1 2009-07-30 14:40:50 +0100 Jan Schmidt <thaytan@noraisin.net>
8 2009-07-29 14:15:53 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
10 * gst-libs/gst/audio/gstaudiofilter.c:
11 audiofilter: Don't assert on slightly different caps
12 Plugins should not assert on incompatible caps, caps negotiation will
15 2009-07-30 13:42:21 +0300 Stefan Kost <ensonic@users.sf.net>
17 * gst/adder/gstadder.c:
18 adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146.
20 2009-07-30 09:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
23 configure: bump Gtk+ requirement of GUI examples from 2.12 to 2.14
24 The gio mount example needs GtkMountOperation, which is new in 2.14.
26 2009-07-27 10:29:27 +0100 Balachandran C <balachandran_c@rediffmail.com>
28 * ext/alsa/gstalsasrc.c:
29 alsasrc: set alsasrc->handle back to NULL when closing device
30 Fixes crashes in gst_alsa_find_device_name() when probing or
31 reading the device-name property (e.g. when doing a dot-file
34 2009-07-24 19:26:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
36 * gst/playback/gststreamselector.c:
37 playbin: rename GType of stream selector pad to avoid clash with input-selector from -bad
38 Rename the GType of the pads of playbin's internal stream selector
39 element so they don't use the same type name as input-selector's
42 2009-07-24 13:39:55 +0100 Jan Schmidt <thaytan@noraisin.net>
77 2009-07-24 13:46:15 +0100 Jan Schmidt <thaytan@noraisin.net>
79 * tests/examples/v4l/.gitignore:
80 ignores: Ignore v4l probing example binary
82 2009-07-24 09:35:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
84 * gst/typefind/gsttypefindfunctions.c:
85 typefind: recognise Kate spu subtitles as well
86 Recognise spu-subtitles, SUB and K-SPU as valid categories for
87 Kate subtitles as well.
89 2009-07-24 00:42:16 +0300 Stefan Kost <ensonic@users.sf.net>
92 Automatic update of common submodule
93 From fedaaee to 94f95e3
95 2009-07-22 14:21:43 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
97 * gst-plugins-base.spec.in:
98 Update spec file with latest changes
100 2009-07-20 17:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
133 * win32/common/_stdint.h:
134 * win32/common/audio-enumtypes.c:
135 * win32/common/config.h:
136 * win32/common/gstrtsp-enumtypes.c:
137 * win32/common/interfaces-enumtypes.c:
138 * win32/common/video-enumtypes.c:
139 0.10.23.3 pre-release
141 2009-07-20 12:51:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
143 * gst/audiotestsrc/gstaudiotestsrc.c:
144 audiotestsrc: call send_event directly
145 We can't call gst_element_send_event() from a streaming thread as it gets the
146 state lock. Instead call the send_event method directly until we have a nice API
150 2009-07-03 04:42:24 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
152 * gst-libs/gst/audio/gstaudiosink.c:
153 audiosink: Add stream-status messages
156 2009-07-03 04:41:05 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
158 * gst-libs/gst/audio/gstaudiosrc.c:
159 audiosrc: Add stream-status messages
162 2009-07-20 10:53:11 +0200 Edward Hervey <bilboed@bilboed.com>
164 * gst/adder/gstadder.c:
165 gstadder: Don't forget to free pending events on flush/dispose.
168 2009-07-12 10:08:12 +0200 Edward Hervey <bilboed@bilboed.com>
170 * tests/check/elements/adder.c:
171 tests/adder: Add stream consistency checking. Fixes #588748
173 2009-07-12 10:07:34 +0200 Edward Hervey <bilboed@bilboed.com>
175 * gst/audiotestsrc/gstaudiotestsrc.c:
176 audiotestsrc: Make sure tags are properly serialized. Fixes #588746
177 We do this by letting the basesrc base class handle the tags.
179 2009-07-13 09:28:54 +0200 Edward Hervey <bilboed@bilboed.com>
181 * gst/adder/gstadder.c:
182 * gst/adder/gstadder.h:
183 adder: Collect incoming tag events and send them after newsegment. Fixes #588747
185 2009-07-16 09:32:46 +0200 Edward Hervey <bilboed@bilboed.com>
187 * ext/vorbis/vorbisdec.c:
188 vorbisdec: Check for empty tag strings. Fixes #588724
190 2009-07-14 17:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
192 * gst/playback/gstqueue2.c:
193 queue2: fix leak and improve buffering
194 Keep track of the max requested position and compare this to the write position
195 in the temp file to get the current amount of buffered data.
196 Fix memleak of all incomming buffers.
199 2009-07-15 17:40:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
201 * gst/playback/Makefile.am:
202 * gst/playback/gstinputselector.c:
203 * gst/playback/gstinputselector.h:
204 * gst/playback/gstplay-marshal.list:
205 * gst/playback/gstplaybin2.c:
206 playbin2: use private copy of input-selector
207 We shouldn't really depend on elements from -bad for stream
208 selection in playbin2, so use a private copy of input-selector
209 until the selector plugin is ready to be moved to -base or -good.
212 2009-07-15 17:26:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
214 * gst/playback/gstinputselector.c:
215 * gst/playback/gstinputselector.h:
216 playback: add private copy of the input-selector from gst-plugins-bad
217 Not hooked up yet though. See #586356.
219 2009-07-14 19:00:36 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
221 * tests/examples/v4l/Makefile.am:
222 examples: fix v4l probe example build
225 2009-07-14 19:00:10 +0100 Jan Schmidt <thaytan@noraisin.net>
259 0.10.23.2 pre-release
261 2009-07-14 16:24:10 +0100 Jan Schmidt <thaytan@noraisin.net>
265 Add Turkish translations
267 2009-07-14 15:31:13 +0100 Jan Schmidt <thaytan@noraisin.net>
269 * tests/check/elements/adder.c:
270 adder: One more attempt to fix the adder test
271 Give up and discard and recreate the alsasrc after checking it can
272 be opened, due to some strange crash inside alsa when we don't.
274 2009-07-14 15:06:41 +0100 Jan Schmidt <thaytan@noraisin.net>
276 * tests/check/elements/adder.c:
277 adder: Perform get_state() in the unit test
278 Wait for the alsasrc to return to NULL after setting it to PAUSED for
279 testing, otherwise it leads to segfaults later on.
281 2009-07-14 14:39:32 +0100 Jan Schmidt <thaytan@noraisin.net>
283 * tests/check/elements/adder.c:
284 adder: Don't fail when alsasrc is unavailable
285 Make the liveadder test succeed silently when it can't be completed
286 either because alsasrc is unavailable, or because the device is
289 2009-07-13 22:51:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
291 * gst-libs/gst/pbutils/descriptions.c:
292 * gst/typefind/gsttypefindfunctions.c:
293 typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest
294 Differentiate subtitle streams and lyrics/cracktastic/complex streams via
295 the category string in the headers. This seems like a useful distinction
296 to make, and also seems more future-proof. See #525743.
298 2009-02-21 13:18:10 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
300 * ext/ogg/gstoggmux.c:
301 oggmux: add Kate caps to the list of accepted types
304 2009-07-13 21:56:46 +0300 Stefan Kost <ensonic@users.sf.net>
306 * gst/playback/gsturidecodebin.c:
307 uridecodebin: treat uri-schemas incasesensitive
308 Treat uri-schemas incasesensitive. This is mandated in rfc2396 section 3.1.
309 Fixes not showing buffering messages e.g. for HTTP://...
311 2009-07-13 21:54:47 +0300 Stefan Kost <ensonic@users.sf.net>
313 * gst-libs/gst/interfaces/navigation.c:
314 navigation: simplify docs
315 Make short-desc short - its used in the toc. Strip uneeded markup.
317 2009-07-13 18:31:15 +0100 Jan Schmidt <thaytan@noraisin.net>
319 * win32/common/libgstnetbuffer.def:
320 * win32/common/libgstvideo.def:
322 Remove methods from video base classes that have moved to -bad.
323 Add gst_netaddress_to_string
325 2009-07-13 17:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
327 * tests/examples/gio/.gitignore:
328 ignores: ignore the giosrc-mounting example binary
330 2009-07-13 17:54:40 +0100 Jan Schmidt <thaytan@noraisin.net>
332 * gst-libs/gst/interfaces/navigation.c:
333 navigation: Add some partial documentation
334 Add a general documentation blurb for the GstNavigation functionality.
335 Still lacks some example code and detail on how to implement it.
337 2009-07-13 17:52:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
339 * gst-libs/gst/pbutils/descriptions.c:
340 pbutils: add description for Siren codec and make two descriptions non-translatable
342 2009-07-13 12:23:20 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
345 Automatic update of common submodule
346 From 5845b63 to fedaaee
348 2009-07-13 18:21:49 +0200 Elliott Sales de Andrade <quantum.analyst at gmail.com>
350 * gst-libs/gst/riff/riff-ids.h:
351 * gst-libs/gst/riff/riff-media.c:
352 riff: add siren to the RIFF parser
353 Add siren7 caps to the RIFF parser.
355 2009-07-13 14:55:59 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
358 * tests/examples/Makefile.am:
359 * tests/examples/v4l/Makefile.am:
360 * tests/examples/v4l/probe.c:
361 v4lsrc: add a simple test case for device probing
363 2009-07-03 11:38:01 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
366 * sys/v4l/Makefile.am:
367 * sys/v4l/gstv4lelement.c:
368 v4lsrc: optional support for device probing with gudev
369 Enumerate v4l devices using gudev if available.
372 2009-07-10 23:24:36 +0100 Stefan Kost <ensonic@users.sf.net>
374 * gst/adder/gstadder.c:
375 adder: add since tags to docs
377 2009-07-10 21:29:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
379 * tests/examples/seek/seek.c:
380 seek: don't automatically start pipeline in DB
381 Keep the pipeline paused when we detect download buffering. The user has to
382 manually start the pipeline for now because we can't estimate when the buffering
383 will finish or when we have underrun.
385 2009-07-10 21:01:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
387 * gst/playback/gstqueue2.c:
388 queue2: flush differently, avoiding deadlocks
389 Don't flush the file by closing and opening it but instead use g_freopen. This
390 avoids a deadlock in shutdown because we emit the temp-location property change
391 with the wrong lock held.
393 2009-07-10 20:25:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
395 * tests/examples/seek/seek.c:
396 seek: add a checkbox for progressive download
398 2009-07-10 20:24:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
400 * gst/playback/gsturidecodebin.c:
401 uridecodebin: Fix template construction
402 Fix the construction of the temporary filename construction as the application
403 name can be NULL and we don't want a separator between the prgname and the
406 2009-07-10 20:04:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
408 * gst/playback/gstplay-enum.c:
409 * gst/playback/gstplay-enum.h:
410 * gst/playback/gstplaybin2.c:
411 playbin2: add support for progressive download
412 Add a new playbin2 flag (initially disabled) to enable progressive download
413 buffering in uridecodebin.
415 2009-07-10 19:59:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
417 * gst/playback/gsturidecodebin.c:
418 uridecodebin: add download property
419 Add a download property that will attempt to configure queue2 into progressive
421 Make sure we only enable download buffering for quicktime and flv formats.
423 2009-07-10 19:49:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
425 * gst/playback/gstqueue2.c:
426 queue2: add temp-template property
427 Add a new temp-template property so that queue2 can securely allocate a
428 temporary filename. Deprecate the temp-location property for setting the
429 location but still use it to notify the allocated temp file.
431 2009-07-10 20:06:28 +0100 Stefan Kost <ensonic@users.sf.net>
433 * gst/adder/gstadder.c:
434 * gst/adder/gstadder.h:
435 adder: add a caps-property to avoid to need to plug a capsfilter afterwards
436 Adder can only handle one common format accross the pads. Thus one needed to add
437 a capsfilter afterwards and manage the caps. Now one can simply set the caps on
440 2009-07-10 18:59:05 +0100 Stefan Kost <ensonic@users.sf.net>
442 * tests/check/elements/adder.c:
443 adder: skip live-seek text if we have no audiosrc, add new test
444 The seek-test needs a real audiosrc. Also add a test that checks that adder is
445 reusable. Finaly handle warnings as warnings to fix a assertion.
447 2009-07-10 19:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
449 * ext/gio/gstgiosink.c:
450 gio: Also post a "not-mounted" message from giosink
452 2009-07-10 17:15:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
454 * tests/examples/gio/giosrc-mounting.c:
455 gio: Remove workaround for playbin2 bug in the sample application
456 The playbin2 bug was #588078.
458 2009-07-10 17:08:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
460 * gst/playback/gstplaybin2.c:
461 playbin2: Make it possible for READY->PAUSED to succeed after it failed the first time
462 If READY->PAUSED failed in the source element we would've swapped
463 the current and next group already. To allow READY->PAUSED to succeed
464 after the first failure we have to swap the current and next group
465 back again. This also ensure that we're again in the same state
466 as before the failed state change and not at the next group.
467 This was especially a problem for playbin2 pipelines that use the
468 new mounting support in giosrc as the source would fail for READY->PAUSED
469 the first time, the application mounts the location and then tries
470 to go READY->PAUSED again (and this time it would succeed).
473 2009-07-10 11:42:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
476 * tests/examples/Makefile.am:
477 * tests/examples/gio/Makefile.am:
478 * tests/examples/gio/giosrc-mounting.c:
479 gio: Add example application that shows how to handle the "not-mounted" message
481 2009-07-10 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
484 gio: Remove the experimental status from the GIO plugin
487 2009-07-10 11:24:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
489 * ext/gio/gstgiosink.c:
490 * ext/gio/gstgiosrc.c:
491 gio: Add documentation for the new "not-mounted" and "file-exists" messages
493 2009-07-09 13:45:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
495 * ext/gio/gstgiobasesrc.c:
496 gio: Make sure that we have the correct stream position when starting
498 2009-07-08 17:24:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
500 * ext/gio/gstgiobasesink.c:
501 gio: Make sure to flush the output stream if it shouldn't be closed
502 Otherwise there might still be unwritten data after the element
505 2009-07-08 17:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
507 * ext/gio/gstgiobasesink.c:
508 * ext/gio/gstgiobasesink.h:
509 * ext/gio/gstgiobasesrc.c:
510 * ext/gio/gstgiobasesrc.h:
511 * ext/gio/gstgiosink.c:
512 * ext/gio/gstgiosrc.c:
513 gio: Don't close the GIO streams for the giostream{src,sink} elements
514 This makes it possible to do something useful with the streams
515 after the element has stopped. Fixes bug #587896.
517 2009-07-08 17:19:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
519 * tests/check/pipelines/gio.c:
520 gio: Try to reuse the pipeline with the same stream objects
522 2009-07-08 17:02:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
524 * ext/gio/gstgiobasesink.c:
525 * ext/gio/gstgiobasesrc.c:
526 gio: Improve the error message if a stream is already closed before usage
528 2009-07-08 16:55:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
530 * ext/gio/gstgiosink.c:
531 gio: Post a custom file-exists message on the bus if the file already exists
532 An application can handle this message, remove the file in question
533 and restart the pipeline again without showing an error.
534 This fixes bug #529300.
536 2009-07-08 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
538 * ext/gio/gstgiosrc.c:
539 gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted
541 2009-07-08 16:50:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
543 * ext/gio/gstgiosink.c:
544 gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink
546 2009-07-08 15:52:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
548 * ext/gio/gstgiosrc.c:
549 gio: Post a custom "not-mounted" message on the bus
550 This allows applications to mount the GFile if possible and restart
551 the pipeline instead of simply giving an error.
553 2009-07-08 15:08:32 +0200 Philip Jägenstedt <philipj@opera.com>
555 * gst/audioconvert/gstchannelmix.c:
556 audioconvert: Fix compilation when debugging is disabled
559 2009-07-07 20:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
561 * ext/gio/gstgiobasesink.c:
562 * ext/gio/gstgiobasesink.h:
563 * ext/gio/gstgiobasesrc.h:
564 * ext/gio/gstgiosink.c:
565 * ext/gio/gstgiosink.h:
566 * ext/gio/gstgiostreamsink.c:
567 * ext/gio/gstgiostreamsink.h:
568 gio: Add vfunc for requesting the stream for the sinks too
570 2009-07-07 20:21:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
572 * ext/gio/gstgiobasesink.c:
573 * ext/gio/gstgiobasesink.h:
574 * ext/gio/gstgiobasesrc.c:
575 * ext/gio/gstgiosink.c:
576 * ext/gio/gstgiosrc.c:
577 * ext/gio/gstgiostreamsink.c:
578 * ext/gio/gstgiostreamsrc.c:
579 gio: Some more random cleanup
581 2009-07-07 20:20:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
584 * ext/gio/gstgiobasesink.c:
585 * ext/gio/gstgiobasesrc.c:
586 * ext/gio/gstgiobasesrc.h:
587 * ext/gio/gstgiosink.c:
588 * ext/gio/gstgiosrc.c:
589 * ext/gio/gstgiosrc.h:
590 * ext/gio/gstgiostreamsink.c:
591 * ext/gio/gstgiostreamsrc.c:
592 * ext/gio/gstgiostreamsrc.h:
593 gio: Update my mail address and copyright
595 2009-07-07 20:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
597 * ext/gio/gstgiobasesrc.c:
598 * ext/gio/gstgiobasesrc.h:
599 * ext/gio/gstgiosrc.c:
600 * ext/gio/gstgiostreamsrc.c:
601 * ext/gio/gstgiostreamsrc.h:
602 gio: General clean up and simplification
603 The GInputStreams are now requested by a vfunc from
604 the subclasses instead of relying that the subclass
605 sets it until it's needed.
606 This might also fix bug #587896.
608 2009-07-06 22:31:12 +0100 Stefan Kost <ensonic@users.sf.net>
610 * gst/adder/gstadder.c:
611 adder: keep sending newsegments after seeking
612 Adder sends with timestamps from 0 upwards. After seeking we need to send
613 new-segments to get correct positions-queries.
615 2009-07-06 20:44:00 +0100 Stefan Kost <ensonic@users.sf.net>
617 * tests/check/elements/adder.c:
618 adder: make test more robust
619 Add audioconverts to the live-seeking test to make it negotiate.
621 2009-06-30 17:19:50 +0300 Stefan Kost <ensonic@users.sf.net>
623 * sys/xvimage/xvimagesink.c:
624 xvimagesink: use core performance log category
626 2009-07-05 21:29:40 +0200 Edward Hervey <bilboed@bilboed.com>
628 * gst/adder/gstadder.c:
629 adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped.
630 This ensures that collectpads' cookie is properly updated so that when the streaming
631 threads will restart and be checking for the flushing status of all pads there will
632 be no inconsistent state.
634 2009-07-05 18:01:38 +0200 Hans-Peter Nilsson <hp@gcc.gnu.org>
636 * ext/pango/gstclockoverlay.c:
637 pango: Call tzset() before localtime_r()
638 POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't
639 required to set the state variables that define the current timezone. Indeed,
640 glibc (at least 2.9) doesn't do this for subsequent calls. The effect is that
641 if the system timezone is changed for a running program between two calls to
642 gst_clock_overlay_render_time, it won't be noticed. For glibc, changing the
643 timezone equals /etc/localtime being modified.
646 2009-07-01 17:33:14 -0700 David Schleef <ds@schleef.org>
649 build: remove spurious schroedinger reference
651 2009-07-01 10:25:43 -0700 David Schleef <ds@schleef.org>
655 * ext/schroedinger/Makefile.am:
656 * ext/schroedinger/gstschro.c:
657 * ext/schroedinger/gstschrodec.c:
658 * ext/schroedinger/gstschroenc.c:
659 * ext/schroedinger/gstschroparse.c:
660 * ext/schroedinger/gstschroutils.c:
661 * ext/schroedinger/gstschroutils.h:
662 * gst-libs/gst/video/Makefile.am:
663 * gst-libs/gst/video/gstbasevideocodec.c:
664 * gst-libs/gst/video/gstbasevideocodec.h:
665 * gst-libs/gst/video/gstbasevideodecoder.c:
666 * gst-libs/gst/video/gstbasevideodecoder.h:
667 * gst-libs/gst/video/gstbasevideoencoder.c:
668 * gst-libs/gst/video/gstbasevideoencoder.h:
669 * gst-libs/gst/video/gstbasevideoparse.c:
670 * gst-libs/gst/video/gstbasevideoparse.h:
671 * gst-libs/gst/video/gstbasevideoutils.c:
672 * gst-libs/gst/video/gstbasevideoutils.h:
673 basevideo: send basevideo back to remedial school
674 Move basevideo classes and schroedinger plugin to -bad.
676 2009-07-01 12:54:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
678 * docs/libs/gst-plugins-base-libs-sections.txt:
679 * gst-libs/gst/netbuffer/gstnetbuffer.h:
680 netaddress: add constant for max len
682 2009-07-01 12:48:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
684 * docs/libs/gst-plugins-base-libs-sections.txt:
685 * gst-libs/gst/netbuffer/gstnetbuffer.c:
686 * gst-libs/gst/netbuffer/gstnetbuffer.h:
687 netbuffer: add gst_netaddress_to_string
688 Add function to serialize a net address to a string.
689 API: GstNetAddress::gst_netaddress_to_string()
691 2009-06-30 18:44:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
693 * gst/playback/gsturidecodebin.c:
694 uridecodebin: make fd:// uri use buffering too
695 fd:// usually operate in push mode only and are thus suitable for buffering.
697 2009-06-30 14:46:38 +0300 Stefan Kost <ensonic@users.sf.net>
699 * gst/playback/gstplaybin2.c:
700 * gst/volume/gstvolume.c:
701 volume: include "1.0=100%" in property description
703 2009-06-30 14:45:51 +0300 Stefan Kost <ensonic@users.sf.net>
705 * gst/playback/gstplaysink.c:
706 playsink: remove unused property defs
708 2009-06-29 17:11:50 +0300 Stefan Kost <ensonic@users.sf.net>
710 * gst-libs/gst/audio/multichannel.c:
711 multichannel: rewrite the new doc comment a bit
712 Its part of the audio lib.
714 2009-06-29 14:34:02 +0100 Jan Schmidt <thaytan@noraisin.net>
716 * gst/playback/gstplaysink.c:
717 playsink: Avoid a segfault when the video sink fails to start
718 Don't attempt to display the subpictures and segfault when the
719 video sink failed to start (and hence the videochain is NULL).
721 2009-06-29 15:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
723 * gst-libs/gst/audio/gstringbuffer.c:
724 * gst-libs/gst/audio/gstringbuffer.h:
725 ringbuffer: add vmethod to clear the ringbuffer
726 Add a vmethod so that subclasses can be notified when they should clear the data
729 2009-06-29 14:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
731 * gst-libs/gst/riff/riff-media.c:
732 riff-media: Fix the fourcc caps property for VC-1/WMVA
733 The caps property for carrying fourccs is 'format', not 'fourcc'
735 2009-06-29 12:20:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
737 * gst-libs/gst/rtsp/gstrtspconnection.c:
738 rtsp: include in.h for FreeBSD compat
741 2009-06-29 12:20:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
743 * win32/common/libgstapp.def:
744 defs: add defs for new appsink buffer-list method
746 2009-06-29 12:14:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
748 * gst-libs/gst/app/gstappsink.c:
749 * gst-libs/gst/app/gstappsink.h:
750 appsink: add docs and signals
751 Add docs for the new callback.
752 Add signals for the new buffer-list support.
754 2009-06-29 10:24:36 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
756 * tests/check/elements/appsink.c:
757 Added unit tests for buffer list support in appsink.
759 2009-06-17 11:12:08 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
761 * gst-libs/gst/app/gstappsink.c:
762 Added buffer list support.
764 2009-06-17 09:23:11 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
766 * gst-libs/gst/app/gstappsink.h:
767 Added buffer list support.
769 2009-06-29 09:36:27 +0200 Peter Kjellerstedt <pkj@axis.com>
771 * gst-libs/gst/sdp/gstsdpmessage.c:
772 sdp: Include winsock2.h after defining WINVER.
773 Similar to bug #587080.
775 2009-06-29 09:31:40 +0200 Peter Kjellerstedt <pkj@axis.com>
777 * gst-libs/gst/rtsp/gstrtspconnection.c:
778 rtsp: Moved a comment.
780 2009-06-27 23:23:02 +0300 Stefan Kost <ensonic@users.sf.net>
782 * gst-libs/gst/audio/audio.c:
783 * gst-libs/gst/audio/multichannel.c:
784 docs: add basic section docs for multichannel and relocate the ones for audio
785 Add section docs for multichannel, so that it has a short desc in the toc too.
786 Move the section docs in adio up, so that the follow the copyright like
789 2009-06-26 21:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
791 * sys/v4l/gstv4lelement.c:
792 * sys/v4l/gstv4lsrc.c:
793 v4l: open/close device in ready.
794 Simillar change like in v4l2src. This allows probing feature in paused, where
795 streaming is noit yet started.
797 2009-06-10 17:05:22 +0300 René Stadler <rene.stadler@nokia.com>
799 * gst/playback/gstplaysink.c:
800 playbin2: fix initial volume handling also when reusing the element
801 This is a follow-up to commit 452988, making it work correctly when the audio
804 2009-06-26 21:48:58 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
806 * gst-libs/gst/rtsp/gstrtspconnection.c:
807 Define WINVER before including any win headers
810 2009-06-27 00:50:54 +0300 René Stadler <mail@renestadler.de>
812 * gst-libs/gst/riff/riff-read.c:
813 riff: prevent crash if rounded up tag size exceeds data size
814 When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
815 and an invalid read past the buffer data follows.
817 2009-06-26 15:17:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
819 * gst-libs/gst/video/gstbasevideocodec.c:
820 basevideocodec: By default don't allow caps changes on the srcpad
821 This fixed playback of Dirac files with schrodec when upstream wants
822 a different width/height, basevideocodec accepts this and then
823 pushes buffers with new caps but content of the old caps.
824 In the best case this will just result in wrong unit size and a
825 failure in basestransform elements.
827 2009-06-26 14:11:21 +0100 Jan Schmidt <thaytan@noraisin.net>
830 autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
831 Check for more automake command variants. Use printf instead of 'echo -n'
834 2009-06-26 13:41:38 +0100 Jan Schmidt <thaytan@noraisin.net>
837 Automatic update of common submodule
838 From f810030 to 5845b63
840 2009-06-26 13:14:02 +0300 Stefan Kost <ensonic@users.sf.net>
842 * gst/playback/gstscreenshot.c:
843 screenshot: don't leak message
845 2009-06-25 12:04:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
847 * gst/typefind/gsttypefindfunctions.c:
848 typefinding: lower the h264 typefinder's probability
849 A NEARLY_CERTAIN is absolutely not warranted given the kind
850 of things it checks for. Even a LIKELY is probably not entirely
853 2009-06-24 15:13:56 +0100 Jan Schmidt <jan.schmidt@sun.com>
856 Automatic update of common submodule
857 From f3bb51b to f810030
859 2009-06-24 09:48:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
861 * gst-libs/gst/pbutils/descriptions.c:
862 pbutils: add description for multipart
863 So we get slightly nicer error messages when multipartdemux is missing.
865 2009-06-23 18:07:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
867 * gst/adder/gstadder.c:
868 adder: only unflush when we flushed before
869 Ass suggested by Stefan Kost:
870 Keep track of when the sinkpad was set to flushing and unflush the pad when an
871 upstream flushing seek failed.
873 2009-06-23 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
875 * gst/playback/gsturidecodebin.c:
876 uridecodebin: fix leak when the source fails to change state
878 2009-06-23 12:40:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
880 * gst/subparse/gstssaparse.c:
881 ssaparse: avoid leaking all buffers
883 2009-06-22 22:18:03 +0300 Stefan Kost <ensonic@users.sf.net>
885 * tests/check/elements/adder.c:
886 adder: test seek handling in adder
887 This tests seeking on an adder that has a normal and a live source connected.
888 Wheter the current behavior is the desired one needs to be discussed still
891 2009-06-22 16:17:10 +0300 Stefan Kost <ensonic@users.sf.net>
893 * sys/ximage/ximagesink.c:
894 * sys/xvimage/xvimagesink.c:
895 x(v)imagesink: pass the xwindow along to not look at the yet unset var.
896 When we call this from xwindow_new, x(v)imagesink->xwindow is not yet set.
898 2009-06-22 11:40:33 +0300 Stefan Kost <ensonic@users.sf.net>
900 * sys/ximage/ximagesink.c:
901 * sys/ximage/ximagesink.h:
902 * sys/xvimage/xvimagesink.c:
903 * sys/xvimage/xvimagesink.h:
904 x(v)imagesink: catch tags and show title in own window
905 Refactor the code that sets the window title. Catch tag-events and use title
906 metadata for the window title.
908 2009-06-21 19:42:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
910 * gst/audiotestsrc/gstaudiotestsrc.c:
911 audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
912 Also make all the function arrays constant.
914 2009-06-21 12:27:37 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
916 * gst/audiotestsrc/gstaudiotestsrc.c:
917 * gst/audiotestsrc/gstaudiotestsrc.h:
918 audiotestsrc: Add support for generating gaussian white noise
919 This patch adds support for stationary white Gaussian noise.
920 The Box-Muller algorithm is used to generate pairs of independent
921 normally-distributed random numbers.
924 2009-06-20 23:46:28 +0100 Jan Schmidt <thaytan@noraisin.net>
926 * gst/ffmpegcolorspace/imgconvert.c:
927 * gst/ffmpegcolorspace/imgconvert_template.h:
928 ffmpegcolorspace: Fix NV12 and NV21 transformations
929 Fix some stride problems, fix the nv12 to nv21 direct transformation,
930 and implement a direct conversion to yuv444 to save CPU.
932 2009-06-20 22:36:21 +0100 Jan Schmidt <thaytan@noraisin.net>
934 * gst/videotestsrc/videotestsrc.c:
935 videotestsrc: Fix NV12 painting for odd strides/heights
937 2009-06-19 22:16:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
939 * ext/cdparanoia/gstcdparanoiasrc.c:
940 cdparanoia: run-time license is LGPL now that we require cdparanoia 0.10.2
941 cdparanoia has an LGPL v2.1 license since 0.10.1 and we now require 0.10.2.
942 Finally fixes #531035.
944 2009-06-19 21:25:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
946 * ext/cdparanoia/gstcdparanoiasrc.c:
947 cdparanoia: try to guess a good cache size if it's set to -1
948 Try to guess from the paranoia-mode setting whether playback or
949 ripping is wanted, and use a smaller cache size if we're likely
950 to be doing playback, to avoid a long startup delay. Since this
951 was the value used in older cdparanoia versions, it should be
952 fine in any case. See #586331.
954 2009-06-19 11:27:40 +1000 Jonathan Matthew <jonathan@d14n.org>
957 * ext/cdparanoia/gstcdparanoiasrc.c:
958 * ext/cdparanoia/gstcdparanoiasrc.h:
959 cdparanoia: expose cache size setting
960 This setting was added in cdparanoia 10.2. The default value is good
961 for audio extraction, but lower values (previous versions of cdparanoia
962 used 150) are better for realtime playback.
965 2009-06-19 17:43:03 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
967 * gst-plugins-base.spec.in:
968 Make build of schro plugin conditional
970 2009-06-19 15:52:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
972 * docs/libs/gst-plugins-base-libs-sections.txt:
973 * gst-libs/gst/rtp/gstbasertppayload.c:
974 * gst-libs/gst/rtp/gstbasertppayload.h:
975 * win32/common/libgstrtp.def:
976 basertppayload: add support for bufferlists
977 Based on patch from Ognyan Tonchev.
980 2009-06-19 15:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
982 * gst-libs/gst/rtp/gstrtpbuffer.c:
983 rtpbuffer: use new convenience functions
984 New core convenience functions makes the list getters and setters trivial.
985 Maybe even too trivial...
987 2009-06-18 19:07:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
989 * win32/common/libgstrtp.def:
990 defs: add new symbol to win32 defs file
991 Based on patches by Ognyan Tonchev.
994 2009-06-18 19:04:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
996 * docs/libs/gst-plugins-base-libs-sections.txt:
997 * gst-libs/gst/rtp/gstrtpbuffer.c:
998 rtp: cleanups, add _list_get_seq() too
999 Clean up the docs a little.
1000 Add missing _list_get_seq method.
1001 Add new symbols to the docs
1003 2009-06-18 18:47:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1005 * gst-libs/gst/rtp/gstrtpbuffer.c:
1006 * win32/common/libgstrtp.def:
1008 Add Since tags to docs
1009 Move some code around
1012 2009-06-18 17:46:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1014 * gst-libs/gst/rtp/gstrtpbuffer.c:
1015 * gst-libs/gst/rtp/gstrtpbuffer.h:
1016 * tests/check/libs/rtp.c:
1017 rtp: add bufferlist support
1019 2009-06-18 18:03:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1021 * gst-libs/gst/rtp/gstrtpbuffer.c:
1022 rtp: pass data to macros instead of GstBuffer
1024 2009-06-18 17:42:10 +0100 Jan Schmidt <thaytan@noraisin.net>
1026 * win32/common/libgstrtsp.def:
1027 win32: Add gst_rtsp_watch_queue_data() to the exports
1028 Fix the tests by exporting the new symbol from the win32 dlls
1030 2009-06-18 18:13:22 +0300 Stefan Kost <ensonic@users.sf.net>
1032 * sys/xvimage/xvimagesink.c:
1033 xvimagesink: appname might be NULL
1034 Don't set title if appname is unknown.
1036 2009-06-18 17:58:06 +0300 Stefan Kost <ensonic@users.sf.net>
1038 * sys/xvimage/xvimagesink.c:
1039 xvimagesink: set window title from application name
1041 2009-06-09 19:14:00 +0200 Peter Kjellerstedt <pkj@axis.com>
1043 * gst-libs/gst/rtsp/gstrtspurl.c:
1044 rtsp: Made the parsing of the RTSP URL scheme more generic.
1046 2009-06-15 13:58:26 +0200 Peter Kjellerstedt <pkj@axis.com>
1048 * gst-libs/gst/rtsp/gstrtspconnection.c:
1049 * gst-libs/gst/rtsp/gstrtspconnection.h:
1050 rtsp: Added gst_rtsp_watch_queue_data().
1051 gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
1052 but allows for queuing any data block for writing (much like
1053 gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)
1054 API: gst_rtsp_watch_queue_data()
1056 2009-06-09 16:37:09 +0200 Peter Kjellerstedt <pkj@axis.com>
1058 * gst-libs/gst/rtsp/gstrtspconnection.c:
1059 rtsp: Only extract the session ID from RTSP responses.
1061 2009-06-09 19:06:57 +0200 Peter Kjellerstedt <pkj@axis.com>
1063 * gst-libs/gst/rtsp/gstrtspurl.c:
1064 rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
1066 2009-06-09 14:31:18 +0200 Peter Kjellerstedt <pkj@axis.com>
1068 * gst-libs/gst/rtsp/gstrtspconnection.c:
1069 rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
1071 2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
1073 * gst-libs/gst/rtsp/gstrtspconnection.c:
1074 rtsp: Improved base64 decoding in fill_bytes().
1075 The base64 decoding in fill_bytes() expected the size of the read data to
1076 be evenly divisible by four (which is true for the base64 encoded data
1077 itself). This did not, however, take whitespace (especially line breaks)
1078 into account and would fail the decoding if any whitespace was present.
1080 2009-06-17 14:00:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1082 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1083 audiosrc: fix get_offset
1084 When we need to jump to the most recently captured sample, jump to where the
1085 next sample will be written instead of to some old data.
1088 2009-06-17 13:18:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1090 * gst-libs/gst/audio/gstbaseaudiosink.c:
1091 audiosink: free the ringbuffer when going to NULL
1092 Unparent and free the ringbuffer when going to NULL, like we do with the
1093 audiosrc element. We can do this now because we correctly manage the time
1096 2009-06-17 13:17:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1098 * gst-libs/gst/audio/gstaudiosink.c:
1099 * gst-libs/gst/audio/gstaudiosrc.c:
1100 audio: correctly handle short read/writes
1102 2009-05-05 15:37:54 +0300 René Stadler <rene.stadler@nokia.com>
1104 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1105 baseaudiosrc: add some extra logging for buffer timestamps
1107 2009-06-17 11:22:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1109 * gst/adder/gstadder.c:
1110 adder: more seeking fixes.
1111 When a seek failed upstream, make sure the adder sinkpad is set unflushing again
1112 so that streaming can continue.
1113 We only have a pending segment when we flushed.
1114 Set the flush_stop_pending flag inside the appropriate locks and before we
1115 attempt to perform the upstream seek.
1116 Add some more comments.
1117 Use the right lock to protect the flags in flush_stop.
1120 2009-06-17 07:24:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1122 * gst/playback/gstdecodebin2.c:
1123 decodebin2: Free iterator after removing all groups
1125 2009-06-16 19:38:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1127 * gst-libs/gst/video/gstvideofilter.c:
1128 videofilter: Add a default get_unit_size function
1129 This returns the correct values for all formats that are handled by
1130 GstVideoFormat and makes all the custom get_unit_size functions in
1131 many elements unnecessary.
1133 2009-06-16 18:57:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1135 * gst-libs/gst/rtsp/gstrtspdefs.c:
1136 * gst-libs/gst/rtsp/gstrtspdefs.h:
1137 rtsp: add Timestamp header field
1140 2009-06-16 18:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1142 * gst/playback/gstplaybin2.c:
1143 playbin2: set smarter target state on uridecodebin
1144 Set the target state of the newly added uridecodebins to somthing else that
1145 PAUSED so that we keep their state in sync with the playsink state.
1148 2009-06-16 18:13:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1150 * gst/playback/gstplaysink.c:
1151 playsink: set the sink flag on the element
1153 2009-06-16 18:09:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1155 * gst/playback/gsturidecodebin.c:
1156 uridecodebin: add debug message
1158 2009-06-16 14:05:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1160 * gst-libs/gst/audio/gstaudiosink.c:
1161 * gst-libs/gst/audio/gstaudiosrc.c:
1162 audiosink, audiosrc: do the class_ref()s in the right class_init functions
1163 Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
1165 2009-06-15 15:39:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1167 * gst-libs/gst/audio/gstaudiosink.c:
1168 * gst-libs/gst/audio/gstaudiosrc.c:
1169 audiosink,audiosrc: ref the audio ring buffer class and type in class_init
1170 Hack around thread-safety issues in GObject and our racy _get_type()
1171 functions (we could easily fix the _get_type() functions, but we still
1172 need to hack around the GObject class races until we require a newer
1173 GLib version, I think).
1175 2009-06-15 12:57:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1177 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1178 audiosrc: return FALSE when receiving a SEEK event
1179 When receiving a seek event, return FALSE as we don't implement seeking.
1181 2009-06-15 11:06:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1183 * tests/examples/seek/seek.c:
1184 Don't use deprecated GTK API
1187 2009-06-15 11:40:00 +0300 Stefan Kost <ensonic@users.sf.net>
1189 * gst/adder/gstadder.c:
1190 adder: send flush_stop when seeking failed
1191 At least do the fix to sent the flush_stop when seeking failed to ensure we
1192 keep no pads flushing. before it was send when the seeking worked which is just
1193 plain wrong and was not the intention.
1195 2009-06-12 15:17:14 +0200 Peter Kjellerstedt <pkj@axis.com>
1197 * gst-libs/gst/rtsp/gstrtspconnection.c:
1198 rtsp: Use a more consistent naming of GstRTSPRec variables.
1200 2009-06-12 15:11:05 +0200 Peter Kjellerstedt <pkj@axis.com>
1202 * gst-libs/gst/rtsp/gstrtspconnection.c:
1203 * gst-libs/gst/rtsp/gstrtspconnection.h:
1204 rtsp: Call message_sent() callback for all sent messages.
1205 Previously the messages_sent() callback was only called for messages
1206 which had a CSeq, which excluded all data messages. Instead of using the
1207 CSeq as ID, use a simple index counter.
1209 2009-06-14 22:13:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1211 * ext/ogg/gstoggdemux.c:
1212 * ext/theora/theoradec.c:
1213 * ext/vorbis/vorbisdec.c:
1214 oggdemux: post/send tags with the container-format tag
1215 For this to work properly, theoradec and vorbisdec need to put
1216 tag events received from upstream into the pending_events list
1217 so they get pushed out after any newsegment event, not before.
1219 2009-06-14 20:30:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1221 * tests/examples/seek/scrubby.c:
1222 * tests/examples/seek/seek.c:
1223 * tests/old/examples/seek/cdplayer.c:
1224 Don't use deprecated GTK API
1227 2009-06-12 16:31:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1229 * gst/adder/gstadder.c:
1230 adder: send flush-stop earlier
1231 When no flush-stop has been sent by upstream, we have to send one ourselves to
1232 continue playback. Do this as soon as the collect function is called instead of
1233 after we possibly pushed segment events (that got then flushed out)
1235 2009-06-12 13:55:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1237 * tests/examples/seek/seek.c:
1238 seek: add shuttle controls
1240 2009-06-12 13:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1242 * tests/examples/seek/stepping2.c:
1243 example: fix compile
1245 2009-06-12 13:52:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1247 * tests/examples/seek/Makefile.am:
1248 examples: build the stepping2 example
1250 2009-06-12 13:52:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1252 * gst/playback/gstplaysink.c:
1253 playsink: update for new step API
1255 2009-06-12 13:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1257 * ext/ogg/gstoggdemux.c:
1258 oggdemux: do reverse seeks more accurate
1259 For reverse seeking with the accurate flag set, try to be more precise by
1260 seeking a little bit after the requested position.
1262 2009-06-11 22:32:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1264 * ext/ogg/gstogmparse.c:
1265 * gst/subparse/gstssaparse.c:
1266 * gst/subparse/gstssaparse.h:
1267 * gst/subparse/gstsubparse.c:
1268 * gst/subparse/gstsubparse.h:
1269 subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC
1270 Make subtitle parsers post a taglist with codec tags, so the application
1271 knows what kind of subtitle a subtitle stream is. Fixes #576552.
1273 2009-06-11 19:12:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1275 * gst-libs/gst/audio/gstringbuffer.c:
1276 ringbuffer: handle border cases in resampler
1278 2009-06-11 13:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
1281 * docs/libs/Makefile.am:
1282 * docs/plugins/Makefile.am:
1283 docs: Update common. Use upload-doc.mak instead of upload.mak
1285 2009-06-11 12:39:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1287 * gst-libs/gst/rtp/gstbasertppayload.c:
1290 2009-06-11 12:17:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1292 * gst-libs/gst/audio/gstbaseaudiosink.c:
1293 baseaudiosink: reset accum when dropping samples
1294 When we are resampling and we drop samples because we paused, reset the accum
1295 counter because it's now invalid.
1297 2009-06-11 11:16:15 +0100 Jan Schmidt <thaytan@noraisin.net>
1299 * docs/libs/gst-plugins-base-libs-sections.txt:
1300 * gst-libs/gst/interfaces/mixer.h:
1301 * gst-libs/gst/video/gstbasevideodecoder.h:
1302 docs: Fix a couple of warnings from the docs build.
1304 2009-06-10 21:36:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1306 * gst-libs/gst/audio/testchannels.c:
1307 Don't include config.h multiple times when build audio testchannel app.
1308 Fixes build problem on win32 (#585075).
1310 2009-06-10 16:56:51 +0100 Jan Schmidt <thaytan@noraisin.net>
1312 * gst/playback/gstplaybin2.c:
1313 * gst/playback/gsturidecodebin.c:
1314 playbin2/uridecodebin: Fix connection-speed propagation
1315 uridecodebin expects the passed connection-speed value in kbps, so we
1316 need to divide the value stored in bps by 1000. Also, lower the upper
1317 limit on the properties to the value that we can actually store in our
1318 internal guint (which is plenty high enough)
1320 2009-06-10 14:37:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1322 * gst/subparse/gstsubparse.c:
1323 * tests/check/elements/subparse.c:
1324 subparse: recognise more subrip timestamp variants
1325 Be even less restrictive in what we accept for .srt timestamps when
1326 typefinding and parsing subrip subtitles and add a unit test for
1327 the 'new' format. Fixes #585197.
1329 2009-06-09 22:00:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1331 * gst-libs/gst/rtsp/gstrtsptransport.h:
1332 rtsp: add some more docs
1334 2009-06-09 18:24:55 +0200 Peter Kjellerstedt <pkj@axis.com>
1336 * gst-libs/gst/rtsp/gstrtspmessage.c:
1337 rtsp: Avoid a compiler warning.
1339 2009-06-09 18:23:28 +0200 Peter Kjellerstedt <pkj@axis.com>
1341 * gst-libs/gst/rtsp/gstrtspdefs.h:
1342 rtsp: Updated documentation for GstRTSPResult.
1343 Moved GST_RTSP_ELAST to be last in the documentation to match the actual
1346 2009-05-20 17:30:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1349 autogen: remove -Wno-portability from here
1350 as it is in configure.ac now.
1352 2009-06-09 16:28:20 +0200 Peter Kjellerstedt <pkj@axis.com>
1354 * gst-libs/gst/rtsp/gstrtspconnection.c:
1355 rtsp: Plug a memory leak.
1356 Free memory related to any partially read and/or written RTSP messages.
1358 2009-06-09 12:09:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1360 * gst-libs/gst/audio/gstbaseaudiosink.c:
1361 baseaudiosink: no need to cause discont when clipping
1362 Remove the discont-when-clipping hack now that basesink provides us with
1363 correctly clipped samples when stepping.
1365 2009-06-08 17:26:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1367 * gst-libs/gst/audio/gstbaseaudiosink.c:
1368 audiosink: don't align when we clip
1369 Don't align samples when they were clipped. Not entirely correct but better than
1372 2009-06-08 16:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1374 * tests/examples/seek/.gitignore:
1375 * tests/examples/seek/stepping2.c:
1376 examples: add stepping example in PLAYING
1377 Add stepping example in PLAYING, audio is a bit distorted because basesink does
1378 not provide good clipping info yet.
1380 2009-06-08 10:25:00 +0200 Edward Hervey <bilboed@bilboed.com>
1382 * gst-libs/gst/pbutils/descriptions.c:
1383 pbutils: Add description for hdv/aux-* formats.
1385 2009-06-07 22:20:33 +0400 LRN <lrn1986@gmail.com>
1387 * ext/schroedinger/Makefile.am:
1388 Added libgstbase to schro's LIBADD
1391 2009-06-06 02:15:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1393 * gst-libs/gst/tag/gstid3tag.c:
1394 libgsttag: don't extract genres from empty ID3v1 tags
1395 If we don't have any other info, don't try to interpret the
1396 genre field. In particular we don't want to interpret a genre
1397 of 0 as 'Blues' if no other fields are set and the entire tag
1400 2009-06-05 18:13:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1402 * gst/playback/gstdecodebin2.c:
1403 decodebin2: make sure varargs are of right type
1404 Explicitly cast the variables to g_object_set to their right types.
1406 2009-06-05 16:49:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1408 * gst/playback/gstdecodebin2.c:
1409 decodebin2: increase stream probing queues
1410 When we are probing for streams, we want to set the queue size in such a way
1411 that we can scan a maximum amount of data without consuming too much memory.
1412 Therefore, remove the time limit on the queue and only stop scanning after 2MB
1416 2009-06-05 14:06:17 +0200 Peter Kjellerstedt <pkj@axis.com>
1418 * gst-libs/gst/rtsp/gstrtspconnection.c:
1421 2009-06-05 14:05:54 +0200 Peter Kjellerstedt <pkj@axis.com>
1423 * gst-libs/gst/rtsp/gstrtspconnection.c:
1424 rtsp: Remove an unused variable.
1426 2009-06-05 13:59:14 +0200 Peter Kjellerstedt <pkj@axis.com>
1428 * gst-libs/gst/rtsp/gstrtspconnection.c:
1429 rtsp: Removed duplicate initialization of conn->writefd.
1431 2009-06-05 13:55:08 +0200 Peter Kjellerstedt <pkj@axis.com>
1433 * gst-libs/gst/rtsp/gstrtspconnection.c:
1434 rtsp: Use #defined status codes.
1436 2009-06-05 13:53:29 +0200 Peter Kjellerstedt <pkj@axis.com>
1438 * gst-libs/gst/rtsp/gstrtspconnection.c:
1439 rtsp: Correct gen_tunnel_reply().
1440 Prevent gen_tunnel_reply() from generating an incomplete response
1441 in case an error response code is given.
1443 2009-06-05 10:57:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1446 * win32/common/_stdint.h:
1447 * win32/common/config.h:
1448 * win32/common/video-enumtypes.c:
1449 configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
1450 See #584835. Also update win32 files while we're at it.
1452 2009-06-04 08:57:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1454 * gst/playback/gstplaybin2.c:
1455 playbin2: API: Add {audio,video,text}-tags-changed signals
1458 2009-06-03 20:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1460 * ext/vorbis/vorbisdec.c:
1461 vorbisdec: don't put invalid bitrate values into the taglist
1462 Bitrates are stored as 32-bit signed integers in the vorbis
1463 identification headers, but seem to be read incorrectly,
1464 namely as unsigned 32-bit integers, into the vorbis structure
1465 members which are of type long, which makes our check for
1466 values <= 0 fail with files that put -1 in there for unset
1469 2009-06-03 15:52:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1471 * tests/examples/seek/.gitignore:
1472 ignore: add new stepping app to ignore
1474 2009-06-03 15:31:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1476 * tests/examples/seek/Makefile.am:
1477 * tests/examples/seek/stepping.c:
1478 examples: add stepping example.
1479 Add an example of using playbin2 and frame stepping to simulate variable rate
1480 playback based on a sine wave.
1482 2009-06-03 12:45:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1484 * gst/playback/gstplaybin2.c:
1485 * gst/playback/gstplaysink.h:
1486 playbin2: also set custom text and subp sinks
1487 Set the custom subpicture and text sinks along with the custom audio and video
1489 Fix a little docs blurb too.
1491 2009-06-02 12:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1493 * gst-libs/gst/rtsp/gstrtspconnection.c:
1494 * gst-libs/gst/rtsp/gstrtspconnection.h:
1495 rtsp: add G_LIKELY because we can
1497 2009-06-02 09:53:05 +0200 Edward Hervey <bilboed@bilboed.com>
1499 * gst/typefind/gsttypefindfunctions.c:
1500 typefindfunctions: Fix caps for ogg typefinder.
1502 2009-05-29 11:10:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1504 * docs/libs/gst-plugins-base-libs-sections.txt:
1505 docs: remove some cruft from -sections.txt file
1507 2009-06-01 11:31:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1509 * gst/playback/gstplaysink.c:
1510 * tests/examples/seek/seek.c:
1511 add framestepping to playbin2 and seek
1513 2009-06-01 09:59:22 +0200 Peter Kjellerstedt <pkj@axis.com>
1515 * gst-libs/gst/rtsp/gstrtspconnection.c:
1516 rtsp: Avoid compiler warnings with -Wextra.
1518 2009-06-01 09:58:27 +0200 Peter Kjellerstedt <pkj@axis.com>
1520 * gst-libs/gst/rtsp/gstrtspconnection.h:
1521 rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
1523 2009-06-01 09:43:04 +0200 Peter Kjellerstedt <pkj@axis.com>
1525 * gst-libs/gst/sdp/gstsdpmessage.c:
1526 sdp: Remove an unused variable.
1528 2009-05-30 14:17:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1530 * gst/ffmpegcolorspace/imgconvert.c:
1531 * gst/ffmpegcolorspace/imgconvert_template.h:
1532 ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale
1534 2009-05-29 00:09:15 +0100 Jan Schmidt <thaytan@noraisin.net>
1536 * gst/playback/gstplaybin2.c:
1537 playbin2: Have playbin recognise PGS subpicture streams
1538 Recognise PGS subpicture streams and connect them to the SPU pad
1539 in playsink. Unfortunately this fails badly with negotiation errors
1540 if the SPU is not recent enough to support the stream. I'm not sure
1541 how to add format negotiation in yet.
1543 2009-05-21 23:11:29 +0100 Jan Schmidt <thaytan@noraisin.net>
1545 * gst/playback/gstdecodebin2.c:
1546 * gst/playback/gsturidecodebin.c:
1547 decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.
1549 2009-05-28 20:37:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1551 * gst/playback/gstplaysink.c:
1552 playbin2: fix volume handling for audio sinks without "volume" property
1553 When using an audio sink without a "volume" property, volume control
1554 would only work for the first song. For the next song, we'd try to
1555 re-use the existing audio chain, but inadvertently set chain->volume
1556 to NULL instead of to the existing volume element.
1558 2009-05-28 17:05:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1560 * gst/playback/gstplaysink.c:
1561 playbin2: cosmetic change to avoid unnecessary line breaks
1562 Looks nicer and works around gst-indent silliness.
1564 2009-05-28 17:21:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1566 * gst/playback/gstplaysink.c:
1567 playbin2: don't lose the ref to the volume element
1568 Only release the ref to the volume element when it is controled by a sink. For
1569 software volume we never have to fear that it will change.
1571 2009-05-28 15:21:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1573 * gst/playback/gstplaybin2.c:
1574 * gst/playback/gstplaysink.c:
1575 playbin2: actually use configured audio/video sinks
1576 playbin2 inadvertently used autoaudiosink and autovideosink up to now,
1577 since it would overwrite the sinks configured via the "audio-sink"
1578 and "video-sink" properties with the stream-specific group sinks when
1579 configuring the outputs. Those are usually NULL however, so that would
1580 overwrite the configured sinks with NULL which makes playbin2 then
1581 default to the auto sinks. Fix this by keeping a reference to each
1582 configured sink in playbin2 and setting up the right sinks depending
1583 on whether there is a stream-specific sink or not.
1586 2009-05-27 17:37:38 +0300 Stefan Kost <ensonic@users.sf.net>
1588 * tests/examples/seek/seek.c:
1589 seek: add volume label and sync with sink volume
1590 Look at the volume and have the pulsemixer open at same time. Unfortunately
1591 playbin2 does not emit notify on volume right, so this polls for now.
1593 2009-05-27 18:12:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1595 * gst/playback/gstdecodebin2.c:
1596 decodebin2: remove leftover elements
1597 Remove all of the elements inside decodebin2 when goint to READY and NULL.
1598 Makes decodebin2 reusable.
1601 2009-05-27 15:36:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1603 * gst/playback/gstplaysink.c:
1604 playbin2; release refs to volume/mute properties
1605 Release the refs to the volume and mute property elemens before setting the
1606 child elements to READY or NULL.
1609 2009-05-27 12:10:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1611 * gst/gdp/gstgdppay.c:
1612 gdppay: set caps on outgoing buffers
1613 Set caps on outgoing buffers because NULL caps confuse basetransform.
1616 2009-05-27 11:08:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1618 * gst-libs/gst/netbuffer/gstnetbuffer.c:
1619 netbuffer: also note the order of IP4 addresses
1620 IP4 addresses are also stored in network byte order. Make a note of this in the
1623 2009-05-26 22:43:34 +0200 Alessandro Decina <alessandro.d@gmail.com>
1625 * ext/theora/theoraparse.c:
1626 theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.
1628 2009-05-26 11:13:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1630 * gst-libs/gst/rtsp/gstrtspconnection.c:
1631 Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14"
1632 This reverts commit 418760cf740332c12c3fd9cf3244af134fa9534b.
1633 We now require GLib 2.16.
1635 2009-05-26 15:18:09 +0100 Jan Schmidt <thaytan@noraisin.net>
1640 2009-05-26 15:37:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1642 * gst-libs/gst/netbuffer/gstnetbuffer.c:
1643 netbuffer: document that the port is network order
1644 Document the fact that we store the port number in network order in
1645 GstNetAddress and that the caller should byteswap appropriately.
1647 2009-05-26 15:23:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1649 * gst/videoscale/gstvideoscale.c:
1650 * gst/videoscale/vs_4tap.c:
1651 * gst/videoscale/vs_4tap.h:
1652 * gst/videoscale/vs_image.c:
1653 * gst/videoscale/vs_image.h:
1654 * gst/videoscale/vs_scanline.c:
1655 * gst/videoscale/vs_scanline.h:
1656 videoscale: Add support for 16 bit grayscale in native endianness
1658 2009-05-26 14:58:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1660 * gst/ffmpegcolorspace/avcodec.h:
1661 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
1662 * gst/ffmpegcolorspace/imgconvert.c:
1663 ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian
1665 2009-05-26 14:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1667 * gst/videotestsrc/videotestsrc.c:
1668 * gst/videotestsrc/videotestsrc.h:
1669 videotestsrc: Add support for 16 bit grayscale in native endianness
1671 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
1673 add can-activate-pull property to baseaudiosink
1674 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
1677 2009-05-26 13:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1679 * ext/ogg/gstoggdemux.c:
1680 oggdemux: fix boundary case for seeking.
1681 When we have exactly 0 bytes left to search, make sure we stop instead of going
1682 into an infinite loop.
1684 2009-05-26 11:11:03 +0200 Bastien Nocera <hadess at hadess.net>
1686 * gst-libs/gst/cdda/Makefile.am:
1687 * gst-libs/gst/cdda/gstcddabasesrc.c:
1688 * gst-libs/gst/cdda/sha1.c:
1689 * gst-libs/gst/cdda/sha1.h:
1690 cddabasesrc: Remove copy of sha1 digest
1691 Remove our copy of sha1 digest now that we depend on glib 2.16.
1694 2009-05-25 17:54:01 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
1696 * gst-plugins-base.spec.in:
1699 2009-05-23 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1701 * gst-libs/gst/video/gstbasevideodecoder.c:
1702 * gst-libs/gst/video/gstbasevideoparse.c:
1703 * gst-libs/gst/video/gstbasevideoutils.c:
1704 * gst-libs/gst/video/gstbasevideoutils.h:
1705 * win32/common/libgstvideo.def:
1706 video: don't expose internal gst_adapter_get_buffer() helper function
1707 If it's really needed it should go into GstAdapter in core.
1709 2009-05-22 21:29:51 -0700 David Schleef <ds@schleef.org>
1711 * gst-libs/gst/video/gstbasevideodecoder.c:
1712 basevideo: Fix memleak
1714 2009-05-22 21:27:58 -0700 David Schleef <ds@schleef.org>
1716 * ext/schroedinger/gstschrodec.c:
1717 * ext/schroedinger/gstschroparse.c:
1718 schro: Fix usage of adapter_masked_scan_uint32
1719 Because *somebody* changed the API without telling me.
1721 2009-05-22 21:25:06 -0700 David Schleef <ds@schleef.org>
1723 * ext/schroedinger/gstschro.c:
1724 schro: Change package name to GST_PACKAGE_NAME
1726 2009-05-22 17:34:10 -0700 David Schleef <ds@schleef.org>
1728 * gst-libs/gst/video/gstbasevideoencoder.c:
1729 basevideo: Add preset interface to encoder
1731 2009-05-22 17:31:14 -0700 David Schleef <ds@schleef.org>
1733 * gst/audioresample/gstaudioresample.c:
1734 Run liboil benchmark multiple times
1735 The statistics function requires multiple runs, otherwise
1736 it causes a divide by zero error.
1738 2009-05-22 19:36:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1740 * m4/gst-fionread.m4:
1741 m4: fix 'suspicious cache value' warning for gst-fionread.m4
1742 .. here as well (should really be moved to common, but I'm too lazy).
1744 2009-05-22 17:41:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1746 * ext/vorbis/vorbisdec.c:
1747 vorbisdec: detect and report errors better
1748 Check the return values of a couple more libvorbis functions and post an error
1749 when something is wrong instead of continuing and crashing.
1751 2009-05-22 15:49:14 +0300 Stefan Kost <ensonic@users.sf.net>
1753 * gst/playback/gstplaysink.c:
1754 playbin2: fix initial volume and mute handling
1755 Use two flags to remember volume/mute changes at times when we don't have the
1756 audiochain yet (e.g. construction). Only set values when they were actualy
1757 changed. This makes pulseaudio's stream restore functional.
1759 2009-05-22 10:19:51 +0100 Jan Schmidt <thaytan@noraisin.net>
1762 Automatic update of common submodule
1763 From d3a8fab to 888e0a2
1765 2009-05-22 09:03:22 +0100 Jan Schmidt <thaytan@noraisin.net>
1767 * win32/common/libgstvideo.def:
1768 win32: Remove gst_adapter_masked_scan_uint32 from the exports
1770 2009-05-21 10:48:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1772 * gst-libs/gst/audio/gstbaseaudiosink.c:
1773 audiosink: improve debug message
1775 2009-05-19 18:10:55 -0700 Michael Smith <msmith@songbirdnest.com>
1777 * gst-libs/gst/tag/gstid3tag.c:
1778 gstid3tag: Don't extract a track number unless present.
1779 In ID3v1, a track number is present only if byte 125 is null AND
1780 byte 126 is non-null. If the track number is not present, don't add
1781 a track number tag with value 0.
1783 2009-05-20 00:48:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1785 * gst-libs/gst/video/gstbasevideoutils.c:
1786 * gst-libs/gst/video/gstbasevideoutils.h:
1787 videoutils: remove adapter methods
1788 Remove adapter methods now that they are in core.
1790 2009-05-20 00:42:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1792 * win32/common/libgstvideo.def:
1793 defs: add new symbols
1795 2009-05-19 17:47:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1798 autogen: pass -Wno-portability to automake to suppress warnings
1801 2009-05-19 02:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1803 * docs/libs/.gitignore:
1804 gitignore: remove bogus *.sgml wildcard - these files are tracked in git
1806 2009-05-19 18:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1808 * gst/tcp/gsttcpclientsrc.c:
1809 tcpclientsrc: this is not a live source
1810 Don't mark us as a live source because we are not.
1812 2009-05-19 18:41:02 +0300 Stefan Kost <ensonic@users.sf.net>
1814 * gst/adder/gstadder.c:
1815 adder: only send flush_stop when seek failed
1816 This is still not the ultimate fix. Added some comment to explain the troubles.
1818 2009-05-19 17:17:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1820 * gst-libs/gst/audio/gstbaseaudiosink.c:
1821 audiosink: return the return value of wait_preroll
1822 Return the value that _wait_preroll() returned instead of always WRONG_STATE.
1824 2009-05-19 16:45:56 +0300 Stefan Kost <ensonic@users.sf.net>
1826 * gst/adder/gstadder.c:
1827 * gst/adder/gstadder.h:
1828 adder: send flush_stop to match flush_start
1829 Adder was relying that something else sends a flush stop. When using adder with
1830 a livesource it was not getting a flush_stop and thus all pads downstream where
1831 keept flushing. Mark a pending flush_stop and send it when we are working on
1832 the new segment back in the streaming thread.
1834 2009-05-19 16:02:44 +0300 Stefan Kost <ensonic@users.sf.net>
1836 * tests/examples/seek/seek.c:
1837 seek: ui improvements
1838 Repaint the window black on expose, as this looks nicer when resizing or using
1839 the expander. Also show time after slider, as this saves a whole line (nice on
1842 2009-04-29 18:36:17 +0300 Stefan Kost <ensonic@users.sf.net>
1844 * gst/playback/gstdecodebin.c:
1845 decodebin: use iterators instead of list
1846 The list api is deprecated. Use threadsafe iterators instead.
1848 2009-05-19 15:35:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1850 * gst/playback/gsturidecodebin.c:
1851 uridecodebin: configure caps on decodebin2
1852 Implement the caps property by setting the configured caps on new decodebin2
1856 2009-05-19 15:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1858 * gst/playback/gstdecodebin2.c:
1859 decodebin2: avoid some _caps_ref in some cases
1860 Only mess with the caps refcount when we configure different caps.
1862 2009-05-19 15:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1864 * gst/playback/gsturidecodebin.c:
1865 uridecodebin: fix potential caps leak
1866 Free the user-configured caps in finalize.
1868 2009-05-19 15:20:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1870 * gst/playback/gsturidecodebin.c:
1871 uridecodebin: add queue after cdda://
1872 Add a queue2 after the raw output pads of certain sources such as those for uris
1874 No tuning of the queue is done yet as the defaults seem to work fine for me.
1877 2009-05-19 12:45:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1879 * ext/ogg/gstoggdemux.c:
1880 oggdemux: don't loop when at EOS
1881 When we try to read the last page, don't try to read past the upper boundary, as
1882 this might cause endless loops.
1885 2009-05-19 11:20:19 +0200 Edward Hervey <bilboed@bilboed.com>
1887 * gst/audioresample/gstaudioresample.c:
1888 audioresample: Don't drain remaining buffers after a flush.
1889 If we were resetted (due to a flush), we can not drain the remaining
1890 buffers since they would be pushed before a valid new newsegment event.
1892 2009-05-18 22:29:07 -0700 Michael Smith <msmith@syncword.(none)>
1894 * ext/theora/theoradec.c:
1895 theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.
1897 2009-05-19 01:13:34 +0300 Stefan Kost <ensonic@users.sf.net>
1899 * gst/adder/gstadder.c:
1900 adder: add more logging and return value checking
1902 2009-05-19 01:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
1904 * gst/adder/gstadder.c:
1905 adder: handle the return value from iterator_fold
1907 2009-05-19 01:03:44 +0300 Stefan Kost <ensonic@users.sf.net>
1909 * gst/adder/gstadder.c:
1910 adder: use the pad in logging as objects
1911 Helps to differenciate between source and sinks pads.
1913 2009-04-21 22:54:19 +0300 Stefan Kost <ensonic@users.sf.net>
1915 * tests/examples/seek/seek.c:
1916 seek: use parser for mp3 and rename variable
1918 2009-05-18 11:08:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1920 * tests/examples/seek/seek.c:
1921 seek: add playbin2 options in expander
1922 Add the playbin2 stream selection options inside an expander to preserve some
1925 2009-02-10 15:29:10 -0800 David Schleef <ds@schleef.org>
1927 * gst/videotestsrc/videotestsrc.c:
1928 videotestsrc: Add support for v210 and v216 formats
1930 2009-05-15 16:21:15 -0700 David Schleef <ds@schleef.org>
1932 * gst-libs/gst/video/gstbasevideocodec.c:
1933 * gst-libs/gst/video/gstbasevideodecoder.c:
1934 * gst-libs/gst/video/gstbasevideoencoder.c:
1935 * gst-libs/gst/video/gstbasevideoparse.c:
1936 video: remove // comments
1938 2009-05-15 16:18:18 -0700 David Schleef <ds@schleef.org>
1940 * gst-libs/gst/video/video.c:
1941 * gst-libs/gst/video/video.h:
1942 video: Add Y444, v210, v216 formats
1944 2009-05-15 16:12:37 -0700 David Schleef <ds@schleef.org>
1948 * ext/schroedinger/Makefile.am:
1949 * ext/schroedinger/gstschro.c:
1950 * ext/schroedinger/gstschrodec.c:
1951 * ext/schroedinger/gstschroenc.c:
1952 * ext/schroedinger/gstschroparse.c:
1953 * ext/schroedinger/gstschroutils.c:
1954 * ext/schroedinger/gstschroutils.h:
1955 schro: Move schro plugin from Schroedinger
1956 Previous history is in Schroedinger. Depends on, and is an example
1957 of using, GstBaseVideo* base classes.
1958 Code was reindented, and an #ifdef HAVE_ENCODER removed.
1960 2009-05-15 10:23:08 -0700 David Schleef <ds@schleef.org>
1962 * gst-libs/gst/video/Makefile.am:
1963 * gst-libs/gst/video/gstbasevideocodec.c:
1964 * gst-libs/gst/video/gstbasevideocodec.h:
1965 * gst-libs/gst/video/gstbasevideodecoder.c:
1966 * gst-libs/gst/video/gstbasevideodecoder.h:
1967 * gst-libs/gst/video/gstbasevideoencoder.c:
1968 * gst-libs/gst/video/gstbasevideoencoder.h:
1969 * gst-libs/gst/video/gstbasevideoparse.c:
1970 * gst-libs/gst/video/gstbasevideoparse.h:
1971 * gst-libs/gst/video/gstbasevideoutils.c:
1972 * gst-libs/gst/video/gstbasevideoutils.h:
1973 video: Copy BaseVideo classes from Schroedinger
1975 2009-05-15 23:05:45 +0200 Arnout Vandecappelle <arnout@mind.be>
1977 * gst/tcp/gstmultifdsink.c:
1978 multifdsink: add num-fds property
1979 multifdsink::num-fds
1981 2009-05-15 20:36:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1983 * gst-libs/gst/pbutils/descriptions.c:
1984 pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
1986 2009-05-14 11:44:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1988 * ext/vorbis/vorbisenc.c:
1989 vorbisenc: Implement Preset interface
1991 2009-05-14 11:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1993 * ext/theora/theoraenc.c:
1994 theoraenc: Implement Preset interface
1996 2009-05-14 11:41:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1998 * ext/ogg/gstoggmux.c:
1999 oggmux: Implement Preset interface
2001 2009-05-14 21:37:22 +0100 Jan Schmidt <thaytan@noraisin.net>
2003 * gst/playback/gstplaysink.c:
2004 playbin2: Fix cdda:// playback
2005 Don't send async-start when the playsink has already been configured
2006 before changing state.
2008 2009-05-14 01:31:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2011 configure: require core CVS for gst_adapter_prev_timestamp()
2012 which is used in the libvisual plugin.
2014 2009-04-22 18:34:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2017 AUTHORS: fix my email
2019 2009-04-22 18:35:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2021 * gst-libs/gst/audio/gstaudioclock.c:
2022 audioclock: make our internal time monotonic
2023 Make the internal time increase monotonically.
2025 2009-05-13 19:27:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2027 * ext/libvisual/visual.c:
2028 visual: remove next_ts variable
2029 We can remove the next_ts variable as we don't use it anymore.
2031 2009-05-13 19:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2033 * ext/libvisual/visual.c:
2034 visual: use new adapter timestamp code
2035 Use the new adapter timestamp tracking code to make things easier and produce
2036 vastly better output timestamps.
2038 2009-05-13 01:35:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2070 po: avoid conflicts of local *.po files with files in git
2071 Make it so that filenames and line numbers are only stored in the *.pot file
2072 (which is not in git), but not in the individual *.po files. This information
2073 is hardly useful for translators in our case, and it should avoid the constant
2074 conflicts of local *.po files with the ones in git which are caused by the
2075 source files changing and the line numbers being updated. This commit might
2076 cause one last merge conflict for you, which you can work around with
2077 "git checkout po/*.po" before merging or pulling. After that there should
2078 (hopefully) not be any more local modifications of these files (unless
2079 someone committed additions or changes to translated strings and the
2080 *.po files haven't been updated yet, that is).
2082 2009-05-12 23:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2084 * tests/check/elements/.gitignore:
2085 * tests/check/elements/audioresample.c:
2086 tests: fix audioresample unit test on big endian architectures
2087 Don't hardcode endianness=1234 in the filtercaps, it will cause
2088 pad link failures which will result in the test timing out.
2090 2009-05-12 17:18:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2092 * gst/audiotestsrc/gstaudiotestsrc.c:
2093 audiotestsrc: fix broken enum nick - it should have a hyphen
2094 The enum nick should be 'sine-table', not 'sine table'. Technically this is
2095 an API/ABI change I guess, but anyone who was using this and didn't report
2098 2009-05-01 01:04:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2100 * gst/audiotestsrc/gstaudiotestsrc.c:
2101 audiotestsrc: seek to the requested byte offset, not the expected byte offset
2103 2009-05-01 01:03:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2105 * gst/audiotestsrc/gstaudiotestsrc.c:
2106 * gst/audiotestsrc/gstaudiotestsrc.h:
2107 audiotestsrc: support more than just one channel
2109 2009-05-12 15:52:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2111 * gst-libs/gst/interfaces/propertyprobe.h:
2112 propertyprobe: Fix typo in the docs
2114 2009-05-12 12:17:55 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
2116 * ext/ogg/gstoggmux.c:
2117 * ext/theora/theora.c:
2118 * ext/vorbis/vorbis.c:
2119 Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder
2121 2009-04-30 16:37:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2123 * gst/videorate/gstvideorate.c:
2124 * gst/videorate/gstvideorate.h:
2125 videorate: handle invalid timestamps better
2126 Handle buffers with -1 timestamps better by keeping track of the en time of the
2127 previous buffer and assuming the -1 timestamp buffer goes right after the
2129 when we have two buffers that are equally good, output the oldest buffer once to
2131 don't try to calculate latency when the input framerate is unknown.
2133 2009-04-28 11:37:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2135 * ext/ogg/gstoggmux.c:
2136 oggmux: small debug statement in DISCONT
2138 2009-04-28 11:24:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2140 * ext/ogg/gstoggdemux.c:
2141 * ext/ogg/gstoggdemux.h:
2142 oggdemux: fix abuse of ogg API, handle broken oggs
2143 When we feed the ogg sync layer, we need to feed it contiguous data even if the
2144 sync layer did not consume all of it yet. This makes sure that it always finds
2145 the next page even for more corrupted files. Use a different read_offset for
2146 this purpose. since we now keep track of the sync layer, we don't have to reset
2147 after finding a start of a page.
2148 Add some more debug info for the error paths.
2149 Only reset the sync layer when we perform a seek operation.
2150 Avoid failure when the next chain has no bos pages but instead simply ignore it.
2151 when we receive unknown page serial numbers mid stream, don't fail but post a
2152 warning and hope that we get back on track later.
2155 2009-04-30 16:41:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2157 * gst/playback/gstdecodebin2.c:
2158 decodebin2: make subpictures a raw output format
2159 Subpictures are a raw format, we want those pads exposed so that playbin2 can do
2160 the subpicture mixing.
2162 2009-04-27 10:15:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2164 * gst-libs/gst/rtp/gstbasertppayload.c:
2165 * gst-libs/gst/rtp/gstbasertppayload.h:
2166 rtpdepay: add some more comments
2168 2009-04-17 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2170 * gst-libs/gst/audio/gstaudioclock.c:
2171 audioclock: make sure values are ever increasing
2173 2009-05-05 17:17:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2175 * gst/playback/gstplaysink.c:
2176 playbin2: make fallback identity silent
2177 Set the signal-handoffs to FALSE and silent to TRUE for the fallback identity
2178 element so that it consumes less CPU.
2180 2009-04-17 10:57:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2182 * gst/playback/gstplaybin2.c:
2183 * gst/playback/gstplaysink.c:
2184 playbin2: handle custom audiosinks differently
2185 Keep track of the autoplugged custom sinks and configure them in the playsink
2186 element when we have collected all streams.
2187 Also make sure that we only select one custom sink.
2188 When unreffing the internal sink, we don't need to change the state to NULL.
2190 2009-05-12 10:36:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2192 * gst/playback/gstplaybin2.c:
2193 * gst/playback/gstplaysink.c:
2194 * gst/playback/gstplaysink.h:
2195 playbin2: unify custom sink get/set functions
2196 Use one function to set/get all of the different sink types.
2197 cleanup up the subpicture chain too.
2198 Allow setting a custom subpicture sink.
2200 2009-05-11 18:29:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2202 * gst-libs/gst/interfaces/tunernorm.h:
2203 interfaces: Seperate some more struct definitions from typedefs
2205 2009-05-11 15:48:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2207 * gst-libs/gst/interfaces/navigation.h:
2208 * gst-libs/gst/interfaces/videoorientation.h:
2209 * gst-libs/gst/interfaces/xoverlay.h:
2210 interfaces: Seperate some more struct definitions from typedefs
2212 2009-05-10 17:28:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2214 * win32/common/libgstinterfaces.def:
2215 Add new functions to win32 exports
2217 2009-05-10 17:28:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2219 * docs/libs/gst-plugins-base-libs-sections.txt:
2220 Add new functions to the docs
2222 2009-05-10 17:25:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2224 * gst-libs/gst/interfaces/mixer.c:
2225 * gst-libs/gst/interfaces/mixer.h:
2226 interfaces: API: Add gst_mixer_get_mixer_type()
2227 This is a convenience function that returns the mixer_type
2228 of the interface struct.
2230 2009-05-10 17:25:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2232 * gst-libs/gst/interfaces/colorbalance.c:
2233 interfaces: Add docs for gst_color_balance_get_balance_type()
2235 2009-05-10 11:17:19 +0200 Marc-Andre Lureau <marcandre.lureau@gmail.com>
2238 Run libtoolize before aclocal
2239 This unbreaks the build in some cases. Fixes bug #582021
2241 2009-05-07 17:38:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2243 * ext/pango/gsttextrender.c:
2244 textrender: Correctly initialize the background for ARGB too
2246 2009-05-07 16:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2248 * ext/pango/gsttextrender.c:
2249 * ext/pango/gsttextrender.h:
2250 textrender: Use libgstvideo functions to create caps
2251 Also check if downstream wants ARGB always when we get
2254 2009-05-07 16:52:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2256 * ext/pango/gsttextrender.c:
2257 textrender: Don't always use ARGB if downstream supports it but take it's preference
2259 2009-05-07 16:48:08 +0200 Kapil Agrawal <kapil@mediamagictechnologies.com>
2261 * ext/pango/gsttextrender.c:
2262 * ext/pango/gsttextrender.h:
2263 textrender: Add support for ARGB and alignment properties
2266 2009-05-07 16:42:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2268 * ext/pango/gsttextrender.c:
2269 textrender: Add ; after GST_BOILERPLATE to fix indention
2271 2009-05-07 15:10:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2273 * gst-libs/gst/tag/gstvorbistag.c:
2274 vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
2276 2009-05-07 14:59:36 +0200 Arnout Vandecappelle <arnout@mind.be>
2278 * gst/typefind/gsttypefindfunctions.c:
2279 typefindfunctions: made mp3_type_find less aggressive
2280 mp3_type_find could suggest already when only a single valid header
2281 was found, if it ran out of data before the end of the next frame.
2282 Therefore, ignore the last found frame if it was incomplete.
2285 2009-05-07 14:48:29 +0200 John Millikin <jmillikin@gmail.com>
2287 * gst-libs/gst/tag/gstvorbistag.c:
2288 vorbistag: Store cover art in vorbiscomments
2291 2009-05-07 06:14:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2293 * gst-libs/gst/interfaces/colorbalance.c:
2294 * gst-libs/gst/interfaces/colorbalance.h:
2295 interfaces: API: Add gst_color_balance_get_balance_type()
2296 This is a convenience function that returns the balance_type
2297 of the interface struct.
2299 2009-05-06 17:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2301 * gst-libs/gst/interfaces/colorbalance.h:
2302 * gst-libs/gst/interfaces/colorbalancechannel.h:
2303 * gst-libs/gst/interfaces/tuner.h:
2304 * gst-libs/gst/interfaces/tunerchannel.h:
2305 interfaces: Separate struct definitions from typedefs
2307 2009-05-06 14:03:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2309 * pkgconfig/gstreamer-app-uninstalled.pc.in:
2310 Fix libdir for uninstalled gstreamer-app library
2312 2009-05-12 01:59:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2314 * gst-libs/gst/pbutils/descriptions.c:
2315 pbutils: add description for APE tag caps
2317 2009-05-12 01:35:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2320 configure: bump core requirement to last release
2321 as that's more likely to be true than that we need
2324 2009-05-12 01:21:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2328 configure: rename CVS -> git in a couple of places
2330 2009-05-12 01:17:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2333 configure: bump GLib requirement to GLib >= 2.16
2334 as per the New Regime (see wiki).
2336 2009-05-01 00:09:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2338 * gst-libs/gst/tag/gsttagdemux.c:
2339 tagdemux: cache events from upstream and re-send them once we have a source pad
2340 Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
2343 2009-05-07 14:07:44 -0700 Michael Smith <msmith@songbirdnest.com>
2345 * gst-libs/gst/riff/riff-media.c:
2346 riff: support UYVY raw 4:2:2 in riff.
2348 2009-05-11 21:20:07 +0100 Jan Schmidt <thaytan@noraisin.net>
2351 Back to development -> 0.10.23.1
2353 2009-04-27 22:42:55 -0700 Michael Smith <msmith@syncword.(none)>
2355 * ext/theora/theoradec.c:
2356 theoradec: fix buffer overrun on 422 decode.
2358 2009-04-27 21:39:01 -0700 Michael Smith <msmith@syncword.(none)>
2360 * ext/theora/theoradec.c:
2361 theoradec: 444 support.
2363 2009-04-27 21:30:04 -0700 Michael Smith <msmith@syncword.(none)>
2365 * ext/theora/theoradec.c:
2366 theoradec: handle 422 images (as YUY2).
2368 2009-04-27 21:01:51 -0700 Michael Smith <msmith@syncword.(none)>
2370 * ext/theora/gsttheoradec.h:
2371 * ext/theora/theoradec.c:
2372 theoradec: rearrange code in preparation for 422 and 444 support.
2374 === release 0.10.23 ===
2376 2009-05-10 23:57:01 +0100 Jan Schmidt <thaytan@noraisin.net>
2382 * docs/plugins/gst-plugins-base-plugins.args:
2383 * docs/plugins/gst-plugins-base-plugins.hierarchy:
2384 * docs/plugins/gst-plugins-base-plugins.interfaces:
2385 * docs/plugins/gst-plugins-base-plugins.prerequisites:
2386 * docs/plugins/gst-plugins-base-plugins.signals:
2387 * docs/plugins/inspect/plugin-adder.xml:
2388 * docs/plugins/inspect/plugin-alsa.xml:
2389 * docs/plugins/inspect/plugin-app.xml:
2390 * docs/plugins/inspect/plugin-audioconvert.xml:
2391 * docs/plugins/inspect/plugin-audiorate.xml:
2392 * docs/plugins/inspect/plugin-audioresample.xml:
2393 * docs/plugins/inspect/plugin-audiotestsrc.xml:
2394 * docs/plugins/inspect/plugin-cdparanoia.xml:
2395 * docs/plugins/inspect/plugin-decodebin.xml:
2396 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
2397 * docs/plugins/inspect/plugin-gdp.xml:
2398 * docs/plugins/inspect/plugin-gio.xml:
2399 * docs/plugins/inspect/plugin-gnomevfs.xml:
2400 * docs/plugins/inspect/plugin-libvisual.xml:
2401 * docs/plugins/inspect/plugin-ogg.xml:
2402 * docs/plugins/inspect/plugin-pango.xml:
2403 * docs/plugins/inspect/plugin-playback.xml:
2404 * docs/plugins/inspect/plugin-queue2.xml:
2405 * docs/plugins/inspect/plugin-subparse.xml:
2406 * docs/plugins/inspect/plugin-tcp.xml:
2407 * docs/plugins/inspect/plugin-theora.xml:
2408 * docs/plugins/inspect/plugin-typefindfunctions.xml:
2409 * docs/plugins/inspect/plugin-uridecodebin.xml:
2410 * docs/plugins/inspect/plugin-video4linux.xml:
2411 * docs/plugins/inspect/plugin-videorate.xml:
2412 * docs/plugins/inspect/plugin-videoscale.xml:
2413 * docs/plugins/inspect/plugin-videotestsrc.xml:
2414 * docs/plugins/inspect/plugin-volume.xml:
2415 * docs/plugins/inspect/plugin-vorbis.xml:
2416 * docs/plugins/inspect/plugin-ximagesink.xml:
2417 * docs/plugins/inspect/plugin-xvimagesink.xml:
2418 * gst-plugins-base.doap:
2419 * win32/common/_stdint.h:
2420 * win32/common/config.h:
2423 2009-05-10 23:56:05 +0100 Jan Schmidt <thaytan@noraisin.net>
2456 2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
2488 * win32/common/_stdint.h:
2489 * win32/common/config.h:
2490 0.10.22.6 pre-release
2492 2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2494 * gst/playback/gstplaysink.c:
2495 playbin2: fix resume after pause
2496 Don't ignore the state change of the children, they might be doing an ASYNC
2499 2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
2532 0.10.22.5 pre-release
2534 2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2536 * gst/tcp/gstmultifdsink.c:
2537 * gst/tcp/gsttcp-marshal.list:
2538 multifdsink: fix signature of the add-full signal
2539 The second parameter is a GstSyncMethod enum, not a boolean.
2541 2009-05-07 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2543 * gst/playback/gstplaysink.c:
2544 playsink: initialize variable too
2546 2009-05-07 14:28:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2548 * gst/playback/gstplaysink.c:
2549 playbin2: make playsink go ASYNC to PAUSED
2550 Make playsink go async to the PAUSED state instead of relying on uridecodebin
2551 for async behaviour in playbin. This solves some problems (mainly with DVD)
2552 where the pipeline would go to PLAYING before preroll completed, failing to
2553 select the audiosink clock.
2556 2009-05-06 16:09:52 +0100 Jan Schmidt <thaytan@noraisin.net>
2588 * win32/common/_stdint.h:
2589 * win32/common/config.h:
2590 0.10.22.4 pre-release
2592 2009-05-06 13:19:34 +0100 Zaheer Merali <zaheerabbas@merali.org>
2594 * ext/theora/theoraenc.c:
2595 * ext/vorbis/vorbisenc.c:
2596 vorbisenc, theoraenc: Ensure gp is computed consistently + clip to segment
2597 With vorbisenc, compute the granulepos with running time and clip incoming
2599 With theoraenc, drop out of segment buffers.
2601 2009-05-01 16:47:53 +0100 Jan Schmidt <thaytan@noraisin.net>
2603 * gst/audioresample/gstaudioresample.c:
2604 audioresample: Fix buffer size transformations
2605 When calculating the input/output buffer sizes in the transform_size function,
2606 take the number of channels into account, so we don't end up calculating
2607 a buffer size that only contains a partial number of audio frames.
2608 Also, when going from output size to input size, round down rather than
2609 up, so as to calculate the minimum number of samples that *might* yield
2610 a buffer of the intended destination size.
2611 Fixes: #580470 and #580952
2613 2009-04-29 16:45:27 +0100 Jan Schmidt <thaytan@noraisin.net>
2615 * ext/vorbis/gstvorbisenc.h:
2616 * ext/vorbis/vorbisenc.c:
2617 vorbisenc: Ensure output buffers fall within the segment
2618 Add the start position of the first segment to the running time
2619 used to generate buffer timestamps in vorbisenc. This avoids generating
2620 buffers which fall outside the initial segment. The element segment
2621 handling requires more extensive fixing, but this at least prevents
2622 regressions. Fixes: #580020
2624 2009-04-29 11:18:42 +0200 Andy Wingo <wingo@oblong.net>
2626 * gst-libs/gst/audio/gstbaseaudiosink.c:
2627 Revert "add can-activate-pull property to baseaudiosink"
2628 This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.
2630 2009-04-29 11:18:33 +0200 Andy Wingo <wingo@oblong.net>
2632 * gst-libs/gst/audio/gstbaseaudiosink.c:
2633 Revert "[baseaudiosink] add docs for can-activate-pull"
2634 This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.
2636 2009-04-28 18:48:33 +0200 Andy Wingo <wingo@oblong.net>
2638 [baseaudiosink] add docs for can-activate-pull
2639 * gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
2642 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
2644 add can-activate-pull property to baseaudiosink
2645 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
2648 2009-04-28 11:32:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2650 * gst/videorate/gstvideorate.c:
2651 * gst/videorate/gstvideorate.h:
2652 videorate: clear discont on duplicated buffers
2653 When videorate duplicates a buffer with a DISCONT flag, it copies the discont on
2654 the first pushed buffer but fails to clear it for subsequent buffers. This
2655 causes theoraenc!oggmux and possibly other elements to consider this a discont
2657 Fix videorate to produce discont as the first buffer and after a flushing seek.
2660 2009-04-24 18:13:00 +0100 Jan Schmidt <thaytan@noraisin.net>
2662 * tests/check/Makefile.am:
2663 check: Disable the playbin2 for this release, as it is a bit racy.
2664 Disable the test, as per the discussion in #580120. Needs re-enabling
2665 after the release, when playbin2 is fixed.
2667 2009-04-23 08:41:19 +0200 Edward Hervey <bilboed@bilboed.com>
2669 * gst/playback/gstdecodebin2.c:
2670 decodebin2: Don't reduce max-size-time of exposed groups. Fixes #579912
2671 The 2s limit is way too small for a lot of files (which have an interleave
2672 in time of between 3 and 5s). Instead, leave it to the initial 5s value
2673 and reduce the other limits (allowing us to stay memory-efficient).
2675 2009-04-21 21:06:59 +0100 Jan Schmidt <thaytan@noraisin.net>
2707 * win32/common/_stdint.h:
2708 * win32/common/config.h:
2709 0.10.22.3 pre-release
2711 2009-04-21 20:41:23 +0100 René Stadler <mail@renestadler.de>
2713 * gst/audioresample/gstaudioresample.c:
2714 audioresample: Fix unused variable in compilation with --disable-gst-debug
2717 2009-04-21 22:12:28 +0100 Jan Schmidt <thaytan@noraisin.net>
2720 Automatic update of common submodule
2721 From b3941ea to 6ab11d1
2723 2009-04-21 20:57:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2725 * gst/playback/gstplaybasebin.c:
2726 playbin: only use raw_decoding_mode when it's true
2727 First check the pad caps if they are raw before setting the raw_decoding_mode to
2728 TRUE. Fixes playback of transport streams and other streams that require large
2732 2009-04-19 18:15:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2734 * gst-libs/gst/cdda/gstcddabasesrc.c:
2735 * tests/check/libs/cddabasesrc.c:
2736 cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core
2737 Don't use REPLACE_ALL merge mode when that's not really what we want,
2738 as now that REPLACE_ALL actually does what it's supposed to do in
2739 core, we drop tags we wanted to keep, such as the various disc id
2740 tags. Add unit test for this as well. Fixes #579463.
2742 2009-04-17 10:34:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2744 * gst-libs/gst/rtsp/gstrtspconnection.c:
2745 rtspconnection: don't use GLib-2.16 API, we require only 2.14
2748 2009-04-17 10:55:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2750 * gst-libs/gst/audio/gstbaseaudiosink.c:
2751 baseaudiosink: don't unparent the ringbuffer
2752 when going to NULL, don't unparent the ringbuffer because we don't support going
2753 back to 0 very well yet.
2756 2009-04-17 10:53:10 +0200 Olivier Crete <tester at tester.ca>
2758 * gst-libs/gst/rtp/gstrtcpbuffer.c:
2759 RTCP: don't fail when retrieving invalid PT
2760 We can't meaningfully assert on valid packet types so just return the type as it
2761 is. Update the comments to reflect this.
2764 2009-04-16 12:12:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2766 * docs/libs/gst-plugins-base-libs-sections.txt:
2767 * gst-libs/gst/app/gstappsink.h:
2768 * gst-libs/gst/app/gstappsrc.h:
2769 app: add trivial cast macros
2770 Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
2771 and add the macros to the standard macros in the docs.
2774 2009-04-16 12:09:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2776 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
2777 pkgconfig: add the app/ directory to Libs
2778 Add the appsrc/appsink directory to the Libs in the uninstalled
2779 pkgconfig file so that one can build against it.
2782 2009-04-15 22:59:31 +0100 Jan Schmidt <thaytan@noraisin.net>
2785 0.10.22.2 pre-release
2787 2009-04-15 22:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
2790 ChangeLog: regenerate changelog with the gen-changelog script
2792 2009-04-16 00:41:13 +0100 Jan Schmidt <thaytan@noraisin.net>
2823 po: Update po files from TP
2825 2009-04-16 00:40:59 +0100 Jan Schmidt <thaytan@noraisin.net>
2827 * win32/common/_stdint.h:
2828 * win32/common/config.h:
2829 * win32/common/gstrtsp-enumtypes.c:
2830 * win32/common/interfaces-enumtypes.c:
2831 * win32/common/interfaces-enumtypes.h:
2832 * win32/common/video-enumtypes.c:
2833 win32: Update win32 build files
2835 2009-04-16 00:31:55 +0100 Jan Schmidt <thaytan@noraisin.net>
2837 * tests/check/libs/video.c:
2838 check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.
2840 2009-04-16 00:31:00 +0100 Jan Schmidt <thaytan@noraisin.net>
2842 * tests/check/elements/playbin2.c:
2843 check: Fix the input uri in playbin2 test.
2844 Don't try and use a random file in wim's home directory as a test input
2846 2009-04-15 15:35:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2848 * gst-libs/gst/video/video.h:
2849 video: Fix typo in the docs
2851 2009-04-15 14:53:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2853 * gst-libs/gst/video/video.c:
2854 * gst-libs/gst/video/video.h:
2855 video: Add support for YVYU YUV colorspace
2857 2009-04-15 00:17:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2859 * docs/libs/gst-plugins-base-libs-docs.sgml:
2860 * gst-libs/gst/fft/gstfft.c:
2861 docs: fix hyperlink and move fft attribution to the right place
2863 2009-04-15 00:02:39 +0300 Stefan Kost <ensonic@users.sf.net>
2865 * gst-libs/gst/audio/gstbaseaudiosink.c:
2866 log: use G_GUINT64_FORMAT instead of llu
2868 2009-04-14 18:31:52 +0200 Josep Torra <n770galaxy at gmail.com>
2870 * gst-libs/gst/rtsp/gstrtspdefs.c:
2871 * gst-libs/gst/rtsp/gstrtspdefs.h:
2872 RTSP: add missing headers for WMS RTSP
2873 Add missing headers related to Windows Media RTSP extension.
2876 2009-04-14 18:16:37 +0200 Olivier Crete <tester at tester.ca>
2878 * docs/design/draft-keyframe-force.txt:
2879 * ext/theora/gsttheoraenc.h:
2880 * ext/theora/theoraenc.c:
2881 theoraenc: implement upstream keyframe force
2882 Implement handling of upstream keyframe forcing.
2883 Update the design documents too.
2886 2009-04-14 17:31:31 +0200 Olivier Crete <tester at tester.ca>
2888 * ext/theora/theoraenc.c:
2889 theoraenc: factor out keyframe forcing
2892 2009-04-14 17:01:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2895 * gst-libs/gst/fft/gstfft.c:
2896 Give credit to Mark Borgerding (kissfft author)
2897 and add myself to AUTHORS as well. Fixes #575638.
2899 2009-04-14 17:04:06 +0200 Jan Urbanski <j.urbanski at students.mimuw.edu.pl>
2901 * gst/tcp/gstmultifdsink.c:
2902 * gst/tcp/gstmultifdsink.h:
2903 multifdsink: add property to resend streamheaders
2904 Adds a new property in multifdsink, resend-streamheader.
2905 If this property is false, the multifdsink will not send the streamheader if
2906 there's already one set for a particular client.
2907 There are some formats in which every stream needs to start with a certain
2908 blob, but you can't inject this blob at leisure. If the producer wants to
2909 change the blob in question and sets in as the streamheader on the outgoing
2910 buffers' caps, new clients of multifdsink will get the new streamheader, but
2911 old clients will break, because they'll see the blob in the middle of the
2913 The property is true by default, so existing code will not see any difference.
2916 2009-04-14 16:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2918 * gst/tcp/gstmultifdsink.c:
2919 * gst/tcp/gstmultifdsink.h:
2920 multifdsink: add property to handle client write
2921 Add a property to disable listening to client writes. This property is usefull
2922 when other code will deal with reading from the client socket.
2923 API: GstMultiFdSink::handle-read property
2925 2009-04-14 16:45:20 +0200 Johann Prieur <johann.prieur at gmail.com>
2927 * docs/libs/gst-plugins-base-libs-sections.txt:
2928 * gst-libs/gst/rtp/gstrtcpbuffer.c:
2929 * gst-libs/gst/rtp/gstrtcpbuffer.h:
2930 * win32/common/libgstrtp.def:
2931 RTCP: add beginnings of Feedback messages
2932 Add the beginnings of parsing and constructing Feedback messages.
2935 2009-04-14 13:51:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2937 * gst/playback/gstplaysink.c:
2938 playbin2: clear the target
2939 Clear the target of our ghostpads before we remove the pad from the element.
2940 This to make sure that the internal pad is not left linked to whatever pad we
2941 were ghosted to. This should only be a problem when we leak the ghostpads.
2942 Also release our subpicture pads.
2945 2009-04-14 12:10:30 +0100 Hannes Bistry <hannesb@gmx.net>
2947 * sys/ximage/ximagesink.c:
2948 ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
2951 2009-04-14 13:16:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2953 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2954 baseaudiosrc: adjust the internal timestamp
2955 Adjust the internal timestamp before comparing it against the adjusted clock
2959 2009-04-14 13:12:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2961 * gst-libs/gst/audio/gstbaseaudiosink.c:
2962 baseaudiosink: use new clock time methods
2963 Use the unadjusted internal clock times to calculate the internal/external
2964 offset when calibrating the clock.
2965 When going to NULL, unparent and free the ringbuffer, like we do in the source
2969 2009-04-14 13:08:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2971 * gst-libs/gst/audio/gstaudioclock.c:
2972 * gst-libs/gst/audio/gstaudioclock.h:
2973 * win32/common/libgstaudio.def:
2974 audioclock: add methods for the internal offset
2975 Add two methods for getting the unadjusted time of the clock and one for
2976 adjusting an internal time. We will need these methods for correctly handling
2977 the time after a gst_audio_clock_reset().
2978 Add a debug category and some debug lines to the audio clock.
2979 API: gst_audio_clock_get_time()
2980 API: gst_audio_clock_adjust()
2981 API: GST_AUDIO_CLOCK_CAST()
2983 2009-04-14 11:34:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2985 * gst/playback/gstdecodebin2.c:
2986 decodebin2: fix up the debugs and warnings
2987 Use _OBJECT variants because we can. Go over some log statements and put them in
2991 2009-04-12 22:26:33 +0200 Luca Ognibene <luca.ognibene at gmail.com>
2993 * gst/tcp/gstmultifdsink.c:
2994 multifdsink: fix error in sync-method
2995 Multifdsink did not handle sync-method=latest-keyframe correctly when the
2996 soft-limit is set to -1 (unlimited).
2999 2009-04-10 21:49:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3001 * gst-libs/gst/audio/gstbaseaudiosink.c:
3002 baseaudiosink: use the internal clock time
3003 We can't assume that the internal clock time is the same as the function we
3004 installed on our provided clock because somebody might have changed it.
3006 2009-04-10 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3008 * tests/examples/seek/seek.c:
3009 seek: handle clock-lost messages
3010 When we receive a clock-lost message we need to pause and play to select a new
3013 2009-04-10 13:44:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3015 * tests/check/Makefile.am:
3016 * tests/check/elements/playbin2.c:
3017 check: add a unit test for playbin2
3018 Add unit test for playbin2 and include the refcount test in #577794.
3020 2009-04-10 13:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3022 * gst/playback/gstplaysink.c:
3023 playbin2: fix refcounting of visualisations
3026 2009-04-10 13:27:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3028 * gst/playback/gstplaysink.c:
3029 playsink: fix refcounting of custom elements
3030 Sink the custom sinks, let other elements we create be sunken by the bin we add
3034 2009-04-10 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3036 * tests/check/elements/appsink.c:
3037 check: fix appsink test
3038 Fix the appsink test now that the method signature changed.
3040 2009-04-10 12:26:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3042 * gst/playback/gstplaybin2.c:
3043 playbin2: handle missing input-selector
3044 Gracefully degrade and disable stream selection when input-selector is
3047 2009-04-09 23:46:17 +0200 Martin Samuelsson <martin.samuelsson at axis.com>
3049 * gst-libs/gst/app/gstappsink.c:
3050 * gst-libs/gst/app/gstappsink.h:
3051 appsink: make callbacks return GstFlowReturn
3052 Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
3053 errors can be reported properly.
3056 2009-04-09 18:04:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3058 * gst-libs/gst/audio/gstringbuffer.c:
3059 * gst-libs/gst/audio/gstringbuffer.h:
3060 ringbuffer: allow for custom commit functions
3061 Allow subclasses to override the commit method.
3063 2009-04-08 18:04:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3065 * gst-libs/gst/audio/gstbaseaudiosink.c:
3066 baseaudiosink: fix a small glitch after pause
3067 After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
3068 the amount of output samples we consumed. We can't do this reliably with the
3069 current API when we are doing trick modes but we can do the right thing for
3072 2009-04-08 16:43:27 +0300 Stefan Kost <ensonic@users.sf.net>
3074 * gst/playback/gstplaysink.c:
3075 playbin2: better error message on sink failure
3076 If we could create the sinks, but the don't work, don't send the missing plugin
3077 message and report that the state-changed failed.
3079 2009-04-07 22:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
3081 * gst-libs/gst/audio/gstaudiofilter.c:
3082 audiofilter: don't leak pad-template
3083 gst_element_class_add_pad_template() does not take ownership.
3085 2009-04-04 21:18:38 +0300 Felipe Contreras <felipe.contreras@gmail.com>
3088 Automatic update of common submodule
3089 From d0ea89e to b3941ea
3091 2009-04-04 16:28:14 +0200 Edward Hervey <bilboed@bilboed.com>
3093 * gst-libs/gst/interfaces/navigation.c:
3094 * sys/v4l/v4lsrc_calls.c:
3095 navigation/v4l: Don't use g_return_val_if_fail for computed/used values.
3097 2009-03-22 09:46:37 +0100 Edward Hervey <bilboed@bilboed.com>
3099 * ext/theora/theoradec.c:
3100 theoradec: return GST_CLOCK_TIME_NONE for negative framecounts.
3101 This fixes most seeking issues when used with gnonlin.
3104 2009-04-04 14:53:42 +0200 Edward Hervey <bilboed@bilboed.com>
3107 Automatic update of common submodule
3108 From f8b3d91 to d0ea89e
3110 2009-04-03 10:51:42 -0700 Michael Smith <msmith@songbirdnest.com>
3112 * gst/playback/gstplaybin2.c:
3113 playbin2: don't leak selector when getting current stream numbers.
3115 2009-04-02 22:28:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3117 * gst-libs/gst/rtsp/gstrtspconnection.c:
3118 rtsp: use fully qualified urls when using a proxy
3119 Use a fully qualified url when specifying the url for tunneled requests through
3123 2009-03-31 00:54:30 +0100 Jan Schmidt <thaytan@noraisin.net>
3125 * docs/libs/gst-plugins-base-libs-sections.txt:
3126 * gst-libs/gst/interfaces/navigation.c:
3127 * gst-libs/gst/interfaces/navigation.h:
3128 * tests/check/Makefile.am:
3129 * tests/check/libs/.gitignore:
3130 * tests/check/libs/navigation.c:
3131 * win32/common/libgstinterfaces.def:
3132 navigation: Extend the navigation interface
3133 Add support for a set of standard commands that can be queried and executed to
3134 support applications like DVD. Add query construction and parsing functions.
3135 Add new messages that can be sent on the bus to provide notifications related
3136 to commands, multiangle changes, and button highlight activity.
3137 Add some helper functions to parse the existing GstNavigation events that
3138 elements might receive.
3139 Document it all and add unit tests.
3141 2009-02-04 17:03:07 +0000 Jan Schmidt <thaytan@noraisin.net>
3143 * gst/playback/gstplaybasebin.c:
3144 * gst/playback/gstplaybasebin.h:
3145 playbin: Add simple 'raw decoding mode'.
3146 Raw decoding mode removes almost all buffering in video and audio queues
3147 when a source providing already decoded video/audio is detected, on the
3148 possibly bogus assumption that such a source should provide sufficient
3149 internal queueing. Fixes playback on some DVDs, and improves it
3152 2009-04-02 09:27:07 +0100 Jan Schmidt <thaytan@noraisin.net>
3154 * tests/check/elements/.gitignore:
3155 ignores: Ignore the videoscale check binary
3157 2009-04-02 12:13:57 +0100 Jan Schmidt <thaytan@noraisin.net>
3159 * win32/common/libgstrtsp.def:
3160 win32: Add gst_rtsp_connection_set_proxy to the win32 exports
3162 2009-04-02 10:42:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3164 * ext/alsa/gstalsamixer.c:
3165 alsamixer: don't forget to release locks in a few places
3168 2009-04-02 11:10:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3170 * gst/videoscale/vs_4tap.c:
3171 videoscale: Don't read over line ends when taking the last Cr or Cb
3173 2009-04-02 10:52:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3175 * gst/videoscale/vs_4tap.c:
3176 videoscale: Don't write to few pixels and don't mix Cr and Cb
3179 2009-04-01 15:15:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3181 * gst/audioresample/gstaudioresample.c:
3182 * tests/check/elements/audioresample.c:
3183 audioresample: fix negotiation so that upstream can actually fixate to downstream's rate
3184 If one side has a preference for a particular sample rate or set of sample rates, we
3185 should honour this in the caps we advertise and transform to and from, so that elements
3186 actually know about the other side's sample rate preference and can negotiate to it
3187 if supported. Also add unit test for this.
3189 2009-03-26 19:34:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3191 * gst/playback/gstplaybin2.c:
3192 docs: add a blurb about redirect messages to playbin2 docs
3194 2009-04-01 09:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3196 * gst-libs/gst/rtsp/gstrtspconnection.c:
3197 rtsp: fix little typo in the comments
3199 2009-03-31 17:52:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3201 * gst-libs/gst/rtsp/gstrtspconnection.c:
3202 rtspconnection: make gst_rtsp_watch_queue_message() thread-safe
3203 People might queue messages from a thread other than the thread in which
3204 the main context which this watch is attached is iterated from, so use
3205 a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
3206 over list nodes just freed in the other thread. This just fixes issues
3207 I've had with gst-rtsp-server. We might need more locking in various
3210 2009-03-31 18:13:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3212 * gst-libs/gst/rtsp/gstrtspconnection.c:
3213 * gst-libs/gst/rtsp/gstrtspmessage.c:
3214 rtsp: clear the entire builder structure
3215 And use structure instead of variable with sizeof when
3216 clearing the rtsp message structure, for clarity.
3218 2009-03-31 17:56:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3220 * gst-libs/gst/rtsp/gstrtspmessage.c:
3221 docs: fix typo in gst_rtsp_message_unset() API docs
3223 2009-03-31 19:00:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3225 * gst-libs/gst/rtsp/gstrtspconnection.c:
3226 * gst-libs/gst/rtsp/gstrtspconnection.h:
3227 rtsp: add support for proxies
3228 Add suport for proxy servers. Currently only used for tunneled HTTP
3229 connections without authentication.
3231 2009-03-31 18:57:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3233 * gst-libs/gst/rtsp/gstrtspmessage.c:
3234 Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
3235 This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.
3237 2009-03-26 18:54:56 +0200 Stefan Kost <ensonic@users.sf.net>
3239 * sys/xvimage/xvimagesink.c:
3240 xvimagesink: use xcontext->depth instead of bits in attr.max_value for colorkey
3241 According to the drivers in http://cgit.freedesktop.org/xorg/driver/ we should
3242 format the colorkey depending on xcontext->depth. This is what they will use to
3243 interprete the value. The max_value in turn is usualy a constant regardless of
3246 2009-03-31 12:22:14 +0300 Stefan Kost <ensonic@users.sf.net>
3248 * gst-libs/gst/rtsp/gstrtspmessage.c:
3249 rtsp: reset whole message (was sizeof pointer instead of sizeof type)
3251 2009-03-31 00:56:18 +0100 Jan Schmidt <thaytan@noraisin.net>
3253 * gst-libs/gst/interfaces/mixer.c:
3254 doc: Fix a typo in the GstMixer docs
3256 2009-03-29 12:01:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3258 * gst/videoscale/vs_scanline.c:
3259 videoscale: Fix linear scaling for one byte components
3262 2009-03-29 11:53:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3264 * gst/videoscale/vs_4tap.c:
3265 videoscale: Fix 4tap scaling of YUYV and friends
3267 2009-03-28 16:08:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3269 * gst/videoscale/vs_image.c:
3270 * gst/videoscale/vs_scanline.c:
3271 * gst/videoscale/vs_scanline.h:
3272 videoscale: Rewrite YUYV (and friends) scaling and don't read/write over line ends
3273 Partially fixes bug #577054, there's just one issue left now.
3275 2009-03-28 12:48:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3277 * tests/check/elements/videoscale.c:
3278 videoscale: Add some more unit tests
3280 2009-03-28 11:51:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3282 * gst/videoscale/gstvideoscale.c:
3283 videoscale: Use bilinear instead of 4tap scaling for heights < 4
3284 Partially fixes bug #577054.
3286 2009-03-28 11:45:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3288 * gst/videoscale/vs_scanline.c:
3289 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY/RGB/RGBA
3290 This case is for upscaling a frame with width=1
3291 Partially fixes bug #577054.
3293 2009-03-28 11:27:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3295 * gst/videoscale/vs_scanline.c:
3296 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
3297 Partially fixes bug #577054.
3299 2009-03-28 10:40:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3301 * gst/videotestsrc/gstvideotestsrc.c:
3302 videotestsrc: Initialize buffer memory with zeroes
3303 This prevents valgrind warnings when accessing the "x" parts
3304 of xRGB and friends in other elements that handle (and can handle)
3305 xRGB like ARGB (for example videoscale).
3307 2009-03-28 10:25:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3309 * tests/check/Makefile.am:
3310 * tests/check/elements/videoscale.c:
3311 videoscale: Add a lot of unit tests
3313 2009-03-28 10:06:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3315 * gst/videoscale/gstvideoscale.c:
3316 videocale: Add support for video/x-raw-gray with bpp=depth=8
3318 2009-03-28 10:01:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3320 * gst/videotestsrc/videotestsrc.c:
3321 videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8
3323 2009-03-28 09:43:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3325 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
3326 ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format
3328 2009-03-27 19:12:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3330 * gst/videoscale/vs_4tap.c:
3331 videoscale: Take the next luma value instead of every second next when scaling UYVY and friends
3333 2009-03-27 19:09:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3335 * gst/videoscale/gstvideoscale.c:
3336 videoscale: Add support for v308 YUV colorspace
3338 2009-03-27 13:15:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3340 * gst/videoscale/vs_4tap.c:
3341 videoscale: Add my copyright to the 4tap scalers
3343 2009-03-27 13:14:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3345 * gst/videoscale/gstvideoscale.c:
3346 videoscale: Enable 4-tap scaling for all supported formats
3348 2009-03-27 13:14:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3350 * gst/videoscale/vs_4tap.c:
3351 * gst/videoscale/vs_4tap.h:
3352 videoscale: Implement 4-tap scaling for RGB565 and RGB555
3354 2009-03-27 10:47:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3356 * gst/videoscale/vs_4tap.c:
3357 * gst/videoscale/vs_4tap.h:
3358 videoscale: Implement 4-tap scaling for UYVY
3360 2009-03-27 09:33:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3362 * gst/videoscale/vs_4tap.c:
3363 * gst/videoscale/vs_4tap.h:
3364 videoscale: Implement 4-tap scaling for YUY2 and YVYU
3366 2009-03-26 22:14:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3368 * gst/videoscale/vs_4tap.c:
3369 * gst/videoscale/vs_4tap.h:
3370 videoscale: Implement 4-tap scaling for RGB and BGR
3372 2009-03-26 22:08:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3374 * gst/videoscale/vs_4tap.c:
3375 * gst/videoscale/vs_4tap.h:
3376 videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats
3378 2009-03-26 11:02:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3380 * ext/pango/gsttextoverlay.c:
3381 textoverlay: Fix drawing of UYVY text borders
3383 2009-03-26 10:36:27 +0100 Zeeshan Ali <zeeshan.ali@nokia.com>
3385 * ext/pango/gsttextoverlay.c:
3386 * ext/pango/gsttextoverlay.h:
3387 textoverlay: Add support for UYVY colorspace
3390 2009-03-25 19:01:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3392 * gst/playback/gstdecodebin2.c:
3393 decodebin2: do some more cleanup
3394 Free the groups when we go to READY.
3395 Allow for NO_PREROLL elements.
3397 2009-03-25 16:37:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3399 * gst-libs/gst/rtsp/gstrtspconnection.c:
3400 rtsp: start CSeq counting from 1 instead of 0
3401 Start counting from 1 instead of 0 as this is what most other clients
3404 2009-03-25 16:35:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3406 * gst-libs/gst/rtsp/gstrtspdefs.c:
3407 * gst-libs/gst/rtsp/gstrtspdefs.h:
3408 rtsp: add ETag and If-Match headers
3409 Add new headers, we need them for RealMedia support.
3411 2009-03-25 14:16:25 +0200 Stefan Kost <ensonic@users.sf.net>
3413 * sys/xvimage/xvimagesink.c:
3414 xvimagesink: scale the colorkey components in case of 16bit visuals
3415 Use a default that won't be scales to 0,0,0
3417 2009-03-25 11:27:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3419 * gst-libs/gst/audio/gstbaseaudiosrc.c:
3420 audiosrc: improve 'Dropped n samples' warning message
3422 2009-03-24 19:41:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3424 * tests/examples/app/appsrc-ra.c:
3425 * tests/examples/app/appsrc-seekable.c:
3426 examples: use new method to set flags
3427 Use the new core method for setting object enum properties by name.
3429 2009-03-24 18:29:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3431 * gst/playback/gstplaysink.c:
3432 * gst/playback/gstplaysink.h:
3433 playbin2: add more support for subpictures
3435 2009-03-24 17:12:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3437 * gst/playback/gstplaybin2.c:
3438 * gst/playback/gstplaysink.c:
3439 * gst/playback/gstplaysink.h:
3440 playbin2: first support for subpictures
3441 Add beginnings of subpicture support.
3443 2009-03-24 15:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3445 * tests/examples/seek/seek.c:
3446 seek: print tags from the different tracks
3448 2009-03-24 12:22:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3450 * gst/playback/gstplaybin2.c:
3451 playbin2: blacklist subpictures for now
3452 Blacklist the subpictures until we add support for them.
3453 Add some small debug info.
3456 2009-03-24 12:19:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3458 * gst/playback/gsturidecodebin.c:
3459 uridecodebin: expose more media types
3460 Expose more media types from a raw source, such as the subpicture and various
3462 Small cleanups and add some more debugging.
3465 2009-03-24 10:42:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3467 * gst/playback/gstplaysink.c:
3468 playbin2: rescan audio sinks for volume/mute
3469 Rescan the audio sinks for the mute and volume properties.
3472 2009-03-23 19:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3474 * gst/playback/gstplaysink.c:
3475 playbin2: fix reuse of the video chains
3476 When reusing playbin with visualisations, reset the async property on the video
3477 sink because some sinks might dynamically recreate their sinks.
3480 2009-03-23 17:37:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3482 * gst/playback/gstplaysink.c:
3483 playbin2: allow dynamic swtiching of subtitles
3484 When we have the textpad configured, enable and disable the subtitles by setting
3485 the silent flag on the overlay element instead of trying to remove elements.
3488 2009-03-23 16:59:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3490 * tests/icles/playbin-text.c:
3491 tests: print some more info in the text example
3492 Print both the position and the running_time when the subtitle becomes available
3495 2009-03-23 16:04:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3497 * gst/playback/gstplaysink.c:
3498 playbin2: fix dynamic switching of visualisations
3499 Fix the switching of visualisations by requesting and releasing the tee request
3503 2009-03-23 16:19:11 +0200 Stefan Kost <ensonic@users.sf.net>
3506 * gst/tcp/gsttcpclientsink.c:
3507 * gst/tcp/gsttcpclientsrc.c:
3508 * gst/tcp/gsttcpserversink.c:
3509 * gst/tcp/gsttcpserversrc.c:
3510 docs: add examples for tcp elements, also use correct section name. Fixes #564139
3511 Updated the examples in the README to actually work. Add them to api docs. Tests
3512 the api-docs and fix the section names to make the docs actualy show up.
3513 The example for "tcpserversrc" needs review (might be an element bug).
3515 2009-03-17 09:14:02 +0200 Stefan Kost <ensonic@users.sf.net>
3517 * gst/videoscale/gstvideoscale.c:
3518 indent: fix damange that gst-indent did some time ago
3520 2009-03-23 15:27:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3522 * gst/playback/gstplaysink.c:
3523 playbin2: fix linking order
3524 Link after doing the state change and unlink before shutting down. Makes the
3525 window for causing races in toggling the visualisations smaller.
3528 2009-03-23 12:26:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3530 * gst/playback/gsturidecodebin.c:
3531 uridecodebin: reset counter
3532 reset the number of pending dynamic operations back to 0 when we reuse
3536 2009-03-23 11:38:53 +0100 Edward Hervey <bilboed@bilboed.com>
3538 * ext/theora/theoradec.c:
3539 theoradec: Use GST_CLOCK_TIME_NONE for invalid positions. Fixes #543591
3540 The problem was that previously we didn't check whether _theora_granule_frame
3541 returned a negative framecount or not, resulting in bogus timestamps.
3543 2009-03-21 09:46:28 +0100 René Stadler <mail@renestadler.de>
3545 * ext/vorbis/vorbisenc.c:
3546 vorbisenc: Set caps on non-header ouput buffers.
3549 2009-03-20 16:13:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3551 * tests/examples/seek/seek.c:
3552 seek: Add some more debug
3553 Add some more info about the selected streams.
3555 2009-03-20 15:47:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3557 * gst/playback/gstdecodebin2.c:
3558 decodebin2: a pad starts out being not drained.
3559 Mark a new pad as not drained until we get EOS on it.
3561 2009-03-20 14:17:19 +0100 LRN <lrn1986 at gmail dot com>
3563 * gst/playback/gstqueue2.c:
3564 win32: fix seeking in large files
3565 Fix Seeking in large files by using the 64-bit seek functions.
3568 2009-03-19 20:31:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3570 * gst/playback/gstdecodebin2.c:
3571 decodebin2: recover from failing to add a pad
3572 When we cannot add a pad to the decodebin2 for some reason, print a warning but
3573 continue adding the remaining pads.
3575 2009-03-19 19:35:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3577 * gst/playback/gstdecodebin2.c:
3578 decodebin2: more cleanups and docs.
3579 Add some more comments and use g_list_prepend().
3581 2009-03-19 19:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3583 * gst/playback/gstdecodebin2.c:
3584 decodebin2: refactoring and race fixes
3585 Refactor some code so that we can take the right locks and in the right order.
3586 Fixes quite a bit of races already.
3588 2009-03-19 19:03:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3590 * gst/playback/gstplaybin2.c:
3591 playbin2: remove the group cond + cleanups
3592 Remove the group GCond that we used for waiting for groups to finish because we
3593 use pad blocking on the selectors and counters instead for waiting for the
3595 remove the obsolete about_to_finish variable set while emiting the
3596 about-to-finish signal and fix some old comments.
3597 We don't need to take the playbin lock when querying the uridecodebin.
3599 2009-03-18 10:45:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3601 * tests/icles/playbin-text.c:
3602 icles: print better error and warning messages
3605 2009-03-17 22:53:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3607 * gst-libs/gst/rtsp/gstrtspbase64.c:
3608 * gst-libs/gst/rtsp/gstrtspbase64.h:
3609 rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode
3610 This also fixes another instance of CVE-2008-4316.
3612 2009-03-17 19:53:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3614 * ext/ogg/gstoggdemux.c:
3615 oggdemux: report -1 for duration in push mode
3616 In push mode we must return TRUE from the duration query with a value of -1
3617 meaning that we know that we don't know the duration.
3619 2009-03-17 19:09:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3621 * gst/playback/gstdecodebin2.c:
3622 decodebin2: add extra dynamic ref for demuxers
3623 When we make a group connected to a demuxer, keep an extra dynamic refcount for
3624 the group which is only decremented when no_more_pads or a multiqueue overrun is
3625 detected. This way we avoid a race between exposing the group while more dynamic
3626 refs are added from new pads.
3629 2009-03-17 15:39:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3631 * gst/playback/gstplaysink.c:
3632 playbin2: sync state of the sink correctly
3633 Sync the state of the newly added chains to the state of the parent sink element
3634 to avoid lost async-start messages. Fixes cdda:// async-done message storm.
3636 2009-03-17 11:54:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3638 * gst/playback/gstplaybin2.c:
3639 playbin2: return NOT_LINKED for unselected streams
3640 When streams are not selected in the selector, return NOT_LINKED so that
3641 upstream elements can skip decoding. Only do this for audio and video pads
3642 because for text streams the overhead is smaller and they could come from
3645 2009-03-17 11:51:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3647 * gst/playback/gstplaysink.c:
3648 playbin: set custom text sink properties
3649 Set the custom sink async=FALSE to not make it participate in preroll because we
3650 are dealing with sparse streams.
3651 Try to set sync=TRUE on the custom text sink.
3653 2009-03-17 11:30:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3655 * tests/icles/playbin-text.c:
3656 example: use appsink instead of fakesink
3657 Use appsink instead of fakesink to get the subtitles.
3658 Make things more pretty.
3660 2009-03-17 11:24:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3662 * tests/icles/.gitignore:
3663 * tests/icles/Makefile.am:
3664 * tests/icles/playbin-text.c:
3665 examples: add example of intercepting subtitles
3666 Add an example of how to install a custom sink for receiving subtitles in
3669 2009-03-17 11:03:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3671 * tests/check/elements/appsink.c:
3672 tests: fix include in the appsink test
3673 Fix dist by doing the right include.
3675 2009-03-16 16:42:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3677 * gst/playback/gstplaybin2.c:
3678 playbin2: don't try to set invalid stream numbers
3679 Fix a problem with setting the stream numbers because we check for the wrong
3683 2009-03-16 16:16:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3685 * gst/playback/gstplaybin2.c:
3686 playbin2: release the shutdown lock
3687 Release the shutdown lock when we wait for other groups to complete or else we
3688 have a deadlock when the other group completes and tries to grab the shutdown
3692 2009-03-16 15:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3694 * tests/examples/app/appsrc-ra.c:
3695 * tests/examples/app/appsrc-seekable.c:
3696 * tests/examples/app/appsrc-stream.c:
3697 * tests/examples/app/appsrc-stream2.c:
3698 examples: fix g_object_set() value type.
3699 Make sure we cast the length value as a gint64 to the vararg g_object_set() just
3700 incase sizeof(gsize) != sizeof(gint64).
3702 2009-03-15 19:57:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3704 * gst/typefind/gsttypefindfunctions.c:
3705 typefinding: make flac typefinder return lower probability for frame headers
3706 The flac frame header typefinder overstates the likelihood of a match, leading
3707 to false positives with e.g. aac streams and PDF files. Reduce probabilty
3708 returned from LIKELY to POSSIBLE for the frame header matchin code.
3711 2009-03-11 12:59:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3713 * gst/typefind/gsttypefindfunctions.c:
3714 typefinding: improve image/bmp typefinder
3715 Detect more variations and also bail out in more cases where the values
3716 don't make sense. Furthermore, add width/height and bpp to the caps,
3719 2009-03-13 15:22:42 +0000 Jan Schmidt <thaytan@noraisin.net>
3721 * tests/check/Makefile.am:
3722 check: Ignore alsamixer in the states test too
3724 2009-03-13 15:22:11 +0000 Jan Schmidt <thaytan@noraisin.net>
3726 * sys/v4l/v4l_calls.c:
3727 v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.
3729 2009-03-13 16:19:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3731 * gst-libs/gst/rtsp/gstrtspconnection.c:
3732 rtsp: fix resolving of hostnames
3733 We were returning a pointer to a stack variable with the resolved hostname,
3735 return a copy of the resolved ip address instead.
3738 2009-03-13 15:29:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3740 * ext/vorbis/vorbisparse.c:
3741 vorbisparse: be smarter when queueing headers
3742 Look at the first buffer byte to see if a buffer is a header instead of counting
3745 2009-03-13 15:27:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3747 * ext/theora/gsttheoraparse.h:
3748 * ext/theora/theoraparse.c:
3749 theoraparse: be smarter when queuing headers
3750 Look at the first byte of the buffer data (if we can) to decide if the packet is
3751 a header packet or not instead of counting packets.
3753 2009-03-13 15:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3755 * ext/ogg/gstoggdemux.c:
3756 oggdemux: add some debug info
3757 Add some debug info to log when the seek worked.
3759 2009-03-13 15:14:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3761 * gst-libs/gst/app/gstappsrc.c:
3762 appsrc: release lock in _eos flushing case
3763 Release the mutex when we are flushing in gst_app_src_end_of_stream()
3766 2009-03-13 11:49:10 +0000 Jan Schmidt <thaytan@noraisin.net>
3768 * ext/vorbis/vorbisdec.c:
3769 vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.
3771 2009-03-13 11:48:28 +0000 Jan Schmidt <thaytan@noraisin.net>
3773 * ext/theora/theoradec.c:
3774 theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.
3776 2009-03-12 18:27:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3778 * gst/playback/gsturidecodebin.c:
3779 playbin2: fix raw elements like cdda://
3780 Fix a fixme with a one liner and make cd playback work again.
3782 2009-03-12 17:47:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3784 * gst/playback/gstplaybin2.c:
3785 * gst/playback/gstplaysink.c:
3786 * gst/playback/gstplaysink.h:
3787 playbin2: improve subtitle handling
3788 Add property to playbin2 to configure a custom sink that receives the raw
3789 subtitle buffers instead of using a textoverlay.
3790 Improve the property finding code to make it more usable.
3791 Use property find code to find async properties in custom sinks that are bins.
3792 Improve text overlay code to gracefully handle missing elements.
3794 2009-02-24 15:58:42 +0000 Jan Schmidt <thaytan@noraisin.net>
3796 * gst-libs/gst/tag/gstvorbistag.c:
3797 vorbistag: Protect memory allocation calculation from overflow.
3798 Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
3800 2009-03-12 11:34:20 +0000 Jan Urbanski <jurbanski@flumotion.com>
3802 * gst-plugins-base.spec.in:
3805 2009-03-11 18:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3807 * gst-libs/gst/rtsp/gstrtspconnection.c:
3808 rtsp: fix parsing of the timeout parameter
3811 2009-03-11 16:20:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3813 * gst-libs/gst/rtsp/gstrtspmessage.c:
3814 rtsp: fix g_return condition
3815 when parsing a data message, we require a data message.
3817 2009-03-11 13:33:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3819 * gst/typefind/gsttypefindfunctions.c:
3820 typefinding: flac typefinder fixes
3821 Use scan context for initial peek as well. Peek 6 bytes in the initial
3822 peek rather than 5 bytes, to match the length of the memcmp we're doing
3823 on that data later. Return immediately when we found caps from looking
3824 at the beginning of the data - no point in continuing to scan the next
3825 64kB for something matching a frame header.
3827 2009-03-11 14:08:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3829 * gst-libs/gst/rtsp/gstrtspmessage.c:
3830 rtsp: free the right string.
3831 Free the key value before we remove the header item from the array. The item we
3832 retrieved from the array is only valid until we remove it from the array.
3834 2009-03-11 14:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3836 * gst-libs/gst/rtsp/gstrtspconnection.c:
3837 rtsp: keep track of amount of decoded bytes
3838 Keep track of the actual amount of decoded bytes, which can be less than 3 when
3839 we decode the last bits of a base64 message.
3841 2009-03-10 21:00:26 +0200 Stefan Kost <ensonic@users.sf.net>
3843 * gst/adder/gstadder.c:
3844 adder: log details in getcaps like in setcaps
3846 2009-03-10 13:11:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3849 win32: update MANIFEST, fixing 'make dist'
3851 2009-03-09 23:12:00 +0000 Jan Schmidt <thaytan@noraisin.net>
3854 Automatic update of common submodule
3855 From 7032163 to f8b3d91
3857 2009-03-09 16:19:40 +0100 Jonathan Matthew <notverysmart at gmail dot com>
3859 * gst/typefind/gsttypefindfunctions.c:
3860 typefind: add photoshop typefind functions
3861 Add photoshop typefind functions.
3864 2009-03-09 15:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3866 * gst/playback/gstdecodebin2.c:
3867 decodebin2: only remove pads that were added
3868 Flag pads that were added so that we can see if we need to remove them later or
3871 2009-03-09 13:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3873 * gst-libs/gst/rtsp/gstrtsptransport.c:
3874 rtsp: only add ports when not using TCP
3875 Only add the port numbers in the transport string when we are using udp or
3878 2009-03-09 13:53:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3880 * gst-libs/gst/rtsp/gstrtspmessage.c:
3881 rtsp: use gstreamer dump mem
3884 2009-03-09 13:51:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3886 * gst-libs/gst/rtsp/gstrtspconnection.c:
3887 rtsp: use glib base64 encoder
3890 2009-03-06 19:28:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
3892 * gst/playback/gstdecodebin2.c:
3893 Unblock blocked ghostpads when shutting down. Fixes #574293.
3895 2009-03-09 10:03:13 +0100 Edward Hervey <bilboed@bilboed.com>
3897 * gst-libs/gst/riff/riff-media.c:
3898 Riff: Add mapping for Fraps video codec.
3899 Found through insanity testrun. Confirmed mapping in libavformat.
3901 2009-03-09 09:07:13 +0100 Edward Hervey <bilboed@bilboed.com>
3903 * gst-libs/gst/riff/riff-media.c:
3904 riff: Add the 'DVR ' mapping for mpeg2video.
