 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
16160 * gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
16161 * gst-libs/gst/fft/gstfft.h:
16162 * gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
16163 (gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
16164 (gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
16165 * gst-libs/gst/fft/gstfftf32.h:
16166 * gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
16167 (gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
16168 (gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
16169 * gst-libs/gst/fft/gstfftf64.h:
16170 * gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
16171 (gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
16172 (gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
16173 * gst-libs/gst/fft/gstffts16.h:
16174 * gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
16175 (gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
16176 (gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
16177 * gst-libs/gst/fft/gstffts32.h:
16178 * gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
16179 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16180 (kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
16181 (kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
16182 * gst-libs/gst/fft/kiss_fft_f32.h:
16183 * gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
16184 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16185 (kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
16186 (kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
16187 * gst-libs/gst/fft/kiss_fft_f64.h:
16188 * gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
16189 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16190 (kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
16191 (kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
16192 * gst-libs/gst/fft/kiss_fft_s16.h:
16193 * gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
16194 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16195 (kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
16196 (kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
16197 * gst-libs/gst/fft/kiss_fft_s32.h:
16198 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
16199 (kiss_fftr_f32), (kiss_fftri_f32):
16200 * gst-libs/gst/fft/kiss_fftr_f32.h:
16201 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
16202 (kiss_fftr_f64), (kiss_fftri_f64):
16203 * gst-libs/gst/fft/kiss_fftr_f64.h:
16204 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
16205 (kiss_fftr_s16), (kiss_fftri_s16):
16206 * gst-libs/gst/fft/kiss_fftr_s16.h:
16207 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
16208 (kiss_fftr_s32), (kiss_fftri_s32):
16209 * gst-libs/gst/fft/kiss_fftr_s32.h:
16210 * gst-libs/gst/fft/kiss_version:
16211 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
16212 * pkgconfig/gstreamer-plugins-base.pc.in:
16213 Add libgstfft, a FFT library based on Kiss FFT which is
16214 BSD licensed. Supported sample formats are int16, int32,
16215 float and double. For those formats a real FFT and IFFT
16216 can be done, different windowing functions can be applied
16217 and functions for extracting the magnitude and phase exist.
16219 * docs/libs/Makefile.am:
16220 * docs/libs/gst-plugins-base-libs-docs.sgml:
16221 * docs/libs/gst-plugins-base-libs-sections.txt:
16222 Integrate libgstfft into the docs.
16223 * tests/check/Makefile.am:
16224 * tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
16225 Add unit tests for libgstfft, currently only testing the FFT.
16226 Unit tests for IFFT will follow soon.
16228 2007-09-05 23:07:40 +0000 Peter Kjellerstedt <pkj@axis.com>
16230 gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
16231 Original commit message from CVS:
16232 Patch by: Peter Kjellerstedt <pkj at axis com>
16233 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
16234 (gst_sdp_message_init), (gst_sdp_message_uninit),
16235 (is_multicast_address), (gst_sdp_message_as_text),
16236 (gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
16237 (gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
16238 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
16239 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
16240 (gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
16241 (gst_sdp_media_init), (gst_sdp_media_uninit),
16242 (gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
16243 (gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
16244 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
16245 (gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
16246 (gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
16247 * gst-libs/gst/sdp/gstsdpmessage.h:
16248 Separate INIT_ARRAY() and related macros into two versions, one for
16249 structures and one for pointers (e.g., INIT_ARRAY() and
16250 INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
16251 lists of emails and phone numbers.
16252 Add missing const as appropriate.
16253 Change all gint to guint since they all actually represent unsigned
16255 Do not use time as a variable name as it shadows the global time().
16256 Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
16257 Actually implement gst_sdp_message_add_time().
16258 Make gst_sdp_message_add_time() take repeat times as an argument.
16259 Store repeat times in GstSDPTime as a GArray rather than as gchar**.
16260 Corrected the definition of gst_sdp_media_get_bandwidth() (was
16261 misspelled as badwidth).
16262 gst-indented and a little clean up. Fixes #471067.
16264 2007-09-05 21:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16266 gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
16267 Original commit message from CVS:
16268 * gst/volume/gstvolume.c: (volume_choose_func),
16269 (volume_process_double), (volume_process_double_clamp),
16270 (volume_process_float_clamp):
16271 Correctly clamp float/double samples in the [-1.0,1.0] range to
16272 prevent weird effects.
16273 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
16274 Add unit tests for all samples types that had none before.
16276 2007-09-05 14:09:15 +0000 Tim-Philipp Müller <tim@centricular.net>
16278 gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
16279 Original commit message from CVS:
16280 * gst-libs/gst/rtp/gstrtpbuffer.c:
16281 Need to include stdlib.h for abs() here too.
16283 2007-09-05 14:01:25 +0000 Tim-Philipp Müller <tim@centricular.net>
16285 gst/playback/gststreaminfo.c: Fix build.
16286 Original commit message from CVS:
16287 * gst/playback/gststreaminfo.c:
16290 2007-09-05 10:32:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16292 gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
16293 Original commit message from CVS:
16294 * gst/playback/gststreaminfo.c:
16295 Clean up some half-disabled code and comment.
16297 2007-09-04 16:18:48 +0000 Wim Taymans <wim.taymans@gmail.com>
16299 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
16300 Original commit message from CVS:
16301 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
16302 (gst_base_rtp_payload_audio_handle_event):
16303 Return FALSE from the event handler to let the parent class handle the
16305 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16306 (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
16307 Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
16308 * gst-libs/gst/rtp/gstbasertppayload.c:
16309 Bump the MTU to 1400.
16311 2007-09-04 01:50:55 +0000 Johan Dahlin <johan@gnome.org>
16313 gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
16314 Original commit message from CVS:
16315 2007-09-03 Johan Dahlin <jdahlin@async.com.br>
16316 * gst/typefind/gsttypefindfunctions.c (plugin_init):
16317 Add an audio/x-nsf typefind function for the nsfdec element.
16319 2007-09-03 20:46:38 +0000 Renato Filho <renato.filho@indt.org.br>
16321 gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
16322 Original commit message from CVS:
16323 * gst/playback/gstplaybasebin.c:
16324 Included "myth://" on stream_uris list for enable buffering to mythtv files
16326 2007-09-03 19:31:11 +0000 Wim Taymans <wim.taymans@gmail.com>
16328 Fix parsing of RB blocks.
16329 Original commit message from CVS:
16330 * docs/libs/gst-plugins-base-libs-sections.txt:
16331 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
16332 (gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
16333 (gst_rtcp_unix_to_ntp):
16334 * gst-libs/gst/rtp/gstrtcpbuffer.h:
16335 Fix parsing of RB blocks.
16337 Added helper functions to convert to/from UNIX and NTP time.
16338 API: gst_rtcp_ntp_to_unix()
16339 API: gst_rtcp_unix_to_ntp()
16340 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
16341 (gst_rtp_buffer_get_header_len),
16342 (gst_rtp_buffer_get_extension_data),
16343 (gst_rtp_buffer_get_payload_subbuffer),
16344 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
16345 (gst_rtp_buffer_ext_timestamp):
16346 * gst-libs/gst/rtp/gstrtpbuffer.h:
16347 Fix some more docs.
16348 Implement handling of packets with extensions.
16349 Fix padding check in _validate().
16350 Added function to get extension data.
16351 API: gst_rtp_buffer_get_header_len()
16352 API: gst_rtp_buffer_get_extension_data()
16354 2007-09-03 19:19:35 +0000 Wim Taymans <wim.taymans@gmail.com>
16356 gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
16357 Original commit message from CVS:
16358 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16359 (gst_base_rtp_depayload_class_init),
16360 (gst_base_rtp_depayload_set_gst_timestamp):
16361 Add some more docs for the queue-delay property and fix a typo in a
16363 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
16366 2007-09-03 19:17:33 +0000 Wim Taymans <wim.taymans@gmail.com>
16368 gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
16369 Original commit message from CVS:
16370 * gst-libs/gst/audio/gstbaseaudiosink.c:
16371 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
16372 (gst_base_audio_sink_change_state):
16373 When skew slaving, try to hover around the middle of a segment so that
16374 we at most drift by half a segment.
16375 If we are aligning in the oposite direction of the clock skew, we don't
16378 2007-08-31 21:07:20 +0000 Wim Taymans <wim.taymans@gmail.com>
16380 gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
16381 Original commit message from CVS:
16382 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16383 (gst_base_rtp_depayload_setcaps),
16384 (gst_base_rtp_depayload_set_gst_timestamp):
16385 Be less silly with the segment start, just apply the clock-base to the
16388 2007-08-31 15:58:30 +0000 Wim Taymans <wim.taymans@gmail.com>
16390 gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
16391 Original commit message from CVS:
16392 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16393 (gst_base_rtp_depayload_class_init),
16394 (gst_base_rtp_depayload_finalize),
16395 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
16396 (gst_base_rtp_depayload_handle_sink_event),
16397 (gst_base_rtp_depayload_set_gst_timestamp),
16398 (gst_base_rtp_depayload_change_state):
16399 * gst-libs/gst/rtp/gstbasertpdepayload.h:
16400 Deprecate the queue handling thread thing and remove the code.
16401 Use new method to calculate the extended timestamp.
16403 2007-08-31 15:21:13 +0000 Wim Taymans <wim.taymans@gmail.com>
16405 gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
16406 Original commit message from CVS:
16407 * gst-libs/gst/rtp/gstrtcpbuffer.c:
16408 (gst_rtcp_packet_sdes_copy_entry):
16409 Use g_strndup which does exactly what we want.
16410 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
16411 (gst_rtp_buffer_ext_timestamp):
16412 * gst-libs/gst/rtp/gstrtpbuffer.h:
16413 Add helper function to compare seqnums.
16414 Add helper function to calculate extended timestamps.
16415 API: gst_rtp_buffer_compare_seqnum()
16416 API: gst_rtp_buffer_ext_timestamp()
16418 2007-08-30 21:59:23 +0000 Wim Taymans <wim.taymans@gmail.com>
16420 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
16421 Original commit message from CVS:
16422 * gst-libs/gst/rtp/gstrtcpbuffer.c:
16423 (gst_rtcp_packet_sdes_get_entry),
16424 (gst_rtcp_packet_sdes_copy_entry):
16425 * gst-libs/gst/rtp/gstrtcpbuffer.h:
16426 Fix and document SDES item data function.
16427 Add new function that makes a proper copy of SDES item data.
16428 API: gst_rtcp_packet_sdes_copy_entry()
16430 2007-08-30 07:29:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16432 The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
16433 Original commit message from CVS:
16436 The tcp and subparse plugins are under gst, but not totaly free of
16437 dependencies. Handle selection inconfigure.ac, so that they show up
16438 on the final list of what is build and what is not. Maybe they should
16439 better be moved to ext.
16441 2007-08-30 06:58:46 +0000 Daniel Díaz <yosoy@danieldiaz.org>
16443 Check if libxml provides HTML parser which subparse needs.
16444 Original commit message from CVS:
16445 Patch by: Daniel Díaz <yosoy@danieldiaz.org>
16448 Check if libxml provides HTML parser which subparse needs.
16451 2007-08-29 14:22:04 +0000 Tim-Philipp Müller <tim@centricular.net>
16453 ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
16454 Original commit message from CVS:
16455 * ext/alsa/gstalsa.c:
16456 Fix typo and compilation on big endian systems.
16458 2007-08-29 12:16:46 +0000 Tim-Philipp Müller <tim@centricular.net>
16460 gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
16461 Original commit message from CVS:
16462 * gst/subparse/gstssaparse.c:
16463 Convert SSA newline codes into actual newline characters (#470766).
16465 2007-08-28 14:58:17 +0000 Tim-Philipp Müller <tim@centricular.net>
16467 API: also add gst_install_plugins_supported() while we're at it (see #470456).
16468 Original commit message from CVS:
16469 * docs/libs/gst-plugins-base-libs-sections.txt:
16470 * gst-libs/gst/pbutils/install-plugins.c:
16471 * gst-libs/gst/pbutils/install-plugins.h:
16472 * tests/check/libs/pbutils.c:
16473 API: also add gst_install_plugins_supported() while we're at it
16476 2007-08-28 14:23:55 +0000 Tim-Philipp Müller <tim@centricular.net>
16478 API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
16479 Original commit message from CVS:
16480 * docs/libs/gst-plugins-base-libs-sections.txt:
16481 * gst-libs/gst/pbutils/missing-plugins.c:
16482 * gst-libs/gst/pbutils/missing-plugins.h:
16483 * tests/check/libs/pbutils.c:
16484 API: add gst_missing_*_installer_detail_new() convenience API so
16485 that applications that know exactly what they're missing can request
16486 installer detail strings for those items directly instead of having
16487 to first create a dummy missing-plugin message and then get the
16488 installer detail string from that. Fixes #470456.
16490 2007-08-27 11:59:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16492 gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
16493 Original commit message from CVS:
16494 * gst/playback/gstdecodebin.c: (close_pad_link):
16495 We need to set up delayed-linking whenever the caps are non-fixed,
16496 not just when there are multiple types - use gst_pad_is_fixed()
16499 2007-08-26 14:14:33 +0000 Tim-Philipp Müller <tim@centricular.net>
16501 gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
16502 Original commit message from CVS:
16503 * gst-libs/gst/pbutils/missing-plugins.c:
16504 (gst_missing_plugin_message_get_installer_detail):
16505 Add missing separator in PID fallback case.
16507 2007-08-24 15:28:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16509 ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
16510 Original commit message from CVS:
16511 * ext/alsa/Makefile.am:
16512 There is no GST_PLUGINS_BASE_LIBS defined.
16513 * ext/alsa/gstalsa.c:
16514 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
16515 * ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
16516 Add support for ALSA 24-bit formats.
16517 snd_pcm_delay can return an error code, especially
16518 during XRUNS. In that case, the best we can do is assume
16520 * gst/audioconvert/Makefile.am:
16521 Add flags from -base before any more-remote dependencies.
16523 2007-08-23 20:45:45 +0000 Davyd <davyd@madeley.id.au>
16525 gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
16526 Original commit message from CVS:
16527 Based on a patch by: Davyd <davyd at madeley dot id dot au>
16528 * gst/volume/gstvolume.c: (volume_choose_func),
16529 (volume_update_real_volume), (gst_volume_set_volume),
16530 (gst_volume_init), (volume_process_int32),
16531 (volume_process_int32_clamp), (volume_process_int24),
16532 (volume_process_int24_clamp), (volume_process_int16),
16533 (volume_process_int16_clamp), (volume_process_int8),
16534 (volume_process_int8_clamp), (volume_update_volume), (plugin_init):
16535 * gst/volume/gstvolume.h:
16536 Add support for int32, int24 and int8 to the volume element.
16539 2007-08-23 12:37:42 +0000 Tim-Philipp Müller <tim@centricular.net>
16541 tests/examples/Makefile.am: Fix even more.
16542 Original commit message from CVS:
16543 * tests/examples/Makefile.am:
16546 2007-08-23 10:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16548 Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
16549 Original commit message from CVS:
16551 * docs/libs/Makefile.am:
16552 * docs/libs/gst-plugins-base-libs-docs.sgml:
16553 * docs/libs/gst-plugins-base-libs-sections.txt:
16554 * ext/gnomevfs/gstgnomevfssrc.c:
16555 * ext/gnomevfs/gstgnomevfssrc.h:
16556 * gst-libs/gst/Makefile.am:
16557 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
16558 * pkgconfig/gstreamer-plugins-base.pc.in:
16559 * sys/v4l/v4lsrc_calls.c:
16560 * tests/examples/Makefile.am:
16561 * win32/common/config.h:
16562 Revert unwanted commit. many thanks to moap. I want a fix for
16563 https://thomas.apestaart.org/moap/trac/ticket/239
16565 2007-08-23 08:33:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16567 Original commit message from CVS:
16568 reviewed by: <delete if not using a buddy>
16569 patch by: <delete if not someone else's patch>
16571 * docs/libs/Makefile.am:
16572 * docs/libs/gst-plugins-base-libs-docs.sgml:
16573 * docs/libs/gst-plugins-base-libs-sections.txt:
16574 * ext/gnomevfs/gstgnomevfssrc.c:
16575 * ext/gnomevfs/gstgnomevfssrc.h:
16576 * gst-libs/gst/Makefile.am:
16577 * gst-libs/gst/audio/gstaudiofilter.h:
16578 * gst/typefind/gsttypefindfunctions.c:
16579 * gst/volume/gstvolume.c:
16580 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
16581 * pkgconfig/gstreamer-plugins-base.pc.in:
16582 * sys/v4l/v4lsrc_calls.c:
16583 * tests/examples/Makefile.am:
16584 * win32/common/config.h:
16586 2007-08-22 15:29:04 +0000 Wim Taymans <wim.taymans@gmail.com>
16588 gst-libs/gst/audio/audio.c: Clarify the docs a little.
16589 Original commit message from CVS:
16590 * gst-libs/gst/audio/audio.c:
16591 Clarify the docs a little.
16593 2007-08-22 11:20:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16595 gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
16596 Original commit message from CVS:
16597 * gst/volume/gstvolume.c:
16598 Enable liboil for float and add more details about problems with
16601 2007-08-21 15:43:24 +0000 Wim Taymans <wim.taymans@gmail.com>
16603 sys/v4l/gstv4lsrc.c: Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
16604 Original commit message from CVS:
16605 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
16606 Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
16608 2007-08-21 12:08:43 +0000 Wim Taymans <wim.taymans@gmail.com>
16610 ext/vorbis/vorbisdec.c: When calculating the first timestamp of the buffers, don't go below 0 and clip the samples be...
16611 Original commit message from CVS:
16612 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
16613 When calculating the first timestamp of the buffers, don't go below 0
16614 and clip the samples because the offset was on the eos page.
16617 2007-08-21 11:42:39 +0000 Wim Taymans <wim.taymans@gmail.com>
16619 ext/ogg/gstoggdemux.c: Also submit the eos page when trying to find the first timestamp.
16620 Original commit message from CVS:
16621 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
16622 (gst_ogg_demux_collect_chain_info):
16623 Also submit the eos page when trying to find the first timestamp.
16626 2007-08-17 15:24:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16628 gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
16629 Original commit message from CVS:
16630 * gst-libs/gst/audio/audio.h:
16631 Use gst_util_uint64_scale() instead of doing the math
16632 with double for GST_FRAMES_TO_CLOCK_TIME() and
16633 GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
16634 prevents rounding errors. Fixes #467667.
16636 2007-08-17 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
16638 gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
16639 Original commit message from CVS:
16640 * gst-libs/gst/rtsp/gstrtspconnection.c:
16641 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
16642 (gst_rtsp_connection_read), (gst_rtsp_connection_poll):
16643 * gst-libs/gst/rtsp/gstrtspconnection.h:
16645 On shutdown, don't read the control socket yet.
16646 Set timeout value correctly in all cases.
16647 Add function to check if the server accepts reads or writes.
16648 API: gst_rtsp_connection_poll()
16649 * gst-libs/gst/rtsp/gstrtspdefs.h:
16650 Fix compilation with -pedantic.
16651 Add enum for _poll.
16653 2007-08-16 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
16655 gst-libs/gst/app/gstappsink.c: Override the preroll vmethod instead of overriding the render method twice.
16656 Original commit message from CVS:
16657 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
16658 Override the preroll vmethod instead of overriding the render method
16661 2007-08-16 16:06:21 +0000 Olivier Crete <tester@tester.ca>
16663 gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
16664 Original commit message from CVS:
16665 Patch by: Olivier Crete <tester at tester ca>
16666 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
16667 (gst_basertppayload_getcaps):
16668 * gst-libs/gst/rtp/gstbasertppayload.h:
16669 Add getcaps vfunc to basertppayload. See #465146.
16671 2007-08-16 11:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
16673 gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
16674 Original commit message from CVS:
16675 * gst/playback/gstplaybasebin.c: (queue_threshold_reached):
16676 Only post buffering messages when we are a stream.
16678 2007-08-15 17:05:45 +0000 Tim-Philipp Müller <tim@centricular.net>
16680 gst-libs/gst/pbutils/: Small docs fix and addition.
16681 Original commit message from CVS:
16682 * gst-libs/gst/pbutils/install-plugins.c:
16683 * gst-libs/gst/pbutils/missing-plugins.c:
16684 Small docs fix and addition.
16686 2007-08-14 17:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
16688 gst-libs/gst/app/gstappsink.c: Don't use new API.
16689 Original commit message from CVS:
16690 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
16693 2007-08-14 17:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
16695 gst-libs/gst/app/gstappsink.*: Make love to appsink.
16696 Original commit message from CVS:
16697 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
16698 (gst_app_sink_class_init), (gst_app_sink_dispose),
16699 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
16700 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
16701 (gst_app_sink_render), (gst_app_sink_get_caps),
16702 (gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
16703 (gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
16704 * gst-libs/gst/app/gstappsink.h:
16705 Make love to appsink.
16706 Make it support pulling of the preroll buffer.
16707 Add docs and debug statements.
16708 Fix some races wrt to EOS handling and stopping.
16710 Implement FLUSHING.
16711 API: gst_app_sink_pull_preroll()
16713 2007-08-13 15:37:29 +0000 Tim-Philipp Müller <tim@centricular.net>
16715 tests/icles/: Add a dumb little test for textoverlay alignments.
16716 Original commit message from CVS:
16717 * tests/icles/.cvsignore:
16718 * tests/icles/Makefile.am:
16719 * tests/icles/test-textoverlay.c:
16720 Add a dumb little test for textoverlay alignments.
16722 2007-08-13 15:26:54 +0000 Dan Williams <dcbw@redhat.com>
16724 ext/pango/gsttextoverlay.*: API: add "line-alignment" property (#459334). Add gtk-doc blurb for "silent" property so ...
16725 Original commit message from CVS:
16726 Patch by: Dan Williams <dcbw redhat com>
16727 * ext/pango/gsttextoverlay.c:
16728 * ext/pango/gsttextoverlay.h:
16729 API: add "line-alignment" property (#459334). Add gtk-doc blurb for
16730 "silent" property so there's a Since tag in the API reference.
16732 2007-08-13 11:21:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
16736 Original commit message from CVS:
16739 2007-08-12 16:30:36 +0000 Wim Taymans <wim.taymans@gmail.com>
16741 gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
16742 Original commit message from CVS:
16743 * gst-libs/gst/rtp/gstbasertppayload.c:
16744 (gst_basertppayload_set_outcaps):
16745 * gst-libs/gst/rtp/gstbasertppayload.h:
16746 Improve caps negotiation so that downstream elements can confiure
16747 certain RTP properties by fixing them on the caps. See #465146.
16750 2007-08-11 12:39:51 +0000 Tim-Philipp Müller <tim@centricular.net>
16752 Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
16753 Original commit message from CVS:
16754 * docs/libs/gst-plugins-base-libs-sections.txt:
16755 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16756 * gst-libs/gst/rtp/gstbasertpdepayload.h:
16757 Mark as deprecated some macros which were presumably meant to be
16758 private API and accidentally exposed in the public header file.
16759 Also actually _init() lock (only works at the moment because the
16760 struct is zeroed out when created and the initial values in the
16761 mutex struct are zeroes too). (#459585)
16763 2007-08-10 17:35:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16765 docs/libs/Makefile.am: Remove cruft and do some cleanups.
16766 Original commit message from CVS:
16767 * docs/libs/Makefile.am:
16768 Remove cruft and do some cleanups.
16769 * docs/libs/gst-plugins-base-libs-docs.sgml:
16770 Prepare for comming gtkdoc features (rebase against online docs).
16772 2007-08-10 13:55:44 +0000 Michael Smith <msmith@xiph.org>
16774 gst/audiorate/gstaudiorate.c: Debug output fixes.
16775 Original commit message from CVS:
16776 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
16777 Debug output fixes.
16778 * tests/check/elements/audiorate.c: (do_perfect_stream_test),
16780 Change the number of buffers used; 500 is too many and leads to
16783 2007-08-10 10:08:05 +0000 Tim-Philipp Müller <tim@centricular.net>
16785 gst/: Printf format fixes (#465028).
16786 Original commit message from CVS:
16787 * gst/playback/gstqueue2.c:
16788 * gst/videorate/gstvideorate.c:
16789 Printf format fixes (#465028).
16791 2007-08-09 15:44:02 +0000 Michael Smith <msmith@xiph.org>
16793 gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
16794 Original commit message from CVS:
16795 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
16796 If we have a large (> 1 second) discontinuity, push a series of
16797 smaller buffers rather than a single very large buffer. Avoids
16798 unreasonably large single buffer allocations when encountering a
16800 * tests/check/elements/audiorate.c: (GST_START_TEST),
16802 Add a test for this.
16804 2007-08-09 12:06:43 +0000 Josep Torra Valles <josep@fluendo.com>
16806 gst/playback/gstplaybasebin.c: Fixes: #465015
16807 Original commit message from CVS:
16808 * gst/playback/gstplaybasebin.c: (group_commit),
16809 (queue_remove_probe), (queue_threshold_reached):
16810 Patch by: Josep Torra Valles <josep@fluendo.com>
16812 Make sure we remove the check_queues buffer probe from the
16813 correct queue to avoid racily going back to "buffering 99%" when
16814 buffering is actually complete.
16815 Also, fix the spelling of Josep's surname in the ChangeLog.
16817 2007-08-09 11:37:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16819 ext/ogg/gstoggmux.c: Do not leak oggmux instance.
16820 Original commit message from CVS:
16821 * ext/ogg/gstoggmux.c:
16822 Do not leak oggmux instance.
16823 * ext/vorbis/vorbisenc.c:
16826 2007-08-09 10:51:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
16828 po/: Updated translations.
16829 Original commit message from CVS:
16835 Updated translations.
16837 2007-08-08 16:07:21 +0000 Yang Hong <hongyang@redflag-linux.com>
16839 ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
16840 Original commit message from CVS:
16841 patch by: Yang Hong <hongyang@redflag-linux.com>
16842 * ext/pango/gsttextoverlay.c:
16843 * ext/pango/gsttextoverlay.h:
16844 Add 'silent' property to GstTimeOverlay. Fixes #462979
16846 2007-08-08 15:05:22 +0000 Josep Torre Valles <josep@fluendo.com>
16848 Add connection-speed property. Fixes #464690.
16849 Original commit message from CVS:
16850 Patch by: Josep Torre Valles <josep@fluendo.com>
16851 * docs/plugins/gst-plugins-base-plugins.args:
16852 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
16853 (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
16854 (gst_uri_decode_bin_get_property), (gen_source_element):
16855 Add connection-speed property. Fixes #464690.
16857 2007-08-07 15:13:46 +0000 Damien Lespiau <damien.lespiau@gmail.com>
16859 Fix compilation on windows. Fixes #464320.
16860 Original commit message from CVS:
16861 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
16863 * gst-libs/gst/rtsp/Makefile.am:
16864 * gst-libs/gst/rtsp/gstrtspconnection.c:
16865 (gst_rtsp_connection_connect):
16866 Fix compilation on windows. Fixes #464320.
16868 2007-08-07 14:14:54 +0000 Josep Torre Valles <josep@fluendo.com>
16870 gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
16871 Original commit message from CVS:
16872 Patch by: Josep Torre Valles <josep@fluendo.com>
16873 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
16874 (gst_play_base_bin_init), (queue_threshold_reached),
16875 (gen_source_element), (setup_substreams),
16876 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
16877 (gst_play_base_bin_get_streaminfo_value_array):
16878 * gst/playback/gstplaybasebin.h:
16879 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
16880 (gst_play_bin_set_property), (gst_play_bin_get_property),
16881 (gst_play_bin_handle_redirect_message):
16882 Move connection-speed property from playbin to playbasebin so that we
16883 can also configure it in source elements that have the connection-speed
16884 property. Fixes #464028.
16885 Add some debug info here and there.
16887 2007-08-06 16:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16889 gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
16890 Original commit message from CVS:
16891 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
16892 Properly respond to conversion queries. Fixes #464079.
16894 2007-08-03 19:53:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16896 gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
16897 Original commit message from CVS:
16898 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
16899 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
16900 (gst_audio_test_src_init_sine_table),
16901 (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
16902 * gst/audiotestsrc/gstaudiotestsrc.h:
16903 Add float/double and int32 support to audiotestsrc. Fixes #460422.
16904 Also set the default volume to the default value specified in the
16907 2007-08-03 19:40:14 +0000 Jens Granseuer <jensgr@gmx.net>
16909 gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
16910 Original commit message from CVS:
16911 Patch by: Jens Granseuer <jensgr at gmx dot net>
16912 * gst/audioconvert/gstaudioquantize.c:
16913 Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
16915 2007-08-03 15:44:01 +0000 Wim Taymans <wim.taymans@gmail.com>
16917 gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
16918 Original commit message from CVS:
16919 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
16920 Add rdt manager for rdt transport.
16921 Fix parsing of RDT transport.
16923 2007-08-03 14:43:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16925 configure.ac: Back to CVS
16926 Original commit message from CVS:
16930 === release 0.10.14 ===
16932 2007-08-03 14:41:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16938 * docs/plugins/gst-plugins-base-plugins.args:
16939 * docs/plugins/inspect/plugin-adder.xml:
16940 * docs/plugins/inspect/plugin-alsa.xml:
16941 * docs/plugins/inspect/plugin-audioconvert.xml:
16942 * docs/plugins/inspect/plugin-audiorate.xml:
16943 * docs/plugins/inspect/plugin-audioresample.xml:
16944 * docs/plugins/inspect/plugin-audiotestsrc.xml:
16945 * docs/plugins/inspect/plugin-cdparanoia.xml:
16946 * docs/plugins/inspect/plugin-decodebin.xml:
16947 * docs/plugins/inspect/plugin-decodebin2.xml:
16948 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
16949 * docs/plugins/inspect/plugin-gdp.xml:
16950 * docs/plugins/inspect/plugin-gnomevfs.xml:
16951 * docs/plugins/inspect/plugin-libvisual.xml:
16952 * docs/plugins/inspect/plugin-ogg.xml:
16953 * docs/plugins/inspect/plugin-pango.xml:
16954 * docs/plugins/inspect/plugin-playbin.xml:
16955 * docs/plugins/inspect/plugin-subparse.xml:
16956 * docs/plugins/inspect/plugin-tcp.xml:
16957 * docs/plugins/inspect/plugin-theora.xml:
16958 * docs/plugins/inspect/plugin-typefindfunctions.xml:
16959 * docs/plugins/inspect/plugin-video4linux.xml:
16960 * docs/plugins/inspect/plugin-videorate.xml:
16961 * docs/plugins/inspect/plugin-videoscale.xml:
16962 * docs/plugins/inspect/plugin-videotestsrc.xml:
16963 * docs/plugins/inspect/plugin-volume.xml:
16964 * docs/plugins/inspect/plugin-vorbis.xml:
16965 * docs/plugins/inspect/plugin-ximagesink.xml:
16966 * docs/plugins/inspect/plugin-xvimagesink.xml:
16967 * gst-plugins-base.doap:
16968 * win32/common/config.h:
16970 Original commit message from CVS:
16973 2007-08-03 14:24:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16991 Original commit message from CVS:
16994 2007-07-27 17:37:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16996 tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
16997 Original commit message from CVS:
16998 * tests/check/libs/audio.c: (GST_START_TEST):
16999 Fix the test to reflect the behaviour of gst_audio_clip_buffer.
17001 2007-07-27 17:10:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17003 gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
17004 Original commit message from CVS:
17005 * gst-libs/gst/audio/audio.c:
17006 When clipping a buffer with no timestamp, assume it is
17007 within the segment without warnings.
17010 2007-07-27 11:16:23 +0000 Wim Taymans <wim.taymans@gmail.com>
17012 gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
17013 Original commit message from CVS:
17014 * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
17015 Fire the signal on the object, not the interface.
17017 2007-07-27 09:17:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17019 gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
17020 Original commit message from CVS:
17021 * gst-libs/gst/rtsp/.cvsignore:
17022 Ber. Don't include the full path, idiot.
17024 2007-07-27 08:29:29 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17026 gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
17027 Original commit message from CVS:
17028 * gst-libs/gst/rtsp/.cvsignore:
17029 Ignore generated files.
17031 2007-07-26 19:57:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17033 gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
17034 Original commit message from CVS:
17035 * gst-libs/gst/interfaces/Makefile.am:
17036 * gst-libs/gst/interfaces/interfaces-marshal.list:
17037 * gst-libs/gst/interfaces/rtspextension.c:
17038 * gst-libs/gst/interfaces/rtspextension.h:
17039 * gst-libs/gst/rtsp/Makefile.am:
17040 * gst-libs/gst/rtsp/gstrtsp.h:
17041 * gst-libs/gst/rtsp/gstrtspextension.c:
17042 (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
17043 (gst_rtsp_extension_detect_server),
17044 (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
17045 (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
17046 (gst_rtsp_extension_configure_stream),
17047 (gst_rtsp_extension_get_transports),
17048 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
17049 * gst-libs/gst/rtsp/gstrtspextension.h:
17050 * gst-libs/gst/rtsp/rtsp-marshal.list:
17051 Move the rtspextension.h interface into gstrtspextension.h
17052 as part of libgstrtsp instead of libgstinterfaces, because it's
17053 only for use within plugins, not applications.
17054 Add stuff to do the enum & marshal generation needed in libgstrtsp now.
17055 Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
17056 signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
17059 2007-07-26 15:48:01 +0000 Wim Taymans <wim.taymans@gmail.com>
17061 gst-libs/gst/interfaces/: Fix marshaller for the send signal.
17062 Original commit message from CVS:
17063 * gst-libs/gst/interfaces/Makefile.am:
17064 * gst-libs/gst/interfaces/interfaces-marshal.list:
17065 * gst-libs/gst/interfaces/rtspextension.c:
17066 (gst_rtsp_extension_iface_init),
17067 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
17068 * gst-libs/gst/interfaces/rtspextension.h:
17069 Fix marshaller for the send signal.
17070 Add URL to stream selection interface method.
17072 2007-07-26 15:35:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17074 gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
17075 Original commit message from CVS:
17076 * gst-libs/gst/riff/Makefile.am:
17077 Pull in our dependencies from -base before those from outside.
17079 2007-07-26 14:33:01 +0000 Wim Taymans <wim.taymans@gmail.com>
17081 API: gst_rtsp_base64_decode_ip()
17082 Original commit message from CVS:
17083 * docs/libs/gst-plugins-base-libs-sections.txt:
17084 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
17085 * gst-libs/gst/rtsp/gstrtspbase64.h:
17086 API: gst_rtsp_base64_decode_ip()
17087 Added function to decode Base64 in-place.
17089 2007-07-26 14:08:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17091 tests/check/libs/.cvsignore: Ignore the mixer test binary.
17092 Original commit message from CVS:
17093 * tests/check/libs/.cvsignore:
17094 Ignore the mixer test binary.
17096 2007-07-26 10:00:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17098 ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
17099 Original commit message from CVS:
17100 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
17101 Gratuitous comment change to trigger a rebuild on the buildbots.
17103 2007-07-25 18:20:36 +0000 Wim Taymans <wim.taymans@gmail.com>
17105 gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
17106 Original commit message from CVS:
17107 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
17108 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
17109 (gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
17110 (gst_sdp_media_get_format), (gst_sdp_media_get_information),
17111 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
17112 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
17113 (gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
17114 (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
17115 (gst_sdp_media_get_attribute_val):
17116 * gst-libs/gst/sdp/gstsdpmessage.h:
17117 Constify args where we can.
17119 2007-07-25 18:18:49 +0000 Wim Taymans <wim.taymans@gmail.com>
17121 gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
17122 Original commit message from CVS:
17123 * gst-libs/gst/interfaces/Makefile.am:
17124 * gst-libs/gst/interfaces/rtspextension.c:
17125 (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
17126 (gst_rtsp_extension_detect_server),
17127 (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
17128 (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
17129 (gst_rtsp_extension_configure_stream),
17130 (gst_rtsp_extension_get_transports),
17131 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
17132 * gst-libs/gst/interfaces/rtspextension.h:
17133 Move interface for RTSP extensions from -good to here.
17134 Added helper methods to invoke interface methods.
17136 2007-07-25 11:22:30 +0000 Wim Taymans <wim.taymans@gmail.com>
17138 Fix some more RTSP docs.
17139 Original commit message from CVS:
17140 * docs/libs/gst-plugins-base-libs-sections.txt:
17141 * gst-libs/gst/rtsp/gstrtspdefs.h:
17142 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
17143 (gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
17144 (gst_rtsp_message_init_response),
17145 (gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
17146 (gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
17147 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
17148 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
17149 (gst_rtsp_message_get_body), (dump_key_value):
17150 * gst-libs/gst/rtsp/gstrtspmessage.h:
17151 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
17152 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
17153 (gst_rtsp_range_parse):
17154 * gst-libs/gst/rtsp/gstrtsprange.h:
17155 * gst-libs/gst/rtsp/gstrtsptransport.c:
17156 * gst-libs/gst/rtsp/gstrtspurl.c:
17157 Fix some more RTSP docs.
17158 Add some missing methods for dealing with messages.
17160 2007-07-24 19:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
17162 Added beginnings of RTSP documentation.
17163 Original commit message from CVS:
17164 * docs/libs/gst-plugins-base-libs-docs.sgml:
17165 * docs/libs/gst-plugins-base-libs-sections.txt:
17166 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
17167 * gst-libs/gst/rtsp/gstrtspbase64.h:
17168 * gst-libs/gst/rtsp/gstrtspconnection.c:
17169 (gst_rtsp_connection_connect), (add_auth_header),
17170 (gst_rtsp_connection_write), (gst_rtsp_connection_send),
17171 (read_body), (gst_rtsp_connection_receive),
17172 (gst_rtsp_connection_next_timeout),
17173 (gst_rtsp_connection_reset_timeout),
17174 (gst_rtsp_connection_set_auth):
17175 * gst-libs/gst/rtsp/gstrtspconnection.h:
17176 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
17177 * gst-libs/gst/rtsp/gstrtspdefs.h:
17178 * gst-libs/gst/rtsp/gstrtspmessage.h:
17179 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
17180 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
17181 (gst_rtsp_range_parse):
17182 * gst-libs/gst/rtsp/gstrtspurl.h:
17183 Added beginnings of RTSP documentation.
17185 2007-07-24 17:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
17187 Document the SDP library.
17188 Original commit message from CVS:
17189 * docs/libs/Makefile.am:
17190 * docs/libs/gst-plugins-base-libs-docs.sgml:
17191 * docs/libs/gst-plugins-base-libs-sections.txt:
17192 * gst-libs/gst/sdp/gstsdp.h:
17193 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
17194 (gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
17195 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
17196 (gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
17197 (gst_sdp_message_get_attribute_val),
17198 (gst_sdp_message_add_attribute), (gst_sdp_media_new),
17199 (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
17200 (gst_sdp_media_get_media), (gst_sdp_media_set_media),
17201 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
17202 (gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
17203 (gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
17204 (gst_sdp_media_get_format), (gst_sdp_media_add_format),
17205 (gst_sdp_media_get_information), (gst_sdp_media_set_information),
17206 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
17207 (gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
17208 (gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
17209 (gst_sdp_media_set_key), (gst_sdp_media_get_key),
17210 (gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
17211 (gst_sdp_media_get_attribute_val_n),
17212 (gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
17213 (print_media), (gst_sdp_message_dump):
17214 * gst-libs/gst/sdp/gstsdpmessage.h:
17215 Document the SDP library.
17216 Add some of the missing SDPMedia methods.
17218 2007-07-24 11:52:56 +0000 Wim Taymans <wim.taymans@gmail.com>
17220 Move SDP and RTSP from helper objects in -good to a reusable library.
17221 Original commit message from CVS:
17223 * gst-libs/gst/Makefile.am:
17224 * gst-libs/gst/rtsp/Makefile.am:
17225 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
17226 * gst-libs/gst/rtsp/gstrtspbase64.h:
17227 * gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
17228 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
17229 (add_auth_header), (add_date_header), (gst_rtsp_connection_write),
17230 (gst_rtsp_connection_send), (read_line), (read_string), (read_key),
17231 (parse_response_status), (parse_request_line), (parse_line),
17232 (gst_rtsp_connection_read), (read_body),
17233 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
17234 (gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
17235 (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
17236 (gst_rtsp_connection_set_auth):
17237 * gst-libs/gst/rtsp/gstrtspconnection.h:
17238 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
17239 (gst_rtsp_strresult), (gst_rtsp_method_as_text),
17240 (gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
17241 (gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
17242 (gst_rtsp_find_method):
17243 * gst-libs/gst/rtsp/gstrtspdefs.h:
17244 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
17245 (gst_rtsp_message_new), (gst_rtsp_message_init),
17246 (gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
17247 (gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
17248 (gst_rtsp_message_init_data), (gst_rtsp_message_unset),
17249 (gst_rtsp_message_free), (gst_rtsp_message_add_header),
17250 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
17251 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
17252 (gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
17253 (gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
17254 (gst_rtsp_message_dump):
17255 * gst-libs/gst/rtsp/gstrtspmessage.h:
17256 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
17257 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
17258 (gst_rtsp_range_parse), (gst_rtsp_range_free):
17259 * gst-libs/gst/rtsp/gstrtsprange.h:
17260 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
17261 (gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
17262 (gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
17263 (range_as_text), (rtsp_transport_mode_as_text),
17264 (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
17265 (gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
17266 (gst_rtsp_transport_free):
17267 * gst-libs/gst/rtsp/gstrtsptransport.h:
17268 * gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
17269 (gst_rtsp_url_free), (gst_rtsp_url_set_port),
17270 (gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
17271 * gst-libs/gst/rtsp/gstrtspurl.h:
17272 * gst-libs/gst/sdp/Makefile.am:
17273 * gst-libs/gst/sdp/gstsdp.h:
17274 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
17275 (gst_sdp_connection_init), (gst_sdp_bandwidth_init),
17276 (gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
17277 (gst_sdp_attribute_init), (gst_sdp_message_new),
17278 (gst_sdp_message_init), (gst_sdp_message_uninit),
17279 (gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
17280 (gst_sdp_media_uninit), (gst_sdp_media_free),
17281 (gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
17282 (gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
17283 (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
17284 (gst_sdp_message_add_zone), (gst_sdp_message_set_key),
17285 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
17286 (gst_sdp_message_get_attribute_val),
17287 (gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
17288 (gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
17289 (gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
17290 (gst_sdp_media_get_attribute_val_n),
17291 (gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
17292 (read_string), (read_string_del), (gst_sdp_parse_line),
17293 (gst_sdp_message_parse_buffer), (print_media),
17294 (gst_sdp_message_dump):
17295 * gst-libs/gst/sdp/gstsdpmessage.h:
17296 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
17297 Move SDP and RTSP from helper objects in -good to a reusable library.
17298 Use a proper gst_ namespace.
17300 2007-07-23 18:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17302 ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
17303 Original commit message from CVS:
17304 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
17305 (vorbis_dec_flush_decode):
17306 Use the new buffer clipping function from gstaudio here.
17308 2007-07-23 18:26:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17310 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
17311 Original commit message from CVS:
17312 * docs/libs/gst-plugins-base-libs-sections.txt:
17313 * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
17314 * gst-libs/gst/audio/audio.h:
17315 * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
17316 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
17317 Also add deprecation guards for gst_audio_structure_set_int() to the
17320 2007-07-23 14:45:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17322 docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
17323 Original commit message from CVS:
17324 * docs/libs/gst-plugins-base-libs-sections.txt:
17327 2007-07-23 11:18:35 +0000 Dan Williams <dcbw@redhat.com>
17329 gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
17330 Original commit message from CVS:
17331 Patch by: Dan Williams <dcbw at redhat dot com>
17332 * gst/playback/gstplaybasebin.c:
17333 (gst_play_base_bin_get_streaminfo_value_array):
17334 Don't return NULL when querying the stream info value array but instead
17335 return an empty array. Fixes #459204.
17337 2007-07-23 10:41:18 +0000 Tim-Philipp Müller <tim@centricular.net>
17339 gst/playback/gsturidecodebin.c: Init debug category before using it.
17340 Original commit message from CVS:
17341 * gst/playback/gsturidecodebin.c:
17342 Init debug category before using it.
17344 2007-07-21 09:56:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17346 gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
17347 Original commit message from CVS:
17348 * gst-libs/gst/interfaces/mixer.h:
17349 Add padding vars in place of the signal pointers
17350 when building with DISABLE_DEPRECATED so that the
17351 interface structure doesn't change size.
17353 2007-07-21 09:21:12 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
17356 Original commit message from CVS:
17357 * docs/libs/gst-plugins-base-libs-sections.txt:
17358 * ext/alsa/gstalsamixer.c:
17359 * ext/alsa/gstalsamixer.h:
17360 * ext/alsa/gstalsamixerelement.c:
17361 * ext/alsa/gstalsamixertrack.c:
17362 * gst-libs/gst/interfaces/mixer.c:
17363 * gst-libs/gst/interfaces/mixer.h:
17364 * gst-libs/gst/interfaces/mixeroptions.c:
17365 * gst-libs/gst/interfaces/mixeroptions.h:
17366 * gst-libs/gst/interfaces/mixertrack.c:
17367 * gst-libs/gst/interfaces/mixertrack.h:
17368 * tests/check/Makefile.am:
17369 * tests/check/libs/mixer.c:
17370 Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
17372 Add support for notifying mixer changes on the message bus, and
17373 implement it in alsamixer.
17374 API: gst_mixer_get_mixer_flags
17375 API: gst_mixer_message_parse_mute_toggled
17376 API: gst_mixer_message_parse_record_toggled
17377 API: gst_mixer_message_parse_volume_changed
17378 API: gst_mixer_message_parse_option_changed
17379 API: GstMixerMessageType
17382 2007-07-20 16:09:03 +0000 Michael Smith <msmith@xiph.org>
17384 sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
17385 Original commit message from CVS:
17386 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
17387 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
17388 xcontext->im_format is only for testing XShm support (as the header
17389 file comments document). Use xvimage->im_format for everything else.
17390 Avoids spurious warnings on buffer allocation before setcaps.
17392 2007-07-20 07:22:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17394 tests/: We should use $(LIBM).
17395 Original commit message from CVS:
17396 * tests/examples/volume/Makefile.am:
17397 * tests/icles/Makefile.am:
17398 We should use $(LIBM).
17400 2007-07-20 06:13:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17402 tests/icles/Makefile.am: This needs -lm.
17403 Original commit message from CVS:
17404 * tests/icles/Makefile.am:
17407 2007-07-18 07:35:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17409 Add stdlib include (free, atoi, exit).
17410 Original commit message from CVS:
17411 * examples/app/appsrc_ex.c:
17412 * examples/switch/switcher.c:
17413 * ext/neon/gstneonhttpsrc.c:
17414 * ext/timidity/gstwildmidi.c:
17415 * ext/x264/gstx264enc.c:
17416 * gst/mve/mveaudioenc.c: (mve_compress_audio):
17417 * gst/rtpmanager/gstrtpclient.c:
17418 * gst/rtpmanager/gstrtpjitterbuffer.c:
17419 * gst/spectrum/demo-audiotest.c:
17420 * gst/spectrum/demo-osssrc.c:
17421 * sys/dvb/gstdvbsrc.c:
17422 Add stdlib include (free, atoi, exit).
17424 2007-07-16 10:10:28 +0000 Wim Taymans <wim.taymans@gmail.com>
17426 gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
17427 Original commit message from CVS:
17428 * gst-libs/gst/rtp/gstbasertppayload.c:
17429 (gst_basertppayload_class_init), (gst_basertppayload_init),
17430 (gst_basertppayload_set_property),
17431 (gst_basertppayload_get_property):
17432 Don't break ABI, restore previous ranges. Keep the default random
17433 selection of timestamp and seqnum offset but as soon as the app sets a
17434 specific value, use that one.
17436 2007-07-14 18:33:15 +0000 Bastien Nocera <hadess@hadess.net>
17438 sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
17439 Original commit message from CVS:
17440 Patch by: Bastien Nocera <hadess at hadess dot net>
17441 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
17442 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
17443 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
17444 * sys/xvimage/xvimagesink.h:
17445 Add option to turn off double-buffering for debugging purposes.
17448 2007-07-14 18:20:41 +0000 Jorn Baayen <jorn@openedhand.com>
17450 sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
17451 Original commit message from CVS:
17452 Patch by: Jorn Baayen <jorn at openedhand dot com>
17453 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
17454 (gst_ximagesink_set_property), (gst_ximagesink_get_property),
17455 (gst_ximagesink_init), (gst_ximagesink_class_init):
17456 * sys/ximage/ximagesink.h:
17457 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
17458 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
17459 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
17460 * sys/xvimage/xvimagesink.h:
17461 add 'handle-expose' property. Useful for video widgets which may want to
17462 be in control of Expose behaviour. Fixes #380625
17464 2007-07-14 17:23:42 +0000 Wim Taymans <wim.taymans@gmail.com>
17466 gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
17467 Original commit message from CVS:
17468 * gst-libs/gst/rtp/gstbasertppayload.c:
17469 (gst_basertppayload_class_init), (gst_basertppayload_init),
17470 (gst_basertppayload_event), (gst_basertppayload_push),
17471 (gst_basertppayload_set_property),
17472 (gst_basertppayload_get_property),
17473 (gst_basertppayload_change_state):
17474 * gst-libs/gst/rtp/gstbasertppayload.h:
17475 Fix ranges of rtp payloader properties so that the full range can be
17476 used in addition to -1 (random).
17477 Fix wrong seqnum reporting in caps.
17480 2007-07-13 18:12:19 +0000 Wim Taymans <wim.taymans@gmail.com>
17482 gst/videorate/gstvideorate.c: Use boilerplate.
17483 Original commit message from CVS:
17484 * gst/videorate/gstvideorate.c: (gst_video_rate_init),
17485 (gst_video_rate_query):
17487 Add latency query, might not be perfect yet but already works a lot
17488 better. Fixes #442557.
17490 2007-07-13 16:05:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17492 sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
17493 Original commit message from CVS:
17494 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
17495 (gst_xvimagesink_setcaps):
17496 * sys/xvimage/xvimagesink.h:
17497 After a caps change, redraw our borders to avoid garbage left there
17498 when the image format changes to a smaller size, like 16:9 -> 4:3
17499 Also, hold the flow_lock a bit longer in the set_caps while we're
17500 fiddling with the xcontext.
17502 2007-07-13 16:02:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17504 Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
17505 Original commit message from CVS:
17508 * tests/Makefile.am:
17509 Remove bogus check for libcheck, since we check for
17510 gstreamer-check and it pulls in the required info from there, and we
17511 weren't actually _using_ the information for libcheck ourselves
17514 2007-07-13 15:52:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17516 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
17517 Original commit message from CVS:
17518 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
17519 (gst_ffmpeg_caps_to_pixfmt):
17520 Fix the r_mask test for RGBA32 on little-endian.
17521 Fix a stupid typo that would have obviously broken
17522 compilation on big-endian, if anyone was testing.
17524 2007-07-12 15:02:43 +0000 Wim Taymans <wim.taymans@gmail.com>
17526 gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
17527 Original commit message from CVS:
17528 * gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
17529 (paint_hline_str4):
17530 * gst/videotestsrc/videotestsrc.h:
17531 Add alpha to the color struct.
17532 Use a default alpha value of 255 instead of 128.
17534 2007-07-12 12:01:20 +0000 Wim Taymans <wim.taymans@gmail.com>
17536 gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
17537 Original commit message from CVS:
17538 * gst/playback/gstplaybasebin.c: (no_more_pads_full),
17540 Clear the dynamic pads counter when starting a new uri. This makes
17541 reusing playbin work again.
17544 2007-07-12 11:13:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17546 configure.ac: Use pkg-config to locate check.
17547 Original commit message from CVS:
17549 Use pkg-config to locate check.
17551 2007-07-11 23:12:12 +0000 Tim-Philipp Müller <tim@centricular.net>
17553 Fix 'make check' build against core CVS.
17554 Original commit message from CVS:
17556 * tests/check/elements/volume.c: (GST_START_TEST):
17557 Fix 'make check' build against core CVS.
17559 2007-07-10 20:46:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17561 gst-libs/gst/: Make gtk-doc happy.
17562 Original commit message from CVS:
17563 * gst-libs/gst/interfaces/propertyprobe.c:
17564 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
17565 * gst-libs/gst/tag/gstvorbistag.c:
17566 Make gtk-doc happy.
17568 2007-07-08 13:07:38 +0000 Tim-Philipp Müller <tim@centricular.net>
17570 gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
17571 Original commit message from CVS:
17572 * gst-libs/gst/audio/gstbaseaudiosink.c:
17573 (gst_base_audio_sink_callback):
17574 Quick hack to make audiosinks stop at EOS when operating in
17575 pull-mode; needs to be fixed properly some day.
17577 2007-07-06 18:19:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17579 docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
17580 Original commit message from CVS:
17581 * docs/libs/gst-plugins-base-libs-sections.txt:
17582 Fix location of includes in the docs.
17584 2007-07-06 11:40:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17586 gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
17587 Original commit message from CVS:
17588 * gst/ffmpegcolorspace/avcodec.h:
17589 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
17590 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
17591 (gst_ffmpegcsp_avpicture_fill):
17592 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
17593 (img_get_alpha_info):
17594 Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
17595 of the existing BGRA32 and RGBA32 formats with the alpha at the other
17596 end of the word. Partially fixes #451908
17598 2007-07-05 08:43:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17600 docs/: Simplify --extra-dir as gtkdoc scans recursively.
17601 Original commit message from CVS:
17602 * docs/libs/Makefile.am:
17603 * docs/plugins/Makefile.am:
17604 Simplify --extra-dir as gtkdoc scans recursively.
17606 2007-07-03 11:52:47 +0000 Wim Taymans <wim.taymans@gmail.com>
17608 gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
17609 Original commit message from CVS:
17610 * gst/adder/gstadder.c: (gst_adder_sink_getcaps),
17611 (gst_adder_request_new_pad):
17612 Make getcaps more robust by not using the proxycaps function. This makes
17613 sure that we don't end up recursively calling getcaps upstream.
17616 2007-06-29 17:21:18 +0000 Wim Taymans <wim.taymans@gmail.com>
17618 gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
17619 Original commit message from CVS:
17620 * gst/audioconvert/audioconvert.c:
17621 Include math.h to fix compilation.
17623 2007-06-29 14:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17625 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
17626 Original commit message from CVS:
17627 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
17628 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
17629 Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
17630 format, as produced by some dc1394 cameras like the iSight.
17631 See http://www.fourcc.org/yuv.php#IYU1
17633 2007-06-28 20:37:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17635 gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
17636 Original commit message from CVS:
17637 * gst/audioconvert/Makefile.am:
17638 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
17639 (check_default), (audio_convert_prepare_context),
17640 (audio_convert_clean_context), (audio_convert_convert):
17641 * gst/audioconvert/audioconvert.h:
17642 * gst/audioconvert/gstaudioconvert.c:
17643 (gst_audio_convert_dithering_get_type),
17644 (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
17645 (gst_audio_convert_init), (gst_audio_convert_set_caps),
17646 (gst_audio_convert_set_property), (gst_audio_convert_get_property):
17647 * gst/audioconvert/gstaudioconvert.h:
17648 * gst/audioconvert/gstaudioquantize.c:
17649 (gst_audio_quantize_setup_noise_shaping),
17650 (gst_audio_quantize_free_noise_shaping),
17651 (gst_audio_quantize_setup_dither),
17652 (gst_audio_quantize_free_dither),
17653 (gst_audio_quantize_setup_quantize_func),
17654 (gst_audio_quantize_setup), (gst_audio_quantize_free):
17655 * gst/audioconvert/gstaudioquantize.h:
17656 Implement dithering and noise shaping in audioconvert. By default now
17657 TPDF dithering (and no noise shaping) will be used when converting
17658 from a higher bit depth to 20 bit depth or smaller, otherwise
17659 everything will be as it is now.
17660 For the last audioconvert in a pipeline it would make sense to
17661 use some kind of noise shaping, enabling it by default for all
17662 conversions would give undesired results though. Fixes #360246.
17663 * tests/check/elements/audioconvert.c: (setup_audioconvert),
17665 Adjust unit test for the new audioconvert.
17667 2007-06-28 11:06:56 +0000 Wim Taymans <wim.taymans@gmail.com>
17669 gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
17670 Original commit message from CVS:
17671 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
17672 Use other metrics as well when estimating the buffer level.
17674 2007-06-28 10:21:19 +0000 Wim Taymans <wim.taymans@gmail.com>
17676 gst/playback/gstplaybasebin.c: Small debug improvement.
17677 Original commit message from CVS:
17678 * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
17679 Small debug improvement.
17680 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
17682 Tweak the rate estimation period.
17683 When calculating the buffer filledness in rate estimation mode, don't
17684 mix it with other metrics.
17686 2007-06-28 09:46:11 +0000 Wim Taymans <wim.taymans@gmail.com>
17688 gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
17689 Original commit message from CVS:
17690 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
17691 (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
17692 When creating the groups, allow for a 5 second, unlimited buffers
17693 preroll phase after which we expose the group.
17694 When the group is exposed, use a small number of buffers up to a 2
17695 second limit. Also disconnect the overrun signal from multiqueue when we
17696 exposed the group because it is not needed anymore.
17698 2007-06-27 22:30:19 +0000 Tim-Philipp Müller <tim@centricular.net>
17700 gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
17701 Original commit message from CVS:
17702 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
17703 Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
17704 to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
17705 (#451707); also, output some debugging info when dealing with
17707 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
17708 Add unit test for the above.
17710 2007-06-27 12:55:20 +0000 Tim-Philipp Müller <tim@centricular.net>
17712 gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
17713 Original commit message from CVS:
17714 * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
17715 Add description for Windows Media RTP caps.
17716 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
17717 Remove RTP fields that don't define the format from caps.
17719 2007-06-27 10:14:03 +0000 Tim-Philipp Müller <tim@centricular.net>
17721 ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
17722 Original commit message from CVS:
17723 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
17724 Skip empty buffers, but not empty header buffers. That way the original
17725 vorbisdec unit test still passes (#451145); also, take into account
17726 that those empty packets might carry a granulepos.
17727 * tests/check/Makefile.am:
17728 * tests/check/elements/vorbisdec.c:
17729 (_create_codebook_header_buffer), (_create_audio_buffer),
17730 (GST_START_TEST), (vorbisdec_suite):
17731 Add unit test that sends an empty packet.
17733 2007-06-27 09:49:51 +0000 Wim Taymans <wim.taymans@gmail.com>
17735 ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
17736 Original commit message from CVS:
17737 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
17738 Don't error out on 0-sized packets, just emit a warning because this is
17739 not a fatal error. Fixes #451145.
17741 2007-06-25 12:43:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17743 docs/plugins/: Update docs with caps info.
17744 Original commit message from CVS:
17745 * docs/plugins/gst-plugins-base-plugins.args:
17746 * docs/plugins/gst-plugins-base-plugins.signals:
17747 * docs/plugins/inspect/plugin-adder.xml:
17748 * docs/plugins/inspect/plugin-alsa.xml:
17749 * docs/plugins/inspect/plugin-audioconvert.xml:
17750 * docs/plugins/inspect/plugin-audiorate.xml:
17751 * docs/plugins/inspect/plugin-audioresample.xml:
17752 * docs/plugins/inspect/plugin-audiotestsrc.xml:
17753 * docs/plugins/inspect/plugin-cdparanoia.xml:
17754 * docs/plugins/inspect/plugin-decodebin.xml:
17755 * docs/plugins/inspect/plugin-decodebin2.xml:
17756 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
17757 * docs/plugins/inspect/plugin-gdp.xml:
17758 * docs/plugins/inspect/plugin-gnomevfs.xml:
17759 * docs/plugins/inspect/plugin-libvisual.xml:
17760 * docs/plugins/inspect/plugin-ogg.xml:
17761 * docs/plugins/inspect/plugin-pango.xml:
17762 * docs/plugins/inspect/plugin-playbin.xml:
17763 * docs/plugins/inspect/plugin-subparse.xml:
17764 * docs/plugins/inspect/plugin-tcp.xml:
17765 * docs/plugins/inspect/plugin-theora.xml:
17766 * docs/plugins/inspect/plugin-typefindfunctions.xml:
17767 * docs/plugins/inspect/plugin-video4linux.xml:
17768 * docs/plugins/inspect/plugin-videorate.xml:
17769 * docs/plugins/inspect/plugin-videoscale.xml:
17770 * docs/plugins/inspect/plugin-videotestsrc.xml:
17771 * docs/plugins/inspect/plugin-volume.xml:
17772 * docs/plugins/inspect/plugin-vorbis.xml:
17773 * docs/plugins/inspect/plugin-ximagesink.xml:
17774 * docs/plugins/inspect/plugin-xvimagesink.xml:
17775 Update docs with caps info.
17777 2007-06-25 12:04:15 +0000 Tim-Philipp Müller <tim@centricular.net>
17779 po/POTFILES.in: Add more files with translatable strings (#450875).
17780 Original commit message from CVS:
17782 Add more files with translatable strings (#450875).
17784 2007-06-23 14:44:07 +0000 Edward Hervey <bilboed@bilboed.com>
17786 ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
17787 Original commit message from CVS:
17788 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
17789 The chain should be freed if we error out here, else it will leak.
17790 * gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
17791 (cleanup_decodebin):
17792 Don't forget to *properly* remove the signals, else it will leak.
17794 2007-06-22 14:25:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17796 MAINTAINERS: Updating all the maintainers files
17797 Original commit message from CVS:
17799 Updating all the maintainers files
17801 2007-06-21 08:34:46 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17803 tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
17804 Original commit message from CVS:
17805 * tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
17807 Destroy and recreate parse-launch based pipeline after stop to be able
17808 to play again. Reorder some code and add more comments.
17810 2007-06-20 11:09:03 +0000 Wim Taymans <wim@fluendo.com>
17812 gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
17813 Original commit message from CVS:
17814 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
17815 When handling a delayed-caps notification case, mark
17816 the group as dynamic so that the nbdynamic count is
17817 incremented and decremented correctly. Fixes: #449156
17818 Patch by: Wim Taymans <wim@fluendo.com>
17820 2007-06-19 19:13:04 +0000 Andy Wingo <wingo@pobox.com>
17823 * gst-libs/gst/audio/gstbaseaudiosink.c:
17824 * win32/common/config.h:
17825 gst-libs/gst/audio/gstbaseaudiosink.c
17826 Original commit message from CVS:
17827 2007-06-19 Andy Wingo <wingo@pobox.com>
17828 * gst-libs/gst/audio/gstbaseaudiosink.c
17829 (gst_base_audio_sink_init): Enable pull-mode operation.
17831 2007-06-19 09:34:35 +0000 Michael Smith <msmith@xiph.org>
17833 gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
17834 Original commit message from CVS:
17835 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
17836 Change minimum rate back to 1000 to allow low-sample-rate wav files
17839 2007-06-17 17:27:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
17841 po/vi.po: Update translations.
17842 Original commit message from CVS:
17844 Update translations.
17846 2007-06-16 03:42:14 +0000 David Schleef <ds@schleef.org>
17848 gst/playback/gstqueue2.c: Fix compile error from ignored return value.
17849 Original commit message from CVS:
17850 * gst/playback/gstqueue2.c:
17851 Fix compile error from ignored return value.
17853 2007-06-15 15:23:36 +0000 Michael Smith <msmith@xiph.org>
17855 gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
17856 Original commit message from CVS:
17857 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
17858 Update tmpbuf for all neccesary rows, not just one, as is required
17862 2007-06-15 11:15:28 +0000 Michael Smith <msmith@xiph.org>
17864 tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
17865 Original commit message from CVS:
17866 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
17867 (eos_buffer_probe):
17868 Add a test that ensures we set DELTA_UNIT on all non-header,
17869 non-video buffers, if we have a video stream.
17870 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
17871 (gst_ogg_mux_process_best_pad):
17872 Move setting delta_pad to earlier, where we inspect all pads, so
17873 that leading audio pages don't get DELTA_UNIT unset if they come
17874 before the first DELTA_UNIT from video pages. Fixes the newly-added
17875 test. Fixes #385527.
17877 2007-06-14 19:53:27 +0000 Tim-Philipp Müller <tim@centricular.net>
17879 tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
17880 Original commit message from CVS:
17881 * tests/check/pipelines/streamheader.c: (streamheader_suite):
17882 Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
17883 fails on the p5-ppc64 build bot and the failure looks like it is due
17884 to the same issue as #348114, ie. a compiler bug.
17886 2007-06-13 18:20:57 +0000 Edward Hervey <bilboed@bilboed.com>
17888 gst/playback/gstqueue2.c: Fix build on MacOSX.
17889 Original commit message from CVS:
17890 * gst/playback/gstqueue2.c: (gst_queue_create_read):
17891 Fix build on MacOSX.
17893 2007-06-13 09:01:32 +0000 Wim Taymans <wim.taymans@gmail.com>
17895 ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
17896 Original commit message from CVS:
17897 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
17898 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
17899 Fix compilation on mingw. Fixes #446972.
17901 2007-06-12 08:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
17903 gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
17904 Original commit message from CVS:
17905 Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
17906 * gst/playback/gstqueue2.c: (update_buffering),
17907 (gst_queue_locked_enqueue):
17908 Fix a division by zero when the max percent is <= 0. Fixes #446572.
17909 also update the buffering status when receiving events. Fixes #446551.
17911 2007-06-11 11:32:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
17913 gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
17914 Original commit message from CVS:
17915 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
17916 * gst/playback/gstqueue2.c: (gst_queue_peer_query),
17917 (gst_queue_handle_src_query):
17918 Wait for preroll before attempting to forward a duration query upstream.
17921 2007-06-07 21:08:38 +0000 Sébastien Moutte <sebastien@moutte.net>
17923 gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
17924 Original commit message from CVS:
17925 * gst-libs/gst/rtp/gstbasertpdepayload.c:
17926 (gst_base_rtp_depayload_set_gst_timestamp):
17927 Use G_GINT64_CONSTANT macro for int64 constant.
17928 * win32/common/libgstinterfaces.def:
17929 * win32/common/libgsttag.def:
17930 Add new exported functions.
17932 2007-06-07 14:25:32 +0000 Tim-Philipp Müller <tim@centricular.net>
17934 ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
17935 Original commit message from CVS:
17936 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
17937 The BOS page of the first Dirac video stream needs to come before
17938 the BOS page of any Vorbis streams or other audio streams, just like
17941 2007-06-07 09:11:27 +0000 Wim Taymans <wim.taymans@gmail.com>
17943 gst/playback/gstqueue2.c: Fix compilation.
17944 Original commit message from CVS:
17945 * gst/playback/gstqueue2.c: (gst_queue_get_range):
17948 2007-06-06 13:36:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
17950 gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
17951 Original commit message from CVS:
17952 Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
17953 * gst/playback/gstqueue2.c: (gst_queue_init),
17954 (gst_queue_handle_sink_event), (gst_queue_chain),
17955 (gst_queue_get_range), (gst_queue_src_checkgetrange_function),
17956 (gst_queue_sink_activate_push), (gst_queue_src_activate_push),
17957 (gst_queue_src_activate_pull):
17958 Add pull based scheduling and fix some deadlocks. Fixes #444523.
17959 Does not yet completely work because duration queries upstream won't
17962 2007-06-06 09:08:50 +0000 Wim Taymans <wim.taymans@gmail.com>
17964 Some more fseeko checks.
17965 Original commit message from CVS:
17967 * gst/playback/gstqueue2.c: (gst_queue_create_read):
17968 Some more fseeko checks.
17970 2007-06-06 08:01:42 +0000 Wim Taymans <wim.taymans@gmail.com>
17972 configure.ac: check for large file support.
17973 Original commit message from CVS:
17975 check for large file support.
17977 2007-06-05 21:36:11 +0000 Sven Arvidsson <sa@whiz.se>
17979 gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
17980 Original commit message from CVS:
17981 Based on a patch by Sven Arvidsson <sa at whiz dot se>:
17982 * gst/subparse/gstsubparse.c: (parse_subrip),
17983 (subviewer_unescape_newlines), (parse_subviewer),
17984 (gst_sub_parse_data_format_autodetect),
17985 (gst_sub_parse_format_autodetect), (gst_subparse_type_find):
17986 * gst/subparse/gstsubparse.h:
17987 Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
17988 * tests/check/elements/subparse.c: (GST_START_TEST),
17990 Add a unit test for both SubViewer formats.
17992 2007-06-05 17:08:04 +0000 Michael Smith <msmith@xiph.org>
17994 gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
17995 Original commit message from CVS:
17996 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
17997 Don't overflow intermediate values when seeking to large time values
18000 2007-06-05 17:02:13 +0000 Wim Taymans <wim.taymans@gmail.com>
18002 gst/playback/gstqueue2.c: Include stdio to define fseeko.
18003 Original commit message from CVS:
18004 * gst/playback/gstqueue2.c: (gst_queue_have_data),
18005 (gst_queue_create_read), (gst_queue_read_item_from_file),
18006 (gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
18007 Include stdio to define fseeko.
18009 2007-06-05 16:37:09 +0000 Edward Hervey <edward@fluendo.com>
18011 sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
18012 Original commit message from CVS:
18013 Patch by: Edward Hervey <edward@fluendo.com>
18014 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
18015 (gst_v4lsrc_query):
18016 Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
18018 2007-06-05 16:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
18020 gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
18021 Original commit message from CVS:
18022 * gst-libs/gst/riff/Makefile.am:
18023 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
18024 Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
18025 our own implementation.
18027 2007-06-05 16:19:30 +0000 Wim Taymans <wim.taymans@gmail.com>
18029 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
18030 Original commit message from CVS:
18031 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18032 (gst_base_rtp_depayload_setcaps),
18033 (gst_base_rtp_depayload_set_gst_timestamp),
18034 (gst_base_rtp_depayload_change_state):
18035 Handle timestamp wraparound.
18037 2007-06-05 16:17:30 +0000 Wim Taymans <wim.taymans@gmail.com>
18039 gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
18040 Original commit message from CVS:
18041 * gst/playback/gsturidecodebin.c: (no_more_pads_full),
18042 (new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
18043 (gst_uri_decode_bin_change_state):
18044 Make sure we name srcpads uniquely even when using different internal
18046 Signal no-more-pads when no more dynamic elements exist.
18047 Remove pads on cleanup.
18049 2007-06-05 16:14:23 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
18051 gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
18052 Original commit message from CVS:
18053 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
18054 * gst/playback/gstqueue2.c: (gst_queue_class_init),
18055 (gst_queue_init), (gst_queue_finalize),
18056 (gst_queue_write_buffer_to_file), (gst_queue_have_data),
18057 (gst_queue_create_read), (gst_queue_read_item_from_file),
18058 (gst_queue_open_temp_location_file),
18059 (gst_queue_close_temp_location_file), (gst_queue_locked_flush),
18060 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
18061 (gst_queue_is_empty), (gst_queue_is_filled),
18062 (gst_queue_change_state), (gst_queue_set_temp_location),
18063 (gst_queue_set_property):
18064 Add support for filebased buffering. Fixes #441264.
18066 2007-06-05 16:05:19 +0000 Wim Taymans <wim.taymans@gmail.com>
18068 gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
18069 Original commit message from CVS:
18070 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
18071 (analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
18072 (caps_notify_group_cb), (gst_decode_group_new),
18073 (gst_decode_group_free):
18074 Add support for delayed caps fixation when autoplugging.
18075 Optimize cases where a multiqueue is not needed/wanted, like right after
18076 anything that is not a demuxer.
18078 2007-06-05 16:02:57 +0000 Wim Taymans <wim.taymans@gmail.com>
18080 ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
18081 Original commit message from CVS:
18082 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
18083 (gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
18084 (gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
18085 consideratly speedup ogg chain detection by not trying to find a base
18086 timestamp for skeleton streams.
18088 2007-06-05 16:00:33 +0000 Wim Taymans <wim.taymans@gmail.com>
18090 gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
18091 Original commit message from CVS:
18092 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
18093 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
18094 (gst_multi_fd_sink_remove_flush),
18095 (gst_multi_fd_sink_remove_client_link),
18096 (gst_multi_fd_sink_handle_client_write),
18097 (gst_multi_fd_sink_handle_clients):
18098 * gst/tcp/gstmultifdsink.h:
18099 Add support for remuve_flush.
18101 2007-06-05 15:59:00 +0000 Wim Taymans <wim.taymans@gmail.com>
18103 Add draft design for forcing keyframes in encoders and implement in theoraenc.
18104 Original commit message from CVS:
18105 * docs/design/draft-keyframe-force.txt:
18106 * ext/theora/theoraenc.c: (theora_enc_sink_event),
18107 (theora_enc_chain):
18108 Add draft design for forcing keyframes in encoders and implement in
18111 2007-06-05 13:22:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18113 configure.ac: Back to CVS
18114 Original commit message from CVS:
18118 === release 0.10.13 ===
18120 2007-06-05 12:50:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18126 * docs/plugins/gst-plugins-base-plugins.args:
18127 * docs/plugins/inspect/plugin-adder.xml:
18128 * docs/plugins/inspect/plugin-alsa.xml:
18129 * docs/plugins/inspect/plugin-audioconvert.xml:
18130 * docs/plugins/inspect/plugin-audiorate.xml:
18131 * docs/plugins/inspect/plugin-audioresample.xml:
18132 * docs/plugins/inspect/plugin-audiotestsrc.xml:
18133 * docs/plugins/inspect/plugin-cdparanoia.xml:
18134 * docs/plugins/inspect/plugin-decodebin.xml:
18135 * docs/plugins/inspect/plugin-decodebin2.xml:
18136 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
18137 * docs/plugins/inspect/plugin-gdp.xml:
18138 * docs/plugins/inspect/plugin-gnomevfs.xml:
18139 * docs/plugins/inspect/plugin-libvisual.xml:
18140 * docs/plugins/inspect/plugin-ogg.xml:
18141 * docs/plugins/inspect/plugin-pango.xml:
18142 * docs/plugins/inspect/plugin-playbin.xml:
18143 * docs/plugins/inspect/plugin-subparse.xml:
18144 * docs/plugins/inspect/plugin-tcp.xml:
18145 * docs/plugins/inspect/plugin-theora.xml:
18146 * docs/plugins/inspect/plugin-typefindfunctions.xml:
18147 * docs/plugins/inspect/plugin-video4linux.xml:
18148 * docs/plugins/inspect/plugin-videorate.xml:
18149 * docs/plugins/inspect/plugin-videoscale.xml:
18150 * docs/plugins/inspect/plugin-videotestsrc.xml:
18151 * docs/plugins/inspect/plugin-volume.xml:
18152 * docs/plugins/inspect/plugin-vorbis.xml:
18153 * docs/plugins/inspect/plugin-ximagesink.xml:
18154 * docs/plugins/inspect/plugin-xvimagesink.xml:
18155 * gst-plugins-base.doap:
18156 * win32/common/config.h:
18157 * win32/vs6/grammar.dsp:
18158 * win32/vs6/gst_plugins_base.dsw:
18159 * win32/vs6/libgstadder.dsp:
18160 * win32/vs6/libgstaudio.dsp:
18161 * win32/vs6/libgstaudioconvert.dsp:
18162 * win32/vs6/libgstaudiorate.dsp:
18163 * win32/vs6/libgstaudioresample.dsp:
18164 * win32/vs6/libgstaudioscale.dsp:
18165 * win32/vs6/libgstaudiotestsrc.dsp:
18166 * win32/vs6/libgstcdda.dsp:
18167 * win32/vs6/libgstdecodebin.dsp:
18168 * win32/vs6/libgstdecodebin2.dsp:
18169 * win32/vs6/libgstdirectsound.dsp:
18170 * win32/vs6/libgstffmpegcolorspace.dsp:
18171 * win32/vs6/libgstgdp.dsp:
18172 * win32/vs6/libgstinterfaces.dsp:
18173 * win32/vs6/libgstnetbuffer.dsp:
18174 * win32/vs6/libgstogg.dsp:
18175 * win32/vs6/libgstpbutils.dsp:
18176 * win32/vs6/libgstplaybin.dsp:
18177 * win32/vs6/libgstriff.dsp:
18178 * win32/vs6/libgstrtp.dsp:
18179 * win32/vs6/libgstsinesrc.dsp:
18180 * win32/vs6/libgstsubparse.dsp:
18181 * win32/vs6/libgsttag.dsp:
18182 * win32/vs6/libgsttheora.dsp:
18183 * win32/vs6/libgsttypefindfunctions.dsp:
18184 * win32/vs6/libgstutils.dsp:
18185 * win32/vs6/libgstvideo.dsp:
18186 * win32/vs6/libgstvideorate.dsp:
18187 * win32/vs6/libgstvideoscale.dsp:
18188 * win32/vs6/libgstvideotestsrc.dsp:
18189 * win32/vs6/libgstvolume.dsp:
18190 * win32/vs6/libgstvorbis.dsp:
18191 Release 0.10.13 "What's going on?"
18192 Original commit message from CVS:
18193 Release 0.10.13 "What's going on?"
18195 2007-06-05 12:32:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18213 Original commit message from CVS:
18216 2007-05-31 17:08:58 +0000 Wim Taymans <wim@fluendo.com>
18218 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...
18219 Original commit message from CVS:
18220 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
18221 In riff, the depth is stored in the size field but it just means that
18222 the least significant bits are cleared. We can therefore just play
18223 the sample as if it had a depth == width. Fixes: #440997
18224 Patch by: Wim Taymans <wim@fluendo.com>
18225 Patch by: Sebastian Dröge <slomo@circular-chaos.org>
18227 2007-05-31 16:36:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18229 gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
18230 Original commit message from CVS:
18231 * gst-libs/gst/floatcast/floatcast.h:
18232 Define inline when needed on win32 builds. Fixes: #441295
18234 2007-05-29 13:38:35 +0000 Wim Taymans <wim.taymans@gmail.com>
18236 gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
18237 Original commit message from CVS:
18238 * gst/playback/gstplaybasebin.c: (queue_overrun),
18239 (no_more_pads_full):
18240 Stop buffering when the group is commited because the queues filled up.
18243 2007-05-25 10:07:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18245 Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
18246 Original commit message from CVS:
18247 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
18248 (gst_alsa_mixer_free), (gst_alsa_mixer_update),
18249 (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
18250 (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
18251 (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
18252 * ext/alsa/gstalsamixer.h:
18253 * ext/alsa/gstalsamixerelement.c:
18254 (gst_alsa_mixer_element_interface_supported),
18255 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
18256 (gst_alsa_mixer_element_set_property),
18257 (gst_alsa_mixer_element_get_property),
18258 (gst_alsa_mixer_element_change_state):
18259 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
18260 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
18261 (gst_mixer_option_changed):
18262 * gst-libs/gst/interfaces/mixer.h:
18263 Revert commits towards #152864 made so far. We'll pick it up again
18264 after the 0.10.13 release.
18266 2007-05-24 16:22:23 +0000 Wim Taymans <wim.taymans@gmail.com>
18268 gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
18269 Original commit message from CVS:
18270 * gst-libs/gst/audio/gstbaseaudiosink.c:
18271 (gst_base_audio_sink_render):
18272 After an interrupt (PAUSED/flush) assume that the next sample should not
18273 be aligned to the previous sample. Fixes #417992.
18275 2007-05-24 15:16:59 +0000 Tim-Philipp Müller <tim@centricular.net>
18277 gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
18278 Original commit message from CVS:
18279 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
18280 Don't add channels and rate fields to the template caps for
18281 audio/x-dts, as wavparse might not always be able to set them,
18282 which would then lead to 'caps are not a real subset of the
18283 template caps' warnings.
18285 2007-05-24 11:15:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18287 gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
18288 Original commit message from CVS:
18289 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
18290 Handle unknown or invalid pads without crashing, as might occur if
18291 a media file like an mp3 is specified as a subtitle file.
18294 2007-05-24 10:19:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18296 gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
18297 Original commit message from CVS:
18298 * gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
18300 Block the subtitle bin output queue before ghosting it and linking,
18301 then unblock after. This avoids spurious not-linked errors caused
18302 by the queue starting up (because it gets linked when it is ghosted).
18305 2007-05-23 15:54:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18307 tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
18308 Original commit message from CVS:
18309 * tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
18310 Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
18311 file. Avoids flukes where the input gets typefound to some valid but
18314 2007-05-22 15:45:19 +0000 Tim-Philipp Müller <tim@centricular.net>
18316 tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
18317 Original commit message from CVS:
18318 * tests/check/Makefile.am:
18319 * tests/check/elements/.cvsignore:
18320 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
18321 (cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
18322 Add unit test for gnomevfssink seeking and position reporting for
18325 2007-05-22 15:30:26 +0000 Mark Nauwelaerts <manauw@skynet.be>
18327 ext/gnomevfs/gstgnomevfssink.*: see #412648.