3905 Found this in 3 files from the insanity suite and mapping is also present
3908 2009-03-09 09:06:40 +0100 Edward Hervey <bilboed@bilboed.com>
3910 * gst/typefind/gsttypefindfunctions.c:
3911 typefind: Use the proper data pointer instead of poking random memory.
3913 2009-03-08 18:17:48 +0100 LRN <lrn1986@gmail.com>
3915 * gst-libs/gst/rtsp/gstrtspconnection.c:
3916 rtsp: fix compilation on windows.
3917 Remove unused variable when building for windows.
3920 2009-03-08 12:03:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3923 Automatic update of common submodule
3924 From ffa738d to 7032163
3926 2009-03-08 11:19:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3929 Automatic update of common submodule
3930 From 3f13e4e to ffa738d
3932 2009-03-07 11:44:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3935 Automatic update of common submodule
3936 From 3c7456b to 3f13e4e
3938 2009-03-07 10:44:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3941 Automatic update of common submodule
3942 From 57c83f2 to 3c7456b
3944 2009-03-06 19:02:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3946 * ext/theora/theoradec.c:
3947 theoradec: parse and use codec_data in the caps
3948 Parse the codec_data in the caps and use this as the headers.
3951 2009-03-06 18:53:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3953 * gst-libs/gst/riff/riff-media.c:
3954 riff: add theora mapping
3955 Add theora mappings. See #574169.
3957 2009-03-06 16:31:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3959 * gst-libs/gst/rtsp/gstrtspconnection.c:
3960 * gst-libs/gst/rtsp/gstrtspconnection.h:
3961 * win32/common/libgstrtsp.def:
3962 rtsp: Add methods for getting the read/write fds
3963 API:gst_rtsp_connection_get_readfd()
3964 API:gst_rtsp_connection_get_writefd()
3966 2009-03-06 10:35:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3969 * win32/common/audio-enumtypes.c:
3970 win32: indent copied *-enumtypes.c files in make win32-update
3972 2009-03-06 10:35:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3975 win32: update MANIFEST
3977 2009-03-06 10:30:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3980 * win32/common/config.h:
3981 win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define
3983 2009-03-06 10:05:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3985 * win32/common/_stdint.h:
3986 * win32/common/config.h:
3987 * win32/common/gstrtsp-enumtypes.c:
3988 * win32/common/interfaces-enumtypes.c:
3989 * win32/common/multichannel-enumtypes.c:
3990 * win32/common/pbutils-enumtypes.c:
3991 * win32/common/video-enumtypes.c:
3992 * win32/common/video-enumtypes.h:
3993 win32: update windows files via make win32-update
3994 Updates win32 files using the new system/hook, and defines HAVE_PROCESS_H,
3995 which fixes the build of pbutils on windows (#574319).
3997 2009-03-06 10:03:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4000 gitignore: ignore more
4002 2009-03-06 10:37:38 +0100 Julien Moutte <julien@fluendo.com>
4004 * gst-libs/gst/rtsp/gstrtspconnection.c:
4005 Fix build on Mac OS X
4007 2009-03-05 15:42:23 -0800 Michael Smith <msmith@songbirdnest.com>
4009 * gst/playback/gstdecodebin2.c:
4010 decodebin2: don't stay connected to notify::caps after negotiation
4011 Disconnect the notify::caps signal in our callback (it'll be re-added
4012 if we're not, in fact, finished getting complete caps). Ensures that
4013 caps changes mid-stream (e.g. from an mp3 that changes from
4014 stereo->mono mid-file) don't cause us to try to add a new pad.
4016 2009-03-05 13:48:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4018 * gst-libs/gst/rtsp/gstrtsprange.c:
4019 rtsp: fix parsing of 'now-' ranges.
4022 2009-03-05 12:43:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4024 * tests/examples/dynamic/.gitignore:
4025 * tests/examples/dynamic/Makefile.am:
4026 * tests/examples/dynamic/sprinkle.c:
4027 * tests/examples/dynamic/sprinkle2.c:
4028 * tests/examples/dynamic/sprinkle3.c:
4029 examples: add some more sprinkle examples
4030 Add some more sprinle examples and add some more comments.
4033 2009-03-05 11:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4035 * docs/plugins/gst-plugins-base-plugins-sections.txt:
4036 docs: add appsrc symbols to standard section
4039 2009-03-05 12:27:16 +0200 Stefan Kost <ensonic@users.sf.net>
4041 * gst/adder/gstadder.c:
4042 adder: add variants for unsigned to fix warnings for unneeded check
4043 For unsigned int out+in can't be < 0.
4045 2009-03-05 10:58:12 +0200 Stefan Kost <ensonic@users.sf.net>
4047 * gst/subparse/gstsubparse.c:
4048 subparse: use the right variable in debug log, encoding is not yet initialized
4050 2009-03-05 10:51:25 +0200 Stefan Kost <ensonic@users.sf.net>
4052 * sys/v4l/v4l_calls.c:
4053 v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix
4055 2009-03-05 10:39:33 +0200 Stefan Kost <ensonic@users.sf.net>
4057 * gst/audioresample/gstaudioresample.c:
4058 audioresample: add missing break in event handling, remove dead code
4060 2009-03-04 16:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4062 * gst-libs/gst/rtsp/gstrtspconnection.c:
4063 rtsp: do some more cleanup in _close
4064 Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
4065 unconnected state as it was allocated.
4067 2009-03-04 16:11:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4069 * gst-libs/gst/rtsp/gstrtspconnection.c:
4070 * gst-libs/gst/rtsp/gstrtspconnection.h:
4071 rtsp: fix the memory management of the url
4072 Constify the url parameter in _create.
4073 Make a copy of the url stored in the connection.
4074 Free the url when the connection is freed.
4076 2009-03-04 12:21:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4078 * docs/libs/gst-plugins-base-libs-sections.txt:
4079 * gst-libs/gst/rtsp/gstrtspconnection.c:
4080 * gst-libs/gst/rtsp/gstrtspconnection.h:
4081 * win32/common/libgstrtsp.def:
4082 RTSP: Add support for server tunneling
4083 Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
4084 that a server can store and match the id against other tunnel requests.
4085 Fix the URI in the tunnel requests so that they contain the absolute uri and the
4086 query string if any instead of just the hostname.
4087 Transparently base64 decode the input stream when tunneling.
4088 Add method to set the connection ip address so that it can be included in the
4090 Add method to connect the two tunnel requests.
4091 Add two callbacks for the async mode to notify a tunnel start and tunnel
4093 Add method to reset the watch after the connection has been tunneled.
4094 Various little refactoring to make more stuff reusable.
4095 API: RTSP::gst_rtsp_connection_set_ip()
4096 API: RTSP::gst_rtsp_connection_get_tunnelid()
4097 API: RTSP::gst_rtsp_connection_do_tunnel()
4098 API: RTSP::gst_rtsp_watch_reset()
4100 2009-03-04 12:18:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4102 * gst-libs/gst/rtsp/gstrtspdefs.c:
4103 * gst-libs/gst/rtsp/gstrtspdefs.h:
4104 rtsp: add new defines for tunneling
4105 Add two more result codes for tunneling support.
4107 2009-03-04 12:12:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4109 * gst-libs/gst/rtsp/gstrtspmessage.h:
4110 rtsp: remove , from last enum member
4111 Remove , from last enum member to improve compatibility with other compilers.
4113 2009-02-28 15:23:20 -0800 LRN <lrn1986@gmail.com>
4115 * gst/subparse/gstsubparse.c:
4116 subparse: Convert regex code to GRegex code
4117 Fixes: #572993. Patch author prefers to use an alias, contact
4118 ds if you actually need a real name.
4119 Signed-off-by: David Schleef <ds@schleef.org>
4121 2009-03-02 16:13:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4123 * gst-libs/gst/rtsp/gstrtspconnection.c:
4124 rtsp: remove debugging g_message
4127 2009-03-02 16:03:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4129 * docs/libs/gst-plugins-base-libs-sections.txt:
4130 * gst-libs/gst/rtsp/gstrtspconnection.c:
4131 * gst-libs/gst/rtsp/gstrtspconnection.h:
4132 * win32/common/libgstrtsp.def:
4133 RTSP: add support for Quicktime tunneled RTSP
4134 Add support for tunneling RTSP over HTTP.
4135 Fix documentation some more.
4137 API: RTSP:gst_rtsp_connection_is_tunneled()
4138 API: RTSP:gst_rtsp_connection_set_tunneled()
4140 2009-03-02 15:48:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4142 * gst-libs/gst/rtsp/gstrtsptransport.h:
4143 * gst-libs/gst/rtsp/gstrtspurl.c:
4144 RTSP: parse rtsph uris as RTSP tunneled over HTTP
4145 Add transport define for RTSP tunneled over HTTP.
4146 Parse rtsph:// uris as tunneled HTTP over TCP.
4147 API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
4150 2009-03-02 12:48:18 +0100 Edward Hervey <bilboed@bilboed.com>
4152 * win32/common/libgstrtsp.def:
4153 win32: Add gst_rtsp_connection_get_url definition
4154 No, I'm not wim's buildslave, seriously.
4156 2009-03-02 10:58:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4158 * gst-libs/gst/rtsp/gstrtspconnection.c:
4159 * gst-libs/gst/rtsp/gstrtspconnection.h:
4160 rtsp: add _get_url method and separate sockets
4161 Add gst_rtsp_connection_get_url() method.
4162 Reserve space for 2 sockets, one for reading and one for writing. Use socket
4163 pointers to select the read and write sockets. This should allow us to implement
4164 tunneling over HTTP soon.
4165 API: RTSP::gst_rtsp_connection_get_url()
4167 2009-03-01 18:31:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4169 * gst-libs/gst/app/gstapp-marshal.list:
4170 app: force automatic rebuild of gstapp-marshal.[ch] after previous change
4171 The previous change to appsrc/appsink requires people to 'make clean'
4172 to get the marshallers rebuilt (causing a build failure otherwise).
4173 Change some lines in the .list file around to force a rebuild of
4174 these files automatically.
4176 2009-02-28 11:07:04 -0800 David Schleef <ds@schleef.org>
4179 Bump glib requirement to 2.14
4181 2009-02-28 19:37:53 +0100 LRN <lrn1986@gmail.com>
4183 * ext/gio/gstgiobasesink.c:
4184 gio: Use correct format modifier for size_t
4187 2009-02-28 19:35:33 +0100 LRN <lrn1986@gmail.com>
4189 * gst-libs/gst/rtsp/gstrtspconnection.c:
4190 rtspconnection: Use correct types for some functions on Win32
4193 2009-02-28 13:11:59 +0100 Edward Hervey <bilboed@bilboed.com>
4195 * gst-libs/gst/rtsp/gstrtspconnection.c:
4196 rtspconnection: Fix warning about using unitialized value.
4198 2009-02-28 12:41:28 +0100 Edward Hervey <bilboed@bilboed.com>
4200 * gst-libs/gst/riff/riff-ids.h:
4201 * gst-libs/gst/riff/riff-media.c:
4202 riff: Add more codec mappings.
4203 This comes mostly from a review of ffmpeg/libavformat/riff.c
4205 2009-02-27 11:14:25 +0200 Stefan Kost <ensonic@users.sf.net>
4207 * ext/alsa/gstalsa.c:
4208 alsa: release pcminfo after the strdup
4210 2009-02-26 17:38:47 +0200 Stefan Kost <ensonic@users.sf.net>
4212 * gst-libs/gst/rtsp/gstrtsprange.c:
4213 rtsprange: don't leak the range in case of parsing error.
4214 Free the gstRTSPTimeRange if we don't return it. Also simplify
4215 gst_rtsp_range_free() as it is valid to pass NULL to g_free().
4217 2009-02-26 16:47:39 +0200 Stefan Kost <ensonic@users.sf.net>
4219 * ext/alsa/gstalsa.c:
4220 alsa: cleanup name lookup.
4221 We can break, once we have a name to make sure, we won't read it ever twice.
4223 2009-02-26 16:09:03 +0200 Stefan Kost <ensonic@users.sf.net>
4225 * gst/subparse/gstsubparse.c:
4226 subparse: don't leak line, if flushing
4228 2009-02-26 16:03:39 +0200 Stefan Kost <ensonic@users.sf.net>
4230 * ext/gio/gstgiosink.c:
4231 giosink: reflow error handling to not leak uri
4233 2009-02-26 15:53:10 +0200 Stefan Kost <ensonic@users.sf.net>
4235 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
4236 * gst/ffmpegcolorspace/imgconvert.c:
4237 ffmpegcolorspace: remove unused code/variables
4239 2009-02-26 12:10:47 +0200 Stefan Kost <ensonic@users.sf.net>
4241 * sys/ximage/ximagesink.c:
4242 ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps
4244 2009-02-26 16:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4246 * docs/libs/gst-plugins-base-libs-sections.txt:
4247 * gst-libs/gst/app/gstappsink.c:
4248 * gst-libs/gst/app/gstappsrc.c:
4249 * gst-libs/gst/app/gstappsrc.h:
4250 * win32/common/libgstapp.def:
4251 app: add callbacks to appsrc, cleanups
4252 Add a uri handler to appsink.
4253 don't emit signals when we have installed callbacks on appsink.
4254 Add callbacks to appsrc to replace the signals.
4255 Add property to disable callbacks in appsrc, default to TRUE for backwards
4256 compatibility but disable when callbacks are installed.
4257 API: GstAppSrc::emit-signals
4258 API: GstAppSrc::gst_app_src_set_emit_signals()
4259 API: GstAppSrc::gst_app_src_get_emit_signals()
4260 API: GstAppSrc::gst_app_src_set_callbacks()
4262 2009-02-26 11:42:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4264 * docs/libs/gst-plugins-base-libs-sections.txt:
4265 * gst-libs/gst/app/gstappsink.h:
4266 * tests/check/elements/appsink.c:
4267 Appsink: add padding for callbacks + docs
4268 Add some padding to the callbacks structure just to be safe.
4269 Remove the now invisible marshaller methods from the docs.
4270 Fix a comment in the unit test.
4272 2009-02-26 09:52:59 +0100 Edward Hervey <bilboed@bilboed.com>
4274 * win32/common/libgstapp.def:
4275 win32: Add new libgstapp symbol
4277 2009-02-26 10:07:21 +0200 Stefan Kost <ensonic@users.sf.net>
4279 * docs/plugins/gst-plugins-base-plugins-sections.txt:
4280 docs: clean section.txt file.
4281 Add appsrc/sink symbols to private, as they are covered in the libs docs.
4283 2009-02-26 10:06:23 +0200 Stefan Kost <ensonic@users.sf.net>
4285 * gst/playback/gstplaybasebin.c:
4286 docs: fix random text after since: tag. Also fix class name to make the docs actual appear.
4288 2009-02-26 09:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
4290 * docs/plugins/gst-plugins-base-plugins.args:
4291 * docs/plugins/gst-plugins-base-plugins.hierarchy:
4292 * docs/plugins/gst-plugins-base-plugins.interfaces:
4293 * docs/plugins/gst-plugins-base-plugins.prerequisites:
4294 * docs/plugins/inspect/plugin-adder.xml:
4295 * docs/plugins/inspect/plugin-alsa.xml:
4296 * docs/plugins/inspect/plugin-app.xml:
4297 * docs/plugins/inspect/plugin-audioconvert.xml:
4298 * docs/plugins/inspect/plugin-audiorate.xml:
4299 * docs/plugins/inspect/plugin-audioresample.xml:
4300 * docs/plugins/inspect/plugin-audiotestsrc.xml:
4301 * docs/plugins/inspect/plugin-cdparanoia.xml:
4302 * docs/plugins/inspect/plugin-decodebin.xml:
4303 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
4304 * docs/plugins/inspect/plugin-gdp.xml:
4305 * docs/plugins/inspect/plugin-gio.xml:
4306 * docs/plugins/inspect/plugin-gnomevfs.xml:
4307 * docs/plugins/inspect/plugin-libvisual.xml:
4308 * docs/plugins/inspect/plugin-ogg.xml:
4309 * docs/plugins/inspect/plugin-pango.xml:
4310 * docs/plugins/inspect/plugin-playback.xml:
4311 * docs/plugins/inspect/plugin-queue2.xml:
4312 * docs/plugins/inspect/plugin-subparse.xml:
4313 * docs/plugins/inspect/plugin-tcp.xml:
4314 * docs/plugins/inspect/plugin-theora.xml:
4315 * docs/plugins/inspect/plugin-typefindfunctions.xml:
4316 * docs/plugins/inspect/plugin-uridecodebin.xml:
4317 * docs/plugins/inspect/plugin-video4linux.xml:
4318 * docs/plugins/inspect/plugin-videorate.xml:
4319 * docs/plugins/inspect/plugin-videoscale.xml:
4320 * docs/plugins/inspect/plugin-videotestsrc.xml:
4321 * docs/plugins/inspect/plugin-volume.xml:
4322 * docs/plugins/inspect/plugin-vorbis.xml:
4323 * docs/plugins/inspect/plugin-ximagesink.xml:
4324 * docs/plugins/inspect/plugin-xvimagesink.xml:
4325 * gst/playback/gstplaybin2.c:
4326 docs: playbin2 has no stream-info
4328 2009-02-26 09:53:03 +0200 Stefan Kost <ensonic@users.sf.net>
4330 * gst-libs/gst/video/video.h:
4331 docs: fix newly added interlace constants and plug holes in video format docs
4333 2009-02-26 09:35:43 +0200 Stefan Kost <ensonic@users.sf.net>
4335 * gst-libs/gst/app/gstappsink.c:
4336 * gst-libs/gst/app/gstappsrc.c:
4337 * gst-libs/gst/audio/gstaudiofilter.c:
4338 * gst-libs/gst/audio/gstringbuffer.c:
4339 * gst-libs/gst/rtp/gstrtcpbuffer.c:
4340 docs: don't put random stuff in tags.
4341 Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
4342 tag to append text again to the documentation body.
4344 2009-02-06 11:10:15 +0200 Stefan Kost <ensonic@users.sf.net>
4346 * sys/ximage/ximagesink.c:
4347 ximagsink: do not access uninitialized height variable.
4348 Exit like in xvimagesink, if we have partial caps.
4350 2009-02-25 20:26:05 -0800 David Schleef <ds@schleef.org>
4354 * win32/common/config.h.in:
4355 Change how win32/common/config.h is updated
4356 Generate win32/common/config.h-new directly from config.h.in,
4357 using shell variables in configure and some hard-coded information.
4358 Change top-level makefile so that 'make win32-update' copies the
4359 generated file to win32/common/config.h, which we keep in source
4360 control. It's kept in source control so that the git tree is
4362 This change is similar to the one recently applied to GStreamer,
4363 except that it adds a few -base specific defines.
4365 2009-02-25 19:40:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4367 * gst-libs/gst/app/Makefile.am:
4368 * gst-libs/gst/app/gstappsink.c:
4369 * gst-libs/gst/app/gstappsrc.c:
4370 * win32/common/libgstapp.def:
4371 app: add win32 .def file and only export functions we want exported
4372 Add a .def file for win32 builds (and make check-exports).
4373 Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes #573165).
4374 Make sure private marshaller functions aren't exported by prefixing them with __gst;
4375 also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
4376 a comment why we're not using glib-genmarshal for this one.
4378 2009-02-25 17:08:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4380 * tests/examples/dynamic/.gitignore:
4381 * tests/examples/dynamic/Makefile.am:
4382 * tests/examples/dynamic/sprinkle.c:
4383 sprinkle: Add another example app
4384 Add an example app that dynamically adds and removes audiotestsrc elements from
4387 2009-02-25 16:25:33 +0100 Peter Kjellerstedt <pkj@axis.com>
4389 * gst-libs/gst/rtsp/gstrtspconnection.c:
4392 2009-02-25 11:31:02 +0100 Peter Kjellerstedt <pkj@axis.com>
4394 * gst-libs/gst/rtsp/gstrtspconnection.c:
4395 * gst/tcp/gstmultifdsink.c:
4396 rtsp, multifdsink: Unify the use of union gst_sockaddr.
4398 2009-02-25 14:22:35 +0000 Jan Schmidt <thaytan@noraisin.net>
4402 build: Update shave init statement for changes in common. Bump common.
4404 2009-02-25 13:16:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4406 * sys/xvimage/xvimagesink.c:
4407 * sys/xvimage/xvimagesink.h:
4408 xvimageink: protect buffer_alloc from shutdown
4409 Use the pool_lock in the buffer_alloc function to detect shutdown. Avoids
4410 crashes when the sink is shutdown.
4412 2009-02-25 12:43:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4414 * gst/playback/gstplaybin2.c:
4415 playbin: use flushing pads instead of fakesink
4416 Use the flushing pads on playsink to terminate on shutdown instead of plugging
4417 fakesinks. this should be a little cheaper.
4419 2009-02-25 12:42:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4421 * gst/playback/gstplaysink.c:
4422 * gst/playback/gstplaysink.h:
4423 playsink: Add FLUSHING pad type
4424 Make it possible to request a flushing pad from the playsink. We can eventually
4425 use these flushing pads to quickly terminate the dataflow when we are shutting
4428 2009-02-25 11:31:52 +0000 Jan Schmidt <thaytan@noraisin.net>
4431 Automatic update of common submodule
4432 From 9cf8c9b to a6ce5c6
4434 2009-02-25 09:52:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4436 * gst-libs/gst/riff/riff-media.c:
4437 riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
4440 2009-02-25 12:07:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4442 * tests/icles/stress-playbin.c:
4443 stress-playbin: print the current uri
4444 Print the current uri so that we can more easily see what uri caused a crash or
4447 2009-02-25 11:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4449 * tests/icles/stress-playbin.c:
4450 Print the errors more clearly
4451 Print some more verbose messages when dealing with errors.
4453 2009-02-25 10:08:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4455 * gst/playback/gstplaybin2.c:
4456 Release the group lock when setting states
4457 Release the group lock while we perform the state changes on the uridecodebins
4458 because that might trigger callbacks that we need to handle with the group lock
4459 taken. Avoids a possible deadly embrace in some id3/flac files.
4462 2009-02-25 10:05:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4464 * gst/playback/gstdecodebin2.c:
4465 Combine finding and creating groups
4466 Combine the search for the current group and optionally creating one into one
4467 function so that we can avoid taking the lock multiple times.
4469 2009-02-25 08:22:00 +0100 Edward Hervey <bilboed@bilboed.com>
4471 * gst/playback/gstplaybin2.c:
4472 Playbin2: Don't leave unused parameters in debug statements.
4473 Fixes build on macosx
4475 2009-02-24 10:33:05 +0100 Edward Hervey <bilboed@bilboed.com>
4477 * gst-libs/gst/riff/riff-media.c:
4478 Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)
4480 2009-02-24 18:43:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4482 * gst/playback/gstplaybin2.c:
4483 Add some G_UNLIKELY because we can
4484 Add a G_UNLIKELY when checking the shutdown variable.
4486 2009-02-24 17:23:58 +0000 Garret D'Amore <garrett.damore@sun.com>
4488 * gst-libs/gst/interfaces/mixer.h:
4489 * gst-libs/gst/interfaces/mixertrack.h:
4490 mixer interface: Add flags to enhance mixer interfaces
4491 This patch adds a few flags to the mixer and mixerctrl interface to
4492 better support OSSv4 (and potentially other backends).
4493 Patch By: Garret D'Amore <garrett.damore@sun.com>
4494 Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>
4495 API: GST_MIXER_FLAG_HAS_WHITELIST, GST_MIXER_FLAG_GROUPING,
4496 API: GST_MIXER_TRACK_NO_RECORD, GST_MIXER_TRACK_NO_MUTE,
4497 API: GST_MIXER_TRACK_WHITELIST
4499 2009-02-24 17:03:08 +0000 Jan Schmidt <thaytan@noraisin.net>
4501 * gst/tcp/gstmultifdsink.c:
4502 multifdsink: Fix strict aliasing error using a union
4504 2009-02-24 16:49:40 +0000 Jan Schmidt <thaytan@noraisin.net>
4506 * gst-libs/gst/rtsp/gstrtspconnection.c:
4507 rtsp: Fix a strict aliasing warning
4508 Fix strict aliasing warnings from casting a sockaddr_storage and
4509 using it as a sockaddr_in6. Use a union instead.
4511 2009-02-24 16:08:49 +0000 Jan Schmidt <thaytan@noraisin.net>
4513 * docs/libs/.gitignore:
4514 * docs/libs/tmpl/.gitignore:
4515 * docs/plugins/.gitignore:
4516 * docs/plugins/tmpl/.gitignore:
4517 Remove .gitignore files from the docs tmpl dirs, that are killed by make clean.
4519 2009-02-24 14:36:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4521 * docs/plugins/Makefile.am:
4522 * ext/vorbis/Makefile.am:
4523 * ext/vorbis/gstvorbisdec.h:
4524 * ext/vorbis/gstvorbisenc.h:
4525 * ext/vorbis/gstvorbisparse.h:
4526 * ext/vorbis/gstvorbistag.h:
4527 * ext/vorbis/vorbis.c:
4528 * ext/vorbis/vorbisdec.c:
4529 * ext/vorbis/vorbisdec.h:
4530 * ext/vorbis/vorbisenc.c:
4531 * ext/vorbis/vorbisenc.h:
4532 * ext/vorbis/vorbisparse.c:
4533 * ext/vorbis/vorbisparse.h:
4534 * ext/vorbis/vorbistag.c:
4535 * ext/vorbis/vorbistag.h:
4536 vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
4538 2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4540 * gst/ffmpegcolorspace/avcodec.h:
4541 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
4542 * gst/ffmpegcolorspace/imgconvert.c:
4543 ffmpegcolorspace: Add conversion from/to YVYU colorspace
4546 2009-02-24 13:42:01 +0100 Jonas Danielsson <jonas.danielsson@axis.com>
4548 * gst/ffmpegcolorspace/imgconvert.c:
4549 ffmpegcolorspace: Add direct UYVY->GRAY8 conversion
4550 The conversion from UYVY to RGB24 and then to GRAY8
4551 is quite slow. Fixes bug #569655.
4553 2009-02-19 17:16:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4555 * gst/playback/gstplaybin2.c:
4556 playbin2: fix deadlock when shutting down. Fixes #572577.
4558 2009-02-19 17:15:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4560 * tests/icles/stress-playbin.c:
4561 stress-playbin: make more flexible, e.g. also useful for playbin2
4563 2009-02-24 12:11:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4565 * gst-libs/gst/rtsp/gstrtspconnection.c:
4566 Match WSAStartup and WSACleanup correctly
4567 Don't randomly call WSAStartup and WSACleanup but instead call the startup when
4568 we create a connection and cleanup when we free it again. Because the internal
4569 datastructure is refcounted, this should not cause any refcounting leaks when
4570 the connection is managed correctly.
4573 2009-02-18 11:59:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4575 * gst/playback/gstplaysink.c:
4576 playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105.
4578 2009-02-23 10:57:42 -0800 David Flynn <davidf@rd.bbc.co.uk>
4580 * pkgconfig/gstreamer-app-uninstalled.pc.in:
4581 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
4582 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
4583 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
4584 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
4585 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
4586 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
4587 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
4588 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
4589 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
4590 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
4591 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
4592 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
4593 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
4594 * pkgconfig/gstreamer-video-uninstalled.pc.in:
4595 Add srcdir to includes for out-of-source builds
4596 When you use gstreamer uninstalled and build outside
4597 the source tree, the includes need to be specified for
4598 both the source tree and the build tree.
4599 Signed-off-by: David Schleef <ds@schleef.org>
4601 2009-02-22 17:23:52 +0000 Jan Schmidt <thaytan@noraisin.net>
4604 * docs/libs/Makefile.am:
4605 * docs/plugins/Makefile.am:
4606 Use shave for the build output
4608 2009-02-23 12:17:07 +0100 Edward Hervey <bilboed@bilboed.com>
4610 * win32/common/libgstrtsp.def:
4611 win32: Add new symbol to libgstrtsp.def
4613 2009-02-23 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4615 * gst-libs/gst/rtsp/gstrtspextension.c:
4616 * gst-libs/gst/rtsp/gstrtspextension.h:
4617 Add method for handling server requests
4618 Add a receive_request so that extensions can react to server requests.
4620 2009-02-22 19:20:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4622 * tests/check/libs/netbuffer.c:
4623 Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)
4625 2009-02-22 19:19:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4627 * ext/theora/theoraparse.c:
4628 theoraparse: Use the correct unref functions
4630 2009-02-22 19:18:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4632 * sys/ximage/ximagesink.c:
4633 * sys/xvimage/xvimagesink.c:
4634 x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()
4636 2009-02-22 19:12:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4638 * gst-libs/gst/tag/gsttagdemux.c:
4639 tagdemux: Unref the actual buffer instead of the memory address of the buffer
4641 2009-02-22 15:47:53 +0000 Jan Schmidt <thaytan@noraisin.net>
4644 Automatic update of common submodule
4645 From 5d7c9cc to 9cf8c9b
4647 2009-02-22 14:49:29 +0100 Edward Hervey <bilboed@bilboed.com>
4649 * win32/common/libgstrtsp.def:
4650 * win32/common/libgstvideo.def:
4651 win32/common: Update .def files for recent API addition
4653 2009-02-22 13:43:35 +0100 Edward Hervey <bilboed@bilboed.com>
4655 * tests/check/libs/rtp.c:
4656 tests: Fix indentation
4658 2009-02-22 13:42:33 +0100 Edward Hervey <bilboed@bilboed.com>
4660 * gst-libs/gst/video/video.c:
4661 libs/video: Fix gst_video_format_new_caps* functions.
4662 Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
4665 2009-02-21 11:13:36 -0800 David Schleef <ds@schleef.org>
4668 Automatic update of common submodule
4669 From 80c627d to 5d7c9cc
4671 2009-02-20 17:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4673 * gst-libs/gst/rtsp/gstrtspmessage.c:
4674 Improve key/value parsing
4675 Improve header field parsing by keeping a ref to the key/value instead of
4676 copying it into a local variable.
4678 2009-02-20 12:35:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4680 * gst-libs/gst/rtsp/gstrtspconnection.c:
4681 Add trailing \0 to message length
4682 We always put a trailing 0 at the end of the message body. Reflect this fact in
4683 the length of the message.
4685 2009-02-20 09:50:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4687 * gst-libs/gst/rtsp/gstrtspconnection.c:
4688 Don't parse headers for data messages
4689 Don't try to parse the headers on a data message because they don't have
4692 2009-02-19 12:18:29 -0800 Benjamin M. Schwartz <bens@alum.mit.edu>
4694 * ext/theora/gsttheoraenc.h:
4695 * ext/theora/theoraenc.c:
4696 theoraenc: Add property for speed level control
4697 Add property "speed-level" to control the amount of motion searching
4698 the encoder does. This is only available in libtheora >= 1.0 and
4699 will silently fail with earlier libraries. Fixes: #572275.
4700 Signed-off-by: David Schleef <ds@schleef.org>
4702 2009-02-19 17:40:45 +0100 Edward Hervey <bilboed@bilboed.com>
4704 * gst-libs/gst/video/video.c:
4705 * gst-libs/gst/video/video.h:
4706 video: Fix 'Since' tags
4708 2009-01-26 10:30:53 +0100 Edward Hervey <bilboed@bilboed.com>
4710 * docs/libs/gst-plugins-base-libs-sections.txt:
4711 * gst-libs/gst/video/video.c:
4712 * gst-libs/gst/video/video.h:
4713 video: Add flags for interlaced video along with convenience methods for interlaced caps.
4714 These three flags allow all know combinations of interlaced formats. They should
4715 only be used when the caps contain 'interlaced=True'.
4716 Fixes #163577 (yes, it's a 4 year old bug).
4718 2009-02-19 15:51:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4720 * docs/libs/gst-plugins-base-libs-sections.txt:
4721 * gst-libs/gst/rtsp/gstrtspconnection.c:
4722 * gst-libs/gst/rtsp/gstrtspconnection.h:
4723 Make RTSPConnection opaque and rename RTSPChannel
4724 Make the RTSPConnection object opaque so that we can extend it in the future.
4725 Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
4727 2009-01-26 10:31:14 +0100 Edward Hervey <bilboed@bilboed.com>
4729 * gst-libs/gst/riff/riff-media.c:
4730 Add some more mappings for h264 in riff
4732 2009-02-19 10:49:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4734 * win32/common/libgstrtsp.def:
4735 Add new RTSP symbols to def files
4736 Add the new RTSP symbols to the windows def file.
4738 2009-02-19 10:44:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4740 * docs/libs/gst-plugins-base-libs-sections.txt:
4741 * gst-libs/gst/app/gstappsink.c:
4742 * gst-libs/gst/app/gstappsink.h:
4743 * tests/check/Makefile.am:
4744 * tests/check/elements/.gitignore:
4745 * tests/check/elements/appsink.c:
4746 Add method to install callbacks on appsink
4747 Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
4749 Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
4750 performant alternative to connecting to the signals.
4751 Add a unit test for appsink.
4752 Clean up some of the appsink docs.
4753 API: GstAppSink::gst_app_sink_set_callbacks()
4755 2009-02-18 18:46:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4757 * docs/libs/gst-plugins-base-libs-sections.txt:
4758 * gst-libs/gst/rtsp/gstrtspconnection.c:
4759 * gst-libs/gst/rtsp/gstrtspconnection.h:
4760 Add RTSP accept method
4761 Add a method to accept a connection on a socket and create a GstRTSPConnection
4763 API: gst_rtsp_connection_accept()
4765 2009-02-18 17:42:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4767 * docs/libs/gst-plugins-base-libs-sections.txt:
4768 * gst-libs/gst/rtsp/gstrtspconnection.c:
4769 * gst-libs/gst/rtsp/gstrtspconnection.h:
4770 Add RTSP channel object for async io
4771 Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
4772 that the connection can be monitored from a maincontext. This allows us to
4773 operate in ASYNC mode, which is handy when building a server.
4774 Rework the old code to use the async code under the hood.
4775 API: gst_rtsp_channel_new()
4776 API: gst_rtsp_channel_unref()
4777 API: gst_rtsp_channel_attach()
4778 API: gst_rtsp_channel_queue_message()
4780 2009-02-15 07:30:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4782 * gst/audioresample/gstaudioresample.c:
4783 audioresample: Add locking to protect the resampling context
4784 When setting the quality/filter-length while PLAYING the
4785 resampling context will be destroyed and created again in
4786 some cases, which will cause crashes in the transform function
4787 if it's called at that time.
4789 2009-02-13 10:10:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4791 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
4792 * gst/videotestsrc/videotestsrc.c:
4793 ffmpegcolorspace/videotestsrc: Use v308 instead of V308
4795 2009-02-12 19:02:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4797 * gst/ffmpegcolorspace/avcodec.h:
4798 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
4799 * gst/ffmpegcolorspace/imgconvert.c:
4800 * gst/ffmpegcolorspace/imgconvert_template.h:
4801 ffmpegcolorspace: Add support for packed 4:4:4 YUV (format=V308)
4802 Only conversions from/to are implemented, which
4803 gives (indirect) support for all possible conversions.
4804 Partially fixes bug #571147.
4806 2009-02-12 18:17:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4808 * gst/videotestsrc/videotestsrc.c:
4809 videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
4810 Partially fixes bug #571147.
4812 2009-02-12 09:18:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4814 * gst-libs/gst/tag/gsttagdemux.c:
4815 tagdemux: don't abort when downstream pulls a buffer of size 0
4816 Pulling a 0-sized buffer is allowed, and we should handle this correctly instead of
4817 aborting. Fixes #571009 (wma file with ID3v2 tag).
4819 2009-02-11 16:39:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4821 * gst-libs/gst/riff/riff-read.c:
4822 riff: error out on nonsensical chunk sizes instead of aborting
4823 When encountering a nonsensical chunk size such as (guint)-1, error out cleanly instead of
4824 continuing and trying to g_memdup() 4GB of data that doesn't exist, which will either abort
4825 in g_malloc() or crash.
4826 Fixes #553295, crash with fuzzed AVI file.
4828 2009-02-11 16:39:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4831 Make git ignore backup files.
4833 2009-02-10 20:38:58 -0800 Michael Smith <msmith@syncword.(none)>
4835 * gst/playback/gstplaybin2.c:
4836 Revert "Remove pad-removed handlers after setting the decodebins to NULL."
4837 This reverts commit b36d8f3e119f9edc5993c08025614ee32642972e.
4838 This brought back some deadlocks. A small leak is better, for now. Need to
4839 figure out a way to fix the leak properly.
4841 2009-02-10 17:16:07 -0800 Michael Smith <msmith@songbirdnest.com>
4843 * gst/playback/gstplaybin2.c:
4844 playbin2: Fix segfault on notify after group change.
4845 If our group has been switched, then we get a selector active-pad
4846 notification, we don't need to notify.
4848 2009-02-10 17:10:33 -0800 Michael Smith <msmith@songbirdnest.com>
4850 * gst/playback/gstplaysink.c:
4851 playbin2: Look for volume/mute properties recursively in audio element.
4852 Rather than only checking for volume property on the audio sink
4853 directly, recursively look for it on sinks within it (if it's a bin).
4854 Allows use of sink-as-volume-control where the application has supplied
4855 an audio-sink bin that includes a real audio sink internally.
4857 2009-02-10 18:29:22 +0000 Christian Schaller <cschalle@crazyhorse.localdomain>
4859 * gst-plugins-base.spec.in:
4860 Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base
4862 2009-02-10 17:39:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4864 * gst/videotestsrc/videotestsrc.c:
4865 videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
4866 Partially fixes bug #571147.
4868 2009-02-10 17:37:06 +0100 Peter Kjellerstedt <pkj@axis.com>
4870 * gst-libs/gst/rtsp/gstrtspmessage.c:
4871 gstrtspmessage: Minor documentation correction.
4872 Corrected documentation about what needs to be freed after calling
4873 gst_rtsp_message_new(), gst_rtsp_message_new_request(),
4874 gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
4876 2009-02-10 11:00:12 +0100 Antoine Tremblay <hexa00@gmail.com>
4878 * ext/alsa/gstalsamixer.c:
4879 alsamixer: Fix race condition that made alsamixer not working properly
4880 This is due to race conditions between functions that
4881 modified the mixer like set_volume and
4882 snd_mixer_handle_events since the handle_events
4883 can now be called at any time.
4884 Fixed by adding locking around any snd_mixer call
4885 since even read functions can modify the mixer stucture, since
4886 alsa likes to clear it's values before reading new ones.
4887 The favorite race condition seemed to be that set_volume
4888 called read_elem (in alsalib) that reset the volumes to
4889 0 and then read them with read_x_volume. This read looped
4890 on each channel and as the race condition occured the
4891 channels value could be anything , most of the time
4892 it was 0. Thus no value was read or only the value of
4893 one channel was and the volume was reset to 0.
4896 2009-02-09 12:02:21 +0100 Edward Hervey <bilboed@bilboed.com>
4899 Bump revision to use for common submodule.
4901 2009-02-05 15:47:00 +0200 Stefan Kost <ensonic@users.sf.net>
4903 * sys/xvimage/xvimagesink.c:
4904 xvimagesink: do not call _xwindow_clear on ready->paused.
4905 Calling clear at that transition does things like stopping xvideo (which is not
4906 running at that time) and also clearing anything what the application might have drawn.
4907 This breaks handle-expose and autopaint-colorkey features.
4909 2009-02-04 17:03:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4911 * docs/libs/gst-plugins-base-libs-sections.txt:
4912 * gst-libs/gst/rtsp/gstrtsprange.c:
4913 * gst-libs/gst/rtsp/gstrtsprange.h:
4914 RTSPRange: Add method to serialize ranges
4915 Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
4916 be used by a server.
4917 API: GstRTSPRange::gst_rtsp_range_to_string()
4919 2009-02-04 13:16:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4921 * gst-libs/gst/rtsp/gstrtspurl.c:
4922 * gst-libs/gst/rtsp/gstrtspurl.h:
4923 GstRTSPUrl: Add some const to methods
4924 Add const to the methods that do not modify the object.
4926 2009-02-04 13:53:30 +0200 Stefan Kost <ensonic@users.sf.net>
4928 * gst/playback/gstplaysink.c:
4929 playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
4930 The flags where present but actually not been taken into account.
4932 2009-02-04 12:06:38 +0200 Stefan Kost <ensonic@users.sf.net>
4934 * gst/audioresample/gstaudioresample.c:
4935 audioresample: Add a proper deprecation comment and also drop G_PARAM_CONSTRUCT.
4936 The comment will ensure that is is marked properly in the docs and the
4937 GParamSpecflag was causing a duplicated initialisation of the same value.
4939 2009-02-04 11:18:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4941 * gst-libs/gst/rtsp/gstrtspconnection.c:
4942 Add more g_return_if_fail() calls
4943 Check that we have a valid file descriptor before entering certain functions in
4944 order to avoid undesirable situations.
4945 Add some more debugging in the connect method.
4947 2009-02-04 10:31:21 +0200 Stefan Kost <ensonic@users.sf.net>
4950 * gst/audioresample/Makefile.am:
4951 * gst/audioresample/gstaudioresample.c:
4952 audioresample: Only pull in liboil if its actualy used.
4953 Liboil still has quite significant startup overhead especialy on embedded
4954 platforms. In audioresample it was only used for the profiling timer.
4956 2009-02-03 15:26:08 +0200 Stefan Kost <ensonic@users.sf.net>
4958 * gst/typefind/gsttypefindfunctions.c:
4959 typefind: Make the flac check more tight to not mistace some aac files for flac. Fixes #570356.
4960 Add comments about the flac format. Tighten the check to not allow values that
4963 2009-02-03 10:52:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4965 * win32/common/libgstrtsp.def:
4967 Add new methods to the windows def file.
4969 2009-02-02 17:25:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4971 * gst-libs/gst/pbutils/install-plugins.c:
4972 * tests/check/libs/pbutils.c:
4973 pbutils: remove duplicate detail strings when calling the external codec installer
4974 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.
4976 2009-02-02 18:05:42 +0200 Stefan Kost <ensonic@users.sf.net>
4978 * gst-libs/gst/audio/gstaudiosink.c:
4979 * gst-libs/gst/audio/gstaudiosink.h:
4980 Add a FIXME 0.11. Make the log message a bit more detailed and add comments.
4982 2009-02-02 15:43:03 +0200 Stefan Kost <ensonic@users.sf.net>
4985 * gst/audioresample/gstaudioresample.c:
4986 Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark.
4988 2009-02-02 13:30:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4990 * sys/ximage/ximagesink.c:
4991 Fix buffer_alloc in ximagesink
4992 Remove some useless debug info that reported wrong image sizes.
4993 When upstream does not accept out suggested size, fall back to allocating an
4994 image of the requested width/height instead of the currently configured size.
4995 The problem is that an image is reused from the pool because the width/height
4996 match but the caps on the new buffer are the requested caps with possibly
4997 different height/width resulting in errors.
4999 2009-02-02 12:54:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5001 * gst/playback/gstdecodebin2.c:
5002 * gst/playback/gsturidecodebin.c:
5003 Fix documentation for autoplug-select
5004 fix the documentation strings for the autoplug-select signal.
5007 2009-02-02 10:09:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5009 * gst-libs/gst/rtsp/gstrtspmessage.c:
5010 Fix string leak in rtspmessage
5011 when we remove a header field from a message we must free the value associated
5012 with the key to avoid a memory leak.
5014 2009-01-31 18:45:47 +0200 Stefan Kost <ensonic@users.sf.net>
5016 * docs/libs/gst-plugins-base-libs-docs.sgml:
5017 Its "Base Library" and not just "Library".
5019 2009-01-31 18:44:32 +0200 Stefan Kost <ensonic@users.sf.net>
5021 * gst-libs/gst/audio/gstaudiofilter.c:
5022 Link to the class, as we can't link to the members yet.
5024 2009-01-30 17:48:23 -0800 Michael Smith <msmith@songbirdnest.com>
5026 * gst/playback/gstplaybin2.c:
5027 Remove pad-removed handlers after setting the decodebins to NULL.
5028 They do needed cleanup; without this we leak selector requestpads.
5030 2009-01-30 17:47:07 -0800 Michael Smith <msmith@songbirdnest.com>
5032 * gst/playback/gstplaybin2.c:
5033 Unref selector request pad even if we no longer have a selector.
5034 During destruction, we won't have a selector any more, but we still need
5035 to unref the pad to avoid leaking it.
5037 2009-01-30 15:23:23 -0800 Michael Smith <msmith@songbirdnest.com>
5039 * gst/playback/gstplaybin2.c:
5040 Unref source in playbin2's finalize method
5042 2009-01-30 12:04:01 -0800 Michael Smith <msmith@songbirdnest.com>
5044 * gst/playback/gstplaysink.c:
5045 Fix more leaks of pads and elements in gstplaysink.
5046 Don't keep extra references to volume and mute elements; we don't need
5048 Ensure we unref pads that we have references to, and release request
5051 2009-01-30 11:04:37 -0800 Michael Smith <msmith@songbirdnest.com>
5053 * gst/playback/gstplaysink.c:
5054 Avoid leaking all playsinks. Fix some internal leaks.
5055 Playsink was holding references to itself. Don't do that, it's not cool.
5056 Also, free all chains in dispose.
5058 2009-01-30 10:54:12 -0800 Michael Smith <msmith@songbirdnest.com>
5060 * gst/playback/gstplaybin2.c:
5061 Unref peer request pad after releasing it, since we hold a reference.
5063 2009-01-30 10:52:52 -0800 Michael Smith <msmith@songbirdnest.com>
5065 * gst/playback/gstplaybin2.c:
5066 Fix caps leak in playbin2.
5068 2009-01-30 10:51:11 -0800 Michael Smith <msmith@songbirdnest.com>
5070 * gst/playback/gstplaybin2.c:
5071 Unref active pad from selector when finding active stream.
5073 2009-01-30 10:49:55 -0800 Michael Smith <msmith@songbirdnest.com>
5075 * gst/playback/gstplaybin2.c:
5076 Free uris when finalizing playbin2 instance.
5078 2009-01-30 10:38:17 -0800 Michael Smith <msmith@songbirdnest.com>
5080 * gst/playback/gsturidecodebin.c:
5081 Unref pads when iterating over them in analyse_source.
5082 Fixes leak of source's srcpad when using uridecodebin.
5084 2009-01-30 22:22:07 +0200 Stefan Kost <ensonic@users.sf.net>
5086 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
5087 Add releaseinfo with online url.
5089 2009-01-30 17:58:15 +0000 Jan Schmidt <jan.schmidt@sun.com>
5091 * gst/playback/gstplaybasebin.c:
5092 Fix compilation warning on Forte
5094 2009-01-30 17:16:39 +0000 Jan Schmidt <jan.schmidt@sun.com>
5096 * gst/adder/gstadder.c:
5097 Don't do void pointer arithmetic.
5099 2009-01-30 17:25:51 +0000 Jan Schmidt <thaytan@noraisin.net>
5104 2009-01-30 08:50:53 +0100 Edward Hervey <bilboed@bilboed.com>
5108 Use a symbolic link for the pre-commit client-side hook
5110 2009-01-30 08:12:42 +0100 Edward Hervey <bilboed@bilboed.com>
5113 Add more files/directories to ignore
5115 2009-01-29 14:00:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5117 * gst-libs/gst/rtsp/gstrtspdefs.c:
5119 Fix some typos in the doc string of the new
5120 gst_rtsp_options_as_string() method.
5122 2009-01-29 11:55:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5124 * docs/libs/gst-plugins-base-libs-sections.txt:
5125 * gst-libs/gst/rtsp/gstrtspconnection.c:
5126 * gst-libs/gst/rtsp/gstrtspmessage.c:
5127 * gst-libs/gst/rtsp/gstrtspmessage.h:
5128 Add new RTSP message method to set header
5129 Add gst_rtsp_message_take_header() that takes ownership of the passed header
5130 value. This allows us to avoid an allocations and memory copy in some
5132 API: GstRTSPMessage::gst_rtsp_message_take_header()
5134 2009-01-29 11:51:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5136 * docs/libs/gst-plugins-base-libs-sections.txt:
5137 Add new method to docs
5138 Add the new gst_rtsp_options_as_text() method to the docs.
5140 2009-01-28 11:48:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5142 * gst-libs/gst/rtsp/gstrtspdefs.c:
5143 * gst-libs/gst/rtsp/gstrtspdefs.h:
5144 Add method to serialize RTSP options
5145 Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
5147 API: GstRTSP::gst_rtsp_options_as_text()
5149 2009-01-26 17:59:37 -0800 Michael Smith <msmith@songbirdnest.com>
5151 * gst/typefind/gsttypefindfunctions.c:
5152 Ensure we have sufficient data when using data scan contexts.
5153 Fixes crashes typefinding things that look like they might contain AAC
5154 data (but probably aren't actually AAC).
5156 2009-01-26 23:32:09 +0000 Jan Schmidt <thaytan@noraisin.net>
5158 * ext/gio/Makefile.am:
5159 Fix include order for gio plugin
5161 2009-01-23 23:59:48 +0000 Jan Schmidt <thaytan@noraisin.net>
5163 * win32/common/config.h:
5164 Update win32 config.h for 0.10.22.1 dev cycle
5166 2009-01-23 23:16:11 +0000 Jan Schmidt <thaytan@noraisin.net>
5169 * docs/libs/.gitignore:
5170 * gst-libs/gst/audio/.gitignore:
5171 * gst-libs/gst/video/.gitignore:
5173 * tests/examples/dynamic/.gitignore:
5174 Extend and clean up git ignores
5176 2009-01-23 12:31:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5179 * docs/plugins/Makefile.am:
5180 * docs/plugins/gst-plugins-base-plugins-sections.txt:
5181 * docs/plugins/gst-plugins-base-plugins.args:
5182 * docs/plugins/gst-plugins-base-plugins.hierarchy:
5183 * docs/plugins/gst-plugins-base-plugins.interfaces:
5184 * docs/plugins/gst-plugins-base-plugins.prerequisites:
5185 * docs/plugins/inspect/plugin-adder.xml:
5186 * docs/plugins/inspect/plugin-alsa.xml:
5187 * docs/plugins/inspect/plugin-app.xml:
5188 * docs/plugins/inspect/plugin-audioconvert.xml:
5189 * docs/plugins/inspect/plugin-audiorate.xml:
5190 * docs/plugins/inspect/plugin-audioresample.xml:
5191 * docs/plugins/inspect/plugin-audiotestsrc.xml:
5192 * docs/plugins/inspect/plugin-cdparanoia.xml:
5193 * docs/plugins/inspect/plugin-decodebin.xml:
5194 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
5195 * docs/plugins/inspect/plugin-gdp.xml:
5196 * docs/plugins/inspect/plugin-gio.xml:
5197 * docs/plugins/inspect/plugin-gnomevfs.xml:
5198 * docs/plugins/inspect/plugin-libvisual.xml:
5199 * docs/plugins/inspect/plugin-ogg.xml:
5200 * docs/plugins/inspect/plugin-pango.xml:
5201 * docs/plugins/inspect/plugin-playback.xml:
5202 * docs/plugins/inspect/plugin-queue2.xml:
5203 * docs/plugins/inspect/plugin-subparse.xml:
5204 * docs/plugins/inspect/plugin-tcp.xml:
5205 * docs/plugins/inspect/plugin-theora.xml:
5206 * docs/plugins/inspect/plugin-typefindfunctions.xml:
5207 * docs/plugins/inspect/plugin-uridecodebin.xml:
5208 * docs/plugins/inspect/plugin-video4linux.xml:
5209 * docs/plugins/inspect/plugin-videorate.xml:
5210 * docs/plugins/inspect/plugin-videoscale.xml:
5211 * docs/plugins/inspect/plugin-videotestsrc.xml:
5212 * docs/plugins/inspect/plugin-volume.xml:
5213 * docs/plugins/inspect/plugin-vorbis.xml:
5214 * docs/plugins/inspect/plugin-ximagesink.xml:
5215 * docs/plugins/inspect/plugin-xvimagesink.xml:
5216 * gst/audioresample/Makefile.am:
5217 * gst/audioresample/README:
5218 * gst/audioresample/arch.h:
5219 * gst/audioresample/buffer.c:
5220 * gst/audioresample/buffer.h:
5221 * gst/audioresample/debug.c:
5222 * gst/audioresample/debug.h:
5223 * gst/audioresample/fixed_arm4.h:
5224 * gst/audioresample/fixed_arm5e.h:
5225 * gst/audioresample/fixed_bfin.h:
5226 * gst/audioresample/fixed_debug.h:
5227 * gst/audioresample/fixed_generic.h:
5228 * gst/audioresample/functable.c:
5229 * gst/audioresample/functable.h:
5230 * gst/audioresample/gstaudioresample.c:
5231 * gst/audioresample/gstaudioresample.h:
5232 * gst/audioresample/resample.c:
5233 * gst/audioresample/resample.h:
5234 * gst/audioresample/resample_chunk.c:
5235 * gst/audioresample/resample_functable.c:
5236 * gst/audioresample/resample_ref.c:
5237 * gst/audioresample/resample_sse.h:
5238 * gst/audioresample/speex_resampler.h:
5239 * gst/audioresample/speex_resampler_double.c:
5240 * gst/audioresample/speex_resampler_float.c:
5241 * gst/audioresample/speex_resampler_int.c:
5242 * gst/audioresample/speex_resampler_wrapper.h:
5243 * gst/speexresample/Makefile.am:
5244 * gst/speexresample/README:
5245 * gst/speexresample/arch.h:
5246 * gst/speexresample/fixed_arm4.h:
5247 * gst/speexresample/fixed_arm5e.h:
5248 * gst/speexresample/fixed_bfin.h:
5249 * gst/speexresample/fixed_debug.h:
5250 * gst/speexresample/fixed_generic.h:
5251 * gst/speexresample/gstspeexresample.c:
5252 * gst/speexresample/gstspeexresample.h:
5253 * gst/speexresample/resample.c:
5254 * gst/speexresample/resample_sse.h:
5255 * gst/speexresample/speex_resampler.h:
5256 * gst/speexresample/speex_resampler_double.c:
5257 * gst/speexresample/speex_resampler_float.c:
5258 * gst/speexresample/speex_resampler_int.c:
5259 * gst/speexresample/speex_resampler_wrapper.h:
5260 * gst/typefind/gsttypefindfunctions.c:
5261 * tests/check/Makefile.am:
5262 * tests/check/elements/audioresample.c:
5263 * tests/check/elements/speexresample.c:
5264 Rename files and types from speexresample to audioresample
5265 Rename files and types from speexresample to audioresample
5266 to finish the move and to prevent any confusion.
5268 2009-01-23 11:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5270 * sys/xvimage/xvimagesink.c:
5271 Add some more debugging to the Xv strides
5272 Add some more debugging to the strides as they are received from the server and
5273 the expected strides.
5275 2009-01-23 11:40:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5277 * gst/typefind/gsttypefindfunctions.c:
5278 Add typefind function for gsm
5279 Because core now supports typefindfactories without a typefind function we can
5280 register a factory fo GSM that will --if all else fails-- assume the file is a
5281 GSM file based on the registered extension.
5284 2009-01-23 11:37:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5286 * gst/playback/gsturidecodebin.c:
5287 Use more performant link function
5288 We can use gst_element_link_pads() instead of the more generic
5289 gst_element_link() function because we know the pads. This saves some cycles
5290 because the more generic function needs to search for possible compatible caps
5293 2009-01-23 11:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5295 * gst-libs/gst/riff/riff-ids.h:
5296 * gst-libs/gst/riff/riff-media.c:
5297 Add more codec ids for RIFF formats
5298 Handle codec ID for various other AAC formats.
5299 Sync the list of possible codec ids with that of ffmpeg.
5302 2009-01-23 11:27:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5304 * ext/theora/theoradec.c:
5305 Use rounded values for image strides and sizes
5306 Round up the height before calculating the expected size and
5307 strides of the output image.
5309 2009-01-23 11:23:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5311 * ext/alsa/gstalsasink.c:
5312 Improve debug message
5313 Improve the debug message when alsa returns an error.
5315 2009-01-23 11:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5317 * gst-libs/gst/app/gstappsrc.c:
5318 Reset queued_bytes counter when flushing
5319 Set the amount of queued bytes in the internal queue back to 0 when we clear the
5323 2009-01-23 10:19:27 +0100 Benjamin Gaignard <benjamin@gaignard.net>
5325 * gst/typefind/gsttypefindfunctions.c:
5326 Add typefinder for Mobile XMF. Fixes bug #568707.
5328 2009-01-23 10:00:11 +0100 Brian Cameron <brian.cameron@sun.com>
5331 Fix linking on Solaris. Fixes bug #568482.
5332 Check for nsl and socket libraries and add them to
5333 LIBS if they're found. They're needed for socket()
5334 and gethostbyname() on Solaris.
5336 2009-01-22 22:09:47 +0000 Jan Schmidt <thaytan@noraisin.net>
5338 * gst/playback/gstplaybasebin.c:
5339 Fix use-after-unref problem noticed by Josep Torra Valles, and run
5342 2009-01-22 17:46:59 +0200 Stefan Kost <ensonic@users.sf.net>
5345 Update common snapshot.
5347 2009-01-22 13:47:24 +0100 Sebastian Dröge <slomo@circular-chaos.org>
5352 2009-01-22 13:12:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5354 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
5356 2009-01-22 10:14:28 +0100 Sebastian Dröge <slomo@circular-chaos.org>
5358 * gst-libs/gst/fft/gstfftf32.c:
5359 * gst-libs/gst/fft/gstfftf64.c:
5360 * gst-libs/gst/fft/gstffts16.c:
5361 * gst-libs/gst/fft/gstffts32.c:
5362 Reduce the number of allocations for creating FFT contexts
5363 Reduce the number of allocations from 2 to 1 for every FFT
5364 context by allocating enough memory for the FFT context
5365 and passing parts of it to the kissfft allocation functions.
5367 2009-01-22 11:32:56 +0000 Jan Schmidt <thaytan@noraisin.net>
5370 Back to devel -> 0.10.22.1
5372 2009-01-22 05:57:53 +0100 Edward Hervey <bilboed@bilboed.com>
5376 Install and use pre-commit indentation hook from common
5378 2009-01-21 13:09:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5380 * gst-libs/gst/rtp/gstrtpbuffer.c:
5381 * tests/check/libs/rtp.c:
5382 Avoid overflows in the padding checks by doing the check slightly
5384 Add a unit test to check for correct behaviour.
5386 2009-01-21 04:31:32 +0100 Edward Hervey <bilboed@bilboed.com>
5389 autogen.sh : Use git submodule
5391 === release 0.10.22 ===
5393 2009-01-19 23:10:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5399 * docs/plugins/gst-plugins-base-plugins.hierarchy:
5400 * docs/plugins/gst-plugins-base-plugins.interfaces:
5401 * docs/plugins/gst-plugins-base-plugins.prerequisites:
5402 * docs/plugins/inspect/plugin-adder.xml:
5403 * docs/plugins/inspect/plugin-alsa.xml:
5404 * docs/plugins/inspect/plugin-app.xml:
5405 * docs/plugins/inspect/plugin-audioconvert.xml:
5406 * docs/plugins/inspect/plugin-audiorate.xml:
5407 * docs/plugins/inspect/plugin-audioresample.xml:
5408 * docs/plugins/inspect/plugin-audiotestsrc.xml:
5409 * docs/plugins/inspect/plugin-cdparanoia.xml:
5410 * docs/plugins/inspect/plugin-decodebin.xml:
5411 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
5412 * docs/plugins/inspect/plugin-gdp.xml:
5413 * docs/plugins/inspect/plugin-gnomevfs.xml:
5414 * docs/plugins/inspect/plugin-libvisual.xml:
5415 * docs/plugins/inspect/plugin-ogg.xml:
5416 * docs/plugins/inspect/plugin-pango.xml:
5417 * docs/plugins/inspect/plugin-playback.xml:
5418 * docs/plugins/inspect/plugin-queue2.xml:
5419 * docs/plugins/inspect/plugin-subparse.xml:
5420 * docs/plugins/inspect/plugin-tcp.xml:
5421 * docs/plugins/inspect/plugin-theora.xml:
5422 * docs/plugins/inspect/plugin-typefindfunctions.xml:
5423 * docs/plugins/inspect/plugin-uridecodebin.xml:
5424 * docs/plugins/inspect/plugin-video4linux.xml:
5425 * docs/plugins/inspect/plugin-videorate.xml:
5426 * docs/plugins/inspect/plugin-videoscale.xml:
5427 * docs/plugins/inspect/plugin-videotestsrc.xml:
5428 * docs/plugins/inspect/plugin-volume.xml:
5429 * docs/plugins/inspect/plugin-vorbis.xml:
5430 * docs/plugins/inspect/plugin-ximagesink.xml:
5431 * docs/plugins/inspect/plugin-xvimagesink.xml:
5432 * gst-plugins-base.doap:
5462 * win32/common/config.h:
5464 Original commit message from CVS:
5467 2009-01-19 22:01:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5499 Original commit message from CVS:
5502 2009-01-16 11:44:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5504 gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
5505 Original commit message from CVS:
5506 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
5507 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
5508 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
5509 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
5510 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
5511 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
5512 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
5513 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
5514 Use correct struct alignment everywhere to prevent unaligned
5515 memory accesses, resulting in SIGBUS on sparc and probably others.
5518 2009-01-16 11:40:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5520 gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
5521 Original commit message from CVS:
5522 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
5523 Forward unknown events upstream to allow latency configuration.
5526 2009-01-13 14:47:19 +0000 Wim Taymans <wim.taymans@gmail.com>
5528 gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
5529 Original commit message from CVS:
5530 * gst/playback/gstplaybin2.c: (groups_set_locked_state):
5531 Provide the right arguments to a debug line.
5533 2009-01-13 06:51:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
5535 sys/xvimage/xvimagesink.c: Don't reset the colorkey when element is reused. Fixes #567511.
5536 Original commit message from CVS:
5537 * sys/xvimage/xvimagesink.c:
5538 Don't reset the colorkey when element is reused. Fixes #567511.
5540 2009-01-09 23:42:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5542 configure.ac: 0.10.21.3 pre-release
5543 Original commit message from CVS:
5545 0.10.21.3 pre-release
5547 2009-01-09 23:13:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5549 gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
5550 Original commit message from CVS:
5551 * gst-libs/gst/app/gstappsink.c:
5552 Store the returned signal id in the right slot when
5553 registering the pull-buffer signal.
5555 Spotted by: Thomas Vander Stichele <thomas at apestaart dot org>
5557 2009-01-09 17:17:50 +0000 Tim-Philipp Müller <tim@centricular.net>
5559 gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
5560 Original commit message from CVS:
5561 * gst-libs/gst/interfaces/mixer.c:
5562 Small docs addition to clarify that one really mustn't free
5563 the constant GList returned (#566812).
5565 2009-01-08 17:18:24 +0000 Wim Taymans <wim.taymans@gmail.com>
5567 Add GType for GstRTSPUrl and expose a copy function because we can.
5568 Original commit message from CVS:
5569 * docs/libs/gst-plugins-base-libs-sections.txt:
5570 * gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
5571 (gst_rtsp_url_get_type), (gst_rtsp_url_copy):
5572 * gst-libs/gst/rtsp/gstrtspurl.h:
5573 * win32/common/libgstrtsp.def:
5574 Add GType for GstRTSPUrl and expose a copy function because we can.
5575 API: gst_rtsp_url_copy()
5578 2009-01-07 18:36:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5580 Add plugin dependency for the GIO and GVfs modules.
5581 Original commit message from CVS:
5583 * ext/gio/gstgio.c: (plugin_init):
5584 Add plugin dependency for the GIO and GVfs modules.
5587 2009-01-07 18:32:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5589 Add plugin dependency for the gnomevfs modules.
5590 Original commit message from CVS:
5592 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
5593 Add plugin dependency for the gnomevfs modules.
5596 2009-01-07 18:30:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5598 win32/common/libgstcdda.def: Add new symbol to the list of exported symbols.
5599 Original commit message from CVS:
5600 * win32/common/libgstcdda.def:
5601 Add new symbol to the list of exported symbols.
5603 2009-01-07 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
5605 gst/playback/gstplaybin2.c: Fix some comments and docs.
5606 Original commit message from CVS:
5607 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
5608 (gst_play_bin_set_uri), (gst_play_bin_set_suburi),
5609 (no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
5610 (activate_group), (deactivate_group), (groups_set_locked_state),
5611 (gst_play_bin_change_state):
5612 Fix some comments and docs.
5613 Post an error message when we fail to link the selector to the sink.
5614 Remove pushing of EOS, this seems unneeded.
5615 Lock the state of deactivated groups so that they don't accidentally
5616 reactivate when the playbin2 state changes.
5617 Reuse uridecodebins.
5618 Unlock and relock state of groups when playbin goes to NULL.
5621 * gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
5622 Only do something in the pad removed callback when we are dealing with
5623 our sourcepads because the sinkpads don't have a ghostpad.
5625 2009-01-07 10:50:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5627 gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
5628 Original commit message from CVS:
5629 * gst-libs/gst/cdda/gstcddabasesrc.c:
5630 * gst-libs/gst/cdda/gstcddabasesrc.h:
5631 Make the GType of GstCDDABaseSrcMode public for bindings.
5634 2009-01-06 18:03:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5636 Use new core API to make registry re-scan the plugin whenever visualisations are added or removed (see #350477).
5637 Original commit message from CVS:
5639 * ext/libvisual/visual.c: (plugin_init):
5640 Use new core API to make registry re-scan the plugin
5641 whenever visualisations are added or removed (see #350477).
5643 2009-01-06 17:30:31 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
5645 gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
5646 Original commit message from CVS:
5647 Patch by: José Alburquerque <jaalburqu svn gnome org>
5648 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
5649 * gst-libs/gst/audio/gstaudioclock.h:
5650 Make gst_audio_clock_new use const gchar* to ease the wrapping of
5651 C++ bindings. Fixes #566723.
5653 2009-01-06 12:16:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5655 Add pkg-config files for libgstapp. Fixes bug #566761.
5656 Original commit message from CVS:
5658 * pkgconfig/Makefile.am:
5659 * pkgconfig/gstreamer-app-uninstalled.pc.in:
5660 * pkgconfig/gstreamer-app.pc.in:
5661 Add pkg-config files for libgstapp. Fixes bug #566761.
5663 2009-01-06 11:10:29 +0000 Tim-Philipp Müller <tim@centricular.net>
5665 gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
5666 Original commit message from CVS:
5667 * gst-libs/gst/app/gstappsink.c:
5668 * gst-libs/gst/app/gstappsink.h:
5669 * gst-libs/gst/app/gstappsrc.c:
5670 * gst-libs/gst/app/gstappsrc.h:
5671 Make debug categories static. Use _element_class_set_details_simple().
5673 2009-01-06 10:56:45 +0000 Tim-Philipp Müller <tim@centricular.net>
5675 gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
5676 Original commit message from CVS:
5677 * gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
5678 (gst_app_sink_class_init), (gst_app_sink_init),
5679 (gst_app_sink_dispose), (gst_app_sink_finalize),
5680 (gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
5681 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
5682 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
5683 (gst_app_sink_render), (gst_app_sink_getcaps),
5684 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
5685 (gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
5686 (gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
5687 (gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
5688 (gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
5689 (gst_app_sink_pull_buffer)::
5690 * gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
5691 * gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
5692 (gst_app_src_class_init), (gst_app_src_init),
5693 (gst_app_src_flush_queued), (gst_app_src_dispose),
5694 (gst_app_src_finalize), (gst_app_src_set_property),
5695 (gst_app_src_get_property), (gst_app_src_unlock),
5696 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
5697 (gst_app_src_is_seekable), (gst_app_src_check_get_range),
5698 (gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
5699 (gst_app_src_set_caps), (gst_app_src_get_caps),
5700 (gst_app_src_set_size), (gst_app_src_get_size),
5701 (gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
5702 (gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
5703 (gst_app_src_set_latencies), (gst_app_src_set_latency),
5704 (gst_app_src_get_latency), (gst_app_src_push_buffer_full),
5705 (gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
5706 * gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
5707 Move private data into a private instance struct. Add padding to
5708 instance and class structures exposed in public headers. Add
5709 Since markers to the gtk-doc blurbs (#566750).
5711 2009-01-06 10:50:37 +0000 Wim Taymans <wim.taymans@gmail.com>
5713 tests/examples/app/appsrc_ex.c: Some comments.
5714 Original commit message from CVS:
5715 * tests/examples/app/appsrc_ex.c: (main):
5717 When pulling a buffer we can get NULL when the element is EOS, don't try
5718 to unref this NULL buffer.
5720 2009-01-06 10:16:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5722 gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
5723 Original commit message from CVS:
5724 * gst-libs/gst/video/Makefile.am:
5725 * gst-libs/gst/video/video.h:
5726 Fix up build flags and include statement for the new generated
5727 enumtypes files, to fix dist.
5729 2009-01-05 23:04:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5731 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
5732 Original commit message from CVS:
5734 * docs/libs/Makefile.am:
5735 * docs/libs/gst-plugins-base-libs-docs.sgml:
5736 * docs/libs/gst-plugins-base-libs-sections.txt:
5737 * docs/plugins/Makefile.am:
5738 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
5739 * docs/plugins/gst-plugins-base-plugins-sections.txt:
5740 * docs/plugins/gst-plugins-base-plugins.args:
5741 * docs/plugins/gst-plugins-base-plugins.hierarchy:
5742 * docs/plugins/gst-plugins-base-plugins.interfaces:
5743 * docs/plugins/gst-plugins-base-plugins.prerequisites:
5744 * docs/plugins/gst-plugins-base-plugins.signals:
5745 * docs/plugins/inspect/plugin-app.xml:
5746 * gst-libs/gst/Makefile.am:
5747 * gst-libs/gst/app/gstappsink.c:
5748 * gst-libs/gst/app/gstappsrc.c:
5749 * tests/examples/Makefile.am:
5750 * tests/examples/app/Makefile.am:
5751 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
5753 2009-01-05 17:13:13 +0000 Wim Taymans <wim.taymans@gmail.com>
5755 gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
5756 Original commit message from CVS:
5757 * gst-libs/gst/audio/gstbaseaudiosink.c:
5758 (gst_base_audio_sink_change_state):
5759 Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
5760 take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
5761 this because the async_play method is deprecated and usually not called
5764 2009-01-05 12:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
5766 gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
5767 Original commit message from CVS:
5768 * gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
5769 Disconnect signal handlers before destroying a previous decodebin so
5770 that we don't end up causing deadlocks. Fixes #566586.
5772 2009-01-05 10:59:35 +0000 Wim Taymans <wim.taymans@gmail.com>
5774 gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
5775 Original commit message from CVS:
5776 * gst/audiotestsrc/gstaudiotestsrc.c:
5777 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
5778 (gst_audio_test_src_check_get_range),
5779 (gst_audio_test_src_set_property),
5780 (gst_audio_test_src_get_property):
5781 * gst/audiotestsrc/gstaudiotestsrc.h:
5782 Add property to control pull/push based scheduling.
5784 2009-01-02 15:04:13 +0000 Alessandro Decina <alessandro.d@gmail.com>
5786 Make the seek and colorkey examples depend on gtk+-x11 as they use
5787 Original commit message from CVS:
5789 * tests/examples/seek/Makefile.am:
5790 * tests/icles/Makefile.am:
5791 Make the seek and colorkey examples depend on gtk+-x11 as they use
5793 Fixes the build with gtk+-quartz.
5795 2008-12-31 16:04:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5797 win32/common/: Add new exports to win32 files.
5798 Original commit message from CVS:
5799 * win32/common/libgstaudio.def:
5800 * win32/common/libgsttag.def:
5801 * win32/common/libgstvideo.def:
5802 Add new exports to win32 files.
5804 2008-12-31 13:31:55 +0000 Edward Hervey <bilboed@bilboed.com>
5806 gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
5807 Original commit message from CVS:
5808 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
5809 * gst-libs/gst/tag/gsttagdemux.h:
5810 Add GType for GstTagDemuxResult enum.
5812 2008-12-31 13:01:30 +0000 Edward Hervey <bilboed@bilboed.com>
5814 gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
5815 Original commit message from CVS:
5816 * gst-libs/gst/video/Makefile.am:
5817 * gst-libs/gst/video/video.h:
5818 Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
5819 This will help bindings to use it.
5821 2008-12-31 11:20:26 +0000 Edward Hervey <bilboed@bilboed.com>
5823 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
5824 Original commit message from CVS:
5825 * gst-libs/gst/audio/Makefile.am:
5826 * gst-libs/gst/audio/audio.c:
5827 * gst-libs/gst/audio/multichannel.h:
5828 * gst-libs/gst/audio/testchannels.c:
5830 * win32/common/audio-enumtypes.c:
5831 (gst_audio_channel_position_get_type),
5832 (gst_ring_buffer_state_get_type),
5833 (gst_ring_buffer_seg_state_get_type),
5834 (gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
5835 * win32/common/audio-enumtypes.h:
5836 * win32/common/multichannel-enumtypes.c:
5837 * win32/common/multichannel-enumtypes.h:
5838 * win32/vs6/grammar.dsp:
5839 * win32/vs6/libgstaudio.dsp:
5840 * win32/vs7/libgstaudio.vcproj:
5841 * win32/vs8/libgstaudio.vcproj:
5842 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
5843 audio- in order to wrap all enums declarations of that library.
5844 This modification should not matter since that header file is not a
5845 public header (it will be included by public headers).
5846 Modify win32 crap^Wfiles accordingly.
5848 2008-12-30 17:55:07 +0000 Edward Hervey <bilboed@bilboed.com>
5850 gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
5851 Original commit message from CVS:
5852 * gst-libs/gst/audio/gstbaseaudiosrc.h:
5853 * gst-libs/gst/audio/gstbaseaudiosink.h:
5854 Complete Sebastien's commit from the 13th by exporting the
5855 _slave_method_get_type() methods.
5857 2008-12-29 16:45:20 +0000 Wim Taymans <wim.taymans@gmail.com>
5859 gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
5860 Original commit message from CVS:
5861 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
5862 (gst_app_src_init), (gst_app_src_set_property),
5863 (gst_app_src_get_property), (gst_app_src_query),
5864 (gst_app_src_set_latencies), (gst_app_src_set_latency),
5865 (gst_app_src_get_latency), (gst_app_src_push_buffer_full):
5866 * gst-libs/gst/app/gstappsrc.h:
5867 Add properties and methods to configure and retrieve the min and max
5870 2008-12-20 17:38:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5872 ext/: Implement URI query. Fixes bug #562949.
5873 Original commit message from CVS:
5874 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_query):
5875 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_class_init),
5876 (gst_gio_base_src_query):
5877 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_query):
5878 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
5879 (gst_gnome_vfs_src_query):
5880 Implement URI query. Fixes bug #562949.
5882 2008-12-20 12:48:43 +0000 Wim Taymans <wim.taymans@gmail.com>
5884 gst/playback/gstplaybin2.c: Add some debug info.
5885 Original commit message from CVS:
5886 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
5887 Add some debug info.
5888 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
5889 (gst_play_sink_reconfigure), (gst_play_sink_request_pad),
5890 (gst_play_sink_release_pad):
5891 Add some more debug info.
5892 Reconfigure the audio chain when we switch between raw and encoded audio
5893 in gapless playback.
5895 2008-12-20 12:45:03 +0000 Wim Taymans <wim.taymans@gmail.com>
5897 gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
5898 Original commit message from CVS:
5899 * gst-libs/gst/audio/gstbaseaudiosink.c:
5900 (gst_base_audio_sink_setcaps):
5901 Pause the write thread before deactivating and releasing the ringbuffer
5902 to avoid a deadlock when we do gapless playback with different sample
5903 rates in playbin2. Fixes #564929.
5905 2008-12-19 13:03:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5907 gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
5908 Original commit message from CVS:
5909 * gst-libs/gst/audio/gstbaseaudiosrc.c:
5910 Make GstAudioSrcSlaveMethod get_type() function non-static
5912 * win32/common/libgstaudio.def:
5913 * win32/common/libgstnetbuffer.def:
5914 Add some missing functions to the list of exported symbols.
5916 2008-12-18 12:37:33 +0000 Andrew Feren <acferen@yahoo.com>
5918 gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
5919 Original commit message from CVS:
5920 Patch by: Andrew Feren <acferen at yahoo dot com>
5921 * gst-libs/gst/netbuffer/gstnetbuffer.c:
5922 (gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
5923 (gst_netaddress_get_address_bytes),
5924 (gst_netaddress_set_address_bytes):
5925 * gst-libs/gst/netbuffer/gstnetbuffer.h:
5926 Make gst_netaddress_get_ip4_address fail for v6 addresses.
5927 Make gst_netaddress_get_ip6_address either fail or return the v4
5928 address as a transitional v6 address.
5929 Add two convenience functions:
5930 API: gst_netaddress_get_address_bytes()
5931 API: gst_netaddress_set_address_bytes()
5934 2008-12-17 13:51:46 +0000 Wim Taymans <wim.taymans@gmail.com>
5936 Add appsrc and appsink documentation.
5937 Original commit message from CVS:
5938 * docs/plugins/Makefile.am:
5939 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
5940 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
5941 * gst-libs/gst/app/gstappsink.c:
5942 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
5943 Add appsrc and appsink documentation.
5945 2008-12-17 08:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
5947 gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
5948 Original commit message from CVS:
5949 * gst/adder/Makefile.am:
5950 * gst/adder/gstadder.c:
5951 Cleanup variable names to make the adder-loop easier to understand.
5952 Also try to use liboil to spee it up, but ifdef it out as it does not
5953 make any change for me (Intel pentim M (sse,sse2) please try on other
5956 2008-12-16 20:16:17 +0000 Wim Taymans <wim.taymans@gmail.com>
5958 Add minimal docs to make the remaining tcp elements show up.
5959 Original commit message from CVS:
5960 * docs/plugins/Makefile.am:
5961 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
5962 * docs/plugins/gst-plugins-base-plugins-sections.txt:
5963 * gst/tcp/gsttcpclientsink.c:
5964 * gst/tcp/gsttcpclientsrc.c:
5965 * gst/tcp/gsttcpserversrc.c:
5966 Add minimal docs to make the remaining tcp elements show up.
5969 2008-12-15 12:02:26 +0000 Wim Taymans <wim.taymans@gmail.com>
5971 examples/app/: Fix example to unref after emiting the push-buffer action.
5972 Original commit message from CVS:
5973 * examples/app/appsrc-ra.c: (feed_data):
5974 * examples/app/appsrc-seekable.c: (feed_data):
5975 * examples/app/appsrc-stream.c: (read_data):
5976 * examples/app/appsrc-stream2.c: (feed_data):
5977 Fix example to unref after emiting the push-buffer action.
5978 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
5979 (gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
5980 (gst_app_src_push_buffer_action):
5981 Don't take the ref on the buffer in push-buffer action because it's too
5982 awkward for bindings. Fixes #564482.
5984 2008-12-13 19:32:13 +0000 Tim-Philipp Müller <tim@centricular.net>
5986 win32/common/config.h: Update to CVS version.
5987 Original commit message from CVS:
5988 * win32/common/config.h:
5989 Update to CVS version.
5990 * win32/common/config.h.in:
5991 Hardcode path to plugin install helper exe, just like we hardcode
5992 the paths in core. Removes another source of VCS conflicts for
5993 people hacking gst-plugins-base on systems with autotools.
5995 2008-12-13 16:21:12 +0000 Edward Hervey <bilboed@bilboed.com>
5997 m4/Makefile.am: And a couple more .m4 that don't exist anymore with gettext 0.17
5998 Original commit message from CVS:
6000 And a couple more .m4 that don't exist anymore with gettext 0.17
6002 2008-12-13 12:41:56 +0000 Edward Hervey <bilboed@bilboed.com>
6004 m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we...
6005 Original commit message from CVS:
6007 inttypes.m4 hasn't been available since gettext-0.15, and since we now
6008 require gettext >= 0.17 ... we can remove it from the list of files to
6011 2008-12-13 06:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6013 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
6014 Original commit message from CVS:
6015 * gst-libs/gst/audio/gstbaseaudiosink.c:
6016 (gst_base_audio_sink_slave_method_get_type),
6017 (gst_base_audio_sink_class_init):
6018 * gst-libs/gst/audio/gstbaseaudiosink.h:
6019 * gst-libs/gst/audio/gstbaseaudiosrc.c:
6020 (gst_base_audio_src_slave_method_get_type),
6021 (gst_base_audio_src_class_init):
6022 * gst-libs/gst/audio/gstbaseaudiosrc.h:
6023 API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
6024 public API. This is needed for the C++ bindings to be able
6025 to use this base classes. Fixes bug #564200, #564206.
6027 2008-12-12 19:41:28 +0000 Edward Hervey <bilboed@bilboed.com>
6029 gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
6030 Original commit message from CVS:
6031 * gst-libs/gst/cdda/gstcddabasesrc.c:
6032 (gst_cdda_base_src_handle_event):
6033 Remove erroneous gst_buffer_ref().
6034 * tests/check/libs/rtp.c: (GST_START_TEST):
6035 Don't forget to unref the buffer once you're done with it.
6037 2008-12-12 13:06:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6039 gst/playback/: XRef to GstXOverlay.
6040 Original commit message from CVS:
6041 * gst/playback/gstplaybin.c:
6042 * gst/playback/gstplaybin2.c:
6043 XRef to GstXOverlay.
6045 2008-12-12 10:54:45 +0000 Edward Hervey <bilboed@bilboed.com>
6047 gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
6048 Original commit message from CVS:
6049 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
6050 Free the factory array when finalizing.
6051 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
6052 Use a GstStaticPadTemplate since the src pad caps are fixed.
6054 2008-12-12 07:17:21 +0000 Edward Hervey <bilboed@bilboed.com>
6056 ext/vorbis/vorbisenc.c: Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with pad templates.
6057 Original commit message from CVS:
6058 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_base_init),
6059 (gst_vorbis_enc_init):
6060 Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with
6063 2008-12-12 07:15:22 +0000 Edward Hervey <bilboed@bilboed.com>
6065 gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
6066 Original commit message from CVS:
6067 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
6068 (gst_riff_create_video_template_caps):
6069 Add mapping for VP6 in avi/riff.
6071 2008-12-11 15:49:12 +0000 Edward Hervey <bilboed@bilboed.com>
6073 gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
6074 Original commit message from CVS:
6075 * gst/subparse/samiparse.c: (sami_context_push_state),
6076 (sami_context_pop_state), (start_sami_element), (end_sami_element):
6077 Some versions of libxml seem to be very picky as to strict formatting
6078 of the input and never 'close' the final </body> tag.
6079 In order to fix that bad behaviour, we trigger the flushing of
6080 remaining data on both </body> and </sami>.
6083 2008-12-11 12:32:03 +0000 Guillaume Emont <guillaume@fluendo.com>
6085 gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
6086 Original commit message from CVS:
6087 Patch by: Guillaume Emont <guillaume at fluendo dot com>
6088 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
6089 Add typefinders for MS Word files and OS X .DS_Store files to
6090 prevent them to be recognized as MPEG files. Fixes bug #564098.
6092 2008-12-11 11:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
6094 gst/playback/gstplaysink.c: Add some more debug info.
6095 Original commit message from CVS:
6096 * gst/playback/gstplaysink.c: (gen_audio_chain),
6097 (gst_play_sink_reconfigure):
6098 Add some more debug info.
6099 Fix linking of just an encoded sink.
6100 Handle failure to create a sink chain more gracefully than crashing.
6102 2008-12-11 10:33:48 +0000 Wim Taymans <wim.taymans@gmail.com>
6104 tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
6105 Original commit message from CVS:
6106 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
6107 Pushing 10 buffers is enough to run the test.
6109 2008-12-11 10:28:43 +0000 Wim Taymans <wim.taymans@gmail.com>
6111 tests/examples/seek/seek.c: Hook up the SKIP seek flag.
6112 Original commit message from CVS:
6113 * tests/examples/seek/seek.c: (do_seek), (stop_cb),
6114 (skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
6116 Hook up the SKIP seek flag.
6118 2008-12-10 18:43:32 +0000 Wim Taymans <wim.taymans@gmail.com>
6120 gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
6121 Original commit message from CVS:
6122 * gst/playback/gstplaybin2.c: (pad_added_cb):
6123 Error out with a missing-plugin error when the input-selector was not
6125 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
6128 2008-12-10 17:39:32 +0000 Wim Taymans <wim.taymans@gmail.com>
6130 gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
6131 Original commit message from CVS:
6132 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
6133 (gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
6134 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
6135 (gst_play_sink_send_event), (gst_play_sink_change_state):
6137 Try to set the selected sink to READY before using it. This will allow
6138 for detection of incompatible formats sooner.
6139 Don't cause a fatal error when conversion elements are missing but post
6140 a missing-element message and a warning instead because things might
6141 still link and run fine.
6142 Simplyfy the construction of audio and video sink chains.
6144 2008-12-10 14:55:10 +0000 Wim Taymans <wim.taymans@gmail.com>
6146 ext/ogg/gstoggdemux.c: Use G_DEFINE_TYPE for the OggPad to get some threadsafe type init from glib.
6147 Original commit message from CVS:
6148 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
6149 (gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
6150 Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
6153 2008-12-10 08:19:13 +0000 Luis Menina <liberforce@freeside.fr>
6155 gst/: Include glib.h instead of a specific GLib header. Including single
6156 Original commit message from CVS:
6157 Patch by: Luis Menina <liberforce at freeside dot fr>
6158 * gst-libs/gst/floatcast/floatcast.h:
6159 * gst/typefind/gsttypefindfunctions.c:
6160 Include glib.h instead of a specific GLib header. Including single
6161 GLib headers is deprecated. Fixes bug #563904.
6163 2008-12-09 18:30:10 +0000 Julien Moutte <julien@moutte.net>
6165 gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
6166 Original commit message from CVS:
6167 2008-12-09 Julien Moutte <julien@fluendo.com>
6168 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
6169 Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
6171 2008-12-09 17:21:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6173 gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
6174 Original commit message from CVS:
6175 * gst-libs/gst/riff/riff-read.c:
6176 Fix handling of odd chunks in riff metadata.
6178 2008-12-08 18:44:22 +0000 Wim Taymans <wim.taymans@gmail.com>
6180 gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
6181 Original commit message from CVS:
6182 * gst/volume/gstvolume.c: (gst_volume_class_init),
6183 (volume_before_transform), (volume_transform_ip):
6184 Use new basetransform vmethod to reconfigure the dynamic properties and
6185 any pending volume/mute changes. Fixes #563508.
6187 2008-12-08 18:12:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6189 configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...
6190 Original commit message from CVS:
6192 First check for "theoraenc theoradec" and if that failed check
6193 for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
6194 deprecate the latter. Also linking on Windows fails with just "theora"
6195 and the version check would fail for the release candidates.
6198 2008-12-08 15:25:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6200 gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
6201 Original commit message from CVS:
6202 * gst/playback/gstdecodebin.c:
6203 * gst/playback/gstdecodebin2.c:
6204 Add basic docs to decodebin and link to decodebin from decodebin2.
6206 2008-12-08 12:08:32 +0000 Olivier Crete <tester@tester.ca>
6208 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
6209 Original commit message from CVS:
6210 Patch by: Olivier Crete <tester at tester ca>
6211 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
6212 * gst-libs/gst/rtp/gstrtcpbuffer.h:
6213 Implement gst_rtcp_packet_remove(). Fixes #563174.
6214 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
6215 Add unit test for some RTCP functions.
6217 2008-12-04 20:09:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6219 configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.
6220 Original commit message from CVS:
6222 Apparently AC_CONFIG_MACRO_DIR breaks when using more
6223 than one macro directory, reverting last change.
6225 2008-12-04 19:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6227 configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros.
6228 Original commit message from CVS:
6230 Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
6233 2008-12-03 17:47:44 +0000 Edward Hervey <bilboed@bilboed.com>
6235 sys/: Clear all flags on buffers returned from the image pool.
6236 Original commit message from CVS:
6237 * sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
6238 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
6239 Clear all flags on buffers returned from the image pool.
6242 2008-12-01 19:36:35 +0000 이문형 <iwings@gmail.com>
6244 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...
6245 Original commit message from CVS:
6246 Patch by: 이문형 <iwings at gmail dot com>
6247 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
6248 Don't forget to release the lock again if we bail out because some
6249 pad is flushing or we've reached EOS, otherwise things will lock up
6250 next time _push_buffer() is called (#562802).
6252 2008-11-29 13:31:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6254 Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
6255 Original commit message from CVS:
6256 Patch by: Cygwin Ports maintainer
6257 <yselkowitz at users dot sourceforge dot net>
6260 Require gettext 0.17 because older versions don't mix with libtool
6261 2.2. At build time an older gettext version will still work.
6264 2008-11-28 13:30:36 +0000 Christian Schaller <uraeus@gnome.org>
6267 * gst/speexresample/Makefile.am:
6269 Original commit message from CVS:
6272 2008-11-28 09:44:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6274 Update documentation of speexresample for the new element name.
6275 Original commit message from CVS:
6276 * docs/plugins/gst-plugins-base-plugins.args:
6277 * docs/plugins/gst-plugins-base-plugins.hierarchy:
6278 * docs/plugins/gst-plugins-base-plugins.interfaces:
6279 * docs/plugins/gst-plugins-base-plugins.prerequisites:
6280 * docs/plugins/inspect/plugin-videorate.xml:
6281 * gst/speexresample/gstspeexresample.c:
6282 Update documentation of speexresample for the new element name.
6284 2008-11-28 09:04:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6286 gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
6287 Original commit message from CVS:
6288 * gst/speexresample/README:
6289 Update README with the latest diff between the Speex resampler
6292 2008-11-28 08:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6294 gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
6295 Original commit message from CVS:
6296 * gst/speexresample/gstspeexresample.c: (plugin_init):
6297 Update the debug category from speex_resample to audioresample.
6299 2008-11-27 19:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6301 Remove audioresample files.
6302 Original commit message from CVS:
6303 * gst/audioresample/Makefile.am:
6304 * gst/audioresample/buffer.c:
6305 * gst/audioresample/buffer.h:
6306 * gst/audioresample/debug.c:
6307 * gst/audioresample/debug.h:
6308 * gst/audioresample/functable.c:
6309 * gst/audioresample/functable.h:
6310 * gst/audioresample/gstaudioresample.c:
6311 * gst/audioresample/gstaudioresample.h:
6312 * gst/audioresample/resample.c:
6313 * gst/audioresample/resample.h:
6314 * gst/audioresample/resample_chunk.c:
6315 * gst/audioresample/resample_functable.c:
6316 * gst/audioresample/resample_ref.c:
6317 * tests/check/elements/audioresample.c:
6318 Remove audioresample files.
6320 2008-11-27 17:04:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6322 docs/plugins/inspect/plugin-audioresample.xml: Regenerated for library filename change.
6323 Original commit message from CVS:
6324 * docs/plugins/inspect/plugin-audioresample.xml:
6325 Regenerated for library filename change.
6327 2008-11-27 16:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6329 Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
6330 Original commit message from CVS:
6332 * docs/plugins/Makefile.am:
6333 * docs/plugins/gst-plugins-base-plugins-sections.txt:
6334 * docs/plugins/gst-plugins-base-plugins.args:
6335 * docs/plugins/gst-plugins-base-plugins.hierarchy:
6336 * docs/plugins/gst-plugins-base-plugins.interfaces:
6337 * docs/plugins/gst-plugins-base-plugins.prerequisites:
6338 * docs/plugins/inspect/plugin-adder.xml:
6339 * docs/plugins/inspect/plugin-alsa.xml:
6340 * docs/plugins/inspect/plugin-audioconvert.xml:
6341 * docs/plugins/inspect/plugin-audiorate.xml:
6342 * docs/plugins/inspect/plugin-audioresample.xml:
6343 * docs/plugins/inspect/plugin-audiotestsrc.xml:
6344 * docs/plugins/inspect/plugin-cdparanoia.xml:
6345 * docs/plugins/inspect/plugin-decodebin.xml:
6346 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
6347 * docs/plugins/inspect/plugin-gdp.xml:
6348 * docs/plugins/inspect/plugin-gio.xml:
6349 * docs/plugins/inspect/plugin-gnomevfs.xml:
6350 * docs/plugins/inspect/plugin-libvisual.xml:
6351 * docs/plugins/inspect/plugin-ogg.xml:
6352 * docs/plugins/inspect/plugin-pango.xml:
6353 * docs/plugins/inspect/plugin-playback.xml:
6354 * docs/plugins/inspect/plugin-queue2.xml:
6355 * docs/plugins/inspect/plugin-subparse.xml:
6356 * docs/plugins/inspect/plugin-tcp.xml:
6357 * docs/plugins/inspect/plugin-theora.xml:
6358 * docs/plugins/inspect/plugin-typefindfunctions.xml:
6359 * docs/plugins/inspect/plugin-uridecodebin.xml:
6360 * docs/plugins/inspect/plugin-video4linux.xml:
6361 * docs/plugins/inspect/plugin-videorate.xml:
6362 * docs/plugins/inspect/plugin-videoscale.xml:
6363 * docs/plugins/inspect/plugin-videotestsrc.xml:
6364 * docs/plugins/inspect/plugin-volume.xml:
6365 * docs/plugins/inspect/plugin-vorbis.xml:
6366 * docs/plugins/inspect/plugin-ximagesink.xml:
6367 * docs/plugins/inspect/plugin-xvimagesink.xml:
6368 * gst/speexresample/gstspeexresample.c: (plugin_init):
6369 * gst/speexresample/Makefile.am:
6370 * tests/check/Makefile.am:
6371 * tests/check/elements/speexresample.c: (setup_speexresample),
6372 (GST_START_TEST), (test_pipeline):
6373 Rename the moved speexresample to audioresample, integrate into the
6374 build system and remove the old audioresample from the build system.
6375 Fixes bug #558124, #385061, #346218, #116051.
6377 2008-11-27 16:47:41 +0000 Wim Taymans <wim.taymans@gmail.com>
6379 gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
6380 Original commit message from CVS:
6381 * gst-libs/gst/audio/gstbaseaudiosrc.c:
6382 (gst_base_audio_src_get_offset), (gst_base_audio_src_create):
6383 Avoid nasty int overflows after about 12 hours and 25 minutes when these
6384 code paths are triggered.
6385 A free beer to Håvard Graff for finding this!
6387 2008-11-27 11:16:44 +0000 이문형 <iwings@gmail.com>
6389 gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
6390 Original commit message from CVS:
6391 Patch by: 이문형 <iwings at gmail dot com>
6392 * gst-libs/gst/rtsp/gstrtspconnection.c:
6393 (gst_rtsp_connection_connect):
6394 A successful gst_poll_wait() doesn't always mean successful connect() on
6395 Windows. We should check errors by calling gst_poll_fd_has_error().
6398 2008-11-25 16:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6400 tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.
6401 Original commit message from CVS:
6402 * tests/check/elements/speexresample.c: (test_pipeline):
6403 Make unit test again faster to prevent timeouts with valgrind.
6405 2008-11-25 15:33:30 +0000 Wim Taymans <wim.taymans@gmail.com>
6407 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
6408 Original commit message from CVS:
6409 * gst-libs/gst/rtp/gstrtcpbuffer.c:
6410 Fix typo in the docs.
6412 2008-11-25 15:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
6414 ext/ogg/gstoggdemux.c: If no stream was found before receiving EOS, post an error message.
6415 Original commit message from CVS:
6416 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
6417 If no stream was found before receiving EOS, post an error message.
6420 2008-11-25 15:14:30 +0000 Wim Taymans <wim.taymans@gmail.com>
6422 ext/theora/: Parse segment events.
6423 Original commit message from CVS:
6424 * ext/theora/gsttheoraenc.h:
6425 * ext/theora/theoraenc.c: (gst_theora_enc_init),
6426 (theora_buffer_from_packet), (theora_push_packet),
6427 (theora_enc_sink_event), (theora_enc_is_discontinuous),
6429 Parse segment events.
6430 Pass incomming buffer timestamps to outgoing buffers.
6431 Use the running_time to construct the granulepos.
6434 2008-11-25 11:00:55 +0000 Wim Taymans <wim.taymans@gmail.com>
6436 gst/playback/gstplaybin2.c: Fix buffer-duration property.
6437 Original commit message from CVS:
6438 * gst/playback/gstplaybin2.c: (activate_group):
6439 Fix buffer-duration property.
6441 2008-11-25 10:32:49 +0000 Wim Taymans <wim.taymans@gmail.com>
6443 gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
6444 Original commit message from CVS:
6445 * gst-libs/gst/audio/gstbaseaudiosink.c:
6446 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
6447 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
6448 (gst_base_audio_sink_change_state):
6449 Really fix audiosink drain handling by keeping track of the running_time
6452 2008-11-24 20:25:24 +0000 Michael Smith <msmith@xiph.org>
6454 gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
6455 Original commit message from CVS:
6456 * gst/playback/gstplaybin2.c:
6457 Add notification of current stream. Add ability to configure buffer
6459 * gst/playback/gsturidecodebin.c:
6460 Add ability to configure buffer sizes for streaming mode.
6463 2008-11-24 20:11:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6465 gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
6466 Original commit message from CVS:
6467 * gst-libs/gst/audio/gstbaseaudiosink.c:
6468 Time is already in running_time. Remove base_time handling. Fixes
6469 audiosinks not draining and thus chopping some audio in the end.
6471 2008-11-24 19:18:59 +0000 David Schleef <ds@schleef.org>
6473 ext/ogg/gstoggmux.*: If we're muxing a dirac stream, flush the page after every picture.
6474 Original commit message from CVS:
6475 * ext/ogg/gstoggmux.c:
6476 * ext/ogg/gstoggmux.h:
6477 If we're muxing a dirac stream, flush the page after every picture.
6479 2008-11-24 12:56:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6481 gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
6482 Original commit message from CVS:
6483 * gst-libs/gst/audio/gstbaseaudiosink.c:
6484 Add one log message to check for audio_drained. Sync one log message
6485 with the condition. Send EOS after draining audio in pull mode.
6487 2008-11-24 12:07:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6489 ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr...
6490 Original commit message from CVS:
6491 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
6492 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
6493 Use gst_buffer_try_new_and_alloc() and fail properly if the
6494 allocation failed. This prevents abort() if downstream elements
6495 request an insane amount of memory.
6497 2008-11-24 12:03:11 +0000 Jon Trowbridge <trow@ximian.com>
6499 gst/volume/gstvolume.*: Cleanup volume, define and use default values.
6500 Original commit message from CVS:
6501 * gst/volume/gstvolume.c: (volume_choose_func),
6502 (volume_update_volume), (gst_volume_set_volume),
6503 (gst_volume_get_volume), (gst_volume_set_mute),
6504 (gst_volume_class_init), (gst_volume_init),
6505 (volume_process_double), (volume_process_float),
6506 (volume_process_int32), (volume_process_int32_clamp),
6507 (volume_process_int24), (volume_process_int24_clamp),
6508 (volume_process_int16), (volume_process_int16_clamp),
6509 (volume_process_int8), (volume_process_int8_clamp), (volume_setup),
6510 (volume_transform_ip), (volume_set_property),
6511 (volume_get_property):
6512 * gst/volume/gstvolume.h:
6513 Cleanup volume, define and use default values.
6514 Recalculate new volume and mute setup before processing. Fixes #561789.
6515 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
6516 Add controller unit test. Patch by: Jonathan Matthew
6517 Fix bogus test that messed with basetransform's internal state.
6519 2008-11-22 15:02:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6521 tests/check/elements/speexresample.c: Make the unit test a bit faster to prevent timeouts, especially with valgrind.
6522 Original commit message from CVS:
6523 * tests/check/elements/speexresample.c: (GST_START_TEST):
6524 Make the unit test a bit faster to prevent timeouts, especially
6527 2008-11-22 14:44:26 +0000 Wim Taymans <wim.taymans@gmail.com>
6529 gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
6530 Original commit message from CVS:
6531 * gst/videorate/gstvideorate.c:
6532 Add jpeg and png image media types to the caps. Fixes #561436.
6534 2008-11-22 14:31:43 +0000 Wim Taymans <wim.taymans@gmail.com>
6536 gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ...
6537 Original commit message from CVS:
6538 * gst/playback/gstplaysink.c: (gen_audio_chain):
6539 Don't post an error when we can't configure the volume but post a
6540 warning instead. Fixes #561780.
6542 2008-11-21 20:32:56 +0000 Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
6544 gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video...
6545 Original commit message from CVS:
6546 Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
6547 * gst/videotestsrc/gstvideotestsrc.c:
6548 * gst/videotestsrc/gstvideotestsrc.h:
6549 * gst/videotestsrc/videotestsrc.c:
6550 * gst/videotestsrc/videotestsrc.h:
6551 Add a zone plate pattern generator based on BBC R&D Report
6552 1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate
6553 kx2=20 ky2=20 kt=1'.
6555 2008-11-21 15:45:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6557 gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
6558 Original commit message from CVS:
6559 * gst/speexresample/gstspeexresample.c:
6560 (gst_speex_resample_class_init), (gst_speex_resample_set_property),
6561 (gst_speex_resample_get_property):
6562 Add a "filter-length" property that maps to the quality values
6563 for compatibilty with audioresample.
6565 2008-11-21 00:04:48 +0000 Michael Smith <msmith@xiph.org>
6567 gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
6568 Original commit message from CVS:
6569 * gst/playback/gstdecodebin2.c:
6570 Fix random fat-fingering making this not compile.
6572 2008-11-20 22:11:38 +0000 Michael Smith <msmith@xiph.org>
6574 gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh...
6575 Original commit message from CVS:
6576 * gst/playback/gstdecodebin2.c:
6577 If the top-level type of the stream is plain text, don't try to decode
6578 it, matching behaviour of decodebin.
6579 * gst/playback/gstplaysink.c:
6580 If we fail to generate a text chain (e.g. due to missing optional
6581 plugins), don't crash.
6583 2008-11-20 22:06:05 +0000 Michael Smith <msmith@xiph.org>
6585 gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
6586 Original commit message from CVS:
6587 * gst-libs/gst/rtsp/gstrtspdefs.c:
6588 Fix win32 build. Oops.
6590 2008-11-20 21:40:49 +0000 Michael Smith <msmith@xiph.org>
6592 gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
6593 Original commit message from CVS:
6594 * gst-libs/gst/rtsp/gstrtspdefs.c:
6595 Use WSAGetLastError() rather than errno/h_errno on win32.
6597 2008-11-20 21:20:27 +0000 Michael Smith <msmith@xiph.org>
6599 gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
6600 Original commit message from CVS:
6601 * gst-libs/gst/riff/riff-media.c:
6602 Support WMA Lossless properly.
6604 2008-11-19 00:24:44 +0000 David Schleef <ds@schleef.org>
6606 gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect...
6607 Original commit message from CVS:
6608 * gst/videotestsrc/gstvideotestsrc.c:
6609 * gst/videotestsrc/gstvideotestsrc.h:
6610 * gst/videotestsrc/videotestsrc.c:
6611 * gst/videotestsrc/videotestsrc.h:
6612 Add "colorspec" property, specifying whether to generate BT.601
6613 or BT.709 video. This only affects YCbCr values, not RGB, since
6614 if you're generating a 709 test pattern, presumably you want
6615 709 RGB primaries, not 601. Also add "smpte75" pattern, which
6616 uses 75% colors instead of 100%, since this is often more useful
6617 for testing (and also follows the SMPTE EG-1 guideline).
6619 2008-11-18 18:08:42 +0000 Alessandro Decina <alessandro.d@gmail.com>
6621 gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2.
6622 Original commit message from CVS:
6623 * gst/playback/gstdecodebin.c:
6624 Add a "sink-caps" property to decodebin like it's done for decodebin2.
6627 2008-11-14 21:44:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
6629 gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
6630 Original commit message from CVS:
6631 * gst/audioresample/gstaudioresample.c:
6632 Guard against a NULL dereference I somehow encountered -
6633 with a FLUSH_STOP arriving either before basetransform _start(),
6635 * gst/typefind/gsttypefindfunctions.c:
6636 Make sure we never jump backwards when typefinding corrupt mov files.
6638 2008-11-14 21:39:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
6640 gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
6641 Original commit message from CVS:
6642 * gst-libs/gst/interfaces/propertyprobe.c:
6643 Fix random type causing a docs warning.
6645 2008-11-14 15:40:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6647 sys/v4l/gstv4l.c: Give it a minimal rank for autovideosrc.
6648 Original commit message from CVS:
6650 Give it a minimal rank for autovideosrc.
6652 2008-11-13 21:11:13 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6654 gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types.
6655 Original commit message from CVS:
6656 * gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
6658 Improve typefinding of ISO JPEG2000 mime types.
6660 2008-11-13 18:18:32 +0000 Wim Taymans <wim.taymans@gmail.com>
6662 sys/xvimage/xvimagesink.*: Avoid typechecking when we do trivial casts.
6663 Original commit message from CVS:
6664 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
6665 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps),
6666 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
6667 * sys/xvimage/xvimagesink.h:
6668 Avoid typechecking when we do trivial casts.
6669 Move error handling out of the main program flow.
6670 Sneak in the display-region caps property, not completely correct yet.
6671 Cache the width/height in buffer_alloc instead of parsing it from the
6674 2008-11-13 17:27:37 +0000 Wim Taymans <wim.taymans@gmail.com>
6676 gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
6677 Original commit message from CVS:
6678 * gst/playback/gstplaybin2.c: (deactivate_group):
6679 don't try to unlink the selector sinkpad when we don't have it yet. This
6680 can happen if an error occured before the group was complete.
6682 2008-11-13 15:37:40 +0000 Wim Taymans <wim.taymans@gmail.com>
6684 gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
6685 Original commit message from CVS:
6686 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
6687 (gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
6688 (gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
6689 (gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
6690 (gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
6691 (gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
6692 (gst_rtp_buffer_get_extension_data),
6693 (gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
6694 (gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
6695 (gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
6696 (gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
6697 (gst_rtp_buffer_get_payload_type),
6698 (gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
6699 (gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
6700 (gst_rtp_buffer_set_timestamp),
6701 (gst_rtp_buffer_get_payload_subbuffer),
6702 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
6703 Avoid expensive type checks we already did as part of the
6704 _validate() function that should be called first.
6706 2008-11-11 16:40:50 +0000 Wim Taymans <wim.taymans@gmail.com>
6708 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some cases where a newsegment event was not sent.
6709 Original commit message from CVS:
6710 * gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
6711 (gst_base_rtp_depayload_push_full),
6712 (gst_base_rtp_depayload_set_gst_timestamp):
6713 Fix some cases where a newsegment event was not sent.
6715 2008-11-11 15:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
6717 gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co...
6718 Original commit message from CVS:
6719 * gst/playback/gstplaybin2.c: (activate_group):
6720 Catch state change errors and stop from the uridecodebin elements
6721 instead of trying to continue in vain.
6723 2008-11-10 14:53:45 +0000 Edward Hervey <bilboed@bilboed.com>
6725 gst/: Wim, you're a bad boy. You don't want people to contact you or what?
6726 Original commit message from CVS:
6727 * gst-libs/gst/app/gstappsink.c:
6728 * gst-libs/gst/app/gstappsrc.c:
6729 * gst/h264parse/gsth264parse.c:
6730 Wim, you're a bad boy. You don't want people to contact you or what?
6732 2008-11-10 14:22:09 +0000 Wim Taymans <wim.taymans@gmail.com>
6734 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
6735 Original commit message from CVS:
6736 * gst-libs/gst/audio/gstbaseaudiosink.c:
6737 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
6738 (gst_base_audio_sink_callback):
6739 Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
6740 for the latency to expire, fixes #559567.
6742 2008-11-10 13:55:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
6744 gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
6745 Original commit message from CVS:
6746 * gst/adder/gstadder.c:
6747 Change author string after seeing output of gst-inspector.
6749 2008-11-10 10:33:26 +0000 Wim Taymans <wim.taymans@gmail.com>
6751 gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559...
6752 Original commit message from CVS:
6753 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
6754 Don't try to do crazy things when we only have a text pad without a
6755 video pad. Fixes #559478.
6757 2008-11-07 17:35:46 +0000 Wim Taymans <wim.taymans@gmail.com>
6759 gst-libs/gst/app/gstappsrc.*: Add is-live property.
6760 Original commit message from CVS:
6761 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
6762 (gst_app_src_init), (gst_app_src_set_property),
6763 (gst_app_src_get_property), (gst_app_src_push_buffer):
6764 * gst-libs/gst/app/gstappsrc.h:
6765 Add is-live property.
6768 2008-11-06 12:14:51 +0000 Wim Taymans <wim.taymans@gmail.com>
6770 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...
6771 Original commit message from CVS:
6772 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
6773 Fix case where we don't have a range for the rates or channels as is the
6774 case with truespeech.
6776 2008-11-05 19:18:25 +0000 Wim Taymans <wim.taymans@gmail.com>
6778 gst/volume/gstvolume.*: Keep negotiated state in a separate variable.
6779 Original commit message from CVS:
6780 * gst/volume/gstvolume.c: (volume_update_real_volume),
6781 (gst_volume_set_volume), (gst_volume_get_volume),
6782 (gst_volume_set_mute), (gst_volume_init), (volume_setup),
6783 (volume_transform_ip), (volume_update_mute),
6784 (volume_update_volume), (volume_get_property):
6785 * gst/volume/gstvolume.h:
6786 Keep negotiated state in a separate variable.
6787 Protect the volume and mute properties with the object lock.
6788 Protect modifying the transform with the transform lock.
6790 2008-11-05 12:20:21 +0000 Wim Taymans <wim.taymans@gmail.com>
6792 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Only convert caps to string when debug is enabled.
6793 Original commit message from CVS:
6794 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
6795 (gst_ffmpeg_pixfmt_to_caps):
6796 Only convert caps to string when debug is enabled.
6798 2008-11-04 18:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
6800 ext/theora/: Copy seqnum.
6801 Original commit message from CVS:
6802 * ext/theora/gsttheoradec.h:
6803 * ext/theora/theoradec.c: (gst_theora_dec_init),
6804 (gst_theora_dec_reset), (theora_dec_src_event),
6805 (theora_dec_sink_event), (theora_handle_type_packet):
6807 Keep events in a pending list, like vorbisdec, instead of trying
6808 to construct a segment event ourselves.
6809 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
6810 (vorbis_dec_src_event), (vorbis_dec_sink_event):
6811 * ext/vorbis/vorbisdec.h:
6814 2008-11-04 17:24:35 +0000 Wim Taymans <wim.taymans@gmail.com>
6816 ext/ogg/gstoggdemux.*: Copy seqnums around to track playback segments and messages.
6817 Original commit message from CVS:
6818 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
6819 (gst_ogg_demux_deactivate_current_chain),
6820 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page),
6821 (gst_ogg_demux_loop):
6822 * ext/ogg/gstoggdemux.h:
6823 Copy seqnums around to track playback segments and messages.
6825 2008-11-04 12:42:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6827 Don't install static libs for plugins. Fixes #550851 for -bad.
6828 Original commit message from CVS:
6829 * ext/alsaspdif/Makefile.am:
6830 * ext/amrwb/Makefile.am:
6831 * ext/apexsink/Makefile.am:
6832 * ext/arts/Makefile.am:
6833 * ext/artsd/Makefile.am:
6834 * ext/audiofile/Makefile.am:
6835 * ext/audioresample/Makefile.am:
6836 * ext/bz2/Makefile.am:
6837 * ext/cdaudio/Makefile.am:
6838 * ext/celt/Makefile.am:
6839 * ext/dc1394/Makefile.am:
6840 * ext/dirac/Makefile.am:
6841 * ext/directfb/Makefile.am:
6842 * ext/divx/Makefile.am:
6843 * ext/dts/Makefile.am:
6844 * ext/faac/Makefile.am:
6845 * ext/faad/Makefile.am:
6846 * ext/gsm/Makefile.am:
6847 * ext/hermes/Makefile.am:
6848 * ext/ivorbis/Makefile.am:
6849 * ext/jack/Makefile.am:
6850 * ext/jp2k/Makefile.am:
6851 * ext/ladspa/Makefile.am:
6852 * ext/lcs/Makefile.am:
6853 * ext/libfame/Makefile.am:
6854 * ext/libmms/Makefile.am:
6855 * ext/metadata/Makefile.am:
6856 * ext/mpeg2enc/Makefile.am:
6857 * ext/mplex/Makefile.am:
6858 * ext/musepack/Makefile.am:
6859 * ext/musicbrainz/Makefile.am:
6860 * ext/mythtv/Makefile.am:
6861 * ext/nas/Makefile.am:
6862 * ext/neon/Makefile.am:
6863 * ext/ofa/Makefile.am:
6864 * ext/polyp/Makefile.am:
6865 * ext/resindvd/Makefile.am:
6866 * ext/sdl/Makefile.am:
6867 * ext/shout/Makefile.am:
6868 * ext/snapshot/Makefile.am:
6869 * ext/sndfile/Makefile.am:
6870 * ext/soundtouch/Makefile.am:
6871 * ext/spc/Makefile.am:
6872 * ext/swfdec/Makefile.am:
6873 * ext/tarkin/Makefile.am:
6874 * ext/theora/Makefile.am:
6875 * ext/timidity/Makefile.am:
6876 * ext/twolame/Makefile.am:
6877 * ext/x264/Makefile.am:
6878 * ext/xine/Makefile.am:
6879 * ext/xvid/Makefile.am:
6880 * gst-libs/gst/app/Makefile.am:
6881 * gst-libs/gst/dshow/Makefile.am:
6882 * gst/aiffparse/Makefile.am:
6883 * gst/app/Makefile.am:
6884 * gst/audiobuffer/Makefile.am:
6885 * gst/bayer/Makefile.am:
6886 * gst/cdxaparse/Makefile.am:
6887 * gst/chart/Makefile.am:
6888 * gst/colorspace/Makefile.am:
6889 * gst/dccp/Makefile.am:
6890 * gst/deinterlace/Makefile.am:
6891 * gst/deinterlace2/Makefile.am:
6892 * gst/dvdspu/Makefile.am:
6893 * gst/festival/Makefile.am:
6894 * gst/filter/Makefile.am:
6895 * gst/flacparse/Makefile.am:
6896 * gst/flv/Makefile.am:
6897 * gst/games/Makefile.am:
6898 * gst/h264parse/Makefile.am:
6899 * gst/librfb/Makefile.am:
6900 * gst/mixmatrix/Makefile.am:
6901 * gst/modplug/Makefile.am:
6902 * gst/mpeg1sys/Makefile.am:
6903 * gst/mpeg4videoparse/Makefile.am:
6904 * gst/mpegdemux/Makefile.am:
6905 * gst/mpegtsmux/Makefile.am:
6906 * gst/mpegvideoparse/Makefile.am:
6907 * gst/mve/Makefile.am:
6908 * gst/nsf/Makefile.am:
6909 * gst/nuvdemux/Makefile.am:
6910 * gst/overlay/Makefile.am:
6911 * gst/passthrough/Makefile.am:
6912 * gst/pcapparse/Makefile.am:
6913 * gst/playondemand/Makefile.am:
6914 * gst/rawparse/Makefile.am:
6915 * gst/real/Makefile.am:
6916 * gst/rtjpeg/Makefile.am:
6917 * gst/rtpmanager/Makefile.am:
6918 * gst/scaletempo/Makefile.am:
6919 * gst/sdp/Makefile.am:
6920 * gst/selector/Makefile.am:
6921 * gst/smooth/Makefile.am:
6922 * gst/smoothwave/Makefile.am:
6923 * gst/speed/Makefile.am:
6924 * gst/speexresample/Makefile.am:
6925 * gst/stereo/Makefile.am:
6926 * gst/subenc/Makefile.am:
6927 * gst/tta/Makefile.am:
6928 * gst/vbidec/Makefile.am:
6929 * gst/videodrop/Makefile.am:
6930 * gst/videosignal/Makefile.am:
6931 * gst/virtualdub/Makefile.am:
6932 * gst/vmnc/Makefile.am:
6933 * gst/y4m/Makefile.am:
6934 * sys/acmenc/Makefile.am:
6935 * sys/cdrom/Makefile.am:
6936 * sys/dshowdecwrapper/Makefile.am:
6937 * sys/dshowsrcwrapper/Makefile.am:
6938 * sys/dvb/Makefile.am:
6939 * sys/dxr3/Makefile.am:
6940 * sys/fbdev/Makefile.am:
6941 * sys/oss4/Makefile.am:
6942 * sys/qcam/Makefile.am:
6943 * sys/qtwrapper/Makefile.am:
6944 * sys/vcd/Makefile.am:
6945 * sys/wininet/Makefile.am:
6946 * win32/common/config.h:
6947 Don't install static libs for plugins. Fixes #550851 for -bad.
6949 2008-11-03 15:30:14 +0000 Matthias Kretz <kretz@kde.org>
6951 ext/alsa/gstalsasink.c: Make all access non-blocking so that we can better handle unplugging of usb devices. Fixes #5...
6952 Original commit message from CVS:
6953 Based on patch by: Matthias Kretz <kretz at kde dot org>
6954 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
6955 (gst_alsasink_prepare), (gst_alsasink_unprepare),
6956 (gst_alsasink_write):
6957 Make all access non-blocking so that we can better handle unplugging
6958 of usb devices. Fixes #559111
6960 2008-11-03 10:49:24 +0000 Damien Lespiau <damien.lespiau@gmail.com>
6962 gst-libs/gst/rtsp/gstrtspconnection.c: Make the next call to poll not depend on previous calls to poll with or withou...
6963 Original commit message from CVS:
6964 Patch by: Damien Lespiau <damien.lespiau gmail com>
6965 * gst-libs/gst/rtsp/gstrtspconnection.c:
6966 (gst_rtsp_connection_write):
6967 Make the next call to poll not depend on previous calls to poll with or
6968 without reading from the active descriptor. Fixes #544293.
6970 2008-11-03 08:55:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6972 gst/speexresample/gstspeexresample.c: Add TODO at the top of the file for enabling SSE/ARM specific optimizations and...
6973 Original commit message from CVS:
6974 * gst/speexresample/gstspeexresample.c:
6975 (gst_speex_resample_convert_buffer):
6976 Add TODO at the top of the file for enabling SSE/ARM specific
6977 optimizations and choosing the fastest implementation at runtime.
6978 Add g_assert_not_reached() at two places that should really never
6981 2008-11-02 09:19:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6983 gst/speexresample/gstspeexresample.c: Fix format string and arguments.
6984 Original commit message from CVS:
6985 * gst/speexresample/gstspeexresample.c:
6986 (gst_speex_resample_check_discont):
6987 Fix format string and arguments.
6988 * gst/speexresample/resample_sse.h:
6991 2008-11-01 19:38:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6993 gst/speexresample/: Add missing headers to Makefile.am.
6994 Original commit message from CVS:
6995 * gst/speexresample/Makefile.am:
6996 * gst/speexresample/gstspeexresample.c:
6997 (gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
6998 (gst_speex_resample_convert_buffer), (_benchmark_int_float),
6999 (_benchmark_int_int), (_benchmark_integer_resampling),
7001 * gst/speexresample/gstspeexresample.h:
7002 * gst/speexresample/resample.c:
7003 * gst/speexresample/speex_resampler_double.c:
7004 * gst/speexresample/speex_resampler_float.c:
7005 * gst/speexresample/speex_resampler_int.c:
7006 * gst/speexresample/speex_resampler_wrapper.h:
7007 Add missing headers to Makefile.am.
7008 Update copyright, years and my mail address.
7009 Benchmark the integer resampling implementation against the
7010 float implementation and use the faster one for 8/16 bit integer
7011 input. On most recent systems the floating point version is faster.
7013 2008-10-31 09:49:57 +0000 Nick Haddad <nick@haddads.net>
7015 gst-libs/gst/riff/: Add support for other fourcc codes that are commonly used for 'uncompressed RGB', including 'RGB ...
7016 Original commit message from CVS:
7017 Patch by: Nick Haddad <nick at haddads dot net>
7018 * gst-libs/gst/riff/riff-ids.h:
7019 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
7020 Add support for other fourcc codes that are commonly used for
7021 'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
7024 2008-10-30 14:55:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7026 gst/speexresample/gstspeexresample.c: The length for the buffer conversion function is the number of audio frames, i....
7027 Original commit message from CVS:
7028 * gst/speexresample/gstspeexresample.c:
7029 (gst_speex_resample_convert_buffer):
7030 The length for the buffer conversion function is the number of
7031 audio frames, i.e. we need to multiply it by the number of channels
7032 to get the number of values. Also spotted by the unit test after
7033 running in valgrind.
7035 2008-10-30 14:46:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7037 tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
7038 Original commit message from CVS:
7039 * tests/check/elements/speexresample.c: (element_message_cb),
7040 (eos_message_cb), (test_pipeline), (GST_START_TEST),
7041 (speexresample_suite):
7042 Add pipeline unit tests for testing all supported formats with
7043 up/downsampling and different in/outrates.
7044 * gst/speexresample/gstspeexresample.c:
7045 (gst_speex_resample_push_drain), (gst_speex_resample_process):
7046 * gst/speexresample/speex_resampler_wrapper.h:
7047 Fix bugs identified by the testsuite.
7049 2008-10-30 13:44:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7051 gst/speexresample/: Add support for int8, int24 and int32 input by converting internally to/from int16 or double.
7052 Original commit message from CVS:
7053 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
7054 (gst_speex_resample_get_funcs),
7055 (gst_speex_resample_transform_size),
7056 (gst_speex_resample_convert_buffer),
7057 (gst_speex_resample_push_drain), (gst_speex_resample_process):
7058 * gst/speexresample/gstspeexresample.h:
7059 * gst/speexresample/speex_resampler_wrapper.h:
7060 Add support for int8, int24 and int32 input by converting internally
7061 to/from int16 or double.
7063 2008-10-30 12:43:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7065 Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
7066 Original commit message from CVS:
7067 * gst/speexresample/Makefile.am:
7068 * gst/speexresample/arch.h:
7069 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
7070 (gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
7071 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
7072 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
7073 (_gcd), (gst_speex_resample_transform_size),
7074 (gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
7075 (gst_speex_resample_process), (gst_speex_resample_transform),
7076 (gst_speex_resample_query), (gst_speex_resample_set_property):
7077 * gst/speexresample/gstspeexresample.h:
7078 * gst/speexresample/resample.c:
7079 * gst/speexresample/speex_resampler.h:
7080 * gst/speexresample/speex_resampler_double.c:
7081 * gst/speexresample/speex_resampler_wrapper.h:
7082 * tests/check/elements/speexresample.c: (setup_speexresample),
7083 (test_perfect_stream_instance), (GST_START_TEST),
7084 (test_discont_stream_instance):
7085 Add support for double samples as input and refactor the usage
7086 of the different compilation flavors of the speex resampler.
7088 2008-10-30 11:43:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7090 gst/audioresample/gstaudioresample.c: Return the result of parent_class->event().
7091 Original commit message from CVS:
7092 * gst/audioresample/gstaudioresample.c:
7093 Return the result of parent_class->event().
7095 2008-10-29 17:02:55 +0000 Wim Taymans <wim.taymans@gmail.com>
7097 gst-libs/gst/app/gstappsink.c: Fix the docs.
7098 Original commit message from CVS:
7099 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
7102 2008-10-29 12:11:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7104 gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
7105 Original commit message from CVS:
7106 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
7107 (gst_speex_resample_get_unit_size),
7108 (gst_speex_resample_push_drain), (gst_speex_resample_event),
7109 (gst_speex_resample_check_discont), (gst_speex_resample_process),
7110 (gst_speex_resample_transform):
7111 * gst/speexresample/gstspeexresample.h:
7112 Rewrite timestamp tracking to make it more robust and guarantee
7114 * tests/check/Makefile.am:
7115 * tests/check/elements/speexresample.c: (setup_speexresample),
7116 (cleanup_speexresample), (fail_unless_perfect_stream),
7117 (test_perfect_stream_instance), (GST_START_TEST),
7118 (test_discont_stream_instance), (live_switch_alloc_only_48000),
7119 (live_switch_get_sink_caps), (live_switch_push),
7120 (speexresample_suite):
7121 Add unit tests for speexresample based on the audioresample unit tests.
7123 2008-10-28 19:30:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7125 gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
7126 Original commit message from CVS:
7127 * gst/speexresample/gstspeexresample.c:
7128 (gst_speex_resample_get_unit_size),
7129 (gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
7130 (gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
7131 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
7132 (gst_speex_resample_push_drain), (gst_speex_resample_event),
7133 (gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
7134 (gst_speex_resample_process), (gst_speex_resample_transform),
7135 (gst_speex_resample_query), (gst_speex_resample_set_property):
7136 * gst/speexresample/gstspeexresample.h:
7137 Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
7138 instead of GST_DEBUG, ...
7140 2008-10-28 16:28:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7142 gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
7143 Original commit message from CVS:
7144 * gst/speexresample/gstspeexresample.c:
7145 (gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
7146 (gst_speex_resample_process):
7147 Fixate to the nearest supported rate instead of the first one.
7149 2008-10-28 16:25:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7151 gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
7152 Original commit message from CVS:
7153 * gst/audioresample/gstaudioresample.c:
7154 (gst_audioresample_class_init), (audioresample_fixate_caps):
7155 Fixate the rate to the nearest supported rate instead of
7156 the first one. Fixes bug #549510.
7158 2008-10-28 11:46:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7160 gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
7161 Original commit message from CVS:
7162 * gst/speexresample/README:
7163 * gst/speexresample/arch.h:
7164 * gst/speexresample/fixed_arm4.h:
7165 * gst/speexresample/fixed_arm5e.h:
7166 * gst/speexresample/fixed_bfin.h:
7167 * gst/speexresample/fixed_debug.h:
7168 * gst/speexresample/fixed_generic.h:
7169 * gst/speexresample/resample.c: (compute_func), (main), (sinc),
7170 (cubic_coef), (resampler_basic_direct_single),
7171 (resampler_basic_direct_double),
7172 (resampler_basic_interpolate_single),
7173 (resampler_basic_interpolate_double), (update_filter),
7174 (speex_resampler_init_frac), (speex_resampler_process_native),
7175 (speex_resampler_magic), (speex_resampler_process_float),
7176 (speex_resampler_process_int),
7177 (speex_resampler_process_interleaved_float),
7178 (speex_resampler_process_interleaved_int),
7179 (speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
7180 (speex_resampler_reset_mem):
7181 * gst/speexresample/speex_resampler.h:
7182 Update Speex resampler with latest version from Speex GIT.
7184 2008-10-27 14:57:34 +0000 Wim Taymans <wim.taymans@gmail.com>
7186 win32/common/libgstaudio.def: Add new symbols.
7187 Original commit message from CVS:
7188 * win32/common/libgstaudio.def:
7191 2008-10-23 09:57:06 +0000 Wim Taymans <wim.taymans@gmail.com>
7193 ext/vorbis/vorbisdec.c: Attempt to make obfuscated code clearer.
7194 Original commit message from CVS:
7195 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
7196 Attempt to make obfuscated code clearer.
7198 2008-10-23 07:11:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7200 Move float endianness conversion macros to core. Second part of bug ##555196.