18328 Original commit message from CVS:
18329 Patch by: Mark Nauwelaerts <manauw at skynet be>
18330 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
18331 (gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
18332 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
18333 * ext/gnomevfs/gstgnomevfssink.h:
18334 Fix position reporting, especially after a seek (from upstream),
18337 2007-05-22 15:04:41 +0000 Tim-Philipp Müller <tim@centricular.net>
18339 ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
18340 Original commit message from CVS:
18341 * ext/cdparanoia/gstcdparanoiasrc.c:
18344 2007-05-22 11:40:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18346 gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
18347 Original commit message from CVS:
18348 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
18349 Specify the full valid range for MP3 samplerates. Fixes a regression
18350 caused by extra header checks since the last release.
18352 2007-05-21 15:32:42 +0000 Mike Smith <msmith@xiph.org>
18354 sys/: Fix a locking-order bug I introduced with my changes the other day.
18355 Original commit message from CVS:
18356 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
18357 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
18358 Fix a locking-order bug I introduced with my changes the other day.
18359 Patch by Mike Smith.
18361 2007-05-21 15:24:21 +0000 Michael Smith <msmith@xiph.org>
18363 ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
18364 Original commit message from CVS:
18365 * ext/theora/theoradec.c: (theora_handle_data_packet):
18366 Don't look inside 0-length packets (which indicate duplicated
18369 2007-05-21 10:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
18372 Original commit message from CVS:
18373 * ext/cdparanoia/gstcdparanoiasrc.c:
18374 (gst_cd_paranoia_src_read_sector):
18375 * gst-libs/gst/audio/gstbaseaudiosrc.c:
18376 (gst_base_audio_src_create):
18378 * ext/theora/theoradec.c: (theora_dec_sink_event):
18380 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18381 (gst_base_rtp_depayload_set_gst_timestamp):
18383 * gst/playback/gstdecodebin.c: (queue_underrun_cb):
18384 And some debug info when a FIXME path is hit.
18386 2007-05-21 09:45:28 +0000 Wim Taymans <wim.taymans@gmail.com>
18388 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
18389 Original commit message from CVS:
18390 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18391 (gst_base_rtp_audio_payload_class_init),
18392 (gst_base_rtp_audio_payload_init),
18393 (gst_base_rtp_audio_payload_finalize),
18394 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
18395 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
18396 (gst_base_rtp_payload_audio_handle_event):
18397 Some cleanups, remove minptime property as it is now in the parent
18399 Override parent class event function.
18400 * gst-libs/gst/rtp/gstbasertppayload.c:
18401 (gst_basertppayload_class_init), (gst_basertppayload_init),
18402 (gst_basertppayload_event), (gst_basertppayload_set_property),
18403 (gst_basertppayload_get_property):
18404 * gst-libs/gst/rtp/gstbasertppayload.h:
18405 Add min-ptime property.
18406 Add handle-event vmethod. Fixes #415001.
18408 2007-05-18 17:10:03 +0000 Christian Schaller <uraeus@gnome.org>
18410 * gst-plugins-base.spec.in:
18412 Original commit message from CVS:
18415 2007-05-18 15:23:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18417 gst-libs/gst/audio/gstbaseaudiosink.c
18418 Original commit message from CVS:
18419 * gst-libs/gst/audio/gstbaseaudiosink.c
18420 (gst_base_audio_sink_change_state):
18421 Fix typo in comment.
18422 * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
18423 free_dynamics, pad_probe, close_pad_link, try_to_link_1,
18424 get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
18426 * gst/playback/gstplaybin.c (gst_play_bin_set_property,
18427 gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
18428 Remove trailing whitespaces in comments.
18429 * gst/volume/Makefile.am:
18432 2007-05-18 15:10:08 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
18435 * gst-libs/gst/interfaces/mixer.h:
18436 gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
18437 Original commit message from CVS:
18438 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
18439 * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
18440 set_option, get_option, _gst_reserved):
18441 Revert reordering functions (keep ABI).
18443 2007-05-17 17:35:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18445 sys/: When we create our own window, indicate that we handle the
18446 Original commit message from CVS:
18447 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
18448 (gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
18449 (gst_ximagesink_show_frame):
18450 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
18451 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
18452 (gst_xvimagesink_show_frame):
18453 When we create our own window, indicate that we handle the
18454 WM_DELETE client message from the window manager, so that it won't
18455 kill our window (and our app) along with it. Handle ClientMessage,
18456 post an error on the bus, and close the window. Further buffers
18457 arriving will result in a FlowError because the window has been
18460 Clean up the X event handling loop and make them the same for
18461 both xvimagesink and ximagesink while I'm at it.
18463 2007-05-17 16:27:32 +0000 Wim Taymans <wim.taymans@gmail.com>
18465 gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
18466 Original commit message from CVS:
18467 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
18468 Make decodebin2 autoplug depayloaders too.
18469 * gst/playback/gsturidecodebin.c: (source_new_pad):
18470 Set the newly created decoder in a usable state when autoplugging a
18471 dynamic source such as RTSP.
18473 2007-05-17 16:11:03 +0000 Tim-Philipp Müller <tim@centricular.net>
18475 gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
18476 Original commit message from CVS:
18477 * gst/playback/gststreaminfo.c: (cb_probe):
18478 Ignore video-codec tag for audio streams and ignore audio-codec tags
18479 for video streams. Should make codec name collection a bit more
18480 robust against sloppy demuxers that send tag events containing both
18481 tags down each pad.
18483 2007-05-17 15:22:44 +0000 Wim Taymans <wim.taymans@gmail.com>
18485 gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
18486 Original commit message from CVS:
18487 * gst/playback/gstqueue2.c: (update_rates):
18488 Tweak the buffering thresholds a little.
18489 Update the buffer size with the previously calculate rate instead of
18490 only when we calculate a new rate so that we get smoother buffering
18492 * gst/playback/Makefile.am:
18493 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
18494 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
18495 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
18496 (gst_uri_decode_bin_get_property), (unknown_type),
18497 (add_element_stream), (no_more_pads_full), (no_more_pads),
18498 (source_no_more_pads), (new_decoded_pad), (array_has_value),
18499 (gen_source_element), (has_all_raw_caps), (analyse_source),
18500 (remove_decoders), (make_decoder), (remove_source),
18501 (source_new_pad), (setup_source), (decoder_query_init),
18502 (decoder_query_duration_fold), (decoder_query_duration_done),
18503 (decoder_query_position_fold), (decoder_query_position_done),
18504 (decoder_query_latency_fold), (decoder_query_latency_done),
18505 (decoder_query_seeking_fold), (decoder_query_seeking_done),
18506 (decoder_query_generic_fold), (gst_uri_decode_bin_query),
18507 (gst_uri_decode_bin_change_state), (plugin_init):
18508 New element that intergrates a source, optional buffering element and
18511 2007-05-17 14:17:17 +0000 Tim-Philipp Müller <tim@centricular.net>
18513 configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
18514 Original commit message from CVS:
18516 Bump libtheora requirement to 1.0alpha5 for the pixformat check
18517 (also has a .pc file, so we don't need the fallback check any
18518 longer). Fixes #438840.
18520 2007-05-17 13:36:11 +0000 Wim Taymans <wim.taymans@gmail.com>
18522 gst/playback/gstqueue2.c: fix build.
18523 Original commit message from CVS:
18524 * gst/playback/gstqueue2.c: (gst_queue_get_type),
18525 (gst_queue_class_init), (gst_queue_finalize), (update_time_level),
18526 (apply_segment), (apply_buffer), (update_buffering),
18527 (reset_rate_timer), (update_rates), (gst_queue_locked_flush),
18528 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
18529 (gst_queue_handle_sink_event), (gst_queue_is_filled),
18530 (gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
18534 2007-05-17 11:57:44 +0000 Wim Taymans <wim.taymans@gmail.com>
18536 gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
18537 Original commit message from CVS:
18538 * gst/playback/Makefile.am:
18539 * gst/playback/gstqueue2.c: (gst_queue_get_type),
18540 (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
18541 (gst_queue_getcaps), (gst_queue_bufferalloc),
18542 (gst_queue_acceptcaps), (update_time_level), (apply_segment),
18543 (apply_buffer), (update_buffering), (reset_rate_timer),
18544 (update_rates), (gst_queue_locked_flush),
18545 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
18546 (gst_queue_handle_sink_event), (gst_queue_is_empty),
18547 (gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
18548 (gst_queue_loop), (gst_queue_handle_src_event),
18549 (gst_queue_handle_src_query), (gst_queue_sink_activate_push),
18550 (gst_queue_src_activate_push), (gst_queue_change_state),
18551 (gst_queue_set_property), (gst_queue_get_property), (plugin_init):
18552 On our way to playbin2 this is the new network queue that does buffering
18553 all by itself using high and low watermarks. It can also measure up and
18554 downstream bandwidth to optimally size the queue.
18556 2007-05-17 11:16:14 +0000 Michael Smith <msmith@xiph.org>
18558 gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
18559 Original commit message from CVS:
18560 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
18561 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
18562 Use the segment->last_stop value to calculate the next timestamp to
18563 generate after a seek; not the segment->start value.
18565 2007-05-15 20:14:06 +0000 David Schleef <ds@schleef.org>
18567 docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
18568 Original commit message from CVS:
18569 * docs/Makefile.am: Install docs even when --disable-gtk-doc
18570 is disabled. This matches the behavior of gtk+. Fixes #349099.
18572 2007-05-15 17:11:09 +0000 Wim Taymans <wim.taymans@gmail.com>
18574 ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
18575 Original commit message from CVS:
18576 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
18577 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
18578 Some more chained streaming ogg timestamp fixes.
18580 2007-05-15 16:46:10 +0000 Wim Taymans <wim.taymans@gmail.com>
18582 ext/ogg/gstoggdemux.c: Add some FIXMEs.
18583 Original commit message from CVS:
18584 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
18585 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
18586 (gst_ogg_demux_handle_page):
18588 Fix chain start/stop segment handling based on patch by
18589 <ahalda at cs dot mcgill dot ca> see #320984.
18591 2007-05-15 15:33:54 +0000 Michael Smith <msmith@xiph.org>
18593 configure.ac: We don't require a C++ compiler. So don't require one.
18594 Original commit message from CVS:
18596 We don't require a C++ compiler. So don't require one.
18598 2007-05-15 15:29:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18601 * ext/alsa/gstalsamixer.c:
18602 ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
18603 Original commit message from CVS:
18604 * ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
18605 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
18606 gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
18607 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
18608 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
18609 gst_alsa_mixer_update_track):
18610 Apply some of the cleanup Tim suggested in #152864 afterwards.
18612 2007-05-15 14:01:26 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
18614 ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
18615 Original commit message from CVS:
18616 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
18617 * ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
18618 _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
18619 gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
18620 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
18621 gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
18622 gst_alsa_mixer_handle_source_callback,
18623 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
18624 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
18625 gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
18626 gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
18627 gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
18628 gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
18629 * ext/alsa/gstalsamixer.h (handle_source, interface, dir):
18630 * ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
18631 gst_alsa_mixer_element_interface_supported,
18632 gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
18633 gst_alsa_mixer_element_set_property,
18634 gst_alsa_mixer_element_get_property,
18635 gst_alsa_mixer_element_change_state):
18636 * ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
18637 * gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
18638 gst_mixer_option_changed):
18639 * gst-libs/gst/interfaces/mixer.h (set_option, get_option,
18640 volume_changed, option_changed, _gst_reserved):
18641 Implement notification for alsamixer. Fixes #152864
18643 2007-05-15 03:53:11 +0000 David Schleef <ds@schleef.org>
18645 gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
18646 Original commit message from CVS:
18647 * gst/videotestsrc/videotestsrc.c:
18648 * gst/videotestsrc/videotestsrc.h:
18649 Add support for video/x-raw-bayer.
18651 2007-05-13 01:06:19 +0000 David Schleef <ds@schleef.org>
18653 sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
18654 Original commit message from CVS:
18655 * sys/xvimage/xvimagesink.c:
18656 Add some sanity checking for the XVImage size returned by X.
18657 Related to #377400.
18659 2007-05-12 16:18:39 +0000 Wim Taymans <wim.taymans@gmail.com>
18661 gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
18662 Original commit message from CVS:
18663 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18664 (gst_base_rtp_depayload_setcaps),
18665 (gst_base_rtp_depayload_set_gst_timestamp):
18666 Parse and use additional caps fields as described in updated
18667 application/x-rtp caps spec.
18669 2007-05-12 16:16:22 +0000 Wim Taymans <wim.taymans@gmail.com>
18671 ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
18672 Original commit message from CVS:
18673 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
18674 (gst_ogg_demux_collect_chain_info):
18675 If there is a stream in a chain without any data packets, ignore the
18676 stream in the total length calculations. Might be related to #436820.
18678 2007-05-11 17:33:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18680 gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
18681 Original commit message from CVS:
18682 * gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
18683 (mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
18684 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
18685 (mpeg_video_type_find), (mpeg_video_stream_type_find),
18687 Consolidate and re-work our mpeg system stream detection to probe
18688 more packets and produce a higher confidence result. Fixes a
18689 regression caused by lowering the typefind probability last year
18690 - related to bug #397810. Remove the redundant MPEG-1 specific
18691 typefind function, as the new one detects both MPEG-1 & MPEG-2
18693 Also cleanup the MPEG elementary and MPEG-TS detection functions a
18695 Tested against my media test directory, with some improvements and
18698 2007-05-10 15:28:13 +0000 Wim Taymans <wim.taymans@gmail.com>
18700 gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
18701 Original commit message from CVS:
18702 * gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
18703 (queue_out_of_data):
18704 Connect to the new queue "pushing" signal instead of the broken
18707 2007-05-09 21:17:40 +0000 Sébastien Moutte <sebastien@moutte.net>
18709 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
18710 Original commit message from CVS:
18711 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18712 (gst_base_rtp_audio_payload_handle_frame_based_buffer):
18713 Move variable declaration before the first instruction.
18714 * gst/videotestsrc/videotestsrc.c:
18715 Define M_PI if it's not defined yet.
18716 * win32/common/libgstrtp.def:
18717 Add new exported functions.
18719 2007-05-09 11:54:32 +0000 Michael Smith <msmith@xiph.org>
18721 ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
18722 Original commit message from CVS:
18723 * ext/theora/theoradec.c: (theora_handle_type_packet):
18724 gst_pad_push_event() does not return a GstFlowReturn!
18726 2007-05-09 11:25:34 +0000 Wim Taymans <wim.taymans@gmail.com>
18728 tests/examples/seek/: Some small cosmetic changes.
18729 Original commit message from CVS:
18730 * tests/examples/seek/scrubby.c: (stop_cb), (main):
18731 * tests/examples/seek/seek.c: (do_seek):
18732 Some small cosmetic changes.
18734 2007-05-08 19:24:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18737 * gst/adder/gstadder.c:
18738 * gst/adder/gstadder.h:
18739 gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
18740 Original commit message from CVS:
18741 * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
18742 gst_adder_change_state):
18743 * gst/adder/gstadder.h (bps, offset, collect_event, segment,
18744 segment_pending, segment_position, segment_rate):
18745 Handle playback-rate on adder.
18747 2007-05-07 11:43:31 +0000 Michael Smith <msmith@xiph.org>
18749 ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
18750 Original commit message from CVS:
18751 * ext/theora/gsttheoradec.h:
18752 * ext/theora/theoradec.c: (gst_theora_dec_reset),
18753 (theora_dec_sink_event), (theora_handle_comment_packet),
18754 (theora_handle_type_packet), (theora_dec_change_state):
18755 Don't push events (newsegment, tags) before initialising the
18757 This is neccesary for seeking to work correctly in gnonlin.
18759 2007-05-04 13:10:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18761 gst/: gst/audiotestsrc/gstaudiotestsrc.c
18762 Original commit message from CVS:
18763 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18764 * gst/adder/gstadder.c:
18765 * gst/audiotestsrc/gstaudiotestsrc.c
18766 (gst_audio_test_src_create_white_noise):
18767 * gst/videotestsrc/gstvideotestsrc.c:
18768 * gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
18769 VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
18770 volume_sink_template, volume_src_template, gst_volume_init,
18771 volume_process_double, volume_process_int16,
18772 volume_process_int16_clamp):
18773 Doc fixes and formatting.
18775 2007-05-04 12:41:21 +0000 Tim-Philipp Müller <tim@centricular.net>
18777 tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
18778 Original commit message from CVS:
18779 * tests/check/Makefile.am:
18780 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
18781 Minimal check for volume's GstController usability; also another
18784 2007-05-04 09:06:38 +0000 Tim-Philipp Müller <tim@centricular.net>
18786 gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
18787 Original commit message from CVS:
18788 * gst-libs/gst/cdda/gstcddabasesrc.c:
18789 (gst_cdda_base_src_add_track):
18790 Fix it so that it (a) makes sense and (b) doesn't break
18791 everything cdda-related including the unit test.
18793 2007-05-04 08:46:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18795 gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
18796 Original commit message from CVS:
18797 * gst-libs/gst/cdda/gstcddabasesrc.c:
18798 (gst_cdda_base_src_add_track):
18799 Fix build when disabling asserts.
18801 2007-05-03 16:29:10 +0000 Tim-Philipp Müller <tim@centricular.net>
18803 sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
18804 Original commit message from CVS:
18805 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
18806 When XShm is not available, we might get row strides that are not
18807 rounded up to multiples of four; this is bad, because virtually
18808 every RGB-processing element in GStreamer assumes rowstrides are
18809 rounded up to multiples of four, so let's allocate at least enough
18810 memory to avoid crashes in this case. The image will still be
18811 displayed distorted though if this happens, so that still needs
18812 fixing (maybe by allocating a bigger image with an 'even' width
18813 and then clipping it appropriately when rendering - something for
18814 Xlib aficionados in any case).
18816 2007-05-03 13:16:21 +0000 Michael Smith <msmith@xiph.org>
18818 gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
18819 Original commit message from CVS:
18820 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
18821 If a buffer doesn't have a timestamp, assume it's contiguous with
18822 the previous buffer, and synthesise timestamps appropriately.
18824 2007-05-03 11:24:00 +0000 Edward Hervey <bilboed@bilboed.com>
18826 tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
18827 Original commit message from CVS:
18828 * tests/check/elements/videorate.c: (GST_START_TEST):
18829 Set buffer timestamp to a valid value in order to test the buffer
18830 really does stay in videorate.
18832 2007-05-03 10:47:22 +0000 Edward Hervey <bilboed@bilboed.com>
18834 gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
18835 Original commit message from CVS:
18836 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
18837 There is no sensible way to handle incoming buffers which don't have a
18838 valid timestamp. We therefore discard them and wait for the next one.
18840 2007-05-01 18:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
18842 gst/playback/: Better error message for text files.
18843 Original commit message from CVS:
18844 * gst/playback/gstdecodebin.c: (type_found), (plugin_init):
18845 * gst/playback/gstdecodebin2.c: (plugin_init):
18846 Better error message for text files.
18848 2007-04-29 14:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
18850 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
18851 Original commit message from CVS:
18852 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
18853 Fix offset bug in generation RR packets.
18855 2007-04-27 15:33:46 +0000 Julien Moutte <julien@moutte.net>
18857 ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
18858 Original commit message from CVS:
18859 2007-04-27 Julien MOUTTE <julien@moutte.net>
18860 * ext/theora/theoradec.c: (_theora_granule_time),
18861 (theora_dec_push_forward), (theora_handle_data_packet),
18862 (theora_dec_decode_buffer): Calculate buffer duration correctly
18863 to generate a perfect stream (#433888).
18864 * gst/audioresample/gstaudioresample.c:
18865 (audioresample_check_discont): Glib provides ABS.
18867 2007-04-27 15:01:40 +0000 Wim Taymans <wim.taymans@gmail.com>
18869 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
18870 Original commit message from CVS:
18871 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
18872 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
18873 (gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
18874 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
18875 (gst_rtcp_packet_bye_set_reason):
18876 * gst-libs/gst/rtp/gstrtcpbuffer.h:
18877 Fix RB block parsing and writing.
18878 Add support for constructing BYE packets.
18880 2007-04-25 08:54:34 +0000 Tim-Philipp Müller <tim@centricular.net>
18882 When posting a warning message because samples were dropped, post something more intelligible than he default error m...
18883 Original commit message from CVS:
18884 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
18885 (gst_base_audio_src_create):
18887 When posting a warning message because samples were dropped, post
18888 something more intelligible than he default error message for clock
18889 errors which is just confusing in this context (#432984).
18891 2007-04-25 08:10:26 +0000 Wim Taymans <wim.taymans@gmail.com>
18893 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
18894 Original commit message from CVS:
18895 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
18896 (gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
18897 (read_packet_header), (gst_rtcp_packet_move_to_next),
18898 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
18899 (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
18900 (gst_rtcp_packet_sdes_get_item_count),
18901 (gst_rtcp_packet_sdes_first_item),
18902 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
18903 (gst_rtcp_packet_sdes_first_entry),
18904 (gst_rtcp_packet_sdes_next_entry),
18905 (gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
18906 (gst_rtcp_packet_sdes_add_entry):
18907 * gst-libs/gst/rtp/gstrtcpbuffer.h:
18908 Implement code to write SR, RR and SDES packets.
18910 2007-04-24 20:45:24 +0000 Christian Kirbach <Christian.Kirbach@googlemail.com>
18912 sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
18913 Original commit message from CVS:
18914 Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
18915 * sys/ximage/ximagesink.c:
18916 Fix build if XShm is not available (#432362).
18918 2007-04-24 18:58:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18920 gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
18921 Original commit message from CVS:
18922 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
18923 Initalize the AudioConvertCtx with zeroes, otherwise it will contain
18924 pointers to random memory which are passed to g_free() when
18925 audio_convert_prepare_context() is called the first time.
18927 2007-04-24 15:00:07 +0000 Dan Williams <dcbw@redhat.com>
18929 gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
18930 Original commit message from CVS:
18931 Patch by: Dan Williams <dcbw redhat com>
18932 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
18933 Don't leak incoming buffer if gst_pad_push() returns a
18934 non-OK flow. Fixes #432755.
18935 * tests/check/elements/videorate.c: (GST_START_TEST),
18937 Unit test for the above by Yours Truly.
18939 2007-04-23 20:04:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18941 gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
18942 Original commit message from CVS:
18943 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
18944 (gst_adder_sink_event), (gst_adder_collected):
18945 Fix non-flushing segmented seeks, Fixes #340060 for me
18947 2007-04-21 15:29:27 +0000 Tim-Philipp Müller <tim@centricular.net>
18950 ChangeLog surgery: add API keyword
18951 Original commit message from CVS:
18952 ChangeLog surgery: add API keyword
18954 2007-04-21 15:25:22 +0000 Olivier Crete <tester@tester.ca>
18956 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
18957 Original commit message from CVS:
18958 Patch by: Olivier Crete <tester at tester ca>
18959 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18960 (gst_base_rtp_audio_payload_class_init),
18961 (gst_base_rtp_audio_payload_init),
18962 (gst_base_rtp_audio_payload_dispose):
18963 Chain up to parent class in dispose function; get rid of
18964 unnecessary 'diposed' flag in private structure (#415001).
18966 2007-04-21 15:10:25 +0000 Tim-Philipp Müller <tim@centricular.net>
18968 Some minor docs fixes and additions; also add missing 'Since' bits.
18969 Original commit message from CVS:
18970 * docs/libs/gst-plugins-base-libs.types:
18971 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18972 (gst_base_rtp_audio_payload_class_init):
18973 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18974 * gst-libs/gst/rtp/gstbasertppayload.c:
18975 Some minor docs fixes and additions; also add missing 'Since' bits.
18977 2007-04-21 14:40:45 +0000 Zeeshan Ali <zeenix@gmail.com>
18979 gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
18980 Original commit message from CVS:
18981 Patch by: Zeeshan Ali <zeenix gmail com>
18982 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18983 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
18984 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
18985 (gst_base_rtp_audio_payload_push):
18986 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
18987 The recently-added gst_base_rtp_audio_payload_push() should take an
18988 object of type GstBaseRTPAudioPayload as first argument (#431672).
18990 2007-04-21 14:14:24 +0000 Tim-Philipp Müller <tim@centricular.net>
18992 gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
18993 Original commit message from CVS:
18994 * gst/audioresample/gstaudioresample.c:
18995 Make more functions static, just because we can.
18997 2007-04-21 13:54:39 +0000 Tim-Philipp Müller <tim@centricular.net>
18999 tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
19000 Original commit message from CVS:
19001 * tests/check/elements/audioresample.c:
19002 Add unit test for audioresample shutdown crasher (#420106).
19004 2007-04-20 10:42:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19006 gst/subparse/: Use GST_DISABLE_XML here
19007 Original commit message from CVS:
19008 * gst/subparse/gstsubparse.c:
19009 * gst/subparse/samiparse.c:
19010 Use GST_DISABLE_XML here
19011 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
19012 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
19013 (gst_xvimagesink_buffer_alloc),
19014 (gst_xvimagesink_navigation_send_event):
19015 * sys/xvimage/xvimagesink.h:
19016 Include stdlib.h when using atoi.
19017 * tests/check/elements/playbin.c: (playbin_suite):
19018 Use GST_DISABLE_REGISTRY here
19020 2007-04-19 16:58:53 +0000 Michael Smith <msmith@xiph.org>
19022 ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
19023 Original commit message from CVS:
19024 * ext/theora/gsttheoraenc.h:
19025 * ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
19026 (theora_enc_sink_event), (theora_enc_change_state):
19027 Track initialisation state; don't try to use encoder state if we're
19028 not initialised (it'll segfault).
19030 2007-04-18 11:06:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19032 tests/check/pipelines/.cvsignore: Fix build.
19033 Original commit message from CVS:
19034 * tests/check/pipelines/.cvsignore:
19037 2007-04-17 10:56:37 +0000 Tim-Philipp Müller <tim@centricular.net>
19039 gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
19040 Original commit message from CVS:
19041 * gst/app/Makefile.am:
19042 Fix CFLAGS and hopefully #430594.
19044 2007-04-17 02:53:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19046 gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
19047 Original commit message from CVS:
19048 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19049 Allow random depths between 1 and 32 instead of only multiplies of 8.
19051 2007-04-17 02:04:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19053 gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
19054 Original commit message from CVS:
19055 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19056 Set the maximum number of channels for PCM and float in the correct
19057 place to have it also used when creating the template caps.
19059 2007-04-17 01:56:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19061 gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
19062 Original commit message from CVS:
19063 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19064 Correctly support 4, 6 and 8 channels with normal PCM and float
19066 Fix the depth and signedness calculation in extensible wav files and
19067 also handle 1, 2, 4, 6, 8 channels here when a file without channel
19069 Add support for float, alaw and mulaw in extensible wav files.
19070 This allows correct playback of all but 5 files from
19071 http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
19072 (gst_riff_create_audio_template_caps):
19073 Add voxware and float formats to the template caps.
19075 2007-04-16 22:20:03 +0000 Vincent Torri <vtorri@univ-evry.fr>
19077 ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
19078 Original commit message from CVS:
19079 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
19080 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
19081 Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
19082 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19083 * gst/audioresample/gstaudioresample.c: (audioresample_do_output):
19084 Use the correct format strings for integer formats.
19086 2007-04-16 21:44:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19088 * gst-plugins-base.doap:
19090 Original commit message from CVS:
19093 2007-04-16 21:42:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19095 * gst-plugins-base.doap:
19097 Original commit message from CVS:
19100 2007-04-15 14:35:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19102 ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
19103 Original commit message from CVS:
19104 * ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
19105 Don't use pad_alloc_buffer_and_set_caps to create a small header
19106 packet, or, worse, to create a big temporary video buffer using the
19109 2007-04-14 12:34:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19111 gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19112 Original commit message from CVS:
19113 * gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
19114 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19115 GST_START_TEST, buffer_probe_cb, GST_START_TEST):
19116 Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
19118 2007-04-13 22:10:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19120 * gst/tcp/gstmultifdsink.c:
19122 Original commit message from CVS:
19125 2007-04-13 21:55:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19128 * tests/check/pipelines/streamheader.c:
19129 tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19130 Original commit message from CVS:
19131 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19132 GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
19133 streamheader_suite):
19134 Add another test set up for failure
19136 2007-04-13 21:09:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19138 * ext/ogg/gstoggmux.c:
19139 * gst/gdp/gstgdpdepay.c:
19141 Original commit message from CVS:
19144 2007-04-13 21:08:11 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19146 tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
19147 Original commit message from CVS:
19148 * tests/check/Makefile.am:
19149 * tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
19150 GST_START_TEST, streamheader_suite, main):
19151 Add a test for the streamheader bug Wim fixed.
19153 2007-04-13 11:42:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19155 ext/theora/theoradec.c: Fix misleading comment.
19156 Original commit message from CVS:
19157 * ext/theora/theoradec.c: (theora_dec_sink_event):
19158 Fix misleading comment.
19160 2007-04-13 06:17:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19162 gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
19163 Original commit message from CVS:
19164 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19165 More sanity checks for the header fields.
19167 2007-04-12 16:36:36 +0000 Tim-Philipp Müller <tim@centricular.net>
19169 gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
19170 Original commit message from CVS:
19171 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
19172 Try encodings from all environment variables, not just those in the
19173 first environment variable that is set.
19175 2007-04-12 15:00:03 +0000 Wim Taymans <wim.taymans@gmail.com>
19177 gst/videorate/gstvideorate.c: Add some debug.
19178 Original commit message from CVS:
19179 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
19180 (gst_video_rate_chain):
19182 * tests/check/elements/videorate.c: (GST_START_TEST),
19184 Added check for videorate changing caps handling. Closes #421834.
19186 2007-04-12 12:57:33 +0000 Michael Smith <msmith@xiph.org>
19188 ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
19189 Original commit message from CVS:
19190 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
19191 Use scale functions to avoid overflow when calculating duration of
19194 2007-04-12 12:19:20 +0000 Tim-Philipp Müller <tim@centricular.net>
19196 API: add gst_tag_freeform_string_to_utf8() (#405072).
19197 Original commit message from CVS:
19198 * docs/libs/gst-plugins-base-libs-sections.txt:
19199 * gst-libs/gst/tag/tag.h:
19200 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
19201 API: add gst_tag_freeform_string_to_utf8() (#405072).
19202 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
19203 Use gst_tag_freeform_string_to_utf8() here.
19205 2007-04-12 10:38:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19207 * gst/tcp/gstmultifdsink.c:
19209 Original commit message from CVS:
19212 2007-04-12 10:03:22 +0000 Wim Taymans <wim.taymans@gmail.com>
19214 gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
19215 Original commit message from CVS:
19216 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
19217 (gst_gdp_pay_sink_event):
19218 Make sure we set the IN_CAPS flag correctly.
19219 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
19220 Get the IN_CAPS flag before we call functions that mess with the flags.
19222 2007-04-10 20:37:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19225 * gst/gdp/gstgdppay.c:
19226 gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
19227 Original commit message from CVS:
19228 * gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
19229 gst_gdp_pay_chain, gst_gdp_pay_sink_event):
19230 Only stamp buffers with offset/offset_end right before they get
19231 pushed. This ensures offset continuity, which was not the case
19233 gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
19235 2007-04-10 20:25:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19237 * gst/gdp/gstgdpdepay.c:
19238 * gst/gdp/gstgdppay.c:
19240 Original commit message from CVS:
19243 2007-04-10 11:23:18 +0000 Christian Schaller <uraeus@gnome.org>
19246 * gst-plugins-base.spec.in:
19247 update spec file for RTP changes
19248 Original commit message from CVS:
19249 update spec file for RTP changes
19251 2007-04-06 12:58:06 +0000 Wim Taymans <wim.taymans@gmail.com>
19253 gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
19254 Original commit message from CVS:
19255 * gst/playback/gstplaybin.c: (add_sink),
19256 (gst_play_bin_change_state):
19257 Activate sync in playbin, we are ready to handle it for live streams.
19259 2007-04-06 09:56:18 +0000 Tim-Philipp Müller <tim@centricular.net>
19261 tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
19262 Original commit message from CVS:
19263 * tests/check/elements/playbin.c:
19264 (test_sink_usage_video_only_stream), (playbin_suite):
19265 Add small test for stream-info-value-array code paths.
19267 2007-04-05 15:44:40 +0000 Wim Taymans <wim.taymans@gmail.com>
19269 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
19270 Original commit message from CVS:
19271 * gst-libs/gst/audio/gstbaseaudiosink.c:
19272 (gst_base_audio_sink_skew_slaving):
19273 Don't try to create invalid calibration parameters by making the
19274 internal time go backwards, instead make external time go forward.
19276 2007-04-05 10:27:06 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
19278 gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
19279 Original commit message from CVS:
19280 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
19281 * gst/playback/gstplaybasebin.c: (add_stream):
19282 Fix leak in add_stream(), when g_value_set_object() increases the
19283 refcount of streaminfo object. Fixes #426250.
19285 2007-04-04 02:45:03 +0000 David Schleef <ds@schleef.org>
19287 gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
19288 Original commit message from CVS:
19289 * gst/videotestsrc/gstvideotestsrc.c:
19290 * gst/videotestsrc/gstvideotestsrc.h:
19291 * gst/videotestsrc/videotestsrc.c:
19292 * gst/videotestsrc/videotestsrc.h:
19293 Add a test pattern called "circular", which has concentric
19294 rings with varying radial frequency. The main purpose of this
19295 pattern is to test fidelity loss in a filter or scaler element.
19296 Notably, this pattern is scale invariant, and is optimally viewed
19297 with a width (and height) of 400.
19299 2007-04-03 11:10:52 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
19301 gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
19302 Original commit message from CVS:
19303 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
19304 * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
19305 (deactivate_free_recursive):
19306 Decodebin2 doesn't unref pads it obtains in some occasions:
19307 - multiqueue src pads, when either connecting further or exposing
19308 - sink pads of new autoplugged elements
19309 - peer pads when recursively freeing elements
19312 2007-03-30 17:05:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19314 gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
19315 Original commit message from CVS:
19316 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19317 Add audio/x-raw-float support, now that audioconvert support
19318 non-native endianness floats.
19320 2007-03-30 15:00:49 +0000 Tim-Philipp Müller <tim@centricular.net>
19322 docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
19323 Original commit message from CVS:
19324 * docs/libs/gst-plugins-base-libs-docs.sgml:
19325 gstreamer-plugins-base.pc doesn't exist, it's
19326 gstreamer-plugins-base-0.10.pc.
19328 2007-03-29 18:42:34 +0000 René Stadler <mail@renestadler.de>
19330 with some minor changes
19331 Original commit message from CVS:
19332 Patch by: René Stadler <mail at renestadler dot de>
19333 with some minor changes
19334 * gst-libs/gst/floatcast/floatcast.h:
19335 Use more efficient float endianness conversion functions that don't
19336 involve 2 function calls per value.
19337 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
19338 (check_default), (audio_convert_prepare_context):
19339 * gst/audioconvert/gstaudioconvert.c:
19340 (gst_audio_convert_parse_caps), (make_lossless_changes):
19341 Support non-native endianness floats as input and output.
19343 * tests/check/elements/audioconvert.c: (verify_convert),
19345 Add unit tests for the non-native endianness float conversions.
19347 2007-03-29 16:23:53 +0000 Wim Taymans <wim.taymans@gmail.com>
19349 gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
19350 Original commit message from CVS:
19351 * gst-libs/gst/rtp/gstbasertpdepayload.c:
19352 (gst_base_rtp_depayload_base_init),
19353 (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
19354 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
19355 (gst_base_rtp_depayload_set_gst_timestamp),
19356 (gst_base_rtp_depayload_change_state),
19357 (gst_base_rtp_depayload_set_property),
19358 (gst_base_rtp_depayload_get_property):
19359 * gst-libs/gst/rtp/gstbasertpdepayload.h:
19360 Add Private structure.
19361 Bring element code to 2007.
19362 Parse clock-base caps param and use it when generating the
19364 Reset variables before going to PAUSED.
19367 2007-03-29 16:20:31 +0000 Wim Taymans <wim.taymans@gmail.com>
19370 Original commit message from CVS:
19371 * docs/libs/gst-plugins-base-libs-docs.sgml:
19372 * docs/libs/gst-plugins-base-libs-sections.txt:
19373 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19374 (gst_base_rtp_audio_payload_get_adapter):
19376 Fix some more docs.
19377 * gst-libs/gst/rtp/Makefile.am:
19378 * gst-libs/gst/rtp/gstrtcpbuffer.c:
19379 (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
19380 (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
19381 (gst_rtcp_buffer_get_packet_count), (read_packet_header),
19382 (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
19383 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
19384 (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
19385 (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
19386 (gst_rtcp_packet_sr_get_sender_info),
19387 (gst_rtcp_packet_sr_set_sender_info),
19388 (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
19389 (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
19390 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
19391 (gst_rtcp_packet_sdes_get_chunk_count),
19392 (gst_rtcp_packet_sdes_first_chunk),
19393 (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
19394 (gst_rtcp_packet_sdes_first_item),
19395 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
19396 (gst_rtcp_packet_bye_get_ssrc_count),
19397 (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
19398 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
19399 (gst_rtcp_packet_bye_get_reason_len),
19400 (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
19401 * gst-libs/gst/rtp/gstrtcpbuffer.h:
19402 Add new helper object for parsing and creating RTCP messages.
19404 2007-03-29 12:07:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19406 gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
19407 Original commit message from CVS:
19408 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19409 PCM samples with width=8 must be always unsigned, no matter what
19412 2007-03-29 11:24:47 +0000 Andy Wingo <wingo@pobox.com>
19414 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
19415 Original commit message from CVS:
19416 2007-03-29 Andy Wingo <wingo@pobox.com>
19417 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
19418 perfect offsets also, not just timestamps.
19419 * tests/check/elements/videorate.c (test_more): Test that given
19420 any incoming offsets, that videorate produces perfect offsets.
19422 2007-03-29 10:19:45 +0000 Wim Taymans <wim.taymans@gmail.com>
19424 gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
19425 Original commit message from CVS:
19426 * gst-libs/gst/riff/riff-ids.h:
19427 Add some more RIFF formats.
19429 2007-03-29 10:17:52 +0000 Wim Taymans <wim.taymans@gmail.com>
19431 gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
19432 Original commit message from CVS:
19433 * gst-libs/gst/rtp/gstrtpbuffer.c:
19434 (gst_rtp_buffer_default_clock_rate):
19435 * gst-libs/gst/rtp/gstrtpbuffer.h:
19436 Fix fixed payload names and docs.
19437 Added method to get the default clock rates of fixed payload types.
19438 API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
19440 2007-03-28 15:24:40 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
19442 tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
19443 Original commit message from CVS:
19444 * tests/check/pipelines/.cvsignore:
19445 Add new vorbisdec test to cvsignore.
19447 2007-03-28 14:50:47 +0000 Wim Taymans <wim.taymans@gmail.com>
19449 gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
19450 Original commit message from CVS:
19451 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
19452 (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
19453 (gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
19454 (gst_base_audio_sink_set_property),
19455 (gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
19456 (clock_convert_external), (gst_base_audio_sink_resample_slaving),
19457 (gst_base_audio_sink_skew_slaving),
19458 (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
19459 (gst_base_audio_sink_async_play):
19460 * gst-libs/gst/audio/gstbaseaudiosink.h:
19461 Store private stuff in GstBaseAudioSinkPrivate.
19462 Add configurable clock slaving modes property.
19463 API:: GstBaseAudioSink::slave-method property
19464 Some more latency reporting tweaks.
19465 Added skew based clock slaving correction and make it the default until
19466 the resampling method is more robust.
19468 2007-03-27 12:44:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19470 gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
19471 Original commit message from CVS:
19472 * gst/audioconvert/audioconvert.c:
19473 Add docs to the integer pack functions and implement proper
19474 rounding. Before we had rounding towards negative infinity, i.e.
19475 always the smaller number was taken. Now we use natural rounding,
19476 i.e. rounding to the nearest integer and to the one with the largest
19477 absolute value for X.5. The old rounding introduced some minor
19478 distortions. Fixes #420079
19479 * tests/check/elements/audioconvert.c: (GST_START_TEST):
19480 Fix one unit test that assumed the old rounding and added unit tests
19481 for checking signed/unsigned int16 <-> signed/unsigned int16 with
19482 depth 8, one for signed int16 <-> unsigned int16 and one for the new
19483 rounding from signed int32 to signed/unsigned int16.
19485 2007-03-27 11:31:17 +0000 Michael Smith <msmith@xiph.org>
19487 gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
19488 Original commit message from CVS:
19489 * gst/audioconvert/gstaudioconvert.c: (strip_width_64),
19490 (gst_audio_convert_transform_caps):
19491 Fix typo in debug line introduced recently, as pointed out on irc.
19493 2007-03-27 10:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
19495 Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
19496 Original commit message from CVS:
19497 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
19498 * tests/check/libs/tag.c: (GST_START_TEST):
19499 Make sure we parse floating-point numbers in vorbis comments
19500 correctly with either '.' or ',' as separator, no matter what
19501 the current locale is. Add unit test for this too.
19503 2007-03-27 09:37:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19505 * tests/check/pipelines/vorbisdec.c:
19507 Original commit message from CVS:
19510 2007-03-26 22:38:19 +0000 René Stadler <mail@renestadler.de>
19512 gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
19513 Original commit message from CVS:
19514 Patch by: René Stadler <mail at renestadler de>
19515 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
19516 When writing out floating-point numbers to vorbis comment tags, always
19517 use the same character as separator no matter what the current locale is
19519 * tests/check/libs/tag.c: (GST_START_TEST):
19520 Add unit tests for replaygain tags in vorbis comments (closes #423055).
19522 2007-03-26 20:56:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19524 ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
19525 Original commit message from CVS:
19526 * ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
19527 vorbis_handle_data_packet):
19528 Correctly set DURATION to generate a timestamp-continuous stream.
19529 One bug left at the end; see
19530 ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
19531 * tests/check/Makefile.am:
19532 * tests/check/pipelines/vorbisenc.c (GST_START_TEST):
19533 Add a test to check this. Without the above patch this test fails.
19535 2007-03-26 11:44:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19537 gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
19538 Original commit message from CVS:
19539 * gst-libs/gst/rtp/Makefile.am:
19540 The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
19542 2007-03-23 15:43:24 +0000 Christian Schaller <uraeus@gnome.org>
19544 * gst-plugins-base.spec.in:
19546 Original commit message from CVS:
19549 2007-03-23 12:32:33 +0000 Michael Smith <msmith@xiph.org>
19551 gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
19552 Original commit message from CVS:
19553 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
19554 (gst_video_rate_reset), (gst_video_rate_chain):
19555 If videorate changes caps, we can no longer use the old buffer
19556 (which may have a different size, incompatible with our caps).
19557 So don't do that; just duplicate the new frame more times.
19559 2007-03-22 17:43:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19561 gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
19562 Original commit message from CVS:
19563 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
19564 Remove playbin's override of the set_clock vmethod. It's irrelevant
19565 after Wim's commit on the 19th.
19567 2007-03-22 14:37:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19569 gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
19570 Original commit message from CVS:
19571 * gst-libs/gst/app/Makefile.am:
19572 Use GST_ALL_LDFLAGS, which actually exists, but maybe David
19573 can confirm that was what he wanted.
19575 2007-03-22 09:26:02 +0000 Wim Taymans <wim.taymans@gmail.com>
19577 ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
19578 Original commit message from CVS:
19579 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
19580 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
19581 * ext/gnomevfs/gstgnomevfssrc.h:
19582 Don't cache file sizes. Fixes #341078.
19584 2007-03-21 11:03:23 +0000 Tim-Philipp Müller <tim@centricular.net>
19586 gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
19587 Original commit message from CVS:
19588 * gst/playback/gstplaybin.c: (add_sink):
19589 Use GST_PTR_FORMAT to log caps.
19591 2007-03-21 10:23:11 +0000 Young-Ho Cha <ganadist@chollian.net>
19593 gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
19594 Original commit message from CVS:
19595 Patch by: Young-Ho Cha <ganadist at chollian net>
19596 * gst/subparse/samiparse.c: (handle_start_font):
19597 Special-case some more colour names that pango doesn't handle by
19598 default. Fixes #420578.
19600 2007-03-20 11:49:55 +0000 Michael Smith <msmith@xiph.org>
19602 ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
19603 Original commit message from CVS:
19604 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
19605 If we get a zero-sized input buffer, don't pass it to libvorbis, as
19606 that marks EOS internally. After that, libvorbis will buffer all
19607 input data, and encode none of it, eventually leading to memory
19610 2007-03-19 10:52:50 +0000 Wim Taymans <wim.taymans@gmail.com>
19612 gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
19613 Original commit message from CVS:
19614 * gst/playback/gstdecodebin.c: (remove_fakesink):
19615 Don't post STATE_DIRTY anymore.
19616 * gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
19617 (gst_play_bin_change_state):
19618 Remove stream_time reset in seek handling, core does that now.
19619 Disable clocking for live pipelines by forcing a NULL clock to the
19620 complete pipeline, core is too smart now for our previous hack.
19621 We can always autoplug in PAUSED now.
19623 2007-03-18 03:14:01 +0000 David Schleef <ds@schleef.org>
19625 REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
19626 Original commit message from CVS:
19627 * REQUIREMENTS: Update this file, change the formatting to make
19628 it more consistent, plus more machine readable.
19630 2007-03-16 17:29:09 +0000 Michael Smith <msmith@xiph.org>
19632 gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
19633 Original commit message from CVS:
19634 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
19635 (strip_width_64), (append_with_other_format):
19636 Previous fix was too simplistic, and broke the tests. Use a better
19637 approach; only strip 64 from widths for integer audio.
19639 2007-03-16 16:42:23 +0000 Michael Smith <msmith@xiph.org>
19641 gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
19642 Original commit message from CVS:
19643 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
19644 (gst_audio_convert_transform_caps):
19645 We don't support 64 bit integer audio, so don't try to claim we can.
19646 Stops us producing caps don't match our template caps.
19649 2007-03-15 10:52:21 +0000 Michael Smith <msmith@xiph.org>
19651 gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
19652 Original commit message from CVS:
19653 * gst/audioresample/gstaudioresample.c:
19654 (audioresample_check_discont), (audioresample_transform):
19655 Don't trigger discontinuities for very small imperfections; a filter
19656 flush will sound bad, and many plugins have rounding errors leading
19659 2007-03-14 21:11:18 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
19661 gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
19662 Original commit message from CVS:
19663 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19664 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
19665 Add min-ptime property to RTP base audio payloader. Patch by
19666 olivier.crete@collabora.co.uk.
19668 Indentation/whitespace/documentation fixes.
19670 2007-03-14 17:16:30 +0000 Julien Moutte <julien@moutte.net>
19672 gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
19673 Original commit message from CVS:
19674 2007-03-14 Julien MOUTTE <julien@moutte.net>
19675 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
19676 (audioresample_transform_size), (audioresample_do_output),
19677 (audioresample_transform), (audioresample_pushthrough): Handle
19678 discontinuous streams.
19679 * gst/audioresample/gstaudioresample.h:
19680 * tests/check/elements/audioresample.c:
19681 (test_discont_stream_instance), (GST_START_TEST),
19682 (audioresample_suite): Add a test for discontinuous streams.
19683 * win32/common/config.h: Updated.
19685 2007-03-14 15:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19687 po/: Update translations from translation project.
19688 Original commit message from CVS:
19702 Update translations from translation project.
19704 2007-03-14 15:05:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19706 * gst/gdp/gstgdpdepay.c:
19708 Original commit message from CVS:
19711 2007-03-14 14:48:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19713 gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
19714 Original commit message from CVS:
19715 * gst/audioresample/debug.h:
19716 * gst/audioresample/resample.c: (resample_init):
19717 Since I really am not interested in a debug line for each sample
19718 being processed, move the library's debugging to its own category,
19721 2007-03-14 14:09:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19723 * gst/audioresample/gstaudioresample.c:
19724 add debugging and reformat docs
19725 Original commit message from CVS:
19726 add debugging and reformat docs
19728 2007-03-12 23:29:07 +0000 Michael Smith <msmith@xiph.org>
19730 ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
19731 Original commit message from CVS:
19732 * ext/theora/theoradec.c: (theora_handle_type_packet):
19733 Since the plugin doesn't support anything other than 4:2:0 right
19734 now, post an error and fail if we get something else. Won't matter
19735 until libtheora supports the other pixel formats, but hopefully
19738 2007-03-12 15:50:35 +0000 Alex Lancaster <alexlan@fedoraproject.org>
19741 I'm too lazy to comment this
19742 Original commit message from CVS:
19743 Mention Patch by: Alex Lancaster in a recent commit.
19745 2007-03-12 11:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19747 examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.
19748 Original commit message from CVS:
19749 * examples/app/.cvsignore:
19750 The buildbot demands .cvsignore files, and I comply.
19752 2007-03-11 00:48:26 +0000 David Schleef <ds@schleef.org>
19754 Add appsrc/appsink example.
19755 Original commit message from CVS:
19757 * examples/Makefile.am:
19758 * examples/app/Makefile.am:
19759 * examples/app/appsrc_ex.c:
19760 Add appsrc/appsink example.
19761 * gst-libs/gst/app/Makefile.am:
19762 * gst-libs/gst/app/gstapp.c:
19763 * gst-libs/gst/app/gstappsink.c:
19764 * gst-libs/gst/app/gstappsink.h:
19765 * gst/app/gstapp.c:
19768 2007-03-10 15:59:33 +0000 Sébastien Moutte <sebastien@moutte.net>
19770 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
19771 Original commit message from CVS:
19772 * gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
19773 Use gst_guint64_to_gdouble for conversion.
19775 Add new files to the win32 MANIFEST.
19776 * win32/common/libgstaudio.def:
19777 * win32/common/libgstpbutils.def:
19778 Add new exported functions.
19779 * win32/vs6/gst_plugins_base.dsw:
19780 * win32/vs6/libgstdecodebin.dsp:
19781 * win32/vs6/libgstplaybin.dsp:
19782 Change the link to libgstpbutils.lib.
19783 * win32/vs6/libgstdecodebin2.dsp:
19784 Add a new project for decodebin2.
19785 * win32/vs6/libgstpbutils.dsp:
19786 Add a new project for pbutils.
19788 2007-03-10 12:18:58 +0000 Tim-Philipp Müller <tim@centricular.net>
19790 gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
19791 Original commit message from CVS:
19792 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
19793 Also accept partial dates with only year and month,
19794 like 1999-12-00 (fixes #410396 even more).
19795 * tests/check/libs/tag.c: (GST_START_TEST):
19796 Add unit test for the above.
19798 2007-03-10 11:21:08 +0000 Tim-Philipp Müller <tim@centricular.net>
19800 tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
19801 Original commit message from CVS:
19802 * tests/check/elements/subparse.c: (GST_START_TEST),
19804 Add unit test for MPL2 subtitle format (#413799).
19806 2007-03-10 11:17:52 +0000 Kamil Pawlowski <kamilpe@gmail.com>
19808 gst/subparse/: Add support for MPL2 subtitle format (#413799).
19809 Original commit message from CVS:
19810 Patch by: Kamil Pawlowski <kamilpe gmail com>
19811 * gst/subparse/Makefile.am:
19812 * gst/subparse/gstsubparse.c:
19813 (gst_sub_parse_data_format_autodetect),
19814 (gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
19815 (gst_subparse_type_find):
19816 * gst/subparse/gstsubparse.h:
19817 * gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
19818 * gst/subparse/mpl2parse.h:
19819 Add support for MPL2 subtitle format (#413799).
19821 2007-03-09 17:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
19823 configure.ac: We require core CVS for the new buffer metadata copy functions.
19824 Original commit message from CVS:
19826 We require core CVS for the new buffer metadata copy functions.
19828 2007-03-09 16:51:13 +0000 Wim Taymans <wim.taymans@gmail.com>
19830 gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
19831 Original commit message from CVS:
19832 * gst-libs/gst/tag/gstid3tag.c:
19833 Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
19836 2007-03-09 16:46:35 +0000 Wim Taymans <wim.taymans@gmail.com>
19838 ext/libvisual/visual.c: Improve adapter usage and comments.
19839 Original commit message from CVS:
19840 * ext/libvisual/visual.c: (gst_visual_sink_setcaps),
19841 (gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
19842 Improve adapter usage and comments.
19844 2007-03-09 16:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
19846 Use new metadata copy function.
19847 Original commit message from CVS:
19848 * ext/pango/gsttextrender.c: (gst_text_render_chain):
19849 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
19850 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
19851 Use new metadata copy function.
19852 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
19853 (gst_ffmpegcsp_transform):
19854 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
19855 Basetransform copied the metadata for us.
19857 2007-03-09 16:28:04 +0000 Tim-Philipp Müller <tim@centricular.net>
19859 ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
19860 Original commit message from CVS:
19861 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
19862 (gst_text_overlay_video_event):
19863 Some more logging. Only accept newsegment events in TIME format and
19864 send a WARNING message if they are not in TIME format.
19865 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
19866 (gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
19867 (gst_sub_parse_chain), (gst_sub_parse_sink_event):
19868 * gst/subparse/gstsubparse.h:
19869 No need to allocate GstSegment structure dynamically, just put it
19870 into the instance structure; ignore newsegment events in BYTE
19871 format and in particular don't let it overwrite our saved TIME
19872 segment from the last seek.
19874 2007-03-09 13:05:04 +0000 Michael Smith <msmith@xiph.org>
19876 gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
19877 Original commit message from CVS:
19878 * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
19879 Replace AC3 typefinder with one that isn't terrible, and actually
19882 2007-03-09 12:22:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19884 gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
19885 Original commit message from CVS:
19886 * gst/audioconvert/gstaudioconvert.c:
19887 (gst_audio_convert_transform):
19888 fix error category and translatable string
19890 2007-03-09 11:23:32 +0000 Tim-Philipp Müller <tim@centricular.net>
19892 pkgconfig/: Fix up utils => pbutils here too.
19893 Original commit message from CVS:
19894 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
19895 * pkgconfig/gstreamer-plugins-base.pc.in:
19896 Fix up utils => pbutils here too.
19898 2007-03-09 10:49:53 +0000 Tim-Philipp Müller <tim@centricular.net>
19900 gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
19901 Original commit message from CVS:
19902 * gst/subparse/gstsubparse.c: (handle_buffer):
19903 Break out of loop in chain function as soon as possible if we get
19904 a non-OK flow return.
19906 2007-03-08 18:26:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19908 tests/check/elements/alsa.c: Unref the mixer if the state change fails too (if the alsa devices are inaccessible, for...
19909 Original commit message from CVS:
19910 * tests/check/elements/alsa.c: (GST_START_TEST):
19911 Unref the mixer if the state change fails too (if the
19912 alsa devices are inaccessible, for example)
19914 2007-03-08 17:49:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19916 tests/check/Makefile.am: Don't test libvisual elements in the states check, because libvisual seems to leak internally.
19917 Original commit message from CVS:
19918 * tests/check/Makefile.am:
19919 Don't test libvisual elements in the states check, because libvisual
19920 seems to leak internally.
19921 Re-enable the alsa and states tests now that there's new suppressions
19923 * tests/check/elements/alsa.c: (GST_START_TEST):
19924 Don't leak the alsamixer we instantiated.
19926 2007-03-08 15:22:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19928 sys/: Move some cleanup stuff from the state change handler into a _reset() function that can be called from _finaliz...
19929 Original commit message from CVS:
19930 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
19931 (gst_ximagesink_change_state), (gst_ximagesink_reset),
19932 (gst_ximagesink_finalize):
19933 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
19934 (gst_xvimagesink_reset), (gst_xvimagesink_finalize):
19935 Move some cleanup stuff from the state change handler into a _reset()
19936 function that can be called from _finalize(). This ensures that things
19937 get freed even if (for some reason) the NULL->READY state transition
19938 fails in the parent class.
19939 Even if a parent state change fails, process our downward state change
19940 logic instead of bailing out early.
19941 Free the correct xcontext pointer in ximagesink's xcontext_clear.
19943 2007-03-08 12:53:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19945 ext/alsa/gstalsasink.c: Extra log line.
19946 Original commit message from CVS:
19947 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
19949 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
19950 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
19951 Use pango_font_description_set_family_static instead of
19952 pango_font_description_set_family to save a string copy (it was
19953 leaking due to the strdup anyway)
19954 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
19955 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
19956 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
19957 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
19958 Chain up in finalize.
19960 2007-03-07 18:50:10 +0000 Tim-Philipp Müller <tim@centricular.net>
19962 gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
19963 Original commit message from CVS:
19964 * gst-libs/gst/interfaces/mixertrack.c:
19965 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
19966 (gst_mixer_track_set_property):
19967 API: add "untranslated-label" property which should be set by
19968 implementations at construct time (#414645).
19969 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
19970 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
19971 Set "untranslated-label" when constructing mixer track objects.
19972 * tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
19973 Unit test to check the above.
19975 2007-03-07 17:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
19977 ext/ogg/gstoggdemux.c: Fix confusing debug message.
19978 Original commit message from CVS:
19979 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
19980 Fix confusing debug message.
19982 2007-03-07 17:12:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19984 gst-plugins-base.doap: update doap file with new version
19985 Original commit message from CVS:
19986 * gst-plugins-base.doap:
19987 update doap file with new version
19989 2007-03-07 17:05:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19991 * gst/tcp/gstmultifdsink.c:
19993 Original commit message from CVS:
19996 2007-03-07 16:56:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19998 configure.ac: Back to CVS
19999 Original commit message from CVS:
20003 === release 0.10.12 ===
20005 2007-03-07 16:46:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20011 * docs/plugins/gst-plugins-base-plugins.args:
20012 * docs/plugins/inspect/plugin-adder.xml:
20013 * docs/plugins/inspect/plugin-alsa.xml:
20014 * docs/plugins/inspect/plugin-audioconvert.xml:
20015 * docs/plugins/inspect/plugin-audiorate.xml:
20016 * docs/plugins/inspect/plugin-audioresample.xml:
20017 * docs/plugins/inspect/plugin-audiotestsrc.xml:
20018 * docs/plugins/inspect/plugin-cdparanoia.xml:
20019 * docs/plugins/inspect/plugin-decodebin.xml:
20020 * docs/plugins/inspect/plugin-decodebin2.xml:
20021 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
20022 * docs/plugins/inspect/plugin-gdp.xml:
20023 * docs/plugins/inspect/plugin-gnomevfs.xml:
20024 * docs/plugins/inspect/plugin-libvisual.xml:
20025 * docs/plugins/inspect/plugin-ogg.xml:
20026 * docs/plugins/inspect/plugin-pango.xml:
20027 * docs/plugins/inspect/plugin-playbin.xml:
20028 * docs/plugins/inspect/plugin-subparse.xml:
20029 * docs/plugins/inspect/plugin-tcp.xml:
20030 * docs/plugins/inspect/plugin-theora.xml:
20031 * docs/plugins/inspect/plugin-typefindfunctions.xml:
20032 * docs/plugins/inspect/plugin-video4linux.xml:
20033 * docs/plugins/inspect/plugin-videorate.xml:
20034 * docs/plugins/inspect/plugin-videoscale.xml:
20035 * docs/plugins/inspect/plugin-videotestsrc.xml:
20036 * docs/plugins/inspect/plugin-volume.xml:
20037 * docs/plugins/inspect/plugin-vorbis.xml:
20038 * docs/plugins/inspect/plugin-ximagesink.xml:
20039 * docs/plugins/inspect/plugin-xvimagesink.xml:
20040 * win32/common/config.h:
20042 Original commit message from CVS:
20045 2007-03-07 15:35:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20064 Original commit message from CVS:
20067 2007-03-06 12:31:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20069 configure.ac: Bump version to 0.10.11.4 pre-release
20070 Original commit message from CVS:
20072 Bump version to 0.10.11.4 pre-release
20074 2007-03-06 12:10:08 +0000 Wim Taymans <wim.taymans@gmail.com>
20076 gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
20077 Original commit message from CVS:
20078 * gst-libs/gst/audio/gstbaseaudiosink.c:
20079 (gst_base_audio_sink_async_play):
20080 Fix regression that made GStreamer skip the first samples of audio.
20083 2007-03-05 11:21:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20085 configure.ac: Bump version to 0.10.11.3 pre-release
20086 Original commit message from CVS:
20088 Bump version to 0.10.11.3 pre-release
20090 2007-03-05 09:35:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20092 po/POTFILES.in: Update paths for the rename from utils to pbutils to fix the build.
20093 Original commit message from CVS:
20095 Update paths for the rename from utils to pbutils to fix the build.
20097 2007-03-05 09:27:55 +0000 Tim-Philipp Müller <tim@centricular.net>
20099 gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
20100 Original commit message from CVS:
20101 * gst-libs/gst/pbutils/Makefile.am:
20102 Change directory to install headers in from gst/utils to gst/pbutils
20105 2007-03-04 23:41:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20107 * tests/check/libs/.gitignore:
20109 Original commit message from CVS:
20112 2007-03-04 23:41:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20114 * win32/common/config.h:
20115 * win32/common/libgstutils.def:
20117 Original commit message from CVS:
20120 2007-03-04 23:39:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20122 rename utils to pbutils
20123 Original commit message from CVS:
20125 * docs/libs/gst-plugins-base-libs-docs.sgml:
20126 * docs/libs/gst-plugins-base-libs-sections.txt:
20127 * gst-libs/gst/Makefile.am:
20128 * gst-libs/gst/interfaces/mixer.c:
20129 * gst-libs/gst/pbutils/Makefile.am:
20130 * gst-libs/gst/pbutils/descriptions.c:
20131 (gst_pb_utils_get_source_description),
20132 (gst_pb_utils_get_sink_description),
20133 (gst_pb_utils_get_decoder_description),
20134 (gst_pb_utils_get_encoder_description),
20135 (gst_pb_utils_get_element_description),
20136 (gst_pb_utils_add_codec_description_to_tag_list),
20137 (gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
20138 * gst-libs/gst/pbutils/descriptions.h:
20139 * gst-libs/gst/pbutils/install-plugins.c:
20140 * gst-libs/gst/pbutils/install-plugins.h:
20141 * gst-libs/gst/pbutils/missing-plugins.c:
20142 (gst_missing_uri_source_message_new),
20143 (gst_missing_uri_sink_message_new),
20144 (gst_missing_element_message_new),
20145 (gst_missing_decoder_message_new),
20146 (gst_missing_encoder_message_new),
20147 (gst_missing_plugin_message_get_description):
20148 * gst-libs/gst/pbutils/missing-plugins.h:
20149 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
20150 * gst-libs/gst/pbutils/pbutils.h:
20151 * gst-libs/gst/utils/Makefile.am:
20152 * gst-libs/gst/utils/base-utils.c:
20153 * gst-libs/gst/utils/base-utils.h:
20154 * gst-libs/gst/utils/descriptions.c:
20155 * gst-libs/gst/utils/descriptions.h:
20156 * gst-libs/gst/utils/install-plugins.c:
20157 * gst-libs/gst/utils/install-plugins.h:
20158 * gst-libs/gst/utils/missing-plugins.c:
20159 * gst-libs/gst/utils/missing-plugins.h:
20160 * gst-plugins-base.spec.in:
20161 * gst/playback/Makefile.am:
20162 * gst/playback/gstdecodebin.c:
20163 * gst/playback/gstdecodebin2.c:
20164 * gst/playback/gstplaybasebin.c: (setup_subtitle),
20165 (gen_source_element):
20166 * gst/playback/gstplaybin.c: (plugin_init):
20167 * tests/check/Makefile.am:
20168 * tests/check/libs/pbutils.c: (GST_START_TEST),
20169 (test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
20170 * tests/check/libs/utils.c:
20171 rename utils to pbutils
20173 2007-03-03 10:23:03 +0000 David Schleef <ds@schleef.org>
20175 gst-libs/gst/app/Makefile.am: Install the headers.
20176 Original commit message from CVS:
20177 * gst-libs/gst/app/Makefile.am:
20178 Install the headers.
20180 2007-03-03 10:10:30 +0000 David Schleef <ds@schleef.org>
20182 gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.
20183 Original commit message from CVS:
20184 * gst-libs/gst/app/Makefile.am:
20185 * gst-libs/gst/app/gstappbuffer.c:
20186 * gst-libs/gst/app/gstappbuffer.h:
20187 * gst-libs/gst/app/gstappsrc.c:
20188 Add GstAppBuffer that includes a callback and closure for
20189 proper handling of data chunks.
20191 2007-03-03 09:06:06 +0000 David Schleef <ds@schleef.org>
20193 gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.
20194 Original commit message from CVS:
20195 * gst-libs/gst/app/gstappsrc.c:
20196 * gst-libs/gst/app/gstappsrc.h:
20197 Hacking to address issues in 413418.
20199 2007-03-03 08:16:57 +0000 David Schleef <ds@schleef.org>
20201 Move the app library to gst-libs/gst/app (duh!)
20202 Original commit message from CVS:
20206 * gst-libs/gst/Makefile.am:
20207 * gst-libs/gst/app/Makefile.am:
20208 * gst-libs/gst/app/gstapp.c:
20209 * gst-libs/gst/app/gstappsrc.c:
20210 * gst-libs/gst/app/gstappsrc.h:
20211 * gst/app/Makefile.am:
20212 * gst/app/gstapp.c:
20213 * gst/app/gstappsrc.c:
20214 * gst/app/gstappsrc.h:
20215 Move the app library to gst-libs/gst/app (duh!)
20217 2007-03-02 12:59:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20219 Add documentation for decodebin2 that indicates that the API is still unstable.
20220 Original commit message from CVS:
20221 * docs/plugins/Makefile.am:
20222 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
20223 * docs/plugins/gst-plugins-base-plugins-sections.txt:
20224 * docs/plugins/inspect/plugin-decodebin2.xml:
20225 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
20226 Add documentation for decodebin2 that indicates that the API
20229 2007-03-01 18:50:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20231 configure.ac: Update to 0.10.11.2 (0.10.12 pre-release)
20232 Original commit message from CVS:
20234 Update to 0.10.11.2 (0.10.12 pre-release)
20236 2007-03-01 17:29:55 +0000 Wim Taymans <wim.taymans@gmail.com>
20238 gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
20239 Original commit message from CVS:
20240 * gst-libs/gst/audio/gstbaseaudiosink.c:
20241 (gst_base_audio_sink_async_play):
20242 base time is irrelevant here.
20244 2007-03-01 17:01:43 +0000 Wim Taymans <wim.taymans@gmail.com>
20246 gst-libs/gst/audio/: Improve debugging.
20247 Original commit message from CVS:
20248 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
20249 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
20251 * gst-libs/gst/audio/gstbaseaudiosink.c:
20252 (gst_base_audio_sink_query), (gst_base_audio_sink_event),
20253 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
20254 Improve latency and clock slaving calculations.
20255 Improve slave clock calibration.
20256 * gst-libs/gst/audio/gstringbuffer.c:
20257 (gst_ring_buffer_commit_full):
20258 When we are asked to render N sample to 0 bytes, return N.
20260 2007-03-01 16:48:45 +0000 Wim Taymans <wim.taymans@gmail.com>
20262 ext/alsa/gstalsasink.*: Remove unused dispose function.
20263 Original commit message from CVS:
20264 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
20265 (gst_alsasink_write), (gst_alsasink_reset):
20266 * ext/alsa/gstalsasink.h:
20267 Remove unused dispose function.
20268 Rename lock to not interfere with alsasrc lock.
20269 * ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
20270 (gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
20271 (gst_alsasrc_read), (gst_alsasrc_reset):
20272 * ext/alsa/gstalsasrc.h:
20273 Implement finalize function.
20274 Use lock to protect alsa access.
20276 Fine tune sw params.
20278 2007-03-01 10:20:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20283 Original commit message from CVS:
20286 2007-02-28 19:27:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20288 configure.ac: Convert to new AG_GST style.
20289 Original commit message from CVS:
20291 Convert to new AG_GST style.
20293 2007-02-28 15:17:20 +0000 Ed Catmur <ed@catmur.co.uk>
20295 gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
20296 Original commit message from CVS:
20297 Patch by: Ed Catmur <ed at catmur dot co dot uk>
20298 * gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
20299 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
20300 Fix race condition when rapidly switching visualisations in playbin.
20303 2007-02-28 15:11:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20305 tests/check/Makefile.am: Include local stuff before system installed things in LDFLAGS and
20306 Original commit message from CVS:
20307 * tests/check/Makefile.am:
20308 Include local stuff before system installed things in LDFLAGS and
20311 2007-02-28 15:10:06 +0000 Wim Taymans <wim.taymans@gmail.com>
20313 ext/ogg/gstoggdemux.c: Improve debugging.
20314 Original commit message from CVS:
20315 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
20318 2007-02-28 15:05:03 +0000 Wim Taymans <wim.taymans@gmail.com>
20320 sys/v4l/: Fix duration and timestamping, taking latency into account.
20321 Original commit message from CVS:
20322 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
20323 (gst_v4lsrc_fixate), (gst_v4lsrc_query):
20324 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
20325 Fix duration and timestamping, taking latency into account.
20326 Implement latency query.
20328 2007-02-28 15:02:25 +0000 Wim Taymans <wim.taymans@gmail.com>
20330 gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
20331 Original commit message from CVS:
20332 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
20333 (gst_audio_clock_new):
20335 * gst-libs/gst/audio/gstbaseaudiosink.c:
20336 (gst_base_audio_sink_init), (gst_base_audio_sink_query):
20337 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
20338 (gst_base_audio_src_query), (gst_base_audio_src_get_offset),
20339 (gst_base_audio_src_create):
20340 Improve latency query code.
20341 Use proper clock names.
20343 2007-02-28 12:57:46 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20345 * tests/check/generic/states.c:
20347 Original commit message from CVS:
20350 2007-02-28 12:44:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20352 tests/check/generic/states.c: Copy the states.c test from core again
20353 Original commit message from CVS:
20354 * tests/check/generic/states.c: (GST_START_TEST):
20355 Copy the states.c test from core again
20356 * tests/check/Makefile.am:
20357 ignore cdio and cdparanoiasrc
20359 2007-02-28 12:08:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20361 gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
20362 Original commit message from CVS:
20363 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
20364 (double_hq), (audio_convert_get_func_index), (check_default),
20365 (audio_convert_prepare_context), (audio_convert_convert):
20366 Also make valgrind happy and avoid copying data in some cases.
20368 2007-02-28 11:58:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20370 * tests/check/generic/states.c:
20372 Original commit message from CVS:
20375 2007-02-28 11:47:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20377 Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
20378 Original commit message from CVS:
20379 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
20380 (double_hq), (audio_convert_get_func_index),
20381 (audio_convert_prepare_context), (audio_convert_convert):
20382 * gst/audioconvert/gstaudioconvert.c:
20383 (gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
20384 (gst_audio_convert_transform_caps):
20385 * tests/check/elements/audioconvert.c: (GST_START_TEST),
20386 (audioconvert_suite):
20387 Don't run inplace if that overwrites source data as we go. Add more
20388 tests. Fixes #339837 even more.
20390 2007-02-27 18:45:37 +0000 Julien Moutte <julien@moutte.net>
20392 tests/examples/seek/seek.c: Fix various seeking bugs (Slider was not updating when doing a non flushing seek, Reverse...
20393 Original commit message from CVS:
20394 2007-02-27 Julien MOUTTE <julien@moutte.net>
20395 * tests/examples/seek/seek.c: (do_seek), (set_update_scale),
20396 (msg_segment_done): Fix various seeking bugs (Slider was not
20397 updating when doing a non flushing seek, Reverse playback
20398 on segment seek was wrong).
20400 2007-02-26 21:01:03 +0000 David Schleef <ds@schleef.org>
20402 Add a new plugin/library to make it easy for apps to shove data into a pipeline.
20403 Original commit message from CVS:
20405 * gst/app/Makefile.am:
20406 * gst/app/gstapp.c:
20407 * gst/app/gstappsrc.c:
20408 * gst/app/gstappsrc.h:
20409 Add a new plugin/library to make it easy for apps to shove
20410 data into a pipeline.
20412 2007-02-26 11:48:49 +0000 Wim Taymans <wim.taymans@gmail.com>
20414 tests/examples/seek/seek.c: When we stop scrubbing, don't leave the pipeline PLAYING when we requested a PAUSED state.
20415 Original commit message from CVS:
20416 * tests/examples/seek/seek.c: (stop_seek):
20417 When we stop scrubbing, don't leave the pipeline PLAYING when we
20418 requested a PAUSED state.
20420 2007-02-25 23:51:03 +0000 René Stadler <mail@renestadler.de>
20422 gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
20423 Original commit message from CVS:
20424 Patch by: René Stadler <mail at renestadler de>
20425 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
20426 Parse date strings in vorbis comments that have an invalid (zero)
20427 month or day (#410396).
20428 * tests/check/libs/tag.c: (GST_START_TEST):
20429 Test case for the above.
20431 2007-02-24 20:12:49 +0000 Loïc Minier <lool+gnome@via.ecp.fr>
20433 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
20434 Original commit message from CVS:
20435 Patch by: Loïc Minier <lool+gnome at via ecp fr>
20437 * ext/alsa/Makefile.am:
20438 * gst/audiotestsrc/Makefile.am:
20439 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
20441 2007-02-23 18:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
20443 gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
20444 Original commit message from CVS:
20445 * gst/playback/gstplaybin.c:
20446 Improve docs: point out that the application needs to assist playbin
20449 2007-02-23 13:10:50 +0000 Tim-Philipp Müller <tim@centricular.net>
20451 Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
20452 Original commit message from CVS:
20453 * gst-libs/gst/utils/install-plugins.c:
20454 * gst-libs/gst/utils/missing-plugins.c:
20455 * tests/check/libs/utils.c: (missing_msg_check_getters):
20456 Change GStreamer marker prefix in detail string from 'gstreamer.net'
20457 to just 'gstreamer'. Document the caps string component of the
20458 decoder/encoder detail a bit better, since not everyone will be
20459 familiar with the GStreamer media type/caps system (but they better
20460 enjoy nested itemized lists).
20462 2007-02-22 12:57:47 +0000 Tim-Philipp Müller <tim@centricular.net>
20464 gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
20465 Original commit message from CVS:
20466 * gst-libs/gst/netbuffer/gstnetbuffer.c:
20467 (notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
20468 Fix copying of GstNetBuffer (would crash before, or at least lead to
20469 invalid memory access, #410772), for now by copying the GstBuffer copy
20470 code from the core over here so we can copy the GstBuffer fields on a
20471 provided buffer instance (of type GstNetBuffer in this case). Would be
20472 better to fix this with some support by the core though (and in the long
20473 run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
20474 * tests/check/Makefile.am:
20475 Enable unit test for GstNetBuffer.
20477 2007-02-22 11:04:10 +0000 Andy Wingo <wingo@pobox.com>
20480 * gst-libs/gst/audio/gstbaseaudiosink.c:
20481 gst-libs/gst/audio/gstbaseaudiosink.c
20482 Original commit message from CVS:
20483 2007-02-22 Andy Wingo <wingo@pobox.com>
20484 * gst-libs/gst/audio/gstbaseaudiosink.c
20485 (gst_base_audio_sink_init): Disable pull-mode activation until we
20486 figure out how to make audio sinks go to PLAYING.
20488 2007-02-22 09:04:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20490 Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
20491 Original commit message from CVS:
20492 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
20493 (double_hq), (audio_convert_get_func_index),
20494 (audio_convert_prepare_context), (audio_convert_convert):
20495 * gst/audioconvert/audioconvert.h:
20496 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
20497 (gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
20498 * gst/audioconvert/gstchannelmix.h:
20499 * tests/check/elements/audioconvert.c: (GST_START_TEST):
20500 Add float as an intermediate format, as well as float mixing. Enable
20501 test that was failing before. Fixes #339837
20503 2007-02-21 16:12:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20505 tests/examples/seek/seek.c: Undo the previous commit: -1 as a stop time implies that the stop time is the end of file...
20506 Original commit message from CVS:
20507 * tests/examples/seek/seek.c: (do_seek):
20508 Undo the previous commit: -1 as a stop time implies that the stop
20509 time is the end of file, clearing any previously configured segment.
20511 2007-02-21 15:36:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20513 tests/examples/seek/seek.c: Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
20514 Original commit message from CVS:
20515 * tests/examples/seek/seek.c: (do_seek):
20516 Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
20518 2007-02-21 13:55:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20520 gst/volume/gstvolume.c: Unbreak volume, value remains gint.
20521 Original commit message from CVS:
20522 * gst/volume/gstvolume.c: (volume_process_int16),
20523 (volume_process_int16_clamp), (volume_set_caps):
20524 Unbreak volume, value remains gint.
20526 2007-02-21 13:08:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20528 gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
20529 Original commit message from CVS:
20530 * gst/volume/gstvolume.c: (volume_choose_func),
20531 (volume_update_real_volume), (gst_volume_set_volume),
20532 (gst_volume_init), (volume_process_double), (volume_process_float),
20533 (volume_process_int16), (volume_process_int16_clamp),
20534 (volume_set_caps), (volume_transform_ip), (volume_update_volume):
20535 * gst/volume/gstvolume.h:
20536 Extend float audio support (double) and some int->uint cleanups.
20538 2007-02-20 15:44:32 +0000 Edward Hervey <bilboed@bilboed.com>
20540 gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
20541 Original commit message from CVS:
20542 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
20543 (multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
20544 (sort_end_pads), (gst_decode_group_expose),
20545 (gst_decode_group_hide):
20546 Don't free groups from the streaming threads. Just put them aside and
20547 free them in dispose.
20549 2007-02-20 11:20:52 +0000 Edward Hervey <bilboed@bilboed.com>
20551 gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
20552 Original commit message from CVS:
20553 * gst/playback/gstdecodebin2.c: (connect_element),
20554 (pad_added_group_cb), (gst_decode_group_check_if_blocked),
20555 (sort_end_pads), (gst_decode_group_expose):
20556 Handle dynamic pads within groups.
20557 Sort pads before exposing them in order to make playbin happy.
20558 There still is a race with the multiqueue filling up. This should be
20562 2007-02-18 21:02:36 +0000 Tim-Philipp Müller <tim@centricular.net>
20564 gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
20565 Original commit message from CVS:
20566 * gst-libs/gst/utils/base-utils.c:
20567 * gst-libs/gst/utils/descriptions.c:
20568 * gst-libs/gst/utils/install-plugins.c:
20569 * gst-libs/gst/utils/missing-plugins.c:
20570 Some more docs (and descriptions for two subtitle formats).
20572 2007-02-16 10:19:45 +0000 Tim-Philipp Müller <tim@centricular.net>
20574 gst-libs/gst/audio/audio.c: Fix documentation.
20575 Original commit message from CVS:
20576 * gst-libs/gst/audio/audio.c:
20579 2007-02-16 10:15:46 +0000 Yves Lefebvre <ivanohe@abacom.com>
20581 gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
20582 Original commit message from CVS:
20583 Patch by: Yves Lefebvre <ivanohe abacom com>
20584 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
20585 Don't leak caps. Fixes #408278.
20587 2007-02-15 15:17:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20589 More docs coverage and some ChangeLog surgery (add missing names)
20590 Original commit message from CVS:
20591 * ext/cdparanoia/gstcdparanoiasrc.h:
20592 * ext/ogg/gstoggdemux.h:
20593 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
20594 (gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
20595 (gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
20596 * gst-libs/gst/audio/audio.h:
20597 * gst-libs/gst/audio/gstaudiofilter.h:
20598 * gst-libs/gst/interfaces/videoorientation.h:
20599 * gst/adder/gstadder.h:
20600 More docs coverage and some ChangeLog surgery (add missing names)
20602 2007-02-15 12:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
20604 sys/: Small constifications.
20605 Original commit message from CVS:
20606 * sys/ximage/ximagesink.c:
20607 (gst_ximagesink_calculate_pixel_aspect_ratio):
20608 * sys/xvimage/xvimagesink.c:
20609 (gst_xvimagesink_calculate_pixel_aspect_ratio):
20610 Small constifications.
20612 2007-02-15 12:06:25 +0000 Wim Taymans <wim.taymans@gmail.com>
20614 gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
20615 Original commit message from CVS:
20616 * gst-libs/gst/audio/gstbaseaudiosink.c:
20617 (gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
20618 (gst_base_audio_sink_render), (gst_base_audio_sink_callback),
20619 (gst_base_audio_sink_async_play),
20620 (gst_base_audio_sink_change_state):
20621 Answer latency query.
20622 Use configured latency when syncing.
20624 * gst-libs/gst/audio/gstbaseaudiosrc.c:
20625 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
20626 (gst_base_audio_src_query), (gst_base_audio_src_change_state):
20627 Fix possible memleak.
20628 Implement latency query.
20631 2007-02-15 11:59:41 +0000 Wim Taymans <wim.taymans@gmail.com>
20633 ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
20634 Original commit message from CVS:
20635 * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
20636 Ignore errors in reset, these are not fatal. They also grab the element
20637 lock which is already taking when this function is called. Fixes
20640 2007-02-13 13:50:56 +0000 Christian Schaller <uraeus@gnome.org>
20642 * gst-plugins-base.spec.in:
20643 add header file for easy codec install
20644 Original commit message from CVS:
20645 add header file for easy codec install
20647 2007-02-13 10:24:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20649 configure.ac: Remove 'tests/examples/xerror/Makefile' from output files again.
20650 Original commit message from CVS:
20652 Remove 'tests/examples/xerror/Makefile' from output files again.
20654 2007-02-13 09:12:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20656 Also crossref against gst-plugins-base-libs.
20657 Original commit message from CVS:
20659 * docs/plugins/Makefile.am:
20660 Also crossref against gst-plugins-base-libs.
20662 2007-02-12 20:42:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20664 Add crossreferences to glib/gobject/gstream docs.
20665 Original commit message from CVS:
20667 * docs/libs/Makefile.am:
20668 * docs/plugins/Makefile.am:
20669 Add crossreferences to glib/gobject/gstream docs.
20670 * gst-libs/gst/audio/audio.h:
20672 * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
20673 Add own debug category.
20675 2007-02-12 11:01:04 +0000 René Stadler <mail@renestadler.de>
20677 gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
20678 Original commit message from CVS:
20679 Patch by: René Stadler <mail at renestadler de>
20680 * gst-libs/gst/tag/gstvorbistag.c:
20681 Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
20684 2007-02-12 10:33:40 +0000 Tim-Philipp Müller <tim@centricular.net>
20686 gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
20687 Original commit message from CVS:
20688 * gst/playback/gstplaybasebin.c: (setup_source):
20689 When we have external subtitles and wait for the subtitle decodebin
20690 to get up and running, we set up a (sync) bus handler for the
20691 subtitle decodebin, so we can stop waiting when it posts an error
20692 message. However, we should do that before we set the subtitle
20693 decodebin's state to playing, otherwise things are racy and we might
20694 miss error messages posted before we had a chance to set up the bus.
20695 This should finally fix totem hanging on .txt pseudo-subtitle files.
20697 2007-02-10 19:27:48 +0000 Sébastien Moutte <sebastien@moutte.net>
20699 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
20700 Original commit message from CVS:
20701 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
20702 Use gst_gdouble_to_guint64 for conversions.
20703 * win32/common/config.h.in:
20704 Add a define for GST_INSTALL_PLUGINS_HELPER
20705 * win32/common/libgstaudio.def:
20706 * win32/common/libgstcdda.def:
20707 * win32/common/libgstnetbuffer.def:
20708 * win32/common/libgstrtp.def:
20709 * win32/common/libgutils.def:
20710 Add new exported functions.
20711 * win32/vs6/gst_plugins_base.dsw:
20712 * win32/vs6/libgstdecodebin.dsp:
20713 * win32/vs6/libgstnetbuffer.dsp:
20714 * win32/vs6/libgstplaybin.dsp:
20715 * win32/vs6/libgstrtp.dsp:
20716 * win32/vs6/libgstvorbis.dsp:
20717 * win32/vs6/libgstcdda.dsp:
20718 * win32/vs6/libgstgdp.dsp:
20719 * win32/vs6/libgstutils.dsp:
20720 Update and add new project files.
20722 2007-02-10 18:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
20724 gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
20725 Original commit message from CVS:
20726 * gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
20727 (subrip_remove_unhandled_tags), (parse_subrip):
20728 For SubRip (.srt) subtitles, ignore all markup tags we don't
20729 handle (like font tags, for example).
20730 * tests/check/elements/subparse.c:
20733 2007-02-09 13:28:01 +0000 Tim-Philipp Müller <tim@centricular.net>
20737 Original commit message from CVS:
20740 2007-02-09 13:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
20742 gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
20743 Original commit message from CVS:
20744 * gst/playback/gstdecodebin.c: (add_fakesink),
20745 (gst_decode_bin_change_state):
20746 * gst/playback/gstdecodebin2.c: (add_fakesink),
20747 (gst_decode_bin_change_state):
20748 Don't error out if there is no fakesink in the READY to NULL state
20749 change, since when decodebin is re-used, we're only adding the
20750 fakesink element in READY to PAUSED.
20751 * tests/check/elements/decodebin.c:
20752 (new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
20754 Minimal unit test to make sure we can use the same decodebin
20755 instance twice (at least with audiotestsrc input).
20757 2007-02-09 09:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
20759 ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
20760 Original commit message from CVS:
20761 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
20762 Try to get devic-name from device string first, and from handle only
20763 as fallback (seems to yield better results and is more robust
20764 against buggy probing code on the application side).
20766 2007-02-08 15:43:26 +0000 Julien Puydt <julien.puydt@laposte.net>
20768 ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
20769 Original commit message from CVS:
20770 Based on patch by: Julien Puydt <julien.puydt at laposte net>
20771 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
20772 (gst_alsa_find_device_name):
20773 * ext/alsa/gstalsa.h:
20774 * ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
20775 * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
20776 Improve device-name detection a bit, especially in the case where
20777 the device is not actually open (#405020, #405024). Move common code
20778 into gstalsa.c instead of duplicating it.
20780 2007-02-07 13:05:01 +0000 Tim-Philipp Müller <tim@centricular.net>
20782 gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
20783 Original commit message from CVS:
20784 * gst/audioconvert/gstaudioconvert.c:
20785 Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
20787 2007-02-06 17:47:32 +0000 Julien Moutte <julien@moutte.net>
20789 sys/xvimage/xvimagesink.*: Implement PropertyProbe Interface for XVAdaptors so that one can choose the adaptor to use...
20790 Original commit message from CVS:
20791 2007-02-06 Julien MOUTTE <julien@moutte.net>
20792 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
20793 (gst_xvimagesink_get_xv_support),
20794 (gst_xvimagesink_xcontext_clear),
20795 (gst_xvimagesink_interface_supported),
20796 (gst_xvimagesink_probe_get_properties),
20797 (gst_xvimagesink_probe_probe_property),
20798 (gst_xvimagesink_probe_needs_probe),
20799 (gst_xvimagesink_probe_get_values),
20800 (gst_xvimagesink_property_probe_interface_init),
20801 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
20802 (gst_xvimagesink_init), (gst_xvimagesink_class_init),
20803 (gst_xvimagesink_get_type):
20804 * sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
20805 for XVAdaptors so that one can choose the adaptor to use with
20806 gstreamer-properties.
20808 2007-02-06 14:00:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20810 gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
20811 Original commit message from CVS:
20812 * gst/audioconvert/gstaudioconvert.c:
20813 Also mention that a conversion from double to float is suboptimal still.
20815 2007-02-06 09:42:05 +0000 Tim-Philipp Müller <tim@centricular.net>
20817 gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
20818 Original commit message from CVS:
20819 * gst-libs/gst/audio/gstaudiofilter.c:
20820 (gst_audio_filter_class_init), (gst_audio_filter_change_state):
20821 Clear our formats structure and free the caps contained in it when
20824 2007-02-05 18:39:51 +0000 Andy Wingo <wingo@pobox.com>
20827 * gst-libs/gst/audio/gstbaseaudiosink.c:
20828 gst-libs/gst/audio/gstbaseaudiosink.c
20829 Original commit message from CVS:
20830 2007-02-05 Andy Wingo <wingo@pobox.com>
20831 * gst-libs/gst/audio/gstbaseaudiosink.c
20832 (gst_base_audio_sink_callback): Update basesink->offset so that we
20833 pull monotonically increasing offsets instead of, um, seeking back
20834 to 0 each time. Fixes alsasrc ! alsasink!
20836 2007-02-05 11:44:52 +0000 Tim-Philipp Müller <tim@centricular.net>
20838 gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
20839 Original commit message from CVS:
20840 * gst/videoscale/gstvideoscale.c:
20841 A width and height of 1 makes us crash, so increase minimum size to
20842 2x2 pixels until someone feels like fixing this (#404512).
20844 2007-02-04 16:23:37 +0000 Tim-Philipp Müller <tim@centricular.net>
20846 tests/check/pipelines/oggmux.c: Add small test to make sure request pads are cleaned up properly even if oggmux never...
20847 Original commit message from CVS:
20848 * tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
20849 Add small test to make sure request pads are cleaned up properly
20850 even if oggmux never changes state out of NULL.
20852 2007-02-04 14:11:51 +0000 Tim-Philipp Müller <tim@centricular.net>
20854 tests/check/libs/utils.c: Fix unit test. Turns out things work much better when you
20855 Original commit message from CVS:
20856 * tests/check/libs/utils.c: (GST_START_TEST):
20857 Fix unit test. Turns out things work much better when you
20858 NULL-terminate string arrays. Should make p5 build bot happy again.
20860 2007-02-03 23:28:45 +0000 Tim-Philipp Müller <tim@centricular.net>
20862 gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
20863 Original commit message from CVS:
20864 * gst-libs/gst/audio/Makefile.am:
20865 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
20866 (gst_audio_filter_template_base_init),
20867 (gst_audio_filter_template_class_init),
20868 (gst_audio_filter_template_init),
20869 (gst_audio_filter_template_set_property),
20870 (gst_audio_filter_template_get_property),
20871 (gst_audio_filter_template_setup),
20872 (gst_audio_filter_template_filter),
20873 (gst_audio_filter_template_filter_inplace), (plugin_init):
20874 Oops, forgot to commit fixed-up example.
20876 2007-02-03 20:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
20878 Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
20879 Original commit message from CVS:
20880 * docs/libs/gst-plugins-base-libs-sections.txt:
20881 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
20882 (gst_audio_filter_class_init), (gst_audio_filter_init),
20883 (gst_audio_filter_set_caps),
20884 (gst_audio_filter_class_add_pad_templates):
20885 * gst-libs/gst/audio/gstaudiofilter.h:
20886 Port GstAudioFilter to 0.10. This change technically breaks
20887 API and ABI (and thus also every library developer's heart),
20888 but seems justifiable on the grounds that the base class was
20889 completely unusable before (ie. would crash immediately when
20890 actually used). Fixes #403963 (and eventually also #403572).
20891 Also document all of this a bit.
20893 2007-02-03 14:26:54 +0000 Tim-Philipp Müller <tim@centricular.net>
20895 Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
20896 Original commit message from CVS:
20897 * gst-libs/gst/utils/install-plugins.c:
20898 (gst_install_plugins_spawn_child):
20899 * tests/check/libs/utils.c:
20900 (test_base_utils_install_plugins_do_callout):
20901 Lowering log level to see why things fail on the p5 build bot;
20902 fix some typos in unit test messages.
20904 2007-02-03 13:59:27 +0000 Tim-Philipp Müller <tim@centricular.net>
20906 tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
20907 Original commit message from CVS:
20908 * tests/check/libs/utils.c:
20909 (test_base_utils_install_plugins_do_callout):
20910 Don't hard-code temp directory for test helper; use GLib functions
20911 to write out file and do error checking etc.
20913 2007-02-02 20:42:08 +0000 Tim-Philipp Müller <tim@centricular.net>
20915 gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
20916 Original commit message from CVS:
20917 * gst-libs/gst/utils/Makefile.am:
20918 * gst-libs/gst/utils/base-utils.h:
20919 * gst-libs/gst/utils/install-plugins.c:
20920 (gst_install_plugins_context_set_xid),
20921 (gst_install_plugins_context_new),
20922 (gst_install_plugins_context_free),
20923 (gst_install_plugins_get_helper),
20924 (gst_install_plugins_spawn_child),
20925 (gst_install_plugins_return_from_status),
20926 (gst_install_plugins_installer_exited),
20927 (gst_install_plugins_async), (gst_install_plugins_sync),
20928 (gst_install_plugins_return_get_name),
20929 (gst_install_plugins_installation_in_progress):
20930 * gst-libs/gst/utils/install-plugins.h:
20931 API: add API for applications to initiate installation of missing
20932 plugins, ie. gst_install_plugins_async() primarily.
20933 Based on libgimme-codec by Ryan Lortie.
20935 Add --with-install-plugins-helper configure option so distros can specify
20936 the path of the helper script or program to call when plugin installation
20937 is requested (distros: please do any argument munging in this helper
20938 script instead of patching GStreamer to pass arguments differently
20939 to another program directly).
20940 * docs/libs/gst-plugins-base-libs-docs.sgml:
20941 * docs/libs/gst-plugins-base-libs-sections.txt:
20942 Build and document new API.
20943 * tests/check/libs/utils.c: (result_cb),
20944 (test_base_utils_install_plugins_do_callout), (GST_START_TEST),
20945 (libgstbaseutils_suite):
20946 Some simple checks for the new API.
20948 2007-02-02 14:44:29 +0000 Tim-Philipp Müller <tim@centricular.net>
20950 tests/check/elements/audioconvert.c: Add small test for 32bit float <=> 64bit float conversion (works only one way so...
20951 Original commit message from CVS:
20952 * tests/check/elements/audioconvert.c: (test_float_conversion):
20953 Add small test for 32bit float <=> 64bit float conversion (works
20954 only one way so far, 32=>64 produces structured noise).
20956 2007-02-02 11:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
20958 gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
20959 Original commit message from CVS:
20960 * gst/audioconvert/gstaudioconvert.c:
20961 (set_structure_widths_32_and_64), (make_lossless_changes):
20962 We don't support floats with a width of 40, 48 or 56 bits.
20964 2007-02-02 09:48:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20966 gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
20967 Original commit message from CVS:
20968 * gst/audioconvert/audioconvert.c: (float), (double),
20969 (audio_convert_get_func_index):
20970 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
20971 (make_lossless_changes):
20972 Support for 64-bit float audio in audioconvert (#339837)
20974 2007-02-01 18:50:08 +0000 Holger Wansing <linux@wansing-online.de>
20976 po/: Add German translation (#352069).
20977 Original commit message from CVS:
20978 Patch by: Holger Wansing <linux wansing-online de>
20981 Add German translation (#352069).
20983 2007-02-01 17:52:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20985 ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
20986 Original commit message from CVS:
20987 reviewed by: Wim Taymans <wim@fluendo.com>
20988 * ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
20989 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
20990 Use newly added GstCollectPads API to free the allocated resources in
20991 the GstOggPad structures (#402393).
20993 2007-01-31 15:58:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20995 gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
20996 Original commit message from CVS:
20997 * gst/playback/gstplaybin.c: (gen_vis_element):
20998 Add audioresample+audioconvert in front of the visualisation
20999 element, so that elements like libvisual 0.4 that don't support all
21000 samplerates can work.
21003 2007-01-30 19:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
21005 gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
21006 Original commit message from CVS:
21007 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
21008 (gst_play_base_bin_get_streaminfo_value_array):
21009 Take some locks and make a copy of the streaminfo value array we
21010 maintain while holding the lock, so that the application can
21011 retrieve the stream-info as a value array in a thread-safe way.
21013 2007-01-30 11:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
21015 gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
21016 Original commit message from CVS:
21017 * gst/audioconvert/gstaudioconvert.c:
21018 Don't fail on 0 sized buffers. Fixes #396835.
21020 2007-01-29 21:13:07 +0000 David Schleef <ds@schleef.org>
21022 gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
21023 Original commit message from CVS:
21024 * gst/typefind/gsttypefindfunctions.c:
21025 Detect BBCD as video/x-dirac, so we can play raw dirac
21028 2007-01-29 18:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
21030 ext/theora/theoraenc.c: Check return value of theora_encode_header(), or we might try to allocate a random number of ...
21031 Original commit message from CVS:
21032 * ext/theora/theoraenc.c: (theora_enc_chain):
21033 Check return value of theora_encode_header(), or we might try to
21034 allocate a random number of bytes. theora_encode_header() can fail
21035 if libtheora has been compiled with encoding support disabled.
21038 2007-01-29 10:53:06 +0000 Wim Taymans <wim.taymans@gmail.com>
21040 tests/check/gst/.cvsignore: Do as buildbot says.
21041 Original commit message from CVS:
21042 * tests/check/gst/.cvsignore:
21043 Do as buildbot says.
21045 2007-01-29 10:25:11 +0000 Wim Taymans <wim.taymans@gmail.com>
21047 ext/libvisual/visual.c: Fix strides in libvisual. Gst uses X strides.
21048 Original commit message from CVS:
21049 * ext/libvisual/visual.c: (gst_visual_src_setcaps):
21050 Fix strides in libvisual. Gst uses X strides.
21051 Inspired by: <ed at catmur dot co dot uk> and
21052 <tim at centricular dot net>
21055 2007-01-27 13:32:24 +0000 Wim Taymans <wim.taymans@gmail.com>
21057 ext/ogg/gstoggdemux.*: Properly propagate streaming errors when we are scanning the file for chains so that we don't ...
21058 Original commit message from CVS:
21059 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
21060 (gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
21061 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
21062 (gst_ogg_demux_perform_seek),
21063 (gst_ogg_demux_bisect_forward_serialno),
21064 (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
21065 (gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
21066 (gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
21067 (gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
21068 * ext/ogg/gstoggdemux.h:
21069 Properly propagate streaming errors when we are scanning the file for
21070 chains so that we don't crash when shut down. Might fix some crashers
21071 when quickly switching oggs in RB such as #332503 and #378436.
21073 2007-01-26 12:44:46 +0000 Tim-Philipp Müller <tim@centricular.net>
21075 ext/gnomevfs/gstgnomevfssrc.c: Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND error code as well.
21076 Original commit message from CVS:
21077 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
21078 Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
21079 error code as well.
21081 2007-01-25 16:02:41 +0000 Wim Taymans <wim.taymans@gmail.com>
21083 gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
21084 Original commit message from CVS:
21085 * gst/playback/gstplaybasebin.c: (remove_source):
21086 Don't try to disconnect a signal from a finalized object.
21088 2007-01-25 14:29:21 +0000 Tim-Philipp Müller <tim@centricular.net>
21090 gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
21091 Original commit message from CVS:
21092 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
21093 Cast lock macro parameters to make sure we're actually accessing the
21094 lock member at the right class level. Free list itself in _dispose()
21095 as well and NULL it in case dispose gets called multiple times.
21097 2007-01-25 14:02:37 +0000 Edward Hervey <bilboed@bilboed.com>
21099 gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
21100 Original commit message from CVS:
21101 * gst/playback/gstdecodebin2.c:
21102 (gst_decode_bin_dispose),(gst_decode_bin_finalize):
21103 Free GstDecodeGroups no longer used.
21104 (gst_decode_group_expose):
21105 Don't unlock too many times !
21106 (deactivate_free_recursive):
21107 Free iterator once we're done with it.
21108 Fix for recursively deactivating elements (stop at ghostpads).
21110 2007-01-25 12:24:18 +0000 Tim-Philipp Müller <tim@centricular.net>
21112 gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
21113 Original commit message from CVS:
21114 * gst/playback/gstplaybin.c: (handoff):
21115 Fix up caps on the frame buffer before we save it and potentially
21116 make it accessible to other threads via g_object_get; also use
21117 gst_buffer_replace() instead of gst_mini_object_replace().
21119 2007-01-25 12:06:59 +0000 Tim-Philipp Müller <tim@centricular.net>
21121 gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
21122 Original commit message from CVS:
21123 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
21124 Make getting the current frame thread-safe.
21126 2007-01-25 11:48:10 +0000 Edward Hervey <bilboed@bilboed.com>
21128 gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
21129 Original commit message from CVS:
21130 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
21131 (gst_decode_group_new), (gst_decode_group_free):
21132 Set queues to bigger sizes to cope with HD contents.
21133 Fix some mutex freeing and add comment about MT safe methods.
21135 2007-01-24 12:51:20 +0000 Tim-Philipp Müller <tim@centricular.net>
21137 ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
21138 Original commit message from CVS:
21139 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
21140 (gst_text_overlay_text_event):
21141 Don't unnecessarily ref (and then leak) upstream events if the text
21142 pad is not linked. Fixes #399948.
21143 * tests/check/gst-plugins-base.supp:
21144 Add suppression for pango on edgy/x86 for textoverlay test.
21146 2007-01-24 12:10:56 +0000 Wim Taymans <wim.taymans@gmail.com>
21148 gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
21149 Original commit message from CVS:
21150 * gst-libs/gst/rtp/gstrtpbuffer.h:
21151 Add some more fixed payloads.
21153 2007-01-23 18:39:45 +0000 Tim-Philipp Müller <tim@centricular.net>
21155 ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
21156 Original commit message from CVS:
21157 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
21158 Error out properly if we get an error from libogg while reading the
21159 BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).
21161 2007-01-23 17:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
21163 gst/playback/gstdecodebin2.c: Don't leak mutex.
21164 Original commit message from CVS:
21165 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
21167 * tests/check/elements/playbin.c:
21168 (test_sink_usage_video_only_stream),
21169 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
21170 (test_suburi_error_wrongproto), (test_missing_urisource_handler),
21171 (test_missing_suburisource_handler),
21172 (test_missing_primary_decoder), (playbin_suite):
21173 Run all tests once with decodebin and once with decodebin2.
21174 One test does not pass yet with decodebin2.
21176 2007-01-23 14:30:28 +0000 Edward Hervey <bilboed@bilboed.com>
21178 ext/ogg/gstoggmux.c: Fix the cases where oggmux doesn't properly figure out that all sinkpads have gone EOS, and ther...
21179 Original commit message from CVS:
21180 * ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
21181 Fix the cases where oggmux doesn't properly figure out that all
21182 sinkpads have gone EOS, and therefore doesn't push out the remaining
21183 buffers and the final EOS event.
21186 2007-01-23 13:19:19 +0000 Julien Moutte <julien@moutte.net>
21188 sys/: Don't lock on navigation event push, just on keysym to string.
21189 Original commit message from CVS:
21190 2007-01-23 Julien MOUTTE <julien@moutte.net>
21191 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
21192 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
21193 Don't lock on navigation event push, just on keysym to string.
21194 Fixes #397673 again.
21196 2007-01-22 17:37:38 +0000 Edward Hervey <bilboed@bilboed.com>
21198 gst/playback/gstdecodebin2.c: Cleanups.
21199 Original commit message from CVS:
21200 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
21201 (get_current_group), (group_demuxer_event_probe),
21202 (gst_decode_group_expose), (deactivate_free_recursive),
21203 (gst_decode_group_free):
21205 Don't forget to emit 'no-more-pads' once a group is exposed.
21206 Cleanup elements from a DecodeGroup once we remove it.
21207 Protect call to gst_decode_group_expose() with the decodebin lock.
21209 2007-01-22 13:16:42 +0000 Julien Moutte <julien@moutte.net>
21211 sys/: Looking at Xorg code i can't figure out if that XKeysymToString function is thread sensible or not. Lock it jus...
21212 Original commit message from CVS:
21213 2007-01-22 Julien MOUTTE <julien@moutte.net>
21214 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
21215 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
21216 Looking at Xorg code i can't figure out if that XKeysymToString
21217 function is thread sensible or not. Lock it just in case as
21218 recommended by Radek Doulik <rodo at ximian dot com>.
21220 2007-01-22 13:10:13 +0000 Julien Moutte <julien@moutte.net>
21222 sys/: Lock that X Call as well. Fixes #397673.
21223 Original commit message from CVS:
21224 2007-01-22 Julien MOUTTE <julien@moutte.net>
21225 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
21226 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
21227 Lock that X Call as well. Fixes #397673.
21229 2007-01-22 12:03:27 +0000 Tim-Philipp Müller <tim@centricular.net>
21231 gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
21232 Original commit message from CVS:
21233 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
21234 Don't go into an endless loop if the file starts with 00 00 01 2X,
21235 like quicktime redirect files might. Fixes #396042.
21236 * tests/check/Makefile.am:
21237 * tests/check/gst/.cvsignore:
21238 * tests/check/gst/typefindfunctions.c: (GST_START_TEST),
21239 (typefindfunctions_suite):
21240 Add unit test for the above.
21242 2007-01-22 10:27:26 +0000 Tim-Philipp Müller <tim@centricular.net>
21244 gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
21245 Original commit message from CVS:
21246 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
21247 On second thought, use "depth" field rather than "bpp" field.
21249 2007-01-22 09:23:01 +0000 Tim-Philipp Müller <tim@centricular.net>
21251 gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
21252 Original commit message from CVS:
21253 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
21254 Camtasia caps apparently need a bpp field (#398875).
21256 2007-01-19 19:09:05 +0000 Tim-Philipp Müller <tim@centricular.net>
21258 gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
21259 Original commit message from CVS:
21260 * gst/playback/gstplaybasebin.c: (setup_subtitle),
21261 (gen_source_element), (gst_play_base_bin_change_state):
21262 Attempt at a better error message in case we don't have the required
21263 URI handler installed; post missing-plugin message also when we're
21264 missing an URI handler for the subtitle URI; clean up properly also
21265 when an error occurs and we never made it to PAUSED state.
21266 * tests/check/elements/playbin.c: (GST_START_TEST),
21268 Check that we're also getting a missing-plugin messsage for a
21269 missing subtitle URI handler (and clean up properly).
21271 2007-01-19 18:47:30 +0000 Tim-Philipp Müller <tim@centricular.net>
21273 gst/playback/gstplaybasebin.c: Plug a few reference leaks.
21274 Original commit message from CVS:
21275 * gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
21276 Plug a few reference leaks.
21278 2007-01-19 12:23:06 +0000 Tim-Philipp Müller <tim@centricular.net>
21280 gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
21281 Original commit message from CVS:
21282 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
21283 Lower probability a bit if the marker isn't right at the start,
21284 to decrease the chance of false positives.
21286 2007-01-19 11:31:50 +0000 Tim-Philipp Müller <tim@centricular.net>
21288 gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
21289 Original commit message from CVS:
21290 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
21291 Small mpeg2 system stream typefinding improvement: make typefinder
21292 probe a bit into the stream instead of just looking for a marker
21293 at the beginning. Fixes #397810.
21295 2007-01-18 16:23:35 +0000 Tim-Philipp Müller <tim@centricular.net>
21297 gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
21298 Original commit message from CVS:
21299 * gst/audioconvert/gstchannelmix.c:
21300 Remove compatibility cruft for prehistoric GLib versions.
21302 2007-01-17 16:11:14 +0000 Tim-Philipp Müller <tim@centricular.net>
21304 gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
21305 Original commit message from CVS:
21306 * gst/playback/Makefile.am:
21307 * gst/playback/gstdecodebin.c: (close_pad_link):
21308 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
21309 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
21310 (gst_play_base_bin_handle_message_func), (unknown_type):
21311 Let decodebin be the element to post missing-plugin messages for
21312 missing decoders (rather than playbin); make playbin implement
21313 GstBin::handle_message so we can suppress missing-plugin messages
21314 for types we're not handling on purpose (don't want to bring up an
21315 installer in those cases).
21317 2007-01-16 19:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21319 gst/: Fix potentially unaligned access (#397207).
21320 Original commit message from CVS:
21321 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
21322 * gst-libs/gst/tag/gstvorbistag.c:
21323 (gst_tag_list_to_vorbiscomment_buffer):
21324 * gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
21325 Fix potentially unaligned access (#397207).
21327 2007-01-16 12:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21329 tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
21330 Original commit message from CVS:
21331 * tests/examples/seek/seek.c: (set_scale), (update_scale),
21332 (do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
21333 (rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
21335 Allow to toggle looping while it plays. Fix callback prototype. Clean
21336 up code a bit more. Add copyright header.
21338 2007-01-16 11:41:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21340 sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
21341 Original commit message from CVS:
21342 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
21343 Red and blue mask was swapped (spotted by Dan Williams).
21345 2007-01-15 13:58:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21347 gst-libs/gst/tag/: Use new beats-per-minute tag from core.
21348 Original commit message from CVS:
21349 * gst-libs/gst/tag/gstid3tag.c:
21350 * gst-libs/gst/tag/gstvorbistag.c:
21351 Use new beats-per-minute tag from core.
21353 2007-01-15 11:30:53 +0000 Tim-Philipp Müller <tim@centricular.net>
21355 po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
21356 Original commit message from CVS:
21358 Add new files with translatable strings, so they actually make it
21359 into the template file one day.
21361 2007-01-12 21:19:35 +0000 Andy Wingo <wingo@pobox.com>
21364 * gst-libs/gst/audio/gstbaseaudiosink.c:
21365 * gst-libs/gst/audio/gstbaseaudiosrc.c:
21366 gst-libs/gst/audio/gstbaseaudiosink.c
21367 Original commit message from CVS:
21368 2007-01-12 Andy Wingo <wingo@pobox.com>
21369 * gst-libs/gst/audio/gstbaseaudiosink.c
21370 (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
21371 (gst_base_audio_sink_activate_pull): Remove the handwavey nego
21372 stuff, as the base class handles this now. Actually tell the ring
21374 (gst_base_audio_sink_callback): Cast the ring buffer correctly.
21375 How did this work before? Maybe I'm not as awesome a programmer as
21377 * gst-libs/gst/audio/gstbaseaudiosrc.c
21378 (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
21381 2007-01-12 18:08:23 +0000 Tim-Philipp Müller <tim@centricular.net>
21383 gst-libs/gst/utils/missing-plugins.c: Remove more fields so that the application can better blacklist formats that ha...
21384 Original commit message from CVS:
21385 * gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
21386 Remove more fields so that the application can better blacklist
21387 formats that have been tried before.
21389 2007-01-12 17:43:40 +0000 Christian Schaller <uraeus@gnome.org>
21391 * gst-plugins-base.spec.in:
21393 Original commit message from CVS:
21396 2007-01-12 12:47:29 +0000 Tim-Philipp Müller <tim@centricular.net>
21398 gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
21399 Original commit message from CVS:
21400 * gst-libs/gst/audio/mixerutils.h:
21401 Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
21402 used when compiling with c++ compilers as well.
21404 2007-01-12 09:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
21406 gst/typefind/gsttypefindfunctions.c: Fix comment.
21407 Original commit message from CVS:
21408 * gst/typefind/gsttypefindfunctions.c:
21411 2007-01-11 13:12:17 +0000 Tim-Philipp Müller <tim@centricular.net>
21413 gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
21414 Original commit message from CVS:
21415 * gst/playback/gstplaybin.c: (post_missing_element_message),
21416 (gen_video_element), (gen_text_element), (gen_audio_element),
21418 Post missing-plugin messages also when we error out because
21419 converters, textoverlay or auto*sinks are missing (#161922).
21421 2007-01-10 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
21423 gst/playback/: Fix the case where we try to ref a NULL element when we delay a link because of unfixed caps.
21424 Original commit message from CVS:
21425 * gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
21426 (is_demuxer_element), (new_caps):
21427 * gst/playback/gstplaybasebin.c: (source_new_pad):
21428 Fix the case where we try to ref a NULL element when we delay a link
21429 because of unfixed caps.
21430 Set the state of autoplugged decodebins to PAUSED.
21431 RTSP now works in playbin, we can remove it from the blacklist.
21433 2007-01-09 14:33:24 +0000 Tim-Philipp Müller <tim@centricular.net>
21435 gst/playback/: Post missing-plugin messages on the bus for missing sources and missing decoders/demuxers/depayloaders...
21436 Original commit message from CVS:
21437 * gst/playback/Makefile.am:
21438 * gst/playback/gstplaybasebin.c: (string_arr_has_str),
21439 (unknown_type), (setup_subtitle), (gen_source_element):
21440 * gst/playback/gstplaybin.c: (plugin_init):
21441 Post missing-plugin messages on the bus for missing sources and
21442 missing decoders/demuxers/depayloaders; fix error code used when
21443 we're missing an URI handler source; for media types that we are not
21444 handling on purpose at the moment, don't print "don't know how to
21445 handle xyz" messages to the terminal or post missing-plugin
21446 messages on the bus.
21447 * tests/check/elements/playbin.c: (create_playbin),
21448 (GST_START_TEST), (gst_codec_src_uri_get_type),
21449 (gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
21450 (gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
21451 (gst_codec_src_init_type), (gst_codec_src_base_init),
21452 (gst_codec_src_create), (gst_codec_src_class_init),
21453 (gst_codec_src_init), (plugin_init), (playbin_suite):
21454 Add some tests for the missing-plugin stuff.
21456 2007-01-09 14:20:08 +0000 Tim-Philipp Müller <tim@centricular.net>
21458 API: add new libgstbaseutils library with functions
21459 Original commit message from CVS:
21461 * gst-libs/gst/Makefile.am:
21462 * gst-libs/gst/utils/Makefile.am:
21463 * gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
21464 * gst-libs/gst/utils/base-utils.h:
21465 * gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
21466 (find_format_info), (caps_are_rtp_caps),
21467 (gst_base_utils_get_source_description),
21468 (gst_base_utils_get_sink_description),
21469 (gst_base_utils_get_decoder_description),
21470 (gst_base_utils_get_encoder_description),
21471 (gst_base_utils_get_element_description),
21472 (gst_base_utils_add_codec_description_to_tag_list),
21473 (gst_base_utils_get_codec_description), (gst_base_utils_list_all):
21474 * gst-libs/gst/utils/descriptions.h:
21475 * gst-libs/gst/utils/missing-plugins.c:
21476 (missing_structure_get_type), (copy_and_clean_caps),
21477 (gst_missing_uri_source_message_new),
21478 (gst_missing_uri_sink_message_new),
21479 (gst_missing_element_message_new),
21480 (gst_missing_decoder_message_new),
21481 (gst_missing_encoder_message_new),
21482 (missing_structure_get_string_detail),
21483 (missing_structure_get_caps_detail),
21484 (gst_missing_plugin_message_get_installer_detail),
21485 (gst_missing_plugin_message_get_description),
21486 (gst_is_missing_plugin_message):
21487 * gst-libs/gst/utils/missing-plugins.h:
21488 API: add new libgstbaseutils library with functions
21489 - to create and parse missing-plugins messages
21490 - that provide (translated) descriptions for caps/decoders/sources/etc.
21492 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
21493 * pkgconfig/gstreamer-plugins-base.pc.in:
21495 * docs/libs/gst-plugins-base-libs-docs.sgml:
21496 * docs/libs/gst-plugins-base-libs-sections.txt:
21497 Generate docs for new lib and API.
21498 * tests/check/Makefile.am:
21499 * tests/check/libs/.cvsignore:
21500 * tests/check/libs/utils.c: (missing_msg_check_getters),
21501 (GST_START_TEST), (libgstbaseutils_suite):
21502 Add some basic unit tests.
21504 2007-01-09 13:35:08 +0000 Tim-Philipp Müller <tim@centricular.net>
21506 ext/ogg/Makefile.am: Dist gstoggdemux.h to fix 'make distcheck'.
21507 Original commit message from CVS:
21508 * ext/ogg/Makefile.am:
21509 Dist gstoggdemux.h to fix 'make distcheck'.
21510 * sys/v4l/Makefile.am:
21511 Fix 'make distcheck' even more.
21513 2007-01-09 12:30:46 +0000 Wim Taymans <wim.taymans@gmail.com>
21516 Original commit message from CVS:
21517 * docs/plugins/Makefile.am:
21518 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
21519 * docs/plugins/gst-plugins-base-plugins-sections.txt:
21520 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
21521 (gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
21522 (gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
21523 (gst_ogg_demux_perform_seek):
21524 * ext/ogg/gstoggdemux.h:
21526 Add some more comments.
21529 2007-01-09 11:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
21531 Small documentation updates/fixes
21532 Original commit message from CVS:
21533 * ext/theora/theoradec.c:
21534 * ext/vorbis/vorbisdec.c:
21535 * gst-libs/gst/audio/gstringbuffer.c:
21536 (gst_ring_buffer_commit_full):
21537 * gst-libs/gst/audio/gstringbuffer.h:
21538 * gst-libs/gst/rtp/gstrtpbuffer.c:
21539 * gst-libs/gst/tag/gstvorbistag.c:
21540 Small documentation updates/fixes
21542 2007-01-09 10:37:01 +0000 Tim-Philipp Müller <tim@centricular.net>
21544 configure.ac: Require core CVS HEAD for Andy's basesrc/sink API additions.
21545 Original commit message from CVS:
21547 Require core CVS HEAD for Andy's basesrc/sink API additions.
21549 2007-01-08 14:01:23 +0000 Günter Thelen <daedalus.inc@gmx.net>
21551 gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
21552 Original commit message from CVS:
21553 Patch by: Günter Thelen <daedalus dot inc at gmx net>
21554 * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
21556 Add typefinder for flac-in-ogg in conformance with the ogg-mapping
21557 on flac.sf.net (there appear to be other versions of the first
21558 ogg page in the wild) (#391365).
21560 2007-01-08 13:32:32 +0000 Tim-Philipp Müller <tim@centricular.net>
21562 configure.ac: Check if localtime_r() is available.
21563 Original commit message from CVS:
21565 Check if localtime_r() is available.
21566 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
21567 If localtime_r() is not available, fall back to localtime(). Should
21568 fix build on MingW (#393310).
21570 2007-01-08 12:30:03 +0000 Tim-Philipp Müller <tim@centricular.net>
21572 gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
21573 Original commit message from CVS:
21574 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
21575 * gst/subparse/gstsubparse.h:
21576 Remove spurious 1000 subtrahend when calculating the timestamp from
21577 the frame number and the frame rate . Also, use the frames/second
21578 value specified in the first line of the file, if one is specified
21579 there. Should fix #357503.
21580 * tests/check/elements/subparse.c: (do_test),
21581 (test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
21583 Add some basic unit tests for the microdvd subtitle format.
21585 2007-01-07 21:53:38 +0000 Young-Ho Cha <ganadist@chollian.net>
21587 sys/xvimage/xvimagesink.c: Fixes : #390076.
21588 Original commit message from CVS:
21589 2007-01-07 Julien MOUTTE <julien@moutte.net>
21590 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
21591 (gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
21592 (gst_xvimagesink_xvimage_put),
21593 (gst_lookup_xv_port_from_adaptor),
21594 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
21595 (gst_xvimagesink_set_xwindow_id),
21596 (gst_xvimagesink_set_event_handling),
21597 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
21598 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
21599 Patch by : Young-Ho Cha <ganadist at chollian dot net>
21601 Add an adaptor property to select a specific XV adaptor.
21602 * sys/xvimage/xvimagesink.h:
21604 2007-01-07 18:50:13 +0000 Julien Moutte <julien@moutte.net>
21606 sys/: Use flow_lock much more to protect every access to xwindow.
21607 Original commit message from CVS:
21608 2007-01-07 Julien MOUTTE <julien@moutte.net>
21609 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
21610 (gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
21611 (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
21612 (gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
21613 (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
21614 (gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
21615 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
21616 (gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
21617 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
21618 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
21619 (gst_xvimagesink_change_state),
21620 (gst_xvimagesink_set_xwindow_id),
21621 (gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
21622 Use flow_lock much more to protect every access to xwindow.
21623 Try to catch erros while creating images in case some drivers
21625 just generating an XError when the requested image is too big.
21626 Should fix : #354698, #384008, #384060.
21627 * tests/icles/stress-xoverlay.c: (cycle_window),
21629 Implement some stress testing of setting window xid.
21631 2007-01-07 10:33:55 +0000 Sébastien Moutte <sebastien@moutte.net>
21633 win32/common/libgsaudio.def: Add new exported function.
21634 Original commit message from CVS:
21635 * win32/common/libgsaudio.def:
21636 Add new exported function.
21637 * win32/common/libgstogg.dsp:
21638 Add gstoggaviparse.c to the build.
21639 * win32/common/libgstvideoscale.dsp:
21640 Add vs_4tap.c to the build.
21641 * win32/common/libgstvorbis.dsp:
21642 Add vorbistag.c to the build.
21644 2007-01-06 17:28:40 +0000 Andy Wingo <wingo@pobox.com>
21647 * gst-libs/gst/audio/gstbaseaudiosink.c:
21648 gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
21649 Original commit message from CVS:
21650 2007-01-06 Andy Wingo <wingo@pobox.com>
21651 * gst-libs/gst/audio/gstbaseaudiosink.c
21652 (gst_base_audio_sink_class_init)
21653 (gst_base_audio_sink_init):
21654 (gst_base_audio_sink_activate_pull): Add an activate_pull function
21655 to baseaudiosink, and tell basesink that we can work in pull mode.
21656 This way the ring buffer thread drives the pipeline directly, if
21657 pull mode is possible. There is some lingering nastiness regarding
21659 (gst_base_audio_sink_callback): Implement the callback to pull
21660 data. This interface is a bit light, though -- it should get a
21661 GstFlowReturn return value at least.
21663 2007-01-05 19:43:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21665 Printf format and missing argument fixes.
21666 Original commit message from CVS:
21667 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
21668 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
21669 * gst/playback/gstdecodebin2.c:
21670 (gst_decode_group_check_if_blocked):
21671 Printf format and missing argument fixes.
21673 2007-01-05 18:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21675 ext/ogg/gstogmparse.c: Activate pads before adding them to the element.
21676 Original commit message from CVS:
21677 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
21678 (gst_ogm_parse_change_state):
21679 Activate pads before adding them to the element.
21681 2007-01-05 16:02:50 +0000 Tim-Philipp Müller <tim@centricular.net>
21683 tests/examples/seek/: Call g_thread_init() first thing in main() (see #391278).
21684 Original commit message from CVS:
21685 * tests/examples/seek/scrubby.c: (main):
21686 * tests/examples/seek/seek.c: (main):
21687 Call g_thread_init() first thing in main() (see #391278).
21689 2007-01-05 12:19:34 +0000 Tim-Philipp Müller <tim@centricular.net>
21691 tests/check/: Add test for GstNetBuffer + gst_buffer_copy(). Disabled for the time being, since it's broken, see #393...
21692 Original commit message from CVS:
21693 * tests/check/Makefile.am:
21694 * tests/check/libs/.cvsignore:
21695 * tests/check/libs/netbuffer.c: (GST_START_TEST),
21697 Add test for GstNetBuffer + gst_buffer_copy(). Disabled
21698 for the time being, since it's broken, see #393099.
21700 2007-01-05 12:13:24 +0000 Tim-Philipp Müller <tim@centricular.net>
21702 tests/check/Makefile.am: Update to use GST_PLUGINS_BASE_CFLAGS as well.
21703 Original commit message from CVS:
21704 * tests/check/Makefile.am:
21705 Update to use GST_PLUGINS_BASE_CFLAGS as well.
21707 2007-01-04 12:49:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21709 configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
21710 Original commit message from CVS:
21712 split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
21713 so that GST_BASE_CFLAGS can go inbetween them, making sure
21714 we use uninstalled gst-libs headers
21715 * docs/libs/Makefile.am:
21716 * ext/alsa/Makefile.am:
21717 * ext/cdparanoia/Makefile.am:
21718 * ext/gnomevfs/Makefile.am:
21719 * ext/libvisual/Makefile.am:
21720 * ext/ogg/Makefile.am:
21721 * ext/theora/Makefile.am:
21722 * ext/vorbis/Makefile.am:
21723 * gst-libs/gst/audio/Makefile.am:
21724 * gst-libs/gst/cdda/Makefile.am:
21725 * gst-libs/gst/interfaces/Makefile.am:
21726 * gst-libs/gst/riff/Makefile.am:
21727 * gst-libs/gst/rtp/Makefile.am:
21728 * gst-libs/gst/tag/Makefile.am:
21729 * gst/adder/Makefile.am:
21730 * gst/audioconvert/Makefile.am:
21731 * gst/audiorate/Makefile.am:
21732 * gst/audioresample/Makefile.am:
21733 * gst/playback/Makefile.am:
21734 * gst/tcp/Makefile.am:
21735 * gst/videoscale/Makefile.am:
21736 * gst/volume/Makefile.am:
21737 * sys/ximage/Makefile.am:
21738 * sys/xvimage/Makefile.am:
21739 * tests/icles/Makefile.am:
21742 2007-01-04 11:30:53 +0000 Julien Moutte <julien@moutte.net>
21744 Add a method to the XOverlay interface to allow disabling of event handling in x[v]imagesink elements. This will let ...
21745 Original commit message from CVS:
21746 2007-01-04 Julien MOUTTE <julien@moutte.net>
21747 * gst-libs/gst/interfaces/xoverlay.c:
21748 (gst_x_overlay_handle_events):
21749 * gst-libs/gst/interfaces/xoverlay.h:
21750 * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
21751 (gst_ximagesink_set_xwindow_id),
21752 (gst_ximagesink_set_event_handling),
21753 (gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
21754 (gst_ximagesink_get_property), (gst_ximagesink_init),
21755 (gst_ximagesink_class_init):
21756 * sys/ximage/ximagesink.h:
21757 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
21758 (gst_xvimagesink_set_xwindow_id),
21759 (gst_xvimagesink_set_event_handling),
21760 (gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
21761 (gst_xvimagesink_get_property), (gst_xvimagesink_init),
21762 (gst_xvimagesink_class_init):
21763 * sys/xvimage/xvimagesink.h:
21764 * tests/icles/stress-xoverlay.c: (toggle_events),
21766 Add a method to the XOverlay interface to allow disabling of
21767 event handling in x[v]imagesink elements. This will let X events
21768 propagate to parent windows which can be usefull in some cases.
21769 Be carefull that the application is then responsible of pushing
21770 navigation events and expose events to the video sink.
21773 2007-01-03 15:45:06 +0000 Tim-Philipp Müller <tim@centricular.net>
21775 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION (fixes #392070).
21776 Original commit message from CVS:
21777 * gst-libs/gst/tag/gstvorbistag.c:
21778 * tests/check/libs/tag.c: (GST_START_TEST):
21779 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
21782 2007-01-01 18:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
21785 Original commit message from CVS:
21787 * docs/Makefile.am:
21788 * docs/design/Makefile.am:
21791 2006-12-27 17:15:35 +0000 Julien Moutte <julien@moutte.net>
21793 docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation typo. Fixes: #390063.
21794 Original commit message from CVS:
21795 2006-12-27 Julien MOUTTE <julien@moutte.net>
21796 * docs/libs/gst-plugins-base-libs-sections.txt: Fix a
21798 typo. Fixes: #390063.
21800 2006-12-27 12:08:13 +0000 Julien Moutte <julien@moutte.net>
21802 sys/: Plug a caps leak.
21803 Original commit message from CVS:
21804 2006-12-27 Julien MOUTTE <julien@moutte.net>
21805 * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
21806 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
21808 * win32/common/config.h: Updated.
21810 2006-12-22 12:10:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21812 tests/check/elements/: Fix the dp tests, but activating the pads for the streamheader tests too and cleaning up condi...
21813 Original commit message from CVS:
21814 * tests/check/elements/gdpdepay.c: (cleanup_gdpdepay),
21815 (setup_gdpdepay_streamheader):
21816 * tests/check/elements/gdppay.c: (cleanup_gdppay),
21817 (setup_gdppay_streamheader):
21818 Fix the dp tests, but activating the pads for the streamheader tests
21819 too and cleaning up conditionaly
21821 2006-12-22 11:09:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21823 gst/ffmpegcolorspace/: Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the other end of the wo...
21824 Original commit message from CVS:
21825 * gst/ffmpegcolorspace/avcodec.h:
21826 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
21827 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
21828 (gst_ffmpegcsp_avpicture_fill):
21829 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
21830 (img_get_alpha_info):
21831 Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
21832 other end of the word. Fixes: #387073.
21833 Add some inconsequential branch hints in a couple of places.
21835 2006-12-21 12:30:11 +0000 Tim-Philipp Müller <tim@centricular.net>
21837 gst/ffmpegcolorspace/gstffmpegcodecmap.c: The "signed" field in raw audio caps is of boolean type, trying to extract ...
21838 Original commit message from CVS:
21839 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
21840 (gst_ffmpeg_caps_to_smpfmt):
21841 The "signed" field in raw audio caps is of boolean type, trying to
21842 extract the value with _get_int() will fail (fix to keep in sync with
21843 the copy in gst-ffmpeg)
21845 2006-12-21 08:12:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21847 tests/check/elements/: consistent pad (de)activation
21848 Original commit message from CVS:
21849 * tests/check/elements/audioresample.c: (cleanup_audioresample):
21850 * tests/check/elements/audiotestsrc.c: (cleanup_audiotestsrc):
21851 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
21852 (cleanup_gdpdepay):
21853 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay):
21854 * tests/check/elements/subparse.c: (teardown_subparse):
21855 * tests/check/elements/textoverlay.c: (cleanup_textoverlay):
21856 * tests/check/elements/videorate.c: (cleanup_videorate):
21857 * tests/check/elements/videotestsrc.c: (cleanup_videotestsrc):
21858 * tests/check/elements/volume.c: (cleanup_volume):
21859 * tests/check/elements/vorbisdec.c: (setup_vorbisdec),
21860 (cleanup_vorbisdec):
21861 * tests/check/elements/vorbistag.c: (setup_vorbistag),
21862 (cleanup_vorbistag):
21863 consistent pad (de)activation
21865 2006-12-20 10:29:58 +0000 Tim-Philipp Müller <tim@centricular.net>
21867 gst/typefind/gsttypefindfunctions.c: Forgot to register the extensions.
21868 Original commit message from CVS:
21869 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
21870 Forgot to register the extensions.
21872 2006-12-20 09:25:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21874 gst/typefind/gsttypefindfunctions.c: Add typefinder for VIVO files (my christmas present to the 90s).
21875 Original commit message from CVS:
21876 * gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
21878 Add typefinder for VIVO files (my christmas present to the 90s).
21880 2006-12-16 13:59:09 +0000 Tim-Philipp Müller <tim@centricular.net>
21882 gst/playback/gstdecodebin.c: Special-case the text/plain media type: we only want to recognise it as a 'raw' decoded ...
21883 Original commit message from CVS:
21884 * gst/playback/gstdecodebin.c: (type_found):
21885 Special-case the text/plain media type: we only want to recognise it
21886 as a 'raw' decoded media type if it comes from a demuxer or subtitle
21887 parser, but not if the entire stream is of text/plain type. If the
21888 entire stream is text/plain, we should just error out.
21889 This fixes playback of audio files with lyrics in totem. Totem can't
21890 distinguish between text files and subtitle files and passes any
21891 .txt file with the same basename as the main file to playbin as
21892 suburi, and playbin will then throw a 'subtitle found, but no video
21893 stream' error, which isn't entirely helpful. See #380342.
21894 Also, with this change we'll show a slightly more correct error
21895 message in case totem passes a playlist file to us (although a
21896 custom error message wording instead of the default text would
21897 probably not be a bad idea either).
21898 Same problem also needs to be fixed for playbin+decodebin2.
21899 * tests/check/Makefile.am:
21900 * tests/check/elements/decodebin.c: (src_handoff_cb),
21901 (decodebin_new_decoded_pad_cb), (GST_START_TEST),
21903 Add simple unit test for decodebin for the above.
21905 2006-12-16 12:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
21907 gst/playback/: Refuse to change state to READY when we failed to create any of the required elements in our instance ...
21908 Original commit message from CVS:
21909 * gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
21910 * gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
21911 Refuse to change state to READY when we failed to create any of the
21912 required elements in our instance init function.
21914 2006-12-15 10:52:23 +0000 Tim-Philipp Müller <tim@centricular.net>
21916 docs/libs/gst-plugins-base-libs-sections.txt: Small docs fixes/updates.
21917 Original commit message from CVS:
21918 * docs/libs/gst-plugins-base-libs-sections.txt:
21919 Small docs fixes/updates.
21920 * gst-libs/gst/video/gstvideosink.h:
21921 Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
21922 from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
21923 removed from the base sink API between 0.9.6 and 0.9.7).
21924 API: add GST_VIDEO_SINK_CAST and use it for the height/width
21925 accessor macros, so we don't do a runtime GObject type check every
21928 2006-12-15 00:20:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21931 Original commit message from CVS:
21933 * gst-plugins-base.doap:
21934 * gst-plugins-base.spec.in:
21937 2006-12-09 15:12:38 +0000 Jens Granseuer <jensgr@gmx.net>
21939 Declare variables at the beginning of a block. Fixes #383195.
21940 Original commit message from CVS:
21941 Patch by: Jens Granseuer <jensgr at gmx net>
21942 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
21943 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
21944 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
21945 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
21946 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
21947 Declare variables at the beginning of a block. Fixes #383195.
21949 2006-12-07 02:38:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21951 configure.ac: Bump version nano - back to CVS.
21952 Original commit message from CVS:
21954 Bump version nano - back to CVS.
21956 === release 0.10.11 ===
21958 2006-12-07 02:30:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21960 configure.ac: releasing 0.10.11, "Dumb things"
21961 Original commit message from CVS:
21962 === release 0.10.11 ===
21963 2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
21965 releasing 0.10.11, "Dumb things"
21967 2006-12-05 12:44:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21969 gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
21970 Original commit message from CVS:
21971 * gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
21972 (close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
21973 Handle the case where an element has multiple pads with
21974 unfixed caps as well as still possibly producing more dynamic
21975 pads by storing each case as a distinct entry in the dynamic list.
21976 Fixes #38223 again.
21978 2006-12-04 13:02:37 +0000 Wim Taymans <wim.taymans@gmail.com>
21980 gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
21981 Original commit message from CVS:
21982 * gst/playback/gstdecodebin.c: (close_pad_link):
21983 Fix #382223, add more dynamic caps handling.
21985 2006-12-01 11:35:57 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
21988 Ignore all pot files
21989 Original commit message from CVS:
21990 Ignore all pot files
21992 2006-12-01 10:36:50 +0000 Michael Smith <msmith@xiph.org>
21994 gst/audiorate/gstaudiorate.c: Delete bad debug code.
21995 Original commit message from CVS:
21996 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
21997 Delete bad debug code.
22000 2006-12-01 10:27:54 +0000 Sergey Scobich <sergey.scobich@gmail.com>
22002 Fix compilation on win32 under VS8
22003 Original commit message from CVS:
22004 * gst/videoscale/vs_4tap.c:
22006 * win32/common/config.h:
22007 * win32/vs8/libgstvideoscale.vcproj:
22008 Fix compilation on win32 under VS8
22009 Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
22010 Partially fixes #381175
22012 2006-11-30 23:46:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22029 Original commit message from CVS:
22032 2006-11-30 12:50:42 +0000 Michael Smith <msmith@xiph.org>
22034 tests/check/pipelines/theoraenc.c: It would be very bad if, after a discont buffer, we thought every single following...
22035 Original commit message from CVS:
22036 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
22038 It would be very bad if, after a discont buffer, we thought every
22039 single following buffer was also discont. So, add to the test to
22040 ensure that this isn't the case.
22041 * ext/theora/theoraenc.c: (theora_enc_is_discontinuous):
22042 ... it was the case. So fix it.
22044 2006-11-28 16:43:18 +0000 Wim Taymans <wim.taymans@gmail.com>
22046 gst/playback/gstplaybasebin.c: Improve debug.
22047 Original commit message from CVS:
22048 * gst/playback/gstplaybasebin.c: (check_queue_event):
22050 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
22051 Fix width and height range from 16 - 4096 to 1 - MAXINT, just like the
22052 padtemplate caps. Refixes #357577.
22054 2006-11-28 16:21:27 +0000 Wim Taymans <wim.taymans@gmail.com>
22056 gst/playback/gstplaybasebin.c: Add event probe to see when EOS is in a queue and we can disable the underrun signals....
22057 Original commit message from CVS:
22058 * gst/playback/gstplaybasebin.c: (check_queue_event),
22059 (queue_threshold_reached), (queue_out_of_data),
22060 (gen_preroll_element):
22061 Add event probe to see when EOS is in a queue and we can disable the
22062 underrun signals. Fixes #357577.
22064 2006-11-28 14:40:39 +0000 Edward Hervey <bilboed@bilboed.com>
22066 gst/playback/: New decodebin2 element.
22067 Original commit message from CVS:
22068 * gst/playback/Makefile.am:
22069 * gst/playback/gstdecodebin2.c: (gst_decode_bin_get_type),
22070 (_gst_boolean_accumulator), (gst_decode_bin_class_init),
22071 (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
22072 (gst_decode_bin_init), (gst_decode_bin_dispose),
22073 (gst_decode_bin_finalize), (gst_decode_bin_set_property),
22074 (gst_decode_bin_get_property), (gst_decode_bin_set_caps),
22075 (gst_decode_bin_get_caps), (gst_decode_bin_autoplug_continue),
22076 (gst_decode_bin_autoplug_sort), (analyze_new_pad), (connect_pad),
22077 (connect_element), (expose_pad), (type_found),
22078 (pad_added_group_cb), (pad_removed_group_cb),
22079 (no_more_pads_group_cb), (pad_added_cb), (pad_removed_cb),
22080 (no_more_pads_cb), (find_compatibles), (is_demuxer_element),
22081 (are_raw_caps), (multi_queue_overrun_cb),
22082 (multi_queue_underrun_cb), (gst_decode_group_new),
22083 (get_current_group), (group_demuxer_event_probe),
22084 (gst_decode_group_control_demuxer_pad),
22085 (gst_decode_group_control_source_pad),
22086 (gst_decode_group_check_if_blocked),
22087 (gst_decode_group_check_if_drained), (gst_decode_group_expose),
22088 (gst_decode_group_hide), (gst_decode_group_free),
22089 (gst_decode_group_set_complete), (source_pad_blocked_cb),
22090 (source_pad_event_probe), (gst_decode_pad_new), (add_fakesink),
22091 (remove_fakesink), (find_sink_pad), (gst_decode_bin_change_state),
22093 New decodebin2 element.
22095 * gst/playback/gstplay-marshal.list:
22096 Added marshallers for new signals in decodebin2
22097 * gst/playback/gstplaybasebin.c: (setup_subtitle), (make_decoder):
22098 Use decodebin2 if *and only if* the USE_DECODEBIN2 environment variable
22101 2006-11-28 10:45:40 +0000 Wim Taymans <wim.taymans@gmail.com>
22103 gst/playback/gstplaybasebin.c: Disable rtsp:// uris for the release, it's not good enough yet.
22104 Original commit message from CVS:
22105 * gst/playback/gstplaybasebin.c: (setup_source),
22106 (gst_play_base_bin_change_state):
22107 Disable rtsp:// uris for the release, it's not good enough yet.
22110 2006-11-26 16:39:41 +0000 Wim Taymans <wim.taymans@gmail.com>
22112 ext/theora/theoradec.c: Implement reverse playback.
22113 Original commit message from CVS:
22114 * ext/theora/theoradec.c: (gst_theora_dec_reset),
22115 (theora_dec_push_forward), (theora_dec_push_reverse),
22116 (theora_handle_data_packet), (theora_dec_decode_buffer),
22117 (theora_dec_flush_decode), (theora_dec_chain_reverse),
22118 (theora_dec_chain_forward), (theora_dec_chain):
22119 Implement reverse playback.
22120 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
22121 (vorbis_dec_decode_buffer), (vorbis_dec_flush_decode),
22122 (vorbis_dec_chain_forward):
22123 Clear buffers used for reverse playback in _reset.
22124 No need to set the eos flag, we clip samples using the segment.
22126 2006-11-24 15:40:58 +0000 Wim Taymans <wim.taymans@gmail.com>
22128 ext/ogg/gstoggdemux.c: Some cleanups.
22129 Original commit message from CVS:
22130 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
22131 (gst_ogg_pad_init), (gst_ogg_pad_dispose), (gst_ogg_pad_reset),
22132 (gst_ogg_pad_stream_out), (gst_ogg_pad_submit_page),
22133 (gst_ogg_chain_reset), (gst_ogg_demux_perform_seek):
22135 Handle continued pages in reverse mode.
22137 2006-11-24 15:39:03 +0000 Wim Taymans <wim.taymans@gmail.com>
22139 ext/vorbis/vorbisdec.c: Small cleanups.
22140 Original commit message from CVS:
22141 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
22142 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
22143 (vorbis_dec_flush_decode):
22145 Don't try to add invalid timestamps.
22146 Clipping will unref the buffer.
22148 2006-11-24 08:56:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22150 gst/: remove obsolete _factory_init protos
22151 Original commit message from CVS:
22152 * gst/adder/gstadder.h:
22153 * gst/audiotestsrc/gstaudiotestsrc.h:
22154 remove obsolete _factory_init protos
22156 2006-11-24 08:35:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22158 sys/xvimage/xvimagesink.c: Fix spacing in debug message.
22159 Original commit message from CVS:
22160 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
22161 Fix spacing in debug message.
22163 2006-11-23 11:07:23 +0000 Wim Taymans <wim.taymans@gmail.com>
22165 ext/ogg/gstoggdemux.c: Don't just ignore return values from _pad_push().
22166 Original commit message from CVS:
22167 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
22168 (gst_ogg_demux_chain):
22169 Don't just ignore return values from _pad_push().
22170 Small debug improvements.
22172 2006-11-23 11:02:11 +0000 Michael Smith <msmith@xiph.org>
22174 ext/ogg/gstoggmux.c: If our incoming buffer is marked as DISCONT, then increment the page number (so that the discont...
22175 Original commit message from CVS:
22176 * ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
22177 If our incoming buffer is marked as DISCONT, then increment the page
22178 number (so that the discontinuity is marked in the final ogg
22179 bitstream) and flush the previous page.
22181 2006-11-22 14:34:03 +0000 Michael Smith <msmith@xiph.org>
22183 ext/theora/: Mark discontinuities of > 3/4 of a frame, reinit encoder.
22184 Original commit message from CVS:
22185 * ext/theora/gsttheoraenc.h:
22186 * ext/theora/theoraenc.c: (gst_theora_enc_init),
22187 (theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
22188 (theora_buffer_from_packet), (theora_enc_is_discontinuous),
22189 (theora_enc_chain), (theora_enc_change_state):
22190 Mark discontinuities of > 3/4 of a frame, reinit encoder.
22191 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
22192 (GST_START_TEST), (theoraenc_suite):
22193 Enable discontinuity test, fix it.
22195 2006-11-21 18:39:34 +0000 Tim-Philipp Müller <tim@centricular.net>
22197 ext/pango/gsttextoverlay.*: Some textoverlay fixes: for one, in the video chain function, actually wait for a text bu...
22198 Original commit message from CVS:
22199 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
22200 (gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
22201 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
22202 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
22203 (gst_text_overlay_change_state):
22204 * ext/pango/gsttextoverlay.h:
22205 Some textoverlay fixes: for one, in the video chain function,
22206 actually wait for a text buffer to come in if there is none at the
22207 moment and there should be one; also, deal more gracefully with
22208 incoming buffers that do not have a timestamp or duration; discard
22209 text buffer when not needed any longer. Fixes #341681.
22210 * tests/check/Makefile.am:
22211 * tests/check/elements/.cvsignore:
22212 * tests/check/elements/textoverlay.c:
22213 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
22214 (setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
22215 (create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
22216 (test_video_waits_for_text_send_text_newsegment_thread),
22217 (test_video_waits_for_text_shutdown_element),
22218 (test_render_continuity_push_video_buffers_thread),
22219 (textoverlay_suite):
22220 Add some unit tests for textoverlay.
22222 2006-11-21 09:29:56 +0000 Tim-Philipp Müller <tim@centricular.net>
22224 gst/typefind/gsttypefindfunctions.c: Avoid integer underflow when the found probability for mp3 is smaller than the '...
22225 Original commit message from CVS:
22226 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
22227 Avoid integer underflow when the found probability for mp3 is
22228 smaller than the 'penalty' we subtract if there's not a clean
22229 mp3 header sync at offset 0.
22231 2006-11-21 08:17:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22233 docs/libs/gst-plugins-base-libs-sections.txt: Add some new symbols to the docs
22234 Original commit message from CVS:
22235 * docs/libs/gst-plugins-base-libs-sections.txt:
22236 Add some new symbols to the docs
22238 2006-11-20 16:44:28 +0000 Tim-Philipp Müller <tim@centricular.net>
22240 tests/check/: Enable ffmpegcolorspace test now that the RGBA32 issue is fixed (for now not for valgrinding though, si...
22241 Original commit message from CVS:
22242 * tests/check/Makefile.am:
22243 * tests/check/elements/ffmpegcolorspace.c:
22244 (ffmpegcolorspace_suite):
22245 Enable ffmpegcolorspace test now that the RGBA32 issue is fixed
22246 (for now not for valgrinding though, since it takes too long).
22248 2006-11-20 15:01:09 +0000 Wim Taymans <wim.taymans@gmail.com>
22250 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix RGBA32 caps. Fixes #357038.
22251 Original commit message from CVS:
22252 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
22253 (gst_ffmpeg_pixfmt_to_caps):
22254 Fix RGBA32 caps. Fixes #357038.
22256 2006-11-20 12:20:39 +0000 Tim-Philipp Müller <tim@centricular.net>
22258 gst-libs/gst/interfaces/mixertrack.h: Add FIXME so we can add some padding here in 0.11
22259 Original commit message from CVS:
22260 * gst-libs/gst/interfaces/mixertrack.h:
22261 Add FIXME so we can add some padding here in 0.11
22263 2006-11-19 17:07:34 +0000 Tim-Philipp Müller <tim@centricular.net>
22265 gst-libs/gst/rtp/gstbasertpaudiopayload.h: Fix GstBaseRTPAudioPayload structure so the whole GObject inheritance busi...
22266 Original commit message from CVS:
22267 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
22268 Fix GstBaseRTPAudioPayload structure so the whole GObject
22269 inheritance business actually works (parent class instance structure
22270 must always come first in the derived class instance structure).
22272 2006-11-16 14:35:30 +0000 Tim-Philipp Müller <tim@centricular.net>
22274 Make sure our checks and the videotestsrc plugin link against the local uninstalled gst libs and not any installed gs...
22275 Original commit message from CVS:
22276 * gst/videotestsrc/Makefile.am:
22277 * tests/check/Makefile.am:
22278 Make sure our checks and the videotestsrc plugin link against the
22279 local uninstalled gst libs and not any installed gst libs that
22280 might happen to exist as well.
22281 * tests/check/elements/adder.c: (message_received),
22282 (test_event_message_received), (test_play_twice_message_received):
22283 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
22284 Fix compiler warnings when compiling against core with disabled
22287 2006-11-16 12:55:08 +0000 Michael Smith <msmith@xiph.org>
22289 gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
22290 Original commit message from CVS:
22291 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
22292 (gst_audio_rate_sink_event), (gst_audio_rate_chain):
22293 Fix audiorate, so that it accurately sets offsets and timestamps.
22294 Doesn't change the fundamental algorithmic decisions; so should be
22296 * tests/check/Makefile.am:
22297 Enable audiorate test now that it passes.
22299 2006-11-15 10:05:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22301 sys/xvimage/xvimagesink.c: clear xv when going to NULL, remove // commented non-existant proto
22302 Original commit message from CVS:
22303 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
22304 clear xv when going to NULL, remove // commented non-existant proto
22305 * tests/examples/seek/seek.c: (main):
22306 add missing tooltip description for scrub and play_scrub
22308 2006-11-14 23:34:19 +0000 David Schleef <ds@schleef.org>
22310 configure.ac: Bump liboil requirement to 0.3.8.
22311 Original commit message from CVS:
22313 Bump liboil requirement to 0.3.8.
22314 * gst-libs/gst/riff/riff-media.c:
22316 * gst/videoscale/vs_image.h:
22317 * gst/videoscale/vs_scanline.h:
22318 Use liboil's stdint.h.
22319 * gst/videotestsrc/videotestsrc.c:
22320 Remove liboil related ifdef's, since they aren't needed now, and
22321 won't work with future versions.
22323 2006-11-14 23:08:38 +0000 David Schleef <ds@schleef.org>
22325 gst/videoscale/: Add a 4-tap image scaler. Theoretically looks much prettier.
22326 Original commit message from CVS:
22327 * gst/videoscale/Makefile.am:
22328 * gst/videoscale/gstvideoscale.c:
22329 * gst/videoscale/gstvideoscale.h:
22330 * gst/videoscale/vs_4tap.c:
22331 * gst/videoscale/vs_4tap.h:
22332 * gst/videoscale/vs_image.c:
22333 * gst/videoscale/vs_image.h:
22334 * gst/videoscale/vs_scanline.c:
22335 * gst/videoscale/vs_scanline.h:
22336 Add a 4-tap image scaler. Theoretically looks much prettier.
22337 The tap calculation could use some improvement.
22339 2006-11-14 11:54:14 +0000 Jan David Mol <j.j.d.mol@tudelft.nl>
22341 Various gsize and gssize printf fixes. Fixes #372507.
22342 Original commit message from CVS:
22343 Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
22344 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
22345 (gst_riff_parse_strf_iavs):
22346 * gst/subparse/gstsubparse.c: (convert_encoding):
22347 * gst/tcp/gstmultifdsink.c:
22348 (gst_multi_fd_sink_handle_client_write):
22349 * gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
22350 (gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
22351 (gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
22352 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
22353 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
22354 (gst_ximagesink_ximage_new):
22355 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
22356 Various gsize and gssize printf fixes. Fixes #372507.
22358 2006-11-13 18:14:48 +0000 Wim Taymans <wim.taymans@gmail.com>
22360 ext/vorbis/vorbisdec.*: First stab at vorbis reverse playback.
22361 Original commit message from CVS:
22362 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
22363 (vorbis_dec_push_forward), (vorbis_dec_push_reverse),
22364 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
22365 (vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
22366 (vorbis_dec_chain_forward), (vorbis_dec_chain):
22367 * ext/vorbis/vorbisdec.h:
22368 First stab at vorbis reverse playback.
22370 2006-11-13 17:30:17 +0000 Wim Taymans <wim.taymans@gmail.com>
22372 gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
22373 Original commit message from CVS:
22374 * gst-libs/gst/audio/gstbaseaudiosink.c:
22375 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
22376 * gst-libs/gst/audio/gstbaseaudiosink.h:
22377 Make the clock sync code more accurate wrt resampling and playback
22378 at different rates.
22379 * gst-libs/gst/audio/gstringbuffer.c:
22380 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
22381 * gst-libs/gst/audio/gstringbuffer.h:
22382 Use better algorithm to interpolate sample rates.
22384 2006-11-13 15:31:01 +0000 Michael Smith <msmith@xiph.org>
22386 ext/ogg/gstoggdemux.c: Improve a debug line slightly.
22387 Original commit message from CVS:
22388 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
22389 Improve a debug line slightly.
22390 * ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
22391 Call gst_riff_init() in plugin_init, to avoid getting errors from
22392 the debug system (unrelated changes to another plugin made this turn
22395 2006-11-10 19:20:21 +0000 Sergey Scobich <sergery.scobich@gmail.com>
22397 win32/common/libgsttag.def: Add missing symbol (#366492).
22398 Original commit message from CVS:
22399 Patch by: Sergey Scobich <sergery.scobich at gmail com>
22400 * win32/common/libgsttag.def:
22401 Add missing symbol (#366492).
22403 2006-11-10 00:52:55 +0000 Tim-Philipp Müller <tim@centricular.net>
22405 gst/playback/gststreamselector.c: Don't unref a NULL pad.
22406 Original commit message from CVS:
22407 * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
22408 Don't unref a NULL pad.
22410 2006-11-09 00:50:00 +0000 Christian Schaller <uraeus@gnome.org>
22412 ext/ogg/gstoggdemux.c: Implement first stab at reverse playback.
22413 Original commit message from CVS:
22414 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
22415 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_perform_seek),
22416 (gst_ogg_demux_handle_page), (gst_ogg_demux_chain),
22417 (gst_ogg_demux_loop_forward), (gst_ogg_demux_loop_reverse),
22418 (gst_ogg_demux_loop):
22419 Implement first stab at reverse playback.
22421 2006-11-07 07:22:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22423 gst-libs/gst/riff/riff-media.c: add h263/h264 variants to the caps, Fixes #363118
22424 Original commit message from CVS:
22425 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
22426 (gst_riff_create_video_template_caps):
22427 add h263/h264 variants to the caps, Fixes #363118
22429 2006-11-06 18:24:59 +0000 Tim-Philipp Müller <tim@centricular.net>
22431 gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
22432 Original commit message from CVS:
22433 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
22434 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
22435 Use g_strerror instead of strerror so we get UTF-8.
22437 2006-11-04 07:25:58 +0000 David Schleef <ds@schleef.org>
22439 ext/ogg/: Add/remove KW-DIRAC header here, since it is ogg-specific.
22440 Original commit message from CVS:
22441 * ext/ogg/gstoggdemux.c:
22442 * ext/ogg/gstoggmux.c:
22443 Add/remove KW-DIRAC header here, since it is ogg-specific.
22445 2006-11-03 15:44:31 +0000 Michael Smith <msmith@xiph.org>
22447 gst/typefind/gsttypefindfunctions.c: Recognise more mpeg4 elementary video streams.
22448 Original commit message from CVS:
22449 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
22450 Recognise more mpeg4 elementary video streams.
22452 2006-11-02 17:26:03 +0000 Edward Hervey <bilboed@bilboed.com>
22454 gst/typefind/gsttypefindfunctions.c: Lower the probability of mp3 typefinding functions if we don't find a valid mp3 ...
22455 Original commit message from CVS:
22456 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
22457 Lower the probability of mp3 typefinding functions if we don't find a
22458 valid mp3 header at the start of the file.
22461 2006-11-02 15:06:36 +0000 Wim Taymans <wim.taymans@gmail.com>
22463 ext/theora/: Document and partially implement an algorithm for doing reverse playback of theora video.
22464 Original commit message from CVS:
22465 * ext/theora/gsttheoradec.h:
22466 * ext/theora/theoradec.c: (gst_theora_dec_init),
22467 (theora_dec_sink_event), (theora_dec_chain_forward),
22468 (theora_dec_flush_decode), (theora_dec_chain_reverse),
22469 (theora_dec_chain):
22470 Document and partially implement an algorithm for doing reverse playback
22473 2006-11-02 14:18:45 +0000 Sergey Scobich <sergey.scobich@gmail.com>
22475 win32/: Misc. VS8 build fixes: fix syntax in config.h, add missing entries to libgsttag.def; add missing dependencies...
22476 Original commit message from CVS:
22477 Patch by: Sergey Scobich <sergey.scobich at gmail com>
22478 * win32/common/config.h:
22479 * win32/common/interfaces-enumtypes.c:
22480 * win32/common/libgsttag.def:
22481 * win32/vs8/gst-plugins-base.sln:
22482 * win32/vs8/libgstaudioresample.vcproj:
22483 * win32/vs8/libgstinterfaces.vcproj:
22484 * win32/vs8/libgstogg.vcproj:
22485 * win32/vs8/libgstriff.vcproj:
22486 * win32/vs8/libgsttag.vcproj:
22487 * win32/vs8/libgsttheora.vcproj:
22488 * win32/vs8/libgstvideoscale.vcproj:
22489 * win32/vs8/libgstvorbis.vcproj:
22490 Misc. VS8 build fixes: fix syntax in config.h, add missing entries
22491 to libgsttag.def; add missing dependencies for some vs8 projects;
22492 re-arrange placement of .def files in vs8 projects (#366334).
22494 2006-11-01 14:08:31 +0000 Tim-Philipp Müller <tim@centricular.net>
22496 ext/ogg/gstogg.c: Remove unused variable.
22497 Original commit message from CVS:
22498 * ext/ogg/gstogg.c:
22499 Remove unused variable.
22500 * ext/ogg/gstoggdemux.c:
22501 Fix Wim's surname in plugin description.
22503 2006-10-31 15:05:33 +0000 Wim Taymans <wim.taymans@gmail.com>
22505 gst-plugins-base.spec.in: spec new .h file. Fixes #368310.
22506 Original commit message from CVS:
22507 * gst-plugins-base.spec.in:
22508 spec new .h file. Fixes #368310.
22510 2006-10-31 14:19:07 +0000 Michael Smith <msmith@xiph.org>
22512 gst/tcp/gstmultifdsink.*: Make using the remove or clear signals threadsafe.
22513 Original commit message from CVS:
22514 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
22515 (gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
22516 (gst_multi_fd_sink_get_stats),
22517 (gst_multi_fd_sink_remove_client_link),
22518 (gst_multi_fd_sink_queue_buffer),
22519 (gst_multi_fd_sink_handle_clients):
22520 * gst/tcp/gstmultifdsink.h:
22521 Make using the remove or clear signals threadsafe.
22522 Make calling get-stats with an invalid fd not segfault.
22525 2006-10-31 10:49:19 +0000 Wim Taymans <wim.taymans@gmail.com>
22527 gst-libs/gst/rtp/: Fix and activate base audio payloader.
22528 Original commit message from CVS:
22529 * gst-libs/gst/rtp/Makefile.am:
22530 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
22531 (gst_base_rtp_audio_payload_init):
22532 Fix and activate base audio payloader.
22534 2006-10-28 17:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
22536 gst/typefind/gsttypefindfunctions.c: Add typefinder for QuickTime Image Files (see #366156).
22537 Original commit message from CVS:
22538 * gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
22540 Add typefinder for QuickTime Image Files (see #366156).
22542 2006-10-28 16:00:51 +0000 Tim-Philipp Müller <tim@centricular.net>
22544 gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
22545 Original commit message from CVS:
22546 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
22547 Another typo fix (#366212).
22549 2006-10-27 17:13:35 +0000 Wim Taymans <wim.taymans@gmail.com>
22551 gst/volume/gstvolume.c: Use stream time to synchronize volume property instead of rather random timestamps. This is n...
22552 Original commit message from CVS:
22553 * gst/volume/gstvolume.c: (volume_transform_ip):
22554 Use stream time to synchronize volume property instead of rather random
22555 timestamps. This is needed when gnonlin does its time shifting.
22557 2006-10-27 16:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
22560 I'm too lazy to comment this
22561 Original commit message from CVS:
22562 *** empty log message ***
22564 2006-10-27 16:45:30 +0000 Mark Nauwelaerts <manauw@skynet.be>
22566 ext/ogg/gstoggmux.c: Remove the pad from the element in release_pad.
22567 Original commit message from CVS:
22568 Patch by: Mark Nauwelaerts <manauw at skynet dot be>
22569 * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
22570 Remove the pad from the element in release_pad.
22572 2006-10-27 11:57:18 +0000 Tim-Philipp Müller <tim@centricular.net>
22574 sys/: Explicitly create our custom buffer classes at a thread-safe location as well, since g_type_class_ref() doesn't...
22575 Original commit message from CVS:
22576 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
22577 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
22578 Explicitly create our custom buffer classes at a thread-safe
22579 location as well, since g_type_class_ref() doesn't seem to be
22580 entirely thread-safe either (#365501; also see #349410).
22582 2006-10-26 10:49:00 +0000 Tim-Philipp Müller <tim@centricular.net>
22584 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...
22585 Original commit message from CVS:
22586 * gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
22587 (gst_riff_parse_info):
22588 If strings in INFO chunk are not UTF-8, do something similar to
22589 what we do for ID3v1 tags: check a number of environment variables
22590 (GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
22591 character sets to try, otherwise try the current locale and/or fall
22592 back on ISO-8859-1. Fixes #360552.
22594 2006-10-23 12:46:41 +0000 Tim-Philipp Müller <tim@centricular.net>
22596 gst/videotestsrc/: Add a bunch of exciting new checkers patterns.
22597 Original commit message from CVS:
22598 * gst/videotestsrc/gstvideotestsrc.c:
22599 (gst_video_test_src_pattern_get_type),
22600 (gst_video_test_src_set_pattern):
22601 * gst/videotestsrc/gstvideotestsrc.h:
22602 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_checkers1),
22603 (gst_video_test_src_checkers2), (gst_video_test_src_checkers4),
22604 (gst_video_test_src_checkers8):
22605 * gst/videotestsrc/videotestsrc.h:
22606 Add a bunch of exciting new checkers patterns.
22608 2006-10-23 12:06:44 +0000 Tim-Philipp Müller <tim@centricular.net>
22610 gst/subparse/: Add support for TMPlayer-type subtitles (#362845).
22611 Original commit message from CVS:
22612 * gst/subparse/Makefile.am:
22613 * gst/subparse/gstsubparse.c:
22614 (gst_sub_parse_data_format_autodetect),
22615 (gst_sub_parse_format_autodetect), (handle_buffer),
22616 (gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
22617 * gst/subparse/gstsubparse.h:
22618 * gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
22620 * gst/subparse/tmplayerparse.h:
22621 Add support for TMPlayer-type subtitles (#362845).
22622 * tests/check/elements/subparse.c: (test_tmplayer_do_test),
22623 (GST_START_TEST), (subparse_suite):
22624 Add some basic unit tests for the above.
22626 2006-10-23 11:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
22628 tests/check/elements/audiorate.c: More tests for audiorate: inject buffers to check behaviour when buffers overlap.
22629 Original commit message from CVS:
22630 * tests/check/elements/audiorate.c: (test_injector_base_init),
22631 (test_injector_class_init), (test_injector_chain),
22632 (test_injector_init), (probe_cb), (do_perfect_stream_test),
22633 (GST_START_TEST), (audiorate_suite):
22634 More tests for audiorate: inject buffers to check behaviour when
22637 2006-10-21 16:39:54 +0000 Tim-Philipp Müller <tim@centricular.net>
22639 tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...
22640 Original commit message from CVS:
22641 * tests/check/Makefile.am:
22642 * tests/check/elements/.cvsignore:
22643 * tests/check/elements/audiorate.c: (probe_cb), (got_buf),
22644 (do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
22645 Add some basic unit tests for audiorate. Disabled at the moment
22646 since it doesn't pass yet (see bug #363119).