7201 Original commit message from CVS:
7202 * docs/libs/gst-plugins-base-libs-sections.txt:
7203 * gst-libs/gst/floatcast/floatcast.h:
7204 Move float endianness conversion macros to core. Second part of
7207 2008-10-22 12:29:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7209 sys/: Don't mark as gtk-doc docs as they aren't public.
7210 Original commit message from CVS:
7211 * sys/ximage/ximagesink.h:
7212 * sys/xvimage/xvimagesink.h:
7213 Don't mark as gtk-doc docs as they aren't public.
7215 2008-10-22 12:25:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7217 Allow setting colorkey if possible. Implement property probe interface for optional X features (autopaint-colorkey, d...
7218 Original commit message from CVS:
7219 * sys/xvimage/xvimagesink.c:
7220 * sys/xvimage/xvimagesink.h:
7221 * tests/icles/Makefile.am:
7222 * tests/icles/test-colorkey.c:
7223 Allow setting colorkey if possible. Implement property probe interface
7224 for optional X features (autopaint-colorkey, double-buffer and
7225 colorkey). Fixes #554533
7227 2008-10-22 12:01:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7229 gst-libs/gst/tag/tags.c: Remove useless buffer size assignment. It already has this value.
7230 Original commit message from CVS:
7231 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
7232 Remove useless buffer size assignment. It already has this value.
7234 2008-10-20 15:35:37 +0000 Wim Taymans <wim.taymans@gmail.com>
7236 gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
7237 Original commit message from CVS:
7238 * gst-libs/gst/audio/gstaudiosink.c:
7239 (gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
7240 (gst_audioringbuffer_activate), (gst_audioringbuffer_release),
7241 (gst_audioringbuffer_stop):
7242 Implement a separate activate functions to start monitoring the segments
7243 or, in pull mode, pulling in data.
7244 * gst-libs/gst/audio/gstbaseaudiosink.c:
7245 (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
7246 (gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
7247 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
7248 (gst_base_audio_sink_activate_pull),
7249 (gst_base_audio_sink_async_play),
7250 (gst_base_audio_sink_change_state):
7251 Implement pad and element convert query function.
7252 Activate the ringbuffer.
7253 Use the segment last_stop value as the offset to pull.
7254 Use new basesink _do_preroll() method to preroll in the pulling thread.
7255 Take appropriate locking in the pulling thread.
7256 * gst-libs/gst/audio/gstringbuffer.h:
7259 2008-10-20 14:08:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7261 gst/typefind/gsttypefindfunctions.c: Improve MXF typefinding a bit by searching for a header partition pack instead o...
7262 Original commit message from CVS:
7263 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
7264 Improve MXF typefinding a bit by searching for a header partition
7265 pack instead of just a general partition pack and checking more
7266 bytes for valid values.
7268 2008-10-20 13:45:55 +0000 Wim Taymans <wim.taymans@gmail.com>
7270 tests/icles/.cvsignore: update ignore file.
7271 Original commit message from CVS:
7272 * tests/icles/.cvsignore:
7274 * tests/icles/Makefile.am:
7275 * tests/icles/test-box.c: (make_pipeline), (main):
7276 Add another interactive command line experimentation suite for
7277 dynamically boxing/cropping/saling an input video.
7279 2008-10-17 13:19:05 +0000 Wim Taymans <wim.taymans@gmail.com>
7281 Add methods to more accuratly control the pulling thread of a ringbuffer.
7282 Original commit message from CVS:
7283 * docs/libs/gst-plugins-base-libs-sections.txt:
7284 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
7285 (gst_ring_buffer_activate), (gst_ring_buffer_is_active):
7286 * gst-libs/gst/audio/gstringbuffer.h:
7287 Add methods to more accuratly control the pulling thread of a
7289 Add format conversion helper code to the ringbuffer.
7290 API: GstRingBuffer:gst_ring_buffer_activate()
7291 API: GstRingBuffer:gst_ring_buffer_is_active()
7292 API: GstRingBuffer:gst_ring_buffer_convert()
7294 2008-10-16 15:44:37 +0000 Wim Taymans <wim.taymans@gmail.com>
7296 gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
7297 Original commit message from CVS:
7298 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
7299 (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
7300 (gst_audioringbuffer_stop):
7301 Signal thread startup earlier so that we can immediatly go into pull
7302 mode when we have to and block on preroll.
7304 2008-10-16 15:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
7306 gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
7307 Original commit message from CVS:
7308 * gst-libs/gst/audio/gstringbuffer.c:
7309 (gst_ring_buffer_prepare_read):
7310 In pull mode we want the callback to prepull a buffer we can preroll on
7311 even when we are not yet playing.
7313 2008-10-16 15:07:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7315 Don't install static libs for plugins. Fixes #550851 for base.
7316 Original commit message from CVS:
7317 * ext/alsa/Makefile.am:
7318 * ext/cdparanoia/Makefile.am:
7319 * ext/gio/Makefile.am:
7320 * ext/gnomevfs/Makefile.am:
7321 * ext/libvisual/Makefile.am:
7322 * ext/ogg/Makefile.am:
7323 * ext/pango/Makefile.am:
7324 * ext/theora/Makefile.am:
7325 * ext/vorbis/Makefile.am:
7326 * gst/adder/Makefile.am:
7327 * gst/audioconvert/Makefile.am:
7328 * gst/audiorate/Makefile.am:
7329 * gst/audioresample/Makefile.am:
7330 * gst/audiotestsrc/Makefile.am:
7331 * gst/ffmpegcolorspace/Makefile.am:
7332 * gst/gdp/Makefile.am:
7333 * gst/playback/Makefile.am:
7334 * gst/subparse/Makefile.am:
7335 * gst/tcp/Makefile.am:
7336 * gst/typefind/Makefile.am:
7337 * gst/videorate/Makefile.am:
7338 * gst/videoscale/Makefile.am:
7339 * gst/videotestsrc/Makefile.am:
7340 * gst/volume/Makefile.am:
7341 * sys/v4l/Makefile.am:
7342 * sys/ximage/Makefile.am:
7343 * sys/xvimage/Makefile.am:
7344 Don't install static libs for plugins. Fixes #550851 for base.
7346 2008-10-16 13:50:00 +0000 Wim Taymans <wim.taymans@gmail.com>
7348 gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe...
7349 Original commit message from CVS:
7350 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
7351 Set the default blocksize to -1 because we will then use the configured
7352 samplesperbuffer to create our output buffer.
7354 2008-10-15 15:28:41 +0000 Edward Hervey <bilboed@bilboed.com>
7356 gst-libs/gst/riff/riff-media.c: Add mappping for the KMVC (Karl Morton's Video) Codec.
7357 Original commit message from CVS:
7358 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
7359 (gst_riff_create_video_template_caps):
7360 Add mappping for the KMVC (Karl Morton's Video) Codec.
7362 2008-10-15 14:25:50 +0000 Edward Hervey <bilboed@bilboed.com>
7364 gst/typefind/gsttypefindfunctions.c: Don't forget to advance the offset of what we're matching against, else we end u...
7365 Original commit message from CVS:
7366 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
7367 Don't forget to advance the offset of what we're matching against, else
7368 we end up in a forever loop.
7370 2008-10-15 11:25:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7372 gst/subparse/gstsubparse.c: Improve typefinding a bit. If we don't have a Unicode charset try GST_SUBTITLE_ENCODING a...
7373 Original commit message from CVS:
7374 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
7375 Improve typefinding a bit. If we don't have a Unicode charset
7376 try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
7378 2008-10-14 11:13:59 +0000 Edward Hervey <bilboed@bilboed.com>
7380 ext/theora/theoradec.c: Fix build on macosx.
7381 Original commit message from CVS:
7382 * ext/theora/theoradec.c: (theora_dec_decode_buffer):
7383 Fix build on macosx.
7385 2008-10-13 11:36:13 +0000 Robin Stocker <robin@nibor.org>
7387 ext/theora/: Parse input caps and make the PAR override the encoded PAR when specified by a container. Fixes #555699.
7388 Original commit message from CVS:
7389 Based on patch by: Robin Stocker <robin at nibor dot org>
7390 * ext/theora/gsttheoradec.h:
7391 * ext/theora/theoradec.c: (gst_theora_dec_init),
7392 (theora_dec_setcaps), (theora_handle_type_packet),
7393 (theora_dec_decode_buffer), (theora_dec_change_state):
7394 Parse input caps and make the PAR override the encoded PAR when
7395 specified by a container. Fixes #555699.
7397 2008-10-13 09:16:59 +0000 Wim Taymans <wim.taymans@gmail.com>
7399 gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY
7400 Original commit message from CVS:
7401 * gst-libs/gst/rtp/gstbasertpdepayload.c:
7402 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
7403 (gst_base_rtp_depayload_set_gst_timestamp),
7404 (gst_base_rtp_depayload_change_state):
7405 * gst-libs/gst/rtp/gstbasertpdepayload.h:
7406 Add some more G_LIKELY
7407 Fail when the setcaps function was not called.
7408 * gst-libs/gst/rtp/gstbasertppayload.c:
7409 (gst_basertppayload_set_outcaps):
7410 Propagate return value of setcaps.
7412 2008-10-13 08:58:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7414 gst/subparse/: Add support for UTF16/UTF32 subtitles as long as the first bytes of the first buffer contain the BOM. ...
7415 Original commit message from CVS:
7416 * gst/subparse/Makefile.am:
7417 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
7418 (gst_sub_parse_class_init), (gst_sub_parse_init),
7419 (gst_convert_to_utf8), (detect_encoding), (convert_encoding),
7420 (get_next_line), (gst_sub_parse_data_format_autodetect),
7421 (feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
7422 (gst_subparse_type_find):
7423 * gst/subparse/gstsubparse.h:
7424 Add support for UTF16/UTF32 subtitles as long as the first bytes of
7425 the first buffer contain the BOM. This also adds support for other
7426 encodings that allow NUL bytes via the encoding property.
7427 Fixes bugs #552237 and #456788.
7429 2008-10-13 08:15:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7431 gst-libs/gst/tag/tags.c: Don't drop the last byte of image tags if they're not an URI list.
7432 Original commit message from CVS:
7433 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
7434 Don't drop the last byte of image tags if they're not an URI list.
7437 2008-10-13 08:00:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7439 gst/typefind/gsttypefindfunctions.c: For looking at the 4th byte we have to get 4 bytes of course and not 3.
7440 Original commit message from CVS:
7441 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
7442 For looking at the 4th byte we have to get 4 bytes of course
7445 2008-10-13 07:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7447 gst/typefind/gsttypefindfunctions.c: Improve FLAC-without-headers typefinding by looking at most of the frame header ...
7448 Original commit message from CVS:
7449 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
7450 Improve FLAC-without-headers typefinding by looking at most of the
7451 frame header and checking if invalid values are used. Should prevent
7452 quite some false positives compared to the old version which only
7453 check if the first 14 bits are set.
7455 2008-10-11 16:27:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7457 sys/xvimage/xvimagesink.c: Don't assert on caps==NULL.
7458 Original commit message from CVS:
7459 * sys/xvimage/xvimagesink.c:
7460 Don't assert on caps==NULL.
7462 2008-10-10 17:13:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7464 Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
7465 Original commit message from CVS:
7466 * gst/subparse/gstsubparse.c:
7467 (gst_sub_parse_data_format_autodetect), (handle_buffer),
7468 (gst_sub_parse_change_state):
7469 * gst/subparse/gstsubparse.h:
7470 * tests/check/elements/subparse.c: (GST_START_TEST):
7471 Add support for subtitle files with UTF-8 BOM at the beginning
7472 by simple stripping it from the first line before passing it
7473 to any parsing code. Fixes bug #555257 and playback of files
7474 created by Gnome Subtitles.
7476 2008-10-10 15:45:15 +0000 Wim Taymans <wim.taymans@gmail.com>
7478 gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place.
7479 Original commit message from CVS:
7480 * gst/audiotestsrc/gstaudiotestsrc.c:
7481 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
7482 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
7483 (gst_audio_test_src_start), (gst_audio_test_src_stop),
7484 (gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
7485 (gst_audio_test_src_create):
7486 * gst/audiotestsrc/gstaudiotestsrc.h:
7487 Define the default property values in the usual place.
7488 Implement start/stop to reset values correctly.
7489 Calculate the sample size only once when we negotiate.
7490 Rename some values to make more sense.
7491 Keep track of our byte range.
7492 Add support for pull based scheduling. Disabled for now until we have
7493 the whole stack working.
7494 Set the BUFFER_OFFSET correctly.
7496 2008-10-10 15:32:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7498 Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
7499 Original commit message from CVS:
7500 Based on a patch by: xavierb at gmail dot com
7501 * gst/subparse/gstsubparse.c:
7502 (gst_sub_parse_data_format_autodetect):
7503 * tests/check/elements/subparse.c: (GST_START_TEST):
7504 Make the detection of the used subtitle a bit less strict
7505 for srt subtitles. Fixes bug #555607.
7507 2008-10-10 15:21:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7509 ext/vorbis/vorbisenc.c: Fix discontinuity detection which was broken by last commit.
7510 Original commit message from CVS:
7511 * ext/vorbis/vorbisenc.c:
7512 (gst_vorbis_enc_buffer_check_discontinuous):
7513 Fix discontinuity detection which was broken by last commit.
7515 2008-10-09 11:18:09 +0000 Tim-Philipp Müller <tim@centricular.net>
7517 configure.ac: Require core CVS for ghostpad API additions used by decodebin2.
7518 Original commit message from CVS:
7520 Require core CVS for ghostpad API additions used by decodebin2.
7522 2008-10-08 15:30:33 +0000 Edward Hervey <bilboed@bilboed.com>
7524 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
7525 Original commit message from CVS:
7526 * gst-libs/gst/audio/gstbaseaudiosrc.c:
7527 (gst_base_audio_src_create):
7528 Fix debug statements (space between '%' and actual format).
7530 2008-10-08 14:44:04 +0000 Wim Taymans <wim.taymans@gmail.com>
7532 gst/playback/gstdecodebin2.c: Remove bogus assert, the decodepad could have been created inside an already existing g...
7533 Original commit message from CVS:
7534 * gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
7535 Remove bogus assert, the decodepad could have been created inside an
7536 already existing group.
7538 2008-10-08 14:01:42 +0000 Andy Wingo <wingo@pobox.com>
7542 Original commit message from CVS:
7545 2008-10-08 14:00:07 +0000 Andy Wingo <wingo@pobox.com>
7547 gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
7548 Original commit message from CVS:
7549 2008-10-08 Andy Wingo <wingo@pobox.com>
7550 * gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
7551 target instead of setting it.
7552 (gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
7553 API for a decode pad. The bugfix is that we set the group in
7554 activate(), not when the pad was created because it might be NULL
7556 (gst_decode_group_control_source_pad, gst_decode_group_expose):
7557 Update to use the API.
7559 2008-10-08 12:49:40 +0000 Andy Wingo <wingo@pobox.com>
7561 gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
7562 Original commit message from CVS:
7563 2008-10-08 Andy Wingo <wingo@pobox.com>
7564 * gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
7565 be a subclass of GstGhostPad.
7566 (analyze_new_pad): So, when emitting the signals that determine
7567 how we do autoplugging, already create the ghost pad and use it as
7568 the pad in the signal arguments. This allows applications to make
7569 a connection between the pad passed in e.g. autoplug-continue, and
7570 the pad passed in new-decoded-pad.
7571 (connect_pad, expose_pad): Update to receive the ghosted decode
7572 pad in the args, retargetting it as necessary if we have to plug
7573 the target pad through a multiqueue.
7574 (gst_decode_group_control_source_pad): Adapt to receive an
7575 already-ghosted pad that just needs activation, blocking, and
7577 (sort_end_pads): Adapt for decode pads actually being pads.
7578 (gst_decode_group_expose): Adapt for decode pads actually being
7579 pads. Rewrite the decode pad names so they appear in order. Adds a
7580 new error case if we couldn't set the name.
7581 (gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
7583 (gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
7584 New API for the decode pad, needed because we shouldn't do these
7585 things inside gst_decode_pad_new(), but after.
7586 (gst_decode_pad_new): Change to actually make the real pad, and
7587 delay the blocking/drainage bits.
7589 2008-10-08 12:12:01 +0000 Daniel Drake <dsd@laptop.org>
7591 ext/ogg/gstoggmux.c: Unref all buffers when clearing collectpads. Fixes bug #546955.
7592 Original commit message from CVS:
7593 Patch by: Daniel Drake <dsd at laptop dot org>
7594 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
7595 Unref all buffers when clearing collectpads. Fixes bug #546955.
7597 2008-10-08 12:08:01 +0000 Klaas <klaas@rivercrew.net>
7599 ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
7600 Original commit message from CVS:
7601 Based on a patch by: Klaas <klaas at rivercrew dot net>
7602 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
7603 (gst_vorbis_enc_buffer_check_discontinuous),
7604 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
7605 * ext/vorbis/vorbisenc.h:
7606 Keep track of the upstream segments and use the running time on that
7607 segment instead of the buffer timestamp everywhere. Fixes bug #525807.
7609 2008-10-08 11:50:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7611 gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
7612 Original commit message from CVS:
7613 * gst/audioconvert/audioconvert.c: (audio_convert_convert):
7614 Prevent overflows with big buffer when calculating the size of
7615 the intermediate buffer by using gst_util_uint64_scale() instead of
7616 plain arithmetics. Fixes bug #552801.
7618 2008-10-08 10:49:15 +0000 Pavel Zeldin <pzeldin@gmail.com>
7620 ext/pango/gstclockoverlay.*: API: Add ability to specify format for date/time display by adding a "time-format" prope...
7621 Original commit message from CVS:
7622 Patch by: Pavel Zeldin <pzeldin at gmail dot com>
7623 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
7624 (gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
7625 (gst_clock_overlay_init), (gst_clock_overlay_set_property),
7626 (gst_clock_overlay_get_property):
7627 * ext/pango/gstclockoverlay.h:
7628 API: Add ability to specify format for date/time display by
7629 adding a "time-format" property.
7632 2008-10-08 09:22:26 +0000 Jan Gerber <j@oil21.org>
7634 gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
7635 Original commit message from CVS:
7636 Patch by: Jan Gerber <j at oil21 dot org>
7637 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
7638 (gst_riff_create_video_template_caps):
7639 Add FFV1 fourcc to support playback of FFMPEG lossless video
7640 in AVI. Fixes bug #555319.
7642 2008-10-08 09:12:36 +0000 Håvard Graff <havard.graff@tandberg.com>
7644 gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
7645 Original commit message from CVS:
7646 Patch by: Håvard Graff <havard dot graff at tandberg dot com>
7647 * gst-libs/gst/audio/gstbaseaudiosrc.c:
7648 (gst_base_audio_src_create):
7649 Implement skew clock slaving. Fixes #552559.
7651 2008-10-08 09:10:23 +0000 Wim Taymans <wim.taymans@gmail.com>
7653 gst-libs/gst/audio/: Fix include of config.h
7654 Original commit message from CVS:
7655 * gst-libs/gst/audio/multichannel.c:
7656 * gst-libs/gst/audio/testchannels.c:
7657 Fix include of config.h
7659 2008-10-06 16:36:20 +0000 Tero Saarni <tero.saarni@gmail.com>
7661 gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
7662 Original commit message from CVS:
7663 Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
7664 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
7665 (print_media), (gst_sdp_message_dump):
7666 Fix parsing of the c= field containing multicast addresses.
7668 Add the connection info to the session or streams.
7669 Fix parsing of the bandwidth.
7670 Add debugging for the connections and bandwidths for a media.
7671 Add debugging for the bandwidth of the session.
7673 2008-10-06 16:31:27 +0000 Wim Taymans <wim.taymans@gmail.com>
7675 gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
7676 Original commit message from CVS:
7677 * gst-libs/gst/rtp/gstbasertppayload.c:
7678 (gst_basertppayload_change_state):
7679 Configure the next seqnum and timestamp in the state change so that they
7680 can be queried soon after.
7682 2008-10-06 16:29:33 +0000 Wim Taymans <wim.taymans@gmail.com>
7684 gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
7685 Original commit message from CVS:
7686 * gst-libs/gst/rtp/gstbasertpdepayload.c:
7687 (gst_base_rtp_depayload_chain):
7688 Improve debugging of the rtptime.
7690 2008-10-05 11:33:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7692 configure.ac: Back to development -> 0.10.21.1
7693 Original commit message from CVS:
7695 Back to development -> 0.10.21.1
7697 2008-10-05 08:18:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7701 Original commit message from CVS:
7704 2008-10-05 08:11:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7706 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
7707 Original commit message from CVS:
7708 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
7710 Add typefinder for MXF.
7712 2008-10-05 08:10:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7714 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
7715 Original commit message from CVS:
7716 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
7718 Add typefinder for MXF.
7720 2008-10-03 15:19:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7722 tests/icles/Makefile.am: Only build test-colorkey if GTK+ is available.
7723 Original commit message from CVS:
7724 * tests/icles/Makefile.am:
7725 Only build test-colorkey if GTK+ is available.
7727 === release 0.10.21 ===
7729 2008-10-03 00:03:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7735 * docs/plugins/gst-plugins-base-plugins.args:
7736 * docs/plugins/gst-plugins-base-plugins.hierarchy:
7737 * docs/plugins/gst-plugins-base-plugins.interfaces:
7738 * docs/plugins/gst-plugins-base-plugins.prerequisites:
7739 * docs/plugins/inspect/plugin-adder.xml:
7740 * docs/plugins/inspect/plugin-alsa.xml:
7741 * docs/plugins/inspect/plugin-audioconvert.xml:
7742 * docs/plugins/inspect/plugin-audiorate.xml:
7743 * docs/plugins/inspect/plugin-audioresample.xml:
7744 * docs/plugins/inspect/plugin-audiotestsrc.xml:
7745 * docs/plugins/inspect/plugin-cdparanoia.xml:
7746 * docs/plugins/inspect/plugin-decodebin.xml:
7747 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
7748 * docs/plugins/inspect/plugin-gdp.xml:
7749 * docs/plugins/inspect/plugin-gio.xml:
7750 * docs/plugins/inspect/plugin-gnomevfs.xml:
7751 * docs/plugins/inspect/plugin-libvisual.xml:
7752 * docs/plugins/inspect/plugin-ogg.xml:
7753 * docs/plugins/inspect/plugin-pango.xml:
7754 * docs/plugins/inspect/plugin-playback.xml:
7755 * docs/plugins/inspect/plugin-queue2.xml:
7756 * docs/plugins/inspect/plugin-subparse.xml:
7757 * docs/plugins/inspect/plugin-tcp.xml:
7758 * docs/plugins/inspect/plugin-theora.xml:
7759 * docs/plugins/inspect/plugin-typefindfunctions.xml:
7760 * docs/plugins/inspect/plugin-uridecodebin.xml:
7761 * docs/plugins/inspect/plugin-video4linux.xml:
7762 * docs/plugins/inspect/plugin-videorate.xml:
7763 * docs/plugins/inspect/plugin-videoscale.xml:
7764 * docs/plugins/inspect/plugin-videotestsrc.xml:
7765 * docs/plugins/inspect/plugin-volume.xml:
7766 * docs/plugins/inspect/plugin-vorbis.xml:
7767 * docs/plugins/inspect/plugin-ximagesink.xml:
7768 * docs/plugins/inspect/plugin-xvimagesink.xml:
7769 * gst-plugins-base.doap:
7770 * win32/common/config.h:
7772 Original commit message from CVS:
7775 2008-10-02 23:44:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7806 Original commit message from CVS:
7809 2008-09-28 22:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7811 configure.ac: 0.10.20.4 pre-release
7812 Original commit message from CVS:
7814 0.10.20.4 pre-release
7816 2008-09-25 10:46:00 +0000 ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>
7818 ext/theora/theoraparse.c: Set the BOS flag on the BOS packet. Fixes #553244.
7819 Original commit message from CVS:
7820 Patch by: ogg.k.ogg.k <ogg dot k dot ogg dot k at googlemail dot com>
7821 * ext/theora/theoraparse.c: (theora_parse_set_streamheader):
7822 Set the BOS flag on the BOS packet. Fixes #553244.
7824 2008-09-23 17:48:14 +0000 Wim Taymans <wim.taymans@gmail.com>
7826 gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
7827 Original commit message from CVS:
7828 * gst-libs/gst/rtsp/gstrtspmessage.c:
7829 (gst_rtsp_message_parse_request),
7830 (gst_rtsp_message_parse_response):
7831 Fix the g_return_val_if_fail() statements.
7833 2008-09-22 17:44:14 +0000 Michael Smith <msmith@xiph.org>
7835 gst-libs/gst/tag/gsttagdemux.c: Fail to activate if there's insufficient data in the file to be usable, preventing an...
7836 Original commit message from CVS:
7837 * gst-libs/gst/tag/gsttagdemux.c:
7838 Fail to activate if there's insufficient data in the file to be usable,
7839 preventing an assertion fail later. Fixes #552960
7841 2008-09-16 15:36:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7843 Commit stuff that should have gone in last week when I made the pre-releases:
7844 Original commit message from CVS:
7845 Commit stuff that should have gone in last week when I made the pre-releases:
7846 2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
7848 0.10.20.2 pre-release
7854 2008-09-15 15:11:18 +0000 Tim-Philipp Müller <tim@centricular.net>
7856 gst/: Recognise Kate subtitle streams (#550582).
7857 Original commit message from CVS:
7858 * gst-libs/gst/pbutils/descriptions.c:
7859 * gst/typefind/gsttypefindfunctions.c:
7860 Recognise Kate subtitle streams (#550582).
7862 2008-09-13 11:04:02 +0000 Tim-Philipp Müller <tim@centricular.net>
7864 gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
7865 Original commit message from CVS:
7866 * gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
7867 Remove trailing comma from enum list, which causes problems
7868 with -pendantic (#550729).
7870 2008-09-05 19:04:47 +0000 Tim-Philipp Müller <tim@centricular.net>
7872 gst-libs/gst/interfaces/propertyprobe.c: More sanity checks for our second-favourite interface.
7873 Original commit message from CVS:
7874 * gst-libs/gst/interfaces/propertyprobe.c:
7875 (gst_property_probe_get_properties),
7876 (gst_property_probe_get_property),
7877 (gst_property_probe_probe_property),
7878 (gst_property_probe_probe_property_name),
7879 (gst_property_probe_needs_probe),
7880 (gst_property_probe_needs_probe_name),
7881 (gst_property_probe_get_values),
7882 (gst_property_probe_get_values_name),
7883 (gst_property_probe_probe_and_get_values),
7884 (gst_property_probe_probe_and_get_values_name):
7885 More sanity checks for our second-favourite interface.
7887 2008-09-05 14:12:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7889 gst-libs/gst/interfaces/propertyprobe.c: Check for NULL pointer, in the hope that this fixes #532864.
7890 Original commit message from CVS:
7891 * gst-libs/gst/interfaces/propertyprobe.c:
7892 Check for NULL pointer, in the hope that this fixes #532864.
7894 2008-09-05 10:24:05 +0000 Tim-Philipp Müller <tim@centricular.net>
7896 sys/xvimage/xvimagesink.c: No really, the next release is 0.10.21 (fix Since: tags in docs).
7897 Original commit message from CVS:
7898 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
7899 No really, the next release is 0.10.21 (fix Since: tags in docs).
7901 2008-09-04 16:25:06 +0000 Wim Taymans <wim.taymans@gmail.com>
7903 gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
7904 Original commit message from CVS:
7905 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
7906 Disable a code path that is now called but causes a deadlock for some
7907 reason and is unneeded.
7909 2008-09-04 13:46:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7911 sys/xvimage/xvimagesink.*: Add a "draw-border" property that can be set to false to disable drawing borders.
7912 Original commit message from CVS:
7913 * sys/xvimage/xvimagesink.c:
7914 * sys/xvimage/xvimagesink.h:
7915 Add a "draw-border" property that can be set to false to disable
7917 * tests/icles/test-colorkey.c:
7918 * tests/icles/Makefile.am:
7919 Add new test application for the colorkey handling.
7921 2008-09-03 14:00:06 +0000 Edward Hervey <bilboed@bilboed.com>
7923 gst-libs/gst/riff/riff-media.c: Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
7924 Original commit message from CVS:
7925 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
7926 Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
7927 This will also be fixed for upcoming gst-ffmpeg release so that once
7928 this release of -base is out, it will work with the latest gst-ffmpeg
7931 2008-09-03 13:27:20 +0000 Edward Hervey <bilboed@bilboed.com>
7933 gst-libs/gst/riff/riff-media.c: Add Truespeech mapping for RIFF formats (AVI/WAV).
7934 Original commit message from CVS:
7935 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
7936 (gst_riff_create_audio_template_caps):
7937 Add Truespeech mapping for RIFF formats (AVI/WAV).
7940 2008-09-03 12:23:44 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
7942 gst/typefind/gsttypefindfunctions.c: Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
7943 Original commit message from CVS:
7944 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
7945 Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
7948 2008-09-03 10:12:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7950 Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
7951 Original commit message from CVS:
7953 * gst/subparse/Makefile.am:
7954 * gst/subparse/gstsubparse.c:
7955 * gst/subparse/samiparse.c:
7956 * tests/check/elements/subparse.c:
7957 Rework last change, so that we build subparse, but just disable the
7958 sami parse functionality, if we're configured to not use xml. In the
7959 tests only the sami test is disabled now.
7961 2008-09-02 15:07:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7963 configure.ac: Disable subparse when xml is disabled. It woundn't work anyway. Fixes test runs.
7964 Original commit message from CVS:
7966 Disable subparse when xml is disabled. It woundn't work anyway. Fixes
7969 2008-09-02 09:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
7971 po/POTFILES.in: Add some more files with strings for translation.
7972 Original commit message from CVS:
7974 Add some more files with strings for translation.
7976 2008-09-02 06:37:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7978 Use new geo location tags from core. Fixes #481169
7979 Original commit message from CVS:
7980 * gst-libs/gst/tag/gstvorbistag.c:
7981 * tests/check/libs/tag.c:
7982 Use new geo location tags from core. Fixes #481169
7984 2008-09-01 16:05:45 +0000 Edward Hervey <bilboed@bilboed.com>
7986 tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
7987 Original commit message from CVS:
7988 * tests/check/elements/audioresample.c: (setup_audioresample),
7989 (fail_unless_perfect_stream), (test_perfect_stream_instance),
7990 (test_discont_stream_instance):
7991 Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
7992 Add debugging for coherence.
7994 2008-08-30 15:55:06 +0000 Jonathan Matthew <notverysmart@gmail.com>
7996 gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
7997 Original commit message from CVS:
7998 Patch by: Jonathan Matthew <notverysmart gmail com>
7999 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
8000 Add typefinder for PDF documents (which is nice to have, since it's a
8001 common format, but also helps prevent false positives). Fixes #549814.
8003 2008-08-27 15:30:16 +0000 Wim Taymans <wim.taymans@gmail.com>
8005 gst/playback/gstplaybin2.c: Fix nasty race where multiple decodebins could start pushing data before we manage to con...
8006 Original commit message from CVS:
8007 * gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
8009 Fix nasty race where multiple decodebins could start pushing data before
8010 we manage to configure the sinks, resulting in not-linked errors in
8011 typical RTSP streaming cases.
8013 2008-08-26 17:24:31 +0000 Wim Taymans <wim.taymans@gmail.com>
8015 gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
8016 Original commit message from CVS:
8017 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
8018 Since we now call stop, we trigger this code path that causes a deadlock
8019 is apparently not needed.
8021 2008-08-26 15:45:36 +0000 Wim Taymans <wim.taymans@gmail.com>
8023 gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
8024 Original commit message from CVS:
8025 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
8026 (gst_ring_buffer_stop):
8027 Also allow the case where the ringbuffer was paused when we try to stop
8028 it so that the basesrc stop function is still called.
8030 2008-08-23 15:25:44 +0000 Mike Ruprecht <cmaiku@gmail.com>
8032 sys/v4l/gstv4lelement.c: Reprobe devices again instead of taking a cached list as new devices could've been plugged i...
8033 Original commit message from CVS:
8034 Patch by: Mike Ruprecht <cmaiku at gmail dot com>
8035 * sys/v4l/gstv4lelement.c: (gst_v4l_class_probe_devices):
8036 Reprobe devices again instead of taking a cached list as new
8037 devices could've been plugged in. Fixes bug #549062.
8039 2008-08-23 15:19:59 +0000 Alessandro Dessina <alessandro@nnva.org>
8041 ext/ogg/gstoggdemux.c: Don't add pads and activate them for skeleton streams. These are already handled inside oggdem...
8042 Original commit message from CVS:
8043 Patch by: Alessandro Dessina <alessandro nnva org>
8044 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
8045 (gst_ogg_demux_activate_chain):
8046 Don't add pads and activate them for skeleton streams. These are already
8047 handled inside oggdemux. Fixes bug #537599.
8049 2008-08-22 15:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
8051 ext/vorbis/vorbisdec.c: Reset variable so that query and convert fail after going back to
8052 Original commit message from CVS:
8053 * ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
8054 Reset variable so that query and convert fail after going back to
8055 READY. Fixes #548898.
8057 2008-08-22 07:24:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8059 ext/vorbis/vorbisenc.c: If a buffer arrives with a timestamp before the timestamp+duration of the previous buffer cli...
8060 Original commit message from CVS:
8061 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
8062 If a buffer arrives with a timestamp before the timestamp+duration
8063 of the previous buffer clip it instead of dropping it completely.
8064 Slight improvement for the unfixable bug #548913.
8066 2008-08-21 14:19:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8068 ext/vorbis/vorbisdec.c: Take the current timestamp instead of timestamp+duration for the offset.
8069 Original commit message from CVS:
8070 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
8071 Take the current timestamp instead of timestamp+duration for the offset.
8072 This offset will later be used for calculating the timestamp and
8073 otherwise vorbisdec will interpolate timestamps wrong if upstream
8074 only sends timestamps and no granulepos.
8076 2008-08-21 11:20:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8078 tests/examples/seek/seek.c: Don't crash when having no visualisations.
8079 Original commit message from CVS:
8080 * tests/examples/seek/seek.c:
8081 Don't crash when having no visualisations.
8083 2008-08-16 20:57:27 +0000 David Schleef <ds@schleef.org>
8085 gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
8086 Original commit message from CVS:
8087 * gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
8088 check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
8091 2008-08-15 07:24:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8093 gst-libs/gst/pbutils/missing-plugins.c: When cleaning up the caps fields also remove "depth" for the same reason we r...
8094 Original commit message from CVS:
8095 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
8096 When cleaning up the caps fields also remove "depth" for the same
8097 reason we remove "width".
8099 2008-08-14 17:14:53 +0000 Tim-Philipp Müller <tim@centricular.net>
8101 gst-libs/gst/pbutils/descriptions.c: Add Lead H.264 here as well.
8102 Original commit message from CVS:
8103 * gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
8104 Add Lead H.264 here as well.
8106 2008-08-14 15:17:31 +0000 Julien Moutte <julien@moutte.net>
8108 gst-libs/gst/riff/riff-media.c: Add Lead H.264 variant.
8109 Original commit message from CVS:
8110 2008-08-14 Julien Moutte <julien@fluendo.com>
8111 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
8112 (gst_riff_create_video_template_caps): Add Lead H.264 variant.
8114 2008-08-13 09:17:38 +0000 Wim Taymans <wim.taymans@gmail.com>
8116 gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
8117 Original commit message from CVS:
8118 * gst-libs/gst/audio/gstbaseaudiosrc.c:
8119 (gst_base_audio_src_create):
8120 When not slaved to another clock also subtract the base_time from our
8121 internal clock time to get the running time.
8123 2008-08-13 00:59:07 +0000 David Schleef <ds@schleef.org>
8125 ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate, since it has no basis in libtheora.
8126 Original commit message from CVS:
8127 * ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
8128 since it has no basis in libtheora.
8130 2008-08-12 06:31:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8132 gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
8133 Original commit message from CVS:
8134 * gst-libs/gst/interfaces/propertyprobe.h:
8135 Remove double "interface" from doc-string.
8136 * gst-libs/gst/interfaces/xoverlay.h:
8138 * gst-libs/gst/riff/riff.c:
8139 Add basic doc blobs.
8141 2008-08-11 15:05:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8143 gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
8144 Original commit message from CVS:
8145 * gst-libs/gst/audio/Makefile.am:
8146 Don't try to build that example anymore.
8148 2008-08-11 14:51:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8150 gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
8151 Original commit message from CVS:
8152 * gst-libs/gst/audio/.cvsignore:
8153 * gst-libs/gst/audio/Makefile.am:
8154 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
8155 * gst-libs/gst/audio/make_filter:
8156 Move audiofiltertemplate to gst-template.
8158 2008-08-11 09:20:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8160 More docs and shuffling. What can we do with the hundreds of #defines.
8161 Original commit message from CVS:
8162 * docs/libs/gst-plugins-base-libs-sections.txt:
8163 * gst-libs/gst/audio/gstaudiosrc.h:
8164 More docs and shuffling. What can we do with the hundreds of #defines.
8166 2008-08-11 08:34:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8168 gst-libs/gst/: Reducing number of dundocumented symbols.
8169 Original commit message from CVS:
8170 * gst-libs/gst/audio/audio.h:
8171 * gst-libs/gst/audio/gstaudiofilter.h:
8172 * gst-libs/gst/audio/gstringbuffer.h:
8173 * gst-libs/gst/interfaces/propertyprobe.h:
8174 * gst-libs/gst/tag/gsttagdemux.h:
8175 Reducing number of dundocumented symbols.
8177 2008-08-11 07:16:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8179 gst-libs/gst/audio/audio.c: Fix doc comment syntax.
8180 Original commit message from CVS:
8181 * gst-libs/gst/audio/audio.c:
8182 Fix doc comment syntax.
8183 * gst-libs/gst/interfaces/propertyprobe.c:
8184 Add more doc-comments and a FIXME: for the signal.
8186 2008-08-07 16:11:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8188 ext/ogg/gstoggmux.*: Don't pretend to support NEWSEGMENT events, instead override the
8189 Original commit message from CVS:
8190 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sink_event),
8191 (gst_ogg_mux_request_new_pad):
8192 * ext/ogg/gstoggmux.h:
8193 Don't pretend to support NEWSEGMENT events, instead override the
8194 GstCollectPads event function to return FALSE on NEWSEGMENT events
8195 and do the normal work for other events.
8196 This prevents elements like flacenc to seek to the start and rewrite
8197 some data which then results in a broken Ogg packet.
8199 2008-08-07 15:58:58 +0000 Frederic Crozat <fcrozat@mandriva.org>
8201 Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
8202 Original commit message from CVS:
8203 Patch by: Frederic Crozat <fcrozat@mandriva.org>
8204 * ext/alsa/gstalsaplugin.c: (plugin_init):
8205 * ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
8206 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
8207 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
8208 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
8209 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
8210 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
8211 * gst/playback/gstdecodebin.c: (plugin_init):
8212 * gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
8213 * gst/playback/gstplayback.c: (plugin_init):
8214 * gst/playback/gstqueue2.c: (plugin_init):
8215 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
8216 * sys/v4l/gstv4l.c: (plugin_init):
8217 Make sure gettext returns translations in UTF-8 encoding rather
8218 than in the current locale encoding (#546822).
8220 2008-08-06 13:12:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8222 gst-libs/gst/pbutils/descriptions.c: Add audio/x-qdm for qtdemux.
8223 Original commit message from CVS:
8224 * gst-libs/gst/pbutils/descriptions.c:
8225 Add audio/x-qdm for qtdemux.
8227 2008-08-05 15:38:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8229 ext/vorbis/vorbisdec.c: Do not leak old taglist.
8230 Original commit message from CVS:
8231 * ext/vorbis/vorbisdec.c:
8232 Do not leak old taglist.
8234 2008-08-04 12:35:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8236 tests/icles/test-scale.c: Include <stdlib.h> for atoi().
8237 Original commit message from CVS:
8238 * tests/icles/test-scale.c:
8239 Include <stdlib.h> for atoi().
8241 2008-08-04 09:11:08 +0000 Andy Wingo <wingo@pobox.com>
8243 gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
8244 Original commit message from CVS:
8245 2008-08-04 Andy Wingo <wingo@pobox.com>
8246 * gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
8249 2008-08-01 13:06:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8251 gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
8252 Original commit message from CVS:
8253 * gst/adder/gstadder.c:
8254 Cleanup lots of empty lines that came from gst-indent going havoc
8255 before I added the INDENT_ON/OFF marker some time agao.
8257 2008-08-01 11:55:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8259 Bump requirement to latest core and use new tag for riff formats.
8260 Original commit message from CVS:
8262 * gst-libs/gst/riff/riff-read.c:
8263 Bump requirement to latest core and use new tag for riff formats.
8266 2008-08-01 11:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
8268 tests/examples/dynamic/: Add example app that dynamically switches between 3 'encoders'.
8269 Original commit message from CVS:
8270 * tests/examples/dynamic/Makefile.am:
8271 * tests/examples/dynamic/codec-select.c: (make_encoder),
8272 (make_pipeline), (do_switch), (my_bus_callback), (main):
8273 Add example app that dynamically switches between 3 'encoders'.
8275 2008-07-31 13:06:13 +0000 Wim Taymans <wim.taymans@gmail.com>
8277 gst/playback/gstplaysink.c: Add some more comments.
8278 Original commit message from CVS:
8279 * gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
8280 Add some more comments.
8282 2008-07-31 12:58:44 +0000 Wim Taymans <wim.taymans@gmail.com>
8284 gst/videotestsrc/gstvideotestsrc.c: Discard buffers of the wrong size after renegotiation, this is perfectly possible...
8285 Original commit message from CVS:
8286 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
8287 (gst_video_test_src_create):
8288 Discard buffers of the wrong size after renegotiation, this is perfectly
8289 possible with things like capsfilter that could suggest caps changes
8290 upstream without knowing the size of the buffer.
8292 2008-07-31 11:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
8294 tests/icles/: Add dynamic rescaling tests for the new basetransform.
8295 Original commit message from CVS:
8296 * tests/icles/.cvsignore:
8297 * tests/icles/Makefile.am:
8298 * tests/icles/test-scale.c: (make_pipeline), (main):
8299 Add dynamic rescaling tests for the new basetransform.
8301 2008-07-30 19:51:36 +0000 Tim-Philipp Müller <tim@centricular.net>
8303 gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
8304 Original commit message from CVS:
8305 * gst/audioconvert/Makefile.am:
8306 Dist recently-added gstfastrandom.h.
8308 2008-07-30 15:29:44 +0000 Edward Hervey <bilboed@bilboed.com>
8310 sys/xvimage/xvimagesink.c: Fix a "may be used uninitialized in this function" which weirdly only appears on macosx (?).
8311 Original commit message from CVS:
8312 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
8313 Fix a "may be used uninitialized in this function" which weirdly only
8314 appears on macosx (?).
8316 2008-07-30 09:02:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8318 gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
8319 Original commit message from CVS:
8320 * gst-libs/gst/riff/riff-ids.h:
8321 Adding acid chunk for tempo and loop information.
8323 2008-07-29 13:01:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8325 sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
8326 Original commit message from CVS:
8327 * sys/xvimage/Makefile.am:
8328 floor() needs linking to $(LIBM).
8330 2008-07-29 12:35:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8332 ext/gnomevfs/gstgnomevfssrc.c: Aggregate short reads and add some comments and debug logging.
8333 Original commit message from CVS:
8334 * ext/gnomevfs/gstgnomevfssrc.c:
8335 Aggregate short reads and add some comments and debug logging.
8338 2008-07-29 10:26:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8340 gst/playback/gstplaybasebin.c: Fix property doc markup (its not a signal).
8341 Original commit message from CVS:
8342 * gst/playback/gstplaybasebin.c:
8343 Fix property doc markup (its not a signal).
8344 * sys/xvimage/xvimagesink.c:
8345 Add since tag for new proeprties (also add sice tags fro the last two
8348 2008-07-29 08:59:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8350 sys/xvimage/xvimagesink.*: Add autofill/colorkey properties. Fixes #538656.
8351 Original commit message from CVS:
8352 * sys/xvimage/xvimagesink.c:
8353 * sys/xvimage/xvimagesink.h:
8354 Add autofill/colorkey properties. Fixes #538656.
8356 2008-07-29 01:58:05 +0000 David Schleef <ds@schleef.org>
8358 sys/xvimage/xvimagesink.c: Fix rounding errors when converting colorbalance values between hardware and object proper...
8359 Original commit message from CVS:
8360 * sys/xvimage/xvimagesink.c:
8361 Fix rounding errors when converting colorbalance values
8362 between hardware and object property ranges. Partial
8363 fix for #537889, however, there still seems to be a small
8364 drift problem that could be totem's fault.
8366 2008-07-28 15:34:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8368 ext/ogg/gstoggdemux.c: Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
8369 Original commit message from CVS:
8370 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
8371 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
8372 Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
8373 This fixes a critical warning.
8375 2008-07-28 13:12:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8377 ext/ogg/gstoggmux.c: Allow muxing of CELT into Ogg streams.
8378 Original commit message from CVS:
8379 * ext/ogg/gstoggmux.c:
8380 Allow muxing of CELT into Ogg streams.
8382 2008-07-28 12:47:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8384 gst/typefind/gsttypefindfunctions.c: Add simple typefinder for the CELT codec (www.celt-codec.org).
8385 Original commit message from CVS:
8386 * gst/typefind/gsttypefindfunctions.c: (celt_type_find),
8388 Add simple typefinder for the CELT codec (www.celt-codec.org).
8390 2008-07-27 11:12:41 +0000 Jan Gerber <j@oil21.org>
8392 ext/ogg/gstoggdemux.c: Fix calculation of the start time from skeleton streams.
8393 Original commit message from CVS:
8394 Patch by: Jan Gerber <j at oil21 dot org>
8395 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
8396 Fix calculation of the start time from skeleton streams.
8399 2008-07-24 13:19:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8401 tests/examples/seek/seek.c: Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
8402 Original commit message from CVS:
8403 * tests/examples/seek/seek.c:
8404 Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
8406 2008-07-23 18:34:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8408 gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
8409 Original commit message from CVS:
8410 * gst/audioconvert/audioconvert.h:
8411 * gst/audioconvert/gstaudioquantize.c:
8412 (gst_audio_quantize_setup_dither),
8413 (gst_audio_quantize_free_dither):
8414 * gst/audioconvert/gstfastrandom.h:
8415 Implement a linear congruential generator as pseudo random number
8416 generator for the dither noise. This is about 2 times faster than
8417 using GLib's mersenne twister. Also this uses only integer math for
8418 generating integers while GLib internally uses floating point math.
8420 2008-07-23 18:27:15 +0000 Michael Smith <msmith@xiph.org>
8422 configure.ac: Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
8423 Original commit message from CVS:
8425 Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
8427 2008-07-23 13:17:31 +0000 Damien Lespiau <damien.lespiau@gmail.com>
8429 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
8430 Original commit message from CVS:
8431 Patch by: Damien Lespiau <damien.lespiau gmail com>
8432 * gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
8433 Use GST_STR_NULL to avoid crashes with libcs that don't
8434 like NULL strings in printf args (such as the win32 one).
8437 2008-07-17 14:21:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8439 sys/xvimage/xvimagesink.c: Oops - set the size of the image used for probing back to 1x1, for consistency with ximage...
8440 Original commit message from CVS:
8441 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
8442 Oops - set the size of the image used for probing back to 1x1, for
8443 consistency with ximagesink
8445 2008-07-17 13:57:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8447 sys/: it's not legal to ask the
8448 Original commit message from CVS:
8449 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
8450 (gst_ximagesink_ximage_new):
8451 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
8452 (gst_xvimagesink_xvimage_new):
8453 Apparently on Solaris and OS/X (at least), it's not legal to ask the
8454 X server to attach to a shared memory segment after we've deleted it,
8455 with the result that MIT-SHM is disabled. Instead, remove it only after
8456 X succeeds in attaching too.
8458 2008-07-17 02:30:24 +0000 David Schleef <ds@schleef.org>
8460 gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second.
8461 Original commit message from CVS:
8462 * gst/audiotestsrc/gstaudiotestsrc.c:
8463 * gst/audiotestsrc/gstaudiotestsrc.h:
8464 Add 'ticks', a 1/30 second sine wave pulse every second.
8466 2008-07-15 22:43:16 +0000 David Schleef <ds@schleef.org>
8468 gst-libs/gst/video/video.c: Revert ABI change.
8469 Original commit message from CVS:
8470 * gst-libs/gst/video/video.c: Revert ABI change.
8472 2008-07-15 13:05:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8474 gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
8475 Original commit message from CVS:
8476 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
8477 Make it impossible to have NULL caps at the point where we set
8478 framerate and other things. Also don't return immediately for "3ivd"
8479 video and let framerate, etc be set. Might fix bug #542508.
8481 2008-07-14 17:06:26 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
8483 gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
8484 Original commit message from CVS:
8485 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
8486 Video format can also be conveniently determined from (many)
8489 2008-07-14 08:18:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8491 gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
8492 Original commit message from CVS:
8493 * gst/playback/gstplaybasebin.c:
8494 * gst/playback/gstplaybasebin.h:
8495 * gst/playback/gstplaybin.c:
8496 * gst/playback/gststreamselector.c:
8497 First stab at integrating DVD subpicture overlay into
8498 playbin. Successfully plugs and plays, but the queues need
8499 shrinking - 3 seconds of video is too much buffering.
8501 2008-07-11 18:06:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8503 gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs.
8504 Original commit message from CVS:
8505 * gst/audioconvert/gstaudioconvert.c:
8506 Remove now obsolete note in the docs.
8508 2008-07-11 06:10:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8510 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
8511 Original commit message from CVS:
8512 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
8513 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
8514 * docs/plugins/gst-plugins-base-plugins-sections.txt:
8515 * docs/plugins/gst-plugins-base-plugins.args:
8516 * docs/plugins/gst-plugins-base-plugins.hierarchy:
8517 * docs/plugins/gst-plugins-base-plugins.interfaces:
8518 * docs/plugins/gst-plugins-base-plugins.prerequisites:
8519 * docs/plugins/gst-plugins-base-plugins.signals:
8520 * docs/plugins/inspect/plugin-adder.xml:
8521 * docs/plugins/inspect/plugin-alsa.xml:
8522 * docs/plugins/inspect/plugin-audioconvert.xml:
8523 * docs/plugins/inspect/plugin-audiorate.xml:
8524 * docs/plugins/inspect/plugin-audioresample.xml:
8525 * docs/plugins/inspect/plugin-audiotestsrc.xml:
8526 * docs/plugins/inspect/plugin-cdparanoia.xml:
8527 * docs/plugins/inspect/plugin-decodebin.xml:
8528 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
8529 * docs/plugins/inspect/plugin-gdp.xml:
8530 * docs/plugins/inspect/plugin-gnomevfs.xml:
8531 * docs/plugins/inspect/plugin-libvisual.xml:
8532 * docs/plugins/inspect/plugin-ogg.xml:
8533 * docs/plugins/inspect/plugin-pango.xml:
8534 * docs/plugins/inspect/plugin-playback.xml:
8535 * docs/plugins/inspect/plugin-queue2.xml:
8536 * docs/plugins/inspect/plugin-subparse.xml:
8537 * docs/plugins/inspect/plugin-tcp.xml:
8538 * docs/plugins/inspect/plugin-theora.xml:
8539 * docs/plugins/inspect/plugin-typefindfunctions.xml:
8540 * docs/plugins/inspect/plugin-uridecodebin.xml:
8541 * docs/plugins/inspect/plugin-video4linux.xml:
8542 * docs/plugins/inspect/plugin-videorate.xml:
8543 * docs/plugins/inspect/plugin-videoscale.xml:
8544 * docs/plugins/inspect/plugin-videotestsrc.xml:
8545 * docs/plugins/inspect/plugin-volume.xml:
8546 * docs/plugins/inspect/plugin-vorbis.xml:
8547 * docs/plugins/inspect/plugin-ximagesink.xml:
8548 * docs/plugins/inspect/plugin-xvimagesink.xml:
8549 * ext/alsa/gstalsamixer.c:
8550 * ext/alsa/gstalsasink.c:
8551 * ext/alsa/gstalsasrc.c:
8552 * ext/gio/gstgiosink.c:
8553 * ext/gio/gstgiosrc.c:
8554 * ext/gio/gstgiostreamsink.c:
8555 * ext/gio/gstgiostreamsrc.c:
8556 * ext/gnomevfs/gstgnomevfssink.c:
8557 * ext/gnomevfs/gstgnomevfssrc.c:
8558 * ext/ogg/gstoggdemux.c:
8559 * ext/ogg/gstoggmux.c:
8560 * ext/pango/gstclockoverlay.c:
8561 * ext/pango/gsttextoverlay.c:
8562 * ext/pango/gsttextrender.c:
8563 * ext/pango/gsttimeoverlay.c:
8564 * ext/theora/theoradec.c:
8565 * ext/theora/theoraenc.c:
8566 * ext/theora/theoraparse.c:
8567 * ext/vorbis/vorbisdec.c:
8568 * ext/vorbis/vorbisenc.c:
8569 * ext/vorbis/vorbisparse.c:
8570 * ext/vorbis/vorbistag.c:
8571 * gst/adder/gstadder.c:
8572 * gst/audioconvert/gstaudioconvert.c:
8573 * gst/audioresample/gstaudioresample.c:
8574 * gst/audiotestsrc/gstaudiotestsrc.c:
8575 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
8576 * gst/gdp/gstgdpdepay.c:
8577 * gst/gdp/gstgdppay.c:
8578 * gst/playback/gstdecodebin2.c:
8579 * gst/playback/gstplaybin.c:
8580 * gst/playback/gstplaybin2.c:
8581 * gst/playback/gstqueue2.c:
8582 * gst/playback/gsturidecodebin.c:
8583 * gst/tcp/gstmultifdsink.c:
8584 * gst/tcp/gsttcpserversink.c:
8585 * gst/videorate/gstvideorate.c:
8586 * gst/videoscale/gstvideoscale.c:
8587 * gst/videotestsrc/gstvideotestsrc.c:
8588 * gst/volume/gstvolume.c:
8589 * sys/ximage/ximagesink.c:
8590 * sys/xvimage/xvimagesink.c:
8591 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
8592 titles. Drop mentining that all our example pipelines are "simple"
8595 2008-07-10 21:06:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8597 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
8598 Original commit message from CVS:
8599 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
8600 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
8601 * docs/plugins/gst-plugins-base-plugins-sections.txt:
8602 * docs/plugins/gst-plugins-base-plugins.args:
8603 * docs/plugins/gst-plugins-base-plugins.hierarchy:
8604 * docs/plugins/gst-plugins-base-plugins.interfaces:
8605 * docs/plugins/gst-plugins-base-plugins.prerequisites:
8606 * docs/plugins/gst-plugins-base-plugins.signals:
8607 * docs/plugins/inspect/plugin-adder.xml:
8608 * docs/plugins/inspect/plugin-alsa.xml:
8609 * docs/plugins/inspect/plugin-audioconvert.xml:
8610 * docs/plugins/inspect/plugin-audiorate.xml:
8611 * docs/plugins/inspect/plugin-audioresample.xml:
8612 * docs/plugins/inspect/plugin-audiotestsrc.xml:
8613 * docs/plugins/inspect/plugin-cdparanoia.xml:
8614 * docs/plugins/inspect/plugin-decodebin.xml:
8615 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
8616 * docs/plugins/inspect/plugin-gdp.xml:
8617 * docs/plugins/inspect/plugin-gnomevfs.xml:
8618 * docs/plugins/inspect/plugin-libvisual.xml:
8619 * docs/plugins/inspect/plugin-ogg.xml:
8620 * docs/plugins/inspect/plugin-pango.xml:
8621 * docs/plugins/inspect/plugin-playback.xml:
8622 * docs/plugins/inspect/plugin-queue2.xml:
8623 * docs/plugins/inspect/plugin-subparse.xml:
8624 * docs/plugins/inspect/plugin-tcp.xml:
8625 * docs/plugins/inspect/plugin-theora.xml:
8626 * docs/plugins/inspect/plugin-typefindfunctions.xml:
8627 * docs/plugins/inspect/plugin-uridecodebin.xml:
8628 * docs/plugins/inspect/plugin-video4linux.xml:
8629 * docs/plugins/inspect/plugin-videorate.xml:
8630 * docs/plugins/inspect/plugin-videoscale.xml:
8631 * docs/plugins/inspect/plugin-videotestsrc.xml:
8632 * docs/plugins/inspect/plugin-volume.xml:
8633 * docs/plugins/inspect/plugin-vorbis.xml:
8634 * docs/plugins/inspect/plugin-ximagesink.xml:
8635 * docs/plugins/inspect/plugin-xvimagesink.xml:
8636 * ext/alsa/gstalsamixer.c:
8637 * ext/alsa/gstalsasink.c:
8638 * ext/alsa/gstalsasrc.c:
8639 * ext/gio/gstgiosink.c:
8640 * ext/gio/gstgiosrc.c:
8641 * ext/gio/gstgiostreamsink.c:
8642 * ext/gio/gstgiostreamsrc.c:
8643 * ext/gnomevfs/gstgnomevfssink.c:
8644 * ext/gnomevfs/gstgnomevfssrc.c:
8645 * ext/ogg/gstoggdemux.c:
8646 * ext/ogg/gstoggmux.c:
8647 * ext/pango/gstclockoverlay.c:
8648 * ext/pango/gsttextoverlay.c:
8649 * ext/pango/gsttextrender.c:
8650 * ext/pango/gsttimeoverlay.c:
8651 * ext/theora/theoradec.c:
8652 * ext/theora/theoraenc.c:
8653 * ext/theora/theoraparse.c:
8654 * ext/vorbis/vorbisdec.c:
8655 * ext/vorbis/vorbisenc.c:
8656 * ext/vorbis/vorbisparse.c:
8657 * ext/vorbis/vorbistag.c:
8658 * gst/adder/gstadder.c:
8659 * gst/audioconvert/gstaudioconvert.c:
8660 * gst/audioresample/gstaudioresample.c:
8661 * gst/audiotestsrc/gstaudiotestsrc.c:
8662 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
8663 * gst/gdp/gstgdpdepay.c:
8664 * gst/gdp/gstgdppay.c:
8665 * gst/playback/gstdecodebin2.c:
8666 * gst/playback/gstplaybin.c:
8667 * gst/playback/gstplaybin2.c:
8668 * gst/playback/gstqueue2.c:
8669 * gst/playback/gsturidecodebin.c:
8670 * gst/tcp/gstmultifdsink.c:
8671 * gst/tcp/gsttcpserversink.c:
8672 * gst/videorate/gstvideorate.c:
8673 * gst/videoscale/gstvideoscale.c:
8674 * gst/videotestsrc/gstvideotestsrc.c:
8675 * gst/volume/gstvolume.c:
8676 * sys/ximage/ximagesink.c:
8677 * sys/xvimage/xvimagesink.c:
8678 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
8679 titles. Drop mentining that all our example pipelines are "simple"
8682 2008-07-07 17:25:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8684 tests/examples/seek/Makefile.am: Fix out of tree build by adding all required CFLAGS.
8685 Original commit message from CVS:
8686 * tests/examples/seek/Makefile.am:
8687 Fix out of tree build by adding all required CFLAGS.
8689 2008-07-07 09:55:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8691 gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
8692 Original commit message from CVS:
8693 * gst/playback/gstdecodebin.c: (add_raw_queue):
8694 And ref the pad before returning it again when linking to the queue
8695 failed. Otherwise we will unref the pad twice later and things break.
8697 2008-07-07 09:48:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8699 gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
8700 Original commit message from CVS:
8701 * gst/playback/gstdecodebin.c: (add_raw_queue):
8702 If linking the raw pad with a queue fails, try it without a queue
8703 instead of failing completely. This should never happen.
8705 2008-07-06 23:22:12 +0000 Evgeniy Stepanov <eugeni.stepanov@gmail.com>
8707 gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
8708 Original commit message from CVS:
8709 Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
8710 * gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
8711 Add a queue after a demuxer if the demuxer outputs raw data. This was
8712 done before only for non-raw data but is required in this case too.
8714 decodebin2 doesn't have this issue because all streams of a group
8715 go through multiqueue.
8717 2008-07-03 09:12:49 +0000 Damien Lespiau <damien.lespiau@gmail.com>
8719 gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
8720 Original commit message from CVS:
8721 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
8722 * gst-libs/gst/sdp/gstsdpmessage.c:
8723 Makes libgstsdp compile with mingw32 by defining the right WINVER so
8724 that getaddrinfo() can be used. Fixes #541358.
8726 2008-07-01 13:22:49 +0000 Wim Taymans <wim.taymans@gmail.com>
8728 gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
8729 Original commit message from CVS:
8730 * gst/videotestsrc/gstvideotestsrc.c:
8731 (gst_video_test_src_class_init), (gst_video_test_src_init),
8732 (gst_video_test_src_set_property),
8733 (gst_video_test_src_get_property), (gst_video_test_src_create):
8734 * gst/videotestsrc/gstvideotestsrc.h:
8735 Cleanups, use default property values as defines.
8736 Add property to enable/disable peer buffer allocation.
8738 2008-06-30 09:46:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8740 tests/check/: Enable unit tests on PPC again as the bugs are now fixed.
8741 Original commit message from CVS:
8742 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
8743 * tests/check/pipelines/streamheader.c: (streamheader_suite):
8744 Enable unit tests on PPC again as the bugs are now fixed.
8746 2008-06-30 09:20:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8748 gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
8749 Original commit message from CVS:
8750 * gst-libs/gst/riff/riff-ids.h:
8751 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
8752 (gst_riff_create_audio_template_caps):
8753 Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
8756 2008-06-30 08:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8758 gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
8759 Original commit message from CVS:
8760 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
8761 (gst_ffmpeg_pixfmt_to_caps):
8762 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
8763 (gst_ffmpegcsp_get_unit_size):
8764 Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
8765 it on other formats. Also adjust the unit size only for that format
8766 to not include the palette. Fixes bug #540497.
8768 2008-06-29 13:45:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8770 gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
8771 Original commit message from CVS:
8772 * gst/adder/gstadder.c:
8773 Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
8775 2008-06-27 07:55:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8777 ChangeLog: ChangeLog surgery.
8778 Original commit message from CVS:
8781 * tests/examples/seek/seek.c:
8782 Move variable into ifdef too.
8784 2008-06-27 07:42:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8786 tests/examples/seek/seek.c: Include config.h and check if we have X. Fixes: #540334.
8787 Original commit message from CVS:
8788 * tests/examples/seek/seek.c:
8789 Include config.h and check if we have X. Fixes: #540334.
8791 2008-06-26 06:03:38 +0000 Sam Morris <sam@robots.org.to.uk>
8793 gst-libs/gst/interfaces/mixertrack.c: API: Add "index" property to GstMixerTrack to differantiate between multiple mi...
8794 Original commit message from CVS:
8795 Patch by: Sam Morris <sam at robots dot org to uk>
8796 * gst-libs/gst/interfaces/mixertrack.c:
8797 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
8798 (gst_mixer_track_set_property):
8799 API: Add "index" property to GstMixerTrack to differantiate between
8800 multiple mixer tracks with the same label.
8801 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
8802 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
8803 Set the "index" property of GstMixerTrack to the index given by ALSA.
8806 2008-06-25 13:15:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8808 tests/examples/seek/: Remove libgstvideo usage. Use gtk_get_option_group instead of gtk_init().
8809 Original commit message from CVS:
8810 * tests/examples/seek/Makefile.am:
8811 * tests/examples/seek/seek.c:
8812 Remove libgstvideo usage. Use gtk_get_option_group instead of
8815 2008-06-24 16:27:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8817 tests/check/Makefile.am: Name the test registry format neutral.
8818 Original commit message from CVS:
8819 * tests/check/Makefile.am:
8820 Name the test registry format neutral.
8822 2008-06-24 16:22:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8824 gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
8825 Original commit message from CVS:
8826 * gst/playback/gstqueue2.c:
8827 Do not double notify. Remove the unsued return value.
8829 2008-06-24 16:15:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8831 ext/alsa/gstalsamixer.c: Also consider "speaker" as a name for master volume. If that doesn't help look for the first...
8832 Original commit message from CVS:
8833 * ext/alsa/gstalsamixer.c:
8834 Also consider "speaker" as a name for master volume. If that doesn't
8835 help look for the first non-mono volume control that also has a
8838 2008-06-24 16:10:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8840 ChangeLog: Forgot to save the ChangeLog :/
8841 Original commit message from CVS:
8843 Forgot to save the ChangeLog :/
8845 2008-06-24 16:05:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8847 tests/examples/seek/: Embedd the xwindow.
8848 Original commit message from CVS:
8849 * tests/examples/seek/Makefile.am:
8850 * tests/examples/seek/seek.c:
8853 2008-06-24 01:14:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8855 sys/ximage/ximagesink.h: When the caps change, make sure to re-draw borders in force-aspect-ratio=true mode.
8856 Original commit message from CVS:
8857 * sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
8858 (gst_ximagesink_setcaps):
8859 * sys/ximage/ximagesink.h:
8860 When the caps change, make sure to re-draw borders in
8861 force-aspect-ratio=true mode.
8862 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
8863 Don't clear the border_draw flag until we actually draw the border.
8864 * tests/check/Makefile.am:
8865 Ignore alsasink/src during the states test too, so it doesn't fail
8866 when running without access to the sound device.
8868 2008-06-22 18:35:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8870 tests/examples/seek/seek.c: Fix crasher when playing a parse-launch line the 2nd time.
8871 Original commit message from CVS:
8872 * tests/examples/seek/seek.c:
8873 Fix crasher when playing a parse-launch line the 2nd time.
8875 2008-06-21 18:56:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
8877 tests/check/pipelines/oggmux.c: Properly ifdef tests to fix compilation.
8878 Original commit message from CVS:
8879 * tests/check/pipelines/oggmux.c:
8880 Properly ifdef tests to fix compilation.
8882 2008-06-21 10:25:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
8886 Original commit message from CVS:
8889 2008-06-20 18:24:24 +0000 Michael Smith <msmith@xiph.org>
8891 gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
8892 Original commit message from CVS:
8893 * gst/playback/gstplay-marshal.list:
8894 * gst/playback/gstplaybin2.c:
8895 Add get-video-pad, get-audio-pad, get-text-pad action signals to
8896 playbin2. This allows the user to get to the selector's sinkpads, and
8897 thus inspect a range of things - caps, tags, etc.
8899 2008-06-20 17:27:03 +0000 Michael Smith <msmith@xiph.org>
8901 gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
8902 Original commit message from CVS:
8903 * gst/playback/gstplaybin2.c:
8904 Use a different constant for the convert-frame signal id.
8907 2008-06-20 17:18:55 +0000 Michael Smith <msmith@xiph.org>
8909 gst/playback/: Fix a whole bunch of typos in comments and log statements.
8910 Original commit message from CVS:
8911 * gst/playback/gstplaybin2.c:
8912 * gst/playback/gstplaysink.c:
8913 Fix a whole bunch of typos in comments and log statements.
8915 2008-06-20 17:02:48 +0000 Michael Smith <msmith@xiph.org>
8917 sys/xvimage/xvimagesink.c: Don't set colour balance values on the Xv port if the user hasn't changed them (via proper...
8918 Original commit message from CVS:
8919 * sys/xvimage/xvimagesink.c:
8920 Don't set colour balance values on the Xv port if the user hasn't
8921 changed them (via properties or the interface). Avoids accumulating
8922 rounding errors for the common case.
8923 Partial fix for bug #537889.
8925 2008-06-20 16:56:18 +0000 Michael Smith <msmith@xiph.org>
8927 gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
8928 Original commit message from CVS:
8929 * gst/playback/gstdecodebin2.c:
8930 Ensure decodebin2 emits 'drained' signal once, and only once, when all
8933 2008-06-20 16:12:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
8936 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...
8937 Original commit message from CVS:
8938 apparently it's an error to specify nc -l -p 3000 - though the short usage
8939 does not make it very clear that you can drop the host arg with -l
8941 2008-06-20 09:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
8943 ext/vorbis/vorbisenc.c: Report the encoder latency. Fixes #538232.
8944 Original commit message from CVS:
8945 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
8946 (gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
8947 Report the encoder latency. Fixes #538232.
8949 2008-06-20 09:19:59 +0000 Wim Taymans <wim.taymans@gmail.com>
8951 gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
8952 Original commit message from CVS:
8953 * gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
8954 (notify_source), (activate_group):
8955 Implement the source property, emit notify when it changes in the
8956 underlying uridecodebin.
8958 2008-06-20 09:14:26 +0000 Wim Taymans <wim.taymans@gmail.com>
8960 tests/examples/seek/seek.c: Free and clear the seek element list so that we don't use invalid references when seeking...
8961 Original commit message from CVS:
8962 * tests/examples/seek/seek.c: (stop_cb):
8963 Free and clear the seek element list so that we don't use invalid
8964 references when seeking after recreating a gst-launch line.
8966 2008-06-20 09:09:37 +0000 Wim Taymans <wim.taymans@gmail.com>
8968 gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
8969 Original commit message from CVS:
8970 * gst-libs/gst/audio/gstbaseaudiosink.c:
8971 (gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
8972 (gst_base_audio_sink_render):
8973 Report latency even if we are not live instead of hiding it.
8974 Take ts-offset and render-delay of the basesink into account when
8976 Rework the clipping code so that we can take the various offsets into
8977 account and still do correct clipping.
8979 2008-06-20 08:52:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8981 configure.ac: Bump verion back to devel -> 0.10.20.1
8982 Original commit message from CVS:
8984 Bump verion back to devel -> 0.10.20.1
8986 2008-06-20 08:47:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8988 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...
8989 Original commit message from CVS:
8990 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
8991 Don't increase the size of non-string image buffers by one as this
8992 might in theory confuse decoders. Still increase it by one for string
8993 image buffers to append '\0'.
8995 2008-06-20 08:45:13 +0000 Antoine Tremblay <hexa00@gmail.com>
8997 gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
8998 Original commit message from CVS:
8999 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
9000 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
9001 Fix a buffer memleak and remove a confusing and wrong debug output.
9004 2008-06-19 11:25:37 +0000 Wim Taymans <wim.taymans@gmail.com>
9006 examples/app/appsink-src.c: Don't use a buffer after unreffing it.
9007 Original commit message from CVS:
9008 * examples/app/appsink-src.c: (on_new_buffer_from_source):
9009 Don't use a buffer after unreffing it.
9011 === release 0.10.20 ===
9013 2008-06-18 14:36:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9019 * docs/plugins/gst-plugins-base-plugins.args:
9020 * docs/plugins/gst-plugins-base-plugins.hierarchy:
9021 * docs/plugins/gst-plugins-base-plugins.interfaces:
9022 * docs/plugins/gst-plugins-base-plugins.prerequisites:
9023 * docs/plugins/inspect/plugin-adder.xml:
9024 * docs/plugins/inspect/plugin-alsa.xml:
9025 * docs/plugins/inspect/plugin-audioconvert.xml:
9026 * docs/plugins/inspect/plugin-audiorate.xml:
9027 * docs/plugins/inspect/plugin-audioresample.xml:
9028 * docs/plugins/inspect/plugin-audiotestsrc.xml:
9029 * docs/plugins/inspect/plugin-cdparanoia.xml:
9030 * docs/plugins/inspect/plugin-decodebin.xml:
9031 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
9032 * docs/plugins/inspect/plugin-gdp.xml:
9033 * docs/plugins/inspect/plugin-gnomevfs.xml:
9034 * docs/plugins/inspect/plugin-libvisual.xml:
9035 * docs/plugins/inspect/plugin-ogg.xml:
9036 * docs/plugins/inspect/plugin-pango.xml:
9037 * docs/plugins/inspect/plugin-playback.xml:
9038 * docs/plugins/inspect/plugin-queue2.xml:
9039 * docs/plugins/inspect/plugin-subparse.xml:
9040 * docs/plugins/inspect/plugin-tcp.xml:
9041 * docs/plugins/inspect/plugin-theora.xml:
9042 * docs/plugins/inspect/plugin-typefindfunctions.xml:
9043 * docs/plugins/inspect/plugin-uridecodebin.xml:
9044 * docs/plugins/inspect/plugin-video4linux.xml:
9045 * docs/plugins/inspect/plugin-videorate.xml:
9046 * docs/plugins/inspect/plugin-videoscale.xml:
9047 * docs/plugins/inspect/plugin-videotestsrc.xml:
9048 * docs/plugins/inspect/plugin-volume.xml:
9049 * docs/plugins/inspect/plugin-vorbis.xml:
9050 * docs/plugins/inspect/plugin-ximagesink.xml:
9051 * docs/plugins/inspect/plugin-xvimagesink.xml:
9052 * gst-plugins-base.doap:
9054 * win32/common/config.h:
9056 Original commit message from CVS:
9059 2008-06-18 14:32:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9088 Original commit message from CVS:
9091 2008-06-18 06:31:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9093 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
9094 Original commit message from CVS:
9095 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
9096 * examples/app/appsrc-ra.c:
9097 * examples/app/appsrc-seekable.c:
9098 * examples/app/appsrc-stream.c:
9099 * examples/app/appsrc-stream2.c:
9100 * ext/directfb/dfbvideosink.h:
9101 * ext/metadata/gstbasemetadata.c:
9102 * ext/metadata/gstbasemetadata.h:
9103 * ext/metadata/metadata.c:
9104 * ext/metadata/metadataexif.c:
9105 * ext/theora/theoradec.h:
9106 * gst/deinterlace2/gstdeinterlace2.h:
9107 * gst/deinterlace2/tvtime/speedy.c:
9108 * gst/deinterlace2/tvtime/speedy.h:
9109 * gst/deinterlace2/tvtime/vfir.c:
9110 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
9113 2008-06-16 14:11:36 +0000 Andy Wingo <wingo@pobox.com>
9115 * gst-libs/gst/app/gstappsrc.c:
9116 gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
9117 Original commit message from CVS:
9118 2008-06-16 Andy Wingo <wingo@pobox.com>
9119 * gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
9120 (gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
9121 G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
9123 2008-06-16 07:30:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9125 Final round of doc updates.
9126 Original commit message from CVS:
9127 * gst/rtpmanager/gstrtpjitterbuffer.c:
9128 * gst/speed/gstspeed.c:
9129 * gst/speexresample/gstspeexresample.c:
9130 * gst/videosignal/gstvideoanalyse.c:
9131 * gst/videosignal/gstvideodetect.c:
9132 * gst/videosignal/gstvideomark.c:
9133 * sys/dvb/gstdvbsrc.c:
9134 * sys/oss4/oss4-mixer.c:
9135 * sys/oss4/oss4-sink.c:
9136 * sys/oss4/oss4-source.c:
9137 * sys/wininet/gstwininetsrc.c:
9138 Final round of doc updates.
9140 2008-06-13 11:59:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9142 docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
9143 Original commit message from CVS:
9144 * docs/plugins/Makefile.am:
9145 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
9146 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
9147 * docs/plugins/gst-plugins-bad-plugins.args:
9148 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
9149 * docs/plugins/gst-plugins-bad-plugins.interfaces:
9150 * docs/plugins/gst-plugins-bad-plugins.prerequisites:
9151 * docs/plugins/gst-plugins-bad-plugins.signals:
9152 * docs/plugins/inspect/plugin-alsaspdif.xml:
9153 * docs/plugins/inspect/plugin-amrwb.xml:
9154 * docs/plugins/inspect/plugin-app.xml:
9155 * docs/plugins/inspect/plugin-bayer.xml:
9156 * docs/plugins/inspect/plugin-bz2.xml:
9157 * docs/plugins/inspect/plugin-cdaudio.xml:
9158 * docs/plugins/inspect/plugin-cdxaparse.xml:
9159 * docs/plugins/inspect/plugin-dtsdec.xml:
9160 * docs/plugins/inspect/plugin-dvb.xml:
9161 * docs/plugins/inspect/plugin-dvdspu.xml:
9162 * docs/plugins/inspect/plugin-faac.xml:
9163 * docs/plugins/inspect/plugin-faad.xml:
9164 * docs/plugins/inspect/plugin-fbdevsink.xml:
9165 * docs/plugins/inspect/plugin-festival.xml:
9166 * docs/plugins/inspect/plugin-filter.xml:
9167 * docs/plugins/inspect/plugin-flvdemux.xml:
9168 * docs/plugins/inspect/plugin-freeze.xml:
9169 * docs/plugins/inspect/plugin-gsm.xml:
9170 * docs/plugins/inspect/plugin-gstinterlace.xml:
9171 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
9172 * docs/plugins/inspect/plugin-h264parse.xml:
9173 * docs/plugins/inspect/plugin-interleave.xml:
9174 * docs/plugins/inspect/plugin-jack.xml:
9175 * docs/plugins/inspect/plugin-ladspa.xml:
9176 * docs/plugins/inspect/plugin-metadata.xml:
9177 * docs/plugins/inspect/plugin-mms.xml:
9178 * docs/plugins/inspect/plugin-modplug.xml:
9179 * docs/plugins/inspect/plugin-mpeg2enc.xml:
9180 * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
9181 * docs/plugins/inspect/plugin-mpegtsparse.xml:
9182 * docs/plugins/inspect/plugin-mpegvideoparse.xml:
9183 * docs/plugins/inspect/plugin-musepack.xml:
9184 * docs/plugins/inspect/plugin-musicbrainz.xml:
9185 * docs/plugins/inspect/plugin-mve.xml:
9186 * docs/plugins/inspect/plugin-mythtv.xml
9187 * docs/plugins/inspect/plugin-nas.xml:
9188 * docs/plugins/inspect/plugin-neon.xml:
9189 * docs/plugins/inspect/plugin-nsfdec.xml:
9190 * docs/plugins/inspect/plugin-nuvdemux.xml:
9191 * docs/plugins/inspect/plugin-oss4.xml
9192 * docs/plugins/inspect/plugin-rawparse.xml:
9193 * docs/plugins/inspect/plugin-real.xml:
9194 * docs/plugins/inspect/plugin-replaygain.xml:
9195 * docs/plugins/inspect/plugin-rfbsrc.xml:
9196 * docs/plugins/inspect/plugin-sdl.xml:
9197 * docs/plugins/inspect/plugin-sdp.xml:
9198 * docs/plugins/inspect/plugin-selector.xml:
9199 * docs/plugins/inspect/plugin-sndfile.xml:
9200 * docs/plugins/inspect/plugin-soundtouch.xml:
9201 * docs/plugins/inspect/plugin-spcdec.xml:
9202 * docs/plugins/inspect/plugin-speed.xml:
9203 * docs/plugins/inspect/plugin-speexresample.xml:
9204 * docs/plugins/inspect/plugin-stereo.xml:
9205 * docs/plugins/inspect/plugin-subenc.xml
9206 * docs/plugins/inspect/plugin-timidity.xml:
9207 * docs/plugins/inspect/plugin-tta.xml:
9208 * docs/plugins/inspect/plugin-vcdsrc.xml:
9209 * docs/plugins/inspect/plugin-videosignal.xml:
9210 * docs/plugins/inspect/plugin-vmnc.xml:
9211 * docs/plugins/inspect/plugin-wildmidi.xml:
9212 * docs/plugins/inspect/plugin-x264.xml:
9213 * docs/plugins/inspect/plugin-xvid.xml:
9214 * docs/plugins/inspect/plugin-y4menc.xml:
9215 * ext/amrwb/gstamrwbdec.c:
9216 * ext/amrwb/gstamrwbenc.c:
9217 * ext/amrwb/gstamrwbparse.c:
9218 * ext/dc1394/gstdc1394.c:
9219 * ext/directfb/dfbvideosink.c:
9220 * ext/ivorbis/vorbisdec.c:
9221 * ext/jack/gstjackaudiosink.c:
9222 * ext/mpeg2enc/gstmpeg2enc.cc:
9223 * ext/mplex/gstmplex.cc:
9224 * ext/musicbrainz/gsttrm.c:
9225 * ext/mythtv/gstmythtvsrc.c:
9226 * ext/theora/theoradec.c:
9227 * ext/timidity/gsttimidity.c:
9228 * ext/timidity/gstwildmidi.c:
9229 * gst-libs/gst/app/gstappsink.c:
9230 * gst/deinterlace/gstdeinterlace.c:
9231 * gst/dvdspu/gstdvdspu.c:
9232 * gst/festival/gstfestival.c:
9233 * gst/freeze/gstfreeze.c:
9234 * gst/interleave/deinterleave.c:
9235 * gst/interleave/interleave.c:
9236 * gst/modplug/gstmodplug.cc:
9237 * gst/nuvdemux/gstnuvdemux.c:
9238 Add missing elements to docs. Fix doc-markup: use convinience syntax
9239 for examples (produces valid docbook), add several refsec2 when we
9240 have several titles. Fix some types.
9242 2008-06-12 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
9244 examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
9245 Original commit message from CVS:
9246 * examples/app/.cvsignore:
9247 * examples/app/Makefile.am:
9248 * examples/app/appsink-src.c: (on_new_buffer_from_source),
9249 (on_source_message), (on_sink_message), (main):
9250 Add beefed up example app from bug #413418. It now also uses appsink
9251 instead of fakesink for more ultimate coolness.
9252 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
9253 (gst_app_src_init), (gst_app_src_set_property),
9254 (gst_app_src_get_property), (gst_app_src_unlock),
9255 (gst_app_src_unlock_stop), (gst_app_src_create),
9256 (gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
9257 (gst_app_src_end_of_stream):
9258 * gst-libs/gst/app/gstappsrc.h:
9259 Add block property to allow push based implementation to block when we
9260 fill up the appsrc queues.
9261 Emit the enough-data signal while releasing our lock.
9263 2008-06-12 14:50:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9265 examples/app/.cvsignore: Ignore more.
9266 Original commit message from CVS:
9267 * examples/app/.cvsignore:
9270 2008-06-12 14:49:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9272 Do not use short_description in section docs for elements. We extract them from element details and there will be war...
9273 Original commit message from CVS:
9274 * ext/dc1394/gstdc1394.c:
9275 * ext/ivorbis/vorbisdec.c:
9276 * ext/jack/gstjackaudiosink.c:
9277 * ext/metadata/gstmetadatademux.c:
9278 * ext/mythtv/gstmythtvsrc.c:
9279 * ext/theora/theoradec.c:
9280 * gst-libs/gst/app/gstappsink.c:
9281 * gst/bayer/gstbayer2rgb.c:
9282 * gst/deinterlace/gstdeinterlace.c:
9283 * gst/rawparse/gstaudioparse.c:
9284 * gst/rawparse/gstvideoparse.c:
9285 * gst/rtpmanager/gstrtpbin.c:
9286 * gst/rtpmanager/gstrtpclient.c:
9287 * gst/rtpmanager/gstrtpjitterbuffer.c:
9288 * gst/rtpmanager/gstrtpptdemux.c:
9289 * gst/rtpmanager/gstrtpsession.c:
9290 * gst/rtpmanager/gstrtpssrcdemux.c:
9291 * gst/selector/gstinputselector.c:
9292 * gst/selector/gstoutputselector.c:
9293 * gst/videosignal/gstvideoanalyse.c:
9294 * gst/videosignal/gstvideodetect.c:
9295 * gst/videosignal/gstvideomark.c:
9296 * sys/oss4/oss4-mixer.c:
9297 * sys/oss4/oss4-sink.c:
9298 * sys/oss4/oss4-source.c:
9299 Do not use short_description in section docs for elements. We extract
9300 them from element details and there will be warnings if they differ.
9301 Also fixing up the ChangeLog order.
9303 2008-06-11 21:17:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9305 configure.ac: 0.10.19.3 pre-release
9306 Original commit message from CVS:
9308 0.10.19.3 pre-release
9310 2008-06-11 20:13:00 +0000 David Schleef <ds@schleef.org>
9312 gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
9313 Original commit message from CVS:
9314 * gst-libs/gst/rtsp/gstrtspconnection.c:
9316 Patch By: David Schleef <ds@schleef.org>
9319 2008-06-11 09:35:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9321 ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
9322 Original commit message from CVS:
9323 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
9324 (gst_gio_base_src_create):
9325 * ext/gio/gstgiobasesrc.h:
9326 Try to read the requested number of bytes, even if the first
9327 read returns less than requested, until nothing is read anymore
9328 or we have the requested amount of bytes. This fixes playback of
9329 files via Samba as Samba only allows to read 64k at once.
9330 Implement a caching algorithm that makes sure that we read at
9331 least 4k of data every time. Some elements will try to read a few
9332 bytes, then seek, read again a few bytes and so on and this is
9333 painfully slow as every operation has to go over DBus if GVfs is
9335 Fixes bug #536849 and #536848.
9336 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
9337 (gst_gio_src_check_get_range):
9338 Override check_get_range() to blacklist http/https URIs
9339 and whitelist file URIs. More to be added on demand.
9341 2008-06-06 16:50:51 +0000 Wim Taymans <wim.taymans@gmail.com>
9343 examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
9344 Original commit message from CVS:
9345 * examples/app/Makefile.am:
9346 * examples/app/appsrc-ra.c: (feed_data), (seek_data),
9347 (found_source), (bus_message), (main):
9348 * examples/app/appsrc-seekable.c: (feed_data), (seek_data),
9349 (found_source), (bus_message), (main):
9350 * examples/app/appsrc-stream2.c: (feed_data), (found_source),
9351 (bus_message), (main):
9352 Added 3 more example application for using appsrc in random-access mode,
9353 pull-mode streaming and pull mode seekable.
9354 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
9355 (gst_app_src_start), (gst_app_src_do_get_size),
9356 (gst_app_src_create):
9357 * gst-libs/gst/app/gstappsrc.h:
9358 Make stream-type property writable.
9359 Unset flushing when starting so that we reuse appsrc.
9360 Inform basesrc about the configured size.
9361 Emit seek-data signal when we are going to a different offset in
9364 2008-06-06 14:19:54 +0000 Wim Taymans <wim.taymans@gmail.com>
9366 examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
9367 Original commit message from CVS:
9368 * examples/app/appsrc-stream.c: (found_source), (main):
9369 Use deep-notify until we can depend on a playbin2 with support for the
9372 2008-06-05 16:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
9374 examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
9375 Original commit message from CVS:
9376 * examples/app/.cvsignore:
9377 * examples/app/Makefile.am:
9378 * examples/app/appsrc-stream.c: (read_data), (start_feed),
9379 (stop_feed), (found_source), (bus_message), (main):
9380 Added an example on how to use appsrc in playbin in streaming mode from
9382 * examples/app/appsrc_ex.c: (main):
9383 Set pipeline to NULL to free queued buffers.
9384 * gst-libs/gst/app/gstapp-marshal.list:
9385 * gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
9386 (gst_app_src_class_init), (gst_app_src_init),
9387 (gst_app_src_flush_queued), (gst_app_src_dispose),
9388 (gst_app_src_set_property), (gst_app_src_get_property),
9389 (gst_app_src_unlock), (gst_app_src_unlock_stop),
9390 (gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
9391 (gst_app_src_check_get_range), (gst_app_src_do_seek),
9392 (gst_app_src_create), (gst_app_src_set_stream_type),
9393 (gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
9394 (gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
9395 (gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
9396 (gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
9397 (gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
9398 * gst-libs/gst/app/gstappsrc.h:
9399 Measure max queue size in bytes instead.
9400 Add support for 3 modes of operation, streaming, seekable and
9401 random-access, making basesrc handle the scheduling modes for each.
9402 Add appsrc:// uri handler so that automatic plugging can be done from
9403 playbin2 or uridecodebin, for example.
9404 Added support for custom segment formats.
9405 Add support for push and pull based operations from the application.
9406 Expand the methods so that errors can be detected.
9407 Flush the queued buffers on seeks and when shutting down.
9408 Add signals to inform the app that a seek must happen.
9410 2008-06-05 09:47:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9412 configure.ac: 0.10.19.2 pre-release
9413 Original commit message from CVS:
9415 0.10.19.2 pre-release
9417 2008-06-04 21:48:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9419 win32/common/: Add new API functions to the dll exports
9420 Original commit message from CVS:
9421 * win32/common/libgstrtsp.def:
9422 * win32/common/libgsttag.def:
9423 Add new API functions to the dll exports
9425 2008-06-04 17:42:38 +0000 Michael Smith <msmith@xiph.org>
9427 gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
9428 Original commit message from CVS:
9429 * gst/playback/gstplaybasebin.c:
9430 Disconnect signals from decodebins we created before we remove it from
9431 playbin, to avoid crashes if the decodebin is eventually disposed after
9432 the playbin itself (possible if the app takes a reference on the
9436 2008-06-04 17:12:40 +0000 Tim-Philipp Müller <tim@centricular.net>
9438 gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
9439 Original commit message from CVS:
9440 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
9441 (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
9442 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
9443 (h264_video_type_find), (mpeg_video_stream_type_find),
9444 (dv_type_find), (mmsh_type_find):
9445 Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
9446 copy caps for no good reason (this may be desirable to make it easier
9447 to detect leaks, but then it should probably be done for all caps
9448 in the typefinder somewhere).
9450 2008-06-04 16:06:49 +0000 Peter Kjellerstedt <pkj@axis.com>
9452 tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
9453 Original commit message from CVS:
9454 * tests/check/Makefile.am:
9455 Do not try to run the check tests for subparse unless it has been
9458 2008-06-04 16:00:26 +0000 Peter Kjellerstedt <pkj@axis.com>
9460 tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
9461 Original commit message from CVS:
9462 * tests/check/pipelines/streamheader.c: (buffer_probe_cb),
9463 (test_multifdsink_gdp_vorbisenc), (streamheader_suite):
9464 Do not try to run a test which requires vorbisenc unless we have
9467 2008-06-04 11:53:53 +0000 Peter Kjellerstedt <pkj@axis.com>
9469 gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
9470 Original commit message from CVS:
9471 * gst-libs/gst/rtsp/gstrtspconnection.c:
9472 (gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
9473 (gst_rtsp_connection_clear_auth_params),
9474 (gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
9475 * gst-libs/gst/rtsp/gstrtspconnection.h:
9476 Add a couple of missing argument guards.
9477 Add a way of setting the DSCP for an RTSP connection.
9478 Add an accessor method for the ip member of GstRTSPConnection as all
9479 members are supposed to be private.
9481 2008-06-04 11:33:23 +0000 Peter Kjellerstedt <pkj@axis.com>
9483 gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
9484 Original commit message from CVS:
9485 * gst/tcp/gstmultifdsink.c: (setup_dscp_client):
9486 Fixed accidental use of IPv4 options for all IPv6 addresses.
9488 2008-06-04 10:18:42 +0000 Tim-Philipp Müller <tim@centricular.net>
9490 gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
9491 Original commit message from CVS:
9492 * gst-libs/gst/interfaces/mixertrack.h:
9493 Document mixer track flags.
9495 2008-06-04 05:58:38 +0000 Antoine Tremblay <hexa00@gmail.com>
9497 gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
9498 Original commit message from CVS:
9499 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
9500 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
9501 Don't set caps on the buffers that contain a copy of the buffer
9502 including the caps of them resulting in an always increasing refcount
9503 of the caps and insanely large caps. Instead include a buffer without
9504 caps in the new caps. Fixes bug #536475.
9506 2008-06-04 05:44:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9508 gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
9509 Original commit message from CVS:
9510 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
9511 Transform a given PAR to a range on the struct with the generic
9512 height/width instead of the struct with the possibly restricted
9515 2008-06-04 04:24:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9517 gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
9518 Original commit message from CVS:
9519 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
9520 Prefer the given format if it contains something stricter than [1,MAX]
9521 for height or width and only put a structure that requires rescaling
9522 as second. This makes it possible to use videoscale in pipelines where
9523 the source can actually produce the wanted height/width but usually
9524 selects a different one from the requested.
9526 2008-06-03 20:01:58 +0000 John Millikin <jmillikin@gmail.com>
9528 gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
9529 Original commit message from CVS:
9530 Based on patch by: John Millikin <jmillikin gmail com>
9531 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
9532 (gst_vorbis_tag_add_coverart):
9533 Retrieve COVERART tags from vorbis comments (#512333)
9535 2008-06-03 19:44:48 +0000 Tim-Philipp Müller <tim@centricular.net>
9537 gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
9538 Original commit message from CVS:
9539 * gst-libs/gst/tag/tag.h:
9540 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
9541 Don't forget to add new enum value here too (should probably use
9542 glib-mkenums here...).
9544 2008-06-03 19:29:06 +0000 Tim-Philipp Müller <tim@centricular.net>
9546 gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
9547 Original commit message from CVS:
9548 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
9549 * gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
9550 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
9551 (gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
9552 (gst_tag_image_data_to_image_buffer):
9553 Add two utility functions to avoid code duplication (#512333):
9554 API: add gst_tag_image_data_to_image_buffer()
9555 API: add gst_tag_list_add_id3_image()
9557 2008-06-03 08:54:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9559 win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
9560 Original commit message from CVS:
9561 * win32/common/libgstaudio.def:
9562 Add gst_audio_check_channel_positions() to the exported symbols.
9564 2008-06-03 08:48:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9566 API: Make gst_audio_check_channel_positions() public.
9567 Original commit message from CVS:
9568 * docs/libs/gst-plugins-base-libs-sections.txt:
9569 * gst-libs/gst/audio/multichannel.c:
9570 (gst_audio_check_channel_positions):
9571 * gst-libs/gst/audio/multichannel.h:
9572 API: Make gst_audio_check_channel_positions() public.
9573 * tests/check/libs/audio.c: (GST_START_TEST):
9574 Add some simple checks for gst_audio_check_channel_positions().
9576 2008-06-02 20:09:14 +0000 Tim-Philipp Müller <tim@centricular.net>
9578 sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.
9579 Original commit message from CVS:
9580 * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
9581 minrange and maxrange are scaled according to the frequency
9584 2008-06-02 18:37:02 +0000 Tim-Philipp Müller <tim@centricular.net>
9586 ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
9587 Original commit message from CVS:
9588 * ext/pango/Makefile.am:
9589 * ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
9590 (gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
9591 Use gstvideo functions to calculate strides and plane offsets. Fixes
9592 rendering issue ('ghost' images of the text on the chroma planes)
9593 with widths or heights that are not multiples of 8 (#506659 and
9594 probably also #485729).
9595 * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
9597 Test with odd height/width too.
9599 2008-06-02 12:20:35 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9601 gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
9602 Original commit message from CVS:
9603 * gst/adder/gstadder.c: (gst_adder_query_duration),
9604 (gst_adder_query_latency):
9605 When using gst_element_iterate_pads() one has to unref every pad
9608 2008-05-31 19:57:57 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9610 gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
9611 Original commit message from CVS:
9612 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9613 (gst_base_audio_src_class_init):
9614 Add a gtk-doc chunk for the new properties to have a Since: indication.
9616 2008-05-31 19:50:59 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9619 ChangeLog surgery, mark API change
9620 Original commit message from CVS:
9621 ChangeLog surgery, mark API change
9623 2008-05-31 18:10:47 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9625 gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
9626 Original commit message from CVS:
9627 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9628 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
9629 (gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
9630 (gst_base_audio_src_change_state):
9631 Provide readable actual-buffer-time and actual-latency-time properties
9632 that reflect the configured ringbuffer values. Fixes #524724.
9634 2008-05-30 15:29:20 +0000 Wim Taymans <wim.taymans@gmail.com>
9636 gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
9637 Original commit message from CVS:
9638 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
9639 (gst_basertppayload_change_state):
9640 Simply converting the running time into an RTP timestamp by scaling it
9641 based on the clock-rate is good enough for making an RTP timestamp. This
9642 has the added benefit that we can later on expose a property with the
9643 RTP timestamp of running time 0, as is needed for RTSP servers to
9644 generate the response of the PLAY request.
9646 2008-05-30 08:42:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9648 gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
9649 Original commit message from CVS:
9650 * gst/audioconvert/gstaudioconvert.c:
9651 (structure_has_fixed_channel_positions),
9652 (gst_audio_convert_transform_caps):
9653 Allow up to 11 positioned channels now that audioconvert can handle
9654 this but add no default positions for > 8 channels.
9655 * tests/check/elements/audioconvert.c: (GST_START_TEST):
9656 Add some unit tests for the above change: Test conversion of
9657 11 positioned channels to stereo and the other way around, test
9658 conversion of 15 unpositioned channels in different ways.
9660 2008-05-29 19:45:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9662 win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
9663 Original commit message from CVS:
9664 * win32/common/libgstaudio.def:
9665 Add gst_audio_clock_reset to the list of exported symbols.
9667 2008-05-29 19:37:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9669 tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
9670 Original commit message from CVS:
9671 * tests/check/elements/vorbisdec.c: (vorbisdec_suite):
9672 Remove wrong_channels_identification_header unit test as we now
9673 support 7 (and more channels).
9675 2008-05-29 12:17:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9677 gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
9678 Original commit message from CVS:
9679 * gst/audioconvert/gstchannelmix.c:
9680 (gst_channel_mix_fill_one_other):
9681 If mixing left or right to center (or the other way around) only take
9682 the complete value if we don't already have the original position in
9685 2008-05-29 11:34:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9687 gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
9688 Original commit message from CVS:
9689 * gst-libs/gst/audio/multichannel.c:
9690 (gst_audio_check_channel_positions),
9691 (gst_audio_set_structure_channel_positions_list),
9692 (gst_audio_fixate_channel_positions):
9693 Allow rear center together with rear left/right and other previously
9694 conflicting channel positions. The reason why they weren't allowed
9695 was the channel mixing implementation in audioconvert.
9696 Also take this into account when fixing channel layouts.
9697 Allow setting channel positions for 1/2 channels when using
9698 gst_audio_set_structure_channel_position().
9699 * gst/audioconvert/gstchannelmix.c:
9700 (gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
9701 (gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
9702 (gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
9703 Major rewrite of the channel mixing.
9704 We now allow previously conflicting channel positions to appear
9705 together (rear center and rear left/right for example).
9707 Rework the way channels are mixed together to take more possible
9708 channel positions into account, properly mix from/to side channels
9709 and don't assume that either center, left&right or nothing of a
9710 specific position is available anymore.
9711 * tests/check/elements/audioconvert.c: (GST_START_TEST):
9712 Adjust unit tests with non-standard 1/2 channel layouts to the more
9713 correct new behaviour.
9714 Add a unit test for 5.1->Stereo downmixing.
9716 2008-05-29 07:02:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9718 ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
9719 Original commit message from CVS:
9720 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
9721 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
9722 Add sane defaults for the 7 and 8 channel layouts as those are
9723 undefined in the Vorbis spec. Use NONE channel layouts when decoding
9724 more than 8 channels instead of erroring out. Fixes bug #535356.
9726 2008-05-28 16:10:20 +0000 Wim Taymans <wim.taymans@gmail.com>
9728 Add theoraparse to the docs and fix some docs.
9729 Original commit message from CVS:
9730 * docs/plugins/Makefile.am:
9731 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
9732 * docs/plugins/gst-plugins-base-plugins-sections.txt:
9733 * ext/theora/theoraparse.c:
9734 Add theoraparse to the docs and fix some docs.
9736 2008-05-28 15:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
9738 gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
9739 Original commit message from CVS:
9740 * gst-libs/gst/cdda/gstcddabasesrc.c:
9741 (gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
9742 Fix EOS condition and track addition check, the track.end sector is
9743 included in the track. Fixes #533265.
9745 2008-05-28 14:49:24 +0000 Mark Nauwelaerts <manauw@skynet.be>
9747 gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
9748 Original commit message from CVS:
9749 Patch by: Mark Nauwelaerts <manauw at skynet be>
9750 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
9751 (gst_video_rate_flush_prev), (gst_video_rate_event),
9752 (gst_video_rate_chain):
9753 * gst/videorate/gstvideorate.h:
9754 React (more) to NEWSEGMENT
9755 Small adjustment in timestamp calculation to prevent mismatches
9758 2008-05-28 11:31:44 +0000 Tim-Philipp Müller <tim@centricular.net>
9760 tests/examples/seek/seek.c: Initialise error to NULL as we should.
9761 Original commit message from CVS:
9762 * tests/examples/seek/seek.c: (make_parselaunch_pipeline):
9763 Initialise error to NULL as we should.
9765 2008-05-28 08:14:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9767 gst/adder/gstadder.c: Implement latency query.
9768 Original commit message from CVS:
9769 * gst/adder/gstadder.c: (gst_adder_query_duration),
9770 (gst_adder_query_latency), (gst_adder_query):
9771 Implement latency query.
9773 2008-05-27 18:10:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9775 gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
9776 Original commit message from CVS:
9777 * gst/adder/gstadder.c: (gst_adder_query_duration):
9778 Correctly resync the iterator if gst_iterator_next() returns
9779 GST_ITERATOR_RESYNC.
9781 2008-05-27 17:14:07 +0000 Tim-Philipp Müller <tim@centricular.net>
9783 win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
9784 Original commit message from CVS:
9785 * win32/vs6/libgstpbutils.dsp:
9786 Add pbutils-enumtypes.c to sources (#518037).
9788 2008-05-27 16:20:17 +0000 Wim Taymans <wim.taymans@gmail.com>
9790 gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
9791 Original commit message from CVS:
9792 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
9793 (gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
9794 * gst-libs/gst/audio/gstaudioclock.h:
9795 Add method to inform the clock that the time starts from 0 again. We use
9796 this info to calculate a clock offset so that the time we report in
9797 internal_time is monotonically increasing, as required by the clock base
9798 class. Fixes #521761.
9799 API: GstAudioClock::gst_audio_clock_reset()
9800 * gst-libs/gst/audio/gstbaseaudiosink.c:
9801 (gst_base_audio_sink_skew_slaving),
9802 (gst_base_audio_sink_change_state):
9803 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9804 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
9805 Reset reported time when we (re)create the ringbuffer.
9807 2008-05-27 16:11:32 +0000 Tim-Philipp Müller <tim@centricular.net>
9809 ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
9810 Original commit message from CVS:
9811 * ext/alsa/gstalsamixertrack.c:
9812 (gst_alsa_mixer_track_update_alsa_capabilities):
9813 Make sure playback volumes aren't accidentally overwritten by
9814 capture volumes if an alsa mixer track has both playback and
9815 capture capabilities: we create two GstMixerTracks in that
9816 case, so make sure we query only the alsa capabilities that
9817 refer to the type of GstMixerTrack we created from the dual
9818 capability alsa element. Should fix issues with Audigy2 sound
9821 2008-05-27 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.net>
9823 tests/check/pipelines/oggmux.c: Don't use deprecated function.
9824 Original commit message from CVS:
9825 * tests/check/pipelines/oggmux.c: (test_pipeline):
9826 Don't use deprecated function.
9828 2008-05-27 10:35:55 +0000 Wim Taymans <wim.taymans@gmail.com>
9830 gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
9831 Original commit message from CVS:
9832 * gst/playback/gstdecodebin2.c:
9833 (gst_decode_group_control_source_pad), (gst_decode_group_expose):
9834 Check for NULL cases and log them, creating ghostpads can, for example,
9835 fail when the pad returns wrong caps.
9836 * gst/playback/gstplaybin2.c: (perform_eos):
9837 When pushing out the EOS event, collect the return value and warn when
9840 2008-05-26 17:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
9842 gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
9843 Original commit message from CVS:
9844 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
9845 (gst_riff_create_video_template_caps):
9846 Add support for DVCPRO.
9848 2008-05-26 10:29:20 +0000 Tim-Philipp Müller <tim@centricular.net>
9850 gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
9851 Original commit message from CVS:
9852 * gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
9853 Change default scaling method from nearest-neighbour to bilinear.
9855 2008-05-26 10:26:00 +0000 Tim-Philipp Müller <tim@centricular.net>
9857 tests/check/libs/video.c: More checks.
9858 Original commit message from CVS:
9859 * tests/check/libs/video.c:
9862 2008-05-25 20:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
9864 Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
9865 Original commit message from CVS:
9866 * gst/subparse/gstsubparse.c: (parser_state_init),
9867 (gst_sub_parse_format_autodetect), (handle_buffer):
9868 * gst/subparse/gstsubparse.h:
9869 * tests/check/elements/subparse.c: (test_tmplayer_style3b):
9870 Limit duration to a maximum of five seconds for tmplayer format where
9871 we can guess the duration only from the timestamp of the next line of
9872 text. We don't want to show a text for eternities just because nothing
9873 else is being said for a while.
9875 2008-05-23 14:14:28 +0000 Wim Taymans <wim.taymans@gmail.com>
9877 gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
9878 Original commit message from CVS:
9879 * gst-libs/gst/rtp/gstbasertpdepayload.c:
9880 (gst_base_rtp_depayload_chain),
9881 (gst_base_rtp_depayload_handle_sink_event),
9882 (gst_base_rtp_depayload_push_full),
9883 (gst_base_rtp_depayload_change_state):
9884 Check sequence numbers, mark input buffers with a discont flag for the
9885 subclass when we detected a gap, drop duplicate buffers. We do this
9886 because one can use the element without a jitterbuffer in front and we
9887 don't want to feed the subclasses invalid or reordered data.
9888 Do an error when the subclass did not provide a process function instead
9890 Some other small cleanups.
9892 2008-05-22 22:35:40 +0000 Tim-Philipp Müller <tim@centricular.net>
9894 gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
9895 Original commit message from CVS:
9896 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
9897 May just as well use the precalculated uvstride here.
9899 2008-05-22 22:09:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9901 Add some documentation comments, and some new headers to be scanned.
9902 Original commit message from CVS:
9903 * docs/plugins/Makefile.am:
9904 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
9905 * docs/plugins/gst-plugins-base-plugins-sections.txt:
9906 * docs/plugins/gst-plugins-base-plugins.args:
9907 * docs/plugins/gst-plugins-base-plugins.hierarchy:
9908 * docs/plugins/gst-plugins-base-plugins.interfaces:
9909 * docs/plugins/gst-plugins-base-plugins.prerequisites:
9910 * docs/plugins/inspect/plugin-adder.xml:
9911 * docs/plugins/inspect/plugin-alsa.xml:
9912 * docs/plugins/inspect/plugin-audioconvert.xml:
9913 * docs/plugins/inspect/plugin-audiorate.xml:
9914 * docs/plugins/inspect/plugin-audioresample.xml:
9915 * docs/plugins/inspect/plugin-audiotestsrc.xml:
9916 * docs/plugins/inspect/plugin-cdparanoia.xml:
9917 * docs/plugins/inspect/plugin-decodebin.xml:
9918 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
9919 * docs/plugins/inspect/plugin-gdp.xml:
9920 * docs/plugins/inspect/plugin-gio.xml:
9921 * docs/plugins/inspect/plugin-gnomevfs.xml:
9922 * docs/plugins/inspect/plugin-libvisual.xml:
9923 * docs/plugins/inspect/plugin-ogg.xml:
9924 * docs/plugins/inspect/plugin-pango.xml:
9925 * docs/plugins/inspect/plugin-playback.xml:
9926 * docs/plugins/inspect/plugin-queue2.xml:
9927 * docs/plugins/inspect/plugin-subparse.xml:
9928 * docs/plugins/inspect/plugin-tcp.xml:
9929 * docs/plugins/inspect/plugin-theora.xml:
9930 * docs/plugins/inspect/plugin-typefindfunctions.xml:
9931 * docs/plugins/inspect/plugin-uridecodebin.xml:
9932 * docs/plugins/inspect/plugin-video4linux.xml:
9933 * docs/plugins/inspect/plugin-videorate.xml:
9934 * docs/plugins/inspect/plugin-videoscale.xml:
9935 * docs/plugins/inspect/plugin-videotestsrc.xml:
9936 * docs/plugins/inspect/plugin-volume.xml:
9937 * docs/plugins/inspect/plugin-vorbis.xml:
9938 * docs/plugins/inspect/plugin-ximagesink.xml:
9939 * docs/plugins/inspect/plugin-xvimagesink.xml:
9940 * ext/cdparanoia/gstcdparanoiasrc.c:
9941 * ext/ogg/gstoggdemux.c:
9942 * ext/ogg/gstoggdemux.h:
9943 * ext/ogg/gstoggmux.c:
9944 * ext/ogg/gstoggmux.h:
9945 * gst/audioconvert/audioconvert.c:
9946 * gst/audioconvert/audioconvert.h:
9947 * gst/audioconvert/gstaudioconvert.h:
9948 * gst/gdp/gstgdpdepay.h:
9949 * gst/gdp/gstgdppay.h:
9950 * gst/playback/gstdecodebin.c:
9951 * gst/playback/gstdecodebin2.c:
9952 * gst/playback/gstplaybin.c:
9953 * gst/playback/gstplaybin2.c:
9954 * gst/playback/gsturidecodebin.c:
9955 * gst/tcp/gstmultifdsink.c:
9956 * gst/tcp/gstmultifdsink.h:
9958 Add some documentation comments, and some new headers to be scanned.
9959 Rename some internal enum declarations (audioconvert's DitherType and
9960 NoiseShapingType, GstUnitType from the TCP elements) to match the
9961 documented GObject type names so that the docs pick them up.
9962 Name the playbin2 docs markups properly so they get picked up. They'll
9963 need renaming back when/if playbin2 becomes playbin.
9964 100% symbol coverage for the plugin docs, booya.
9966 2008-05-22 18:30:15 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
9968 gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
9969 Original commit message from CVS:
9970 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
9971 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
9972 Fix generation of NV12/NV21 frames. Fixes bug #532454.
9974 2008-05-22 11:59:33 +0000 Sjoerd Simons <sjoerd@luon.net>
9976 gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
9977 Original commit message from CVS:
9978 Patch by: Sjoerd Simons <sjoerd at luon dot net>
9979 * gst/playback/gstdecodebin.c: (remove_fakesink):
9980 Lock the fakesink before setting the state to NULL and removing it from
9981 the bin so that a concurrent state change cannot interfere.
9984 2008-05-21 17:09:42 +0000 Felipe Contreras <felipe.contreras@nokia.com>
9986 docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
9987 Original commit message from CVS:
9989 Fix installing plugin documentation when gtk-doc is disabled.
9991 2008-05-21 17:01:16 +0000 Felipe Contreras <felipe.contreras@nokia.com>
9993 gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
9994 Original commit message from CVS:
9995 * gst-libs/gst/rtsp/Makefile.am:
9996 Distribute, don't install md5.h
9998 2008-05-21 16:47:58 +0000 Julien Moutte <julien@moutte.net>
10000 gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
10001 Original commit message from CVS:
10002 2008-05-21 Julien Moutte <julien@fluendo.com>
10003 * gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
10004 instead of SOL_IP, works on more platforms.
10005 * gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
10008 2008-05-21 16:44:15 +0000 Wim Taymans <wim.taymans@gmail.com>
10010 Some debug and comment fixes.
10011 Original commit message from CVS:
10012 * ext/vorbis/vorbisdec.c:
10013 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
10014 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
10015 Some debug and comment fixes.
10016 * tests/examples/dynamic/addstream.c: (main):
10019 2008-05-21 16:36:50 +0000 Wim Taymans <wim.taymans@gmail.com>
10021 Don't use bad gst_element_get_pad().
10022 Original commit message from CVS:
10023 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
10024 * gst/playback/decodetest.c: (new_decoded_pad_cb):
10025 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
10026 (try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
10027 (cleanup_decodebin):
10028 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
10029 (connect_element), (gst_decode_group_control_demuxer_pad):
10030 * gst/playback/gstplaybasebin.c: (queue_remove_probe),
10031 (queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
10033 * gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
10034 (gst_play_bin_set_property), (handoff), (gen_video_element),
10035 (gen_text_element), (gen_audio_element), (gen_vis_element),
10036 (remove_sinks), (add_sink), (setup_sinks):
10037 * gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
10038 * gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
10039 (gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
10040 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
10041 (gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
10042 (gen_video_chain), (gen_text_chain), (gen_audio_chain),
10043 (gen_vis_chain), (gst_play_sink_reconfigure),
10044 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
10045 (gst_play_sink_request_pad):
10046 * gst/playback/gsturidecodebin.c: (type_found), (setup_source):
10047 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
10049 * gst/playback/test6.c: (new_decoded_pad_cb):
10050 * tests/check/elements/audioconvert.c: (GST_START_TEST):
10051 * tests/check/elements/audiorate.c: (test_injector_chain),
10052 (do_perfect_stream_test):
10053 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
10054 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
10055 * tests/check/elements/gnomevfssink.c:
10056 * tests/check/elements/textoverlay.c:
10057 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
10058 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
10059 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
10060 * tests/check/pipelines/oggmux.c: (test_pipeline):
10061 * tests/check/pipelines/streamheader.c: (GST_START_TEST):
10062 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
10063 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
10064 * tests/examples/seek/scrubby.c: (make_wav_pipeline):
10065 * tests/examples/seek/seek.c: (make_mod_pipeline),
10066 (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
10067 (make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
10068 (make_theora_pipeline), (make_vorbis_theora_pipeline),
10069 (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
10070 (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
10071 (update_fill), (msg_buffering):
10072 Don't use bad gst_element_get_pad().
10074 2008-05-21 14:35:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10076 gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
10077 Original commit message from CVS:
10078 * gst-libs/gst/riff/riff-media.c:
10079 Fix wrong method name in docs. Fix calculation of strf fields for
10081 * gst-libs/gst/riff/riff-read.c:
10082 Whitespace fix and removing double ';'.
10084 2008-05-21 11:52:30 +0000 Wim Taymans <wim.taymans@gmail.com>
10086 docs/design/part-playbin2.txt: Add some leftover doc.
10087 Original commit message from CVS:
10088 * docs/design/part-playbin2.txt:
10089 Add some leftover doc.
10091 2008-05-21 11:36:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10093 gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
10094 Original commit message from CVS:
10095 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
10096 Fix copy & paste error in last commit.
10098 2008-05-21 11:30:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10100 gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
10101 Original commit message from CVS:
10102 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
10103 Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
10104 other channel positions when source has SIDE channels and dest doesn't
10105 or the other way around.
10107 2008-05-21 11:29:25 +0000 Henrik Eriksson <henriken@axis.com>
10109 gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
10110 Original commit message from CVS:
10111 Patch by: Henrik Eriksson <henriken at axis dot com>
10112 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
10113 (gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
10114 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
10115 (gst_multi_fd_sink_get_property):
10116 * gst/tcp/gstmultifdsink.h:
10117 Add support for DSCP QOS. Fixes #469933.
10119 2008-05-21 07:46:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10121 tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
10122 Original commit message from CVS:
10123 * tests/check/elements/audioconvert.c: (GST_START_TEST):
10124 Add another test that checks if conversion between standard 1 and 2
10125 channel layouts with and without positions set is working.
10127 2008-05-21 07:39:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10129 gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
10130 Original commit message from CVS:
10131 * gst-libs/gst/audio/multichannel.c:
10132 (gst_audio_check_channel_positions):
10133 Allow non-standard 2 channel layouts.
10134 * tests/check/elements/audioconvert.c: (GST_START_TEST):
10135 Add some tests for converting and remapping non-standard 1 and 2
10138 2008-05-21 07:28:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10140 gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
10141 Original commit message from CVS:
10142 * gst/audioconvert/gstchannelmix.c:
10143 (gst_channel_mix_fill_normalize):
10144 Prevent division by zero if the channel mix matrix contains only
10147 2008-05-21 06:45:22 +0000 Antoine Tremblay <hexa00@gmail.com>
10149 gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
10150 Original commit message from CVS:
10151 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
10152 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
10153 Close a buffer memory leak. Fixes bug #534071.
10155 2008-05-21 06:39:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10157 gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
10158 Original commit message from CVS:
10159 * gst-libs/gst/rtsp/gstrtsptransport.h:
10160 Make the GstRTSPTransport struct members public as there are no
10161 setters/getters and it's supposed to be changed directly.
10164 2008-05-21 05:48:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10166 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...
10167 Original commit message from CVS:
10168 * gst/adder/gstadder.c:
10169 Adder also doesn't support audio/x-raw-int with width!=depth so don't
10170 claim this on the pad template caps.
10172 2008-05-20 16:26:53 +0000 Wim Taymans <wim.taymans@gmail.com>
10174 gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
10175 Original commit message from CVS:
10176 * gst-libs/gst/audio/gstbaseaudiosink.c:
10177 (gst_base_audio_sink_sync_latency):
10178 We can only use our optimal calibration if we prerolled before the
10181 2008-05-20 14:35:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10183 configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
10184 Original commit message from CVS:
10186 Require core CVS for GstBaseSrc buffer caps setting magic.
10188 2008-05-20 12:26:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10190 gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
10191 Original commit message from CVS:
10192 * gst/audioconvert/gstaudioconvert.c:
10193 (gst_audio_convert_fixate_channels):
10194 Fix logic in last commit.
10196 2008-05-20 12:15:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10198 gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
10199 Original commit message from CVS:
10200 * gst/audioconvert/gstaudioconvert.c:
10201 (gst_audio_convert_fixate_channels):
10202 Passthrough the channel positions if the number of output channels is
10203 the same as the number of input channels, the input had a channel
10204 layout and downstream requests no special one. We did this already for
10205 > 2 channels but now it's also done for 1 channel. Fixes bug #533617.
10207 2008-05-20 11:13:27 +0000 Wim Taymans <wim.taymans@gmail.com>
10209 ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
10210 Original commit message from CVS:
10211 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
10212 (gst_gnome_vfs_src_finalize),
10213 (gst_gnome_vfs_src_received_headers_callback),
10214 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
10215 * ext/gnomevfs/gstgnomevfssrc.h:
10216 Set the ICY caps on the srcpad from where they get picked up by the base
10217 class now and set on the outgoing buffers.
10218 * gst-libs/gst/audio/gstbaseaudiosrc.c:
10219 (gst_base_audio_src_create):
10220 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
10221 BaseSrc now sets the caps on outgoing buffers automatically.
10223 2008-05-20 11:09:06 +0000 Wim Taymans <wim.taymans@gmail.com>
10225 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
10226 Original commit message from CVS:
10227 * gst-libs/gst/audio/gstbaseaudiosink.c:
10228 (gst_base_audio_sink_resample_slaving),
10229 (gst_base_audio_sink_skew_slaving),
10230 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
10231 (gst_base_audio_sink_async_play),
10232 (gst_base_audio_sink_change_state):
10233 Change the way in which the ringbuffer is started when dealing with a
10234 slaved clock and latency. We now sync to the clock until we reach
10235 upstream latency before starting the ringbuffer. This has the effect
10236 that we can accurately align the master and slave clocks and let the
10237 rate correction code take care of the initial drift or rounding errors
10238 instead of leaving them uncorrected with the old approach.
10240 2008-05-20 08:12:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10242 gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
10243 Original commit message from CVS:
10244 * gst/audioconvert/gstaudioconvert.c:
10245 (gst_audio_convert_fixate_channels):
10246 Correctly set the default channel positions when converting to 8
10249 2008-05-19 16:13:25 +0000 Tim-Philipp Müller <tim@centricular.net>
10251 configure.ac: Error out if we don't have the required version of core.
10252 Original commit message from CVS:
10254 Error out if we don't have the required version of core.
10256 2008-05-19 15:59:40 +0000 Tim-Philipp Müller <tim@centricular.net>
10258 gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
10259 Original commit message from CVS:
10260 * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
10261 Use data scan helper in aac typefinder and stop scanning
10262 for headers when we've found a type. Also fix potential invalid
10263 memory access when calculating the frame length.
10265 2008-05-19 14:09:08 +0000 Tim-Philipp Müller <tim@centricular.net>
10267 gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
10268 Original commit message from CVS:
10269 * gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
10270 (mpeg_sys_is_valid_pack):
10271 Don't modify scan context when we return FALSE in ensure_data, so
10272 it's possible to continue scanning, and we don't end up with a NULL
10273 data pointer and a positive size, which might bite us the next time
10274 we're called. Small constification.
10276 2008-05-16 21:12:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10278 gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
10279 Original commit message from CVS:
10280 * gst/adder/gstadder.c:
10281 Adder doesn't support 24 bit samples so don't claim it supports them
10282 in the pad template caps.
10284 2008-05-14 20:28:02 +0000 Wim Taymans <wim.taymans@gmail.com>
10286 gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
10287 Original commit message from CVS:
10288 * gst-libs/gst/rtp/gstbasertpdepayload.c:
10289 (gst_base_rtp_depayload_chain):
10290 Validate the RTP packet before further processing it. It's just too
10291 dangerous to accept random packets and people are not forced to use a
10292 jitterbuffer or session manager to filter out the bad packets.
10293 * gst-libs/gst/rtp/gstrtpbuffer.c:
10294 (gst_rtp_buffer_set_extension_data),
10295 (gst_rtp_buffer_get_payload_subbuffer):
10297 When setting extension data in a buffer that is too small, we fail and
10298 we should not set the extension bit.
10299 Change GST_WARNINGS into g_warning because they really are
10300 programming errors.
10301 * tests/check/libs/rtp.c: (GST_START_TEST):
10302 Catch the g_warnings now in the unit tests and that fact that failing to
10303 set extension data left the extension bit untouched.
10305 2008-05-14 13:57:41 +0000 Tim-Philipp Müller <tim@centricular.net>
10307 gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
10308 Original commit message from CVS:
10309 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
10310 Revert previous change which made basetransform handle buffer_alloc
10311 and which breaks things badly in the non-passthrough case since it
10312 returned buffers with a different (ie. sometimes smaller) size than
10313 the size requested.
10315 2008-05-14 13:43:12 +0000 Bernard B <b-gnome@largestprime.net>
10317 gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
10318 Original commit message from CVS:
10319 Patch by: Bernard B <b-gnome at largestprime dot net>
10320 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
10321 Fix seqnum compare function for bordercase values and fix the docs
10322 again. Fixes #533075.
10323 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
10324 Add a testcase for seqnum compare function.
10326 2008-05-14 10:58:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10328 gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
10329 Original commit message from CVS:
10330 * gst/adder/gstadder.c: (gst_adder_setcaps),
10331 (gst_adder_class_init):
10332 Correctly declare the supported endianness on the pad templates
10333 and check for correct endianness in the set caps function. Adder
10334 only supports native endianness.
10335 Also use gst_element_class_set_details_simple().
10337 2008-05-14 09:12:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10339 sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.
10340 Original commit message from CVS:
10341 * sys/xvimage/xvimagesink.c:
10342 Better debug logging in port value handling. Merging separate port
10343 value loops into one.
10345 2008-05-13 16:02:19 +0000 Hannes Bistry <hannesb@gmx.de>
10347 gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
10348 Original commit message from CVS:
10349 Patch by: Hannes Bistry <hannesb at gmx dot de>
10350 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
10351 * gst/tcp/gsttcpserversink.c:
10352 (gst_tcp_server_sink_handle_server_read),
10353 (gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
10354 Fix regression in clientsrc because we did not add the fd to the poll
10355 set anymore. Fixes #532364.
10356 Do some cleanups here and there.
10358 2008-05-13 13:04:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10360 gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
10361 Original commit message from CVS:
10362 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
10363 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
10364 * gst/playback/gstplay-marshal.list:
10365 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
10366 Use correct marshallers. GstCaps are a boxed type and no GObject
10369 2008-05-13 11:37:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10371 win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.
10372 Original commit message from CVS:
10373 * win32/common/libgstrtsp.def:
10374 Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
10377 2008-05-13 10:59:49 +0000 Sjoerd Simons <sjoerd@luon.net>
10379 tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
10380 Original commit message from CVS:
10381 Patch by: Sjoerd Simons <sjoerd at luon dot net>
10382 * tests/check/elements/audioresample.c:
10383 (live_switch_alloc_only_48000), (live_switch_get_sink_caps),
10384 (live_switch_push), (GST_START_TEST):
10385 Add unit test for the latest basetransform negotiation changes.
10388 2008-05-13 09:14:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10390 gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
10391 Original commit message from CVS:
10392 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
10393 Fix nv12<->nv21 conversion if stride is larger than width.
10395 2008-05-13 07:28:21 +0000 j^ <j@oil21.org>
10397 ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...
10398 Original commit message from CVS:
10399 Patch by: j^ <j at oil21 dot org>
10400 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
10401 (gst_ogg_pad_parse_skeleton_fisbone):
10402 * ext/ogg/gstoggdemux.h:
10403 Parse presentation time from skeleton streams and use it as offset
10404 for the timestamps. Fixes bug #530068.
10406 2008-05-12 08:45:11 +0000 Wim Taymans <wim.taymans@gmail.com>
10408 gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
10409 Original commit message from CVS:
10410 * gst-libs/gst/audio/gstbaseaudiosink.c:
10411 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
10412 Revert previous patch that attempted to more accurately calculate the
10413 initial offset between master and slave clock. The best thing we can do
10414 in general is take the time of both clocks as the diff since we don't
10415 know when the actual preroll happened.
10417 2008-05-11 19:52:59 +0000 Tim-Philipp Müller <tim@centricular.net>
10419 gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
10420 Original commit message from CVS:
10421 * gst-libs/gst/pbutils/install-plugins.c:
10422 Fix docs: type and missing word.
10424 2008-05-10 20:16:21 +0000 Tim-Philipp Müller <tim@centricular.net>
10426 gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
10427 Original commit message from CVS:
10428 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
10429 Don't do lots of 4-byte peeks, but use the 'new' data scan helper
10430 for this instead; don't check if we've found enough markers after
10431 each and every step, it's enough to do that only if we've actually
10432 found a new marker.
10433 Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
10435 2008-05-10 18:19:17 +0000 Tim-Philipp Müller <tim@centricular.net>
10437 gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
10438 Original commit message from CVS:
10439 * gst/typefind/gsttypefindfunctions.c:
10440 (DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
10441 (data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
10442 (mpeg_video_stream_type_find):
10443 Move scan helper thingy to the beginning of the file so we can use
10444 it in other typefind functions. Rename it to something more
10445 generic. Also improve handling of things towards the end of the
10446 typefind data: peek as much as we can if we know the size of the
10447 data, rather than just min_size.
10449 2008-05-09 21:42:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10451 Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
10452 Original commit message from CVS:
10453 * docs/libs/gst-plugins-base-libs-sections.txt:
10454 * gst-libs/gst/interfaces/colorbalance.c:
10455 * gst-libs/gst/interfaces/colorbalance.h:
10456 * gst-libs/gst/interfaces/colorbalancechannel.c:
10457 * gst-libs/gst/interfaces/colorbalancechannel.h:
10458 * gst-libs/gst/interfaces/tuner.c:
10459 * gst-libs/gst/interfaces/tunerchannel.c:
10460 * gst-libs/gst/interfaces/tunerchannel.h:
10461 * gst-libs/gst/interfaces/tunernorm.c:
10462 * gst-libs/gst/interfaces/tunernorm.h:
10463 * gst-libs/gst/video/video.c:
10464 * gst-libs/gst/video/video.h:
10465 Document the GstTuner and GstColorBalance interfaces, and some
10466 other random API functions that needed it. 70% symbol coverage, woo.
10468 2008-05-09 16:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
10470 gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
10471 Original commit message from CVS:
10472 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
10473 Choose to allocate one less segment but require one additional segment
10475 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
10476 No need to increment the number of segments in the source.
10477 * gst-libs/gst/audio/gstbaseaudiosink.c:
10478 (gst_base_audio_sink_get_time), (clock_convert_external),
10479 (gst_base_audio_sink_resample_slaving),
10480 (gst_base_audio_sink_skew_slaving),
10481 (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
10482 (gst_base_audio_sink_async_play):
10483 Remove adding latency when returning the internal time while subtracting
10484 it again when we use the value a little later.
10485 When calculating the end timestamp, we are making a rounding error
10486 with the current algorithm. Ensure that we don't accumulate these
10487 rounding errors when aligning samples by not resampling at all if we
10488 don't need to. Fixes #419351.
10489 Make the initial calibration of the clock slaving a little more
10490 predictable and accurate. Also handle the case where we don't do
10493 2008-05-09 08:34:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10495 gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
10496 Original commit message from CVS:
10497 Based on a patch by:
10498 Björn Benderius <bjoern dot benderius at axis dot com>
10499 * gst/ffmpegcolorspace/avcodec.h:
10500 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
10501 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
10502 (gst_ffmpegcsp_avpicture_fill):
10503 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
10504 * gst/ffmpegcolorspace/imgconvert_template.h:
10505 Add conversions from/to NV12 and NV21 and conversions between those
10506 two formats. Fixes bug #532166.