22648 2006-10-20 17:02:19 +0000 Tim-Philipp Müller <tim@centricular.net>
22650 gst/subparse/gstsubparse.c: Add missing closing tags for markup and fix broken markup, otherwise pango won't render a...
22651 Original commit message from CVS:
22652 * gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
22653 (parse_subrip), (handle_buffer):
22654 Add missing closing tags for markup and fix broken markup,
22655 otherwise pango won't render anything (fixes #357531). Also,
22656 make sure the text we send out is always NUL-terminated
22657 (better safe than sorry etc.).
22658 * tests/check/elements/subparse.c: (test_srt_do_test),
22660 Some more tests for .srt incl. tests for the above stuff.
22662 2006-10-20 13:56:55 +0000 Stefan Kost <ensonic@users.sf.net>
22664 sys/: Try to redraw borders only when needed. Apparently this consumes resources on small devices... :-O (#363607)
22665 Original commit message from CVS:
22666 2006-10-20 Julien MOUTTE <julien@moutte.net>
22667 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
22668 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
22669 Patch by: Stefan Kost <ensonic@users.sf.net>
22670 Try to redraw borders only when needed. Apparently this consumes
22671 resources on small devices... :-O (#363607)
22673 2006-10-20 13:54:19 +0000 Michael Smith <msmith@xiph.org>
22675 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...
22676 Original commit message from CVS:
22677 * gst/tcp/gstmultifdsink.c:
22678 (gst_multi_fd_sink_client_queue_buffer):
22679 If caps change, then update the client's idea of the caps so that we
22680 don't end up re-sending streamheaders for every single buffer after
22683 2006-10-20 12:31:02 +0000 Michael Smith <msmith@xiph.org>
22685 ext/ogg/gstoggparse.c: Set caps on pushed buffers; fix up refcounting of caps objects.
22686 Original commit message from CVS:
22687 * ext/ogg/gstoggparse.c: (gst_ogg_parse_dispose),
22688 (gst_ogg_parse_append_header), (gst_ogg_parse_chain):
22689 Set caps on pushed buffers; fix up refcounting of caps objects.
22691 2006-10-19 14:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
22693 gst/typefind/gsttypefindfunctions.c: Typefind mmsh header data packet to application/x-mmsh (#362625).
22694 Original commit message from CVS:
22695 * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
22697 Typefind mmsh header data packet to application/x-mmsh (#362625).
22699 2006-10-19 09:17:48 +0000 Tim-Philipp Müller <tim@centricular.net>
22701 tests/check/: Add very simple unit test for subparse.
22702 Original commit message from CVS:
22703 * tests/check/Makefile.am:
22704 * tests/check/elements/.cvsignore:
22705 * tests/check/elements/subparse.c: (buffer_from_static_string),
22706 (setup_subparse), (teardown_subparse), (test_srt_do_test),
22707 (GST_START_TEST), (subparse_suite):
22708 Add very simple unit test for subparse.
22710 2006-10-19 09:00:21 +0000 Tim-Philipp Müller <tim@centricular.net>
22712 gst/subparse/gstsubparse.c: Strip trailing newlines from subtitle text output.
22713 Original commit message from CVS:
22714 * gst/subparse/gstsubparse.c: (strip_trailing_newlines),
22716 Strip trailing newlines from subtitle text output.
22718 2006-10-18 18:40:12 +0000 Tim-Philipp Müller <tim@centricular.net>
22720 gst/subparse/gstsubparse.c: Fix memleak; clear subparse->textbuf n state change function.
22721 Original commit message from CVS:
22722 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
22723 (gst_sub_parse_change_state):
22724 Fix memleak; clear subparse->textbuf n state change function.
22726 2006-10-18 15:13:59 +0000 Tim-Philipp Müller <tim@centricular.net>
22728 gst/subparse/gstsubparse.c: Don't require subrip (.srt) files to start with a chunk number of 1.
22729 Original commit message from CVS:
22730 * gst/subparse/gstsubparse.c:
22731 (gst_sub_parse_data_format_autodetect):
22732 Don't require subrip (.srt) files to start with a chunk number of 1.
22734 2006-10-18 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
22736 gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
22737 Original commit message from CVS:
22738 * gst-libs/gst/audio/gstbaseaudiosink.c:
22739 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
22740 * gst-libs/gst/audio/gstbaseaudiosink.h:
22741 Extract rate from the NEWSEGMENT event.
22742 Use commit_full to also take rate adjustment into account when writing
22743 samples to the ringbuffer.
22744 * gst-libs/gst/audio/gstringbuffer.c:
22745 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
22746 (gst_ring_buffer_read):
22747 * gst-libs/gst/audio/gstringbuffer.h:
22748 Added _commit_full() to also take rate into account.
22749 Use simple interpolation algorithm to resample audio.
22750 API: gst_ring_buffer_commit_full()
22751 * tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
22752 * tests/examples/seek/seek.c: (segment_done):
22753 Don't try to seek with 0.0 rate, just pause instead.
22754 Remove bogus debug line.
22756 2006-10-18 12:57:54 +0000 Tim-Philipp Müller <tim@centricular.net>
22758 gst/playback/gstplaybasebin.c: Catch async errors when starting up the subtitle bin, so we can stop waiting and conti...
22759 Original commit message from CVS:
22760 * gst/playback/gstplaybasebin.c: (subbin_startup_sync_msg),
22762 Catch async errors when starting up the subtitle bin, so we can
22763 stop waiting and continue with the main film instead of hanging
22764 forever. Fixes #339366.
22765 * tests/check/elements/playbin.c: (playbin_suite):
22766 Enable unit test for the above.
22768 2006-10-18 09:53:03 +0000 Tim-Philipp Müller <tim@centricular.net>
22770 tests/check/: Some small and basic unit tests for playbin; not very useful yet, but at least a start.
22771 Original commit message from CVS:
22772 * tests/check/Makefile.am:
22773 * tests/check/elements/.cvsignore:
22774 * tests/check/elements/playbin.c: (GST_START_TEST),
22775 (gst_red_video_src_uri_get_type),
22776 (gst_red_video_src_uri_get_protocols),
22777 (gst_red_video_src_uri_get_uri), (gst_red_video_src_uri_set_uri),
22778 (gst_red_video_src_uri_handler_init),
22779 (gst_red_video_src_init_type), (gst_red_video_src_base_init),
22780 (gst_red_video_src_create), (gst_red_video_src_class_init),
22781 (gst_red_video_src_init), (plugin_init), (playbin_suite):
22782 Some small and basic unit tests for playbin; not very useful yet,
22783 but at least a start.
22785 2006-10-18 09:46:35 +0000 Tim-Philipp Müller <tim@centricular.net>
22787 gst/playback/gstplaybin.c: The old pad activation spiel.
22788 Original commit message from CVS:
22789 * gst/playback/gstplaybin.c: (setup_sinks):
22790 The old pad activation spiel.
22792 2006-10-18 09:31:49 +0000 Tim-Philipp Müller <tim@centricular.net>
22794 gst/playback/gstplaybasebin.c: Don't hang forever if the subbin already fails to start up in the state change to PAUS...
22795 Original commit message from CVS:
22796 * gst/playback/gstplaybasebin.c: (setup_source):
22797 Don't hang forever if the subbin already fails to start up in
22798 the state change to PAUSED (#339366).
22800 2006-10-17 17:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
22802 gst-libs/gst/interfaces/tuner.c: Fix some function guards, add some more function guards.
22803 Original commit message from CVS:
22804 * gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
22805 (gst_tuner_set_channel), (gst_tuner_get_channel),
22806 (gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
22807 (gst_tuner_set_frequency), (gst_tuner_get_frequency),
22808 (gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
22809 (gst_tuner_find_channel_by_name):
22810 Fix some function guards, add some more function guards.
22812 2006-10-17 11:34:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22814 gst/playback/gstdecodebin.c: Don't return a pad from get_our_ghost_pad unless it is actually the one we want.
22815 Original commit message from CVS:
22816 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
22817 (remove_element_chain):
22818 Don't return a pad from get_our_ghost_pad unless it is actually the
22820 Change a cast in remove_element_chain slightly.
22822 2006-10-13 15:20:29 +0000 Julien Moutte <julien@moutte.net>
22824 tests/examples/seek/seek.c: Segment seeking needs to use the rate and set stop to -1.
22825 Original commit message from CVS:
22826 2006-10-13 Julien MOUTTE <julien@moutte.net>
22827 * tests/examples/seek/seek.c: (do_seek), (start_seek),
22828 (rate_spinbutton_changed_cb), (segment_done),
22829 (msg_state_changed):
22830 Segment seeking needs to use the rate and set stop to -1.
22832 2006-10-13 14:15:42 +0000 Ville Syrjala <ville.syrjala@movial.fi>
22834 gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
22835 Original commit message from CVS:
22836 * gst-libs/gst/audio/gstbaseaudiosink.c:
22837 (gst_base_audio_sink_setcaps):
22838 Don't crash when ringbuffer is not yet created.
22839 Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
22841 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
22842 * gst/playback/gststreamselector.c:
22843 (gst_stream_selector_request_new_pad):
22844 Activate pads befre adding them to running elements.
22846 2006-10-13 11:25:10 +0000 Julien Moutte <julien@moutte.net>
22848 tests/examples/seek/seek.c: Stop the scale updater when we start grabing the slider. Don't wait for the pipeline to b...
22849 Original commit message from CVS:
22850 2006-10-13 Julien MOUTTE <julien@moutte.net>
22851 * tests/examples/seek/seek.c: (do_seek), (start_seek),
22852 (rate_spinbutton_changed_cb), (msg_state_changed): Stop the
22854 updater when we start grabing the slider. Don't wait for the
22855 pipeline to be PAUSED.
22857 2006-10-13 08:57:52 +0000 Tim-Philipp Müller <tim@centricular.net>
22859 gst-libs/gst/interfaces/mixer.c: Guard mixer interface functions against bogus arguments.
22860 Original commit message from CVS:
22861 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
22862 (gst_mixer_set_volume), (gst_mixer_get_volume),
22863 (gst_mixer_set_mute), (gst_mixer_set_option),
22864 (gst_mixer_get_option), (gst_mixer_mute_toggled),
22865 (gst_mixer_record_toggled), (gst_mixer_volume_changed),
22866 (gst_mixer_option_changed):
22867 Guard mixer interface functions against bogus arguments.
22869 2006-10-12 19:39:07 +0000 Julien Moutte <julien@moutte.net>
22871 tests/examples/seek/seek.c: Use state-changed messages to trigger start/stop of scale update timer. Indeed the scale ...
22872 Original commit message from CVS:
22873 2006-10-12 Julien MOUTTE <julien@moutte.net>
22874 * tests/examples/seek/seek.c: (do_seek), (start_seek),
22876 (play_cb), (pause_cb), (stop_cb),
22877 (rate_spinbutton_changed_cb),
22878 (msg_state_changed), (main): Use state-changed messages to
22880 start/stop of scale update timer. Indeed the scale slider was
22881 jumping here and there because the update timer was activated
22882 before seek completed. This fixes instant applying of rate
22884 by pressing the spinbutton like a crazy man !
22886 2006-10-12 19:09:06 +0000 Sebastien Cote <sebas642@yahoo.ca>
22888 gst-libs/gst/rtp/gstbasertppayload.c: Fix two small memory leaks (#361456).
22889 Original commit message from CVS:
22890 Patch by: Sebastien Cote <sebas642 at yahoo.ca>
22891 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
22892 (gst_basertppayload_finalize):
22893 Fix two small memory leaks (#361456).
22895 2006-10-10 18:56:01 +0000 Julien Moutte <julien@moutte.net>
22897 tests/examples/seek/seek.c: When changing spinbutton we try to change the rate on the fly.
22898 Original commit message from CVS:
22899 2006-10-10 Julien MOUTTE <julien@moutte.net>
22900 * tests/examples/seek/seek.c: (do_seek),
22901 (rate_spinbutton_changed_cb): When changing spinbutton we try
22902 to change the rate on the fly.
22904 2006-10-10 16:50:06 +0000 Wim Taymans <wim.taymans@gmail.com>
22906 gst-libs/gst/riff/: Add WMS caps.
22907 Original commit message from CVS:
22908 * gst-libs/gst/riff/riff-ids.h:
22909 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
22910 (gst_riff_create_audio_template_caps):
22913 2006-10-10 12:49:03 +0000 Josep Torre Valles <josep@fluendo.com>
22915 ext/gnomevfs/: Fix URI interface implementation return type.
22916 Original commit message from CVS:
22917 2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
22918 Patch by: Josep Torre Valles <josep@fluendo.com>
22919 * ext/gnomevfs/gstgnomevfssink.c:
22920 * ext/gnomevfs/gstgnomevfssrc.c:
22921 Fix URI interface implementation return type.
22922 * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
22923 Fix what looks like a copy/paste issue when assigning values.
22924 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
22925 (gst_audio_filter_template_get_type):
22926 Cast to prevent Forte warnings.
22927 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
22928 Fix URI interface implementation return type.
22929 gst_pad_query_position requires a signed integer pointer as
22930 3rd parameter, GstClockTime is unsigned.
22931 * gst/audioconvert/audioconvert.c:
22932 Fix integer overflow when treated as signed.
22933 * gst/audioresample/resample.c: (resample_add_input_data):
22934 Cast to prevent warnings on Forte.
22935 * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
22936 Fix integer overflow when treated as signed.
22937 * gst/ffmpegcolorspace/imgconvert_template.h:
22938 Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
22939 * gst/playback/gstdecodebin.c: (queue_filled_cb),
22940 (cleanup_decodebin):
22941 Who initialises a guint to -1!
22942 Cast function pointers to prevent warnings on Forte.
22943 * gst/playback/gstplaybasebin.c: (queue_deadlock_check),
22944 (queue_threshold_reached):
22945 Cast function pointers correctly to prevent warnings on Forte.
22946 * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
22947 Cast function pointers correctly to prevent warnings on Forte.
22948 * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
22949 Obvious change to unsigned, 0xEF > max signed char.
22950 * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
22951 GstClockTime is unsigned, initialise correctly.
22952 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
22953 Cast so pointer arithemetic doesn't cause warnings on Forte.
22954 * gst/videorate/gstvideorate.c:
22955 Use correct return value.
22956 * tests/examples/seek/scrubby.c:
22957 GstClockTime is unsigned, initialise correctly.
22959 2006-10-10 11:20:03 +0000 Ferenc Gerlits <fgerlits@gmail.com>
22961 gst/typefind/gsttypefindfunctions.c: Recognise XML files and XML-like files shorter than 256 bytes as well (fixes #35...
22962 Original commit message from CVS:
22963 Patch by: Ferenc Gerlits <fgerlits at gmail com>
22964 * gst/typefind/gsttypefindfunctions.c:
22965 Recognise XML files and XML-like files shorter than 256 bytes as
22966 well (fixes #359237).
22968 2006-10-09 15:01:30 +0000 Edgard Lima <edgard.lima@indt.org.br>
22972 * gst/typefind/gsttypefindfunctions.c:
22973 Added typefind functions to video/x-nuv media.
22974 Original commit message from CVS:
22975 Added typefind functions to video/x-nuv media.
22977 2006-10-08 16:59:31 +0000 Tim-Philipp Müller <tim@centricular.net>
22979 gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
22980 Original commit message from CVS:
22981 * gst-libs/gst/interfaces/xoverlay.c:
22982 (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
22983 Some more guards against invalid input.
22985 2006-10-07 18:35:39 +0000 Julien Moutte <julien@moutte.net>
22987 ext/pango/gsttextoverlay.c: Useless goto.
22988 Original commit message from CVS:
22989 2006-10-07 Julien MOUTTE <julien@moutte.net>
22990 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
22992 * tests/examples/seek/seek.c: (do_seek),
22993 (rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
22994 seek example to experiment with rates != 1.0 (reverse playback
22997 2006-10-06 19:20:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22999 gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
23000 Original commit message from CVS:
23001 * gst-libs/gst/interfaces/xoverlay.c:
23002 Unref message in doc-example (spotted by Robert McQueen)
23004 2006-10-06 17:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
23006 gst/typefind/gsttypefindfunctions.c: printf fix.
23007 Original commit message from CVS:
23008 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
23009 (mpeg1_parse_header), (mpeg1_sys_type_find):
23012 2006-10-06 14:37:43 +0000 Wim Taymans <wim.taymans@gmail.com>
23014 gst/playback/: Activate dynamic pads before adding them to the element.
23015 Original commit message from CVS:
23016 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
23018 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
23019 Activate dynamic pads before adding them to the element.
23021 2006-10-06 14:04:53 +0000 Michael Smith <msmith@xiph.org>
23023 gst-libs/gst/floatcast/floatcast.h: Fix obviously-bogus macros; use the correct types.
23024 Original commit message from CVS:
23025 * gst-libs/gst/floatcast/floatcast.h:
23026 Fix obviously-bogus macros; use the correct types.
23028 2006-10-06 13:34:46 +0000 Wim Taymans <wim.taymans@gmail.com>
23030 gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
23031 Original commit message from CVS:
23032 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23033 (gst_base_rtp_depayload_change_state):
23034 Also call parent state change function to activate pads.
23035 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
23036 (mpeg1_parse_header), (mpeg1_sys_type_find):
23037 Add some more debug info in mpeg typefinding.
23039 2006-10-06 12:57:10 +0000 Michael Smith <msmith@xiph.org>
23041 ext/theora/theoradec.c: Zero byte theora packets are valid and well-defined; don't warn on them.
23042 Original commit message from CVS:
23043 * ext/theora/theoradec.c: (theora_dec_chain):
23044 Zero byte theora packets are valid and well-defined; don't warn on
23047 2006-10-06 10:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23049 gst/tcp/gstmultifdsink.c: API: add dropped_buffers to the get-stats GValueArray
23050 Original commit message from CVS:
23051 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
23052 (gst_multi_fd_sink_get_stats), (find_limits),
23053 (gst_multi_fd_sink_queue_buffer):
23054 API: add dropped_buffers to the get-stats GValueArray
23056 2006-10-05 15:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
23058 Printf format fixes.
23059 Original commit message from CVS:
23060 * ext/alsa/gstalsadeviceprobe.c:
23061 (gst_alsa_device_property_probe_get_values):
23062 * ext/alsa/gstalsasink.c: (set_hwparams):
23063 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
23064 (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
23065 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
23066 (gst_ogg_mux_process_best_pad):
23067 * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
23068 (gst_ogg_parse_chain):
23069 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
23070 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
23071 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
23072 (gst_vorbis_enc_buffer_check_discontinuous):
23073 * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
23074 * gst-libs/gst/audio/gstbaseaudiosink.c:
23075 (gst_base_audio_sink_render):
23076 * gst-libs/gst/cdda/gstcddabasesrc.c:
23077 (gst_cdda_base_src_handle_track_seek):
23078 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23079 (gst_base_rtp_depayload_push_full):
23080 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
23081 * gst/audioresample/resample.c: (resample_input_pushthrough):
23082 * gst/playback/gstplaybasebin.c: (queue_out_of_data):
23083 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
23084 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
23085 (wavpack_type_find):
23086 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
23087 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
23088 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
23089 * tests/check/elements/volume.c: (GST_START_TEST):
23090 Printf format fixes.
23092 2006-10-04 13:18:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23094 gst/tcp/gsttcp.c: Fix a simple mistake (see the docs)
23095 Original commit message from CVS:
23096 * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
23097 Fix a simple mistake (see the docs)
23100 2006-10-04 13:15:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23102 * win32/common/config.h:
23104 Original commit message from CVS:
23107 2006-10-03 12:11:45 +0000 Tim-Philipp Müller <tim@centricular.net>
23109 docs/plugins/: Add vorbistag element to docs; update version numbers to 0.10.10.1.
23110 Original commit message from CVS:
23111 * docs/plugins/Makefile.am:
23112 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
23113 * docs/plugins/gst-plugins-base-plugins-sections.txt:
23114 * docs/plugins/gst-plugins-base-plugins.args:
23115 * docs/plugins/gst-plugins-base-plugins.hierarchy:
23116 * docs/plugins/inspect/plugin-adder.xml:
23117 * docs/plugins/inspect/plugin-alsa.xml:
23118 * docs/plugins/inspect/plugin-audioconvert.xml:
23119 * docs/plugins/inspect/plugin-audiorate.xml:
23120 * docs/plugins/inspect/plugin-audioresample.xml:
23121 * docs/plugins/inspect/plugin-audiotestsrc.xml:
23122 * docs/plugins/inspect/plugin-cdparanoia.xml:
23123 * docs/plugins/inspect/plugin-decodebin.xml:
23124 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
23125 * docs/plugins/inspect/plugin-gdp.xml:
23126 * docs/plugins/inspect/plugin-gnomevfs.xml:
23127 * docs/plugins/inspect/plugin-libvisual.xml:
23128 * docs/plugins/inspect/plugin-ogg.xml:
23129 * docs/plugins/inspect/plugin-pango.xml:
23130 * docs/plugins/inspect/plugin-playbin.xml:
23131 * docs/plugins/inspect/plugin-subparse.xml:
23132 * docs/plugins/inspect/plugin-tcp.xml:
23133 * docs/plugins/inspect/plugin-theora.xml:
23134 * docs/plugins/inspect/plugin-typefindfunctions.xml:
23135 * docs/plugins/inspect/plugin-video4linux.xml:
23136 * docs/plugins/inspect/plugin-videorate.xml:
23137 * docs/plugins/inspect/plugin-videoscale.xml:
23138 * docs/plugins/inspect/plugin-videotestsrc.xml:
23139 * docs/plugins/inspect/plugin-volume.xml:
23140 * docs/plugins/inspect/plugin-vorbis.xml:
23141 * docs/plugins/inspect/plugin-ximagesink.xml:
23142 * docs/plugins/inspect/plugin-xvimagesink.xml:
23143 Add vorbistag element to docs; update version numbers to 0.10.10.1.
23145 2006-10-03 11:51:48 +0000 James Doc Livingston <doclivingston@gmail.com>
23147 ext/vorbis/: Add new vorbistag element which derives from vorbisparse and is essentially the same as well, only that ...
23148 Original commit message from CVS:
23149 Patch by: James "Doc" Livingston <doclivingston at gmail com>
23150 * ext/vorbis/Makefile.am:
23151 * ext/vorbis/vorbis.c: (plugin_init):
23152 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_class_init),
23153 (vorbis_parse_parse_packet), (vorbis_parse_chain):
23154 * ext/vorbis/vorbisparse.h:
23155 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_base_init),
23156 (gst_vorbis_tag_class_init), (gst_vorbis_tag_init),
23157 (gst_vorbis_tag_parse_packet):
23158 * ext/vorbis/vorbistag.h:
23159 Add new vorbistag element which derives from vorbisparse
23160 and is essentially the same as well, only that it implements
23161 the GstTagSetter interface and can modify the stream's
23162 vorbiscomment on the fly (#335635).
23163 * tests/check/Makefile.am:
23164 * tests/check/elements/.cvsignore:
23165 * tests/check/elements/vorbistag.c: (setup_vorbistag),
23166 (cleanup_vorbistag), (buffer_probe), (start_pipeline),
23167 (get_buffer), (stop_pipeline), (_create_codebook_header_buffer),
23168 (_create_audio_buffer), (GST_START_TEST), (vorbistag_suite):
23169 Add unit test for new vorbistag element.
23171 2006-10-03 10:36:38 +0000 Tim-Philipp Müller <tim@centricular.net>
23173 ext/vorbis/vorbisparse.c: Set BOS flag in packet structure to fix 'jump depends on unitialized value' errors in valgr...
23174 Original commit message from CVS:
23175 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init),
23176 (vorbis_parse_push_headers), (vorbis_parse_chain):
23177 Set BOS flag in packet structure to fix 'jump depends
23178 on unitialized value' errors in valgrind; various minor
23181 2006-09-30 15:30:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23183 gst/playback/gstdecodebin.c: Fix typo in a debug statement.
23184 Original commit message from CVS:
23185 * gst/playback/gstdecodebin.c: (close_pad_link):
23186 Fix typo in a debug statement.
23187 * gst/playback/gstplaybasebin.c: (probe_triggered),
23188 (new_decoded_pad_full), (new_decoded_pad), (subs_new_decoded_pad),
23189 (gen_source_element), (source_new_pad), (analyse_source),
23191 When handling no_more_pads in new_decoded_pad, make sure to treat
23192 subtitle pads correctly. Fixes playback with subtitle files.
23193 Move a recurring message to LOG level.
23194 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
23195 The maximum value for the Xv colorkey on this Radeon is 0xFFFFFFFF,
23196 which ends up as -1 when cast to an int. Make the logic handle the
23197 max value as an unsigned mask and only change the colorkey when it's
23198 a value we recognise.
23200 2006-09-30 00:14:20 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
23202 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
23203 Original commit message from CVS:
23204 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23205 Removed empty * between paragraphs
23207 2006-09-29 23:50:53 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
23209 gst-libs/gst/rtp/: Moved some documentation into .c file
23210 Original commit message from CVS:
23211 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23212 * gst-libs/gst/rtp/README:
23213 Moved some documentation into .c file
23215 2006-09-29 17:35:01 +0000 Wim Taymans <wim.taymans@gmail.com>
23217 gst/playback/gstdecodebin.c: Fix compilation.
23218 Original commit message from CVS:
23219 * gst/playback/gstdecodebin.c: (no_more_pads):
23222 2006-09-29 16:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
23224 gst/playback/gstdecodebin.c: Remove g_print
23225 Original commit message from CVS:
23226 * gst/playback/gstdecodebin.c: (new_caps):
23228 * gst/playback/gstplaybin.c:
23231 2006-09-29 15:16:32 +0000 Tim-Philipp Müller <tim@centricular.net>
23233 tests/check/Makefile.am: Re-enable cddabasesrc test to see if it works again now.
23234 Original commit message from CVS:
23235 * tests/check/Makefile.am:
23236 Re-enable cddabasesrc test to see if it works again
23239 2006-09-29 13:46:45 +0000 Wim Taymans <wim.taymans@gmail.com>
23241 gst/playback/gstplaybasebin.c: Handle invalid URIs a bit more gracefully.
23242 Original commit message from CVS:
23243 * gst/playback/gstplaybasebin.c: (setup_subtitle),
23244 (gen_source_element):
23245 Handle invalid URIs a bit more gracefully.
23247 2006-09-29 12:54:28 +0000 Tim-Philipp Müller <tim@centricular.net>
23249 tests/check/pipelines/oggmux.c: Remove obsolete comment.
23250 Original commit message from CVS:
23251 * tests/check/pipelines/oggmux.c:
23252 Remove obsolete comment.
23254 2006-09-29 10:43:05 +0000 James Doc Livingston <doclivingston@gmail.com>
23256 ext/ogg/gstoggmux.c: Commit patch from James "Doc" Livingston, adds proper EOS handling in oggmux. GStreamer can, for...
23257 Original commit message from CVS:
23258 * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
23259 (gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
23260 (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
23261 (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
23262 (gst_ogg_mux_collected):
23263 Commit patch from James "Doc" Livingston, adds proper EOS handling
23264 in oggmux. GStreamer can, for the first time ever, create a valid
23266 * tests/check/pipelines/oggmux.c: (check_chain_final_state),
23268 Reenable tests now that they pass.
23270 2006-09-29 08:20:03 +0000 Wim Taymans <wim.taymans@gmail.com>
23272 gst/tcp/gstmultifdsink.c: Stop reading commands when EOF (we read 0) as well.
23273 Original commit message from CVS:
23274 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
23275 Stop reading commands when EOF (we read 0) as well.
23277 2006-09-28 15:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
23279 gst/playback/gstdecodebin.c: Implement delayed caps linking needed for element with a lot of different caps on the sr...
23280 Original commit message from CVS:
23281 * gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
23282 (close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
23283 (find_dynamic), (unlinked), (close_link):
23284 Implement delayed caps linking needed for element with a lot of
23285 different caps on the src pads that get fixed at runtime.
23286 Improve management of dynamic elements.
23287 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
23288 (group_destroy), (group_commit), (check_queue), (queue_overrun),
23289 (gen_preroll_element), (remove_groups), (unknown_type),
23290 (add_element_stream), (no_more_pads_full), (no_more_pads),
23291 (sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
23292 (new_decoded_pad), (setup_subtitle), (array_has_value),
23293 (gen_source_element), (source_new_pad), (has_all_raw_caps),
23294 (analyse_source), (remove_decoders), (make_decoder),
23295 (remove_source), (setup_source), (finish_source), (prepare_output),
23296 (gst_play_base_bin_change_state):
23297 * gst/playback/gstplaybasebin.h:
23298 Use more _CAST instead of full type checking casts.
23299 Small cleanups, plug some leaks.
23300 Handle dynamic sources.
23301 Add some helper functions to create lists of strings used for
23302 blacklisting and other stuff.
23303 Refactor some code dealing with analysing the source.
23304 Re-enable sources without pads (like cd:// or other selfcontained
23307 2006-09-28 15:08:15 +0000 Wim Taymans <wim.taymans@gmail.com>
23309 gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
23310 Original commit message from CVS:
23311 * gst-libs/gst/audio/gstbaseaudiosink.c:
23312 (gst_base_audio_sink_render):
23313 When we have a timestamp, we can still perform clipping.
23314 When we have no clock, we must play the sample ASAP.
23316 2006-09-28 11:46:26 +0000 Wim Taymans <wim.taymans@gmail.com>
23318 gst/audiorate/gstaudiorate.c: Set caps on outgoing buffers.
23319 Original commit message from CVS:
23320 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
23321 Set caps on outgoing buffers.
23322 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
23323 (gst_video_rate_event), (gst_video_rate_chain):
23324 * gst/videorate/gstvideorate.h:
23325 Fix videorate some more. Fixes #357977
23327 2006-09-28 11:34:05 +0000 Tim-Philipp Müller <tim@centricular.net>
23329 tests/check/elements/adder.c: Don't set timeout to 6 seconds when we're running in valgrind ... (and how is 6 seconds...
23330 Original commit message from CVS:
23331 * tests/check/elements/adder.c: (adder_suite):
23332 Don't set timeout to 6 seconds when we're running
23333 in valgrind ... (and how is 6 seconds longer than
23334 the default anyway?)
23336 2006-09-28 10:49:56 +0000 Wim Taymans <wim.taymans@gmail.com>
23338 gst/audiorate/gstaudiorate.c: Keep sink and src segment to keep track of time and support more input formats.
23339 Original commit message from CVS:
23340 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
23341 (gst_audio_rate_sink_event), (gst_audio_rate_convert),
23342 (gst_audio_rate_convert_segments), (gst_audio_rate_chain):
23343 Keep sink and src segment to keep track of time and support more
23345 Fix bogus next_offset and run_time calculation, don't understand how
23346 this could have worked before. Fixes #357976.
23347 Remove some unneeded vars.
23349 2006-09-28 09:41:20 +0000 Tim-Philipp Müller <tim@centricular.net>
23351 gst/playback/gstplaybin.c: Only remove visualisation from visbin if there is a visbin (or: don't throw warnings when ...
23352 Original commit message from CVS:
23353 * gst/playback/gstplaybin.c: (remove_sinks):
23354 Only remove visualisation from visbin if there is a visbin (or:
23355 don't throw warnings when closing totem without playing a file).
23357 2006-09-27 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
23359 gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
23360 Original commit message from CVS:
23361 * gst-libs/gst/audio/gstbaseaudiosink.c:
23362 (gst_base_audio_sink_render):
23363 Add some more info in a WARNING.
23364 * gst-libs/gst/audio/gstbaseaudiosrc.c:
23365 (gst_base_audio_src_create):
23366 Handle PAUSE in create function, use new -core addition to
23367 wait for playing. Fixes pausing and resuming capture from an
23369 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
23370 (gst_ring_buffer_read):
23371 Constify some more.
23372 Caller supports interrupted reads now.
23374 2006-09-27 13:29:49 +0000 Christian Schaller <uraeus@gnome.org>
23376 * gst-plugins-base.spec.in:
23377 add new header file to spec
23378 Original commit message from CVS:
23379 add new header file to spec
23381 2006-09-27 12:55:45 +0000 Tim-Philipp Müller <tim@centricular.net>
23383 tests/check/Makefile.am: Another attempt to make the gen64 buildbot happy.
23384 Original commit message from CVS:
23385 * tests/check/Makefile.am:
23386 Another attempt to make the gen64 buildbot happy.
23388 2006-09-27 11:58:17 +0000 Jonathan Matthew <jonathan@kaolin.wh9.net>
23390 ext/libvisual/visual.c: Libvisual plugin was not passing audio data to libvisual 0.4.0 correctly. Fixes #357800
23391 Original commit message from CVS:
23392 Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
23393 * ext/libvisual/visual.c: (gst_visual_clear_actors),
23394 (gst_visual_chain), (gst_visual_change_state):
23395 Libvisual plugin was not passing audio data to libvisual 0.4.0
23396 correctly. Fixes #357800
23398 2006-09-27 11:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
23400 tests/check/pipelines/simple-launch-lines.c: Add timeout to _get_state() so we see which pipeline it is that causes t...
23401 Original commit message from CVS:
23402 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
23403 Add timeout to _get_state() so we see which pipeline it is
23404 that causes trouble on the gen64 build bot.
23406 2006-09-27 11:06:54 +0000 Wim Taymans <wim.taymans@gmail.com>
23408 gst-libs/gst/rtp/gstbasertpdepayload.c: the source pad always uses fixed caps.
23409 Original commit message from CVS:
23410 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23411 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
23412 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
23413 (gst_base_rtp_depayload_set_gst_timestamp):
23414 the source pad always uses fixed caps.
23416 2006-09-27 11:05:08 +0000 Wim Taymans <wim.taymans@gmail.com>
23418 Added docs for the audio libs.
23419 Original commit message from CVS:
23420 * docs/libs/gst-plugins-base-libs-docs.sgml:
23421 * docs/libs/gst-plugins-base-libs-sections.txt:
23422 * gst-libs/gst/audio/gstaudioclock.c:
23423 * gst-libs/gst/audio/gstaudioclock.h:
23424 * gst-libs/gst/audio/gstaudiosink.c:
23425 * gst-libs/gst/audio/gstaudiosink.h:
23426 * gst-libs/gst/audio/gstaudiosrc.c:
23427 * gst-libs/gst/audio/gstbaseaudiosink.c:
23428 (gst_base_audio_sink_render):
23429 * gst-libs/gst/audio/gstbaseaudiosink.h:
23430 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
23431 * gst-libs/gst/audio/gstbaseaudiosrc.h:
23432 * gst-libs/gst/audio/gstringbuffer.h:
23433 Added docs for the audio libs.
23435 2006-09-27 10:59:24 +0000 Tim-Philipp Müller <tim@centricular.net>
23437 tests/check/Makefile.am: Temporarily disable test that fails on the bots for unknown reasons.
23438 Original commit message from CVS:
23439 * tests/check/Makefile.am:
23440 Temporarily disable test that fails on the bots for unknown reasons.
23442 2006-09-27 00:13:29 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
23444 gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
23445 Original commit message from CVS:
23446 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23447 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
23448 Moved AudioCodecType into priv
23449 Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
23451 2006-09-25 15:47:25 +0000 Wim Taymans <wim.taymans@gmail.com>
23453 gst/playback/gstdecodebin.c: Cleanups and small leak fixes.
23454 Original commit message from CVS:
23455 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
23456 (add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
23457 (is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
23459 Cleanups and small leak fixes.
23460 Added Depayloaders to valid list of autopluggable elements.
23462 2006-09-25 13:24:59 +0000 Wim Taymans <wim.taymans@gmail.com>
23464 gst/playback/gstplaybin.c: Detect NO_PREROLL state change returns and disable clock distribution to the sinks so that...
23465 Original commit message from CVS:
23466 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
23467 (gst_play_bin_vis_blocked), (gst_play_bin_set_property),
23468 (gen_video_element), (gen_text_element), (gen_audio_element),
23469 (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
23470 (gst_play_bin_set_clock_func), (gst_play_bin_change_state):
23471 Detect NO_PREROLL state change returns and disable clock distribution to
23472 the sinks so that sync is disabled.
23473 Avoid some type checking and do simple casts instead.
23474 Small cleanups, fix some FIXMEs.
23475 Be more robust when linking user specified elements, catch an report
23476 errors. Fixes #357404.
23477 Fix some leaks in the error paths.
23479 2006-09-25 12:55:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23482 ChangeLog surgery for missing bug-number
23483 Original commit message from CVS:
23484 ChangeLog surgery for missing bug-number
23486 2006-09-25 11:28:15 +0000 Peter Kjellerstedt <pkj@axis.com>
23488 gst/playback/test.c: Fix compilation with uClibc and -Werror (#357591).
23489 Original commit message from CVS:
23490 Patch by: Peter Kjellerstedt <pkj at axis com>
23491 * gst/playback/test.c:
23492 Fix compilation with uClibc and -Werror (#357591).
23494 2006-09-25 10:21:31 +0000 Tim-Philipp Müller <tim@centricular.net>
23496 gst-libs/gst/tag/gstvorbistag.c: Parse dates that are followed by a time as well (#357532).
23497 Original commit message from CVS:
23498 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
23499 Parse dates that are followed by a time as well (#357532).
23500 * tests/check/libs/tag.c: (test_vorbis_tags):
23501 Add unit test for this.
23503 2006-09-23 15:24:55 +0000 Tim-Philipp Müller <tim@centricular.net>
23505 gst/: A few array const-ifications.
23506 Original commit message from CVS:
23507 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
23508 (gst_audio_convert_transform_caps):
23509 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
23510 * gst/videotestsrc/videotestsrc.h:
23511 A few array const-ifications.
23513 2006-09-23 15:02:51 +0000 Tim-Philipp Müller <tim@centricular.net>
23515 tests/check/Makefile.am: See if this makes the build bots happy.
23516 Original commit message from CVS:
23517 * tests/check/Makefile.am:
23518 See if this makes the build bots happy.
23519 * tests/check/libs/cddabasesrc.c:
23522 2006-09-23 14:30:53 +0000 Young-Ho Cha <ganadist@chollian.net>
23524 gst/subparse/samiparse.c: More case-insensitivity for certain tags; recognise entities with decimal codes as special ...
23525 Original commit message from CVS:
23526 Patch by: Young-Ho Cha <ganadist at chollian dot net>
23527 * gst/subparse/samiparse.c: (handle_start_font),
23528 (fix_invalid_entities):
23529 More case-insensitivity for certain tags; recognise entities with
23530 decimal codes as special entities as well (#357330).
23532 2006-09-23 13:32:07 +0000 Tim-Philipp Müller <tim@centricular.net>
23534 gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
23535 Original commit message from CVS:
23536 * gst-libs/gst/Makefile.am:
23537 Need to build tag directory before cdda.
23539 2006-09-23 13:21:07 +0000 Tim-Philipp Müller <tim@centricular.net>
23541 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
23542 Original commit message from CVS:
23543 * docs/libs/gst-plugins-base-libs-sections.txt:
23544 * gst-libs/gst/cdda/Makefile.am:
23545 * gst-libs/gst/cdda/gstcddabasesrc.c:
23546 (gst_cdda_base_src_base_init):
23547 * gst-libs/gst/cdda/gstcddabasesrc.h:
23548 * gst-libs/gst/tag/tag.h:
23549 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
23550 (gst_tag_register_musicbrainz_tags):
23551 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
23552 depend on libgsttag. This is required so we can extract/read tags like
23553 DISCID without depending on libgstcddabasesrc (which used to register
23555 * gst-libs/gst/tag/gstvorbistag.c:
23556 Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
23557 tags (also see #347848).
23558 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
23559 Log vorbis comments we are actually writing. Const-ify array.
23561 2006-09-23 08:53:30 +0000 Wim Taymans <wim.taymans@gmail.com>
23563 gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
23564 Original commit message from CVS:
23565 * gst/playback/gstplaybasebin.c: (gen_preroll_element):
23566 Improve buffering a bit by avoiding a deadlock because we cannot assume
23567 the underrun is always called.
23569 2006-09-23 08:51:14 +0000 Young-Ho Cha <ganadist@chollian.net>
23571 gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
23572 Original commit message from CVS:
23573 Patch by: Young-Ho Cha <ganadist at chollian dot net>
23574 * gst-libs/gst/riff/riff-ids.h:
23575 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
23576 (gst_riff_create_audio_template_caps):
23577 Added MPEG-4 AAC and id and caps. Fixes #357289
23578 Added WMA9 Lossless id.
23580 2006-09-22 14:50:01 +0000 Tim-Philipp Müller <tim@centricular.net>
23582 ext/gnomevfs/gstgnomevfssrc.c: Fix misleading docs addition.
23583 Original commit message from CVS:
23584 * ext/gnomevfs/gstgnomevfssrc.c:
23585 Fix misleading docs addition.
23586 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
23587 Get rid of compiler warning the right way.
23589 2006-09-22 14:13:34 +0000 Wim Taymans <wim.taymans@gmail.com>
23591 gst-libs/gst/rtp/gstbasertpdepayload.*: Small cleanups.
23592 Original commit message from CVS:
23593 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23594 (gst_base_rtp_depayload_finalize),
23595 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
23596 (gst_base_rtp_depayload_push_full),
23597 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
23598 (gst_base_rtp_depayload_process),
23599 (gst_base_rtp_depayload_set_gst_timestamp),
23600 (gst_base_rtp_depayload_queue_release):
23601 * gst-libs/gst/rtp/gstbasertpdepayload.h:
23604 Refactored the process method and added methods to push from the process
23606 Use _scale functions.
23607 API: gst_base_rtp_depayload_push_ts
23608 API: gst_base_rtp_depayload_push
23609 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
23610 timestamps are uint.
23612 2006-09-22 11:59:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23614 gst-libs/gst/interfaces/xoverlay.c: Remove unused statement from doc example.
23615 Original commit message from CVS:
23616 * gst-libs/gst/interfaces/xoverlay.c:
23617 Remove unused statement from doc example.
23619 2006-09-22 09:52:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23621 * gst/videorate/gstvideorate.c:
23623 Original commit message from CVS:
23626 2006-09-21 13:49:47 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23628 gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
23629 Original commit message from CVS:
23630 * gst-libs/gst/interfaces/videoorientation.c:
23631 (gst_video_orientation_iface_init),
23632 (gst_video_orientation_get_hflip),
23633 (gst_video_orientation_get_vflip),
23634 (gst_video_orientation_get_hcenter),
23635 (gst_video_orientation_get_vcenter),
23636 (gst_video_orientation_set_hflip),
23637 (gst_video_orientation_set_vflip),
23638 (gst_video_orientation_set_hcenter),
23639 (gst_video_orientation_set_vcenter):
23640 Add since tags to new API docs, ChangeLog surgery (forgot API keyword
23643 2006-09-21 09:27:47 +0000 Tim-Philipp Müller <tim@centricular.net>
23645 tests/check/: but disable for now since it doesn't pass (something wrong with
23646 Original commit message from CVS:
23647 * tests/check/Makefile.am:
23648 * tests/check/elements/.cvsignore:
23649 * tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
23650 (create_rgb_conversions), (rgb_conversion_free),
23651 (right_shift_colour), (fix_expected_colour), (check_rgb_buf),
23652 (got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
23653 Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
23654 but disable for now since it doesn't pass (something wrong with
23657 2006-09-21 07:01:48 +0000 Wim Taymans <wim.taymans@gmail.com>
23659 gst/playback/gstplaybasebin.c: Refactor handling of overrun detection.
23660 Original commit message from CVS:
23661 * gst/playback/gstplaybasebin.c: (group_commit),
23662 (queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
23663 (queue_out_of_data), (gen_preroll_element),
23664 (preroll_remove_overrun), (probe_triggered):
23665 Refactor handling of overrun detection.
23666 Separate handling of group completion and deadlock detection when doing
23667 network buffering. This should fix some deadlocks that were not detected
23668 because the group was completed.
23669 Add more comments, improve debugging.
23671 2006-09-21 05:31:00 +0000 Wim Taymans <wim.taymans@gmail.com>
23673 tests/check/: Some more compilation fixes.
23674 Original commit message from CVS:
23675 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
23676 * tests/check/libs/audio.c:
23677 Some more compilation fixes.
23679 2006-09-21 05:12:18 +0000 Wim Taymans <wim.taymans@gmail.com>
23681 gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
23682 Original commit message from CVS:
23683 * gst-libs/gst/audio/gstringbuffer.c:
23684 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
23685 (gst_ring_buffer_read):
23686 Early morning compilation fix.
23688 2006-09-20 18:09:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23692 Original commit message from CVS:
23695 2006-09-20 17:04:57 +0000 Wim Taymans <wim.taymans@gmail.com>
23697 tests/check/: Fix some warnings.
23698 Original commit message from CVS:
23699 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
23700 * tests/check/elements/multifdsink.c: (GST_START_TEST):
23701 * tests/check/elements/videorate.c: (GST_START_TEST):
23702 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
23703 * tests/check/pipelines/oggmux.c: (eos_buffer_probe):
23706 2006-09-20 10:59:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23708 sys/xvimage/xvimagesink.c: change colorkey behaviour back according to #354773 comment 6/7
23709 Original commit message from CVS:
23710 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
23711 (gst_xvimagesink_get_times):
23712 change colorkey behaviour back according to #354773 comment 6/7
23714 2006-09-20 10:42:34 +0000 Tim-Philipp Müller <tim@centricular.net>
23717 ChangeLog surgery: remove junk
23718 Original commit message from CVS:
23719 ChangeLog surgery: remove junk
23721 2006-09-19 11:31:06 +0000 Michael Smith <msmith@xiph.org>
23723 gst/tcp/gstmultifdsink.*: Implement stubbed out properties unit-type, units-soft-max, units-max, to allow specifying ...
23724 Original commit message from CVS:
23725 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
23726 (gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
23727 (gst_multi_fd_sink_recover_client),
23728 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
23729 (gst_multi_fd_sink_get_property):
23730 * gst/tcp/gstmultifdsink.h:
23731 Implement stubbed out properties unit-type, units-soft-max,
23732 units-max, to allow specifying maximum sizes in units other than
23736 2006-09-19 10:23:03 +0000 Wim Taymans <wim.taymans@gmail.com>
23738 gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
23739 Original commit message from CVS:
23740 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
23741 (gst_riff_create_audio_template_caps):
23742 Reorder the audio formats a bit for clarity.
23743 Detect and create caps for MSGSM and MSN (WAV49).
23745 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
23746 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
23747 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
23748 Small cleanups, move error handling out of normal flow for clarity.
23750 2006-09-18 15:59:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23752 Add new interface to control video orientation (fixes #354908)
23753 Original commit message from CVS:
23754 * docs/libs/gst-plugins-base-libs-docs.sgml:
23755 * docs/libs/gst-plugins-base-libs.types:
23756 * gst-libs/gst/interfaces/Makefile.am:
23757 * gst-libs/gst/interfaces/videoorientation.c:
23758 (gst_video_orientation_get_type),
23759 (gst_video_orientation_iface_init),
23760 (gst_video_orientation_get_hflip),
23761 (gst_video_orientation_get_vflip),
23762 (gst_video_orientation_get_hcenter),
23763 (gst_video_orientation_get_vcenter),
23764 (gst_video_orientation_set_hflip),
23765 (gst_video_orientation_set_vflip),
23766 (gst_video_orientation_set_hcenter),
23767 (gst_video_orientation_set_vcenter):
23768 * gst-libs/gst/interfaces/videoorientation.h:
23769 Add new interface to control video orientation (fixes #354908)
23771 2006-09-18 15:48:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23773 gst/videotestsrc/gstvideotestsrc.c: Use G_UNLIKELY in _create and log one more detail.
23774 Original commit message from CVS:
23775 * gst/videotestsrc/gstvideotestsrc.c:
23776 Use G_UNLIKELY in _create and log one more detail.
23777 (gst_video_test_src_get_times), (gst_video_test_src_create):
23778 * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
23779 Use gst_util_uint64_scale_int in _get_times().
23781 2006-09-18 15:00:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23783 sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
23784 Original commit message from CVS:
23785 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
23786 Give better warning message (add object and detail).
23788 2006-09-18 14:42:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23790 sys/xvimage/xvimagesink.c: xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes #354773), use gst_util...
23791 Original commit message from CVS:
23792 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
23793 (gst_xvimagesink_get_times):
23794 xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
23795 #354773), use gst_util_uint64_scale_int in _get_times()
23797 2006-09-18 14:21:45 +0000 Michael Smith <msmith@xiph.org>
23799 ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
23800 Original commit message from CVS:
23801 * ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
23802 Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
23803 always true, leading to dropping all timestamps.
23805 2006-09-18 11:40:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23807 ext/libvisual/visual.c: update to work also with libvisual 0.4 API
23808 Original commit message from CVS:
23809 * ext/libvisual/visual.c: (gst_vis_src_negotiate),
23810 (gst_visual_chain), (gst_visual_change_state):
23811 update to work also with libvisual 0.4 API
23812 * tools/gst-launch-ext.1.in:
23813 * tools/gst-visualise.1.in:
23814 remove references to old man-pages
23815 * tests/examples/seek/seek.c: (main):
23816 add real meadi-buttons, add tool-tips for the seek-options, arrange
23817 seek options in a table
23819 2006-09-18 10:57:28 +0000 Michael Smith <msmith@xiph.org>
23821 ext/ogg/gstoggmux.c: Don't generate out-of-order timestamps from oggmux, instead clamp output timestamps to be >= the...
23822 Original commit message from CVS:
23823 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
23824 (gst_ogg_mux_push_buffer):
23825 Don't generate out-of-order timestamps from oggmux, instead clamp
23826 output timestamps to be >= the previously output ts.
23829 2006-09-18 10:18:22 +0000 Michael Smith <msmith@xiph.org>
23831 gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
23832 Original commit message from CVS:
23833 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
23834 (gst_multi_fd_sink_class_init):
23835 Updates, fixes, and typo corrections for multifdsink. No functional
23838 2006-09-17 21:58:06 +0000 Michael Smith <msmith@xiph.org>
23840 gst/typefind/gsttypefindfunctions.c: Don't crash on truncated files - check that we got an 8 byte buffer before tryin...
23841 Original commit message from CVS:
23842 * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
23843 Don't crash on truncated files - check that we got an 8 byte buffer
23844 before trying to memcmp it.
23846 2006-09-17 20:32:09 +0000 Tim-Philipp Müller <tim@centricular.net>
23848 gst/playback/gstplaybasebin.c: Make stream-switching appear instant to the application (ie. make sure that a g_object...
23849 Original commit message from CVS:
23850 * gst/playback/gstplaybasebin.c: (get_active_source):
23851 Make stream-switching appear instant to the application
23852 (ie. make sure that a g_object_get on 'current-foo' returns
23853 the stream previously set with g_object_set(). Totem needs
23854 this to update stream-related meta-info (like audio-codec)
23855 correctly when switching streams.
23857 2006-09-17 20:14:43 +0000 Tim-Philipp Müller <tim@centricular.net>
23859 ext/alsa/gstalsamixer.c: Try harder to guess which mixer track is the master mixer track (instead of just taking the ...
23860 Original commit message from CVS:
23861 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
23862 (gst_alsa_mixer_ensure_track_list):
23863 Try harder to guess which mixer track is the master mixer
23864 track (instead of just taking the first one that has a pvolume).
23867 2006-09-17 11:24:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23869 gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
23870 Original commit message from CVS:
23871 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
23872 (gst_audio_convert_transform_caps):
23873 Get structure-name just once.
23875 2006-09-16 22:30:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23877 tests/check/: Fix big batch of compiler warnings.
23878 Original commit message from CVS:
23879 * tests/check/elements/audioresample.c: (GST_START_TEST):
23880 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
23881 * tests/check/elements/volume.c: (GST_START_TEST):
23882 * tests/check/elements/vorbisdec.c: (GST_START_TEST):
23883 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
23884 (test_pipeline), (GST_START_TEST):
23885 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
23886 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
23887 Fix big batch of compiler warnings.
23889 2006-09-16 21:54:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23891 ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
23892 Original commit message from CVS:
23893 * ext/gnomevfs/gstgnomevfssrc.c:
23894 Add docs about icydemux usage in connection with gnomevfssrc
23895 * ext/libvisual/visual.c:
23896 * ext/ogg/gstoggaviparse.c:
23897 * ext/ogg/gstoggdemux.c:
23898 * ext/ogg/gstoggmux.c:
23899 * ext/ogg/gstoggparse.c:
23900 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
23901 * gst-libs/gst/audio/gstaudiosink.c:
23902 * gst-libs/gst/audio/gstaudiosrc.c:
23903 * gst/audiorate/gstaudiorate.c:
23904 More G_OBJECT macro fixing.
23905 * gst/audiotestsrc/gstaudiotestsrc.h:
23906 Fix wrong info in header due to copy & paste
23908 2006-09-15 14:53:44 +0000 Wim Taymans <wim.taymans@gmail.com>
23910 gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
23911 Original commit message from CVS:
23912 * gst-libs/gst/audio/gstbaseaudiosink.c:
23913 (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
23914 * gst-libs/gst/audio/gstbaseaudiosrc.c:
23915 (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
23916 (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
23917 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
23918 Do the delay calculation in the source/sink base classes as this is
23919 specific for the capture/playback mode.
23920 Try to fixate a bit better, like round depth up to a multiple of 8
23922 Handle underruns correctly by marking DISCONT on buffers and adjusting
23923 timestamps to handle the gap.
23924 Set offset/offset_end correctly on buffers.
23925 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
23926 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
23927 (gst_ring_buffer_read):
23928 Remove resync and underrun recovery from the ringbuffer.
23929 Fix ringbuffer read code on under/overrun.
23931 2006-09-15 11:17:02 +0000 Wim Taymans <wim.taymans@gmail.com>
23933 gst/playback/gstplaybasebin.*: Don't use a 0 low watermark when buffering, it is catching starvation way too late. In...
23934 Original commit message from CVS:
23935 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
23936 (gst_play_base_bin_init), (fill_buffer), (check_queue),
23937 (queue_threshold_reached), (gst_play_base_bin_set_property),
23938 (gst_play_base_bin_get_property):
23939 * gst/playback/gstplaybasebin.h:
23940 Don't use a 0 low watermark when buffering, it is catching starvation
23941 way too late. Instead, use a 3 second queue with 30 and 95
23942 percent low/high watermarks.
23943 Added queue-min-threshold property to configure low watermark.
23944 Use new _buffering message API.
23945 Make queue_threshold variable big enough to store a uint64 time value.
23946 API: playbin::queue-min-threshold property.
23948 2006-09-15 09:13:50 +0000 Wim Taymans <wim.taymans@gmail.com>
23950 configure.ac: We require 0.10.10.1 now because of _wait_preroll().
23951 Original commit message from CVS:
23953 We require 0.10.10.1 now because of _wait_preroll().
23954 * gst-libs/gst/audio/gstbaseaudiosink.c:
23955 (gst_base_audio_sink_render):
23956 Use gst_base_sink_wait_preroll().
23958 2006-09-15 09:09:00 +0000 Wim Taymans <wim.taymans@gmail.com>
23960 ext/alsa/: Use DEBUG_OBJECT more.
23961 Original commit message from CVS:
23962 * ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
23963 * ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
23964 Use DEBUG_OBJECT more.
23966 === release 0.10.10 ===
23968 2006-09-14 20:09:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23975 * docs/plugins/gst-plugins-base-plugins.args:
23976 * docs/plugins/inspect/plugin-adder.xml:
23977 * docs/plugins/inspect/plugin-alsa.xml:
23978 * docs/plugins/inspect/plugin-audioconvert.xml:
23979 * docs/plugins/inspect/plugin-audiorate.xml:
23980 * docs/plugins/inspect/plugin-audioresample.xml:
23981 * docs/plugins/inspect/plugin-audiotestsrc.xml:
23982 * docs/plugins/inspect/plugin-cdparanoia.xml:
23983 * docs/plugins/inspect/plugin-decodebin.xml:
23984 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
23985 * docs/plugins/inspect/plugin-gdp.xml:
23986 * docs/plugins/inspect/plugin-gnomevfs.xml:
23987 * docs/plugins/inspect/plugin-libvisual.xml:
23988 * docs/plugins/inspect/plugin-ogg.xml:
23989 * docs/plugins/inspect/plugin-pango.xml:
23990 * docs/plugins/inspect/plugin-playbin.xml:
23991 * docs/plugins/inspect/plugin-subparse.xml:
23992 * docs/plugins/inspect/plugin-tcp.xml:
23993 * docs/plugins/inspect/plugin-theora.xml:
23994 * docs/plugins/inspect/plugin-typefindfunctions.xml:
23995 * docs/plugins/inspect/plugin-video4linux.xml:
23996 * docs/plugins/inspect/plugin-videorate.xml:
23997 * docs/plugins/inspect/plugin-videoscale.xml:
23998 * docs/plugins/inspect/plugin-videotestsrc.xml:
23999 * docs/plugins/inspect/plugin-volume.xml:
24000 * docs/plugins/inspect/plugin-vorbis.xml:
24001 * docs/plugins/inspect/plugin-ximagesink.xml:
24002 * docs/plugins/inspect/plugin-xvimagesink.xml:
24003 * ext/theora/theoraparse.c:
24004 * gst-libs/gst/rtp/gstrtpbuffer.c:
24005 * gst/playback/gstplaybin.c:
24006 * tests/check/Makefile.am:
24007 * win32/common/config.h:
24009 Original commit message from CVS:
24012 2006-09-09 16:08:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24015 * win32/common/config.h:
24017 Original commit message from CVS:
24020 2006-09-07 19:01:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24023 update bug in changelog
24024 Original commit message from CVS:
24025 update bug in changelog
24027 2006-09-07 19:00:33 +0000 Michael Smith <msmith@fluendo.com>
24029 Fix implementation of sync-method 'next-keyframe'
24030 Original commit message from CVS:
24031 patch by: Michael Smith <msmith at fluendo dot com>
24032 * gst/tcp/gstmultifdsink.c: (is_sync_frame),
24033 (gst_multi_fd_sink_client_queue_buffer),
24034 (gst_multi_fd_sink_new_client):
24035 * tests/check/elements/multifdsink.c: (GST_START_TEST),
24036 (multifdsink_suite):
24037 Fix implementation of sync-method 'next-keyframe'
24039 2006-09-07 15:00:08 +0000 Wim Taymans <wim@fluendo.com>
24041 ext/gnomevfs/gstgnomevfssrc.c: This patch removes the RANDOM flag that was incorrectly introduced with revision 1.91....
24042 Original commit message from CVS:
24043 patch by: Wim Taymans <wim at fluendo dot com>
24044 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
24045 This patch removes the RANDOM flag that was incorrectly introduced with
24046 revision 1.91. Fixes #354590
24048 2006-09-07 14:56:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24051 * win32/common/config.h:
24053 Original commit message from CVS:
24056 2006-09-07 14:56:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24073 Original commit message from CVS:
24076 2006-09-05 09:12:25 +0000 Tim-Philipp Müller <tim@centricular.net>
24078 tests/check/Makefile.am: Random variation in Makefile line to see if it makes the gen64-base-full bot any happier.
24079 Original commit message from CVS:
24080 * tests/check/Makefile.am:
24081 Random variation in Makefile line to see if it makes the
24082 gen64-base-full bot any happier.
24084 2006-09-04 19:04:35 +0000 Tim-Philipp Müller <tim@centricular.net>
24086 tests/check/pipelines/oggmux.c: Disable test that fails at the moment (killed after timeout).
24087 Original commit message from CVS:
24088 * tests/check/pipelines/oggmux.c: (oggmux_suite):
24089 Disable test that fails at the moment (killed after timeout).
24091 2006-09-04 18:19:06 +0000 James Livingston <doclivingston@gmail.com>
24093 tests/check/: Add simple unit test for oggmux from #337026 with checking for the
24094 Original commit message from CVS:
24095 Patch by: James Livingston <doclivingston at gmail.com>
24096 * tests/check/Makefile.am:
24097 * tests/check/pipelines/.cvsignore:
24098 * tests/check/pipelines/oggmux.c: (get_page_codec),
24099 (check_chain_final_state), (fail_if_audio), (validate_ogg_page),
24100 (eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
24101 (test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
24102 (test_theora_vorbis), (oggmux_suite):
24103 Add simple unit test for oggmux from #337026 with checking for the
24104 EOS flags disabled for the time being.
24106 2006-09-04 09:13:01 +0000 Alessandro Dessina <alessandro@nnva.org>
24108 ext/ogg/gstoggmux.c: Add cmml caps to oggmux. Fixes #353912
24109 Original commit message from CVS:
24110 patch by: Alessandro Dessina <alessandro nnva org>
24111 * ext/ogg/gstoggmux.c:
24112 Add cmml caps to oggmux. Fixes #353912
24114 2006-09-02 13:20:59 +0000 Tim-Philipp Müller <tim@centricular.net>
24116 tests/check/elements/videotestsrc.c: Returning a return value often helps. In this case, we don't need the return val...
24117 Original commit message from CVS:
24118 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
24119 Returning a return value often helps. In this case, we
24120 don't need the return value anyway, so just get rid of it.
24121 Should make build bots much happier.
24123 2006-09-02 12:59:48 +0000 Tim-Philipp Müller <tim@centricular.net>
24125 gst/videotestsrc/videotestsrc.*: Add support for AYUV and the various RGBA formats. Initialise fields of paintinfo st...
24126 Original commit message from CVS:
24127 * gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
24128 (paint_get_structure), (gst_video_test_src_get_size),
24129 (gst_video_test_src_smpte), (gst_video_test_src_snow),
24130 (gst_video_test_src_unicolor), (paint_setup_AYUV),
24131 (paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
24132 (paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
24133 * gst/videotestsrc/videotestsrc.h:
24134 Add support for AYUV and the various RGBA formats. Initialise
24135 fields of paintinfo structs allocated on the stack.
24136 * tests/check/elements/videotestsrc.c: (right_shift_colour),
24137 (fix_expected_colour), (check_rgb_buf), (got_buf_cb),
24138 (GST_START_TEST), (videotestsrc_suite):
24139 Add unit tests for videotestsrc's RGB output.
24141 2006-09-01 16:12:35 +0000 Tim-Philipp Müller <tim@centricular.net>
24143 gst/videotestsrc/: Add more uni-colour patterns ("white", "red", "green", and "blue").
24144 Original commit message from CVS:
24145 * gst/videotestsrc/gstvideotestsrc.c:
24146 (gst_video_test_src_pattern_get_type),
24147 (gst_video_test_src_set_pattern):
24148 * gst/videotestsrc/gstvideotestsrc.h:
24149 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
24150 (gst_video_test_src_black), (gst_video_test_src_white),
24151 (gst_video_test_src_red), (gst_video_test_src_green),
24152 (gst_video_test_src_blue):
24153 * gst/videotestsrc/videotestsrc.h:
24154 Add more uni-colour patterns ("white", "red", "green", and "blue").
24156 2006-09-01 10:07:05 +0000 Tim-Philipp Müller <tim@centricular.net>
24158 gst/videotestsrc/videotestsrc.c: Fix stride for YVYU, should be word-aligned (#353658).
24159 Original commit message from CVS:
24160 * gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
24161 Fix stride for YVYU, should be word-aligned (#353658).
24163 2006-08-31 14:37:33 +0000 Tim-Philipp Müller <tim@centricular.net>
24165 gst/adder/gstadder.c: Fix build.
24166 Original commit message from CVS:
24167 * gst/adder/gstadder.c: (gst_adder_src_event):
24170 2006-08-31 12:39:17 +0000 Edward Hervey <bilboed@bilboed.com>
24172 gst/adder/gstadder.*: Remember the start position asked in the incoming seeks, so we can output GST_EVENT_NEW_SEGMENT...
24173 Original commit message from CVS:
24174 * gst/adder/gstadder.c: (forward_event_func),
24175 (gst_adder_src_event), (gst_adder_collected),
24176 (gst_adder_change_state):
24177 * gst/adder/gstadder.h:
24178 Remember the start position asked in the incoming seeks, so we can
24179 output GST_EVENT_NEW_SEGMENT with a correct position value (instead
24180 of assuming it will always be 0).
24182 2006-08-31 12:31:00 +0000 Edward Hervey <bilboed@bilboed.com>
24184 ext/ogg/gstoggdemux.c: Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
24185 Original commit message from CVS:
24186 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
24187 (gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
24188 (gst_ogg_demux_loop):
24189 Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
24191 2006-08-30 17:22:27 +0000 Tim-Philipp Müller <tim@centricular.net>
24193 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Return FALSE instead of returning a random false unit size when the forma...
24194 Original commit message from CVS:
24195 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
24196 (gst_ffmpegcsp_get_unit_size):
24197 Return FALSE instead of returning a random false unit
24198 size when the format isn't known/supported (even if
24199 this shouldn't happen under normal circumstances).
24201 2006-08-29 15:23:46 +0000 Tim-Philipp Müller <tim@centricular.net>
24203 ext/gnomevfs/gstgnomevfssrc.c: Try harder to get the size from a uri by using _info_uri() when _info_from_handle() do...
24204 Original commit message from CVS:
24205 Patch by: Tim-Philipp Müller <tim at centricular dot net>
24206 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
24207 (gst_gnome_vfs_src_start):
24208 Try harder to get the size from a uri by using _info_uri() when
24209 _info_from_handle() does not give us enough info.
24210 Also follow symlinks when getting the size.
24211 Partially Fixes #332864.
24213 2006-08-29 11:50:51 +0000 Viktor Peters <viktor.peters@gmail.com>
24215 ext/alsa/: Improve and fix mixer track handling, in particular better handling of alsa's pvolume/pswitch/cvolume/cswi...
24216 Original commit message from CVS:
24217 Patch by: Viktor Peters <viktor dot peters at gmail dot com>
24218 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
24219 (gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
24220 (gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
24221 (gst_alsa_mixer_set_record):
24222 * ext/alsa/gstalsamixertrack.c:
24223 (gst_alsa_mixer_track_update_alsa_capabilities),
24224 (alsa_track_has_cap), (gst_alsa_mixer_track_new),
24225 (gst_alsa_mixer_track_update):
24226 * ext/alsa/gstalsamixertrack.h:
24227 Improve and fix mixer track handling, in particular better handling
24228 of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
24229 track objects for tracks that have both capture and playback volume
24230 (and label them differently as well so they're not mistakenly
24231 assumed to be duplicates); classify mixer tracks that only affect
24232 the audible volume of something (rather than the capture volume)
24233 as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
24234 for capture tracks to correspond to alsa-pswitch alsa-cswitch
24235 (following the meaning documented in the mixer interface header
24236 file); add support for alsa's exclusive cswitch groups; update/sync
24237 state/flags better if mixer settings are changed by another
24238 application. Fixes #336075.
24240 2006-08-29 10:58:43 +0000 Tim-Philipp Müller <tim@centricular.net>
24242 gst/playback/gstplaybin.c: Improve docs: add section about BUFFERING messages sent by playbin.
24243 Original commit message from CVS:
24244 * gst/playback/gstplaybin.c:
24245 Improve docs: add section about BUFFERING messages sent by playbin.
24247 2006-08-29 10:51:12 +0000 Michael Smith <msmith@xiph.org>
24249 ext/vorbis/vorbisenc.c: Ignore explicit DISCONT marked on buffers (which is often spurious, particularly when using m...
24250 Original commit message from CVS:
24251 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
24252 (gst_vorbis_enc_buffer_check_discontinuous),
24253 (gst_vorbis_enc_chain):
24254 Ignore explicit DISCONT marked on buffers (which is often spurious,
24255 particularly when using multiple segments), in favour of solely
24256 using the timestamps/durations.
24258 2006-08-29 10:32:34 +0000 Edward Hervey <bilboed@bilboed.com>
24260 gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
24261 Original commit message from CVS:
24262 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
24263 Don't rely on incoming buffers offset anymore, since it is completely
24264 broken when using multiple segments.
24265 Instead convert the incoming buffers timestamp to running time, and
24266 then convert that value to the offsets.
24267 Also inform GstSegment of the last outputted stop position, which is
24268 needed if we received several segments with an unknown stop value.
24270 2006-08-29 08:03:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24272 ext/vorbis/vorbisenc.c: fix buffer unreffing on a header push failure
24273 Original commit message from CVS:
24274 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
24275 fix buffer unreffing on a header push failure
24277 2006-08-28 16:17:13 +0000 Wim Taymans <wim.taymans@gmail.com>
24279 gst/audiorate/gstaudiorate.c: Make the metadata of the buffer writable before changing its flags.
24280 Original commit message from CVS:
24281 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
24282 (gst_audio_rate_chain):
24283 Make the metadata of the buffer writable before changing its
24286 2006-08-28 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
24289 Fix changelog with bugzilla bug it fixed.
24290 Original commit message from CVS:
24291 Fix changelog with bugzilla bug it fixed.
24293 2006-08-28 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
24295 gst/audiorate/gstaudiorate.c: Fix audiorate some more.
24296 Original commit message from CVS:
24297 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
24298 (gst_audio_rate_setcaps), (gst_audio_rate_init),
24299 (gst_audio_rate_sink_event), (gst_audio_rate_src_event),
24300 (gst_audio_rate_chain), (gst_audio_rate_change_state):
24301 Fix audiorate some more.
24302 Reset and resync counters on flush and READY.
24303 Handle the DISCONT flag correctly.
24304 Use GstSegment to track position.
24305 Fail when not negotiated.
24307 2006-08-25 16:48:28 +0000 Michael Smith <msmith@xiph.org>
24309 gst/tcp/gstmultifdsink.c: Fix spelling.
24310 Original commit message from CVS:
24311 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
24313 Remove accidently included debug line.
24315 2006-08-25 16:39:38 +0000 Wim Taymans <wim.taymans@gmail.com>
24317 gst/tcp/gstmultifdsink.c: Small cleanups.
24318 Original commit message from CVS:
24319 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
24321 If a buffer is received with no caps, make the buffer metadata
24322 writable and set the caps, making sure that we don't screw up the
24325 2006-08-25 16:19:55 +0000 Michael Smith <msmith@xiph.org>
24327 gst/gdp/gstgdppay.c: Fix memory leaks and misleading debug messages, add a couple of comments.
24328 Original commit message from CVS:
24329 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
24330 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
24331 Fix memory leaks and misleading debug messages, add a couple of
24333 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
24334 (gst_multi_fd_sink_render):
24335 Do not use gst_buffer_make_writable() in a basesink render method,
24336 as it may incorrectly unref the buffer. Instead, use convoluted
24337 dance to avoid copying the buffer except when we need to.
24339 2006-08-25 09:54:56 +0000 Michael Smith <msmith@xiph.org>
24341 ext/vorbis/vorbisenc.c: Allow very small discontinuities in the timestamps. These we can't do anything useful with an...
24342 Original commit message from CVS:
24343 * ext/vorbis/vorbisenc.c:
24344 (gst_vorbis_enc_buffer_check_discontinuous):
24345 Allow very small discontinuities in the timestamps. These we can't
24346 do anything useful with anyway (because vorbis's timestamps have
24347 only sample granularity), and are commonly produced by elements with
24348 minor bugs. Allow up to 1/2 a sample out.
24351 2006-08-24 11:18:56 +0000 Wim Taymans <wim.taymans@gmail.com>
24353 tests/examples/seek/seek.c: Add a checkbox to enable play scrubbing. Makes it possible to disable normal scrubbing.
24354 Original commit message from CVS:
24355 * tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
24356 (play_scrub_toggle_cb), (main):
24357 Add a checkbox to enable play scrubbing. Makes it possible to disable
24360 2006-08-23 19:37:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24362 tests/check/elements/.cvsignore: make buildbot happy
24363 Original commit message from CVS:
24364 * tests/check/elements/.cvsignore:
24365 make buildbot happy
24367 2006-08-23 16:43:03 +0000 Tim-Philipp Müller <tim@centricular.net>
24369 ext/ogg/gstogmparse.c: Refactor ogm parse, do better input checking, misc. clean-ups.
24370 Original commit message from CVS:
24371 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
24372 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
24373 (gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
24374 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
24375 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
24376 (gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
24377 (gst_ogm_text_parse_strip_trailing_zeroes),
24378 (gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
24379 (gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
24380 Refactor ogm parse, do better input checking, misc. clean-ups.
24381 Cache incoming events and push them once the source pad has
24382 been created. Don't pass unterminated strings to sscanf().
24383 Strip trailing zeroes from subtitle text output, since they
24384 are not valid UTF-8. Don't push vorbiscomment packets on
24385 the subtitle text pad. Output perfect streams if possible.
24387 2006-08-23 15:27:38 +0000 Wim Taymans <wim.taymans@gmail.com>
24389 tests/check/libs/cddabasesrc.c: Waits for tasks to settle down so that we clean up correctly for valgrind.
24390 Original commit message from CVS:
24391 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
24392 Waits for tasks to settle down so that we clean up correctly for
24395 2006-08-23 15:11:56 +0000 Tim-Philipp Müller <tim@centricular.net>
24397 tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
24398 Original commit message from CVS:
24399 * tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
24400 Unit test fixes: \377 is more likely to fit into 8 bits than \777;
24401 actually return return value in taglists_are_equal.
24403 2006-08-23 12:14:20 +0000 Tim-Philipp Müller <tim@centricular.net>
24405 ext/ogg/gstogmparse.c: Fix crash due to broken bitstream parsing on x86-64: can't make any assumptions about sizeof(s...
24406 Original commit message from CVS:
24407 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
24408 Fix crash due to broken bitstream parsing on x86-64: can't make
24409 any assumptions about sizeof(struct) due to alignment/packing
24410 differences on different architectures. Fixes #351790.
24412 2006-08-22 16:31:47 +0000 Wim Taymans <wim.taymans@gmail.com>
24414 gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
24415 Original commit message from CVS:
24416 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
24417 (gst_riff_parse_chunk), (gst_riff_parse_file_header),
24418 (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
24419 (gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
24420 (gst_riff_parse_info):
24421 Protect public functions against bad input.
24425 2006-08-22 15:50:36 +0000 Tim-Philipp Müller <tim@centricular.net>
24427 gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
24428 Original commit message from CVS:
24429 * gst-libs/gst/riff/riff-ids.h:
24430 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24431 Add voxware audio IDs (even if we can't play it) (#351795).
24433 2006-08-22 15:11:52 +0000 Tim-Philipp Müller <tim@centricular.net>
24435 gst-libs/gst/riff/riff-media.c: Const-ify some arrays and use G_N_ELEMENTS instead of wasting oodles of RAM on termin...
24436 Original commit message from CVS:
24437 * gst-libs/gst/riff/riff-media.c:
24438 (gst_riff_create_video_template_caps),
24439 (gst_riff_create_audio_template_caps),
24440 (gst_riff_create_iavs_template_caps):
24441 Const-ify some arrays and use G_N_ELEMENTS instead
24442 of wasting oodles of RAM on terminator bits.
24444 2006-08-22 08:27:07 +0000 Tim-Philipp Müller <tim@centricular.net>
24446 And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
24447 Original commit message from CVS:
24448 * gst-libs/gst/tag/gstvorbistag.c:
24449 (gst_tag_list_to_vorbiscomment_buffer):
24450 * tests/check/libs/tag.c: (GST_START_TEST):
24451 And the same for _to_vorbiscomment_buffer(): allow
24452 id_data_len == 0 for speex.
24454 2006-08-21 19:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24458 Original commit message from CVS:
24461 2006-08-21 19:01:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24463 Move GDP plugin to -base from -bad. Closes #347783.
24464 Original commit message from CVS:
24466 * docs/plugins/Makefile.am:
24467 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
24468 * docs/plugins/gst-plugins-base-plugins-sections.txt:
24469 * docs/plugins/inspect/plugin-gdp.xml:
24470 * gst/gdp/Makefile.am:
24471 * tests/check/Makefile.am:
24472 Move GDP plugin to -base from -bad. Closes #347783.
24474 2006-08-21 18:34:46 +0000 Tim-Philipp Müller <tim@centricular.net>
24476 gst-libs/gst/tag/gstvorbistag.c: Allow id_data_len == 0 (needed for vorbis comments in Speex files).
24477 Original commit message from CVS:
24478 * gst-libs/gst/tag/gstvorbistag.c:
24479 (gst_tag_list_from_vorbiscomment_buffer):
24480 Allow id_data_len == 0 (needed for vorbis comments in Speex files).
24481 Also add some checks to make sure we don't memcmp() beyond the end of
24482 vorbiscomment buffer if the ID to check for is larger than the buffer.
24483 * tests/check/libs/tag.c: (GST_START_TEST):
24484 Some more tests for gst_tag_list_from_vorbiscomment_buffer().
24486 2006-08-21 16:39:25 +0000 Tim-Philipp Müller <tim@centricular.net>
24488 ext/vorbis/vorbisenc.c: Use vorbis comment utility functions from libgsttag instead of re-inventing the wheel (partia...
24489 Original commit message from CVS:
24490 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
24491 (gst_vorbis_enc_set_metadata):
24492 Use vorbis comment utility functions from libgsttag
24493 instead of re-inventing the wheel (partially fixes #347091).
24495 2006-08-21 11:42:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24497 tests/check/elements/audioconvert.c: Fix leaks. Wait for state transitions that might happen ASYNC, as well as some t...
24498 Original commit message from CVS:
24499 * tests/check/elements/audioconvert.c: (GST_START_TEST):
24500 Fix leaks. Wait for state transitions that might happen ASYNC, as well
24501 as some that won't.
24503 2006-08-21 10:32:51 +0000 Wim Taymans <wim.taymans@gmail.com>
24505 docs/libs/: Don't try to GObject scan the netbuffer as it's not a GObject.
24506 Original commit message from CVS:
24507 * docs/libs/Makefile.am:
24508 * docs/libs/gst-plugins-base-libs-sections.txt:
24509 * docs/libs/gst-plugins-base-libs.types:
24510 Don't try to GObject scan the netbuffer as it's not a GObject.
24512 * gst-libs/gst/netbuffer/gstnetbuffer.c:
24513 * gst-libs/gst/netbuffer/gstnetbuffer.h:
24514 Document GstNetBuffer.
24516 2006-08-21 08:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24518 tests/check/elements/audioconvert.c: Add testcase for caps-size-explosion
24519 Original commit message from CVS:
24520 * tests/check/elements/audioconvert.c: (GST_START_TEST),
24521 (audioconvert_suite):
24522 Add testcase for caps-size-explosion
24524 2006-08-20 13:05:44 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24526 gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size
24527 Original commit message from CVS:
24528 * gst/audioconvert/gstaudioconvert.c:
24529 (gst_audio_convert_get_unit_size), (set_structure_widths):
24530 Lower debug, use g_assert in _get_unit_size
24531 * gst/audioresample/gstaudioresample.c:
24532 (audioresample_get_unit_size):
24533 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
24534 (gst_ffmpegcsp_get_unit_size):
24535 * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
24536 use g_assert in _get_unit_size
24538 2006-08-18 21:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
24541 ChangeLog surgery: fix bug number
24542 Original commit message from CVS:
24543 ChangeLog surgery: fix bug number
24545 2006-08-18 16:43:26 +0000 Wim Taymans <wim.taymans@gmail.com>
24547 Document GstRTPBuffer.
24548 Original commit message from CVS:
24549 * docs/libs/gst-plugins-base-libs-sections.txt:
24550 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
24551 (gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
24552 (gst_rtp_buffer_get_payload_buffer):
24553 * gst-libs/gst/rtp/gstrtpbuffer.h:
24554 Document GstRTPBuffer.
24555 Added function to efficiently strip payload headers.
24556 API: gst_rtp_buffer_get_payload_subbuffer()
24558 2006-08-17 16:52:06 +0000 Tim-Philipp Müller <tim@centricular.net>
24560 gst-libs/gst/tag/gstvorbistag.c: Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT tags and deserialise...
24561 Original commit message from CVS:
24562 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
24563 (gst_tag_to_vorbis_comments):
24564 Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
24565 tags and deserialise them properly as well (#351768).
24566 Add some more gtk-doc blurbs and also some g_return_if_fail().
24567 * tests/check/libs/tag.c: (GST_START_TEST),
24568 (back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
24571 2006-08-17 15:43:40 +0000 Wim Taymans <wim.taymans@gmail.com>
24573 ext/ogg/: Added ogg-in-avi parser element. Fixes #140139.
24574 Original commit message from CVS:
24575 * ext/ogg/Makefile.am:
24576 * ext/ogg/gstogg.c: (plugin_init):
24577 * ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
24578 (gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
24579 (gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
24580 (gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
24581 (gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
24582 (gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
24583 Added ogg-in-avi parser element. Fixes #140139.
24584 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
24585 Fixed a bug in oggdemux debug code.
24586 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
24587 (gst_riff_create_audio_template_caps):
24588 Recognise Ogg in the AVI extensible wave format.
24590 2006-08-17 10:00:00 +0000 Tim-Philipp Müller <tim@centricular.net>
24592 gst-libs/gst/cdda/gstcddabasesrc.c: Make buffer durations add up (duration should be next_ts-ts for perfect streams)....