10508 2008-05-08 17:35:44 +0000 Edward Hervey <bilboed@bilboed.com>
10510 gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
10511 Original commit message from CVS:
10512 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
10513 Abort the h264 typefinding as soon as _peek() doesn't return anything,
10514 which happens for example with files smaller than 128kb.
10516 2008-05-08 14:46:27 +0000 Wouter Cloetens <zombie@e2big.org>
10518 gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
10519 Original commit message from CVS:
10520 Patch by: Wouter Cloetens <zombie at e2big dot org>
10521 * gst-libs/gst/rtsp/Makefile.am:
10522 * gst-libs/gst/rtsp/gstrtspconnection.c:
10523 (gst_rtsp_connection_create), (md5_digest_to_hex_string),
10524 (auth_digest_compute_hex_urp), (auth_digest_compute_response),
10525 (add_auth_header), (gst_rtsp_connection_free),
10526 (gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
10527 (gst_rtsp_connection_set_auth_param),
10528 (gst_rtsp_connection_clear_auth_params):
10529 * gst-libs/gst/rtsp/gstrtspconnection.h:
10530 Add Digest authorization support for RTSP connections. See #532065.
10531 * gst-libs/gst/rtsp/md5.c:
10532 * gst-libs/gst/rtsp/md5.h:
10533 Yeap, another md5 implementation until we can depend on a glib that has
10536 2008-05-08 06:20:42 +0000 Sjoerd Simons <sjoerd@luon.net>
10538 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
10539 Original commit message from CVS:
10540 Patch by: Sjoerd Simons <sjoerd at luon dot net>
10541 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
10542 Let audioresample use the buffer allocation of basetransform instead
10544 * tests/check/elements/audioresample.c: (alloc_only_48000),
10545 (GST_START_TEST), (audioresample_suite):
10546 Add unit test for the recent basetransform bugfix, where upstream
10547 changes caps to something that can't be passed through anymore.
10549 2008-05-07 19:50:27 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
10551 win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
10552 Original commit message from CVS:
10553 * win32/common/config.h.in:
10554 Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
10555 use the real thing than having "???" unconditionally.
10557 2008-05-07 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
10559 gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
10560 Original commit message from CVS:
10561 * gst-libs/gst/audio/gstbaseaudiosink.c:
10562 (gst_base_audio_sink_query):
10563 Report the latency with the new seglatency parameter.
10564 * gst-libs/gst/audio/gstringbuffer.c:
10565 (gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
10566 (gst_ring_buffer_acquire):
10567 * gst-libs/gst/audio/gstringbuffer.h:
10568 Add new field to the ringbufferspec to specify the expected latency
10569 between the underlying device read/write pointer, this is needed
10570 when writing sinks that sit a little closer to the hardware.
10571 Add some more docs for other fields.
10573 2008-05-07 10:38:23 +0000 Wim Taymans <wim.taymans@gmail.com>
10575 gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
10576 Original commit message from CVS:
10577 * gst-libs/gst/app/.cvsignore:
10578 * gst-libs/gst/app/Makefile.am:
10579 * gst-libs/gst/app/gstapp-marshal.list:
10580 Add marshal.list, make it compile and add to cvsignore.
10581 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
10582 (gst_app_sink_stop):
10584 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
10585 (gst_app_src_init), (gst_app_src_set_property),
10586 (gst_app_src_get_property), (gst_app_src_unlock),
10587 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
10588 (gst_app_src_create), (gst_app_src_set_caps),
10589 (gst_app_src_get_caps), (gst_app_src_set_size),
10590 (gst_app_src_get_size), (gst_app_src_set_seekable),
10591 (gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
10592 (gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
10593 (gst_app_src_end_of_stream):
10594 * gst-libs/gst/app/gstappsrc.h:
10595 Beat appsrc in shape, add signals and actions.
10597 Add properties for caps, size, seekability and max-buffers.
10598 Fix unlock/stop code.
10600 2008-05-06 12:35:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10602 gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
10603 Original commit message from CVS:
10604 * gst/volume/gstvolume.c: (volume_transform_ip):
10605 Return NOT_NEGOTIATED if we didn't set a process function yet for some
10606 reason instead of crashing later. Might fix bug #509125.
10608 2008-05-06 12:12:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10610 gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
10611 Original commit message from CVS:
10612 Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
10613 * gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
10614 * gst/audioconvert/audioconvert.h:
10615 * gst/audioconvert/gstaudioconvert.c:
10616 (gst_audio_convert_parse_caps),
10617 (structure_has_fixed_channel_positions),
10618 (gst_audio_convert_transform_caps):
10619 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
10620 Add support for more than 8 channels and NONE channel layouts. For
10621 more than 8 channels no channel conversion is supported yet, only
10622 format conversions are supported. Fixes bug #398033.
10623 * tests/check/elements/audioconvert.c: (verify_convert),
10624 (GST_START_TEST), (audioconvert_suite):
10625 Add some unit tests by Tim for checking the NONE channel layouts
10626 and more than 8 channels and add some more unit tests for channel
10629 2008-05-06 10:16:49 +0000 Wim Taymans <wim.taymans@gmail.com>
10631 gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
10632 Original commit message from CVS:
10633 * gst/playback/gstdecodebin2.c: (connect_pad):
10634 When autoplugging fails, set the element back to NULL before
10637 2008-05-06 09:59:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10639 win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols.
10640 Original commit message from CVS:
10641 * win32/common/libgstaudio.def:
10642 Add gst_base_audio_src_[sg]et_slave_method() to the exported
10645 2008-05-05 12:33:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10647 gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
10648 Original commit message from CVS:
10649 * gst/subparse/samiparse.c: (handle_start_sync),
10650 (end_sami_element), (characters_sami):
10651 Remove trailing, leading and double whitespaces.
10652 Correctly timestamp buffers and output the last buffer too.
10653 * tests/check/elements/subparse.c: (GST_START_TEST),
10655 Add a simple unit test for SAMI parsing.
10657 2008-05-05 11:14:48 +0000 Young-Ho Cha <ganadist@chollian.net>
10659 gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
10660 Original commit message from CVS:
10661 Patch by: Young-Ho Cha <ganadist at chollian dot net>
10662 * gst/subparse/samiparse.c: (handle_start_sync),
10663 (start_sami_element), (end_sami_element), (characters_sami),
10664 (sami_context_reset):
10665 Only output characters inside the "sync" elements. There could be
10666 other elements like "style" that have some content but should
10667 not be printed. Fixes bug #467911.
10669 2008-05-05 10:27:45 +0000 Wim Taymans <wim.taymans@gmail.com>
10671 gst-libs/gst/app/gstappsink.*: Start some docs.
10672 Original commit message from CVS:
10673 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
10674 (gst_app_sink_init), (gst_app_sink_set_property),
10675 (gst_app_sink_get_property), (gst_app_sink_unlock_start),
10676 (gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
10677 (gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
10678 (gst_app_sink_preroll), (gst_app_sink_render),
10679 (gst_app_sink_set_caps), (gst_app_sink_set_drop),
10680 (gst_app_sink_get_drop):
10681 * gst-libs/gst/app/gstappsink.h:
10683 Add property to drop buffers when the queue is filled
10684 Fix unlocking and flushing when the queues are filled.
10686 2008-05-05 10:03:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10688 gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
10689 Original commit message from CVS:
10690 * gst/playback/gstplaybasebin.c: (set_audio_mute),
10691 (set_active_source):
10692 * gst/playback/gstplaybasebin.h:
10693 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
10694 (playbin_set_audio_mute):
10695 Allow setting -1 as current-audio to mute the current audio stream,
10696 similar to what is done for subtitles. Fixes bug #342294.
10698 2008-05-05 07:41:03 +0000 Edward Hervey <bilboed@bilboed.com>
10700 gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
10701 Original commit message from CVS:
10702 * gst-libs/gst/pbutils/descriptions.c: (formats):
10703 It's SorensOn and not SorensEn.
10705 2008-05-04 15:23:36 +0000 Tim-Philipp Müller <tim@centricular.net>
10707 gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
10708 Original commit message from CVS:
10709 * gst-libs/gst/pbutils/descriptions.c: (formats):
10710 Fix description of video/x-flash-video.
10712 2008-05-04 15:02:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10714 Remove some unused code.
10715 Original commit message from CVS:
10716 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
10717 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
10718 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
10719 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
10720 Remove some unused code.
10721 * gst/audioconvert/gstaudioquantize.c:
10722 (gst_audio_quantize_free_noise_shaping):
10723 Don't return before freeing the noise shaping history.
10725 2008-05-03 16:00:04 +0000 Tim-Philipp Müller <tim@centricular.net>
10727 tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
10728 Original commit message from CVS:
10729 * tests/check/elements/subparse.c: (do_test),
10730 (test_tmplayer_style3b), (subparse_suite):
10731 Add unit test for the tmplayer variant from bug #530962.
10733 2008-05-03 15:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
10735 gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
10736 Original commit message from CVS:
10737 * gst/subparse/gstsubparse.c: (handle_buffer),
10738 (gst_sub_parse_sink_event):
10739 * gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
10740 (tmplayer_parse_line):
10741 Fix parsing of tmplayer subtitle variant where every single line contains
10742 text and there isn't an empty line after each line to determine the
10743 duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
10744 making sure that we push out the last line of text without a duration if
10745 there's still text left in the buffer at the end.
10747 2008-05-03 15:39:04 +0000 Tim-Philipp Müller <tim@centricular.net>
10749 gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
10750 Original commit message from CVS:
10751 * gst/subparse/gstsubparse.c: (feed_textbuf):
10752 Fix detection of discontinuities based on the buffer offset (doesn't work
10753 so well if no buffer offset is set) and also check for the DISCONT buffer
10754 flag. This keeps the parser state from being reset after each buffer in
10757 2008-05-03 12:09:16 +0000 Tim-Philipp Müller <tim@centricular.net>
10759 gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
10760 Original commit message from CVS:
10761 * gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
10762 Further fine-tuning: don't absolutely require sequence or GOP headers
10763 (as introduced in the previous commit), but adjust the typefind
10764 probabilities returned accordingly if we don't see them. Also make sure
10765 picture header and first slice are somewhat close to each other (which
10766 is not perfect but still better than requiring a fixed offset or having
10769 2008-05-02 12:13:08 +0000 Wim Taymans <wim.taymans@gmail.com>
10771 gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
10772 Original commit message from CVS:
10773 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
10774 (gst_basertppayload_sink_setcaps),
10775 (gst_basertppayload_sink_getcaps):
10776 Rename the setcaps/getcaps function internally to make it clear that
10777 they are called for the sink pad.
10779 2008-05-02 12:11:07 +0000 Wim Taymans <wim.taymans@gmail.com>
10781 gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
10782 Original commit message from CVS:
10783 * gst-libs/gst/rtp/gstbasertpdepayload.c:
10784 (gst_base_rtp_depayload_class_init),
10785 (gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
10786 (gst_base_rtp_depayload_packet_lost),
10787 (gst_base_rtp_depayload_set_gst_timestamp):
10788 * gst-libs/gst/rtp/gstbasertpdepayload.h:
10789 Catch packet-lost events from the jitterbuffer and convert them into a
10790 vmethod call (lost-packet) so that depayloaders can do something smart.
10791 Also add a default packet-lost function that sends out a segment update
10794 2008-05-02 11:13:05 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10796 gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
10797 Original commit message from CVS:
10798 * gst/playback/test4.c:
10799 * gst/playback/test5.c:
10800 * gst/playback/test6.c:
10801 * gst/playback/test7.c:
10802 Also include config.h when relying on defines from it. Fixes the
10803 build. Its been a please to serve :)
10805 2008-05-02 10:54:51 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
10808 * gst/videotestsrc/videotestsrc.c:
10809 Add support for NV12 and NV21 in videotestsrc
10810 Original commit message from CVS:
10811 * gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
10812 (paint_setup_NV21), (paint_hline_NV12_NV21):
10813 Add support for NV12 and NV21 in videotestsrc
10815 2008-05-02 10:02:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10817 gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
10818 Original commit message from CVS:
10819 * gst/videoscale/gstvideoscale.c:
10820 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
10821 * gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
10822 (vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
10823 (vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
10824 (vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
10825 (vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
10826 (vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
10827 (vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
10828 (vs_image_scale_linear_RGB555):
10829 Support 1x1 images as input and output as for example the BBC HQ new
10830 streams have 1x1 GIFs in the playlists for some reason.
10832 2008-05-01 19:11:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10834 gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
10835 Original commit message from CVS:
10836 * gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
10838 If we can't activate one of the decoders we plugged in (such as,
10839 say, musepackdec) for some reason (it might not support push mode,
10840 for example), remove any pad probes that close_pad_link() might
10841 have set up. This makes sure we later don't try to remove a probe
10842 for a pad that doesn't exist any longer, and avoids nast warnings
10843 and probably other things too.
10845 2008-04-30 20:54:56 +0000 Tim-Philipp Müller <tim@centricular.net>
10847 gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
10848 Original commit message from CVS:
10849 * gst/typefind/gsttypefindfunctions.c:
10850 (mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
10852 Rework mpeg video stream typefinding a bit more: make sure sequence,
10853 GOP, picture and slice headers appear in the order they should and
10854 that we've in fact at least had one of each; fix picture header
10855 detection; decouple picture and slice header check - don't assume
10856 they're at a fixed offset, there may be extra data in between. Also,
10857 announce varying degrees of probability depending on what we found
10858 exactly (multiple pictures, at least one picture, just sequence and
10859 GOP headers). Finally, in _ensure_data(), take into account that we
10860 might be typefinding smaller amounts of data, such as the first
10861 buffer of a stream, so fall back to the minimum size needed as long
10862 as that's available, instead of erroring out if there's less than
10863 2kB of data. Fixes #526173. Conveniently also doesn't recognise the
10864 fuzzed file from #399342 as valid.
10866 2008-04-30 17:06:45 +0000 Michael Smith <msmith@xiph.org>
10868 ext/theora/theoradec.c: Cool kids don't divide by zero.
10869 Original commit message from CVS:
10870 * ext/theora/theoradec.c:
10871 Cool kids don't divide by zero.
10872 Treat PAR of x:0 as 1:1.
10875 2008-04-30 14:37:52 +0000 Tim-Philipp Müller <tim@centricular.net>
10877 gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
10878 Original commit message from CVS:
10879 * gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
10880 (mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
10881 (mpeg_video_stream_type_find):
10882 Refactor a bit: use context structure to track parsing offset and size of
10883 available data and make the code a bit clearer. Fixes bad memory access
10886 2008-04-28 22:18:49 +0000 Michael Smith <msmith@xiph.org>
10888 gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
10889 Original commit message from CVS:
10890 * gst/playback/test4.c:
10891 * gst/playback/test5.c:
10892 * gst/playback/test6.c:
10893 * gst/tcp/gstmultifdsink.c:
10894 Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
10897 2008-04-28 08:51:38 +0000 Wim Taymans <wim.taymans@gmail.com>
10899 gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
10900 Original commit message from CVS:
10901 * gst-libs/gst/audio/gstbaseaudiosink.h:
10903 * gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
10904 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
10905 (gst_base_audio_src_set_slave_method),
10906 (gst_base_audio_src_get_slave_method),
10907 (gst_base_audio_src_set_property),
10908 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
10909 * gst-libs/gst/audio/gstbaseaudiosrc.h:
10910 Add property and methods for selecting the clock slave method in the
10911 source, like in the sink.
10912 We only implement "none" and "re-timestamp" for now.
10913 API: gst_base_audio_src_set_slave_method()
10914 API: gst_base_audio_src_get_slave_method()
10916 2008-04-25 18:18:47 +0000 Wim Taymans <wim.taymans@gmail.com>
10918 gst-libs/gst/app/gstappsink.*: Add more docs.
10919 Original commit message from CVS:
10920 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
10921 (gst_app_sink_init), (gst_app_sink_set_property),
10922 (gst_app_sink_get_property), (gst_app_sink_event),
10923 (gst_app_sink_preroll), (gst_app_sink_render),
10924 (gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
10925 (gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
10926 (gst_app_sink_pull_buffer):
10927 * gst-libs/gst/app/gstappsink.h:
10929 Add signals for when preroll and render buffers are available.
10930 Add property to control signal emission.
10931 Add property to control the max queue size.
10933 2008-04-25 07:37:09 +0000 Wim Taymans <wim.taymans@gmail.com>
10935 gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
10936 Original commit message from CVS:
10937 * gst-libs/gst/rtp/gstrtpbuffer.c:
10938 Fix the docs about the seqnum compare function, it returns a difference.
10940 2008-04-24 09:27:35 +0000 Edward Hervey <bilboed@bilboed.com>
10942 ext/alsa/gstalsadeviceprobe.c: Don't return before freeing up the allocated structures.
10943 Original commit message from CVS:
10944 * ext/alsa/gstalsadeviceprobe.c:
10945 (gst_alsa_get_device_list): Don't return before freeing up
10946 the allocated structures.
10948 2008-04-24 08:19:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10950 gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
10951 Original commit message from CVS:
10952 * gst/playback/gstplaybin.c:
10953 Remove obsolete streaminfo code and fix a leak. Fixes #529546
10955 2008-04-23 13:50:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10957 ext/ogg/gstoggdemux.c: Revert the event part, that should not go in.
10958 Original commit message from CVS:
10959 * ext/ogg/gstoggdemux.c:
10960 Revert the event part, that should not go in.
10962 2008-04-23 13:45:29 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10964 ext/ogg/gstoggdemux.c: Don't leak GstPluginFeatures when filtering.
10965 Original commit message from CVS:
10966 * ext/ogg/gstoggdemux.c:
10967 Don't leak GstPluginFeatures when filtering.
10969 2008-04-23 08:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10971 sys/xvimage/xvimagesink.c: Add some logging for cases when grabbing the xv failed.
10972 Original commit message from CVS:
10973 * sys/xvimage/xvimagesink.c:
10974 Add some logging for cases when grabbing the xv failed.
10976 2008-04-22 06:18:04 +0000 David Schleef <ds@schleef.org>
10978 ext/ogg/gstoggmux.c: Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos packet. Should conform to what we cu...
10979 Original commit message from CVS:
10980 * ext/ogg/gstoggmux.c:
10981 Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos
10982 packet. Should conform to what we currently think is the
10983 final Ogg/Dirac muxing spec.
10985 2008-04-22 06:13:43 +0000 David Schleef <ds@schleef.org>
10987 sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display. Dark g...
10988 Original commit message from CVS:
10989 * sys/xvimage/xvimagesink.c:
10990 Fix typo that causes the overlay keying color to bright green
10991 on a 16-bit display. Dark grey good. Bright green bad.
10993 2008-04-21 13:47:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10995 ext/gnomevfs/gstgnomevfsuri.c: Add FIXME comment about using uri-list for source and sink.
10996 Original commit message from CVS:
10997 * ext/gnomevfs/gstgnomevfsuri.c:
10998 Add FIXME comment about using uri-list for source and sink.
11000 2008-04-20 11:42:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11002 ext/ogg/gstogmparse.c: GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to vaargs functions to gin...
11003 Original commit message from CVS:
11004 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
11005 GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to
11006 vaargs functions to gint. Otherwise the fractions will get 0 set
11007 instead of the correct value on big endian systems. Fixes bug #529018.
11009 2008-04-20 10:17:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11011 ext/gnomevfs/: Get the list of supported URI schemes in a threadsafe way and use the same list for the source and sink.
11012 Original commit message from CVS:
11013 * ext/gnomevfs/gstgnomevfssink.c:
11014 (gst_gnome_vfs_sink_uri_get_protocols):
11015 * ext/gnomevfs/gstgnomevfssrc.c:
11016 (gst_gnome_vfs_src_uri_get_protocols):
11017 * ext/gnomevfs/gstgnomevfsuri.c: (_internal_get_supported_uris),
11018 (gst_gnomevfs_get_supported_uris):
11019 Get the list of supported URI schemes in a threadsafe way and use the
11020 same list for the source and sink.
11022 2008-04-20 10:11:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11024 ext/gio/gstgio.c: Don't generate a new supported protocols list on each call but cache it. It's supposed to be static...
11025 Original commit message from CVS:
11026 * ext/gio/gstgio.c: (_internal_get_supported_protocols),
11027 (gst_gio_get_supported_protocols):
11028 Don't generate a new supported protocols list on each call but cache
11029 it. It's supposed to be static anyway, this way we only leak it once
11031 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
11032 (gst_gio_sink_class_init), (gst_gio_sink_finalize),
11033 (gst_gio_sink_set_property), (gst_gio_sink_get_property),
11034 (gst_gio_sink_start):
11035 * ext/gio/gstgiosink.h:
11036 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
11037 (gst_gio_src_class_init), (gst_gio_src_finalize),
11038 (gst_gio_src_set_property), (gst_gio_src_get_property),
11039 (gst_gio_src_start):
11040 * ext/gio/gstgiosrc.h:
11041 API: Add "file" properties where one can set a GFile as source/destination.
11042 Add locking to the properties and use gst_element_class_set_details_simple()
11043 instead of a static GstElementDetails struct.
11045 2008-04-19 20:06:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11047 gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
11048 Original commit message from CVS:
11049 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
11051 Add "mpp" and "mp+" as possible extensions for MusePack files.
11052 Add typefinding for MusePack StreamVersion 8 files and include the
11053 stream version in the caps.
11055 2008-04-19 16:33:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11057 gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
11058 Original commit message from CVS:
11059 * gst-libs/gst/rtp/gstrtppayloads.c:
11060 (gst_rtp_payload_info_for_name):
11061 Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
11063 2008-04-18 17:10:43 +0000 Tim-Philipp Müller <tim@centricular.net>
11065 configure.ac: Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level (NB: this only affects compilation of some...
11066 Original commit message from CVS:
11068 Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level
11069 (NB: this only affects compilation of some of the examples).
11070 Remove some configure.ac cruft that's not needed any longer.
11072 2008-04-18 14:54:01 +0000 Edward Hervey <bilboed@bilboed.com>
11074 gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
11075 Original commit message from CVS:
11076 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
11077 Don't validate the payload if there isn't any.
11080 2008-04-17 07:33:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11082 gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
11083 Original commit message from CVS:
11084 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
11085 Use g_atomic_int_set() instead of gst_atomic_int_set().
11087 2008-04-17 07:29:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11089 ext/gio/gstgio.c: Return NULL instead of a gchar * array with one NULL element if we don't get any supported URI sche...
11090 Original commit message from CVS:
11091 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11092 Return NULL instead of a gchar * array with one NULL element if we
11093 don't get any supported URI schemes from GIO.
11095 2008-04-15 19:06:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11097 gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
11098 Original commit message from CVS:
11099 * gst/audiotestsrc/gstaudiotestsrc.c:
11100 Remove cpp style commented old code.
11102 2008-04-15 19:02:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11104 gst/playback/gstdecodebin2.c: Fix signal docs.
11105 Original commit message from CVS:
11106 * gst/playback/gstdecodebin2.c:
11109 2008-04-14 17:58:19 +0000 Tim-Philipp Müller <tim@centricular.net>
11111 ext/pango/gsttextoverlay.c: Fix textoverlay unit test again by making the supposed default value for the wait-text pr...
11112 Original commit message from CVS:
11113 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
11114 (gst_text_overlay_init):
11115 Fix textoverlay unit test again by making the supposed default
11116 value for the wait-text property the actual default value.
11117 Also fix Since: tag for new property.
11119 2008-04-11 17:13:52 +0000 Tim-Philipp Müller <tim@centricular.net>
11121 gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
11122 Original commit message from CVS:
11123 * gst-libs/gst/video/video.c: (gst_video_format_new_caps),
11124 (gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
11125 (gst_video_format_get_pixel_stride),
11126 (gst_video_format_get_component_width),
11127 (gst_video_format_get_component_height),
11128 (gst_video_format_get_component_offset), (gst_video_format_get_size),
11129 (gst_video_format_convert):
11130 Add guards to these functions to ensure sane input values.
11131 * tests/check/libs/video.c:
11132 Fix unit test not to create caps with width=0 and height=0.
11134 2008-04-11 01:25:01 +0000 Wim Taymans <wim.taymans@gmail.com>
11136 docs/design/draft-keyframe-force.txt: Fix typo.
11137 Original commit message from CVS:
11138 * docs/design/draft-keyframe-force.txt:
11140 * gst/playback/gstqueue2.c: (update_buffering),
11141 (gst_queue_handle_src_query):
11142 Set buffering mode in the messages.
11143 Set buffering percent in the query.
11144 * tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
11145 (do_stream_buffering), (do_download_buffering), (msg_buffering):
11146 Do some more fancy things based on the buffering method in use.
11148 2008-04-09 21:42:24 +0000 Wim Taymans <wim.taymans@gmail.com>
11150 tests/examples/seek/seek.c: Add basic download reports to seek using the new buffering API.
11151 Original commit message from CVS:
11152 * tests/examples/seek/seek.c: (update_fill), (set_update_fill),
11153 (play_cb), (pause_cb), (stop_cb), (msg_state_changed),
11154 (msg_buffering), (main):
11155 Add basic download reports to seek using the new buffering API.
11157 2008-04-09 21:40:17 +0000 Wim Taymans <wim.taymans@gmail.com>
11159 gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
11160 Original commit message from CVS:
11161 * gst/playback/gstqueue2.c: (update_buffering),
11162 (gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
11163 (gst_queue_src_checkgetrange_function):
11164 Include extra buffering stats in the buffering message.
11165 Implement BUFFERING query.
11166 * gst/playback/gsturidecodebin.c: (do_async_start),
11167 (do_async_done), (type_found), (setup_streaming), (setup_source),
11168 (gst_uri_decode_bin_change_state):
11169 Only add decodebin2 when the type is found in streaming mode.
11170 Make uridecodebin async to PAUSED even when we don't have decodebin2
11173 2008-04-09 08:38:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11175 ext/gio/gstgio.c: Filter cdda from the supported URI schemes. We can't support musicbrainz tags and everything else o...
11176 Original commit message from CVS:
11177 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11178 Filter cdda from the supported URI schemes. We can't support
11179 musicbrainz tags and everything else one expects from a cdda source
11180 with GIO. Fixes bug #526794.
11182 2008-04-07 22:37:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11184 * sys/xvimage/xvimagesink.c:
11185 Fix calculation of 'expected size' for YV12 buffers.
11186 Original commit message from CVS:
11187 2008-04-07 Jan Schmidt <jan.schmidt@sun.com>
11188 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
11189 (gst_xvimagesink_buffer_alloc):
11190 Fix calculation of 'expected size' for YV12 buffers.
11191 Be a little more verbose in the debug output for buffer-alloc'ed
11192 buffers which turn out to have the wrong size.
11194 2008-04-07 22:26:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11197 Fix calculation of 'expected size' for YV12 buffers.
11198 Original commit message from CVS:
11199 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
11200 (gst_xvimagesink_buffer_alloc):
11201 Fix calculation of 'expected size' for YV12 buffers.
11202 Be a little more verbose in the debug output for buffer-alloc'ed
11203 buffers which turn out to have the wrong size.
11205 2008-04-07 10:50:11 +0000 Tim-Philipp Müller <tim@centricular.net>
11207 Merge other changes from 0.10.19 release branch.
11208 Original commit message from CVS:
11211 * gst-plugins-base.doap:
11212 Merge other changes from 0.10.19 release branch.
11214 2008-04-06 20:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
11216 gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
11217 Original commit message from CVS:
11218 * gst-libs/gst/audio/gstbaseaudiosink.c:
11219 (gst_base_audio_sink_class_init):
11220 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11221 (gst_base_audio_src_class_init):
11222 * gst/playback/gstplayback.c: (plugin_init):
11223 * gst/volume/gstvolume.c: (plugin_init):
11224 Work around missing bits of thread-safety on older GLibs some
11225 more to avoid assertions when starting up multiple playbin
11226 objects concurrently (see #512382).
11228 2008-04-06 17:19:39 +0000 Tim-Philipp Müller <tim@centricular.net>
11230 gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
11231 Original commit message from CVS:
11232 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
11233 Remove some more fields.
11235 2008-04-06 08:56:07 +0000 Damien Lespiau <damien.lespiau@gmail.com>
11237 configure.ac: Actually build dlls when cross-compiling with mingw32.
11238 Original commit message from CVS:
11239 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
11241 Actually build dlls when cross-compiling with mingw32.
11244 2008-04-03 23:01:11 +0000 Tim-Philipp Müller <tim@centricular.net>
11246 configure.ac: Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
11247 Original commit message from CVS:
11249 Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
11251 2008-04-03 16:10:53 +0000 Wim Taymans <wim.taymans@gmail.com>
11253 tests/examples/seek/seek.c: Add statusbar.
11254 Original commit message from CVS:
11255 * tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
11256 (msg_buffering), (connect_bus_signals), (main):
11258 Add buffering support with feedback in the statusbar.
11260 2008-04-03 15:58:37 +0000 Tim-Philipp Müller <tim@centricular.net>
11262 ext/ogg/gstoggmux.c: Fix sample pipeline description.
11263 Original commit message from CVS:
11264 * ext/ogg/gstoggmux.c:
11265 Fix sample pipeline description.
11267 2008-04-03 14:58:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11269 docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
11270 Original commit message from CVS:
11271 * docs/plugins/Makefile.am:
11272 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
11273 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
11274 * docs/plugins/gst-plugins-base-plugins-sections.txt:
11275 Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
11276 * docs/plugins/gst-plugins-base-plugins.args:
11277 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11278 * docs/plugins/gst-plugins-base-plugins.interfaces:
11279 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11280 * docs/plugins/inspect/plugin-adder.xml:
11281 * docs/plugins/inspect/plugin-alsa.xml:
11282 * docs/plugins/inspect/plugin-audioconvert.xml:
11283 * docs/plugins/inspect/plugin-audiorate.xml:
11284 * docs/plugins/inspect/plugin-audioresample.xml:
11285 * docs/plugins/inspect/plugin-audiotestsrc.xml:
11286 * docs/plugins/inspect/plugin-cdparanoia.xml:
11287 * docs/plugins/inspect/plugin-decodebin.xml:
11288 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
11289 * docs/plugins/inspect/plugin-gdp.xml:
11290 * docs/plugins/inspect/plugin-gnomevfs.xml:
11291 * docs/plugins/inspect/plugin-libvisual.xml:
11292 * docs/plugins/inspect/plugin-ogg.xml:
11293 * docs/plugins/inspect/plugin-pango.xml:
11294 * docs/plugins/inspect/plugin-playback.xml:
11295 * docs/plugins/inspect/plugin-queue2.xml:
11296 * docs/plugins/inspect/plugin-subparse.xml:
11297 * docs/plugins/inspect/plugin-tcp.xml:
11298 * docs/plugins/inspect/plugin-theora.xml:
11299 * docs/plugins/inspect/plugin-typefindfunctions.xml:
11300 * docs/plugins/inspect/plugin-uridecodebin.xml:
11301 * docs/plugins/inspect/plugin-video4linux.xml:
11302 * docs/plugins/inspect/plugin-videorate.xml:
11303 * docs/plugins/inspect/plugin-videoscale.xml:
11304 * docs/plugins/inspect/plugin-videotestsrc.xml:
11305 * docs/plugins/inspect/plugin-volume.xml:
11306 * docs/plugins/inspect/plugin-vorbis.xml:
11307 * docs/plugins/inspect/plugin-ximagesink.xml:
11308 * docs/plugins/inspect/plugin-xvimagesink.xml:
11309 Update introspection data.
11310 * ext/ogg/gstoggmux.c:
11312 * gst/playback/gstdecodebin2.c:
11313 Don't use gtk-doc style comment start for private stuff, but make it
11314 formatted like this for consistency.
11316 2008-04-03 12:16:04 +0000 Wim Taymans <wim.taymans@gmail.com>
11318 gst/playback/gstdecodebin2.c: Remove fakesink hack, we can now implement this more elegantly.
11319 Original commit message from CVS:
11320 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
11321 (gst_decode_bin_init), (gst_decode_bin_dispose),
11322 (gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
11323 (gst_decode_bin_set_property), (gst_decode_bin_get_property),
11324 (analyze_new_pad), (connect_pad), (expose_pad),
11325 (gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
11326 (gst_decode_group_expose), (gst_decode_group_free),
11327 (do_async_start), (do_async_done), (gst_decode_bin_change_state):
11328 Remove fakesink hack, we can now implement this more elegantly.
11329 Added property to bypass typefinding.
11330 Removed underrun callback and demuxer pad probe, we now use the srcpad
11331 probe to expose groups.
11332 API::sink-caps property
11333 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
11334 Guard against multiple emissions of the no_more_pads signal, which
11335 happens when we are dealing with chained oggs.
11336 * gst/playback/gsturidecodebin.c: (remove_decoders),
11337 (make_decoder), (type_found), (setup_streaming), (source_new_pad),
11339 For streams, use our own typefind element and plug our queue after it.
11340 We will need this to determine the type of buffering to use for the
11343 2008-04-03 10:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
11345 gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
11346 Original commit message from CVS:
11347 * gst-libs/gst/audio/gstbaseaudiosink.c:
11348 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
11349 Guard against over and underflows because of clock slaving.
11350 When we are using our own clock, still compensate for any calibrations
11351 that we might have done to our clock.
11353 2008-04-03 10:22:33 +0000 Wim Taymans <wim.taymans@gmail.com>
11355 ext/theora/theoradec.c: Don't try to do anything fancy with the return code from pushing an event, it does not have e...
11356 Original commit message from CVS:
11357 * ext/theora/theoradec.c: (theora_handle_type_packet),
11358 (theora_dec_chain):
11359 Don't try to do anything fancy with the return code from pushing an
11360 event, it does not have enough information to turn it into a
11363 2008-04-03 10:19:43 +0000 Wim Taymans <wim.taymans@gmail.com>
11365 ext/ogg/gstoggdemux.c: Add small debug line.
11366 Original commit message from CVS:
11367 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_reset),
11368 (gst_ogg_demux_chain_elem_pad):
11369 Add small debug line.
11370 Pass return code from the internal decoder instead of the too generic
11373 2008-04-03 06:39:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11375 gst-libs/gst/cdda/: Use GLib's base64 implementation instead of our own.
11376 Original commit message from CVS:
11377 * gst-libs/gst/cdda/Makefile.am:
11378 * gst-libs/gst/cdda/base64.c:
11379 * gst-libs/gst/cdda/base64.h:
11380 * gst-libs/gst/cdda/gstcddabasesrc.c:
11381 (gst_cddabasesrc_calculate_musicbrainz_discid):
11382 Use GLib's base64 implementation instead of our own.
11384 2008-04-02 15:41:50 +0000 Wim Taymans <wim.taymans@gmail.com>
11386 ext/ogg/gstoggdemux.c: Refix oggdemux, we only have a problem if we failed to find a chain and we are not EOF.
11387 Original commit message from CVS:
11388 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
11389 (gst_ogg_demux_read_chain):
11390 Refix oggdemux, we only have a problem if we failed to find a chain and
11393 2008-04-02 15:07:01 +0000 Victor STINNER <victor.stinner@haypocalc.com>
11395 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...
11396 Original commit message from CVS:
11397 Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
11398 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
11399 (gst_ogg_demux_read_chain):
11400 When we fail to find a BOS page and we and up with no chain, error out
11401 properly instead of segfaulting. Fixes #525665.
11403 2008-04-02 14:58:05 +0000 Wim Taymans <wim.taymans@gmail.com>
11405 ext/ogg/gstoggdemux.c: The new-pad-group sequence is add-pads, no-more-pads, add-pads, no-more-pads...
11406 Original commit message from CVS:
11407 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
11408 (gst_ogg_demux_read_chain), (gst_ogg_demux_handle_page):
11409 The new-pad-group sequence is add-pads, no-more-pads, add-pads,
11412 2008-04-02 11:08:05 +0000 Wim Taymans <wim.taymans@gmail.com>
11414 gst/playback/gstqueue2.c: Update the estimated input data when we push out a buffer.
11415 Original commit message from CVS:
11416 * gst/playback/gstqueue2.c: (update_out_rates),
11417 (gst_queue_open_temp_location_file),
11418 (gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
11419 (gst_queue_handle_src_query), (gst_queue_set_property):
11420 Update the estimated input data when we push out a buffer.
11421 Add some debug info about the temp file.
11422 Only forward src events when we are not using a temp file.
11423 Don't block the duration query, we need to find something better.
11424 Don't leak the temp filename.
11426 2008-04-01 14:01:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11428 configure.ac: Require GLib 2.12 and liboil 0.3.14.
11429 Original commit message from CVS:
11431 Require GLib 2.12 and liboil 0.3.14.
11432 * gst/volume/gstvolume.c: (volume_process_double):
11433 Unconditionally use liboil 0.3.14 function.
11435 2008-03-31 16:08:45 +0000 Wim Taymans <wim.taymans@gmail.com>
11437 gst-libs/gst/riff/riff-media.c: ms-gsm can have arbitrarty sample rates. See #481354.
11438 Original commit message from CVS:
11439 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
11440 ms-gsm can have arbitrarty sample rates. See #481354.
11442 2008-03-28 16:22:35 +0000 Wim Taymans <wim.taymans@gmail.com>
11444 gst-libs/gst/riff/riff-media.c: MP4S is generic MPEG-4, not a microsoft variant.
11445 Original commit message from CVS:
11446 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
11447 MP4S is generic MPEG-4, not a microsoft variant.
11449 2008-03-27 15:26:38 +0000 Michael Smith <msmith@xiph.org>
11451 gst/gdp/gstgdpdepay.c: Check the body CRC (if set) when depayloading.
11452 Original commit message from CVS:
11453 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
11454 Check the body CRC (if set) when depayloading.
11457 2008-03-24 17:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
11459 ext/pango/gsttextoverlay.c: Fix Since: version for new property.
11460 Original commit message from CVS:
11461 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
11462 Fix Since: version for new property.
11464 2008-03-24 16:40:08 +0000 Wim Taymans <wim.taymans@gmail.com>
11466 gst-libs/gst/rtsp/gstrtspconnection.c: Don't error when poll_wait returns EAGAIN.
11467 Original commit message from CVS:
11468 * gst-libs/gst/rtsp/gstrtspconnection.c:
11469 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
11470 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
11471 Don't error when poll_wait returns EAGAIN.
11473 2008-03-24 14:08:22 +0000 Wim Taymans <wim.taymans@gmail.com>
11475 gst/playback/gstqueue2.c: The queue is never filled when there are no buffers in the queue at all.
11476 Original commit message from CVS:
11477 * gst/playback/gstqueue2.c: (gst_queue_is_filled):
11478 The queue is never filled when there are no buffers in the queue at all.
11481 2008-03-24 12:26:30 +0000 Wim Taymans <wim.taymans@gmail.com>
11483 gst/playback/gstplaybin2.c: Update some docs.
11484 Original commit message from CVS:
11485 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
11486 (init_group), (free_group), (gst_play_bin_init),
11487 (gst_play_bin_finalize), (gst_play_bin_set_uri),
11488 (gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
11489 (gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
11490 (gst_play_bin_set_current_video_stream),
11491 (gst_play_bin_set_current_audio_stream),
11492 (gst_play_bin_set_current_text_stream),
11493 (gst_play_bin_set_encoding), (gst_play_bin_set_property),
11494 (gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
11495 (no_more_pads_cb), (perform_eos), (autoplug_select_cb),
11496 (activate_group), (deactivate_group), (setup_next_source),
11497 (save_current_group), (gst_play_bin_change_state):
11499 Add new locks and conds to protect pipeline creation and group
11501 Implement the sub-uri property.
11502 Keep track of pending uridecodebin creation and configure the output
11503 pipeline after all streams are configured.
11504 Propagate subtitle encoding to the uridecodebins.
11505 Implement getting the video/audio/visualisation elements.
11506 Use input-selector for stream switching.
11507 If we are asked to do visualisation, prefer to autoplug raw sinks
11508 instead of sinks that accept encoded data.
11510 2008-03-24 12:15:26 +0000 Wim Taymans <wim.taymans@gmail.com>
11512 gst/playback/gstplaysink.*: Add methods to get audio/video/vis elements.
11513 Original commit message from CVS:
11514 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
11515 (gst_play_sink_init), (gst_play_sink_dispose),
11516 (gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
11517 (gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
11518 (gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
11519 (gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
11520 (gst_play_sink_set_volume), (gst_play_sink_get_volume),
11521 (gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
11522 (gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
11523 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
11524 (gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
11525 * gst/playback/gstplaysink.h:
11526 Add methods to get audio/video/vis elements.
11527 Add methods to set the font description for the overlay.
11528 Remove properties, we're using this element with its methods only.
11529 Add support for subtitles.
11530 Rearrange the locking a bit to not use the object lock for protecting
11531 the pipeline construction.
11532 Try to use the volume and mute property on the sink when its available.
11533 Implement the mute option with volume when the sink does not have a mute
11535 Only add volume element when the sink has no volume property.
11536 Only do visualisations with raw audio pads.
11538 2008-03-24 12:03:02 +0000 Wim Taymans <wim.taymans@gmail.com>
11540 ext/pango/gsttextoverlay.*: Add property to configure waiting for text on the textpad or not, with the default behavi...
11541 Original commit message from CVS:
11542 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
11543 (gst_text_overlay_init), (gst_text_overlay_set_property),
11544 (gst_text_overlay_get_property), (gst_text_overlay_src_event),
11545 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
11546 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
11547 (gst_text_overlay_change_state):
11548 * ext/pango/gsttextoverlay.h:
11549 Add property to configure waiting for text on the textpad or not, with
11550 the default behaviour being the old one (always wait for text before
11551 rendering the video). This default behaviour is usually not the best one
11552 because the text stream can very sparse and could require queueing a lot
11554 Fix the flushing and EOS handing so that we don't mix up their meaning.
11556 2008-03-24 11:54:02 +0000 Wim Taymans <wim.taymans@gmail.com>
11558 gst/playback/gsturidecodebin.c: Add a readonly source property and notify.
11559 Original commit message from CVS:
11560 * gst/playback/gsturidecodebin.c:
11561 (gst_uri_decode_bin_autoplug_factories),
11562 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
11563 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
11564 (gst_uri_decode_bin_set_property),
11565 (gst_uri_decode_bin_get_property), (no_more_pads_full),
11566 (new_decoded_pad_cb), (gen_source_element), (remove_decoders),
11567 (proxy_autoplug_factories_signal), (make_decoder),
11568 (source_new_pad), (setup_source):
11569 Add a readonly source property and notify.
11570 Add new lock for protecting the construction of the pipeline.
11571 Keep track of the decodebins we plugged.
11572 Correctly proxy the autoplug signal so that it actually continues.
11573 Proxy subtitle-encoding to the decodebins.
11575 2008-03-24 11:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
11577 tests/examples/seek/seek.c: Rearrange some buttons in playbin2 and make some other boxes insensitive when needed.
11578 Original commit message from CVS:
11579 * tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
11580 (text_toggle_cb), (update_streams), (main):
11581 Rearrange some buttons in playbin2 and make some other boxes insensitive
11583 Add language codes to subtitle selection boxes when we gind the right
11584 tags for the streams.
11586 2008-03-24 11:36:08 +0000 Wim Taymans <wim.taymans@gmail.com>
11588 gst/playback/gstdecodebin2.c: Protect caps property with the object lock.
11589 Original commit message from CVS:
11590 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
11591 (gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
11592 (gst_decode_bin_set_subs_encoding),
11593 (gst_decode_bin_get_subs_encoding),
11594 (gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
11595 (deactivate_free_recursive):
11596 Protect caps property with the object lock.
11597 Protect encoding property with the object lock.
11598 Keep list of elements we added that have the subtitle-encoding property.
11599 Distribute the subtitle-encoding to all of the elements when it
11602 2008-03-24 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
11604 gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
11605 Original commit message from CVS:
11606 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
11607 Small debug improvement.
11608 * gst-libs/gst/audio/gstbaseaudiosink.c:
11609 (gst_base_audio_sink_render):
11610 Fix bug in determining the sample start/stop position, we want to base
11611 this decision on the fact that we are going forwards or backwards, not
11612 slower or faster. This fixes some ugly resync warnings when playing at
11615 2008-03-23 13:41:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11617 ext/gio/gstgio.c: Correctly set the supported URI schemes and don't leave some schemes in the middle or at the start ...
11618 Original commit message from CVS:
11619 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11620 Correctly set the supported URI schemes and don't leave
11621 some schemes in the middle or at the start at NULL.
11623 2008-03-23 13:12:41 +0000 Tim-Philipp Müller <tim@centricular.net>
11625 tests/check/elements/gdpdepay.c: Make test compile without unused function/variable warnings on PPC.
11626 Original commit message from CVS:
11627 * tests/check/elements/gdpdepay.c:
11628 Make test compile without unused function/variable warnings on PPC.
11630 2008-03-22 15:00:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11632 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
11633 Original commit message from CVS:
11635 * ext/alsa/gstalsamixerelement.c:
11636 (gst_alsa_mixer_element_class_init):
11637 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
11638 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
11639 * ext/cdparanoia/gstcdparanoiasrc.c:
11640 (gst_cd_paranoia_src_class_init):
11641 * ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
11642 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
11643 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
11644 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
11645 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
11646 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
11647 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
11648 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
11649 * ext/pango/gsttextrender.c: (gst_text_render_class_init):
11650 * ext/theora/theoradec.c: (gst_theora_dec_class_init):
11651 * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
11652 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
11653 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
11654 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
11655 (gst_audio_filter_template_class_init):
11656 * gst-libs/gst/audio/gstbaseaudiosink.c:
11657 (gst_base_audio_sink_class_init):
11658 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11659 (gst_base_audio_src_class_init):
11660 * gst-libs/gst/cdda/gstcddabasesrc.c:
11661 (gst_cdda_base_src_class_init):
11662 * gst-libs/gst/interfaces/mixertrack.c:
11663 (gst_mixer_track_class_init):
11664 * gst-libs/gst/rtp/gstbasertpdepayload.c:
11665 (gst_base_rtp_depayload_class_init):
11666 * gst-libs/gst/rtp/gstbasertppayload.c:
11667 (gst_basertppayload_class_init):
11668 * gst/audioconvert/gstaudioconvert.c:
11669 (gst_audio_convert_class_init):
11670 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
11671 * gst/audioresample/gstaudioresample.c:
11672 (gst_audioresample_class_init):
11673 * gst/audiotestsrc/gstaudiotestsrc.c:
11674 (gst_audio_test_src_class_init):
11675 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
11676 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
11677 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
11678 (preroll_unlinked):
11679 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
11680 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
11681 * gst/playback/gstplaysink.c: (gst_play_sink_class_init):
11682 * gst/playback/gstqueue2.c: (gst_queue_class_init):
11683 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
11684 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
11685 (gst_stream_selector_class_init):
11686 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
11687 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
11688 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
11689 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
11690 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
11691 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
11692 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
11693 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
11694 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
11695 * gst/videotestsrc/gstvideotestsrc.c:
11696 (gst_video_test_src_class_init):
11697 * gst/volume/gstvolume.c: (gst_volume_class_init):
11698 * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
11699 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
11700 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
11701 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
11702 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
11703 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
11704 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
11705 static strings (i.e. all). This gives us less memory usage,
11706 fewer allocations and thus less memory defragmentation. Depend
11707 on core CVS for this. Fixes bug #523806.
11709 2008-03-22 14:13:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11711 ext/gio/gstgio.c: Filter http and https protocols. GIO/GVfs handles them but it's impossible to implement iradio/icec...
11712 Original commit message from CVS:
11713 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11714 Filter http and https protocols. GIO/GVfs handles them but it's
11715 impossible to implement iradio/icecast with it. Better use
11716 souphttpsrc or something else for this.
11717 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
11718 If getting the file informations by a query fails try it with the
11719 seek-to-end trick too.
11721 2008-03-21 16:46:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11723 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...
11724 Original commit message from CVS:
11725 * gst/volume/gstvolume.c: (gst_volume_interface_supported),
11726 (gst_volume_base_init), (gst_volume_class_init),
11727 (volume_process_double), (volume_process_float),
11728 (volume_transform_ip), (plugin_init):
11729 memset buffers to zero if we get a GAP buffer. We usually see a
11730 buffer as one unit so let's handle it as one and don't care about
11731 volume changes while processing one buffer.
11732 Also clean up some stuff a bit.
11734 2008-03-21 15:58:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11736 gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
11737 Original commit message from CVS:
11738 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
11739 (gst_audio_convert_create_silence_buffer),
11740 (gst_audio_convert_transform):
11741 Make audioconvert GAP-aware by outputting silence buffers when the
11742 input has the GAP flag set. This is up to 8x faster.
11743 Based on a patch by Stefan Kost. Fixes bug #517813.
11745 2008-03-21 15:54:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11747 gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
11748 Original commit message from CVS:
11749 * gst/volume/gstvolume.c: (volume_process_double):
11750 Use oil_scalarmultiply_f64_ns() for double processing when it's
11751 available at compile time.
11753 2008-03-21 13:27:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11755 configure.ac: Fix lrint/lrintf checks to actually work. These functions are in libm on Linux at least so try to link ...
11756 Original commit message from CVS:
11758 Fix lrint/lrintf checks to actually work. These functions are
11759 in libm on Linux at least so try to link to it.
11761 2008-03-21 00:36:20 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11763 configure.ac: Back to development - 0.10.18.1
11764 Original commit message from CVS:
11766 Back to development - 0.10.18.1
11768 === release 0.10.18 ===
11770 2008-03-21 00:26:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11776 * docs/plugins/gst-plugins-base-plugins.args:
11777 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11778 * docs/plugins/gst-plugins-base-plugins.interfaces:
11779 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11780 * docs/plugins/gst-plugins-base-plugins.signals:
11781 * docs/plugins/inspect/plugin-adder.xml:
11782 * docs/plugins/inspect/plugin-alsa.xml:
11783 * docs/plugins/inspect/plugin-audioconvert.xml:
11784 * docs/plugins/inspect/plugin-audiorate.xml:
11785 * docs/plugins/inspect/plugin-audioresample.xml:
11786 * docs/plugins/inspect/plugin-audiotestsrc.xml:
11787 * docs/plugins/inspect/plugin-cdparanoia.xml:
11788 * docs/plugins/inspect/plugin-decodebin.xml:
11789 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
11790 * docs/plugins/inspect/plugin-gdp.xml:
11791 * docs/plugins/inspect/plugin-gnomevfs.xml:
11792 * docs/plugins/inspect/plugin-libvisual.xml:
11793 * docs/plugins/inspect/plugin-ogg.xml:
11794 * docs/plugins/inspect/plugin-pango.xml:
11795 * docs/plugins/inspect/plugin-playback.xml:
11796 * docs/plugins/inspect/plugin-queue2.xml:
11797 * docs/plugins/inspect/plugin-subparse.xml:
11798 * docs/plugins/inspect/plugin-tcp.xml:
11799 * docs/plugins/inspect/plugin-theora.xml:
11800 * docs/plugins/inspect/plugin-typefindfunctions.xml:
11801 * docs/plugins/inspect/plugin-uridecodebin.xml:
11802 * docs/plugins/inspect/plugin-video4linux.xml:
11803 * docs/plugins/inspect/plugin-videorate.xml:
11804 * docs/plugins/inspect/plugin-videoscale.xml:
11805 * docs/plugins/inspect/plugin-videotestsrc.xml:
11806 * docs/plugins/inspect/plugin-volume.xml:
11807 * docs/plugins/inspect/plugin-vorbis.xml:
11808 * docs/plugins/inspect/plugin-ximagesink.xml:
11809 * docs/plugins/inspect/plugin-xvimagesink.xml:
11810 * gst-plugins-base.doap:
11812 * win32/common/config.h:
11814 Original commit message from CVS:
11817 2008-03-21 00:16:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11844 Original commit message from CVS:
11847 2008-03-18 12:19:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11849 0.10.17.4 pre-release
11850 Original commit message from CVS:
11852 * win32/common/config.h:
11853 0.10.17.4 pre-release
11855 2008-03-18 11:20:05 +0000 Wim Taymans <wim.taymans@gmail.com>
11857 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL when trying to print strings that could be NULL because this might...
11858 Original commit message from CVS:
11859 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
11860 Use GST_STR_NULL when trying to print strings that could be NULL because
11861 this might crash on some platforms. See #520808.
11863 2008-03-18 11:10:12 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
11865 gst-libs/gst/rtsp/gstrtspconnection.c: Generic Windows fixes that makes libgstrtsp work on Windows when coupled with ...
11866 Original commit message from CVS:
11867 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
11868 * gst-libs/gst/rtsp/gstrtspconnection.c:
11869 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
11870 (read_line), (gst_rtsp_connection_read_internal):
11871 Generic Windows fixes that makes libgstrtsp work on Windows when
11872 coupled with the new GstPoll API. See #520808.
11874 2008-03-17 22:06:56 +0000 Milosz Derezynski <internalerror@gmail.com>
11876 ext/gio/gstgiobasesrc.c: If seeking to a new position succeeds don't simply return from create() without creating a b...
11877 Original commit message from CVS:
11878 Patch by: Milosz Derezynski <internalerror at gmail dot com>
11879 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
11880 If seeking to a new position succeeds don't simply return from
11881 create() without creating a buffer. Do this only in the case
11882 seeking to the new position fails. Fixes bug #523054.
11884 2008-03-17 10:32:28 +0000 Tim-Philipp Müller <tim@centricular.net>
11886 gst-libs/gst/video/video.c: Fix gst_video_format_parse_caps() for RGB caps with alpha channel (#522635).
11887 Original commit message from CVS:
11888 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
11889 (gst_video_format_from_rgba32_masks):
11890 Fix gst_video_format_parse_caps() for RGB caps with alpha channel
11892 * tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
11893 Add unit test for the RGB caps parsing and creation, checking for
11894 internal consistency of the new API and consistency of the API with
11895 the old GST_VIDEO_CAPS_* defines.
11897 2008-03-14 18:42:35 +0000 David Schleef <ds@schleef.org>
11899 gst/videotestsrc/videotestsrc.c: Oops, revert last change because -base is in freeze.
11900 Original commit message from CVS:
11901 * gst/videotestsrc/videotestsrc.c: Oops, revert last change
11902 because -base is in freeze.
11904 2008-03-14 17:33:09 +0000 William M. Brack <wbrack@mmm.hk>
11906 gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
11907 Original commit message from CVS:
11908 Patch by: William M. Brack
11909 * gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
11911 2008-03-14 09:54:44 +0000 Wim Taymans <wim.taymans@gmail.com>
11913 gst/playback/gststreamselector.*: Revert change that caused regression until a real fix is found.
11914 Original commit message from CVS:
11915 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
11916 (gst_selector_pad_chain):
11917 * gst/playback/gststreamselector.h:
11918 Revert change that caused regression until a real fix is found.
11921 2008-03-12 12:39:13 +0000 Michael Smith <msmith@xiph.org>
11923 gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
11924 Original commit message from CVS:
11925 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
11926 * gst-libs/gst/audio/gstringbuffer.h:
11927 Rename recently added buffer types to make more sense.
11928 * ext/alsa/gstalsasink.c: (alsasink_parse_spec),
11929 (gst_alsasink_write):
11930 Adapt for above API changes.
11933 2008-03-11 13:23:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11935 win32/common/libgstnetbuffer.def: Add new symbol gst_netaddress_equal. Fixes bug #521743.
11936 Original commit message from CVS:
11937 * win32/common/libgstnetbuffer.def:
11938 Add new symbol gst_netaddress_equal. Fixes bug #521743.
11940 2008-03-11 00:25:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11942 0.10.17.3 pre-release
11943 Original commit message from CVS:
11945 * win32/common/config.h:
11946 0.10.17.3 pre-release
11948 2008-03-10 17:19:56 +0000 Wim Taymans <wim.taymans@gmail.com>
11950 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
11951 Original commit message from CVS:
11952 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11953 (gst_base_audio_src_create):
11954 Fix duration when no clock was provided. Fixes #520300.
11956 2008-03-07 18:17:44 +0000 Olivier Crete <tester@tester.ca>
11958 Add trivial function to compare GstNetAddress. See #520626.
11959 Original commit message from CVS:
11960 Patch by: Olivier Crete <tester at tester ca>
11961 * docs/libs/gst-plugins-base-libs-sections.txt:
11962 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
11963 * gst-libs/gst/netbuffer/gstnetbuffer.h:
11964 Add trivial function to compare GstNetAddress. See #520626.
11965 API: GstNetBuffer::gst_netaddress_equal
11967 2008-03-07 16:10:51 +0000 Wim Taymans <wim.taymans@gmail.com>
11969 gst/tcp/gstmultifdsink.c: Update mode property docs, it's deprecated now.
11970 Original commit message from CVS:
11971 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
11972 Update mode property docs, it's deprecated now.
11974 2008-03-07 15:48:51 +0000 Wim Taymans <wim.taymans@gmail.com>
11976 gst/: Remove GstPollMode from gstpoll constructor.
11977 Original commit message from CVS:
11978 * gst-libs/gst/rtsp/gstrtspconnection.c:
11979 (gst_rtsp_connection_create):
11980 * gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
11981 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
11982 * gst/tcp/gstmultifdsink.h:
11983 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
11984 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
11985 Remove GstPollMode from gstpoll constructor.
11987 2008-03-04 00:26:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11989 0.10.17.2 pre-release
11990 Original commit message from CVS:
11992 * win32/common/config.h:
11993 0.10.17.2 pre-release
11995 2008-03-03 23:59:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11997 gst/Makefile.am: GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean them twice
11998 Original commit message from CVS:
12000 GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
12002 * win32/common/libgstinterfaces.def:
12003 * win32/common/libgstrtp.def:
12004 Add new API to the defs
12006 2008-03-03 16:11:50 +0000 Mersad Jelacic <mersad@axis.com>
12008 gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
12009 Original commit message from CVS:
12010 Patch by: Mersad Jelacic <mersad at axis dot com>
12011 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
12012 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
12013 API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
12014 possible to specify the sample size in bits. (#509637)
12016 2008-03-03 13:59:19 +0000 Tim-Philipp Müller <tim@centricular.net>
12018 tests/check/libs/mixer.c: Add a few simple checks for the new message types.
12019 Original commit message from CVS:
12020 * tests/check/libs/mixer.c:
12021 Add a few simple checks for the new message types.
12023 2008-03-03 13:56:38 +0000 Tim-Philipp Müller <tim@centricular.net>
12025 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
12026 Original commit message from CVS:
12027 * docs/libs/gst-plugins-base-libs-sections.txt:
12028 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
12029 (gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
12030 (gst_mixer_message_get_type),
12031 (gst_mixer_message_parse_option_changed),
12032 (gst_mixer_message_parse_options_list_changed):
12033 * gst-libs/gst/interfaces/mixer.h: (GstMixerType),
12034 (GST_MIXER_MESSAGE_OPTION_CHANGED),
12035 (GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
12036 (GST_MIXER_MESSAGE_MIXER_CHANGED):
12037 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
12038 and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
12040 2008-03-03 13:50:18 +0000 Tim-Philipp Müller <tim@centricular.net>
12042 gst-libs/gst/interfaces/mixeroptions.*: API: add GstMixerOptions::get_values vfunc (#519906)
12043 Original commit message from CVS:
12044 * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
12045 (gst_mixer_options_get_values):
12046 * gst-libs/gst/interfaces/mixeroptions.h:
12047 (GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
12048 (_GstMixerOptions), (_GstMixerOptionsClass):
12049 API: add GstMixerOptions::get_values vfunc (#519906)
12051 2008-03-03 12:01:15 +0000 Peter Kjellerstedt <pkj@axis.com>
12053 configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#4...
12054 Original commit message from CVS:
12056 Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
12057 plug-ins are included/excluded. (#498222)
12059 2008-03-03 06:22:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12061 gst/typefind/gsttypefindfunctions.c: Add typefinder for IMelody files, using audio/x-imelody.
12062 Original commit message from CVS:
12063 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
12064 Add typefinder for IMelody files, using audio/x-imelody.
12067 2008-03-03 06:04:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12069 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
12070 Original commit message from CVS:
12071 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
12072 * ext/alsa/gstalsasink.c: (set_hwparams):
12073 * ext/alsa/gstalsasrc.c: (set_hwparams):
12074 * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
12075 * ext/ogg/gstoggmux.h:
12076 * ext/ogg/gstogmparse.c:
12077 * gst-libs/gst/audio/audio.c:
12078 * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
12079 * gst-libs/gst/pbutils/missing-plugins.c:
12080 (gst_missing_uri_sink_message_new),
12081 (gst_missing_element_message_new),
12082 (gst_missing_decoder_message_new),
12083 (gst_missing_encoder_message_new):
12084 * gst-libs/gst/rtp/gstbasertppayload.c:
12085 * gst-libs/gst/rtp/gstrtcpbuffer.c:
12086 (gst_rtcp_packet_bye_get_reason):
12087 * gst/audioconvert/gstaudioconvert.c:
12088 * gst/audioresample/gstaudioresample.c:
12089 * gst/ffmpegcolorspace/imgconvert.c:
12090 * gst/playback/test.c: (gen_video_element), (gen_audio_element):
12091 * gst/typefind/gsttypefindfunctions.c:
12092 * gst/videoscale/vs_4tap.c:
12093 * gst/videoscale/vs_4tap.h:
12094 * sys/v4l/gstv4lelement.c:
12095 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
12096 * sys/v4l/v4l_calls.c:
12097 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
12098 (gst_v4lsrc_try_capture):
12099 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
12100 (gst_ximagesink_ximage_new):
12101 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
12102 (gst_xvimagesink_xvimage_new):
12103 * tests/check/elements/audioconvert.c:
12104 * tests/check/elements/audioresample.c:
12105 (fail_unless_perfect_stream):
12106 * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
12107 * tests/check/elements/decodebin.c:
12108 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
12109 (setup_gdpdepay_streamheader):
12110 * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
12111 (setup_gdppay_streamheader):
12112 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
12113 * tests/check/elements/multifdsink.c: (setup_multifdsink):
12114 * tests/check/elements/textoverlay.c:
12115 * tests/check/elements/videorate.c: (setup_videorate):
12116 * tests/check/elements/videotestsrc.c: (setup_videotestsrc):
12117 * tests/check/elements/volume.c: (setup_volume):
12118 * tests/check/elements/vorbisdec.c: (setup_vorbisdec):
12119 * tests/check/elements/vorbistag.c:
12120 * tests/check/generic/clock-selection.c:
12121 * tests/check/generic/states.c: (setup), (teardown):
12122 * tests/check/libs/cddabasesrc.c:
12123 * tests/check/libs/video.c:
12124 * tests/check/pipelines/gio.c:
12125 * tests/check/pipelines/oggmux.c:
12126 * tests/check/pipelines/simple-launch-lines.c:
12127 (simple_launch_lines_suite):
12128 * tests/check/pipelines/streamheader.c:
12129 * tests/check/pipelines/theoraenc.c:
12130 * tests/check/pipelines/vorbisdec.c:
12131 * tests/check/pipelines/vorbisenc.c:
12132 * tests/examples/seek/scrubby.c:
12133 * tests/examples/seek/seek.c: (query_positions_elems),
12134 (query_positions_pads):
12135 * tests/icles/stress-xoverlay.c: (myclock):
12136 Correct all relevant warnings found by the sparse semantic code
12137 analyzer. This include marking several symbols static, using
12138 NULL instead of 0 for pointers and using "foo (void)" instead
12139 of "foo ()" for declarations.
12140 * win32/common/libgstrtp.def:
12141 Add gst_rtp_buffer_set_extension_data to the symbol definition file.
12143 2008-03-02 18:43:15 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
12145 gst/playback/gstplaybin2.c: Make the function signature of the _get_*_tags() functions match the signature of the vfu...
12146 Original commit message from CVS:
12147 Patch by: José Alburquerque <jaalburqu svn gnome org>
12148 * gst/playback/gstplaybin2.c:
12149 Make the function signature of the _get_*_tags() functions match
12150 the signature of the vfuncs they implement, ie. return a
12151 GstTagList rather than a GstStructure, which is more correct,
12152 even if one is typedef'ed to the other (#518940).
12154 2008-03-02 18:32:36 +0000 Tim-Philipp Müller <tim@centricular.net>
12156 gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
12157 Original commit message from CVS:
12158 * gst-libs/gst/rtsp/gstrtspconnection.c:
12159 Don't include unix headers unconditionally (fixes #518037).
12161 2008-03-02 18:24:37 +0000 Tim-Philipp Müller <tim@centricular.net>
12163 tests/check/libs/video.c: Add unit test that makes sure that the strides, offsets and sizes returned for the various ...
12164 Original commit message from CVS:
12165 * tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
12166 (fourcc_list_struct), (fourcc_list), (fourcc_get_size),
12167 (paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
12168 (paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
12169 (paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
12170 (paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
12171 (gst_video_format_is_packed), (video_format_is_packed):
12172 Add unit test that makes sure that the strides, offsets and
12173 sizes returned for the various YUV formats by the new video API
12174 match the old reference implementation in videotestsrc.
12176 2008-03-02 18:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
12178 gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
12179 Original commit message from CVS:
12180 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
12181 (gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
12182 (gst_video_format_is_rgb), (gst_video_format_is_yuv),
12183 (gst_video_format_has_alpha), (gst_video_format_get_row_stride),
12184 (gst_video_format_get_pixel_stride),
12185 (gst_video_format_get_component_width),
12186 (gst_video_format_get_component_height),
12187 (gst_video_format_get_component_offset), (gst_video_format_get_size):
12188 * gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
12189 (GST_VIDEO_FORMAT_Y42B):
12190 API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
12192 2008-03-02 18:07:10 +0000 Tim-Philipp Müller <tim@centricular.net>
12194 gst-libs/gst/video/video.c: YV12 is I420 with swapped components 1 and 2, so the offset of component 1 for I420 shoul...
12195 Original commit message from CVS:
12196 * gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
12197 YV12 is I420 with swapped components 1 and 2, so the offset of
12198 component 1 for I420 should be the offset for component 2 for YV12
12201 2008-02-29 21:48:00 +0000 Rene Stadler <mail@renestadler.de>
12203 sys/v4l/gstv4lelement.c: Add missing semicolon to fix indentation.
12204 Original commit message from CVS:
12205 * sys/v4l/gstv4lelement.c:
12206 Add missing semicolon to fix indentation.
12208 2008-02-29 18:44:36 +0000 Julien Moutte <julien@moutte.net>
12210 ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
12211 Original commit message from CVS:
12212 2008-02-29 Julien Moutte <julien@fluendo.com>
12213 * ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
12214 (gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
12216 if we can do SPDIF output.
12217 * ext/alsa/gstalsa.h:
12218 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
12219 (gst_alsasink_prepare), (gst_alsasink_close),
12220 (gst_alsasink_write):
12221 * ext/alsa/gstalsasink.h: Initial support for SPDIF.
12222 * gst-libs/gst/audio/gstringbuffer.c:
12223 (gst_ring_buffer_parse_caps):
12224 * gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
12226 to support AC3, EC3 and IEC958 buffers.
12228 2008-02-29 17:59:16 +0000 Tim-Philipp Müller <tim@centricular.net>
12230 gst-libs/gst/interfaces/mixer.c: De-cruft and fix message type assertions (NULL is not a really valid mixer message t...
12231 Original commit message from CVS:
12232 * gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
12233 (gst_mixer_message_parse_mute_toggled),
12234 (gst_mixer_message_parse_record_toggled),
12235 (gst_mixer_message_parse_volume_changed),
12236 (gst_mixer_message_parse_option_changed):
12237 De-cruft and fix message type assertions (NULL is not a really
12238 valid mixer message type string).
12240 2008-02-29 14:52:02 +0000 Wim Taymans <wim.taymans@gmail.com>
12242 ext/libvisual/visual.c: When negotiating, actually start from a format that we can support instead of from the too ge...
12243 Original commit message from CVS:
12244 * ext/libvisual/visual.c: (gst_vis_src_negotiate):
12245 When negotiating, actually start from a format that we can support
12246 instead of from the too generic template.
12248 2008-02-29 12:26:48 +0000 Wim Taymans <wim.taymans@gmail.com>
12250 gst/playback/gstplaybin2.c: Enable vis setting.
12251 Original commit message from CVS:
12252 * gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
12253 Enable vis setting.
12254 * gst/playback/gstplaysink.c: (gst_play_sink_init),
12255 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
12256 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
12258 Implement vis switching while playing.
12260 2008-02-29 00:04:57 +0000 David Schleef <ds@schleef.org>
12262 gst-libs/gst/riff/riff-media.c: Add Dirac mapping
12263 Original commit message from CVS:
12264 * gst-libs/gst/riff/riff-media.c: Add Dirac mapping
12266 2008-02-28 10:54:14 +0000 Peter Kjellerstedt <pkj@axis.com>
12268 gst/tcp/: Removed fdset and stress test, they are now known as GstPoll in core.
12269 Original commit message from CVS:
12270 Patch by: Peter Kjellerstedt <pkj at axis com>
12271 * gst/tcp/Makefile.am:
12272 * gst/tcp/fdsetstress.c:
12273 * gst/tcp/gstfdset.c:
12274 * gst/tcp/gstfdset.h:
12275 Removed fdset and stress test, they are now known as GstPoll in
12277 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
12278 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
12279 (gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
12280 (gst_multi_fd_sink_handle_client_write),
12281 (gst_multi_fd_sink_queue_buffer),
12282 (gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
12283 (gst_multi_fd_sink_stop):
12284 * gst/tcp/gstmultifdsink.h:
12285 * gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
12286 (gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
12287 (gst_tcp_gdp_read_caps):
12288 * gst/tcp/gsttcp.h:
12289 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
12290 (gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
12291 (gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
12292 * gst/tcp/gsttcpclientsink.h:
12293 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
12294 (gst_tcp_client_src_create), (gst_tcp_client_src_start),
12295 (gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
12296 * gst/tcp/gsttcpclientsrc.h:
12297 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
12298 (gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
12299 * gst/tcp/gsttcpserversink.h:
12300 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
12301 (gst_tcp_server_src_create), (gst_tcp_server_src_start),
12302 (gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
12303 * gst/tcp/gsttcpserversrc.h:
12304 Port to GstPoll. See #505417.
12306 2008-02-28 09:54:14 +0000 Wim Taymans <wim.taymans@gmail.com>
12309 Patch Changelog a bit to give credit and refer to the relevant bug.
12310 Original commit message from CVS:
12311 Patch Changelog a bit to give credit and refer to the
12314 2008-02-28 09:50:52 +0000 Wim Taymans <wim.taymans@gmail.com>
12316 gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection.
12317 Original commit message from CVS:
12318 * gst-libs/gst/rtsp/gstrtspconnection.c:
12319 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
12320 (gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
12321 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
12322 (gst_rtsp_connection_free), (gst_rtsp_connection_poll),
12323 (gst_rtsp_connection_flush):
12324 * gst-libs/gst/rtsp/gstrtspconnection.h:
12325 Use GstPoll for the rtsp connection.
12327 2008-02-27 12:19:31 +0000 Wim Taymans <wim.taymans@gmail.com>
12329 tests/examples/seek/seek.c: Add combo box for visualisations, populate it with a factory list of all visualisation pl...
12330 Original commit message from CVS:
12331 * tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
12332 (init_visualization_features), (vis_combo_cb), (shot_cb), (main):
12333 Add combo box for visualisations, populate it with a factory list
12334 of all visualisation plugins, configure vis plugin instance in
12337 2008-02-27 10:55:03 +0000 Wim Taymans <wim.taymans@gmail.com>
12339 tests/check/libs/rtp.c: Add check for RTP buffer defaults, padding and marker bit API.
12340 Original commit message from CVS:
12341 * tests/check/libs/rtp.c: (GST_START_TEST):
12342 Add check for RTP buffer defaults, padding and marker bit API.
12344 2008-02-27 10:42:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12346 gst-libs/gst/cdda/sha1.c: Use memcpy() instead of upcasting a byte array to long *. This fixes an unaligned memory ac...
12347 Original commit message from CVS:
12348 * gst-libs/gst/cdda/sha1.c: (sha_transform):
12349 Use memcpy() instead of upcasting a byte array to long *. This
12350 fixes an unaligned memory access, resulting in SIGBUS on IA64.
12351 This should be ported to GCheckSum once we can use GLib 2.16.
12352 Partially fixes bug #500833.
12354 2008-02-27 10:23:27 +0000 Tim-Philipp Müller <tim@centricular.net>
12356 gst-libs/gst/tag/gsttagdemux.c: Push tag event after the newsegment event. Log the pointer of the buffer we're actual...
12357 Original commit message from CVS:
12358 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
12359 Push tag event after the newsegment event. Log the pointer of
12360 the buffer we're actually going to push rather than the buffer
12361 we're feeding to _make_metadata_writable().
12363 2008-02-25 07:21:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12365 gst/typefind/gsttypefindfunctions.c: Comment smoke typefinder for now. The smokedec plugin needs one frame per buffer...
12366 Original commit message from CVS:
12367 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
12368 Comment smoke typefinder for now. The smokedec plugin needs one
12369 frame per buffer but we have no parser yet, thus it simply crashes
12370 in most situations.
12372 2008-02-25 06:48:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12374 gst/typefind/gsttypefindfunctions.c: Add typefinder for the smoke video codec. Copied from the jpeg plugin.
12375 Original commit message from CVS:
12376 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
12377 Add typefinder for the smoke video codec. Copied from the jpeg plugin.
12379 2008-02-25 06:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12381 gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
12382 Original commit message from CVS:
12383 * gst/typefind/gsttypefindfunctions.c: (mid_type_find),
12385 Add midi typefinder, copied from the timidity plugin.
12387 2008-02-23 09:51:26 +0000 Tomasz Sałaciński <tsalacinski@gmail.com>
12389 Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
12390 Original commit message from CVS:
12391 Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
12392 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
12393 * tests/check/elements/subparse.c: (test_microdvd_with_italics),
12395 Forward slashes at the beginning and end of a line also signify
12396 italics (Fixes: #518162).
12398 2008-02-22 06:38:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12400 tests/check/gst-plugins-base.supp: Add a suppression for a cached value in GIO that wasn't moved while moving gio fro...
12401 Original commit message from CVS:
12402 * tests/check/gst-plugins-base.supp:
12403 Add a suppression for a cached value in GIO that wasn't moved
12404 while moving gio from -bad to -base.
12406 2008-02-22 05:27:24 +0000 Brian Cameron <brian.cameron@sun.com>
12408 configure.ac: Don't hardcode -Wall and -Werror for configure checks, this fails with non-GCC compilers. Fixes bug #51...
12409 Original commit message from CVS:
12410 Patch by: Brian Cameron <brian dot cameron at sun dot com>
12412 Don't hardcode -Wall and -Werror for configure checks, this fails
12413 with non-GCC compilers. Fixes bug #517991.
12415 2008-02-21 08:05:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12417 gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
12418 Original commit message from CVS:
12419 * gst/audiotestsrc/gstaudiotestsrc.c:
12420 Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
12422 2008-02-20 15:37:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12424 ext/gnomevfs/gstgnomevfssink.c: Return FALSE when seeking for a new segment fails instead of silently ignoring the fa...
12425 Original commit message from CVS:
12426 * ext/gnomevfs/gstgnomevfssink.c:
12427 (gst_gnome_vfs_sink_handle_event):
12428 Return FALSE when seeking for a new segment fails instead
12429 of silently ignoring the failure and appending every buffer
12430 that comes for the new segment.
12432 2008-02-20 11:52:28 +0000 Wim Taymans <wim.taymans@gmail.com>
12434 gst/playback/gstplaysink.c: Recursively search the sink element for a last-frame property so that we can also find th...
12435 Original commit message from CVS:
12436 * gst/playback/gstplaysink.c: (find_property),
12437 (gst_play_sink_find_property), (gen_video_chain),
12438 (gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
12439 Recursively search the sink element for a last-frame property so that we
12440 can also find the property in autovideosink and friends that don't
12441 always proxy the internal sink properties.
12443 2008-02-19 20:42:09 +0000 Tim-Philipp Müller <tim@centricular.net>
12445 gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
12446 Original commit message from CVS:
12447 * gst-libs/gst/audio/multichannel.c:
12448 (GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
12449 (gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
12450 (gst_audio_set_structure_channel_positions_list),
12451 (add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
12452 (gst_audio_fixate_channel_positions):
12453 Fix confusing terminology in docs and code: structure fields are
12454 'fields' and not 'properties'.
12456 2008-02-19 20:36:58 +0000 Tim-Philipp Müller <tim@centricular.net>
12458 gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
12459 Original commit message from CVS:
12460 * gst-libs/gst/audio/multichannel.c:
12461 (gst_audio_check_channel_positions), (add_list_to_struct):
12462 Give more useful warning messages if one of the channel
12463 layout enums passed to us is invalid and if the "channels"
12464 field in the caps has a GType we don't expect.
12466 2008-02-19 20:22:09 +0000 Tim-Philipp Müller <tim@centricular.net>
12468 gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
12469 Original commit message from CVS:
12470 * gst-libs/gst/audio/multichannel.c:
12471 Fix typo in docs blurb.
12473 2008-02-19 16:16:55 +0000 Josep Torra Valles <josep@fluendo.com>
12475 gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS typefind lookup to fix typefinding on HD clips.
12476 Original commit message from CVS:
12477 2008-02-19 Julien Moutte <julien@fluendo.com>
12478 Patch by: Josep Torra Valles <josep@fluendo.com>
12479 * gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
12480 typefind lookup to fix typefinding on HD clips.
12482 2008-02-19 15:50:37 +0000 Tim-Philipp Müller <tim@centricular.net>
12484 gst/playback/gstscreenshot.*: Fix up copyright (I rewrote the GStreamer-0.10 code for this from scratch back in the d...
12485 Original commit message from CVS:
12486 * gst/playback/gstscreenshot.c:
12487 * gst/playback/gstscreenshot.h:
12488 Fix up copyright (I rewrote the GStreamer-0.10 code for
12489 this from scratch back in the days).
12491 2008-02-19 15:02:33 +0000 Wim Taymans <wim.taymans@gmail.com>
12493 gst/playback/: Add screenshot conversion code from totem.
12494 Original commit message from CVS:
12495 * gst/playback/Makefile.am:
12496 * gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
12497 (create_element), (gst_play_frame_conv_convert):
12498 * gst/playback/gstscreenshot.h:
12499 Add screenshot conversion code from totem.
12500 * gst/playback/gstplay-marshal.list:
12501 * gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
12502 (gst_play_bin_class_init), (gst_play_bin_convert_frame),
12503 (gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
12504 Implement frame property to get a color-unconverted snapshot.
12505 Implement convert-frame action signal to get a converted snapshot image.
12506 Configure connection speed in uridecodebin.
12507 Document some more properties.
12508 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
12509 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
12510 (gst_play_sink_get_last_frame):
12511 * gst/playback/gstplaysink.h:
12512 Use last-buffer property of the video sink to get a video snapshot.
12513 * tests/examples/seek/seek.c: (shot_cb), (main):
12514 Add snapshot button for playbin2 and use the frame property to save the
12515 frame as a png in the current directory.
12517 2008-02-19 11:45:56 +0000 Josep Torra Valles <josep@fluendo.com>
12519 gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
12520 Original commit message from CVS:
12521 Patch by: Josep Torra Valles <josep at fluendo dot com>
12522 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
12524 Add typefinding support for h264 elementary streams.
12527 2008-02-18 13:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12529 configure.ac: Require CVS of core for new API in collectpads.
12530 Original commit message from CVS:
12532 Require CVS of core for new API in collectpads.
12533 * gst/adder/gstadder.c:
12534 Use new API to make adder sparse stream aware.
12536 2008-02-18 11:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
12538 gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
12539 Original commit message from CVS:
12540 * gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
12542 Get the object data correct so that we can remove our channels
12544 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
12545 (gen_vis_chain), (gst_play_sink_reconfigure),
12546 (gst_play_sink_request_pad):
12547 Add option to disable async behaviour in the sinks when possible. This
12548 makes it possible to avoid an audio queue when dealing with
12550 Add option to add a queue for the audio path.
12551 * tests/examples/seek/seek.c: (clear_streams), (update_streams),
12553 Disable the vis checkbox to match the defaults of playbin2.
12554 Only get the stream info when we need to.
12556 2008-02-17 05:15:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12558 ext/gio/: Don't use async operations as they require a running main loop.
12559 Original commit message from CVS:
12560 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
12561 (gst_gio_base_sink_set_stream):
12562 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
12563 (gst_gio_base_src_set_stream):
12564 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
12565 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
12566 Don't use async operations as they require a running main loop.
12567 This makes us block again when closing streams and unable
12568 to mount the enclosing volume of an URI if it isn't yet.
12570 2008-02-15 18:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
12572 gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
12573 Original commit message from CVS:
12574 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
12575 (gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
12576 (gen_vis_chain), (gst_play_sink_reconfigure),
12577 (gst_play_sink_request_pad):
12578 Move tee in front of the audio and vis pipelines.
12579 Add queue for audio for now.
12580 Add visualisation support.
12581 * tests/examples/seek/seek.c: (main):
12582 Visualisation is by default disabled.
12584 2008-02-15 11:58:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12586 ext/gio/: Improve debugging a bit.
12587 Original commit message from CVS:
12588 * ext/gio/gstgiobasesink.c: (close_stream_cb):
12589 * ext/gio/gstgiobasesrc.c: (close_stream_cb):
12590 Improve debugging a bit.
12591 * ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
12592 * ext/gio/gstgiosink.h:
12593 * ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
12594 * ext/gio/gstgiosrc.h:
12595 Try to mount the enclosing volume of a GFile if it isn't mounted
12596 yet. This requires us to wait for an async operation to finish, done
12597 with an nested GMainLoop. Authentication is not supported yet, will
12600 2008-02-14 18:24:42 +0000 Wim Taymans <wim.taymans@gmail.com>
12602 gst/playback/: Add mute property.
12603 Original commit message from CVS:
12604 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
12605 (gst_play_bin_set_property), (gst_play_bin_get_property),
12606 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
12607 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
12608 (gst_play_sink_get_mute), (gen_audio_chain):
12609 * gst/playback/gstplaysink.h:
12611 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
12612 (gst_selector_pad_chain):
12613 * gst/playback/gststreamselector.h:
12614 Make sure we forward the event only once.
12615 * tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
12616 Add and implement the mute button for playbin2.
12618 2008-02-13 14:34:55 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
12620 ext/alsa/gstalsasink.c: Add some more debug info.
12621 Original commit message from CVS:
12622 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
12623 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
12624 Add some more debug info.
12625 Make sure we never return a negative delay. Fixes #516246.
12627 2008-02-12 20:09:07 +0000 Tim-Philipp Müller <tim@centricular.net>
12629 ext/alsa/gstalsasink.c: Revert patch that makes the sink hold the object lock when calling snd_pcm_delay(), since it ...
12630 Original commit message from CVS:
12631 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
12632 Revert patch that makes the sink hold the object lock when
12633 calling snd_pcm_delay(), since it breaks playback for me.
12635 2008-02-12 19:50:36 +0000 Julien Moutte <julien@moutte.net>
12637 tests/examples/seek/seek.c: Add some seek flags when changing rate.
12638 Original commit message from CVS:
12639 2008-02-12 Julien Moutte <julien@fluendo.com>
12640 * tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
12641 some seek flags when changing rate.
12643 2008-02-12 14:51:26 +0000 Wim Taymans <wim.taymans@gmail.com>
12645 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
12646 Original commit message from CVS:
12647 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
12648 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
12649 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
12650 Fix potential leaks.
12651 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
12652 Fix leak when there is no function configured.
12654 2008-02-12 11:36:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12656 sys/v4l/v4lsrc_calls.c: Correctly chain up the finalize method.
12657 Original commit message from CVS:
12658 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
12659 (gst_v4lsrc_buffer_finalize):
12660 Correctly chain up the finalize method.
12662 2008-02-12 09:24:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12664 ext/gio/: Add documentation and example code for giostreamsink/giostreamsrc.
12665 Original commit message from CVS:
12666 * ext/gio/gstgiostreamsink.c:
12667 * ext/gio/gstgiostreamsrc.c:
12668 Add documentation and example code for giostreamsink/giostreamsrc.
12669 * tests/check/pipelines/gio.c: (GST_START_TEST):
12670 Ask the GMemoryOutputStream for the data instead of assuming that
12671 the pointer to the data stayed the same. It could've been realloc'ed.
12673 2008-02-12 08:55:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12675 ext/gio/: Make the documentation of giosink/giosrc complete, large parts are based on the gnomevfssink/gnomevfssrc docs.
12676 Original commit message from CVS:
12677 * ext/gio/gstgiosink.c:
12678 * ext/gio/gstgiosrc.c:
12679 Make the documentation of giosink/giosrc complete, large parts
12680 are based on the gnomevfssink/gnomevfssrc docs.
12682 2008-02-12 08:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12684 docs/plugins/: Add the GIO documentation again and while at that run make update.
12685 Original commit message from CVS:
12686 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
12687 * docs/plugins/gst-plugins-base-plugins-sections.txt:
12688 * docs/plugins/gst-plugins-base-plugins.args:
12689 * docs/plugins/gst-plugins-base-plugins.hierarchy:
12690 * docs/plugins/gst-plugins-base-plugins.interfaces:
12691 * docs/plugins/gst-plugins-base-plugins.prerequisites:
12692 * docs/plugins/gst-plugins-base-plugins.signals:
12693 * docs/plugins/inspect/plugin-adder.xml:
12694 * docs/plugins/inspect/plugin-audioconvert.xml:
12695 * docs/plugins/inspect/plugin-audiorate.xml:
12696 * docs/plugins/inspect/plugin-audioresample.xml:
12697 * docs/plugins/inspect/plugin-decodebin.xml:
12698 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
12699 * docs/plugins/inspect/plugin-gdp.xml:
12700 * docs/plugins/inspect/plugin-gio.xml:
12701 * docs/plugins/inspect/plugin-gnomevfs.xml:
12702 * docs/plugins/inspect/plugin-libvisual.xml:
12703 * docs/plugins/inspect/plugin-ogg.xml:
12704 * docs/plugins/inspect/plugin-pango.xml:
12705 * docs/plugins/inspect/plugin-playback.xml:
12706 * docs/plugins/inspect/plugin-queue2.xml:
12707 * docs/plugins/inspect/plugin-subparse.xml:
12708 * docs/plugins/inspect/plugin-theora.xml:
12709 * docs/plugins/inspect/plugin-uridecodebin.xml:
12710 * docs/plugins/inspect/plugin-videorate.xml:
12711 * docs/plugins/inspect/plugin-videoscale.xml:
12712 * docs/plugins/inspect/plugin-volume.xml:
12713 * docs/plugins/inspect/plugin-vorbis.xml:
12714 Add the GIO documentation again and while at that run make update.
12716 2008-02-11 20:23:44 +0000 Tim-Philipp Müller <tim@centricular.net>
12718 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...
12719 Original commit message from CVS:
12720 * ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
12721 * ext/alsa/gstalsasink.c: (set_swparams):
12722 * ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
12723 Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
12724 against libasound >= 1.0.16, since it's been deprecated in
12725 0.10.16, and alignment is always 1 then, apparently. (#512899)
12727 2008-02-11 18:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
12729 gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
12730 Original commit message from CVS:
12731 * gst/playback/gstplaybin.c: (gen_audio_element):
12732 * gst/playback/gstplaysink.c: (gen_audio_chain):
12733 Handle case where we can't create the volume element a bit
12736 2008-02-11 18:02:13 +0000 Tim-Philipp Müller <tim@centricular.net>
12738 ext/gnomevfs/: Add support for https protocol. Fixes #510229.
12739 Original commit message from CVS:
12740 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
12741 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
12742 Add support for https protocol. Fixes #510229.
12744 2008-02-11 17:03:18 +0000 Alan Peevers <peeves@pacbell.net>
12746 ext/alsa/gstalsasink.c: Take appropriate lock when calling alsa methods.
12747 Original commit message from CVS:
12748 2008-02-11 Julien Moutte <julien@fluendo.com>
12749 Patch by: Alan Peevers <peeves@pacbell.net>
12750 * ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
12751 lock when calling alsa methods.
12753 2008-02-11 13:03:13 +0000 Tim-Philipp Müller <tim@centricular.net>
12755 gst/typefind/gsttypefindfunctions.c: Bump rank of jpeg and png typefinders, which will return maximum probability in ...
12756 Original commit message from CVS:
12757 * gst/typefind/gsttypefindfunctions.c:
12758 Bump rank of jpeg and png typefinders, which will return maximum
12759 probability in the most common cases (thus short-circuiting more
12760 expensive typefinders like the mp3 one for these two quite common
12763 2008-02-11 09:48:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12765 ext/theora/theoraparse.c: Fix long description of the theora parser to be more verbose than just the type name.
12766 Original commit message from CVS:
12767 * ext/theora/theoraparse.c:
12768 Fix long description of the theora parser to be more verbose than just
12771 2008-02-11 06:47:50 +0000 Branko Čibej <brane@xbc.nu>
12773 sys/xvimage/xvimagesink.c: Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
12774 Original commit message from CVS:
12775 Patch by: Branko Čibej <brane at xbc dot nu>
12776 * sys/xvimage/xvimagesink.c:
12777 Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
12780 2008-02-09 10:41:36 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
12782 gst/playback/gstplaybasebin.c: Set is_dynamic as True if there are elements with both request and sometimes src pad t...
12783 Original commit message from CVS:
12784 * gst/playback/gstplaybasebin.c:
12785 Set is_dynamic as True if there are elements with both request
12786 and sometimes src pad templates instead of breaking out when it
12787 finds the first pad template that is a src.
12789 2008-02-08 18:17:51 +0000 Wim Taymans <wim.taymans@gmail.com>
12791 tests/examples/seek/seek.c: Add some stream switching and volume gui for playbin2.
12792 Original commit message from CVS:
12793 * tests/examples/seek/seek.c: (stop_cb), (clear_streams),
12794 (update_streams), (video_combo_cb), (audio_combo_cb),
12795 (text_combo_cb), (volume_spinbutton_changed_cb), (main):
12796 Add some stream switching and volume gui for playbin2.
12798 2008-02-08 17:47:37 +0000 Wim Taymans <wim.taymans@gmail.com>
12800 gst/playback/gstplay-marshal.list: Added marshal for streamselector Tags.
12801 Original commit message from CVS:
12802 * gst/playback/gstplay-marshal.list:
12803 Added marshal for streamselector Tags.
12804 * gst/playback/gstplaybasebin.c: (set_active_source):
12805 Streamselector now selects pads based on the pad object instead of its
12807 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
12808 (init_group), (gst_play_bin_init), (get_group), (get_tags),
12809 (gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
12810 (gst_play_bin_get_text_tags),
12811 (gst_play_bin_set_current_video_stream),
12812 (gst_play_bin_set_current_audio_stream),
12813 (gst_play_bin_set_current_text_stream),
12814 (gst_play_bin_set_property), (gst_play_bin_get_property),
12815 (pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
12816 Remove option to mute streams with the current-a/v/t property, we have
12817 this functionality in the flags.
12818 Add signals to notify when the number of A/V/T channels changed.
12819 Add action signals to get tags for the A/V/T streams.
12820 Implement setting the current A/V/T stream.
12821 Rearrange some things to simplify stream selection.
12823 * gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
12824 (gst_play_sink_get_volume), (gst_play_sink_set_property),
12825 (gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
12826 (activate_vis), (gst_play_sink_reconfigure):
12827 * gst/playback/gstplaysink.h:
12828 Add and implement volume setting methods.
12829 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
12830 (gst_selector_pad_finalize), (gst_selector_pad_get_property),
12831 (gst_selector_pad_event), (gst_stream_selector_class_init),
12832 (gst_stream_selector_init), (gst_stream_selector_finalize),
12833 (gst_stream_selector_set_property),
12834 (gst_stream_selector_get_property),
12835 (gst_stream_selector_get_linked_pad),
12836 (gst_stream_selector_request_new_pad):
12837 * gst/playback/gststreamselector.h:
12838 Add pad properties for tags and status of pads.
12840 Make active pad selection based on pad object instead of name.
12842 2008-02-08 16:10:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12844 configure.ac: Revert last change as we now check in gtk-doc.m4 for sed.
12845 Original commit message from CVS:
12847 Revert last change as we now check in gtk-doc.m4 for sed.
12849 2008-02-08 14:54:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12851 configure.ac: Find and subst SED when building the docs.
12852 Original commit message from CVS:
12854 Find and subst SED when building the docs.
12856 2008-02-08 14:34:41 +0000 Julien Moutte <julien@moutte.net>
12858 tests/examples/seek/seek.c: Make sure bus signals are reconnected when pressing STOP and then PLAY again for a parse ...
12859 Original commit message from CVS:
12860 2008-02-08 Julien Moutte <julien@fluendo.com>
12861 * tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
12862 (main): Make sure bus signals are reconnected when pressing STOP
12863 and then PLAY again for a parse launch pipeline. Fix a ref leak
12865 * win32/common/config.h: Updated.
12867 2008-02-08 00:57:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12869 configure.ac: Make DISABLE_DEPRECATED defined *only* during CVS, not during pre-releases or releases.
12870 Original commit message from CVS:
12872 Make DISABLE_DEPRECATED defined *only* during CVS, not during
12873 pre-releases or releases.
12875 2008-02-08 00:45:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12877 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is reporting
12878 Original commit message from CVS:
12880 * ext/gio/Makefile.am:
12881 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
12884 2008-02-07 23:40:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12886 docs/plugins/Makefile.am: Add the headers which need scanning for the GIO plugin. The rest of the docs still need mig...
12887 Original commit message from CVS:
12888 * docs/plugins/Makefile.am:
12889 Add the headers which need scanning for the GIO plugin. The rest of
12890 the docs still need migrating.
12892 2008-02-07 23:22:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12894 Add gio in a few more places.
12895 Original commit message from CVS:
12897 * tests/check/Makefile.am:
12898 * tests/check/pipelines/.cvsignore:
12899 Add gio in a few more places.
12901 2008-02-07 23:18:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12903 Move gio plugin from -bad and mark as experimental.
12904 Original commit message from CVS:
12907 * tests/check/Makefile.am:
12908 Move gio plugin from -bad and mark as experimental.
12910 2008-02-07 22:39:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12912 gst-libs/gst/interfaces/: Comment out a couple of other things which break the build when
12913 Original commit message from CVS:
12914 * gst-libs/gst/interfaces/mixeroptions.c:
12915 * gst-libs/gst/interfaces/mixertrack.c:
12916 Comment out a couple of other things which break the build when
12917 GST_DISABLE_DEPRECATED isn't on but -Werror is.
12919 2008-02-07 18:28:29 +0000 Tim-Philipp Müller <tim@centricular.net>
12921 docs/libs/gst-plugins-base-libs-sections.txt: Fix pbutils header.
12922 Original commit message from CVS:
12923 * docs/libs/gst-plugins-base-libs-sections.txt:
12924 Fix pbutils header.
12926 2008-02-07 18:07:41 +0000 Christian Schaller <uraeus@gnome.org>
12928 * gst-plugins-base.spec.in:
12929 commit spec file update which includes all the split .pc files
12930 Original commit message from CVS:
12931 commit spec file update which includes all the split .pc files
12933 2008-02-07 12:17:49 +0000 Wim Taymans <wim.taymans@gmail.com>
12935 gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
12936 Original commit message from CVS:
12937 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
12938 Fix compiler warning.
12940 2008-02-07 11:00:45 +0000 Peter Kjellerstedt <pkj@axis.com>
12942 gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
12943 Original commit message from CVS:
12944 Patch by: Peter Kjellerstedt <pkj at axis com>
12945 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
12946 Clear the addrinfo struct using memset. Fixes #514937.
12948 2008-02-06 15:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
12950 gst/tcp/gstfdset.h: Remove unused field to same some memory.
12951 Original commit message from CVS:
12952 * gst/tcp/gstfdset.h:
12953 Remove unused field to same some memory.
12954 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
12955 Mark action signals as such.
12957 2008-02-06 13:35:58 +0000 Michael Smith <msmith@xiph.org>
12959 ext/theora/theoradec.c: Increment granulepos for new-bitstream versions appropriately.
12960 Original commit message from CVS:
12961 * ext/theora/theoradec.c: (_theora_granule_frame),
12963 Increment granulepos for new-bitstream versions appropriately.
12966 2008-02-04 11:51:31 +0000 Wim Taymans <wim.taymans@gmail.com>
12968 tests/examples/seek/seek.c: Remove obsolete stream_time reset after flushing seek, core does that automatically now.
12969 Original commit message from CVS:
12970 * tests/examples/seek/seek.c: (do_seek),
12971 (rate_spinbutton_changed_cb), (update_streams), (main):
12972 Remove obsolete stream_time reset after flushing seek, core does that
12974 Improve accuracy of speed spinbutton.
12975 Only do playbin2 stuff when we actually use it.
12977 2008-02-02 17:29:32 +0000 Tim-Philipp Müller <tim@centricular.net>
12979 tests/check/Makefile.am: Revert previous change of the test environment's GST_PLUGIN_PATH.
12980 Original commit message from CVS:
12981 * tests/check/Makefile.am:
12982 Revert previous change of the test environment's GST_PLUGIN_PATH.
12983 The problem is not with the plugins, but with element factories
12984 and only occurs if elements are split out from existing plugins
12985 or if plugins change name (see #512740).
12987 2008-02-02 15:32:23 +0000 Tim-Philipp Müller <tim@centricular.net>
12989 tests/check/Makefile.am: Fix the tests environment's GST_PLUGIN_PATH: we want the directory with the core's plugins f...
12990 Original commit message from CVS:
12991 * tests/check/Makefile.am:
12992 Fix the tests environment's GST_PLUGIN_PATH: we want the directory
12993 with the core's plugins first and our local build directories last,
12994 since we might be building against an installed core, and that
12995 core's plugin directory may contain older or other versions of
12996 our own -base plugins, but we really do want to test our local
12997 ones (if there are multiple plugins or element factories with the
12998 same name, those inspected last will trump those read in earlier).
12999 Fixes #512740 for the most part.
13001 2008-02-02 07:13:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13003 Use gmtime_r if available as gmtime is not MT-safe.
13004 Original commit message from CVS:
13006 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
13007 Use gmtime_r if available as gmtime is not MT-safe.
13010 2008-02-02 06:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13012 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
13013 Original commit message from CVS:
13014 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
13015 Cast glong to time_t as time_t might have a different type on
13016 other platforms, like FreeBSD, and we get a compiler warning
13017 otherwise. Fixes bug #511825.
13019 2008-02-01 16:44:21 +0000 Wim Taymans <wim.taymans@gmail.com>
13021 gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
13022 Original commit message from CVS:
13023 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
13024 (get_group), (get_n_pads), (gst_play_bin_get_property),
13025 (pad_added_cb), (no_more_pads_cb), (perform_eos),
13026 (autoplug_select_cb), (deactivate_group):
13027 Remove stream-info, we going for something easier.
13028 Refactor getting the current group.
13029 Implement getting the number of audio/video/text streams.
13030 * gst/playback/gststreamselector.c:
13031 (gst_stream_selector_class_init), (gst_stream_selector_init),
13032 (gst_stream_selector_get_property),
13033 (gst_stream_selector_request_new_pad),
13034 (gst_stream_selector_release_pad):
13035 * gst/playback/gststreamselector.h:
13036 Add property for number of pads.
13037 * tests/examples/seek/seek.c: (set_scale), (update_flag),
13038 (vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
13039 (text_toggle_cb), (update_streams), (msg_async_done),
13040 (msg_state_changed), (main):
13041 Block slider callback when updating the slider position.
13042 Add gui elements for controlling playbin2.
13043 Add callback for async_done that updates position/duration.
13045 2008-02-01 12:56:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13047 docs/plugins/: First round of plugin docs cleansups.
13048 Original commit message from CVS:
13049 * docs/plugins/Makefile.am:
13050 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
13051 * docs/plugins/gst-plugins-base-plugins-sections.txt:
13052 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13053 * docs/plugins/gst-plugins-base-plugins.interfaces:
13054 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13055 First round of plugin docs cleansups.
13056 * docs/plugins/inspect/plugin-adder.xml:
13057 * docs/plugins/inspect/plugin-alsa.xml:
13058 * docs/plugins/inspect/plugin-audioconvert.xml:
13059 * docs/plugins/inspect/plugin-audiorate.xml:
13060 * docs/plugins/inspect/plugin-audioresample.xml:
13061 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13062 * docs/plugins/inspect/plugin-cdparanoia.xml:
13063 * docs/plugins/inspect/plugin-decodebin.xml:
13064 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13065 * docs/plugins/inspect/plugin-gdp.xml:
13066 * docs/plugins/inspect/plugin-gnomevfs.xml:
13067 * docs/plugins/inspect/plugin-libvisual.xml:
13068 * docs/plugins/inspect/plugin-ogg.xml:
13069 * docs/plugins/inspect/plugin-pango.xml:
13070 * docs/plugins/inspect/plugin-subparse.xml:
13071 * docs/plugins/inspect/plugin-tcp.xml:
13072 * docs/plugins/inspect/plugin-theora.xml:
13073 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13074 * docs/plugins/inspect/plugin-video4linux.xml:
13075 * docs/plugins/inspect/plugin-videorate.xml:
13076 * docs/plugins/inspect/plugin-videoscale.xml:
13077 * docs/plugins/inspect/plugin-videotestsrc.xml:
13078 * docs/plugins/inspect/plugin-volume.xml:
13079 * docs/plugins/inspect/plugin-vorbis.xml:
13080 * docs/plugins/inspect/plugin-ximagesink.xml:
13081 * docs/plugins/inspect/plugin-xvimagesink.xml:
13083 * ext/ogg/Makefile.am:
13084 * ext/ogg/gstoggmux.c:
13085 * ext/ogg/gstoggmux.h:
13086 Add header for oggmux. the c-file needs a doc blob still.
13088 2008-02-01 11:09:16 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13090 Add gst_rtp_buffer_set_extension_data()
13091 Original commit message from CVS:
13092 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
13093 * gst-libs/gst/rtp/gstrtpbuffer.c:
13094 (gst_rtp_buffer_set_extension_data):
13095 * gst-libs/gst/rtp/gstrtpbuffer.h:
13096 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
13097 Add gst_rtp_buffer_set_extension_data()
13098 Add a unit test for this addition. Fixes #511478.
13099 API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
13101 2008-01-31 17:18:46 +0000 Wim Taymans <wim.taymans@gmail.com>
13103 gst-libs/gst/app/gstappsink.c: Really clean up the queue instead of just unreffing all buffers in it.
13104 Original commit message from CVS:
13105 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
13106 Really clean up the queue instead of just unreffing all buffers
13108 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
13109 (gst_app_src_class_init), (gst_app_src_init),
13110 (gst_app_src_dispose), (gst_app_src_finalize):
13111 Fix dispose/finalize.
13113 2008-01-30 15:34:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13115 ext/gio/: Use async variants of the close stream functions to prevent blocking for a long time there and add some mor...
13116 Original commit message from CVS:
13117 * ext/gio/gstgiobasesink.c: (close_stream_cb),
13118 (gst_gio_base_sink_stop), (gst_gio_base_sink_event),
13119 (gst_gio_base_sink_render), (gst_gio_base_sink_set_stream):
13120 * ext/gio/gstgiobasesrc.c: (close_stream_cb),
13121 (gst_gio_base_src_stop), (gst_gio_base_src_create),
13122 (gst_gio_base_src_set_stream):
13123 Use async variants of the close stream functions to prevent blocking
13124 for a long time there and add some more sanity checks for a correct
13127 2008-01-30 14:42:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13129 configure.ac: Back to CVS
13130 Original commit message from CVS:
13134 === release 0.10.17 ===
13136 2008-01-30 14:19:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13142 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13143 * docs/plugins/inspect/plugin-adder.xml:
13144 * docs/plugins/inspect/plugin-alsa.xml:
13145 * docs/plugins/inspect/plugin-audioconvert.xml:
13146 * docs/plugins/inspect/plugin-audiorate.xml:
13147 * docs/plugins/inspect/plugin-audioresample.xml:
13148 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13149 * docs/plugins/inspect/plugin-cdparanoia.xml:
13150 * docs/plugins/inspect/plugin-decodebin.xml:
13151 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13152 * docs/plugins/inspect/plugin-gdp.xml:
13153 * docs/plugins/inspect/plugin-gnomevfs.xml:
13154 * docs/plugins/inspect/plugin-libvisual.xml:
13155 * docs/plugins/inspect/plugin-ogg.xml:
13156 * docs/plugins/inspect/plugin-pango.xml:
13157 * docs/plugins/inspect/plugin-subparse.xml:
13158 * docs/plugins/inspect/plugin-tcp.xml:
13159 * docs/plugins/inspect/plugin-theora.xml:
13160 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13161 * docs/plugins/inspect/plugin-video4linux.xml:
13162 * docs/plugins/inspect/plugin-videorate.xml:
13163 * docs/plugins/inspect/plugin-videoscale.xml:
13164 * docs/plugins/inspect/plugin-videotestsrc.xml:
13165 * docs/plugins/inspect/plugin-volume.xml:
13166 * docs/plugins/inspect/plugin-vorbis.xml:
13167 * docs/plugins/inspect/plugin-ximagesink.xml:
13168 * docs/plugins/inspect/plugin-xvimagesink.xml:
13169 * gst-plugins-base.doap:
13170 * win32/common/config.h:
13172 Original commit message from CVS:
13175 2008-01-30 13:45:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13177 gst-libs/gst/interfaces/: Also remove the conditional registration of the signals that disappeared with the ABI chang...
13178 Original commit message from CVS:
13179 * gst-libs/gst/interfaces/mixeroptions.c:
13180 * gst-libs/gst/interfaces/mixertrack.c:
13181 Also remove the conditional registration of the signals
13182 that disappeared with the ABI change in 0.10.14
13184 2008-01-30 12:28:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13186 gst-libs/gst/rtsp/gstrtspconnection.c: Revert patch to gstrtspconnection.c for brown paper bag release of -base. Re-o...
13187 Original commit message from CVS:
13188 * gst-libs/gst/rtsp/gstrtspconnection.c:
13189 Revert patch to gstrtspconnection.c for brown paper bag
13190 release of -base. Re-opens: #511825
13192 2008-01-30 12:20:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13194 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
13195 Original commit message from CVS:
13196 * gst-libs/gst/interfaces/mixeroptions.h:
13197 * gst-libs/gst/interfaces/mixertrack.h:
13198 Change the way these deprecated function pointers are removed
13199 so that the compiled ABI is unconditionally smaller. This
13200 sets in stone an ABI break that actually occurred when the
13201 things were deprecated in 0.10.14, which seems to be the best
13202 fix as the only known users are oss-mixer and sunaudio-mixer in
13206 2008-01-30 12:19:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13208 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
13209 Original commit message from CVS:
13210 * gst-libs/gst/interfaces/mixeroptions.h:
13211 * gst-libs/gst/interfaces/mixertrack.h:
13212 Change the way these deprecated function pointers are removed
13213 so that the compiled ABI is unconditionally smaller. This
13214 sets in stone an ABI break that actually occurred when the
13215 things were deprecated in 0.10.14, which seems to be the best
13216 fix as the only known users are oss-mixer and sunaudio-mixer in
13219 2008-01-30 11:43:53 +0000 Tim-Philipp Müller <tim@centricular.net>
13221 win32/common/libgstpbutils.def: Export the two new _get_type() functions which are needed by the python bindings.
13222 Original commit message from CVS:
13223 * win32/common/libgstpbutils.def:
13224 Export the two new _get_type() functions which are needed
13225 by the python bindings.
13227 2008-01-29 09:59:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13229 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
13230 Original commit message from CVS:
13231 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
13232 Cast glong to time_t as time_t might have a different type on
13233 other platforms, like FreeBSD, and we get a compiler warning
13234 otherwise. Fixes bug #511825.
13236 2008-01-29 09:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13238 gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
13239 Original commit message from CVS:
13240 * gst-libs/gst/audio/gstaudiofilter.c:
13241 (gst_audio_filter_class_init):
13242 Initialize the GstRingerBuffer class to get it's debug category
13243 initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
13244 category and otherwise we get some g_critical(). Fixes bug #512334.
13246 2008-01-28 23:35:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13248 configure.ac: Back to CVS
13249 Original commit message from CVS:
13253 === release 0.10.16 ===
13255 2008-01-28 23:31:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13261 * docs/plugins/gst-plugins-base-plugins.args:
13262 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13263 * docs/plugins/gst-plugins-base-plugins.interfaces:
13264 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13265 * docs/plugins/gst-plugins-base-plugins.signals:
13266 * docs/plugins/inspect/plugin-adder.xml:
13267 * docs/plugins/inspect/plugin-alsa.xml:
13268 * docs/plugins/inspect/plugin-audioconvert.xml:
13269 * docs/plugins/inspect/plugin-audiorate.xml:
13270 * docs/plugins/inspect/plugin-audioresample.xml:
13271 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13272 * docs/plugins/inspect/plugin-cdparanoia.xml:
13273 * docs/plugins/inspect/plugin-decodebin.xml:
13274 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13275 * docs/plugins/inspect/plugin-gdp.xml:
13276 * docs/plugins/inspect/plugin-gnomevfs.xml:
13277 * docs/plugins/inspect/plugin-libvisual.xml:
13278 * docs/plugins/inspect/plugin-ogg.xml:
13279 * docs/plugins/inspect/plugin-pango.xml:
13280 * docs/plugins/inspect/plugin-subparse.xml:
13281 * docs/plugins/inspect/plugin-tcp.xml:
13282 * docs/plugins/inspect/plugin-theora.xml:
13283 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13284 * docs/plugins/inspect/plugin-video4linux.xml:
13285 * docs/plugins/inspect/plugin-videorate.xml:
13286 * docs/plugins/inspect/plugin-videoscale.xml:
13287 * docs/plugins/inspect/plugin-videotestsrc.xml:
13288 * docs/plugins/inspect/plugin-volume.xml:
13289 * docs/plugins/inspect/plugin-vorbis.xml:
13290 * docs/plugins/inspect/plugin-ximagesink.xml:
13291 * docs/plugins/inspect/plugin-xvimagesink.xml:
13292 * gst-plugins-base.doap:
13293 * win32/common/config.h:
13295 Original commit message from CVS:
13298 2008-01-28 22:15:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13324 Original commit message from CVS:
13327 2008-01-22 15:37:49 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13329 gst-libs/gst/rtp/gstrtpbuffer.c: Fix typos and wrong extension check. Fixes #511274.
13330 Original commit message from CVS:
13331 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
13332 * gst-libs/gst/rtp/gstrtpbuffer.c:
13333 (gst_rtp_buffer_get_extension_data):
13334 Fix typos and wrong extension check. Fixes #511274.
13336 2008-01-18 00:03:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13338 po/sk.po: Oops - add new sk.po mentioned in the LINGUAS I just committed
13339 Original commit message from CVS:
13341 Oops - add new sk.po mentioned in the LINGUAS I just committed
13343 2008-01-17 22:31:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13345 po/LINGUAS: Add ca translation to the disted list.
13346 Original commit message from CVS:
13348 Add ca translation to the disted list.
13349 * win32/vs6/libgstsdp.dsp:
13350 Convert line endings to CRLF
13352 2008-01-17 21:58:53 +0000 Sébastien Moutte <sebastien@moutte.net>
13354 win32/MANIFEST: Add win32/vs6/libgstrtsp.dsp to MANIFEST
13355 Original commit message from CVS:
13357 Add win32/vs6/libgstrtsp.dsp to MANIFEST
13359 2008-01-16 05:40:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13361 Update for API changes in GIO and require GIO 2.15.2 for this.
13362 Original commit message from CVS:
13364 * tests/check/pipelines/gio.c: (GST_START_TEST):
13365 Update for API changes in GIO and require GIO 2.15.2 for this.
13367 2008-01-14 22:20:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13369 win32/common/: Add new API declarations
13370 Original commit message from CVS:
13371 * win32/common/libgstsdp.def:
13372 * win32/common/libgstvideo.def:
13373 Add new API declarations
13375 2008-01-14 17:00:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13377 ext/theora/: Take a 2nd stab at handling libtheora granulepos changes in the decoder and parser by inspecting the bit...
13378 Original commit message from CVS:
13379 * ext/theora/gsttheoradec.h:
13380 * ext/theora/gsttheoraparse.h:
13381 * ext/theora/theoradec.c:
13382 * ext/theora/theoraparse.c:
13383 Take a 2nd stab at handling libtheora granulepos changes in the decoder
13384 and parser by inspecting the bitstream version of the incoming data.
13386 2008-01-14 13:11:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13388 Provide one pkg-config file for every gst-plugins-base library.
13389 Original commit message from CVS:
13391 * pkgconfig/Makefile.am:
13392 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
13393 * pkgconfig/gstreamer-audio.pc.in:
13394 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
13395 * pkgconfig/gstreamer-cdda.pc.in:
13396 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
13397 * pkgconfig/gstreamer-fft.pc.in:
13398 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
13399 * pkgconfig/gstreamer-floatcast.pc.in:
13400 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
13401 * pkgconfig/gstreamer-interfaces.pc.in:
13402 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
13403 * pkgconfig/gstreamer-netbuffer.pc.in:
13404 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
13405 * pkgconfig/gstreamer-pbutils.pc.in:
13406 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
13407 * pkgconfig/gstreamer-riff.pc.in:
13408 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
13409 * pkgconfig/gstreamer-rtp.pc.in:
13410 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
13411 * pkgconfig/gstreamer-rtsp.pc.in:
13412 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
13413 * pkgconfig/gstreamer-sdp.pc.in:
13414 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
13415 * pkgconfig/gstreamer-tag.pc.in:
13416 * pkgconfig/gstreamer-video-uninstalled.pc.in:
13417 * pkgconfig/gstreamer-video.pc.in:
13418 Provide one pkg-config file for every gst-plugins-base library.
13419 This makes linking to those libraries much more intuitive and
13420 provides standard pkg-config behaviour for them. Fixes bug #499697.
13422 2008-01-14 01:19:34 +0000 David Schleef <ds@schleef.org>
13424 gst/videoscale/vs_4tap.c: Fix valgrind error on 4tap scaling method.
13425 Original commit message from CVS:
13426 * gst/videoscale/vs_4tap.c:
13427 Fix valgrind error on 4tap scaling method.
13429 2008-01-13 21:40:45 +0000 Sébastien Moutte <sebastien@moutte.net>
13431 gst-libs/gst/sdp/gstsdpmessage.c: Include Winsock2.h for VS6 and use a different way initialize hints structure so it...
13432 Original commit message from CVS:
13433 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
13434 Include Winsock2.h for VS6 and use a different way initialize
13435 hints structure so it can build with VS6.
13437 * win32/vs6/libgstsdp.dsp:
13438 * win32/common/libgstsdp.def:
13439 Add new files for libgstsdp.
13440 * win32/vs6/grammar.dsp:
13441 Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
13442 * win32/vs6/gst_plugins_base.dsw:
13443 * win32/vs6/libgstdecodebin.dsp:
13444 * win32/vs6/libgstdecodebin2.dsp:
13445 * win32/vs6/libgstplaybin.dsp:
13446 * win32/vs6/libgstvolume.dsp:
13447 Add new dependencies to the link list.
13449 2008-01-13 17:24:49 +0000 Julien Moutte <julien@moutte.net>
13451 win32/common/: Update/Add generated files in the win32 build directory.
13452 Original commit message from CVS:
13453 2008-01-13 Julien Moutte <julien@fluendo.com>
13454 * win32/common/config.h:
13455 * win32/common/gstrtsp-enumtypes.c: (gst_rtsp_result_get_type),
13456 (gst_rtsp_event_get_type), (gst_rtsp_family_get_type),
13457 (gst_rtsp_state_get_type), (gst_rtsp_version_get_type),
13458 (gst_rtsp_method_get_type), (gst_rtsp_auth_method_get_type),
13459 (gst_rtsp_header_field_get_type),
13460 (gst_rtsp_status_code_get_type):
13461 * win32/common/interfaces-enumtypes.c:
13462 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
13463 (gst_mixer_message_type_get_type), (gst_mixer_flags_get_type),
13464 (gst_mixer_track_flags_get_type),
13465 (gst_tuner_channel_flags_get_type):
13466 * win32/common/multichannel-enumtypes.c:
13467 (gst_audio_channel_position_get_type):
13468 * win32/common/pbutils-enumtypes.c:
13469 (gst_install_plugins_return_get_type):
13470 * win32/common/pbutils-enumtypes.h: Update/Add generated files
13471 in the win32 build directory.
13473 2008-01-12 23:24:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13475 tests/check/Makefile.am: Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
13476 Original commit message from CVS:
13477 * tests/check/Makefile.am:
13478 Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
13479 * tests/check/elements/audiorate.c: (do_perfect_stream_test):
13480 * tests/check/elements/playbin.c:
13481 * tests/check/libs/mixer.c: (test_element_interface_supported),
13482 (gst_implements_interface_init):
13483 * tests/check/libs/rtp.c: (GST_START_TEST):
13484 Fix various assignment type mismatches.
13486 2008-01-12 23:08:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13488 Add test to see if hstrerror is available or if we need libresolv (Solaris) for it, then use it in libgstrtsp.
13489 Original commit message from CVS:
13491 * gst-libs/gst/rtsp/Makefile.am:
13492 Add test to see if hstrerror is available or if we need libresolv
13493 (Solaris) for it, then use it in libgstrtsp.
13495 2008-01-12 14:54:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13497 gst-libs/gst/tag/Makefile.am: Fix include path order
13498 Original commit message from CVS:
13499 * gst-libs/gst/tag/Makefile.am:
13500 Fix include path order
13502 2008-01-11 17:15:23 +0000 Tim-Philipp Müller <tim@centricular.net>
13504 * gst-libs/gst/pbutils/.gitignore:
13505 Ignore more and make buildbot happy
13506 Original commit message from CVS:
13507 Ignore more and make buildbot happy
13509 2008-01-11 16:18:10 +0000 Edward Hervey <bilboed@bilboed.com>
13511 gst-libs/gst/pbutils/install-plugins.*: Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping for bi...
13512 Original commit message from CVS:
13513 * gst-libs/gst/pbutils/install-plugins.c:
13514 (gst_install_plugins_context_copy),
13515 (gst_install_plugins_context_get_type):
13516 * gst-libs/gst/pbutils/install-plugins.h:
13517 Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
13520 2008-01-11 15:48:11 +0000 Michael Smith <msmith@xiph.org>
13522 ext/theora/theoradec.c: Adapt for post-alpha meaning of granulepos, when we have a newer version of libtheora.
13523 Original commit message from CVS:
13524 * ext/theora/theoradec.c: (gst_theora_dec_class_init),
13525 (_theora_granule_frame), (_theora_granule_start_time),
13526 (theora_dec_sink_convert), (theora_dec_decode_buffer):
13527 Adapt for post-alpha meaning of granulepos, when we
13528 have a newer version of libtheora.
13529 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
13530 (theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
13531 (theora_enc_is_discontinuous), (theora_enc_chain):
13533 * tests/check/Makefile.am:
13534 Link libtheora into theoraenc test so we can check which version of
13535 libtheora we're testing against.
13536 * tests/check/pipelines/theoraenc.c: (check_libtheora),
13537 (check_buffer_granulepos),
13538 (check_buffer_granulepos_from_starttime), (GST_START_TEST),
13540 Adapt tests to check the values that are now defined for theora; make
13541 the tests backwards-adapt the passed values if we're running against an
13545 2008-01-10 17:55:53 +0000 Tim-Philipp Müller <tim@centricular.net>
13547 gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
13548 Original commit message from CVS:
13549 * gst-libs/gst/audio/gstbaseaudiosink.c:
13550 (gst_base_audio_sink_class_init):
13551 * gst-libs/gst/audio/gstbaseaudiosrc.c:
13552 (gst_base_audio_src_class_init):
13553 Ref audio clock class from a thread-safe context to make sure
13554 we're not bit by GObjects lack of thread-safety here (#349410),
13555 however unlikely that may be in practice.
13557 2008-01-10 12:22:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13559 autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We...
13560 Original commit message from CVS:
13562 Add -Wno-portability to the automake parameters to stop warnings
13563 about GNU make extensions being used. We require GNU make in almost
13564 every Makefile anyway.
13566 Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
13567 at the same time is required for per target flags.
13569 2008-01-08 21:10:02 +0000 Tim-Philipp Müller <tim@centricular.net>
13571 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...
13572 Original commit message from CVS:
13573 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
13574 Post an error message if we can't pull as many bytes as we need
13575 for the tag. This makes sure the user gets to see a proper error
13576 message if a file with a partial ID3 tag is fed to decodebin, and
13577 not a 'no ID3 tag demuxer' error, which would be confusing
13580 2008-01-08 20:59:20 +0000 Tim-Philipp Müller <tim@centricular.net>
13582 gst-libs/gst/pbutils/descriptions.c: Add description strings for ID3, APE, and ICY tags.
13583 Original commit message from CVS:
13584 * gst-libs/gst/pbutils/descriptions.c: (formats):
13585 Add description strings for ID3, APE, and ICY tags.
13587 2008-01-08 20:48:00 +0000 Tim-Philipp Müller <tim@centricular.net>
13589 gst/playback/gstdecodebin.c: Make sure we error out correctly if we can't activate one of the elements we've added. ...
13590 Original commit message from CVS:
13591 * gst/playback/gstdecodebin.c: (try_to_link_1):
13592 Make sure we error out correctly if we can't activate one of
13593 the elements we've added. Fixes #508138.
13595 2008-01-07 13:59:43 +0000 Bastien Nocera <hadess@hadess.net>
13597 ext/alsa/gstalsamixer.c: Use snd_mixer_selem_set_{playback|capture}_volume_all() if the volume is the same for all ch...
13598 Original commit message from CVS:
13599 Patch by: Bastien Nocera <hadess at hadess net>
13600 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
13601 (check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
13602 Use snd_mixer_selem_set_{playback|capture}_volume_all() if
13603 the volume is the same for all channels. This works around
13604 some problem in alsa that leaves us with inconsistent state
13605 for some reason (#486840).
13607 2008-01-07 13:19:50 +0000 Jerone Young <jerone@gmail.com>
13609 ext/alsa/gstalsamixer.c: If there's no mixer track by the name of 'Master' or 'Front', check if there's one called 'P...
13610 Original commit message from CVS:
13611 Patch by: Jerone Young <jerone at gmail com>
13612 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
13613 If there's no mixer track by the name of 'Master' or 'Front',
13614 check if there's one called 'PCM' before trying the generic
13615 fallback logic (fixes #506928, where we pick 'Mic' as master
13616 track for the AD1984 card in a Thinkpad T61/X61 laptop).
13618 2008-01-07 11:40:04 +0000 Wim Taymans <wim.taymans@gmail.com>
13620 gst/playback/gstplay-enum.*: Add enums for configuration flags.
13621 Original commit message from CVS:
13622 * gst/playback/gstplay-enum.c:
13623 (register_gst_autoplug_select_result),
13624 (gst_autoplug_select_result_get_type), (register_gst_play_flags),
13625 (gst_play_flags_get_type):
13626 * gst/playback/gstplay-enum.h:
13627 Add enums for configuration flags.
13628 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
13629 (init_group), (gst_play_bin_init), (gst_play_bin_set_property),
13630 (gst_play_bin_get_property), (no_more_pads_cb),
13631 (autoplug_select_cb), (gst_play_bin_change_state):
13632 Merge mode with flags.
13633 Add more property getters/setters, defaults and docs.
13634 Add properties to get number of audio/video/text streams.
13635 Create sink object in _init so that we can always rely on it being
13637 * gst/playback/gstplaysink.c: (gst_play_sink_init),
13638 (gen_video_chain), (gen_audio_chain), (gen_vis_chain),
13639 (activate_vis), (gst_play_sink_reconfigure),
13640 (gst_play_sink_set_flags), (gst_play_sink_get_flags),
13641 (gst_play_sink_change_state):
13642 * gst/playback/gstplaysink.h:
13643 Use flags to configure the sink pipelines.
13644 Add tee before audio pipeline so that we can use it for visualisations.
13645 Start working on integrating visualisations.
13646 Remove mode, we can do everything with the flags now.
13647 Add method to configue the sink pipeline.
13649 2008-01-06 16:36:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13651 Update to GMemoryInputStream API changes in GLib SVN and require gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
13652 Original commit message from CVS:
13654 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
13655 * tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
13656 Update to GMemoryInputStream API changes in GLib SVN and require
13657 gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
13658 We can also report the duration for every GSeekable, not only
13659 GFileInputStream and GMemoryInputStream.
13661 2008-01-06 14:39:19 +0000 Tim-Philipp Müller <tim@centricular.net>
13663 tests/check/pipelines/theoraenc.c: Turn these functions into macros so we can see right away where the failure occured.
13664 Original commit message from CVS:
13665 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
13666 (check_buffer_timestamp), (check_buffer_duration):
13667 Turn these functions into macros so we can see right away
13668 where the failure occured.
13670 2008-01-05 22:25:05 +0000 Julien Moutte <julien@moutte.net>
13672 sys/xvimage/xvimagesink.c: Add debugging information to understand how X calculates the stride for XvImages.
13673 Original commit message from CVS:
13674 2008-01-05 Julien Moutte <julien@fluendo.com>
13675 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
13676 debugging information to understand how X calculates the stride
13679 2008-01-03 20:33:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13681 gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
13682 Original commit message from CVS:
13683 * gst/volume/Makefile.am:
13684 * gst/volume/gstvolume.c: (volume_choose_func),
13685 (gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
13687 * gst/volume/gstvolume.h:
13688 Use GstAudioFilter as base class for the volume element instead of
13689 plain GstBaseTransform.
13691 2008-01-03 07:17:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13693 gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
13694 Original commit message from CVS:
13695 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
13696 Don't set element details for the abstract GstAudioFilter class.
13698 2008-01-02 12:09:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13700 gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
13701 Original commit message from CVS:
13702 * gst-libs/gst/audio/gstaudiofilter.c:
13703 (gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
13704 Implement get_unit_size() vmethod of GstBaseTransform.
13706 2008-01-01 12:53:48 +0000 Edward Hervey <bilboed@bilboed.com>
13708 gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
13709 Original commit message from CVS:
13710 * gst-libs/gst/pbutils/Makefile.am:
13711 * gst-libs/gst/pbutils/pbutils.h:
13712 Use glib-enum generator to have a proper enum GType for
13713 GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
13715 2008-01-01 01:21:47 +0000 David Schleef <ds@schleef.org>
13717 tests/check/: Reenable theoraenc test, which fails on the buildbot but not locally.
13718 Original commit message from CVS:
13719 * tests/check/Makefile.am:
13720 * tests/check/pipelines/theoraenc.c:
13721 Reenable theoraenc test, which fails on the buildbot but
13724 2007-12-31 21:31:01 +0000 David Schleef <ds@schleef.org>
13726 docs/: Add *-undeclared.txt to fix buildbot.
13727 Original commit message from CVS:
13728 * docs/libs/.cvsignore:
13729 * docs/plugins/.cvsignore:
13730 Add *-undeclared.txt to fix buildbot.
13732 2007-12-31 20:45:28 +0000 David Schleef <ds@schleef.org>
13734 tests/check/Makefile.am: Second attempt at disabling theoraenc test long enough to get buildbot to compile -base.
13735 Original commit message from CVS:
13736 * tests/check/Makefile.am:
13737 Second attempt at disabling theoraenc test long enough to
13738 get buildbot to compile -base.
13740 2007-12-31 20:21:20 +0000 David Schleef <ds@schleef.org>
13742 tests/check/pipelines/theoraenc.c: Disable theoraenc test long enough to get the buildbot to compile a recent -base.