24593 Original commit message from CVS:
24594 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
24595 Make buffer durations add up (duration should be next_ts-ts for
24596 perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
24598 * tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
24599 (test_buffer_timestamps), (cddabasesrc_suite):
24600 Add unit test for the above.
24601 * tests/check/Makefile.am:
24602 Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
24603 to see what happens.
24605 2006-08-16 11:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
24607 ext/alsa/: Avoid setting and using a NULL device name.
24608 Original commit message from CVS:
24609 * ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
24610 (gst_alsasink_open):
24611 * ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
24612 (gst_alsasrc_open):
24613 Avoid setting and using a NULL device name.
24614 Print more info when we fail to open a device.
24616 2006-08-16 11:28:57 +0000 Tim-Philipp Müller <tim@centricular.net>
24618 API: add gst_tag_parse_extended_comment() (#351426).
24619 Original commit message from CVS:
24620 * docs/libs/gst-plugins-base-libs-sections.txt:
24621 * gst-libs/gst/tag/tag.h:
24622 * gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
24623 API: add gst_tag_parse_extended_comment() (#351426).
24624 * tests/check/Makefile.am:
24625 * tests/check/libs/.cvsignore:
24626 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
24627 Add unit test for gst_tag_parse_extended_comment().
24629 2006-08-15 19:20:16 +0000 Tim-Philipp Müller <tim@centricular.net>
24631 sys/: Fix leak (#351502).
24632 Original commit message from CVS:
24633 * sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
24634 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
24635 Fix leak (#351502).
24637 2006-08-15 17:21:33 +0000 Tim-Philipp Müller <tim@centricular.net>
24640 Original commit message from CVS:
24641 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
24642 * docs/plugins/gst-plugins-base-plugins-sections.txt:
24643 * docs/plugins/gst-plugins-base-plugins.args:
24644 * gst/playback/gstplaybin.c:
24646 * docs/plugins/inspect/plugin-adder.xml:
24647 * docs/plugins/inspect/plugin-alsa.xml:
24648 * docs/plugins/inspect/plugin-audioconvert.xml:
24649 * docs/plugins/inspect/plugin-audiorate.xml:
24650 * docs/plugins/inspect/plugin-audioresample.xml:
24651 * docs/plugins/inspect/plugin-audiotestsrc.xml:
24652 * docs/plugins/inspect/plugin-cdparanoia.xml:
24653 * docs/plugins/inspect/plugin-decodebin.xml:
24654 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
24655 * docs/plugins/inspect/plugin-gnomevfs.xml:
24656 * docs/plugins/inspect/plugin-ogg.xml:
24657 * docs/plugins/inspect/plugin-pango.xml:
24658 * docs/plugins/inspect/plugin-playbin.xml:
24659 * docs/plugins/inspect/plugin-subparse.xml:
24660 * docs/plugins/inspect/plugin-tcp.xml:
24661 * docs/plugins/inspect/plugin-theora.xml:
24662 * docs/plugins/inspect/plugin-typefindfunctions.xml:
24663 * docs/plugins/inspect/plugin-video4linux.xml:
24664 * docs/plugins/inspect/plugin-videorate.xml:
24665 * docs/plugins/inspect/plugin-videoscale.xml:
24666 * docs/plugins/inspect/plugin-videotestsrc.xml:
24667 * docs/plugins/inspect/plugin-volume.xml:
24668 * docs/plugins/inspect/plugin-vorbis.xml:
24669 * docs/plugins/inspect/plugin-ximagesink.xml:
24670 * docs/plugins/inspect/plugin-xvimagesink.xml:
24671 Update to CVS version.
24673 2006-08-14 17:54:01 +0000 Tim-Philipp Müller <tim@centricular.net>
24675 gst/playback/gstplaybin.c: Add "connection-speed" property; re-order redirect messages with multiple redirect locatio...
24676 Original commit message from CVS:
24677 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
24678 (gst_play_bin_set_property), (gst_play_bin_get_property),
24679 (value_list_append_structure_list),
24680 (gst_play_bin_handle_redirect_message),
24681 (gst_play_bin_handle_message):
24682 Add "connection-speed" property; re-order redirect messages with
24683 multiple redirect locations depending on the minimum bitrate if
24684 that information is available and a connection speed is set
24687 2006-08-14 11:41:04 +0000 Tim-Philipp Müller <tim@centricular.net>
24689 gst/playback/gstplaybin.c: Update max volume to the same value that the volume element uses.
24690 Original commit message from CVS:
24691 * gst/playback/gstplaybin.c:
24692 Update max volume to the same value that the volume element uses.
24694 2006-08-14 10:50:15 +0000 Wim Taymans <wim.taymans@gmail.com>
24696 ext/alsa/gstalsamixer.c: Less uglyness..
24697 Original commit message from CVS:
24698 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
24701 2006-08-14 10:49:10 +0000 Wim Taymans <wim.taymans@gmail.com>
24703 ext/ogg/gstoggdemux.c: Add some more debug info.
24704 Original commit message from CVS:
24705 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
24706 (gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
24707 (gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
24708 Add some more debug info.
24709 Don't crash when a seek failed.
24710 Actually return the result of the seek instead of TRUE.
24711 Ignore multiple BOS pages with the same serial so that we don't create
24712 the same stream multiple times.
24713 Post an error when we fail to do the initial seek.
24715 2006-08-13 14:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
24717 ext/alsa/gstalsa.c: Small code cleanup.
24718 Original commit message from CVS:
24719 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
24720 (gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
24721 Small code cleanup.
24722 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
24723 (gst_alsa_mixer_new):
24724 Remove hack that always set the device to hw:0*.
24725 Properly find the card name for whatever device was configured.
24726 Do some better debugging.
24728 * ext/alsa/gstalsamixerelement.c:
24729 (gst_alsa_mixer_element_set_property),
24730 (gst_alsa_mixer_element_change_state):
24732 Handle setting of a NULL device name better.
24734 2006-08-11 15:53:43 +0000 Wim Taymans <wim.taymans@gmail.com>
24736 gst/adder/gstadder.c: Don't clip float values. Fixes #350900.
24737 Original commit message from CVS:
24738 * gst/adder/gstadder.c:
24739 Don't clip float values. Fixes #350900.
24741 2006-08-11 15:33:17 +0000 Andy Wingo <wingo@pobox.com>
24743 gst/tcp/gsttcp.c: Really fix the build?
24744 Original commit message from CVS:
24745 2006-08-11 Andy Wingo <wingo@pobox.com>
24746 * gst/tcp/gsttcp.c: Really fix the build?
24748 2006-08-11 15:29:56 +0000 Andy Wingo <wingo@pobox.com>
24750 gst/tcp/gsttcp.h: For now, always disable deprecation here -- fixes the build.
24751 Original commit message from CVS:
24752 2006-08-11 Andy Wingo <wingo@pobox.com>
24753 * gst/tcp/gsttcp.h: For now, always disable deprecation here --
24756 2006-08-10 13:01:31 +0000 Tim-Philipp Müller <tim@centricular.net>
24758 gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field.
24759 Original commit message from CVS:
24760 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
24761 Float caps shouldn't have a "signed" field.
24763 2006-08-10 08:56:22 +0000 Tim-Philipp Müller <tim@centricular.net>
24765 ext/ogg/gstoggdemux.c: Implement SEEKING query in its most basic form, so that we can at least check if we're seekabl...
24766 Original commit message from CVS:
24767 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
24768 Implement SEEKING query in its most basic form, so that we can
24769 at least check if we're seekable or not (#350655).
24771 2006-08-09 14:42:58 +0000 Tim-Philipp Müller <tim@centricular.net>
24773 gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
24774 Original commit message from CVS:
24775 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
24776 The checks here are not even close to anything that would
24777 justify MAXIMUM probability, lowering to POSSIBLE until someone
24778 fixes the checks (case at hand: quicktime redirection files
24779 might start with 00 00 01 XX and pass the checks here just
24780 fine, see #350399).
24782 2006-08-08 13:57:29 +0000 Edward Hervey <bilboed@bilboed.com>
24784 tests/check/elements/gdpdepay.c: I forgot to include the file containing the #define :)
24785 Original commit message from CVS:
24786 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
24787 I forgot to include the file containing the #define :)
24788 Now includes "config.h"
24790 2006-08-08 13:45:44 +0000 Edward Hervey <bilboed@bilboed.com>
24792 tests/check/elements/gdpdepay.c: Ignore test known to fail on PPC64. See #348114.
24793 Original commit message from CVS:
24794 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
24795 Ignore test known to fail on PPC64. See #348114.
24797 2006-08-08 08:41:13 +0000 Sjoerd Simons <sjoerd@luon.net>
24799 gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
24800 Original commit message from CVS:
24801 Patch by: Sjoerd Simons <sjoerd at luon net>
24802 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
24803 Better detection for multipart/x-mixed-replace: accept leading
24804 whitespaces before the boundary marker as well (as our very own
24805 multipartmux used to produce) (#349068).
24807 2006-08-07 08:26:03 +0000 Young-Ho Cha <ganadist@chollian.net>
24809 gst-libs/gst/riff/: Detect DTS audio streams (#350157).
24810 Original commit message from CVS:
24811 Patch by: Young-Ho Cha <ganadist at chollian net>
24812 * gst-libs/gst/riff/riff-ids.h:
24813 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
24814 (gst_riff_create_audio_template_caps):
24815 Detect DTS audio streams (#350157).
24817 2006-08-05 17:08:05 +0000 Andy Wingo <wingo@pobox.com>
24819 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (gst_theora_parse_class_init) (theora_parse_dispose, theora_par...
24820 Original commit message from CVS:
24821 2006-08-05 Andy Wingo <wingo@pobox.com>
24822 * ext/theora/gsttheoraparse.h:
24823 * ext/theora/theoraparse.c (gst_theora_parse_class_init)
24824 (theora_parse_dispose, theora_parse_set_property)
24825 (theora_parse_get_property, theora_parse_munge_granulepos)
24826 (theora_parse_push_buffer, theora_parse_change_state): Add a
24827 property 'synchronization-points' to fix badly synchronized oggs.
24829 2006-08-04 13:20:23 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
24831 gst/gdp/gstgdpdepay.c: Fix event parsing by gdpdepay. Fixes #349916.
24832 Original commit message from CVS:
24833 2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
24834 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
24835 Fix event parsing by gdpdepay. Fixes #349916.
24837 2006-08-03 15:04:42 +0000 Tim-Philipp Müller <tim@centricular.net>
24839 tests/check/: Add a few tests for the channel position stuff in libgstaudio.
24840 Original commit message from CVS:
24841 * tests/check/Makefile.am:
24842 * tests/check/libs/.cvsignore:
24843 * tests/check/libs/audio.c: (structure_contains_channel_positions),
24844 (fixed_caps_have_channel_positions), (GST_START_TEST),
24845 (audio_suite), (main):
24846 Add a few tests for the channel position stuff in libgstaudio.
24848 2006-08-03 14:16:06 +0000 Tim-Philipp Müller <tim@centricular.net>
24850 ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
24851 Original commit message from CVS:
24852 * ext/alsa/gstalsa.c: (caps_add_channel_configuration),
24853 (gst_alsa_detect_channels):
24854 * ext/alsa/gstalsasink.c:
24855 Add support for cards that (only) do more than 8 channels,
24856 like the Delta 44 (#345188).
24857 * gst-libs/gst/audio/multichannel.c:
24858 (gst_audio_check_channel_positions):
24859 * gst-libs/gst/audio/multichannel.h:
24860 API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
24861 unspecified channel position and cannot be combined with any
24862 of the other audio channel positions; adjust position layout
24863 checks accordingly (#345188).
24865 2006-08-03 11:15:53 +0000 Tim-Philipp Müller <tim@centricular.net>
24867 gst/typefind/gsttypefindfunctions.c: Recognise ancient RealAudio files (see #349779).
24868 Original commit message from CVS:
24869 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
24870 Recognise ancient RealAudio files (see #349779).
24872 2006-08-03 09:01:25 +0000 Jens Granseuer <jensgr@gmx.net>
24874 gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
24875 Original commit message from CVS:
24876 Patch by: Jens Granseuer <jensgr at gmx net>
24877 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
24878 Add typefinder for Interplay's MVE format (#348973).
24880 2006-08-02 17:03:29 +0000 Marcel Moreaux <marcelm@luon.net>
24882 gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
24883 Original commit message from CVS:
24884 Patch by: Marcel Moreaux <marcelm at luon dot net>
24885 * gst-libs/gst/rtp/gstbasertpdepayload.c:
24886 (gst_base_rtp_depayload_add_to_queue):
24887 * gst-libs/gst/rtp/gstbasertpdepayload.h:
24888 Handle RTP sequence number rollover.
24889 Disable jitterbuffer by default.
24891 2006-08-02 16:56:19 +0000 Wim Taymans <wim.taymans@gmail.com>
24893 gst/gdp/gstgdpdepay.c: Disable seeking.
24894 Original commit message from CVS:
24895 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
24896 (gst_gdp_depay_finalize), (gst_gdp_depay_sink_event),
24897 (gst_gdp_depay_src_event), (gst_gdp_depay_chain),
24898 (gst_gdp_depay_change_state):
24901 Clear adapter on disconts.
24902 Clear caps when going to READY instead of NULL
24903 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
24904 (gst_gdp_pay_init), (gst_gdp_pay_finalize), (gst_gdp_pay_reset),
24905 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
24906 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
24907 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
24908 (gst_gdp_pay_sink_event), (gst_gdp_pay_src_event),
24909 (gst_gdp_pay_change_state):
24910 * gst/gdp/gstgdppay.h:
24911 Reset payloader when going to READY.
24912 Fix leaked buffers in ->queue on push errors.
24915 Create packetizer in _init, free in _finalize.
24917 2006-07-31 08:48:36 +0000 Wim Taymans <wim.taymans@gmail.com>
24919 gst/gdp/gstgdpdepay.c: Consume all events except EOS because we generate events from the gdp payload instead. Fixes #...
24920 Original commit message from CVS:
24921 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
24922 (gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
24923 Consume all events except EOS because we generate events from
24924 the gdp payload instead. Fixes #349204
24926 2006-07-28 17:17:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24928 gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
24929 Original commit message from CVS:
24930 * gst/audioresample/gstaudioresample.c: (audioresample_stop),
24931 (audioresample_set_caps):
24932 Don't leak references to the incoming caps. Clean them up when
24934 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
24935 (gst_video_scale_finalize):
24936 Don't leak our temporary pixel buffer.
24937 * tests/check/Makefile.am:
24938 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
24939 (GST_START_TEST), (simple_launch_lines_suite):
24940 Fix leaks and re-enable the test for valgrind checking.
24942 2006-07-28 16:39:31 +0000 Sjoerd Simons <sjoerd@luon.net>
24944 gst/typefind/gsttypefindfunctions.c: Add typefind function for multipart/x-mixed-replace (#348916).
24945 Original commit message from CVS:
24946 Patch by: Sjoerd Simons <sjoerd at luon net>
24947 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
24949 Add typefind function for multipart/x-mixed-replace (#348916).
24951 2006-07-28 14:14:58 +0000 Wim Taymans <wim.taymans@gmail.com>
24953 gst/adder/gstadder.c: Fix leak in duration query.
24954 Original commit message from CVS:
24955 * gst/adder/gstadder.c: (gst_adder_setcaps),
24956 (gst_adder_query_duration):
24957 Fix leak in duration query.
24958 Reflow some docs and notes.
24960 2006-07-28 13:42:48 +0000 Michael Smith <msmith@xiph.org>
24962 tests/check/pipelines/vorbisenc.c: Enable Andy's extra vorbisenc test, now that it passes. Also fix one aspect of it.
24963 Original commit message from CVS:
24964 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
24966 Enable Andy's extra vorbisenc test, now that it passes. Also fix one
24969 2006-07-28 12:48:21 +0000 Michael Smith <msmith@xiph.org>
24971 ext/vorbis/vorbisenc.*: Handle discontinuities in the input vorbis stream correctly, so that the output is properly t...
24972 Original commit message from CVS:
24973 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
24974 (gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
24975 (gst_vorbis_enc_push_buffer),
24976 (gst_vorbis_enc_buffer_check_discontinuous),
24977 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
24978 * ext/vorbis/vorbisenc.h:
24979 Handle discontinuities in the input vorbis stream correctly,
24980 so that the output is properly timestamped (and has good granulepos
24981 values). Needs some oggmux fixes too.
24983 2006-07-27 10:52:52 +0000 Kai Vehmanen <kv2004@eca.cx>
24985 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
24986 Original commit message from CVS:
24987 patch by: Kai Vehmanen <kv2004 eca cx>
24988 * gst-libs/gst/rtp/gstbasertpdepayload.c:
24989 (gst_base_rtp_depayload_chain),
24990 (gst_base_rtp_depayload_handle_sink_event),
24991 (gst_base_rtp_depayload_change_state):
24992 Don't send multiple newsegments with different formats.
24995 2006-07-26 15:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
24997 ext/ogg/gstoggdemux.c: Make seeking in ogg more accurate again by doing the more correct granuletime to stream time c...
24998 Original commit message from CVS:
24999 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
25000 (gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
25001 Make seeking in ogg more accurate again by doing the more correct
25002 granuletime to stream time conversion.
25004 2006-07-26 10:59:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25006 gst/tcp/gstmultifdsink.c: debug a little more understandably do not use goto as a substitute for break, especially if...
25007 Original commit message from CVS:
25008 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
25009 (gst_multi_fd_sink_new_client):
25010 debug a little more understandably
25011 do not use goto as a substitute for break, especially if
25012 break is also being used
25014 2006-07-26 10:55:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25016 * gst/tcp/gsttcp.c:
25017 move a recurring normal event to LOG, where it should be
25018 Original commit message from CVS:
25019 move a recurring normal event to LOG, where it should be
25021 2006-07-26 10:54:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25023 * ext/vorbis/vorbisdec.c:
25025 Original commit message from CVS:
25028 2006-07-26 10:52:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25030 gst/gdp/gstgdpdepay.c: proxying get/set caps is the wrong thing to do, since we really do change caps quite fundament...
25031 Original commit message from CVS:
25032 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
25033 proxying get/set caps is the wrong thing to do, since we really
25034 do change caps quite fundamentally
25035 * tests/check/elements/gdpdepay.c:
25036 * tests/check/elements/gdppay.c:
25037 remove declaration of buffers, it's already done in gstcheck.h
25039 2006-07-26 10:31:06 +0000 Tim-Philipp Müller <tim@centricular.net>
25041 gst/playback/: Remove GLib-2.6 compatibility cruft.
25042 Original commit message from CVS:
25043 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
25044 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
25045 Remove GLib-2.6 compatibility cruft.
25047 2006-07-24 16:47:10 +0000 Wim Taymans <wim.taymans@gmail.com>
25049 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
25050 Original commit message from CVS:
25051 * gst-libs/gst/audio/gstbaseaudiosink.c:
25052 (gst_base_audio_sink_render):
25053 Don't try to align a sample to an unknown value.
25055 2006-07-24 15:14:17 +0000 Wim Taymans <wim.taymans@gmail.com>
25057 gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
25058 Original commit message from CVS:
25059 * gst-libs/gst/audio/gstbaseaudiosink.c:
25060 (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
25061 When the audio clock is slaved to another clock, never try to align
25062 samples but trust the rate interpolation algorithm.
25064 2006-07-24 14:34:42 +0000 Wim Taymans <wim.taymans@gmail.com>
25066 ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
25067 Original commit message from CVS:
25068 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
25069 Don't try to calculate silence samples, base class does this much
25071 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
25072 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
25073 (gst_ring_buffer_acquire):
25074 Calculate silence samples correctly.
25075 * gst-libs/gst/audio/gstringbuffer.h:
25078 2006-07-22 17:01:12 +0000 Tim-Philipp Müller <tim@centricular.net>
25080 gst/typefind/gsttypefindfunctions.c: Limit search for the first markup tag to the first few kB of the file. If we don...
25081 Original commit message from CVS:
25082 * gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
25083 Limit search for the first markup tag to the first few kB of
25084 the file. If we don't find one there, it's highly unlikely that
25085 this is an XML(-ish) file.
25087 2006-07-21 17:04:06 +0000 Andy Wingo <wingo@pobox.com>
25089 tests/check/pipelines/theoraenc.c (test_discontinuity): Similar test to the one in vorbisenc. Also commented out.
25090 Original commit message from CVS:
25091 2006-07-21 Andy Wingo <wingo@pobox.com>
25092 * tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
25093 test to the one in vorbisenc. Also commented out.
25095 2006-07-21 16:54:19 +0000 Andy Wingo <wingo@pobox.com>
25097 tests/check/pipelines/vorbisenc.c: New test, commented out until Mike lands some elite vorbisenc patches.
25098 Original commit message from CVS:
25099 2006-07-21 Andy Wingo <wingo@pobox.com>
25100 * tests/check/pipelines/vorbisenc.c:
25101 (test_discontinuity): New test, commented out until Mike lands
25102 some elite vorbisenc patches.
25104 2006-07-21 15:59:24 +0000 Andy Wingo <wingo@pobox.com>
25106 tests/check/pipelines/: Port to bufferstraw.
25107 Original commit message from CVS:
25108 2006-07-21 Andy Wingo <wingo@pobox.com>
25109 * tests/check/pipelines/vorbisenc.c:
25110 * tests/check/pipelines/theoraenc.c: Port to bufferstraw.
25111 Bufferstraw was actually factored out of these tests. Now we share
25114 2006-07-21 11:03:28 +0000 Wim Taymans <wim.taymans@gmail.com>
25116 ext/theora/theoradec.c: Better clipping.
25117 Original commit message from CVS:
25118 * ext/theora/theoradec.c: (clip_buffer):
25121 2006-07-21 10:43:54 +0000 Wim Taymans <wim.taymans@gmail.com>
25123 gst-libs/gst/audio/gstaudiosink.c: Fix leak.
25124 Original commit message from CVS:
25125 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
25126 (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
25127 (gst_audioringbuffer_release), (gst_audioringbuffer_stop):
25129 Avoid type casting when we can.
25130 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
25133 2006-07-20 16:57:29 +0000 Tim-Philipp Müller <tim@centricular.net>
25135 ext/alsa/gstalsamixerelement.c: Make state change fail if the specified device can't be opened for some reason.
25136 Original commit message from CVS:
25137 * ext/alsa/gstalsamixerelement.c:
25138 (gst_alsa_mixer_element_change_state):
25139 Make state change fail if the specified device can't be opened
25142 2006-07-20 10:42:21 +0000 Wim Taymans <wim.taymans@gmail.com>
25144 gst/playback/test.c: Example of a small audio/video player using decodebin.
25145 Original commit message from CVS:
25146 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
25147 (cb_newpad), (main):
25148 Example of a small audio/video player using decodebin.
25150 2006-07-20 05:56:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25152 gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
25153 Original commit message from CVS:
25154 * gst-libs/gst/riff/riff-ids.h:
25155 Add 'fact' chunk id
25157 2006-07-19 18:20:43 +0000 Wim Taymans <wim.taymans@gmail.com>
25159 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't assert when not negotiated but post a meaningfull error message. Fixes ...
25160 Original commit message from CVS:
25161 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25162 (gst_base_rtp_depayload_chain),
25163 (gst_base_rtp_depayload_change_state):
25164 Don't assert when not negotiated but post a meaningfull
25165 error message. Fixes #347918.
25166 * gst-libs/gst/rtp/gstbasertppayload.c:
25167 Add comment about better default MTU size.
25168 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
25169 Small cleanups, start docs.
25171 2006-07-19 14:46:36 +0000 Martin Szulecki <compiz@sukimashita.com>
25173 sys/v4l/gstv4lelement.c: If "device-name" is requested and the device is not open, try to temporarily open it to obta...
25174 Original commit message from CVS:
25175 Patch by: Martin Szulecki
25176 * sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
25177 If "device-name" is requested and the device is not
25178 open, try to temporarily open it to obtain this
25179 information (#342494).
25181 2006-07-19 12:25:00 +0000 Tim-Philipp Müller <tim@centricular.net>
25183 gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
25184 Original commit message from CVS:
25185 * gst-libs/gst/tag/gstid3tag.c:
25186 Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
25187 * gst-libs/gst/tag/gsttageditingprivate.h:
25188 * gst-libs/gst/tag/gstvorbistag.c:
25189 Some more random const-ifications.
25191 2006-07-18 19:48:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25193 gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
25194 Original commit message from CVS:
25195 * gst-libs/gst/riff/riff-ids.h:
25196 * gst-libs/gst/riff/riff-media.c:
25197 (gst_riff_create_video_template_caps):
25198 Add more FOURCCs (sort list to make stuff easier to find),
25199 add comment what those 16 bytes in struct _gst_riff_strh according to
25202 2006-07-17 14:17:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25204 gst/gdp/: remove parent_class setting, BOILERPLATE does this fix typo in comment
25205 Original commit message from CVS:
25206 2006-07-17 Thomas Vander Stichele <thomas at apestaart dot org>
25207 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
25208 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
25209 remove parent_class setting, BOILERPLATE does this
25210 (gst_gdp_pay_reset_streamheader):
25211 fix typo in comment
25213 2006-07-17 13:48:10 +0000 Tim-Philipp Müller <tim@centricular.net>
25215 gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
25216 Original commit message from CVS:
25217 * gst-libs/gst/audio/multichannel.c:
25218 (gst_audio_check_channel_positions),
25219 (gst_audio_fixate_channel_positions):
25220 Const-ify two arrays.
25222 2006-07-17 12:33:42 +0000 Tim-Philipp Müller <tim@centricular.net>
25224 ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
25225 Original commit message from CVS:
25226 * ext/alsa/gstalsa.c: (caps_add_channel_configuration):
25227 Fix typo, so that alsasink also advertises 8 channels
25228 if that's supported (tags: can, worms, open, alsa, ph34r).
25230 2006-07-17 12:01:04 +0000 Wim Taymans <wim.taymans@gmail.com>
25232 ext/ogg/gstoggdemux.c: *sigh*, when is the compiler going to warn when the comments are out-of-sync with the code.. R...
25233 Original commit message from CVS:
25234 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
25235 (gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
25236 *sigh*, when is the compiler going to warn when the comments
25237 are out-of-sync with the code.. Refix case of busted theora
25238 headers with 0 granule pos.
25240 2006-07-14 17:56:59 +0000 Wim Taymans <wim.taymans@gmail.com>
25242 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix 99% cpu load by waiting for absolute times on the clock. Fixes #347300.
25243 Original commit message from CVS:
25244 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25245 (gst_base_rtp_depayload_wait),
25246 (gst_base_rtp_depayload_change_state),
25247 (gst_base_rtp_depayload_set_property),
25248 (gst_base_rtp_depayload_get_property):
25249 Fix 99% cpu load by waiting for absolute times on the
25250 clock. Fixes #347300.
25252 2006-07-14 17:07:08 +0000 Andy Wingo <wingo@pobox.com>
25254 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (theora_parse_drain_event_queue) (theora_parse_push_headers, th...
25255 Original commit message from CVS:
25256 2006-07-14 Andy Wingo <wingo@pobox.com>
25257 * ext/theora/gsttheoraparse.h:
25258 * ext/theora/theoraparse.c (theora_parse_drain_event_queue)
25259 (theora_parse_push_headers, theora_parse_clear_queue)
25260 (theora_parse_drain_queue_prematurely, )
25261 (theora_parse_sink_event, theora_parse_change_state): Queue events
25262 until we initialized our state, like in vorbisparse.
25264 2006-07-14 16:45:17 +0000 Iain * <iaingnome@gmail.com>
25266 ext/vorbis/vorbisparse.h: ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue) (vorbis_parse_push_headers, vorbi...
25267 Original commit message from CVS:
25268 2006-07-14 Andy Wingo <wingo@pobox.com>
25269 * ext/vorbis/vorbisparse.h:
25270 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
25271 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
25272 (vorbis_parse_drain_queue_prematurely, )
25273 (vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
25274 until we have initialized our state. Fixes seeking after an
25276 2006-07-14 Andy Wingo <wingo@pobox.com>
25277 Patch by: Iain * <iaingnome@gmail.com>
25278 * ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.
25280 2006-07-14 15:52:39 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25282 configure.ac: Bump nano back to CVS
25283 Original commit message from CVS:
25285 Bump nano back to CVS
25287 === release 0.10.9 ===
25289 2006-07-14 15:51:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25291 configure.ac: releasing 0.10.9, "I walk the line"
25292 Original commit message from CVS:
25293 2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
25295 releasing 0.10.9, "I walk the line"
25297 2006-07-14 14:12:40 +0000 Michael Smith <msmith@xiph.org>
25299 tests/check/pipelines/vorbisenc.c: Move a g_cond_signal to earlier to avoid sometimes deadlocking (commonly happens w...
25300 Original commit message from CVS:
25301 * tests/check/pipelines/vorbisenc.c: (stop_pipeline):
25302 Move a g_cond_signal to earlier to avoid sometimes deadlocking
25303 (commonly happens when running this test under valgrind) when trying
25304 to remove the buffer probe.
25306 2006-07-14 10:34:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25308 * gst/gdp/Makefile.am:
25309 build as a plugin, not a lib
25310 Original commit message from CVS:
25311 build as a plugin, not a lib
25313 2006-07-13 16:43:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25315 sys/xvimage/xvimagesink.c: Fix missing g_unlock from the previous commit
25316 Original commit message from CVS:
25317 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
25318 Fix missing g_unlock from the previous commit
25320 2006-07-13 16:34:04 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25322 sys/: Implement a locking order to ensure we always take the object lock before the x_lock and never vice-versa.
25323 Original commit message from CVS:
25324 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
25325 (gst_ximagesink_change_state):
25326 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
25327 (gst_xvimagesink_change_state):
25328 Implement a locking order to ensure we always take the object lock
25329 before the x_lock and never vice-versa.
25331 2006-07-13 15:25:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25333 docs/plugins/: add more plugins and elements to docs
25334 Original commit message from CVS:
25335 * docs/plugins/Makefile.am:
25336 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
25337 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
25338 add more plugins and elements to docs
25339 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
25340 fix segfaults due to wrong g_free
25342 * gst/gdp/gstgdppay.c:
25345 2006-07-13 14:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25347 gst/playback/gstdecodebin.c: Fix a caps leak when linking (#347304)
25348 Original commit message from CVS:
25349 * gst/playback/gstdecodebin.c: (find_compatibles):
25350 Fix a caps leak when linking (#347304)
25351 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
25352 (gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
25353 (gst_ximagesink_change_state):
25354 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
25355 (gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
25356 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
25357 (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
25358 Don't leak shared memory resources. Use the object lock to protect
25359 against the xcontext disappearing while returning a buffer from the
25360 pipeline. (#347304)
25362 2006-07-12 14:20:43 +0000 Edward Hervey <bilboed@bilboed.com>
25364 ext/vorbis/vorbisdec.c: gst_tag_list_merge() returns a new object. Take that into account when using it. This avoids ...
25365 Original commit message from CVS:
25366 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
25367 (vorbis_handle_comment_packet):
25368 gst_tag_list_merge() returns a new object. Take that into account when
25369 using it. This avoids memleak.
25370 Revert previous commit which is not needed.
25372 2006-07-12 13:30:20 +0000 Edward Hervey <bilboed@bilboed.com>
25374 ext/vorbis/vorbisdec.c: Reset the decoder in finalize so that all fields get cleared.
25375 Original commit message from CVS:
25376 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
25377 Reset the decoder in finalize so that all fields get cleared.
25379 2006-07-12 13:24:19 +0000 Wim Taymans <wim.taymans@gmail.com>
25381 gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
25382 Original commit message from CVS:
25383 * gst-libs/gst/audio/gstbaseaudiosrc.c:
25384 (gst_base_audio_src_set_clock),
25385 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
25386 Don't try to post an error message when setting the clock fails
25387 as this can happen when adding an element to a bin which will then
25388 deadlock. Fixes #347296.
25390 2006-07-12 13:04:15 +0000 Edward Hervey <bilboed@bilboed.com>
25392 ext/vorbis/vorbisdec.c: Post tag messages on the bus even if we're not initialized.
25393 Original commit message from CVS:
25394 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
25395 (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
25396 (vorbis_handle_type_packet):
25397 Post tag messages on the bus even if we're not initialized.
25398 If we're not initialized, we still postpone the event pushing of tags.
25400 2006-07-12 11:28:37 +0000 Wim Taymans <wim.taymans@gmail.com>
25402 Revert last two changes that broke the freeze.
25403 Original commit message from CVS:
25404 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
25405 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
25406 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
25407 Revert last two changes that broke the freeze.
25409 2006-07-12 10:59:55 +0000 Wim Taymans <wim.taymans@gmail.com>
25411 ext/alsa/gstalsasink.c: basesink calculates silence sample correctly for us.
25412 Original commit message from CVS:
25413 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
25414 basesink calculates silence sample correctly for us.
25416 2006-07-12 10:58:42 +0000 Wim Taymans <wim.taymans@gmail.com>
25418 gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
25419 Original commit message from CVS:
25420 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
25421 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
25422 Calculate correct silence samples so we don't fill our ringbuffer
25425 2006-07-12 10:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
25427 ext/vorbis/vorbisdec.*: Delay sending events (newsegment, tags) until the decoder is properly initialized.
25428 Original commit message from CVS:
25429 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
25430 (gst_vorbis_dec_reset), (vorbis_dec_sink_event),
25431 (vorbis_handle_comment_packet), (vorbis_handle_type_packet):
25432 * ext/vorbis/vorbisdec.h:
25433 Delay sending events (newsegment, tags) until the decoder is properly
25437 2006-07-11 22:40:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25454 Original commit message from CVS:
25457 2006-07-11 21:04:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25459 tests/check/elements/audioconvert.c: Patch from #347221 adding a test for audioconvert channel remappings.
25460 Original commit message from CVS:
25461 * tests/check/elements/audioconvert.c: (get_float_mc_caps),
25462 (get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
25463 Patch from #347221 adding a test for audioconvert
25464 channel remappings.
25466 2006-07-11 12:03:25 +0000 Tim-Philipp Müller <tim@centricular.net>
25468 gst/subparse/gstssaparse.c: Don't include the terminating NUL in the buffer size, it's only there for extra paranoia ...
25469 Original commit message from CVS:
25470 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
25471 (gst_ssa_parse_parse_line):
25472 Don't include the terminating NUL in the buffer size,
25473 it's only there for extra paranoia (would add random
25474 '*' characters at the end of each subtitle since the
25475 terminator itself is not valid UTF-8 technically).
25476 Also fix indenting after boilerplate macro.
25478 2006-07-10 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.net>
25480 gst/playback/gstdecodebin.c: Also emit 'unknown-type' signal (which should really be called unhandled-type) if we fou...
25481 Original commit message from CVS:
25482 * gst/playback/gstdecodebin.c: (close_pad_link):
25483 Also emit 'unknown-type' signal (which should really be
25484 called unhandled-type) if we found potential decoders/demuxers
25485 in the registry but none of them worked in the end (as in the
25486 case where the plugins don't exist any longer but are still
25487 listed in the registry). Fixes #329798.
25489 2006-07-08 13:48:58 +0000 Andy Wingo <wingo@pobox.com>
25492 * ext/theora/theoraparse.c:
25493 theoraparse.c (theora_parse_push_buffer)
25494 Original commit message from CVS:
25495 2006-07-08 Andy Wingo <wingo@pobox.com>
25496 * theoraparse.c (theora_parse_push_buffer)
25497 (theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
25498 Add some more debugging. Fix granulepos reconstruction in the face
25499 of discontinuities.
25501 2006-07-06 15:54:50 +0000 Wim Taymans <wim.taymans@gmail.com>
25503 gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
25504 Original commit message from CVS:
25505 * gst-libs/gst/audio/gstbaseaudiosink.c:
25506 (gst_base_audio_sink_class_init),
25507 (gst_base_audio_sink_provide_clock):
25508 Use gobject_class instead of G_OBJECT_CLASS (klass)
25509 * gst-libs/gst/audio/gstbaseaudiosrc.c:
25510 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
25511 (gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
25512 (gst_base_audio_src_get_time),
25513 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
25514 (gst_base_audio_src_create_ringbuffer):
25515 Fix latency and buffer-time constants and properties ala basesink.
25516 Implement pull based scheduling. Fixes #346527.
25517 Set default blocksize in GstBaseSrc to 0, we default to pushing out
25519 Refuse slaving to another clock instead of silently not working.
25520 Only provide a clock when we are actually able to do so.
25521 Various small cleanups and compiler hints.
25523 2006-07-06 13:23:07 +0000 Lutz Mueller <lutz@topfrose.de>
25525 gst/typefind/gsttypefindfunctions.c: Add typefinding for text/html (#346581).
25526 Original commit message from CVS:
25527 Patch by: Lutz Mueller <lutz at topfrose de>
25528 * gst/typefind/gsttypefindfunctions.c: (html_type_find),
25530 Add typefinding for text/html (#346581).
25532 2006-07-06 13:12:02 +0000 Tim-Philipp Müller <tim@centricular.net>
25534 gst/typefind/gsttypefindfunctions.c: Fix SMIL typefinding, make xml_check_first_element() more useful.
25535 Original commit message from CVS:
25536 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
25537 (xml_check_first_element), (xml_type_find), (smil_type_find):
25538 Fix SMIL typefinding, make xml_check_first_element() more
25541 2006-07-06 13:04:24 +0000 Tim-Philipp Müller <tim@centricular.net>
25543 gst/playback/gstplaybasebin.*: Protect list of elements with a subtitle-encoding property and the subtitle encoding m...
25544 Original commit message from CVS:
25545 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
25546 (gst_play_base_bin_finalize), (decodebin_element_added_cb),
25547 (decodebin_element_removed_cb), (gst_play_base_bin_set_property):
25548 * gst/playback/gstplaybasebin.h:
25549 Protect list of elements with a subtitle-encoding property and
25550 the subtitle encoding member itself with a lock of their own
25551 instead of using the object lock. This prevents a dead-lock in
25552 the element-remove callback in some circumstances when shutting
25555 2006-07-05 20:11:13 +0000 Sébastien Moutte <sebastien@moutte.net>
25557 win32/common/libgsttag.def: Export some new functions.
25558 Original commit message from CVS:
25559 * win32/common/libgsttag.def:
25560 Export some new functions.
25561 * win32/vs6/libgstogg.dsp:
25562 Add a link to libgsttag-0.10.lib.
25564 2006-07-04 16:50:21 +0000 Tim-Philipp Müller <tim@centricular.net>
25566 ext/alsa/gstalsamixertrack.c: Some const-ification.
25567 Original commit message from CVS:
25568 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
25569 Some const-ification.
25571 2006-07-04 14:06:03 +0000 Wim Taymans <wim.taymans@gmail.com>
25573 gst/playback/gstplaybasebin.c: Improve checking if we are dealing with a stream. Added some more uris that need buffe...
25574 Original commit message from CVS:
25575 * gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
25576 Improve checking if we are dealing with a stream. Added some
25577 more uris that need buffering.
25579 2006-07-03 10:43:31 +0000 Edward Hervey <bilboed@bilboed.com>
25581 ext/vorbis/vorbisdec.c: Remove unused variable.
25582 Original commit message from CVS:
25583 * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
25584 Remove unused variable.
25586 2006-07-02 21:48:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25588 Makefile.am: include lcov.mak
25589 Original commit message from CVS:
25593 add GCOV_LIBS to GST_LIBS
25595 2006-07-02 11:08:58 +0000 Michael Sheldon <webmaster@mikeasoft.com>
25597 ext/alsa/gstalsasrc.c: Add 32 bps to template caps and increase channels range from [1,2] to [1,MAX]. See #346326.
25598 Original commit message from CVS:
25599 Patch by: Michael Sheldon <webmaster at mikeasoft com>
25600 * ext/alsa/gstalsasrc.c:
25601 Add 32 bps to template caps and increase channels range
25602 from [1,2] to [1,MAX]. See #346326.
25604 2006-06-30 12:04:51 +0000 Tim-Philipp Müller <tim@centricular.net>
25606 gst-libs/gst/riff/riff-media.c: Recognise 'WMVA' video codec fourcc (#345879).
25607 Original commit message from CVS:
25608 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
25609 Recognise 'WMVA' video codec fourcc (#345879).
25611 2006-06-29 12:21:06 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
25613 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
25614 Original commit message from CVS:
25615 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
25616 Fixed nasty memory leak
25618 2006-06-26 13:19:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25620 gst/tcp/gsttcp.c: fix logging
25621 Original commit message from CVS:
25622 * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
25623 (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
25626 2006-06-23 16:45:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25628 gst/playback/gstdecodebin.c: Protect remove_fakesink using a mutex, so that we don't try and remove the fakesink simu...
25629 Original commit message from CVS:
25630 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
25631 (gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
25632 (remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
25633 Protect remove_fakesink using a mutex, so that we don't try and
25634 remove the fakesink simultaneously from multiple threads.
25635 When going from READY to PAUSED, restore the fakesink, so that
25636 it is there when decodebin gets reused.
25638 2006-06-23 09:53:09 +0000 Tim-Philipp Müller <tim@centricular.net>
25640 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
25641 Original commit message from CVS:
25642 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
25643 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25644 * gst-libs/gst/rtp/gstbasertppayload.c:
25645 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
25646 * gst/tcp/gstmultifdsink.c:
25647 * gst/tcp/gsttcpclientsink.c:
25648 * gst/tcp/gsttcpclientsrc.c:
25649 * gst/tcp/gsttcpserversink.c:
25650 * gst/tcp/gsttcpserversrc.c:
25651 * gst/videorate/gstvideorate.c:
25652 * gst/videotestsrc/gstvideotestsrc.c:
25653 * sys/v4l/gstv4ljpegsrc.c:
25654 * sys/v4l/gstv4lmjpegsink.c:
25655 * sys/v4l/gstv4lsrc.c:
25656 * tests/examples/seek/scrubby.c:
25657 * tests/examples/seek/seek.c:
25658 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
25660 2006-06-23 09:09:44 +0000 Tim-Philipp Müller <tim@centricular.net>
25662 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) and fix one GObject boilerplate macro.
25663 Original commit message from CVS:
25664 * ext/directfb/dfbvideosink.c:
25665 * ext/gsm/gstgsmdec.c:
25666 * ext/gsm/gstgsmenc.c:
25667 * ext/libmms/gstmms.c:
25668 * ext/neon/gstneonhttpsrc.c:
25669 * ext/theora/theoradec.c:
25670 * gst/freeze/gstfreeze.c:
25671 * gst/gdp/gstgdpdepay.c:
25672 * gst/gdp/gstgdppay.c:
25673 * sys/glsink/glimagesink.c:
25674 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
25675 and fix one GObject boilerplate macro.
25677 2006-06-22 12:13:31 +0000 Tim-Philipp Müller <tim@centricular.net>
25679 gst-libs/gst/tag/tags.c: Second field in GEnumValue shouldn't be a description, but a stringified version of the enum...
25680 Original commit message from CVS:
25681 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
25682 Second field in GEnumValue shouldn't be a description,
25683 but a stringified version of the enum value.
25685 2006-06-22 12:03:14 +0000 Wim Taymans <wim.taymans@gmail.com>
25687 sys/ximage/ximagesink.c: Avoid type checking in buffer casts.
25688 Original commit message from CVS:
25689 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
25690 (gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
25691 (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
25692 Avoid type checking in buffer casts.
25693 Avoid caps copy in buffer_alloc when we can.
25694 Use pad_peer_accept.
25696 2006-06-22 11:01:41 +0000 Tim-Philipp Müller <tim@centricular.net>
25698 gst-libs/gst/tag/tag.h: Oops, make that 'Since: 0.10.9'.
25699 Original commit message from CVS:
25700 * gst-libs/gst/tag/tag.h:
25701 Oops, make that 'Since: 0.10.9'.
25703 2006-06-22 10:55:05 +0000 Tim-Philipp Müller <tim@centricular.net>
25705 API: add GstTagImageType enum to describe images contained in image tags (#345641).
25706 Original commit message from CVS:
25707 * docs/libs/gst-plugins-base-libs-sections.txt:
25708 * gst-libs/gst/tag/tag.h:
25709 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
25710 (gst_tag_image_type_get_type):
25711 API: add GstTagImageType enum to describe images contained
25712 in image tags (#345641).
25714 2006-06-22 10:31:22 +0000 Tim-Philipp Müller <tim@centricular.net>
25716 gst/tcp/gstmultifdsink.c: Fix warnings with gst-inspect: "buffers-min" property should be of G_TYPE_INT and not G_TYP...
25717 Original commit message from CVS:
25718 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
25719 Fix warnings with gst-inspect: "buffers-min" property
25720 should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
25721 typo in property description.
25723 2006-06-22 10:10:51 +0000 Cody Russell <bratsche@gnome.org>
25725 gst/: Avoid unnecessary class cast check in class_init functions (#337747).
25726 Original commit message from CVS:
25727 Patch by: Cody Russell <bratsche at gnome org>
25728 * gst/audioresample/gstaudioresample.c:
25729 (gst_audioresample_class_init):
25730 * gst/playback/gststreamselector.c:
25731 (gst_stream_selector_class_init):
25732 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
25733 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
25734 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
25735 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
25736 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
25737 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
25738 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
25739 * gst/videotestsrc/gstvideotestsrc.c:
25740 (gst_video_test_src_class_init):
25741 * gst/volume/gstvolume.c: (gst_volume_class_init):
25742 Avoid unnecessary class cast check in class_init
25743 functions (#337747).
25745 2006-06-21 18:39:07 +0000 Tim-Philipp Müller <tim@centricular.net>
25747 ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
25748 Original commit message from CVS:
25749 * ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
25750 (gst_text_overlay_video_chain):
25751 g_markup_escape_text() REALLY doesn't like non-UTF8 input
25752 and doesn't validate its input either (and neither did
25753 textoverlay it seems). Let's do that then and fix #345206.
25755 2006-06-19 17:12:57 +0000 Wim Taymans <wim.taymans@gmail.com>
25757 gst/tcp/gstmultifdsink.*: Added shiny new burst-on-connect methods.
25758 Original commit message from CVS:
25759 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
25760 (gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
25761 (gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
25762 (gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
25763 (find_syncframe), (find_limits), (assign_value),
25764 (count_burst_unit), (gst_multi_fd_sink_new_client),
25765 (gst_multi_fd_sink_handle_client_write),
25766 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
25767 (gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
25768 (gst_multi_fd_sink_change_state):
25769 * gst/tcp/gstmultifdsink.h:
25770 Added shiny new burst-on-connect methods.
25771 Add properties to control the minimal amount of data queued.
25773 API: bytes-min property
25774 API: time-min property
25775 API: buffers-min property
25776 API: burst-unit property
25777 API: burst-value property
25778 API: add-full signal
25779 * gst/tcp/gsttcp-marshal.list:
25780 Added new marshaller code for the new signal.
25781 * tests/check/elements/multifdsink.c: (GST_START_TEST),
25782 (multifdsink_suite):
25783 Added testcases for new burst methods.
25785 2006-06-19 11:35:47 +0000 Christian Schaller <uraeus@gnome.org>
25787 * gst-plugins-base.spec.in:
25788 update for latest changes
25789 Original commit message from CVS:
25790 update for latest changes
25792 2006-06-19 09:57:50 +0000 Edward Hervey <bilboed@bilboed.com>
25794 ext/theora/theoradec.c: Implement clipping for accurate seeking.
25795 Original commit message from CVS:
25796 * ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
25797 Implement clipping for accurate seeking.
25800 2006-06-19 09:08:05 +0000 Philip Jaegenstedt <philip@lysator.liu.se>
25802 gst/videoscale/gstvideoscale.c: Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
25803 Original commit message from CVS:
25804 Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
25805 * gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
25806 (gst_video_scale_transform):
25807 Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
25809 2006-06-17 14:18:41 +0000 Tim-Philipp Müller <tim@centricular.net>
25813 Original commit message from CVS:
25816 2006-06-17 14:13:03 +0000 Tim-Philipp Müller <tim@centricular.net>
25818 configure.ac: Fix --disable-extern (can't set conditionals conditionally, #343602).
25819 Original commit message from CVS:
25821 Fix --disable-extern (can't set conditionals conditionally,
25824 2006-06-16 15:43:23 +0000 Tim-Philipp Müller <tim@centricular.net>
25826 tests/check/elements/audioresample.c: Add test case for bug #342789 fixed below.
25827 Original commit message from CVS:
25828 * tests/check/elements/audioresample.c: (test_reuse),
25829 (audioresample_suite):
25830 Add test case for bug #342789 fixed below.
25832 2006-06-16 15:17:44 +0000 Tim-Philipp Müller <tim@centricular.net>
25834 gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
25835 Original commit message from CVS:
25836 * gst/audioresample/gstaudioresample.c:
25837 (gst_audioresample_class_init), (gst_audioresample_init),
25838 (audioresample_start), (audioresample_stop),
25839 (gst_audioresample_set_property), (gst_audioresample_get_property):
25840 Implement GstBaseTransform::start and ::stop so that audioresample
25841 can clear its internal state properly and be reused insted of
25842 causing non-negotiated errors with playbin under some circumstances
25844 * tests/check/elements/audioresample.c: (setup_audioresample),
25845 (cleanup_audioresample):
25846 Need to set element state here so that ::start and ::stop are
25849 2006-06-16 13:59:29 +0000 Young-Ho Cha <ganadist@chollian.net>
25851 gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
25852 Original commit message from CVS:
25853 Patch by: Young-Ho Cha <ganadist at chollian dot net>
25854 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
25855 Parse extra data better, apparently it's right behind
25856 the normal strf header size. Fixes #343500.
25858 2006-06-16 11:04:21 +0000 Wim Taymans <wim.taymans@gmail.com>
25860 ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
25861 Original commit message from CVS:
25862 * ext/alsa/gstalsasink.c: (set_hwparams):
25863 If we fail to set the buffer_time and period_time alsa
25864 parameters, post a warning and leave alsa select a
25865 default instead of failing. Fixes #342085
25867 2006-06-16 10:30:25 +0000 Tim-Philipp Müller <tim@centricular.net>
25870 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
25871 Original commit message from CVS:
25872 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
25874 2006-06-16 10:20:10 +0000 Tim-Philipp Müller <tim@centricular.net>
25876 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.
25877 Original commit message from CVS:
25878 * docs/libs/gst-plugins-base-libs-sections.txt:
25879 * gst-libs/gst/cdda/gstcddabasesrc.h:
25880 Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
25881 out in the header file and shouldn't be listed in the docs.
25882 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
25883 Fix it so that it doesn't crash in the debug statement.
25885 2006-06-16 10:02:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25887 docs/libs/: add remaining symbols into correct setions
25888 Original commit message from CVS:
25889 * docs/libs/Makefile.am:
25890 * docs/libs/gst-plugins-base-libs-docs.sgml:
25891 * docs/libs/gst-plugins-base-libs-sections.txt:
25892 * docs/libs/gst-plugins-base-libs.types:
25893 add remaining symbols into correct setions
25894 * gst-libs/gst/audio/gstringbuffer.c:
25895 fix incomplete docs
25896 * gst-libs/gst/audio/gstringbuffer.h:
25897 comment out not yet implemented function
25898 * gst-libs/gst/floatcast/floatcast.h:
25899 * gst-libs/gst/netbuffer/gstnetbuffer.c:
25900 add short descriptions
25901 * gst-libs/gst/interfaces/propertyprobe.c:
25902 fix return value docs
25903 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
25904 simplify debug logging
25905 * gst-libs/gst/riff/riff-read.h:
25906 sync function prototype and docs
25907 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
25908 remove left over symbol
25910 2006-06-16 09:56:41 +0000 Tim-Philipp Müller <tim@centricular.net>
25912 Use GST_PLUGIN_DOCS macro in configure.ac, add
25913 Original commit message from CVS:
25916 * docs/Makefile.am:
25917 Use GST_PLUGIN_DOCS macro in configure.ac, add
25918 --enable-plugin-docs default to autogen.sh and use
25919 ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
25921 2006-06-15 15:27:49 +0000 Wim Taymans <wim.taymans@gmail.com>
25923 ext/ogg/gstoggdemux.c: Combine GstFlowReturn from the source pads to give a meaningfull result to the upstream peer o...
25924 Original commit message from CVS:
25925 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
25926 (gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
25927 (gst_ogg_demux_loop):
25928 Combine GstFlowReturn from the source pads to give a
25929 meaningfull result to the upstream peer or to stop the
25930 processing task in case of errors.
25932 2006-06-14 14:49:33 +0000 Tim-Philipp Müller <tim@centricular.net>
25934 gst/playback/gststreaminfo.c: Try GST_TAG_CODEC as fallback when extracting the codec name; more debug info.
25935 Original commit message from CVS:
25936 * gst/playback/gststreaminfo.c: (cb_probe):
25937 Try GST_TAG_CODEC as fallback when extracting the
25938 codec name; more debug info.
25940 2006-06-14 14:34:28 +0000 Tim-Philipp Müller <tim@centricular.net>
25942 ext/ogg/: Extract language tags from ogm subtitle streams, so that the subtitle menu choices are labelled correctly in
25943 Original commit message from CVS:
25944 * ext/ogg/Makefile.am:
25945 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
25946 Extract language tags from ogm subtitle streams, so that
25947 the subtitle menu choices are labelled correctly in
25948 Totem (fixes #344708).
25950 2006-06-14 09:13:54 +0000 Alessandro Decina <alessandro@nnva.org>
25952 ext/ogg/gstoggmux.c: Fix various leaks. Fixes #343699.
25953 Original commit message from CVS:
25954 Patch by: Alessandro Decina <alessandro at nnva dot org>
25955 * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
25956 (gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
25957 (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
25958 (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
25959 Fix various leaks. Fixes #343699.
25960 Add x-smoke mime type.
25962 2006-06-14 08:17:45 +0000 Tim-Philipp Müller <tim@centricular.net>
25964 gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
25965 Original commit message from CVS:
25966 * gst-libs/gst/riff/riff-ids.h:
25967 Add IDs for 'bext' chunks (see #343837).
25969 2006-06-12 12:44:38 +0000 Young-Ho Cha <ganadist@chollian.net>
25971 gst/subparse/samiparse.c: Honour font face tags in SAMI subtitles (#344503).
25972 Original commit message from CVS:
25973 Patch by: Young-Ho Cha <ganadist at chollian net>
25974 * gst/subparse/samiparse.c: (sami_context_pop_state),
25975 (handle_start_font), (end_sami_element):
25976 Honour font face tags in SAMI subtitles (#344503).
25978 2006-06-11 20:41:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25980 po/POTFILES.in: add missing files containing translatable strings
25981 Original commit message from CVS:
25983 add missing files containing translatable strings
25985 2006-06-11 19:55:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25987 docs/libs/tmpl/.cvsignore: we don't want those *.sgml files in CVS either
25988 Original commit message from CVS:
25989 * docs/libs/tmpl/.cvsignore:
25990 we don't want those *.sgml files in CVS either
25992 2006-06-11 19:44:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25995 Original commit message from CVS:
25996 * docs/libs/.cvsignore:
25997 * tests/check/elements/.cvsignore:
25998 * tests/check/libs/.cvsignore:
26001 2006-06-11 18:33:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26003 docs/libs/Makefile.am: also commiting the changed Makefile.am (added more libs to the doc-build)
26004 Original commit message from CVS:
26005 * docs/libs/Makefile.am:
26006 also commiting the changed Makefile.am (added more libs to the
26009 2006-06-11 17:08:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26011 docs/libs/: first batch of reordering things, add index & hierarchy
26012 Original commit message from CVS:
26013 * docs/libs/gst-plugins-base-libs-docs.sgml:
26014 * docs/libs/gst-plugins-base-libs-sections.txt:
26015 * docs/libs/gst-plugins-base-libs.types:
26016 first batch of reordering things, add index & hierarchy
26018 2006-06-11 14:08:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26021 * ext/alsa/Makefile.am:
26022 * ext/cdparanoia/Makefile.am:
26023 * ext/gnomevfs/Makefile.am:
26024 * ext/libvisual/Makefile.am:
26025 * ext/ogg/Makefile.am:
26026 * ext/pango/Makefile.am:
26027 * ext/theora/Makefile.am:
26028 * ext/vorbis/Makefile.am:
26029 * sys/v4l/Makefile.am:
26030 * sys/ximage/Makefile.am:
26031 * sys/xvimage/Makefile.am:
26032 further clean up build
26033 Original commit message from CVS:
26034 further clean up build
26036 2006-06-11 12:14:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26038 configure.ac: use GST_PKG_CHECK_MODULES, cleans up output
26039 Original commit message from CVS:
26041 use GST_PKG_CHECK_MODULES, cleans up output
26043 2006-06-11 12:10:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26046 * win32/common/config.h:
26048 Original commit message from CVS:
26051 2006-06-10 18:52:03 +0000 Tim-Philipp Müller <tim@centricular.net>
26053 ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
26054 Original commit message from CVS:
26055 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
26056 Add support for burn:// URIs (#343385); const-ify things a bit,
26057 use G_N_ELEMENTS instead of hard-coded array size.
26059 2006-06-10 18:25:07 +0000 Young-Ho Cha <ganadist@chollian.net>
26061 gst/subparse/samiparse.c: Fix up broken entities before passing them to libxml *sigh*. (#343303).
26062 Original commit message from CVS:
26063 Patch by: Young-Ho Cha <ganadist at chollian net>
26064 * gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
26065 Fix up broken entities before passing them to libxml *sigh*.
26068 2006-06-09 18:52:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26073 Original commit message from CVS:
26076 === release 0.10.8 ===
26078 2006-06-09 18:49:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26084 * docs/plugins/gst-plugins-base-plugins.args:
26085 * docs/plugins/inspect/plugin-adder.xml:
26086 * docs/plugins/inspect/plugin-alsa.xml:
26087 * docs/plugins/inspect/plugin-audioconvert.xml:
26088 * docs/plugins/inspect/plugin-audiorate.xml:
26089 * docs/plugins/inspect/plugin-audioresample.xml:
26090 * docs/plugins/inspect/plugin-audiotestsrc.xml:
26091 * docs/plugins/inspect/plugin-cdparanoia.xml:
26092 * docs/plugins/inspect/plugin-decodebin.xml:
26093 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
26094 * docs/plugins/inspect/plugin-gnomevfs.xml:
26095 * docs/plugins/inspect/plugin-libvisual.xml:
26096 * docs/plugins/inspect/plugin-ogg.xml:
26097 * docs/plugins/inspect/plugin-pango.xml:
26098 * docs/plugins/inspect/plugin-playbin.xml:
26099 * docs/plugins/inspect/plugin-subparse.xml:
26100 * docs/plugins/inspect/plugin-tcp.xml:
26101 * docs/plugins/inspect/plugin-theora.xml:
26102 * docs/plugins/inspect/plugin-typefindfunctions.xml:
26103 * docs/plugins/inspect/plugin-video4linux.xml:
26104 * docs/plugins/inspect/plugin-videorate.xml:
26105 * docs/plugins/inspect/plugin-videoscale.xml:
26106 * docs/plugins/inspect/plugin-videotestsrc.xml:
26107 * docs/plugins/inspect/plugin-volume.xml:
26108 * docs/plugins/inspect/plugin-vorbis.xml:
26109 * docs/plugins/inspect/plugin-ximagesink.xml:
26110 * docs/plugins/inspect/plugin-xvimagesink.xml:
26111 * win32/common/config.h:
26113 Original commit message from CVS:
26116 2006-06-07 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26118 0.10.7.2 prerelease
26119 Original commit message from CVS:
26135 * win32/common/config.h:
26136 0.10.7.2 prerelease
26138 2006-06-07 11:03:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26140 move last template doc snippets to source code and delete them
26141 Original commit message from CVS:
26142 * docs/libs/tmpl/gstaudio.sgml:
26143 * docs/libs/tmpl/gstcolorbalance.sgml:
26144 * docs/libs/tmpl/gstmixer.sgml:
26145 * docs/libs/tmpl/gstringbuffer.sgml:
26146 * docs/libs/tmpl/gsttuner.sgml:
26147 * docs/libs/tmpl/gstxoverlay.sgml:
26148 * gst-libs/gst/audio/audio.c:
26149 * gst-libs/gst/audio/gstringbuffer.c:
26150 * gst-libs/gst/interfaces/colorbalance.c:
26151 * gst-libs/gst/interfaces/mixer.c:
26152 * gst-libs/gst/interfaces/tuner.c:
26153 * gst-libs/gst/interfaces/xoverlay.c:
26154 move last template doc snippets to source code and delete them
26156 2006-06-06 16:26:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26158 * gst/gdp/gstgdppay.c:
26160 Original commit message from CVS:
26163 2006-06-06 14:39:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26165 configure.ac: enable building of GDP elements
26166 Original commit message from CVS:
26168 enable building of GDP elements
26169 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
26170 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
26171 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
26172 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
26173 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
26174 (gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
26175 (gst_gdp_pay_change_state):
26176 * gst/gdp/gstgdppay.h:
26179 2006-06-06 11:13:18 +0000 Michael Smith <msmith@xiph.org>
26181 ext/theora/theoraparse.c: Mark DELTA_UNIT on non-keyframes.
26182 Original commit message from CVS:
26183 * ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
26184 (theora_parse_drain_queue):
26185 Mark DELTA_UNIT on non-keyframes.
26187 2006-06-03 21:06:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26189 gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
26190 Original commit message from CVS:
26191 * gst-libs/gst/audio/gstbaseaudiosink.c:
26192 (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
26193 * gst-libs/gst/audio/gstbaseaudiosink.h:
26194 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
26195 (gst_ring_buffer_samples_done):
26196 * gst-libs/gst/audio/gstringbuffer.h:
26197 Document better the fact that latency_time and buffer_time are values
26198 stored in microseconds, and not the usual GStreamer nanoseconds.
26199 Change the variables (compatibly) that store them from GstClockTime
26200 to guint64 to make it more clear that they're not storing clock times.
26201 Also, remove the bogus property description that says the user can
26202 specify -1 to get the default value, since that's never been the case.
26203 When computing the default segment size for the ring buffer, make it
26204 an integer number of samples.
26205 When the sub-class indicates a delay greater than the number of
26206 samples we've written return 0 from the audio sink get_time method.
26208 2006-06-02 17:01:02 +0000 Michael Smith <msmith@xiph.org>
26210 tests/check/: Don't busy-wait in tests; this was causing test timeouts very frequently when running under valgrind.
26211 Original commit message from CVS:
26212 * tests/check/elements/audioconvert.c: (set_channel_positions),
26213 (get_float_mc_caps), (get_int_mc_caps):
26214 * tests/check/elements/audioresample.c:
26215 * tests/check/elements/audiotestsrc.c: (GST_START_TEST):
26216 * tests/check/elements/videorate.c:
26217 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
26218 * tests/check/elements/volume.c:
26219 * tests/check/elements/vorbisdec.c:
26220 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
26221 Don't busy-wait in tests; this was causing test timeouts very
26222 frequently when running under valgrind.
26224 2006-06-02 16:45:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26226 * gst/gdp/gstgdpdepay.c:
26227 * gst/gdp/gstgdppay.h:
26229 Original commit message from CVS:
26232 2006-06-02 16:35:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26234 * tests/check/elements/multifdsink.c:
26235 fail_if_can_read is racy
26236 Original commit message from CVS:
26237 fail_if_can_read is racy
26239 2006-06-02 16:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26241 gst/tcp/: make multifdsink properly deal with streamheader:
26242 Original commit message from CVS:
26244 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
26245 (gst_multi_fd_sink_remove_client_link),
26246 (gst_multi_fd_sink_client_queue_caps),
26247 (gst_multi_fd_sink_client_queue_buffer),
26248 (gst_multi_fd_sink_handle_client_write),
26249 (gst_multi_fd_sink_render):
26250 * gst/tcp/gstmultifdsink.h:
26251 make multifdsink properly deal with streamheader:
26252 - streamheader is taken from caps
26253 - buffers marked with IN_CAPS are not sent
26254 - streamheaders are sent, on connection, from the caps of the
26255 buffer where the client gets positioned to
26256 - further streamheader changes are done every time the client
26257 will receive a buffer with different caps
26258 * tests/check/elements/multifdsink.c: (GST_START_TEST),
26259 (gst_multifdsink_create_streamheader):
26262 2006-06-02 15:06:59 +0000 Michael Smith <msmith@xiph.org>
26264 ext/vorbis/vorbisdec.c: Reinstate limit on channel count. Vorbis does not define the meaning of > 6 channels, so they...
26265 Original commit message from CVS:
26266 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
26267 Reinstate limit on channel count. Vorbis does not define the meaning
26268 of > 6 channels, so they're just independent channels. Gstreamer
26269 currently has no mechanism to represent N independent channels.
26271 2006-06-02 14:23:34 +0000 Michael Smith <msmith@xiph.org>
26273 ext/vorbis/vorbisdec.c: Don't arbitrarily restrict channel counts and rate in vorbis.
26274 Original commit message from CVS:
26275 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
26276 Don't arbitrarily restrict channel counts and rate in vorbis.
26277 In terms of effects likely on real-world files, this fixes 96kHz
26278 playback of vorbis.
26280 2006-06-02 14:19:18 +0000 Michael Smith <msmith@xiph.org>
26282 gst/audioconvert/audioconvert.c: More correct float->int conversion.
26283 Original commit message from CVS:
26284 * gst/audioconvert/audioconvert.c: (float):
26285 More correct float->int conversion.
26287 2006-06-02 14:07:42 +0000 Michael Smith <msmith@xiph.org>
26289 ext/ogg/gstoggdemux.c: Don't accidently send GST_CLOCK_TIME_NONE as a new segment start value. Fixes g-critical on tr...
26290 Original commit message from CVS:
26291 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
26292 Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
26293 value. Fixes g-critical on trying to play back ogg containing
26296 2006-06-02 10:34:12 +0000 Wim Taymans <wim.taymans@gmail.com>
26298 gst/playback/gstplaybasebin.*: Make the subtitle detection work from any thread so we don't deadlock. Fixes #343397.
26299 Original commit message from CVS:
26300 * gst/playback/gstplaybasebin.c: (group_create), (group_commit),
26302 * gst/playback/gstplaybasebin.h:
26303 Make the subtitle detection work from any thread so we don't
26304 deadlock. Fixes #343397.
26306 2006-06-02 10:28:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26308 gst/gdp/gstgdppay.c: add crc-header and crc-payload properties don't error out on some things that are recoverable
26309 Original commit message from CVS:
26310 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
26311 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
26312 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
26313 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
26314 (gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
26315 (gst_gdp_pay_get_property):
26316 add crc-header and crc-payload properties
26317 don't error out on some things that are recoverable
26318 * tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
26321 2006-06-02 09:17:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26323 * gst/tcp/gsttcp.c:
26324 show type number when packet is of the wrong type
26325 Original commit message from CVS:
26326 show type number when packet is of the wrong type
26328 2006-06-01 23:04:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26330 gst/volume/Makefile.am: Seriously, it's not *that* hard to get compilation right. Even a drunk can do it ! Add LIBOI...
26331 Original commit message from CVS:
26332 * gst/volume/Makefile.am:
26333 Seriously, it's not *that* hard to get compilation right. Even
26334 a drunk can do it ! Add LIBOIL CFLAGS and LIBS
26336 2006-06-01 22:00:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26338 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26339 Original commit message from CVS:
26340 * ext/alsaspdif/alsaspdifsink.h:
26341 * ext/amrwb/gstamrwbdec.h:
26342 * ext/amrwb/gstamrwbenc.h:
26343 * ext/amrwb/gstamrwbparse.h:
26344 * ext/arts/gst_arts.h:
26345 * ext/artsd/gstartsdsink.h:
26346 * ext/audiofile/gstafparse.h:
26347 * ext/audiofile/gstafsink.h:
26348 * ext/audiofile/gstafsrc.h:
26349 * ext/audioresample/gstaudioresample.h:
26350 * ext/bz2/gstbz2dec.h:
26351 * ext/bz2/gstbz2enc.h:
26352 * ext/dirac/gstdiracdec.h:
26353 * ext/directfb/dfbvideosink.h:
26354 * ext/divx/gstdivxdec.h:
26355 * ext/divx/gstdivxenc.h:
26356 * ext/dts/gstdtsdec.h:
26357 * ext/faac/gstfaac.h:
26358 * ext/gsm/gstgsmdec.h:
26359 * ext/gsm/gstgsmenc.h:
26360 * ext/ivorbis/vorbisenc.h:
26361 * ext/libfame/gstlibfame.h:
26362 * ext/nas/nassink.h:
26363 * ext/neon/gstneonhttpsrc.h:
26364 * ext/polyp/polypsink.h:
26365 * ext/sdl/sdlaudiosink.h:
26366 * ext/sdl/sdlvideosink.h:
26367 * ext/shout/gstshout.h:
26368 * ext/snapshot/gstsnapshot.h:
26369 * ext/sndfile/gstsf.h:
26370 * ext/swfdec/gstswfdec.h:
26371 * ext/tarkin/gsttarkindec.h:
26372 * ext/tarkin/gsttarkinenc.h:
26373 * ext/theora/theoradec.h:
26374 * ext/wavpack/gstwavpackdec.h:
26375 * ext/wavpack/gstwavpackparse.h:
26376 * ext/xine/gstxine.h:
26377 * ext/xvid/gstxviddec.h:
26378 * ext/xvid/gstxvidenc.h:
26379 * gst/cdxaparse/gstcdxaparse.h:
26380 * gst/cdxaparse/gstcdxastrip.h:
26381 * gst/colorspace/gstcolorspace.h:
26382 * gst/festival/gstfestival.h:
26383 * gst/freeze/gstfreeze.h:
26384 * gst/gdp/gstgdpdepay.h:
26385 * gst/gdp/gstgdppay.h:
26386 * gst/modplug/gstmodplug.h:
26387 * gst/mpeg1sys/gstmpeg1systemencode.h:
26388 * gst/mpeg1videoparse/gstmp1videoparse.h:
26389 * gst/mpeg2sub/gstmpeg2subt.h:
26390 * gst/mpegaudioparse/gstmpegaudioparse.h:
26391 * gst/multifilesink/gstmultifilesink.h:
26392 * gst/overlay/gstoverlay.h:
26393 * gst/playondemand/gstplayondemand.h:
26394 * gst/qtdemux/qtdemux.h:
26395 * gst/rtjpeg/gstrtjpegdec.h:
26396 * gst/rtjpeg/gstrtjpegenc.h:
26397 * gst/smooth/gstsmooth.h:
26398 * gst/smoothwave/gstsmoothwave.h:
26399 * gst/spectrum/gstspectrum.h:
26400 * gst/speed/gstspeed.h:
26401 * gst/stereo/gststereo.h:
26402 * gst/switch/gstswitch.h:
26403 * gst/tta/gstttadec.h:
26404 * gst/tta/gstttaparse.h:
26405 * gst/videodrop/gstvideodrop.h:
26406 * gst/xingheader/gstxingmux.h:
26407 * sys/directdraw/gstdirectdrawsink.h:
26408 * sys/directsound/gstdirectsoundsink.h:
26409 * sys/dxr3/dxr3audiosink.h:
26410 * sys/dxr3/dxr3spusink.h:
26411 * sys/dxr3/dxr3videosink.h:
26412 * sys/qcam/gstqcamsrc.h:
26413 * sys/vcd/vcdsrc.h:
26414 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26416 2006-06-01 20:39:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26418 gst/volume/gstvolume.*: rewrite the passthrough check, split _int16 and _int16_clamp, fix another property desc., rem...
26419 Original commit message from CVS:
26420 * gst/volume/gstvolume.c: (volume_choose_func),
26421 (volume_update_real_volume), (gst_volume_class_init),
26422 (gst_volume_init), (volume_process_float), (volume_process_int16),
26423 (volume_process_int16_clamp), (volume_set_caps),
26424 (volume_transform_ip), (plugin_init):
26425 * gst/volume/gstvolume.h:
26426 rewrite the passthrough check, split _int16 and _int16_clamp, fix
26427 another property desc., remove unused param from process function
26428 * tests/check/elements/volume.c: (volume_suite):
26429 reactivate the passthrough test
26431 2006-06-01 19:19:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26433 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26434 Original commit message from CVS:
26435 * ext/alsa/gstalsamixerelement.h:
26436 * ext/alsa/gstalsamixeroptions.h:
26437 * ext/alsa/gstalsamixertrack.h:
26438 * ext/gnomevfs/gstgnomevfssink.h:
26439 * ext/gnomevfs/gstgnomevfssrc.h:
26440 * ext/theora/gsttheoradec.h:
26441 * ext/theora/gsttheoraenc.h:
26442 * ext/theora/gsttheoraparse.h:
26443 * ext/vorbis/vorbisparse.h:
26444 * gst-libs/gst/audio/gstaudioclock.h:
26445 * gst-libs/gst/audio/gstaudiofilter.h:
26446 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
26447 * gst/audioconvert/gstaudioconvert.h:
26448 * gst/audioresample/gstaudioresample.h:
26449 * gst/audiotestsrc/gstaudiotestsrc.h:
26450 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
26451 * gst/playback/gststreamselector.h:
26452 * gst/tcp/gstmultifdsink.h:
26453 * gst/tcp/gsttcpclientsink.h:
26454 * gst/tcp/gsttcpclientsrc.h:
26455 * gst/tcp/gsttcpserversink.h:
26456 * gst/tcp/gsttcpserversrc.h:
26457 * gst/videorate/gstvideorate.h:
26458 * gst/videoscale/gstvideoscale.h:
26459 * gst/videotestsrc/gstvideotestsrc.h:
26460 * gst/volume/gstvolume.h:
26461 * sys/v4l/gstv4ljpegsrc.h:
26462 * sys/v4l/gstv4lmjpegsink.h:
26463 * sys/v4l/gstv4lmjpegsrc.h:
26464 * sys/v4l/gstv4lsrc.h:
26465 * sys/ximage/ximagesink.h:
26466 * sys/xvimage/xvimagesink.h:
26467 * tests/old/testsuite/alsa/sinesrc.h:
26468 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26470 2006-05-31 16:56:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26472 * tests/check/elements/multifdsink.c:
26473 remove wrong commit
26474 Original commit message from CVS:
26475 remove wrong commit
26477 2006-05-31 16:21:48 +0000 Wim Taymans <wim.taymans@gmail.com>
26479 ext/libvisual/visual.c: Handle DISCONT.
26480 Original commit message from CVS:
26481 * ext/libvisual/visual.c: (gst_visual_reset),
26482 (gst_visual_sink_setcaps), (gst_visual_sink_event),
26483 (gst_visual_src_event), (get_buffer), (gst_visual_chain):
26485 Use running time before doing QoS.
26488 2006-05-31 14:17:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26490 docs/libs/Makefile.am: set a magic variable to indicate we know the docs are incomplete
26491 Original commit message from CVS:
26492 * docs/libs/Makefile.am:
26493 set a magic variable to indicate we know the docs are incomplete
26495 2006-05-30 20:33:59 +0000 Sébastien Moutte <sebastien@moutte.net>
26497 win32/common/libgstvideo.def: export gst_video_calculate_display_ratio
26498 Original commit message from CVS:
26499 * win32/common/libgstvideo.def:
26500 export gst_video_calculate_display_ratio
26501 * win32/vs6/libgstvideoscale.dsp:
26502 add link to libgstvideo-0.10.lib
26504 2006-05-30 19:00:39 +0000 Tim-Philipp Müller <tim@centricular.net>
26506 gst/playback/gstplaybasebin.c: Throw a more comprehensible error for rtsp:// URIs (rather than erroring out with a ne...
26507 Original commit message from CVS:
26508 * gst/playback/gstplaybasebin.c: (gen_source_element):
26509 Throw a more comprehensible error for rtsp:// URIs (rather
26510 than erroring out with a negotiation error later on) until
26511 we fix playbin to handle rtspsrc etc.
26513 2006-05-30 16:09:36 +0000 Wim Taymans <wim.taymans@gmail.com>
26515 ext/pango/gsttextoverlay.c: Added some FIXMEs.
26516 Original commit message from CVS:
26517 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
26518 (gst_text_overlay_text_event):
26521 2006-05-30 16:07:50 +0000 Wim Taymans <wim.taymans@gmail.com>
26523 gst/adder/gstadder.*: Implement release_request_pad.
26524 Original commit message from CVS:
26525 * gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
26526 (gst_adder_request_new_pad), (gst_adder_release_pad):
26527 * gst/adder/gstadder.h:
26528 Implement release_request_pad.
26529 Make padcounter atomic.
26530 * tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
26531 Added check for release_pad in adder.
26533 2006-05-30 16:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
26535 ext/ogg/gstoggdemux.c: Fix build again.
26536 Original commit message from CVS:
26537 * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
26540 2006-05-30 14:59:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26542 ext/ogg/gstoggdemux.c: add more debugging clean up printf formats for granulepos and serialno
26543 Original commit message from CVS:
26544 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
26545 (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
26546 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
26547 (gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
26548 (gst_ogg_demux_seek), (gst_ogg_demux_get_data),
26549 (gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
26550 (gst_ogg_demux_bisect_forward_serialno),
26551 (gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
26552 (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
26554 clean up printf formats for granulepos and serialno
26556 2006-05-30 14:31:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26558 * tests/check/elements/multifdsink.c:
26559 * tests/check/generic/states.c:
26560 properly fail if we can't make an element
26561 Original commit message from CVS:
26562 properly fail if we can't make an element
26564 2006-05-30 13:22:58 +0000 Michael Smith <msmith@xiph.org>
26566 ext/vorbis/vorbisenc.*: Multi-channel caps negotiation, so we can do proper multichannel vorbis encoding, negotiated ...
26567 Original commit message from CVS:
26568 * ext/vorbis/vorbisenc.c: (raw_caps_factory),
26569 (gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
26570 (gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
26571 (gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
26572 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
26573 * ext/vorbis/vorbisenc.h:
26574 Multi-channel caps negotiation, so we can do proper multichannel
26575 vorbis encoding, negotiated through audioconvert.
26577 2006-05-30 11:45:52 +0000 Wim Taymans <wim.taymans@gmail.com>
26579 tests/check/elements/adder.c: Added check to show that #339935 is fixed with ongoing adder and collectpads fixes.
26580 Original commit message from CVS:
26581 * tests/check/elements/adder.c: (test_event_message_received),
26582 (test_play_twice_message_received), (GST_START_TEST),
26584 Added check to show that #339935 is fixed with ongoing
26585 adder and collectpads fixes.
26587 2006-05-29 17:19:48 +0000 Wim Taymans <wim.taymans@gmail.com>
26589 gst/adder/gstadder.c: Don't leak pad name.
26590 Original commit message from CVS:
26591 * gst/adder/gstadder.c: (gst_adder_request_new_pad):
26592 Don't leak pad name.
26594 2006-05-29 15:49:53 +0000 Wim Taymans <wim.taymans@gmail.com>
26596 gst/adder/gstadder.c: Fix adder seeking.
26597 Original commit message from CVS:
26598 * gst/adder/gstadder.c: (gst_adder_query_duration),
26599 (forward_event_func), (forward_event), (gst_adder_src_event):
26601 Make query/seeking code threadsafe.
26602 * tests/check/Makefile.am:
26603 * tests/check/elements/adder.c: (test_event_message_received),
26604 (GST_START_TEST), (test_play_twice_message_received):
26605 Fix adder test case.
26607 2006-05-29 13:21:00 +0000 Young-Ho Cha <ganadist@chollian.net>
26609 gst/playback/gstplaybasebin.*: Add 'subtitle-encoding' property to playbin, so applications can force a subtitle enco...
26610 Original commit message from CVS:
26611 Patch by: Young-Ho Cha <ganadist at chollian net>
26612 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
26613 (gst_play_base_bin_init), (gst_play_base_bin_dispose),
26614 (set_encoding_element), (decodebin_element_added_cb),
26615 (decodebin_element_removed_cb), (setup_subtitle), (setup_source),
26616 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
26617 * gst/playback/gstplaybasebin.h:
26618 Add 'subtitle-encoding' property to playbin, so applications can
26619 force a subtitle encoding for non-UTF8 subtitles (#342268).
26620 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
26621 (gst_sub_parse_set_property):
26622 Rename recently-added 'encoding' property to 'subtitle-encoding'
26623 (so it can be proxied by playbin/decodebin in a generic way
26624 with less danger of false positives).
26626 2006-05-29 11:04:48 +0000 Michael Smith <msmith@xiph.org>
26628 gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf...