13743 Original commit message from CVS:
13744 * tests/check/pipelines/theoraenc.c:
13745 Disable theoraenc test long enough to get the buildbot to
13746 compile a recent -base.
13748 2007-12-31 13:17:29 +0000 Wim Taymans <wim.taymans@gmail.com>
13750 tests/examples/seek/seek.c: Make sure we reset the slider value to 0.0 without racing against a possible g_idle that ...
13751 Original commit message from CVS:
13752 * tests/examples/seek/seek.c: (stop_cb):
13753 Make sure we reset the slider value to 0.0 without racing against a
13754 possible g_idle that sets it to something else.
13756 2007-12-31 00:32:53 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13758 sys/ximage/ximagesink.c: fix typo
13759 Original commit message from CVS:
13760 * sys/ximage/ximagesink.c:
13763 2007-12-30 19:21:16 +0000 Wim Taymans <wim.taymans@gmail.com>
13765 gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
13766 Original commit message from CVS:
13767 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
13768 * gst-libs/gst/rtsp/gstrtspdefs.h:
13769 Add Location header so that we can start implementing redirects.
13772 2007-12-29 20:55:39 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13774 gst/subparse/gstssaparse.c: combine if's
13775 Original commit message from CVS:
13776 * gst/subparse/gstssaparse.c:
13779 2007-12-29 19:23:59 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13781 gst/subparse/gstssaparse.c: remove duplicate log message
13782 Original commit message from CVS:
13783 * gst/subparse/gstssaparse.c:
13784 remove duplicate log message
13786 2007-12-29 17:29:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13788 Update to latest API changes in GLib/GIO and require at least gio-2.0 2.15.0 for this.
13789 Original commit message from CVS:
13791 * ext/gio/gstgio.c:
13792 * ext/gio/gstgio.h:
13793 * ext/gio/gstgiobasesink.h:
13794 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
13795 * ext/gio/gstgiobasesrc.h:
13796 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
13797 * ext/gio/gstgiosink.h:
13798 * ext/gio/gstgiosrc.h:
13799 * ext/gio/gstgiostreamsink.h:
13800 * ext/gio/gstgiostreamsrc.h:
13801 * tests/check/pipelines/gio.c:
13802 Update to latest API changes in GLib/GIO and require at least
13803 gio-2.0 2.15.0 for this.
13804 * ext/gio/Makefile.am:
13805 Add GST_PLUGIN_LDFLAGS to LDFLAGS.
13807 2007-12-29 16:23:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13809 ext/libvisual/visual.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached()...
13810 Original commit message from CVS:
13811 * ext/libvisual/visual.c: (gst_visual_chain):
13812 Fix 'xyz may be used uninitialized' compiler warnings caused
13813 by broken g_assert_not_reached() macro in GLib-2.15.x and don't
13814 abort() in any case but properly report the error.
13816 2007-12-28 09:00:27 +0000 Wim Taymans <wim.taymans@gmail.com>
13818 gst/playback/gstplaybin2.c: Code cleanups.
13819 Original commit message from CVS:
13820 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
13821 (gst_play_bin_finalize), (gst_play_bin_set_uri),
13822 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
13823 (gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
13824 (autoplug_select_cb), (activate_group), (deactivate_group),
13825 (setup_next_source), (save_current_group),
13826 (gst_play_bin_change_state):
13828 Remove next-uri, we can use the uri property just fine.
13830 Unref uridecodebin when switching.
13831 Fix going to READY.
13832 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
13833 (gst_play_sink_init), (gst_play_sink_dispose),
13834 (gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
13835 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
13836 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
13837 (gst_play_sink_set_property), (gst_play_sink_get_property),
13838 (gen_video_chain), (gen_text_element), (gen_audio_chain),
13839 (gen_vis_element), (gst_play_sink_get_mode),
13840 (gst_play_sink_set_mode), (gst_play_sink_set_flags),
13841 (gst_play_sink_get_flags), (gst_play_sink_request_pad),
13842 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
13843 (gst_play_sink_change_state):
13844 * gst/playback/gstplaysink.h:
13845 Add some locking to make things threadsafe.
13846 * gst/playback/test7.c: (about_to_finish_cb):
13849 2007-12-22 12:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
13851 gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
13852 Original commit message from CVS:
13853 * gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
13854 (gst_video_scale_get_property), (gst_video_scale_transform_caps),
13855 (gst_video_scale_transform):
13856 Don't claim to be able to handle/transform caps that can't really
13857 be handled by the currently selected scaling method (here: RGB or
13858 packed YUV with 4-tap method). Also add locking to method property.
13859 * tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
13860 (test_basetransform_based):
13861 Some test pipelines for the above (not entirely valgrind clean yet
13864 2007-12-22 05:19:00 +0000 David Schleef <ds@schleef.org>
13866 gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
13867 Original commit message from CVS:
13868 * gst-libs/gst/video/video.c:
13869 * gst-libs/gst/video/video.h:
13870 Add additional RGBA and RGB-24 video formats.
13872 2007-12-21 22:46:56 +0000 Tim-Philipp Müller <tim@centricular.net>
13874 tests/check/: Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be deprecated in the future (see #498924).
13875 Original commit message from CVS:
13876 * tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
13877 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
13878 (test_suburi_error_wrongproto), (test_missing_primary_decoder):
13879 * tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
13880 (cddabasesrc_suite):
13881 Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
13882 deprecated in the future (see #498924).
13884 2007-12-21 22:26:47 +0000 Tim-Philipp Müller <tim@centricular.net>
13886 gst/playback/gststreamselector.c: Don't leak event.
13887 Original commit message from CVS:
13888 * gst/playback/gststreamselector.c: (gst_selector_pad_event):
13891 2007-12-20 19:43:25 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13893 gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
13894 Original commit message from CVS:
13895 * gst-libs/gst/riff/riff-read.c:
13896 Use GST_ROUND_UP_2 macro
13898 2007-12-20 17:13:37 +0000 Tim-Philipp Müller <tim@centricular.net>
13900 gst/playback/.cvsignore: Ignore more.
13901 Original commit message from CVS:
13902 * gst/playback/.cvsignore:
13905 2007-12-20 10:41:29 +0000 Tim-Philipp Müller <tim@centricular.net>
13907 Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
13908 Original commit message from CVS:
13909 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
13910 * gst/playback/gstplaybasebin.c: (set_subtitles_visible),
13911 (set_active_source):
13912 * gst/playback/gstplaybasebin.h:
13913 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
13914 (setup_sinks), (playbin_set_subtitles_visible):
13915 Make switching off of subtitles work. To avoid all kind of
13916 problems with unlinking of the subtitle input, we just keep
13917 the subtitle inputs linked as they are and tell textoverlay
13918 not to render them. Fixes #373011.
13919 Other subtitle switching issues (esp. when there are both
13920 external and in-stream subtitles) remain. They'll be solved
13923 2007-12-18 16:21:35 +0000 Wim Taymans <wim.taymans@gmail.com>
13925 gst/playback/gststreamselector.c: Init the pad segment too.
13926 Original commit message from CVS:
13927 * gst/playback/gststreamselector.c: (gst_selector_pad_init):
13928 Init the pad segment too.
13930 2007-12-18 15:56:51 +0000 Wim Taymans <wim.taymans@gmail.com>
13932 gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
13933 Original commit message from CVS:
13934 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
13935 (gst_audioringbuffer_open_device),
13936 (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
13937 (gst_audioringbuffer_release), (gst_audioringbuffer_start),
13938 (gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
13939 (gst_audio_sink_create_ringbuffer):
13940 Improve debug output.
13941 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
13942 (gst_ring_buffer_pause), (gst_ring_buffer_delay):
13943 Prevent some functions from doing things and failing when the
13944 ringbuffer is not yet acquired.
13946 2007-12-18 15:32:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13948 gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
13949 Original commit message from CVS:
13950 * gst-libs/gst/interfaces/interfaces.h:
13951 Also remove interfaces.h from CVS as it is not needed anymore.
13953 2007-12-18 15:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13955 gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
13956 Original commit message from CVS:
13957 * gst-libs/gst/interfaces/Makefile.am:
13958 interfaces.h is not used anymore so remove it from the build
13961 2007-12-18 01:01:23 +0000 David Schleef <ds@schleef.org>
13963 gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
13964 Original commit message from CVS:
13965 * gst/videotestsrc/gstvideotestsrc.c:
13966 * gst/videotestsrc/gstvideotestsrc.h:
13967 Add a "blink" pattern. Turn on the pain. Apologies. It's useful
13968 for testing vertical refresh synchronization.
13970 2007-12-18 00:13:26 +0000 David Schleef <ds@schleef.org>
13972 Add new GstVideFormat enum and write a bunch of helper functions based around it.
13973 Original commit message from CVS:
13974 * docs/libs/gst-plugins-base-libs-sections.txt:
13975 * gst-libs/gst/video/video.c:
13976 * gst-libs/gst/video/video.h:
13977 Add new GstVideFormat enum and write a bunch of helper functions
13980 2007-12-17 23:41:14 +0000 Tim-Philipp Müller <tim@centricular.net>
13982 Makefile.am: Use new common/win32.mak.
13983 Original commit message from CVS:
13985 Use new common/win32.mak.
13987 2007-12-17 16:44:51 +0000 Wim Taymans <wim.taymans@gmail.com>
13989 gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
13990 Original commit message from CVS:
13991 * gst-libs/gst/audio/gstbaseaudiosrc.c:
13992 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
13994 When going from PLAYING to PAUSED, pause the ringbuffer before calling
13995 the parent state change function, just like the audiosink, because the
13996 parent waits for the element to finish its processing before completing
13997 the state change. This makes going to PAUSED a lot snappier.
13998 When going from READY to PAUSED, don't allow the ringbuffer to start
14001 2007-12-17 00:01:00 +0000 Edward Hervey <bilboed@bilboed.com>
14003 gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
14004 Original commit message from CVS:
14005 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
14006 Yet another fix for broken software that produce files with an empty
14007 blockalign field. Instead of completely failing, make a second attempt
14008 at guessing the width/depth by looking at strf->size.
14010 2007-12-16 23:52:58 +0000 Tim-Philipp Müller <tim@centricular.net>
14012 gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
14013 Original commit message from CVS:
14014 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
14015 (gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
14016 * gst-libs/gst/pbutils/install-plugins.c:
14017 (gst_install_plugins_spawn_child), (gst_install_plugins_supported):
14018 * gst-libs/gst/pbutils/missing-plugins.c:
14019 (gst_missing_plugin_message_get_installer_detail),
14020 (gst_missing_encoder_installer_detail_new):
14021 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
14022 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
14023 Turn a few g_assert_not_reached() into g_return_val_if_reached() to
14024 avoid compiler warnings (#503930).
14026 2007-12-16 23:46:16 +0000 Edward Hervey <bilboed@bilboed.com>
14028 gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
14029 Original commit message from CVS:
14030 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
14031 Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
14032 for jpeg video streams.
14033 Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
14034 for the above modification.
14036 2007-12-15 17:27:48 +0000 Tim-Philipp Müller <tim@centricular.net>
14038 gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
14039 Original commit message from CVS:
14040 * gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
14041 (gst_x_overlay_handle_events):
14042 More guards (we don't want klass to end up being NULL).
14044 2007-12-15 03:40:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14046 Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
14047 Original commit message from CVS:
14049 * gst/volume/gstvolume.c: (gst_volume_init):
14050 Use new gst_base_transform_set_gap_aware() function as volume
14051 correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
14054 2007-12-14 19:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
14056 tests/examples/seek/seek.c: Don't go to READY on EOS as this avoids testing of seeking and restarting after EOS, use ...
14057 Original commit message from CVS:
14058 * tests/examples/seek/seek.c: (msg_segment_done), (main):
14059 Don't go to READY on EOS as this avoids testing of seeking and
14060 restarting after EOS, use the stop button when you want to READY.
14061 Don't try to do a flushing seek in segment-done, it does not make
14062 sense to use this for gapless playback and is not needed.
14064 2007-12-14 18:46:12 +0000 Wim Taymans <wim.taymans@gmail.com>
14066 gst/playback/gstqueue2.c: Use separate timers for input and output rates.
14067 Original commit message from CVS:
14068 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
14069 (reset_rate_timer), (update_in_rates), (update_out_rates),
14070 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
14071 (gst_queue_chain), (gst_queue_loop):
14072 Use separate timers for input and output rates.
14073 Pause measuring the output rate when we block for more data.
14076 2007-12-14 16:23:06 +0000 Christian Schaller <uraeus@gnome.org>
14078 * gst/speexresample/Makefile.am:
14079 update spec file and add two missing files for disting
14080 Original commit message from CVS:
14081 update spec file and add two missing files for disting
14083 2007-12-14 09:24:55 +0000 Wim Taymans <wim.taymans@gmail.com>
14085 gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
14086 Original commit message from CVS:
14087 * gst/playback/gstqueue2.c: (gst_queue_chain):
14088 Pause the timer to measure the input rate when we block because the
14089 queue is filled. See #503262.
14091 2007-12-13 15:54:00 +0000 Peter Kjellerstedt <pkj@axis.com>
14093 gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
14094 Original commit message from CVS:
14095 Patch by: Peter Kjellerstedt <pkj at axis com>
14096 * gst-libs/gst/rtsp/gstrtspconnection.c:
14097 (gst_rtsp_connection_free):
14098 Close control sockets. Fixes #503440.
14100 2007-12-13 12:31:38 +0000 Wim Taymans <wim.taymans@gmail.com>
14102 gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
14103 Original commit message from CVS:
14104 * gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
14105 Expose the right pad in the right place with the right element.
14107 2007-12-13 11:40:10 +0000 Tim-Philipp Müller <tim@centricular.net>
14109 gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
14110 Original commit message from CVS:
14111 * gst-libs/gst/pbutils/descriptions.c: (formats):
14112 Add description for 'private' dts caps (who come up with that name?).
14114 2007-12-13 10:10:35 +0000 Tim-Philipp Müller <tim@centricular.net>
14116 Makefile.am: Add check-exports target and run it with 'make check'.
14117 Original commit message from CVS:
14119 Add check-exports target and run it with 'make check'.
14121 Be stricter about what we export in our libraries: change regexp so that
14122 we only export _gst_foo(), but not __gst_foo().
14123 * gst-libs/gst/cdda/base64.h: (rfc822_binary):
14124 * gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
14125 Change internal functions to __gst_foo so they dont' get exported.
14126 * win32/common/libgstaudio.def:
14127 Add missing symbols.
14129 2007-12-11 21:18:57 +0000 David Schleef <ds@schleef.org>
14132 ChangeLog: remove conflict markers
14133 Original commit message from CVS:
14134 ChangeLog: remove conflict markers
14136 2007-12-11 17:14:13 +0000 Tim-Philipp Müller <tim@centricular.net>
14138 ext/gnomevfs/: Use gst_tag_freeform_string_to_utf8() here, which also takes into account any character sets specified...
14139 Original commit message from CVS:
14140 * ext/gnomevfs/Makefile.am:
14141 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
14142 Use gst_tag_freeform_string_to_utf8() here, which also takes
14143 into account any character sets specified by the user via
14144 environment variables.
14146 2007-12-10 15:21:41 +0000 Wim Taymans <wim.taymans@gmail.com>
14148 gst/audioconvert/Makefile.am: Also link to libm.
14149 Original commit message from CVS:
14150 * gst/audioconvert/Makefile.am:
14153 2007-12-10 15:13:55 +0000 Wim Taymans <wim.taymans@gmail.com>
14155 gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
14156 Original commit message from CVS:
14157 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
14158 No need for floating point operations here. avoids having to link
14159 against the math library too.
14161 2007-12-10 11:16:25 +0000 Tim-Philipp Müller <tim@centricular.net>
14163 Add one or two missing formats. Generate ADPCM description dynamically depending on layout/format.
14164 Original commit message from CVS:
14165 * gst-libs/gst/pbutils/descriptions.c: (formats),
14166 (format_info_get_desc):
14167 * tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
14169 Add one or two missing formats. Generate ADPCM description
14170 dynamically depending on layout/format.
14172 2007-12-09 04:28:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14174 configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
14175 Original commit message from CVS:
14177 Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
14179 2007-12-08 18:38:39 +0000 Robin Stocker <robin.stocker@gmx.ch>
14181 gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
14182 Original commit message from CVS:
14183 Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
14184 * gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
14185 Some .srt files start with chunk number 0 and not chunk number 1,
14186 recognise and accept those as well (fixes #502497).
14187 * tests/check/elements/subparse.c: (srt_input), (srt_input0),
14189 Add unit test for the above.
14191 2007-12-06 12:08:21 +0000 Wim Taymans <wim.taymans@gmail.com>
14193 gst/playback/gstplay-enum.*: Add missing files.
14194 Original commit message from CVS:
14195 * gst/playback/gstplay-enum.c:
14196 (register_gst_autoplug_select_result),
14197 (gst_autoplug_select_result_get_type):
14198 * gst/playback/gstplay-enum.h:
14201 2007-12-05 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
14203 gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
14204 Original commit message from CVS:
14205 * gst/playback/Makefile.am:
14206 Group decodebin2 and uridecodebin into the same plugin so that they
14207 can share the GEnumType.
14208 * gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
14209 (_gst_select_accumulator), (gst_decode_bin_class_init),
14210 (gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
14211 (gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
14212 (analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
14213 Add signal to sort factories instead of the more awkward autoplug-select
14215 Modify autoplug_select so that we can try, skip or expose the
14216 autopluggin of an element on a pad.
14217 * gst/playback/gstfactorylists.c: (compare_ranks),
14218 (decoders_filter), (sinks_filter), (gst_factory_list_is_type),
14219 (element_filter), (gst_factory_list_get_elements),
14220 (gst_factory_list_debug), (gst_factory_list_filter):
14221 * gst/playback/gstfactorylists.h:
14222 Simplify the API, allow getting elements based on mask.
14223 * gst/playback/gstplay-marshal.list:
14224 Add some more marshallers.
14225 * gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
14226 (gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
14227 (autoplug_select_cb), (activate_group):
14228 Add support for managing non-raw sinks by providing a custom element and
14229 sink list to decodebin2.
14230 Try to plug non-raw sinks when decodebin2 using autoplug-select of
14232 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
14233 (gst_play_sink_set_mode), (gst_play_sink_request_pad):
14234 * gst/playback/gstplaysink.h:
14235 Add support for raw and non-raw sinks.
14236 Add support to force sinks selected by playbin2.
14237 Don't plug raw converters for non-raw sinks.
14238 * gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
14239 (_gst_select_accumulator), (gst_uri_decode_bin_class_init),
14240 (proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
14242 Use right accumulators.
14245 2007-12-03 13:47:00 +0000 Wim Taymans <wim.taymans@gmail.com>
14247 gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
14248 Original commit message from CVS:
14249 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
14250 Use runnning time as the base time instead of the timestamp.
14251 Spotted by Saur on IRC.
14253 2007-12-03 11:32:30 +0000 Edward Hervey <bilboed@bilboed.com>
14255 gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
14256 Original commit message from CVS:
14257 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
14258 Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
14260 2007-12-03 10:58:14 +0000 Wim Taymans <wim.taymans@gmail.com>
14262 ext/ogg/gstoggdemux.c: If we find a new serial number but it does not contain a BOS page, make sure we initialize the...
14263 Original commit message from CVS:
14264 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
14265 (gst_ogg_demux_read_chain):
14266 If we find a new serial number but it does not contain a BOS page, make
14267 sure we initialize the chain to NULL because else we will try to scan it
14268 and crash. Fixes #500763
14270 2007-11-30 17:47:15 +0000 Wim Taymans <wim.taymans@gmail.com>
14272 gst/playback/: Refactor some common code to filter factories and check caps compat.
14273 Original commit message from CVS:
14274 * gst/playback/Makefile.am:
14275 * gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
14276 (get_feature_array), (decoders_filter), (sinks_filter),
14277 (gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
14278 (gst_factory_list_filter):
14279 * gst/playback/gstfactorylists.h:
14280 Refactor some common code to filter factories and check caps compat.
14281 * gst/playback/gstdecodebin.c:
14282 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
14283 (gst_decode_bin_init), (gst_decode_bin_dispose),
14284 (gst_decode_bin_autoplug_continue),
14285 (gst_decode_bin_autoplug_factories),
14286 (gst_decode_bin_autoplug_select), (analyze_new_pad),
14287 (find_compatibles):
14288 * gst/playback/gstplaybin.c:
14289 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
14290 (gst_play_bin_init), (gst_play_bin_finalize),
14291 (autoplug_factories_cb), (activate_group):
14292 * gst/playback/gstqueue2.c:
14293 * gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
14294 (proxy_autoplug_continue_signal),
14295 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
14296 (proxy_drained_signal):
14297 Add some more debug info and use factor filtering code.
14299 2007-11-26 13:19:46 +0000 Julien Moutte <julien@moutte.net>
14301 configure.ac: Add QuickTime Wrapper plug-in.
14302 Original commit message from CVS:
14303 2007-11-26 Julien Moutte <julien@fluendo.com>
14304 * configure.ac: Add QuickTime Wrapper plug-in.
14305 * gst/speexresample/gstspeexresample.c:
14306 (gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
14307 build on Mac OS X Leopard. Incorrect printf format arguments.
14309 * sys/qtwrapper/Makefile.am:
14310 * sys/qtwrapper/audiodecoders.c:
14311 (qtwrapper_audio_decoder_base_init),
14312 (qtwrapper_audio_decoder_class_init),
14313 (qtwrapper_audio_decoder_init),
14314 (clear_AudioStreamBasicDescription), (fill_indesc_mp3),
14315 (fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
14316 (make_samr_magic_cookie), (open_decoder),
14317 (qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
14318 (qtwrapper_audio_decoder_chain),
14319 (qtwrapper_audio_decoder_sink_event),
14320 (qtwrapper_audio_decoders_register):
14321 * sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
14323 * sys/qtwrapper/codecmapping.h:
14324 * sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
14325 (image_description_for_mp4v), (image_description_from_stsd_buffer),
14326 (image_description_from_codec_data):
14327 * sys/qtwrapper/imagedescription.h:
14328 * sys/qtwrapper/qtutils.c: (get_name_info_from_component),
14329 (get_output_info_from_component), (dump_avcc_atom),
14330 (dump_image_description), (dump_codec_decompress_params),
14331 (addSInt32ToDictionary), (dump_cvpixel_buffer),
14332 (DestroyAudioBufferList), (AllocateAudioBufferList):
14333 * sys/qtwrapper/qtutils.h:
14334 * sys/qtwrapper/qtwrapper.c: (plugin_init):
14335 * sys/qtwrapper/qtwrapper.h:
14336 * sys/qtwrapper/videodecoders.c:
14337 (qtwrapper_video_decoder_base_init),
14338 (qtwrapper_video_decoder_class_init),
14339 (qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
14340 (fill_image_description), (new_image_description), (close_decoder),
14341 (open_decoder), (qtwrapper_video_decoder_sink_setcaps),
14342 (decompressCb), (qtwrapper_video_decoder_chain),
14343 (qtwrapper_video_decoder_sink_event),
14344 (qtwrapper_video_decoders_register): Initial import of QuickTime
14345 wrapper jointly developped by Songbird authors (Pioneers of the
14346 Inevitable) and Fluendo.
14348 2007-11-26 12:25:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14350 gst/: Add GAP-flag support.
14351 Original commit message from CVS:
14352 * gst/audiotestsrc/gstaudiotestsrc.c:
14353 * gst/volume/gstvolume.c:
14354 * gst/volume/gstvolume.h:
14355 Add GAP-flag support.
14357 2007-11-26 08:43:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14359 gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
14360 Original commit message from CVS:
14361 * gst/speexresample/README:
14362 * gst/speexresample/arch.h:
14363 * gst/speexresample/resample.c: (resampler_basic_direct_single),
14364 (resampler_basic_direct_double),
14365 (resampler_basic_interpolate_single),
14366 (resampler_basic_interpolate_double),
14367 (speex_resampler_process_native), (speex_resampler_process_float),
14368 (speex_resampler_process_int),
14369 (speex_resampler_process_interleaved_float),
14370 (speex_resampler_process_interleaved_int),
14371 (speex_resampler_get_input_latency),
14372 (speex_resampler_get_output_latency):
14373 * gst/speexresample/speex_resampler.h:
14374 Update speex resampler to latest SVN. We're now down to only the
14375 changes noted in README again.
14376 * gst/speexresample/speex_resampler_wrapper.h:
14377 * gst/speexresample/gstspeexresample.c:
14378 (gst_speex_resample_push_drain), (gst_speex_resample_query):
14379 Adjust to API changes.
14381 2007-11-24 15:02:01 +0000 Julien Moutte <julien@moutte.net>
14383 tests/examples/seek/seek.c: Increase the range of the rate selector as I would like to test QOS behavior at higher fo...
14384 Original commit message from CVS:
14385 2007-11-24 Julien MOUTTE <julien@moutte.net>
14386 * tests/examples/seek/seek.c: (main): Increase the range of the
14387 rate selector as I would like to test QOS behavior at higher
14388 forward and reverse playback speed like say 64x.
14390 2007-11-23 10:21:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14392 gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
14393 Original commit message from CVS:
14394 * gst/speexresample/gstspeexresample.c:
14395 (gst_speex_resample_update_state):
14396 Only post the latency message if we have a resampler state already.
14398 2007-11-23 10:21:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14400 gst/audioresample/gstaudioresample.c: Implement latency query.
14401 Original commit message from CVS:
14402 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
14403 (audioresample_query), (audioresample_query_type),
14404 (gst_audioresample_set_property):
14405 Implement latency query.
14407 2007-11-23 10:01:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14409 gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
14410 Original commit message from CVS:
14411 * gst/speexresample/gstspeexresample.c:
14412 (gst_speex_resample_update_state):
14413 Also post GST_MESSAGE_LATENCY if the latency changes.
14415 2007-11-23 08:48:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14417 gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
14418 Original commit message from CVS:
14419 * gst/speexresample/resample.c: (speex_resampler_get_latency),
14420 (speex_resampler_drain_float), (speex_resampler_drain_int),
14421 (speex_resampler_drain_interleaved_float),
14422 (speex_resampler_drain_interleaved_int):
14423 * gst/speexresample/speex_resampler.h:
14424 * gst/speexresample/speex_resampler_wrapper.h:
14425 Add functions to push the remaining samples and to get the latency
14426 of the resampler. These will get added to Speex SVN in this or a
14427 slightly changed form at some point too and should get merged then
14429 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
14430 (gst_speex_resample_init_state),
14431 (gst_speex_resample_transform_size),
14432 (gst_speex_resample_push_drain), (gst_speex_resample_event),
14433 (gst_speex_fix_output_buffer), (gst_speex_resample_process),
14434 (gst_speex_resample_query), (gst_speex_resample_query_type):
14435 Drop the prepending zeroes and output the remaining samples on EOS.
14436 Also properly implement the latency query for this. speexresample
14437 should be completely ready for production use now.
14439 2007-11-21 18:02:21 +0000 Wim Taymans <wim.taymans@gmail.com>
14441 gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
14442 Original commit message from CVS:
14443 * gst-libs/gst/audio/gstbaseaudiosink.c:
14444 (gst_base_audio_sink_drain):
14445 Our EOS time contains the base_time, _wait_eos() expects a running_time
14446 so we have to subtract the base_time again before calling the function.
14447 This fixes an EOS regression where the base_time was added twice and EOS
14448 took longer and longer in certain situations.
14451 2007-11-21 13:04:17 +0000 Wim Taymans <wim.taymans@gmail.com>
14453 Expose methods for some object properties so that subclasses can more easily configure them.
14454 Original commit message from CVS:
14455 * docs/libs/gst-plugins-base-libs-sections.txt:
14456 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
14457 (gst_base_audio_sink_set_provide_clock),
14458 (gst_base_audio_sink_get_provide_clock),
14459 (gst_base_audio_sink_set_slave_method),
14460 (gst_base_audio_sink_get_slave_method),
14461 (gst_base_audio_sink_set_property),
14462 (gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
14463 (gst_base_audio_sink_none_slaving),
14464 (gst_base_audio_sink_handle_slaving):
14465 * gst-libs/gst/audio/gstbaseaudiosink.h:
14466 Expose methods for some object properties so that subclasses can more
14467 easily configure them.
14468 Added slave method none, that completely disables slaving to the
14470 API: gst_base_audio_sink_set_provide_clock()
14471 API: gst_base_audio_sink_get_provide_clock()
14472 API: gst_base_audio_sink_set_slave_method()
14473 API: gst_base_audio_sink_get_slave_method()
14474 * gst-libs/gst/audio/gstbaseaudiosrc.c:
14475 (gst_base_audio_src_set_provide_clock),
14476 (gst_base_audio_src_get_provide_clock),
14477 (gst_base_audio_src_set_property),
14478 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
14479 * gst-libs/gst/audio/gstbaseaudiosrc.h:
14480 Expose methods for some object properties so that subclasses can more
14481 easily configure them.
14482 API: gst_base_audio_src_set_provide_clock()
14483 API: gst_base_audio_src_get_provide_clock()
14485 2007-11-21 10:18:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14487 gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
14488 Original commit message from CVS:
14489 * gst/speexresample/README:
14490 Add README explaining where the resampling code was taken from
14491 and which changes were done.
14492 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
14494 Use g_malloc() and friends instead of malloc() to achieve higher
14495 portability and define the functions inline.
14496 * gst/speexresample/speex_resampler.h:
14497 Add back some useless preprocessor stuff to keep the diff between
14498 our version and the one from the Speex SVN repository lower.
14500 2007-11-20 20:23:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14502 gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
14503 Original commit message from CVS:
14504 * gst/speexresample/gstspeexresample.c:
14505 (gst_speex_fix_output_buffer), (gst_speex_resample_transform):
14506 Some small cleanup and addition of a TODO item.
14508 2007-11-20 12:56:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14510 gst/speexresample/Makefile.am: Add missing file.
14511 Original commit message from CVS:
14512 * gst/speexresample/Makefile.am:
14515 2007-11-20 07:53:56 +0000 Joe Peterson <lavajoe@gentoo.org>
14517 gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
14518 Original commit message from CVS:
14519 Patch by: Joe Peterson <lavajoe at gentoo dot org>
14520 * gst-libs/gst/sdp/gstsdpmessage.c:
14521 Fix compilation on FreeBSD (Gentoo). Fixes #498228.
14523 2007-11-20 07:47:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14525 Add speexresample to the docs and while at that do a make update.
14526 Original commit message from CVS:
14527 * docs/plugins/Makefile.am:
14528 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
14529 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
14530 * docs/plugins/gst-plugins-bad-plugins.args:
14531 * docs/plugins/gst-plugins-bad-plugins.signals:
14532 * docs/plugins/inspect/plugin-bz2.xml:
14533 * docs/plugins/inspect/plugin-cdxaparse.xml:
14534 * docs/plugins/inspect/plugin-dtsdec.xml:
14535 * docs/plugins/inspect/plugin-equalizer.xml:
14536 * docs/plugins/inspect/plugin-faac.xml:
14537 * docs/plugins/inspect/plugin-faad.xml:
14538 * docs/plugins/inspect/plugin-filter.xml:
14539 * docs/plugins/inspect/plugin-freeze.xml:
14540 * docs/plugins/inspect/plugin-gio.xml:
14541 * docs/plugins/inspect/plugin-gsm.xml:
14542 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
14543 * docs/plugins/inspect/plugin-h264parse.xml:
14544 * docs/plugins/inspect/plugin-modplug.xml:
14545 * docs/plugins/inspect/plugin-mpeg2enc.xml:
14546 * docs/plugins/inspect/plugin-musepack.xml:
14547 * docs/plugins/inspect/plugin-musicbrainz.xml:
14548 * docs/plugins/inspect/plugin-nsfdec.xml:
14549 * docs/plugins/inspect/plugin-replaygain.xml:
14550 * docs/plugins/inspect/plugin-soundtouch.xml:
14551 * docs/plugins/inspect/plugin-spcdec.xml:
14552 * docs/plugins/inspect/plugin-spectrum.xml:
14553 * docs/plugins/inspect/plugin-speed.xml:
14554 * docs/plugins/inspect/plugin-tta.xml:
14555 * docs/plugins/inspect/plugin-videosignal.xml:
14556 * docs/plugins/inspect/plugin-xingheader.xml:
14557 * docs/plugins/inspect/plugin-xvid.xml:
14558 * gst/speexresample/gstspeexresample.h:
14559 Add speexresample to the docs and while at that do a make update.
14561 2007-11-20 07:30:30 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14563 gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
14564 Original commit message from CVS:
14565 * gst/speexresample/gstspeexresample.c:
14566 (gst_speex_fix_output_buffer), (gst_speex_resample_process):
14567 If the resampler gives less output samples than expected
14568 adjust the output buffer and print a warning.
14570 2007-11-20 07:02:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14572 Add resample element based on the Speex resampling algorithm.
14573 Original commit message from CVS:
14575 * gst/speexresample/arch.h:
14576 * gst/speexresample/fixed_generic.h:
14577 * gst/speexresample/gstspeexresample.c:
14578 (gst_speex_resample_base_init), (gst_speex_resample_class_init),
14579 (gst_speex_resample_init), (gst_speex_resample_start),
14580 (gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
14581 (gst_speex_resample_transform_caps),
14582 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
14583 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
14584 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
14585 (gst_speex_resample_event), (gst_speex_resample_check_discont),
14586 (gst_speex_resample_process), (gst_speex_resample_transform),
14587 (gst_speex_resample_set_property),
14588 (gst_speex_resample_get_property), (plugin_init):
14589 * gst/speexresample/gstspeexresample.h:
14590 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
14591 (speex_free), (compute_func), (main), (sinc), (cubic_coef),
14592 (resampler_basic_direct_single), (resampler_basic_direct_double),
14593 (resampler_basic_interpolate_single),
14594 (resampler_basic_interpolate_double), (update_filter),
14595 (speex_resampler_init), (speex_resampler_init_frac),
14596 (speex_resampler_destroy), (speex_resampler_process_native),
14597 (speex_resampler_process_float), (speex_resampler_process_int),
14598 (speex_resampler_process_interleaved_float),
14599 (speex_resampler_process_interleaved_int),
14600 (speex_resampler_set_rate), (speex_resampler_get_rate),
14601 (speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
14602 (speex_resampler_set_quality), (speex_resampler_get_quality),
14603 (speex_resampler_set_input_stride),
14604 (speex_resampler_get_input_stride),
14605 (speex_resampler_set_output_stride),
14606 (speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
14607 (speex_resampler_reset_mem), (speex_resampler_strerror):
14608 * gst/speexresample/speex_resampler.h:
14609 * gst/speexresample/speex_resampler_float.c:
14610 * gst/speexresample/speex_resampler_int.c:
14611 * gst/speexresample/speex_resampler_wrapper.h:
14612 Add resample element based on the Speex resampling algorithm.
14614 2007-11-19 12:30:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14616 tests/check/libs/fft.c: Fix scaling to really have dB instead of something else.
14617 Original commit message from CVS:
14618 * tests/check/libs/fft.c: (GST_START_TEST):
14619 Fix scaling to really have dB instead of something else.
14621 2007-11-19 12:08:16 +0000 Julien Moutte <julien@moutte.net>
14623 tests/examples/seek/seek.c: There's a nice macro to check
14624 Original commit message from CVS:
14625 2007-11-19 Julien MOUTTE <julien@moutte.net>
14626 * tests/examples/seek/seek.c: (main): There's a nice macro to
14628 GTK version, use it.
14630 2007-11-19 11:59:20 +0000 Julien Moutte <julien@moutte.net>
14632 tests/examples/seek/seek.c: Try to support stable version of GTK.
14633 Original commit message from CVS:
14634 2007-11-19 Julien MOUTTE <julien@moutte.net>
14635 * tests/examples/seek/seek.c: (main): Try to support stable version
14638 2007-11-17 15:25:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14640 gst/playback/: Fix the build + little README update.
14641 Original commit message from CVS:
14642 * gst/playback/README:
14643 * gst/playback/test7.c:
14644 Fix the build + little README update.
14646 2007-11-16 16:02:45 +0000 Wim Taymans <wim.taymans@gmail.com>
14648 tests/examples/seek/seek.c: Add playbin2 seek pipeline.
14649 Original commit message from CVS:
14650 * tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
14651 Add playbin2 seek pipeline.
14653 2007-11-16 15:44:48 +0000 Wim Taymans <wim.taymans@gmail.com>
14655 gst/playback/: Add playbin2.
14656 Original commit message from CVS:
14657 * gst/playback/Makefile.am:
14658 * gst/playback/gstplayback.c: (plugin_init):
14659 * gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
14660 (eos_cb), (about_to_finish_cb), (main):
14662 Added gapless playback example.
14663 * gst/playback/gstplaybasebin.c:
14664 * gst/playback/gstplaybasebin.h:
14665 * gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
14666 * gst/playback/gstqueue2.c:
14667 * gst/playback/test.c:
14668 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
14670 * gst/playback/gststreaminfo.h:
14672 * gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
14673 (gst_play_bin_class_init), (init_group), (gst_play_bin_init),
14674 (gst_play_bin_dispose), (gst_play_bin_set_uri),
14675 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
14676 (gst_play_bin_get_property), (gst_play_bin_handle_message),
14677 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
14678 (drained_cb), (unlink_group), (activate_group),
14679 (setup_next_source), (gst_play_bin_change_state),
14680 (gst_play_bin2_plugin_init):
14681 Added raw first version of playbin2. Does chained oggs and gapless
14682 playback fine. No support for raw sinks yet. No visualisations or
14684 * gst/playback/gstplaysink.c: (gst_play_sink_get_type),
14685 (gst_play_sink_class_init), (gst_play_sink_init),
14686 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
14687 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
14688 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
14689 (gst_play_sink_set_property), (gst_play_sink_get_property),
14690 (post_missing_element_message), (free_chain), (add_chain),
14691 (activate_chain), (gen_video_chain), (gen_text_element),
14692 (gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
14693 (gst_play_sink_set_mode), (gst_play_sink_request_pad),
14694 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
14695 (gst_play_sink_send_event), (gst_play_sink_change_state):
14696 * gst/playback/gstplaysink.h:
14697 Added Element that abstracts the sinks and their pipelines for playbin2.
14699 2007-11-16 15:05:07 +0000 Wim Taymans <wim.taymans@gmail.com>
14701 gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
14702 Original commit message from CVS:
14703 * gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
14704 (gst_selector_pad_class_init), (gst_selector_pad_init),
14705 (gst_selector_pad_finalize), (gst_selector_pad_reset),
14706 (gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
14707 (gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
14708 (gst_selector_pad_chain), (gst_stream_selector_get_type),
14709 (gst_stream_selector_base_init), (gst_stream_selector_class_init),
14710 (gst_stream_selector_init), (gst_stream_selector_set_property),
14711 (gst_stream_selector_get_linked_pad),
14712 (gst_stream_selector_getcaps),
14713 (gst_stream_selector_is_active_sinkpad),
14714 (gst_stream_selector_activate_sinkpad),
14715 (gst_stream_selector_get_linked_pads),
14716 (gst_stream_selector_request_new_pad),
14717 (gst_stream_selector_release_pad):
14718 * gst/playback/gststreamselector.h:
14719 Improve streamselector, make it select and unselect the current pad more
14721 Subclass GstPad for the sinkpads of the selector.
14722 Handle segments more correctly.
14723 Fix caps negotiation.
14724 Implement release_pad.
14726 2007-11-16 12:51:44 +0000 Wim Taymans <wim.taymans@gmail.com>
14728 gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
14729 Original commit message from CVS:
14730 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
14731 (gst_decode_group_check_if_drained), (source_pad_event_probe),
14733 Add drained signal fired when decodebin finishes decoding the data.
14734 Remove deprecated STATE_DIRTY message.
14735 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
14736 (unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
14737 (analyse_source), (proxy_drained_signal), (make_decoder),
14738 (source_new_pad), (value_list_append_structure_list),
14739 (handle_redirect_message), (handle_message):
14740 Proxy the new drained signal.
14741 Handle pad removed from decodebin.
14742 Handle redirect messages by sorting multiple redirections based on the
14745 2007-11-16 11:22:09 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
14747 gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
14748 Original commit message from CVS:
14749 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
14750 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
14751 Fix leaking headers. Fixes #496761.
14753 2007-11-16 11:16:58 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
14755 sys/: Don't leak the PAR on errors. Fixes #496731.
14756 Original commit message from CVS:
14757 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
14758 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
14759 (gst_ximagesink_change_state):
14760 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
14761 Don't leak the PAR on errors. Fixes #496731.
14763 2007-11-16 10:14:34 +0000 Tim-Philipp Müller <tim@centricular.net>
14765 gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
14766 Original commit message from CVS:
14767 * gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
14768 (gst_tag_from_id3_user_tag):
14769 Add mapping for audio cd discid tags, so we can extract
14770 them from tags as well (see #347848). Also compare identifiers
14771 in ID3v2 TXXX frames in a case-insensitive way to increase
14772 compatibility when reading tags (discid vs. DiscID vs. DiscId).
14774 2007-11-16 01:21:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14776 gst-plugins-base.doap: Oops, fix the release name.
14777 Original commit message from CVS:
14778 * gst-plugins-base.doap:
14779 Oops, fix the release name.
14781 2007-11-16 00:44:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14783 gst-plugins-base.doap: Add 0.10.15 release
14784 Original commit message from CVS:
14785 * gst-plugins-base.doap:
14786 Add 0.10.15 release
14788 2007-11-16 00:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14790 configure.ac: Back to CVS
14791 Original commit message from CVS:
14795 === release 0.10.15 ===
14797 2007-11-16 00:14:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14799 configure.ac: releasing 0.10.15, "No need to argue"
14800 Original commit message from CVS:
14801 === release 0.10.15 ===
14802 2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
14804 releasing 0.10.15, "No need to argue"
14806 2007-11-16 00:04:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14831 Original commit message from CVS:
14834 2007-11-15 21:40:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14836 win32/vs6/libgstfft.dsp: Convert line endings to DOS.
14837 Original commit message from CVS:
14838 * win32/vs6/libgstfft.dsp:
14839 Convert line endings to DOS.
14841 2007-11-15 21:14:04 +0000 Sébastien Moutte <sebastien@moutte.net>
14843 win32/: Add a project file for fft plugin and remove socket based plugin which don't build from the workspace.* win32...
14844 Original commit message from CVS:
14845 * win32/vs6/gst_plugins_base.dsw:
14846 * win32/vs6/libgstfft.dsp:
14848 Add a project file for fft plugin and remove socket
14849 based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
14850 * win32/vs6/libgstrtp.dsp:
14851 * win32/vs6/libgsttag.dsp:
14852 Convert line endings back to DOS.
14855 2007-11-14 12:27:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14857 win32/vs6/: Convert line endings back to DOS
14858 Original commit message from CVS:
14859 * win32/vs6/libgstinterfaces.dsp:
14860 * win32/vs6/libgstrtsp.dsp:
14861 Convert line endings back to DOS
14863 2007-11-14 11:08:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14865 gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
14866 Original commit message from CVS:
14867 * gst-libs/gst/fft/kiss_fft_f32.h:
14868 * gst-libs/gst/fft/kiss_fft_f64.h:
14869 * gst-libs/gst/fft/kiss_fft_s16.h:
14870 * gst-libs/gst/fft/kiss_fft_s32.h:
14871 Don't include malloc.h which doesn't exist on Mac OSX.
14872 Instead, pull in glib.h and use g_malloc/g_free for
14873 consistency. Fixes: #496548
14875 2007-11-09 15:54:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14877 gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
14878 Original commit message from CVS:
14879 * gst/playback/gstdecodebin2.c:
14880 Dont leak ghostpad. Fixes #475451.
14882 2007-11-09 12:21:52 +0000 Wim Taymans <wim.taymans@gmail.com>
14884 Update some more docs and comments.
14885 Original commit message from CVS:
14886 * docs/design/design-decodebin.txt:
14887 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
14888 Update some more docs and comments.
14890 2007-11-07 16:47:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14892 Require GIO >= 0.1.2 and adjust unit test for an API change.
14893 Original commit message from CVS:
14895 * tests/check/pipelines/gio.c: (GST_START_TEST):
14896 Require GIO >= 0.1.2 and adjust unit test for an API change.
14898 2007-11-07 15:18:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14900 ext/gio/gstgio.h: Add macro to check if a stream supports seeking.
14901 Original commit message from CVS:
14902 * ext/gio/gstgio.h:
14903 Add macro to check if a stream supports seeking.
14904 * ext/gio/Makefile.am:
14905 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
14906 (gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
14907 (gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
14908 (gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
14909 (gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
14910 (gst_gio_base_sink_render), (gst_gio_base_sink_query),
14911 (gst_gio_base_sink_set_stream):
14912 * ext/gio/gstgiobasesink.h:
14913 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
14914 (gst_gio_base_src_class_init), (gst_gio_base_src_init),
14915 (gst_gio_base_src_finalize), (gst_gio_base_src_start),
14916 (gst_gio_base_src_stop), (gst_gio_base_src_get_size),
14917 (gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
14918 (gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
14919 (gst_gio_base_src_create), (gst_gio_base_src_set_stream):
14920 * ext/gio/gstgiobasesrc.h:
14921 Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
14922 base classes that only require a GInputStream or GOutputStream to
14924 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
14925 (gst_gio_sink_class_init), (gst_gio_sink_init),
14926 (gst_gio_sink_finalize), (gst_gio_sink_start):
14927 * ext/gio/gstgiosink.h:
14928 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
14929 (gst_gio_src_class_init), (gst_gio_src_init),
14930 (gst_gio_src_finalize), (gst_gio_src_start):
14931 * ext/gio/gstgiosrc.h:
14932 Use the newly created base classes here.
14933 * ext/gio/gstgio.c: (plugin_init):
14934 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
14935 (gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
14936 (gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
14937 (gst_gio_stream_sink_get_property):
14938 * ext/gio/gstgiostreamsink.h:
14939 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
14940 (gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
14941 (gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
14942 (gst_gio_stream_src_get_property):
14943 * ext/gio/gstgiostreamsrc.h:
14944 Implement GstGioStreamSink and GstGioStreamSrc that have a property
14945 to set the GInputStream/GOutputStream that should be used.
14946 * tests/check/Makefile.am:
14947 * tests/check/pipelines/.cvsignore:
14948 * tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
14949 (gio_testsuite), (main):
14950 Add unit test for giostreamsrc and giostreamsink.
14952 2007-11-07 11:48:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14954 ext/gio/gstgio.c: Remove nowadays unnecessary workaround for a crash.
14955 Original commit message from CVS:
14956 * ext/gio/gstgio.c: (plugin_init):
14957 Remove nowadays unnecessary workaround for a crash.
14958 * ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
14959 (gst_gio_sink_start), (gst_gio_sink_stop),
14960 (gst_gio_sink_unlock_stop):
14961 * ext/gio/gstgiosink.h:
14962 * ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
14963 (gst_gio_src_stop), (gst_gio_src_unlock_stop):
14964 * ext/gio/gstgiosrc.h:
14965 Make the finalize function safer, clean up everything that could stay
14967 Reset the cancellable instead of creating a new one after cancelling
14969 Don't store the GFile in the element, it's only necessary for creating
14972 2007-11-06 23:35:39 +0000 Sebastien Moutte <sebastien@moutte.net>
14974 gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
14975 Original commit message from CVS:
14976 Patch by: Sebastien Moutte <sebastien moutte net>
14977 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
14978 (gst_rtcp_unix_to_ntp):
14979 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
14980 Fix some C99-isms and and a missing function that some versions of
14981 MSVC don't like too much (#494346).
14982 * win32/vs6/gst_plugins_base.dsw:
14983 * win32/vs6/libgstaudio.dsp:
14984 * win32/vs6/libgstrtp.dsp:
14985 * win32/vs6/libgsttag.dsp:
14986 Update vs6 projects files (#494346).
14988 2007-11-06 16:38:49 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
14990 win32/common/: More missing symbols to export (fixes #493986).
14991 Original commit message from CVS:
14992 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
14993 * win32/common/libgstaudio.def:
14994 * win32/common/libgstcdda.def:
14995 * win32/common/libgstinterfaces.def:
14996 * win32/common/libgstnetbuffer.def:
14997 * win32/common/libgstpbutils.def:
14998 * win32/common/libgstrtp.def:
14999 * win32/common/libgstrtsp.def:
15000 * win32/common/libgsttag.def:
15001 * win32/common/libgstvideo.def:
15002 More missing symbols to export (fixes #493986).
15004 2007-11-06 11:58:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15006 Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
15007 Original commit message from CVS:
15008 * docs/libs/gst-plugins-base-libs-sections.txt:
15009 * gst-libs/gst/fft/gstfftf32.c:
15010 * gst-libs/gst/fft/gstfftf32.h:
15011 * gst-libs/gst/fft/gstfftf64.c:
15012 * gst-libs/gst/fft/gstfftf64.h:
15013 * gst-libs/gst/fft/gstffts16.c:
15014 * gst-libs/gst/fft/gstffts16.h:
15015 * gst-libs/gst/fft/gstffts32.c:
15016 * gst-libs/gst/fft/gstffts32.h:
15017 * tests/check/libs/fft.c: (GST_START_TEST):
15018 Remove the magnitude and phase calculation functions as these have
15019 very special use cases and can't even be used for the spectrum
15020 element. Also adjust the docs to mention some properties of the used
15021 FFT implemention, i.e. how the values are scaled. Fixes #492098.
15023 2007-11-06 11:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
15025 gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
15026 Original commit message from CVS:
15027 * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
15029 Avoid crash when there are external subtitles (fixes #491722).
15031 2007-11-03 10:39:21 +0000 Tim-Philipp Müller <tim@centricular.net>
15033 ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
15034 Original commit message from CVS:
15035 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
15036 * ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
15037 'Could not open resource for writing' is not an acceptable
15038 error message when we can't open the audio device (see #492334),
15039 even less so when we're trying to open it to record something.
15041 2007-11-02 21:03:01 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15043 win32/common/libgstrtp.def: Add some more missing symbols (#492813).
15044 Original commit message from CVS:
15045 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15046 * win32/common/libgstrtp.def:
15047 Add some more missing symbols (#492813).
15049 2007-11-02 14:59:06 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
15051 tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
15052 Original commit message from CVS:
15053 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
15054 * tests/check/elements/audioconvert.c: (verify_convert):
15055 Add check to make sure that the out caps have a channel layout
15056 set on them where they should have one.
15058 2007-11-01 13:28:59 +0000 Vincent Torri <vtorri@univ-evry.fr>
15060 gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
15061 Original commit message from CVS:
15062 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
15063 * gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
15064 * gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
15065 Include our own _stdint.h instead of sys/types.h, makes MingW happy
15067 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
15068 Use _pipe directly, GLib doesn't have a pipe() macro any longer
15069 (it disappeared in GLib 2.14.0) (#492306).
15070 * gst-libs/gst/sdp/Makefile.am:
15071 * gst-libs/gst/sdp/gstsdpmessage.c:
15072 Fix includes and LIBS for win32/Mingw (#492306).
15073 * tests/examples/dynamic/addstream.c (pause_play_stream):
15074 Use more portable g_usleep() instead of sleep() (#492306).
15076 2007-11-01 12:51:57 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15078 gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
15079 Original commit message from CVS:
15080 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15081 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
15082 (gst_ring_buffer_parse_caps):
15083 Return NULL instead of an enum that happens to be 0, fixes warning
15085 * gst-libs/gst/audio/gstringbuffer.h:
15086 No trailing commas in enum list (for gcc-2.9x).
15087 * gst/videotestsrc/videotestsrc.c: (random_char):
15088 Make information loss explicit instead of implicitly truncating to
15089 eight bits via the return value. Fixes runtime error on MSVC when
15090 using the debug CRT (#492114).
15091 * win32/common/config.h.in:
15092 Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
15093 * win32/common/libgstinterfaces.def:
15094 * win32/common/libgstrtp.def:
15095 Export a few more symbols (#492114).
15097 2007-11-01 08:06:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15099 gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
15100 Original commit message from CVS:
15101 * gst-libs/gst/audio/audio.c:
15102 * gst-libs/gst/audio/audio.h:
15103 Readd the deprecation guards, but preserve compilability.
15105 2007-10-31 17:54:48 +0000 Tim-Philipp Müller <tim@centricular.net>
15107 gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
15108 Original commit message from CVS:
15109 * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
15110 (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
15111 Preserve channel layout when fixating the number of channels in the
15112 output caps, or make sure there's a suitable channel position layout
15113 set on the caps if required. Fixes #430677.
15115 2007-10-31 17:32:22 +0000 Tim-Philipp Müller <tim@centricular.net>
15117 tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
15118 Original commit message from CVS:
15119 * tests/check/elements/decodebin.c: (test_text_plain_streams):
15120 Make sure the pipeline really operates in push mode as it should
15123 2007-10-31 15:30:15 +0000 Tim-Philipp Müller <tim@centricular.net>
15125 gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
15126 Original commit message from CVS:
15127 * gst-libs/gst/audio/audio.h:
15128 Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
15129 compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
15130 (ie. normal cvs builds) will fail.
15132 2007-10-31 12:47:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15134 tell gtk-doc about the deprecation guard. Apply more doc fixes.
15135 Original commit message from CVS:
15136 * docs/libs/Makefile.am:
15137 * gst-libs/gst/audio/audio.c:
15138 * gst-libs/gst/audio/audio.h:
15139 * gst-libs/gst/interfaces/mixer.c:
15140 tell gtk-doc about the deprecation guard. Apply more doc fixes.
15142 2007-10-31 12:30:28 +0000 Tim-Philipp Müller <tim@centricular.net>
15144 tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
15145 Original commit message from CVS:
15146 * tests/check/libs/audio.c: (init_value_to_channel_layout),
15147 (test_channel_layout_value_intersect), (audio_suite):
15148 Add simple unit test to make sure GstValue intersection
15149 of channel layouts works the way I think it does.
15151 2007-10-30 20:32:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15153 Fix the docs according to what gtk-doc complained about.
15154 Original commit message from CVS:
15155 * docs/libs/gst-plugins-base-libs-sections.txt:
15156 * gst-libs/gst/audio/gstaudiofilter.h:
15157 * gst-libs/gst/interfaces/mixer.h:
15158 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15159 * gst-libs/gst/rtp/gstbasertpdepayload.h:
15160 * gst-libs/gst/sdp/gstsdpmessage.c:
15161 Fix the docs according to what gtk-doc complained about.
15163 2007-10-30 19:46:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15165 tests/icles/stress-playbin.c: Fix the build.
15166 Original commit message from CVS:
15167 * tests/icles/stress-playbin.c:
15170 2007-10-30 15:54:46 +0000 Tim-Philipp Müller <tim@centricular.net>
15172 gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
15173 Original commit message from CVS:
15174 * gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
15175 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
15176 Post nice/more useful error message if we don't have a decoder for
15179 2007-10-30 15:07:58 +0000 Wim Taymans <wim.taymans@gmail.com>
15181 gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
15182 Original commit message from CVS:
15183 * gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
15184 Be a bit more useful, unblock the pads after we fired the no-more-pads
15185 signal so that we can use the signal to inspect and connect all pads
15186 without having to keep extra state outside of decodebin.
15188 2007-10-30 15:00:06 +0000 Wim Taymans <wim.taymans@gmail.com>
15190 gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
15191 Original commit message from CVS:
15192 * gst/playback/gsturidecodebin.c:
15193 (gst_uri_decode_bin_autoplug_continue),
15194 (gst_uri_decode_bin_class_init), (no_more_pads_full):
15195 Implement default signal handler so that we return TRUE when nothing is
15198 2007-10-28 11:53:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15200 gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
15201 Original commit message from CVS:
15202 * gst-libs/gst/riff/riff-media.c:
15203 (gst_riff_wavext_add_channel_layout),
15204 (gst_riff_wave_add_default_channel_layout),
15205 (gst_riff_wavext_get_default_channel_mask),
15206 (gst_riff_create_audio_caps):
15207 Use the ALSA channel layout as default for wav files without channel
15208 layout information. This fixes playback of chan-id.wav on 5.1 systems
15209 for example. Also refactor the channel layout setting a bit and add
15210 more default channel orders. Fixes #489010.
15212 2007-10-28 11:46:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15215 Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
15216 Original commit message from CVS:
15217 (gst_riff_wavext_add_channel_layout),
15218 (gst_riff_wave_add_default_channel_layout),
15219 (gst_riff_wavext_get_default_channel_mask),
15220 (gst_riff_create_audio_caps):
15221 Use the ALSA channel layout as default for wav files without channel
15222 layout information. This fixes playback of chan-id.wav on 5.1 systems
15223 for example. Also refactor the channel layout setting a bit and add
15224 more default channel orders. Fixes #489010.
15226 2007-10-26 18:57:33 +0000 Tim-Philipp Müller <tim@centricular.net>
15228 tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
15229 Original commit message from CVS:
15230 * tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
15231 GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
15232 -DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
15235 2007-10-26 12:07:14 +0000 Christian Schaller <uraeus@gnome.org>
15237 * gst-plugins-base.spec.in:
15239 Original commit message from CVS:
15242 2007-10-25 17:36:49 +0000 Wim Taymans <wim.taymans@gmail.com>
15244 gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
15245 Original commit message from CVS:
15246 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
15247 (gst_decode_bin_dispose), (gst_decode_bin_set_caps),
15248 (gst_decode_bin_set_subs_encoding),
15249 (gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
15250 (gst_decode_bin_get_property), (analyze_new_pad):
15251 Move subtitle encoding property to decodebin2 so that it can set the
15252 property value on all elements that it autoplugs and that require it.
15253 Make caps refcounting more consistent in get/set.
15254 * gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
15255 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
15256 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
15257 (gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
15258 (proxy_autoplug_continue_signal),
15259 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
15261 Proxy properties and relevant signals from the internal decodebin.
15262 Make properties MT safe.
15264 2007-10-25 15:10:59 +0000 Tim-Philipp Müller <tim@centricular.net>
15266 gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
15267 Original commit message from CVS:
15268 * gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
15269 * gst-libs/gst/tag/tags.c:
15270 Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
15271 GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
15272 * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
15273 Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
15274 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
15275 (gst_tag_to_vorbis_comments):
15276 Map new SORTNAME tags (these tags aren't even semi-official, so I'm
15277 just mapping everything I found in the wild) (#414539).
15279 2007-10-24 11:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
15281 gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
15282 Original commit message from CVS:
15283 Inspired by patch of: René Stadler <mail at renestadler dot de>
15284 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
15285 (gst_decode_bin_autoplug_continue),
15286 (gst_decode_bin_autoplug_factories),
15287 (gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
15288 (find_compatibles):
15289 * gst/playback/gstplay-marshal.list:
15290 Remove the autoplug-sort signal and replace it with a binding friendly
15291 autoplug-select signal.
15292 Add an autoplug-factories signal that can be used to generate a list of
15293 factories to try to autoplug.
15294 Add the GstPad to the autoplugging signal args as it might be needed to
15295 make a good factory selection.
15296 Fix up the marshallers for this. Fixes #407282.
15298 2007-10-23 14:23:14 +0000 Tim-Philipp Müller <tim@centricular.net>
15300 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...
15301 Original commit message from CVS:
15302 * gst-libs/gst/tag/gsttagdemux.c:
15303 Don't abort with an assertion if we receive a seek event with
15304 a start type of NONE (see launchpad bug #155878).
15306 2007-10-22 10:21:46 +0000 Wim Taymans <wim.taymans@gmail.com>
15308 sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
15309 Original commit message from CVS:
15310 * sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
15311 (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
15312 (gst_ximagesink_change_state), (gst_ximagesink_reset):
15313 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
15314 (gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
15315 (gst_xvimagesink_change_state), (gst_xvimagesink_reset):
15316 Make sure that before we clean up the X resources, we shutdown and join
15318 Also make sure the event thread does not shut down immediatly after
15319 startup because the running variable is not yet correctly set.
15322 2007-10-16 16:48:38 +0000 Wim Taymans <wim.taymans@gmail.com>
15324 gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
15325 Original commit message from CVS:
15326 * gst/playback/gstdecodebin.c: (new_pad), (type_found):
15327 Make the window for a race in typefind and shutting down smaller until
15328 we figure out the right locking here. Avoids #485753 usually.
15329 * gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
15330 Remove unneeded lock causing a race in typefind and shutting down.
15332 * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
15333 Also remove sinks when going to NULL because we might not complete the
15334 state change to PAUSED, causing the PAUSED->READY state change not to
15337 2007-10-16 15:33:31 +0000 Wim Taymans <wim.taymans@gmail.com>
15339 gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
15340 Original commit message from CVS:
15341 * gst-libs/gst/audio/gstbaseaudiosink.c:
15342 (gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
15343 Also explicitly release the ringbuffer when going to NULL because it
15344 is required in the setcaps function, before the state change to PAUSED
15347 2007-10-16 14:58:53 +0000 Tim-Philipp Müller <tim@centricular.net>
15349 tests/icles/: Does what it says on the tin.
15350 Original commit message from CVS:
15351 * tests/icles/.cvsignore:
15352 * tests/icles/Makefile.am:
15353 * tests/icles/stress-playbin.c:
15354 Does what it says on the tin.
15356 2007-10-15 11:38:39 +0000 Wim Taymans <wim.taymans@gmail.com>
15358 gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
15359 Original commit message from CVS:
15360 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
15361 Fix queue negotiation. See #486758.
15363 2007-10-12 10:52:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15365 Actual code change to go along with:
15366 Original commit message from CVS:
15367 Actual code change to go along with:
15368 2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
15369 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
15370 (gst_xvimagesink_xwindow_new),
15371 (gst_xvimagesink_update_colorbalance),
15372 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
15373 Fix handling of some of the X atoms. If the last parameter is True,
15374 XInternAtom won't create the atom if it doesn't exist, and therefore
15375 might return None. This causes X errors on Xv implementations that
15376 don't provide the colour balance attributes.
15378 2007-10-12 10:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15381 Remove stray character from the changelog.
15382 Original commit message from CVS:
15383 Remove stray character from the changelog.
15385 2007-10-12 10:33:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15388 I'm too lazy to comment this
15389 Original commit message from CVS:
15390 *** empty log message ***
15392 2007-10-11 18:24:09 +0000 Tim-Philipp Müller <tim@centricular.net>
15394 Extract vorbis comment LICENSE tags correctly.
15395 Original commit message from CVS:
15396 * gst-libs/gst/tag/gstvorbistag.c:
15397 * tests/check/libs/tag.c:
15398 Extract vorbis comment LICENSE tags correctly.
15400 2007-10-11 16:12:21 +0000 Jason Kivlighn <jkivlighn@gmail.com>
15402 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
15403 Original commit message from CVS:
15404 Patch by: Jason Kivlighn <jkivlighn gmail com>
15405 * gst-libs/gst/tag/gstid3tag.c:
15406 * tests/check/libs/tag.c:
15407 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
15409 2007-10-10 17:01:51 +0000 Tim-Philipp Müller <tim@centricular.net>
15411 gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
15412 Original commit message from CVS:
15413 * gst-libs/gst/tag/gsttagdemux.c:
15414 Don't error out when a buggy downstream element doesn't
15415 handle the newsegment event we send properly (especially
15416 not without posting a meaningful error message on the
15417 bus). See bug #471370 and launchpad bug #136264.
15419 2007-10-10 15:36:56 +0000 Wim Taymans <wim.taymans@gmail.com>
15421 gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
15422 Original commit message from CVS:
15423 * gst-libs/gst/audio/gstbaseaudiosink.c:
15424 (gst_base_audio_sink_drain):
15425 Use new basesink method to make our EOS drain interruptable.
15427 2007-10-10 09:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15429 gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
15430 Original commit message from CVS:
15431 * gst-libs/gst/rtp/gstrtppayloads.c:
15432 Fix silly search-replace oversight.
15434 2007-10-09 09:57:17 +0000 Laurent Glayal <spglegle@yahoo.fr>
15436 gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
15437 Original commit message from CVS:
15438 Patch by: Laurent Glayal <spglegle at yahoo dot fr>
15439 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
15440 (gst_basertppayload_set_outcaps):
15441 Fix caps memleak. Fixes #484989.
15443 2007-10-08 18:04:34 +0000 Wim Taymans <wim.taymans@gmail.com>
15445 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
15446 Original commit message from CVS:
15447 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15448 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
15451 2007-10-08 18:02:53 +0000 Wim Taymans <wim.taymans@gmail.com>
15453 gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
15454 Original commit message from CVS:
15455 * gst-libs/gst/audio/gstbaseaudiosrc.c:
15456 (gst_base_audio_src_create):
15457 Also handle the case where there is no clock set on the audio source,
15458 like in the unit tests.
15460 2007-10-08 17:40:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15462 gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
15463 Original commit message from CVS:
15464 * gst-libs/gst/rtp/gstrtppayloads.c:
15465 Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
15466 to avoid compiler warnings
15468 2007-10-08 17:12:32 +0000 Wim Taymans <wim.taymans@gmail.com>
15470 gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
15471 Original commit message from CVS:
15472 * gst/playback/gstdecodebin.c: (type_found),
15473 (gst_decode_bin_change_state):
15474 * gst/playback/gstdecodebin2.c: (type_found),
15475 (gst_decode_bin_change_state):
15476 Don't disconnect the have_type signal because we never reconnect it
15477 later on. Instead keep a variable to see if we already detected a type.
15479 2007-10-08 10:47:26 +0000 Wim Taymans <wim.taymans@gmail.com>
15481 gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
15482 Original commit message from CVS:
15483 * gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
15484 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
15486 Unlink the signal handler when we found the type, we're not going to do
15487 anything sensible with more type_found signals anyway.
15489 2007-10-08 06:07:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15491 ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
15492 Original commit message from CVS:
15493 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
15494 Use GIO function to get a list of supported URI schemes instead of
15495 hard coding something.
15497 2007-10-06 16:49:55 +0000 Tim-Philipp Müller <tim@centricular.net>
15499 gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
15500 Original commit message from CVS:
15501 * gst-libs/gst/tag/gsttagdemux.c:
15504 2007-10-06 15:04:53 +0000 Tim-Philipp Müller <tim@centricular.net>
15506 gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
15507 Original commit message from CVS:
15508 * gst-libs/gst/tag/Makefile.am:
15509 * gst-libs/gst/tag/gsttagdemux.c:
15510 * gst-libs/gst/tag/gsttagdemux.h:
15511 API: add GstTagDemux base class for simple tag demuxers.
15512 * docs/libs/gst-plugins-base-libs-docs.sgml:
15513 * docs/libs/gst-plugins-base-libs-sections.txt:
15514 Add GstTagDemux to docs.
15516 2007-10-05 07:49:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15518 gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
15519 Original commit message from CVS:
15520 * gst-libs/gst/rtp/gstrtpbuffer.c:
15521 (gst_rtp_buffer_get_payload_subbuffer):
15522 Fix bug introduced with last commit which inverted the logic and
15523 caused all buffers to be dropped. Fixes #483620.
15524 Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
15526 2007-10-04 06:50:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15528 gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
15529 Original commit message from CVS:
15530 * gst-libs/gst/rtp/gstrtpbuffer.c:
15531 Replace g_return_if_val (as it could be disabled), with regular return
15534 2007-10-03 14:51:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15536 tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
15537 Original commit message from CVS:
15538 * tests/check/pipelines/simple-launch-lines.c:
15539 Print message name and not just number.
15541 2007-10-02 11:11:13 +0000 Wim Taymans <wim.taymans@gmail.com>
15543 gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
15544 Original commit message from CVS:
15545 * gst-libs/gst/audio/gstbaseaudiosink.c:
15546 (gst_base_audio_sink_async_play):
15547 When slaved to the clock, don't try to align a sample with the previous
15548 one when going to PLAYING again.
15550 2007-10-02 09:04:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15552 tests/examples/snapshot/snapshot.c: Fix the build.
15553 Original commit message from CVS:
15554 * tests/examples/snapshot/snapshot.c:
15557 2007-10-02 07:43:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15559 ext/gio/gstgiosink.c: Update to API changes in GIO.
15560 Original commit message from CVS:
15561 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
15562 Update to API changes in GIO.
15564 2007-10-01 16:33:00 +0000 Wim Taymans <wim.taymans@gmail.com>
15566 gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
15567 Original commit message from CVS:
15568 * gst-libs/gst/sdp/gstsdpmessage.h:
15569 Add RFC 3556 bandwidth modifiers.
15571 2007-10-01 13:37:31 +0000 Wim Taymans <wim.taymans@gmail.com>
15573 Update documentation.
15574 Original commit message from CVS:
15575 * docs/libs/gst-plugins-base-libs-docs.sgml:
15576 * docs/libs/gst-plugins-base-libs-sections.txt:
15577 * gst-libs/gst/rtp/gstrtppayloads.c:
15578 Update documentation.
15580 2007-10-01 13:22:14 +0000 Wim Taymans <wim.taymans@gmail.com>
15582 gst-libs/gst/rtp/: Added new file and header to deal with payload info.
15583 Original commit message from CVS:
15584 * gst-libs/gst/rtp/Makefile.am:
15585 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
15586 (gst_rtp_payload_info_for_name):
15587 * gst-libs/gst/rtp/gstrtppayloads.h:
15588 Added new file and header to deal with payload info.
15589 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
15590 (gst_rtp_buffer_default_clock_rate):
15591 * gst-libs/gst/rtp/gstrtpbuffer.h:
15592 Payload specific stuff is move to new headers.
15593 Implement _default_clock rate using the new payload function.
15594 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
15595 (gst_sdp_parse_line):
15596 * gst-libs/gst/sdp/gstsdpmessage.h:
15597 Add some more comments.
15599 2007-10-01 10:22:46 +0000 Wim Taymans <wim.taymans@gmail.com>
15601 gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
15602 Original commit message from CVS:
15603 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
15604 (sdp_check_header), (sdp_type_find), (plugin_init):
15605 Add typefind function for application/sdp.
15606 Remove some old dirac typefind code that was ifdeffed out.
15608 2007-09-29 12:04:02 +0000 Sébastien Moutte <sebastien@moutte.net>
15610 win32/common/libgstaudio.def: Add new exported functions.
15611 Original commit message from CVS:
15612 * win32/common/libgstaudio.def:
15613 Add new exported functions.
15614 * win32/vs6/grammar.dsp:
15615 Add autogeneration and copy of some autegenerated files from win32/common
15617 * win32/vs6/libgstaudioconvert.dsp:
15618 Add gstaudioquantize.c to the build.
15619 * win32/vs6/libgstinterfaces.dsp:
15620 Add videoorientation.c to the build.
15621 * win32/vs6/libgstriff.dsp:
15622 Add libgsttag to the link libraries list.
15623 * win32/vs6/libgstvolume.dsp:
15624 Add liboil to the link.
15625 * win32/vs6/gst_plugins_base.dsw:
15626 * win32/vs6/libgstrtsp.dsp:
15627 * win32/common/libgstrtsp.def:
15628 Add files to build libgstrtsp library.
15630 2007-09-29 07:01:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15632 ext/gio/: Some minor cleanup and allow setting the location only when the element is not playing or paused.
15633 Original commit message from CVS:
15634 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
15635 (gst_gio_sink_set_property), (gst_gio_sink_render):
15636 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
15637 (gst_gio_src_set_property):
15638 Some minor cleanup and allow setting the location only when the
15639 element is not playing or paused.
15641 2007-09-26 15:14:37 +0000 Wim Taymans <wim.taymans@gmail.com>
15643 tests/examples/snapshot/snapshot.c: Print error when pipeline failed to construct.
15644 Original commit message from CVS:
15645 * tests/examples/snapshot/snapshot.c: (main):
15646 Print error when pipeline failed to construct.
15648 2007-09-25 19:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
15650 Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
15651 Original commit message from CVS:
15653 * gst-libs/gst/tag/gstid3tag.c:
15654 * gst-libs/gst/tag/gstvorbistag.c:
15655 Add mappings for the new GST_TAG_COMPOSER for vorbis comments
15658 2007-09-25 11:54:09 +0000 Tim-Philipp Müller <tim@centricular.net>
15660 gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
15661 Original commit message from CVS:
15662 * gst-libs/gst/floatcast/floatcast.h:
15663 Don't include config.h in an installed public header, this
15664 might break compilation of applications that don't have such
15665 a header and doesn't necessarily do what it's supposed to do
15666 anyway (ie. check for the lrint/lrintf defines) (#442065).
15667 Add docs for the various macros and document how this header
15668 has to be used (link against libm, etc.); add a few FIXMEs;
15669 include math.h for non-c99 code path. Based on patch by
15672 2007-09-25 07:50:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15674 configure.ac: Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead of duplicating these macros in confi...
15675 Original commit message from CVS:
15677 Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
15678 of duplicating these macros in configure.ac.
15680 2007-09-22 17:58:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15682 po/: Updated translations to 0.10.14
15683 Original commit message from CVS:
15687 Updated translations to 0.10.14
15689 2007-09-22 17:57:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15693 Original commit message from CVS:
15696 2007-09-22 17:56:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15698 po/pl.po: Added Polish translation.
15699 Original commit message from CVS:
15700 translated by: Jakub Bogusz <qboosh@pld-linux.org>
15702 Added Polish translation.
15704 2007-09-22 17:55:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15706 po/fi.po: Added Finnish translation.
15707 Original commit message from CVS:
15708 translated by: Ilkka Tuohela <hile@iki.fi>
15710 Added Finnish translation.
15712 2007-09-22 17:54:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15714 po/es.po: Added Spanish translation.
15715 Original commit message from CVS:
15716 translated by: Jorge González González <aloriel@gmail.com>
15718 Added Spanish translation.
15720 2007-09-22 17:53:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15722 po/da.po: Added Danish translation.
15723 Original commit message from CVS:
15724 translated by: Mogens Jaeger <mogens@jaeger.tf>
15726 Added Danish translation.
15728 2007-09-22 17:52:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15730 po/zh_CN.po: Added Chinese (simplified) translation.
15731 Original commit message from CVS:
15732 translated by: Funda Wang <fundawang@linux.net.cn>
15734 Added Chinese (simplified) translation.
15736 2007-09-22 17:51:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15738 po/bg.po: Added Bulgarian translation.
15739 Original commit message from CVS:
15740 translated by: Alexander Shopov <ash@contact.bg>
15742 Added Bulgarian translation.
15744 2007-09-21 18:00:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15746 docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy.
15747 Original commit message from CVS:
15748 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
15750 * ext/gio/gstgiosink.h:
15751 * ext/gio/gstgiosrc.h:
15752 Mark private fields of the instance structs private.
15754 2007-09-21 17:31:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15756 docs/plugins/: Add the GIO plugin to the docs and do a make update while doing that.
15757 Original commit message from CVS:
15758 * docs/plugins/Makefile.am:
15759 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
15760 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
15761 * docs/plugins/gst-plugins-bad-plugins.args:
15762 * docs/plugins/gst-plugins-bad-plugins.signals:
15763 * docs/plugins/inspect/plugin-bz2.xml:
15764 * docs/plugins/inspect/plugin-cdxaparse.xml:
15765 * docs/plugins/inspect/plugin-dfbvideosink.xml:
15766 * docs/plugins/inspect/plugin-dtsdec.xml:
15767 * docs/plugins/inspect/plugin-equalizer.xml:
15768 * docs/plugins/inspect/plugin-faac.xml:
15769 * docs/plugins/inspect/plugin-faad.xml:
15770 * docs/plugins/inspect/plugin-filter.xml:
15771 * docs/plugins/inspect/plugin-freeze.xml:
15772 * docs/plugins/inspect/plugin-gio.xml:
15773 * docs/plugins/inspect/plugin-gsm.xml:
15774 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
15775 * docs/plugins/inspect/plugin-h264parse.xml:
15776 * docs/plugins/inspect/plugin-modplug.xml:
15777 * docs/plugins/inspect/plugin-mpeg2enc.xml:
15778 * docs/plugins/inspect/plugin-musepack.xml:
15779 * docs/plugins/inspect/plugin-musicbrainz.xml:
15780 * docs/plugins/inspect/plugin-nsfdec.xml:
15781 * docs/plugins/inspect/plugin-replaygain.xml:
15782 * docs/plugins/inspect/plugin-soundtouch.xml:
15783 * docs/plugins/inspect/plugin-spcdec.xml:
15784 * docs/plugins/inspect/plugin-spectrum.xml:
15785 * docs/plugins/inspect/plugin-speed.xml:
15786 * docs/plugins/inspect/plugin-tta.xml:
15787 * docs/plugins/inspect/plugin-videosignal.xml:
15788 * docs/plugins/inspect/plugin-xingheader.xml:
15789 * docs/plugins/inspect/plugin-xvid.xml:
15790 Add the GIO plugin to the docs and do a make update
15792 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
15793 Fix a small memleak.
15795 2007-09-21 17:07:56 +0000 René Stadler <mail@renestadler.de>
15797 Add a GIO/GVFS plugin with source and sink elements. This will only be enabled when --enable-experimental is given to...
15798 Original commit message from CVS:
15799 Patch by: René Stadler <mail at renestadler dot de>
15802 * ext/gio/Makefile.am:
15803 * ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek),
15804 (gst_gio_get_supported_protocols),
15805 (gst_gio_uri_handler_get_type_sink),
15806 (gst_gio_uri_handler_get_type_src),
15807 (gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri),
15808 (gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init),
15809 (gst_gio_uri_handler_do_init), (plugin_init):
15810 * ext/gio/gstgio.h:
15811 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
15812 (gst_gio_sink_class_init), (gst_gio_sink_init),
15813 (gst_gio_sink_finalize), (gst_gio_sink_set_property),
15814 (gst_gio_sink_get_property), (gst_gio_sink_start),
15815 (gst_gio_sink_stop), (gst_gio_sink_unlock),
15816 (gst_gio_sink_unlock_stop), (gst_gio_sink_event),
15817 (gst_gio_sink_render), (gst_gio_sink_query):
15818 * ext/gio/gstgiosink.h:
15819 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
15820 (gst_gio_src_class_init), (gst_gio_src_init),
15821 (gst_gio_src_finalize), (gst_gio_src_set_property),
15822 (gst_gio_src_get_property), (gst_gio_src_start),
15823 (gst_gio_src_stop), (gst_gio_src_get_size),
15824 (gst_gio_src_is_seekable), (gst_gio_src_unlock),
15825 (gst_gio_src_unlock_stop), (gst_gio_src_check_get_range),
15826 (gst_gio_src_create):
15827 * ext/gio/gstgiosrc.h:
15828 Add a GIO/GVFS plugin with source and sink elements. This will
15829 only be enabled when --enable-experimental is given to configure
15830 for now as the GIO API is not stable yet. Fixes #476916.
15832 2007-09-21 14:37:26 +0000 Wim Taymans <wim.taymans@gmail.com>
15834 gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
15835 Original commit message from CVS:
15836 * gst/playback/gstqueue2.c: (gst_queue_push_one):
15837 Fix compilation wrt printf arguments.
15839 2007-09-20 17:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
15841 examples/app/appsrc_ex.c: Fix compilation after changing the name of a method.
15842 Original commit message from CVS:
15843 * examples/app/appsrc_ex.c: (main):
15844 Fix compilation after changing the name of a method.
15846 2007-09-20 14:09:24 +0000 Wim Taymans <wim.taymans@gmail.com>
15848 Add simple snapshot example program using appsink.
15849 Original commit message from CVS:
15851 * tests/examples/Makefile.am:
15852 * tests/examples/snapshot/.cvsignore:
15853 * tests/examples/snapshot/Makefile.am:
15854 * tests/examples/snapshot/snapshot.c: (main):
15855 Add simple snapshot example program using appsink.
15857 2007-09-20 13:59:50 +0000 Wim Taymans <wim.taymans@gmail.com>
15859 gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
15860 Original commit message from CVS:
15861 * gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
15862 (gst_app_sink_class_init), (gst_app_sink_init),
15863 (gst_app_sink_dispose), (gst_app_sink_finalize),
15864 (gst_app_sink_set_property), (gst_app_sink_get_property),
15865 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
15866 (gst_app_sink_event), (gst_app_sink_getcaps),
15867 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
15868 (gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
15869 (gst_app_sink_pull_buffer):
15870 * gst-libs/gst/app/gstappsink.h:
15871 Add properties, signals and actions to access the element even without
15872 linking to the library.
15873 Fix some method names and signatures.
15875 2007-09-20 10:37:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15877 tests/check/generic/states.c: Improved state change unit test.
15878 Original commit message from CVS:
15879 * tests/check/generic/states.c:
15880 Improved state change unit test.
15882 2007-09-19 18:16:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15884 Ignore registries in any format.
15885 Original commit message from CVS:
15886 * docs/plugins/.cvsignore:
15887 * tests/check/.cvsignore:
15888 Ignore registries in any format.
15890 2007-09-19 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
15892 gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
15893 Original commit message from CVS:
15894 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15895 (gst_base_rtp_depayload_chain),
15896 (gst_base_rtp_depayload_set_gst_timestamp):
15897 Only copy timestamp on outgoing packets if the depayloader did not set
15899 Also copy duration on outgoing packets.
15901 2007-09-19 15:55:08 +0000 Wim Taymans <wim.taymans@gmail.com>
15903 gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
15904 Original commit message from CVS:
15905 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
15906 (gst_basertppayload_set_outcaps):
15907 Fix compilation because of missing %d in printf.
15908 When fixating caps, fixate what we can and throw away all remaining
15909 unfixed caps, subclasses should do something smart if they need to.
15911 2007-09-19 12:04:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15913 ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
15914 Original commit message from CVS:
15915 * ext/gnomevfs/gstgnomevfssrc.c:
15916 Improve debug logs a bit and be more verbose if things go wrong.
15918 2007-09-17 17:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15920 Fix a bunch of compile warnings shown with Forte.
15921 Original commit message from CVS:
15922 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
15923 (gst_text_overlay_set_property):
15924 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
15925 * gst-libs/gst/audio/gstbaseaudiosink.c:
15926 (gst_base_audio_sink_render):
15927 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
15928 (gst_rtcp_unix_to_ntp):
15929 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
15930 * gst/playback/gstqueue2.c:
15931 * tests/examples/seek/seek.c: (set_scale):
15932 Fix a bunch of compile warnings shown with Forte.
15933 * gst/audiorate/gstaudiorate.c:
15934 Always pull in config.h before including any system headers.
15936 2007-09-17 16:22:17 +0000 Wim Taymans <wim.taymans@gmail.com>
15938 gst/playback/gstqueue2.c: Also fix #476514 for queue2.
15939 Original commit message from CVS:
15940 * gst/playback/gstqueue2.c: (update_buffering),
15941 (gst_queue_locked_flush), (gst_queue_locked_enqueue),
15942 (gst_queue_handle_sink_event), (gst_queue_chain),
15943 (gst_queue_push_one), (gst_queue_sink_activate_push),
15944 (gst_queue_src_activate_push), (gst_queue_src_activate_pull):
15945 Also fix #476514 for queue2.
15947 2007-09-16 19:31:06 +0000 Wim Taymans <wim.taymans@gmail.com>
15949 gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
15950 Original commit message from CVS:
15951 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15952 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
15953 (gst_base_rtp_depayload_chain),
15954 (gst_base_rtp_depayload_handle_sink_event),
15955 (gst_base_rtp_depayload_push_full),
15956 (gst_base_rtp_depayload_set_gst_timestamp),
15957 (gst_base_rtp_depayload_change_state):
15958 Remove code to deal with RTP to GST time conversion, we now just copy
15959 the GST timestamp we receive to the outgoing buffers.
15960 Handle segment and flushes correctly.
15961 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
15962 When we have no valid input timestamp, use the previous rtp timestamp on
15963 the outgoing RTP packet instead of the RTP base time.
15965 2007-09-16 01:56:21 +0000 David Schleef <ds@schleef.org>
15967 ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
15968 Original commit message from CVS:
15969 * ext/alsa/gstalsa.c:
15970 * ext/alsa/gstalsadeviceprobe.c:
15971 * ext/alsa/gstalsamixer.c:
15972 * ext/alsa/gstalsasink.c:
15973 * ext/alsa/gstalsasrc.c:
15974 Change alsa alloca's to malloc to fix warnings on gcc-4.2.
15976 2007-09-15 18:41:27 +0000 Wim Taymans <wim.taymans@gmail.com>
15978 gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
15979 Original commit message from CVS:
15980 * gst-libs/gst/rtp/gstbasertppayload.c:
15981 (gst_basertppayload_set_outcaps), (gst_basertppayload_push):
15982 Add some debug info when negotiating caps.
15984 2007-09-15 00:29:11 +0000 Wim Taymans <wim.taymans@gmail.com>
15986 gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
15987 Original commit message from CVS:
15988 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
15989 A buffer with an empty payload is also a valid buffer.
15991 2007-09-14 20:52:00 +0000 Wim Taymans <wim.taymans@gmail.com>
15993 gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
15994 Original commit message from CVS:
15995 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
15996 (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
15997 (gst_basertppayload_change_state):
15998 Make sure we start our RTP timestamp from the random base RTP
15999 timestamp even if the buffer timestamp starts from some random value.
16001 2007-09-14 16:56:16 +0000 Wim Taymans <wim.taymans@gmail.com>
16003 Add simple exmple app to demonstrate starting and pausing live and non-live bins in a PLAYING pipeline.
16004 Original commit message from CVS:
16006 * tests/examples/Makefile.am:
16007 * tests/examples/dynamic/.cvsignore:
16008 * tests/examples/dynamic/Makefile.am:
16009 * tests/examples/dynamic/addstream.c: (create_stream),
16010 (pause_play_stream), (message_received), (eos_message_received),
16011 (perform_step), (main):
16012 Add simple exmple app to demonstrate starting and pausing live and
16013 non-live bins in a PLAYING pipeline.
16015 2007-09-14 10:42:00 +0000 Julien Moutte <julien@moutte.net>
16017 gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
16018 Original commit message from CVS:
16019 2007-09-14 Julien MOUTTE <julien@moutte.net>
16020 * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
16021 typefind for QCP files (RFC #3625)
16023 2007-09-13 22:52:09 +0000 Wim Taymans <wim.taymans@gmail.com>
16025 gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
16026 Original commit message from CVS:
16027 * gst-libs/gst/audio/gstbaseaudiosink.c:
16028 (gst_base_audio_sink_init):
16029 Disable pull mode scheduling, we're not ready for it yet and it subtly
16030 breaks a lot of things.
16032 2007-09-12 17:35:52 +0000 Tim-Philipp Müller <tim@centricular.net>
16034 tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
16035 Original commit message from CVS:
16036 * tests/check/elements/libvisual.c:
16037 Test all libvisual plugins, not just the first one; this reproduces
16038 bug #450336 quite easily. Looks like a problem with the 'jess'
16041 2007-09-12 17:15:12 +0000 Tim-Philipp Müller <tim@centricular.net>
16043 tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
16044 Original commit message from CVS:
16045 * tests/check/Makefile.am:
16046 * tests/check/elements/.cvsignore:
16047 * tests/check/elements/libvisual.c:
16048 Add basic libvisual test case in an attempt to reproduce bug #450336.
16049 Doesn't reproduce that bug, but some other crasher instead (invalid
16050 free), at least with make elements/libvisual.forever and the bumscope
16051 plugin on x86-64/gutsy. Leaving test disabled for now.
16053 2007-09-12 08:38:21 +0000 Peter Kjellerstedt <pkj@axis.com>
16055 gst/: Printf format fixes (#476128).
16056 Original commit message from CVS:
16057 Patch by: Peter Kjellerstedt <pkj at axis com>
16058 * gst-libs/gst/app/gstappsink.c:
16059 * gst/flv/gstflvdemux.c:
16060 * gst/flv/gstflvparse.c:
16061 * gst/interleave/deinterleave.c:
16062 * gst/switch/gstswitch.c:
16063 Printf format fixes (#476128).
16065 2007-09-11 19:07:57 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
16067 gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
16068 Original commit message from CVS:
16069 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
16070 * gst-libs/gst/rtsp/gstrtspconnection.c:
16071 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
16072 (read_body), (gst_rtsp_connection_receive):
16073 Make sure we can not cancel in the middle of receiving a message.
16076 2007-09-11 11:29:12 +0000 Josep Torra Valles <josep@fluendo.com>
16078 gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
16079 Original commit message from CVS:
16080 Patch by: Josep Torra Valles <josep@fluendo.com>
16081 * gst/playback/gstplaybasebin.c:
16082 Increase upper limit for audio queue a bit; fixes preroll problem
16083 with playbin and decodebin2 when playing a quicktime trailer with
16084 multichannel audio via http (#464666).
16086 2007-09-10 22:10:54 +0000 Wim Taymans <wim.taymans@gmail.com>
16088 gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
16089 Original commit message from CVS:
16090 * gst-libs/gst/audio/gstbaseaudiosrc.c:
16091 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
16092 (gst_base_audio_src_provide_clock),
16093 (gst_base_audio_src_set_property),
16094 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
16095 * gst-libs/gst/audio/gstbaseaudiosrc.h:
16096 Allow othe clocks than the internal clock to be used for the pipeline.
16097 Add property to disable clock provide.
16098 API: GstBaseAudioSrc::provide-clock
16100 2007-09-10 12:05:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16102 gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
16103 Original commit message from CVS:
16104 * gst/playback/gstdecodebin2.c:
16105 Don't leak request pads. Fixes #475395.
16107 2007-09-09 10:25:43 +0000 René Stadler <mail@renestadler.de>
16109 sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
16110 Original commit message from CVS:
16111 Patch by: René Stadler <mail at renestadler dot de>
16112 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
16113 (gst_ximage_buffer_class_init):
16114 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
16115 (gst_xvimage_buffer_class_init):
16116 Correctly chain up finalize with the parent class to prevent
16117 memory leaks. Fixes #474880.
16119 2007-09-09 04:08:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16121 Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
16122 Original commit message from CVS:
16123 * gst/volume/gstvolume.c: (volume_choose_func):
16124 * tests/check/elements/volume.c: (GST_START_TEST):
16125 Revert the latest change: floating point samples are allowed to
16126 have any value, not only values in the range [-1,1]. Thanks to Andy
16127 Wingo for noticing.
16128 Also fix processing of int32 samples with volumes > 4 by making the
16129 unity value smaller which prevents overflows.
16131 2007-09-07 17:37:03 +0000 Tim-Philipp Müller <tim@centricular.net>
16133 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
16134 Original commit message from CVS:
16135 * gst-libs/gst/rtp/gstrtpbuffer.c:
16136 * tests/check/libs/rtp.c:
16137 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
16139 2007-09-07 16:46:05 +0000 Haakon Sporsheim <haakon.sporsheim@tandberg.com>
16141 gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
16142 Original commit message from CVS:
16143 Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
16144 * gst-libs/gst/rtp/gstrtpbuffer.c:
16145 Fix up GstRTPHeader helper struct so that compilers will not under
16146 any circumstances add padding in between our fields, as currently
16147 happens with MSVC on win32, because that would lead to us sending
16148 out RTP payloads with broken RTP headers (#471194).
16149 Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
16150 * tests/check/Makefile.am:
16151 * tests/check/libs/.cvsignore:
16152 * tests/check/libs/rtp.c:
16153 Add some simple unit tests for GstRTPBuffer. Some are disabled
16154 because the code tested still needs fixing (set_csrc() does not work).
16156 2007-09-07 15:05:24 +0000 Christian Schaller <uraeus@gnome.org>
16158 * gst-plugins-base.spec.in:
16159 update spec file to include latest RTSP libraries and headers and more
16160 Original commit message from CVS:
16161 update spec file to include latest RTSP libraries and headers and more
16163 2007-09-07 12:41:01 +0000 Tim-Philipp Müller <tim@centricular.net>
16165 win32/: Add rtsp enumtypes (#474384) and update others.
16166 Original commit message from CVS:
16168 * win32/common/gstrtsp-enumtypes.c:
16169 * win32/common/gstrtsp-enumtypes.h:
16170 * win32/common/interfaces-enumtypes.c:
16171 * win32/common/interfaces-enumtypes.h:
16172 * win32/common/multichannel-enumtypes.c:
16173 Add rtsp enumtypes (#474384) and update others.
16175 2007-09-06 20:31:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16177 configure.ac: Fix configure check for HAVE_LIBXML_HTML.
16178 Original commit message from CVS:
16180 Fix configure check for HAVE_LIBXML_HTML.
16182 2007-09-06 12:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
16184 tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
16185 Original commit message from CVS:
16186 * tests/check/libs/.cvsignore:
16187 Ignore more, in case the build bots work again one day.
16189 2007-09-06 07:00:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16191 Add libgstfft, a FFT library based on Kiss FFT which is
16192 Original commit message from CVS:
16193 Reviewed by: Stefan Kost <ensonic@users.sf.net>
16195 * gst-libs/gst/Makefile.am:
16196 * gst-libs/gst/fft/Makefile.am:
16197 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
16198 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
16199 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
16200 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
16201 * gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
16202 * gst-libs/gst/fft/gstfft.h:
16203 * gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
16204 (gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
16205 (gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
16206 * gst-libs/gst/fft/gstfftf32.h:
16207 * gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
16208 (gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
16209 (gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
16210 * gst-libs/gst/fft/gstfftf64.h:
16211 * gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
16212 (gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
16213 (gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
16214 * gst-libs/gst/fft/gstffts16.h:
16215 * gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
16216 (gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
16217 (gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
16218 * gst-libs/gst/fft/gstffts32.h:
16219 * gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
16220 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16221 (kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
16222 (kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
16223 * gst-libs/gst/fft/kiss_fft_f32.h:
16224 * gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
16225 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16226 (kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
16227 (kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
16228 * gst-libs/gst/fft/kiss_fft_f64.h:
16229 * gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
16230 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16231 (kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
16232 (kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
16233 * gst-libs/gst/fft/kiss_fft_s16.h:
16234 * gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
16235 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16236 (kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
16237 (kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
16238 * gst-libs/gst/fft/kiss_fft_s32.h:
16239 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
16240 (kiss_fftr_f32), (kiss_fftri_f32):
16241 * gst-libs/gst/fft/kiss_fftr_f32.h:
16242 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
16243 (kiss_fftr_f64), (kiss_fftri_f64):
16244 * gst-libs/gst/fft/kiss_fftr_f64.h:
16245 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
16246 (kiss_fftr_s16), (kiss_fftri_s16):
16247 * gst-libs/gst/fft/kiss_fftr_s16.h:
16248 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
16249 (kiss_fftr_s32), (kiss_fftri_s32):
16250 * gst-libs/gst/fft/kiss_fftr_s32.h:
16251 * gst-libs/gst/fft/kiss_version:
16252 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
16253 * pkgconfig/gstreamer-plugins-base.pc.in:
16254 Add libgstfft, a FFT library based on Kiss FFT which is
16255 BSD licensed. Supported sample formats are int16, int32,
16256 float and double. For those formats a real FFT and IFFT
16257 can be done, different windowing functions can be applied
16258 and functions for extracting the magnitude and phase exist.
16260 * docs/libs/Makefile.am:
16261 * docs/libs/gst-plugins-base-libs-docs.sgml:
16262 * docs/libs/gst-plugins-base-libs-sections.txt:
16263 Integrate libgstfft into the docs.
16264 * tests/check/Makefile.am:
16265 * tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
16266 Add unit tests for libgstfft, currently only testing the FFT.
16267 Unit tests for IFFT will follow soon.
16269 2007-09-05 23:07:40 +0000 Peter Kjellerstedt <pkj@axis.com>
16271 gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
16272 Original commit message from CVS:
16273 Patch by: Peter Kjellerstedt <pkj at axis com>
16274 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
16275 (gst_sdp_message_init), (gst_sdp_message_uninit),
16276 (is_multicast_address), (gst_sdp_message_as_text),
16277 (gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
16278 (gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
16279 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
16280 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
16281 (gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
16282 (gst_sdp_media_init), (gst_sdp_media_uninit),
16283 (gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
16284 (gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
16285 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
16286 (gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
16287 (gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
16288 * gst-libs/gst/sdp/gstsdpmessage.h:
16289 Separate INIT_ARRAY() and related macros into two versions, one for
16290 structures and one for pointers (e.g., INIT_ARRAY() and
16291 INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
16292 lists of emails and phone numbers.
16293 Add missing const as appropriate.
16294 Change all gint to guint since they all actually represent unsigned
16296 Do not use time as a variable name as it shadows the global time().
16297 Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
16298 Actually implement gst_sdp_message_add_time().
16299 Make gst_sdp_message_add_time() take repeat times as an argument.
16300 Store repeat times in GstSDPTime as a GArray rather than as gchar**.
16301 Corrected the definition of gst_sdp_media_get_bandwidth() (was
16302 misspelled as badwidth).
16303 gst-indented and a little clean up. Fixes #471067.
16305 2007-09-05 21:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16307 gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
16308 Original commit message from CVS:
16309 * gst/volume/gstvolume.c: (volume_choose_func),
16310 (volume_process_double), (volume_process_double_clamp),
16311 (volume_process_float_clamp):
16312 Correctly clamp float/double samples in the [-1.0,1.0] range to
16313 prevent weird effects.
16314 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
16315 Add unit tests for all samples types that had none before.
16317 2007-09-05 14:09:15 +0000 Tim-Philipp Müller <tim@centricular.net>
16319 gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
16320 Original commit message from CVS:
16321 * gst-libs/gst/rtp/gstrtpbuffer.c:
16322 Need to include stdlib.h for abs() here too.
16324 2007-09-05 14:01:25 +0000 Tim-Philipp Müller <tim@centricular.net>
16326 gst/playback/gststreaminfo.c: Fix build.
16327 Original commit message from CVS:
16328 * gst/playback/gststreaminfo.c:
16331 2007-09-05 10:32:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16333 gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
16334 Original commit message from CVS:
16335 * gst/playback/gststreaminfo.c:
16336 Clean up some half-disabled code and comment.
16338 2007-09-04 16:18:48 +0000 Wim Taymans <wim.taymans@gmail.com>
16340 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
16341 Original commit message from CVS:
16342 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
16343 (gst_base_rtp_payload_audio_handle_event):
16344 Return FALSE from the event handler to let the parent class handle the
16346 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16347 (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
16348 Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
16349 * gst-libs/gst/rtp/gstbasertppayload.c:
16350 Bump the MTU to 1400.
16352 2007-09-04 01:50:55 +0000 Johan Dahlin <johan@gnome.org>
16354 gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
16355 Original commit message from CVS:
16356 2007-09-03 Johan Dahlin <jdahlin@async.com.br>
16357 * gst/typefind/gsttypefindfunctions.c (plugin_init):
16358 Add an audio/x-nsf typefind function for the nsfdec element.
16360 2007-09-03 20:46:38 +0000 Renato Filho <renato.filho@indt.org.br>
16362 gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
16363 Original commit message from CVS:
16364 * gst/playback/gstplaybasebin.c:
16365 Included "myth://" on stream_uris list for enable buffering to mythtv files
16367 2007-09-03 19:31:11 +0000 Wim Taymans <wim.taymans@gmail.com>
16369 Fix parsing of RB blocks.
16370 Original commit message from CVS:
16371 * docs/libs/gst-plugins-base-libs-sections.txt:
16372 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
16373 (gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
16374 (gst_rtcp_unix_to_ntp):
16375 * gst-libs/gst/rtp/gstrtcpbuffer.h:
16376 Fix parsing of RB blocks.
16378 Added helper functions to convert to/from UNIX and NTP time.
16379 API: gst_rtcp_ntp_to_unix()
16380 API: gst_rtcp_unix_to_ntp()
16381 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
16382 (gst_rtp_buffer_get_header_len),
16383 (gst_rtp_buffer_get_extension_data),
16384 (gst_rtp_buffer_get_payload_subbuffer),
16385 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
16386 (gst_rtp_buffer_ext_timestamp):
16387 * gst-libs/gst/rtp/gstrtpbuffer.h:
16388 Fix some more docs.
16389 Implement handling of packets with extensions.
16390 Fix padding check in _validate().
16391 Added function to get extension data.
16392 API: gst_rtp_buffer_get_header_len()
16393 API: gst_rtp_buffer_get_extension_data()
16395 2007-09-03 19:19:35 +0000 Wim Taymans <wim.taymans@gmail.com>
16397 gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
16398 Original commit message from CVS:
16399 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16400 (gst_base_rtp_depayload_class_init),
16401 (gst_base_rtp_depayload_set_gst_timestamp):
16402 Add some more docs for the queue-delay property and fix a typo in a
16404 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
16407 2007-09-03 19:17:33 +0000 Wim Taymans <wim.taymans@gmail.com>
16409 gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
16410 Original commit message from CVS:
16411 * gst-libs/gst/audio/gstbaseaudiosink.c:
16412 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
16413 (gst_base_audio_sink_change_state):
16414 When skew slaving, try to hover around the middle of a segment so that
16415 we at most drift by half a segment.
16416 If we are aligning in the oposite direction of the clock skew, we don't
16419 2007-08-31 21:07:20 +0000 Wim Taymans <wim.taymans@gmail.com>
16421 gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
16422 Original commit message from CVS:
16423 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16424 (gst_base_rtp_depayload_setcaps),
16425 (gst_base_rtp_depayload_set_gst_timestamp):
16426 Be less silly with the segment start, just apply the clock-base to the
16429 2007-08-31 15:58:30 +0000 Wim Taymans <wim.taymans@gmail.com>
16431 gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
16432 Original commit message from CVS:
16433 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16434 (gst_base_rtp_depayload_class_init),
16435 (gst_base_rtp_depayload_finalize),
16436 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
16437 (gst_base_rtp_depayload_handle_sink_event),
16438 (gst_base_rtp_depayload_set_gst_timestamp),
16439 (gst_base_rtp_depayload_change_state):
16440 * gst-libs/gst/rtp/gstbasertpdepayload.h:
16441 Deprecate the queue handling thread thing and remove the code.
16442 Use new method to calculate the extended timestamp.
16444 2007-08-31 15:21:13 +0000 Wim Taymans <wim.taymans@gmail.com>
16446 gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
16447 Original commit message from CVS:
16448 * gst-libs/gst/rtp/gstrtcpbuffer.c:
16449 (gst_rtcp_packet_sdes_copy_entry):
16450 Use g_strndup which does exactly what we want.
16451 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
16452 (gst_rtp_buffer_ext_timestamp):
16453 * gst-libs/gst/rtp/gstrtpbuffer.h:
16454 Add helper function to compare seqnums.
16455 Add helper function to calculate extended timestamps.
16456 API: gst_rtp_buffer_compare_seqnum()
16457 API: gst_rtp_buffer_ext_timestamp()
16459 2007-08-30 21:59:23 +0000 Wim Taymans <wim.taymans@gmail.com>
16461 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
16462 Original commit message from CVS:
16463 * gst-libs/gst/rtp/gstrtcpbuffer.c:
16464 (gst_rtcp_packet_sdes_get_entry),
16465 (gst_rtcp_packet_sdes_copy_entry):
16466 * gst-libs/gst/rtp/gstrtcpbuffer.h:
16467 Fix and document SDES item data function.
16468 Add new function that makes a proper copy of SDES item data.
16469 API: gst_rtcp_packet_sdes_copy_entry()
16471 2007-08-30 07:29:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16473 The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
16474 Original commit message from CVS:
16477 The tcp and subparse plugins are under gst, but not totaly free of
16478 dependencies. Handle selection inconfigure.ac, so that they show up
16479 on the final list of what is build and what is not. Maybe they should
16480 better be moved to ext.
16482 2007-08-30 06:58:46 +0000 Daniel Díaz <yosoy@danieldiaz.org>
16484 Check if libxml provides HTML parser which subparse needs.
16485 Original commit message from CVS:
16486 Patch by: Daniel Díaz <yosoy@danieldiaz.org>
16489 Check if libxml provides HTML parser which subparse needs.
16492 2007-08-29 14:22:04 +0000 Tim-Philipp Müller <tim@centricular.net>
16494 ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
16495 Original commit message from CVS:
16496 * ext/alsa/gstalsa.c:
16497 Fix typo and compilation on big endian systems.
16499 2007-08-29 12:16:46 +0000 Tim-Philipp Müller <tim@centricular.net>
16501 gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
16502 Original commit message from CVS:
16503 * gst/subparse/gstssaparse.c:
16504 Convert SSA newline codes into actual newline characters (#470766).
16506 2007-08-28 14:58:17 +0000 Tim-Philipp Müller <tim@centricular.net>
16508 API: also add gst_install_plugins_supported() while we're at it (see #470456).
16509 Original commit message from CVS:
16510 * docs/libs/gst-plugins-base-libs-sections.txt:
16511 * gst-libs/gst/pbutils/install-plugins.c:
16512 * gst-libs/gst/pbutils/install-plugins.h:
16513 * tests/check/libs/pbutils.c:
16514 API: also add gst_install_plugins_supported() while we're at it
16517 2007-08-28 14:23:55 +0000 Tim-Philipp Müller <tim@centricular.net>
16519 API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
16520 Original commit message from CVS:
16521 * docs/libs/gst-plugins-base-libs-sections.txt:
16522 * gst-libs/gst/pbutils/missing-plugins.c:
16523 * gst-libs/gst/pbutils/missing-plugins.h:
16524 * tests/check/libs/pbutils.c:
16525 API: add gst_missing_*_installer_detail_new() convenience API so
16526 that applications that know exactly what they're missing can request
16527 installer detail strings for those items directly instead of having
16528 to first create a dummy missing-plugin message and then get the
16529 installer detail string from that. Fixes #470456.
16531 2007-08-27 11:59:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16533 gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
16534 Original commit message from CVS:
16535 * gst/playback/gstdecodebin.c: (close_pad_link):
16536 We need to set up delayed-linking whenever the caps are non-fixed,
16537 not just when there are multiple types - use gst_pad_is_fixed()
16540 2007-08-26 14:14:33 +0000 Tim-Philipp Müller <tim@centricular.net>
16542 gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
16543 Original commit message from CVS:
16544 * gst-libs/gst/pbutils/missing-plugins.c:
16545 (gst_missing_plugin_message_get_installer_detail):
16546 Add missing separator in PID fallback case.
16548 2007-08-24 15:28:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16550 ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
16551 Original commit message from CVS:
16552 * ext/alsa/Makefile.am:
16553 There is no GST_PLUGINS_BASE_LIBS defined.
16554 * ext/alsa/gstalsa.c:
16555 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
16556 * ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
16557 Add support for ALSA 24-bit formats.
16558 snd_pcm_delay can return an error code, especially
16559 during XRUNS. In that case, the best we can do is assume
16561 * gst/audioconvert/Makefile.am:
16562 Add flags from -base before any more-remote dependencies.
16564 2007-08-23 20:45:45 +0000 Davyd <davyd@madeley.id.au>
16566 gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
16567 Original commit message from CVS:
16568 Based on a patch by: Davyd <davyd at madeley dot id dot au>
16569 * gst/volume/gstvolume.c: (volume_choose_func),
16570 (volume_update_real_volume), (gst_volume_set_volume),
16571 (gst_volume_init), (volume_process_int32),
16572 (volume_process_int32_clamp), (volume_process_int24),
16573 (volume_process_int24_clamp), (volume_process_int16),
16574 (volume_process_int16_clamp), (volume_process_int8),
16575 (volume_process_int8_clamp), (volume_update_volume), (plugin_init):
16576 * gst/volume/gstvolume.h:
16577 Add support for int32, int24 and int8 to the volume element.
16580 2007-08-23 12:37:42 +0000 Tim-Philipp Müller <tim@centricular.net>
16582 tests/examples/Makefile.am: Fix even more.
16583 Original commit message from CVS:
16584 * tests/examples/Makefile.am:
16587 2007-08-23 10:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16589 Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
16590 Original commit message from CVS:
16592 * docs/libs/Makefile.am:
16593 * docs/libs/gst-plugins-base-libs-docs.sgml:
16594 * docs/libs/gst-plugins-base-libs-sections.txt:
16595 * ext/gnomevfs/gstgnomevfssrc.c:
16596 * ext/gnomevfs/gstgnomevfssrc.h:
16597 * gst-libs/gst/Makefile.am:
16598 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
16599 * pkgconfig/gstreamer-plugins-base.pc.in:
16600 * sys/v4l/v4lsrc_calls.c:
16601 * tests/examples/Makefile.am:
16602 * win32/common/config.h:
16603 Revert unwanted commit. many thanks to moap. I want a fix for
16604 https://thomas.apestaart.org/moap/trac/ticket/239
16606 2007-08-23 08:33:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16608 Original commit message from CVS:
16609 reviewed by: <delete if not using a buddy>
16610 patch by: <delete if not someone else's patch>
16612 * docs/libs/Makefile.am:
16613 * docs/libs/gst-plugins-base-libs-docs.sgml:
16614 * docs/libs/gst-plugins-base-libs-sections.txt:
16615 * ext/gnomevfs/gstgnomevfssrc.c:
16616 * ext/gnomevfs/gstgnomevfssrc.h:
16617 * gst-libs/gst/Makefile.am:
16618 * gst-libs/gst/audio/gstaudiofilter.h:
16619 * gst/typefind/gsttypefindfunctions.c:
16620 * gst/volume/gstvolume.c:
16621 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
16622 * pkgconfig/gstreamer-plugins-base.pc.in:
16623 * sys/v4l/v4lsrc_calls.c:
16624 * tests/examples/Makefile.am:
16625 * win32/common/config.h:
16627 2007-08-22 15:29:04 +0000 Wim Taymans <wim.taymans@gmail.com>
16629 gst-libs/gst/audio/audio.c: Clarify the docs a little.
16630 Original commit message from CVS:
16631 * gst-libs/gst/audio/audio.c:
16632 Clarify the docs a little.
16634 2007-08-22 11:20:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16636 gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
16637 Original commit message from CVS:
16638 * gst/volume/gstvolume.c:
16639 Enable liboil for float and add more details about problems with
16642 2007-08-21 15:43:24 +0000 Wim Taymans <wim.taymans@gmail.com>
16644 sys/v4l/gstv4lsrc.c: Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
16645 Original commit message from CVS:
16646 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
16647 Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
16649 2007-08-21 12:08:43 +0000 Wim Taymans <wim.taymans@gmail.com>
16651 ext/vorbis/vorbisdec.c: When calculating the first timestamp of the buffers, don't go below 0 and clip the samples be...
16652 Original commit message from CVS:
16653 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
16654 When calculating the first timestamp of the buffers, don't go below 0
16655 and clip the samples because the offset was on the eos page.
16658 2007-08-21 11:42:39 +0000 Wim Taymans <wim.taymans@gmail.com>
16660 ext/ogg/gstoggdemux.c: Also submit the eos page when trying to find the first timestamp.
16661 Original commit message from CVS:
16662 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
16663 (gst_ogg_demux_collect_chain_info):
16664 Also submit the eos page when trying to find the first timestamp.
16667 2007-08-17 15:24:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16669 gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
16670 Original commit message from CVS:
16671 * gst-libs/gst/audio/audio.h:
16672 Use gst_util_uint64_scale() instead of doing the math
16673 with double for GST_FRAMES_TO_CLOCK_TIME() and
16674 GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
16675 prevents rounding errors. Fixes #467667.
16677 2007-08-17 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
16679 gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
16680 Original commit message from CVS:
16681 * gst-libs/gst/rtsp/gstrtspconnection.c:
16682 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
16683 (gst_rtsp_connection_read), (gst_rtsp_connection_poll):
16684 * gst-libs/gst/rtsp/gstrtspconnection.h:
16686 On shutdown, don't read the control socket yet.
16687 Set timeout value correctly in all cases.
16688 Add function to check if the server accepts reads or writes.
16689 API: gst_rtsp_connection_poll()
16690 * gst-libs/gst/rtsp/gstrtspdefs.h:
16691 Fix compilation with -pedantic.
16692 Add enum for _poll.
16694 2007-08-16 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
16696 gst-libs/gst/app/gstappsink.c: Override the preroll vmethod instead of overriding the render method twice.
16697 Original commit message from CVS:
16698 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
16699 Override the preroll vmethod instead of overriding the render method
16702 2007-08-16 16:06:21 +0000 Olivier Crete <tester@tester.ca>
16704 gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
16705 Original commit message from CVS:
16706 Patch by: Olivier Crete <tester at tester ca>
16707 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
16708 (gst_basertppayload_getcaps):
16709 * gst-libs/gst/rtp/gstbasertppayload.h:
16710 Add getcaps vfunc to basertppayload. See #465146.
16712 2007-08-16 11:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
16714 gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
16715 Original commit message from CVS:
16716 * gst/playback/gstplaybasebin.c: (queue_threshold_reached):
16717 Only post buffering messages when we are a stream.
16719 2007-08-15 17:05:45 +0000 Tim-Philipp Müller <tim@centricular.net>
16721 gst-libs/gst/pbutils/: Small docs fix and addition.
16722 Original commit message from CVS:
16723 * gst-libs/gst/pbutils/install-plugins.c:
16724 * gst-libs/gst/pbutils/missing-plugins.c:
16725 Small docs fix and addition.
16727 2007-08-14 17:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
16729 gst-libs/gst/app/gstappsink.c: Don't use new API.
16730 Original commit message from CVS:
16731 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
16734 2007-08-14 17:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
16736 gst-libs/gst/app/gstappsink.*: Make love to appsink.
16737 Original commit message from CVS:
16738 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
16739 (gst_app_sink_class_init), (gst_app_sink_dispose),
16740 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
16741 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
16742 (gst_app_sink_render), (gst_app_sink_get_caps),
16743 (gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
16744 (gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
16745 * gst-libs/gst/app/gstappsink.h:
16746 Make love to appsink.
16747 Make it support pulling of the preroll buffer.
16748 Add docs and debug statements.
16749 Fix some races wrt to EOS handling and stopping.
16751 Implement FLUSHING.
16752 API: gst_app_sink_pull_preroll()
16754 2007-08-13 15:37:29 +0000 Tim-Philipp Müller <tim@centricular.net>
16756 tests/icles/: Add a dumb little test for textoverlay alignments.
16757 Original commit message from CVS:
16758 * tests/icles/.cvsignore:
16759 * tests/icles/Makefile.am:
16760 * tests/icles/test-textoverlay.c:
16761 Add a dumb little test for textoverlay alignments.
16763 2007-08-13 15:26:54 +0000 Dan Williams <dcbw@redhat.com>
16765 ext/pango/gsttextoverlay.*: API: add "line-alignment" property (#459334). Add gtk-doc blurb for "silent" property so ...
16766 Original commit message from CVS:
16767 Patch by: Dan Williams <dcbw redhat com>
16768 * ext/pango/gsttextoverlay.c:
16769 * ext/pango/gsttextoverlay.h:
16770 API: add "line-alignment" property (#459334). Add gtk-doc blurb for
16771 "silent" property so there's a Since tag in the API reference.
16773 2007-08-13 11:21:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
16777 Original commit message from CVS:
16780 2007-08-12 16:30:36 +0000 Wim Taymans <wim.taymans@gmail.com>
16782 gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
16783 Original commit message from CVS:
16784 * gst-libs/gst/rtp/gstbasertppayload.c:
16785 (gst_basertppayload_set_outcaps):
16786 * gst-libs/gst/rtp/gstbasertppayload.h:
16787 Improve caps negotiation so that downstream elements can confiure
16788 certain RTP properties by fixing them on the caps. See #465146.
16791 2007-08-11 12:39:51 +0000 Tim-Philipp Müller <tim@centricular.net>
16793 Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
16794 Original commit message from CVS:
16795 * docs/libs/gst-plugins-base-libs-sections.txt:
16796 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16797 * gst-libs/gst/rtp/gstbasertpdepayload.h:
16798 Mark as deprecated some macros which were presumably meant to be
16799 private API and accidentally exposed in the public header file.
16800 Also actually _init() lock (only works at the moment because the
16801 struct is zeroed out when created and the initial values in the
16802 mutex struct are zeroes too). (#459585)
16804 2007-08-10 17:35:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16806 docs/libs/Makefile.am: Remove cruft and do some cleanups.
16807 Original commit message from CVS:
16808 * docs/libs/Makefile.am:
16809 Remove cruft and do some cleanups.
16810 * docs/libs/gst-plugins-base-libs-docs.sgml:
16811 Prepare for comming gtkdoc features (rebase against online docs).
16813 2007-08-10 13:55:44 +0000 Michael Smith <msmith@xiph.org>
16815 gst/audiorate/gstaudiorate.c: Debug output fixes.
16816 Original commit message from CVS:
16817 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
16818 Debug output fixes.
16819 * tests/check/elements/audiorate.c: (do_perfect_stream_test),
16821 Change the number of buffers used; 500 is too many and leads to
16824 2007-08-10 10:08:05 +0000 Tim-Philipp Müller <tim@centricular.net>
16826 gst/: Printf format fixes (#465028).
16827 Original commit message from CVS:
16828 * gst/playback/gstqueue2.c:
16829 * gst/videorate/gstvideorate.c:
16830 Printf format fixes (#465028).
16832 2007-08-09 15:44:02 +0000 Michael Smith <msmith@xiph.org>
16834 gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
16835 Original commit message from CVS:
16836 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
16837 If we have a large (> 1 second) discontinuity, push a series of
16838 smaller buffers rather than a single very large buffer. Avoids
16839 unreasonably large single buffer allocations when encountering a
16841 * tests/check/elements/audiorate.c: (GST_START_TEST),
16843 Add a test for this.
16845 2007-08-09 12:06:43 +0000 Josep Torra Valles <josep@fluendo.com>
16847 gst/playback/gstplaybasebin.c: Fixes: #465015
16848 Original commit message from CVS:
16849 * gst/playback/gstplaybasebin.c: (group_commit),
16850 (queue_remove_probe), (queue_threshold_reached):
16851 Patch by: Josep Torra Valles <josep@fluendo.com>
16853 Make sure we remove the check_queues buffer probe from the
16854 correct queue to avoid racily going back to "buffering 99%" when
16855 buffering is actually complete.
16856 Also, fix the spelling of Josep's surname in the ChangeLog.
16858 2007-08-09 11:37:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16860 ext/ogg/gstoggmux.c: Do not leak oggmux instance.
16861 Original commit message from CVS:
16862 * ext/ogg/gstoggmux.c:
16863 Do not leak oggmux instance.
16864 * ext/vorbis/vorbisenc.c:
16867 2007-08-09 10:51:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
16869 po/: Updated translations.
16870 Original commit message from CVS:
16876 Updated translations.
16878 2007-08-08 16:07:21 +0000 Yang Hong <hongyang@redflag-linux.com>
16880 ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
16881 Original commit message from CVS:
16882 patch by: Yang Hong <hongyang@redflag-linux.com>
16883 * ext/pango/gsttextoverlay.c:
16884 * ext/pango/gsttextoverlay.h:
16885 Add 'silent' property to GstTimeOverlay. Fixes #462979
16887 2007-08-08 15:05:22 +0000 Josep Torre Valles <josep@fluendo.com>
16889 Add connection-speed property. Fixes #464690.
16890 Original commit message from CVS:
16891 Patch by: Josep Torre Valles <josep@fluendo.com>
16892 * docs/plugins/gst-plugins-base-plugins.args:
16893 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
16894 (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
16895 (gst_uri_decode_bin_get_property), (gen_source_element):
16896 Add connection-speed property. Fixes #464690.
16898 2007-08-07 15:13:46 +0000 Damien Lespiau <damien.lespiau@gmail.com>
16900 Fix compilation on windows. Fixes #464320.
16901 Original commit message from CVS:
16902 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
16904 * gst-libs/gst/rtsp/Makefile.am:
16905 * gst-libs/gst/rtsp/gstrtspconnection.c:
16906 (gst_rtsp_connection_connect):
16907 Fix compilation on windows. Fixes #464320.
16909 2007-08-07 14:14:54 +0000 Josep Torre Valles <josep@fluendo.com>
16911 gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
16912 Original commit message from CVS:
16913 Patch by: Josep Torre Valles <josep@fluendo.com>
16914 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
16915 (gst_play_base_bin_init), (queue_threshold_reached),
16916 (gen_source_element), (setup_substreams),
16917 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
16918 (gst_play_base_bin_get_streaminfo_value_array):
16919 * gst/playback/gstplaybasebin.h:
16920 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
16921 (gst_play_bin_set_property), (gst_play_bin_get_property),
16922 (gst_play_bin_handle_redirect_message):
16923 Move connection-speed property from playbin to playbasebin so that we
16924 can also configure it in source elements that have the connection-speed
16925 property. Fixes #464028.
16926 Add some debug info here and there.
16928 2007-08-06 16:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16930 gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
16931 Original commit message from CVS:
16932 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
16933 Properly respond to conversion queries. Fixes #464079.
16935 2007-08-03 19:53:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16937 gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
16938 Original commit message from CVS:
16939 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
16940 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
16941 (gst_audio_test_src_init_sine_table),
16942 (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
16943 * gst/audiotestsrc/gstaudiotestsrc.h:
16944 Add float/double and int32 support to audiotestsrc. Fixes #460422.
16945 Also set the default volume to the default value specified in the
16948 2007-08-03 19:40:14 +0000 Jens Granseuer <jensgr@gmx.net>
16950 gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
16951 Original commit message from CVS:
16952 Patch by: Jens Granseuer <jensgr at gmx dot net>
16953 * gst/audioconvert/gstaudioquantize.c:
16954 Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
16956 2007-08-03 15:44:01 +0000 Wim Taymans <wim.taymans@gmail.com>
16958 gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
16959 Original commit message from CVS:
16960 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
16961 Add rdt manager for rdt transport.
16962 Fix parsing of RDT transport.
16964 2007-08-03 14:43:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16966 configure.ac: Back to CVS
16967 Original commit message from CVS:
16971 === release 0.10.14 ===
16973 2007-08-03 14:41:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16979 * docs/plugins/gst-plugins-base-plugins.args:
16980 * docs/plugins/inspect/plugin-adder.xml:
16981 * docs/plugins/inspect/plugin-alsa.xml:
16982 * docs/plugins/inspect/plugin-audioconvert.xml:
16983 * docs/plugins/inspect/plugin-audiorate.xml:
16984 * docs/plugins/inspect/plugin-audioresample.xml:
16985 * docs/plugins/inspect/plugin-audiotestsrc.xml:
16986 * docs/plugins/inspect/plugin-cdparanoia.xml:
16987 * docs/plugins/inspect/plugin-decodebin.xml:
16988 * docs/plugins/inspect/plugin-decodebin2.xml:
16989 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
16990 * docs/plugins/inspect/plugin-gdp.xml:
16991 * docs/plugins/inspect/plugin-gnomevfs.xml:
16992 * docs/plugins/inspect/plugin-libvisual.xml:
16993 * docs/plugins/inspect/plugin-ogg.xml:
16994 * docs/plugins/inspect/plugin-pango.xml:
16995 * docs/plugins/inspect/plugin-playbin.xml:
16996 * docs/plugins/inspect/plugin-subparse.xml:
16997 * docs/plugins/inspect/plugin-tcp.xml:
16998 * docs/plugins/inspect/plugin-theora.xml:
16999 * docs/plugins/inspect/plugin-typefindfunctions.xml:
17000 * docs/plugins/inspect/plugin-video4linux.xml:
17001 * docs/plugins/inspect/plugin-videorate.xml:
17002 * docs/plugins/inspect/plugin-videoscale.xml:
17003 * docs/plugins/inspect/plugin-videotestsrc.xml:
17004 * docs/plugins/inspect/plugin-volume.xml:
17005 * docs/plugins/inspect/plugin-vorbis.xml:
17006 * docs/plugins/inspect/plugin-ximagesink.xml:
17007 * docs/plugins/inspect/plugin-xvimagesink.xml:
17008 * gst-plugins-base.doap:
17009 * win32/common/config.h:
17011 Original commit message from CVS:
17014 2007-08-03 14:24:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17032 Original commit message from CVS:
17035 2007-07-27 17:37:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17037 tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
17038 Original commit message from CVS:
17039 * tests/check/libs/audio.c: (GST_START_TEST):
17040 Fix the test to reflect the behaviour of gst_audio_clip_buffer.
17042 2007-07-27 17:10:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17044 gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
17045 Original commit message from CVS:
17046 * gst-libs/gst/audio/audio.c:
17047 When clipping a buffer with no timestamp, assume it is
17048 within the segment without warnings.
17051 2007-07-27 11:16:23 +0000 Wim Taymans <wim.taymans@gmail.com>
17053 gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
17054 Original commit message from CVS:
17055 * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
17056 Fire the signal on the object, not the interface.
17058 2007-07-27 09:17:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17060 gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
17061 Original commit message from CVS:
17062 * gst-libs/gst/rtsp/.cvsignore:
17063 Ber. Don't include the full path, idiot.
17065 2007-07-27 08:29:29 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17067 gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
17068 Original commit message from CVS:
17069 * gst-libs/gst/rtsp/.cvsignore:
17070 Ignore generated files.
17072 2007-07-26 19:57:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17074 gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
17075 Original commit message from CVS:
17076 * gst-libs/gst/interfaces/Makefile.am:
17077 * gst-libs/gst/interfaces/interfaces-marshal.list:
17078 * gst-libs/gst/interfaces/rtspextension.c:
17079 * gst-libs/gst/interfaces/rtspextension.h:
17080 * gst-libs/gst/rtsp/Makefile.am:
17081 * gst-libs/gst/rtsp/gstrtsp.h:
17082 * gst-libs/gst/rtsp/gstrtspextension.c:
17083 (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
17084 (gst_rtsp_extension_detect_server),
17085 (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
17086 (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
17087 (gst_rtsp_extension_configure_stream),
17088 (gst_rtsp_extension_get_transports),
17089 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
17090 * gst-libs/gst/rtsp/gstrtspextension.h:
17091 * gst-libs/gst/rtsp/rtsp-marshal.list:
17092 Move the rtspextension.h interface into gstrtspextension.h
17093 as part of libgstrtsp instead of libgstinterfaces, because it's
17094 only for use within plugins, not applications.
17095 Add stuff to do the enum & marshal generation needed in libgstrtsp now.
17096 Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
17097 signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
17100 2007-07-26 15:48:01 +0000 Wim Taymans <wim.taymans@gmail.com>
17102 gst-libs/gst/interfaces/: Fix marshaller for the send signal.
17103 Original commit message from CVS:
17104 * gst-libs/gst/interfaces/Makefile.am:
17105 * gst-libs/gst/interfaces/interfaces-marshal.list:
17106 * gst-libs/gst/interfaces/rtspextension.c:
17107 (gst_rtsp_extension_iface_init),
17108 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
17109 * gst-libs/gst/interfaces/rtspextension.h:
17110 Fix marshaller for the send signal.
17111 Add URL to stream selection interface method.
17113 2007-07-26 15:35:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17115 gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
17116 Original commit message from CVS:
17117 * gst-libs/gst/riff/Makefile.am:
17118 Pull in our dependencies from -base before those from outside.
17120 2007-07-26 14:33:01 +0000 Wim Taymans <wim.taymans@gmail.com>
17122 API: gst_rtsp_base64_decode_ip()
17123 Original commit message from CVS:
17124 * docs/libs/gst-plugins-base-libs-sections.txt:
17125 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
17126 * gst-libs/gst/rtsp/gstrtspbase64.h:
17127 API: gst_rtsp_base64_decode_ip()
17128 Added function to decode Base64 in-place.
17130 2007-07-26 14:08:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17132 tests/check/libs/.cvsignore: Ignore the mixer test binary.