26629 Original commit message from CVS:
26630 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
26631 (append_with_other_format), (set_structure_widths),
26632 (gst_audio_convert_transform_caps):
26633 Patch from #341562: give more specific audio caps in get_caps, so
26634 that basetransform can make better decisions on what caps to
26637 2006-05-28 20:04:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26639 tests/check/elements/volume.c: make it compile again
26640 Original commit message from CVS:
26641 * tests/check/elements/volume.c:
26642 make it compile again
26644 2006-05-28 19:56:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26646 tests/check/elements/volume.c: disable test until #343196 gets resolved
26647 Original commit message from CVS:
26648 * tests/check/elements/volume.c: (volume_suite):
26649 disable test until #343196 gets resolved
26651 2006-05-28 19:42:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26653 gst/adder/gstadder.c: Make it easier to copy&paste
26654 Original commit message from CVS:
26655 * gst/adder/gstadder.c: (gst_adder_get_type):
26656 Make it easier to copy&paste
26657 * gst/volume/Makefile.am:
26658 * gst/volume/gstvolume.c: (volume_update_real_volume),
26659 (gst_volume_set_volume), (gst_volume_set_mute),
26660 (gst_volume_class_init), (volume_process_int16), (volume_set_caps),
26661 (volume_transform_ip), (volume_update_mute),
26662 (volume_update_volume):
26663 * gst/volume/gstvolume.h:
26664 Add own debug category, move duplicate code to helper function, fix
26665 property texts, add more comments and prepare ffor liboil-goodness
26666 * tests/check/Makefile.am:
26667 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
26668 add test for mute and passtrough case, be a bit more verbose to track
26670 * tests/check/generic/states.c: (GST_START_TEST):
26671 catch elements that fail to instantiate
26673 2006-05-28 09:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
26675 tests/check/pipelines/: Comment out tests using parse_launch() if core was built without parsing capabilities.
26676 Original commit message from CVS:
26677 * tests/check/pipelines/simple-launch-lines.c:
26678 * tests/check/pipelines/theoraenc.c:
26679 * tests/check/pipelines/vorbisenc.c:
26680 Comment out tests using parse_launch() if core was built without
26681 parsing capabilities.
26683 2006-05-27 13:34:03 +0000 Edward Hervey <bilboed@bilboed.com>
26685 tests/check/Makefile.am: Extra bonus points for whoever explains to ensonic that you are meant to test unit tests tho...
26686 Original commit message from CVS:
26687 * tests/check/Makefile.am:
26688 Extra bonus points for whoever explains to ensonic that you are meant
26689 to test unit tests thoroughly before commiting them, especially if
26690 you know it's going to break.
26691 De-activated element/adder tests.
26693 2006-05-27 13:09:16 +0000 Edward Hervey <bilboed@bilboed.com>
26695 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Marking caps conversion issues as GST_WARNING is way too verbose,
26696 Original commit message from CVS:
26697 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
26698 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
26699 Marking caps conversion issues as GST_WARNING is way too verbose,
26700 Moving them to GST_LOG.
26702 2006-05-27 11:26:18 +0000 Tim-Philipp Müller <tim@centricular.net>
26704 README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from...
26705 Original commit message from CVS:
26707 Replace current README (containing the release notes from
26708 some 0.9.x version) with a proper README taken from the core.
26710 2006-05-26 15:52:23 +0000 Wim Taymans <wim.taymans@gmail.com>
26712 ext/vorbis/vorbisdec.c: Small cleanups.
26713 Original commit message from CVS:
26714 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
26715 (vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
26716 (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
26717 (vorbis_dec_change_state):
26720 Clip output samples to segment boundaries.
26722 2006-05-26 11:17:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26724 sys/ximage/ximagesink.c: Improve the errors produced on bad output, including some human readable description strings.
26725 Original commit message from CVS:
26726 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
26727 (gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
26728 Improve the errors produced on bad output, including some human
26729 readable description strings.
26730 Handle the (theoretical for ximagesink) case where the XServer
26731 has a different idea about the size required for a particular
26732 frame and gives us too small a memory allocation.
26734 2006-05-26 10:18:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26737 Mention bugs fixed by previous commit
26738 Original commit message from CVS:
26739 Mention bugs fixed by previous commit
26741 2006-05-26 09:40:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26743 sys/xvimage/xvimagesink.c: Improve the errors produced on bad output, including some human readable description strings.
26744 Original commit message from CVS:
26745 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
26746 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
26747 (gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
26748 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
26749 Improve the errors produced on bad output, including some human
26750 readable description strings.
26751 Handle RGB Xv formats properly by transforming them into our
26752 big-endian caps description.
26753 Use gst_caps_truncate to ensure that we never try and choose a
26754 non-fixed caps in buffer_alloc.
26755 Handle the case where the XServer has a different idea about the size
26756 required for a particular frame and gives us too small a memory
26758 Use -1 to indicate 'no image format', because 0 is a valid XServer
26759 image format number.
26760 Put RGB Xv formats at the end of the caps, so that we always prefer
26762 Iterate the available Xv Encodings to determine the maximum width and
26763 height, and then return that in our caps.
26765 2006-05-25 16:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26767 gst/playback/gstdecodebin.c: When there is only one unfinished pad and it receives an event that doesn't match our re...
26768 Original commit message from CVS:
26769 * gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
26770 When there is only one unfinished pad and it receives an event that
26771 doesn't match our requirements, we need to set alldone=FALSE so that
26772 the fakesink is not removed yet.
26774 2006-05-25 09:32:31 +0000 Tim-Philipp Müller <tim@centricular.net>
26776 ext/ogg/gstoggdemux.c: Use gst_type_find_helper_for_buffer() to find the type of stream from the first packet.
26777 Original commit message from CVS:
26778 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
26779 Use gst_type_find_helper_for_buffer() to find the type
26780 of stream from the first packet.
26782 Bump requirements to core CVS (needed for vorbis
26783 typefinding to work).
26785 2006-05-24 08:34:53 +0000 Edward Hervey <bilboed@bilboed.com>
26787 gst/typefind/gsttypefindfunctions.c: Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
26788 Original commit message from CVS:
26789 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
26790 Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
26791 Else they play perfectly fine with qtdemux.
26793 2006-05-23 20:38:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26795 make more debug catagories static
26796 Original commit message from CVS:
26797 * ext/theora/theoradec.c:
26798 * ext/theora/theoraenc.c:
26799 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
26800 * gst/audiorate/gstaudiorate.c:
26801 make more debug catagories static
26802 * tests/check/Makefile.am:
26803 * tests/check/elements/adder.c: (message_received),
26804 (test_event_message_received), (GST_START_TEST),
26805 (test_play_twice_message_received), (adder_suite):
26806 added test case for using element twice, extra bonus points for anyone
26807 who can make these test run reliably
26809 2006-05-23 15:18:40 +0000 Tim-Philipp Müller <tim@centricular.net>
26811 ext/theora/theoradec.c: Make work with time-stamped input buffers that do not have a granulepos in BUFFER_OFFSET_END ...
26812 Original commit message from CVS:
26813 * ext/theora/theoradec.c: (theora_dec_chain):
26814 Make work with time-stamped input buffers that do not
26815 have a granulepos in BUFFER_OFFSET_END (like theora
26816 buffers coming from matroskademux). Fixes #342448.
26818 2006-05-22 15:53:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26820 gst/gdp/: Handle error cases when calling functions do downwards state change after parent's change_state
26821 Original commit message from CVS:
26822 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain),
26823 (gst_gdp_depay_change_state):
26824 * gst/gdp/gstgdpdepay.h:
26825 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader),
26826 (gst_gdp_pay_chain), (gst_gdp_pay_sink_event),
26827 (gst_gdp_pay_change_state):
26828 * gst/gdp/gstgdppay.h:
26829 Handle error cases when calling functions
26830 do downwards state change after parent's change_state
26831 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
26832 * tests/check/elements/gdppay.c: (GST_START_TEST):
26835 2006-05-22 13:25:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26837 adding GDP payloader and depayloader. Build integration will follow later when the GDP issues for core are sorted out.
26838 Original commit message from CVS:
26839 * gst/gdp/Makefile.am:
26840 * gst/gdp/gstgdp.c: (plugin_init):
26841 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_base_init),
26842 (gst_gdp_depay_class_init), (gst_gdp_depay_init),
26843 (gst_gdp_depay_finalize), (gst_gdp_depay_chain),
26844 (gst_gdp_depay_change_state), (gst_gdp_depay_plugin_init):
26845 * gst/gdp/gstgdpdepay.h:
26846 * gst/gdp/gstgdppay.c: (gst_gdp_pay_base_init),
26847 (gst_gdp_pay_class_init), (gst_gdp_pay_init),
26848 (gst_gdp_pay_dispose), (gst_gdp_stamp_buffer),
26849 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
26850 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
26851 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
26852 (gst_gdp_pay_sink_event), (gst_gdp_pay_change_state),
26853 (gst_gdp_pay_plugin_init):
26854 * gst/gdp/gstgdppay.h:
26855 * tests/check/Makefile.am:
26856 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
26857 (cleanup_gdpdepay), (gdpdepay_push_per_byte), (GST_START_TEST),
26858 (setup_gdpdepay_streamheader), (gdpdepay_suite), (main):
26859 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay),
26860 (GST_START_TEST), (setup_gdppay_streamheader), (gdppay_suite),
26862 adding GDP payloader and depayloader. Build integration will
26863 follow later when the GDP issues for core are sorted out.
26865 2006-05-22 11:42:03 +0000 Peter Kjellerstedt <pkj@axis.com>
26867 gst/tcp/Makefile.am: fdstresstest doesn't need Gtk+, fix compilation if gtk is not available (#342566).
26868 Original commit message from CVS:
26869 Patch by: Peter Kjellerstedt <pkj at axis com>
26870 * gst/tcp/Makefile.am:
26871 fdstresstest doesn't need Gtk+, fix compilation if
26872 gtk is not available (#342566).
26874 2006-05-19 17:57:56 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
26876 gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
26877 Original commit message from CVS:
26878 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
26880 Removed redundant floor()
26882 2006-05-19 15:00:43 +0000 Tim-Philipp Müller <tim@centricular.net>
26884 gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
26885 Original commit message from CVS:
26886 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
26887 On second thought, just skip JUNK chunks automatically, so
26888 the caller doesn't have to handle this. Fixes #342345.
26889 Also, return GST_FLOW_UNEXPECTED if we get a short read,
26890 not GST_FLOW_ERROR.
26892 2006-05-19 13:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
26894 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...
26895 Original commit message from CVS:
26896 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
26897 Don't bail out on JUNK chunks with a size of 0 (would try to
26898 pull_range 0 bytes before, which sources don't like too much).
26901 2006-05-19 13:02:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26903 Use the gstutil scaling function to preserve 64 bits while calculating output width and height from the display-aspec...
26904 Original commit message from CVS:
26905 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
26906 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
26907 Use the gstutil scaling function to preserve 64 bits while calculating
26908 output width and height from the display-aspect-ratio. (A continuation
26911 2006-05-19 11:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26913 sys/xvimage/xvimagesink.*: When performing buffer allocations, remember the caps and image format we return so that i...
26914 Original commit message from CVS:
26915 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
26916 (gst_xvimagesink_buffer_alloc):
26917 * sys/xvimage/xvimagesink.h:
26918 When performing buffer allocations, remember the caps and image format
26919 we return so that if the same caps are asked for next time we can
26920 return them immediately without doing any caps intersections.
26922 2006-05-18 23:00:02 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
26924 gst-libs/gst/rtp/README: Some new documentation
26925 Original commit message from CVS:
26926 2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
26927 * gst-libs/gst/rtp/README:
26928 Some new documentation
26929 * gst-libs/gst/rtp/gstrtpbuffer.h:
26930 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
26931 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
26932 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
26933 New RTP audio base payloader class. Supports frame or sample based codecs.
26934 Not enabled in Makefile.am until approved.
26936 2006-05-18 20:30:26 +0000 Tim-Philipp Müller <tim@centricular.net>
26938 tests/check/elements/alsa.c: Fix test case: don't try to free NULL GValueArray when there are no devices.
26939 Original commit message from CVS:
26940 * tests/check/elements/alsa.c: (test_device_property_probe):
26941 Fix test case: don't try to free NULL GValueArray when there
26944 2006-05-18 19:21:53 +0000 Tim-Philipp Müller <tim@centricular.net>
26946 tests/check/: Add simple test that runs a device property probe on alsasrc, alsasink and alsamixer. Disable valgrind ...
26947 Original commit message from CVS:
26948 * tests/check/Makefile.am:
26949 * tests/check/elements/alsa.c: (test_device_property_probe),
26950 (alsa_suite), (main):
26951 Add simple test that runs a device property probe on alsasrc,
26952 alsasink and alsamixer. Disable valgrind check for now (too
26953 many leaks in libasound, and valgrind ignored my suppressions
26956 2006-05-18 17:19:39 +0000 Martin Szulecki <gnomebugzilla@sukimashita.com>
26958 ext/alsa/: Clean up and simplify alsa device probing. Make it actually work for multiple classes. Don't cache results...
26959 Original commit message from CVS:
26960 * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
26961 (gst_alsa_device_property_probe_probe_property),
26962 (gst_alsa_device_property_probe_needs_probe),
26963 (gst_alsa_device_property_probe_get_values),
26964 (gst_alsa_type_add_device_property_probe_interface):
26965 * ext/alsa/gstalsadeviceprobe.h:
26966 * ext/alsa/gstalsamixerelement.c:
26967 (gst_alsa_mixer_element_init_interfaces):
26968 * ext/alsa/gstalsamixerelement.h:
26969 Clean up and simplify alsa device probing. Make it actually work
26970 for multiple classes. Don't cache results any longer.
26971 * ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
26972 (gst_alsasink_init):
26973 * ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
26974 (gst_alsasrc_interface_supported), (gst_implements_interface_init),
26975 (gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
26976 Make alsasink and alsasrc implement the GstPropertyProbe interface
26977 for device probing (#342181).
26978 Patch by: Martin Szulecki <gnomebugzilla at sukimashita com>
26980 2006-05-18 10:05:23 +0000 Tim-Philipp Müller <tim@centricular.net>
26982 gst/subparse/samiparse.c: Don't ignore return value of strtol (++compiler_happiness).
26983 Original commit message from CVS:
26984 * gst/subparse/samiparse.c: (handle_start_font):
26985 Don't ignore return value of strtol (++compiler_happiness).
26987 2006-05-17 17:49:10 +0000 Young-Ho Cha <ganadist@chollian.net>
26989 gst/subparse/gstsubparse.*: Add 'encoding' property (#341681).
26990 Original commit message from CVS:
26991 Patch by: Young-Ho Cha <ganadist chollian net>
26992 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
26993 (gst_sub_parse_class_init), (gst_sub_parse_init),
26994 (gst_sub_parse_set_property), (gst_sub_parse_get_property),
26995 (convert_encoding):
26996 * gst/subparse/gstsubparse.h:
26997 Add 'encoding' property (#341681).
26998 * gst/subparse/samiparse.c: (characters_sami):
26999 Output is pango markup, so we need to escape text
27000 between tags (#342143).
27002 2006-05-16 17:34:14 +0000 Tim-Philipp Müller <tim@centricular.net>
27004 gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
27005 Original commit message from CVS:
27006 * gst-libs/gst/audio/multichannel.c:
27007 (gst_audio_check_channel_positions):
27008 It's okay to have caps with channels=1 and a channel position
27009 different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
27010 (deinterleavers might want to keep the position in the caps,
27011 so that they can be re-interleaved again properly later).
27012 Leave check for unexpected 2-channel layouts intact for now.
27014 2006-05-16 16:28:10 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
27016 gst/tcp/gsttcp.c: Return GST_FLOW_UNEXPECTED when we have an eos on the socket so basesrc can do its job correctly.
27017 Original commit message from CVS:
27018 2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
27019 * gst/tcp/gsttcp.c: (gst_tcp_socket_read):
27020 Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
27021 basesrc can do its job correctly.
27023 2006-05-16 15:52:17 +0000 Tim-Philipp Müller <tim@centricular.net>
27025 ext/alsa/: Refactor and improve caps probing code: probe signedness when we probe the supported formats/widths; set e...
27026 Original commit message from CVS:
27027 * ext/alsa/Makefile.am:
27028 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
27029 (gst_alsa_detect_formats), (get_channel_free_structure),
27030 (caps_add_channel_configuration), (gst_alsa_detect_channels),
27031 (gst_alsa_probe_supported_formats):
27032 * ext/alsa/gstalsa.h:
27033 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
27034 Refactor and improve caps probing code: probe signedness
27035 when we probe the supported formats/widths; set endianness
27036 to the one we actually probed for (ie. cpu endianness).
27037 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
27038 (gst_alsasrc_close):
27039 * ext/alsa/gstalsasrc.h:
27040 Implement caps probing for alsasrc.
27042 2006-05-15 17:42:19 +0000 Wim Taymans <wim.taymans@gmail.com>
27044 ext/theora/theoradec.c: Cleanups, add some G_LIKELY.
27045 Original commit message from CVS:
27046 * ext/theora/theoradec.c: (gst_theora_dec_reset),
27047 (theora_dec_src_query), (theora_dec_src_event),
27048 (theora_dec_sink_event), (theora_handle_comment_packet),
27049 (theora_handle_data_packet), (theora_dec_change_state):
27050 Cleanups, add some G_LIKELY.
27051 Use segment helpers instead of our own wrong code.
27052 Clear queued buffers on seek and READY.
27053 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
27054 (vorbis_dec_convert), (vorbis_dec_src_query),
27055 (vorbis_dec_src_event), (vorbis_dec_sink_event),
27056 (vorbis_handle_comment_packet), (vorbis_dec_push),
27057 (vorbis_handle_data_packet), (vorbis_dec_chain),
27058 (vorbis_dec_change_state):
27059 * ext/vorbis/vorbisdec.h:
27060 Remove old useless packetno variable.
27061 Do position query properly.
27063 Do cleanup of queued buffers in new helper function
27066 2006-05-15 17:17:22 +0000 Tim-Philipp Müller <tim@centricular.net>
27068 ext/alsa/gstalsasink.c: Query supported sample rates. Fixes #341732.
27069 Original commit message from CVS:
27070 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
27071 Query supported sample rates. Fixes #341732.
27073 2006-05-15 17:01:02 +0000 Julien Moutte <julien@moutte.net>
27075 gst/playback/gstdecodebin.c: Make decodebin reusable when going from PAUSE_TO_READY and then back to PAUSED.
27076 Original commit message from CVS:
27077 2006-05-15 Julien MOUTTE <julien@moutte.net>
27078 * gst/playback/gstdecodebin.c: (cleanup_decodebin),
27079 (gst_decode_bin_change_state): Make decodebin reusable
27080 when going from PAUSE_TO_READY and then back to PAUSED.
27083 2006-05-15 16:49:31 +0000 Wim Taymans <wim.taymans@gmail.com>
27085 ext/vorbis/vorbisdec.c: Cleanups. Use refcounting and DEBUG_OBJECT.
27086 Original commit message from CVS:
27087 * ext/vorbis/vorbisdec.c: (vorbis_get_query_types),
27088 (vorbis_dec_convert), (vorbis_dec_src_query),
27089 (vorbis_dec_sink_query), (vorbis_dec_src_event),
27090 (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
27091 (vorbis_dec_clean_queued), (vorbis_dec_push),
27092 (vorbis_handle_data_packet), (vorbis_dec_change_state):
27093 Cleanups. Use refcounting and DEBUG_OBJECT.
27094 Reset segment on flush, use code methods instead of our
27096 Fix potential memleak.
27098 2006-05-15 16:46:44 +0000 Tim-Philipp Müller <tim@centricular.net>
27100 ext/alsa/gstalsasink.*: Don't leak allocated snd_output_t structure if there's more than one alsasink instance at a t...
27101 Original commit message from CVS:
27102 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
27103 (gst_alsasink_init):
27104 * ext/alsa/gstalsasink.h:
27105 Don't leak allocated snd_output_t structure if there's
27106 more than one alsasink instance at a time (#341873).
27107 Also fix GObject macros in header file.
27109 2006-05-15 15:31:30 +0000 Tim-Philipp Müller <tim@centricular.net>
27111 gst/subparse/gstsubparse.c: Don't use libxml functions in the typefinding code.
27112 Original commit message from CVS:
27113 * gst/subparse/gstsubparse.c:
27114 (gst_sub_parse_data_format_autodetect):
27115 Don't use libxml functions in the typefinding code.
27117 2006-05-15 15:01:08 +0000 Wim Taymans <wim.taymans@gmail.com>
27119 ext/ogg/gstoggdemux.c: Fix seeking performance in the case where a non-header packet has a 0 granulepos (busted theor...
27120 Original commit message from CVS:
27121 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
27122 Fix seeking performance in the case where a non-header
27123 packet has a 0 granulepos (busted theora case).
27126 2006-05-15 14:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
27128 gst/subparse/gstsubparse.c: Improve SAMI typefinding: handle case where there are whitespaces or newlines in front of...
27129 Original commit message from CVS:
27130 * gst/subparse/gstsubparse.c:
27131 (gst_sub_parse_data_format_autodetect):
27132 Improve SAMI typefinding: handle case where there are
27133 whitespaces or newlines in front of the first <SAMI>
27136 2006-05-15 12:18:13 +0000 Tim-Philipp Müller <tim@centricular.net>
27138 configure.ac: Build video4linux plugin even if there's no XVIDEO, just without implementing the GstXOverlay interface...
27139 Original commit message from CVS:
27141 Build video4linux plugin even if there's no XVIDEO, just
27142 without implementing the GstXOverlay interface (#334002).
27144 2006-05-15 10:17:04 +0000 Tim-Philipp Müller <tim@centricular.net>
27146 Add tentative support for libvisual-0.4 (#336881).
27147 Original commit message from CVS:
27149 * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
27151 Add tentative support for libvisual-0.4 (#336881).
27153 2006-05-15 09:41:03 +0000 Young-Ho Cha <ganadist@chollian.net>
27155 gst/subparse/samiparse.c: Need to map "silver" colour explicitly (#169936).
27156 Original commit message from CVS:
27157 Patch by: Young-Ho Cha <ganadist at chollian net>
27158 * gst/subparse/samiparse.c: (handle_start_font):
27159 Need to map "silver" colour explicitly (#169936).
27161 2006-05-15 09:14:35 +0000 Young-Ho Cha <ganadist@chollian.net>
27163 gst/subparse/: Add support for SAMI subtitles (#169936).
27164 Original commit message from CVS:
27165 Patch by: Young-Ho Cha <ganadist at chollian net>
27166 * gst/subparse/Makefile.am:
27167 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
27168 (parser_state_dispose), (gst_sub_parse_data_format_autodetect),
27169 (gst_sub_parse_format_autodetect), (feed_textbuf),
27170 (gst_subparse_type_find), (plugin_init):
27171 * gst/subparse/gstsubparse.h:
27172 * gst/subparse/samiparse.c:
27173 * gst/subparse/samiparse.h:
27174 Add support for SAMI subtitles (#169936).
27176 2006-05-14 21:18:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27178 * win32/common/config.h:
27180 Original commit message from CVS:
27183 2006-05-14 21:18:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27186 fix mistakes in README
27187 Original commit message from CVS:
27188 fix mistakes in README
27190 2006-05-14 18:15:17 +0000 Michael Smith <msmith@xiph.org>
27192 gst/audioconvert/gstchannelmix.c: Fix #341696: crash when mixing L+R+C to mono or stereo.
27193 Original commit message from CVS:
27194 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
27195 Fix #341696: crash when mixing L+R+C to mono or stereo.
27196 * tests/check/Makefile.am:
27197 * tests/check/elements/audioconvert.c: (set_channel_positions),
27198 (get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
27199 (audioconvert_suite):
27200 Add test for the above, including some generic framework bits for
27201 testing multichannel things.
27203 2006-05-14 16:05:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27207 Original commit message from CVS:
27210 === release 0.10.7 ===
27212 2006-05-14 16:00:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27214 configure.ac: releasing 0.10.7, "Leave the gun"
27215 Original commit message from CVS:
27216 2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
27218 releasing 0.10.7, "Leave the gun"
27220 2006-05-14 15:55:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27238 Original commit message from CVS:
27241 2006-05-12 22:22:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27244 Original commit message from CVS:
27245 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
27246 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
27249 2006-05-12 21:30:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27251 Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
27252 Original commit message from CVS:
27253 * docs/libs/gst-plugins-base-libs-docs.sgml:
27254 * docs/libs/gst-plugins-base-libs-sections.txt:
27255 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
27256 * gst-libs/gst/video/video.h:
27257 * gst/videoscale/Makefile.am:
27258 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
27259 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
27260 * tests/check/Makefile.am:
27261 * tests/check/libs/video.c: (GST_START_TEST), (video_suite),
27263 Fix integer overflow problem with pixel-aspect-ratio calculations
27264 in videoscale and xvimagesink (#341542)
27266 2006-05-12 16:56:52 +0000 Tim-Philipp Müller <tim@centricular.net>
27268 gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
27269 Original commit message from CVS:
27270 * gst-libs/gst/tag/gstid3tag.c:
27271 Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
27273 2006-05-12 10:39:08 +0000 Sébastien Moutte <sebastien@moutte.net>
27275 win32/MANIFEST: update win32 files listing
27276 Original commit message from CVS:
27278 update win32 files listing
27280 2006-05-11 21:47:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27282 * tests/check/elements/multifdsink.c:
27283 disable failing check on gentoo64
27284 Original commit message from CVS:
27285 disable failing check on gentoo64
27287 2006-05-11 21:35:44 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27289 * tests/check/elements/multifdsink.c:
27290 disable failing check on gentoo64
27291 Original commit message from CVS:
27292 disable failing check on gentoo64
27294 2006-05-11 21:20:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27296 * tests/check/elements/multifdsink.c:
27297 macros show the correct line
27298 Original commit message from CVS:
27299 macros show the correct line
27301 2006-05-11 21:04:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27303 * tests/check/elements/multifdsink.c:
27304 macros show the correct line
27305 Original commit message from CVS:
27306 macros show the correct line
27308 2006-05-11 21:01:05 +0000 Sjoerd Simons <sjoerd@luon.net>
27310 gst/playback/gstplaybasebin.*: API: GstPlayBaseBin::stream-info-value-array property use a more bindings-friendly way...
27311 Original commit message from CVS:
27312 2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
27313 patch by: Sjoerd Simons (sjoerd@luon.net)
27314 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
27315 (group_create), (group_destroy), (add_stream),
27316 (gst_play_base_bin_get_property),
27317 (gst_play_base_bin_get_streaminfo_value_array):
27318 * gst/playback/gstplaybasebin.h:
27319 API: GstPlayBaseBin::stream-info-value-array property
27320 use a more bindings-friendly way of exposing streaminfo
27321 using a GValueArray. Tested in ipython.
27324 2006-05-11 19:44:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27326 * tests/check/elements/multifdsink.c:
27327 fix some type warnings
27328 Original commit message from CVS:
27329 fix some type warnings
27331 2006-05-11 19:38:22 +0000 Wim Taymans <wim.taymans@gmail.com>
27333 gst/playback/gstdecodebin.c: Also catch queue underruns but don't do anything yet.
27334 Original commit message from CVS:
27335 * gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
27336 (queue_underrun_cb), (queue_filled_cb):
27337 Also catch queue underruns but don't do anything yet.
27338 Refactor and comment queue enlarging code a bit.
27339 * gst/playback/gstplaybasebin.c: (queue_overrun),
27340 (queue_threshold_reached), (queue_out_of_data),
27341 (gen_preroll_element):
27342 If a queue over/underruns check that we don't create nasty
27343 deadlocks when the min-threshold is not reached but the
27344 max-bytes is. In those cases disable max-bytes when we
27345 know that the queue is fed timed data.
27348 2006-05-11 18:06:18 +0000 Tim-Philipp Müller <tim@centricular.net>
27350 gst/playback/gstplaybin.c: Make playbin automatically plug an 'audioresample' element before the audio sink as well. ...
27351 Original commit message from CVS:
27352 * gst/playback/gstplaybin.c: (gen_audio_element):
27353 Make playbin automatically plug an 'audioresample'
27354 element before the audio sink as well. This solves
27355 problems with sinks that only accept a very specific
27356 sample rate, like esdsink (e.g. #340379).
27358 2006-05-11 16:04:28 +0000 Tim-Philipp Müller <tim@centricular.net>
27360 gst/playback/gstplaybasebin.c: Make http sources send special headers so that we receive icecast metadata if the http...
27361 Original commit message from CVS:
27362 * gst/playback/gstplaybasebin.c: (gen_source_element):
27363 Make http sources send special headers so that we receive
27364 icecast metadata if the http stream is an icecast stream
27365 (otherwise the server will just ignore them). This also
27366 means that from now on users will need the 'icydemux'
27367 element from gst-plugins-good installed if they want to
27368 listen to icecast radio streams. (#341432, #333657).
27370 2006-05-11 12:34:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27372 * gst/tcp/gstmultifdsink.c:
27374 Original commit message from CVS:
27377 2006-05-11 11:40:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27379 gst/tcp/gstmultifdsink.c: remove stupid example from docs - it should come with a simple
27380 Original commit message from CVS:
27381 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
27382 (gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
27383 remove stupid example from docs - it should come with a simple
27386 * tests/check/elements/multifdsink.c: (wait_bytes_served),
27387 (fail_if_can_read), (GST_START_TEST),
27388 (gst_multifdsink_create_streamheader), (multifdsink_suite):
27389 add a test for changing streamheader which exposes a bug in
27392 2006-05-11 10:33:46 +0000 Michael Smith <msmith@xiph.org>
27394 ext/gnomevfs/gstgnomevfssrc.*: Don't set icy-caps unless we have a sane interval value. Move interval to a local vari...
27395 Original commit message from CVS:
27396 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
27397 (gst_gnome_vfs_src_received_headers_callback):
27398 * ext/gnomevfs/gstgnomevfssrc.h:
27399 Don't set icy-caps unless we have a sane interval value. Move
27400 interval to a local variable; we never use it outside this function.
27402 2006-05-11 10:14:20 +0000 Wim Taymans <wim.taymans@gmail.com>
27404 sys/: Register special buffer types along with the objects so that they are not registered at runtime from N differen...
27405 Original commit message from CVS:
27406 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
27407 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
27408 Register special buffer types along with the objects so
27409 that they are not registered at runtime from N different
27410 streaming threads since they are not threadsafe.
27412 2006-05-10 18:31:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27414 * tests/check/elements/multifdsink.c:
27415 set caps and plug leaks
27416 Original commit message from CVS:
27417 set caps and plug leaks
27419 2006-05-10 18:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27421 tests/check/elements/multifdsink.c: add two more tests, one doing streamheader
27422 Original commit message from CVS:
27423 * tests/check/elements/multifdsink.c: (wait_bytes_served),
27424 (GST_START_TEST), (fail_unless_read), (multifdsink_suite):
27425 add two more tests, one doing streamheader
27427 2006-05-10 16:34:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27429 gst/tcp/gstmultifdsink.c: clean up the bufqueue when shutting down
27430 Original commit message from CVS:
27431 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
27432 clean up the bufqueue when shutting down
27433 * tests/check/Makefile.am:
27434 * tests/check/elements/multifdsink.c: (setup_multifdsink),
27435 (cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
27437 add a test for the leak that was just fixed
27439 2006-05-10 15:16:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27441 * gst/tcp/gstmultifdsink.c:
27443 Original commit message from CVS:
27446 2006-05-10 15:14:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27448 * gst/tcp/gstmultifdsink.c:
27449 * gst/tcp/gstmultifdsink.h:
27451 Original commit message from CVS:
27454 2006-05-10 11:54:36 +0000 Wim Taymans <wim.taymans@gmail.com>
27456 gst/adder/gstadder.*: Updated some docs. Added comments and FIXMEs all over the place.
27457 Original commit message from CVS:
27458 * gst/adder/gstadder.c: (gst_adder_setcaps),
27459 (gst_adder_query_duration), (gst_adder_query), (forward_event),
27460 (gst_adder_src_event), (gst_adder_sink_event),
27461 (gst_adder_class_init), (gst_adder_finalize),
27462 (gst_adder_request_new_pad), (gst_adder_collected):
27463 * gst/adder/gstadder.h:
27464 Updated some docs. Added comments and FIXMEs all over the place.
27465 Improve debugging info.
27466 Fix leak on finalize by not calling the parent.
27467 Implement duration query.
27468 Make event forwarding threadsafe.
27469 Correctly send NEWSEGMENT at start and after flush.
27470 Handle EOS correctly.
27471 Post error when not negotiated.
27472 * tests/check/elements/adder.c: (GST_START_TEST):
27473 Added FIXME in the test.
27475 2006-05-09 19:24:46 +0000 Tim-Philipp Müller <tim@centricular.net>
27477 Const-ify GEnumValue and GFlagsValue arrays. Use
27478 Original commit message from CVS:
27479 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
27480 (gst_text_overlay_halign_get_type),
27481 (gst_text_overlay_wrap_mode_get_type):
27482 * ext/theora/theoradec.c: (theora_handle_type_packet),
27483 (theora_handle_data_packet):
27484 * ext/theora/theoraenc.c: (gst_border_mode_get_type),
27485 (theora_enc_sink_setcaps), (theora_enc_chain):
27486 * gst-libs/gst/cdda/gstcddabasesrc.c:
27487 (gst_cdda_base_src_mode_get_type):
27488 * gst/audiotestsrc/gstaudiotestsrc.c:
27489 (gst_audiostestsrc_wave_get_type):
27490 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
27491 * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
27492 * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
27493 (gst_sync_method_get_type), (gst_unit_type_get_type),
27494 (gst_client_status_get_type):
27495 * gst/videoscale/gstvideoscale.c:
27496 (gst_video_scale_method_get_type):
27497 * gst/videotestsrc/gstvideotestsrc.c:
27498 (gst_video_test_src_pattern_get_type):
27499 * gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
27500 (paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
27501 (paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
27502 (paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
27503 (paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
27504 (paint_setup_RGB565), (paint_setup_xRGB1555):
27505 Const-ify GEnumValue and GFlagsValue arrays. Use
27506 GST_ROUND_UP_* macros instead of home-made ones.
27508 2006-05-09 17:40:41 +0000 Tim-Philipp Müller <tim@centricular.net>
27510 configure.ac: Require core CVS for the new newsegment stuff.
27511 Original commit message from CVS:
27513 Require core CVS for the new newsegment stuff.
27515 2006-05-09 17:30:48 +0000 Sjoerd Simons <sjoerd@luon.net>
27517 gst/tcp/gstmultifdsink.c: Register nick for enum value (#341160).
27518 Original commit message from CVS:
27519 Patch by: Sjoerd Simons <sjoerd at luon net>
27520 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
27521 Register nick for enum value (#341160).
27523 2006-05-09 16:46:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27525 gst/typefind/gsttypefindfunctions.c: backout typefind patch #340375
27526 Original commit message from CVS:
27527 * gst/typefind/gsttypefindfunctions.c: (m4a_type_find),
27529 backout typefind patch #340375
27530 * tests/check/elements/adder.c: (message_received),
27531 (GST_START_TEST), (adder_suite):
27532 redo, signal-handling of test
27534 2006-05-09 16:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
27536 gst/adder/gstadder.*: Remove bogus segment merging and forwarding, we don't care about timestamps anyway and we just ...
27537 Original commit message from CVS:
27538 * gst/adder/gstadder.c: (gst_adder_request_new_pad),
27539 (gst_adder_collected):
27540 * gst/adder/gstadder.h:
27541 Remove bogus segment merging and forwarding, we don't
27542 care about timestamps anyway and we just produce a
27544 Also create a nice NEWSEGMENT event when we start.
27545 Use _scale_int some more.
27547 2006-05-09 11:59:13 +0000 Edward Hervey <bilboed@bilboed.com>
27549 tests/icles/stress-xoverlay.c: Fix if core was built without parsing support.
27550 Original commit message from CVS:
27551 * tests/icles/stress-xoverlay.c:
27552 Fix if core was built without parsing support.
27554 2006-05-09 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
27556 gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
27557 Original commit message from CVS:
27558 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
27559 Add SEDG (Samsung MPEG-4) fourcc.
27561 2006-05-09 11:31:47 +0000 Edward Hervey <bilboed@bilboed.com>
27563 tests/examples/volume/volume.c: Fox if core was built without parsing support.
27564 Original commit message from CVS:
27565 * tests/examples/volume/volume.c:
27566 Fox if core was built without parsing support.
27567 * tests/examples/seek/seek.c:
27568 Disable the parse_launch example if core was built without parsing
27571 2006-05-09 11:21:24 +0000 Edward Hervey <bilboed@bilboed.com>
27573 tests/examples/seek/seek.c: Disable the parse_launch example if core was built without parsing support.
27574 Original commit message from CVS:
27575 * tests/examples/seek/seek.c:
27576 Disable the parse_launch example if core was built without parsing
27579 2006-05-08 15:51:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27581 * docs/libs/tmpl/gstcolorbalance.sgml:
27582 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
27583 * gst/tcp/gstmultifdsink.c:
27584 * gst/videoscale/gstvideoscale.c:
27585 doc reparagraphing and DEBUG_FUNCPTRing
27586 Original commit message from CVS:
27587 doc reparagraphing and DEBUG_FUNCPTRing
27589 2006-05-08 11:51:43 +0000 Edward Hervey <bilboed@bilboed.com>
27591 autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize
27592 Original commit message from CVS:
27593 * autogen.sh: (CONFIGURE_DEF_OPT):
27594 libtoolize on Darwin/MacOSX is called glibtoolize
27596 2006-05-07 17:39:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27598 tests/check/: Disable the adder test, until the build-slaves posses the kindness to either like it or to give valid r...
27599 Original commit message from CVS:
27600 * tests/check/Makefile.am:
27601 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
27602 Disable the adder test, until the build-slaves posses the kindness to
27603 either like it or to give valid reason for not doing so
27605 2006-05-07 17:25:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27607 tests/check/elements/adder.c: Shuffle NULL state change around and raise timeout more
27608 Original commit message from CVS:
27609 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
27611 Shuffle NULL state change around and raise timeout more
27613 2006-05-07 17:07:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27615 gst/typefind/gsttypefindfunctions.c: Add typefind to distinguish between "audio/x-m4a" and new type "video/mp4". Fixe...
27616 Original commit message from CVS:
27617 * gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
27618 (mp4_type_find), (plugin_init):
27619 Add typefind to distinguish between "audio/x-m4a" and new type
27620 "video/mp4". Fixes #340375
27621 * tests/check/elements/adder.c: (adder_suite):
27622 Raise timeout to make buildbot happy
27624 2006-05-07 16:39:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27626 Add sink-event handling to adder. It tries to merge incomming newsegment-events. Added test to check if segment_done ...
27627 Original commit message from CVS:
27628 * gst/adder/gstadder.c: (gst_adder_sink_event),
27629 (gst_adder_request_new_pad), (gst_adder_change_state):
27630 * gst/adder/gstadder.h:
27631 * tests/check/Makefile.am:
27632 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
27633 (adder_suite), (main):
27634 Add sink-event handling to adder. It tries to merge incomming
27635 newsegment-events. Added test to check if segment_done is comming
27638 2006-05-05 16:34:15 +0000 Andy Wingo <wingo@pobox.com>
27641 * ext/theora/theoraparse.c:
27642 * ext/vorbis/vorbisparse.c:
27643 ext/theora/theoraparse.c (gst_theora_parse_init) ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
27644 Original commit message from CVS:
27645 2006-05-05 Andy Wingo <wingo@pobox.com>
27646 * ext/theora/theoraparse.c (gst_theora_parse_init)
27647 (theora_parse_src_convert, theora_parse_src_query):
27648 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
27649 (vorbis_parse_convert, vorbis_parse_src_query): Add convert and
27650 query functions on the source pads of the theora and vorbis parse
27651 elements. Fixes position querying when doing a remux.
27653 2006-05-05 13:46:37 +0000 Michael Smith <msmith@xiph.org>
27655 ext/theora/theoraparse.c: Fix flushing.
27656 Original commit message from CVS:
27657 * ext/theora/theoraparse.c: (parse_granulepos),
27658 (theora_parse_drain_queue_prematurely),
27659 (theora_parse_queue_buffer), (theora_parse_sink_event):
27661 Fix invalid granulepos outputs when starting with a non-keyframe.
27663 2006-05-05 12:37:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27665 gst/typefind/gsttypefindfunctions.c: Rearrange MPEG system stream detection, fixing some memleaks in the process.
27666 Original commit message from CVS:
27667 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
27668 (mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
27669 Rearrange MPEG system stream detection, fixing some memleaks in the
27671 Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
27672 they clean up their data correctly.
27673 Remove unused ogganx caps and move the 'is_annodex' check to inside
27674 the 'is_ogg' if statement.
27676 2006-05-05 11:33:37 +0000 Wim Taymans <wim.taymans@gmail.com>
27678 gst/playback/gstdecodebin.c: Properly remove ghostpads. Fixes #340392
27679 Original commit message from CVS:
27680 * gst/playback/gstdecodebin.c: (cleanup_decodebin):
27681 Properly remove ghostpads. Fixes #340392
27683 2006-05-04 18:43:58 +0000 David Schleef <ds@schleef.org>
27685 gst/typefind/gsttypefindfunctions.c:
27686 Original commit message from CVS:
27687 * gst/typefind/gsttypefindfunctions.c:
27689 2006-05-03 16:32:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27691 gst/typefind/gsttypefindfunctions.c: When typefinding an MP3 in push-based mode, don't penalise the probability down ...
27692 Original commit message from CVS:
27693 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
27694 (mpeg_ts_probe_headers), (mpeg_ts_type_find):
27695 When typefinding an MP3 in push-based mode, don't penalise the
27696 probability down to 74% when we found 5 valid frames just because we
27697 can't peek the end of the file.
27698 Make the probability for detecting MPEG Transport Streams based on the
27699 number of sequential headers we successfully detected.
27701 2006-05-03 15:52:46 +0000 Wim Taymans <wim.taymans@gmail.com>
27703 ext/vorbis/vorbisdec.c: Still produce an error when we receive an empty packet.
27704 Original commit message from CVS:
27705 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
27706 (vorbis_dec_push), (vorbis_dec_chain):
27707 Still produce an error when we receive an empty packet.
27709 2006-05-03 15:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
27711 ext/ogg/gstoggdemux.c: Mark buffers with DISCONT after seek and after activating new chains.
27712 Original commit message from CVS:
27713 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
27714 (gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
27715 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
27716 Mark buffers with DISCONT after seek and after activating new
27718 * ext/theora/gsttheoradec.h:
27719 * ext/theora/theoradec.c: (gst_theora_dec_reset),
27720 (theora_get_query_types), (theora_dec_sink_event),
27721 (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
27722 (theora_dec_change_state):
27724 Detect and mark DISCONT buffers.
27725 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
27726 (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
27727 (vorbis_dec_change_state):
27728 * ext/vorbis/vorbisdec.h:
27730 Detect and mark DISCONT buffers.
27731 Don't crash on 0 sized buffers.
27733 2006-05-03 08:58:13 +0000 Wim Taymans <wim.taymans@gmail.com>
27735 gst/volume/gstvolume.c: Increase "volume" property to 10.0. Fixes #340369.
27736 Original commit message from CVS:
27737 * gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
27738 (volume_transform_ip):
27739 Increase "volume" property to 10.0. Fixes #340369.
27740 Set the process function to NULL when capsnego fails so that
27741 we properly error out.
27743 2006-05-02 18:15:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27745 gst/playback/: free cpas using gst_caps_unref, don't leak caps-strings
27746 Original commit message from CVS:
27747 * gst/playback/gstplaybin.c: (add_sink):
27748 * gst/playback/test.c: (main):
27749 * gst/playback/test5.c: (dump_element_stats):
27750 * gst/playback/test6.c: (main):
27751 free cpas using gst_caps_unref, don't leak caps-strings
27753 2006-05-02 06:33:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27755 * gst-libs/gst/rtp/gstbasertppayload.c:
27757 Original commit message from CVS:
27760 2006-05-01 19:08:40 +0000 Tim-Philipp Müller <tim@centricular.net>
27762 gst/typefind/gsttypefindfunctions.c: Refine musepack typefinding a bit. Return MAXIMUM probability when we detect str...
27763 Original commit message from CVS:
27764 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
27766 Refine musepack typefinding a bit. Return MAXIMUM
27767 probability when we detect stream version 7 to make
27768 sure the mpeg audio typefinder doesn't trump us.
27770 2006-04-29 16:25:58 +0000 Tim-Philipp Müller <tim@centricular.net>
27772 gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
27773 Original commit message from CVS:
27774 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
27775 Protect against unexpected NULL strf_data buffer.
27777 2006-04-29 13:09:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27779 tests/check/elements/audioconvert.c: interpret the out[] buffer in the order the bytes are actually put in, which is ...
27780 Original commit message from CVS:
27781 * tests/check/elements/audioconvert.c: (verify_convert),
27783 interpret the out[] buffer in the order the bytes are actually
27784 put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
27785 Other tests should use BYTE_ORDER since the array is filled in
27788 2006-04-29 12:10:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27790 * tests/check/elements/audioconvert.c:
27791 dump expected data when audioconvert test fails
27792 Original commit message from CVS:
27793 dump expected data when audioconvert test fails
27795 2006-04-29 11:55:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27797 tests/check/elements/audioconvert.c: when a test fails, give an indication of which it is
27798 Original commit message from CVS:
27799 * tests/check/elements/audioconvert.c: (verify_convert),
27801 when a test fails, give an indication of which it is
27803 2006-04-29 09:48:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27805 * ext/ogg/gstoggmux.c:
27806 * ext/theora/theoraenc.c:
27807 add another include
27808 Original commit message from CVS:
27809 add another include
27811 2006-04-29 01:24:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27813 * gst/subparse/gstssaparse.c:
27814 atoi() needs stdlib.h
27815 Original commit message from CVS:
27816 atoi() needs stdlib.h
27818 2006-04-29 01:18:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27820 * gst/playback/test4.c:
27821 * gst/playback/test5.c:
27822 * gst/playback/test6.c:
27823 exit needs stdlib.h
27824 Original commit message from CVS:
27825 exit needs stdlib.h
27827 2006-04-29 01:10:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27829 gst-libs/gst/cdda/gstcddabasesrc.c: compile fix; strtol() needs <stdlib.h>
27830 Original commit message from CVS:
27831 * gst-libs/gst/cdda/gstcddabasesrc.c:
27832 compile fix; strtol() needs <stdlib.h>
27834 2006-04-29 01:04:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27838 * docs/Makefile.am:
27839 * docs/libs/Makefile.am:
27840 * docs/libs/tmpl/gstcolorbalance.sgml:
27841 * docs/plugins/Makefile.am:
27843 use common upload.mak
27844 Original commit message from CVS:
27845 use common upload.mak
27847 2006-04-28 19:46:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27849 make GstElementDetails const
27850 Original commit message from CVS:
27851 * ext/alsa/gstalsamixerelement.c:
27852 * ext/alsa/gstalsasrc.c:
27853 * ext/cdparanoia/gstcdparanoiasrc.c:
27854 * ext/gnomevfs/gstgnomevfssink.c:
27855 * ext/gnomevfs/gstgnomevfssrc.c:
27856 * ext/ogg/gstoggdemux.c:
27857 * ext/ogg/gstoggmux.c:
27858 * ext/ogg/gstoggparse.c:
27859 * ext/ogg/gstogmparse.c:
27860 * ext/pango/gstclockoverlay.c:
27861 * ext/pango/gsttextoverlay.c:
27862 * ext/pango/gsttextrender.c:
27863 * ext/pango/gsttimeoverlay.c:
27864 * ext/theora/theoradec.c:
27865 * ext/theora/theoraenc.c:
27866 * ext/vorbis/vorbisdec.c:
27867 * ext/vorbis/vorbisenc.c:
27868 * gst-libs/gst/audio/gstaudiofilter.c:
27869 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
27870 * gst/audioconvert/gstaudioconvert.c:
27871 * gst/audiorate/gstaudiorate.c:
27872 * gst/audioresample/gstaudioresample.c:
27873 * gst/audiotestsrc/gstaudiotestsrc.c:
27874 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
27875 * gst/playback/gstdecodebin.c:
27876 * gst/playback/gstplaybin.c:
27877 * gst/playback/gststreamselector.c:
27878 * gst/subparse/gstsubparse.c:
27879 * gst/tcp/gstmultifdsink.c:
27880 * gst/tcp/gsttcpclientsink.c:
27881 * gst/tcp/gsttcpclientsrc.c:
27882 * gst/tcp/gsttcpserversink.c:
27883 * gst/tcp/gsttcpserversrc.c:
27884 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
27885 * gst/videorate/gstvideorate.c:
27886 * gst/videoscale/gstvideoscale.c:
27887 * gst/videotestsrc/gstvideotestsrc.c:
27888 * gst/volume/gstvolume.c:
27889 * sys/v4l/gstv4ljpegsrc.c:
27890 * sys/v4l/gstv4lmjpegsink.c:
27891 * sys/v4l/gstv4lmjpegsrc.c:
27892 * sys/v4l/gstv4lsrc.c:
27893 * sys/ximage/ximagesink.c:
27894 * sys/xvimage/xvimagesink.c:
27895 * tests/check/libs/cddabasesrc.c:
27896 make GstElementDetails const
27898 2006-04-28 19:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27900 gst/adder/gstadder.c: send events from src-pad to all sink-pads fixes #338657
27901 Original commit message from CVS:
27902 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
27904 send events from src-pad to all sink-pads fixes #338657
27906 2006-04-28 19:08:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27908 ext/alsa/gstalsasink.c: query witdh capabilities from alsa, fixes #338919
27909 Original commit message from CVS:
27910 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
27911 (alsasink_parse_spec):
27912 query witdh capabilities from alsa, fixes #338919
27914 2006-04-28 15:31:28 +0000 Wim Taymans <wim.taymans@gmail.com>
27916 gst/tcp/gstmultifdsink.*: Fix race condition in multifdsink that can lead to spurious duplicate clients. this patch a...
27917 Original commit message from CVS:
27918 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
27919 (gst_multi_fd_sink_remove_client_link):
27920 * gst/tcp/gstmultifdsink.h:
27921 Fix race condition in multifdsink that can lead to spurious
27922 duplicate clients. this patch adds a new signal that is fired when
27923 multifdsink has removed all references to the fd.
27925 Updated documentation.
27926 API: client-fd-removed signal added
27928 2006-04-28 15:24:00 +0000 Michael Smith <msmith@xiph.org>
27930 gst/tcp/gstmultifdsink.c: When asking g_value_array_new to prealloc elements, we may as well ask for the right number...
27931 Original commit message from CVS:
27932 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
27933 When asking g_value_array_new to prealloc elements, we may as well
27934 ask for the right number of elements.
27936 2006-04-28 15:08:09 +0000 Wim Taymans <wim.taymans@gmail.com>
27938 gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
27939 Original commit message from CVS:
27940 * gst-libs/gst/audio/gstbaseaudiosink.c:
27941 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
27942 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
27943 patch to make timestamp checking more tollerant to rounding
27944 errors given that real discontinuities are to be marked on
27945 buffers. Fixes some asf files and #338778.
27946 Also avoid some crashers when we receive an event in the
27949 2006-04-28 15:01:58 +0000 Michael Smith <msmith@xiph.org>
27951 ext/gnomevfs/gstgnomevfssrc.*: Remove ICY handling (mostly) from gnomevfssrc, in favour of proper shared support with...
27952 Original commit message from CVS:
27953 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
27954 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
27955 (gst_gnome_vfs_src_get_property),
27956 (gst_gnome_vfs_src_send_additional_headers_callback),
27957 (gst_gnome_vfs_src_received_headers_callback),
27958 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
27959 (gst_gnome_vfs_src_stop):
27960 * ext/gnomevfs/gstgnomevfssrc.h:
27961 Remove ICY handling (mostly) from gnomevfssrc, in favour of
27962 proper shared support within icydemux.
27964 2006-04-28 14:49:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27966 gst/videorate/gstvideorate.c: fix up docs fix a leak when no caps negotiated fix counting of input frames
27967 Original commit message from CVS:
27968 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
27969 (gst_video_rate_swap_prev), (gst_video_rate_chain):
27971 fix a leak when no caps negotiated
27972 fix counting of input frames
27973 * tests/check/elements/.cvsignore:
27974 * tests/check/elements/videorate.c: (assert_videorate_stats),
27975 (GST_START_TEST), (videorate_suite):
27976 add tests for these
27978 2006-04-28 14:48:11 +0000 Wim Taymans <wim.taymans@gmail.com>
27980 gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
27981 Original commit message from CVS:
27982 * gst-libs/gst/audio/gstringbuffer.c:
27983 (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
27984 (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
27985 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
27986 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
27987 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
27988 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
27989 (gst_ring_buffer_commit), (gst_ring_buffer_read),
27990 (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
27991 (gst_ring_buffer_clear), (gst_ring_buffer_may_start):
27992 Check arguments passed to public functions instead of
27995 2006-04-28 14:37:46 +0000 Wim Taymans <wim.taymans@gmail.com>
27997 gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
27998 Original commit message from CVS:
27999 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
28000 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
28001 GstBaseAudioSrc must be live or it does not work.
28002 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
28003 Don't set live to TRUE as this is the default in the parentclass.
28005 2006-04-28 14:37:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28007 * win32/common/config.h:
28009 Original commit message from CVS:
28012 2006-04-28 14:33:45 +0000 Wim Taymans <wim.taymans@gmail.com>
28014 gst/videoscale/gstvideoscale.c: Videoscale doesn't pass on pixel-aspect ratio. Handle all fixation cases better. Fixe...
28015 Original commit message from CVS:
28016 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps),
28017 (gst_video_scale_fixate_caps), (gst_video_scale_src_event):
28018 Videoscale doesn't pass on pixel-aspect ratio. Handle all
28019 fixation cases better. Fixes #338991
28021 2006-04-28 14:24:38 +0000 Wim Taymans <wim.taymans@gmail.com>
28023 gst/videotestsrc/gstvideotestsrc.c: Handle 0/1 framerate correctly Fixes #331901.
28024 Original commit message from CVS:
28025 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
28026 Handle 0/1 framerate correctly Fixes #331901.
28028 2006-04-28 14:22:16 +0000 Wim Taymans <wim.taymans@gmail.com>
28030 tests/check/elements/audioconvert.c: Added check for correct clipping when doing float samples in audioconvert.
28031 Original commit message from CVS:
28032 * tests/check/elements/audioconvert.c: (get_float_caps),
28033 (GST_START_TEST), (audioconvert_suite):
28034 Added check for correct clipping when doing float samples
28037 2006-04-28 14:19:49 +0000 Wim Taymans <wim.taymans@gmail.com>
28039 gst/videorate/gstvideorate.c: Print more debugging info.
28040 Original commit message from CVS:
28041 * gst/videorate/gstvideorate.c: (gst_video_rate_event),
28042 (gst_video_rate_chain):
28043 Print more debugging info.
28045 2006-04-28 14:17:00 +0000 Wim Taymans <wim.taymans@gmail.com>
28047 gst/audioresample/gstaudioresample.c: Add support for other formats audioresample can handle such as 32 bits in and f...
28048 Original commit message from CVS:
28049 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
28050 (resample_set_state_from_caps):
28051 Add support for other formats audioresample can handle such as
28052 32 bits in and float and 64 bits float. Fixes #301759
28054 2006-04-28 14:12:28 +0000 Wim Taymans <wim.taymans@gmail.com>
28056 gst/audioconvert/audioconvert.c: correctly clip float samples > 1.0. Fixes #338718
28057 Original commit message from CVS:
28058 * gst/audioconvert/audioconvert.c: (float):
28059 correctly clip float samples > 1.0. Fixes #338718
28061 2006-04-28 13:35:34 +0000 Young-Ho Cha <ganadist@chollian.net>
28063 ext/pango/gsttextoverlay.c: Don't strip newlines from the text. Also, center lines within multi-line paragraphs (#339...
28064 Original commit message from CVS:
28065 Patch by: Young-Ho Cha <ganadist at chollian net>
28066 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
28067 (gst_text_overlay_render_text):
28068 Don't strip newlines from the text. Also, center lines
28069 within multi-line paragraphs (#339405).
28071 2006-04-28 12:15:33 +0000 Tim-Philipp Müller <tim@centricular.net>
28073 gst/typefind/gsttypefindfunctions.c: Fix wavpack typefinding to work in more cases (don't peek for chunks of multiple...
28074 Original commit message from CVS:
28075 * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
28076 Fix wavpack typefinding to work in more cases (don't peek
28077 for chunks of multiple hundred kBs at once, but process
28078 things step-by-step in smaller units). Fixes #339786.
28080 2006-04-28 10:58:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28085 Original commit message from CVS:
28088 === release 0.10.6 ===
28090 2006-04-28 10:53:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28096 * docs/plugins/gst-plugins-base-plugins.signals:
28097 * docs/plugins/inspect/plugin-adder.xml:
28098 * docs/plugins/inspect/plugin-alsa.xml:
28099 * docs/plugins/inspect/plugin-audioconvert.xml:
28100 * docs/plugins/inspect/plugin-audiorate.xml:
28101 * docs/plugins/inspect/plugin-audioresample.xml:
28102 * docs/plugins/inspect/plugin-audiotestsrc.xml:
28103 * docs/plugins/inspect/plugin-cdparanoia.xml:
28104 * docs/plugins/inspect/plugin-decodebin.xml:
28105 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
28106 * docs/plugins/inspect/plugin-gnomevfs.xml:
28107 * docs/plugins/inspect/plugin-libvisual.xml:
28108 * docs/plugins/inspect/plugin-ogg.xml:
28109 * docs/plugins/inspect/plugin-pango.xml:
28110 * docs/plugins/inspect/plugin-playbin.xml:
28111 * docs/plugins/inspect/plugin-subparse.xml:
28112 * docs/plugins/inspect/plugin-tcp.xml:
28113 * docs/plugins/inspect/plugin-theora.xml:
28114 * docs/plugins/inspect/plugin-typefindfunctions.xml:
28115 * docs/plugins/inspect/plugin-video4linux.xml:
28116 * docs/plugins/inspect/plugin-videorate.xml:
28117 * docs/plugins/inspect/plugin-videoscale.xml:
28118 * docs/plugins/inspect/plugin-videotestsrc.xml:
28119 * docs/plugins/inspect/plugin-volume.xml:
28120 * docs/plugins/inspect/plugin-vorbis.xml:
28121 * docs/plugins/inspect/plugin-ximagesink.xml:
28122 * docs/plugins/inspect/plugin-xvimagesink.xml:
28125 Original commit message from CVS:
28128 2006-04-28 10:42:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28131 * win32/common/config.h:
28132 dist more win32 files
28133 Original commit message from CVS:
28134 dist more win32 files
28136 2006-04-28 10:41:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28153 Original commit message from CVS:
28156 2006-04-27 00:19:29 +0000 David Schleef <ds@schleef.org>
28158 gst/videoscale/gstvideoscale.c: Add call to oil_init().
28159 Original commit message from CVS:
28160 * gst/videoscale/gstvideoscale.c: Add call to oil_init().
28163 2006-04-26 17:20:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28167 * win32/common/config.h:
28169 Original commit message from CVS:
28172 2006-04-26 17:17:39 +0000 Wim Taymans <wim.taymans@gmail.com>
28174 ext/ogg/gstoggdemux.c: make sure correct newsegments are sent, so that the decoder and the demuxer agree on timestamp...
28175 Original commit message from CVS:
28176 2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
28177 patch by: Wim Taymans
28178 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
28179 (gst_ogg_demux_perform_seek):
28180 make sure correct newsegments are sent, so that the decoder
28181 and the demuxer agree on timestamps. Fixes playback of a lot
28182 of Ogg files that do not start from 0. Fixes #339833.
28184 2006-04-26 16:44:20 +0000 Edward Hervey <edward@fluendo.com>
28186 Fix an infinite loop if frames are passed in with wrongly ordered timestamps. Fixes #339013.
28187 Original commit message from CVS:
28188 Patch by: Edward Hervey <edward@fluendo.com>
28189 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
28190 * tests/check/Makefile.am:
28191 * tests/check/elements/videorate.c: (assert_videorate_stats),
28192 (setup_videorate), (cleanup_videorate), (GST_START_TEST),
28193 (videorate_suite), (main):
28194 Fix an infinite loop if frames are passed in with wrongly ordered
28195 timestamps. Fixes #339013.
28197 2006-04-26 13:55:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28200 * win32/common/config.h:
28202 Original commit message from CVS:
28205 2006-04-22 21:25:41 +0000 Tim-Philipp Müller <tim@centricular.net>
28207 gst/typefind/gsttypefindfunctions.c: fix typefinding on some ISO files. Fixes #339212.
28208 Original commit message from CVS:
28209 Patch by: Tim-Philipp Müller <tim at centricular dot net>
28210 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
28211 fix typefinding on some ISO files. Fixes #339212.
28213 2006-04-22 21:19:06 +0000 Tim-Philipp Müller <tim@centricular.net>
28215 gst-libs/gst/riff/riff-media.c: add another H264 fourcc. Fixes #339047.
28216 Original commit message from CVS:
28217 Patch by: Tim-Philipp Müller <tim at centricular dot net>
28218 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
28219 add another H264 fourcc. Fixes #339047.
28221 2006-04-22 21:12:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28223 gst/playback/gststreamselector.c: Restore old StreamSelector behaviour.
28224 Original commit message from CVS:
28225 Patch by: Jan Schmidt
28226 * gst/playback/gststreamselector.c:
28227 (gst_stream_selector_bufferalloc):
28228 Restore old StreamSelector behaviour.
28231 2006-04-13 09:26:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28234 * gst-libs/gst/rtp/Makefile.am:
28235 * gst-libs/gst/rtp/gstrtpbuffer.h:
28236 reverting rtp patches to fix freeze break on -base as explained on the list
28237 Original commit message from CVS:
28238 reverting rtp patches to fix freeze break on -base as explained on the list
28240 2006-04-13 03:55:12 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28242 gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
28243 Original commit message from CVS:
28244 2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28245 * gst-libs/gst/rtp/gstrtpbuffer.h:
28246 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
28247 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
28248 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
28249 New RTP audio base payloader class. Supports frame or sample based codecs
28251 2006-04-12 11:04:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28269 update libtool versioning
28270 Original commit message from CVS:
28271 update libtool versioning
28273 2006-04-12 10:58:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28277 * win32/common/config.h:
28279 Original commit message from CVS:
28282 2006-04-11 17:31:29 +0000 Antoine Tremblay <hexa00@gmail.com>
28284 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
28285 Original commit message from CVS:
28286 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
28287 * gst-libs/gst/rtp/gstbasertpdepayload.c:
28288 (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
28289 Fix some memory leaks: on finalize, free buffers left in the queue
28290 before destroying the queue; in _push(), unref rtp_buf even if
28291 the process vfunc returned a NULL buffer as output buffer (#337548);
28292 demote some recuring debug messages to LOG level.
28294 2006-04-11 15:01:51 +0000 Christian Schaller <uraeus@gnome.org>
28296 * gst-plugins-base.spec.in:
28297 fix version number macro
28298 Original commit message from CVS:
28299 fix version number macro
28301 2006-04-11 14:42:33 +0000 Wim Taymans <wim.taymans@gmail.com>
28303 ext/ogg/gstoggdemux.c: More cleanups.
28304 Original commit message from CVS:
28305 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
28306 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
28307 (gst_ogg_chain_free), (gst_ogg_demux_sink_event),
28308 (gst_ogg_demux_loop):
28310 Respect segment stop when emiting EOS or SEGMENT_DONE.
28313 2006-04-11 10:45:32 +0000 Tim-Philipp Müller <tim@centricular.net>
28315 gst/playback/gststreamselector.c: Don't leak pad name.
28316 Original commit message from CVS:
28317 * gst/playback/gststreamselector.c:
28318 (gst_stream_selector_get_property):
28319 Don't leak pad name.
28321 2006-04-11 09:42:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28324 Mention bug #336617 closed by recent commit
28325 Original commit message from CVS:
28326 Mention bug #336617 closed by recent commit
28328 2006-04-10 20:32:46 +0000 Michael Smith <msmith@xiph.org>
28330 tests/check/: so that FC4 buildslaves can pass.
28331 Original commit message from CVS:
28332 * tests/check/Makefile.am:
28333 * tests/check/gst-plugins-base.supp:
28334 Suppress an old libtheora bug (fixed in more recent versions), so
28335 that FC4 buildslaves can pass.
28337 2006-04-10 19:13:30 +0000 Wim Taymans <wim.taymans@gmail.com>
28339 ext/ogg/gstoggdemux.c: Don't leak events.
28340 Original commit message from CVS:
28341 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
28342 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
28343 (gst_ogg_demux_init), (gst_ogg_demux_finalize),
28344 (gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
28345 (gst_ogg_demux_loop):
28347 Remember what error we got when finding chains, if we
28348 were shutdown, that would not be an error.
28350 2006-04-10 17:05:46 +0000 Wim Taymans <wim.taymans@gmail.com>
28352 gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
28353 Original commit message from CVS:
28354 * gst-libs/gst/audio/gstbaseaudiosink.c:
28355 (gst_base_audio_sink_event):
28356 Starting the ringbuffer when we did not acquire it can cause
28357 a deadlock, is pointless and causes nasty things for
28359 Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
28361 2006-04-10 15:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
28363 ext/ogg/gstoggdemux.c: Add some more debugging.
28364 Original commit message from CVS:
28365 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
28366 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
28367 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
28368 (gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
28369 (gst_ogg_demux_deactivate_current_chain),
28370 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
28371 (gst_ogg_demux_bisect_forward_serialno),
28372 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
28373 Add some more debugging.
28375 2006-04-10 14:52:10 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28378 * ext/theora/theoraenc.c:
28380 Original commit message from CVS:
28383 2006-04-10 10:29:21 +0000 Wim Taymans <wim.taymans@gmail.com>
28385 ext/theora/theoradec.c: Some more debug info.
28386 Original commit message from CVS:
28387 * ext/theora/theoradec.c: (theora_dec_src_event),
28388 (theora_handle_data_packet):
28389 Some more debug info.
28390 * tests/examples/seek/seek.c: (start_seek), (main):
28391 Print element messages too.
28393 2006-04-09 17:14:22 +0000 Sébastien Moutte <sebastien@moutte.net>
28395 gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer sta...
28396 Original commit message from CVS:
28397 * gst/audioresample/debug.h:
28398 replace debug macros with variable number of parameters
28399 by a simple alias to gstreamer standard debug macros
28400 (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
28401 supported by MSVC 6.0 and 7.1)
28402 * gst/audioresample/resample.h:
28403 define M_PI and rint for WIN32
28404 * win32/common/libgstaudio.def:
28405 * win32/common/libgstriff.def:
28406 * win32/common/libgsttag.def:
28407 * win32/common/libgstvideo.def:
28408 add new exported functions
28410 update project files
28412 2006-04-08 21:02:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28414 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
28415 Original commit message from CVS:
28416 * ext/alsa/gstalsamixeroptions.c:
28417 (gst_alsa_mixer_options_class_init):
28418 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
28419 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
28420 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
28421 * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
28422 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
28423 * gst-libs/gst/audio/gstaudiofilter.c:
28424 (gst_audio_filter_class_init):
28425 * gst-libs/gst/audio/gstaudiosink.c:
28426 (gst_audioringbuffer_class_init):
28427 * gst-libs/gst/audio/gstaudiosrc.c:
28428 (gst_audioringbuffer_class_init):
28429 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
28430 * gst-libs/gst/interfaces/colorbalancechannel.c:
28431 (gst_color_balance_channel_class_init):
28432 * gst-libs/gst/interfaces/mixeroptions.c:
28433 (gst_mixer_options_class_init):
28434 * gst-libs/gst/interfaces/mixertrack.c:
28435 (gst_mixer_track_class_init):
28436 * gst-libs/gst/interfaces/tunerchannel.c:
28437 (gst_tuner_channel_class_init):
28438 * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
28439 * gst-libs/gst/netbuffer/gstnetbuffer.c:
28440 (gst_netbuffer_class_init):
28441 * gst-libs/gst/rtp/gstbasertppayload.c:
28442 (gst_basertppayload_class_init):
28443 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
28444 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
28445 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
28446 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
28447 * gst/playback/gststreamselector.c:
28448 (gst_stream_selector_class_init):
28449 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
28450 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
28451 * sys/v4l/gstv4lcolorbalance.c:
28452 (gst_v4l_color_balance_channel_class_init):
28453 * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
28454 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
28455 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
28456 * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
28457 (gst_v4l_tuner_norm_class_init):
28458 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
28459 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
28460 * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
28461 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
28463 2006-04-08 18:09:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28465 Fix broken GObject macros
28466 Original commit message from CVS:
28467 * ext/pango/gsttextrender.h:
28468 * gst-libs/gst/audio/gstaudiosink.h:
28469 * gst-libs/gst/audio/gstaudiosrc.h:
28470 * gst-libs/gst/audio/gstbaseaudiosink.h:
28471 * gst-libs/gst/audio/gstbaseaudiosrc.h:
28472 * gst-libs/gst/audio/gstringbuffer.h:
28473 * gst-libs/gst/rtp/gstbasertpdepayload.h:
28474 * gst-libs/gst/rtp/gstbasertppayload.h:
28475 * gst-libs/gst/video/gstvideofilter.h:
28476 * gst-libs/gst/video/gstvideosink.h:
28477 * gst/playback/gstplaybasebin.h:
28478 * gst/tcp/gstmultifdsink.h:
28479 * sys/v4l/gstv4lelement.h:
28480 Fix broken GObject macros
28482 2006-04-08 16:21:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28484 ext/alsa/gstalsasink.c: More debug to trace why my USB headset is not working with gst
28485 Original commit message from CVS:
28486 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
28487 More debug to trace why my USB headset is not working with gst
28489 2006-04-07 17:18:11 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28491 gst/playback/gstplaybasebin.c: Clean up our group elements properly in the case where it never got committed - it sti...
28492 Original commit message from CVS:
28493 * gst/playback/gstplaybasebin.c: (group_destroy):
28494 Clean up our group elements properly in the case where it never
28495 got committed - it still got added unconditionally to the bin.
28497 2006-04-07 15:14:32 +0000 Wim Taymans <wim.taymans@gmail.com>
28499 ext/theora/theoradec.c: Unref unhandled events.
28500 Original commit message from CVS:
28501 * ext/theora/theoradec.c: (theora_dec_sink_event),
28502 (theora_handle_data_packet), (theora_dec_chain):
28503 Unref unhandled events.
28504 Protect against empty buffers.
28505 Perform QoS on running time.
28507 2006-04-07 13:24:54 +0000 Michael Smith <msmith@xiph.org>
28509 ext/vorbis/vorbisenc.c: Remove leaks from vorbisenc.
28510 Original commit message from CVS:
28511 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
28512 (gst_vorbis_enc_chain):
28513 Remove leaks from vorbisenc.
28514 Mostly minor changes, the only significant one is that now the
28515 buffers we set as 'streamheader' on the caps are copies of the
28516 original buffers, to avoid circular refcounting problems.
28518 2006-04-07 09:51:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28520 gst/playback/gstplaybasebin.c: Don't remove our mute-probe if someone else already did so.
28521 Original commit message from CVS:
28522 * gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
28523 Don't remove our mute-probe if someone else already did so.
28524 Don't set a 2nd one if there is already one pending on the pad.
28525 * gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
28527 When a seek fails, ensure that playbin is still set back to playing.
28528 * gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
28529 (mpeg_ts_type_find), (plugin_init):
28530 Add a typefind function for mpeg-ts streams.
28532 2006-04-06 11:40:45 +0000 Andy Wingo <wingo@pobox.com>
28535 * gst/audiotestsrc/gstaudiotestsrc.c:
28536 * gst/videorate/gstvideorate.c:
28537 gst/videorate/gstvideorate.c (gst_video_rate_reset)
28538 Original commit message from CVS:
28539 2006-04-06 Andy Wingo <wingo@pobox.com>
28540 * gst/videorate/gstvideorate.c (gst_video_rate_reset)
28541 (gst_video_rate_init): Caps-related parameters should not be reset
28542 by a flush -- move their inits to the instance init function.
28543 (gst_video_rate_flush_prev): Don't complain if gst_pad_push
28544 is not OK, just return the result.
28545 * gst/audiotestsrc/gstaudiotestsrc.c
28546 (gst_audio_test_src_class_init)
28547 (gst_audio_test_src_get_times): Re-enable is-live=true, as was
28548 broken by Stefan's commit on 24 March.
28550 2006-04-06 10:50:14 +0000 Andy Wingo <wingo@pobox.com>
28552 ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on buffers being pushed out. Fixes oggmux ! multifdsink.
28553 Original commit message from CVS:
28554 2006-04-06 Andy Wingo <wingo@pobox.com>
28555 * ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
28556 buffers being pushed out. Fixes oggmux ! multifdsink.
28558 2006-04-05 13:05:25 +0000 Tim-Philipp Müller <tim@centricular.net>
28560 ext/vorbis/: Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make vorbisenc adhere to the official nomenclature; u...
28561 Original commit message from CVS:
28562 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
28563 (gst_vorbis_dec_init), (vorbis_dec_finalize):
28564 * ext/vorbis/vorbisdec.h:
28565 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_add_interfaces),
28566 (gst_vorbis_enc_base_init), (gst_vorbis_enc_class_init),
28567 (gst_vorbis_enc_sink_setcaps), (gst_vorbis_enc_convert_src),
28568 (gst_vorbis_enc_convert_sink), (gst_vorbis_enc_get_query_types),
28569 (gst_vorbis_enc_src_query), (gst_vorbis_enc_sink_query),
28570 (gst_vorbis_enc_init), (gst_vorbis_enc_get_tag_value),
28571 (gst_vorbis_enc_metadata_set1), (gst_vorbis_enc_set_metadata),
28572 (gst_vorbis_enc_setup), (gst_vorbis_enc_clear),
28573 (gst_vorbis_enc_buffer_from_packet),
28574 (gst_vorbis_enc_buffer_from_header_packet),
28575 (gst_vorbis_enc_push_buffer), (gst_vorbis_enc_push_packet),
28576 (gst_vorbis_enc_set_header_on_caps), (gst_vorbis_enc_sink_event),
28577 (gst_vorbis_enc_chain), (gst_vorbis_enc_output_buffers),
28578 (gst_vorbis_enc_get_property), (gst_vorbis_enc_set_property),
28579 (gst_vorbis_enc_change_state):
28580 * ext/vorbis/vorbisenc.h:
28581 Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make
28582 vorbisenc adhere to the official nomenclature; use boilerplate
28585 2006-04-04 11:20:24 +0000 Andy Wingo <wingo@pobox.com>
28587 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Whoops, fix bug introduced. Bad hacker!
28588 Original commit message from CVS:
28589 2006-04-04 Andy Wingo <wingo@pobox.com>
28590 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
28591 Whoops, fix bug introduced. Bad hacker!
28593 2006-04-04 11:15:00 +0000 Andy Wingo <wingo@pobox.com>
28595 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Properly handle the case where you get EOS before any buffe...
28596 Original commit message from CVS:
28597 2006-04-04 Andy Wingo <wingo@pobox.com>
28598 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
28599 Properly handle the case where you get EOS before any buffers are
28600 received. Use gst_buffer_make_metadata_writable where appropriate.
28602 2006-04-04 10:16:46 +0000 Andy Wingo <wingo@pobox.com>
28604 ext/theora/theoradec.c (theora_handle_data_packet): This value is often negative -- make it signed so as not to wrap ...
28605 Original commit message from CVS:
28606 2006-04-04 Andy Wingo <wingo@pobox.com>
28607 * ext/theora/theoradec.c (theora_handle_data_packet): This value
28608 is often negative -- make it signed so as not to wrap around.
28609 Fixes segfaults introduced on 9 March.
28611 2006-04-03 16:43:10 +0000 Wim Taymans <wim.taymans@gmail.com>
28613 ext/theora/: Don't try to store a gdouble in a gboolean.
28614 Original commit message from CVS:
28615 * ext/theora/gsttheoradec.h:
28616 * ext/theora/theoradec.c: (theora_dec_src_event):
28617 Don't try to store a gdouble in a gboolean.
28620 2006-04-03 12:55:18 +0000 Michael Smith <msmith@xiph.org>
28622 ext/ogg/gstoggmux.c: Oggmux sucks.
28623 Original commit message from CVS:
28624 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
28626 Make it suck slightly less by writing out the final page.
28627 Still can't encode a vorbis-in-ogg file correctly, though.
28629 2006-04-03 08:49:06 +0000 Andy Wingo <wingo@pobox.com>
28631 ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove a g_print.
28632 Original commit message from CVS:
28633 2006-04-03 Andy Wingo <wingo@pobox.com>
28634 * ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
28637 2006-04-03 08:32:21 +0000 Andy Wingo <wingo@pobox.com>
28639 ext/theora/theora.c (plugin_init): Register theoraparse.
28640 Original commit message from CVS:
28641 2006-04-03 Andy Wingo <wingo@pobox.com>
28642 * ext/theora/theora.c (plugin_init): Register theoraparse.
28643 * ext/theora/gsttheoraparse.h:
28644 * ext/theora/theoraparse.c: New files implementing a theora
28645 parser. Now we can properly remux ogg/theora+vorbis, yay.
28647 2006-04-03 08:28:58 +0000 Andy Wingo <wingo@pobox.com>
28649 ext/vorbis/vorbisparse.c: Add some docs and a copyright.
28650 Original commit message from CVS:
28651 2006-04-03 Andy Wingo <wingo@pobox.com>
28652 * ext/vorbis/vorbisparse.c: Add some docs and a copyright.
28654 2006-04-01 15:34:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28658 don't use AS_LIBTOOL_TAGS, it doesn't work
28659 Original commit message from CVS:
28660 don't use AS_LIBTOOL_TAGS, it doesn't work
28662 2006-04-01 11:41:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28665 * ext/pango/gsttextoverlay.c:
28666 * sys/v4l/gstv4lsrc.c:
28667 remove BT8x8 from description, works for more devices
28668 Original commit message from CVS:
28669 remove BT8x8 from description, works for more devices
28671 2006-04-01 11:21:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28673 gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798)
28674 Original commit message from CVS:
28675 * gst/audiotestsrc/gstaudiotestsrc.c:
28676 Fixed the sample pipeline (see #323798)
28678 2006-04-01 09:50:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28680 use AS_VERSION and AS_NANO more cleanups
28681 Original commit message from CVS:
28683 * win32/common/config.h:
28684 * win32/common/config.h.in:
28685 use AS_VERSION and AS_NANO
28688 2006-03-31 17:08:41 +0000 Andy Wingo <wingo@pobox.com>
28690 ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix uninitialized variable return that would happen.
28691 Original commit message from CVS:
28692 2006-03-31 Andy Wingo <wingo@pobox.com>
28693 * ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
28694 uninitialized variable return that would happen.
28696 2006-03-31 16:57:47 +0000 Andy Wingo <wingo@pobox.com>
28698 ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix uninitialized variable return that would never happen.
28699 Original commit message from CVS:
28700 2006-03-31 Andy Wingo <wingo@pobox.com>
28701 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
28702 uninitialized variable return that would never happen.
28704 2006-03-31 16:43:43 +0000 Andy Wingo <wingo@pobox.com>
28706 ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
28707 Original commit message from CVS:
28708 2006-03-31 Andy Wingo <wingo@pobox.com>
28709 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
28710 (vorbis_parse_sink_event): Add an event function to flush our
28711 state on a seek, and to drain buffers on a premature EOS.
28712 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
28713 (vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
28714 (vorbis_parse_chain, vorbis_parse_queue_buffer)
28715 (vorbis_parse_drain_queue): Queue up buffers until we can set
28716 their timestamps and granulepos values.
28717 * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
28718 and keep track of data needed for deriving granulepos and
28719 timestamps for buffers.
28721 2006-03-30 11:05:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28723 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
28724 * pkgconfig/gstreamer-plugins-base.pc.in:
28725 expose pluginsdir so gonlin can use it for tests
28726 Original commit message from CVS:
28727 expose pluginsdir so gonlin can use it for tests
28729 2006-03-30 10:03:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28731 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
28732 * pkgconfig/gstreamer-plugins-base.pc.in:
28733 add ccda to libraries
28734 Original commit message from CVS:
28735 add ccda to libraries
28737 2006-03-29 14:00:08 +0000 j^ <j@bootlab.org>
28739 better/unified long descriptions
28740 Original commit message from CVS:
28741 Patch by: j^ <j at bootlab dot org>
28742 * ext/alsa/gstalsamixerelement.c:
28743 (gst_alsa_mixer_element_class_init):
28744 * ext/alsa/gstalsasink.c:
28745 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
28746 * ext/ogg/gstoggdemux.c:
28747 * ext/ogg/gstoggmux.c:
28748 * ext/ogg/gstoggparse.c:
28749 * ext/pango/gstclockoverlay.c:
28750 * ext/pango/gsttextoverlay.c:
28751 * ext/pango/gsttextrender.c:
28752 * ext/pango/gsttimeoverlay.c:
28753 * ext/theora/theoradec.c:
28754 * ext/theora/theoraenc.c:
28755 * ext/vorbis/vorbisdec.c:
28756 * ext/vorbis/vorbisenc.c:
28757 * gst/audioconvert/gstaudioconvert.c:
28758 * gst/subparse/gstsubparse.c:
28759 * gst/tcp/gstmultifdsink.c:
28760 * gst/tcp/gsttcpclientsink.c:
28761 * gst/tcp/gsttcpclientsrc.c:
28762 * gst/tcp/gsttcpserversink.c:
28763 * gst/tcp/gsttcpserversrc.c:
28764 better/unified long descriptions
28767 2006-03-29 13:54:24 +0000 Wim Taymans <wim.taymans@gmail.com>
28769 tests/examples/seek/seek.c: Don't let double and tripple clicks mess up our state.