17133 Original commit message from CVS:
17134 * tests/check/libs/.cvsignore:
17135 Ignore the mixer test binary.
17137 2007-07-26 10:00:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17139 ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
17140 Original commit message from CVS:
17141 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
17142 Gratuitous comment change to trigger a rebuild on the buildbots.
17144 2007-07-25 18:20:36 +0000 Wim Taymans <wim.taymans@gmail.com>
17146 gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
17147 Original commit message from CVS:
17148 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
17149 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
17150 (gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
17151 (gst_sdp_media_get_format), (gst_sdp_media_get_information),
17152 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
17153 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
17154 (gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
17155 (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
17156 (gst_sdp_media_get_attribute_val):
17157 * gst-libs/gst/sdp/gstsdpmessage.h:
17158 Constify args where we can.
17160 2007-07-25 18:18:49 +0000 Wim Taymans <wim.taymans@gmail.com>
17162 gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
17163 Original commit message from CVS:
17164 * gst-libs/gst/interfaces/Makefile.am:
17165 * gst-libs/gst/interfaces/rtspextension.c:
17166 (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
17167 (gst_rtsp_extension_detect_server),
17168 (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
17169 (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
17170 (gst_rtsp_extension_configure_stream),
17171 (gst_rtsp_extension_get_transports),
17172 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
17173 * gst-libs/gst/interfaces/rtspextension.h:
17174 Move interface for RTSP extensions from -good to here.
17175 Added helper methods to invoke interface methods.
17177 2007-07-25 11:22:30 +0000 Wim Taymans <wim.taymans@gmail.com>
17179 Fix some more RTSP docs.
17180 Original commit message from CVS:
17181 * docs/libs/gst-plugins-base-libs-sections.txt:
17182 * gst-libs/gst/rtsp/gstrtspdefs.h:
17183 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
17184 (gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
17185 (gst_rtsp_message_init_response),
17186 (gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
17187 (gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
17188 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
17189 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
17190 (gst_rtsp_message_get_body), (dump_key_value):
17191 * gst-libs/gst/rtsp/gstrtspmessage.h:
17192 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
17193 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
17194 (gst_rtsp_range_parse):
17195 * gst-libs/gst/rtsp/gstrtsprange.h:
17196 * gst-libs/gst/rtsp/gstrtsptransport.c:
17197 * gst-libs/gst/rtsp/gstrtspurl.c:
17198 Fix some more RTSP docs.
17199 Add some missing methods for dealing with messages.
17201 2007-07-24 19:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
17203 Added beginnings of RTSP documentation.
17204 Original commit message from CVS:
17205 * docs/libs/gst-plugins-base-libs-docs.sgml:
17206 * docs/libs/gst-plugins-base-libs-sections.txt:
17207 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
17208 * gst-libs/gst/rtsp/gstrtspbase64.h:
17209 * gst-libs/gst/rtsp/gstrtspconnection.c:
17210 (gst_rtsp_connection_connect), (add_auth_header),
17211 (gst_rtsp_connection_write), (gst_rtsp_connection_send),
17212 (read_body), (gst_rtsp_connection_receive),
17213 (gst_rtsp_connection_next_timeout),
17214 (gst_rtsp_connection_reset_timeout),
17215 (gst_rtsp_connection_set_auth):
17216 * gst-libs/gst/rtsp/gstrtspconnection.h:
17217 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
17218 * gst-libs/gst/rtsp/gstrtspdefs.h:
17219 * gst-libs/gst/rtsp/gstrtspmessage.h:
17220 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
17221 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
17222 (gst_rtsp_range_parse):
17223 * gst-libs/gst/rtsp/gstrtspurl.h:
17224 Added beginnings of RTSP documentation.
17226 2007-07-24 17:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
17228 Document the SDP library.
17229 Original commit message from CVS:
17230 * docs/libs/Makefile.am:
17231 * docs/libs/gst-plugins-base-libs-docs.sgml:
17232 * docs/libs/gst-plugins-base-libs-sections.txt:
17233 * gst-libs/gst/sdp/gstsdp.h:
17234 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
17235 (gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
17236 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
17237 (gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
17238 (gst_sdp_message_get_attribute_val),
17239 (gst_sdp_message_add_attribute), (gst_sdp_media_new),
17240 (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
17241 (gst_sdp_media_get_media), (gst_sdp_media_set_media),
17242 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
17243 (gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
17244 (gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
17245 (gst_sdp_media_get_format), (gst_sdp_media_add_format),
17246 (gst_sdp_media_get_information), (gst_sdp_media_set_information),
17247 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
17248 (gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
17249 (gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
17250 (gst_sdp_media_set_key), (gst_sdp_media_get_key),
17251 (gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
17252 (gst_sdp_media_get_attribute_val_n),
17253 (gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
17254 (print_media), (gst_sdp_message_dump):
17255 * gst-libs/gst/sdp/gstsdpmessage.h:
17256 Document the SDP library.
17257 Add some of the missing SDPMedia methods.
17259 2007-07-24 11:52:56 +0000 Wim Taymans <wim.taymans@gmail.com>
17261 Move SDP and RTSP from helper objects in -good to a reusable library.
17262 Original commit message from CVS:
17264 * gst-libs/gst/Makefile.am:
17265 * gst-libs/gst/rtsp/Makefile.am:
17266 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
17267 * gst-libs/gst/rtsp/gstrtspbase64.h:
17268 * gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
17269 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
17270 (add_auth_header), (add_date_header), (gst_rtsp_connection_write),
17271 (gst_rtsp_connection_send), (read_line), (read_string), (read_key),
17272 (parse_response_status), (parse_request_line), (parse_line),
17273 (gst_rtsp_connection_read), (read_body),
17274 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
17275 (gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
17276 (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
17277 (gst_rtsp_connection_set_auth):
17278 * gst-libs/gst/rtsp/gstrtspconnection.h:
17279 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
17280 (gst_rtsp_strresult), (gst_rtsp_method_as_text),
17281 (gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
17282 (gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
17283 (gst_rtsp_find_method):
17284 * gst-libs/gst/rtsp/gstrtspdefs.h:
17285 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
17286 (gst_rtsp_message_new), (gst_rtsp_message_init),
17287 (gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
17288 (gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
17289 (gst_rtsp_message_init_data), (gst_rtsp_message_unset),
17290 (gst_rtsp_message_free), (gst_rtsp_message_add_header),
17291 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
17292 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
17293 (gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
17294 (gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
17295 (gst_rtsp_message_dump):
17296 * gst-libs/gst/rtsp/gstrtspmessage.h:
17297 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
17298 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
17299 (gst_rtsp_range_parse), (gst_rtsp_range_free):
17300 * gst-libs/gst/rtsp/gstrtsprange.h:
17301 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
17302 (gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
17303 (gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
17304 (range_as_text), (rtsp_transport_mode_as_text),
17305 (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
17306 (gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
17307 (gst_rtsp_transport_free):
17308 * gst-libs/gst/rtsp/gstrtsptransport.h:
17309 * gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
17310 (gst_rtsp_url_free), (gst_rtsp_url_set_port),
17311 (gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
17312 * gst-libs/gst/rtsp/gstrtspurl.h:
17313 * gst-libs/gst/sdp/Makefile.am:
17314 * gst-libs/gst/sdp/gstsdp.h:
17315 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
17316 (gst_sdp_connection_init), (gst_sdp_bandwidth_init),
17317 (gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
17318 (gst_sdp_attribute_init), (gst_sdp_message_new),
17319 (gst_sdp_message_init), (gst_sdp_message_uninit),
17320 (gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
17321 (gst_sdp_media_uninit), (gst_sdp_media_free),
17322 (gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
17323 (gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
17324 (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
17325 (gst_sdp_message_add_zone), (gst_sdp_message_set_key),
17326 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
17327 (gst_sdp_message_get_attribute_val),
17328 (gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
17329 (gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
17330 (gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
17331 (gst_sdp_media_get_attribute_val_n),
17332 (gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
17333 (read_string), (read_string_del), (gst_sdp_parse_line),
17334 (gst_sdp_message_parse_buffer), (print_media),
17335 (gst_sdp_message_dump):
17336 * gst-libs/gst/sdp/gstsdpmessage.h:
17337 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
17338 Move SDP and RTSP from helper objects in -good to a reusable library.
17339 Use a proper gst_ namespace.
17341 2007-07-23 18:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17343 ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
17344 Original commit message from CVS:
17345 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
17346 (vorbis_dec_flush_decode):
17347 Use the new buffer clipping function from gstaudio here.
17349 2007-07-23 18:26:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17351 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
17352 Original commit message from CVS:
17353 * docs/libs/gst-plugins-base-libs-sections.txt:
17354 * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
17355 * gst-libs/gst/audio/audio.h:
17356 * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
17357 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
17358 Also add deprecation guards for gst_audio_structure_set_int() to the
17361 2007-07-23 14:45:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17363 docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
17364 Original commit message from CVS:
17365 * docs/libs/gst-plugins-base-libs-sections.txt:
17368 2007-07-23 11:18:35 +0000 Dan Williams <dcbw@redhat.com>
17370 gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
17371 Original commit message from CVS:
17372 Patch by: Dan Williams <dcbw at redhat dot com>
17373 * gst/playback/gstplaybasebin.c:
17374 (gst_play_base_bin_get_streaminfo_value_array):
17375 Don't return NULL when querying the stream info value array but instead
17376 return an empty array. Fixes #459204.
17378 2007-07-23 10:41:18 +0000 Tim-Philipp Müller <tim@centricular.net>
17380 gst/playback/gsturidecodebin.c: Init debug category before using it.
17381 Original commit message from CVS:
17382 * gst/playback/gsturidecodebin.c:
17383 Init debug category before using it.
17385 2007-07-21 09:56:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17387 gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
17388 Original commit message from CVS:
17389 * gst-libs/gst/interfaces/mixer.h:
17390 Add padding vars in place of the signal pointers
17391 when building with DISABLE_DEPRECATED so that the
17392 interface structure doesn't change size.
17394 2007-07-21 09:21:12 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
17397 Original commit message from CVS:
17398 * docs/libs/gst-plugins-base-libs-sections.txt:
17399 * ext/alsa/gstalsamixer.c:
17400 * ext/alsa/gstalsamixer.h:
17401 * ext/alsa/gstalsamixerelement.c:
17402 * ext/alsa/gstalsamixertrack.c:
17403 * gst-libs/gst/interfaces/mixer.c:
17404 * gst-libs/gst/interfaces/mixer.h:
17405 * gst-libs/gst/interfaces/mixeroptions.c:
17406 * gst-libs/gst/interfaces/mixeroptions.h:
17407 * gst-libs/gst/interfaces/mixertrack.c:
17408 * gst-libs/gst/interfaces/mixertrack.h:
17409 * tests/check/Makefile.am:
17410 * tests/check/libs/mixer.c:
17411 Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
17413 Add support for notifying mixer changes on the message bus, and
17414 implement it in alsamixer.
17415 API: gst_mixer_get_mixer_flags
17416 API: gst_mixer_message_parse_mute_toggled
17417 API: gst_mixer_message_parse_record_toggled
17418 API: gst_mixer_message_parse_volume_changed
17419 API: gst_mixer_message_parse_option_changed
17420 API: GstMixerMessageType
17423 2007-07-20 16:09:03 +0000 Michael Smith <msmith@xiph.org>
17425 sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
17426 Original commit message from CVS:
17427 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
17428 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
17429 xcontext->im_format is only for testing XShm support (as the header
17430 file comments document). Use xvimage->im_format for everything else.
17431 Avoids spurious warnings on buffer allocation before setcaps.
17433 2007-07-20 07:22:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17435 tests/: We should use $(LIBM).
17436 Original commit message from CVS:
17437 * tests/examples/volume/Makefile.am:
17438 * tests/icles/Makefile.am:
17439 We should use $(LIBM).
17441 2007-07-20 06:13:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17443 tests/icles/Makefile.am: This needs -lm.
17444 Original commit message from CVS:
17445 * tests/icles/Makefile.am:
17448 2007-07-18 07:35:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17450 Add stdlib include (free, atoi, exit).
17451 Original commit message from CVS:
17452 * examples/app/appsrc_ex.c:
17453 * examples/switch/switcher.c:
17454 * ext/neon/gstneonhttpsrc.c:
17455 * ext/timidity/gstwildmidi.c:
17456 * ext/x264/gstx264enc.c:
17457 * gst/mve/mveaudioenc.c: (mve_compress_audio):
17458 * gst/rtpmanager/gstrtpclient.c:
17459 * gst/rtpmanager/gstrtpjitterbuffer.c:
17460 * gst/spectrum/demo-audiotest.c:
17461 * gst/spectrum/demo-osssrc.c:
17462 * sys/dvb/gstdvbsrc.c:
17463 Add stdlib include (free, atoi, exit).
17465 2007-07-16 10:10:28 +0000 Wim Taymans <wim.taymans@gmail.com>
17467 gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
17468 Original commit message from CVS:
17469 * gst-libs/gst/rtp/gstbasertppayload.c:
17470 (gst_basertppayload_class_init), (gst_basertppayload_init),
17471 (gst_basertppayload_set_property),
17472 (gst_basertppayload_get_property):
17473 Don't break ABI, restore previous ranges. Keep the default random
17474 selection of timestamp and seqnum offset but as soon as the app sets a
17475 specific value, use that one.
17477 2007-07-14 18:33:15 +0000 Bastien Nocera <hadess@hadess.net>
17479 sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
17480 Original commit message from CVS:
17481 Patch by: Bastien Nocera <hadess at hadess dot net>
17482 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
17483 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
17484 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
17485 * sys/xvimage/xvimagesink.h:
17486 Add option to turn off double-buffering for debugging purposes.
17489 2007-07-14 18:20:41 +0000 Jorn Baayen <jorn@openedhand.com>
17491 sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
17492 Original commit message from CVS:
17493 Patch by: Jorn Baayen <jorn at openedhand dot com>
17494 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
17495 (gst_ximagesink_set_property), (gst_ximagesink_get_property),
17496 (gst_ximagesink_init), (gst_ximagesink_class_init):
17497 * sys/ximage/ximagesink.h:
17498 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
17499 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
17500 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
17501 * sys/xvimage/xvimagesink.h:
17502 add 'handle-expose' property. Useful for video widgets which may want to
17503 be in control of Expose behaviour. Fixes #380625
17505 2007-07-14 17:23:42 +0000 Wim Taymans <wim.taymans@gmail.com>
17507 gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
17508 Original commit message from CVS:
17509 * gst-libs/gst/rtp/gstbasertppayload.c:
17510 (gst_basertppayload_class_init), (gst_basertppayload_init),
17511 (gst_basertppayload_event), (gst_basertppayload_push),
17512 (gst_basertppayload_set_property),
17513 (gst_basertppayload_get_property),
17514 (gst_basertppayload_change_state):
17515 * gst-libs/gst/rtp/gstbasertppayload.h:
17516 Fix ranges of rtp payloader properties so that the full range can be
17517 used in addition to -1 (random).
17518 Fix wrong seqnum reporting in caps.
17521 2007-07-13 18:12:19 +0000 Wim Taymans <wim.taymans@gmail.com>
17523 gst/videorate/gstvideorate.c: Use boilerplate.
17524 Original commit message from CVS:
17525 * gst/videorate/gstvideorate.c: (gst_video_rate_init),
17526 (gst_video_rate_query):
17528 Add latency query, might not be perfect yet but already works a lot
17529 better. Fixes #442557.
17531 2007-07-13 16:05:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17533 sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
17534 Original commit message from CVS:
17535 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
17536 (gst_xvimagesink_setcaps):
17537 * sys/xvimage/xvimagesink.h:
17538 After a caps change, redraw our borders to avoid garbage left there
17539 when the image format changes to a smaller size, like 16:9 -> 4:3
17540 Also, hold the flow_lock a bit longer in the set_caps while we're
17541 fiddling with the xcontext.
17543 2007-07-13 16:02:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17545 Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
17546 Original commit message from CVS:
17549 * tests/Makefile.am:
17550 Remove bogus check for libcheck, since we check for
17551 gstreamer-check and it pulls in the required info from there, and we
17552 weren't actually _using_ the information for libcheck ourselves
17555 2007-07-13 15:52:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17557 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
17558 Original commit message from CVS:
17559 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
17560 (gst_ffmpeg_caps_to_pixfmt):
17561 Fix the r_mask test for RGBA32 on little-endian.
17562 Fix a stupid typo that would have obviously broken
17563 compilation on big-endian, if anyone was testing.
17565 2007-07-12 15:02:43 +0000 Wim Taymans <wim.taymans@gmail.com>
17567 gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
17568 Original commit message from CVS:
17569 * gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
17570 (paint_hline_str4):
17571 * gst/videotestsrc/videotestsrc.h:
17572 Add alpha to the color struct.
17573 Use a default alpha value of 255 instead of 128.
17575 2007-07-12 12:01:20 +0000 Wim Taymans <wim.taymans@gmail.com>
17577 gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
17578 Original commit message from CVS:
17579 * gst/playback/gstplaybasebin.c: (no_more_pads_full),
17581 Clear the dynamic pads counter when starting a new uri. This makes
17582 reusing playbin work again.
17585 2007-07-12 11:13:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17587 configure.ac: Use pkg-config to locate check.
17588 Original commit message from CVS:
17590 Use pkg-config to locate check.
17592 2007-07-11 23:12:12 +0000 Tim-Philipp Müller <tim@centricular.net>
17594 Fix 'make check' build against core CVS.
17595 Original commit message from CVS:
17597 * tests/check/elements/volume.c: (GST_START_TEST):
17598 Fix 'make check' build against core CVS.
17600 2007-07-10 20:46:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17602 gst-libs/gst/: Make gtk-doc happy.
17603 Original commit message from CVS:
17604 * gst-libs/gst/interfaces/propertyprobe.c:
17605 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
17606 * gst-libs/gst/tag/gstvorbistag.c:
17607 Make gtk-doc happy.
17609 2007-07-08 13:07:38 +0000 Tim-Philipp Müller <tim@centricular.net>
17611 gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
17612 Original commit message from CVS:
17613 * gst-libs/gst/audio/gstbaseaudiosink.c:
17614 (gst_base_audio_sink_callback):
17615 Quick hack to make audiosinks stop at EOS when operating in
17616 pull-mode; needs to be fixed properly some day.
17618 2007-07-06 18:19:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17620 docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
17621 Original commit message from CVS:
17622 * docs/libs/gst-plugins-base-libs-sections.txt:
17623 Fix location of includes in the docs.
17625 2007-07-06 11:40:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17627 gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
17628 Original commit message from CVS:
17629 * gst/ffmpegcolorspace/avcodec.h:
17630 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
17631 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
17632 (gst_ffmpegcsp_avpicture_fill):
17633 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
17634 (img_get_alpha_info):
17635 Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
17636 of the existing BGRA32 and RGBA32 formats with the alpha at the other
17637 end of the word. Partially fixes #451908
17639 2007-07-05 08:43:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17641 docs/: Simplify --extra-dir as gtkdoc scans recursively.
17642 Original commit message from CVS:
17643 * docs/libs/Makefile.am:
17644 * docs/plugins/Makefile.am:
17645 Simplify --extra-dir as gtkdoc scans recursively.
17647 2007-07-03 11:52:47 +0000 Wim Taymans <wim.taymans@gmail.com>
17649 gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
17650 Original commit message from CVS:
17651 * gst/adder/gstadder.c: (gst_adder_sink_getcaps),
17652 (gst_adder_request_new_pad):
17653 Make getcaps more robust by not using the proxycaps function. This makes
17654 sure that we don't end up recursively calling getcaps upstream.
17657 2007-06-29 17:21:18 +0000 Wim Taymans <wim.taymans@gmail.com>
17659 gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
17660 Original commit message from CVS:
17661 * gst/audioconvert/audioconvert.c:
17662 Include math.h to fix compilation.
17664 2007-06-29 14:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17666 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
17667 Original commit message from CVS:
17668 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
17669 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
17670 Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
17671 format, as produced by some dc1394 cameras like the iSight.
17672 See http://www.fourcc.org/yuv.php#IYU1
17674 2007-06-28 20:37:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17676 gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
17677 Original commit message from CVS:
17678 * gst/audioconvert/Makefile.am:
17679 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
17680 (check_default), (audio_convert_prepare_context),
17681 (audio_convert_clean_context), (audio_convert_convert):
17682 * gst/audioconvert/audioconvert.h:
17683 * gst/audioconvert/gstaudioconvert.c:
17684 (gst_audio_convert_dithering_get_type),
17685 (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
17686 (gst_audio_convert_init), (gst_audio_convert_set_caps),
17687 (gst_audio_convert_set_property), (gst_audio_convert_get_property):
17688 * gst/audioconvert/gstaudioconvert.h:
17689 * gst/audioconvert/gstaudioquantize.c:
17690 (gst_audio_quantize_setup_noise_shaping),
17691 (gst_audio_quantize_free_noise_shaping),
17692 (gst_audio_quantize_setup_dither),
17693 (gst_audio_quantize_free_dither),
17694 (gst_audio_quantize_setup_quantize_func),
17695 (gst_audio_quantize_setup), (gst_audio_quantize_free):
17696 * gst/audioconvert/gstaudioquantize.h:
17697 Implement dithering and noise shaping in audioconvert. By default now
17698 TPDF dithering (and no noise shaping) will be used when converting
17699 from a higher bit depth to 20 bit depth or smaller, otherwise
17700 everything will be as it is now.
17701 For the last audioconvert in a pipeline it would make sense to
17702 use some kind of noise shaping, enabling it by default for all
17703 conversions would give undesired results though. Fixes #360246.
17704 * tests/check/elements/audioconvert.c: (setup_audioconvert),
17706 Adjust unit test for the new audioconvert.
17708 2007-06-28 11:06:56 +0000 Wim Taymans <wim.taymans@gmail.com>
17710 gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
17711 Original commit message from CVS:
17712 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
17713 Use other metrics as well when estimating the buffer level.
17715 2007-06-28 10:21:19 +0000 Wim Taymans <wim.taymans@gmail.com>
17717 gst/playback/gstplaybasebin.c: Small debug improvement.
17718 Original commit message from CVS:
17719 * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
17720 Small debug improvement.
17721 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
17723 Tweak the rate estimation period.
17724 When calculating the buffer filledness in rate estimation mode, don't
17725 mix it with other metrics.
17727 2007-06-28 09:46:11 +0000 Wim Taymans <wim.taymans@gmail.com>
17729 gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
17730 Original commit message from CVS:
17731 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
17732 (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
17733 When creating the groups, allow for a 5 second, unlimited buffers
17734 preroll phase after which we expose the group.
17735 When the group is exposed, use a small number of buffers up to a 2
17736 second limit. Also disconnect the overrun signal from multiqueue when we
17737 exposed the group because it is not needed anymore.
17739 2007-06-27 22:30:19 +0000 Tim-Philipp Müller <tim@centricular.net>
17741 gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
17742 Original commit message from CVS:
17743 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
17744 Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
17745 to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
17746 (#451707); also, output some debugging info when dealing with
17748 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
17749 Add unit test for the above.
17751 2007-06-27 12:55:20 +0000 Tim-Philipp Müller <tim@centricular.net>
17753 gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
17754 Original commit message from CVS:
17755 * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
17756 Add description for Windows Media RTP caps.
17757 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
17758 Remove RTP fields that don't define the format from caps.
17760 2007-06-27 10:14:03 +0000 Tim-Philipp Müller <tim@centricular.net>
17762 ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
17763 Original commit message from CVS:
17764 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
17765 Skip empty buffers, but not empty header buffers. That way the original
17766 vorbisdec unit test still passes (#451145); also, take into account
17767 that those empty packets might carry a granulepos.
17768 * tests/check/Makefile.am:
17769 * tests/check/elements/vorbisdec.c:
17770 (_create_codebook_header_buffer), (_create_audio_buffer),
17771 (GST_START_TEST), (vorbisdec_suite):
17772 Add unit test that sends an empty packet.
17774 2007-06-27 09:49:51 +0000 Wim Taymans <wim.taymans@gmail.com>
17776 ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
17777 Original commit message from CVS:
17778 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
17779 Don't error out on 0-sized packets, just emit a warning because this is
17780 not a fatal error. Fixes #451145.
17782 2007-06-25 12:43:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17784 docs/plugins/: Update docs with caps info.
17785 Original commit message from CVS:
17786 * docs/plugins/gst-plugins-base-plugins.args:
17787 * docs/plugins/gst-plugins-base-plugins.signals:
17788 * docs/plugins/inspect/plugin-adder.xml:
17789 * docs/plugins/inspect/plugin-alsa.xml:
17790 * docs/plugins/inspect/plugin-audioconvert.xml:
17791 * docs/plugins/inspect/plugin-audiorate.xml:
17792 * docs/plugins/inspect/plugin-audioresample.xml:
17793 * docs/plugins/inspect/plugin-audiotestsrc.xml:
17794 * docs/plugins/inspect/plugin-cdparanoia.xml:
17795 * docs/plugins/inspect/plugin-decodebin.xml:
17796 * docs/plugins/inspect/plugin-decodebin2.xml:
17797 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
17798 * docs/plugins/inspect/plugin-gdp.xml:
17799 * docs/plugins/inspect/plugin-gnomevfs.xml:
17800 * docs/plugins/inspect/plugin-libvisual.xml:
17801 * docs/plugins/inspect/plugin-ogg.xml:
17802 * docs/plugins/inspect/plugin-pango.xml:
17803 * docs/plugins/inspect/plugin-playbin.xml:
17804 * docs/plugins/inspect/plugin-subparse.xml:
17805 * docs/plugins/inspect/plugin-tcp.xml:
17806 * docs/plugins/inspect/plugin-theora.xml:
17807 * docs/plugins/inspect/plugin-typefindfunctions.xml:
17808 * docs/plugins/inspect/plugin-video4linux.xml:
17809 * docs/plugins/inspect/plugin-videorate.xml:
17810 * docs/plugins/inspect/plugin-videoscale.xml:
17811 * docs/plugins/inspect/plugin-videotestsrc.xml:
17812 * docs/plugins/inspect/plugin-volume.xml:
17813 * docs/plugins/inspect/plugin-vorbis.xml:
17814 * docs/plugins/inspect/plugin-ximagesink.xml:
17815 * docs/plugins/inspect/plugin-xvimagesink.xml:
17816 Update docs with caps info.
17818 2007-06-25 12:04:15 +0000 Tim-Philipp Müller <tim@centricular.net>
17820 po/POTFILES.in: Add more files with translatable strings (#450875).
17821 Original commit message from CVS:
17823 Add more files with translatable strings (#450875).
17825 2007-06-23 14:44:07 +0000 Edward Hervey <bilboed@bilboed.com>
17827 ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
17828 Original commit message from CVS:
17829 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
17830 The chain should be freed if we error out here, else it will leak.
17831 * gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
17832 (cleanup_decodebin):
17833 Don't forget to *properly* remove the signals, else it will leak.
17835 2007-06-22 14:25:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17837 MAINTAINERS: Updating all the maintainers files
17838 Original commit message from CVS:
17840 Updating all the maintainers files
17842 2007-06-21 08:34:46 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17844 tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
17845 Original commit message from CVS:
17846 * tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
17848 Destroy and recreate parse-launch based pipeline after stop to be able
17849 to play again. Reorder some code and add more comments.
17851 2007-06-20 11:09:03 +0000 Wim Taymans <wim@fluendo.com>
17853 gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
17854 Original commit message from CVS:
17855 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
17856 When handling a delayed-caps notification case, mark
17857 the group as dynamic so that the nbdynamic count is
17858 incremented and decremented correctly. Fixes: #449156
17859 Patch by: Wim Taymans <wim@fluendo.com>
17861 2007-06-19 19:13:04 +0000 Andy Wingo <wingo@pobox.com>
17864 * gst-libs/gst/audio/gstbaseaudiosink.c:
17865 * win32/common/config.h:
17866 gst-libs/gst/audio/gstbaseaudiosink.c
17867 Original commit message from CVS:
17868 2007-06-19 Andy Wingo <wingo@pobox.com>
17869 * gst-libs/gst/audio/gstbaseaudiosink.c
17870 (gst_base_audio_sink_init): Enable pull-mode operation.
17872 2007-06-19 09:34:35 +0000 Michael Smith <msmith@xiph.org>
17874 gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
17875 Original commit message from CVS:
17876 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
17877 Change minimum rate back to 1000 to allow low-sample-rate wav files
17880 2007-06-17 17:27:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
17882 po/vi.po: Update translations.
17883 Original commit message from CVS:
17885 Update translations.
17887 2007-06-16 03:42:14 +0000 David Schleef <ds@schleef.org>
17889 gst/playback/gstqueue2.c: Fix compile error from ignored return value.
17890 Original commit message from CVS:
17891 * gst/playback/gstqueue2.c:
17892 Fix compile error from ignored return value.
17894 2007-06-15 15:23:36 +0000 Michael Smith <msmith@xiph.org>
17896 gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
17897 Original commit message from CVS:
17898 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
17899 Update tmpbuf for all neccesary rows, not just one, as is required
17903 2007-06-15 11:15:28 +0000 Michael Smith <msmith@xiph.org>
17905 tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
17906 Original commit message from CVS:
17907 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
17908 (eos_buffer_probe):
17909 Add a test that ensures we set DELTA_UNIT on all non-header,
17910 non-video buffers, if we have a video stream.
17911 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
17912 (gst_ogg_mux_process_best_pad):
17913 Move setting delta_pad to earlier, where we inspect all pads, so
17914 that leading audio pages don't get DELTA_UNIT unset if they come
17915 before the first DELTA_UNIT from video pages. Fixes the newly-added
17916 test. Fixes #385527.
17918 2007-06-14 19:53:27 +0000 Tim-Philipp Müller <tim@centricular.net>
17920 tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
17921 Original commit message from CVS:
17922 * tests/check/pipelines/streamheader.c: (streamheader_suite):
17923 Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
17924 fails on the p5-ppc64 build bot and the failure looks like it is due
17925 to the same issue as #348114, ie. a compiler bug.
17927 2007-06-13 18:20:57 +0000 Edward Hervey <bilboed@bilboed.com>
17929 gst/playback/gstqueue2.c: Fix build on MacOSX.
17930 Original commit message from CVS:
17931 * gst/playback/gstqueue2.c: (gst_queue_create_read):
17932 Fix build on MacOSX.
17934 2007-06-13 09:01:32 +0000 Wim Taymans <wim.taymans@gmail.com>
17936 ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
17937 Original commit message from CVS:
17938 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
17939 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
17940 Fix compilation on mingw. Fixes #446972.
17942 2007-06-12 08:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
17944 gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
17945 Original commit message from CVS:
17946 Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
17947 * gst/playback/gstqueue2.c: (update_buffering),
17948 (gst_queue_locked_enqueue):
17949 Fix a division by zero when the max percent is <= 0. Fixes #446572.
17950 also update the buffering status when receiving events. Fixes #446551.
17952 2007-06-11 11:32:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
17954 gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
17955 Original commit message from CVS:
17956 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
17957 * gst/playback/gstqueue2.c: (gst_queue_peer_query),
17958 (gst_queue_handle_src_query):
17959 Wait for preroll before attempting to forward a duration query upstream.
17962 2007-06-07 21:08:38 +0000 Sébastien Moutte <sebastien@moutte.net>
17964 gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
17965 Original commit message from CVS:
17966 * gst-libs/gst/rtp/gstbasertpdepayload.c:
17967 (gst_base_rtp_depayload_set_gst_timestamp):
17968 Use G_GINT64_CONSTANT macro for int64 constant.
17969 * win32/common/libgstinterfaces.def:
17970 * win32/common/libgsttag.def:
17971 Add new exported functions.
17973 2007-06-07 14:25:32 +0000 Tim-Philipp Müller <tim@centricular.net>
17975 ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
17976 Original commit message from CVS:
17977 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
17978 The BOS page of the first Dirac video stream needs to come before
17979 the BOS page of any Vorbis streams or other audio streams, just like
17982 2007-06-07 09:11:27 +0000 Wim Taymans <wim.taymans@gmail.com>
17984 gst/playback/gstqueue2.c: Fix compilation.
17985 Original commit message from CVS:
17986 * gst/playback/gstqueue2.c: (gst_queue_get_range):
17989 2007-06-06 13:36:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
17991 gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
17992 Original commit message from CVS:
17993 Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
17994 * gst/playback/gstqueue2.c: (gst_queue_init),
17995 (gst_queue_handle_sink_event), (gst_queue_chain),
17996 (gst_queue_get_range), (gst_queue_src_checkgetrange_function),
17997 (gst_queue_sink_activate_push), (gst_queue_src_activate_push),
17998 (gst_queue_src_activate_pull):
17999 Add pull based scheduling and fix some deadlocks. Fixes #444523.
18000 Does not yet completely work because duration queries upstream won't
18003 2007-06-06 09:08:50 +0000 Wim Taymans <wim.taymans@gmail.com>
18005 Some more fseeko checks.
18006 Original commit message from CVS:
18008 * gst/playback/gstqueue2.c: (gst_queue_create_read):
18009 Some more fseeko checks.
18011 2007-06-06 08:01:42 +0000 Wim Taymans <wim.taymans@gmail.com>
18013 configure.ac: check for large file support.
18014 Original commit message from CVS:
18016 check for large file support.
18018 2007-06-05 21:36:11 +0000 Sven Arvidsson <sa@whiz.se>
18020 gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
18021 Original commit message from CVS:
18022 Based on a patch by Sven Arvidsson <sa at whiz dot se>:
18023 * gst/subparse/gstsubparse.c: (parse_subrip),
18024 (subviewer_unescape_newlines), (parse_subviewer),
18025 (gst_sub_parse_data_format_autodetect),
18026 (gst_sub_parse_format_autodetect), (gst_subparse_type_find):
18027 * gst/subparse/gstsubparse.h:
18028 Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
18029 * tests/check/elements/subparse.c: (GST_START_TEST),
18031 Add a unit test for both SubViewer formats.
18033 2007-06-05 17:08:04 +0000 Michael Smith <msmith@xiph.org>
18035 gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
18036 Original commit message from CVS:
18037 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
18038 Don't overflow intermediate values when seeking to large time values
18041 2007-06-05 17:02:13 +0000 Wim Taymans <wim.taymans@gmail.com>
18043 gst/playback/gstqueue2.c: Include stdio to define fseeko.
18044 Original commit message from CVS:
18045 * gst/playback/gstqueue2.c: (gst_queue_have_data),
18046 (gst_queue_create_read), (gst_queue_read_item_from_file),
18047 (gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
18048 Include stdio to define fseeko.
18050 2007-06-05 16:37:09 +0000 Edward Hervey <edward@fluendo.com>
18052 sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
18053 Original commit message from CVS:
18054 Patch by: Edward Hervey <edward@fluendo.com>
18055 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
18056 (gst_v4lsrc_query):
18057 Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
18059 2007-06-05 16:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
18061 gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
18062 Original commit message from CVS:
18063 * gst-libs/gst/riff/Makefile.am:
18064 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
18065 Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
18066 our own implementation.
18068 2007-06-05 16:19:30 +0000 Wim Taymans <wim.taymans@gmail.com>
18070 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
18071 Original commit message from CVS:
18072 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18073 (gst_base_rtp_depayload_setcaps),
18074 (gst_base_rtp_depayload_set_gst_timestamp),
18075 (gst_base_rtp_depayload_change_state):
18076 Handle timestamp wraparound.
18078 2007-06-05 16:17:30 +0000 Wim Taymans <wim.taymans@gmail.com>
18080 gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
18081 Original commit message from CVS:
18082 * gst/playback/gsturidecodebin.c: (no_more_pads_full),
18083 (new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
18084 (gst_uri_decode_bin_change_state):
18085 Make sure we name srcpads uniquely even when using different internal
18087 Signal no-more-pads when no more dynamic elements exist.
18088 Remove pads on cleanup.
18090 2007-06-05 16:14:23 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
18092 gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
18093 Original commit message from CVS:
18094 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
18095 * gst/playback/gstqueue2.c: (gst_queue_class_init),
18096 (gst_queue_init), (gst_queue_finalize),
18097 (gst_queue_write_buffer_to_file), (gst_queue_have_data),
18098 (gst_queue_create_read), (gst_queue_read_item_from_file),
18099 (gst_queue_open_temp_location_file),
18100 (gst_queue_close_temp_location_file), (gst_queue_locked_flush),
18101 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
18102 (gst_queue_is_empty), (gst_queue_is_filled),
18103 (gst_queue_change_state), (gst_queue_set_temp_location),
18104 (gst_queue_set_property):
18105 Add support for filebased buffering. Fixes #441264.
18107 2007-06-05 16:05:19 +0000 Wim Taymans <wim.taymans@gmail.com>
18109 gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
18110 Original commit message from CVS:
18111 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
18112 (analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
18113 (caps_notify_group_cb), (gst_decode_group_new),
18114 (gst_decode_group_free):
18115 Add support for delayed caps fixation when autoplugging.
18116 Optimize cases where a multiqueue is not needed/wanted, like right after
18117 anything that is not a demuxer.
18119 2007-06-05 16:02:57 +0000 Wim Taymans <wim.taymans@gmail.com>
18121 ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
18122 Original commit message from CVS:
18123 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
18124 (gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
18125 (gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
18126 consideratly speedup ogg chain detection by not trying to find a base
18127 timestamp for skeleton streams.
18129 2007-06-05 16:00:33 +0000 Wim Taymans <wim.taymans@gmail.com>
18131 gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
18132 Original commit message from CVS:
18133 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
18134 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
18135 (gst_multi_fd_sink_remove_flush),
18136 (gst_multi_fd_sink_remove_client_link),
18137 (gst_multi_fd_sink_handle_client_write),
18138 (gst_multi_fd_sink_handle_clients):
18139 * gst/tcp/gstmultifdsink.h:
18140 Add support for remuve_flush.
18142 2007-06-05 15:59:00 +0000 Wim Taymans <wim.taymans@gmail.com>
18144 Add draft design for forcing keyframes in encoders and implement in theoraenc.
18145 Original commit message from CVS:
18146 * docs/design/draft-keyframe-force.txt:
18147 * ext/theora/theoraenc.c: (theora_enc_sink_event),
18148 (theora_enc_chain):
18149 Add draft design for forcing keyframes in encoders and implement in
18152 2007-06-05 13:22:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18154 configure.ac: Back to CVS
18155 Original commit message from CVS:
18159 === release 0.10.13 ===
18161 2007-06-05 12:50:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18167 * docs/plugins/gst-plugins-base-plugins.args:
18168 * docs/plugins/inspect/plugin-adder.xml:
18169 * docs/plugins/inspect/plugin-alsa.xml:
18170 * docs/plugins/inspect/plugin-audioconvert.xml:
18171 * docs/plugins/inspect/plugin-audiorate.xml:
18172 * docs/plugins/inspect/plugin-audioresample.xml:
18173 * docs/plugins/inspect/plugin-audiotestsrc.xml:
18174 * docs/plugins/inspect/plugin-cdparanoia.xml:
18175 * docs/plugins/inspect/plugin-decodebin.xml:
18176 * docs/plugins/inspect/plugin-decodebin2.xml:
18177 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
18178 * docs/plugins/inspect/plugin-gdp.xml:
18179 * docs/plugins/inspect/plugin-gnomevfs.xml:
18180 * docs/plugins/inspect/plugin-libvisual.xml:
18181 * docs/plugins/inspect/plugin-ogg.xml:
18182 * docs/plugins/inspect/plugin-pango.xml:
18183 * docs/plugins/inspect/plugin-playbin.xml:
18184 * docs/plugins/inspect/plugin-subparse.xml:
18185 * docs/plugins/inspect/plugin-tcp.xml:
18186 * docs/plugins/inspect/plugin-theora.xml:
18187 * docs/plugins/inspect/plugin-typefindfunctions.xml:
18188 * docs/plugins/inspect/plugin-video4linux.xml:
18189 * docs/plugins/inspect/plugin-videorate.xml:
18190 * docs/plugins/inspect/plugin-videoscale.xml:
18191 * docs/plugins/inspect/plugin-videotestsrc.xml:
18192 * docs/plugins/inspect/plugin-volume.xml:
18193 * docs/plugins/inspect/plugin-vorbis.xml:
18194 * docs/plugins/inspect/plugin-ximagesink.xml:
18195 * docs/plugins/inspect/plugin-xvimagesink.xml:
18196 * gst-plugins-base.doap:
18197 * win32/common/config.h:
18198 * win32/vs6/grammar.dsp:
18199 * win32/vs6/gst_plugins_base.dsw:
18200 * win32/vs6/libgstadder.dsp:
18201 * win32/vs6/libgstaudio.dsp:
18202 * win32/vs6/libgstaudioconvert.dsp:
18203 * win32/vs6/libgstaudiorate.dsp:
18204 * win32/vs6/libgstaudioresample.dsp:
18205 * win32/vs6/libgstaudioscale.dsp:
18206 * win32/vs6/libgstaudiotestsrc.dsp:
18207 * win32/vs6/libgstcdda.dsp:
18208 * win32/vs6/libgstdecodebin.dsp:
18209 * win32/vs6/libgstdecodebin2.dsp:
18210 * win32/vs6/libgstdirectsound.dsp:
18211 * win32/vs6/libgstffmpegcolorspace.dsp:
18212 * win32/vs6/libgstgdp.dsp:
18213 * win32/vs6/libgstinterfaces.dsp:
18214 * win32/vs6/libgstnetbuffer.dsp:
18215 * win32/vs6/libgstogg.dsp:
18216 * win32/vs6/libgstpbutils.dsp:
18217 * win32/vs6/libgstplaybin.dsp:
18218 * win32/vs6/libgstriff.dsp:
18219 * win32/vs6/libgstrtp.dsp:
18220 * win32/vs6/libgstsinesrc.dsp:
18221 * win32/vs6/libgstsubparse.dsp:
18222 * win32/vs6/libgsttag.dsp:
18223 * win32/vs6/libgsttheora.dsp:
18224 * win32/vs6/libgsttypefindfunctions.dsp:
18225 * win32/vs6/libgstutils.dsp:
18226 * win32/vs6/libgstvideo.dsp:
18227 * win32/vs6/libgstvideorate.dsp:
18228 * win32/vs6/libgstvideoscale.dsp:
18229 * win32/vs6/libgstvideotestsrc.dsp:
18230 * win32/vs6/libgstvolume.dsp:
18231 * win32/vs6/libgstvorbis.dsp:
18232 Release 0.10.13 "What's going on?"
18233 Original commit message from CVS:
18234 Release 0.10.13 "What's going on?"
18236 2007-06-05 12:32:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18254 Original commit message from CVS:
18257 2007-05-31 17:08:58 +0000 Wim Taymans <wim@fluendo.com>
18259 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...
18260 Original commit message from CVS:
18261 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
18262 In riff, the depth is stored in the size field but it just means that
18263 the least significant bits are cleared. We can therefore just play
18264 the sample as if it had a depth == width. Fixes: #440997
18265 Patch by: Wim Taymans <wim@fluendo.com>
18266 Patch by: Sebastian Dröge <slomo@circular-chaos.org>
18268 2007-05-31 16:36:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18270 gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
18271 Original commit message from CVS:
18272 * gst-libs/gst/floatcast/floatcast.h:
18273 Define inline when needed on win32 builds. Fixes: #441295
18275 2007-05-29 13:38:35 +0000 Wim Taymans <wim.taymans@gmail.com>
18277 gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
18278 Original commit message from CVS:
18279 * gst/playback/gstplaybasebin.c: (queue_overrun),
18280 (no_more_pads_full):
18281 Stop buffering when the group is commited because the queues filled up.
18284 2007-05-25 10:07:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18286 Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
18287 Original commit message from CVS:
18288 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
18289 (gst_alsa_mixer_free), (gst_alsa_mixer_update),
18290 (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
18291 (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
18292 (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
18293 * ext/alsa/gstalsamixer.h:
18294 * ext/alsa/gstalsamixerelement.c:
18295 (gst_alsa_mixer_element_interface_supported),
18296 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
18297 (gst_alsa_mixer_element_set_property),
18298 (gst_alsa_mixer_element_get_property),
18299 (gst_alsa_mixer_element_change_state):
18300 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
18301 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
18302 (gst_mixer_option_changed):
18303 * gst-libs/gst/interfaces/mixer.h:
18304 Revert commits towards #152864 made so far. We'll pick it up again
18305 after the 0.10.13 release.
18307 2007-05-24 16:22:23 +0000 Wim Taymans <wim.taymans@gmail.com>
18309 gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
18310 Original commit message from CVS:
18311 * gst-libs/gst/audio/gstbaseaudiosink.c:
18312 (gst_base_audio_sink_render):
18313 After an interrupt (PAUSED/flush) assume that the next sample should not
18314 be aligned to the previous sample. Fixes #417992.
18316 2007-05-24 15:16:59 +0000 Tim-Philipp Müller <tim@centricular.net>
18318 gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
18319 Original commit message from CVS:
18320 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
18321 Don't add channels and rate fields to the template caps for
18322 audio/x-dts, as wavparse might not always be able to set them,
18323 which would then lead to 'caps are not a real subset of the
18324 template caps' warnings.
18326 2007-05-24 11:15:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18328 gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
18329 Original commit message from CVS:
18330 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
18331 Handle unknown or invalid pads without crashing, as might occur if
18332 a media file like an mp3 is specified as a subtitle file.
18335 2007-05-24 10:19:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18337 gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
18338 Original commit message from CVS:
18339 * gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
18341 Block the subtitle bin output queue before ghosting it and linking,
18342 then unblock after. This avoids spurious not-linked errors caused
18343 by the queue starting up (because it gets linked when it is ghosted).
18346 2007-05-23 15:54:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18348 tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
18349 Original commit message from CVS:
18350 * tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
18351 Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
18352 file. Avoids flukes where the input gets typefound to some valid but
18355 2007-05-22 15:45:19 +0000 Tim-Philipp Müller <tim@centricular.net>
18357 tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
18358 Original commit message from CVS:
18359 * tests/check/Makefile.am:
18360 * tests/check/elements/.cvsignore:
18361 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
18362 (cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
18363 Add unit test for gnomevfssink seeking and position reporting for
18366 2007-05-22 15:30:26 +0000 Mark Nauwelaerts <manauw@skynet.be>
18368 ext/gnomevfs/gstgnomevfssink.*: see #412648.
18369 Original commit message from CVS:
18370 Patch by: Mark Nauwelaerts <manauw at skynet be>
18371 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
18372 (gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
18373 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
18374 * ext/gnomevfs/gstgnomevfssink.h:
18375 Fix position reporting, especially after a seek (from upstream),
18378 2007-05-22 15:04:41 +0000 Tim-Philipp Müller <tim@centricular.net>
18380 ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
18381 Original commit message from CVS:
18382 * ext/cdparanoia/gstcdparanoiasrc.c:
18385 2007-05-22 11:40:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18387 gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
18388 Original commit message from CVS:
18389 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
18390 Specify the full valid range for MP3 samplerates. Fixes a regression
18391 caused by extra header checks since the last release.
18393 2007-05-21 15:32:42 +0000 Mike Smith <msmith@xiph.org>
18395 sys/: Fix a locking-order bug I introduced with my changes the other day.
18396 Original commit message from CVS:
18397 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
18398 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
18399 Fix a locking-order bug I introduced with my changes the other day.
18400 Patch by Mike Smith.
18402 2007-05-21 15:24:21 +0000 Michael Smith <msmith@xiph.org>
18404 ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
18405 Original commit message from CVS:
18406 * ext/theora/theoradec.c: (theora_handle_data_packet):
18407 Don't look inside 0-length packets (which indicate duplicated
18410 2007-05-21 10:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
18413 Original commit message from CVS:
18414 * ext/cdparanoia/gstcdparanoiasrc.c:
18415 (gst_cd_paranoia_src_read_sector):
18416 * gst-libs/gst/audio/gstbaseaudiosrc.c:
18417 (gst_base_audio_src_create):
18419 * ext/theora/theoradec.c: (theora_dec_sink_event):
18421 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18422 (gst_base_rtp_depayload_set_gst_timestamp):
18424 * gst/playback/gstdecodebin.c: (queue_underrun_cb):
18425 And some debug info when a FIXME path is hit.
18427 2007-05-21 09:45:28 +0000 Wim Taymans <wim.taymans@gmail.com>
18429 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
18430 Original commit message from CVS:
18431 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18432 (gst_base_rtp_audio_payload_class_init),
18433 (gst_base_rtp_audio_payload_init),
18434 (gst_base_rtp_audio_payload_finalize),
18435 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
18436 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
18437 (gst_base_rtp_payload_audio_handle_event):
18438 Some cleanups, remove minptime property as it is now in the parent
18440 Override parent class event function.
18441 * gst-libs/gst/rtp/gstbasertppayload.c:
18442 (gst_basertppayload_class_init), (gst_basertppayload_init),
18443 (gst_basertppayload_event), (gst_basertppayload_set_property),
18444 (gst_basertppayload_get_property):
18445 * gst-libs/gst/rtp/gstbasertppayload.h:
18446 Add min-ptime property.
18447 Add handle-event vmethod. Fixes #415001.
18449 2007-05-18 17:10:03 +0000 Christian Schaller <uraeus@gnome.org>
18451 * gst-plugins-base.spec.in:
18453 Original commit message from CVS:
18456 2007-05-18 15:23:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18458 gst-libs/gst/audio/gstbaseaudiosink.c
18459 Original commit message from CVS:
18460 * gst-libs/gst/audio/gstbaseaudiosink.c
18461 (gst_base_audio_sink_change_state):
18462 Fix typo in comment.
18463 * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
18464 free_dynamics, pad_probe, close_pad_link, try_to_link_1,
18465 get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
18467 * gst/playback/gstplaybin.c (gst_play_bin_set_property,
18468 gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
18469 Remove trailing whitespaces in comments.
18470 * gst/volume/Makefile.am:
18473 2007-05-18 15:10:08 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
18476 * gst-libs/gst/interfaces/mixer.h:
18477 gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
18478 Original commit message from CVS:
18479 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
18480 * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
18481 set_option, get_option, _gst_reserved):
18482 Revert reordering functions (keep ABI).
18484 2007-05-17 17:35:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18486 sys/: When we create our own window, indicate that we handle the
18487 Original commit message from CVS:
18488 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
18489 (gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
18490 (gst_ximagesink_show_frame):
18491 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
18492 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
18493 (gst_xvimagesink_show_frame):
18494 When we create our own window, indicate that we handle the
18495 WM_DELETE client message from the window manager, so that it won't
18496 kill our window (and our app) along with it. Handle ClientMessage,
18497 post an error on the bus, and close the window. Further buffers
18498 arriving will result in a FlowError because the window has been
18501 Clean up the X event handling loop and make them the same for
18502 both xvimagesink and ximagesink while I'm at it.
18504 2007-05-17 16:27:32 +0000 Wim Taymans <wim.taymans@gmail.com>
18506 gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
18507 Original commit message from CVS:
18508 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
18509 Make decodebin2 autoplug depayloaders too.
18510 * gst/playback/gsturidecodebin.c: (source_new_pad):
18511 Set the newly created decoder in a usable state when autoplugging a
18512 dynamic source such as RTSP.
18514 2007-05-17 16:11:03 +0000 Tim-Philipp Müller <tim@centricular.net>
18516 gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
18517 Original commit message from CVS:
18518 * gst/playback/gststreaminfo.c: (cb_probe):
18519 Ignore video-codec tag for audio streams and ignore audio-codec tags
18520 for video streams. Should make codec name collection a bit more
18521 robust against sloppy demuxers that send tag events containing both
18522 tags down each pad.
18524 2007-05-17 15:22:44 +0000 Wim Taymans <wim.taymans@gmail.com>
18526 gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
18527 Original commit message from CVS:
18528 * gst/playback/gstqueue2.c: (update_rates):
18529 Tweak the buffering thresholds a little.
18530 Update the buffer size with the previously calculate rate instead of
18531 only when we calculate a new rate so that we get smoother buffering
18533 * gst/playback/Makefile.am:
18534 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
18535 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
18536 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
18537 (gst_uri_decode_bin_get_property), (unknown_type),
18538 (add_element_stream), (no_more_pads_full), (no_more_pads),
18539 (source_no_more_pads), (new_decoded_pad), (array_has_value),
18540 (gen_source_element), (has_all_raw_caps), (analyse_source),
18541 (remove_decoders), (make_decoder), (remove_source),
18542 (source_new_pad), (setup_source), (decoder_query_init),
18543 (decoder_query_duration_fold), (decoder_query_duration_done),
18544 (decoder_query_position_fold), (decoder_query_position_done),
18545 (decoder_query_latency_fold), (decoder_query_latency_done),
18546 (decoder_query_seeking_fold), (decoder_query_seeking_done),
18547 (decoder_query_generic_fold), (gst_uri_decode_bin_query),
18548 (gst_uri_decode_bin_change_state), (plugin_init):
18549 New element that intergrates a source, optional buffering element and
18552 2007-05-17 14:17:17 +0000 Tim-Philipp Müller <tim@centricular.net>
18554 configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
18555 Original commit message from CVS:
18557 Bump libtheora requirement to 1.0alpha5 for the pixformat check
18558 (also has a .pc file, so we don't need the fallback check any
18559 longer). Fixes #438840.
18561 2007-05-17 13:36:11 +0000 Wim Taymans <wim.taymans@gmail.com>
18563 gst/playback/gstqueue2.c: fix build.
18564 Original commit message from CVS:
18565 * gst/playback/gstqueue2.c: (gst_queue_get_type),
18566 (gst_queue_class_init), (gst_queue_finalize), (update_time_level),
18567 (apply_segment), (apply_buffer), (update_buffering),
18568 (reset_rate_timer), (update_rates), (gst_queue_locked_flush),
18569 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
18570 (gst_queue_handle_sink_event), (gst_queue_is_filled),
18571 (gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
18575 2007-05-17 11:57:44 +0000 Wim Taymans <wim.taymans@gmail.com>
18577 gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
18578 Original commit message from CVS:
18579 * gst/playback/Makefile.am:
18580 * gst/playback/gstqueue2.c: (gst_queue_get_type),
18581 (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
18582 (gst_queue_getcaps), (gst_queue_bufferalloc),
18583 (gst_queue_acceptcaps), (update_time_level), (apply_segment),
18584 (apply_buffer), (update_buffering), (reset_rate_timer),
18585 (update_rates), (gst_queue_locked_flush),
18586 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
18587 (gst_queue_handle_sink_event), (gst_queue_is_empty),
18588 (gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
18589 (gst_queue_loop), (gst_queue_handle_src_event),
18590 (gst_queue_handle_src_query), (gst_queue_sink_activate_push),
18591 (gst_queue_src_activate_push), (gst_queue_change_state),
18592 (gst_queue_set_property), (gst_queue_get_property), (plugin_init):
18593 On our way to playbin2 this is the new network queue that does buffering
18594 all by itself using high and low watermarks. It can also measure up and
18595 downstream bandwidth to optimally size the queue.
18597 2007-05-17 11:16:14 +0000 Michael Smith <msmith@xiph.org>
18599 gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
18600 Original commit message from CVS:
18601 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
18602 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
18603 Use the segment->last_stop value to calculate the next timestamp to
18604 generate after a seek; not the segment->start value.
18606 2007-05-15 20:14:06 +0000 David Schleef <ds@schleef.org>
18608 docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
18609 Original commit message from CVS:
18610 * docs/Makefile.am: Install docs even when --disable-gtk-doc
18611 is disabled. This matches the behavior of gtk+. Fixes #349099.
18613 2007-05-15 17:11:09 +0000 Wim Taymans <wim.taymans@gmail.com>
18615 ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
18616 Original commit message from CVS:
18617 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
18618 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
18619 Some more chained streaming ogg timestamp fixes.
18621 2007-05-15 16:46:10 +0000 Wim Taymans <wim.taymans@gmail.com>
18623 ext/ogg/gstoggdemux.c: Add some FIXMEs.
18624 Original commit message from CVS:
18625 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
18626 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
18627 (gst_ogg_demux_handle_page):
18629 Fix chain start/stop segment handling based on patch by
18630 <ahalda at cs dot mcgill dot ca> see #320984.
18632 2007-05-15 15:33:54 +0000 Michael Smith <msmith@xiph.org>
18634 configure.ac: We don't require a C++ compiler. So don't require one.
18635 Original commit message from CVS:
18637 We don't require a C++ compiler. So don't require one.
18639 2007-05-15 15:29:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18642 * ext/alsa/gstalsamixer.c:
18643 ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
18644 Original commit message from CVS:
18645 * ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
18646 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
18647 gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
18648 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
18649 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
18650 gst_alsa_mixer_update_track):
18651 Apply some of the cleanup Tim suggested in #152864 afterwards.
18653 2007-05-15 14:01:26 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
18655 ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
18656 Original commit message from CVS:
18657 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
18658 * ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
18659 _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
18660 gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
18661 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
18662 gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
18663 gst_alsa_mixer_handle_source_callback,
18664 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
18665 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
18666 gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
18667 gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
18668 gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
18669 gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
18670 * ext/alsa/gstalsamixer.h (handle_source, interface, dir):
18671 * ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
18672 gst_alsa_mixer_element_interface_supported,
18673 gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
18674 gst_alsa_mixer_element_set_property,
18675 gst_alsa_mixer_element_get_property,
18676 gst_alsa_mixer_element_change_state):
18677 * ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
18678 * gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
18679 gst_mixer_option_changed):
18680 * gst-libs/gst/interfaces/mixer.h (set_option, get_option,
18681 volume_changed, option_changed, _gst_reserved):
18682 Implement notification for alsamixer. Fixes #152864
18684 2007-05-15 03:53:11 +0000 David Schleef <ds@schleef.org>
18686 gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
18687 Original commit message from CVS:
18688 * gst/videotestsrc/videotestsrc.c:
18689 * gst/videotestsrc/videotestsrc.h:
18690 Add support for video/x-raw-bayer.
18692 2007-05-13 01:06:19 +0000 David Schleef <ds@schleef.org>
18694 sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
18695 Original commit message from CVS:
18696 * sys/xvimage/xvimagesink.c:
18697 Add some sanity checking for the XVImage size returned by X.
18698 Related to #377400.
18700 2007-05-12 16:18:39 +0000 Wim Taymans <wim.taymans@gmail.com>
18702 gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
18703 Original commit message from CVS:
18704 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18705 (gst_base_rtp_depayload_setcaps),
18706 (gst_base_rtp_depayload_set_gst_timestamp):
18707 Parse and use additional caps fields as described in updated
18708 application/x-rtp caps spec.
18710 2007-05-12 16:16:22 +0000 Wim Taymans <wim.taymans@gmail.com>
18712 ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
18713 Original commit message from CVS:
18714 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
18715 (gst_ogg_demux_collect_chain_info):
18716 If there is a stream in a chain without any data packets, ignore the
18717 stream in the total length calculations. Might be related to #436820.
18719 2007-05-11 17:33:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18721 gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
18722 Original commit message from CVS:
18723 * gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
18724 (mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
18725 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
18726 (mpeg_video_type_find), (mpeg_video_stream_type_find),
18728 Consolidate and re-work our mpeg system stream detection to probe
18729 more packets and produce a higher confidence result. Fixes a
18730 regression caused by lowering the typefind probability last year
18731 - related to bug #397810. Remove the redundant MPEG-1 specific
18732 typefind function, as the new one detects both MPEG-1 & MPEG-2
18734 Also cleanup the MPEG elementary and MPEG-TS detection functions a
18736 Tested against my media test directory, with some improvements and
18739 2007-05-10 15:28:13 +0000 Wim Taymans <wim.taymans@gmail.com>
18741 gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
18742 Original commit message from CVS:
18743 * gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
18744 (queue_out_of_data):
18745 Connect to the new queue "pushing" signal instead of the broken
18748 2007-05-09 21:17:40 +0000 Sébastien Moutte <sebastien@moutte.net>
18750 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
18751 Original commit message from CVS:
18752 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18753 (gst_base_rtp_audio_payload_handle_frame_based_buffer):
18754 Move variable declaration before the first instruction.
18755 * gst/videotestsrc/videotestsrc.c:
18756 Define M_PI if it's not defined yet.
18757 * win32/common/libgstrtp.def:
18758 Add new exported functions.
18760 2007-05-09 11:54:32 +0000 Michael Smith <msmith@xiph.org>
18762 ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
18763 Original commit message from CVS:
18764 * ext/theora/theoradec.c: (theora_handle_type_packet):
18765 gst_pad_push_event() does not return a GstFlowReturn!
18767 2007-05-09 11:25:34 +0000 Wim Taymans <wim.taymans@gmail.com>
18769 tests/examples/seek/: Some small cosmetic changes.
18770 Original commit message from CVS:
18771 * tests/examples/seek/scrubby.c: (stop_cb), (main):
18772 * tests/examples/seek/seek.c: (do_seek):
18773 Some small cosmetic changes.
18775 2007-05-08 19:24:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18778 * gst/adder/gstadder.c:
18779 * gst/adder/gstadder.h:
18780 gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
18781 Original commit message from CVS:
18782 * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
18783 gst_adder_change_state):
18784 * gst/adder/gstadder.h (bps, offset, collect_event, segment,
18785 segment_pending, segment_position, segment_rate):
18786 Handle playback-rate on adder.
18788 2007-05-07 11:43:31 +0000 Michael Smith <msmith@xiph.org>
18790 ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
18791 Original commit message from CVS:
18792 * ext/theora/gsttheoradec.h:
18793 * ext/theora/theoradec.c: (gst_theora_dec_reset),
18794 (theora_dec_sink_event), (theora_handle_comment_packet),
18795 (theora_handle_type_packet), (theora_dec_change_state):
18796 Don't push events (newsegment, tags) before initialising the
18798 This is neccesary for seeking to work correctly in gnonlin.
18800 2007-05-04 13:10:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18802 gst/: gst/audiotestsrc/gstaudiotestsrc.c
18803 Original commit message from CVS:
18804 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18805 * gst/adder/gstadder.c:
18806 * gst/audiotestsrc/gstaudiotestsrc.c
18807 (gst_audio_test_src_create_white_noise):
18808 * gst/videotestsrc/gstvideotestsrc.c:
18809 * gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
18810 VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
18811 volume_sink_template, volume_src_template, gst_volume_init,
18812 volume_process_double, volume_process_int16,
18813 volume_process_int16_clamp):
18814 Doc fixes and formatting.
18816 2007-05-04 12:41:21 +0000 Tim-Philipp Müller <tim@centricular.net>
18818 tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
18819 Original commit message from CVS:
18820 * tests/check/Makefile.am:
18821 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
18822 Minimal check for volume's GstController usability; also another
18825 2007-05-04 09:06:38 +0000 Tim-Philipp Müller <tim@centricular.net>
18827 gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
18828 Original commit message from CVS:
18829 * gst-libs/gst/cdda/gstcddabasesrc.c:
18830 (gst_cdda_base_src_add_track):
18831 Fix it so that it (a) makes sense and (b) doesn't break
18832 everything cdda-related including the unit test.
18834 2007-05-04 08:46:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18836 gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
18837 Original commit message from CVS:
18838 * gst-libs/gst/cdda/gstcddabasesrc.c:
18839 (gst_cdda_base_src_add_track):
18840 Fix build when disabling asserts.
18842 2007-05-03 16:29:10 +0000 Tim-Philipp Müller <tim@centricular.net>
18844 sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
18845 Original commit message from CVS:
18846 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
18847 When XShm is not available, we might get row strides that are not
18848 rounded up to multiples of four; this is bad, because virtually
18849 every RGB-processing element in GStreamer assumes rowstrides are
18850 rounded up to multiples of four, so let's allocate at least enough
18851 memory to avoid crashes in this case. The image will still be
18852 displayed distorted though if this happens, so that still needs
18853 fixing (maybe by allocating a bigger image with an 'even' width
18854 and then clipping it appropriately when rendering - something for
18855 Xlib aficionados in any case).
18857 2007-05-03 13:16:21 +0000 Michael Smith <msmith@xiph.org>
18859 gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
18860 Original commit message from CVS:
18861 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
18862 If a buffer doesn't have a timestamp, assume it's contiguous with
18863 the previous buffer, and synthesise timestamps appropriately.
18865 2007-05-03 11:24:00 +0000 Edward Hervey <bilboed@bilboed.com>
18867 tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
18868 Original commit message from CVS:
18869 * tests/check/elements/videorate.c: (GST_START_TEST):
18870 Set buffer timestamp to a valid value in order to test the buffer
18871 really does stay in videorate.
18873 2007-05-03 10:47:22 +0000 Edward Hervey <bilboed@bilboed.com>
18875 gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
18876 Original commit message from CVS:
18877 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
18878 There is no sensible way to handle incoming buffers which don't have a
18879 valid timestamp. We therefore discard them and wait for the next one.
18881 2007-05-01 18:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
18883 gst/playback/: Better error message for text files.
18884 Original commit message from CVS:
18885 * gst/playback/gstdecodebin.c: (type_found), (plugin_init):
18886 * gst/playback/gstdecodebin2.c: (plugin_init):
18887 Better error message for text files.
18889 2007-04-29 14:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
18891 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
18892 Original commit message from CVS:
18893 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
18894 Fix offset bug in generation RR packets.
18896 2007-04-27 15:33:46 +0000 Julien Moutte <julien@moutte.net>
18898 ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
18899 Original commit message from CVS:
18900 2007-04-27 Julien MOUTTE <julien@moutte.net>
18901 * ext/theora/theoradec.c: (_theora_granule_time),
18902 (theora_dec_push_forward), (theora_handle_data_packet),
18903 (theora_dec_decode_buffer): Calculate buffer duration correctly
18904 to generate a perfect stream (#433888).
18905 * gst/audioresample/gstaudioresample.c:
18906 (audioresample_check_discont): Glib provides ABS.
18908 2007-04-27 15:01:40 +0000 Wim Taymans <wim.taymans@gmail.com>
18910 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
18911 Original commit message from CVS:
18912 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
18913 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
18914 (gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
18915 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
18916 (gst_rtcp_packet_bye_set_reason):
18917 * gst-libs/gst/rtp/gstrtcpbuffer.h:
18918 Fix RB block parsing and writing.
18919 Add support for constructing BYE packets.
18921 2007-04-25 08:54:34 +0000 Tim-Philipp Müller <tim@centricular.net>
18923 When posting a warning message because samples were dropped, post something more intelligible than he default error m...
18924 Original commit message from CVS:
18925 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
18926 (gst_base_audio_src_create):
18928 When posting a warning message because samples were dropped, post
18929 something more intelligible than he default error message for clock
18930 errors which is just confusing in this context (#432984).
18932 2007-04-25 08:10:26 +0000 Wim Taymans <wim.taymans@gmail.com>
18934 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
18935 Original commit message from CVS:
18936 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
18937 (gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
18938 (read_packet_header), (gst_rtcp_packet_move_to_next),
18939 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
18940 (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
18941 (gst_rtcp_packet_sdes_get_item_count),
18942 (gst_rtcp_packet_sdes_first_item),
18943 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
18944 (gst_rtcp_packet_sdes_first_entry),
18945 (gst_rtcp_packet_sdes_next_entry),
18946 (gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
18947 (gst_rtcp_packet_sdes_add_entry):
18948 * gst-libs/gst/rtp/gstrtcpbuffer.h:
18949 Implement code to write SR, RR and SDES packets.
18951 2007-04-24 20:45:24 +0000 Christian Kirbach <Christian.Kirbach@googlemail.com>
18953 sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
18954 Original commit message from CVS:
18955 Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
18956 * sys/ximage/ximagesink.c:
18957 Fix build if XShm is not available (#432362).
18959 2007-04-24 18:58:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18961 gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
18962 Original commit message from CVS:
18963 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
18964 Initalize the AudioConvertCtx with zeroes, otherwise it will contain
18965 pointers to random memory which are passed to g_free() when
18966 audio_convert_prepare_context() is called the first time.
18968 2007-04-24 15:00:07 +0000 Dan Williams <dcbw@redhat.com>
18970 gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
18971 Original commit message from CVS:
18972 Patch by: Dan Williams <dcbw redhat com>
18973 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
18974 Don't leak incoming buffer if gst_pad_push() returns a
18975 non-OK flow. Fixes #432755.
18976 * tests/check/elements/videorate.c: (GST_START_TEST),
18978 Unit test for the above by Yours Truly.
18980 2007-04-23 20:04:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18982 gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
18983 Original commit message from CVS:
18984 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
18985 (gst_adder_sink_event), (gst_adder_collected):
18986 Fix non-flushing segmented seeks, Fixes #340060 for me
18988 2007-04-21 15:29:27 +0000 Tim-Philipp Müller <tim@centricular.net>
18991 ChangeLog surgery: add API keyword
18992 Original commit message from CVS:
18993 ChangeLog surgery: add API keyword
18995 2007-04-21 15:25:22 +0000 Olivier Crete <tester@tester.ca>
18997 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
18998 Original commit message from CVS:
18999 Patch by: Olivier Crete <tester at tester ca>
19000 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19001 (gst_base_rtp_audio_payload_class_init),
19002 (gst_base_rtp_audio_payload_init),
19003 (gst_base_rtp_audio_payload_dispose):
19004 Chain up to parent class in dispose function; get rid of
19005 unnecessary 'diposed' flag in private structure (#415001).
19007 2007-04-21 15:10:25 +0000 Tim-Philipp Müller <tim@centricular.net>
19009 Some minor docs fixes and additions; also add missing 'Since' bits.
19010 Original commit message from CVS:
19011 * docs/libs/gst-plugins-base-libs.types:
19012 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19013 (gst_base_rtp_audio_payload_class_init):
19014 * gst-libs/gst/rtp/gstbasertpdepayload.c:
19015 * gst-libs/gst/rtp/gstbasertppayload.c:
19016 Some minor docs fixes and additions; also add missing 'Since' bits.
19018 2007-04-21 14:40:45 +0000 Zeeshan Ali <zeenix@gmail.com>
19020 gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
19021 Original commit message from CVS:
19022 Patch by: Zeeshan Ali <zeenix gmail com>
19023 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19024 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
19025 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
19026 (gst_base_rtp_audio_payload_push):
19027 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
19028 The recently-added gst_base_rtp_audio_payload_push() should take an
19029 object of type GstBaseRTPAudioPayload as first argument (#431672).
19031 2007-04-21 14:14:24 +0000 Tim-Philipp Müller <tim@centricular.net>
19033 gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
19034 Original commit message from CVS:
19035 * gst/audioresample/gstaudioresample.c:
19036 Make more functions static, just because we can.
19038 2007-04-21 13:54:39 +0000 Tim-Philipp Müller <tim@centricular.net>
19040 tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
19041 Original commit message from CVS:
19042 * tests/check/elements/audioresample.c:
19043 Add unit test for audioresample shutdown crasher (#420106).
19045 2007-04-20 10:42:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19047 gst/subparse/: Use GST_DISABLE_XML here
19048 Original commit message from CVS:
19049 * gst/subparse/gstsubparse.c:
19050 * gst/subparse/samiparse.c:
19051 Use GST_DISABLE_XML here
19052 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
19053 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
19054 (gst_xvimagesink_buffer_alloc),
19055 (gst_xvimagesink_navigation_send_event):
19056 * sys/xvimage/xvimagesink.h:
19057 Include stdlib.h when using atoi.
19058 * tests/check/elements/playbin.c: (playbin_suite):
19059 Use GST_DISABLE_REGISTRY here
19061 2007-04-19 16:58:53 +0000 Michael Smith <msmith@xiph.org>
19063 ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
19064 Original commit message from CVS:
19065 * ext/theora/gsttheoraenc.h:
19066 * ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
19067 (theora_enc_sink_event), (theora_enc_change_state):
19068 Track initialisation state; don't try to use encoder state if we're
19069 not initialised (it'll segfault).
19071 2007-04-18 11:06:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19073 tests/check/pipelines/.cvsignore: Fix build.
19074 Original commit message from CVS:
19075 * tests/check/pipelines/.cvsignore:
19078 2007-04-17 10:56:37 +0000 Tim-Philipp Müller <tim@centricular.net>
19080 gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
19081 Original commit message from CVS:
19082 * gst/app/Makefile.am:
19083 Fix CFLAGS and hopefully #430594.
19085 2007-04-17 02:53:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19087 gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
19088 Original commit message from CVS:
19089 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19090 Allow random depths between 1 and 32 instead of only multiplies of 8.
19092 2007-04-17 02:04:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19094 gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
19095 Original commit message from CVS:
19096 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19097 Set the maximum number of channels for PCM and float in the correct
19098 place to have it also used when creating the template caps.
19100 2007-04-17 01:56:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19102 gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
19103 Original commit message from CVS:
19104 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19105 Correctly support 4, 6 and 8 channels with normal PCM and float
19107 Fix the depth and signedness calculation in extensible wav files and
19108 also handle 1, 2, 4, 6, 8 channels here when a file without channel
19110 Add support for float, alaw and mulaw in extensible wav files.
19111 This allows correct playback of all but 5 files from
19112 http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
19113 (gst_riff_create_audio_template_caps):
19114 Add voxware and float formats to the template caps.
19116 2007-04-16 22:20:03 +0000 Vincent Torri <vtorri@univ-evry.fr>
19118 ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
19119 Original commit message from CVS:
19120 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
19121 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
19122 Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
19123 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19124 * gst/audioresample/gstaudioresample.c: (audioresample_do_output):
19125 Use the correct format strings for integer formats.
19127 2007-04-16 21:44:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19129 * gst-plugins-base.doap:
19131 Original commit message from CVS:
19134 2007-04-16 21:42:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19136 * gst-plugins-base.doap:
19138 Original commit message from CVS:
19141 2007-04-15 14:35:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19143 ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
19144 Original commit message from CVS:
19145 * ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
19146 Don't use pad_alloc_buffer_and_set_caps to create a small header
19147 packet, or, worse, to create a big temporary video buffer using the
19150 2007-04-14 12:34:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19152 gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19153 Original commit message from CVS:
19154 * gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
19155 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19156 GST_START_TEST, buffer_probe_cb, GST_START_TEST):
19157 Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
19159 2007-04-13 22:10:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19161 * gst/tcp/gstmultifdsink.c:
19163 Original commit message from CVS:
19166 2007-04-13 21:55:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19169 * tests/check/pipelines/streamheader.c:
19170 tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19171 Original commit message from CVS:
19172 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19173 GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
19174 streamheader_suite):
19175 Add another test set up for failure
19177 2007-04-13 21:09:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19179 * ext/ogg/gstoggmux.c:
19180 * gst/gdp/gstgdpdepay.c:
19182 Original commit message from CVS:
19185 2007-04-13 21:08:11 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19187 tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
19188 Original commit message from CVS:
19189 * tests/check/Makefile.am:
19190 * tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
19191 GST_START_TEST, streamheader_suite, main):
19192 Add a test for the streamheader bug Wim fixed.
19194 2007-04-13 11:42:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19196 ext/theora/theoradec.c: Fix misleading comment.
19197 Original commit message from CVS:
19198 * ext/theora/theoradec.c: (theora_dec_sink_event):
19199 Fix misleading comment.
19201 2007-04-13 06:17:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19203 gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
19204 Original commit message from CVS:
19205 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19206 More sanity checks for the header fields.
19208 2007-04-12 16:36:36 +0000 Tim-Philipp Müller <tim@centricular.net>
19210 gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
19211 Original commit message from CVS:
19212 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
19213 Try encodings from all environment variables, not just those in the
19214 first environment variable that is set.
19216 2007-04-12 15:00:03 +0000 Wim Taymans <wim.taymans@gmail.com>
19218 gst/videorate/gstvideorate.c: Add some debug.
19219 Original commit message from CVS:
19220 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
19221 (gst_video_rate_chain):
19223 * tests/check/elements/videorate.c: (GST_START_TEST),
19225 Added check for videorate changing caps handling. Closes #421834.
19227 2007-04-12 12:57:33 +0000 Michael Smith <msmith@xiph.org>
19229 ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
19230 Original commit message from CVS:
19231 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
19232 Use scale functions to avoid overflow when calculating duration of
19235 2007-04-12 12:19:20 +0000 Tim-Philipp Müller <tim@centricular.net>
19237 API: add gst_tag_freeform_string_to_utf8() (#405072).
19238 Original commit message from CVS:
19239 * docs/libs/gst-plugins-base-libs-sections.txt:
19240 * gst-libs/gst/tag/tag.h:
19241 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
19242 API: add gst_tag_freeform_string_to_utf8() (#405072).
19243 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
19244 Use gst_tag_freeform_string_to_utf8() here.
19246 2007-04-12 10:38:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19248 * gst/tcp/gstmultifdsink.c:
19250 Original commit message from CVS:
19253 2007-04-12 10:03:22 +0000 Wim Taymans <wim.taymans@gmail.com>
19255 gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
19256 Original commit message from CVS:
19257 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
19258 (gst_gdp_pay_sink_event):
19259 Make sure we set the IN_CAPS flag correctly.
19260 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
19261 Get the IN_CAPS flag before we call functions that mess with the flags.
19263 2007-04-10 20:37:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19266 * gst/gdp/gstgdppay.c:
19267 gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
19268 Original commit message from CVS:
19269 * gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
19270 gst_gdp_pay_chain, gst_gdp_pay_sink_event):
19271 Only stamp buffers with offset/offset_end right before they get
19272 pushed. This ensures offset continuity, which was not the case
19274 gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
19276 2007-04-10 20:25:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19278 * gst/gdp/gstgdpdepay.c:
19279 * gst/gdp/gstgdppay.c:
19281 Original commit message from CVS:
19284 2007-04-10 11:23:18 +0000 Christian Schaller <uraeus@gnome.org>
19287 * gst-plugins-base.spec.in:
19288 update spec file for RTP changes
19289 Original commit message from CVS:
19290 update spec file for RTP changes
19292 2007-04-06 12:58:06 +0000 Wim Taymans <wim.taymans@gmail.com>
19294 gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
19295 Original commit message from CVS:
19296 * gst/playback/gstplaybin.c: (add_sink),
19297 (gst_play_bin_change_state):
19298 Activate sync in playbin, we are ready to handle it for live streams.
19300 2007-04-06 09:56:18 +0000 Tim-Philipp Müller <tim@centricular.net>
19302 tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
19303 Original commit message from CVS:
19304 * tests/check/elements/playbin.c:
19305 (test_sink_usage_video_only_stream), (playbin_suite):
19306 Add small test for stream-info-value-array code paths.
19308 2007-04-05 15:44:40 +0000 Wim Taymans <wim.taymans@gmail.com>
19310 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
19311 Original commit message from CVS:
19312 * gst-libs/gst/audio/gstbaseaudiosink.c:
19313 (gst_base_audio_sink_skew_slaving):
19314 Don't try to create invalid calibration parameters by making the
19315 internal time go backwards, instead make external time go forward.
19317 2007-04-05 10:27:06 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
19319 gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
19320 Original commit message from CVS:
19321 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
19322 * gst/playback/gstplaybasebin.c: (add_stream):
19323 Fix leak in add_stream(), when g_value_set_object() increases the
19324 refcount of streaminfo object. Fixes #426250.
19326 2007-04-04 02:45:03 +0000 David Schleef <ds@schleef.org>
19328 gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
19329 Original commit message from CVS:
19330 * gst/videotestsrc/gstvideotestsrc.c:
19331 * gst/videotestsrc/gstvideotestsrc.h:
19332 * gst/videotestsrc/videotestsrc.c:
19333 * gst/videotestsrc/videotestsrc.h:
19334 Add a test pattern called "circular", which has concentric
19335 rings with varying radial frequency. The main purpose of this
19336 pattern is to test fidelity loss in a filter or scaler element.
19337 Notably, this pattern is scale invariant, and is optimally viewed
19338 with a width (and height) of 400.
19340 2007-04-03 11:10:52 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
19342 gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
19343 Original commit message from CVS:
19344 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
19345 * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
19346 (deactivate_free_recursive):
19347 Decodebin2 doesn't unref pads it obtains in some occasions:
19348 - multiqueue src pads, when either connecting further or exposing
19349 - sink pads of new autoplugged elements
19350 - peer pads when recursively freeing elements
19353 2007-03-30 17:05:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19355 gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
19356 Original commit message from CVS:
19357 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19358 Add audio/x-raw-float support, now that audioconvert support
19359 non-native endianness floats.
19361 2007-03-30 15:00:49 +0000 Tim-Philipp Müller <tim@centricular.net>
19363 docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
19364 Original commit message from CVS:
19365 * docs/libs/gst-plugins-base-libs-docs.sgml:
19366 gstreamer-plugins-base.pc doesn't exist, it's
19367 gstreamer-plugins-base-0.10.pc.
19369 2007-03-29 18:42:34 +0000 René Stadler <mail@renestadler.de>
19371 with some minor changes
19372 Original commit message from CVS:
19373 Patch by: René Stadler <mail at renestadler dot de>
19374 with some minor changes
19375 * gst-libs/gst/floatcast/floatcast.h:
19376 Use more efficient float endianness conversion functions that don't
19377 involve 2 function calls per value.
19378 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
19379 (check_default), (audio_convert_prepare_context):
19380 * gst/audioconvert/gstaudioconvert.c:
19381 (gst_audio_convert_parse_caps), (make_lossless_changes):
19382 Support non-native endianness floats as input and output.
19384 * tests/check/elements/audioconvert.c: (verify_convert),
19386 Add unit tests for the non-native endianness float conversions.
19388 2007-03-29 16:23:53 +0000 Wim Taymans <wim.taymans@gmail.com>
19390 gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
19391 Original commit message from CVS:
19392 * gst-libs/gst/rtp/gstbasertpdepayload.c:
19393 (gst_base_rtp_depayload_base_init),
19394 (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
19395 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
19396 (gst_base_rtp_depayload_set_gst_timestamp),
19397 (gst_base_rtp_depayload_change_state),
19398 (gst_base_rtp_depayload_set_property),
19399 (gst_base_rtp_depayload_get_property):
19400 * gst-libs/gst/rtp/gstbasertpdepayload.h:
19401 Add Private structure.
19402 Bring element code to 2007.
19403 Parse clock-base caps param and use it when generating the
19405 Reset variables before going to PAUSED.
19408 2007-03-29 16:20:31 +0000 Wim Taymans <wim.taymans@gmail.com>
19411 Original commit message from CVS:
19412 * docs/libs/gst-plugins-base-libs-docs.sgml:
19413 * docs/libs/gst-plugins-base-libs-sections.txt:
19414 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19415 (gst_base_rtp_audio_payload_get_adapter):
19417 Fix some more docs.
19418 * gst-libs/gst/rtp/Makefile.am:
19419 * gst-libs/gst/rtp/gstrtcpbuffer.c:
19420 (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
19421 (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
19422 (gst_rtcp_buffer_get_packet_count), (read_packet_header),
19423 (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
19424 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
19425 (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
19426 (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
19427 (gst_rtcp_packet_sr_get_sender_info),
19428 (gst_rtcp_packet_sr_set_sender_info),
19429 (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
19430 (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
19431 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
19432 (gst_rtcp_packet_sdes_get_chunk_count),
19433 (gst_rtcp_packet_sdes_first_chunk),
19434 (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
19435 (gst_rtcp_packet_sdes_first_item),
19436 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
19437 (gst_rtcp_packet_bye_get_ssrc_count),
19438 (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
19439 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
19440 (gst_rtcp_packet_bye_get_reason_len),
19441 (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
19442 * gst-libs/gst/rtp/gstrtcpbuffer.h:
19443 Add new helper object for parsing and creating RTCP messages.
19445 2007-03-29 12:07:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19447 gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
19448 Original commit message from CVS:
19449 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19450 PCM samples with width=8 must be always unsigned, no matter what
19453 2007-03-29 11:24:47 +0000 Andy Wingo <wingo@pobox.com>
19455 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
19456 Original commit message from CVS:
19457 2007-03-29 Andy Wingo <wingo@pobox.com>
19458 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
19459 perfect offsets also, not just timestamps.
19460 * tests/check/elements/videorate.c (test_more): Test that given
19461 any incoming offsets, that videorate produces perfect offsets.
19463 2007-03-29 10:19:45 +0000 Wim Taymans <wim.taymans@gmail.com>
19465 gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
19466 Original commit message from CVS:
19467 * gst-libs/gst/riff/riff-ids.h:
19468 Add some more RIFF formats.
19470 2007-03-29 10:17:52 +0000 Wim Taymans <wim.taymans@gmail.com>
19472 gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
19473 Original commit message from CVS:
19474 * gst-libs/gst/rtp/gstrtpbuffer.c:
19475 (gst_rtp_buffer_default_clock_rate):
19476 * gst-libs/gst/rtp/gstrtpbuffer.h:
19477 Fix fixed payload names and docs.
19478 Added method to get the default clock rates of fixed payload types.
19479 API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
19481 2007-03-28 15:24:40 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
19483 tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
19484 Original commit message from CVS:
19485 * tests/check/pipelines/.cvsignore:
19486 Add new vorbisdec test to cvsignore.
19488 2007-03-28 14:50:47 +0000 Wim Taymans <wim.taymans@gmail.com>
19490 gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
19491 Original commit message from CVS:
19492 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
19493 (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
19494 (gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
19495 (gst_base_audio_sink_set_property),
19496 (gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
19497 (clock_convert_external), (gst_base_audio_sink_resample_slaving),
19498 (gst_base_audio_sink_skew_slaving),
19499 (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
19500 (gst_base_audio_sink_async_play):
19501 * gst-libs/gst/audio/gstbaseaudiosink.h:
19502 Store private stuff in GstBaseAudioSinkPrivate.
19503 Add configurable clock slaving modes property.
19504 API:: GstBaseAudioSink::slave-method property
19505 Some more latency reporting tweaks.
19506 Added skew based clock slaving correction and make it the default until
19507 the resampling method is more robust.
19509 2007-03-27 12:44:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19511 gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
19512 Original commit message from CVS:
19513 * gst/audioconvert/audioconvert.c:
19514 Add docs to the integer pack functions and implement proper
19515 rounding. Before we had rounding towards negative infinity, i.e.
19516 always the smaller number was taken. Now we use natural rounding,
19517 i.e. rounding to the nearest integer and to the one with the largest
19518 absolute value for X.5. The old rounding introduced some minor
19519 distortions. Fixes #420079
19520 * tests/check/elements/audioconvert.c: (GST_START_TEST):
19521 Fix one unit test that assumed the old rounding and added unit tests
19522 for checking signed/unsigned int16 <-> signed/unsigned int16 with
19523 depth 8, one for signed int16 <-> unsigned int16 and one for the new
19524 rounding from signed int32 to signed/unsigned int16.
19526 2007-03-27 11:31:17 +0000 Michael Smith <msmith@xiph.org>
19528 gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
19529 Original commit message from CVS:
19530 * gst/audioconvert/gstaudioconvert.c: (strip_width_64),
19531 (gst_audio_convert_transform_caps):
19532 Fix typo in debug line introduced recently, as pointed out on irc.
19534 2007-03-27 10:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
19536 Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
19537 Original commit message from CVS:
19538 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
19539 * tests/check/libs/tag.c: (GST_START_TEST):
19540 Make sure we parse floating-point numbers in vorbis comments
19541 correctly with either '.' or ',' as separator, no matter what
19542 the current locale is. Add unit test for this too.
19544 2007-03-27 09:37:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19546 * tests/check/pipelines/vorbisdec.c:
19548 Original commit message from CVS:
19551 2007-03-26 22:38:19 +0000 René Stadler <mail@renestadler.de>
19553 gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
19554 Original commit message from CVS:
19555 Patch by: René Stadler <mail at renestadler de>
19556 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
19557 When writing out floating-point numbers to vorbis comment tags, always
19558 use the same character as separator no matter what the current locale is
19560 * tests/check/libs/tag.c: (GST_START_TEST):
19561 Add unit tests for replaygain tags in vorbis comments (closes #423055).
19563 2007-03-26 20:56:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19565 ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
19566 Original commit message from CVS:
19567 * ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
19568 vorbis_handle_data_packet):
19569 Correctly set DURATION to generate a timestamp-continuous stream.
19570 One bug left at the end; see
19571 ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
19572 * tests/check/Makefile.am:
19573 * tests/check/pipelines/vorbisenc.c (GST_START_TEST):
19574 Add a test to check this. Without the above patch this test fails.
19576 2007-03-26 11:44:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19578 gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
19579 Original commit message from CVS:
19580 * gst-libs/gst/rtp/Makefile.am:
19581 The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
19583 2007-03-23 15:43:24 +0000 Christian Schaller <uraeus@gnome.org>
19585 * gst-plugins-base.spec.in:
19587 Original commit message from CVS:
19590 2007-03-23 12:32:33 +0000 Michael Smith <msmith@xiph.org>
19592 gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
19593 Original commit message from CVS:
19594 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
19595 (gst_video_rate_reset), (gst_video_rate_chain):
19596 If videorate changes caps, we can no longer use the old buffer
19597 (which may have a different size, incompatible with our caps).
19598 So don't do that; just duplicate the new frame more times.
19600 2007-03-22 17:43:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19602 gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
19603 Original commit message from CVS:
19604 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
19605 Remove playbin's override of the set_clock vmethod. It's irrelevant
19606 after Wim's commit on the 19th.
19608 2007-03-22 14:37:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19610 gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
19611 Original commit message from CVS:
19612 * gst-libs/gst/app/Makefile.am:
19613 Use GST_ALL_LDFLAGS, which actually exists, but maybe David
19614 can confirm that was what he wanted.
19616 2007-03-22 09:26:02 +0000 Wim Taymans <wim.taymans@gmail.com>
19618 ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
19619 Original commit message from CVS:
19620 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
19621 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
19622 * ext/gnomevfs/gstgnomevfssrc.h:
19623 Don't cache file sizes. Fixes #341078.
19625 2007-03-21 11:03:23 +0000 Tim-Philipp Müller <tim@centricular.net>
19627 gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
19628 Original commit message from CVS:
19629 * gst/playback/gstplaybin.c: (add_sink):
19630 Use GST_PTR_FORMAT to log caps.
19632 2007-03-21 10:23:11 +0000 Young-Ho Cha <ganadist@chollian.net>
19634 gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
19635 Original commit message from CVS:
19636 Patch by: Young-Ho Cha <ganadist at chollian net>
19637 * gst/subparse/samiparse.c: (handle_start_font):
19638 Special-case some more colour names that pango doesn't handle by
19639 default. Fixes #420578.
19641 2007-03-20 11:49:55 +0000 Michael Smith <msmith@xiph.org>
19643 ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
19644 Original commit message from CVS:
19645 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
19646 If we get a zero-sized input buffer, don't pass it to libvorbis, as
19647 that marks EOS internally. After that, libvorbis will buffer all
19648 input data, and encode none of it, eventually leading to memory
19651 2007-03-19 10:52:50 +0000 Wim Taymans <wim.taymans@gmail.com>
19653 gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
19654 Original commit message from CVS:
19655 * gst/playback/gstdecodebin.c: (remove_fakesink):
19656 Don't post STATE_DIRTY anymore.
19657 * gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
19658 (gst_play_bin_change_state):
19659 Remove stream_time reset in seek handling, core does that now.
19660 Disable clocking for live pipelines by forcing a NULL clock to the
19661 complete pipeline, core is too smart now for our previous hack.
19662 We can always autoplug in PAUSED now.
19664 2007-03-18 03:14:01 +0000 David Schleef <ds@schleef.org>
19666 REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
19667 Original commit message from CVS:
19668 * REQUIREMENTS: Update this file, change the formatting to make
19669 it more consistent, plus more machine readable.
19671 2007-03-16 17:29:09 +0000 Michael Smith <msmith@xiph.org>
19673 gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
19674 Original commit message from CVS:
19675 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
19676 (strip_width_64), (append_with_other_format):
19677 Previous fix was too simplistic, and broke the tests. Use a better
19678 approach; only strip 64 from widths for integer audio.
19680 2007-03-16 16:42:23 +0000 Michael Smith <msmith@xiph.org>
19682 gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
19683 Original commit message from CVS:
19684 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
19685 (gst_audio_convert_transform_caps):
19686 We don't support 64 bit integer audio, so don't try to claim we can.
19687 Stops us producing caps don't match our template caps.
19690 2007-03-15 10:52:21 +0000 Michael Smith <msmith@xiph.org>
19692 gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
19693 Original commit message from CVS:
19694 * gst/audioresample/gstaudioresample.c:
19695 (audioresample_check_discont), (audioresample_transform):
19696 Don't trigger discontinuities for very small imperfections; a filter
19697 flush will sound bad, and many plugins have rounding errors leading
19700 2007-03-14 21:11:18 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
19702 gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
19703 Original commit message from CVS:
19704 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19705 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
19706 Add min-ptime property to RTP base audio payloader. Patch by
19707 olivier.crete@collabora.co.uk.
19709 Indentation/whitespace/documentation fixes.
19711 2007-03-14 17:16:30 +0000 Julien Moutte <julien@moutte.net>
19713 gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
19714 Original commit message from CVS:
19715 2007-03-14 Julien MOUTTE <julien@moutte.net>
19716 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
19717 (audioresample_transform_size), (audioresample_do_output),
19718 (audioresample_transform), (audioresample_pushthrough): Handle
19719 discontinuous streams.
19720 * gst/audioresample/gstaudioresample.h:
19721 * tests/check/elements/audioresample.c:
19722 (test_discont_stream_instance), (GST_START_TEST),
19723 (audioresample_suite): Add a test for discontinuous streams.
19724 * win32/common/config.h: Updated.
19726 2007-03-14 15:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19728 po/: Update translations from translation project.
19729 Original commit message from CVS:
19743 Update translations from translation project.
19745 2007-03-14 15:05:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19747 * gst/gdp/gstgdpdepay.c:
19749 Original commit message from CVS:
19752 2007-03-14 14:48:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19754 gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
19755 Original commit message from CVS:
19756 * gst/audioresample/debug.h:
19757 * gst/audioresample/resample.c: (resample_init):
19758 Since I really am not interested in a debug line for each sample
19759 being processed, move the library's debugging to its own category,
19762 2007-03-14 14:09:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19764 * gst/audioresample/gstaudioresample.c:
19765 add debugging and reformat docs
19766 Original commit message from CVS:
19767 add debugging and reformat docs
19769 2007-03-12 23:29:07 +0000 Michael Smith <msmith@xiph.org>
19771 ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
19772 Original commit message from CVS:
19773 * ext/theora/theoradec.c: (theora_handle_type_packet):
19774 Since the plugin doesn't support anything other than 4:2:0 right
19775 now, post an error and fail if we get something else. Won't matter
19776 until libtheora supports the other pixel formats, but hopefully
19779 2007-03-12 15:50:35 +0000 Alex Lancaster <alexlan@fedoraproject.org>
19782 I'm too lazy to comment this
19783 Original commit message from CVS:
19784 Mention Patch by: Alex Lancaster in a recent commit.
19786 2007-03-12 11:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19788 examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.
19789 Original commit message from CVS:
19790 * examples/app/.cvsignore:
19791 The buildbot demands .cvsignore files, and I comply.
19793 2007-03-11 00:48:26 +0000 David Schleef <ds@schleef.org>
19795 Add appsrc/appsink example.
19796 Original commit message from CVS:
19798 * examples/Makefile.am:
19799 * examples/app/Makefile.am:
19800 * examples/app/appsrc_ex.c:
19801 Add appsrc/appsink example.
19802 * gst-libs/gst/app/Makefile.am:
19803 * gst-libs/gst/app/gstapp.c:
19804 * gst-libs/gst/app/gstappsink.c:
19805 * gst-libs/gst/app/gstappsink.h:
19806 * gst/app/gstapp.c:
19809 2007-03-10 15:59:33 +0000 Sébastien Moutte <sebastien@moutte.net>
19811 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
19812 Original commit message from CVS:
19813 * gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
19814 Use gst_guint64_to_gdouble for conversion.
19816 Add new files to the win32 MANIFEST.
19817 * win32/common/libgstaudio.def:
19818 * win32/common/libgstpbutils.def:
19819 Add new exported functions.
19820 * win32/vs6/gst_plugins_base.dsw:
19821 * win32/vs6/libgstdecodebin.dsp:
19822 * win32/vs6/libgstplaybin.dsp:
19823 Change the link to libgstpbutils.lib.
19824 * win32/vs6/libgstdecodebin2.dsp:
19825 Add a new project for decodebin2.
19826 * win32/vs6/libgstpbutils.dsp:
19827 Add a new project for pbutils.
19829 2007-03-10 12:18:58 +0000 Tim-Philipp Müller <tim@centricular.net>
19831 gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
19832 Original commit message from CVS:
19833 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
19834 Also accept partial dates with only year and month,
19835 like 1999-12-00 (fixes #410396 even more).
19836 * tests/check/libs/tag.c: (GST_START_TEST):
19837 Add unit test for the above.
19839 2007-03-10 11:21:08 +0000 Tim-Philipp Müller <tim@centricular.net>
19841 tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
19842 Original commit message from CVS:
19843 * tests/check/elements/subparse.c: (GST_START_TEST),
19845 Add unit test for MPL2 subtitle format (#413799).
19847 2007-03-10 11:17:52 +0000 Kamil Pawlowski <kamilpe@gmail.com>
19849 gst/subparse/: Add support for MPL2 subtitle format (#413799).
19850 Original commit message from CVS:
19851 Patch by: Kamil Pawlowski <kamilpe gmail com>
19852 * gst/subparse/Makefile.am:
19853 * gst/subparse/gstsubparse.c:
19854 (gst_sub_parse_data_format_autodetect),
19855 (gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
19856 (gst_subparse_type_find):
19857 * gst/subparse/gstsubparse.h:
19858 * gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
19859 * gst/subparse/mpl2parse.h:
19860 Add support for MPL2 subtitle format (#413799).
19862 2007-03-09 17:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
19864 configure.ac: We require core CVS for the new buffer metadata copy functions.
19865 Original commit message from CVS:
19867 We require core CVS for the new buffer metadata copy functions.
19869 2007-03-09 16:51:13 +0000 Wim Taymans <wim.taymans@gmail.com>
19871 gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
19872 Original commit message from CVS:
19873 * gst-libs/gst/tag/gstid3tag.c:
19874 Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
19877 2007-03-09 16:46:35 +0000 Wim Taymans <wim.taymans@gmail.com>
19879 ext/libvisual/visual.c: Improve adapter usage and comments.
19880 Original commit message from CVS:
19881 * ext/libvisual/visual.c: (gst_visual_sink_setcaps),
19882 (gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
19883 Improve adapter usage and comments.
19885 2007-03-09 16:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
19887 Use new metadata copy function.
19888 Original commit message from CVS:
19889 * ext/pango/gsttextrender.c: (gst_text_render_chain):
19890 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
19891 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
19892 Use new metadata copy function.
19893 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
19894 (gst_ffmpegcsp_transform):
19895 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
19896 Basetransform copied the metadata for us.
19898 2007-03-09 16:28:04 +0000 Tim-Philipp Müller <tim@centricular.net>
19900 ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
19901 Original commit message from CVS:
19902 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
19903 (gst_text_overlay_video_event):
19904 Some more logging. Only accept newsegment events in TIME format and
19905 send a WARNING message if they are not in TIME format.
19906 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
19907 (gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
19908 (gst_sub_parse_chain), (gst_sub_parse_sink_event):
19909 * gst/subparse/gstsubparse.h:
19910 No need to allocate GstSegment structure dynamically, just put it
19911 into the instance structure; ignore newsegment events in BYTE
19912 format and in particular don't let it overwrite our saved TIME
19913 segment from the last seek.
19915 2007-03-09 13:05:04 +0000 Michael Smith <msmith@xiph.org>
19917 gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
19918 Original commit message from CVS:
19919 * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
19920 Replace AC3 typefinder with one that isn't terrible, and actually
19923 2007-03-09 12:22:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19925 gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
19926 Original commit message from CVS:
19927 * gst/audioconvert/gstaudioconvert.c:
19928 (gst_audio_convert_transform):
19929 fix error category and translatable string
19931 2007-03-09 11:23:32 +0000 Tim-Philipp Müller <tim@centricular.net>
19933 pkgconfig/: Fix up utils => pbutils here too.
19934 Original commit message from CVS:
19935 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
19936 * pkgconfig/gstreamer-plugins-base.pc.in:
19937 Fix up utils => pbutils here too.
19939 2007-03-09 10:49:53 +0000 Tim-Philipp Müller <tim@centricular.net>
19941 gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
19942 Original commit message from CVS:
19943 * gst/subparse/gstsubparse.c: (handle_buffer):
19944 Break out of loop in chain function as soon as possible if we get
19945 a non-OK flow return.
19947 2007-03-08 18:26:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19949 tests/check/elements/alsa.c: Unref the mixer if the state change fails too (if the alsa devices are inaccessible, for...
19950 Original commit message from CVS:
19951 * tests/check/elements/alsa.c: (GST_START_TEST):
19952 Unref the mixer if the state change fails too (if the
19953 alsa devices are inaccessible, for example)
19955 2007-03-08 17:49:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19957 tests/check/Makefile.am: Don't test libvisual elements in the states check, because libvisual seems to leak internally.
19958 Original commit message from CVS:
19959 * tests/check/Makefile.am:
19960 Don't test libvisual elements in the states check, because libvisual
19961 seems to leak internally.
19962 Re-enable the alsa and states tests now that there's new suppressions
19964 * tests/check/elements/alsa.c: (GST_START_TEST):
19965 Don't leak the alsamixer we instantiated.
19967 2007-03-08 15:22:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19969 sys/: Move some cleanup stuff from the state change handler into a _reset() function that can be called from _finaliz...
19970 Original commit message from CVS:
19971 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
19972 (gst_ximagesink_change_state), (gst_ximagesink_reset),
19973 (gst_ximagesink_finalize):
19974 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
19975 (gst_xvimagesink_reset), (gst_xvimagesink_finalize):
19976 Move some cleanup stuff from the state change handler into a _reset()
19977 function that can be called from _finalize(). This ensures that things
19978 get freed even if (for some reason) the NULL->READY state transition
19979 fails in the parent class.
19980 Even if a parent state change fails, process our downward state change
19981 logic instead of bailing out early.
19982 Free the correct xcontext pointer in ximagesink's xcontext_clear.
19984 2007-03-08 12:53:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19986 ext/alsa/gstalsasink.c: Extra log line.
19987 Original commit message from CVS:
19988 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
19990 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
19991 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
19992 Use pango_font_description_set_family_static instead of
19993 pango_font_description_set_family to save a string copy (it was
19994 leaking due to the strdup anyway)
19995 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
19996 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
19997 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
19998 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
19999 Chain up in finalize.
20001 2007-03-07 18:50:10 +0000 Tim-Philipp Müller <tim@centricular.net>
20003 gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
20004 Original commit message from CVS:
20005 * gst-libs/gst/interfaces/mixertrack.c:
20006 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
20007 (gst_mixer_track_set_property):
20008 API: add "untranslated-label" property which should be set by
20009 implementations at construct time (#414645).
20010 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
20011 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
20012 Set "untranslated-label" when constructing mixer track objects.
20013 * tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
20014 Unit test to check the above.
20016 2007-03-07 17:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
20018 ext/ogg/gstoggdemux.c: Fix confusing debug message.
20019 Original commit message from CVS:
20020 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
20021 Fix confusing debug message.
20023 2007-03-07 17:12:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20025 gst-plugins-base.doap: update doap file with new version
20026 Original commit message from CVS:
20027 * gst-plugins-base.doap:
20028 update doap file with new version
20030 2007-03-07 17:05:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20032 * gst/tcp/gstmultifdsink.c:
20034 Original commit message from CVS:
20037 2007-03-07 16:56:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20039 configure.ac: Back to CVS
20040 Original commit message from CVS:
20044 === release 0.10.12 ===
20046 2007-03-07 16:46:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20052 * docs/plugins/gst-plugins-base-plugins.args:
20053 * docs/plugins/inspect/plugin-adder.xml:
20054 * docs/plugins/inspect/plugin-alsa.xml:
20055 * docs/plugins/inspect/plugin-audioconvert.xml:
20056 * docs/plugins/inspect/plugin-audiorate.xml:
20057 * docs/plugins/inspect/plugin-audioresample.xml:
20058 * docs/plugins/inspect/plugin-audiotestsrc.xml:
20059 * docs/plugins/inspect/plugin-cdparanoia.xml:
20060 * docs/plugins/inspect/plugin-decodebin.xml:
20061 * docs/plugins/inspect/plugin-decodebin2.xml:
20062 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
20063 * docs/plugins/inspect/plugin-gdp.xml:
20064 * docs/plugins/inspect/plugin-gnomevfs.xml:
20065 * docs/plugins/inspect/plugin-libvisual.xml:
20066 * docs/plugins/inspect/plugin-ogg.xml:
20067 * docs/plugins/inspect/plugin-pango.xml:
20068 * docs/plugins/inspect/plugin-playbin.xml:
20069 * docs/plugins/inspect/plugin-subparse.xml:
20070 * docs/plugins/inspect/plugin-tcp.xml:
20071 * docs/plugins/inspect/plugin-theora.xml:
20072 * docs/plugins/inspect/plugin-typefindfunctions.xml:
20073 * docs/plugins/inspect/plugin-video4linux.xml:
20074 * docs/plugins/inspect/plugin-videorate.xml:
20075 * docs/plugins/inspect/plugin-videoscale.xml:
20076 * docs/plugins/inspect/plugin-videotestsrc.xml:
20077 * docs/plugins/inspect/plugin-volume.xml:
20078 * docs/plugins/inspect/plugin-vorbis.xml:
20079 * docs/plugins/inspect/plugin-ximagesink.xml:
20080 * docs/plugins/inspect/plugin-xvimagesink.xml:
20081 * win32/common/config.h:
20083 Original commit message from CVS:
20086 2007-03-07 15:35:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20105 Original commit message from CVS:
20108 2007-03-06 12:31:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20110 configure.ac: Bump version to 0.10.11.4 pre-release
20111 Original commit message from CVS:
20113 Bump version to 0.10.11.4 pre-release
20115 2007-03-06 12:10:08 +0000 Wim Taymans <wim.taymans@gmail.com>
20117 gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
20118 Original commit message from CVS:
20119 * gst-libs/gst/audio/gstbaseaudiosink.c:
20120 (gst_base_audio_sink_async_play):
20121 Fix regression that made GStreamer skip the first samples of audio.
20124 2007-03-05 11:21:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20126 configure.ac: Bump version to 0.10.11.3 pre-release
20127 Original commit message from CVS:
20129 Bump version to 0.10.11.3 pre-release
20131 2007-03-05 09:35:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20133 po/POTFILES.in: Update paths for the rename from utils to pbutils to fix the build.
20134 Original commit message from CVS:
20136 Update paths for the rename from utils to pbutils to fix the build.
20138 2007-03-05 09:27:55 +0000 Tim-Philipp Müller <tim@centricular.net>
20140 gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
20141 Original commit message from CVS:
20142 * gst-libs/gst/pbutils/Makefile.am:
20143 Change directory to install headers in from gst/utils to gst/pbutils
20146 2007-03-04 23:41:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20148 * tests/check/libs/.gitignore:
20150 Original commit message from CVS:
20153 2007-03-04 23:41:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20155 * win32/common/config.h:
20156 * win32/common/libgstutils.def:
20158 Original commit message from CVS:
20161 2007-03-04 23:39:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20163 rename utils to pbutils
20164 Original commit message from CVS:
20166 * docs/libs/gst-plugins-base-libs-docs.sgml:
20167 * docs/libs/gst-plugins-base-libs-sections.txt:
20168 * gst-libs/gst/Makefile.am:
20169 * gst-libs/gst/interfaces/mixer.c:
20170 * gst-libs/gst/pbutils/Makefile.am:
20171 * gst-libs/gst/pbutils/descriptions.c:
20172 (gst_pb_utils_get_source_description),
20173 (gst_pb_utils_get_sink_description),
20174 (gst_pb_utils_get_decoder_description),
20175 (gst_pb_utils_get_encoder_description),
20176 (gst_pb_utils_get_element_description),
20177 (gst_pb_utils_add_codec_description_to_tag_list),
20178 (gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
20179 * gst-libs/gst/pbutils/descriptions.h:
20180 * gst-libs/gst/pbutils/install-plugins.c:
20181 * gst-libs/gst/pbutils/install-plugins.h:
20182 * gst-libs/gst/pbutils/missing-plugins.c:
20183 (gst_missing_uri_source_message_new),
20184 (gst_missing_uri_sink_message_new),
20185 (gst_missing_element_message_new),
20186 (gst_missing_decoder_message_new),
20187 (gst_missing_encoder_message_new),
20188 (gst_missing_plugin_message_get_description):
20189 * gst-libs/gst/pbutils/missing-plugins.h:
20190 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
20191 * gst-libs/gst/pbutils/pbutils.h:
20192 * gst-libs/gst/utils/Makefile.am:
20193 * gst-libs/gst/utils/base-utils.c:
20194 * gst-libs/gst/utils/base-utils.h:
20195 * gst-libs/gst/utils/descriptions.c:
20196 * gst-libs/gst/utils/descriptions.h:
20197 * gst-libs/gst/utils/install-plugins.c:
20198 * gst-libs/gst/utils/install-plugins.h:
20199 * gst-libs/gst/utils/missing-plugins.c:
20200 * gst-libs/gst/utils/missing-plugins.h:
20201 * gst-plugins-base.spec.in:
20202 * gst/playback/Makefile.am:
20203 * gst/playback/gstdecodebin.c:
20204 * gst/playback/gstdecodebin2.c:
20205 * gst/playback/gstplaybasebin.c: (setup_subtitle),
20206 (gen_source_element):
20207 * gst/playback/gstplaybin.c: (plugin_init):
20208 * tests/check/Makefile.am:
20209 * tests/check/libs/pbutils.c: (GST_START_TEST),
20210 (test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
20211 * tests/check/libs/utils.c:
20212 rename utils to pbutils
20214 2007-03-03 10:23:03 +0000 David Schleef <ds@schleef.org>
20216 gst-libs/gst/app/Makefile.am: Install the headers.
20217 Original commit message from CVS:
20218 * gst-libs/gst/app/Makefile.am:
20219 Install the headers.
20221 2007-03-03 10:10:30 +0000 David Schleef <ds@schleef.org>
20223 gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.
20224 Original commit message from CVS:
20225 * gst-libs/gst/app/Makefile.am:
20226 * gst-libs/gst/app/gstappbuffer.c:
20227 * gst-libs/gst/app/gstappbuffer.h:
20228 * gst-libs/gst/app/gstappsrc.c:
20229 Add GstAppBuffer that includes a callback and closure for
20230 proper handling of data chunks.
20232 2007-03-03 09:06:06 +0000 David Schleef <ds@schleef.org>
20234 gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.
20235 Original commit message from CVS:
20236 * gst-libs/gst/app/gstappsrc.c:
20237 * gst-libs/gst/app/gstappsrc.h:
20238 Hacking to address issues in 413418.
20240 2007-03-03 08:16:57 +0000 David Schleef <ds@schleef.org>
20242 Move the app library to gst-libs/gst/app (duh!)
20243 Original commit message from CVS:
20247 * gst-libs/gst/Makefile.am:
20248 * gst-libs/gst/app/Makefile.am:
20249 * gst-libs/gst/app/gstapp.c:
20250 * gst-libs/gst/app/gstappsrc.c:
20251 * gst-libs/gst/app/gstappsrc.h:
20252 * gst/app/Makefile.am:
20253 * gst/app/gstapp.c:
20254 * gst/app/gstappsrc.c:
20255 * gst/app/gstappsrc.h:
20256 Move the app library to gst-libs/gst/app (duh!)
20258 2007-03-02 12:59:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20260 Add documentation for decodebin2 that indicates that the API is still unstable.
20261 Original commit message from CVS:
20262 * docs/plugins/Makefile.am:
20263 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
20264 * docs/plugins/gst-plugins-base-plugins-sections.txt:
20265 * docs/plugins/inspect/plugin-decodebin2.xml:
20266 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
20267 Add documentation for decodebin2 that indicates that the API
20270 2007-03-01 18:50:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20272 configure.ac: Update to 0.10.11.2 (0.10.12 pre-release)
20273 Original commit message from CVS:
20275 Update to 0.10.11.2 (0.10.12 pre-release)
20277 2007-03-01 17:29:55 +0000 Wim Taymans <wim.taymans@gmail.com>
20279 gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
20280 Original commit message from CVS:
20281 * gst-libs/gst/audio/gstbaseaudiosink.c:
20282 (gst_base_audio_sink_async_play):
20283 base time is irrelevant here.
20285 2007-03-01 17:01:43 +0000 Wim Taymans <wim.taymans@gmail.com>
20287 gst-libs/gst/audio/: Improve debugging.
20288 Original commit message from CVS:
20289 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
20290 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
20292 * gst-libs/gst/audio/gstbaseaudiosink.c:
20293 (gst_base_audio_sink_query), (gst_base_audio_sink_event),
20294 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
20295 Improve latency and clock slaving calculations.
20296 Improve slave clock calibration.
20297 * gst-libs/gst/audio/gstringbuffer.c:
20298 (gst_ring_buffer_commit_full):
20299 When we are asked to render N sample to 0 bytes, return N.
20301 2007-03-01 16:48:45 +0000 Wim Taymans <wim.taymans@gmail.com>
20303 ext/alsa/gstalsasink.*: Remove unused dispose function.
20304 Original commit message from CVS:
20305 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
20306 (gst_alsasink_write), (gst_alsasink_reset):
20307 * ext/alsa/gstalsasink.h:
20308 Remove unused dispose function.
20309 Rename lock to not interfere with alsasrc lock.
20310 * ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
20311 (gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
20312 (gst_alsasrc_read), (gst_alsasrc_reset):
20313 * ext/alsa/gstalsasrc.h:
20314 Implement finalize function.
20315 Use lock to protect alsa access.
20317 Fine tune sw params.
20319 2007-03-01 10:20:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20324 Original commit message from CVS:
20327 2007-02-28 19:27:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20329 configure.ac: Convert to new AG_GST style.
20330 Original commit message from CVS:
20332 Convert to new AG_GST style.
20334 2007-02-28 15:17:20 +0000 Ed Catmur <ed@catmur.co.uk>
20336 gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
20337 Original commit message from CVS:
20338 Patch by: Ed Catmur <ed at catmur dot co dot uk>
20339 * gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
20340 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
20341 Fix race condition when rapidly switching visualisations in playbin.
20344 2007-02-28 15:11:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20346 tests/check/Makefile.am: Include local stuff before system installed things in LDFLAGS and
20347 Original commit message from CVS:
20348 * tests/check/Makefile.am:
20349 Include local stuff before system installed things in LDFLAGS and
20352 2007-02-28 15:10:06 +0000 Wim Taymans <wim.taymans@gmail.com>
20354 ext/ogg/gstoggdemux.c: Improve debugging.
20355 Original commit message from CVS:
20356 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
20359 2007-02-28 15:05:03 +0000 Wim Taymans <wim.taymans@gmail.com>
20361 sys/v4l/: Fix duration and timestamping, taking latency into account.
20362 Original commit message from CVS:
20363 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
20364 (gst_v4lsrc_fixate), (gst_v4lsrc_query):
20365 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
20366 Fix duration and timestamping, taking latency into account.
20367 Implement latency query.
20369 2007-02-28 15:02:25 +0000 Wim Taymans <wim.taymans@gmail.com>
20371 gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
20372 Original commit message from CVS:
20373 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
20374 (gst_audio_clock_new):
20376 * gst-libs/gst/audio/gstbaseaudiosink.c:
20377 (gst_base_audio_sink_init), (gst_base_audio_sink_query):
20378 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
20379 (gst_base_audio_src_query), (gst_base_audio_src_get_offset),
20380 (gst_base_audio_src_create):
20381 Improve latency query code.
20382 Use proper clock names.
20384 2007-02-28 12:57:46 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20386 * tests/check/generic/states.c:
20388 Original commit message from CVS:
20391 2007-02-28 12:44:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20393 tests/check/generic/states.c: Copy the states.c test from core again
20394 Original commit message from CVS:
20395 * tests/check/generic/states.c: (GST_START_TEST):
20396 Copy the states.c test from core again
20397 * tests/check/Makefile.am:
20398 ignore cdio and cdparanoiasrc
20400 2007-02-28 12:08:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20402 gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
20403 Original commit message from CVS:
20404 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
20405 (double_hq), (audio_convert_get_func_index), (check_default),
20406 (audio_convert_prepare_context), (audio_convert_convert):
20407 Also make valgrind happy and avoid copying data in some cases.
20409 2007-02-28 11:58:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20411 * tests/check/generic/states.c:
20413 Original commit message from CVS:
20416 2007-02-28 11:47:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20418 Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
20419 Original commit message from CVS:
20420 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
20421 (double_hq), (audio_convert_get_func_index),
20422 (audio_convert_prepare_context), (audio_convert_convert):
20423 * gst/audioconvert/gstaudioconvert.c:
20424 (gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
20425 (gst_audio_convert_transform_caps):
20426 * tests/check/elements/audioconvert.c: (GST_START_TEST),
20427 (audioconvert_suite):
20428 Don't run inplace if that overwrites source data as we go. Add more
20429 tests. Fixes #339837 even more.
20431 2007-02-27 18:45:37 +0000 Julien Moutte <julien@moutte.net>
20433 tests/examples/seek/seek.c: Fix various seeking bugs (Slider was not updating when doing a non flushing seek, Reverse...
20434 Original commit message from CVS:
20435 2007-02-27 Julien MOUTTE <julien@moutte.net>
20436 * tests/examples/seek/seek.c: (do_seek), (set_update_scale),
20437 (msg_segment_done): Fix various seeking bugs (Slider was not
20438 updating when doing a non flushing seek, Reverse playback
20439 on segment seek was wrong).
20441 2007-02-26 21:01:03 +0000 David Schleef <ds@schleef.org>
20443 Add a new plugin/library to make it easy for apps to shove data into a pipeline.
20444 Original commit message from CVS:
20446 * gst/app/Makefile.am:
20447 * gst/app/gstapp.c:
20448 * gst/app/gstappsrc.c:
20449 * gst/app/gstappsrc.h:
20450 Add a new plugin/library to make it easy for apps to shove
20451 data into a pipeline.
20453 2007-02-26 11:48:49 +0000 Wim Taymans <wim.taymans@gmail.com>
20455 tests/examples/seek/seek.c: When we stop scrubbing, don't leave the pipeline PLAYING when we requested a PAUSED state.
20456 Original commit message from CVS:
20457 * tests/examples/seek/seek.c: (stop_seek):
20458 When we stop scrubbing, don't leave the pipeline PLAYING when we
20459 requested a PAUSED state.
20461 2007-02-25 23:51:03 +0000 René Stadler <mail@renestadler.de>
20463 gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
20464 Original commit message from CVS:
20465 Patch by: René Stadler <mail at renestadler de>
20466 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
20467 Parse date strings in vorbis comments that have an invalid (zero)
20468 month or day (#410396).
20469 * tests/check/libs/tag.c: (GST_START_TEST):
20470 Test case for the above.
20472 2007-02-24 20:12:49 +0000 Loïc Minier <lool+gnome@via.ecp.fr>
20474 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
20475 Original commit message from CVS:
20476 Patch by: Loïc Minier <lool+gnome at via ecp fr>
20478 * ext/alsa/Makefile.am:
20479 * gst/audiotestsrc/Makefile.am:
20480 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
20482 2007-02-23 18:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
20484 gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
20485 Original commit message from CVS:
20486 * gst/playback/gstplaybin.c:
20487 Improve docs: point out that the application needs to assist playbin
20490 2007-02-23 13:10:50 +0000 Tim-Philipp Müller <tim@centricular.net>
20492 Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
20493 Original commit message from CVS:
20494 * gst-libs/gst/utils/install-plugins.c:
20495 * gst-libs/gst/utils/missing-plugins.c:
20496 * tests/check/libs/utils.c: (missing_msg_check_getters):
20497 Change GStreamer marker prefix in detail string from 'gstreamer.net'
20498 to just 'gstreamer'. Document the caps string component of the
20499 decoder/encoder detail a bit better, since not everyone will be
20500 familiar with the GStreamer media type/caps system (but they better
20501 enjoy nested itemized lists).
20503 2007-02-22 12:57:47 +0000 Tim-Philipp Müller <tim@centricular.net>
20505 gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
20506 Original commit message from CVS:
20507 * gst-libs/gst/netbuffer/gstnetbuffer.c:
20508 (notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
20509 Fix copying of GstNetBuffer (would crash before, or at least lead to
20510 invalid memory access, #410772), for now by copying the GstBuffer copy
20511 code from the core over here so we can copy the GstBuffer fields on a
20512 provided buffer instance (of type GstNetBuffer in this case). Would be
20513 better to fix this with some support by the core though (and in the long
20514 run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
20515 * tests/check/Makefile.am:
20516 Enable unit test for GstNetBuffer.
20518 2007-02-22 11:04:10 +0000 Andy Wingo <wingo@pobox.com>
20521 * gst-libs/gst/audio/gstbaseaudiosink.c:
20522 gst-libs/gst/audio/gstbaseaudiosink.c
20523 Original commit message from CVS:
20524 2007-02-22 Andy Wingo <wingo@pobox.com>
20525 * gst-libs/gst/audio/gstbaseaudiosink.c
20526 (gst_base_audio_sink_init): Disable pull-mode activation until we
20527 figure out how to make audio sinks go to PLAYING.
20529 2007-02-22 09:04:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20531 Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
20532 Original commit message from CVS:
20533 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
20534 (double_hq), (audio_convert_get_func_index),
20535 (audio_convert_prepare_context), (audio_convert_convert):
20536 * gst/audioconvert/audioconvert.h:
20537 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
20538 (gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
20539 * gst/audioconvert/gstchannelmix.h:
20540 * tests/check/elements/audioconvert.c: (GST_START_TEST):
20541 Add float as an intermediate format, as well as float mixing. Enable
20542 test that was failing before. Fixes #339837
20544 2007-02-21 16:12:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20546 tests/examples/seek/seek.c: Undo the previous commit: -1 as a stop time implies that the stop time is the end of file...
20547 Original commit message from CVS:
20548 * tests/examples/seek/seek.c: (do_seek):
20549 Undo the previous commit: -1 as a stop time implies that the stop
20550 time is the end of file, clearing any previously configured segment.
20552 2007-02-21 15:36:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20554 tests/examples/seek/seek.c: Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
20555 Original commit message from CVS:
20556 * tests/examples/seek/seek.c: (do_seek):
20557 Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
20559 2007-02-21 13:55:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20561 gst/volume/gstvolume.c: Unbreak volume, value remains gint.
20562 Original commit message from CVS:
20563 * gst/volume/gstvolume.c: (volume_process_int16),
20564 (volume_process_int16_clamp), (volume_set_caps):
20565 Unbreak volume, value remains gint.
20567 2007-02-21 13:08:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20569 gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
20570 Original commit message from CVS:
20571 * gst/volume/gstvolume.c: (volume_choose_func),
20572 (volume_update_real_volume), (gst_volume_set_volume),
20573 (gst_volume_init), (volume_process_double), (volume_process_float),
20574 (volume_process_int16), (volume_process_int16_clamp),
20575 (volume_set_caps), (volume_transform_ip), (volume_update_volume):
20576 * gst/volume/gstvolume.h:
20577 Extend float audio support (double) and some int->uint cleanups.
20579 2007-02-20 15:44:32 +0000 Edward Hervey <bilboed@bilboed.com>
20581 gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
20582 Original commit message from CVS:
20583 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
20584 (multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
20585 (sort_end_pads), (gst_decode_group_expose),
20586 (gst_decode_group_hide):
20587 Don't free groups from the streaming threads. Just put them aside and
20588 free them in dispose.
20590 2007-02-20 11:20:52 +0000 Edward Hervey <bilboed@bilboed.com>
20592 gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
20593 Original commit message from CVS:
20594 * gst/playback/gstdecodebin2.c: (connect_element),
20595 (pad_added_group_cb), (gst_decode_group_check_if_blocked),
20596 (sort_end_pads), (gst_decode_group_expose):
20597 Handle dynamic pads within groups.
20598 Sort pads before exposing them in order to make playbin happy.
20599 There still is a race with the multiqueue filling up. This should be
20603 2007-02-18 21:02:36 +0000 Tim-Philipp Müller <tim@centricular.net>
20605 gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
20606 Original commit message from CVS:
20607 * gst-libs/gst/utils/base-utils.c:
20608 * gst-libs/gst/utils/descriptions.c:
20609 * gst-libs/gst/utils/install-plugins.c:
20610 * gst-libs/gst/utils/missing-plugins.c:
20611 Some more docs (and descriptions for two subtitle formats).
20613 2007-02-16 10:19:45 +0000 Tim-Philipp Müller <tim@centricular.net>
20615 gst-libs/gst/audio/audio.c: Fix documentation.
20616 Original commit message from CVS:
20617 * gst-libs/gst/audio/audio.c:
20620 2007-02-16 10:15:46 +0000 Yves Lefebvre <ivanohe@abacom.com>
20622 gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
20623 Original commit message from CVS:
20624 Patch by: Yves Lefebvre <ivanohe abacom com>
20625 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
20626 Don't leak caps. Fixes #408278.
20628 2007-02-15 15:17:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20630 More docs coverage and some ChangeLog surgery (add missing names)
20631 Original commit message from CVS:
20632 * ext/cdparanoia/gstcdparanoiasrc.h:
20633 * ext/ogg/gstoggdemux.h:
20634 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
20635 (gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
20636 (gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
20637 * gst-libs/gst/audio/audio.h:
20638 * gst-libs/gst/audio/gstaudiofilter.h:
20639 * gst-libs/gst/interfaces/videoorientation.h:
20640 * gst/adder/gstadder.h:
20641 More docs coverage and some ChangeLog surgery (add missing names)
20643 2007-02-15 12:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
20645 sys/: Small constifications.
20646 Original commit message from CVS:
20647 * sys/ximage/ximagesink.c:
20648 (gst_ximagesink_calculate_pixel_aspect_ratio):
20649 * sys/xvimage/xvimagesink.c:
20650 (gst_xvimagesink_calculate_pixel_aspect_ratio):
20651 Small constifications.
20653 2007-02-15 12:06:25 +0000 Wim Taymans <wim.taymans@gmail.com>
20655 gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
20656 Original commit message from CVS:
20657 * gst-libs/gst/audio/gstbaseaudiosink.c:
20658 (gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
20659 (gst_base_audio_sink_render), (gst_base_audio_sink_callback),
20660 (gst_base_audio_sink_async_play),
20661 (gst_base_audio_sink_change_state):
20662 Answer latency query.
20663 Use configured latency when syncing.
20665 * gst-libs/gst/audio/gstbaseaudiosrc.c:
20666 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
20667 (gst_base_audio_src_query), (gst_base_audio_src_change_state):
20668 Fix possible memleak.
20669 Implement latency query.
20672 2007-02-15 11:59:41 +0000 Wim Taymans <wim.taymans@gmail.com>
20674 ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
20675 Original commit message from CVS:
20676 * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
20677 Ignore errors in reset, these are not fatal. They also grab the element
20678 lock which is already taking when this function is called. Fixes
20681 2007-02-13 13:50:56 +0000 Christian Schaller <uraeus@gnome.org>
20683 * gst-plugins-base.spec.in:
20684 add header file for easy codec install
20685 Original commit message from CVS:
20686 add header file for easy codec install
20688 2007-02-13 10:24:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20690 configure.ac: Remove 'tests/examples/xerror/Makefile' from output files again.
20691 Original commit message from CVS:
20693 Remove 'tests/examples/xerror/Makefile' from output files again.
20695 2007-02-13 09:12:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20697 Also crossref against gst-plugins-base-libs.
20698 Original commit message from CVS:
20700 * docs/plugins/Makefile.am:
20701 Also crossref against gst-plugins-base-libs.
20703 2007-02-12 20:42:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20705 Add crossreferences to glib/gobject/gstream docs.
20706 Original commit message from CVS:
20708 * docs/libs/Makefile.am:
20709 * docs/plugins/Makefile.am:
20710 Add crossreferences to glib/gobject/gstream docs.
20711 * gst-libs/gst/audio/audio.h:
20713 * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
20714 Add own debug category.
20716 2007-02-12 11:01:04 +0000 René Stadler <mail@renestadler.de>
20718 gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
20719 Original commit message from CVS:
20720 Patch by: René Stadler <mail at renestadler de>
20721 * gst-libs/gst/tag/gstvorbistag.c:
20722 Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
20725 2007-02-12 10:33:40 +0000 Tim-Philipp Müller <tim@centricular.net>
20727 gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
20728 Original commit message from CVS:
20729 * gst/playback/gstplaybasebin.c: (setup_source):
20730 When we have external subtitles and wait for the subtitle decodebin
20731 to get up and running, we set up a (sync) bus handler for the
20732 subtitle decodebin, so we can stop waiting when it posts an error
20733 message. However, we should do that before we set the subtitle
20734 decodebin's state to playing, otherwise things are racy and we might
20735 miss error messages posted before we had a chance to set up the bus.
20736 This should finally fix totem hanging on .txt pseudo-subtitle files.
20738 2007-02-10 19:27:48 +0000 Sébastien Moutte <sebastien@moutte.net>
20740 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
20741 Original commit message from CVS:
20742 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
20743 Use gst_gdouble_to_guint64 for conversions.
20744 * win32/common/config.h.in:
20745 Add a define for GST_INSTALL_PLUGINS_HELPER
20746 * win32/common/libgstaudio.def:
20747 * win32/common/libgstcdda.def:
20748 * win32/common/libgstnetbuffer.def:
20749 * win32/common/libgstrtp.def:
20750 * win32/common/libgutils.def:
20751 Add new exported functions.
20752 * win32/vs6/gst_plugins_base.dsw:
20753 * win32/vs6/libgstdecodebin.dsp:
20754 * win32/vs6/libgstnetbuffer.dsp:
20755 * win32/vs6/libgstplaybin.dsp:
20756 * win32/vs6/libgstrtp.dsp:
20757 * win32/vs6/libgstvorbis.dsp:
20758 * win32/vs6/libgstcdda.dsp:
20759 * win32/vs6/libgstgdp.dsp:
20760 * win32/vs6/libgstutils.dsp:
20761 Update and add new project files.
20763 2007-02-10 18:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
20765 gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
20766 Original commit message from CVS:
20767 * gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
20768 (subrip_remove_unhandled_tags), (parse_subrip):
20769 For SubRip (.srt) subtitles, ignore all markup tags we don't
20770 handle (like font tags, for example).
20771 * tests/check/elements/subparse.c:
20774 2007-02-09 13:28:01 +0000 Tim-Philipp Müller <tim@centricular.net>
20778 Original commit message from CVS:
20781 2007-02-09 13:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
20783 gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
20784 Original commit message from CVS:
20785 * gst/playback/gstdecodebin.c: (add_fakesink),
20786 (gst_decode_bin_change_state):
20787 * gst/playback/gstdecodebin2.c: (add_fakesink),
20788 (gst_decode_bin_change_state):
20789 Don't error out if there is no fakesink in the READY to NULL state
20790 change, since when decodebin is re-used, we're only adding the
20791 fakesink element in READY to PAUSED.
20792 * tests/check/elements/decodebin.c:
20793 (new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
20795 Minimal unit test to make sure we can use the same decodebin
20796 instance twice (at least with audiotestsrc input).
20798 2007-02-09 09:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
20800 ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
20801 Original commit message from CVS:
20802 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
20803 Try to get devic-name from device string first, and from handle only
20804 as fallback (seems to yield better results and is more robust
20805 against buggy probing code on the application side).
20807 2007-02-08 15:43:26 +0000 Julien Puydt <julien.puydt@laposte.net>
20809 ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
20810 Original commit message from CVS:
20811 Based on patch by: Julien Puydt <julien.puydt at laposte net>
20812 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
20813 (gst_alsa_find_device_name):
20814 * ext/alsa/gstalsa.h:
20815 * ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
20816 * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
20817 Improve device-name detection a bit, especially in the case where
20818 the device is not actually open (#405020, #405024). Move common code
20819 into gstalsa.c instead of duplicating it.
20821 2007-02-07 13:05:01 +0000 Tim-Philipp Müller <tim@centricular.net>
20823 gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
20824 Original commit message from CVS:
20825 * gst/audioconvert/gstaudioconvert.c:
20826 Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
20828 2007-02-06 17:47:32 +0000 Julien Moutte <julien@moutte.net>
20830 sys/xvimage/xvimagesink.*: Implement PropertyProbe Interface for XVAdaptors so that one can choose the adaptor to use...
20831 Original commit message from CVS:
20832 2007-02-06 Julien MOUTTE <julien@moutte.net>
20833 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
20834 (gst_xvimagesink_get_xv_support),
20835 (gst_xvimagesink_xcontext_clear),
20836 (gst_xvimagesink_interface_supported),
20837 (gst_xvimagesink_probe_get_properties),
20838 (gst_xvimagesink_probe_probe_property),
20839 (gst_xvimagesink_probe_needs_probe),
20840 (gst_xvimagesink_probe_get_values),
20841 (gst_xvimagesink_property_probe_interface_init),
20842 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
20843 (gst_xvimagesink_init), (gst_xvimagesink_class_init),
20844 (gst_xvimagesink_get_type):
20845 * sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
20846 for XVAdaptors so that one can choose the adaptor to use with
20847 gstreamer-properties.
20849 2007-02-06 14:00:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20851 gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
20852 Original commit message from CVS:
20853 * gst/audioconvert/gstaudioconvert.c:
20854 Also mention that a conversion from double to float is suboptimal still.
20856 2007-02-06 09:42:05 +0000 Tim-Philipp Müller <tim@centricular.net>
20858 gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
20859 Original commit message from CVS:
20860 * gst-libs/gst/audio/gstaudiofilter.c:
20861 (gst_audio_filter_class_init), (gst_audio_filter_change_state):
20862 Clear our formats structure and free the caps contained in it when
20865 2007-02-05 18:39:51 +0000 Andy Wingo <wingo@pobox.com>
20868 * gst-libs/gst/audio/gstbaseaudiosink.c:
20869 gst-libs/gst/audio/gstbaseaudiosink.c
20870 Original commit message from CVS:
20871 2007-02-05 Andy Wingo <wingo@pobox.com>
20872 * gst-libs/gst/audio/gstbaseaudiosink.c
20873 (gst_base_audio_sink_callback): Update basesink->offset so that we
20874 pull monotonically increasing offsets instead of, um, seeking back
20875 to 0 each time. Fixes alsasrc ! alsasink!
20877 2007-02-05 11:44:52 +0000 Tim-Philipp Müller <tim@centricular.net>
20879 gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
20880 Original commit message from CVS:
20881 * gst/videoscale/gstvideoscale.c:
20882 A width and height of 1 makes us crash, so increase minimum size to
20883 2x2 pixels until someone feels like fixing this (#404512).
20885 2007-02-04 16:23:37 +0000 Tim-Philipp Müller <tim@centricular.net>
20887 tests/check/pipelines/oggmux.c: Add small test to make sure request pads are cleaned up properly even if oggmux never...
20888 Original commit message from CVS:
20889 * tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
20890 Add small test to make sure request pads are cleaned up properly
20891 even if oggmux never changes state out of NULL.
20893 2007-02-04 14:11:51 +0000 Tim-Philipp Müller <tim@centricular.net>
20895 tests/check/libs/utils.c: Fix unit test. Turns out things work much better when you
20896 Original commit message from CVS:
20897 * tests/check/libs/utils.c: (GST_START_TEST):
20898 Fix unit test. Turns out things work much better when you
20899 NULL-terminate string arrays. Should make p5 build bot happy again.
20901 2007-02-03 23:28:45 +0000 Tim-Philipp Müller <tim@centricular.net>
20903 gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
20904 Original commit message from CVS:
20905 * gst-libs/gst/audio/Makefile.am:
20906 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
20907 (gst_audio_filter_template_base_init),
20908 (gst_audio_filter_template_class_init),
20909 (gst_audio_filter_template_init),
20910 (gst_audio_filter_template_set_property),
20911 (gst_audio_filter_template_get_property),
20912 (gst_audio_filter_template_setup),
20913 (gst_audio_filter_template_filter),
20914 (gst_audio_filter_template_filter_inplace), (plugin_init):
20915 Oops, forgot to commit fixed-up example.
20917 2007-02-03 20:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
20919 Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
20920 Original commit message from CVS:
20921 * docs/libs/gst-plugins-base-libs-sections.txt:
20922 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
20923 (gst_audio_filter_class_init), (gst_audio_filter_init),
20924 (gst_audio_filter_set_caps),
20925 (gst_audio_filter_class_add_pad_templates):
20926 * gst-libs/gst/audio/gstaudiofilter.h:
20927 Port GstAudioFilter to 0.10. This change technically breaks
20928 API and ABI (and thus also every library developer's heart),
20929 but seems justifiable on the grounds that the base class was
20930 completely unusable before (ie. would crash immediately when
20931 actually used). Fixes #403963 (and eventually also #403572).
20932 Also document all of this a bit.
20934 2007-02-03 14:26:54 +0000 Tim-Philipp Müller <tim@centricular.net>
20936 Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
20937 Original commit message from CVS:
20938 * gst-libs/gst/utils/install-plugins.c:
20939 (gst_install_plugins_spawn_child):
20940 * tests/check/libs/utils.c:
20941 (test_base_utils_install_plugins_do_callout):
20942 Lowering log level to see why things fail on the p5 build bot;
20943 fix some typos in unit test messages.
20945 2007-02-03 13:59:27 +0000 Tim-Philipp Müller <tim@centricular.net>
20947 tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
20948 Original commit message from CVS:
20949 * tests/check/libs/utils.c:
20950 (test_base_utils_install_plugins_do_callout):
20951 Don't hard-code temp directory for test helper; use GLib functions
20952 to write out file and do error checking etc.
20954 2007-02-02 20:42:08 +0000 Tim-Philipp Müller <tim@centricular.net>
20956 gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
20957 Original commit message from CVS:
20958 * gst-libs/gst/utils/Makefile.am:
20959 * gst-libs/gst/utils/base-utils.h:
20960 * gst-libs/gst/utils/install-plugins.c:
20961 (gst_install_plugins_context_set_xid),
20962 (gst_install_plugins_context_new),
20963 (gst_install_plugins_context_free),
20964 (gst_install_plugins_get_helper),
20965 (gst_install_plugins_spawn_child),
20966 (gst_install_plugins_return_from_status),
20967 (gst_install_plugins_installer_exited),
20968 (gst_install_plugins_async), (gst_install_plugins_sync),
20969 (gst_install_plugins_return_get_name),
20970 (gst_install_plugins_installation_in_progress):
20971 * gst-libs/gst/utils/install-plugins.h:
20972 API: add API for applications to initiate installation of missing
20973 plugins, ie. gst_install_plugins_async() primarily.
20974 Based on libgimme-codec by Ryan Lortie.
20976 Add --with-install-plugins-helper configure option so distros can specify
20977 the path of the helper script or program to call when plugin installation
20978 is requested (distros: please do any argument munging in this helper
20979 script instead of patching GStreamer to pass arguments differently
20980 to another program directly).
20981 * docs/libs/gst-plugins-base-libs-docs.sgml:
20982 * docs/libs/gst-plugins-base-libs-sections.txt:
20983 Build and document new API.
20984 * tests/check/libs/utils.c: (result_cb),
20985 (test_base_utils_install_plugins_do_callout), (GST_START_TEST),
20986 (libgstbaseutils_suite):
20987 Some simple checks for the new API.
20989 2007-02-02 14:44:29 +0000 Tim-Philipp Müller <tim@centricular.net>
20991 tests/check/elements/audioconvert.c: Add small test for 32bit float <=> 64bit float conversion (works only one way so...
20992 Original commit message from CVS:
20993 * tests/check/elements/audioconvert.c: (test_float_conversion):
20994 Add small test for 32bit float <=> 64bit float conversion (works
20995 only one way so far, 32=>64 produces structured noise).
20997 2007-02-02 11:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
20999 gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
21000 Original commit message from CVS:
21001 * gst/audioconvert/gstaudioconvert.c:
21002 (set_structure_widths_32_and_64), (make_lossless_changes):
21003 We don't support floats with a width of 40, 48 or 56 bits.
21005 2007-02-02 09:48:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21007 gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
21008 Original commit message from CVS:
21009 * gst/audioconvert/audioconvert.c: (float), (double),
21010 (audio_convert_get_func_index):
21011 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
21012 (make_lossless_changes):
21013 Support for 64-bit float audio in audioconvert (#339837)
21015 2007-02-01 18:50:08 +0000 Holger Wansing <linux@wansing-online.de>
21017 po/: Add German translation (#352069).
21018 Original commit message from CVS:
21019 Patch by: Holger Wansing <linux wansing-online de>
21022 Add German translation (#352069).
21024 2007-02-01 17:52:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21026 ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
21027 Original commit message from CVS:
21028 reviewed by: Wim Taymans <wim@fluendo.com>
21029 * ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
21030 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
21031 Use newly added GstCollectPads API to free the allocated resources in
21032 the GstOggPad structures (#402393).
21034 2007-01-31 15:58:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21036 gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
21037 Original commit message from CVS:
21038 * gst/playback/gstplaybin.c: (gen_vis_element):
21039 Add audioresample+audioconvert in front of the visualisation
21040 element, so that elements like libvisual 0.4 that don't support all
21041 samplerates can work.
21044 2007-01-30 19:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
21046 gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
21047 Original commit message from CVS:
21048 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
21049 (gst_play_base_bin_get_streaminfo_value_array):
21050 Take some locks and make a copy of the streaminfo value array we
21051 maintain while holding the lock, so that the application can
21052 retrieve the stream-info as a value array in a thread-safe way.
21054 2007-01-30 11:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
21056 gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
21057 Original commit message from CVS:
21058 * gst/audioconvert/gstaudioconvert.c:
21059 Don't fail on 0 sized buffers. Fixes #396835.
21061 2007-01-29 21:13:07 +0000 David Schleef <ds@schleef.org>
21063 gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
21064 Original commit message from CVS:
21065 * gst/typefind/gsttypefindfunctions.c:
21066 Detect BBCD as video/x-dirac, so we can play raw dirac
21069 2007-01-29 18:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
21071 ext/theora/theoraenc.c: Check return value of theora_encode_header(), or we might try to allocate a random number of ...
21072 Original commit message from CVS:
21073 * ext/theora/theoraenc.c: (theora_enc_chain):
21074 Check return value of theora_encode_header(), or we might try to
21075 allocate a random number of bytes. theora_encode_header() can fail
21076 if libtheora has been compiled with encoding support disabled.
21079 2007-01-29 10:53:06 +0000 Wim Taymans <wim.taymans@gmail.com>
21081 tests/check/gst/.cvsignore: Do as buildbot says.
21082 Original commit message from CVS:
21083 * tests/check/gst/.cvsignore:
21084 Do as buildbot says.
21086 2007-01-29 10:25:11 +0000 Wim Taymans <wim.taymans@gmail.com>
21088 ext/libvisual/visual.c: Fix strides in libvisual. Gst uses X strides.
21089 Original commit message from CVS:
21090 * ext/libvisual/visual.c: (gst_visual_src_setcaps):
21091 Fix strides in libvisual. Gst uses X strides.
21092 Inspired by: <ed at catmur dot co dot uk> and
21093 <tim at centricular dot net>
21096 2007-01-27 13:32:24 +0000 Wim Taymans <wim.taymans@gmail.com>
21098 ext/ogg/gstoggdemux.*: Properly propagate streaming errors when we are scanning the file for chains so that we don't ...
21099 Original commit message from CVS:
21100 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
21101 (gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
21102 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
21103 (gst_ogg_demux_perform_seek),
21104 (gst_ogg_demux_bisect_forward_serialno),
21105 (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
21106 (gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
21107 (gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
21108 (gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
21109 * ext/ogg/gstoggdemux.h:
21110 Properly propagate streaming errors when we are scanning the file for
21111 chains so that we don't crash when shut down. Might fix some crashers
21112 when quickly switching oggs in RB such as #332503 and #378436.
21114 2007-01-26 12:44:46 +0000 Tim-Philipp Müller <tim@centricular.net>
21116 ext/gnomevfs/gstgnomevfssrc.c: Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND error code as well.
21117 Original commit message from CVS:
21118 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
21119 Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
21120 error code as well.
21122 2007-01-25 16:02:41 +0000 Wim Taymans <wim.taymans@gmail.com>
21124 gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
21125 Original commit message from CVS:
21126 * gst/playback/gstplaybasebin.c: (remove_source):
21127 Don't try to disconnect a signal from a finalized object.
21129 2007-01-25 14:29:21 +0000 Tim-Philipp Müller <tim@centricular.net>
21131 gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
21132 Original commit message from CVS:
21133 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
21134 Cast lock macro parameters to make sure we're actually accessing the
21135 lock member at the right class level. Free list itself in _dispose()
21136 as well and NULL it in case dispose gets called multiple times.
21138 2007-01-25 14:02:37 +0000 Edward Hervey <bilboed@bilboed.com>
21140 gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
21141 Original commit message from CVS:
21142 * gst/playback/gstdecodebin2.c:
21143 (gst_decode_bin_dispose),(gst_decode_bin_finalize):
21144 Free GstDecodeGroups no longer used.
21145 (gst_decode_group_expose):
21146 Don't unlock too many times !
21147 (deactivate_free_recursive):
21148 Free iterator once we're done with it.
21149 Fix for recursively deactivating elements (stop at ghostpads).
21151 2007-01-25 12:24:18 +0000 Tim-Philipp Müller <tim@centricular.net>
21153 gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
21154 Original commit message from CVS:
21155 * gst/playback/gstplaybin.c: (handoff):
21156 Fix up caps on the frame buffer before we save it and potentially
21157 make it accessible to other threads via g_object_get; also use
21158 gst_buffer_replace() instead of gst_mini_object_replace().
21160 2007-01-25 12:06:59 +0000 Tim-Philipp Müller <tim@centricular.net>
21162 gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
21163 Original commit message from CVS:
21164 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
21165 Make getting the current frame thread-safe.
21167 2007-01-25 11:48:10 +0000 Edward Hervey <bilboed@bilboed.com>
21169 gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
21170 Original commit message from CVS:
21171 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
21172 (gst_decode_group_new), (gst_decode_group_free):
21173 Set queues to bigger sizes to cope with HD contents.
21174 Fix some mutex freeing and add comment about MT safe methods.
21176 2007-01-24 12:51:20 +0000 Tim-Philipp Müller <tim@centricular.net>
21178 ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
21179 Original commit message from CVS:
21180 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
21181 (gst_text_overlay_text_event):
21182 Don't unnecessarily ref (and then leak) upstream events if the text
21183 pad is not linked. Fixes #399948.
21184 * tests/check/gst-plugins-base.supp:
21185 Add suppression for pango on edgy/x86 for textoverlay test.
21187 2007-01-24 12:10:56 +0000 Wim Taymans <wim.taymans@gmail.com>
21189 gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
21190 Original commit message from CVS:
21191 * gst-libs/gst/rtp/gstrtpbuffer.h:
21192 Add some more fixed payloads.
21194 2007-01-23 18:39:45 +0000 Tim-Philipp Müller <tim@centricular.net>
21196 ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
21197 Original commit message from CVS:
21198 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
21199 Error out properly if we get an error from libogg while reading the
21200 BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).
21202 2007-01-23 17:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
21204 gst/playback/gstdecodebin2.c: Don't leak mutex.
21205 Original commit message from CVS:
21206 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
21208 * tests/check/elements/playbin.c:
21209 (test_sink_usage_video_only_stream),
21210 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
21211 (test_suburi_error_wrongproto), (test_missing_urisource_handler),
21212 (test_missing_suburisource_handler),
21213 (test_missing_primary_decoder), (playbin_suite):
21214 Run all tests once with decodebin and once with decodebin2.
21215 One test does not pass yet with decodebin2.
21217 2007-01-23 14:30:28 +0000 Edward Hervey <bilboed@bilboed.com>
21219 ext/ogg/gstoggmux.c: Fix the cases where oggmux doesn't properly figure out that all sinkpads have gone EOS, and ther...
21220 Original commit message from CVS:
21221 * ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
21222 Fix the cases where oggmux doesn't properly figure out that all
21223 sinkpads have gone EOS, and therefore doesn't push out the remaining
21224 buffers and the final EOS event.
21227 2007-01-23 13:19:19 +0000 Julien Moutte <julien@moutte.net>
21229 sys/: Don't lock on navigation event push, just on keysym to string.
21230 Original commit message from CVS:
21231 2007-01-23 Julien MOUTTE <julien@moutte.net>
21232 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
21233 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
21234 Don't lock on navigation event push, just on keysym to string.
21235 Fixes #397673 again.
21237 2007-01-22 17:37:38 +0000 Edward Hervey <bilboed@bilboed.com>
21239 gst/playback/gstdecodebin2.c: Cleanups.
21240 Original commit message from CVS:
21241 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
21242 (get_current_group), (group_demuxer_event_probe),
21243 (gst_decode_group_expose), (deactivate_free_recursive),
21244 (gst_decode_group_free):
21246 Don't forget to emit 'no-more-pads' once a group is exposed.
21247 Cleanup elements from a DecodeGroup once we remove it.
21248 Protect call to gst_decode_group_expose() with the decodebin lock.
21250 2007-01-22 13:16:42 +0000 Julien Moutte <julien@moutte.net>
21252 sys/: Looking at Xorg code i can't figure out if that XKeysymToString function is thread sensible or not. Lock it jus...
21253 Original commit message from CVS:
21254 2007-01-22 Julien MOUTTE <julien@moutte.net>
21255 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
21256 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
21257 Looking at Xorg code i can't figure out if that XKeysymToString
21258 function is thread sensible or not. Lock it just in case as
21259 recommended by Radek Doulik <rodo at ximian dot com>.
21261 2007-01-22 13:10:13 +0000 Julien Moutte <julien@moutte.net>
21263 sys/: Lock that X Call as well. Fixes #397673.
21264 Original commit message from CVS:
21265 2007-01-22 Julien MOUTTE <julien@moutte.net>
21266 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
21267 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
21268 Lock that X Call as well. Fixes #397673.
21270 2007-01-22 12:03:27 +0000 Tim-Philipp Müller <tim@centricular.net>
21272 gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
21273 Original commit message from CVS:
21274 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
21275 Don't go into an endless loop if the file starts with 00 00 01 2X,
21276 like quicktime redirect files might. Fixes #396042.
21277 * tests/check/Makefile.am:
21278 * tests/check/gst/.cvsignore:
21279 * tests/check/gst/typefindfunctions.c: (GST_START_TEST),
21280 (typefindfunctions_suite):
21281 Add unit test for the above.
21283 2007-01-22 10:27:26 +0000 Tim-Philipp Müller <tim@centricular.net>
21285 gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
21286 Original commit message from CVS:
21287 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
21288 On second thought, use "depth" field rather than "bpp" field.
21290 2007-01-22 09:23:01 +0000 Tim-Philipp Müller <tim@centricular.net>
21292 gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
21293 Original commit message from CVS:
21294 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
21295 Camtasia caps apparently need a bpp field (#398875).
21297 2007-01-19 19:09:05 +0000 Tim-Philipp Müller <tim@centricular.net>
21299 gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
21300 Original commit message from CVS:
21301 * gst/playback/gstplaybasebin.c: (setup_subtitle),
21302 (gen_source_element), (gst_play_base_bin_change_state):
21303 Attempt at a better error message in case we don't have the required
21304 URI handler installed; post missing-plugin message also when we're
21305 missing an URI handler for the subtitle URI; clean up properly also
21306 when an error occurs and we never made it to PAUSED state.
21307 * tests/check/elements/playbin.c: (GST_START_TEST),
21309 Check that we're also getting a missing-plugin messsage for a
21310 missing subtitle URI handler (and clean up properly).
21312 2007-01-19 18:47:30 +0000 Tim-Philipp Müller <tim@centricular.net>
21314 gst/playback/gstplaybasebin.c: Plug a few reference leaks.
21315 Original commit message from CVS:
21316 * gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
21317 Plug a few reference leaks.
21319 2007-01-19 12:23:06 +0000 Tim-Philipp Müller <tim@centricular.net>
21321 gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
21322 Original commit message from CVS:
21323 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
21324 Lower probability a bit if the marker isn't right at the start,
21325 to decrease the chance of false positives.
21327 2007-01-19 11:31:50 +0000 Tim-Philipp Müller <tim@centricular.net>
21329 gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
21330 Original commit message from CVS:
21331 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
21332 Small mpeg2 system stream typefinding improvement: make typefinder
21333 probe a bit into the stream instead of just looking for a marker
21334 at the beginning. Fixes #397810.
21336 2007-01-18 16:23:35 +0000 Tim-Philipp Müller <tim@centricular.net>
21338 gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
21339 Original commit message from CVS:
21340 * gst/audioconvert/gstchannelmix.c:
21341 Remove compatibility cruft for prehistoric GLib versions.
21343 2007-01-17 16:11:14 +0000 Tim-Philipp Müller <tim@centricular.net>
21345 gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
21346 Original commit message from CVS:
21347 * gst/playback/Makefile.am:
21348 * gst/playback/gstdecodebin.c: (close_pad_link):
21349 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
21350 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
21351 (gst_play_base_bin_handle_message_func), (unknown_type):
21352 Let decodebin be the element to post missing-plugin messages for
21353 missing decoders (rather than playbin); make playbin implement
21354 GstBin::handle_message so we can suppress missing-plugin messages
21355 for types we're not handling on purpose (don't want to bring up an
21356 installer in those cases).
21358 2007-01-16 19:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21360 gst/: Fix potentially unaligned access (#397207).
21361 Original commit message from CVS:
21362 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
21363 * gst-libs/gst/tag/gstvorbistag.c:
21364 (gst_tag_list_to_vorbiscomment_buffer):
21365 * gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
21366 Fix potentially unaligned access (#397207).
21368 2007-01-16 12:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21370 tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
21371 Original commit message from CVS:
21372 * tests/examples/seek/seek.c: (set_scale), (update_scale),
21373 (do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
21374 (rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
21376 Allow to toggle looping while it plays. Fix callback prototype. Clean
21377 up code a bit more. Add copyright header.
21379 2007-01-16 11:41:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21381 sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
21382 Original commit message from CVS:
21383 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
21384 Red and blue mask was swapped (spotted by Dan Williams).
21386 2007-01-15 13:58:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21388 gst-libs/gst/tag/: Use new beats-per-minute tag from core.
21389 Original commit message from CVS:
21390 * gst-libs/gst/tag/gstid3tag.c:
21391 * gst-libs/gst/tag/gstvorbistag.c:
21392 Use new beats-per-minute tag from core.
21394 2007-01-15 11:30:53 +0000 Tim-Philipp Müller <tim@centricular.net>
21396 po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
21397 Original commit message from CVS:
21399 Add new files with translatable strings, so they actually make it
21400 into the template file one day.
21402 2007-01-12 21:19:35 +0000 Andy Wingo <wingo@pobox.com>
21405 * gst-libs/gst/audio/gstbaseaudiosink.c:
21406 * gst-libs/gst/audio/gstbaseaudiosrc.c:
21407 gst-libs/gst/audio/gstbaseaudiosink.c
21408 Original commit message from CVS:
21409 2007-01-12 Andy Wingo <wingo@pobox.com>
21410 * gst-libs/gst/audio/gstbaseaudiosink.c
21411 (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
21412 (gst_base_audio_sink_activate_pull): Remove the handwavey nego
21413 stuff, as the base class handles this now. Actually tell the ring
21415 (gst_base_audio_sink_callback): Cast the ring buffer correctly.
21416 How did this work before? Maybe I'm not as awesome a programmer as
21418 * gst-libs/gst/audio/gstbaseaudiosrc.c
21419 (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
21422 2007-01-12 18:08:23 +0000 Tim-Philipp Müller <tim@centricular.net>
21424 gst-libs/gst/utils/missing-plugins.c: Remove more fields so that the application can better blacklist formats that ha...
21425 Original commit message from CVS:
21426 * gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
21427 Remove more fields so that the application can better blacklist
21428 formats that have been tried before.
21430 2007-01-12 17:43:40 +0000 Christian Schaller <uraeus@gnome.org>
21432 * gst-plugins-base.spec.in:
21434 Original commit message from CVS:
21437 2007-01-12 12:47:29 +0000 Tim-Philipp Müller <tim@centricular.net>
21439 gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
21440 Original commit message from CVS:
21441 * gst-libs/gst/audio/mixerutils.h:
21442 Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
21443 used when compiling with c++ compilers as well.
21445 2007-01-12 09:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
21447 gst/typefind/gsttypefindfunctions.c: Fix comment.
21448 Original commit message from CVS:
21449 * gst/typefind/gsttypefindfunctions.c:
21452 2007-01-11 13:12:17 +0000 Tim-Philipp Müller <tim@centricular.net>
21454 gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
21455 Original commit message from CVS:
21456 * gst/playback/gstplaybin.c: (post_missing_element_message),
21457 (gen_video_element), (gen_text_element), (gen_audio_element),
21459 Post missing-plugin messages also when we error out because
21460 converters, textoverlay or auto*sinks are missing (#161922).
21462 2007-01-10 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
21464 gst/playback/: Fix the case where we try to ref a NULL element when we delay a link because of unfixed caps.
21465 Original commit message from CVS:
21466 * gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
21467 (is_demuxer_element), (new_caps):
21468 * gst/playback/gstplaybasebin.c: (source_new_pad):
21469 Fix the case where we try to ref a NULL element when we delay a link
21470 because of unfixed caps.
21471 Set the state of autoplugged decodebins to PAUSED.
21472 RTSP now works in playbin, we can remove it from the blacklist.
21474 2007-01-09 14:33:24 +0000 Tim-Philipp Müller <tim@centricular.net>
21476 gst/playback/: Post missing-plugin messages on the bus for missing sources and missing decoders/demuxers/depayloaders...
21477 Original commit message from CVS:
21478 * gst/playback/Makefile.am:
21479 * gst/playback/gstplaybasebin.c: (string_arr_has_str),
21480 (unknown_type), (setup_subtitle), (gen_source_element):
21481 * gst/playback/gstplaybin.c: (plugin_init):
21482 Post missing-plugin messages on the bus for missing sources and
21483 missing decoders/demuxers/depayloaders; fix error code used when
21484 we're missing an URI handler source; for media types that we are not
21485 handling on purpose at the moment, don't print "don't know how to
21486 handle xyz" messages to the terminal or post missing-plugin
21487 messages on the bus.
21488 * tests/check/elements/playbin.c: (create_playbin),
21489 (GST_START_TEST), (gst_codec_src_uri_get_type),
21490 (gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
21491 (gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
21492 (gst_codec_src_init_type), (gst_codec_src_base_init),
21493 (gst_codec_src_create), (gst_codec_src_class_init),
21494 (gst_codec_src_init), (plugin_init), (playbin_suite):
21495 Add some tests for the missing-plugin stuff.
21497 2007-01-09 14:20:08 +0000 Tim-Philipp Müller <tim@centricular.net>
21499 API: add new libgstbaseutils library with functions
21500 Original commit message from CVS:
21502 * gst-libs/gst/Makefile.am:
21503 * gst-libs/gst/utils/Makefile.am:
21504 * gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
21505 * gst-libs/gst/utils/base-utils.h:
21506 * gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
21507 (find_format_info), (caps_are_rtp_caps),
21508 (gst_base_utils_get_source_description),
21509 (gst_base_utils_get_sink_description),
21510 (gst_base_utils_get_decoder_description),
21511 (gst_base_utils_get_encoder_description),
21512 (gst_base_utils_get_element_description),
21513 (gst_base_utils_add_codec_description_to_tag_list),
21514 (gst_base_utils_get_codec_description), (gst_base_utils_list_all):
21515 * gst-libs/gst/utils/descriptions.h:
21516 * gst-libs/gst/utils/missing-plugins.c:
21517 (missing_structure_get_type), (copy_and_clean_caps),
21518 (gst_missing_uri_source_message_new),
21519 (gst_missing_uri_sink_message_new),
21520 (gst_missing_element_message_new),
21521 (gst_missing_decoder_message_new),
21522 (gst_missing_encoder_message_new),
21523 (missing_structure_get_string_detail),
21524 (missing_structure_get_caps_detail),
21525 (gst_missing_plugin_message_get_installer_detail),
21526 (gst_missing_plugin_message_get_description),
21527 (gst_is_missing_plugin_message):
21528 * gst-libs/gst/utils/missing-plugins.h:
21529 API: add new libgstbaseutils library with functions
21530 - to create and parse missing-plugins messages
21531 - that provide (translated) descriptions for caps/decoders/sources/etc.
21533 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
21534 * pkgconfig/gstreamer-plugins-base.pc.in:
21536 * docs/libs/gst-plugins-base-libs-docs.sgml:
21537 * docs/libs/gst-plugins-base-libs-sections.txt:
21538 Generate docs for new lib and API.
21539 * tests/check/Makefile.am:
21540 * tests/check/libs/.cvsignore:
21541 * tests/check/libs/utils.c: (missing_msg_check_getters),
21542 (GST_START_TEST), (libgstbaseutils_suite):
21543 Add some basic unit tests.
21545 2007-01-09 13:35:08 +0000 Tim-Philipp Müller <tim@centricular.net>
21547 ext/ogg/Makefile.am: Dist gstoggdemux.h to fix 'make distcheck'.
21548 Original commit message from CVS:
21549 * ext/ogg/Makefile.am:
21550 Dist gstoggdemux.h to fix 'make distcheck'.
21551 * sys/v4l/Makefile.am:
21552 Fix 'make distcheck' even more.
21554 2007-01-09 12:30:46 +0000 Wim Taymans <wim.taymans@gmail.com>
21557 Original commit message from CVS:
21558 * docs/plugins/Makefile.am:
21559 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
21560 * docs/plugins/gst-plugins-base-plugins-sections.txt:
21561 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
21562 (gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
21563 (gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
21564 (gst_ogg_demux_perform_seek):
21565 * ext/ogg/gstoggdemux.h:
21567 Add some more comments.
21570 2007-01-09 11:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
21572 Small documentation updates/fixes
21573 Original commit message from CVS:
21574 * ext/theora/theoradec.c:
21575 * ext/vorbis/vorbisdec.c:
21576 * gst-libs/gst/audio/gstringbuffer.c:
21577 (gst_ring_buffer_commit_full):
21578 * gst-libs/gst/audio/gstringbuffer.h:
21579 * gst-libs/gst/rtp/gstrtpbuffer.c:
21580 * gst-libs/gst/tag/gstvorbistag.c:
21581 Small documentation updates/fixes
21583 2007-01-09 10:37:01 +0000 Tim-Philipp Müller <tim@centricular.net>
21585 configure.ac: Require core CVS HEAD for Andy's basesrc/sink API additions.
21586 Original commit message from CVS:
21588 Require core CVS HEAD for Andy's basesrc/sink API additions.
21590 2007-01-08 14:01:23 +0000 Günter Thelen <daedalus.inc@gmx.net>
21592 gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
21593 Original commit message from CVS:
21594 Patch by: Günter Thelen <daedalus dot inc at gmx net>
21595 * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
21597 Add typefinder for flac-in-ogg in conformance with the ogg-mapping
21598 on flac.sf.net (there appear to be other versions of the first
21599 ogg page in the wild) (#391365).
21601 2007-01-08 13:32:32 +0000 Tim-Philipp Müller <tim@centricular.net>
21603 configure.ac: Check if localtime_r() is available.
21604 Original commit message from CVS:
21606 Check if localtime_r() is available.
21607 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
21608 If localtime_r() is not available, fall back to localtime(). Should
21609 fix build on MingW (#393310).
21611 2007-01-08 12:30:03 +0000 Tim-Philipp Müller <tim@centricular.net>
21613 gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
21614 Original commit message from CVS:
21615 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
21616 * gst/subparse/gstsubparse.h:
21617 Remove spurious 1000 subtrahend when calculating the timestamp from
21618 the frame number and the frame rate . Also, use the frames/second
21619 value specified in the first line of the file, if one is specified
21620 there. Should fix #357503.
21621 * tests/check/elements/subparse.c: (do_test),
21622 (test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
21624 Add some basic unit tests for the microdvd subtitle format.
21626 2007-01-07 21:53:38 +0000 Young-Ho Cha <ganadist@chollian.net>
21628 sys/xvimage/xvimagesink.c: Fixes : #390076.
21629 Original commit message from CVS:
21630 2007-01-07 Julien MOUTTE <julien@moutte.net>
21631 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
21632 (gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
21633 (gst_xvimagesink_xvimage_put),
21634 (gst_lookup_xv_port_from_adaptor),
21635 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
21636 (gst_xvimagesink_set_xwindow_id),
21637 (gst_xvimagesink_set_event_handling),
21638 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
21639 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
21640 Patch by : Young-Ho Cha <ganadist at chollian dot net>
21642 Add an adaptor property to select a specific XV adaptor.
21643 * sys/xvimage/xvimagesink.h:
21645 2007-01-07 18:50:13 +0000 Julien Moutte <julien@moutte.net>
21647 sys/: Use flow_lock much more to protect every access to xwindow.
21648 Original commit message from CVS:
21649 2007-01-07 Julien MOUTTE <julien@moutte.net>
21650 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
21651 (gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
21652 (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
21653 (gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
21654 (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
21655 (gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
21656 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
21657 (gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
21658 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
21659 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
21660 (gst_xvimagesink_change_state),
21661 (gst_xvimagesink_set_xwindow_id),
21662 (gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
21663 Use flow_lock much more to protect every access to xwindow.
21664 Try to catch erros while creating images in case some drivers
21666 just generating an XError when the requested image is too big.
21667 Should fix : #354698, #384008, #384060.
21668 * tests/icles/stress-xoverlay.c: (cycle_window),
21670 Implement some stress testing of setting window xid.
21672 2007-01-07 10:33:55 +0000 Sébastien Moutte <sebastien@moutte.net>
21674 win32/common/libgsaudio.def: Add new exported function.
21675 Original commit message from CVS:
21676 * win32/common/libgsaudio.def:
21677 Add new exported function.
21678 * win32/common/libgstogg.dsp:
21679 Add gstoggaviparse.c to the build.
21680 * win32/common/libgstvideoscale.dsp:
21681 Add vs_4tap.c to the build.
21682 * win32/common/libgstvorbis.dsp:
21683 Add vorbistag.c to the build.
21685 2007-01-06 17:28:40 +0000 Andy Wingo <wingo@pobox.com>
21688 * gst-libs/gst/audio/gstbaseaudiosink.c:
21689 gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
21690 Original commit message from CVS:
21691 2007-01-06 Andy Wingo <wingo@pobox.com>
21692 * gst-libs/gst/audio/gstbaseaudiosink.c
21693 (gst_base_audio_sink_class_init)
21694 (gst_base_audio_sink_init):
21695 (gst_base_audio_sink_activate_pull): Add an activate_pull function
21696 to baseaudiosink, and tell basesink that we can work in pull mode.
21697 This way the ring buffer thread drives the pipeline directly, if
21698 pull mode is possible. There is some lingering nastiness regarding
21700 (gst_base_audio_sink_callback): Implement the callback to pull
21701 data. This interface is a bit light, though -- it should get a
21702 GstFlowReturn return value at least.
21704 2007-01-05 19:43:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21706 Printf format and missing argument fixes.
21707 Original commit message from CVS:
21708 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
21709 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
21710 * gst/playback/gstdecodebin2.c:
21711 (gst_decode_group_check_if_blocked):
21712 Printf format and missing argument fixes.
21714 2007-01-05 18:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21716 ext/ogg/gstogmparse.c: Activate pads before adding them to the element.
21717 Original commit message from CVS:
21718 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
21719 (gst_ogm_parse_change_state):
21720 Activate pads before adding them to the element.
21722 2007-01-05 16:02:50 +0000 Tim-Philipp Müller <tim@centricular.net>
21724 tests/examples/seek/: Call g_thread_init() first thing in main() (see #391278).
21725 Original commit message from CVS:
21726 * tests/examples/seek/scrubby.c: (main):
21727 * tests/examples/seek/seek.c: (main):
21728 Call g_thread_init() first thing in main() (see #391278).
21730 2007-01-05 12:19:34 +0000 Tim-Philipp Müller <tim@centricular.net>
21732 tests/check/: Add test for GstNetBuffer + gst_buffer_copy(). Disabled for the time being, since it's broken, see #393...
21733 Original commit message from CVS:
21734 * tests/check/Makefile.am:
21735 * tests/check/libs/.cvsignore:
21736 * tests/check/libs/netbuffer.c: (GST_START_TEST),
21738 Add test for GstNetBuffer + gst_buffer_copy(). Disabled
21739 for the time being, since it's broken, see #393099.
21741 2007-01-05 12:13:24 +0000 Tim-Philipp Müller <tim@centricular.net>
21743 tests/check/Makefile.am: Update to use GST_PLUGINS_BASE_CFLAGS as well.
21744 Original commit message from CVS:
21745 * tests/check/Makefile.am:
21746 Update to use GST_PLUGINS_BASE_CFLAGS as well.
21748 2007-01-04 12:49:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21750 configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
21751 Original commit message from CVS:
21753 split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
21754 so that GST_BASE_CFLAGS can go inbetween them, making sure
21755 we use uninstalled gst-libs headers
21756 * docs/libs/Makefile.am:
21757 * ext/alsa/Makefile.am:
21758 * ext/cdparanoia/Makefile.am:
21759 * ext/gnomevfs/Makefile.am:
21760 * ext/libvisual/Makefile.am:
21761 * ext/ogg/Makefile.am:
21762 * ext/theora/Makefile.am:
21763 * ext/vorbis/Makefile.am:
21764 * gst-libs/gst/audio/Makefile.am:
21765 * gst-libs/gst/cdda/Makefile.am:
21766 * gst-libs/gst/interfaces/Makefile.am:
21767 * gst-libs/gst/riff/Makefile.am:
21768 * gst-libs/gst/rtp/Makefile.am:
21769 * gst-libs/gst/tag/Makefile.am:
21770 * gst/adder/Makefile.am:
21771 * gst/audioconvert/Makefile.am:
21772 * gst/audiorate/Makefile.am:
21773 * gst/audioresample/Makefile.am:
21774 * gst/playback/Makefile.am:
21775 * gst/tcp/Makefile.am:
21776 * gst/videoscale/Makefile.am:
21777 * gst/volume/Makefile.am:
21778 * sys/ximage/Makefile.am:
21779 * sys/xvimage/Makefile.am:
21780 * tests/icles/Makefile.am:
21783 2007-01-04 11:30:53 +0000 Julien Moutte <julien@moutte.net>
21785 Add a method to the XOverlay interface to allow disabling of event handling in x[v]imagesink elements. This will let ...
21786 Original commit message from CVS:
21787 2007-01-04 Julien MOUTTE <julien@moutte.net>
21788 * gst-libs/gst/interfaces/xoverlay.c:
21789 (gst_x_overlay_handle_events):
21790 * gst-libs/gst/interfaces/xoverlay.h:
21791 * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
21792 (gst_ximagesink_set_xwindow_id),
21793 (gst_ximagesink_set_event_handling),
21794 (gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
21795 (gst_ximagesink_get_property), (gst_ximagesink_init),
21796 (gst_ximagesink_class_init):
21797 * sys/ximage/ximagesink.h:
21798 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
21799 (gst_xvimagesink_set_xwindow_id),
21800 (gst_xvimagesink_set_event_handling),
21801 (gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
21802 (gst_xvimagesink_get_property), (gst_xvimagesink_init),
21803 (gst_xvimagesink_class_init):
21804 * sys/xvimage/xvimagesink.h:
21805 * tests/icles/stress-xoverlay.c: (toggle_events),
21807 Add a method to the XOverlay interface to allow disabling of
21808 event handling in x[v]imagesink elements. This will let X events
21809 propagate to parent windows which can be usefull in some cases.
21810 Be carefull that the application is then responsible of pushing
21811 navigation events and expose events to the video sink.
21814 2007-01-03 15:45:06 +0000 Tim-Philipp Müller <tim@centricular.net>
21816 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION (fixes #392070).
21817 Original commit message from CVS:
21818 * gst-libs/gst/tag/gstvorbistag.c:
21819 * tests/check/libs/tag.c: (GST_START_TEST):
21820 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
21823 2007-01-01 18:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
21826 Original commit message from CVS:
21828 * docs/Makefile.am:
21829 * docs/design/Makefile.am:
21832 2006-12-27 17:15:35 +0000 Julien Moutte <julien@moutte.net>
21834 docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation typo. Fixes: #390063.
21835 Original commit message from CVS:
21836 2006-12-27 Julien MOUTTE <julien@moutte.net>
21837 * docs/libs/gst-plugins-base-libs-sections.txt: Fix a
21839 typo. Fixes: #390063.
21841 2006-12-27 12:08:13 +0000 Julien Moutte <julien@moutte.net>
21843 sys/: Plug a caps leak.
21844 Original commit message from CVS:
21845 2006-12-27 Julien MOUTTE <julien@moutte.net>
21846 * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
21847 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
21849 * win32/common/config.h: Updated.
21851 2006-12-22 12:10:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21853 tests/check/elements/: Fix the dp tests, but activating the pads for the streamheader tests too and cleaning up condi...
21854 Original commit message from CVS:
21855 * tests/check/elements/gdpdepay.c: (cleanup_gdpdepay),
21856 (setup_gdpdepay_streamheader):
21857 * tests/check/elements/gdppay.c: (cleanup_gdppay),
21858 (setup_gdppay_streamheader):
21859 Fix the dp tests, but activating the pads for the streamheader tests
21860 too and cleaning up conditionaly
21862 2006-12-22 11:09:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21864 gst/ffmpegcolorspace/: Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the other end of the wo...
21865 Original commit message from CVS:
21866 * gst/ffmpegcolorspace/avcodec.h:
21867 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
21868 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
21869 (gst_ffmpegcsp_avpicture_fill):
21870 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
21871 (img_get_alpha_info):
21872 Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
21873 other end of the word. Fixes: #387073.
21874 Add some inconsequential branch hints in a couple of places.
21876 2006-12-21 12:30:11 +0000 Tim-Philipp Müller <tim@centricular.net>
21878 gst/ffmpegcolorspace/gstffmpegcodecmap.c: The "signed" field in raw audio caps is of boolean type, trying to extract ...
21879 Original commit message from CVS:
21880 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
21881 (gst_ffmpeg_caps_to_smpfmt):
21882 The "signed" field in raw audio caps is of boolean type, trying to
21883 extract the value with _get_int() will fail (fix to keep in sync with
21884 the copy in gst-ffmpeg)
21886 2006-12-21 08:12:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21888 tests/check/elements/: consistent pad (de)activation
21889 Original commit message from CVS:
21890 * tests/check/elements/audioresample.c: (cleanup_audioresample):
21891 * tests/check/elements/audiotestsrc.c: (cleanup_audiotestsrc):
21892 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
21893 (cleanup_gdpdepay):
21894 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay):
21895 * tests/check/elements/subparse.c: (teardown_subparse):
21896 * tests/check/elements/textoverlay.c: (cleanup_textoverlay):
21897 * tests/check/elements/videorate.c: (cleanup_videorate):
21898 * tests/check/elements/videotestsrc.c: (cleanup_videotestsrc):
21899 * tests/check/elements/volume.c: (cleanup_volume):
21900 * tests/check/elements/vorbisdec.c: (setup_vorbisdec),
21901 (cleanup_vorbisdec):
21902 * tests/check/elements/vorbistag.c: (setup_vorbistag),
21903 (cleanup_vorbistag):
21904 consistent pad (de)activation
21906 2006-12-20 10:29:58 +0000 Tim-Philipp Müller <tim@centricular.net>
21908 gst/typefind/gsttypefindfunctions.c: Forgot to register the extensions.
21909 Original commit message from CVS:
21910 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
21911 Forgot to register the extensions.
21913 2006-12-20 09:25:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21915 gst/typefind/gsttypefindfunctions.c: Add typefinder for VIVO files (my christmas present to the 90s).
21916 Original commit message from CVS:
21917 * gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
21919 Add typefinder for VIVO files (my christmas present to the 90s).
21921 2006-12-16 13:59:09 +0000 Tim-Philipp Müller <tim@centricular.net>
21923 gst/playback/gstdecodebin.c: Special-case the text/plain media type: we only want to recognise it as a 'raw' decoded ...
21924 Original commit message from CVS:
21925 * gst/playback/gstdecodebin.c: (type_found):
21926 Special-case the text/plain media type: we only want to recognise it
21927 as a 'raw' decoded media type if it comes from a demuxer or subtitle
21928 parser, but not if the entire stream is of text/plain type. If the
21929 entire stream is text/plain, we should just error out.
21930 This fixes playback of audio files with lyrics in totem. Totem can't
21931 distinguish between text files and subtitle files and passes any
21932 .txt file with the same basename as the main file to playbin as
21933 suburi, and playbin will then throw a 'subtitle found, but no video
21934 stream' error, which isn't entirely helpful. See #380342.
21935 Also, with this change we'll show a slightly more correct error
21936 message in case totem passes a playlist file to us (although a
21937 custom error message wording instead of the default text would
21938 probably not be a bad idea either).
21939 Same problem also needs to be fixed for playbin+decodebin2.
21940 * tests/check/Makefile.am:
21941 * tests/check/elements/decodebin.c: (src_handoff_cb),
21942 (decodebin_new_decoded_pad_cb), (GST_START_TEST),
21944 Add simple unit test for decodebin for the above.
21946 2006-12-16 12:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
21948 gst/playback/: Refuse to change state to READY when we failed to create any of the required elements in our instance ...
21949 Original commit message from CVS:
21950 * gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
21951 * gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
21952 Refuse to change state to READY when we failed to create any of the
21953 required elements in our instance init function.
21955 2006-12-15 10:52:23 +0000 Tim-Philipp Müller <tim@centricular.net>
21957 docs/libs/gst-plugins-base-libs-sections.txt: Small docs fixes/updates.
21958 Original commit message from CVS:
21959 * docs/libs/gst-plugins-base-libs-sections.txt:
21960 Small docs fixes/updates.
21961 * gst-libs/gst/video/gstvideosink.h:
21962 Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
21963 from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
21964 removed from the base sink API between 0.9.6 and 0.9.7).
21965 API: add GST_VIDEO_SINK_CAST and use it for the height/width
21966 accessor macros, so we don't do a runtime GObject type check every
21969 2006-12-15 00:20:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21972 Original commit message from CVS:
21974 * gst-plugins-base.doap:
21975 * gst-plugins-base.spec.in:
21978 2006-12-09 15:12:38 +0000 Jens Granseuer <jensgr@gmx.net>
21980 Declare variables at the beginning of a block. Fixes #383195.
21981 Original commit message from CVS:
21982 Patch by: Jens Granseuer <jensgr at gmx net>
21983 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
21984 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
21985 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
21986 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
21987 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
21988 Declare variables at the beginning of a block. Fixes #383195.
21990 2006-12-07 02:38:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21992 configure.ac: Bump version nano - back to CVS.
21993 Original commit message from CVS:
21995 Bump version nano - back to CVS.
21997 === release 0.10.11 ===
21999 2006-12-07 02:30:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22001 configure.ac: releasing 0.10.11, "Dumb things"
22002 Original commit message from CVS:
22003 === release 0.10.11 ===
22004 2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
22006 releasing 0.10.11, "Dumb things"
22008 2006-12-05 12:44:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22010 gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
22011 Original commit message from CVS:
22012 * gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
22013 (close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
22014 Handle the case where an element has multiple pads with
22015 unfixed caps as well as still possibly producing more dynamic
22016 pads by storing each case as a distinct entry in the dynamic list.
22017 Fixes #38223 again.
22019 2006-12-04 13:02:37 +0000 Wim Taymans <wim.taymans@gmail.com>
22021 gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
22022 Original commit message from CVS:
22023 * gst/playback/gstdecodebin.c: (close_pad_link):
22024 Fix #382223, add more dynamic caps handling.
22026 2006-12-01 11:35:57 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
22029 Ignore all pot files
22030 Original commit message from CVS:
22031 Ignore all pot files
22033 2006-12-01 10:36:50 +0000 Michael Smith <msmith@xiph.org>
22035 gst/audiorate/gstaudiorate.c: Delete bad debug code.
22036 Original commit message from CVS:
22037 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
22038 Delete bad debug code.
22041 2006-12-01 10:27:54 +0000 Sergey Scobich <sergey.scobich@gmail.com>
22043 Fix compilation on win32 under VS8
22044 Original commit message from CVS:
22045 * gst/videoscale/vs_4tap.c:
22047 * win32/common/config.h:
22048 * win32/vs8/libgstvideoscale.vcproj:
22049 Fix compilation on win32 under VS8
22050 Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
22051 Partially fixes #381175
22053 2006-11-30 23:46:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22070 Original commit message from CVS:
22073 2006-11-30 12:50:42 +0000 Michael Smith <msmith@xiph.org>
22075 tests/check/pipelines/theoraenc.c: It would be very bad if, after a discont buffer, we thought every single following...
22076 Original commit message from CVS:
22077 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
22079 It would be very bad if, after a discont buffer, we thought every
22080 single following buffer was also discont. So, add to the test to
22081 ensure that this isn't the case.
22082 * ext/theora/theoraenc.c: (theora_enc_is_discontinuous):
22083 ... it was the case. So fix it.
22085 2006-11-28 16:43:18 +0000 Wim Taymans <wim.taymans@gmail.com>
22087 gst/playback/gstplaybasebin.c: Improve debug.
22088 Original commit message from CVS:
22089 * gst/playback/gstplaybasebin.c: (check_queue_event):
22091 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
22092 Fix width and height range from 16 - 4096 to 1 - MAXINT, just like the
22093 padtemplate caps. Refixes #357577.
22095 2006-11-28 16:21:27 +0000 Wim Taymans <wim.taymans@gmail.com>
22097 gst/playback/gstplaybasebin.c: Add event probe to see when EOS is in a queue and we can disable the underrun signals....
22098 Original commit message from CVS:
22099 * gst/playback/gstplaybasebin.c: (check_queue_event),
22100 (queue_threshold_reached), (queue_out_of_data),
22101 (gen_preroll_element):
22102 Add event probe to see when EOS is in a queue and we can disable the
22103 underrun signals. Fixes #357577.
22105 2006-11-28 14:40:39 +0000 Edward Hervey <bilboed@bilboed.com>
22107 gst/playback/: New decodebin2 element.
22108 Original commit message from CVS:
22109 * gst/playback/Makefile.am:
22110 * gst/playback/gstdecodebin2.c: (gst_decode_bin_get_type),
22111 (_gst_boolean_accumulator), (gst_decode_bin_class_init),
22112 (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
22113 (gst_decode_bin_init), (gst_decode_bin_dispose),
22114 (gst_decode_bin_finalize), (gst_decode_bin_set_property),
22115 (gst_decode_bin_get_property), (gst_decode_bin_set_caps),
22116 (gst_decode_bin_get_caps), (gst_decode_bin_autoplug_continue),
22117 (gst_decode_bin_autoplug_sort), (analyze_new_pad), (connect_pad),
22118 (connect_element), (expose_pad), (type_found),
22119 (pad_added_group_cb), (pad_removed_group_cb),
22120 (no_more_pads_group_cb), (pad_added_cb), (pad_removed_cb),
22121 (no_more_pads_cb), (find_compatibles), (is_demuxer_element),
22122 (are_raw_caps), (multi_queue_overrun_cb),
22123 (multi_queue_underrun_cb), (gst_decode_group_new),
22124 (get_current_group), (group_demuxer_event_probe),
22125 (gst_decode_group_control_demuxer_pad),
22126 (gst_decode_group_control_source_pad),
22127 (gst_decode_group_check_if_blocked),
22128 (gst_decode_group_check_if_drained), (gst_decode_group_expose),
22129 (gst_decode_group_hide), (gst_decode_group_free),
22130 (gst_decode_group_set_complete), (source_pad_blocked_cb),
22131 (source_pad_event_probe), (gst_decode_pad_new), (add_fakesink),
22132 (remove_fakesink), (find_sink_pad), (gst_decode_bin_change_state),
22134 New decodebin2 element.
22136 * gst/playback/gstplay-marshal.list:
22137 Added marshallers for new signals in decodebin2
22138 * gst/playback/gstplaybasebin.c: (setup_subtitle), (make_decoder):
22139 Use decodebin2 if *and only if* the USE_DECODEBIN2 environment variable
22142 2006-11-28 10:45:40 +0000 Wim Taymans <wim.taymans@gmail.com>
22144 gst/playback/gstplaybasebin.c: Disable rtsp:// uris for the release, it's not good enough yet.
22145 Original commit message from CVS:
22146 * gst/playback/gstplaybasebin.c: (setup_source),
22147 (gst_play_base_bin_change_state):
22148 Disable rtsp:// uris for the release, it's not good enough yet.
22151 2006-11-26 16:39:41 +0000 Wim Taymans <wim.taymans@gmail.com>
22153 ext/theora/theoradec.c: Implement reverse playback.
22154 Original commit message from CVS:
22155 * ext/theora/theoradec.c: (gst_theora_dec_reset),
22156 (theora_dec_push_forward), (theora_dec_push_reverse),
22157 (theora_handle_data_packet), (theora_dec_decode_buffer),
22158 (theora_dec_flush_decode), (theora_dec_chain_reverse),
22159 (theora_dec_chain_forward), (theora_dec_chain):
22160 Implement reverse playback.
22161 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
22162 (vorbis_dec_decode_buffer), (vorbis_dec_flush_decode),
22163 (vorbis_dec_chain_forward):
22164 Clear buffers used for reverse playback in _reset.
22165 No need to set the eos flag, we clip samples using the segment.
22167 2006-11-24 15:40:58 +0000 Wim Taymans <wim.taymans@gmail.com>
22169 ext/ogg/gstoggdemux.c: Some cleanups.
22170 Original commit message from CVS:
22171 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
22172 (gst_ogg_pad_init), (gst_ogg_pad_dispose), (gst_ogg_pad_reset),
22173 (gst_ogg_pad_stream_out), (gst_ogg_pad_submit_page),
22174 (gst_ogg_chain_reset), (gst_ogg_demux_perform_seek):
22176 Handle continued pages in reverse mode.
22178 2006-11-24 15:39:03 +0000 Wim Taymans <wim.taymans@gmail.com>
22180 ext/vorbis/vorbisdec.c: Small cleanups.
22181 Original commit message from CVS:
22182 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
22183 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
22184 (vorbis_dec_flush_decode):
22186 Don't try to add invalid timestamps.
22187 Clipping will unref the buffer.
22189 2006-11-24 08:56:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22191 gst/: remove obsolete _factory_init protos
22192 Original commit message from CVS:
22193 * gst/adder/gstadder.h:
22194 * gst/audiotestsrc/gstaudiotestsrc.h:
22195 remove obsolete _factory_init protos
22197 2006-11-24 08:35:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22199 sys/xvimage/xvimagesink.c: Fix spacing in debug message.
22200 Original commit message from CVS:
22201 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
22202 Fix spacing in debug message.
22204 2006-11-23 11:07:23 +0000 Wim Taymans <wim.taymans@gmail.com>
22206 ext/ogg/gstoggdemux.c: Don't just ignore return values from _pad_push().
22207 Original commit message from CVS:
22208 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
22209 (gst_ogg_demux_chain):
22210 Don't just ignore return values from _pad_push().
22211 Small debug improvements.
22213 2006-11-23 11:02:11 +0000 Michael Smith <msmith@xiph.org>
22215 ext/ogg/gstoggmux.c: If our incoming buffer is marked as DISCONT, then increment the page number (so that the discont...
22216 Original commit message from CVS:
22217 * ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
22218 If our incoming buffer is marked as DISCONT, then increment the page
22219 number (so that the discontinuity is marked in the final ogg
22220 bitstream) and flush the previous page.
22222 2006-11-22 14:34:03 +0000 Michael Smith <msmith@xiph.org>
22224 ext/theora/: Mark discontinuities of > 3/4 of a frame, reinit encoder.
22225 Original commit message from CVS:
22226 * ext/theora/gsttheoraenc.h:
22227 * ext/theora/theoraenc.c: (gst_theora_enc_init),
22228 (theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
22229 (theora_buffer_from_packet), (theora_enc_is_discontinuous),
22230 (theora_enc_chain), (theora_enc_change_state):
22231 Mark discontinuities of > 3/4 of a frame, reinit encoder.
22232 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
22233 (GST_START_TEST), (theoraenc_suite):
22234 Enable discontinuity test, fix it.
22236 2006-11-21 18:39:34 +0000 Tim-Philipp Müller <tim@centricular.net>
22238 ext/pango/gsttextoverlay.*: Some textoverlay fixes: for one, in the video chain function, actually wait for a text bu...
22239 Original commit message from CVS:
22240 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
22241 (gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
22242 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
22243 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
22244 (gst_text_overlay_change_state):
22245 * ext/pango/gsttextoverlay.h:
22246 Some textoverlay fixes: for one, in the video chain function,
22247 actually wait for a text buffer to come in if there is none at the
22248 moment and there should be one; also, deal more gracefully with
22249 incoming buffers that do not have a timestamp or duration; discard
22250 text buffer when not needed any longer. Fixes #341681.
22251 * tests/check/Makefile.am:
22252 * tests/check/elements/.cvsignore:
22253 * tests/check/elements/textoverlay.c:
22254 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
22255 (setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
22256 (create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
22257 (test_video_waits_for_text_send_text_newsegment_thread),
22258 (test_video_waits_for_text_shutdown_element),
22259 (test_render_continuity_push_video_buffers_thread),
22260 (textoverlay_suite):
22261 Add some unit tests for textoverlay.
22263 2006-11-21 09:29:56 +0000 Tim-Philipp Müller <tim@centricular.net>
22265 gst/typefind/gsttypefindfunctions.c: Avoid integer underflow when the found probability for mp3 is smaller than the '...
22266 Original commit message from CVS:
22267 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
22268 Avoid integer underflow when the found probability for mp3 is
22269 smaller than the 'penalty' we subtract if there's not a clean
22270 mp3 header sync at offset 0.
22272 2006-11-21 08:17:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22274 docs/libs/gst-plugins-base-libs-sections.txt: Add some new symbols to the docs
22275 Original commit message from CVS:
22276 * docs/libs/gst-plugins-base-libs-sections.txt:
22277 Add some new symbols to the docs
22279 2006-11-20 16:44:28 +0000 Tim-Philipp Müller <tim@centricular.net>
22281 tests/check/: Enable ffmpegcolorspace test now that the RGBA32 issue is fixed (for now not for valgrinding though, si...
22282 Original commit message from CVS:
22283 * tests/check/Makefile.am:
22284 * tests/check/elements/ffmpegcolorspace.c:
22285 (ffmpegcolorspace_suite):
22286 Enable ffmpegcolorspace test now that the RGBA32 issue is fixed
22287 (for now not for valgrinding though, since it takes too long).
22289 2006-11-20 15:01:09 +0000 Wim Taymans <wim.taymans@gmail.com>
22291 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix RGBA32 caps. Fixes #357038.
22292 Original commit message from CVS:
22293 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
22294 (gst_ffmpeg_pixfmt_to_caps):
22295 Fix RGBA32 caps. Fixes #357038.
22297 2006-11-20 12:20:39 +0000 Tim-Philipp Müller <tim@centricular.net>
22299 gst-libs/gst/interfaces/mixertrack.h: Add FIXME so we can add some padding here in 0.11
22300 Original commit message from CVS:
22301 * gst-libs/gst/interfaces/mixertrack.h:
22302 Add FIXME so we can add some padding here in 0.11
22304 2006-11-19 17:07:34 +0000 Tim-Philipp Müller <tim@centricular.net>
22306 gst-libs/gst/rtp/gstbasertpaudiopayload.h: Fix GstBaseRTPAudioPayload structure so the whole GObject inheritance busi...
22307 Original commit message from CVS:
22308 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
22309 Fix GstBaseRTPAudioPayload structure so the whole GObject
22310 inheritance business actually works (parent class instance structure
22311 must always come first in the derived class instance structure).
22313 2006-11-16 14:35:30 +0000 Tim-Philipp Müller <tim@centricular.net>
22315 Make sure our checks and the videotestsrc plugin link against the local uninstalled gst libs and not any installed gs...
22316 Original commit message from CVS:
22317 * gst/videotestsrc/Makefile.am:
22318 * tests/check/Makefile.am:
22319 Make sure our checks and the videotestsrc plugin link against the
22320 local uninstalled gst libs and not any installed gst libs that
22321 might happen to exist as well.
22322 * tests/check/elements/adder.c: (message_received),
22323 (test_event_message_received), (test_play_twice_message_received):
22324 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
22325 Fix compiler warnings when compiling against core with disabled
22328 2006-11-16 12:55:08 +0000 Michael Smith <msmith@xiph.org>
22330 gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
22331 Original commit message from CVS:
22332 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
22333 (gst_audio_rate_sink_event), (gst_audio_rate_chain):
22334 Fix audiorate, so that it accurately sets offsets and timestamps.
22335 Doesn't change the fundamental algorithmic decisions; so should be
22337 * tests/check/Makefile.am:
22338 Enable audiorate test now that it passes.
22340 2006-11-15 10:05:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22342 sys/xvimage/xvimagesink.c: clear xv when going to NULL, remove // commented non-existant proto
22343 Original commit message from CVS:
22344 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
22345 clear xv when going to NULL, remove // commented non-existant proto
22346 * tests/examples/seek/seek.c: (main):
22347 add missing tooltip description for scrub and play_scrub
22349 2006-11-14 23:34:19 +0000 David Schleef <ds@schleef.org>
22351 configure.ac: Bump liboil requirement to 0.3.8.
22352 Original commit message from CVS:
22354 Bump liboil requirement to 0.3.8.
22355 * gst-libs/gst/riff/riff-media.c:
22357 * gst/videoscale/vs_image.h:
22358 * gst/videoscale/vs_scanline.h:
22359 Use liboil's stdint.h.
22360 * gst/videotestsrc/videotestsrc.c:
22361 Remove liboil related ifdef's, since they aren't needed now, and
22362 won't work with future versions.
22364 2006-11-14 23:08:38 +0000 David Schleef <ds@schleef.org>
22366 gst/videoscale/: Add a 4-tap image scaler. Theoretically looks much prettier.
22367 Original commit message from CVS:
22368 * gst/videoscale/Makefile.am:
22369 * gst/videoscale/gstvideoscale.c:
22370 * gst/videoscale/gstvideoscale.h:
22371 * gst/videoscale/vs_4tap.c:
22372 * gst/videoscale/vs_4tap.h:
22373 * gst/videoscale/vs_image.c:
22374 * gst/videoscale/vs_image.h:
22375 * gst/videoscale/vs_scanline.c:
22376 * gst/videoscale/vs_scanline.h:
22377 Add a 4-tap image scaler. Theoretically looks much prettier.
22378 The tap calculation could use some improvement.
22380 2006-11-14 11:54:14 +0000 Jan David Mol <j.j.d.mol@tudelft.nl>
22382 Various gsize and gssize printf fixes. Fixes #372507.
22383 Original commit message from CVS:
22384 Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
22385 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
22386 (gst_riff_parse_strf_iavs):
22387 * gst/subparse/gstsubparse.c: (convert_encoding):
22388 * gst/tcp/gstmultifdsink.c:
22389 (gst_multi_fd_sink_handle_client_write):
22390 * gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
22391 (gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
22392 (gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
22393 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
22394 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
22395 (gst_ximagesink_ximage_new):
22396 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
22397 Various gsize and gssize printf fixes. Fixes #372507.
22399 2006-11-13 18:14:48 +0000 Wim Taymans <wim.taymans@gmail.com>
22401 ext/vorbis/vorbisdec.*: First stab at vorbis reverse playback.
22402 Original commit message from CVS:
22403 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
22404 (vorbis_dec_push_forward), (vorbis_dec_push_reverse),
22405 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
22406 (vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
22407 (vorbis_dec_chain_forward), (vorbis_dec_chain):
22408 * ext/vorbis/vorbisdec.h:
22409 First stab at vorbis reverse playback.
22411 2006-11-13 17:30:17 +0000 Wim Taymans <wim.taymans@gmail.com>
22413 gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
22414 Original commit message from CVS:
22415 * gst-libs/gst/audio/gstbaseaudiosink.c:
22416 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
22417 * gst-libs/gst/audio/gstbaseaudiosink.h:
22418 Make the clock sync code more accurate wrt resampling and playback
22419 at different rates.
22420 * gst-libs/gst/audio/gstringbuffer.c:
22421 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
22422 * gst-libs/gst/audio/gstringbuffer.h:
22423 Use better algorithm to interpolate sample rates.
22425 2006-11-13 15:31:01 +0000 Michael Smith <msmith@xiph.org>
22427 ext/ogg/gstoggdemux.c: Improve a debug line slightly.
22428 Original commit message from CVS:
22429 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
22430 Improve a debug line slightly.
22431 * ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
22432 Call gst_riff_init() in plugin_init, to avoid getting errors from
22433 the debug system (unrelated changes to another plugin made this turn
22436 2006-11-10 19:20:21 +0000 Sergey Scobich <sergery.scobich@gmail.com>
22438 win32/common/libgsttag.def: Add missing symbol (#366492).
22439 Original commit message from CVS:
22440 Patch by: Sergey Scobich <sergery.scobich at gmail com>
22441 * win32/common/libgsttag.def:
22442 Add missing symbol (#366492).
22444 2006-11-10 00:52:55 +0000 Tim-Philipp Müller <tim@centricular.net>
22446 gst/playback/gststreamselector.c: Don't unref a NULL pad.
22447 Original commit message from CVS:
22448 * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
22449 Don't unref a NULL pad.
22451 2006-11-09 00:50:00 +0000 Christian Schaller <uraeus@gnome.org>
22453 ext/ogg/gstoggdemux.c: Implement first stab at reverse playback.
22454 Original commit message from CVS:
22455 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
22456 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_perform_seek),
22457 (gst_ogg_demux_handle_page), (gst_ogg_demux_chain),
22458 (gst_ogg_demux_loop_forward), (gst_ogg_demux_loop_reverse),
22459 (gst_ogg_demux_loop):
22460 Implement first stab at reverse playback.
22462 2006-11-07 07:22:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22464 gst-libs/gst/riff/riff-media.c: add h263/h264 variants to the caps, Fixes #363118
22465 Original commit message from CVS:
22466 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
22467 (gst_riff_create_video_template_caps):
22468 add h263/h264 variants to the caps, Fixes #363118
22470 2006-11-06 18:24:59 +0000 Tim-Philipp Müller <tim@centricular.net>
22472 gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
22473 Original commit message from CVS:
22474 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
22475 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
22476 Use g_strerror instead of strerror so we get UTF-8.
22478 2006-11-04 07:25:58 +0000 David Schleef <ds@schleef.org>
22480 ext/ogg/: Add/remove KW-DIRAC header here, since it is ogg-specific.
22481 Original commit message from CVS:
22482 * ext/ogg/gstoggdemux.c:
22483 * ext/ogg/gstoggmux.c:
22484 Add/remove KW-DIRAC header here, since it is ogg-specific.
22486 2006-11-03 15:44:31 +0000 Michael Smith <msmith@xiph.org>
22488 gst/typefind/gsttypefindfunctions.c: Recognise more mpeg4 elementary video streams.
22489 Original commit message from CVS:
22490 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
22491 Recognise more mpeg4 elementary video streams.
22493 2006-11-02 17:26:03 +0000 Edward Hervey <bilboed@bilboed.com>
22495 gst/typefind/gsttypefindfunctions.c: Lower the probability of mp3 typefinding functions if we don't find a valid mp3 ...
22496 Original commit message from CVS:
22497 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
22498 Lower the probability of mp3 typefinding functions if we don't find a
22499 valid mp3 header at the start of the file.
22502 2006-11-02 15:06:36 +0000 Wim Taymans <wim.taymans@gmail.com>
22504 ext/theora/: Document and partially implement an algorithm for doing reverse playback of theora video.
22505 Original commit message from CVS:
22506 * ext/theora/gsttheoradec.h:
22507 * ext/theora/theoradec.c: (gst_theora_dec_init),
22508 (theora_dec_sink_event), (theora_dec_chain_forward),
22509 (theora_dec_flush_decode), (theora_dec_chain_reverse),
22510 (theora_dec_chain):
22511 Document and partially implement an algorithm for doing reverse playback
22514 2006-11-02 14:18:45 +0000 Sergey Scobich <sergey.scobich@gmail.com>
22516 win32/: Misc. VS8 build fixes: fix syntax in config.h, add missing entries to libgsttag.def; add missing dependencies...
22517 Original commit message from CVS:
22518 Patch by: Sergey Scobich <sergey.scobich at gmail com>
22519 * win32/common/config.h:
22520 * win32/common/interfaces-enumtypes.c:
22521 * win32/common/libgsttag.def:
22522 * win32/vs8/gst-plugins-base.sln:
22523 * win32/vs8/libgstaudioresample.vcproj:
22524 * win32/vs8/libgstinterfaces.vcproj:
22525 * win32/vs8/libgstogg.vcproj:
22526 * win32/vs8/libgstriff.vcproj:
22527 * win32/vs8/libgsttag.vcproj:
22528 * win32/vs8/libgsttheora.vcproj:
22529 * win32/vs8/libgstvideoscale.vcproj:
22530 * win32/vs8/libgstvorbis.vcproj:
22531 Misc. VS8 build fixes: fix syntax in config.h, add missing entries
22532 to libgsttag.def; add missing dependencies for some vs8 projects;
22533 re-arrange placement of .def files in vs8 projects (#366334).
22535 2006-11-01 14:08:31 +0000 Tim-Philipp Müller <tim@centricular.net>
22537 ext/ogg/gstogg.c: Remove unused variable.
22538 Original commit message from CVS:
22539 * ext/ogg/gstogg.c:
22540 Remove unused variable.
22541 * ext/ogg/gstoggdemux.c:
22542 Fix Wim's surname in plugin description.
22544 2006-10-31 15:05:33 +0000 Wim Taymans <wim.taymans@gmail.com>
22546 gst-plugins-base.spec.in: spec new .h file. Fixes #368310.
22547 Original commit message from CVS:
22548 * gst-plugins-base.spec.in:
22549 spec new .h file. Fixes #368310.
22551 2006-10-31 14:19:07 +0000 Michael Smith <msmith@xiph.org>
22553 gst/tcp/gstmultifdsink.*: Make using the remove or clear signals threadsafe.
22554 Original commit message from CVS:
22555 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
22556 (gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
22557 (gst_multi_fd_sink_get_stats),
22558 (gst_multi_fd_sink_remove_client_link),
22559 (gst_multi_fd_sink_queue_buffer),
22560 (gst_multi_fd_sink_handle_clients):
22561 * gst/tcp/gstmultifdsink.h:
22562 Make using the remove or clear signals threadsafe.
22563 Make calling get-stats with an invalid fd not segfault.
22566 2006-10-31 10:49:19 +0000 Wim Taymans <wim.taymans@gmail.com>
22568 gst-libs/gst/rtp/: Fix and activate base audio payloader.
22569 Original commit message from CVS:
22570 * gst-libs/gst/rtp/Makefile.am:
22571 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
22572 (gst_base_rtp_audio_payload_init):
22573 Fix and activate base audio payloader.
22575 2006-10-28 17:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
22577 gst/typefind/gsttypefindfunctions.c: Add typefinder for QuickTime Image Files (see #366156).
22578 Original commit message from CVS:
22579 * gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
22581 Add typefinder for QuickTime Image Files (see #366156).
22583 2006-10-28 16:00:51 +0000 Tim-Philipp Müller <tim@centricular.net>
22585 gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
22586 Original commit message from CVS:
22587 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
22588 Another typo fix (#366212).
22590 2006-10-27 17:13:35 +0000 Wim Taymans <wim.taymans@gmail.com>
22592 gst/volume/gstvolume.c: Use stream time to synchronize volume property instead of rather random timestamps. This is n...
22593 Original commit message from CVS:
22594 * gst/volume/gstvolume.c: (volume_transform_ip):
22595 Use stream time to synchronize volume property instead of rather random
22596 timestamps. This is needed when gnonlin does its time shifting.
22598 2006-10-27 16:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
22601 I'm too lazy to comment this
22602 Original commit message from CVS:
22603 *** empty log message ***
22605 2006-10-27 16:45:30 +0000 Mark Nauwelaerts <manauw@skynet.be>
22607 ext/ogg/gstoggmux.c: Remove the pad from the element in release_pad.
22608 Original commit message from CVS:
22609 Patch by: Mark Nauwelaerts <manauw at skynet dot be>
22610 * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
22611 Remove the pad from the element in release_pad.
22613 2006-10-27 11:57:18 +0000 Tim-Philipp Müller <tim@centricular.net>
22615 sys/: Explicitly create our custom buffer classes at a thread-safe location as well, since g_type_class_ref() doesn't...
22616 Original commit message from CVS:
22617 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
22618 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
22619 Explicitly create our custom buffer classes at a thread-safe
22620 location as well, since g_type_class_ref() doesn't seem to be
22621 entirely thread-safe either (#365501; also see #349410).
22623 2006-10-26 10:49:00 +0000 Tim-Philipp Müller <tim@centricular.net>
22625 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...
22626 Original commit message from CVS:
22627 * gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
22628 (gst_riff_parse_info):
22629 If strings in INFO chunk are not UTF-8, do something similar to
22630 what we do for ID3v1 tags: check a number of environment variables
22631 (GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
22632 character sets to try, otherwise try the current locale and/or fall
22633 back on ISO-8859-1. Fixes #360552.
22635 2006-10-23 12:46:41 +0000 Tim-Philipp Müller <tim@centricular.net>
22637 gst/videotestsrc/: Add a bunch of exciting new checkers patterns.
22638 Original commit message from CVS:
22639 * gst/videotestsrc/gstvideotestsrc.c:
22640 (gst_video_test_src_pattern_get_type),
22641 (gst_video_test_src_set_pattern):
22642 * gst/videotestsrc/gstvideotestsrc.h:
22643 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_checkers1),
22644 (gst_video_test_src_checkers2), (gst_video_test_src_checkers4),
22645 (gst_video_test_src_checkers8):
22646 * gst/videotestsrc/videotestsrc.h:
22647 Add a bunch of exciting new checkers patterns.
22649 2006-10-23 12:06:44 +0000 Tim-Philipp Müller <tim@centricular.net>
22651 gst/subparse/: Add support for TMPlayer-type subtitles (#362845).
22652 Original commit message from CVS:
22653 * gst/subparse/Makefile.am:
22654 * gst/subparse/gstsubparse.c:
22655 (gst_sub_parse_data_format_autodetect),
22656 (gst_sub_parse_format_autodetect), (handle_buffer),
22657 (gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
22658 * gst/subparse/gstsubparse.h:
22659 * gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
22661 * gst/subparse/tmplayerparse.h:
22662 Add support for TMPlayer-type subtitles (#362845).
22663 * tests/check/elements/subparse.c: (test_tmplayer_do_test),
22664 (GST_START_TEST), (subparse_suite):
22665 Add some basic unit tests for the above.
22667 2006-10-23 11:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
22669 tests/check/elements/audiorate.c: More tests for audiorate: inject buffers to check behaviour when buffers overlap.
22670 Original commit message from CVS:
22671 * tests/check/elements/audiorate.c: (test_injector_base_init),
22672 (test_injector_class_init), (test_injector_chain),
22673 (test_injector_init), (probe_cb), (do_perfect_stream_test),
22674 (GST_START_TEST), (audiorate_suite):
22675 More tests for audiorate: inject buffers to check behaviour when
22678 2006-10-21 16:39:54 +0000 Tim-Philipp Müller <tim@centricular.net>
22680 tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...
22681 Original commit message from CVS:
22682 * tests/check/Makefile.am:
22683 * tests/check/elements/.cvsignore:
22684 * tests/check/elements/audiorate.c: (probe_cb), (got_buf),
22685 (do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
22686 Add some basic unit tests for audiorate. Disabled at the moment
22687 since it doesn't pass yet (see bug #363119).
22689 2006-10-20 17:02:19 +0000 Tim-Philipp Müller <tim@centricular.net>
22691 gst/subparse/gstsubparse.c: Add missing closing tags for markup and fix broken markup, otherwise pango won't render a...
22692 Original commit message from CVS:
22693 * gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
22694 (parse_subrip), (handle_buffer):
22695 Add missing closing tags for markup and fix broken markup,
22696 otherwise pango won't render anything (fixes #357531). Also,
22697 make sure the text we send out is always NUL-terminated
22698 (better safe than sorry etc.).
22699 * tests/check/elements/subparse.c: (test_srt_do_test),
22701 Some more tests for .srt incl. tests for the above stuff.
22703 2006-10-20 13:56:55 +0000 Stefan Kost <ensonic@users.sf.net>
22705 sys/: Try to redraw borders only when needed. Apparently this consumes resources on small devices... :-O (#363607)
22706 Original commit message from CVS:
22707 2006-10-20 Julien MOUTTE <julien@moutte.net>
22708 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
22709 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
22710 Patch by: Stefan Kost <ensonic@users.sf.net>
22711 Try to redraw borders only when needed. Apparently this consumes
22712 resources on small devices... :-O (#363607)
22714 2006-10-20 13:54:19 +0000 Michael Smith <msmith@xiph.org>
22716 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...
22717 Original commit message from CVS:
22718 * gst/tcp/gstmultifdsink.c:
22719 (gst_multi_fd_sink_client_queue_buffer):
22720 If caps change, then update the client's idea of the caps so that we
22721 don't end up re-sending streamheaders for every single buffer after
22724 2006-10-20 12:31:02 +0000 Michael Smith <msmith@xiph.org>
22726 ext/ogg/gstoggparse.c: Set caps on pushed buffers; fix up refcounting of caps objects.
22727 Original commit message from CVS:
22728 * ext/ogg/gstoggparse.c: (gst_ogg_parse_dispose),
22729 (gst_ogg_parse_append_header), (gst_ogg_parse_chain):
22730 Set caps on pushed buffers; fix up refcounting of caps objects.
22732 2006-10-19 14:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
22734 gst/typefind/gsttypefindfunctions.c: Typefind mmsh header data packet to application/x-mmsh (#362625).
22735 Original commit message from CVS:
22736 * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
22738 Typefind mmsh header data packet to application/x-mmsh (#362625).
22740 2006-10-19 09:17:48 +0000 Tim-Philipp Müller <tim@centricular.net>
22742 tests/check/: Add very simple unit test for subparse.
22743 Original commit message from CVS:
22744 * tests/check/Makefile.am:
22745 * tests/check/elements/.cvsignore:
22746 * tests/check/elements/subparse.c: (buffer_from_static_string),
22747 (setup_subparse), (teardown_subparse), (test_srt_do_test),
22748 (GST_START_TEST), (subparse_suite):
22749 Add very simple unit test for subparse.
22751 2006-10-19 09:00:21 +0000 Tim-Philipp Müller <tim@centricular.net>
22753 gst/subparse/gstsubparse.c: Strip trailing newlines from subtitle text output.
22754 Original commit message from CVS:
22755 * gst/subparse/gstsubparse.c: (strip_trailing_newlines),
22757 Strip trailing newlines from subtitle text output.
22759 2006-10-18 18:40:12 +0000 Tim-Philipp Müller <tim@centricular.net>
22761 gst/subparse/gstsubparse.c: Fix memleak; clear subparse->textbuf n state change function.
22762 Original commit message from CVS:
22763 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
22764 (gst_sub_parse_change_state):
22765 Fix memleak; clear subparse->textbuf n state change function.
22767 2006-10-18 15:13:59 +0000 Tim-Philipp Müller <tim@centricular.net>
22769 gst/subparse/gstsubparse.c: Don't require subrip (.srt) files to start with a chunk number of 1.
22770 Original commit message from CVS:
22771 * gst/subparse/gstsubparse.c:
22772 (gst_sub_parse_data_format_autodetect):
22773 Don't require subrip (.srt) files to start with a chunk number of 1.
22775 2006-10-18 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
22777 gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
22778 Original commit message from CVS:
22779 * gst-libs/gst/audio/gstbaseaudiosink.c:
22780 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
22781 * gst-libs/gst/audio/gstbaseaudiosink.h:
22782 Extract rate from the NEWSEGMENT event.
22783 Use commit_full to also take rate adjustment into account when writing
22784 samples to the ringbuffer.
22785 * gst-libs/gst/audio/gstringbuffer.c:
22786 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
22787 (gst_ring_buffer_read):
22788 * gst-libs/gst/audio/gstringbuffer.h:
22789 Added _commit_full() to also take rate into account.
22790 Use simple interpolation algorithm to resample audio.
22791 API: gst_ring_buffer_commit_full()
22792 * tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
22793 * tests/examples/seek/seek.c: (segment_done):
22794 Don't try to seek with 0.0 rate, just pause instead.
22795 Remove bogus debug line.
22797 2006-10-18 12:57:54 +0000 Tim-Philipp Müller <tim@centricular.net>
22799 gst/playback/gstplaybasebin.c: Catch async errors when starting up the subtitle bin, so we can stop waiting and conti...
22800 Original commit message from CVS:
22801 * gst/playback/gstplaybasebin.c: (subbin_startup_sync_msg),
22803 Catch async errors when starting up the subtitle bin, so we can
22804 stop waiting and continue with the main film instead of hanging
22805 forever. Fixes #339366.
22806 * tests/check/elements/playbin.c: (playbin_suite):
22807 Enable unit test for the above.
22809 2006-10-18 09:53:03 +0000 Tim-Philipp Müller <tim@centricular.net>
22811 tests/check/: Some small and basic unit tests for playbin; not very useful yet, but at least a start.
22812 Original commit message from CVS:
22813 * tests/check/Makefile.am:
22814 * tests/check/elements/.cvsignore:
22815 * tests/check/elements/playbin.c: (GST_START_TEST),
22816 (gst_red_video_src_uri_get_type),
22817 (gst_red_video_src_uri_get_protocols),
22818 (gst_red_video_src_uri_get_uri), (gst_red_video_src_uri_set_uri),
22819 (gst_red_video_src_uri_handler_init),
22820 (gst_red_video_src_init_type), (gst_red_video_src_base_init),
22821 (gst_red_video_src_create), (gst_red_video_src_class_init),
22822 (gst_red_video_src_init), (plugin_init), (playbin_suite):
22823 Some small and basic unit tests for playbin; not very useful yet,
22824 but at least a start.
22826 2006-10-18 09:46:35 +0000 Tim-Philipp Müller <tim@centricular.net>
22828 gst/playback/gstplaybin.c: The old pad activation spiel.
22829 Original commit message from CVS:
22830 * gst/playback/gstplaybin.c: (setup_sinks):
22831 The old pad activation spiel.
22833 2006-10-18 09:31:49 +0000 Tim-Philipp Müller <tim@centricular.net>
22835 gst/playback/gstplaybasebin.c: Don't hang forever if the subbin already fails to start up in the state change to PAUS...
22836 Original commit message from CVS:
22837 * gst/playback/gstplaybasebin.c: (setup_source):
22838 Don't hang forever if the subbin already fails to start up in
22839 the state change to PAUSED (#339366).
22841 2006-10-17 17:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
22843 gst-libs/gst/interfaces/tuner.c: Fix some function guards, add some more function guards.
22844 Original commit message from CVS:
22845 * gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
22846 (gst_tuner_set_channel), (gst_tuner_get_channel),
22847 (gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
22848 (gst_tuner_set_frequency), (gst_tuner_get_frequency),
22849 (gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
22850 (gst_tuner_find_channel_by_name):
22851 Fix some function guards, add some more function guards.
22853 2006-10-17 11:34:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22855 gst/playback/gstdecodebin.c: Don't return a pad from get_our_ghost_pad unless it is actually the one we want.
22856 Original commit message from CVS:
22857 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
22858 (remove_element_chain):
22859 Don't return a pad from get_our_ghost_pad unless it is actually the
22861 Change a cast in remove_element_chain slightly.
22863 2006-10-13 15:20:29 +0000 Julien Moutte <julien@moutte.net>
22865 tests/examples/seek/seek.c: Segment seeking needs to use the rate and set stop to -1.
22866 Original commit message from CVS:
22867 2006-10-13 Julien MOUTTE <julien@moutte.net>
22868 * tests/examples/seek/seek.c: (do_seek), (start_seek),
22869 (rate_spinbutton_changed_cb), (segment_done),
22870 (msg_state_changed):
22871 Segment seeking needs to use the rate and set stop to -1.
22873 2006-10-13 14:15:42 +0000 Ville Syrjala <ville.syrjala@movial.fi>
22875 gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
22876 Original commit message from CVS:
22877 * gst-libs/gst/audio/gstbaseaudiosink.c:
22878 (gst_base_audio_sink_setcaps):
22879 Don't crash when ringbuffer is not yet created.
22880 Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
22882 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
22883 * gst/playback/gststreamselector.c:
22884 (gst_stream_selector_request_new_pad):
22885 Activate pads befre adding them to running elements.
22887 2006-10-13 11:25:10 +0000 Julien Moutte <julien@moutte.net>
22889 tests/examples/seek/seek.c: Stop the scale updater when we start grabing the slider. Don't wait for the pipeline to b...
22890 Original commit message from CVS:
22891 2006-10-13 Julien MOUTTE <julien@moutte.net>
22892 * tests/examples/seek/seek.c: (do_seek), (start_seek),
22893 (rate_spinbutton_changed_cb), (msg_state_changed): Stop the
22895 updater when we start grabing the slider. Don't wait for the
22896 pipeline to be PAUSED.
22898 2006-10-13 08:57:52 +0000 Tim-Philipp Müller <tim@centricular.net>
22900 gst-libs/gst/interfaces/mixer.c: Guard mixer interface functions against bogus arguments.
22901 Original commit message from CVS:
22902 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
22903 (gst_mixer_set_volume), (gst_mixer_get_volume),
22904 (gst_mixer_set_mute), (gst_mixer_set_option),
22905 (gst_mixer_get_option), (gst_mixer_mute_toggled),
22906 (gst_mixer_record_toggled), (gst_mixer_volume_changed),
22907 (gst_mixer_option_changed):
22908 Guard mixer interface functions against bogus arguments.
22910 2006-10-12 19:39:07 +0000 Julien Moutte <julien@moutte.net>
22912 tests/examples/seek/seek.c: Use state-changed messages to trigger start/stop of scale update timer. Indeed the scale ...
22913 Original commit message from CVS:
22914 2006-10-12 Julien MOUTTE <julien@moutte.net>
22915 * tests/examples/seek/seek.c: (do_seek), (start_seek),
22917 (play_cb), (pause_cb), (stop_cb),
22918 (rate_spinbutton_changed_cb),
22919 (msg_state_changed), (main): Use state-changed messages to
22921 start/stop of scale update timer. Indeed the scale slider was
22922 jumping here and there because the update timer was activated
22923 before seek completed. This fixes instant applying of rate
22925 by pressing the spinbutton like a crazy man !
22927 2006-10-12 19:09:06 +0000 Sebastien Cote <sebas642@yahoo.ca>
22929 gst-libs/gst/rtp/gstbasertppayload.c: Fix two small memory leaks (#361456).
22930 Original commit message from CVS:
22931 Patch by: Sebastien Cote <sebas642 at yahoo.ca>
22932 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
22933 (gst_basertppayload_finalize):
22934 Fix two small memory leaks (#361456).
22936 2006-10-10 18:56:01 +0000 Julien Moutte <julien@moutte.net>
22938 tests/examples/seek/seek.c: When changing spinbutton we try to change the rate on the fly.
22939 Original commit message from CVS:
22940 2006-10-10 Julien MOUTTE <julien@moutte.net>
22941 * tests/examples/seek/seek.c: (do_seek),
22942 (rate_spinbutton_changed_cb): When changing spinbutton we try
22943 to change the rate on the fly.
22945 2006-10-10 16:50:06 +0000 Wim Taymans <wim.taymans@gmail.com>
22947 gst-libs/gst/riff/: Add WMS caps.
22948 Original commit message from CVS:
22949 * gst-libs/gst/riff/riff-ids.h:
22950 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
22951 (gst_riff_create_audio_template_caps):
22954 2006-10-10 12:49:03 +0000 Josep Torre Valles <josep@fluendo.com>
22956 ext/gnomevfs/: Fix URI interface implementation return type.
22957 Original commit message from CVS:
22958 2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
22959 Patch by: Josep Torre Valles <josep@fluendo.com>
22960 * ext/gnomevfs/gstgnomevfssink.c:
22961 * ext/gnomevfs/gstgnomevfssrc.c:
22962 Fix URI interface implementation return type.
22963 * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
22964 Fix what looks like a copy/paste issue when assigning values.
22965 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
22966 (gst_audio_filter_template_get_type):
22967 Cast to prevent Forte warnings.
22968 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
22969 Fix URI interface implementation return type.
22970 gst_pad_query_position requires a signed integer pointer as
22971 3rd parameter, GstClockTime is unsigned.
22972 * gst/audioconvert/audioconvert.c:
22973 Fix integer overflow when treated as signed.
22974 * gst/audioresample/resample.c: (resample_add_input_data):
22975 Cast to prevent warnings on Forte.
22976 * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
22977 Fix integer overflow when treated as signed.
22978 * gst/ffmpegcolorspace/imgconvert_template.h:
22979 Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
22980 * gst/playback/gstdecodebin.c: (queue_filled_cb),
22981 (cleanup_decodebin):
22982 Who initialises a guint to -1!
22983 Cast function pointers to prevent warnings on Forte.
22984 * gst/playback/gstplaybasebin.c: (queue_deadlock_check),
22985 (queue_threshold_reached):
22986 Cast function pointers correctly to prevent warnings on Forte.
22987 * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
22988 Cast function pointers correctly to prevent warnings on Forte.
22989 * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
22990 Obvious change to unsigned, 0xEF > max signed char.
22991 * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
22992 GstClockTime is unsigned, initialise correctly.
22993 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
22994 Cast so pointer arithemetic doesn't cause warnings on Forte.
22995 * gst/videorate/gstvideorate.c:
22996 Use correct return value.
22997 * tests/examples/seek/scrubby.c:
22998 GstClockTime is unsigned, initialise correctly.
23000 2006-10-10 11:20:03 +0000 Ferenc Gerlits <fgerlits@gmail.com>
23002 gst/typefind/gsttypefindfunctions.c: Recognise XML files and XML-like files shorter than 256 bytes as well (fixes #35...
23003 Original commit message from CVS:
23004 Patch by: Ferenc Gerlits <fgerlits at gmail com>
23005 * gst/typefind/gsttypefindfunctions.c:
23006 Recognise XML files and XML-like files shorter than 256 bytes as
23007 well (fixes #359237).
23009 2006-10-09 15:01:30 +0000 Edgard Lima <edgard.lima@indt.org.br>
23013 * gst/typefind/gsttypefindfunctions.c:
23014 Added typefind functions to video/x-nuv media.
23015 Original commit message from CVS:
23016 Added typefind functions to video/x-nuv media.
23018 2006-10-08 16:59:31 +0000 Tim-Philipp Müller <tim@centricular.net>
23020 gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
23021 Original commit message from CVS:
23022 * gst-libs/gst/interfaces/xoverlay.c:
23023 (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
23024 Some more guards against invalid input.
23026 2006-10-07 18:35:39 +0000 Julien Moutte <julien@moutte.net>
23028 ext/pango/gsttextoverlay.c: Useless goto.
23029 Original commit message from CVS:
23030 2006-10-07 Julien MOUTTE <julien@moutte.net>
23031 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
23033 * tests/examples/seek/seek.c: (do_seek),
23034 (rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
23035 seek example to experiment with rates != 1.0 (reverse playback
23038 2006-10-06 19:20:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23040 gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
23041 Original commit message from CVS:
23042 * gst-libs/gst/interfaces/xoverlay.c:
23043 Unref message in doc-example (spotted by Robert McQueen)
23045 2006-10-06 17:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
23047 gst/typefind/gsttypefindfunctions.c: printf fix.
23048 Original commit message from CVS:
23049 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
23050 (mpeg1_parse_header), (mpeg1_sys_type_find):
23053 2006-10-06 14:37:43 +0000 Wim Taymans <wim.taymans@gmail.com>
23055 gst/playback/: Activate dynamic pads before adding them to the element.
23056 Original commit message from CVS:
23057 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
23059 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
23060 Activate dynamic pads before adding them to the element.
23062 2006-10-06 14:04:53 +0000 Michael Smith <msmith@xiph.org>
23064 gst-libs/gst/floatcast/floatcast.h: Fix obviously-bogus macros; use the correct types.
23065 Original commit message from CVS:
23066 * gst-libs/gst/floatcast/floatcast.h:
23067 Fix obviously-bogus macros; use the correct types.
23069 2006-10-06 13:34:46 +0000 Wim Taymans <wim.taymans@gmail.com>
23071 gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
23072 Original commit message from CVS:
23073 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23074 (gst_base_rtp_depayload_change_state):
23075 Also call parent state change function to activate pads.
23076 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
23077 (mpeg1_parse_header), (mpeg1_sys_type_find):
23078 Add some more debug info in mpeg typefinding.
23080 2006-10-06 12:57:10 +0000 Michael Smith <msmith@xiph.org>
23082 ext/theora/theoradec.c: Zero byte theora packets are valid and well-defined; don't warn on them.
23083 Original commit message from CVS:
23084 * ext/theora/theoradec.c: (theora_dec_chain):
23085 Zero byte theora packets are valid and well-defined; don't warn on
23088 2006-10-06 10:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23090 gst/tcp/gstmultifdsink.c: API: add dropped_buffers to the get-stats GValueArray
23091 Original commit message from CVS:
23092 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
23093 (gst_multi_fd_sink_get_stats), (find_limits),
23094 (gst_multi_fd_sink_queue_buffer):
23095 API: add dropped_buffers to the get-stats GValueArray
23097 2006-10-05 15:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
23099 Printf format fixes.
23100 Original commit message from CVS:
23101 * ext/alsa/gstalsadeviceprobe.c:
23102 (gst_alsa_device_property_probe_get_values):
23103 * ext/alsa/gstalsasink.c: (set_hwparams):
23104 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
23105 (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
23106 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
23107 (gst_ogg_mux_process_best_pad):
23108 * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
23109 (gst_ogg_parse_chain):
23110 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
23111 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
23112 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
23113 (gst_vorbis_enc_buffer_check_discontinuous):
23114 * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
23115 * gst-libs/gst/audio/gstbaseaudiosink.c:
23116 (gst_base_audio_sink_render):
23117 * gst-libs/gst/cdda/gstcddabasesrc.c:
23118 (gst_cdda_base_src_handle_track_seek):
23119 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23120 (gst_base_rtp_depayload_push_full):
23121 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
23122 * gst/audioresample/resample.c: (resample_input_pushthrough):
23123 * gst/playback/gstplaybasebin.c: (queue_out_of_data):
23124 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
23125 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
23126 (wavpack_type_find):
23127 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
23128 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
23129 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
23130 * tests/check/elements/volume.c: (GST_START_TEST):
23131 Printf format fixes.
23133 2006-10-04 13:18:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23135 gst/tcp/gsttcp.c: Fix a simple mistake (see the docs)
23136 Original commit message from CVS:
23137 * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
23138 Fix a simple mistake (see the docs)
23141 2006-10-04 13:15:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23143 * win32/common/config.h:
23145 Original commit message from CVS:
23148 2006-10-03 12:11:45 +0000 Tim-Philipp Müller <tim@centricular.net>
23150 docs/plugins/: Add vorbistag element to docs; update version numbers to 0.10.10.1.
23151 Original commit message from CVS:
23152 * docs/plugins/Makefile.am:
23153 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
23154 * docs/plugins/gst-plugins-base-plugins-sections.txt:
23155 * docs/plugins/gst-plugins-base-plugins.args:
23156 * docs/plugins/gst-plugins-base-plugins.hierarchy:
23157 * docs/plugins/inspect/plugin-adder.xml:
23158 * docs/plugins/inspect/plugin-alsa.xml:
23159 * docs/plugins/inspect/plugin-audioconvert.xml:
23160 * docs/plugins/inspect/plugin-audiorate.xml:
23161 * docs/plugins/inspect/plugin-audioresample.xml:
23162 * docs/plugins/inspect/plugin-audiotestsrc.xml:
23163 * docs/plugins/inspect/plugin-cdparanoia.xml:
23164 * docs/plugins/inspect/plugin-decodebin.xml:
23165 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
23166 * docs/plugins/inspect/plugin-gdp.xml:
23167 * docs/plugins/inspect/plugin-gnomevfs.xml:
23168 * docs/plugins/inspect/plugin-libvisual.xml:
23169 * docs/plugins/inspect/plugin-ogg.xml:
23170 * docs/plugins/inspect/plugin-pango.xml:
23171 * docs/plugins/inspect/plugin-playbin.xml:
23172 * docs/plugins/inspect/plugin-subparse.xml:
23173 * docs/plugins/inspect/plugin-tcp.xml:
23174 * docs/plugins/inspect/plugin-theora.xml:
23175 * docs/plugins/inspect/plugin-typefindfunctions.xml:
23176 * docs/plugins/inspect/plugin-video4linux.xml:
23177 * docs/plugins/inspect/plugin-videorate.xml:
23178 * docs/plugins/inspect/plugin-videoscale.xml:
23179 * docs/plugins/inspect/plugin-videotestsrc.xml:
23180 * docs/plugins/inspect/plugin-volume.xml:
23181 * docs/plugins/inspect/plugin-vorbis.xml:
23182 * docs/plugins/inspect/plugin-ximagesink.xml:
23183 * docs/plugins/inspect/plugin-xvimagesink.xml:
23184 Add vorbistag element to docs; update version numbers to 0.10.10.1.
23186 2006-10-03 11:51:48 +0000 James Doc Livingston <doclivingston@gmail.com>
23188 ext/vorbis/: Add new vorbistag element which derives from vorbisparse and is essentially the same as well, only that ...
23189 Original commit message from CVS:
23190 Patch by: James "Doc" Livingston <doclivingston at gmail com>
23191 * ext/vorbis/Makefile.am:
23192 * ext/vorbis/vorbis.c: (plugin_init):
23193 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_class_init),
23194 (vorbis_parse_parse_packet), (vorbis_parse_chain):
23195 * ext/vorbis/vorbisparse.h:
23196 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_base_init),
23197 (gst_vorbis_tag_class_init), (gst_vorbis_tag_init),
23198 (gst_vorbis_tag_parse_packet):
23199 * ext/vorbis/vorbistag.h:
23200 Add new vorbistag element which derives from vorbisparse
23201 and is essentially the same as well, only that it implements
23202 the GstTagSetter interface and can modify the stream's
23203 vorbiscomment on the fly (#335635).
23204 * tests/check/Makefile.am:
23205 * tests/check/elements/.cvsignore:
23206 * tests/check/elements/vorbistag.c: (setup_vorbistag),
23207 (cleanup_vorbistag), (buffer_probe), (start_pipeline),
23208 (get_buffer), (stop_pipeline), (_create_codebook_header_buffer),
23209 (_create_audio_buffer), (GST_START_TEST), (vorbistag_suite):
23210 Add unit test for new vorbistag element.
23212 2006-10-03 10:36:38 +0000 Tim-Philipp Müller <tim@centricular.net>
23214 ext/vorbis/vorbisparse.c: Set BOS flag in packet structure to fix 'jump depends on unitialized value' errors in valgr...
23215 Original commit message from CVS:
23216 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init),
23217 (vorbis_parse_push_headers), (vorbis_parse_chain):
23218 Set BOS flag in packet structure to fix 'jump depends
23219 on unitialized value' errors in valgrind; various minor
23222 2006-09-30 15:30:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23224 gst/playback/gstdecodebin.c: Fix typo in a debug statement.
23225 Original commit message from CVS:
23226 * gst/playback/gstdecodebin.c: (close_pad_link):
23227 Fix typo in a debug statement.
23228 * gst/playback/gstplaybasebin.c: (probe_triggered),
23229 (new_decoded_pad_full), (new_decoded_pad), (subs_new_decoded_pad),
23230 (gen_source_element), (source_new_pad), (analyse_source),
23232 When handling no_more_pads in new_decoded_pad, make sure to treat
23233 subtitle pads correctly. Fixes playback with subtitle files.
23234 Move a recurring message to LOG level.
23235 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
23236 The maximum value for the Xv colorkey on this Radeon is 0xFFFFFFFF,
23237 which ends up as -1 when cast to an int. Make the logic handle the
23238 max value as an unsigned mask and only change the colorkey when it's
23239 a value we recognise.
23241 2006-09-30 00:14:20 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
23243 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
23244 Original commit message from CVS:
23245 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23246 Removed empty * between paragraphs
23248 2006-09-29 23:50:53 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
23250 gst-libs/gst/rtp/: Moved some documentation into .c file
23251 Original commit message from CVS:
23252 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23253 * gst-libs/gst/rtp/README:
23254 Moved some documentation into .c file
23256 2006-09-29 17:35:01 +0000 Wim Taymans <wim.taymans@gmail.com>
23258 gst/playback/gstdecodebin.c: Fix compilation.
23259 Original commit message from CVS:
23260 * gst/playback/gstdecodebin.c: (no_more_pads):
23263 2006-09-29 16:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
23265 gst/playback/gstdecodebin.c: Remove g_print
23266 Original commit message from CVS:
23267 * gst/playback/gstdecodebin.c: (new_caps):
23269 * gst/playback/gstplaybin.c:
23272 2006-09-29 15:16:32 +0000 Tim-Philipp Müller <tim@centricular.net>
23274 tests/check/Makefile.am: Re-enable cddabasesrc test to see if it works again now.
23275 Original commit message from CVS:
23276 * tests/check/Makefile.am:
23277 Re-enable cddabasesrc test to see if it works again
23280 2006-09-29 13:46:45 +0000 Wim Taymans <wim.taymans@gmail.com>
23282 gst/playback/gstplaybasebin.c: Handle invalid URIs a bit more gracefully.
23283 Original commit message from CVS:
23284 * gst/playback/gstplaybasebin.c: (setup_subtitle),
23285 (gen_source_element):
23286 Handle invalid URIs a bit more gracefully.
23288 2006-09-29 12:54:28 +0000 Tim-Philipp Müller <tim@centricular.net>
23290 tests/check/pipelines/oggmux.c: Remove obsolete comment.
23291 Original commit message from CVS:
23292 * tests/check/pipelines/oggmux.c:
23293 Remove obsolete comment.
23295 2006-09-29 10:43:05 +0000 James Doc Livingston <doclivingston@gmail.com>
23297 ext/ogg/gstoggmux.c: Commit patch from James "Doc" Livingston, adds proper EOS handling in oggmux. GStreamer can, for...
23298 Original commit message from CVS:
23299 * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
23300 (gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
23301 (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
23302 (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
23303 (gst_ogg_mux_collected):
23304 Commit patch from James "Doc" Livingston, adds proper EOS handling
23305 in oggmux. GStreamer can, for the first time ever, create a valid
23307 * tests/check/pipelines/oggmux.c: (check_chain_final_state),
23309 Reenable tests now that they pass.
23311 2006-09-29 08:20:03 +0000 Wim Taymans <wim.taymans@gmail.com>
23313 gst/tcp/gstmultifdsink.c: Stop reading commands when EOF (we read 0) as well.
23314 Original commit message from CVS:
23315 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
23316 Stop reading commands when EOF (we read 0) as well.
23318 2006-09-28 15:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
23320 gst/playback/gstdecodebin.c: Implement delayed caps linking needed for element with a lot of different caps on the sr...
23321 Original commit message from CVS:
23322 * gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
23323 (close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
23324 (find_dynamic), (unlinked), (close_link):
23325 Implement delayed caps linking needed for element with a lot of
23326 different caps on the src pads that get fixed at runtime.
23327 Improve management of dynamic elements.
23328 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
23329 (group_destroy), (group_commit), (check_queue), (queue_overrun),
23330 (gen_preroll_element), (remove_groups), (unknown_type),
23331 (add_element_stream), (no_more_pads_full), (no_more_pads),
23332 (sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
23333 (new_decoded_pad), (setup_subtitle), (array_has_value),
23334 (gen_source_element), (source_new_pad), (has_all_raw_caps),
23335 (analyse_source), (remove_decoders), (make_decoder),
23336 (remove_source), (setup_source), (finish_source), (prepare_output),
23337 (gst_play_base_bin_change_state):
23338 * gst/playback/gstplaybasebin.h:
23339 Use more _CAST instead of full type checking casts.
23340 Small cleanups, plug some leaks.
23341 Handle dynamic sources.
23342 Add some helper functions to create lists of strings used for
23343 blacklisting and other stuff.
23344 Refactor some code dealing with analysing the source.
23345 Re-enable sources without pads (like cd:// or other selfcontained
23348 2006-09-28 15:08:15 +0000 Wim Taymans <wim.taymans@gmail.com>
23350 gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
23351 Original commit message from CVS:
23352 * gst-libs/gst/audio/gstbaseaudiosink.c:
23353 (gst_base_audio_sink_render):
23354 When we have a timestamp, we can still perform clipping.
23355 When we have no clock, we must play the sample ASAP.
23357 2006-09-28 11:46:26 +0000 Wim Taymans <wim.taymans@gmail.com>
23359 gst/audiorate/gstaudiorate.c: Set caps on outgoing buffers.
23360 Original commit message from CVS:
23361 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
23362 Set caps on outgoing buffers.
23363 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
23364 (gst_video_rate_event), (gst_video_rate_chain):
23365 * gst/videorate/gstvideorate.h:
23366 Fix videorate some more. Fixes #357977
23368 2006-09-28 11:34:05 +0000 Tim-Philipp Müller <tim@centricular.net>
23370 tests/check/elements/adder.c: Don't set timeout to 6 seconds when we're running in valgrind ... (and how is 6 seconds...
23371 Original commit message from CVS:
23372 * tests/check/elements/adder.c: (adder_suite):
23373 Don't set timeout to 6 seconds when we're running
23374 in valgrind ... (and how is 6 seconds longer than
23375 the default anyway?)
23377 2006-09-28 10:49:56 +0000 Wim Taymans <wim.taymans@gmail.com>
23379 gst/audiorate/gstaudiorate.c: Keep sink and src segment to keep track of time and support more input formats.
23380 Original commit message from CVS:
23381 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
23382 (gst_audio_rate_sink_event), (gst_audio_rate_convert),
23383 (gst_audio_rate_convert_segments), (gst_audio_rate_chain):
23384 Keep sink and src segment to keep track of time and support more
23386 Fix bogus next_offset and run_time calculation, don't understand how
23387 this could have worked before. Fixes #357976.
23388 Remove some unneeded vars.
23390 2006-09-28 09:41:20 +0000 Tim-Philipp Müller <tim@centricular.net>
23392 gst/playback/gstplaybin.c: Only remove visualisation from visbin if there is a visbin (or: don't throw warnings when ...
23393 Original commit message from CVS:
23394 * gst/playback/gstplaybin.c: (remove_sinks):
23395 Only remove visualisation from visbin if there is a visbin (or:
23396 don't throw warnings when closing totem without playing a file).
23398 2006-09-27 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
23400 gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
23401 Original commit message from CVS:
23402 * gst-libs/gst/audio/gstbaseaudiosink.c:
23403 (gst_base_audio_sink_render):
23404 Add some more info in a WARNING.
23405 * gst-libs/gst/audio/gstbaseaudiosrc.c:
23406 (gst_base_audio_src_create):
23407 Handle PAUSE in create function, use new -core addition to
23408 wait for playing. Fixes pausing and resuming capture from an
23410 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
23411 (gst_ring_buffer_read):
23412 Constify some more.
23413 Caller supports interrupted reads now.
23415 2006-09-27 13:29:49 +0000 Christian Schaller <uraeus@gnome.org>
23417 * gst-plugins-base.spec.in:
23418 add new header file to spec
23419 Original commit message from CVS:
23420 add new header file to spec
23422 2006-09-27 12:55:45 +0000 Tim-Philipp Müller <tim@centricular.net>
23424 tests/check/Makefile.am: Another attempt to make the gen64 buildbot happy.
23425 Original commit message from CVS:
23426 * tests/check/Makefile.am:
23427 Another attempt to make the gen64 buildbot happy.
23429 2006-09-27 11:58:17 +0000 Jonathan Matthew <jonathan@kaolin.wh9.net>
23431 ext/libvisual/visual.c: Libvisual plugin was not passing audio data to libvisual 0.4.0 correctly. Fixes #357800
23432 Original commit message from CVS:
23433 Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
23434 * ext/libvisual/visual.c: (gst_visual_clear_actors),
23435 (gst_visual_chain), (gst_visual_change_state):
23436 Libvisual plugin was not passing audio data to libvisual 0.4.0
23437 correctly. Fixes #357800
23439 2006-09-27 11:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
23441 tests/check/pipelines/simple-launch-lines.c: Add timeout to _get_state() so we see which pipeline it is that causes t...
23442 Original commit message from CVS:
23443 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
23444 Add timeout to _get_state() so we see which pipeline it is
23445 that causes trouble on the gen64 build bot.
23447 2006-09-27 11:06:54 +0000 Wim Taymans <wim.taymans@gmail.com>
23449 gst-libs/gst/rtp/gstbasertpdepayload.c: the source pad always uses fixed caps.
23450 Original commit message from CVS:
23451 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23452 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
23453 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
23454 (gst_base_rtp_depayload_set_gst_timestamp):
23455 the source pad always uses fixed caps.
23457 2006-09-27 11:05:08 +0000 Wim Taymans <wim.taymans@gmail.com>
23459 Added docs for the audio libs.
23460 Original commit message from CVS:
23461 * docs/libs/gst-plugins-base-libs-docs.sgml:
23462 * docs/libs/gst-plugins-base-libs-sections.txt:
23463 * gst-libs/gst/audio/gstaudioclock.c:
23464 * gst-libs/gst/audio/gstaudioclock.h:
23465 * gst-libs/gst/audio/gstaudiosink.c:
23466 * gst-libs/gst/audio/gstaudiosink.h:
23467 * gst-libs/gst/audio/gstaudiosrc.c:
23468 * gst-libs/gst/audio/gstbaseaudiosink.c:
23469 (gst_base_audio_sink_render):
23470 * gst-libs/gst/audio/gstbaseaudiosink.h:
23471 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
23472 * gst-libs/gst/audio/gstbaseaudiosrc.h:
23473 * gst-libs/gst/audio/gstringbuffer.h:
23474 Added docs for the audio libs.
23476 2006-09-27 10:59:24 +0000 Tim-Philipp Müller <tim@centricular.net>
23478 tests/check/Makefile.am: Temporarily disable test that fails on the bots for unknown reasons.
23479 Original commit message from CVS:
23480 * tests/check/Makefile.am:
23481 Temporarily disable test that fails on the bots for unknown reasons.
23483 2006-09-27 00:13:29 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
23485 gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
23486 Original commit message from CVS:
23487 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23488 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
23489 Moved AudioCodecType into priv
23490 Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
23492 2006-09-25 15:47:25 +0000 Wim Taymans <wim.taymans@gmail.com>
23494 gst/playback/gstdecodebin.c: Cleanups and small leak fixes.
23495 Original commit message from CVS:
23496 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
23497 (add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
23498 (is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
23500 Cleanups and small leak fixes.
23501 Added Depayloaders to valid list of autopluggable elements.
23503 2006-09-25 13:24:59 +0000 Wim Taymans <wim.taymans@gmail.com>
23505 gst/playback/gstplaybin.c: Detect NO_PREROLL state change returns and disable clock distribution to the sinks so that...
23506 Original commit message from CVS:
23507 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
23508 (gst_play_bin_vis_blocked), (gst_play_bin_set_property),
23509 (gen_video_element), (gen_text_element), (gen_audio_element),
23510 (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
23511 (gst_play_bin_set_clock_func), (gst_play_bin_change_state):
23512 Detect NO_PREROLL state change returns and disable clock distribution to
23513 the sinks so that sync is disabled.
23514 Avoid some type checking and do simple casts instead.
23515 Small cleanups, fix some FIXMEs.
23516 Be more robust when linking user specified elements, catch an report
23517 errors. Fixes #357404.
23518 Fix some leaks in the error paths.
23520 2006-09-25 12:55:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23523 ChangeLog surgery for missing bug-number
23524 Original commit message from CVS:
23525 ChangeLog surgery for missing bug-number
23527 2006-09-25 11:28:15 +0000 Peter Kjellerstedt <pkj@axis.com>
23529 gst/playback/test.c: Fix compilation with uClibc and -Werror (#357591).
23530 Original commit message from CVS:
23531 Patch by: Peter Kjellerstedt <pkj at axis com>
23532 * gst/playback/test.c:
23533 Fix compilation with uClibc and -Werror (#357591).
23535 2006-09-25 10:21:31 +0000 Tim-Philipp Müller <tim@centricular.net>
23537 gst-libs/gst/tag/gstvorbistag.c: Parse dates that are followed by a time as well (#357532).
23538 Original commit message from CVS:
23539 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
23540 Parse dates that are followed by a time as well (#357532).
23541 * tests/check/libs/tag.c: (test_vorbis_tags):
23542 Add unit test for this.
23544 2006-09-23 15:24:55 +0000 Tim-Philipp Müller <tim@centricular.net>
23546 gst/: A few array const-ifications.
23547 Original commit message from CVS:
23548 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
23549 (gst_audio_convert_transform_caps):
23550 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
23551 * gst/videotestsrc/videotestsrc.h:
23552 A few array const-ifications.
23554 2006-09-23 15:02:51 +0000 Tim-Philipp Müller <tim@centricular.net>
23556 tests/check/Makefile.am: See if this makes the build bots happy.
23557 Original commit message from CVS:
23558 * tests/check/Makefile.am:
23559 See if this makes the build bots happy.
23560 * tests/check/libs/cddabasesrc.c:
23563 2006-09-23 14:30:53 +0000 Young-Ho Cha <ganadist@chollian.net>
23565 gst/subparse/samiparse.c: More case-insensitivity for certain tags; recognise entities with decimal codes as special ...
23566 Original commit message from CVS:
23567 Patch by: Young-Ho Cha <ganadist at chollian dot net>
23568 * gst/subparse/samiparse.c: (handle_start_font),
23569 (fix_invalid_entities):
23570 More case-insensitivity for certain tags; recognise entities with
23571 decimal codes as special entities as well (#357330).
23573 2006-09-23 13:32:07 +0000 Tim-Philipp Müller <tim@centricular.net>
23575 gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
23576 Original commit message from CVS:
23577 * gst-libs/gst/Makefile.am:
23578 Need to build tag directory before cdda.
23580 2006-09-23 13:21:07 +0000 Tim-Philipp Müller <tim@centricular.net>
23582 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
23583 Original commit message from CVS:
23584 * docs/libs/gst-plugins-base-libs-sections.txt:
23585 * gst-libs/gst/cdda/Makefile.am:
23586 * gst-libs/gst/cdda/gstcddabasesrc.c:
23587 (gst_cdda_base_src_base_init):
23588 * gst-libs/gst/cdda/gstcddabasesrc.h:
23589 * gst-libs/gst/tag/tag.h:
23590 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
23591 (gst_tag_register_musicbrainz_tags):
23592 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
23593 depend on libgsttag. This is required so we can extract/read tags like
23594 DISCID without depending on libgstcddabasesrc (which used to register
23596 * gst-libs/gst/tag/gstvorbistag.c:
23597 Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
23598 tags (also see #347848).
23599 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
23600 Log vorbis comments we are actually writing. Const-ify array.
23602 2006-09-23 08:53:30 +0000 Wim Taymans <wim.taymans@gmail.com>
23604 gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
23605 Original commit message from CVS:
23606 * gst/playback/gstplaybasebin.c: (gen_preroll_element):
23607 Improve buffering a bit by avoiding a deadlock because we cannot assume
23608 the underrun is always called.
23610 2006-09-23 08:51:14 +0000 Young-Ho Cha <ganadist@chollian.net>
23612 gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
23613 Original commit message from CVS:
23614 Patch by: Young-Ho Cha <ganadist at chollian dot net>
23615 * gst-libs/gst/riff/riff-ids.h:
23616 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
23617 (gst_riff_create_audio_template_caps):
23618 Added MPEG-4 AAC and id and caps. Fixes #357289
23619 Added WMA9 Lossless id.
23621 2006-09-22 14:50:01 +0000 Tim-Philipp Müller <tim@centricular.net>
23623 ext/gnomevfs/gstgnomevfssrc.c: Fix misleading docs addition.
23624 Original commit message from CVS:
23625 * ext/gnomevfs/gstgnomevfssrc.c:
23626 Fix misleading docs addition.
23627 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
23628 Get rid of compiler warning the right way.
23630 2006-09-22 14:13:34 +0000 Wim Taymans <wim.taymans@gmail.com>
23632 gst-libs/gst/rtp/gstbasertpdepayload.*: Small cleanups.
23633 Original commit message from CVS:
23634 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23635 (gst_base_rtp_depayload_finalize),
23636 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
23637 (gst_base_rtp_depayload_push_full),
23638 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
23639 (gst_base_rtp_depayload_process),
23640 (gst_base_rtp_depayload_set_gst_timestamp),
23641 (gst_base_rtp_depayload_queue_release):
23642 * gst-libs/gst/rtp/gstbasertpdepayload.h:
23645 Refactored the process method and added methods to push from the process
23647 Use _scale functions.
23648 API: gst_base_rtp_depayload_push_ts
23649 API: gst_base_rtp_depayload_push
23650 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
23651 timestamps are uint.
23653 2006-09-22 11:59:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23655 gst-libs/gst/interfaces/xoverlay.c: Remove unused statement from doc example.
23656 Original commit message from CVS:
23657 * gst-libs/gst/interfaces/xoverlay.c:
23658 Remove unused statement from doc example.
23660 2006-09-22 09:52:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23662 * gst/videorate/gstvideorate.c:
23664 Original commit message from CVS:
23667 2006-09-21 13:49:47 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23669 gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
23670 Original commit message from CVS:
23671 * gst-libs/gst/interfaces/videoorientation.c:
23672 (gst_video_orientation_iface_init),
23673 (gst_video_orientation_get_hflip),
23674 (gst_video_orientation_get_vflip),
23675 (gst_video_orientation_get_hcenter),
23676 (gst_video_orientation_get_vcenter),
23677 (gst_video_orientation_set_hflip),
23678 (gst_video_orientation_set_vflip),
23679 (gst_video_orientation_set_hcenter),
23680 (gst_video_orientation_set_vcenter):
23681 Add since tags to new API docs, ChangeLog surgery (forgot API keyword
23684 2006-09-21 09:27:47 +0000 Tim-Philipp Müller <tim@centricular.net>
23686 tests/check/: but disable for now since it doesn't pass (something wrong with
23687 Original commit message from CVS:
23688 * tests/check/Makefile.am:
23689 * tests/check/elements/.cvsignore:
23690 * tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
23691 (create_rgb_conversions), (rgb_conversion_free),
23692 (right_shift_colour), (fix_expected_colour), (check_rgb_buf),
23693 (got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
23694 Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
23695 but disable for now since it doesn't pass (something wrong with
23698 2006-09-21 07:01:48 +0000 Wim Taymans <wim.taymans@gmail.com>
23700 gst/playback/gstplaybasebin.c: Refactor handling of overrun detection.
23701 Original commit message from CVS:
23702 * gst/playback/gstplaybasebin.c: (group_commit),
23703 (queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
23704 (queue_out_of_data), (gen_preroll_element),
23705 (preroll_remove_overrun), (probe_triggered):
23706 Refactor handling of overrun detection.
23707 Separate handling of group completion and deadlock detection when doing
23708 network buffering. This should fix some deadlocks that were not detected
23709 because the group was completed.
23710 Add more comments, improve debugging.
23712 2006-09-21 05:31:00 +0000 Wim Taymans <wim.taymans@gmail.com>
23714 tests/check/: Some more compilation fixes.
23715 Original commit message from CVS:
23716 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
23717 * tests/check/libs/audio.c:
23718 Some more compilation fixes.
23720 2006-09-21 05:12:18 +0000 Wim Taymans <wim.taymans@gmail.com>
23722 gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
23723 Original commit message from CVS:
23724 * gst-libs/gst/audio/gstringbuffer.c:
23725 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
23726 (gst_ring_buffer_read):
23727 Early morning compilation fix.
23729 2006-09-20 18:09:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23733 Original commit message from CVS:
23736 2006-09-20 17:04:57 +0000 Wim Taymans <wim.taymans@gmail.com>
23738 tests/check/: Fix some warnings.
23739 Original commit message from CVS:
23740 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
23741 * tests/check/elements/multifdsink.c: (GST_START_TEST):
23742 * tests/check/elements/videorate.c: (GST_START_TEST):
23743 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
23744 * tests/check/pipelines/oggmux.c: (eos_buffer_probe):
23747 2006-09-20 10:59:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23749 sys/xvimage/xvimagesink.c: change colorkey behaviour back according to #354773 comment 6/7
23750 Original commit message from CVS:
23751 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
23752 (gst_xvimagesink_get_times):
23753 change colorkey behaviour back according to #354773 comment 6/7
23755 2006-09-20 10:42:34 +0000 Tim-Philipp Müller <tim@centricular.net>
23758 ChangeLog surgery: remove junk
23759 Original commit message from CVS:
23760 ChangeLog surgery: remove junk
23762 2006-09-19 11:31:06 +0000 Michael Smith <msmith@xiph.org>
23764 gst/tcp/gstmultifdsink.*: Implement stubbed out properties unit-type, units-soft-max, units-max, to allow specifying ...
23765 Original commit message from CVS:
23766 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
23767 (gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
23768 (gst_multi_fd_sink_recover_client),
23769 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
23770 (gst_multi_fd_sink_get_property):
23771 * gst/tcp/gstmultifdsink.h:
23772 Implement stubbed out properties unit-type, units-soft-max,
23773 units-max, to allow specifying maximum sizes in units other than
23777 2006-09-19 10:23:03 +0000 Wim Taymans <wim.taymans@gmail.com>
23779 gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
23780 Original commit message from CVS:
23781 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
23782 (gst_riff_create_audio_template_caps):
23783 Reorder the audio formats a bit for clarity.
23784 Detect and create caps for MSGSM and MSN (WAV49).
23786 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
23787 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
23788 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
23789 Small cleanups, move error handling out of normal flow for clarity.
23791 2006-09-18 15:59:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23793 Add new interface to control video orientation (fixes #354908)
23794 Original commit message from CVS:
23795 * docs/libs/gst-plugins-base-libs-docs.sgml:
23796 * docs/libs/gst-plugins-base-libs.types:
23797 * gst-libs/gst/interfaces/Makefile.am:
23798 * gst-libs/gst/interfaces/videoorientation.c:
23799 (gst_video_orientation_get_type),
23800 (gst_video_orientation_iface_init),
23801 (gst_video_orientation_get_hflip),
23802 (gst_video_orientation_get_vflip),
23803 (gst_video_orientation_get_hcenter),
23804 (gst_video_orientation_get_vcenter),
23805 (gst_video_orientation_set_hflip),
23806 (gst_video_orientation_set_vflip),
23807 (gst_video_orientation_set_hcenter),
23808 (gst_video_orientation_set_vcenter):
23809 * gst-libs/gst/interfaces/videoorientation.h:
23810 Add new interface to control video orientation (fixes #354908)
23812 2006-09-18 15:48:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23814 gst/videotestsrc/gstvideotestsrc.c: Use G_UNLIKELY in _create and log one more detail.
23815 Original commit message from CVS:
23816 * gst/videotestsrc/gstvideotestsrc.c:
23817 Use G_UNLIKELY in _create and log one more detail.
23818 (gst_video_test_src_get_times), (gst_video_test_src_create):
23819 * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
23820 Use gst_util_uint64_scale_int in _get_times().
23822 2006-09-18 15:00:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23824 sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
23825 Original commit message from CVS:
23826 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
23827 Give better warning message (add object and detail).
23829 2006-09-18 14:42:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23831 sys/xvimage/xvimagesink.c: xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes #354773), use gst_util...
23832 Original commit message from CVS:
23833 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
23834 (gst_xvimagesink_get_times):
23835 xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
23836 #354773), use gst_util_uint64_scale_int in _get_times()
23838 2006-09-18 14:21:45 +0000 Michael Smith <msmith@xiph.org>
23840 ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
23841 Original commit message from CVS:
23842 * ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
23843 Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
23844 always true, leading to dropping all timestamps.