28770 Original commit message from CVS:
28771 * tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
28773 Don't let double and tripple clicks mess up our state.
28775 2006-03-28 13:13:43 +0000 Tim-Philipp Müller <tim@centricular.net>
28777 gst/playback/gstplaybin.c: Error out gracefully when we can't create any of the usual conversion elements for some re...
28778 Original commit message from CVS:
28779 * gst/playback/gstplaybin.c: (gen_video_element),
28780 (gen_text_element), (gen_audio_element), (gen_vis_element):
28781 Error out gracefully when we can't create any of the usual
28782 conversion elements for some reason. Also, don't try to
28783 create an audioscale (sic) element that's not used anyway.
28785 2006-03-28 10:21:52 +0000 Tim-Philipp Müller <tim@centricular.net>
28787 gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
28788 Original commit message from CVS:
28789 * gst/playback/gstplaybasebin.c: (setup_source):
28790 Don't post RESOURCE_NOT_FOUND error when we can't find a source
28791 element for a particular protocol, that's confusing for users.
28792 Instead, post a RESOURCE_FAILED error, so that our own error
28793 message is actually shown in totem etc. (#336303).
28795 2006-03-27 16:36:46 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
28797 ext/gnomevfs/gstgnomevfssrc.c: Fix some minor memory leaks (#336194).
28798 Original commit message from CVS:
28799 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
28800 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
28801 (gst_gnome_vfs_src_get_icy_metadata):
28802 Fix some minor memory leaks (#336194).
28804 2006-03-27 16:15:00 +0000 Tim-Philipp Müller <tim@centricular.net>
28806 ext/gnomevfs/: Make gnomevfssink accept filenames as well as URIs for the "location" property, just like gnomevfssrc ...
28807 Original commit message from CVS:
28808 * ext/gnomevfs/gstgnomevfs.c:
28809 (gst_gnome_vfs_location_to_uri_string):
28810 * ext/gnomevfs/gstgnomevfs.h:
28811 * ext/gnomevfs/gstgnomevfssink.c:
28812 (gst_gnome_vfs_sink_set_property):
28813 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
28814 Make gnomevfssink accept filenames as well as URIs for the
28815 "location" property, just like gnomevfssrc does (and
28816 filesrc/filesink do) (#336190).
28818 2006-03-24 20:35:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28820 tests/check/generic/clock-selection.c: set to NULL before unreffing, fixes a valgrind leak.
28821 Original commit message from CVS:
28822 * tests/check/generic/clock-selection.c: (GST_START_TEST):
28823 set to NULL before unreffing, fixes a valgrind leak.
28824 Why was this not triggering the error that an object needs to
28825 be NULL before unreffing ?
28826 * win32/common/config.h:
28829 2006-03-24 17:57:39 +0000 Tim-Philipp Müller <tim@centricular.net>
28831 gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
28832 Original commit message from CVS:
28833 * gst/subparse/gstsubparse.c: (convert_encoding),
28834 (gst_sub_parse_change_state):
28835 * gst/subparse/gstsubparse.h:
28836 Text subtitle files may or may not be UTF-8. If it's not, we
28837 don't really want to see '?' characters in place of non-ASCII
28838 characters like accented characters. So let's assume the input
28839 is UTF-8 until we come across text that is clearly not. If it's
28840 not UTF-8, we don't really know what it is, so try the following:
28841 (a) see whether the GST_SUBTITLE_ENCODING environment variable
28842 is set; if not, check (b) if the current locale encoding is
28843 non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
28844 the current locale encoding is UTF-8 and the environment variable
28845 was not set to any particular encoding. Not perfect, but better
28846 than nothing (and better than before, I think) (fixes #172848).
28848 2006-03-24 17:39:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28850 * docs/plugins/tmpl/.gitignore:
28851 * tests/check/libs/.gitignore:
28852 * tests/check/pipelines/.gitignore:
28853 * tests/examples/volume/.gitignore:
28855 Original commit message from CVS:
28858 2006-03-24 17:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28860 configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
28861 Original commit message from CVS:
28862 2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
28864 update core requirement to 0.10.4.1 because of async_playback
28865 vmethod on GstBaseSink
28867 2006-03-24 17:11:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28869 use DEBUG_FUNCPTR for collectpads
28870 Original commit message from CVS:
28871 * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
28872 * gst/adder/gstadder.c: (gst_adder_init):
28873 use DEBUG_FUNCPTR for collectpads
28875 2006-03-24 14:11:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28878 don't go through check-torture if no check installed
28879 Original commit message from CVS:
28880 don't go through check-torture if no check installed
28882 2006-03-24 10:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28884 Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
28885 Original commit message from CVS:
28886 * docs/plugins/Makefile.am:
28887 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
28888 * docs/plugins/gst-plugins-base-plugins-sections.txt:
28889 * ext/cdparanoia/gstcdparanoiasrc.c:
28890 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
28891 (gst_gnome_vfs_sink_class_init):
28892 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
28893 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
28894 * ext/ogg/gstoggmux.c:
28895 * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
28896 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
28897 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
28898 * ext/pango/gsttextoverlay.c:
28899 * ext/pango/gsttextrender.c:
28900 * ext/theora/theoradec.c:
28901 * ext/theora/theoraenc.c:
28902 * ext/vorbis/vorbisdec.c:
28903 * ext/vorbis/vorbisenc.c:
28904 * gst-libs/gst/audio/gstaudiofilter.c:
28905 (gst_audio_filter_base_init):
28906 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
28907 (gst_audio_filter_template_base_init):
28908 * gst/adder/gstadder.c: (gst_adder_get_type):
28909 * gst/adder/gstadder.h:
28910 * gst/audioconvert/gstaudioconvert.c:
28911 * gst/audiotestsrc/gstaudiotestsrc.c:
28912 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
28913 (gst_audio_test_src_create):
28914 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
28915 * gst/playback/gstdecodebin.c:
28916 * gst/playback/gstplaybin.c:
28917 * gst/playback/gststreamselector.c:
28918 (gst_stream_selector_base_init):
28919 * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
28920 * gst/volume/gstvolume.c:
28921 * sys/v4l/gstv4lmjpegsink.c:
28922 * sys/v4l/gstv4lmjpegsrc.c:
28923 * tests/check/libs/cddabasesrc.c:
28924 * tests/old/examples/gob/gst-identity2.gob:
28925 Add docs for adder, use GST_ELEMENT_DETAILS macro,
28926 define GstElementDetails at the top
28928 2006-03-23 21:48:18 +0000 Sébastien Moutte <sebastien@moutte.net>
28930 win32/common/libgstinterfaces.def: Add a lot of export functions for gst-python
28931 Original commit message from CVS:
28932 * win32/common/libgstinterfaces.def:
28933 Add a lot of export functions for gst-python
28934 * win32/common/libgstinterfaces.dsp:
28935 Add a missing include folder in the project configuration
28937 2006-03-23 16:58:03 +0000 Wim Taymans <wim.taymans@gmail.com>
28939 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
28940 Original commit message from CVS:
28941 * gst-libs/gst/audio/gstbaseaudiosrc.c:
28942 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
28943 (gst_base_audio_src_change_state):
28944 Fix audio sources, forgot to make the ringbuffer
28947 2006-03-23 16:29:58 +0000 Wim Taymans <wim.taymans@gmail.com>
28949 gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
28950 Original commit message from CVS:
28951 * gst-libs/gst/audio/gstbaseaudiosrc.c:
28952 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
28953 (gst_base_audio_src_change_state):
28954 unparent instead of unref the ringbuffer.
28956 2006-03-23 16:24:23 +0000 Wim Taymans <wim.taymans@gmail.com>
28958 gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
28959 Original commit message from CVS:
28960 * gst-libs/gst/audio/gstbaseaudiosink.c:
28961 (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
28962 (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
28963 Implement new async_play vmethod to start slaving and allow
28964 playback start in case of async PLAY state changes.
28965 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
28966 Enable QoS with new method in base class.
28968 2006-03-23 11:18:19 +0000 Julien MOUTTE <julien@moutte.net>
28970 gst/videotestsrc/gstvideotestsrc.c: Partially handle 0 framerate, only EOS after the first frame is missing.
28971 Original commit message from CVS:
28972 Patch by: Julien MOUTTE <julien at moutte dot net>
28973 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
28974 (gst_video_test_src_do_seek), (gst_video_test_src_create):
28975 Partially handle 0 framerate, only EOS after the first frame
28978 2006-03-23 09:38:59 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
28980 gst/: Patch for support of YVU9 AVI files (#334822)
28981 Original commit message from CVS:
28982 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
28983 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
28984 (gst_riff_create_video_template_caps):
28985 * gst/ffmpegcolorspace/avcodec.h:
28986 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
28987 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
28988 (gst_ffmpegcsp_avpicture_fill):
28989 * gst/ffmpegcolorspace/imgconvert.c:
28990 Patch for support of YVU9 AVI files (#334822)
28992 2006-03-22 15:29:25 +0000 Edward Hervey <bilboed@bilboed.com>
28994 docs/design/design-decodebin.txt: Added design document for new decodebin text/x-pango-markup is also a default targe...
28995 Original commit message from CVS:
28996 * docs/design/design-decodebin.txt:
28997 Added design document for new decodebin
28998 (Target Caps): text/x-pango-markup is also a default target caps.
29000 2006-03-22 15:11:47 +0000 Edward Hervey <bilboed@bilboed.com>
29002 docs/design/design-decodebin.txt: Added design document for new decodebin
29003 Original commit message from CVS:
29004 * docs/design/design-decodebin.txt:
29005 Added design document for new decodebin
29007 2006-03-22 12:33:09 +0000 Wim Taymans <wim.taymans@gmail.com>
29009 gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
29010 Original commit message from CVS:
29011 * gst-libs/gst/audio/gstbaseaudiosink.c:
29012 (gst_base_audio_sink_dispose):
29013 Since we _parent the ringbuffer, we also need to
29014 _unparent instead of a plain _unref.
29016 2006-03-22 12:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
29018 tests/examples/seek/seek.c: Add scrub checkbox.
29019 Original commit message from CVS:
29020 * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
29021 (stop_seek), (scrub_toggle_cb), (main):
29022 Add scrub checkbox.
29024 2006-03-21 17:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
29026 ext/ogg/gstoggparse.c: Fix very inefficient usage of linked lists (#335365).
29027 Original commit message from CVS:
29028 * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
29029 (gst_ogg_parse_chain):
29030 Fix very inefficient usage of linked lists (#335365).
29032 2006-03-21 14:26:01 +0000 Edward Hervey <bilboed@bilboed.com>
29034 gcc 4.1 unreferenced pointer fixes.
29035 Original commit message from CVS:
29036 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
29037 * gst/playback/gstplaybin.c: (handoff):
29038 * gst/playback/gststreamselector.c:
29039 (gst_stream_selector_set_property):
29040 gcc 4.1 unreferenced pointer fixes.
29041 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
29042 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
29043 gst_buffer_ref() now takes a GstBuffer*.
29045 2006-03-20 18:09:41 +0000 Julien Moutte <julien@moutte.net>
29047 sys/xvimage/xvimagesink.c: Fix a memleak reported by Jan Schmidt.
29048 Original commit message from CVS:
29049 2006-03-20 Julien MOUTTE <julien@moutte.net>
29050 * sys/xvimage/xvimagesink.c:
29051 (gst_xvimagesink_get_format_from_caps): Fix a memleak reported
29054 2006-03-19 11:37:46 +0000 Tim-Philipp Müller <tim@centricular.net>
29056 gst/typefind/gsttypefindfunctions.c: Can't do tag preferences via probability, as tags would then lose against types ...
29057 Original commit message from CVS:
29058 * gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
29059 (id3v1_type_find), (apetag_type_find), (plugin_init):
29060 Can't do tag preferences via probability, as tags would then
29061 lose against types that are recognised with MAXIMUM probability
29062 (like .wav); so let all tag typefinders return MAXIMUM themselves
29063 and order them via the rank. Split ID3v1 and ID3v2 typefinders so
29064 that we can prefer APE to ID3v1 (fixes #335028).
29066 2006-03-17 17:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
29068 gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
29069 Original commit message from CVS:
29070 * gst-libs/gst/audio/gstbaseaudiosink.c:
29071 (gst_base_audio_sink_change_state):
29072 * gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
29073 (gst_ring_buffer_may_start):
29074 * gst-libs/gst/audio/gstringbuffer.h:
29075 Only start playback if we are playing.
29076 should fix #330748.
29078 2006-03-17 13:11:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29080 Revert accidental commits to these files.
29081 Original commit message from CVS:
29082 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
29083 * win32/common/config.h:
29084 Revert accidental commits to these files.
29086 2006-03-16 20:01:03 +0000 Michal Benes <michal.benes@xeris.cz>
29088 tests/Makefile.am: Don't try to build tests in tests/icles if we don't have X (#323852)
29089 Original commit message from CVS:
29090 Patch by: Michal Benes <michal dot benes at xeris dot cz>
29091 * tests/Makefile.am:
29092 Don't try to build tests in tests/icles if we
29093 don't have X (#323852)
29095 2006-03-16 13:08:01 +0000 Tim-Philipp Müller <tim@centricular.net>
29097 gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
29098 Original commit message from CVS:
29099 * gst-libs/gst/tag/gstid3tag.c:
29100 Add TXXX frame identifiers for replaygain stuff as used
29101 by some taggers (see #323721).
29103 2006-03-16 10:22:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29105 gst/playback/gststreamselector.c: Preserve the existing buggy streamselector behaviour by performing a fallback buffe...
29106 Original commit message from CVS:
29107 * gst/playback/gststreamselector.c:
29108 (gst_stream_selector_set_property),
29109 (gst_stream_selector_bufferalloc):
29110 Preserve the existing buggy streamselector behaviour by performing
29111 a fallback buffer allocation when downstream isn't linked yet.
29112 This should really be fixed in playbin by blocking pads until it's
29114 Also, use gst_pad_alloc_buffer instead of
29115 gst_pad_alloc_buffer_and_set.
29117 2006-03-15 22:40:08 +0000 Tim-Philipp Müller <tim@centricular.net>
29119 gst-libs/gst/tag/gstid3tag.c: Don't crash on unknown ID3v2 TXXX frames.
29120 Original commit message from CVS:
29121 * gst-libs/gst/tag/gstid3tag.c:
29122 Don't crash on unknown ID3v2 TXXX frames.
29124 2006-03-15 17:59:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29126 ext/alsa/gstalsasink.c: Chain up to the parent finalize method.
29127 Original commit message from CVS:
29128 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
29129 Chain up to the parent finalize method.
29130 Add 32-bit sample size to the template caps.
29131 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
29132 (gst_riff_create_video_template_caps):
29133 Add the fourcc that the VMWare codec uses.
29134 * gst/playback/gststreamselector.c:
29135 (gst_stream_selector_set_property),
29136 (gst_stream_selector_bufferalloc),
29137 (gst_stream_selector_request_new_pad):
29138 For the active pad, forward buffer-alloc requests, otherwise
29139 return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
29140 having to memcpy every frame when used by playbin.
29141 * gst/tcp/gstmultifdsink.c:
29142 (gst_multi_fd_sink_handle_client_write):
29143 Get negotiated caps from the sink pad, rather than the sink
29146 2006-03-15 17:11:34 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
29148 ext/gnomevfs/gstgnomevfssrc.c: Don't forget to set src->callbacks_pushed to FALSE again when popping them, otherwise ...
29149 Original commit message from CVS:
29150 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
29151 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_pop_callbacks):
29152 Don't forget to set src->callbacks_pushed to FALSE again when
29153 popping them, otherwise re-activation in a different mode won't
29156 2006-03-15 11:30:29 +0000 Sebastien Moutte <sebastien@moutte.net>
29158 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
29159 Original commit message from CVS:
29160 Patch by: Sebastien Moutte <sebastien moutte net>
29161 * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
29162 (gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
29163 (gst_ffmpeg_smpfmt_to_caps):
29164 Replace __VA_ARGS__ caps creation macros with varargs functions.
29165 Makes things compile on MSVC (#320765), looks nicer, and we can
29166 tell the compiler to check for the NULL terminator.
29168 2006-03-14 15:13:04 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
29170 gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
29171 Original commit message from CVS:
29172 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
29173 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29174 Make sure the buffer we copy into is really always big
29175 enough, this time for real (#333488).
29177 2006-03-14 13:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
29179 gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
29180 Original commit message from CVS:
29181 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29182 Add support for 24bpp DIB (#305279).
29184 2006-03-14 11:11:59 +0000 Wim Taymans <wim.taymans@gmail.com>
29186 gst/: Re-enable QoS after the release.
29187 Original commit message from CVS:
29188 * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
29189 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
29190 * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
29191 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
29192 (gst_video_scale_init), (gst_video_scale_src_event):
29193 Re-enable QoS after the release.
29194 Rework videoscale to use the base class src_event handler.
29196 2006-03-14 09:51:01 +0000 Tim-Philipp Müller <tim@centricular.net>
29198 configure.ac: back to CVS.
29199 Original commit message from CVS:
29203 === release 0.10.5 ===
29205 2006-03-13 19:50:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29211 * docs/plugins/inspect/plugin-adder.xml:
29212 * docs/plugins/inspect/plugin-alsa.xml:
29213 * docs/plugins/inspect/plugin-audioconvert.xml:
29214 * docs/plugins/inspect/plugin-audiorate.xml:
29215 * docs/plugins/inspect/plugin-audioresample.xml:
29216 * docs/plugins/inspect/plugin-audiotestsrc.xml:
29217 * docs/plugins/inspect/plugin-cdparanoia.xml:
29218 * docs/plugins/inspect/plugin-decodebin.xml:
29219 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
29220 * docs/plugins/inspect/plugin-gnomevfs.xml:
29221 * docs/plugins/inspect/plugin-libvisual.xml:
29222 * docs/plugins/inspect/plugin-ogg.xml:
29223 * docs/plugins/inspect/plugin-pango.xml:
29224 * docs/plugins/inspect/plugin-playbin.xml:
29225 * docs/plugins/inspect/plugin-subparse.xml:
29226 * docs/plugins/inspect/plugin-tcp.xml:
29227 * docs/plugins/inspect/plugin-theora.xml:
29228 * docs/plugins/inspect/plugin-typefindfunctions.xml:
29229 * docs/plugins/inspect/plugin-video4linux.xml:
29230 * docs/plugins/inspect/plugin-videorate.xml:
29231 * docs/plugins/inspect/plugin-videoscale.xml:
29232 * docs/plugins/inspect/plugin-videotestsrc.xml:
29233 * docs/plugins/inspect/plugin-volume.xml:
29234 * docs/plugins/inspect/plugin-vorbis.xml:
29235 * docs/plugins/inspect/plugin-ximagesink.xml:
29236 * docs/plugins/inspect/plugin-xvimagesink.xml:
29237 * win32/common/config.h:
29239 Original commit message from CVS:
29242 2006-03-13 17:28:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29259 Original commit message from CVS:
29262 2006-03-13 11:17:19 +0000 Tim-Philipp Müller <tim@centricular.net>
29264 docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
29265 Original commit message from CVS:
29266 * docs/plugins/Makefile.am:
29267 Part of previous cdparanoiasrc docs fixes, forgot to commit.
29269 2006-03-12 14:56:31 +0000 Tim-Philipp Müller <tim@centricular.net>
29271 docs/plugins/: Add cdparanoiasrc to docs.
29272 Original commit message from CVS:
29273 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29274 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29275 * docs/plugins/gst-plugins-base-plugins.hierarchy:
29276 Add cdparanoiasrc to docs.
29277 * gst-libs/gst/cdda/gstcddabasesrc.c:
29278 More GstCddaBaseSrc docs.
29280 2006-03-12 13:47:22 +0000 Tim-Philipp Müller <tim@centricular.net>
29282 Add new API to libgsttag: gst_tag_from_id3_user_tag().
29283 Original commit message from CVS:
29284 * docs/libs/gst-plugins-base-libs-sections.txt:
29285 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
29286 * gst-libs/gst/tag/tag.h:
29287 Add new API to libgsttag: gst_tag_from_id3_user_tag().
29289 2006-03-11 19:47:16 +0000 Tim-Philipp Müller <tim@centricular.net>
29291 gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
29292 Original commit message from CVS:
29293 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29294 NULL-terminate array of mpeg4 video file extensions.
29295 Fixes crash on PPC (#334226).
29297 2006-03-11 16:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
29299 ext/gnomevfs/gstgnomevfssrc.c: gnome_vfs_uri_is_local() alone is not a good indicator whether we can operate in pull-...
29300 Original commit message from CVS:
29301 * ext/gnomevfs/gstgnomevfssrc.c:
29302 (gst_gnome_vfs_src_check_get_range):
29303 gnome_vfs_uri_is_local() alone is not a good indicator
29304 whether we can operate in pull-mode with a specific URI,
29305 as it returns FALSE for file:// URIs that point to an
29306 NFS-mounted path. Be more conservative here: whitelist
29307 local files, blacklist http URIs and use the old
29308 mechanism for anything else (fixes #334216).
29310 2006-03-10 19:15:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29312 configure.ac: back to trunk
29313 Original commit message from CVS:
29317 === release 0.10.4 ===
29319 2006-03-10 19:05:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29325 * docs/plugins/gst-plugins-base-plugins.args:
29326 * docs/plugins/inspect/plugin-adder.xml:
29327 * docs/plugins/inspect/plugin-alsa.xml:
29328 * docs/plugins/inspect/plugin-audioconvert.xml:
29329 * docs/plugins/inspect/plugin-audiorate.xml:
29330 * docs/plugins/inspect/plugin-audioresample.xml:
29331 * docs/plugins/inspect/plugin-audiotestsrc.xml:
29332 * docs/plugins/inspect/plugin-cdparanoia.xml:
29333 * docs/plugins/inspect/plugin-decodebin.xml:
29334 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
29335 * docs/plugins/inspect/plugin-gnomevfs.xml:
29336 * docs/plugins/inspect/plugin-libvisual.xml:
29337 * docs/plugins/inspect/plugin-ogg.xml:
29338 * docs/plugins/inspect/plugin-pango.xml:
29339 * docs/plugins/inspect/plugin-playbin.xml:
29340 * docs/plugins/inspect/plugin-subparse.xml:
29341 * docs/plugins/inspect/plugin-tcp.xml:
29342 * docs/plugins/inspect/plugin-theora.xml:
29343 * docs/plugins/inspect/plugin-typefindfunctions.xml:
29344 * docs/plugins/inspect/plugin-video4linux.xml:
29345 * docs/plugins/inspect/plugin-videorate.xml:
29346 * docs/plugins/inspect/plugin-videoscale.xml:
29347 * docs/plugins/inspect/plugin-videotestsrc.xml:
29348 * docs/plugins/inspect/plugin-volume.xml:
29349 * docs/plugins/inspect/plugin-vorbis.xml:
29350 * docs/plugins/inspect/plugin-ximagesink.xml:
29351 * docs/plugins/inspect/plugin-xvimagesink.xml:
29353 * win32/common/config.h:
29355 Original commit message from CVS:
29358 2006-03-10 12:37:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29360 gst-libs/gst/video/gstvideosink.c: Disable max-lateness by setting it to -1 for now, so that we can bed QoS stuff in ...
29361 Original commit message from CVS:
29362 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
29363 Disable max-lateness by setting it to -1 for now, so that
29364 we can bed QoS stuff in thoroughly between now and the next
29367 2006-03-10 11:09:23 +0000 Fabrizio <fabrizio.ge@tiscali.it>
29369 gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
29370 Original commit message from CVS:
29371 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29372 Make sure we don't read beyond the palette buffer in case of
29373 broken or manipulated files (#333488, patch by: Fabrizio
29376 2006-03-10 10:44:02 +0000 Edward Hervey <bilboed@bilboed.com>
29378 gst/typefind/gsttypefindfunctions.c: Fix for variable not initialized.
29379 Original commit message from CVS:
29380 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
29381 Fix for variable not initialized.
29383 2006-03-09 19:02:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29386 * docs/libs/tmpl/gstringbuffer.sgml:
29401 * win32/common/config.h:
29403 Original commit message from CVS:
29406 2006-03-09 17:58:00 +0000 Wim Taymans <wim.taymans@gmail.com>
29408 ext/libvisual/visual.c: Small cleanups.
29409 Original commit message from CVS:
29410 * ext/libvisual/visual.c: (gst_visual_get_type),
29411 (gst_visual_src_setcaps), (gst_vis_src_negotiate),
29412 (gst_visual_chain):
29414 * ext/theora/gsttheoradec.h:
29415 * ext/theora/theoradec.c: (gst_theora_dec_init),
29416 (gst_theora_dec_reset), (_theora_granule_time),
29417 (theora_dec_src_convert), (theora_dec_sink_convert),
29418 (theora_dec_src_query), (theora_dec_src_event),
29419 (theora_dec_sink_event), (theora_handle_comment_packet),
29420 (theora_handle_header_packet), (theora_dec_push),
29421 (theora_handle_data_packet), (theora_dec_chain),
29422 (theora_dec_change_state):
29425 2006-03-09 17:50:59 +0000 Wim Taymans <wim.taymans@gmail.com>
29427 ext/gnomevfs/gstgnomevfssrc.c: Some cleanups.
29428 Original commit message from CVS:
29429 * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
29430 (audiocast_register_listener), (gst_gnome_vfs_src_start):
29433 2006-03-09 17:45:39 +0000 Wim Taymans <wim.taymans@gmail.com>
29435 ext/ogg/gstoggdemux.c: Don't try to activate NULL chains.
29436 Original commit message from CVS:
29437 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
29438 Don't try to activate NULL chains.
29440 2006-03-09 16:30:41 +0000 Tim-Philipp Müller <tim@centricular.net>
29442 gst/typefind/gsttypefindfunctions.c: Fix invalid memory access to region before peek'd data (#332964).
29443 Original commit message from CVS:
29444 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
29445 Fix invalid memory access to region before peek'd data (#332964).
29447 2006-03-09 15:05:03 +0000 Christophe Fergeau <teuf@gnome.org>
29450 Original commit message from CVS:
29451 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init):
29452 * ext/pango/gsttextrender.c: (gst_text_render_init):
29453 * gst/adder/gstadder.c: (gst_adder_init):
29454 Don't leak padtemplates, patch by Christophe Fergeau,
29457 2006-03-09 12:56:35 +0000 Tim-Philipp Müller <tim@centricular.net>
29459 gst/subparse/gstsubparse.c: Fix invalid memory access: make sure string passed to regexec() is NUL-termianted.
29460 Original commit message from CVS:
29461 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
29462 Fix invalid memory access: make sure string passed to
29463 regexec() is NUL-termianted.
29465 2006-03-09 12:37:59 +0000 Tim-Philipp Müller <tim@centricular.net>
29467 gst/typefind/gsttypefindfunctions.c: Refactor mpeg/audio typefinding to make it more maintainable and easier to fine-...
29468 Original commit message from CVS:
29469 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
29471 Refactor mpeg/audio typefinding to make it more maintainable
29472 and easier to fine-tune. Make probing into middle of the file
29473 work properly (fixes #333900, also see #152688).
29475 2006-03-09 11:10:03 +0000 Tim-Philipp Müller <tim@centricular.net>
29477 gst/typefind/gsttypefindfunctions.c: Remove part from previous commit that was bogus: g_utf8_validate() does in fact ...
29478 Original commit message from CVS:
29479 * gst/typefind/gsttypefindfunctions.c:
29480 (utf8_type_find_have_valid_utf8_at_offset):
29481 Remove part from previous commit that was bogus:
29482 g_utf8_validate() does in fact not accept embedded
29483 zeroes, so we don't need to check for those (thanks
29484 to Mike for the hint).
29486 2006-03-08 17:11:29 +0000 Tim-Philipp Müller <tim@centricular.net>
29488 gst/typefind/gsttypefindfunctions.c: Make plain/text typefinder more conservative: firstly, check for embedded zeroes...
29489 Original commit message from CVS:
29490 * gst/typefind/gsttypefindfunctions.c:
29491 (utf8_type_find_count_embedded_zeroes),
29492 (utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
29493 Make plain/text typefinder more conservative: firstly, check
29494 for embedded zeroes, which are perfectly valid UTF-8 characters,
29495 but also a fairly good sign that something is not a plain text
29496 file; secondly, probe into the middle of the file if possible.
29497 If we can't probe into the middle, limit the probability value
29498 to be returned to TYPE_FIND_POSSIBLE (see #333900).
29500 2006-03-08 11:34:45 +0000 Michael Smith <msmith@xiph.org>
29502 gst/typefind/gsttypefindfunctions.c: Make typefind function name for mpeg4 video unique.
29503 Original commit message from CVS:
29504 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29505 Make typefind function name for mpeg4 video unique.
29507 2006-03-08 09:53:31 +0000 Wim Taymans <wim.taymans@gmail.com>
29509 ext/libvisual/visual.c: Cleanups, post nice errors.
29510 Original commit message from CVS:
29511 * ext/libvisual/visual.c: (gst_visual_init),
29512 (gst_visual_clear_actors), (gst_visual_dispose),
29513 (gst_visual_reset), (gst_visual_src_setcaps),
29514 (gst_visual_sink_setcaps), (gst_vis_src_negotiate),
29515 (gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
29516 (gst_visual_chain), (gst_visual_change_state):
29517 Cleanups, post nice errors.
29518 Handle sink and src events.
29519 Implement simple QoS.
29520 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
29521 Use new basesink methods to configure max-lateness.
29523 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
29524 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
29525 Debug statement cleanups.
29526 * gst/volume/gstvolume.c: (gst_volume_class_init):
29529 2006-03-08 09:50:23 +0000 Tim-Philipp Müller <tim@centricular.net>
29531 ext/pango/gsttextoverlay.c: Revert API/ABI break from March 1. Keep 'halign' and 'valign' as string type properties, ...
29532 Original commit message from CVS:
29533 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
29534 (gst_text_overlay_init), (gst_text_overlay_set_property),
29535 (gst_text_overlay_get_property):
29536 Revert API/ABI break from March 1. Keep 'halign' and 'valign'
29537 as string type properties, but mark them deprecated. Add
29538 'halignment' and 'valignment' properties that use enums
29539 instead of strings.
29541 2006-03-08 09:37:12 +0000 Fabrizio <fabrizio.ge@tiscali.it>
29543 gst-libs/gst/riff/riff-media.c: Allow palettes with less than 256 colours in AVI files
29544 Original commit message from CVS:
29545 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29546 Allow palettes with less than 256 colours in AVI files
29547 (#333488, patch by: Fabrizio Gennari).
29549 2006-03-07 21:56:09 +0000 Julien Moutte <julien@moutte.net>
29551 ext/pango/gsttextoverlay.c: Fix wrong EOS handling on text pad. We were releasing the queued text buffer when we shou...
29552 Original commit message from CVS:
29553 2006-03-07 Julien MOUTTE <julien@moutte.net>
29554 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
29555 (gst_text_overlay_video_event): Fix wrong EOS handling on text
29556 pad. We were releasing the queued text buffer when we should keep
29557 it until video pad gets EOS or discard the text buffer because it's
29558 too old. That was eating the last subtitle buffer. Add some more
29561 2006-03-07 17:28:36 +0000 Tim-Philipp Müller <tim@centricular.net>
29563 ext/pango/gsttextoverlay.c: Fix invalid memory access (we can't access a buffer after it's been pushed downstream wit...
29564 Original commit message from CVS:
29565 * ext/pango/gsttextoverlay.c: (gst_text_overlay_render_text),
29566 (gst_text_overlay_video_chain):
29567 Fix invalid memory access (we can't access a buffer after it's been
29568 pushed downstream without taking a reference); fix memory leak (if
29569 there's no text to render, bail out before allocating stuff).
29571 2006-03-07 15:08:15 +0000 Tim-Philipp Müller <tim@centricular.net>
29573 ext/pango/gsttextoverlay.*: If input is plain text, escape it before passing it to pango_layout_set_markup().
29574 Original commit message from CVS:
29575 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
29576 (gst_text_overlay_setcaps_txt), (gst_text_overlay_video_chain):
29577 * ext/pango/gsttextoverlay.h:
29578 If input is plain text, escape it before passing it to
29579 pango_layout_set_markup().
29581 2006-03-07 13:01:21 +0000 Tim-Philipp Müller <tim@centricular.net>
29583 gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
29584 Original commit message from CVS:
29585 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
29586 Don't ignore flow return from gst_pad_push().
29588 2006-03-07 12:49:03 +0000 Christophe Fergeau <teuf@gnome.org>
29590 Don't leak references returned by gst_pad_get_parent()
29591 Original commit message from CVS:
29592 * ext/libvisual/visual.c: (gst_visual_getcaps),
29593 (gst_visual_src_setcaps), (gst_visual_sink_setcaps):
29594 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
29595 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
29596 (gst_vorbisenc_convert_sink):
29597 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
29598 (gst_audio_duration_from_pad_buffer):
29599 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
29600 (gst_audio_filter_chain):
29601 * gst-libs/gst/rtp/gstbasertpdepayload.c:
29602 (gst_base_rtp_depayload_setcaps):
29603 * gst-libs/gst/video/video.c: (gst_video_frame_rate),
29604 (gst_video_get_size):
29605 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
29606 Don't leak references returned by gst_pad_get_parent()
29607 (#333663, based on patch by: Christophe Fergeau).
29609 2006-03-06 20:52:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29611 ext/gnomevfs/gstgnomevfssink.c: change location param details
29612 Original commit message from CVS:
29613 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
29614 change location param details
29615 * gst/volume/gstvolume.c: (plugin_init):
29616 correct plugin description
29618 2006-03-06 20:07:55 +0000 Tim-Philipp Müller <tim@centricular.net>
29620 ext/gnomevfs/gstgnomevfssrc.c: Override GstBaseSrc::check_get_range() in order to avoid opening the resource just to ...
29621 Original commit message from CVS:
29622 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
29623 (gst_gnome_vfs_src_check_get_range):
29624 Override GstBaseSrc::check_get_range() in order to avoid opening
29625 the resource just to check whether we can operate in pull-mode or
29626 not - we can predict that pretty well from the URI alone. Should
29627 fix problems with last.fm (#331690). (Requires latest core CVS).
29629 2006-03-06 16:18:51 +0000 Wim Taymans <wim.taymans@gmail.com>
29631 gst-libs/gst/video/gstvideosink.c: Throw away frames that are later than 20 ms.
29632 Original commit message from CVS:
29633 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
29634 (gst_video_sink_class_init):
29635 Throw away frames that are later than 20 ms.
29637 2006-03-06 14:14:47 +0000 Fabrizio <fabrizio.ge@tiscali.it>
29639 gst-libs/gst/riff/riff-media.c:
29640 Original commit message from CVS:
29641 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
29642 Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).
29644 2006-03-05 23:39:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29646 ext/ogg/gstoggmux.c: put Theora BOS pages before others. This hardcodes the Ogg/Theora I profile, but hey.
29647 Original commit message from CVS:
29648 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
29649 (gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
29650 put Theora BOS pages before others. This hardcodes
29651 the Ogg/Theora I profile, but hey.
29653 2006-03-05 23:06:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29655 * ext/ogg/gstoggmux.c:
29656 changed more than 5 lines
29657 Original commit message from CVS:
29658 changed more than 5 lines
29660 2006-03-05 22:57:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29662 ogg muxing of vorbis and theora now has pages ordered correctly again, even with delays.
29663 Original commit message from CVS:
29664 ogg muxing of vorbis and theora now has pages ordered correctly again,
29667 updated with some examples
29668 * ext/theora/theoraenc.c: (granulepos_to_timestamp),
29669 (granulepos_add), (theora_buffer_from_packet):
29670 * ext/vorbis/vorbisenc.c: (granulepos_to_timestamp_offset),
29671 (granulepos_to_timestamp), (gst_vorbisenc_buffer_from_packet),
29672 (gst_vorbisenc_chain):
29673 implement strategy from ext/ogg/README
29674 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
29675 (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
29676 (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_compare_pads),
29677 (gst_ogg_mux_queue_pads), (gst_ogg_mux_collected):
29678 Fix muxer so that oggz-validate is happy with all streams;
29679 except for no eos mark, and the BOS page ordering
29680 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
29681 (check_buffer_granulepos):
29682 * tests/check/pipelines/vorbisenc.c: (check_buffer_granulepos):
29683 update tests to check for OFFSET being set as requested
29684 fixed type of granulepos, it's not a ClockTime
29686 2006-03-05 21:34:23 +0000 Julien Moutte <julien@moutte.net>
29688 sys/xvimage/xvimagesink.c: Check that the xvimage we are creating has a correct size before returning it. (#3...
29689 Original commit message from CVS:
29690 2006-03-05 Julien MOUTTE <julien@moutte.net>
29691 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
29692 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
29693 Check that the xvimage we are creating has a correct size before returning it. (#314897)
29695 2006-03-05 13:44:05 +0000 Tim-Philipp Müller <tim@centricular.net>
29697 gst/typefind/gsttypefindfunctions.c: Give id3 and ape tag typefinders a rank slightly higher than PRIMARY to ensure t...
29698 Original commit message from CVS:
29699 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29700 Give id3 and ape tag typefinders a rank slightly higher
29701 than PRIMARY to ensure they're always run before any of
29702 the other typefinders (in particular wav and mp3) (#324186).
29704 2006-03-05 13:08:37 +0000 Tim-Philipp Müller <tim@centricular.net>
29706 gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
29707 Original commit message from CVS:
29708 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29709 Add support for '3IVD' fourcc (#333403).
29711 2006-03-04 14:35:10 +0000 Tim-Philipp Müller <tim@centricular.net>
29713 configure.ac: Bump requirements to GStreamer CVS for the new error enum.
29714 Original commit message from CVS:
29716 Bump requirements to GStreamer CVS for the new error enum.
29717 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
29718 Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
29719 space left on the device (fixes #333352).
29721 2006-03-03 23:53:50 +0000 Sébastien Moutte <sebastien@moutte.net>
29723 win32/vs6: add a project file for libgstvolume update the workspace
29724 Original commit message from CVS:
29726 add a project file for libgstvolume
29727 update the workspace
29729 2006-03-03 15:26:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29732 * ext/ogg/gstoggmux.c:
29734 Original commit message from CVS:
29737 2006-03-03 15:22:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29739 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
29740 Original commit message from CVS:
29741 2006-03-03 Thomas Vander Stichele <thomas at apestaart dot org>
29742 * ext/theora/theoraenc.c: (theora_set_header_on_caps):
29743 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
29745 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
29746 Set IN_CAPS on header buffers
29748 2006-03-02 18:23:55 +0000 Wim Taymans <wim.taymans@gmail.com>
29750 docs/plugins/: Add audioresample to docs.
29751 Original commit message from CVS:
29752 * docs/plugins/Makefile.am:
29753 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29754 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29755 Add audioresample to docs.
29756 * gst/audioconvert/gstaudioconvert.c:
29758 * gst/audioresample/gstaudioresample.c:
29759 (gst_audioresample_base_init), (gst_audioresample_class_init),
29760 (gst_audioresample_init), (gst_audioresample_dispose),
29761 (audioresample_get_unit_size), (audioresample_transform_caps),
29762 (resample_set_state_from_caps), (audioresample_transform_size),
29763 (audioresample_set_caps), (audioresample_event),
29764 (audioresample_do_output), (audioresample_transform),
29765 (audioresample_pushthrough), (gst_audioresample_set_property),
29766 (gst_audioresample_get_property), (plugin_init):
29767 * gst/audioresample/gstaudioresample.h:
29769 Small code cleanups.
29771 2006-03-02 18:12:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29773 * gst/videorate/Makefile.am:
29775 Original commit message from CVS:
29778 2006-03-02 17:48:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29780 * ext/ogg/gstoggmux.c:
29781 debug using the actual GstPad, that allows us to see the serialno in the padname
29782 Original commit message from CVS:
29783 debug using the actual GstPad, that allows us to see the serialno in the padname
29785 2006-03-02 17:46:36 +0000 Wim Taymans <wim.taymans@gmail.com>
29787 docs/plugins/: Added videoscale to docs.
29788 Original commit message from CVS:
29789 * docs/plugins/Makefile.am:
29790 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29791 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29792 Added videoscale to docs.
29793 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
29794 (gst_video_rate_swap_prev), (gst_video_rate_event),
29795 (gst_video_rate_chain):
29797 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
29798 (gst_video_scale_init), (gst_video_scale_prepare_size),
29799 (gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
29800 (gst_video_scale_fixate_caps), (gst_video_scale_transform):
29801 * gst/videoscale/gstvideoscale.h:
29802 Added docs, examples.
29803 Some code cleanups.
29804 Post errors instead of g_warning.
29806 2006-03-02 17:30:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29808 * ext/ogg/gstoggmux.c:
29809 clean up debug messages
29810 Original commit message from CVS:
29811 clean up debug messages
29813 2006-03-02 17:15:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29815 * ext/ogg/gstoggmux.c:
29816 extra debugging from older version, makes it easier to compare
29817 Original commit message from CVS:
29818 extra debugging from older version, makes it easier to compare
29820 2006-03-02 17:04:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29822 * ext/ogg/gstoggmux.c:
29823 some space cleanup and debug fixes
29824 Original commit message from CVS:
29825 some space cleanup and debug fixes
29827 2006-03-02 16:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
29829 docs/: Added some more docs to libs and plugins.
29830 Original commit message from CVS:
29831 * docs/libs/gst-plugins-base-libs-docs.sgml:
29832 * docs/libs/gst-plugins-base-libs-sections.txt:
29833 * docs/libs/gst-plugins-base-libs.types:
29834 * docs/plugins/Makefile.am:
29835 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29836 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29837 Added some more docs to libs and plugins.
29838 * gst-libs/gst/audio/gstringbuffer.c:
29839 (gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
29840 * gst-libs/gst/audio/gstringbuffer.h:
29841 Document ringbuffer some more.
29842 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
29843 (gst_video_rate_setcaps), (gst_video_rate_reset),
29844 (gst_video_rate_init), (gst_video_rate_flush_prev),
29845 (gst_video_rate_swap_prev), (gst_video_rate_event),
29846 (gst_video_rate_chain), (gst_video_rate_change_state):
29847 * gst/videorate/gstvideorate.h:
29848 Fix videorate to use segments.
29849 Make it work with 0/1 framerates (closes #331903)
29850 Handle EOS correctly.
29853 2006-03-02 13:13:00 +0000 Tim-Philipp Müller <tim@centricular.net>
29855 ext/ogg/gstogmparse.c: In state change function, first chain up to parent class, then handle downwards state change s...
29856 Original commit message from CVS:
29857 * ext/ogg/gstogmparse.c: (gst_ogm_parse_class_init),
29858 (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
29859 (gst_ogm_text_parse_init), (gst_ogm_parse_change_state):
29860 In state change function, first chain up to parent class,
29861 then handle downwards state change stuff. Remove some
29862 commented out cruft from 0.8 code.
29864 2006-03-02 12:35:59 +0000 Tim-Philipp Müller <tim@centricular.net>
29866 ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
29867 Original commit message from CVS:
29868 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
29869 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
29870 (gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
29871 (gst_ogm_parse_chain):
29872 Don't remove/re-add source pad if the new caps are the same as
29873 the old caps anyway (#333042). When removing source pad, don't
29874 unref it afterwards - we didn't ref it when adding. Sprinkle some
29875 GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
29876 after using gst_pad_get_parent(). Return downstream flow return
29877 value in chain function.
29879 2006-03-02 11:28:23 +0000 Wim Taymans <wim.taymans@gmail.com>
29881 docs/plugins/: Fix hierarchy, added some more elements to the docs.
29882 Original commit message from CVS:
29883 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29884 * docs/plugins/gst-plugins-base-plugins.args:
29885 * docs/plugins/gst-plugins-base-plugins.hierarchy:
29886 * docs/plugins/gst-plugins-base-plugins.interfaces:
29887 * docs/plugins/gst-plugins-base-plugins.signals:
29888 Fix hierarchy, added some more elements to the docs.
29889 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
29890 (gst_ffmpegcsp_get_type):
29891 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
29892 Fix docs for ffmpegcolorspace.
29894 2006-03-01 19:24:44 +0000 Tim-Philipp Müller <tim@centricular.net>
29896 gst/typefind/gsttypefindfunctions.c: Some typefinding fine-tuning:
29897 Original commit message from CVS:
29898 * gst/typefind/gsttypefindfunctions.c: (id3_type_find),
29899 (apetag_type_find), (ape_type_find), (plugin_init):
29900 Some typefinding fine-tuning:
29901 - rank ID3/APE tags in order of preference via probabilities, so that
29902 ID3v2 > APEv2 > APEv1 > ID3v1.
29903 - three or four bytes don't really justify MAXIMUM probability,
29904 change those to 'very likely' (musepack and monkeysaudio).
29906 2006-03-01 18:25:18 +0000 Wim Taymans <wim.taymans@gmail.com>
29909 Original commit message from CVS:
29910 * docs/plugins/Makefile.am:
29911 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29912 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29913 * ext/alsa/gstalsamixer.c:
29914 * ext/alsa/gstalsamixer.h:
29915 * ext/alsa/gstalsamixerelement.c:
29916 (gst_alsa_mixer_element_class_init), (gst_alsa_mixer_element_init):
29917 * ext/alsa/gstalsamixerelement.h:
29918 * ext/alsa/gstalsasink.c:
29919 * ext/alsa/gstalsasink.h:
29920 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
29921 (gst_alsasrc_init):
29922 * ext/alsa/gstalsasrc.h:
29924 Small code cleanups.
29926 2006-03-01 17:52:45 +0000 Wim Taymans <wim.taymans@gmail.com>
29928 ext/theora/Makefile.am: Dist new header too,
29929 Original commit message from CVS:
29930 * ext/theora/Makefile.am:
29931 Dist new header too,
29933 2006-03-01 17:39:28 +0000 Wim Taymans <wim.taymans@gmail.com>
29935 Fix some more docs.
29936 Original commit message from CVS:
29937 * docs/plugins/Makefile.am:
29938 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29939 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29940 * ext/gnomevfs/gstgnomevfssink.h:
29941 * ext/gnomevfs/gstgnomevfssrc.h:
29942 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
29943 * ext/vorbis/vorbisdec.h:
29944 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
29945 * ext/vorbis/vorbisenc.h:
29946 * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
29947 (vorbis_parse_chain), (vorbis_parse_change_state):
29948 * ext/vorbis/vorbisparse.h:
29949 * gst/audioconvert/gstaudioconvert.h:
29950 * gst/tcp/gsttcpserversink.h:
29951 * gst/videotestsrc/gstvideotestsrc.c:
29952 * gst/videotestsrc/gstvideotestsrc.h:
29953 * gst/volume/gstvolume.c:
29954 * gst/volume/gstvolume.h:
29955 Fix some more docs.
29956 Added docs for vorbisdec and vorbisparse.
29959 2006-03-01 16:24:37 +0000 Wim Taymans <wim.taymans@gmail.com>
29961 Updated/added documentation.
29962 Original commit message from CVS:
29963 * docs/plugins/Makefile.am:
29964 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29965 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29966 * ext/pango/gstclockoverlay.h:
29967 * ext/pango/gsttextoverlay.h:
29968 * ext/pango/gsttextrender.h:
29969 * ext/pango/gsttimeoverlay.h:
29970 * ext/theora/gsttheoradec.h:
29971 * ext/theora/gsttheoraenc.h:
29972 * ext/theora/theoradec.c:
29973 * ext/theora/theoraenc.c:
29974 * gst/audioconvert/gstaudioconvert.h:
29975 * gst/audiotestsrc/gstaudiotestsrc.h:
29976 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
29977 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
29978 * gst/tcp/gstmultifdsink.h:
29979 Updated/added documentation.
29980 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
29981 (gst_text_overlay_halign_get_type),
29982 (gst_text_overlay_wrap_mode_get_type),
29983 (gst_text_overlay_base_init), (gst_text_overlay_class_init),
29984 (gst_text_overlay_init), (gst_text_overlay_set_property),
29985 (gst_text_overlay_get_property):
29986 Fix up properties to be enums instead of string to make bindings,
29987 introspection and automatic GUI creation possible.
29988 Add getters for the properties.
29990 2006-02-28 21:21:07 +0000 Sébastien Moutte <sebastien@moutte.net>
29992 gst/audiotestsrc/gstaudiotestsrc.c: added defines of M_PI and M_PI_2
29993 Original commit message from CVS:
29994 * gst/audiotestsrc/gstaudiotestsrc.c:
29995 added defines of M_PI and M_PI_2
29996 * gst/ffmpegcolorspace/avcodec.h:
29997 removed #include "stdint.h" for win32 as _stdint.h is
29998 autogenerated to win32/common
29999 * win32/common/libgstaudio.def:
30000 * win32/common/libgsttag.def:
30003 some project files bugs corrected
30005 project files are reset to the default vs7 configuration
30006 (they link to msvcr71.dll using default optimizations)
30008 2006-02-28 19:08:12 +0000 Wim Taymans <wim.taymans@gmail.com>
30010 ext/gnomevfs/gstgnomevfssink.c: Fix some docs.
30011 Original commit message from CVS:
30012 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
30015 2006-02-28 13:52:04 +0000 Edward Hervey <bilboed@bilboed.com>
30017 ext/alsa/gstalsasrc.c: Set proper class on the ElementDetails:
30018 Original commit message from CVS:
30019 * ext/alsa/gstalsasrc.c:
30020 Set proper class on the ElementDetails:
30021 Source/Audio instead of Src/Audio
30023 2006-02-28 12:19:11 +0000 Edward Hervey <bilboed@bilboed.com>
30025 gst/videoscale/vs_scanline.c: Revert optimization in videoscale. It should go in liboil and have an appropriate liboi...
30026 Original commit message from CVS:
30027 * gst/videoscale/vs_scanline.c:
30028 (vs_scanline_resample_nearest_RGBA):
30029 Revert optimization in videoscale. It should go in liboil and have
30030 an appropriate liboil function.
30032 2006-02-28 11:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
30034 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
30035 Original commit message from CVS:
30036 * gst-libs/gst/audio/gstbaseaudiosink.c:
30037 (gst_base_audio_sink_provide_clock):
30038 Don't try to provide a clock in the NULL state.
30040 2006-02-28 11:04:47 +0000 Wim Taymans <wim.taymans@gmail.com>
30042 ext/ogg/gstoggdemux.c: Use GstSegment infrastructure to remove duplicated code and handle more seek cases correctly.
30043 Original commit message from CVS:
30044 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
30045 (gst_ogg_pad_event), (gst_ogg_pad_internal_chain),
30046 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
30047 (gst_ogg_demux_deactivate_current_chain),
30048 (gst_ogg_demux_activate_chain), (gst_ogg_demux_do_seek),
30049 (gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_info),
30050 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
30051 (gst_ogg_demux_loop), (gst_ogg_demux_change_state):
30052 Use GstSegment infrastructure to remove duplicated code
30053 and handle more seek cases correctly.
30055 2006-02-28 10:39:19 +0000 Wim Taymans <wim.taymans@gmail.com>
30057 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Don't ignore return code from ffmpeg convert function.
30058 Original commit message from CVS:
30059 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30060 (gst_ffmpegcsp_transform):
30061 Don't ignore return code from ffmpeg convert function.
30062 * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
30063 Split out some long statements to ease debugging.
30065 2006-02-27 12:08:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30067 ext/libvisual/visual.c: Don't use gst_pad_use_fixed_caps, because it prevents downstream from being able to renegotia...
30068 Original commit message from CVS:
30069 * ext/libvisual/visual.c: (gst_visual_init),
30070 (gst_vis_src_negotiate), (get_buffer), (plugin_init):
30071 Don't use gst_pad_use_fixed_caps, because it prevents downstream from
30072 being able to renegotiate the size. Instead, use the negotiation
30073 algorithm from the goom plugin to pick an initial output caps.
30074 Also, allow theoretical libvisual plugins that might support non-GL
30075 output even if they also do GL.
30077 2006-02-26 21:05:46 +0000 Julien Moutte <julien@moutte.net>
30079 ext/libvisual/visual.c: Load only non GL plugins. Fix some memleaks and possible negotiation issues.
30080 Original commit message from CVS:
30081 2006-02-26 Julien MOUTTE <julien@moutte.net>
30082 * ext/libvisual/visual.c: (gst_visual_init),
30083 (gst_visual_src_setcaps), (get_buffer), (gst_visual_chain),
30084 (plugin_init): Load only non GL plugins. Fix some memleaks and
30085 possible negotiation issues.
30087 2006-02-24 23:19:44 +0000 Julien Moutte <julien@moutte.net>
30089 gst-libs/gst/tag/tag.h: Adding Annodex tags here.
30090 Original commit message from CVS:
30091 2006-02-25 Julien MOUTTE <julien@moutte.net>
30092 * gst-libs/gst/tag/tag.h: Adding Annodex tags here.
30094 2006-02-24 18:55:27 +0000 Michael Smith <msmith@xiph.org>
30096 gst/typefind/gsttypefindfunctions.c: Fix CMML type find function to not require a specific minor version of the CMML ...
30097 Original commit message from CVS:
30098 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
30099 (cmml_type_find), (plugin_init):
30100 Fix CMML type find function to not require a specific minor version
30101 of the CMML header.
30102 Add an MPEG4 video elementary stream typefind function.
30104 2006-02-24 17:31:53 +0000 Michael Smith <msmith@xiph.org>
30106 ext/ogg/gstoggdemux.c: Annodex support in ogg demuxer. Doesn't do very much without the other annodex patches (to come).
30107 Original commit message from CVS:
30108 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
30109 (gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
30110 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
30111 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
30112 (gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
30113 (gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
30114 Annodex support in ogg demuxer. Doesn't do very much without the
30115 other annodex patches (to come).
30117 2006-02-24 16:21:34 +0000 Tim-Philipp Müller <tim@centricular.net>
30119 gst-libs/gst/riff/riff-media.c:
30120 Original commit message from CVS:
30121 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
30122 Pick up palette for MS video v1 (#327028, patch by:
30123 Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
30125 2006-02-24 13:54:04 +0000 Tim-Philipp Müller <tim@centricular.net>
30127 gst/ffmpegcolorspace/gstffmpegcolorspace.c: The 'palette_data' field from incoming RGB caps shouldn't be proxied on o...
30128 Original commit message from CVS:
30129 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30130 (gst_ffmpegcsp_caps_remove_format_info),
30131 (gst_ffmpegcsp_get_unit_size):
30132 The 'palette_data' field from incoming RGB caps shouldn't be
30133 proxied on outgoing YUV caps; also, restrict unit size
30134 adjustment in case of paletted data only to the unit that
30135 actually has a palette. Fixes #330711.
30137 2006-02-24 12:18:14 +0000 Tim-Philipp Müller <tim@centricular.net>
30139 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Plug some memory leaks.
30140 Original commit message from CVS:
30141 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30142 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
30143 (gst_ffmpegcsp_finalize), (gst_ffmpegcsp_class_init),
30144 (gst_ffmpegcsp_get_unit_size):
30145 Plug some memory leaks.
30147 2006-02-24 10:18:52 +0000 Tim-Philipp Müller <tim@centricular.net>
30149 sys/: Add some _CFLAGS and _LIBS that seem to be missing and/or required for Cygwin (see #317048).
30150 Original commit message from CVS:
30151 * sys/ximage/Makefile.am:
30152 * sys/xvimage/Makefile.am:
30153 Add some _CFLAGS and _LIBS that seem to be missing
30154 and/or required for Cygwin (see #317048).
30156 2006-02-24 00:07:18 +0000 Tim-Philipp Müller <tim@centricular.net>
30159 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
30160 Original commit message from CVS:
30161 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
30163 2006-02-22 18:46:46 +0000 Wim Taymans <wim.taymans@gmail.com>
30165 ext/alsa/gstalsasrc.c: Fix description as pointed out by caugier.
30166 Original commit message from CVS:
30167 * ext/alsa/gstalsasrc.c:
30168 Fix description as pointed out by caugier.
30170 2006-02-22 10:29:22 +0000 Edward Hervey <bilboed@bilboed.com>
30172 gst/typefind/gsttypefindfunctions.c: Better 3gp typefinding.
30173 Original commit message from CVS:
30174 Reviewed by : Edward Hervey <edward@fluendo.com>
30175 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
30177 Better 3gp typefinding.
30179 2006-02-21 12:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
30181 ext/gnomevfs/gstgnomevfssrc.c: Don't send EOS event here, the base class will send one for us.
30182 Original commit message from CVS:
30183 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
30184 Don't send EOS event here, the base class will send one for us.
30185 * gst/playback/gstplaybasebin.c: (prepare_output):
30186 Subpictures without video stream aren't allowed either.
30187 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
30188 Fix debug statement copy'n'paste-o.
30190 2006-02-21 12:05:18 +0000 Tim-Philipp Müller <tim@centricular.net>
30192 ext/alsa/gstalsamixer.c: Fix issues with mixer keeping state when muting/unmuting and when changing the volume whilst...
30193 Original commit message from CVS:
30194 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
30195 Fix issues with mixer keeping state when muting/unmuting
30196 and when changing the volume whilst muted (see #331763
30199 2006-02-20 18:27:06 +0000 Tim-Philipp Müller <tim@centricular.net>
30201 gst/subparse/gstsubparse.c: Set right caps given that we send escaped text. Also, honour <i></i>, <b></b> and <u></u>...
30202 Original commit message from CVS:
30203 * gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
30204 (parse_subrip), (gst_sub_parse_format_autodetect):
30205 Set right caps given that we send escaped text. Also,
30206 honour <i></i>, <b></b> and <u></u> markers that can be found
30207 in .srt files (fixes #310202).
30209 2006-02-20 16:21:14 +0000 Tim-Philipp Müller <tim@centricular.net>
30211 gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
30212 Original commit message from CVS:
30213 * gst-libs/gst/audio/mixerutils.c:
30214 (element_factory_rank_compare_func):
30215 Make order in which elements are tried more determinable.
30217 2006-02-20 15:57:51 +0000 Julien Moutte <julien@moutte.net>
30219 gst/playback/gstdecodebin.c: Make decodebin reusable by fixing remove_element_chain first and then introduce a cleane...
30220 Original commit message from CVS:
30221 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
30222 (remove_element_chain), (cleanup_decodebin),
30223 (gst_decode_bin_change_state): Make decodebin reusable by
30224 fixing remove_element_chain first and then introduce a
30225 cleaner in state change to ->NULL. (Closes #331678)
30226 ------------------------------------------------------
30228 2006-02-19 14:32:35 +0000 Wim Taymans <wim.taymans@gmail.com>
30230 ext/gnomevfs/gstgnomevfssink.c: use 0666 mask when creating files so umask gets applied correctly. Fixes #331295.
30231 Original commit message from CVS:
30232 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
30233 use 0666 mask when creating files so umask gets applied
30234 correctly. Fixes #331295.
30236 2006-02-19 14:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
30238 gst/subparse/: Add very basic parser for SSA subtitle streams (as often found in matroska files).
30239 Original commit message from CVS:
30240 * gst/subparse/Makefile.am:
30241 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
30242 (gst_ssa_parse_dispose), (gst_ssa_parse_init),
30243 (gst_ssa_parse_class_init), (gst_ssa_parse_src_event),
30244 (gst_ssa_parse_sink_event), (gst_ssa_parse_setcaps),
30245 (gst_ssa_parse_remove_override_codes), (gst_ssa_parse_parse_line),
30246 (gst_ssa_parse_chain), (gst_ssa_parse_change_state):
30247 * gst/subparse/gstssaparse.h:
30248 * gst/subparse/gstsubparse.c: (plugin_init):
30249 Add very basic parser for SSA subtitle streams (as often
30250 found in matroska files).
30252 2006-02-19 14:09:40 +0000 Tim-Philipp Müller <tim@centricular.net>
30254 gst/playback/gstdecodebin.c: That should be text/x-pango-markup, not text/x-pango-layout.
30255 Original commit message from CVS:
30256 * gst/playback/gstdecodebin.c: (mimetype_is_raw):
30257 That should be text/x-pango-markup, not text/x-pango-layout.
30259 2006-02-19 12:41:03 +0000 Julien Moutte <julien@moutte.net>
30261 ext/pango/gsttextoverlay.c: Polishing.
30262 Original commit message from CVS:
30263 2006-02-19 Julien MOUTTE <julien@moutte.net>
30264 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize):
30267 2006-02-19 12:05:23 +0000 Julien Moutte <julien@moutte.net>
30269 ext/pango/gsttextoverlay.c: Fix state change deadlock.
30270 Original commit message from CVS:
30271 2006-02-19 Julien MOUTTE <julien@moutte.net>
30272 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
30273 (gst_text_overlay_finalize), (gst_text_overlay_init),
30274 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
30275 (gst_text_overlay_render_text),
30276 (gst_text_overlay_text_pad_link),
30277 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
30278 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
30279 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
30280 Fix state change deadlock.
30282 2006-02-19 11:56:28 +0000 Julien Moutte <julien@moutte.net>
30284 ext/pango/gsttextoverlay.*: Fix seeking both for muxed formats and subtitles files.
30285 Original commit message from CVS:
30286 2006-02-19 Julien MOUTTE <julien@moutte.net>
30287 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
30288 (gst_text_overlay_finalize), (gst_text_overlay_init),
30289 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
30290 (gst_text_overlay_render_text),
30291 (gst_text_overlay_text_pad_link),
30292 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
30293 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
30294 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
30295 * ext/pango/gsttextoverlay.h: Fix seeking both for muxed formats
30296 and subtitles files.
30298 2006-02-19 00:40:38 +0000 Julien Moutte <julien@moutte.net>
30300 gst/playback/gstdecodebin.c: pango layout should be considered as row.
30301 Original commit message from CVS:
30302 2006-02-19 Julien MOUTTE <julien@moutte.net>
30303 * gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
30304 should be considered as row.
30306 2006-02-19 00:25:16 +0000 Julien Moutte <julien@moutte.net>
30308 gst/playback/gststreaminfo.*: Introduce language informations.
30309 Original commit message from CVS:
30310 2006-02-19 Julien MOUTTE <julien@moutte.net>
30311 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
30313 * gst/playback/gststreaminfo.h: Introduce language informations.
30315 2006-02-18 22:41:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30317 sys/: Set shared memory segments to be deleted as soon as we have attached, that way they get cleaned up automaticall...
30318 Original commit message from CVS:
30319 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
30320 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
30321 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
30322 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
30323 Set shared memory segments to be deleted as soon as we have attached,
30324 that way they get cleaned up automatically if we crash.
30326 2006-02-18 19:53:48 +0000 Julien Moutte <julien@moutte.net>
30328 ext/pango/: Those functions are called with lock held.
30329 Original commit message from CVS:
30330 2006-02-18 Julien MOUTTE <julien@moutte.net>
30331 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_get_text):
30332 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_get_text): Those
30333 functions are called with lock held.
30335 2006-02-18 19:51:47 +0000 Julien Moutte <julien@moutte.net>
30339 Original commit message from CVS:
30342 2006-02-18 19:10:35 +0000 Julien Moutte <julien@moutte.net>
30344 ext/pango/gsttextoverlay.c: Refactoring of textoverlay without collectpads. This now supports sparse subtitles coming...
30345 Original commit message from CVS:
30346 2006-02-18 Julien MOUTTE <julien@moutte.net>
30347 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
30348 (gst_text_overlay_finalize), (gst_text_overlay_init),
30349 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
30350 (gst_text_overlay_render_text),
30351 (gst_text_overlay_text_pad_link),
30352 (gst_text_overlay_text_pad_unlink),
30353 (gst_text_overlay_text_event),
30354 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
30355 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
30356 (gst_text_overlay_change_state): Refactoring of textoverlay
30357 without collectpads. This now supports sparse subtitles coming
30358 from a demuxer instead of a sub file. Seeking is still broken
30359 though. Need to discuss with wtay some more on how to handle
30361 * ext/pango/gsttextoverlay.h:
30362 * gst/playback/gstplaybin.c: (setup_sinks): Support linking with
30363 subtitles coming from the demuxer.
30365 2006-02-17 19:31:12 +0000 Wim Taymans <wim.taymans@gmail.com>
30367 ext/vorbis/vorbisenc.c: Use some more scaling functions.
30368 Original commit message from CVS:
30369 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
30370 (gst_vorbisenc_convert_sink):
30371 Use some more scaling functions.
30373 2006-02-17 16:12:11 +0000 Tim-Philipp Müller <tim@centricular.net>
30375 ext/cdparanoia/gstcdparanoiasrc.*: Add back 'transport-error' and 'uncorrected-error' signals and make them actually ...
30376 Original commit message from CVS:
30377 * ext/cdparanoia/gstcdparanoiasrc.c:
30378 (gst_cd_paranoia_src_class_init), (gst_cd_paranoia_dummy_callback),
30379 (gst_cd_paranoia_paranoia_callback),
30380 (gst_cd_paranoia_src_signal_is_being_watched),
30381 (gst_cd_paranoia_src_read_sector):
30382 * ext/cdparanoia/gstcdparanoiasrc.h:
30383 Add back 'transport-error' and 'uncorrected-error' signals and
30384 make them actually be fired when bad stuff happens (#319340).
30386 2006-02-17 14:07:01 +0000 Wim Taymans <wim.taymans@gmail.com>
30388 gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
30389 Original commit message from CVS:
30390 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
30391 (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
30392 (gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
30393 (gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
30394 (gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
30395 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
30396 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
30397 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
30398 (gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
30399 (gst_ring_buffer_clear):
30401 Added some G_LIKELY.
30403 2006-02-17 10:15:52 +0000 Wim Taymans <wim.taymans@gmail.com>
30405 gst-libs/gst/audio/TODO: Update TODO
30406 Original commit message from CVS:
30407 * gst-libs/gst/audio/TODO:
30409 * gst-libs/gst/audio/gstbaseaudiosink.c:
30410 (gst_base_audio_sink_get_offset):
30411 When trying to play samples ASAP and we don't have a
30412 previous sample, try to play at position 0 instead of
30413 an invalid position.
30415 2006-02-17 09:24:56 +0000 Wim Taymans <wim.taymans@gmail.com>
30417 ext/alsa/gstalsasink.c: Also release lock when we get an error in _reset(); fix an error message.
30418 Original commit message from CVS:
30419 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
30420 (gst_alsasink_reset):
30421 Also release lock when we get an error in _reset();
30422 fix an error message.
30424 2006-02-16 21:01:23 +0000 Tim-Philipp Müller <tim@centricular.net>
30426 ext/alsa/gstalsasink.*: Add support for more than 2 channels (#326720).
30427 Original commit message from CVS:
30428 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
30429 (gst_alsasink_init), (get_channel_free_structure),
30430 (caps_add_channel_configuration), (gst_alsasink_getcaps),
30431 (gst_alsasink_close):
30432 * ext/alsa/gstalsasink.h:
30433 Add support for more than 2 channels (#326720).
30435 2006-02-16 20:19:51 +0000 Tim-Philipp Müller <tim@centricular.net>
30437 gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
30438 Original commit message from CVS:
30439 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
30440 Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
30441 with 4 or 6 channels, assume a default channel layout to make things
30442 work (not sure there's anything else we can do in those cases).
30444 2006-02-16 19:18:46 +0000 Tim-Philipp Müller <tim@centricular.net>
30446 gst-libs/gst/audio/multichannel.c: Minor docs fix.
30447 Original commit message from CVS:
30448 * gst-libs/gst/audio/multichannel.c:
30450 * gst-libs/gst/riff/Makefile.am:
30451 * gst-libs/gst/riff/riff-ids.h:
30452 * gst-libs/gst/riff/riff-media.c:
30453 (gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
30454 Add support for WAVEFORMATEX, eg. PCM audio with more than two
30455 channels and a channel layout map.
30457 2006-02-16 17:06:46 +0000 Edward Hervey <bilboed@bilboed.com>
30459 gst/videoscale/vs_scanline.c: C-level optimization of the RGBA nearest neighbour function.
30460 Original commit message from CVS:
30461 Reviewed by Edward Hervey <edward@fluendo.com>
30462 * gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
30463 C-level optimization of the RGBA nearest neighbour function.
30464 Eventually this might end up in liboil with vectorized versions.
30466 2006-02-16 11:44:43 +0000 Tim-Philipp Müller <tim@centricular.net>
30468 gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
30469 Original commit message from CVS:
30470 * gst-libs/gst/audio/multichannel.c:
30471 (gst_audio_get_channel_positions):
30472 When we have more than 2 channels, but no channel layout is
30473 specified in the caps, return some default channel layout
30474 to the caller and warn about about a possibly buggy element
30475 (could be buggy filtercaps as well of course) (#317038).
30477 2006-02-16 09:29:38 +0000 Tim-Philipp Müller <tim@centricular.net>
30479 pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: Add gst-libs/gst/cdda to list of lib search paths.
30480 Original commit message from CVS:
30481 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
30482 Add gst-libs/gst/cdda to list of lib search paths.
30484 2006-02-15 12:20:47 +0000 Andy Wingo <wingo@pobox.com>
30486 ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating timestamp, update timestamp_end as well. Fixes a bugaboo. ...
30487 Original commit message from CVS:
30488 2006-02-15 Andy Wingo <wingo@pobox.com>
30489 * ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating
30490 timestamp, update timestamp_end as well. Fixes a bugaboo. I hope
30491 to the Lord Jesus that I do not have to touch the ogg muxer ever
30494 2006-02-15 12:07:57 +0000 Edward Hervey <bilboed@bilboed.com>
30496 gst/typefind/gsttypefindfunctions.c: quicktime movie files can also contain 'uuid' atoms.
30497 Original commit message from CVS:
30498 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
30499 quicktime movie files can also contain 'uuid' atoms.
30501 2006-02-14 18:52:52 +0000 Tim-Philipp Müller <tim@centricular.net>
30503 gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun...
30504 Original commit message from CVS:
30505 * gst/audioconvert/plugin.c: (plugin_init):
30506 Register the GstAudioChannelPosition enum type with the type
30507 system in the plugin_init function, so that it is known before
30508 any element actually makes use of multi-channel stuff. This is
30509 required for example if one wants to be able to deserialise/use
30510 a caps string with channel positions before any pipeline has
30511 been setup and started, like with gst-launch.
30513 2006-02-14 13:45:35 +0000 Wim Taymans <wim.taymans@gmail.com>
30515 gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
30516 Original commit message from CVS:
30517 * gst-libs/gst/audio/gstringbuffer.c:
30518 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
30519 (gst_ring_buffer_samples_done), (wait_segment),
30520 (gst_ring_buffer_commit), (gst_ring_buffer_clear):
30521 Add some compiler G_(UN_)LIKELY help.
30522 SIGNAL the ringbuffer waiters when going to PAUSED as well to
30523 make sure they can exit their functions. Should fix #330748
30525 2006-02-13 20:49:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30527 Windows does not have long long; copy the generated _stdint.h
30528 Original commit message from CVS:
30532 * win32/common/_stdint.h:
30533 Windows does not have long long; copy the generated _stdint.h
30534 * win32/common/interfaces-enumtypes.c:
30535 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
30536 (gst_mixer_track_flags_get_type),
30537 (gst_tuner_channel_flags_get_type):
30538 * win32/common/multichannel-enumtypes.c:
30539 (gst_audio_channel_position_get_type):
30542 2006-02-13 18:49:02 +0000 Wim Taymans <wim.taymans@gmail.com>
30544 gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
30545 Original commit message from CVS:
30546 * gst-libs/gst/audio/gstbaseaudiosink.c:
30547 (gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
30548 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
30549 Always sync on first sample we receive when starting.
30551 2006-02-13 15:59:48 +0000 Wim Taymans <wim.taymans@gmail.com>
30553 gst/playback/gstplaybin.c: Update vis bin docs.
30554 Original commit message from CVS:
30555 * gst/playback/gstplaybin.c: (gen_vis_element):
30556 Update vis bin docs.
30557 Move queue after tee so we don't queue video buffers but
30558 audio samples instead. Fixes problems where the video queue
30559 is filled and the audio queue empty.
30561 2006-02-13 15:17:34 +0000 Tim-Philipp Müller <tim@centricular.net>
30563 gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
30564 Original commit message from CVS:
30565 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
30566 No need to push an EOS event here, GstBaseSrc will do that for us
30567 when we return FLOW_UNEXPECTED.
30569 2006-02-12 14:54:55 +0000 Wim Taymans <wim.taymans@gmail.com>
30571 gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
30572 Original commit message from CVS:
30573 * gst-libs/gst/audio/gstbaseaudiosink.c:
30574 (gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
30575 (gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
30576 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
30577 Use scale functions when possible.
30578 Fix error messages.
30579 Free clockid when after waiting for EOS.
30580 Use G_(UN_)LIKLY when it makes sense.
30581 Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
30583 2006-02-12 14:26:55 +0000 Edward Hervey <bilboed@bilboed.com>
30585 gst/playback/gstplaybasebin.c: Remove stray semi-colon (fixes #330888).
30586 Original commit message from CVS:
30587 * gst/playback/gstplaybasebin.c: (prepare_output):
30588 Remove stray semi-colon (fixes #330888).
30590 2006-02-11 23:35:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30592 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...
30593 Original commit message from CVS:
30594 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
30595 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
30596 Fix up the XShm call testing so that we catch errors, and don't
30597 cause new ones by attempting to detach from a segment we failed
30598 to attach to. Fixes #312439.
30600 2006-02-10 11:29:55 +0000 Edward Hervey <bilboed@bilboed.com>
30602 gst/typefind/gsttypefindfunctions.c: Added flv file typefind (video/x-flv).
30603 Original commit message from CVS:
30604 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
30605 Added flv file typefind (video/x-flv).
30607 2006-02-10 10:53:33 +0000 Edward Hervey <bilboed@bilboed.com>
30609 gst-libs/gst/riff/riff-media.c: Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
30610 Original commit message from CVS:
30611 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
30612 (gst_riff_create_video_template_caps):
30613 Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
30614 Also added the caps to the default set of riff video caps.
30616 2006-02-09 19:05:23 +0000 Andy Wingo <wingo@pobox.com>
30618 ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start time and the end time of the last packet in the page.
30619 Original commit message from CVS:
30620 2006-02-09 Andy Wingo <wingo@pobox.com>
30621 * ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start
30622 time and the end time of the last packet in the page.
30623 (gst_ogg_mux_pad_queue_page): In addition to setting the timestamp
30624 on the pages in our queue, set the duration as well. Reflow a
30626 (gst_ogg_mux_collected): Keep track of GstOggPad->timestamp_end.
30627 Fixes bad muxing order.
30629 2006-02-09 17:04:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30631 gst-libs/gst/rtp/gstbasertppayload.c: update seqnum before setting it on the packet; this makes sure that the timesta...
30632 Original commit message from CVS:
30633 * gst-libs/gst/rtp/gstbasertppayload.c:
30634 (gst_basertppayload_setcaps), (gst_basertppayload_push):
30635 update seqnum before setting it on the packet; this makes sure
30636 that the timestamp and seqnum properties match after pushing
30639 2006-02-09 12:16:35 +0000 Andy Wingo <wingo@pobox.com>
30643 Original commit message from CVS:
30646 2006-02-09 11:46:03 +0000 Andy Wingo <wingo@pobox.com>
30648 * gst-libs/gst/audio/gstringbuffer.c:
30649 * win32/common/config.h:
30651 Original commit message from CVS:
30654 2006-02-09 11:36:18 +0000 Andy Wingo <wingo@pobox.com>
30656 gst-libs/gst/audio/gstringbuffer.c
30657 Original commit message from CVS:
30658 2006-02-09 Andy Wingo <wingo@pobox.com>
30659 * gst-libs/gst/audio/gstringbuffer.c
30660 (gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
30661 overflow after 13.5 hours of recording. Kapow!
30662 * ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
30663 the buffer size -- we don't care about underrun/overrun reporting
30664 right now, just need to return a useful value.
30666 2006-02-09 11:21:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30668 configure.ac: Back to CVS
30669 Original commit message from CVS:
30673 === release 0.10.3 ===
30675 2006-02-09 11:18:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30681 * docs/plugins/inspect/plugin-adder.xml:
30682 * docs/plugins/inspect/plugin-alsa.xml:
30683 * docs/plugins/inspect/plugin-audioconvert.xml:
30684 * docs/plugins/inspect/plugin-audiorate.xml:
30685 * docs/plugins/inspect/plugin-audioresample.xml:
30686 * docs/plugins/inspect/plugin-audiotestsrc.xml:
30687 * docs/plugins/inspect/plugin-cdparanoia.xml:
30688 * docs/plugins/inspect/plugin-decodebin.xml:
30689 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
30690 * docs/plugins/inspect/plugin-gnomevfs.xml:
30691 * docs/plugins/inspect/plugin-libvisual.xml:
30692 * docs/plugins/inspect/plugin-ogg.xml:
30693 * docs/plugins/inspect/plugin-pango.xml:
30694 * docs/plugins/inspect/plugin-playbin.xml:
30695 * docs/plugins/inspect/plugin-subparse.xml:
30696 * docs/plugins/inspect/plugin-tcp.xml:
30697 * docs/plugins/inspect/plugin-theora.xml:
30698 * docs/plugins/inspect/plugin-typefindfunctions.xml:
30699 * docs/plugins/inspect/plugin-video4linux.xml:
30700 * docs/plugins/inspect/plugin-videorate.xml:
30701 * docs/plugins/inspect/plugin-videoscale.xml:
30702 * docs/plugins/inspect/plugin-videotestsrc.xml:
30703 * docs/plugins/inspect/plugin-volume.xml:
30704 * docs/plugins/inspect/plugin-vorbis.xml:
30705 * docs/plugins/inspect/plugin-ximagesink.xml:
30706 * docs/plugins/inspect/plugin-xvimagesink.xml:
30707 * win32/common/config.h:
30709 Original commit message from CVS:
30712 2006-02-08 18:37:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30714 configure.ac: Drat. Bump libtool version number for new API.
30715 Original commit message from CVS:
30717 Drat. Bump libtool version number for new API.
30718 Prelease 0.10.2.3 (of 0.10.3)
30720 2006-02-08 15:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30722 0.10.2.2 prerelease (of 0.10.3).
30723 Original commit message from CVS:
30725 * win32/common/config.h:
30726 0.10.2.2 prerelease (of 0.10.3).
30728 2006-02-08 15:50:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30730 gst/tcp/gsttcpclientsrc.c: Revert Andy's newsegment change pending a more correct fix.
30731 Original commit message from CVS:
30732 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
30733 Revert Andy's newsegment change pending a more correct
30736 2006-02-08 12:46:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30753 Original commit message from CVS:
30756 2006-02-08 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30758 * gst/tcp/gstmultifdsink.c:
30760 Original commit message from CVS:
30763 2006-02-08 09:20:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30765 gst/typefind/gsttypefindfunctions.c: detect more files as 3gp group and reorder the iso file formats
30766 Original commit message from CVS:
30768 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
30769 (qt_type_find), (plugin_init):
30770 detect more files as 3gp
30771 group and reorder the iso file formats
30773 2006-02-07 18:32:00 +0000 Tim-Philipp Müller <tim@centricular.net>
30775 ext/vorbis/vorbis.c: Register musicbrainz tags, so apps don't have to.
30776 Original commit message from CVS:
30777 * ext/vorbis/vorbis.c: (plugin_init):
30778 Register musicbrainz tags, so apps don't have to.
30780 2006-02-07 17:44:53 +0000 Tim-Philipp Müller <tim@centricular.net>
30782 gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
30783 Original commit message from CVS:
30784 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
30785 (gst_tag_to_vorbis_tag):
30786 Make sure we called gst_tag_register_musicbrainz_tags()
30787 before possibly mapping a vorbiscomment string from/to a
30790 2006-02-07 16:16:41 +0000 Tim-Philipp Müller <tim@centricular.net>
30792 gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
30793 Original commit message from CVS:
30794 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
30795 In case we can't find the required number of consecutive
30796 mpeg audio frames to positively identify an MPEG audio
30797 stream, check if there's at least a valid mpeg audio
30798 frame right at offset 0 and if so suggest mpeg/audio
30799 caps with a very low probability (#153004).