23846 2006-09-18 11:40:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23848 ext/libvisual/visual.c: update to work also with libvisual 0.4 API
23849 Original commit message from CVS:
23850 * ext/libvisual/visual.c: (gst_vis_src_negotiate),
23851 (gst_visual_chain), (gst_visual_change_state):
23852 update to work also with libvisual 0.4 API
23853 * tools/gst-launch-ext.1.in:
23854 * tools/gst-visualise.1.in:
23855 remove references to old man-pages
23856 * tests/examples/seek/seek.c: (main):
23857 add real meadi-buttons, add tool-tips for the seek-options, arrange
23858 seek options in a table
23860 2006-09-18 10:57:28 +0000 Michael Smith <msmith@xiph.org>
23862 ext/ogg/gstoggmux.c: Don't generate out-of-order timestamps from oggmux, instead clamp output timestamps to be >= the...
23863 Original commit message from CVS:
23864 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
23865 (gst_ogg_mux_push_buffer):
23866 Don't generate out-of-order timestamps from oggmux, instead clamp
23867 output timestamps to be >= the previously output ts.
23870 2006-09-18 10:18:22 +0000 Michael Smith <msmith@xiph.org>
23872 gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
23873 Original commit message from CVS:
23874 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
23875 (gst_multi_fd_sink_class_init):
23876 Updates, fixes, and typo corrections for multifdsink. No functional
23879 2006-09-17 21:58:06 +0000 Michael Smith <msmith@xiph.org>
23881 gst/typefind/gsttypefindfunctions.c: Don't crash on truncated files - check that we got an 8 byte buffer before tryin...
23882 Original commit message from CVS:
23883 * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
23884 Don't crash on truncated files - check that we got an 8 byte buffer
23885 before trying to memcmp it.
23887 2006-09-17 20:32:09 +0000 Tim-Philipp Müller <tim@centricular.net>
23889 gst/playback/gstplaybasebin.c: Make stream-switching appear instant to the application (ie. make sure that a g_object...
23890 Original commit message from CVS:
23891 * gst/playback/gstplaybasebin.c: (get_active_source):
23892 Make stream-switching appear instant to the application
23893 (ie. make sure that a g_object_get on 'current-foo' returns
23894 the stream previously set with g_object_set(). Totem needs
23895 this to update stream-related meta-info (like audio-codec)
23896 correctly when switching streams.
23898 2006-09-17 20:14:43 +0000 Tim-Philipp Müller <tim@centricular.net>
23900 ext/alsa/gstalsamixer.c: Try harder to guess which mixer track is the master mixer track (instead of just taking the ...
23901 Original commit message from CVS:
23902 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
23903 (gst_alsa_mixer_ensure_track_list):
23904 Try harder to guess which mixer track is the master mixer
23905 track (instead of just taking the first one that has a pvolume).
23908 2006-09-17 11:24:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23910 gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
23911 Original commit message from CVS:
23912 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
23913 (gst_audio_convert_transform_caps):
23914 Get structure-name just once.
23916 2006-09-16 22:30:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23918 tests/check/: Fix big batch of compiler warnings.
23919 Original commit message from CVS:
23920 * tests/check/elements/audioresample.c: (GST_START_TEST):
23921 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
23922 * tests/check/elements/volume.c: (GST_START_TEST):
23923 * tests/check/elements/vorbisdec.c: (GST_START_TEST):
23924 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
23925 (test_pipeline), (GST_START_TEST):
23926 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
23927 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
23928 Fix big batch of compiler warnings.
23930 2006-09-16 21:54:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23932 ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
23933 Original commit message from CVS:
23934 * ext/gnomevfs/gstgnomevfssrc.c:
23935 Add docs about icydemux usage in connection with gnomevfssrc
23936 * ext/libvisual/visual.c:
23937 * ext/ogg/gstoggaviparse.c:
23938 * ext/ogg/gstoggdemux.c:
23939 * ext/ogg/gstoggmux.c:
23940 * ext/ogg/gstoggparse.c:
23941 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
23942 * gst-libs/gst/audio/gstaudiosink.c:
23943 * gst-libs/gst/audio/gstaudiosrc.c:
23944 * gst/audiorate/gstaudiorate.c:
23945 More G_OBJECT macro fixing.
23946 * gst/audiotestsrc/gstaudiotestsrc.h:
23947 Fix wrong info in header due to copy & paste
23949 2006-09-15 14:53:44 +0000 Wim Taymans <wim.taymans@gmail.com>
23951 gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
23952 Original commit message from CVS:
23953 * gst-libs/gst/audio/gstbaseaudiosink.c:
23954 (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
23955 * gst-libs/gst/audio/gstbaseaudiosrc.c:
23956 (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
23957 (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
23958 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
23959 Do the delay calculation in the source/sink base classes as this is
23960 specific for the capture/playback mode.
23961 Try to fixate a bit better, like round depth up to a multiple of 8
23963 Handle underruns correctly by marking DISCONT on buffers and adjusting
23964 timestamps to handle the gap.
23965 Set offset/offset_end correctly on buffers.
23966 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
23967 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
23968 (gst_ring_buffer_read):
23969 Remove resync and underrun recovery from the ringbuffer.
23970 Fix ringbuffer read code on under/overrun.
23972 2006-09-15 11:17:02 +0000 Wim Taymans <wim.taymans@gmail.com>
23974 gst/playback/gstplaybasebin.*: Don't use a 0 low watermark when buffering, it is catching starvation way too late. In...
23975 Original commit message from CVS:
23976 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
23977 (gst_play_base_bin_init), (fill_buffer), (check_queue),
23978 (queue_threshold_reached), (gst_play_base_bin_set_property),
23979 (gst_play_base_bin_get_property):
23980 * gst/playback/gstplaybasebin.h:
23981 Don't use a 0 low watermark when buffering, it is catching starvation
23982 way too late. Instead, use a 3 second queue with 30 and 95
23983 percent low/high watermarks.
23984 Added queue-min-threshold property to configure low watermark.
23985 Use new _buffering message API.
23986 Make queue_threshold variable big enough to store a uint64 time value.
23987 API: playbin::queue-min-threshold property.
23989 2006-09-15 09:13:50 +0000 Wim Taymans <wim.taymans@gmail.com>
23991 configure.ac: We require 0.10.10.1 now because of _wait_preroll().
23992 Original commit message from CVS:
23994 We require 0.10.10.1 now because of _wait_preroll().
23995 * gst-libs/gst/audio/gstbaseaudiosink.c:
23996 (gst_base_audio_sink_render):
23997 Use gst_base_sink_wait_preroll().
23999 2006-09-15 09:09:00 +0000 Wim Taymans <wim.taymans@gmail.com>
24001 ext/alsa/: Use DEBUG_OBJECT more.
24002 Original commit message from CVS:
24003 * ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
24004 * ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
24005 Use DEBUG_OBJECT more.
24007 === release 0.10.10 ===
24009 2006-09-14 20:09:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24016 * docs/plugins/gst-plugins-base-plugins.args:
24017 * docs/plugins/inspect/plugin-adder.xml:
24018 * docs/plugins/inspect/plugin-alsa.xml:
24019 * docs/plugins/inspect/plugin-audioconvert.xml:
24020 * docs/plugins/inspect/plugin-audiorate.xml:
24021 * docs/plugins/inspect/plugin-audioresample.xml:
24022 * docs/plugins/inspect/plugin-audiotestsrc.xml:
24023 * docs/plugins/inspect/plugin-cdparanoia.xml:
24024 * docs/plugins/inspect/plugin-decodebin.xml:
24025 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
24026 * docs/plugins/inspect/plugin-gdp.xml:
24027 * docs/plugins/inspect/plugin-gnomevfs.xml:
24028 * docs/plugins/inspect/plugin-libvisual.xml:
24029 * docs/plugins/inspect/plugin-ogg.xml:
24030 * docs/plugins/inspect/plugin-pango.xml:
24031 * docs/plugins/inspect/plugin-playbin.xml:
24032 * docs/plugins/inspect/plugin-subparse.xml:
24033 * docs/plugins/inspect/plugin-tcp.xml:
24034 * docs/plugins/inspect/plugin-theora.xml:
24035 * docs/plugins/inspect/plugin-typefindfunctions.xml:
24036 * docs/plugins/inspect/plugin-video4linux.xml:
24037 * docs/plugins/inspect/plugin-videorate.xml:
24038 * docs/plugins/inspect/plugin-videoscale.xml:
24039 * docs/plugins/inspect/plugin-videotestsrc.xml:
24040 * docs/plugins/inspect/plugin-volume.xml:
24041 * docs/plugins/inspect/plugin-vorbis.xml:
24042 * docs/plugins/inspect/plugin-ximagesink.xml:
24043 * docs/plugins/inspect/plugin-xvimagesink.xml:
24044 * ext/theora/theoraparse.c:
24045 * gst-libs/gst/rtp/gstrtpbuffer.c:
24046 * gst/playback/gstplaybin.c:
24047 * tests/check/Makefile.am:
24048 * win32/common/config.h:
24050 Original commit message from CVS:
24053 2006-09-09 16:08:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24056 * win32/common/config.h:
24058 Original commit message from CVS:
24061 2006-09-07 19:01:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24064 update bug in changelog
24065 Original commit message from CVS:
24066 update bug in changelog
24068 2006-09-07 19:00:33 +0000 Michael Smith <msmith@fluendo.com>
24070 Fix implementation of sync-method 'next-keyframe'
24071 Original commit message from CVS:
24072 patch by: Michael Smith <msmith at fluendo dot com>
24073 * gst/tcp/gstmultifdsink.c: (is_sync_frame),
24074 (gst_multi_fd_sink_client_queue_buffer),
24075 (gst_multi_fd_sink_new_client):
24076 * tests/check/elements/multifdsink.c: (GST_START_TEST),
24077 (multifdsink_suite):
24078 Fix implementation of sync-method 'next-keyframe'
24080 2006-09-07 15:00:08 +0000 Wim Taymans <wim@fluendo.com>
24082 ext/gnomevfs/gstgnomevfssrc.c: This patch removes the RANDOM flag that was incorrectly introduced with revision 1.91....
24083 Original commit message from CVS:
24084 patch by: Wim Taymans <wim at fluendo dot com>
24085 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
24086 This patch removes the RANDOM flag that was incorrectly introduced with
24087 revision 1.91. Fixes #354590
24089 2006-09-07 14:56:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24092 * win32/common/config.h:
24094 Original commit message from CVS:
24097 2006-09-07 14:56:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24114 Original commit message from CVS:
24117 2006-09-05 09:12:25 +0000 Tim-Philipp Müller <tim@centricular.net>
24119 tests/check/Makefile.am: Random variation in Makefile line to see if it makes the gen64-base-full bot any happier.
24120 Original commit message from CVS:
24121 * tests/check/Makefile.am:
24122 Random variation in Makefile line to see if it makes the
24123 gen64-base-full bot any happier.
24125 2006-09-04 19:04:35 +0000 Tim-Philipp Müller <tim@centricular.net>
24127 tests/check/pipelines/oggmux.c: Disable test that fails at the moment (killed after timeout).
24128 Original commit message from CVS:
24129 * tests/check/pipelines/oggmux.c: (oggmux_suite):
24130 Disable test that fails at the moment (killed after timeout).
24132 2006-09-04 18:19:06 +0000 James Livingston <doclivingston@gmail.com>
24134 tests/check/: Add simple unit test for oggmux from #337026 with checking for the
24135 Original commit message from CVS:
24136 Patch by: James Livingston <doclivingston at gmail.com>
24137 * tests/check/Makefile.am:
24138 * tests/check/pipelines/.cvsignore:
24139 * tests/check/pipelines/oggmux.c: (get_page_codec),
24140 (check_chain_final_state), (fail_if_audio), (validate_ogg_page),
24141 (eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
24142 (test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
24143 (test_theora_vorbis), (oggmux_suite):
24144 Add simple unit test for oggmux from #337026 with checking for the
24145 EOS flags disabled for the time being.
24147 2006-09-04 09:13:01 +0000 Alessandro Dessina <alessandro@nnva.org>
24149 ext/ogg/gstoggmux.c: Add cmml caps to oggmux. Fixes #353912
24150 Original commit message from CVS:
24151 patch by: Alessandro Dessina <alessandro nnva org>
24152 * ext/ogg/gstoggmux.c:
24153 Add cmml caps to oggmux. Fixes #353912
24155 2006-09-02 13:20:59 +0000 Tim-Philipp Müller <tim@centricular.net>
24157 tests/check/elements/videotestsrc.c: Returning a return value often helps. In this case, we don't need the return val...
24158 Original commit message from CVS:
24159 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
24160 Returning a return value often helps. In this case, we
24161 don't need the return value anyway, so just get rid of it.
24162 Should make build bots much happier.
24164 2006-09-02 12:59:48 +0000 Tim-Philipp Müller <tim@centricular.net>
24166 gst/videotestsrc/videotestsrc.*: Add support for AYUV and the various RGBA formats. Initialise fields of paintinfo st...
24167 Original commit message from CVS:
24168 * gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
24169 (paint_get_structure), (gst_video_test_src_get_size),
24170 (gst_video_test_src_smpte), (gst_video_test_src_snow),
24171 (gst_video_test_src_unicolor), (paint_setup_AYUV),
24172 (paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
24173 (paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
24174 * gst/videotestsrc/videotestsrc.h:
24175 Add support for AYUV and the various RGBA formats. Initialise
24176 fields of paintinfo structs allocated on the stack.
24177 * tests/check/elements/videotestsrc.c: (right_shift_colour),
24178 (fix_expected_colour), (check_rgb_buf), (got_buf_cb),
24179 (GST_START_TEST), (videotestsrc_suite):
24180 Add unit tests for videotestsrc's RGB output.
24182 2006-09-01 16:12:35 +0000 Tim-Philipp Müller <tim@centricular.net>
24184 gst/videotestsrc/: Add more uni-colour patterns ("white", "red", "green", and "blue").
24185 Original commit message from CVS:
24186 * gst/videotestsrc/gstvideotestsrc.c:
24187 (gst_video_test_src_pattern_get_type),
24188 (gst_video_test_src_set_pattern):
24189 * gst/videotestsrc/gstvideotestsrc.h:
24190 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
24191 (gst_video_test_src_black), (gst_video_test_src_white),
24192 (gst_video_test_src_red), (gst_video_test_src_green),
24193 (gst_video_test_src_blue):
24194 * gst/videotestsrc/videotestsrc.h:
24195 Add more uni-colour patterns ("white", "red", "green", and "blue").
24197 2006-09-01 10:07:05 +0000 Tim-Philipp Müller <tim@centricular.net>
24199 gst/videotestsrc/videotestsrc.c: Fix stride for YVYU, should be word-aligned (#353658).
24200 Original commit message from CVS:
24201 * gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
24202 Fix stride for YVYU, should be word-aligned (#353658).
24204 2006-08-31 14:37:33 +0000 Tim-Philipp Müller <tim@centricular.net>
24206 gst/adder/gstadder.c: Fix build.
24207 Original commit message from CVS:
24208 * gst/adder/gstadder.c: (gst_adder_src_event):
24211 2006-08-31 12:39:17 +0000 Edward Hervey <bilboed@bilboed.com>
24213 gst/adder/gstadder.*: Remember the start position asked in the incoming seeks, so we can output GST_EVENT_NEW_SEGMENT...
24214 Original commit message from CVS:
24215 * gst/adder/gstadder.c: (forward_event_func),
24216 (gst_adder_src_event), (gst_adder_collected),
24217 (gst_adder_change_state):
24218 * gst/adder/gstadder.h:
24219 Remember the start position asked in the incoming seeks, so we can
24220 output GST_EVENT_NEW_SEGMENT with a correct position value (instead
24221 of assuming it will always be 0).
24223 2006-08-31 12:31:00 +0000 Edward Hervey <bilboed@bilboed.com>
24225 ext/ogg/gstoggdemux.c: Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
24226 Original commit message from CVS:
24227 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
24228 (gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
24229 (gst_ogg_demux_loop):
24230 Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
24232 2006-08-30 17:22:27 +0000 Tim-Philipp Müller <tim@centricular.net>
24234 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Return FALSE instead of returning a random false unit size when the forma...
24235 Original commit message from CVS:
24236 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
24237 (gst_ffmpegcsp_get_unit_size):
24238 Return FALSE instead of returning a random false unit
24239 size when the format isn't known/supported (even if
24240 this shouldn't happen under normal circumstances).
24242 2006-08-29 15:23:46 +0000 Tim-Philipp Müller <tim@centricular.net>
24244 ext/gnomevfs/gstgnomevfssrc.c: Try harder to get the size from a uri by using _info_uri() when _info_from_handle() do...
24245 Original commit message from CVS:
24246 Patch by: Tim-Philipp Müller <tim at centricular dot net>
24247 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
24248 (gst_gnome_vfs_src_start):
24249 Try harder to get the size from a uri by using _info_uri() when
24250 _info_from_handle() does not give us enough info.
24251 Also follow symlinks when getting the size.
24252 Partially Fixes #332864.
24254 2006-08-29 11:50:51 +0000 Viktor Peters <viktor.peters@gmail.com>
24256 ext/alsa/: Improve and fix mixer track handling, in particular better handling of alsa's pvolume/pswitch/cvolume/cswi...
24257 Original commit message from CVS:
24258 Patch by: Viktor Peters <viktor dot peters at gmail dot com>
24259 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
24260 (gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
24261 (gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
24262 (gst_alsa_mixer_set_record):
24263 * ext/alsa/gstalsamixertrack.c:
24264 (gst_alsa_mixer_track_update_alsa_capabilities),
24265 (alsa_track_has_cap), (gst_alsa_mixer_track_new),
24266 (gst_alsa_mixer_track_update):
24267 * ext/alsa/gstalsamixertrack.h:
24268 Improve and fix mixer track handling, in particular better handling
24269 of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
24270 track objects for tracks that have both capture and playback volume
24271 (and label them differently as well so they're not mistakenly
24272 assumed to be duplicates); classify mixer tracks that only affect
24273 the audible volume of something (rather than the capture volume)
24274 as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
24275 for capture tracks to correspond to alsa-pswitch alsa-cswitch
24276 (following the meaning documented in the mixer interface header
24277 file); add support for alsa's exclusive cswitch groups; update/sync
24278 state/flags better if mixer settings are changed by another
24279 application. Fixes #336075.
24281 2006-08-29 10:58:43 +0000 Tim-Philipp Müller <tim@centricular.net>
24283 gst/playback/gstplaybin.c: Improve docs: add section about BUFFERING messages sent by playbin.
24284 Original commit message from CVS:
24285 * gst/playback/gstplaybin.c:
24286 Improve docs: add section about BUFFERING messages sent by playbin.
24288 2006-08-29 10:51:12 +0000 Michael Smith <msmith@xiph.org>
24290 ext/vorbis/vorbisenc.c: Ignore explicit DISCONT marked on buffers (which is often spurious, particularly when using m...
24291 Original commit message from CVS:
24292 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
24293 (gst_vorbis_enc_buffer_check_discontinuous),
24294 (gst_vorbis_enc_chain):
24295 Ignore explicit DISCONT marked on buffers (which is often spurious,
24296 particularly when using multiple segments), in favour of solely
24297 using the timestamps/durations.
24299 2006-08-29 10:32:34 +0000 Edward Hervey <bilboed@bilboed.com>
24301 gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
24302 Original commit message from CVS:
24303 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
24304 Don't rely on incoming buffers offset anymore, since it is completely
24305 broken when using multiple segments.
24306 Instead convert the incoming buffers timestamp to running time, and
24307 then convert that value to the offsets.
24308 Also inform GstSegment of the last outputted stop position, which is
24309 needed if we received several segments with an unknown stop value.
24311 2006-08-29 08:03:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24313 ext/vorbis/vorbisenc.c: fix buffer unreffing on a header push failure
24314 Original commit message from CVS:
24315 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
24316 fix buffer unreffing on a header push failure
24318 2006-08-28 16:17:13 +0000 Wim Taymans <wim.taymans@gmail.com>
24320 gst/audiorate/gstaudiorate.c: Make the metadata of the buffer writable before changing its flags.
24321 Original commit message from CVS:
24322 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
24323 (gst_audio_rate_chain):
24324 Make the metadata of the buffer writable before changing its
24327 2006-08-28 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
24330 Fix changelog with bugzilla bug it fixed.
24331 Original commit message from CVS:
24332 Fix changelog with bugzilla bug it fixed.
24334 2006-08-28 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
24336 gst/audiorate/gstaudiorate.c: Fix audiorate some more.
24337 Original commit message from CVS:
24338 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
24339 (gst_audio_rate_setcaps), (gst_audio_rate_init),
24340 (gst_audio_rate_sink_event), (gst_audio_rate_src_event),
24341 (gst_audio_rate_chain), (gst_audio_rate_change_state):
24342 Fix audiorate some more.
24343 Reset and resync counters on flush and READY.
24344 Handle the DISCONT flag correctly.
24345 Use GstSegment to track position.
24346 Fail when not negotiated.
24348 2006-08-25 16:48:28 +0000 Michael Smith <msmith@xiph.org>
24350 gst/tcp/gstmultifdsink.c: Fix spelling.
24351 Original commit message from CVS:
24352 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
24354 Remove accidently included debug line.
24356 2006-08-25 16:39:38 +0000 Wim Taymans <wim.taymans@gmail.com>
24358 gst/tcp/gstmultifdsink.c: Small cleanups.
24359 Original commit message from CVS:
24360 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
24362 If a buffer is received with no caps, make the buffer metadata
24363 writable and set the caps, making sure that we don't screw up the
24366 2006-08-25 16:19:55 +0000 Michael Smith <msmith@xiph.org>
24368 gst/gdp/gstgdppay.c: Fix memory leaks and misleading debug messages, add a couple of comments.
24369 Original commit message from CVS:
24370 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
24371 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
24372 Fix memory leaks and misleading debug messages, add a couple of
24374 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
24375 (gst_multi_fd_sink_render):
24376 Do not use gst_buffer_make_writable() in a basesink render method,
24377 as it may incorrectly unref the buffer. Instead, use convoluted
24378 dance to avoid copying the buffer except when we need to.
24380 2006-08-25 09:54:56 +0000 Michael Smith <msmith@xiph.org>
24382 ext/vorbis/vorbisenc.c: Allow very small discontinuities in the timestamps. These we can't do anything useful with an...
24383 Original commit message from CVS:
24384 * ext/vorbis/vorbisenc.c:
24385 (gst_vorbis_enc_buffer_check_discontinuous):
24386 Allow very small discontinuities in the timestamps. These we can't
24387 do anything useful with anyway (because vorbis's timestamps have
24388 only sample granularity), and are commonly produced by elements with
24389 minor bugs. Allow up to 1/2 a sample out.
24392 2006-08-24 11:18:56 +0000 Wim Taymans <wim.taymans@gmail.com>
24394 tests/examples/seek/seek.c: Add a checkbox to enable play scrubbing. Makes it possible to disable normal scrubbing.
24395 Original commit message from CVS:
24396 * tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
24397 (play_scrub_toggle_cb), (main):
24398 Add a checkbox to enable play scrubbing. Makes it possible to disable
24401 2006-08-23 19:37:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24403 tests/check/elements/.cvsignore: make buildbot happy
24404 Original commit message from CVS:
24405 * tests/check/elements/.cvsignore:
24406 make buildbot happy
24408 2006-08-23 16:43:03 +0000 Tim-Philipp Müller <tim@centricular.net>
24410 ext/ogg/gstogmparse.c: Refactor ogm parse, do better input checking, misc. clean-ups.
24411 Original commit message from CVS:
24412 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
24413 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
24414 (gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
24415 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
24416 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
24417 (gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
24418 (gst_ogm_text_parse_strip_trailing_zeroes),
24419 (gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
24420 (gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
24421 Refactor ogm parse, do better input checking, misc. clean-ups.
24422 Cache incoming events and push them once the source pad has
24423 been created. Don't pass unterminated strings to sscanf().
24424 Strip trailing zeroes from subtitle text output, since they
24425 are not valid UTF-8. Don't push vorbiscomment packets on
24426 the subtitle text pad. Output perfect streams if possible.
24428 2006-08-23 15:27:38 +0000 Wim Taymans <wim.taymans@gmail.com>
24430 tests/check/libs/cddabasesrc.c: Waits for tasks to settle down so that we clean up correctly for valgrind.
24431 Original commit message from CVS:
24432 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
24433 Waits for tasks to settle down so that we clean up correctly for
24436 2006-08-23 15:11:56 +0000 Tim-Philipp Müller <tim@centricular.net>
24438 tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
24439 Original commit message from CVS:
24440 * tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
24441 Unit test fixes: \377 is more likely to fit into 8 bits than \777;
24442 actually return return value in taglists_are_equal.
24444 2006-08-23 12:14:20 +0000 Tim-Philipp Müller <tim@centricular.net>
24446 ext/ogg/gstogmparse.c: Fix crash due to broken bitstream parsing on x86-64: can't make any assumptions about sizeof(s...
24447 Original commit message from CVS:
24448 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
24449 Fix crash due to broken bitstream parsing on x86-64: can't make
24450 any assumptions about sizeof(struct) due to alignment/packing
24451 differences on different architectures. Fixes #351790.
24453 2006-08-22 16:31:47 +0000 Wim Taymans <wim.taymans@gmail.com>
24455 gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
24456 Original commit message from CVS:
24457 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
24458 (gst_riff_parse_chunk), (gst_riff_parse_file_header),
24459 (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
24460 (gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
24461 (gst_riff_parse_info):
24462 Protect public functions against bad input.
24466 2006-08-22 15:50:36 +0000 Tim-Philipp Müller <tim@centricular.net>
24468 gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
24469 Original commit message from CVS:
24470 * gst-libs/gst/riff/riff-ids.h:
24471 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24472 Add voxware audio IDs (even if we can't play it) (#351795).
24474 2006-08-22 15:11:52 +0000 Tim-Philipp Müller <tim@centricular.net>
24476 gst-libs/gst/riff/riff-media.c: Const-ify some arrays and use G_N_ELEMENTS instead of wasting oodles of RAM on termin...
24477 Original commit message from CVS:
24478 * gst-libs/gst/riff/riff-media.c:
24479 (gst_riff_create_video_template_caps),
24480 (gst_riff_create_audio_template_caps),
24481 (gst_riff_create_iavs_template_caps):
24482 Const-ify some arrays and use G_N_ELEMENTS instead
24483 of wasting oodles of RAM on terminator bits.
24485 2006-08-22 08:27:07 +0000 Tim-Philipp Müller <tim@centricular.net>
24487 And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
24488 Original commit message from CVS:
24489 * gst-libs/gst/tag/gstvorbistag.c:
24490 (gst_tag_list_to_vorbiscomment_buffer):
24491 * tests/check/libs/tag.c: (GST_START_TEST):
24492 And the same for _to_vorbiscomment_buffer(): allow
24493 id_data_len == 0 for speex.
24495 2006-08-21 19:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24499 Original commit message from CVS:
24502 2006-08-21 19:01:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24504 Move GDP plugin to -base from -bad. Closes #347783.
24505 Original commit message from CVS:
24507 * docs/plugins/Makefile.am:
24508 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
24509 * docs/plugins/gst-plugins-base-plugins-sections.txt:
24510 * docs/plugins/inspect/plugin-gdp.xml:
24511 * gst/gdp/Makefile.am:
24512 * tests/check/Makefile.am:
24513 Move GDP plugin to -base from -bad. Closes #347783.
24515 2006-08-21 18:34:46 +0000 Tim-Philipp Müller <tim@centricular.net>
24517 gst-libs/gst/tag/gstvorbistag.c: Allow id_data_len == 0 (needed for vorbis comments in Speex files).
24518 Original commit message from CVS:
24519 * gst-libs/gst/tag/gstvorbistag.c:
24520 (gst_tag_list_from_vorbiscomment_buffer):
24521 Allow id_data_len == 0 (needed for vorbis comments in Speex files).
24522 Also add some checks to make sure we don't memcmp() beyond the end of
24523 vorbiscomment buffer if the ID to check for is larger than the buffer.
24524 * tests/check/libs/tag.c: (GST_START_TEST):
24525 Some more tests for gst_tag_list_from_vorbiscomment_buffer().
24527 2006-08-21 16:39:25 +0000 Tim-Philipp Müller <tim@centricular.net>
24529 ext/vorbis/vorbisenc.c: Use vorbis comment utility functions from libgsttag instead of re-inventing the wheel (partia...
24530 Original commit message from CVS:
24531 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
24532 (gst_vorbis_enc_set_metadata):
24533 Use vorbis comment utility functions from libgsttag
24534 instead of re-inventing the wheel (partially fixes #347091).
24536 2006-08-21 11:42:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24538 tests/check/elements/audioconvert.c: Fix leaks. Wait for state transitions that might happen ASYNC, as well as some t...
24539 Original commit message from CVS:
24540 * tests/check/elements/audioconvert.c: (GST_START_TEST):
24541 Fix leaks. Wait for state transitions that might happen ASYNC, as well
24542 as some that won't.
24544 2006-08-21 10:32:51 +0000 Wim Taymans <wim.taymans@gmail.com>
24546 docs/libs/: Don't try to GObject scan the netbuffer as it's not a GObject.
24547 Original commit message from CVS:
24548 * docs/libs/Makefile.am:
24549 * docs/libs/gst-plugins-base-libs-sections.txt:
24550 * docs/libs/gst-plugins-base-libs.types:
24551 Don't try to GObject scan the netbuffer as it's not a GObject.
24553 * gst-libs/gst/netbuffer/gstnetbuffer.c:
24554 * gst-libs/gst/netbuffer/gstnetbuffer.h:
24555 Document GstNetBuffer.
24557 2006-08-21 08:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24559 tests/check/elements/audioconvert.c: Add testcase for caps-size-explosion
24560 Original commit message from CVS:
24561 * tests/check/elements/audioconvert.c: (GST_START_TEST),
24562 (audioconvert_suite):
24563 Add testcase for caps-size-explosion
24565 2006-08-20 13:05:44 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24567 gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size
24568 Original commit message from CVS:
24569 * gst/audioconvert/gstaudioconvert.c:
24570 (gst_audio_convert_get_unit_size), (set_structure_widths):
24571 Lower debug, use g_assert in _get_unit_size
24572 * gst/audioresample/gstaudioresample.c:
24573 (audioresample_get_unit_size):
24574 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
24575 (gst_ffmpegcsp_get_unit_size):
24576 * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
24577 use g_assert in _get_unit_size
24579 2006-08-18 21:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
24582 ChangeLog surgery: fix bug number
24583 Original commit message from CVS:
24584 ChangeLog surgery: fix bug number
24586 2006-08-18 16:43:26 +0000 Wim Taymans <wim.taymans@gmail.com>
24588 Document GstRTPBuffer.
24589 Original commit message from CVS:
24590 * docs/libs/gst-plugins-base-libs-sections.txt:
24591 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
24592 (gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
24593 (gst_rtp_buffer_get_payload_buffer):
24594 * gst-libs/gst/rtp/gstrtpbuffer.h:
24595 Document GstRTPBuffer.
24596 Added function to efficiently strip payload headers.
24597 API: gst_rtp_buffer_get_payload_subbuffer()
24599 2006-08-17 16:52:06 +0000 Tim-Philipp Müller <tim@centricular.net>
24601 gst-libs/gst/tag/gstvorbistag.c: Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT tags and deserialise...
24602 Original commit message from CVS:
24603 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
24604 (gst_tag_to_vorbis_comments):
24605 Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
24606 tags and deserialise them properly as well (#351768).
24607 Add some more gtk-doc blurbs and also some g_return_if_fail().
24608 * tests/check/libs/tag.c: (GST_START_TEST),
24609 (back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
24612 2006-08-17 15:43:40 +0000 Wim Taymans <wim.taymans@gmail.com>
24614 ext/ogg/: Added ogg-in-avi parser element. Fixes #140139.
24615 Original commit message from CVS:
24616 * ext/ogg/Makefile.am:
24617 * ext/ogg/gstogg.c: (plugin_init):
24618 * ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
24619 (gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
24620 (gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
24621 (gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
24622 (gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
24623 (gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
24624 Added ogg-in-avi parser element. Fixes #140139.
24625 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
24626 Fixed a bug in oggdemux debug code.
24627 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
24628 (gst_riff_create_audio_template_caps):
24629 Recognise Ogg in the AVI extensible wave format.
24631 2006-08-17 10:00:00 +0000 Tim-Philipp Müller <tim@centricular.net>
24633 gst-libs/gst/cdda/gstcddabasesrc.c: Make buffer durations add up (duration should be next_ts-ts for perfect streams)....
24634 Original commit message from CVS:
24635 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
24636 Make buffer durations add up (duration should be next_ts-ts for
24637 perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
24639 * tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
24640 (test_buffer_timestamps), (cddabasesrc_suite):
24641 Add unit test for the above.
24642 * tests/check/Makefile.am:
24643 Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
24644 to see what happens.
24646 2006-08-16 11:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
24648 ext/alsa/: Avoid setting and using a NULL device name.
24649 Original commit message from CVS:
24650 * ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
24651 (gst_alsasink_open):
24652 * ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
24653 (gst_alsasrc_open):
24654 Avoid setting and using a NULL device name.
24655 Print more info when we fail to open a device.
24657 2006-08-16 11:28:57 +0000 Tim-Philipp Müller <tim@centricular.net>
24659 API: add gst_tag_parse_extended_comment() (#351426).
24660 Original commit message from CVS:
24661 * docs/libs/gst-plugins-base-libs-sections.txt:
24662 * gst-libs/gst/tag/tag.h:
24663 * gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
24664 API: add gst_tag_parse_extended_comment() (#351426).
24665 * tests/check/Makefile.am:
24666 * tests/check/libs/.cvsignore:
24667 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
24668 Add unit test for gst_tag_parse_extended_comment().
24670 2006-08-15 19:20:16 +0000 Tim-Philipp Müller <tim@centricular.net>
24672 sys/: Fix leak (#351502).
24673 Original commit message from CVS:
24674 * sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
24675 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
24676 Fix leak (#351502).
24678 2006-08-15 17:21:33 +0000 Tim-Philipp Müller <tim@centricular.net>
24681 Original commit message from CVS:
24682 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
24683 * docs/plugins/gst-plugins-base-plugins-sections.txt:
24684 * docs/plugins/gst-plugins-base-plugins.args:
24685 * gst/playback/gstplaybin.c:
24687 * docs/plugins/inspect/plugin-adder.xml:
24688 * docs/plugins/inspect/plugin-alsa.xml:
24689 * docs/plugins/inspect/plugin-audioconvert.xml:
24690 * docs/plugins/inspect/plugin-audiorate.xml:
24691 * docs/plugins/inspect/plugin-audioresample.xml:
24692 * docs/plugins/inspect/plugin-audiotestsrc.xml:
24693 * docs/plugins/inspect/plugin-cdparanoia.xml:
24694 * docs/plugins/inspect/plugin-decodebin.xml:
24695 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
24696 * docs/plugins/inspect/plugin-gnomevfs.xml:
24697 * docs/plugins/inspect/plugin-ogg.xml:
24698 * docs/plugins/inspect/plugin-pango.xml:
24699 * docs/plugins/inspect/plugin-playbin.xml:
24700 * docs/plugins/inspect/plugin-subparse.xml:
24701 * docs/plugins/inspect/plugin-tcp.xml:
24702 * docs/plugins/inspect/plugin-theora.xml:
24703 * docs/plugins/inspect/plugin-typefindfunctions.xml:
24704 * docs/plugins/inspect/plugin-video4linux.xml:
24705 * docs/plugins/inspect/plugin-videorate.xml:
24706 * docs/plugins/inspect/plugin-videoscale.xml:
24707 * docs/plugins/inspect/plugin-videotestsrc.xml:
24708 * docs/plugins/inspect/plugin-volume.xml:
24709 * docs/plugins/inspect/plugin-vorbis.xml:
24710 * docs/plugins/inspect/plugin-ximagesink.xml:
24711 * docs/plugins/inspect/plugin-xvimagesink.xml:
24712 Update to CVS version.
24714 2006-08-14 17:54:01 +0000 Tim-Philipp Müller <tim@centricular.net>
24716 gst/playback/gstplaybin.c: Add "connection-speed" property; re-order redirect messages with multiple redirect locatio...
24717 Original commit message from CVS:
24718 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
24719 (gst_play_bin_set_property), (gst_play_bin_get_property),
24720 (value_list_append_structure_list),
24721 (gst_play_bin_handle_redirect_message),
24722 (gst_play_bin_handle_message):
24723 Add "connection-speed" property; re-order redirect messages with
24724 multiple redirect locations depending on the minimum bitrate if
24725 that information is available and a connection speed is set
24728 2006-08-14 11:41:04 +0000 Tim-Philipp Müller <tim@centricular.net>
24730 gst/playback/gstplaybin.c: Update max volume to the same value that the volume element uses.
24731 Original commit message from CVS:
24732 * gst/playback/gstplaybin.c:
24733 Update max volume to the same value that the volume element uses.
24735 2006-08-14 10:50:15 +0000 Wim Taymans <wim.taymans@gmail.com>
24737 ext/alsa/gstalsamixer.c: Less uglyness..
24738 Original commit message from CVS:
24739 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
24742 2006-08-14 10:49:10 +0000 Wim Taymans <wim.taymans@gmail.com>
24744 ext/ogg/gstoggdemux.c: Add some more debug info.
24745 Original commit message from CVS:
24746 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
24747 (gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
24748 (gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
24749 Add some more debug info.
24750 Don't crash when a seek failed.
24751 Actually return the result of the seek instead of TRUE.
24752 Ignore multiple BOS pages with the same serial so that we don't create
24753 the same stream multiple times.
24754 Post an error when we fail to do the initial seek.
24756 2006-08-13 14:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
24758 ext/alsa/gstalsa.c: Small code cleanup.
24759 Original commit message from CVS:
24760 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
24761 (gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
24762 Small code cleanup.
24763 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
24764 (gst_alsa_mixer_new):
24765 Remove hack that always set the device to hw:0*.
24766 Properly find the card name for whatever device was configured.
24767 Do some better debugging.
24769 * ext/alsa/gstalsamixerelement.c:
24770 (gst_alsa_mixer_element_set_property),
24771 (gst_alsa_mixer_element_change_state):
24773 Handle setting of a NULL device name better.
24775 2006-08-11 15:53:43 +0000 Wim Taymans <wim.taymans@gmail.com>
24777 gst/adder/gstadder.c: Don't clip float values. Fixes #350900.
24778 Original commit message from CVS:
24779 * gst/adder/gstadder.c:
24780 Don't clip float values. Fixes #350900.
24782 2006-08-11 15:33:17 +0000 Andy Wingo <wingo@pobox.com>
24784 gst/tcp/gsttcp.c: Really fix the build?
24785 Original commit message from CVS:
24786 2006-08-11 Andy Wingo <wingo@pobox.com>
24787 * gst/tcp/gsttcp.c: Really fix the build?
24789 2006-08-11 15:29:56 +0000 Andy Wingo <wingo@pobox.com>
24791 gst/tcp/gsttcp.h: For now, always disable deprecation here -- fixes the build.
24792 Original commit message from CVS:
24793 2006-08-11 Andy Wingo <wingo@pobox.com>
24794 * gst/tcp/gsttcp.h: For now, always disable deprecation here --
24797 2006-08-10 13:01:31 +0000 Tim-Philipp Müller <tim@centricular.net>
24799 gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field.
24800 Original commit message from CVS:
24801 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
24802 Float caps shouldn't have a "signed" field.
24804 2006-08-10 08:56:22 +0000 Tim-Philipp Müller <tim@centricular.net>
24806 ext/ogg/gstoggdemux.c: Implement SEEKING query in its most basic form, so that we can at least check if we're seekabl...
24807 Original commit message from CVS:
24808 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
24809 Implement SEEKING query in its most basic form, so that we can
24810 at least check if we're seekable or not (#350655).
24812 2006-08-09 14:42:58 +0000 Tim-Philipp Müller <tim@centricular.net>
24814 gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
24815 Original commit message from CVS:
24816 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
24817 The checks here are not even close to anything that would
24818 justify MAXIMUM probability, lowering to POSSIBLE until someone
24819 fixes the checks (case at hand: quicktime redirection files
24820 might start with 00 00 01 XX and pass the checks here just
24821 fine, see #350399).
24823 2006-08-08 13:57:29 +0000 Edward Hervey <bilboed@bilboed.com>
24825 tests/check/elements/gdpdepay.c: I forgot to include the file containing the #define :)
24826 Original commit message from CVS:
24827 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
24828 I forgot to include the file containing the #define :)
24829 Now includes "config.h"
24831 2006-08-08 13:45:44 +0000 Edward Hervey <bilboed@bilboed.com>
24833 tests/check/elements/gdpdepay.c: Ignore test known to fail on PPC64. See #348114.
24834 Original commit message from CVS:
24835 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
24836 Ignore test known to fail on PPC64. See #348114.
24838 2006-08-08 08:41:13 +0000 Sjoerd Simons <sjoerd@luon.net>
24840 gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
24841 Original commit message from CVS:
24842 Patch by: Sjoerd Simons <sjoerd at luon net>
24843 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
24844 Better detection for multipart/x-mixed-replace: accept leading
24845 whitespaces before the boundary marker as well (as our very own
24846 multipartmux used to produce) (#349068).
24848 2006-08-07 08:26:03 +0000 Young-Ho Cha <ganadist@chollian.net>
24850 gst-libs/gst/riff/: Detect DTS audio streams (#350157).
24851 Original commit message from CVS:
24852 Patch by: Young-Ho Cha <ganadist at chollian net>
24853 * gst-libs/gst/riff/riff-ids.h:
24854 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
24855 (gst_riff_create_audio_template_caps):
24856 Detect DTS audio streams (#350157).
24858 2006-08-05 17:08:05 +0000 Andy Wingo <wingo@pobox.com>
24860 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (gst_theora_parse_class_init) (theora_parse_dispose, theora_par...
24861 Original commit message from CVS:
24862 2006-08-05 Andy Wingo <wingo@pobox.com>
24863 * ext/theora/gsttheoraparse.h:
24864 * ext/theora/theoraparse.c (gst_theora_parse_class_init)
24865 (theora_parse_dispose, theora_parse_set_property)
24866 (theora_parse_get_property, theora_parse_munge_granulepos)
24867 (theora_parse_push_buffer, theora_parse_change_state): Add a
24868 property 'synchronization-points' to fix badly synchronized oggs.
24870 2006-08-04 13:20:23 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
24872 gst/gdp/gstgdpdepay.c: Fix event parsing by gdpdepay. Fixes #349916.
24873 Original commit message from CVS:
24874 2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
24875 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
24876 Fix event parsing by gdpdepay. Fixes #349916.
24878 2006-08-03 15:04:42 +0000 Tim-Philipp Müller <tim@centricular.net>
24880 tests/check/: Add a few tests for the channel position stuff in libgstaudio.
24881 Original commit message from CVS:
24882 * tests/check/Makefile.am:
24883 * tests/check/libs/.cvsignore:
24884 * tests/check/libs/audio.c: (structure_contains_channel_positions),
24885 (fixed_caps_have_channel_positions), (GST_START_TEST),
24886 (audio_suite), (main):
24887 Add a few tests for the channel position stuff in libgstaudio.
24889 2006-08-03 14:16:06 +0000 Tim-Philipp Müller <tim@centricular.net>
24891 ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
24892 Original commit message from CVS:
24893 * ext/alsa/gstalsa.c: (caps_add_channel_configuration),
24894 (gst_alsa_detect_channels):
24895 * ext/alsa/gstalsasink.c:
24896 Add support for cards that (only) do more than 8 channels,
24897 like the Delta 44 (#345188).
24898 * gst-libs/gst/audio/multichannel.c:
24899 (gst_audio_check_channel_positions):
24900 * gst-libs/gst/audio/multichannel.h:
24901 API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
24902 unspecified channel position and cannot be combined with any
24903 of the other audio channel positions; adjust position layout
24904 checks accordingly (#345188).
24906 2006-08-03 11:15:53 +0000 Tim-Philipp Müller <tim@centricular.net>
24908 gst/typefind/gsttypefindfunctions.c: Recognise ancient RealAudio files (see #349779).
24909 Original commit message from CVS:
24910 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
24911 Recognise ancient RealAudio files (see #349779).
24913 2006-08-03 09:01:25 +0000 Jens Granseuer <jensgr@gmx.net>
24915 gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
24916 Original commit message from CVS:
24917 Patch by: Jens Granseuer <jensgr at gmx net>
24918 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
24919 Add typefinder for Interplay's MVE format (#348973).
24921 2006-08-02 17:03:29 +0000 Marcel Moreaux <marcelm@luon.net>
24923 gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
24924 Original commit message from CVS:
24925 Patch by: Marcel Moreaux <marcelm at luon dot net>
24926 * gst-libs/gst/rtp/gstbasertpdepayload.c:
24927 (gst_base_rtp_depayload_add_to_queue):
24928 * gst-libs/gst/rtp/gstbasertpdepayload.h:
24929 Handle RTP sequence number rollover.
24930 Disable jitterbuffer by default.
24932 2006-08-02 16:56:19 +0000 Wim Taymans <wim.taymans@gmail.com>
24934 gst/gdp/gstgdpdepay.c: Disable seeking.
24935 Original commit message from CVS:
24936 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
24937 (gst_gdp_depay_finalize), (gst_gdp_depay_sink_event),
24938 (gst_gdp_depay_src_event), (gst_gdp_depay_chain),
24939 (gst_gdp_depay_change_state):
24942 Clear adapter on disconts.
24943 Clear caps when going to READY instead of NULL
24944 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
24945 (gst_gdp_pay_init), (gst_gdp_pay_finalize), (gst_gdp_pay_reset),
24946 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
24947 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
24948 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
24949 (gst_gdp_pay_sink_event), (gst_gdp_pay_src_event),
24950 (gst_gdp_pay_change_state):
24951 * gst/gdp/gstgdppay.h:
24952 Reset payloader when going to READY.
24953 Fix leaked buffers in ->queue on push errors.
24956 Create packetizer in _init, free in _finalize.
24958 2006-07-31 08:48:36 +0000 Wim Taymans <wim.taymans@gmail.com>
24960 gst/gdp/gstgdpdepay.c: Consume all events except EOS because we generate events from the gdp payload instead. Fixes #...
24961 Original commit message from CVS:
24962 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
24963 (gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
24964 Consume all events except EOS because we generate events from
24965 the gdp payload instead. Fixes #349204
24967 2006-07-28 17:17:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24969 gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
24970 Original commit message from CVS:
24971 * gst/audioresample/gstaudioresample.c: (audioresample_stop),
24972 (audioresample_set_caps):
24973 Don't leak references to the incoming caps. Clean them up when
24975 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
24976 (gst_video_scale_finalize):
24977 Don't leak our temporary pixel buffer.
24978 * tests/check/Makefile.am:
24979 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
24980 (GST_START_TEST), (simple_launch_lines_suite):
24981 Fix leaks and re-enable the test for valgrind checking.
24983 2006-07-28 16:39:31 +0000 Sjoerd Simons <sjoerd@luon.net>
24985 gst/typefind/gsttypefindfunctions.c: Add typefind function for multipart/x-mixed-replace (#348916).
24986 Original commit message from CVS:
24987 Patch by: Sjoerd Simons <sjoerd at luon net>
24988 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
24990 Add typefind function for multipart/x-mixed-replace (#348916).
24992 2006-07-28 14:14:58 +0000 Wim Taymans <wim.taymans@gmail.com>
24994 gst/adder/gstadder.c: Fix leak in duration query.
24995 Original commit message from CVS:
24996 * gst/adder/gstadder.c: (gst_adder_setcaps),
24997 (gst_adder_query_duration):
24998 Fix leak in duration query.
24999 Reflow some docs and notes.
25001 2006-07-28 13:42:48 +0000 Michael Smith <msmith@xiph.org>
25003 tests/check/pipelines/vorbisenc.c: Enable Andy's extra vorbisenc test, now that it passes. Also fix one aspect of it.
25004 Original commit message from CVS:
25005 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
25007 Enable Andy's extra vorbisenc test, now that it passes. Also fix one
25010 2006-07-28 12:48:21 +0000 Michael Smith <msmith@xiph.org>
25012 ext/vorbis/vorbisenc.*: Handle discontinuities in the input vorbis stream correctly, so that the output is properly t...
25013 Original commit message from CVS:
25014 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
25015 (gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
25016 (gst_vorbis_enc_push_buffer),
25017 (gst_vorbis_enc_buffer_check_discontinuous),
25018 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
25019 * ext/vorbis/vorbisenc.h:
25020 Handle discontinuities in the input vorbis stream correctly,
25021 so that the output is properly timestamped (and has good granulepos
25022 values). Needs some oggmux fixes too.
25024 2006-07-27 10:52:52 +0000 Kai Vehmanen <kv2004@eca.cx>
25026 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
25027 Original commit message from CVS:
25028 patch by: Kai Vehmanen <kv2004 eca cx>
25029 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25030 (gst_base_rtp_depayload_chain),
25031 (gst_base_rtp_depayload_handle_sink_event),
25032 (gst_base_rtp_depayload_change_state):
25033 Don't send multiple newsegments with different formats.
25036 2006-07-26 15:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
25038 ext/ogg/gstoggdemux.c: Make seeking in ogg more accurate again by doing the more correct granuletime to stream time c...
25039 Original commit message from CVS:
25040 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
25041 (gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
25042 Make seeking in ogg more accurate again by doing the more correct
25043 granuletime to stream time conversion.
25045 2006-07-26 10:59:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25047 gst/tcp/gstmultifdsink.c: debug a little more understandably do not use goto as a substitute for break, especially if...
25048 Original commit message from CVS:
25049 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
25050 (gst_multi_fd_sink_new_client):
25051 debug a little more understandably
25052 do not use goto as a substitute for break, especially if
25053 break is also being used
25055 2006-07-26 10:55:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25057 * gst/tcp/gsttcp.c:
25058 move a recurring normal event to LOG, where it should be
25059 Original commit message from CVS:
25060 move a recurring normal event to LOG, where it should be
25062 2006-07-26 10:54:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25064 * ext/vorbis/vorbisdec.c:
25066 Original commit message from CVS:
25069 2006-07-26 10:52:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25071 gst/gdp/gstgdpdepay.c: proxying get/set caps is the wrong thing to do, since we really do change caps quite fundament...
25072 Original commit message from CVS:
25073 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
25074 proxying get/set caps is the wrong thing to do, since we really
25075 do change caps quite fundamentally
25076 * tests/check/elements/gdpdepay.c:
25077 * tests/check/elements/gdppay.c:
25078 remove declaration of buffers, it's already done in gstcheck.h
25080 2006-07-26 10:31:06 +0000 Tim-Philipp Müller <tim@centricular.net>
25082 gst/playback/: Remove GLib-2.6 compatibility cruft.
25083 Original commit message from CVS:
25084 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
25085 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
25086 Remove GLib-2.6 compatibility cruft.
25088 2006-07-24 16:47:10 +0000 Wim Taymans <wim.taymans@gmail.com>
25090 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
25091 Original commit message from CVS:
25092 * gst-libs/gst/audio/gstbaseaudiosink.c:
25093 (gst_base_audio_sink_render):
25094 Don't try to align a sample to an unknown value.
25096 2006-07-24 15:14:17 +0000 Wim Taymans <wim.taymans@gmail.com>
25098 gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
25099 Original commit message from CVS:
25100 * gst-libs/gst/audio/gstbaseaudiosink.c:
25101 (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
25102 When the audio clock is slaved to another clock, never try to align
25103 samples but trust the rate interpolation algorithm.
25105 2006-07-24 14:34:42 +0000 Wim Taymans <wim.taymans@gmail.com>
25107 ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
25108 Original commit message from CVS:
25109 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
25110 Don't try to calculate silence samples, base class does this much
25112 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
25113 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
25114 (gst_ring_buffer_acquire):
25115 Calculate silence samples correctly.
25116 * gst-libs/gst/audio/gstringbuffer.h:
25119 2006-07-22 17:01:12 +0000 Tim-Philipp Müller <tim@centricular.net>
25121 gst/typefind/gsttypefindfunctions.c: Limit search for the first markup tag to the first few kB of the file. If we don...
25122 Original commit message from CVS:
25123 * gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
25124 Limit search for the first markup tag to the first few kB of
25125 the file. If we don't find one there, it's highly unlikely that
25126 this is an XML(-ish) file.
25128 2006-07-21 17:04:06 +0000 Andy Wingo <wingo@pobox.com>
25130 tests/check/pipelines/theoraenc.c (test_discontinuity): Similar test to the one in vorbisenc. Also commented out.
25131 Original commit message from CVS:
25132 2006-07-21 Andy Wingo <wingo@pobox.com>
25133 * tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
25134 test to the one in vorbisenc. Also commented out.
25136 2006-07-21 16:54:19 +0000 Andy Wingo <wingo@pobox.com>
25138 tests/check/pipelines/vorbisenc.c: New test, commented out until Mike lands some elite vorbisenc patches.
25139 Original commit message from CVS:
25140 2006-07-21 Andy Wingo <wingo@pobox.com>
25141 * tests/check/pipelines/vorbisenc.c:
25142 (test_discontinuity): New test, commented out until Mike lands
25143 some elite vorbisenc patches.
25145 2006-07-21 15:59:24 +0000 Andy Wingo <wingo@pobox.com>
25147 tests/check/pipelines/: Port to bufferstraw.
25148 Original commit message from CVS:
25149 2006-07-21 Andy Wingo <wingo@pobox.com>
25150 * tests/check/pipelines/vorbisenc.c:
25151 * tests/check/pipelines/theoraenc.c: Port to bufferstraw.
25152 Bufferstraw was actually factored out of these tests. Now we share
25155 2006-07-21 11:03:28 +0000 Wim Taymans <wim.taymans@gmail.com>
25157 ext/theora/theoradec.c: Better clipping.
25158 Original commit message from CVS:
25159 * ext/theora/theoradec.c: (clip_buffer):
25162 2006-07-21 10:43:54 +0000 Wim Taymans <wim.taymans@gmail.com>
25164 gst-libs/gst/audio/gstaudiosink.c: Fix leak.
25165 Original commit message from CVS:
25166 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
25167 (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
25168 (gst_audioringbuffer_release), (gst_audioringbuffer_stop):
25170 Avoid type casting when we can.
25171 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
25174 2006-07-20 16:57:29 +0000 Tim-Philipp Müller <tim@centricular.net>
25176 ext/alsa/gstalsamixerelement.c: Make state change fail if the specified device can't be opened for some reason.
25177 Original commit message from CVS:
25178 * ext/alsa/gstalsamixerelement.c:
25179 (gst_alsa_mixer_element_change_state):
25180 Make state change fail if the specified device can't be opened
25183 2006-07-20 10:42:21 +0000 Wim Taymans <wim.taymans@gmail.com>
25185 gst/playback/test.c: Example of a small audio/video player using decodebin.
25186 Original commit message from CVS:
25187 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
25188 (cb_newpad), (main):
25189 Example of a small audio/video player using decodebin.
25191 2006-07-20 05:56:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25193 gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
25194 Original commit message from CVS:
25195 * gst-libs/gst/riff/riff-ids.h:
25196 Add 'fact' chunk id
25198 2006-07-19 18:20:43 +0000 Wim Taymans <wim.taymans@gmail.com>
25200 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't assert when not negotiated but post a meaningfull error message. Fixes ...
25201 Original commit message from CVS:
25202 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25203 (gst_base_rtp_depayload_chain),
25204 (gst_base_rtp_depayload_change_state):
25205 Don't assert when not negotiated but post a meaningfull
25206 error message. Fixes #347918.
25207 * gst-libs/gst/rtp/gstbasertppayload.c:
25208 Add comment about better default MTU size.
25209 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
25210 Small cleanups, start docs.
25212 2006-07-19 14:46:36 +0000 Martin Szulecki <compiz@sukimashita.com>
25214 sys/v4l/gstv4lelement.c: If "device-name" is requested and the device is not open, try to temporarily open it to obta...
25215 Original commit message from CVS:
25216 Patch by: Martin Szulecki
25217 * sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
25218 If "device-name" is requested and the device is not
25219 open, try to temporarily open it to obtain this
25220 information (#342494).
25222 2006-07-19 12:25:00 +0000 Tim-Philipp Müller <tim@centricular.net>
25224 gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
25225 Original commit message from CVS:
25226 * gst-libs/gst/tag/gstid3tag.c:
25227 Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
25228 * gst-libs/gst/tag/gsttageditingprivate.h:
25229 * gst-libs/gst/tag/gstvorbistag.c:
25230 Some more random const-ifications.
25232 2006-07-18 19:48:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25234 gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
25235 Original commit message from CVS:
25236 * gst-libs/gst/riff/riff-ids.h:
25237 * gst-libs/gst/riff/riff-media.c:
25238 (gst_riff_create_video_template_caps):
25239 Add more FOURCCs (sort list to make stuff easier to find),
25240 add comment what those 16 bytes in struct _gst_riff_strh according to
25243 2006-07-17 14:17:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25245 gst/gdp/: remove parent_class setting, BOILERPLATE does this fix typo in comment
25246 Original commit message from CVS:
25247 2006-07-17 Thomas Vander Stichele <thomas at apestaart dot org>
25248 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
25249 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
25250 remove parent_class setting, BOILERPLATE does this
25251 (gst_gdp_pay_reset_streamheader):
25252 fix typo in comment
25254 2006-07-17 13:48:10 +0000 Tim-Philipp Müller <tim@centricular.net>
25256 gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
25257 Original commit message from CVS:
25258 * gst-libs/gst/audio/multichannel.c:
25259 (gst_audio_check_channel_positions),
25260 (gst_audio_fixate_channel_positions):
25261 Const-ify two arrays.
25263 2006-07-17 12:33:42 +0000 Tim-Philipp Müller <tim@centricular.net>
25265 ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
25266 Original commit message from CVS:
25267 * ext/alsa/gstalsa.c: (caps_add_channel_configuration):
25268 Fix typo, so that alsasink also advertises 8 channels
25269 if that's supported (tags: can, worms, open, alsa, ph34r).
25271 2006-07-17 12:01:04 +0000 Wim Taymans <wim.taymans@gmail.com>
25273 ext/ogg/gstoggdemux.c: *sigh*, when is the compiler going to warn when the comments are out-of-sync with the code.. R...
25274 Original commit message from CVS:
25275 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
25276 (gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
25277 *sigh*, when is the compiler going to warn when the comments
25278 are out-of-sync with the code.. Refix case of busted theora
25279 headers with 0 granule pos.
25281 2006-07-14 17:56:59 +0000 Wim Taymans <wim.taymans@gmail.com>
25283 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix 99% cpu load by waiting for absolute times on the clock. Fixes #347300.
25284 Original commit message from CVS:
25285 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25286 (gst_base_rtp_depayload_wait),
25287 (gst_base_rtp_depayload_change_state),
25288 (gst_base_rtp_depayload_set_property),
25289 (gst_base_rtp_depayload_get_property):
25290 Fix 99% cpu load by waiting for absolute times on the
25291 clock. Fixes #347300.
25293 2006-07-14 17:07:08 +0000 Andy Wingo <wingo@pobox.com>
25295 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (theora_parse_drain_event_queue) (theora_parse_push_headers, th...
25296 Original commit message from CVS:
25297 2006-07-14 Andy Wingo <wingo@pobox.com>
25298 * ext/theora/gsttheoraparse.h:
25299 * ext/theora/theoraparse.c (theora_parse_drain_event_queue)
25300 (theora_parse_push_headers, theora_parse_clear_queue)
25301 (theora_parse_drain_queue_prematurely, )
25302 (theora_parse_sink_event, theora_parse_change_state): Queue events
25303 until we initialized our state, like in vorbisparse.
25305 2006-07-14 16:45:17 +0000 Iain * <iaingnome@gmail.com>
25307 ext/vorbis/vorbisparse.h: ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue) (vorbis_parse_push_headers, vorbi...
25308 Original commit message from CVS:
25309 2006-07-14 Andy Wingo <wingo@pobox.com>
25310 * ext/vorbis/vorbisparse.h:
25311 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
25312 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
25313 (vorbis_parse_drain_queue_prematurely, )
25314 (vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
25315 until we have initialized our state. Fixes seeking after an
25317 2006-07-14 Andy Wingo <wingo@pobox.com>
25318 Patch by: Iain * <iaingnome@gmail.com>
25319 * ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.
25321 2006-07-14 15:52:39 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25323 configure.ac: Bump nano back to CVS
25324 Original commit message from CVS:
25326 Bump nano back to CVS
25328 === release 0.10.9 ===
25330 2006-07-14 15:51:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25332 configure.ac: releasing 0.10.9, "I walk the line"
25333 Original commit message from CVS:
25334 2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
25336 releasing 0.10.9, "I walk the line"
25338 2006-07-14 14:12:40 +0000 Michael Smith <msmith@xiph.org>
25340 tests/check/pipelines/vorbisenc.c: Move a g_cond_signal to earlier to avoid sometimes deadlocking (commonly happens w...
25341 Original commit message from CVS:
25342 * tests/check/pipelines/vorbisenc.c: (stop_pipeline):
25343 Move a g_cond_signal to earlier to avoid sometimes deadlocking
25344 (commonly happens when running this test under valgrind) when trying
25345 to remove the buffer probe.
25347 2006-07-14 10:34:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25349 * gst/gdp/Makefile.am:
25350 build as a plugin, not a lib
25351 Original commit message from CVS:
25352 build as a plugin, not a lib
25354 2006-07-13 16:43:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25356 sys/xvimage/xvimagesink.c: Fix missing g_unlock from the previous commit
25357 Original commit message from CVS:
25358 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
25359 Fix missing g_unlock from the previous commit
25361 2006-07-13 16:34:04 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25363 sys/: Implement a locking order to ensure we always take the object lock before the x_lock and never vice-versa.
25364 Original commit message from CVS:
25365 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
25366 (gst_ximagesink_change_state):
25367 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
25368 (gst_xvimagesink_change_state):
25369 Implement a locking order to ensure we always take the object lock
25370 before the x_lock and never vice-versa.
25372 2006-07-13 15:25:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25374 docs/plugins/: add more plugins and elements to docs
25375 Original commit message from CVS:
25376 * docs/plugins/Makefile.am:
25377 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
25378 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
25379 add more plugins and elements to docs
25380 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
25381 fix segfaults due to wrong g_free
25383 * gst/gdp/gstgdppay.c:
25386 2006-07-13 14:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25388 gst/playback/gstdecodebin.c: Fix a caps leak when linking (#347304)
25389 Original commit message from CVS:
25390 * gst/playback/gstdecodebin.c: (find_compatibles):
25391 Fix a caps leak when linking (#347304)
25392 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
25393 (gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
25394 (gst_ximagesink_change_state):
25395 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
25396 (gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
25397 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
25398 (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
25399 Don't leak shared memory resources. Use the object lock to protect
25400 against the xcontext disappearing while returning a buffer from the
25401 pipeline. (#347304)
25403 2006-07-12 14:20:43 +0000 Edward Hervey <bilboed@bilboed.com>
25405 ext/vorbis/vorbisdec.c: gst_tag_list_merge() returns a new object. Take that into account when using it. This avoids ...
25406 Original commit message from CVS:
25407 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
25408 (vorbis_handle_comment_packet):
25409 gst_tag_list_merge() returns a new object. Take that into account when
25410 using it. This avoids memleak.
25411 Revert previous commit which is not needed.
25413 2006-07-12 13:30:20 +0000 Edward Hervey <bilboed@bilboed.com>
25415 ext/vorbis/vorbisdec.c: Reset the decoder in finalize so that all fields get cleared.
25416 Original commit message from CVS:
25417 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
25418 Reset the decoder in finalize so that all fields get cleared.
25420 2006-07-12 13:24:19 +0000 Wim Taymans <wim.taymans@gmail.com>
25422 gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
25423 Original commit message from CVS:
25424 * gst-libs/gst/audio/gstbaseaudiosrc.c:
25425 (gst_base_audio_src_set_clock),
25426 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
25427 Don't try to post an error message when setting the clock fails
25428 as this can happen when adding an element to a bin which will then
25429 deadlock. Fixes #347296.
25431 2006-07-12 13:04:15 +0000 Edward Hervey <bilboed@bilboed.com>
25433 ext/vorbis/vorbisdec.c: Post tag messages on the bus even if we're not initialized.
25434 Original commit message from CVS:
25435 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
25436 (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
25437 (vorbis_handle_type_packet):
25438 Post tag messages on the bus even if we're not initialized.
25439 If we're not initialized, we still postpone the event pushing of tags.
25441 2006-07-12 11:28:37 +0000 Wim Taymans <wim.taymans@gmail.com>
25443 Revert last two changes that broke the freeze.
25444 Original commit message from CVS:
25445 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
25446 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
25447 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
25448 Revert last two changes that broke the freeze.
25450 2006-07-12 10:59:55 +0000 Wim Taymans <wim.taymans@gmail.com>
25452 ext/alsa/gstalsasink.c: basesink calculates silence sample correctly for us.
25453 Original commit message from CVS:
25454 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
25455 basesink calculates silence sample correctly for us.
25457 2006-07-12 10:58:42 +0000 Wim Taymans <wim.taymans@gmail.com>
25459 gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
25460 Original commit message from CVS:
25461 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
25462 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
25463 Calculate correct silence samples so we don't fill our ringbuffer
25466 2006-07-12 10:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
25468 ext/vorbis/vorbisdec.*: Delay sending events (newsegment, tags) until the decoder is properly initialized.
25469 Original commit message from CVS:
25470 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
25471 (gst_vorbis_dec_reset), (vorbis_dec_sink_event),
25472 (vorbis_handle_comment_packet), (vorbis_handle_type_packet):
25473 * ext/vorbis/vorbisdec.h:
25474 Delay sending events (newsegment, tags) until the decoder is properly
25478 2006-07-11 22:40:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25495 Original commit message from CVS:
25498 2006-07-11 21:04:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25500 tests/check/elements/audioconvert.c: Patch from #347221 adding a test for audioconvert channel remappings.
25501 Original commit message from CVS:
25502 * tests/check/elements/audioconvert.c: (get_float_mc_caps),
25503 (get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
25504 Patch from #347221 adding a test for audioconvert
25505 channel remappings.
25507 2006-07-11 12:03:25 +0000 Tim-Philipp Müller <tim@centricular.net>
25509 gst/subparse/gstssaparse.c: Don't include the terminating NUL in the buffer size, it's only there for extra paranoia ...
25510 Original commit message from CVS:
25511 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
25512 (gst_ssa_parse_parse_line):
25513 Don't include the terminating NUL in the buffer size,
25514 it's only there for extra paranoia (would add random
25515 '*' characters at the end of each subtitle since the
25516 terminator itself is not valid UTF-8 technically).
25517 Also fix indenting after boilerplate macro.
25519 2006-07-10 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.net>
25521 gst/playback/gstdecodebin.c: Also emit 'unknown-type' signal (which should really be called unhandled-type) if we fou...
25522 Original commit message from CVS:
25523 * gst/playback/gstdecodebin.c: (close_pad_link):
25524 Also emit 'unknown-type' signal (which should really be
25525 called unhandled-type) if we found potential decoders/demuxers
25526 in the registry but none of them worked in the end (as in the
25527 case where the plugins don't exist any longer but are still
25528 listed in the registry). Fixes #329798.
25530 2006-07-08 13:48:58 +0000 Andy Wingo <wingo@pobox.com>
25533 * ext/theora/theoraparse.c:
25534 theoraparse.c (theora_parse_push_buffer)
25535 Original commit message from CVS:
25536 2006-07-08 Andy Wingo <wingo@pobox.com>
25537 * theoraparse.c (theora_parse_push_buffer)
25538 (theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
25539 Add some more debugging. Fix granulepos reconstruction in the face
25540 of discontinuities.
25542 2006-07-06 15:54:50 +0000 Wim Taymans <wim.taymans@gmail.com>
25544 gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
25545 Original commit message from CVS:
25546 * gst-libs/gst/audio/gstbaseaudiosink.c:
25547 (gst_base_audio_sink_class_init),
25548 (gst_base_audio_sink_provide_clock):
25549 Use gobject_class instead of G_OBJECT_CLASS (klass)
25550 * gst-libs/gst/audio/gstbaseaudiosrc.c:
25551 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
25552 (gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
25553 (gst_base_audio_src_get_time),
25554 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
25555 (gst_base_audio_src_create_ringbuffer):
25556 Fix latency and buffer-time constants and properties ala basesink.
25557 Implement pull based scheduling. Fixes #346527.
25558 Set default blocksize in GstBaseSrc to 0, we default to pushing out
25560 Refuse slaving to another clock instead of silently not working.
25561 Only provide a clock when we are actually able to do so.
25562 Various small cleanups and compiler hints.
25564 2006-07-06 13:23:07 +0000 Lutz Mueller <lutz@topfrose.de>
25566 gst/typefind/gsttypefindfunctions.c: Add typefinding for text/html (#346581).
25567 Original commit message from CVS:
25568 Patch by: Lutz Mueller <lutz at topfrose de>
25569 * gst/typefind/gsttypefindfunctions.c: (html_type_find),
25571 Add typefinding for text/html (#346581).
25573 2006-07-06 13:12:02 +0000 Tim-Philipp Müller <tim@centricular.net>
25575 gst/typefind/gsttypefindfunctions.c: Fix SMIL typefinding, make xml_check_first_element() more useful.
25576 Original commit message from CVS:
25577 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
25578 (xml_check_first_element), (xml_type_find), (smil_type_find):
25579 Fix SMIL typefinding, make xml_check_first_element() more
25582 2006-07-06 13:04:24 +0000 Tim-Philipp Müller <tim@centricular.net>
25584 gst/playback/gstplaybasebin.*: Protect list of elements with a subtitle-encoding property and the subtitle encoding m...
25585 Original commit message from CVS:
25586 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
25587 (gst_play_base_bin_finalize), (decodebin_element_added_cb),
25588 (decodebin_element_removed_cb), (gst_play_base_bin_set_property):
25589 * gst/playback/gstplaybasebin.h:
25590 Protect list of elements with a subtitle-encoding property and
25591 the subtitle encoding member itself with a lock of their own
25592 instead of using the object lock. This prevents a dead-lock in
25593 the element-remove callback in some circumstances when shutting
25596 2006-07-05 20:11:13 +0000 Sébastien Moutte <sebastien@moutte.net>
25598 win32/common/libgsttag.def: Export some new functions.
25599 Original commit message from CVS:
25600 * win32/common/libgsttag.def:
25601 Export some new functions.
25602 * win32/vs6/libgstogg.dsp:
25603 Add a link to libgsttag-0.10.lib.
25605 2006-07-04 16:50:21 +0000 Tim-Philipp Müller <tim@centricular.net>
25607 ext/alsa/gstalsamixertrack.c: Some const-ification.
25608 Original commit message from CVS:
25609 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
25610 Some const-ification.
25612 2006-07-04 14:06:03 +0000 Wim Taymans <wim.taymans@gmail.com>
25614 gst/playback/gstplaybasebin.c: Improve checking if we are dealing with a stream. Added some more uris that need buffe...
25615 Original commit message from CVS:
25616 * gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
25617 Improve checking if we are dealing with a stream. Added some
25618 more uris that need buffering.
25620 2006-07-03 10:43:31 +0000 Edward Hervey <bilboed@bilboed.com>
25622 ext/vorbis/vorbisdec.c: Remove unused variable.
25623 Original commit message from CVS:
25624 * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
25625 Remove unused variable.
25627 2006-07-02 21:48:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25629 Makefile.am: include lcov.mak
25630 Original commit message from CVS:
25634 add GCOV_LIBS to GST_LIBS
25636 2006-07-02 11:08:58 +0000 Michael Sheldon <webmaster@mikeasoft.com>
25638 ext/alsa/gstalsasrc.c: Add 32 bps to template caps and increase channels range from [1,2] to [1,MAX]. See #346326.
25639 Original commit message from CVS:
25640 Patch by: Michael Sheldon <webmaster at mikeasoft com>
25641 * ext/alsa/gstalsasrc.c:
25642 Add 32 bps to template caps and increase channels range
25643 from [1,2] to [1,MAX]. See #346326.
25645 2006-06-30 12:04:51 +0000 Tim-Philipp Müller <tim@centricular.net>
25647 gst-libs/gst/riff/riff-media.c: Recognise 'WMVA' video codec fourcc (#345879).
25648 Original commit message from CVS:
25649 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
25650 Recognise 'WMVA' video codec fourcc (#345879).
25652 2006-06-29 12:21:06 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
25654 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
25655 Original commit message from CVS:
25656 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
25657 Fixed nasty memory leak
25659 2006-06-26 13:19:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25661 gst/tcp/gsttcp.c: fix logging
25662 Original commit message from CVS:
25663 * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
25664 (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
25667 2006-06-23 16:45:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25669 gst/playback/gstdecodebin.c: Protect remove_fakesink using a mutex, so that we don't try and remove the fakesink simu...
25670 Original commit message from CVS:
25671 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
25672 (gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
25673 (remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
25674 Protect remove_fakesink using a mutex, so that we don't try and
25675 remove the fakesink simultaneously from multiple threads.
25676 When going from READY to PAUSED, restore the fakesink, so that
25677 it is there when decodebin gets reused.
25679 2006-06-23 09:53:09 +0000 Tim-Philipp Müller <tim@centricular.net>
25681 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
25682 Original commit message from CVS:
25683 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
25684 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25685 * gst-libs/gst/rtp/gstbasertppayload.c:
25686 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
25687 * gst/tcp/gstmultifdsink.c:
25688 * gst/tcp/gsttcpclientsink.c:
25689 * gst/tcp/gsttcpclientsrc.c:
25690 * gst/tcp/gsttcpserversink.c:
25691 * gst/tcp/gsttcpserversrc.c:
25692 * gst/videorate/gstvideorate.c:
25693 * gst/videotestsrc/gstvideotestsrc.c:
25694 * sys/v4l/gstv4ljpegsrc.c:
25695 * sys/v4l/gstv4lmjpegsink.c:
25696 * sys/v4l/gstv4lsrc.c:
25697 * tests/examples/seek/scrubby.c:
25698 * tests/examples/seek/seek.c:
25699 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
25701 2006-06-23 09:09:44 +0000 Tim-Philipp Müller <tim@centricular.net>
25703 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) and fix one GObject boilerplate macro.
25704 Original commit message from CVS:
25705 * ext/directfb/dfbvideosink.c:
25706 * ext/gsm/gstgsmdec.c:
25707 * ext/gsm/gstgsmenc.c:
25708 * ext/libmms/gstmms.c:
25709 * ext/neon/gstneonhttpsrc.c:
25710 * ext/theora/theoradec.c:
25711 * gst/freeze/gstfreeze.c:
25712 * gst/gdp/gstgdpdepay.c:
25713 * gst/gdp/gstgdppay.c:
25714 * sys/glsink/glimagesink.c:
25715 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
25716 and fix one GObject boilerplate macro.
25718 2006-06-22 12:13:31 +0000 Tim-Philipp Müller <tim@centricular.net>
25720 gst-libs/gst/tag/tags.c: Second field in GEnumValue shouldn't be a description, but a stringified version of the enum...
25721 Original commit message from CVS:
25722 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
25723 Second field in GEnumValue shouldn't be a description,
25724 but a stringified version of the enum value.
25726 2006-06-22 12:03:14 +0000 Wim Taymans <wim.taymans@gmail.com>
25728 sys/ximage/ximagesink.c: Avoid type checking in buffer casts.
25729 Original commit message from CVS:
25730 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
25731 (gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
25732 (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
25733 Avoid type checking in buffer casts.
25734 Avoid caps copy in buffer_alloc when we can.
25735 Use pad_peer_accept.
25737 2006-06-22 11:01:41 +0000 Tim-Philipp Müller <tim@centricular.net>
25739 gst-libs/gst/tag/tag.h: Oops, make that 'Since: 0.10.9'.
25740 Original commit message from CVS:
25741 * gst-libs/gst/tag/tag.h:
25742 Oops, make that 'Since: 0.10.9'.
25744 2006-06-22 10:55:05 +0000 Tim-Philipp Müller <tim@centricular.net>
25746 API: add GstTagImageType enum to describe images contained in image tags (#345641).
25747 Original commit message from CVS:
25748 * docs/libs/gst-plugins-base-libs-sections.txt:
25749 * gst-libs/gst/tag/tag.h:
25750 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
25751 (gst_tag_image_type_get_type):
25752 API: add GstTagImageType enum to describe images contained
25753 in image tags (#345641).
25755 2006-06-22 10:31:22 +0000 Tim-Philipp Müller <tim@centricular.net>
25757 gst/tcp/gstmultifdsink.c: Fix warnings with gst-inspect: "buffers-min" property should be of G_TYPE_INT and not G_TYP...
25758 Original commit message from CVS:
25759 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
25760 Fix warnings with gst-inspect: "buffers-min" property
25761 should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
25762 typo in property description.
25764 2006-06-22 10:10:51 +0000 Cody Russell <bratsche@gnome.org>
25766 gst/: Avoid unnecessary class cast check in class_init functions (#337747).
25767 Original commit message from CVS:
25768 Patch by: Cody Russell <bratsche at gnome org>
25769 * gst/audioresample/gstaudioresample.c:
25770 (gst_audioresample_class_init):
25771 * gst/playback/gststreamselector.c:
25772 (gst_stream_selector_class_init):
25773 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
25774 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
25775 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
25776 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
25777 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
25778 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
25779 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
25780 * gst/videotestsrc/gstvideotestsrc.c:
25781 (gst_video_test_src_class_init):
25782 * gst/volume/gstvolume.c: (gst_volume_class_init):
25783 Avoid unnecessary class cast check in class_init
25784 functions (#337747).
25786 2006-06-21 18:39:07 +0000 Tim-Philipp Müller <tim@centricular.net>
25788 ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
25789 Original commit message from CVS:
25790 * ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
25791 (gst_text_overlay_video_chain):
25792 g_markup_escape_text() REALLY doesn't like non-UTF8 input
25793 and doesn't validate its input either (and neither did
25794 textoverlay it seems). Let's do that then and fix #345206.
25796 2006-06-19 17:12:57 +0000 Wim Taymans <wim.taymans@gmail.com>
25798 gst/tcp/gstmultifdsink.*: Added shiny new burst-on-connect methods.
25799 Original commit message from CVS:
25800 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
25801 (gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
25802 (gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
25803 (gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
25804 (find_syncframe), (find_limits), (assign_value),
25805 (count_burst_unit), (gst_multi_fd_sink_new_client),
25806 (gst_multi_fd_sink_handle_client_write),
25807 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
25808 (gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
25809 (gst_multi_fd_sink_change_state):
25810 * gst/tcp/gstmultifdsink.h:
25811 Added shiny new burst-on-connect methods.
25812 Add properties to control the minimal amount of data queued.
25814 API: bytes-min property
25815 API: time-min property
25816 API: buffers-min property
25817 API: burst-unit property
25818 API: burst-value property
25819 API: add-full signal
25820 * gst/tcp/gsttcp-marshal.list:
25821 Added new marshaller code for the new signal.
25822 * tests/check/elements/multifdsink.c: (GST_START_TEST),
25823 (multifdsink_suite):
25824 Added testcases for new burst methods.
25826 2006-06-19 11:35:47 +0000 Christian Schaller <uraeus@gnome.org>
25828 * gst-plugins-base.spec.in:
25829 update for latest changes
25830 Original commit message from CVS:
25831 update for latest changes
25833 2006-06-19 09:57:50 +0000 Edward Hervey <bilboed@bilboed.com>
25835 ext/theora/theoradec.c: Implement clipping for accurate seeking.
25836 Original commit message from CVS:
25837 * ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
25838 Implement clipping for accurate seeking.
25841 2006-06-19 09:08:05 +0000 Philip Jaegenstedt <philip@lysator.liu.se>
25843 gst/videoscale/gstvideoscale.c: Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
25844 Original commit message from CVS:
25845 Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
25846 * gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
25847 (gst_video_scale_transform):
25848 Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
25850 2006-06-17 14:18:41 +0000 Tim-Philipp Müller <tim@centricular.net>
25854 Original commit message from CVS:
25857 2006-06-17 14:13:03 +0000 Tim-Philipp Müller <tim@centricular.net>
25859 configure.ac: Fix --disable-extern (can't set conditionals conditionally, #343602).
25860 Original commit message from CVS:
25862 Fix --disable-extern (can't set conditionals conditionally,
25865 2006-06-16 15:43:23 +0000 Tim-Philipp Müller <tim@centricular.net>
25867 tests/check/elements/audioresample.c: Add test case for bug #342789 fixed below.
25868 Original commit message from CVS:
25869 * tests/check/elements/audioresample.c: (test_reuse),
25870 (audioresample_suite):
25871 Add test case for bug #342789 fixed below.
25873 2006-06-16 15:17:44 +0000 Tim-Philipp Müller <tim@centricular.net>
25875 gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
25876 Original commit message from CVS:
25877 * gst/audioresample/gstaudioresample.c:
25878 (gst_audioresample_class_init), (gst_audioresample_init),
25879 (audioresample_start), (audioresample_stop),
25880 (gst_audioresample_set_property), (gst_audioresample_get_property):
25881 Implement GstBaseTransform::start and ::stop so that audioresample
25882 can clear its internal state properly and be reused insted of
25883 causing non-negotiated errors with playbin under some circumstances
25885 * tests/check/elements/audioresample.c: (setup_audioresample),
25886 (cleanup_audioresample):
25887 Need to set element state here so that ::start and ::stop are
25890 2006-06-16 13:59:29 +0000 Young-Ho Cha <ganadist@chollian.net>
25892 gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
25893 Original commit message from CVS:
25894 Patch by: Young-Ho Cha <ganadist at chollian dot net>
25895 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
25896 Parse extra data better, apparently it's right behind
25897 the normal strf header size. Fixes #343500.
25899 2006-06-16 11:04:21 +0000 Wim Taymans <wim.taymans@gmail.com>
25901 ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
25902 Original commit message from CVS:
25903 * ext/alsa/gstalsasink.c: (set_hwparams):
25904 If we fail to set the buffer_time and period_time alsa
25905 parameters, post a warning and leave alsa select a
25906 default instead of failing. Fixes #342085
25908 2006-06-16 10:30:25 +0000 Tim-Philipp Müller <tim@centricular.net>
25911 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
25912 Original commit message from CVS:
25913 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
25915 2006-06-16 10:20:10 +0000 Tim-Philipp Müller <tim@centricular.net>
25917 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.
25918 Original commit message from CVS:
25919 * docs/libs/gst-plugins-base-libs-sections.txt:
25920 * gst-libs/gst/cdda/gstcddabasesrc.h:
25921 Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
25922 out in the header file and shouldn't be listed in the docs.
25923 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
25924 Fix it so that it doesn't crash in the debug statement.
25926 2006-06-16 10:02:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25928 docs/libs/: add remaining symbols into correct setions
25929 Original commit message from CVS:
25930 * docs/libs/Makefile.am:
25931 * docs/libs/gst-plugins-base-libs-docs.sgml:
25932 * docs/libs/gst-plugins-base-libs-sections.txt:
25933 * docs/libs/gst-plugins-base-libs.types:
25934 add remaining symbols into correct setions
25935 * gst-libs/gst/audio/gstringbuffer.c:
25936 fix incomplete docs
25937 * gst-libs/gst/audio/gstringbuffer.h:
25938 comment out not yet implemented function
25939 * gst-libs/gst/floatcast/floatcast.h:
25940 * gst-libs/gst/netbuffer/gstnetbuffer.c:
25941 add short descriptions
25942 * gst-libs/gst/interfaces/propertyprobe.c:
25943 fix return value docs
25944 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
25945 simplify debug logging
25946 * gst-libs/gst/riff/riff-read.h:
25947 sync function prototype and docs
25948 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
25949 remove left over symbol
25951 2006-06-16 09:56:41 +0000 Tim-Philipp Müller <tim@centricular.net>
25953 Use GST_PLUGIN_DOCS macro in configure.ac, add
25954 Original commit message from CVS:
25957 * docs/Makefile.am:
25958 Use GST_PLUGIN_DOCS macro in configure.ac, add
25959 --enable-plugin-docs default to autogen.sh and use
25960 ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
25962 2006-06-15 15:27:49 +0000 Wim Taymans <wim.taymans@gmail.com>
25964 ext/ogg/gstoggdemux.c: Combine GstFlowReturn from the source pads to give a meaningfull result to the upstream peer o...
25965 Original commit message from CVS:
25966 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
25967 (gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
25968 (gst_ogg_demux_loop):
25969 Combine GstFlowReturn from the source pads to give a
25970 meaningfull result to the upstream peer or to stop the
25971 processing task in case of errors.
25973 2006-06-14 14:49:33 +0000 Tim-Philipp Müller <tim@centricular.net>
25975 gst/playback/gststreaminfo.c: Try GST_TAG_CODEC as fallback when extracting the codec name; more debug info.
25976 Original commit message from CVS:
25977 * gst/playback/gststreaminfo.c: (cb_probe):
25978 Try GST_TAG_CODEC as fallback when extracting the
25979 codec name; more debug info.
25981 2006-06-14 14:34:28 +0000 Tim-Philipp Müller <tim@centricular.net>
25983 ext/ogg/: Extract language tags from ogm subtitle streams, so that the subtitle menu choices are labelled correctly in
25984 Original commit message from CVS:
25985 * ext/ogg/Makefile.am:
25986 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
25987 Extract language tags from ogm subtitle streams, so that
25988 the subtitle menu choices are labelled correctly in
25989 Totem (fixes #344708).
25991 2006-06-14 09:13:54 +0000 Alessandro Decina <alessandro@nnva.org>
25993 ext/ogg/gstoggmux.c: Fix various leaks. Fixes #343699.
25994 Original commit message from CVS:
25995 Patch by: Alessandro Decina <alessandro at nnva dot org>
25996 * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
25997 (gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
25998 (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
25999 (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
26000 Fix various leaks. Fixes #343699.
26001 Add x-smoke mime type.
26003 2006-06-14 08:17:45 +0000 Tim-Philipp Müller <tim@centricular.net>
26005 gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
26006 Original commit message from CVS:
26007 * gst-libs/gst/riff/riff-ids.h:
26008 Add IDs for 'bext' chunks (see #343837).
26010 2006-06-12 12:44:38 +0000 Young-Ho Cha <ganadist@chollian.net>
26012 gst/subparse/samiparse.c: Honour font face tags in SAMI subtitles (#344503).
26013 Original commit message from CVS:
26014 Patch by: Young-Ho Cha <ganadist at chollian net>
26015 * gst/subparse/samiparse.c: (sami_context_pop_state),
26016 (handle_start_font), (end_sami_element):
26017 Honour font face tags in SAMI subtitles (#344503).
26019 2006-06-11 20:41:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26021 po/POTFILES.in: add missing files containing translatable strings
26022 Original commit message from CVS:
26024 add missing files containing translatable strings
26026 2006-06-11 19:55:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26028 docs/libs/tmpl/.cvsignore: we don't want those *.sgml files in CVS either
26029 Original commit message from CVS:
26030 * docs/libs/tmpl/.cvsignore:
26031 we don't want those *.sgml files in CVS either
26033 2006-06-11 19:44:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26036 Original commit message from CVS:
26037 * docs/libs/.cvsignore:
26038 * tests/check/elements/.cvsignore:
26039 * tests/check/libs/.cvsignore:
26042 2006-06-11 18:33:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26044 docs/libs/Makefile.am: also commiting the changed Makefile.am (added more libs to the doc-build)
26045 Original commit message from CVS:
26046 * docs/libs/Makefile.am:
26047 also commiting the changed Makefile.am (added more libs to the
26050 2006-06-11 17:08:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26052 docs/libs/: first batch of reordering things, add index & hierarchy
26053 Original commit message from CVS:
26054 * docs/libs/gst-plugins-base-libs-docs.sgml:
26055 * docs/libs/gst-plugins-base-libs-sections.txt:
26056 * docs/libs/gst-plugins-base-libs.types:
26057 first batch of reordering things, add index & hierarchy
26059 2006-06-11 14:08:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26062 * ext/alsa/Makefile.am:
26063 * ext/cdparanoia/Makefile.am:
26064 * ext/gnomevfs/Makefile.am:
26065 * ext/libvisual/Makefile.am:
26066 * ext/ogg/Makefile.am:
26067 * ext/pango/Makefile.am:
26068 * ext/theora/Makefile.am:
26069 * ext/vorbis/Makefile.am:
26070 * sys/v4l/Makefile.am:
26071 * sys/ximage/Makefile.am:
26072 * sys/xvimage/Makefile.am:
26073 further clean up build
26074 Original commit message from CVS:
26075 further clean up build
26077 2006-06-11 12:14:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26079 configure.ac: use GST_PKG_CHECK_MODULES, cleans up output
26080 Original commit message from CVS:
26082 use GST_PKG_CHECK_MODULES, cleans up output
26084 2006-06-11 12:10:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26087 * win32/common/config.h:
26089 Original commit message from CVS:
26092 2006-06-10 18:52:03 +0000 Tim-Philipp Müller <tim@centricular.net>
26094 ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
26095 Original commit message from CVS:
26096 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
26097 Add support for burn:// URIs (#343385); const-ify things a bit,
26098 use G_N_ELEMENTS instead of hard-coded array size.
26100 2006-06-10 18:25:07 +0000 Young-Ho Cha <ganadist@chollian.net>
26102 gst/subparse/samiparse.c: Fix up broken entities before passing them to libxml *sigh*. (#343303).
26103 Original commit message from CVS:
26104 Patch by: Young-Ho Cha <ganadist at chollian net>
26105 * gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
26106 Fix up broken entities before passing them to libxml *sigh*.
26109 2006-06-09 18:52:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26114 Original commit message from CVS:
26117 === release 0.10.8 ===
26119 2006-06-09 18:49:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26125 * docs/plugins/gst-plugins-base-plugins.args:
26126 * docs/plugins/inspect/plugin-adder.xml:
26127 * docs/plugins/inspect/plugin-alsa.xml:
26128 * docs/plugins/inspect/plugin-audioconvert.xml:
26129 * docs/plugins/inspect/plugin-audiorate.xml:
26130 * docs/plugins/inspect/plugin-audioresample.xml:
26131 * docs/plugins/inspect/plugin-audiotestsrc.xml:
26132 * docs/plugins/inspect/plugin-cdparanoia.xml:
26133 * docs/plugins/inspect/plugin-decodebin.xml:
26134 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
26135 * docs/plugins/inspect/plugin-gnomevfs.xml:
26136 * docs/plugins/inspect/plugin-libvisual.xml:
26137 * docs/plugins/inspect/plugin-ogg.xml:
26138 * docs/plugins/inspect/plugin-pango.xml:
26139 * docs/plugins/inspect/plugin-playbin.xml:
26140 * docs/plugins/inspect/plugin-subparse.xml:
26141 * docs/plugins/inspect/plugin-tcp.xml:
26142 * docs/plugins/inspect/plugin-theora.xml:
26143 * docs/plugins/inspect/plugin-typefindfunctions.xml:
26144 * docs/plugins/inspect/plugin-video4linux.xml:
26145 * docs/plugins/inspect/plugin-videorate.xml:
26146 * docs/plugins/inspect/plugin-videoscale.xml:
26147 * docs/plugins/inspect/plugin-videotestsrc.xml:
26148 * docs/plugins/inspect/plugin-volume.xml:
26149 * docs/plugins/inspect/plugin-vorbis.xml:
26150 * docs/plugins/inspect/plugin-ximagesink.xml:
26151 * docs/plugins/inspect/plugin-xvimagesink.xml:
26152 * win32/common/config.h:
26154 Original commit message from CVS:
26157 2006-06-07 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26159 0.10.7.2 prerelease
26160 Original commit message from CVS:
26176 * win32/common/config.h:
26177 0.10.7.2 prerelease
26179 2006-06-07 11:03:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26181 move last template doc snippets to source code and delete them
26182 Original commit message from CVS:
26183 * docs/libs/tmpl/gstaudio.sgml:
26184 * docs/libs/tmpl/gstcolorbalance.sgml:
26185 * docs/libs/tmpl/gstmixer.sgml:
26186 * docs/libs/tmpl/gstringbuffer.sgml:
26187 * docs/libs/tmpl/gsttuner.sgml:
26188 * docs/libs/tmpl/gstxoverlay.sgml:
26189 * gst-libs/gst/audio/audio.c:
26190 * gst-libs/gst/audio/gstringbuffer.c:
26191 * gst-libs/gst/interfaces/colorbalance.c:
26192 * gst-libs/gst/interfaces/mixer.c:
26193 * gst-libs/gst/interfaces/tuner.c:
26194 * gst-libs/gst/interfaces/xoverlay.c:
26195 move last template doc snippets to source code and delete them
26197 2006-06-06 16:26:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26199 * gst/gdp/gstgdppay.c:
26201 Original commit message from CVS:
26204 2006-06-06 14:39:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26206 configure.ac: enable building of GDP elements
26207 Original commit message from CVS:
26209 enable building of GDP elements
26210 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
26211 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
26212 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
26213 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
26214 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
26215 (gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
26216 (gst_gdp_pay_change_state):
26217 * gst/gdp/gstgdppay.h:
26220 2006-06-06 11:13:18 +0000 Michael Smith <msmith@xiph.org>
26222 ext/theora/theoraparse.c: Mark DELTA_UNIT on non-keyframes.
26223 Original commit message from CVS:
26224 * ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
26225 (theora_parse_drain_queue):
26226 Mark DELTA_UNIT on non-keyframes.
26228 2006-06-03 21:06:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26230 gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
26231 Original commit message from CVS:
26232 * gst-libs/gst/audio/gstbaseaudiosink.c:
26233 (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
26234 * gst-libs/gst/audio/gstbaseaudiosink.h:
26235 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
26236 (gst_ring_buffer_samples_done):
26237 * gst-libs/gst/audio/gstringbuffer.h:
26238 Document better the fact that latency_time and buffer_time are values
26239 stored in microseconds, and not the usual GStreamer nanoseconds.
26240 Change the variables (compatibly) that store them from GstClockTime
26241 to guint64 to make it more clear that they're not storing clock times.
26242 Also, remove the bogus property description that says the user can
26243 specify -1 to get the default value, since that's never been the case.
26244 When computing the default segment size for the ring buffer, make it
26245 an integer number of samples.
26246 When the sub-class indicates a delay greater than the number of
26247 samples we've written return 0 from the audio sink get_time method.
26249 2006-06-02 17:01:02 +0000 Michael Smith <msmith@xiph.org>
26251 tests/check/: Don't busy-wait in tests; this was causing test timeouts very frequently when running under valgrind.
26252 Original commit message from CVS:
26253 * tests/check/elements/audioconvert.c: (set_channel_positions),
26254 (get_float_mc_caps), (get_int_mc_caps):
26255 * tests/check/elements/audioresample.c:
26256 * tests/check/elements/audiotestsrc.c: (GST_START_TEST):
26257 * tests/check/elements/videorate.c:
26258 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
26259 * tests/check/elements/volume.c:
26260 * tests/check/elements/vorbisdec.c:
26261 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
26262 Don't busy-wait in tests; this was causing test timeouts very
26263 frequently when running under valgrind.
26265 2006-06-02 16:45:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26267 * gst/gdp/gstgdpdepay.c:
26268 * gst/gdp/gstgdppay.h:
26270 Original commit message from CVS:
26273 2006-06-02 16:35:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26275 * tests/check/elements/multifdsink.c:
26276 fail_if_can_read is racy
26277 Original commit message from CVS:
26278 fail_if_can_read is racy
26280 2006-06-02 16:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26282 gst/tcp/: make multifdsink properly deal with streamheader:
26283 Original commit message from CVS:
26285 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
26286 (gst_multi_fd_sink_remove_client_link),
26287 (gst_multi_fd_sink_client_queue_caps),
26288 (gst_multi_fd_sink_client_queue_buffer),
26289 (gst_multi_fd_sink_handle_client_write),
26290 (gst_multi_fd_sink_render):
26291 * gst/tcp/gstmultifdsink.h:
26292 make multifdsink properly deal with streamheader:
26293 - streamheader is taken from caps
26294 - buffers marked with IN_CAPS are not sent
26295 - streamheaders are sent, on connection, from the caps of the
26296 buffer where the client gets positioned to
26297 - further streamheader changes are done every time the client
26298 will receive a buffer with different caps
26299 * tests/check/elements/multifdsink.c: (GST_START_TEST),
26300 (gst_multifdsink_create_streamheader):
26303 2006-06-02 15:06:59 +0000 Michael Smith <msmith@xiph.org>
26305 ext/vorbis/vorbisdec.c: Reinstate limit on channel count. Vorbis does not define the meaning of > 6 channels, so they...
26306 Original commit message from CVS:
26307 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
26308 Reinstate limit on channel count. Vorbis does not define the meaning
26309 of > 6 channels, so they're just independent channels. Gstreamer
26310 currently has no mechanism to represent N independent channels.
26312 2006-06-02 14:23:34 +0000 Michael Smith <msmith@xiph.org>
26314 ext/vorbis/vorbisdec.c: Don't arbitrarily restrict channel counts and rate in vorbis.
26315 Original commit message from CVS:
26316 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
26317 Don't arbitrarily restrict channel counts and rate in vorbis.
26318 In terms of effects likely on real-world files, this fixes 96kHz
26319 playback of vorbis.
26321 2006-06-02 14:19:18 +0000 Michael Smith <msmith@xiph.org>
26323 gst/audioconvert/audioconvert.c: More correct float->int conversion.
26324 Original commit message from CVS:
26325 * gst/audioconvert/audioconvert.c: (float):
26326 More correct float->int conversion.
26328 2006-06-02 14:07:42 +0000 Michael Smith <msmith@xiph.org>
26330 ext/ogg/gstoggdemux.c: Don't accidently send GST_CLOCK_TIME_NONE as a new segment start value. Fixes g-critical on tr...
26331 Original commit message from CVS:
26332 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
26333 Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
26334 value. Fixes g-critical on trying to play back ogg containing
26337 2006-06-02 10:34:12 +0000 Wim Taymans <wim.taymans@gmail.com>
26339 gst/playback/gstplaybasebin.*: Make the subtitle detection work from any thread so we don't deadlock. Fixes #343397.
26340 Original commit message from CVS:
26341 * gst/playback/gstplaybasebin.c: (group_create), (group_commit),
26343 * gst/playback/gstplaybasebin.h:
26344 Make the subtitle detection work from any thread so we don't
26345 deadlock. Fixes #343397.
26347 2006-06-02 10:28:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26349 gst/gdp/gstgdppay.c: add crc-header and crc-payload properties don't error out on some things that are recoverable
26350 Original commit message from CVS:
26351 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
26352 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
26353 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
26354 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
26355 (gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
26356 (gst_gdp_pay_get_property):
26357 add crc-header and crc-payload properties
26358 don't error out on some things that are recoverable
26359 * tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
26362 2006-06-02 09:17:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26364 * gst/tcp/gsttcp.c:
26365 show type number when packet is of the wrong type
26366 Original commit message from CVS:
26367 show type number when packet is of the wrong type
26369 2006-06-01 23:04:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26371 gst/volume/Makefile.am: Seriously, it's not *that* hard to get compilation right. Even a drunk can do it ! Add LIBOI...
26372 Original commit message from CVS:
26373 * gst/volume/Makefile.am:
26374 Seriously, it's not *that* hard to get compilation right. Even
26375 a drunk can do it ! Add LIBOIL CFLAGS and LIBS
26377 2006-06-01 22:00:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26379 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26380 Original commit message from CVS:
26381 * ext/alsaspdif/alsaspdifsink.h:
26382 * ext/amrwb/gstamrwbdec.h:
26383 * ext/amrwb/gstamrwbenc.h:
26384 * ext/amrwb/gstamrwbparse.h:
26385 * ext/arts/gst_arts.h:
26386 * ext/artsd/gstartsdsink.h:
26387 * ext/audiofile/gstafparse.h:
26388 * ext/audiofile/gstafsink.h:
26389 * ext/audiofile/gstafsrc.h:
26390 * ext/audioresample/gstaudioresample.h:
26391 * ext/bz2/gstbz2dec.h:
26392 * ext/bz2/gstbz2enc.h:
26393 * ext/dirac/gstdiracdec.h:
26394 * ext/directfb/dfbvideosink.h:
26395 * ext/divx/gstdivxdec.h:
26396 * ext/divx/gstdivxenc.h:
26397 * ext/dts/gstdtsdec.h:
26398 * ext/faac/gstfaac.h:
26399 * ext/gsm/gstgsmdec.h:
26400 * ext/gsm/gstgsmenc.h:
26401 * ext/ivorbis/vorbisenc.h:
26402 * ext/libfame/gstlibfame.h:
26403 * ext/nas/nassink.h:
26404 * ext/neon/gstneonhttpsrc.h:
26405 * ext/polyp/polypsink.h:
26406 * ext/sdl/sdlaudiosink.h:
26407 * ext/sdl/sdlvideosink.h:
26408 * ext/shout/gstshout.h:
26409 * ext/snapshot/gstsnapshot.h:
26410 * ext/sndfile/gstsf.h:
26411 * ext/swfdec/gstswfdec.h:
26412 * ext/tarkin/gsttarkindec.h:
26413 * ext/tarkin/gsttarkinenc.h:
26414 * ext/theora/theoradec.h:
26415 * ext/wavpack/gstwavpackdec.h:
26416 * ext/wavpack/gstwavpackparse.h:
26417 * ext/xine/gstxine.h:
26418 * ext/xvid/gstxviddec.h:
26419 * ext/xvid/gstxvidenc.h:
26420 * gst/cdxaparse/gstcdxaparse.h:
26421 * gst/cdxaparse/gstcdxastrip.h:
26422 * gst/colorspace/gstcolorspace.h:
26423 * gst/festival/gstfestival.h:
26424 * gst/freeze/gstfreeze.h:
26425 * gst/gdp/gstgdpdepay.h:
26426 * gst/gdp/gstgdppay.h:
26427 * gst/modplug/gstmodplug.h:
26428 * gst/mpeg1sys/gstmpeg1systemencode.h:
26429 * gst/mpeg1videoparse/gstmp1videoparse.h:
26430 * gst/mpeg2sub/gstmpeg2subt.h:
26431 * gst/mpegaudioparse/gstmpegaudioparse.h:
26432 * gst/multifilesink/gstmultifilesink.h:
26433 * gst/overlay/gstoverlay.h:
26434 * gst/playondemand/gstplayondemand.h:
26435 * gst/qtdemux/qtdemux.h:
26436 * gst/rtjpeg/gstrtjpegdec.h:
26437 * gst/rtjpeg/gstrtjpegenc.h:
26438 * gst/smooth/gstsmooth.h:
26439 * gst/smoothwave/gstsmoothwave.h:
26440 * gst/spectrum/gstspectrum.h:
26441 * gst/speed/gstspeed.h:
26442 * gst/stereo/gststereo.h:
26443 * gst/switch/gstswitch.h:
26444 * gst/tta/gstttadec.h:
26445 * gst/tta/gstttaparse.h:
26446 * gst/videodrop/gstvideodrop.h:
26447 * gst/xingheader/gstxingmux.h:
26448 * sys/directdraw/gstdirectdrawsink.h:
26449 * sys/directsound/gstdirectsoundsink.h:
26450 * sys/dxr3/dxr3audiosink.h:
26451 * sys/dxr3/dxr3spusink.h:
26452 * sys/dxr3/dxr3videosink.h:
26453 * sys/qcam/gstqcamsrc.h:
26454 * sys/vcd/vcdsrc.h:
26455 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26457 2006-06-01 20:39:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26459 gst/volume/gstvolume.*: rewrite the passthrough check, split _int16 and _int16_clamp, fix another property desc., rem...
26460 Original commit message from CVS:
26461 * gst/volume/gstvolume.c: (volume_choose_func),
26462 (volume_update_real_volume), (gst_volume_class_init),
26463 (gst_volume_init), (volume_process_float), (volume_process_int16),
26464 (volume_process_int16_clamp), (volume_set_caps),
26465 (volume_transform_ip), (plugin_init):
26466 * gst/volume/gstvolume.h:
26467 rewrite the passthrough check, split _int16 and _int16_clamp, fix
26468 another property desc., remove unused param from process function
26469 * tests/check/elements/volume.c: (volume_suite):
26470 reactivate the passthrough test
26472 2006-06-01 19:19:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26474 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26475 Original commit message from CVS:
26476 * ext/alsa/gstalsamixerelement.h:
26477 * ext/alsa/gstalsamixeroptions.h:
26478 * ext/alsa/gstalsamixertrack.h:
26479 * ext/gnomevfs/gstgnomevfssink.h:
26480 * ext/gnomevfs/gstgnomevfssrc.h:
26481 * ext/theora/gsttheoradec.h:
26482 * ext/theora/gsttheoraenc.h:
26483 * ext/theora/gsttheoraparse.h:
26484 * ext/vorbis/vorbisparse.h:
26485 * gst-libs/gst/audio/gstaudioclock.h:
26486 * gst-libs/gst/audio/gstaudiofilter.h:
26487 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
26488 * gst/audioconvert/gstaudioconvert.h:
26489 * gst/audioresample/gstaudioresample.h:
26490 * gst/audiotestsrc/gstaudiotestsrc.h:
26491 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
26492 * gst/playback/gststreamselector.h:
26493 * gst/tcp/gstmultifdsink.h:
26494 * gst/tcp/gsttcpclientsink.h:
26495 * gst/tcp/gsttcpclientsrc.h:
26496 * gst/tcp/gsttcpserversink.h:
26497 * gst/tcp/gsttcpserversrc.h:
26498 * gst/videorate/gstvideorate.h:
26499 * gst/videoscale/gstvideoscale.h:
26500 * gst/videotestsrc/gstvideotestsrc.h:
26501 * gst/volume/gstvolume.h:
26502 * sys/v4l/gstv4ljpegsrc.h:
26503 * sys/v4l/gstv4lmjpegsink.h:
26504 * sys/v4l/gstv4lmjpegsrc.h:
26505 * sys/v4l/gstv4lsrc.h:
26506 * sys/ximage/ximagesink.h:
26507 * sys/xvimage/xvimagesink.h:
26508 * tests/old/testsuite/alsa/sinesrc.h:
26509 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26511 2006-05-31 16:56:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26513 * tests/check/elements/multifdsink.c:
26514 remove wrong commit
26515 Original commit message from CVS:
26516 remove wrong commit
26518 2006-05-31 16:21:48 +0000 Wim Taymans <wim.taymans@gmail.com>
26520 ext/libvisual/visual.c: Handle DISCONT.
26521 Original commit message from CVS:
26522 * ext/libvisual/visual.c: (gst_visual_reset),
26523 (gst_visual_sink_setcaps), (gst_visual_sink_event),
26524 (gst_visual_src_event), (get_buffer), (gst_visual_chain):
26526 Use running time before doing QoS.
26529 2006-05-31 14:17:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26531 docs/libs/Makefile.am: set a magic variable to indicate we know the docs are incomplete
26532 Original commit message from CVS:
26533 * docs/libs/Makefile.am:
26534 set a magic variable to indicate we know the docs are incomplete
26536 2006-05-30 20:33:59 +0000 Sébastien Moutte <sebastien@moutte.net>
26538 win32/common/libgstvideo.def: export gst_video_calculate_display_ratio
26539 Original commit message from CVS:
26540 * win32/common/libgstvideo.def:
26541 export gst_video_calculate_display_ratio
26542 * win32/vs6/libgstvideoscale.dsp:
26543 add link to libgstvideo-0.10.lib
26545 2006-05-30 19:00:39 +0000 Tim-Philipp Müller <tim@centricular.net>
26547 gst/playback/gstplaybasebin.c: Throw a more comprehensible error for rtsp:// URIs (rather than erroring out with a ne...
26548 Original commit message from CVS:
26549 * gst/playback/gstplaybasebin.c: (gen_source_element):
26550 Throw a more comprehensible error for rtsp:// URIs (rather
26551 than erroring out with a negotiation error later on) until
26552 we fix playbin to handle rtspsrc etc.
26554 2006-05-30 16:09:36 +0000 Wim Taymans <wim.taymans@gmail.com>
26556 ext/pango/gsttextoverlay.c: Added some FIXMEs.
26557 Original commit message from CVS:
26558 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
26559 (gst_text_overlay_text_event):
26562 2006-05-30 16:07:50 +0000 Wim Taymans <wim.taymans@gmail.com>
26564 gst/adder/gstadder.*: Implement release_request_pad.
26565 Original commit message from CVS:
26566 * gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
26567 (gst_adder_request_new_pad), (gst_adder_release_pad):
26568 * gst/adder/gstadder.h:
26569 Implement release_request_pad.
26570 Make padcounter atomic.
26571 * tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
26572 Added check for release_pad in adder.
26574 2006-05-30 16:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
26576 ext/ogg/gstoggdemux.c: Fix build again.
26577 Original commit message from CVS:
26578 * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
26581 2006-05-30 14:59:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26583 ext/ogg/gstoggdemux.c: add more debugging clean up printf formats for granulepos and serialno
26584 Original commit message from CVS:
26585 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
26586 (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
26587 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
26588 (gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
26589 (gst_ogg_demux_seek), (gst_ogg_demux_get_data),
26590 (gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
26591 (gst_ogg_demux_bisect_forward_serialno),
26592 (gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
26593 (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
26595 clean up printf formats for granulepos and serialno
26597 2006-05-30 14:31:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26599 * tests/check/elements/multifdsink.c:
26600 * tests/check/generic/states.c:
26601 properly fail if we can't make an element
26602 Original commit message from CVS:
26603 properly fail if we can't make an element
26605 2006-05-30 13:22:58 +0000 Michael Smith <msmith@xiph.org>
26607 ext/vorbis/vorbisenc.*: Multi-channel caps negotiation, so we can do proper multichannel vorbis encoding, negotiated ...
26608 Original commit message from CVS:
26609 * ext/vorbis/vorbisenc.c: (raw_caps_factory),
26610 (gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
26611 (gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
26612 (gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
26613 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
26614 * ext/vorbis/vorbisenc.h:
26615 Multi-channel caps negotiation, so we can do proper multichannel
26616 vorbis encoding, negotiated through audioconvert.
26618 2006-05-30 11:45:52 +0000 Wim Taymans <wim.taymans@gmail.com>
26620 tests/check/elements/adder.c: Added check to show that #339935 is fixed with ongoing adder and collectpads fixes.
26621 Original commit message from CVS:
26622 * tests/check/elements/adder.c: (test_event_message_received),
26623 (test_play_twice_message_received), (GST_START_TEST),
26625 Added check to show that #339935 is fixed with ongoing
26626 adder and collectpads fixes.
26628 2006-05-29 17:19:48 +0000 Wim Taymans <wim.taymans@gmail.com>
26630 gst/adder/gstadder.c: Don't leak pad name.
26631 Original commit message from CVS:
26632 * gst/adder/gstadder.c: (gst_adder_request_new_pad):
26633 Don't leak pad name.
26635 2006-05-29 15:49:53 +0000 Wim Taymans <wim.taymans@gmail.com>
26637 gst/adder/gstadder.c: Fix adder seeking.
26638 Original commit message from CVS:
26639 * gst/adder/gstadder.c: (gst_adder_query_duration),
26640 (forward_event_func), (forward_event), (gst_adder_src_event):
26642 Make query/seeking code threadsafe.
26643 * tests/check/Makefile.am:
26644 * tests/check/elements/adder.c: (test_event_message_received),
26645 (GST_START_TEST), (test_play_twice_message_received):
26646 Fix adder test case.
26648 2006-05-29 13:21:00 +0000 Young-Ho Cha <ganadist@chollian.net>
26650 gst/playback/gstplaybasebin.*: Add 'subtitle-encoding' property to playbin, so applications can force a subtitle enco...
26651 Original commit message from CVS:
26652 Patch by: Young-Ho Cha <ganadist at chollian net>
26653 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
26654 (gst_play_base_bin_init), (gst_play_base_bin_dispose),
26655 (set_encoding_element), (decodebin_element_added_cb),
26656 (decodebin_element_removed_cb), (setup_subtitle), (setup_source),
26657 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
26658 * gst/playback/gstplaybasebin.h:
26659 Add 'subtitle-encoding' property to playbin, so applications can
26660 force a subtitle encoding for non-UTF8 subtitles (#342268).
26661 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
26662 (gst_sub_parse_set_property):
26663 Rename recently-added 'encoding' property to 'subtitle-encoding'
26664 (so it can be proxied by playbin/decodebin in a generic way
26665 with less danger of false positives).
26667 2006-05-29 11:04:48 +0000 Michael Smith <msmith@xiph.org>
26669 gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf...
26670 Original commit message from CVS:
26671 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
26672 (append_with_other_format), (set_structure_widths),
26673 (gst_audio_convert_transform_caps):
26674 Patch from #341562: give more specific audio caps in get_caps, so
26675 that basetransform can make better decisions on what caps to
26678 2006-05-28 20:04:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26680 tests/check/elements/volume.c: make it compile again
26681 Original commit message from CVS:
26682 * tests/check/elements/volume.c:
26683 make it compile again
26685 2006-05-28 19:56:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26687 tests/check/elements/volume.c: disable test until #343196 gets resolved
26688 Original commit message from CVS:
26689 * tests/check/elements/volume.c: (volume_suite):
26690 disable test until #343196 gets resolved
26692 2006-05-28 19:42:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26694 gst/adder/gstadder.c: Make it easier to copy&paste
26695 Original commit message from CVS:
26696 * gst/adder/gstadder.c: (gst_adder_get_type):
26697 Make it easier to copy&paste
26698 * gst/volume/Makefile.am:
26699 * gst/volume/gstvolume.c: (volume_update_real_volume),
26700 (gst_volume_set_volume), (gst_volume_set_mute),
26701 (gst_volume_class_init), (volume_process_int16), (volume_set_caps),
26702 (volume_transform_ip), (volume_update_mute),
26703 (volume_update_volume):
26704 * gst/volume/gstvolume.h:
26705 Add own debug category, move duplicate code to helper function, fix
26706 property texts, add more comments and prepare ffor liboil-goodness
26707 * tests/check/Makefile.am:
26708 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
26709 add test for mute and passtrough case, be a bit more verbose to track
26711 * tests/check/generic/states.c: (GST_START_TEST):
26712 catch elements that fail to instantiate
26714 2006-05-28 09:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
26716 tests/check/pipelines/: Comment out tests using parse_launch() if core was built without parsing capabilities.
26717 Original commit message from CVS:
26718 * tests/check/pipelines/simple-launch-lines.c:
26719 * tests/check/pipelines/theoraenc.c:
26720 * tests/check/pipelines/vorbisenc.c:
26721 Comment out tests using parse_launch() if core was built without
26722 parsing capabilities.
26724 2006-05-27 13:34:03 +0000 Edward Hervey <bilboed@bilboed.com>
26726 tests/check/Makefile.am: Extra bonus points for whoever explains to ensonic that you are meant to test unit tests tho...
26727 Original commit message from CVS:
26728 * tests/check/Makefile.am:
26729 Extra bonus points for whoever explains to ensonic that you are meant
26730 to test unit tests thoroughly before commiting them, especially if
26731 you know it's going to break.
26732 De-activated element/adder tests.
26734 2006-05-27 13:09:16 +0000 Edward Hervey <bilboed@bilboed.com>
26736 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Marking caps conversion issues as GST_WARNING is way too verbose,
26737 Original commit message from CVS:
26738 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
26739 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
26740 Marking caps conversion issues as GST_WARNING is way too verbose,
26741 Moving them to GST_LOG.
26743 2006-05-27 11:26:18 +0000 Tim-Philipp Müller <tim@centricular.net>
26745 README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from...
26746 Original commit message from CVS:
26748 Replace current README (containing the release notes from
26749 some 0.9.x version) with a proper README taken from the core.
26751 2006-05-26 15:52:23 +0000 Wim Taymans <wim.taymans@gmail.com>
26753 ext/vorbis/vorbisdec.c: Small cleanups.
26754 Original commit message from CVS:
26755 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
26756 (vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
26757 (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
26758 (vorbis_dec_change_state):
26761 Clip output samples to segment boundaries.
26763 2006-05-26 11:17:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26765 sys/ximage/ximagesink.c: Improve the errors produced on bad output, including some human readable description strings.
26766 Original commit message from CVS:
26767 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
26768 (gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
26769 Improve the errors produced on bad output, including some human
26770 readable description strings.
26771 Handle the (theoretical for ximagesink) case where the XServer
26772 has a different idea about the size required for a particular
26773 frame and gives us too small a memory allocation.
26775 2006-05-26 10:18:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26778 Mention bugs fixed by previous commit
26779 Original commit message from CVS:
26780 Mention bugs fixed by previous commit
26782 2006-05-26 09:40:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26784 sys/xvimage/xvimagesink.c: Improve the errors produced on bad output, including some human readable description strings.
26785 Original commit message from CVS:
26786 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
26787 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
26788 (gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
26789 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
26790 Improve the errors produced on bad output, including some human
26791 readable description strings.
26792 Handle RGB Xv formats properly by transforming them into our
26793 big-endian caps description.
26794 Use gst_caps_truncate to ensure that we never try and choose a
26795 non-fixed caps in buffer_alloc.
26796 Handle the case where the XServer has a different idea about the size
26797 required for a particular frame and gives us too small a memory
26799 Use -1 to indicate 'no image format', because 0 is a valid XServer
26800 image format number.
26801 Put RGB Xv formats at the end of the caps, so that we always prefer
26803 Iterate the available Xv Encodings to determine the maximum width and
26804 height, and then return that in our caps.
26806 2006-05-25 16:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26808 gst/playback/gstdecodebin.c: When there is only one unfinished pad and it receives an event that doesn't match our re...
26809 Original commit message from CVS:
26810 * gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
26811 When there is only one unfinished pad and it receives an event that
26812 doesn't match our requirements, we need to set alldone=FALSE so that
26813 the fakesink is not removed yet.
26815 2006-05-25 09:32:31 +0000 Tim-Philipp Müller <tim@centricular.net>
26817 ext/ogg/gstoggdemux.c: Use gst_type_find_helper_for_buffer() to find the type of stream from the first packet.
26818 Original commit message from CVS:
26819 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
26820 Use gst_type_find_helper_for_buffer() to find the type
26821 of stream from the first packet.
26823 Bump requirements to core CVS (needed for vorbis
26824 typefinding to work).
26826 2006-05-24 08:34:53 +0000 Edward Hervey <bilboed@bilboed.com>
26828 gst/typefind/gsttypefindfunctions.c: Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
26829 Original commit message from CVS:
26830 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
26831 Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
26832 Else they play perfectly fine with qtdemux.
26834 2006-05-23 20:38:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26836 make more debug catagories static
26837 Original commit message from CVS:
26838 * ext/theora/theoradec.c:
26839 * ext/theora/theoraenc.c:
26840 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
26841 * gst/audiorate/gstaudiorate.c:
26842 make more debug catagories static
26843 * tests/check/Makefile.am:
26844 * tests/check/elements/adder.c: (message_received),
26845 (test_event_message_received), (GST_START_TEST),
26846 (test_play_twice_message_received), (adder_suite):
26847 added test case for using element twice, extra bonus points for anyone
26848 who can make these test run reliably
26850 2006-05-23 15:18:40 +0000 Tim-Philipp Müller <tim@centricular.net>
26852 ext/theora/theoradec.c: Make work with time-stamped input buffers that do not have a granulepos in BUFFER_OFFSET_END ...
26853 Original commit message from CVS:
26854 * ext/theora/theoradec.c: (theora_dec_chain):
26855 Make work with time-stamped input buffers that do not
26856 have a granulepos in BUFFER_OFFSET_END (like theora
26857 buffers coming from matroskademux). Fixes #342448.
26859 2006-05-22 15:53:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26861 gst/gdp/: Handle error cases when calling functions do downwards state change after parent's change_state
26862 Original commit message from CVS:
26863 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain),
26864 (gst_gdp_depay_change_state):
26865 * gst/gdp/gstgdpdepay.h:
26866 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader),
26867 (gst_gdp_pay_chain), (gst_gdp_pay_sink_event),
26868 (gst_gdp_pay_change_state):
26869 * gst/gdp/gstgdppay.h:
26870 Handle error cases when calling functions
26871 do downwards state change after parent's change_state
26872 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
26873 * tests/check/elements/gdppay.c: (GST_START_TEST):
26876 2006-05-22 13:25:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26878 adding GDP payloader and depayloader. Build integration will follow later when the GDP issues for core are sorted out.
26879 Original commit message from CVS:
26880 * gst/gdp/Makefile.am:
26881 * gst/gdp/gstgdp.c: (plugin_init):
26882 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_base_init),
26883 (gst_gdp_depay_class_init), (gst_gdp_depay_init),
26884 (gst_gdp_depay_finalize), (gst_gdp_depay_chain),
26885 (gst_gdp_depay_change_state), (gst_gdp_depay_plugin_init):
26886 * gst/gdp/gstgdpdepay.h:
26887 * gst/gdp/gstgdppay.c: (gst_gdp_pay_base_init),
26888 (gst_gdp_pay_class_init), (gst_gdp_pay_init),
26889 (gst_gdp_pay_dispose), (gst_gdp_stamp_buffer),
26890 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
26891 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
26892 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
26893 (gst_gdp_pay_sink_event), (gst_gdp_pay_change_state),
26894 (gst_gdp_pay_plugin_init):
26895 * gst/gdp/gstgdppay.h:
26896 * tests/check/Makefile.am:
26897 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
26898 (cleanup_gdpdepay), (gdpdepay_push_per_byte), (GST_START_TEST),
26899 (setup_gdpdepay_streamheader), (gdpdepay_suite), (main):
26900 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay),
26901 (GST_START_TEST), (setup_gdppay_streamheader), (gdppay_suite),
26903 adding GDP payloader and depayloader. Build integration will
26904 follow later when the GDP issues for core are sorted out.
26906 2006-05-22 11:42:03 +0000 Peter Kjellerstedt <pkj@axis.com>
26908 gst/tcp/Makefile.am: fdstresstest doesn't need Gtk+, fix compilation if gtk is not available (#342566).
26909 Original commit message from CVS:
26910 Patch by: Peter Kjellerstedt <pkj at axis com>
26911 * gst/tcp/Makefile.am:
26912 fdstresstest doesn't need Gtk+, fix compilation if
26913 gtk is not available (#342566).
26915 2006-05-19 17:57:56 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
26917 gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
26918 Original commit message from CVS:
26919 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
26921 Removed redundant floor()
26923 2006-05-19 15:00:43 +0000 Tim-Philipp Müller <tim@centricular.net>
26925 gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
26926 Original commit message from CVS:
26927 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
26928 On second thought, just skip JUNK chunks automatically, so
26929 the caller doesn't have to handle this. Fixes #342345.
26930 Also, return GST_FLOW_UNEXPECTED if we get a short read,
26931 not GST_FLOW_ERROR.
26933 2006-05-19 13:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
26935 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...
26936 Original commit message from CVS:
26937 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
26938 Don't bail out on JUNK chunks with a size of 0 (would try to
26939 pull_range 0 bytes before, which sources don't like too much).
26942 2006-05-19 13:02:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26944 Use the gstutil scaling function to preserve 64 bits while calculating output width and height from the display-aspec...
26945 Original commit message from CVS:
26946 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
26947 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
26948 Use the gstutil scaling function to preserve 64 bits while calculating
26949 output width and height from the display-aspect-ratio. (A continuation
26952 2006-05-19 11:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26954 sys/xvimage/xvimagesink.*: When performing buffer allocations, remember the caps and image format we return so that i...
26955 Original commit message from CVS:
26956 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
26957 (gst_xvimagesink_buffer_alloc):
26958 * sys/xvimage/xvimagesink.h:
26959 When performing buffer allocations, remember the caps and image format
26960 we return so that if the same caps are asked for next time we can
26961 return them immediately without doing any caps intersections.
26963 2006-05-18 23:00:02 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
26965 gst-libs/gst/rtp/README: Some new documentation
26966 Original commit message from CVS:
26967 2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
26968 * gst-libs/gst/rtp/README:
26969 Some new documentation
26970 * gst-libs/gst/rtp/gstrtpbuffer.h:
26971 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
26972 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
26973 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
26974 New RTP audio base payloader class. Supports frame or sample based codecs.
26975 Not enabled in Makefile.am until approved.
26977 2006-05-18 20:30:26 +0000 Tim-Philipp Müller <tim@centricular.net>
26979 tests/check/elements/alsa.c: Fix test case: don't try to free NULL GValueArray when there are no devices.
26980 Original commit message from CVS:
26981 * tests/check/elements/alsa.c: (test_device_property_probe):
26982 Fix test case: don't try to free NULL GValueArray when there
26985 2006-05-18 19:21:53 +0000 Tim-Philipp Müller <tim@centricular.net>
26987 tests/check/: Add simple test that runs a device property probe on alsasrc, alsasink and alsamixer. Disable valgrind ...
26988 Original commit message from CVS:
26989 * tests/check/Makefile.am:
26990 * tests/check/elements/alsa.c: (test_device_property_probe),
26991 (alsa_suite), (main):
26992 Add simple test that runs a device property probe on alsasrc,
26993 alsasink and alsamixer. Disable valgrind check for now (too
26994 many leaks in libasound, and valgrind ignored my suppressions
26997 2006-05-18 17:19:39 +0000 Martin Szulecki <gnomebugzilla@sukimashita.com>
26999 ext/alsa/: Clean up and simplify alsa device probing. Make it actually work for multiple classes. Don't cache results...
27000 Original commit message from CVS:
27001 * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
27002 (gst_alsa_device_property_probe_probe_property),
27003 (gst_alsa_device_property_probe_needs_probe),
27004 (gst_alsa_device_property_probe_get_values),
27005 (gst_alsa_type_add_device_property_probe_interface):
27006 * ext/alsa/gstalsadeviceprobe.h:
27007 * ext/alsa/gstalsamixerelement.c:
27008 (gst_alsa_mixer_element_init_interfaces):
27009 * ext/alsa/gstalsamixerelement.h:
27010 Clean up and simplify alsa device probing. Make it actually work
27011 for multiple classes. Don't cache results any longer.
27012 * ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
27013 (gst_alsasink_init):
27014 * ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
27015 (gst_alsasrc_interface_supported), (gst_implements_interface_init),
27016 (gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
27017 Make alsasink and alsasrc implement the GstPropertyProbe interface
27018 for device probing (#342181).
27019 Patch by: Martin Szulecki <gnomebugzilla at sukimashita com>
27021 2006-05-18 10:05:23 +0000 Tim-Philipp Müller <tim@centricular.net>
27023 gst/subparse/samiparse.c: Don't ignore return value of strtol (++compiler_happiness).
27024 Original commit message from CVS:
27025 * gst/subparse/samiparse.c: (handle_start_font):
27026 Don't ignore return value of strtol (++compiler_happiness).
27028 2006-05-17 17:49:10 +0000 Young-Ho Cha <ganadist@chollian.net>
27030 gst/subparse/gstsubparse.*: Add 'encoding' property (#341681).
27031 Original commit message from CVS:
27032 Patch by: Young-Ho Cha <ganadist chollian net>
27033 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
27034 (gst_sub_parse_class_init), (gst_sub_parse_init),
27035 (gst_sub_parse_set_property), (gst_sub_parse_get_property),
27036 (convert_encoding):
27037 * gst/subparse/gstsubparse.h:
27038 Add 'encoding' property (#341681).
27039 * gst/subparse/samiparse.c: (characters_sami):
27040 Output is pango markup, so we need to escape text
27041 between tags (#342143).
27043 2006-05-16 17:34:14 +0000 Tim-Philipp Müller <tim@centricular.net>
27045 gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
27046 Original commit message from CVS:
27047 * gst-libs/gst/audio/multichannel.c:
27048 (gst_audio_check_channel_positions):
27049 It's okay to have caps with channels=1 and a channel position
27050 different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