30801 2006-02-07 15:52:26 +0000 Andy Wingo <wingo@pobox.com>
30803 gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to a TIME segment if we get timestamped buffers. Requir...
30804 Original commit message from CVS:
30805 2006-02-07 Andy Wingo <wingo@pobox.com>
30806 * gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
30807 a TIME segment if we get timestamped buffers. Requires recent
30808 fixes in core to work properly.
30810 2006-02-07 14:57:46 +0000 Tim-Philipp Müller <tim@centricular.net>
30812 gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
30813 Original commit message from CVS:
30814 * gst/playback/gstplaybasebin.c: (prepare_output):
30815 Don't print the URI as part of the error message, it
30816 makes error dialogs look rather ugly, especially if
30817 the URI is very long or has characters in it that
30820 2006-02-07 13:11:31 +0000 Tim-Philipp Müller <tim@centricular.net>
30822 gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
30823 Original commit message from CVS:
30824 * gst/playback/gstplaybasebin.c: (prepare_output):
30825 Error out if we have only text or subtitles, but nothing
30826 else. Also error out if we have subtitles but no video
30829 2006-02-07 11:44:39 +0000 Tim-Philipp Müller <tim@centricular.net>
30831 ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
30832 Original commit message from CVS:
30833 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
30834 Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
30835 Post an error message on the bus when we encounter an
30836 error, which will hopefully be more meaningful than the
30837 'Internal Flow Error' message users get to see if we
30838 just return GST_FLOW_ERROR.
30840 2006-02-07 11:28:04 +0000 Andy Wingo <wingo@pobox.com>
30842 configure.ac (GST_MAJORMINOR): Update core version req to 0.10.2.2, for the collectpads API addition (#330244).
30843 Original commit message from CVS:
30844 2006-02-07 Andy Wingo <wingo@pobox.com>
30845 * configure.ac (GST_MAJORMINOR): Update core version req to
30846 0.10.2.2, for the collectpads API addition (#330244).
30848 2006-02-06 19:09:26 +0000 Tim-Philipp Müller <tim@centricular.net>
30850 ext/gnomevfs/gstgnomevfs.c: Return FALSE from plugin_init() when GnomeVFS can't be initialised for some reason (#3284...
30851 Original commit message from CVS:
30852 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
30853 Return FALSE from plugin_init() when GnomeVFS can't
30854 be initialised for some reason (#328423).
30856 2006-02-06 13:26:54 +0000 Julien Moutte <julien@moutte.net>
30858 ext/pango/gsttextoverlay.c: Stick to seeking theory until i find the bug.
30859 Original commit message from CVS:
30860 2006-02-06 Julien MOUTTE <julien@moutte.net>
30861 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
30862 Stick to seeking theory until i find the bug.
30863 * gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
30865 2006-02-06 12:38:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30867 Make theoraenc and the tests leak free. Like, really.
30868 Original commit message from CVS:
30869 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
30870 (theora_enc_finalize), (theora_enc_sink_setcaps),
30871 (theora_set_header_on_caps), (theora_enc_chain),
30872 (theora_enc_change_state):
30873 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
30874 Make theoraenc and the tests leak free. Like, really.
30876 2006-02-05 23:31:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30878 Add a finalize method to ensure we clean up state even if someone omitted the state change back to NULL.
30879 Original commit message from CVS:
30880 (theora_enc_finalize), (theora_enc_sink_setcaps):
30881 Add a finalize method to ensure we clean up state even if
30882 someone omitted the state change back to NULL.
30883 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
30884 (gst_vorbisenc_chain):
30885 Free some more leaked bits.
30886 * tests/check/pipelines/theoraenc.c: (start_pipeline),
30888 Wait for state changes to happen if they're ASYNC.
30889 This ought to teach those fancy pants buildbots a lesson.
30891 2006-02-05 22:47:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30893 gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
30894 Original commit message from CVS:
30895 * gst-libs/gst/tag/gstid3tag.c:
30896 Add mapping for ID3 International Standard Recording Code
30899 2006-02-05 22:44:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30901 ext/vorbis/vorbisenc.c: Don't leak tag names.
30902 Original commit message from CVS:
30903 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
30904 Don't leak tag names.
30906 2006-02-05 18:22:01 +0000 Tim-Philipp Müller <tim@centricular.net>
30908 Split libgsttag docs into multiple sections.
30909 Original commit message from CVS:
30910 * docs/libs/gst-plugins-base-libs-docs.sgml:
30911 * docs/libs/gst-plugins-base-libs-sections.txt:
30912 * gst-libs/gst/tag/gstid3tag.c:
30913 * gst-libs/gst/tag/gstvorbistag.c:
30914 * gst-libs/gst/tag/tags.c:
30915 Split libgsttag docs into multiple sections.
30917 2006-02-05 18:01:33 +0000 Tim-Philipp Müller <tim@centricular.net>
30919 Add libgsttag to the docs.
30920 Original commit message from CVS:
30921 * docs/libs/Makefile.am:
30922 * docs/libs/gst-plugins-base-libs-docs.sgml:
30923 * docs/libs/gst-plugins-base-libs-sections.txt:
30924 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
30925 * gst-libs/gst/tag/gstvorbistag.c:
30926 * gst-libs/gst/tag/tag.h:
30927 * gst-libs/gst/tag/tags.c:
30928 Add libgsttag to the docs.
30930 2006-02-05 17:21:23 +0000 Julien Moutte <julien@moutte.net>
30932 ext/pango/gsttextoverlay.c: Fix clockoverlay.
30933 Original commit message from CVS:
30934 2006-02-05 Julien MOUTTE <julien@moutte.net>
30935 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
30936 (gst_text_overlay_init), (gst_text_overlay_src_event),
30937 (gst_text_overlay_collected): Fix clockoverlay.
30939 2006-02-05 17:15:17 +0000 Tim-Philipp Müller <tim@centricular.net>
30941 docs/libs/compiling.sgml: Fix typo: it's pkg-config, not pkg-gconfig
30942 Original commit message from CVS:
30943 * docs/libs/compiling.sgml:
30944 Fix typo: it's pkg-config, not pkg-gconfig
30945 * docs/libs/gst-plugins-base-libs-docs.sgml:
30946 * docs/libs/gst-plugins-base-libs-sections.txt:
30947 * docs/libs/tmpl/gstgconf.sgml:
30948 There is no libgstgconf in 0.10, remove it
30951 2006-02-05 16:03:48 +0000 Julien Moutte <julien@moutte.net>
30953 docs/libs/tmpl/gstcolorbalance.sgml: Updated.
30954 Original commit message from CVS:
30955 2006-02-05 Julien MOUTTE <julien@moutte.net>
30956 * docs/libs/tmpl/gstcolorbalance.sgml: Updated.
30957 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
30958 (gst_text_overlay_src_event), (gst_text_overlay_collected):
30959 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
30960 (gst_sub_parse_class_init), (gst_sub_parse_init),
30961 (gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
30962 (parse_mpsub), (parser_state_init), (handle_buffer),
30963 (gst_sub_parse_chain), (gst_sub_parse_sink_event),
30965 * gst/subparse/gstsubparse.h: Introduce seeking code.
30967 2006-02-05 15:14:06 +0000 Tim-Philipp Müller <tim@centricular.net>
30969 gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
30970 Original commit message from CVS:
30971 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
30972 Add comment about LANGUAGE tag inconsistency (we want
30973 ISO-639-1, but extract three-letter identifiers?)
30975 Add two translatable files.
30977 2006-02-05 14:59:28 +0000 Tim-Philipp Müller <tim@centricular.net>
30979 gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
30980 Original commit message from CVS:
30981 * gst-libs/gst/tag/Makefile.am:
30982 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
30983 * gst-libs/gst/tag/tag.h:
30984 * gst-libs/gst/tag/tags.c:
30985 (gst_tag_register_musicbrainz_tags_internal),
30986 (gst_tag_register_musicbrainz_tags):
30987 Forward-port some tags stuff from the 0.8 branch. This is
30988 mostly the addition of musicbrainz tags and their mapping
30989 to vorbistags, and a vorbistag mapping of the language tag.
30991 2006-02-05 12:06:25 +0000 Julien Moutte <julien@moutte.net>
30993 gst/playback/gstplaybin.c: Fix broken code refactoring.
30994 Original commit message from CVS:
30995 2006-02-05 Julien MOUTTE <julien@moutte.net>
30996 * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
30999 2006-02-05 03:05:41 +0000 David Schleef <ds@schleef.org>
31001 Add Dirac typefinding and add dirac format to oggmux.
31002 Original commit message from CVS:
31003 * ext/ogg/gstoggmux.c:
31004 * gst/typefind/gsttypefindfunctions.c:
31005 Add Dirac typefinding and add dirac format to oggmux.
31007 2006-02-04 07:49:03 +0000 Michael Smith <msmith@xiph.org>
31010 Improve error message for liboil missingness.
31011 Original commit message from CVS:
31012 Improve error message for liboil missingness.
31014 2006-02-03 19:23:41 +0000 Tim-Philipp Müller <tim@centricular.net>
31016 gst/playback/gstdecodebin.c: Don't put essential function call into g_return_*() macro, otherwise it'll all be replac...
31017 Original commit message from CVS:
31018 * gst/playback/gstdecodebin.c: (try_to_link_1):
31019 Don't put essential function call into
31020 g_return_*() macro, otherwise it'll all be
31021 replaced by NOOPs when compiling with
31022 G_DISABLE_CHECKS defined.
31024 2006-02-03 17:45:44 +0000 Edgard Lima <edgard.lima@indt.org.br>
31027 * ext/ogg/gstoggdemux.c:
31028 * ext/ogg/gstoggparse.c:
31029 * gst/tcp/gsttcpserversink.c:
31030 * sys/v4l/v4lsrc_calls.c:
31031 * sys/v4l/v4lsrc_calls.h:
31032 Just make it compile with --disable-gst-debug.
31033 Original commit message from CVS:
31034 Just make it compile with --disable-gst-debug.
31036 2006-02-03 12:51:47 +0000 Wim Taymans <wim.taymans@gmail.com>
31038 ext/alsa/gstalsasink.*: Add lock to protect alsa calls.
31039 Original commit message from CVS:
31040 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
31041 (gst_alsasink_class_init), (gst_alsasink_init),
31042 (gst_alsasink_write), (gst_alsasink_reset):
31043 * ext/alsa/gstalsasink.h:
31044 Add lock to protect alsa calls.
31045 Implement reset to flush samples ASAP, does not work
31048 2006-02-02 18:18:31 +0000 Wim Taymans <wim.taymans@gmail.com>
31050 gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
31051 Original commit message from CVS:
31052 * gst-libs/gst/audio/gstbaseaudiosink.c:
31053 (gst_base_audio_sink_provide_clock):
31054 Ugh.. getting late I guess...
31056 2006-02-02 18:13:26 +0000 Wim Taymans <wim.taymans@gmail.com>
31058 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
31059 Original commit message from CVS:
31060 * gst-libs/gst/audio/gstbaseaudiosink.c:
31061 (gst_base_audio_sink_provide_clock),
31062 (gst_base_audio_sink_set_property),
31063 (gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
31064 Don't try to provide a clock when we are not negotiated since
31065 we might not be able to make it run.
31067 2006-02-02 17:51:48 +0000 Tim-Philipp Müller <tim@centricular.net>
31069 gst/playback/gstdecodebin.c: Unlinking two source pads is ... hard.
31070 Original commit message from CVS:
31071 * gst/playback/gstdecodebin.c: (try_to_link_1):
31072 Unlinking two source pads is ... hard.
31074 2006-02-02 12:14:35 +0000 Wim Taymans <wim.taymans@gmail.com>
31076 gst-libs/gst/audio/TODO: Updated.
31077 Original commit message from CVS:
31078 * gst-libs/gst/audio/TODO:
31080 * gst-libs/gst/audio/gstbaseaudiosink.c:
31081 (gst_base_audio_sink_drain), (gst_base_audio_sink_event):
31082 On EOS, wait till the last sample is played before posting EOS.
31084 2006-02-02 08:53:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31086 * tests/check/pipelines/theoraenc.c:
31087 comment on my understanding
31088 Original commit message from CVS:
31089 comment on my understanding
31091 2006-02-02 08:47:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31094 * tests/check/pipelines/theoraenc.c:
31095 reformat to fit 80 chars
31096 Original commit message from CVS:
31097 reformat to fit 80 chars
31099 2006-02-02 00:04:37 +0000 Kai Vehmanen <kv2004@eca.cx>
31101 gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
31102 Original commit message from CVS:
31103 2006-02-01 Philippe Kalaf <burger at speedy dot org>
31104 * gst-libs/gst/rtp/gstbasertpdepayload.c:
31105 Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
31106 setting queue_delay to zero. Also avoid thread being started if
31107 queue_delay is zero.
31109 2006-02-01 14:51:29 +0000 Tim-Philipp Müller <tim@centricular.net>
31111 gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
31112 Original commit message from CVS:
31113 * gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
31114 Make test work again by connecting fakesinks to each decoded pad,
31115 which makes the pipeline wait until each fakesink has a buffer
31116 queued before going to PAUSED state. At that point we know the
31117 decodebin pads are negotiated.
31119 2006-02-01 11:59:47 +0000 Tim-Philipp Müller <tim@centricular.net>
31121 gst/: Pass unhandled queries to the parent class's query function.
31122 Original commit message from CVS:
31123 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
31124 (gst_cdda_base_src_handle_event):
31125 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
31126 Pass unhandled queries to the parent class's query function.
31128 2006-02-01 11:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
31130 Pass unhandled queries upstream instead of just dropping them (#326447). Also, fix supported query types list for som...
31131 Original commit message from CVS:
31132 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
31133 (gst_ogg_pad_src_query):
31134 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
31135 * ext/theora/theoradec.c: (theora_dec_src_query),
31136 (theora_dec_sink_query):
31137 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
31138 (vorbis_dec_sink_query):
31139 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
31140 (gst_vorbisenc_sink_query):
31141 * gst/adder/gstadder.c: (gst_adder_query):
31142 Pass unhandled queries upstream instead of just
31143 dropping them (#326447). Also, fix supported
31144 query types list for some elements.
31146 2006-02-01 09:58:15 +0000 Tim-Philipp Müller <tim@centricular.net>
31148 gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
31149 Original commit message from CVS:
31150 * gst/typefind/gsttypefindfunctions.c: (au_type_find),
31151 (paris_type_find), (ilbc_type_find), (plugin_init):
31152 Fix typefinding for audio/x-au, audio/x-paris and
31153 audio/iLBC-sh. We cannot use the START_WITH macros
31154 here, because there can only be one typefind factory
31155 with the same name (caps), so the second one would
31156 replace the first one and the first one would never
31157 be called when doing typefinding (see #161712).
31159 2006-01-31 19:25:10 +0000 Wim Taymans <wim.taymans@gmail.com>
31161 ext/vorbis/vorbisdec.c: Use scale_int when we can, add some more scaling.
31162 Original commit message from CVS:
31163 * ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
31164 (vorbis_handle_header_packet), (vorbis_dec_push),
31165 (vorbis_handle_data_packet):
31166 Use scale_int when we can, add some more scaling.
31167 Check packettype before parsing it.
31169 2006-01-31 17:44:35 +0000 Wim Taymans <wim.taymans@gmail.com>
31171 ext/theora/theoradec.c: Call right _scale functions.
31172 Original commit message from CVS:
31173 * ext/theora/theoradec.c: (_theora_granule_time),
31174 (theora_dec_src_convert), (theora_dec_sink_convert):
31175 Call right _scale functions.
31176 Use parameter instead of some other random value.
31178 2006-01-31 17:27:00 +0000 Wim Taymans <wim.taymans@gmail.com>
31180 ext/theora/theoradec.c: Use higher precision timestamps calculation.
31181 Original commit message from CVS:
31182 * ext/theora/theoradec.c: (_theora_granule_frame),
31183 (_theora_granule_time), (_inc_granulepos),
31184 (theora_dec_src_convert), (theora_dec_sink_convert),
31185 (theora_handle_type_packet), (theora_handle_data_packet),
31186 (theora_dec_chain):
31187 Use higher precision timestamps calculation.
31188 Convert some other conversions to _scale.
31190 2006-01-31 17:19:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31192 gst/: initialize gst_controller before using
31193 Original commit message from CVS:
31194 * gst/audiotestsrc/gstaudiotestsrc.c:
31195 (gst_audio_test_src_create_sine_table), (plugin_init):
31196 * gst/volume/gstvolume.c: (plugin_init):
31197 initialize gst_controller before using
31199 2006-01-31 16:26:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31201 tests/check/pipelines/: Define constant using G_GINT64_CONSTANT to avoid errors when passing it around - otherwise it...
31202 Original commit message from CVS:
31203 * tests/check/pipelines/theoraenc.c:
31204 * tests/check/pipelines/vorbisenc.c:
31205 Define constant using G_GINT64_CONSTANT to avoid errors when
31206 passing it around - otherwise it gets truncated to 32 bits.
31207 Fixes failing tests.
31209 2006-01-31 15:36:13 +0000 Andy Wingo <wingo@pobox.com>
31211 sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the caps being set doesn't have a framerate value. Basic...
31212 Original commit message from CVS:
31213 2006-01-31 Andy Wingo <wingo@pobox.com>
31214 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
31215 caps being set doesn't have a framerate value. Basically a stopgap
31217 * ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
31218 technically correct enough to put into core though.
31219 (gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
31220 DURATION. Fixes theoraenc ! oggmux.
31221 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
31222 fraction, not double.
31224 2006-01-31 12:23:35 +0000 Christian Schaller <uraeus@gnome.org>
31226 * gst-plugins-base.spec.in:
31227 update with latest files
31228 Original commit message from CVS:
31229 update with latest files
31231 2006-01-30 23:42:54 +0000 Sébastien Moutte <sebastien@moutte.net>
31233 win32/vs7: add vs7 project files created by Sergey Scobich
31234 Original commit message from CVS:
31236 add vs7 project files created by Sergey Scobich
31238 2006-01-30 22:18:53 +0000 Sébastien Moutte <sebastien@moutte.net>
31240 win32/vs8: add vs8 project files created by Sergey Scobich
31241 Original commit message from CVS:
31243 add vs8 project files created by Sergey Scobich
31245 2006-01-30 19:22:22 +0000 Andy Wingo <wingo@pobox.com>
31247 ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare timestamp + duration, not just timestamp -- ogg pages should ...
31248 Original commit message from CVS:
31249 2006-01-30 Andy Wingo <wingo@pobox.com>
31250 * ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
31251 timestamp + duration, not just timestamp -- ogg pages should be
31252 ordered by stop time. Necessary fix given the change in vorbis
31255 2006-01-30 19:21:07 +0000 Andy Wingo <wingo@pobox.com>
31258 * ext/theora/gsttheoraenc.h:
31259 * ext/theora/theoraenc.c:
31260 * tests/check/pipelines/theoraenc.c:
31261 ext/theora/theoraenc.c (theora_enc_sink_setcaps)
31262 Original commit message from CVS:
31263 2006-01-30 Andy Wingo <wingo@pobox.com>
31264 * ext/theora/theoraenc.c (theora_enc_sink_setcaps)
31265 (gst_theora_enc_init): Pull the granule shift out of the encoder.
31266 (granulepos_add): New function, handles the messiness of adjusting
31268 (theora_buffer_from_packet):
31269 (theora_enc_chain):
31270 (theora_enc_sink_event): Use granulepos_add, not +.
31271 * tests/check/pipelines/theoraenc.c
31272 (check_buffer_granulepos_from_starttime): Just check the frame
31273 count, not the actual granulepos -- we can't dictate to the
31274 encoder when it should be placing keyframes.
31276 2006-01-30 18:17:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31278 ext/gnomevfs/gstgnomevfssrc.c: SERVICE_NOT_AVAILABLE happens for example when you're trying to play an http:// stream...
31279 Original commit message from CVS:
31280 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
31281 SERVICE_NOT_AVAILABLE happens for example when you're trying to
31282 play an http:// stream from a server that's not serving
31284 2006-01-30 17:08:11 +0000 Andy Wingo <wingo@pobox.com>
31286 tests/check/pipelines/: Totally remove the UINT64_CONSTANT macro, doesn't appear to be needed or available.
31287 Original commit message from CVS:
31288 2006-01-30 Andy Wingo <wingo@pobox.com>
31289 * tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
31290 * tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
31291 remove the UINT64_CONSTANT macro, doesn't appear to be needed or
31294 2006-01-30 17:01:54 +0000 Andy Wingo <wingo@pobox.com>
31296 ext/theora/: Same changes as were done to vorbisenc, although theoraenc was timestamping correctly. Added handling of...
31297 Original commit message from CVS:
31298 2006-01-30 Andy Wingo <wingo@pobox.com>
31299 * ext/theora/gsttheoraenc.h:
31300 * ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
31301 although theoraenc was timestamping correctly. Added handling of
31302 streams that start with nonzero timestamps.
31303 * tests/check/Makefile.am:
31304 * tests/check/pipelines/theoraenc.c: New file, basically does same
31305 tests as vorbisenc.
31306 * tests/check/pipelines/vorbisenc.c: I claim these bugs.
31308 2006-01-30 16:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
31310 gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
31311 Original commit message from CVS:
31312 * gst-libs/gst/audio/gstaudiosink.c:
31313 (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
31314 (gst_audioringbuffer_pause):
31315 Implement pause that does not wait for completion.
31316 * gst-libs/gst/audio/gstbaseaudiosink.c:
31317 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
31318 Don't drop buffers when going to PAUSED but perform preroll on
31319 remaining samples now that core base class supports this.
31320 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
31321 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
31322 (gst_ring_buffer_commit):
31323 Pause should not signal waiters.
31324 Implement return value of _commit correctly.
31326 2006-01-30 15:01:28 +0000 Andy Wingo <wingo@pobox.com>
31328 tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
31329 Original commit message from CVS:
31330 2006-01-30 Andy Wingo <wingo@pobox.com>
31331 * tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
31332 * ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
31333 updated to timestamp from the first sample, not the last.
31334 (gst_vorbisenc_buffer_from_header_packet): New function, takes
31335 special care of granulepos and timestamp for header packets.
31336 (gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
31337 when the first buffer has a nonzero timestamp.
31338 * ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
31339 (GstVorbisEnc.subgranule_offset): New members. Take care of the
31340 case when the first audio buffer we get has a nonzero timestamp.
31341 (GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
31342 properly timestamp vorbis buffers with the time of the first
31343 sample, not the last.
31344 * ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
31345 vorbis_granule_time_copy -- now it takes the granule/subgranule
31346 offset into account.
31347 * tests/check/pipelines/vorbisenc.c: New test for correctness of
31348 timestamps, durations, and granulepos on buffers produced by
31351 2006-01-30 14:42:28 +0000 Eric Jonas <jonas@mit.edu>
31353 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Patch from Eric Jonas to support conversions to/from UYVY (Fixes: #324626)
31354 Original commit message from CVS:
31355 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
31356 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
31357 Patch from Eric Jonas to support conversions to/from UYVY
31360 2006-01-30 08:11:14 +0000 Julien Moutte <julien@moutte.net>
31362 gst/playback/: Implement subtitles.
31363 Original commit message from CVS:
31364 2006-01-30 Julien MOUTTE <julien@moutte.net>
31365 * gst/playback/gstplaybasebin.c: (group_commit),
31367 (setup_subtitle), (setup_source), (set_active_source):
31368 * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
31369 (gen_text_element), (gen_audio_element), (gen_vis_element),
31370 (remove_sinks), (add_sink), (setup_sinks): Implement subtitles.
31372 2006-01-29 19:13:39 +0000 Sébastien Moutte <sebastien@moutte.net>
31374 gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
31375 Original commit message from CVS:
31376 * gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
31377 * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
31378 use of gst_guint64_to_gdouble to be compliant with vs6
31379 * gst/playback/gstdecodebin.c: (try_to_link_1)
31380 * gst/videorate/videorate.c: (gst_video_rate_blank_data)
31381 use of G_GINT64_CONSTANT for int64 constants
31382 * win32/common/libgstinterfaces.def:
31383 export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
31385 update and add new project files
31387 2006-01-29 18:21:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31389 add a win32-update rule like in core, and copy over enumtypes files
31390 Original commit message from CVS:
31393 * win32/common/interfaces-enumtypes.c:
31394 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
31395 (gst_mixer_track_flags_get_type),
31396 (gst_tuner_channel_flags_get_type):
31397 * win32/common/interfaces-enumtypes.h:
31398 * win32/common/multichannel-enumtypes.c:
31399 (gst_audio_channel_position_get_type):
31400 * win32/common/multichannel-enumtypes.h:
31401 add a win32-update rule like in core, and copy over enumtypes files
31403 2006-01-29 18:07:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31406 generate win32/common/config.h
31407 Original commit message from CVS:
31408 generate win32/common/config.h
31410 2006-01-29 18:05:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31412 win32/: add config files just like in core
31413 Original commit message from CVS:
31415 * win32/common/config.h:
31416 * win32/common/config.h.in:
31417 add config files just like in core
31419 2006-01-28 18:22:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31421 ext/alsa/: Update all error messages. All of them should either use the default translated message, or actually prov...
31422 Original commit message from CVS:
31423 * ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
31424 (set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
31425 (gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
31426 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
31427 (set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
31428 (gst_alsasrc_unprepare), (gst_alsasrc_read):
31429 Update all error messages. All of them should either use
31430 the default translated message, or actually provide a
31431 translatable string.
31432 Make the string for channel count problems meaningful.
31434 2006-01-28 18:19:18 +0000 Tim-Philipp Müller <tim@centricular.net>
31436 gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
31437 Original commit message from CVS:
31438 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
31439 Make gcc-4.1 happy (part of #327357).
31441 2006-01-28 16:35:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31443 sys/v4l/v4l_calls.c: check for and throw RESOURCE_BUSY
31444 Original commit message from CVS:
31445 * sys/v4l/v4l_calls.c: (gst_v4l_open):
31446 check for and throw RESOURCE_BUSY
31448 2006-01-28 02:13:14 +0000 David Schleef <ds@schleef.org>
31450 gst/videoscale/vs_scanline.c: Oops, *that's* why I never checked in this change -- it requires liboil features not in...
31451 Original commit message from CVS:
31452 * gst/videoscale/vs_scanline.c: Oops, *that's* why I never
31453 checked in this change -- it requires liboil features not
31454 in 0.3.6. Revert parts.
31456 2006-01-27 23:40:19 +0000 David Schleef <ds@schleef.org>
31458 update liboil requirement to 0.3.6
31459 Original commit message from CVS:
31461 * configure.ac: update liboil requirement to 0.3.6
31462 * gst/videoscale/Makefile.am:
31463 * gst/videoscale/vs_scanline.c: liboilify
31465 2006-01-27 17:00:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31467 ext/libvisual/visual.c: When pad_alloc returns a GstFlowReturn other than GST_FLOW_OK, make sure it is passed upstream.
31468 Original commit message from CVS:
31469 * ext/libvisual/visual.c: (get_buffer):
31470 When pad_alloc returns a GstFlowReturn other
31471 than GST_FLOW_OK, make sure it is passed upstream.
31473 2006-01-27 01:36:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31475 ext/alsa/gstalsasink.c: Free the device name string.
31476 Original commit message from CVS:
31477 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
31478 (gst_alsasink_class_init):
31479 Free the device name string.
31480 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
31481 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
31482 (gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
31483 Don't remove a pad from the collectpads structure until it
31484 is released - it's a request pad, and may receive data again
31485 if the element gets moved back to PLAYING state.
31486 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
31487 Ensure we turn on double buffering on the Xv port, and
31488 set the colour key to something dark and mysterious that
31491 2006-01-27 01:06:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31493 ext/: - a library should not call setlocale. see Libraries node in gettext manual
31494 Original commit message from CVS:
31495 * ext/alsa/gstalsaplugin.c: (plugin_init):
31496 * ext/cdparanoia/gstcdparanoiasrc.c:
31497 (gst_cd_paranoia_src_base_init), (plugin_init):
31498 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
31499 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
31500 - a library should not call setlocale. see Libraries node in
31502 - make sure all plugins that use translation do bindtextdomain
31503 to point to the localedir
31504 * gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
31505 (setup_sinks), (plugin_init):
31506 all this, and check for NULL when creating sinks
31508 2006-01-26 23:21:31 +0000 Julien Moutte <julien@moutte.net>
31510 gst/subparse/gstsubparse.c: Make typefinding of subtitles work again.
31511 Original commit message from CVS:
31512 2006-01-27 Julien MOUTTE <julien@moutte.net>
31513 * gst/subparse/gstsubparse.c: (gst_subparse_type_find),
31514 (plugin_init): Make typefinding of subtitles work again.
31516 2006-01-26 20:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
31518 gst/typefind/gsttypefindfunctions.c: Backport a bunch of typefinding fixes from the 0.8 branch.
31519 Original commit message from CVS:
31520 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
31521 (mp3_type_frame_length_from_header), (mp3_type_find),
31522 (wavpack_type_find), (m4a_type_find), (ircam_type_find),
31524 Backport a bunch of typefinding fixes from the 0.8 branch.
31525 Also, improve wavpack typefinding: if we can't peek the
31526 entire wavpack block, try to parse the bits we can get and
31527 see if we find what we're looking for in those.
31529 2006-01-26 19:17:38 +0000 Julien Moutte <julien@moutte.net>
31531 sys/: Handle some more cases of pixel aspect ratio.
31532 Original commit message from CVS:
31533 2006-01-26 Julien MOUTTE <julien@moutte.net>
31534 * sys/ximage/ximagesink.c:
31535 (gst_ximagesink_calculate_pixel_aspect_ratio):
31536 * sys/xvimage/xvimagesink.c:
31537 (gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
31538 more cases of pixel aspect ratio.
31540 2006-01-26 13:09:24 +0000 Edward Hervey <bilboed@bilboed.com>
31542 gst/playback/gstdecodebin.c: Also consider the flush-start and tag events as unblockers for the pad probes.
31543 Original commit message from CVS:
31544 * gst/playback/gstdecodebin.c: (pad_probe):
31545 Also consider the flush-start and tag events as unblockers
31546 for the pad probes.
31548 2006-01-26 12:32:58 +0000 Julien Moutte <julien@moutte.net>
31550 gst/playback/gstplaybin.c: On the fly visualisation switch, works disabling, enabling as well but it won't be able to...
31551 Original commit message from CVS:
31552 2006-01-26 Julien MOUTTE <julien@moutte.net>
31553 * gst/playback/gstplaybin.c: (gst_play_bin_init),
31554 (gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
31555 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
31556 On the fly visualisation switch, works disabling, enabling as
31557 well but it won't be able to enable vis in a playbin that was
31558 created with no visualisation.
31560 2006-01-25 10:50:32 +0000 Wim Taymans <wim.taymans@gmail.com>
31562 gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
31563 Original commit message from CVS:
31564 * gst-libs/gst/audio/gstbaseaudiosink.c:
31565 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
31566 Undo previous commit, it breaks resume after pause.
31568 2006-01-25 09:27:01 +0000 Wim Taymans <wim.taymans@gmail.com>
31570 gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
31571 Original commit message from CVS:
31572 * gst-libs/gst/audio/gstbaseaudiosink.c:
31573 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
31574 (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
31576 Post error when caps cannot be parsed.
31577 Resync on discontinuity in the stream.
31578 Clip samples to segment boundaries.
31579 return WRONG_STATE sooner when we are flushing.
31580 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
31581 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
31582 Make audiosrc operate in TIME.
31583 Set TIMESTAMP and DURATION on buffers.
31585 2006-01-24 21:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
31587 tests/examples/seek/seek.c: Output tag messages as well.
31588 Original commit message from CVS:
31589 * tests/examples/seek/seek.c: (main):
31590 Output tag messages as well.
31592 2006-01-23 15:05:24 +0000 Edward Hervey <bilboed@bilboed.com>
31594 gst/playback/gstdecodebin.c: Replace GstPadBlockCallback with pad probes that detect first buffer AND eos before remo...
31595 Original commit message from CVS:
31596 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
31597 (free_pad_probes), (remove_fakesink), (pad_probe),
31598 (close_pad_link), (gst_decode_bin_change_state):
31599 Replace GstPadBlockCallback with pad probes that detect
31600 first buffer AND eos before removing fakesink.
31601 Fixes hang with demuxers doing EOS while pre-rolling.
31604 2006-01-23 10:10:36 +0000 Jens Granseuer <jensgr@gmx.net>
31606 GCC 2.95 fixes (#328263).
31607 Original commit message from CVS:
31608 2006-01-23 Andy Wingo <wingo@pobox.com>
31609 * ext/alsa/gstalsasink.c:
31610 * gst-libs/gst/rtp/gstbasertpdepayload.c:
31611 (gst_base_rtp_depayload_setcaps),
31612 (gst_base_rtp_depayload_add_to_queue),
31613 (gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
31614 Patch by: Jens Granseuer <jensgr at gmx dot net>
31616 2006-01-22 17:24:02 +0000 Julien Moutte <julien@moutte.net>
31618 sys/: Playbin keeps some ref to some frames. We might get a frame destroyed after changing state to
31619 Original commit message from CVS:
31620 2006-01-22 Julien MOUTTE <julien@moutte.net>
31621 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
31622 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
31623 (gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
31624 frames. We might get a frame destroyed after changing state to
31625 NULL, adding a safety check on xcontext.
31627 2006-01-22 14:50:53 +0000 Tim-Philipp Müller <tim@centricular.net>
31629 gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
31630 Original commit message from CVS:
31631 * gst-libs/gst/interfaces/xoverlay.c:
31632 Fix prepare-xwindow-id code example in the docs - we need to
31633 ignore all messages that aren't element messages as well.
31635 2006-01-21 22:40:03 +0000 Julien Moutte <julien@moutte.net>
31637 sys/xvimage/xvimagesink.c: I think one day i'll completely undestand how caps negotiation is supposed to work. This r...
31638 Original commit message from CVS:
31639 2006-01-21 Julien MOUTTE <julien@moutte.net>
31640 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
31641 I think one day i'll completely undestand how caps negotiation
31642 is supposed to work. This refactoring handles buffer_alloc
31643 called with caps we can't handle. We definitely don't want a
31644 set_caps with those caps, so we define and allocate a buffer
31645 we would like to receive.
31647 2006-01-20 19:10:26 +0000 Christian Schaller <uraeus@gnome.org>
31651 up automake requirement to 1.7
31652 Original commit message from CVS:
31653 up automake requirement to 1.7
31655 2006-01-19 10:59:51 +0000 Tim-Philipp Müller <tim@centricular.net>
31657 gst/playback/gstplaybasebin.c: Free iterator when done.
31658 Original commit message from CVS:
31659 * gst/playback/gstplaybasebin.c: (setup_source):
31660 Free iterator when done.
31662 2006-01-17 11:43:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31664 gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
31665 Original commit message from CVS:
31666 * gst-libs/gst/audio/gstbaseaudiosink.c:
31667 (gst_base_audio_sink_render):
31668 Fix playback of non-synchronised streams by assuming a rate
31669 of 1.0 instead of a random one.
31670 Makes this work again:
31671 gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
31672 endianness=(int)4321, signed=(boolean)true, width=(int)16,
31673 depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
31674 audioresample ! alsasink
31676 2006-01-16 21:01:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31680 Original commit message from CVS:
31683 === release 0.10.2 ===
31685 2006-01-16 20:59:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31691 * docs/plugins/gst-plugins-base-plugins.args:
31692 * docs/plugins/inspect/plugin-adder.xml:
31693 * docs/plugins/inspect/plugin-alsa.xml:
31694 * docs/plugins/inspect/plugin-audioconvert.xml:
31695 * docs/plugins/inspect/plugin-audiorate.xml:
31696 * docs/plugins/inspect/plugin-audioresample.xml:
31697 * docs/plugins/inspect/plugin-audiotestsrc.xml:
31698 * docs/plugins/inspect/plugin-cdparanoia.xml:
31699 * docs/plugins/inspect/plugin-decodebin.xml:
31700 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
31701 * docs/plugins/inspect/plugin-gnomevfs.xml:
31702 * docs/plugins/inspect/plugin-libvisual.xml:
31703 * docs/plugins/inspect/plugin-ogg.xml:
31704 * docs/plugins/inspect/plugin-pango.xml:
31705 * docs/plugins/inspect/plugin-playbin.xml:
31706 * docs/plugins/inspect/plugin-subparse.xml:
31707 * docs/plugins/inspect/plugin-tcp.xml:
31708 * docs/plugins/inspect/plugin-theora.xml:
31709 * docs/plugins/inspect/plugin-typefindfunctions.xml:
31710 * docs/plugins/inspect/plugin-video4linux.xml:
31711 * docs/plugins/inspect/plugin-videorate.xml:
31712 * docs/plugins/inspect/plugin-videoscale.xml:
31713 * docs/plugins/inspect/plugin-videotestsrc.xml:
31714 * docs/plugins/inspect/plugin-volume.xml:
31715 * docs/plugins/inspect/plugin-vorbis.xml:
31716 * docs/plugins/inspect/plugin-ximagesink.xml:
31717 * docs/plugins/inspect/plugin-xvimagesink.xml:
31719 Original commit message from CVS:
31722 2006-01-16 16:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31724 gst/playback/: Comment out broken code that connects to the state-changed signal.
31725 Original commit message from CVS:
31726 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
31727 * gst/playback/gststreamselector.c:
31728 (gst_stream_selector_set_property):
31729 Comment out broken code that connects to the state-changed signal.
31730 At this point, changing current stream selection is broken, but
31731 stuff like gst-launch playbin current-audio=1 works and filters
31732 to the chosen stream.
31734 2006-01-16 15:31:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31736 ext/vorbis/vorbisdec.c: Fix #327216 (null dereference in vorbisdec)
31737 Original commit message from CVS:
31738 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
31739 Fix #327216 (null dereference in vorbisdec)
31741 2006-01-16 15:19:55 +0000 Tim-Philipp Müller <tim@centricular.net>
31743 ext/theora/theoradec.c: Post taglist actually on bus instead of just freeing it (fixes #327114 and totem bug #327080).
31744 Original commit message from CVS:
31745 * ext/theora/theoradec.c: (theora_handle_comment_packet):
31746 Post taglist actually on bus instead of just freeing it
31747 (fixes #327114 and totem bug #327080).
31748 * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
31749 Use gst_element_found_tags_for_pad(), so that the tags
31750 are sent downstream as an event as well.
31752 2006-01-15 10:06:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31754 sys/: move all regularly occurring messages to GST_LOG level add some more object logs
31755 Original commit message from CVS:
31756 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
31757 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
31758 (gst_ximagesink_buffer_alloc):
31759 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
31760 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
31761 (gst_xvimagesink_buffer_alloc):
31762 move all regularly occurring messages to GST_LOG level
31763 add some more object logs
31765 2006-01-14 22:59:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31783 Original commit message from CVS:
31786 2006-01-14 20:46:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31788 ext/ogg/gstoggmux.c: fix a silly segfault
31789 Original commit message from CVS:
31790 2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
31791 * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
31792 fix a silly segfault
31794 2006-01-14 12:52:22 +0000 Tim-Philipp Müller <tim@centricular.net>
31796 Add docs for mixerutils stuff.
31797 Original commit message from CVS:
31798 * docs/libs/gst-plugins-base-libs-docs.sgml:
31799 * docs/libs/gst-plugins-base-libs-sections.txt:
31800 * gst-libs/gst/audio/mixerutils.c:
31801 * gst-libs/gst/audio/mixerutils.h:
31802 Add docs for mixerutils stuff.
31804 2006-01-13 17:17:07 +0000 Tim-Philipp Müller <tim@centricular.net>
31806 gst/playback/gstplaybasebin.c: Fix playback for sources that emit raw audio or raw video streams (e.g.: cd audio sour...
31807 Original commit message from CVS:
31808 * gst/playback/gstplaybasebin.c: (setup_source):
31809 Fix playback for sources that emit raw audio or
31810 raw video streams (e.g.: cd audio sources) (#325984).
31812 2006-01-13 16:45:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31814 gst-libs/gst/audio/mixerutils.c: actually save the element we create
31815 Original commit message from CVS:
31816 * gst-libs/gst/audio/mixerutils.c:
31817 (gst_audio_mixer_filter_do_filter):
31818 actually save the element we create
31820 2006-01-13 16:17:50 +0000 Christian Schaller <uraeus@gnome.org>
31822 * gst-plugins-base.spec.in:
31823 remove version suffix
31824 Original commit message from CVS:
31825 remove version suffix
31827 2006-01-12 14:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
31829 gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
31830 Original commit message from CVS:
31831 * gst-libs/gst/cdda/gstcddabasesrc.c:
31832 (gst_cdda_base_src_handle_track_seek):
31833 No need to post a tag message on the bus when seeking
31834 within the same track, only post it when the current
31837 2006-01-11 18:30:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31839 gst/playback/: Reenable stream selection. These mechanisms need a complete overhaul in the face of 0.8->0.10 changes ...
31840 Original commit message from CVS:
31841 * gst/playback/gstplaybasebin.c: (group_destroy),
31842 (probe_triggered), (new_decoded_pad), (mute_group_type),
31843 (set_active_source):
31844 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
31845 * gst/playback/gststreamselector.c:
31846 (gst_stream_selector_base_init),
31847 (gst_stream_selector_set_property),
31848 (gst_stream_selector_request_new_pad):
31849 Reenable stream selection. These mechanisms need a complete overhaul
31850 in the face of 0.8->0.10 changes though.
31852 2006-01-11 18:03:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31854 ext/ogg/gstoggdemux.c: Change the pad template to src_%d to match the pads that are created from it. decodebin needs ...
31855 Original commit message from CVS:
31856 * ext/ogg/gstoggdemux.c:
31857 Change the pad template to src_%d to match the pads that
31858 are created from it. decodebin needs this information in order
31859 to decide that oggdemux is capable of producing multiple pads
31860 (and hence needs queues inserted).
31861 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
31862 (gst_ogg_mux_collected):
31863 Make debug output more useful by using GST_PTR_FORMAT.
31865 2006-01-11 17:38:35 +0000 Christian Schaller <uraeus@gnome.org>
31867 * gst-plugins-base.spec.in:
31868 update spec.in file
31869 Original commit message from CVS:
31870 update spec.in file
31872 2006-01-11 15:11:20 +0000 Tim-Philipp Müller <tim@centricular.net>
31874 gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
31875 Original commit message from CVS:
31876 Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
31877 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
31878 Set depth and width for alaw/mulaw (fixes #326601).
31880 2006-01-10 23:58:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31882 tests/icles/Makefile.am: don't build the tests if we don't have the libs
31883 Original commit message from CVS:
31884 * tests/icles/Makefile.am:
31885 don't build the tests if we don't have the libs
31887 2006-01-10 18:06:56 +0000 Tim-Philipp Müller <tim@centricular.net>
31889 ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
31890 Original commit message from CVS:
31891 * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
31892 (gst_cd_paranoia_paranoia_callback):
31893 Don't try to free NULL pointers.
31895 2006-01-10 15:47:48 +0000 Edward Hervey <bilboed@bilboed.com>
31897 gst/audiorate/gstaudiorate.c: Add debugging category.
31898 Original commit message from CVS:
31899 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
31900 (gst_audio_rate_change_state), (plugin_init):
31901 Add debugging category.
31903 Add case for incoming buffers without valid offset/offset_end.
31905 2006-01-10 12:25:59 +0000 Michael Smith <msmith@xiph.org>
31907 gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
31908 Original commit message from CVS:
31909 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
31910 Don't leak GCond in audio sources.
31912 2006-01-10 11:49:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31914 gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
31915 Original commit message from CVS:
31916 * gst/playback/gstplaybin.c: (gen_audio_element):
31917 Don't leak an autoaudiosink/alsasink when we generate
31918 a new audio element. (old code, I guess)
31920 2006-01-10 11:04:21 +0000 Michael Smith <msmith@xiph.org>
31922 gst/audiorate/gstaudiorate.c: Support float audio in audiorate.
31923 Original commit message from CVS:
31924 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
31925 Support float audio in audiorate.
31926 Use width rather than depth for selecting sample width.
31928 2006-01-10 10:06:53 +0000 Tim-Philipp Müller <tim@centricular.net>
31930 gst/videotestsrc/videotestsrc.h: Use GLib types here (that way we don't have to include the generated _stdint.h heade...
31931 Original commit message from CVS:
31932 * gst/videotestsrc/videotestsrc.h:
31933 Use GLib types here (that way we don't have to include the
31934 generated _stdint.h header, which makes life easier for win32
31935 folks that don't use autotools for the build) (#325990, patch
31936 by: Sergey Scobich).
31938 2006-01-10 09:38:44 +0000 Tim-Philipp Müller <tim@centricular.net>
31940 gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
31941 Original commit message from CVS:
31942 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
31943 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
31944 (gst_ring_buffer_pause), (wait_segment):
31945 * gst-libs/gst/audio/gstringbuffer.h:
31946 Name (private) union, makes Forte compiler happy (this time
31947 for real) (#324900).
31949 2006-01-09 10:52:33 +0000 Tim-Philipp Müller <tim@centricular.net>
31951 gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
31952 Original commit message from CVS:
31953 * gst-libs/gst/audio/Makefile.am:
31954 Link against libgstinterfaces, needed for mixer
31955 and property probe stuff.
31957 2006-01-09 10:46:52 +0000 Edward Hervey <bilboed@bilboed.com>
31959 gst-libs/gst/Makefile.am:
31960 Original commit message from CVS:
31961 * gst-libs/gst/Makefile.am:
31963 2006-01-09 09:38:34 +0000 Tim-Philipp Müller <tim@centricular.net>
31965 gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
31966 Original commit message from CVS:
31967 * gst-libs/gst/audio/Makefile.am:
31968 * gst-libs/gst/audio/mixerutils.c:
31969 (gst_audio_mixer_filter_do_filter),
31970 (gst_audio_mixer_filter_check_element),
31971 (gst_audio_mixer_filter_probe_feature),
31972 (element_factory_rank_compare_func),
31973 (gst_audio_default_registry_mixer_filter):
31974 * gst-libs/gst/audio/mixerutils.h:
31975 Add gst_audio_default_registry_mixer_filter() utility
31978 2006-01-03 17:33:16 +0000 Michael Smith <msmith@xiph.org>
31980 gst/audioresample/resample.h: As before, but for o_buf
31981 Original commit message from CVS:
31982 * gst/audioresample/resample.h:
31983 As before, but for o_buf
31985 2006-01-03 17:27:13 +0000 Michael Smith <msmith@xiph.org>
31987 gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
31988 Original commit message from CVS:
31989 * gst/audioresample/resample.h:
31990 Declare struct _ResampleState.buffer as unsigned char *, not void *,
31991 since we do arithmetic on it.
31993 2006-01-02 23:37:38 +0000 Tim-Philipp Müller <tim@centricular.net>
31995 gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
31996 Original commit message from CVS:
31997 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
31998 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
31999 (gst_ring_buffer_pause), (wait_segment):
32000 * gst-libs/gst/audio/gstringbuffer.h:
32001 Sun's Forte compiler doesn't seem to like anonymous structs,
32002 so use same setup as in GstBaseSrc (fixes #324900).
32004 2005-12-30 14:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32006 move old example to tests/examples/volume/volune.c
32007 Original commit message from CVS:
32009 * gst/volume/Makefile.am:
32010 * gst/volume/demo.c:
32011 move old example to tests/examples/volume/volune.c
32012 * tests/examples/Makefile.am:
32013 * tests/examples/seek/seek.c: (main):
32014 change window-close event from "delete-event" to "destroy"
32015 * tests/examples/volume/Makefile.am:
32016 * tests/examples/volume/volume.c: (value_changed_callback),
32017 (setup_gui), (message_received), (eos_message_received), (main):
32018 fix event handling and bus usage
32020 2005-12-29 20:37:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32022 gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...
32023 Original commit message from CVS:
32024 * gst/audiotestsrc/gstaudiotestsrc.c:
32025 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
32026 (gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
32027 (gst_audio_test_src_query), (gst_audio_test_src_create_sine),
32028 (gst_audio_test_src_create_square),
32029 (gst_audio_test_src_create_saw),
32030 (gst_audio_test_src_create_triangle),
32031 (gst_audio_test_src_create_silence),
32032 (gst_audio_test_src_create_white_noise),
32033 (gst_audio_test_src_create_pink_noise),
32034 (gst_audio_test_src_init_sine_table),
32035 (gst_audio_test_src_create_sine_table),
32036 (gst_audio_test_src_change_wave),
32037 (gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
32038 (gst_audio_test_src_create), (gst_audio_test_src_set_property):
32039 * gst/audiotestsrc/gstaudiotestsrc.h:
32040 update to basesrc changes, implement segmented seeking and eos handling,
32041 add a 'sine-tab' waveform for performance critical playback
32043 2005-12-29 16:17:55 +0000 Tim-Philipp Müller <tim@centricular.net>
32045 po/POTFILES.in: ... and this time the other modified file that I missed last time.
32046 Original commit message from CVS:
32048 ... and this time the other modified file that I missed last time.
32050 2005-12-29 14:31:49 +0000 Michael Smith <msmith@xiph.org>
32052 gst/playback/gstdecodebin.c: Fix non-C89 variable declaration not at the start of a block. Should help some compilers.
32053 Original commit message from CVS:
32054 * gst/playback/gstdecodebin.c: (new_pad):
32055 Fix non-C89 variable declaration not at the start of a block. Should
32056 help some compilers.
32058 2005-12-29 12:43:22 +0000 Tim-Philipp Müller <tim@centricular.net>
32060 tests/check/Makefile.am: And now fix 'make distcheck' (builddir != srcdir)
32061 Original commit message from CVS:
32062 * tests/check/Makefile.am:
32063 And now fix 'make distcheck' (builddir != srcdir)
32065 2005-12-29 12:22:24 +0000 Tim-Philipp Müller <tim@centricular.net>
32067 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia plugin again (there are still fixes required to pla...
32068 Original commit message from CVS:
32070 * ext/cdparanoia/Makefile.am:
32071 * ext/cdparanoia/gstcdparanoia.c:
32072 * ext/cdparanoia/gstcdparanoia.h:
32073 * ext/cdparanoia/gstcdparanoiasrc.c:
32074 (gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
32075 (gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
32076 (gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
32077 (gst_cd_paranoia_paranoia_callback),
32078 (gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
32079 (gst_cd_paranoia_src_set_property),
32080 (gst_cd_paranoia_src_get_property), (plugin_init):
32081 * ext/cdparanoia/gstcdparanoiasrc.h:
32082 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
32083 plugin again (there are still fixes required to playbin to make
32084 cdda:// uris work there).
32086 2005-12-29 12:13:57 +0000 Tim-Philipp Müller <tim@centricular.net>
32088 tests/check/Makefile.am: Fix test case compilation.
32089 Original commit message from CVS:
32090 * tests/check/Makefile.am:
32091 Fix test case compilation.
32093 2005-12-29 11:49:11 +0000 Tim-Philipp Müller <tim@centricular.net>
32095 gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
32096 Original commit message from CVS:
32097 * gst-libs/gst/cdda/gstcddabasesrc.c:
32098 (gst_cdda_base_src_update_duration),
32099 (gst_cdda_base_src_calculate_cddb_id):
32100 An integer is not a string. Fix access to uninitialised variable.
32101 * tests/check/Makefile.am:
32102 Add cddabasesrc unit test; also actually enable the vorbis test.
32103 * tests/check/generic/states.c:
32104 Blacklist new cd audio elements as well.
32105 * tests/check/libs/cddabasesrc.c:
32106 Unit test for GstCddaBaseSrc (discid calculation mostly).
32108 2005-12-28 18:19:25 +0000 Tim-Philipp Müller <tim@centricular.net>
32110 docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
32111 Original commit message from CVS:
32112 * docs/libs/Makefile.am:
32113 * docs/libs/gst-plugins-base-libs-docs.sgml:
32114 * docs/libs/gst-plugins-base-libs-sections.txt:
32115 * docs/libs/gst-plugins-base-libs.types:
32116 Add docs for libgstcdda/GstCddaBaseSrc.
32117 * gst-libs/gst/interfaces/mixertrack.h:
32118 Do one struct member per line with a semicolon at the end, that way
32119 even gtk-doc might parse it without complaining.
32121 2005-12-28 18:06:50 +0000 Tim-Philipp Müller <tim@centricular.net>
32123 Add new libgstcdda with GstCddaBaseSrc class.
32124 Original commit message from CVS:
32126 * gst-libs/gst/Makefile.am:
32127 * gst-libs/gst/cdda/Makefile.am:
32128 * gst-libs/gst/cdda/base64.c:
32129 * gst-libs/gst/cdda/base64.h:
32130 * gst-libs/gst/cdda/gstcddabasesrc.c:
32131 (gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
32132 (gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
32133 (gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
32134 (gst_cdda_base_src_get_property),
32135 (gst_cdda_base_src_get_track_from_sector),
32136 (gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
32137 (gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
32138 (gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
32139 (gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
32140 (gst_cdda_base_src_uri_get_protocols),
32141 (gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
32142 (gst_cdda_base_src_uri_handler_init),
32143 (gst_cdda_base_src_setup_interfaces),
32144 (gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
32145 (cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
32146 (lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
32147 (gst_cdda_base_src_add_tags),
32148 (gst_cdda_base_src_add_index_associations),
32149 (gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
32150 (gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
32151 (gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
32152 (gst_cdda_base_src_create):
32153 * gst-libs/gst/cdda/gstcddabasesrc.h:
32154 * gst-libs/gst/cdda/sha1.c:
32155 * gst-libs/gst/cdda/sha1.h:
32156 Add new libgstcdda with GstCddaBaseSrc class.
32158 2005-12-28 14:59:41 +0000 Tim-Philipp Müller <tim@centricular.net>
32160 ext/gnomevfs/gstgnomevfssink.h: Use GstBaseSinkClass as parent_class member for class struct, not
32161 Original commit message from CVS:
32162 * ext/gnomevfs/gstgnomevfssink.h:
32163 Use GstBaseSinkClass as parent_class member for class struct, not
32166 2005-12-27 22:29:43 +0000 Tim-Philipp Müller <tim@centricular.net>
32168 gst/videotestsrc/gstvideotestsrc.c: Add start method to reset running time and number of frames sent
32169 Original commit message from CVS:
32170 * gst/videotestsrc/gstvideotestsrc.c:
32171 (gst_video_test_src_class_init), (gst_video_test_src_start):
32172 Add start method to reset running time and number of frames sent
32173 when starting up (fixes #324696; patch by: Michal Benes).
32175 2005-12-27 21:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
32177 docs/plugins/: Add docs stuff for gnomevfssrc and gnomevfssink.
32178 Original commit message from CVS:
32179 * docs/plugins/Makefile.am:
32180 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
32181 * docs/plugins/gst-plugins-base-plugins-sections.txt:
32182 * docs/plugins/gst-plugins-base-plugins.args:
32183 * docs/plugins/gst-plugins-base-plugins.hierarchy:
32184 * docs/plugins/gst-plugins-base-plugins.signals:
32185 Add docs stuff for gnomevfssrc and gnomevfssink.
32186 * ext/gnomevfs/gstgnomevfssrc.c:
32187 Fix example pipeline in gtk-doc blurb.
32189 2005-12-27 21:42:23 +0000 Tim-Philipp Müller <tim@centricular.net>
32191 ext/gnomevfs/: Port gnomevfssink; add gtk-doc blurb.
32192 Original commit message from CVS:
32193 * ext/gnomevfs/Makefile.am:
32194 * ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
32195 (gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
32196 (gst_gnome_vfs_handle_get_type), (plugin_init):
32197 * ext/gnomevfs/gstgnomevfs.h:
32198 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
32199 (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
32200 (gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
32201 (gst_gnome_vfs_sink_set_property),
32202 (gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
32203 (gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
32204 (gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
32205 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
32206 (gst_gnome_vfs_sink_uri_get_type),
32207 (gst_gnome_vfs_sink_uri_get_protocols),
32208 (gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
32209 (gst_gnome_vfs_sink_uri_handler_init):
32210 * ext/gnomevfs/gstgnomevfssink.h:
32211 Port gnomevfssink; add gtk-doc blurb.
32212 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
32213 (gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
32214 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
32215 (gst_gnome_vfs_src_uri_get_type),
32216 (gst_gnome_vfs_src_uri_get_protocols),
32217 (gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
32218 (gst_gnome_vfs_src_uri_handler_init),
32219 (gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
32220 (gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
32221 (gst_gnome_vfs_src_send_additional_headers_callback),
32222 (gst_gnome_vfs_src_received_headers_callback),
32223 (gst_gnome_vfs_src_push_callbacks),
32224 (gst_gnome_vfs_src_pop_callbacks),
32225 (gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
32226 (gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
32227 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
32228 * ext/gnomevfs/gstgnomevfssrc.h:
32229 s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
32230 file; add gtk-doc blurb with example pipelines.
32232 2005-12-23 18:16:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32236 Original commit message from CVS:
32239 === release 0.10.1 ===
32241 2005-12-23 18:08:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32247 * docs/libs/tmpl/gstcolorbalance.sgml:
32248 * docs/plugins/gst-plugins-base-plugins.args:
32249 * docs/plugins/gst-plugins-base-plugins.signals:
32250 * docs/plugins/inspect/plugin-adder.xml:
32251 * docs/plugins/inspect/plugin-alsa.xml:
32252 * docs/plugins/inspect/plugin-audioconvert.xml:
32253 * docs/plugins/inspect/plugin-audiorate.xml:
32254 * docs/plugins/inspect/plugin-audioresample.xml:
32255 * docs/plugins/inspect/plugin-audiotestsrc.xml:
32256 * docs/plugins/inspect/plugin-decodebin.xml:
32257 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
32258 * docs/plugins/inspect/plugin-gnomevfs.xml:
32259 * docs/plugins/inspect/plugin-libvisual.xml:
32260 * docs/plugins/inspect/plugin-ogg.xml:
32261 * docs/plugins/inspect/plugin-pango.xml:
32262 * docs/plugins/inspect/plugin-playbin.xml:
32263 * docs/plugins/inspect/plugin-subparse.xml:
32264 * docs/plugins/inspect/plugin-tcp.xml:
32265 * docs/plugins/inspect/plugin-theora.xml:
32266 * docs/plugins/inspect/plugin-typefindfunctions.xml:
32267 * docs/plugins/inspect/plugin-video4linux.xml:
32268 * docs/plugins/inspect/plugin-videorate.xml:
32269 * docs/plugins/inspect/plugin-videoscale.xml:
32270 * docs/plugins/inspect/plugin-videotestsrc.xml:
32271 * docs/plugins/inspect/plugin-volume.xml:
32272 * docs/plugins/inspect/plugin-vorbis.xml:
32273 * docs/plugins/inspect/plugin-ximagesink.xml:
32274 * docs/plugins/inspect/plugin-xvimagesink.xml:
32276 Original commit message from CVS:
32279 2005-12-21 20:59:52 +0000 Edgard Lima <edgard.lima@indt.org.br>
32282 * gst/typefind/gsttypefindfunctions.c:
32283 iLBC30 and iLBC20 added to typefind.
32284 Original commit message from CVS:
32285 iLBC30 and iLBC20 added to typefind.
32287 2005-12-20 15:57:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32291 * docs/libs/tmpl/gstcolorbalance.sgml:
32307 Original commit message from CVS:
32310 2005-12-20 12:24:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32312 * gst-libs/gst/audio/gstbaseaudiosink.c:
32313 * gst-libs/gst/audio/gstbaseaudiosrc.c:
32314 stop making fun of older compilers
32315 Original commit message from CVS:
32316 stop making fun of older compilers
32318 2005-12-20 12:00:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32320 gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
32321 Original commit message from CVS:
32322 * gst-libs/gst/audio/gstbaseaudiosink.c:
32323 (gst_base_audio_sink_class_init):
32324 * gst-libs/gst/audio/gstbaseaudiosrc.c:
32325 (gst_base_audio_src_class_init):
32326 update strings, values are in microseconds
32327 change the default sink buffer time to something that is smaller
32328 (to help software volume mixing have a slightly lower delay) but
32329 still be acceptable on Wim's laptop
32331 2005-12-20 10:13:05 +0000 Edward Hervey <bilboed@bilboed.com>
32333 gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
32334 Original commit message from CVS:
32335 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
32336 Made a quack, forgot to add DUCK to the riff video template.
32338 2005-12-19 15:00:38 +0000 Edward Hervey <bilboed@bilboed.com>
32340 ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
32341 Original commit message from CVS:
32342 * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
32343 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
32344 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
32345 (gst_ogm_parse_chain):
32346 Make sure pads are initialized correctly.
32347 * gst-libs/gst/riff/riff-ids.h:
32348 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
32349 (gst_riff_create_video_template_caps):
32350 Add a whole bunch of FOURCC <=> MimeType.
32351 Extend the riff video pad template to support the newly added fourcc.
32353 2005-12-18 15:04:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32355 ext/ogg/gstoggdemux.c: Extra debug output when activating/deactivating chains.
32356 Original commit message from CVS:
32357 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
32358 (gst_ogg_demux_activate_chain):
32359 Extra debug output when activating/deactivating chains.
32360 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
32361 (is_demuxer_element), (try_to_link_1), (remove_element_chain),
32363 Remove a queue from our list when it becomes unlinked.
32364 Don't add queues to elements in class 'Demux' if they
32365 can only produce one pad
32367 2005-12-18 10:54:33 +0000 Julien Moutte <julien@moutte.net>
32369 gst-libs/gst/video/gstvideosink.c: Add a debug category.
32370 Original commit message from CVS:
32371 2005-12-18 Julien MOUTTE <julien@moutte.net>
32372 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
32373 (gst_video_sink_get_type): Add a debug category.
32375 2005-12-18 00:56:07 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
32377 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
32378 Original commit message from CVS:
32379 2005-12-17 Philippe Khalaf <burger@speedy.org>
32380 * gst-libs/gst/rtp/gstbasertpdepayload.c:
32381 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
32382 Handle downstream newsegment by sending our own newsegment before the
32383 next buffer to be released. (#323900)
32385 2005-12-18 00:41:10 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
32387 gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
32388 Original commit message from CVS:
32389 2005-12-17 Philippe Khalaf <burger@speedy.org>
32390 * gst-libs/gst/rtp/gstbasertpdepayload.c:
32391 (gst_base_rtp_depayload_set_gst_timestamp):
32392 add queue delay to new segment as well (as opposed to just the first
32393 buffer). (bug #322347)
32395 2005-12-16 22:00:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32397 ext/libvisual/visual.c: change some char* into char[]
32398 Original commit message from CVS:
32399 * ext/libvisual/visual.c: (make_valid_name):
32400 change some char* into char[]
32401 * gst/audiotestsrc/gstaudiotestsrc.c:
32402 (gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
32403 (gst_audio_test_src_create):
32404 * gst/audiotestsrc/gstaudiotestsrc.h:
32405 prepare to handle EOS and SEGMENT_DONE
32407 2005-12-16 12:32:37 +0000 Tim-Philipp Müller <tim@centricular.net>
32409 tests/check/generic/states.c: Blacklist cdparanoia element in state test.
32410 Original commit message from CVS:
32411 * tests/check/generic/states.c: (GST_START_TEST):
32412 Blacklist cdparanoia element in state test.
32414 2005-12-16 11:25:51 +0000 Benjamin Pineau <ben.pineau@gmail.com>
32416 gst/tcp/: Add <string.h> includes for memset and FD_ZERO (fixes #323878;
32417 Original commit message from CVS:
32418 * gst/tcp/gsttcp.c:
32419 * gst/tcp/gsttcpclientsink.c:
32420 * gst/tcp/gsttcpserversink.c:
32421 * gst/tcp/gsttcpserversrc.c:
32422 Add <string.h> includes for memset and FD_ZERO (fixes #323878;
32423 patch by: Benjamin Pineau).
32425 2005-12-15 14:43:38 +0000 Michael Smith <msmith@xiph.org>
32427 gst/videorate/gstvideorate.c: Fix timestamping for videorate when the first buffer it sees has a non-zero timestamp. ...
32428 Original commit message from CVS:
32429 * gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
32430 (gst_video_rate_chain):
32431 Fix timestamping for videorate when the first buffer it sees has a
32432 non-zero timestamp. Fix some misleading debug output.
32434 2005-12-15 10:30:14 +0000 Michael Smith <msmith@xiph.org>
32436 gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
32437 Original commit message from CVS:
32438 * gst/audioresample/gstaudioresample.c:
32439 Don't leak all input buffers to audioresample.
32441 2005-12-15 10:15:10 +0000 Tim-Philipp Müller <tim@centricular.net>
32443 ext/pango/gsttextoverlay.c: Don't operate on empty text buffers. Strip newlines and tabs only from the end of the tex...
32444 Original commit message from CVS:
32445 * ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
32446 Don't operate on empty text buffers. Strip newlines and
32447 tabs only from the end of the text, but leave them intact
32448 in the middle. Fix typo in gtk-doc description.
32450 2005-12-15 09:48:19 +0000 Tim-Philipp Müller <tim@centricular.net>
32452 gst/playback/: Make sure the video frame buffer we return to apps via the "frame" property always has caps set on it....
32453 Original commit message from CVS:
32454 * gst/playback/gstplaybasebin.c:
32455 * gst/playback/gstplaybin.c: (handoff):
32456 Make sure the video frame buffer we return to apps via the
32457 "frame" property always has caps set on it. Modify
32458 _gst_gvalue_set_object() macro to handle NULL objects
32461 2005-12-14 20:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32463 gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
32464 Original commit message from CVS:
32465 * gst/audiotestsrc/gstaudiotestsrc.c:
32466 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
32467 (gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
32468 (gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
32469 (gst_audio_test_src_create):
32470 * gst/audiotestsrc/gstaudiotestsrc.h:
32471 Adjust to some recent api changes and add wtays new cool seeking
32474 2005-12-14 17:58:48 +0000 Tim-Philipp Müller <tim@centricular.net>
32476 ext/alsa/: Helper functions to add device probing via the GstPropertyProbe interface to a class.
32477 Original commit message from CVS:
32478 * ext/alsa/Makefile.am:
32479 * ext/alsa/gstalsadeviceprobe.c:
32480 * ext/alsa/gstalsadeviceprobe.h:
32481 Helper functions to add device probing via the GstPropertyProbe
32482 interface to a class.
32483 * ext/alsa/gstalsamixer.h:
32484 Comment out GST_ALSA_MIXER, it returns a struct that's not
32486 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
32487 Add some debug info.
32488 * ext/alsa/gstalsamixerelement.c:
32489 (gst_alsa_mixer_element_interface_supported),
32490 (gst_implements_interface_init),
32491 (gst_alsa_mixer_element_init_interfaces),
32492 (gst_alsa_mixer_element_class_init),
32493 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
32494 (gst_alsa_mixer_element_set_property),
32495 (gst_alsa_mixer_element_get_property),
32496 (gst_alsa_mixer_element_change_state):
32497 * ext/alsa/gstalsamixerelement.h:
32498 Add 'device' and 'device-name' properties. Add GstPropertyProbe
32499 for device handling (gnome-volume-control will need that).
32501 2005-12-12 20:31:24 +0000 Christian Schaller <uraeus@gnome.org>
32505 * gst-plugins-base.spec.in:
32506 updates to activate cdparanoia plugin
32507 Original commit message from CVS:
32508 updates to activate cdparanoia plugin
32510 2005-12-12 19:13:09 +0000 Michael Smith <msmith@xiph.org>
32512 ext/ogg/gstoggdemux.c: Use the correct function to free list of typefind factories.
32513 Original commit message from CVS:
32514 * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
32515 Use the correct function to free list of typefind factories.
32517 2005-12-12 15:09:55 +0000 Wim Taymans <wim.taymans@gmail.com>
32519 gst/videotestsrc/gstvideotestsrc.*: Implement seeking in videotestsrc.
32520 Original commit message from CVS:
32521 * gst/videotestsrc/gstvideotestsrc.c:
32522 (gst_video_test_src_class_init), (gst_video_test_src_init),
32523 (gst_video_test_src_parse_caps), (gst_video_test_src_query),
32524 (gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
32525 (gst_video_test_src_create):
32526 * gst/videotestsrc/gstvideotestsrc.h:
32527 Implement seeking in videotestsrc.
32530 2005-12-12 15:06:46 +0000 Wim Taymans <wim.taymans@gmail.com>
32532 ext/cdparanoia/: Partially ported cdparanoia now that basesrc can support a plugin like this..
32533 Original commit message from CVS:
32534 * ext/cdparanoia/Makefile.am:
32535 * ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
32536 (gst_paranoia_endian_get_type), (_do_init),
32537 (cdparanoia_class_init), (cdparanoia_init),
32538 (cdparanoia_set_property), (cdparanoia_get_property),
32539 (cdparanoia_do_seek), (cdparanoia_is_seekable),
32540 (cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
32541 (cdparanoia_convert), (cdparanoia_get_query_types),
32542 (cdparanoia_query), (cdparanoia_set_index),
32543 (cdparanoia_uri_set_uri):
32544 * ext/cdparanoia/gstcdparanoia.h:
32545 Partially ported cdparanoia now that basesrc can support a
32548 2005-12-12 13:03:50 +0000 Wim Taymans <wim.taymans@gmail.com>
32550 tests/examples/seek/scrubby.c: Set higher priority for bus events so they don't get reordered with gtk gui events.
32551 Original commit message from CVS:
32552 * tests/examples/seek/scrubby.c: (main):
32553 Set higher priority for bus events so they don't get reordered with
32555 * tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
32556 (flush_toggle_cb), (main):
32557 Added checkbox do disable flushing seeks.
32558 Disable scrubbing when doing non flushing seeks.
32560 2005-12-12 09:52:37 +0000 Tim-Philipp Müller <tim@centricular.net>
32562 gst/subparse/gstsubparse.c: Implement some sort of event handling that doesn't rely on g_return_if_fail; make sure we...
32563 Original commit message from CVS:
32564 * gst/subparse/gstsubparse.c: (gst_sub_parse_init),
32565 (gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
32566 (parser_state_init), (handle_buffer), (gst_sub_parse_chain),
32567 (gst_sub_parse_sink_event), (gst_sub_parse_change_state):
32568 Implement some sort of event handling that doesn't rely on
32569 g_return_if_fail; make sure we always push the last chunk of an
32570 .srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
32571 state change function; remove some old cruft. Seeking is still
32572 rather unlikely to work though.
32573 * tools/.cvsignore:
32576 2005-12-11 22:26:08 +0000 Julien Moutte <julien@moutte.net>
32578 sys/xvimage/xvimagesink.c: Fixed a leak of the current image reference when cleaning up.
32579 Original commit message from CVS:
32580 2005-12-11 Julien MOUTTE <julien@moutte.net>
32581 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
32582 Fixed a leak of the current image reference when cleaning up.
32583 Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
32585 2005-12-09 10:23:42 +0000 Michael Smith <msmith@xiph.org>
32587 tools/: Remove gst-launch-ext. It doesn't work, and is no longer particularly useful.
32588 Original commit message from CVS:
32589 * tools/Makefile.am:
32590 * tools/gst-launch-ext-m.m:
32591 Remove gst-launch-ext. It doesn't work, and is no longer
32592 particularly useful.
32594 2005-12-08 18:53:57 +0000 Luca Ognibene <luogni@tin.it>
32596 ext/ogg/gstogmparse.c: don't pass random values to ogmparse convert function.
32597 Original commit message from CVS:
32598 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
32599 don't pass random values to ogmparse convert function.
32600 Make seeking possible in the exile1.ogm file.
32602 2005-12-07 18:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
32604 gst/playback/: Work around refcount problem with g_value_set_object() that occur if the core has been compiled agains...
32605 Original commit message from CVS:
32606 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
32607 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
32608 Work around refcount problem with g_value_set_object() that occur
32609 if the core has been compiled against GLib-2.6 (g_value_set_object()
32610 will only g_object_ref() the element, but the caller will
32611 gst_object_unref() it and bad things will happen due to the way
32612 GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
32613 totem for people on FC4 using Thomas's 0.10 RPMs.
32615 2005-12-07 11:34:37 +0000 Edward Hervey <bilboed@bilboed.com>
32617 Time to welcome ogm to 0.10 :)
32618 Original commit message from CVS:
32619 Time to welcome ogm to 0.10 :)
32620 * ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
32621 (gst_ogg_pad_typefind):
32622 Oggdemux can now properly typefind elements with dynamic pads.
32623 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
32624 Properly set caps on src pad, and set caps on outgoing buffers.
32626 2005-12-06 19:42:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32629 * ext/alsa/gstalsamixer.h:
32630 * ext/alsa/gstalsamixerelement.h:
32631 * ext/alsa/gstalsamixeroptions.h:
32632 * ext/alsa/gstalsamixertrack.h:
32633 * ext/alsa/gstalsasink.c:
32634 * ext/alsa/gstalsasink.h:
32635 * ext/alsa/gstalsasrc.c:
32636 * ext/alsa/gstalsasrc.h:
32637 * ext/cdparanoia/gstcdparanoia.h:
32638 * ext/gnomevfs/gstgnomevfsuri.h:
32639 * ext/ogg/gstoggdemux.c:
32640 * ext/ogg/gstoggmux.c:
32641 * ext/pango/gsttextoverlay.h:
32642 * ext/theora/theoradec.c:
32643 * ext/theora/theoraenc.c:
32644 * ext/vorbis/vorbisdec.h:
32645 * ext/vorbis/vorbisenc.c:
32646 * ext/vorbis/vorbisenc.h:
32647 * ext/vorbis/vorbisparse.h:
32648 * gst-libs/gst/audio/gstaudioclock.h:
32649 * gst-libs/gst/audio/gstaudiosink.c:
32650 * gst-libs/gst/audio/gstaudiosink.h:
32651 * gst-libs/gst/audio/gstaudiosrc.c:
32652 * gst-libs/gst/audio/gstaudiosrc.h:
32653 * gst-libs/gst/audio/gstbaseaudiosink.c:
32654 * gst-libs/gst/audio/gstbaseaudiosink.h:
32655 * gst-libs/gst/audio/gstbaseaudiosrc.c:
32656 * gst-libs/gst/audio/gstbaseaudiosrc.h:
32657 * gst-libs/gst/audio/gstringbuffer.h:
32658 * gst-libs/gst/audio/multichannel.h:
32659 * gst-libs/gst/floatcast/floatcast.h:
32660 * gst-libs/gst/interfaces/colorbalance.c:
32661 * gst-libs/gst/interfaces/colorbalance.h:
32662 * gst-libs/gst/interfaces/colorbalancechannel.h:
32663 * gst-libs/gst/interfaces/mixer.h:
32664 * gst-libs/gst/interfaces/mixeroptions.h:
32665 * gst-libs/gst/interfaces/mixertrack.h:
32666 * gst-libs/gst/interfaces/navigation.h:
32667 * gst-libs/gst/interfaces/propertyprobe.h:
32668 * gst-libs/gst/interfaces/tuner.h:
32669 * gst-libs/gst/interfaces/tunerchannel.h:
32670 * gst-libs/gst/interfaces/tunernorm.h:
32671 * gst-libs/gst/interfaces/xoverlay.h:
32672 * gst-libs/gst/netbuffer/gstnetbuffer.h:
32673 * gst-libs/gst/riff/riff-ids.h:
32674 * gst-libs/gst/riff/riff-media.h:
32675 * gst-libs/gst/riff/riff-read.h:
32676 * gst-libs/gst/rtp/gstbasertpdepayload.h:
32677 * gst-libs/gst/rtp/gstbasertppayload.c:
32678 * gst-libs/gst/rtp/gstbasertppayload.h:
32679 * gst-libs/gst/rtp/gstrtpbuffer.c:
32680 * gst-libs/gst/rtp/gstrtpbuffer.h:
32681 * gst-libs/gst/tag/gsttageditingprivate.h:
32682 * gst-libs/gst/tag/gstvorbistag.c:
32683 * gst-libs/gst/tag/tag.h:
32684 * gst-libs/gst/video/video.h:
32685 * gst/adder/gstadder.c:
32686 * gst/adder/gstadder.h:
32687 * gst/audioconvert/audioconvert.c:
32688 * gst/audioconvert/audioconvert.h:
32689 * gst/audioconvert/gstaudioconvert.c:
32690 * gst/audioconvert/gstchannelmix.c:
32691 * gst/audioconvert/gstchannelmix.h:
32692 * gst/audiorate/gstaudiorate.c:
32693 * gst/audioresample/buffer.h:
32694 * gst/audioresample/functable.h:
32695 * gst/audioresample/gstaudioresample.c:
32696 * gst/audioresample/resample.h:
32697 * gst/ffmpegcolorspace/avcodec.h:
32698 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
32699 * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
32700 * gst/ffmpegcolorspace/imgconvert.c:
32701 * gst/ffmpegcolorspace/imgconvert_template.h:
32702 * gst/playback/gstdecodebin.c:
32703 * gst/playback/gstplaybasebin.h:
32704 * gst/playback/gstplaybin.c:
32705 * gst/playback/gststreaminfo.h:
32706 * gst/tcp/gstfdset.c:
32707 * gst/tcp/gstfdset.h:
32708 * gst/tcp/gstmultifdsink.c:
32709 * gst/tcp/gstmultifdsink.h:
32710 * gst/tcp/gsttcp.h:
32711 * gst/tcp/gsttcpclientsrc.c:
32712 * gst/tcp/gsttcpclientsrc.h:
32713 * gst/tcp/gsttcpplugin.h:
32714 * gst/tcp/gsttcpserversink.c:
32715 * gst/tcp/gsttcpserversrc.c:
32716 * gst/typefind/gsttypefindfunctions.c:
32717 * gst/videorate/gstvideorate.c:
32718 * gst/videotestsrc/gstvideotestsrc.h:
32719 * gst/videotestsrc/videotestsrc.h:
32720 * sys/v4l/gstv4lcolorbalance.h:
32721 * sys/v4l/gstv4ltuner.h:
32722 * sys/v4l/gstv4lxoverlay.h:
32723 * sys/v4l/v4l_calls.h:
32724 * sys/v4l/videodev_mjpeg.h:
32725 * tests/check/elements/audioconvert.c:
32726 * tests/check/elements/audioresample.c:
32727 * tests/check/elements/audiotestsrc.c:
32728 * tests/check/elements/videotestsrc.c:
32729 * tests/check/elements/volume.c:
32730 * tests/examples/seek/scrubby.c:
32731 * tests/examples/seek/seek.c:
32733 Original commit message from CVS:
32736 2005-12-06 19:28:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32738 * docs/libs/tmpl/gstaudio.sgml:
32739 * docs/libs/tmpl/gstcolorbalance.sgml:
32740 * docs/libs/tmpl/gstgconf.sgml:
32741 * docs/libs/tmpl/gstmixer.sgml:
32742 * docs/libs/tmpl/gstringbuffer.sgml:
32743 * docs/libs/tmpl/gsttuner.sgml:
32744 * docs/libs/tmpl/gstxoverlay.sgml:
32745 put back stability level
32746 Original commit message from CVS:
32747 put back stability level
32749 2005-12-05 18:11:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32753 Original commit message from CVS:
32756 === release 0.10.0 ===
32758 2005-12-05 18:02:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32764 * docs/libs/tmpl/gstcolorbalance.sgml:
32765 * docs/plugins/inspect/plugin-adder.xml:
32766 * docs/plugins/inspect/plugin-alsa.xml:
32767 * docs/plugins/inspect/plugin-audioconvert.xml:
32768 * docs/plugins/inspect/plugin-audiorate.xml:
32769 * docs/plugins/inspect/plugin-audioresample.xml:
32770 * docs/plugins/inspect/plugin-audiotestsrc.xml:
32771 * docs/plugins/inspect/plugin-decodebin.xml:
32772 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
32773 * docs/plugins/inspect/plugin-gnomevfs.xml:
32774 * docs/plugins/inspect/plugin-libvisual.xml:
32775 * docs/plugins/inspect/plugin-ogg.xml:
32776 * docs/plugins/inspect/plugin-pango.xml:
32777 * docs/plugins/inspect/plugin-playbin.xml:
32778 * docs/plugins/inspect/plugin-subparse.xml:
32779 * docs/plugins/inspect/plugin-tcp.xml:
32780 * docs/plugins/inspect/plugin-theora.xml:
32781 * docs/plugins/inspect/plugin-typefindfunctions.xml:
32782 * docs/plugins/inspect/plugin-video4linux.xml:
32783 * docs/plugins/inspect/plugin-videorate.xml:
32784 * docs/plugins/inspect/plugin-videoscale.xml:
32785 * docs/plugins/inspect/plugin-videotestsrc.xml:
32786 * docs/plugins/inspect/plugin-volume.xml:
32787 * docs/plugins/inspect/plugin-vorbis.xml:
32788 * docs/plugins/inspect/plugin-ximagesink.xml:
32789 * docs/plugins/inspect/plugin-xvimagesink.xml:
32791 Original commit message from CVS: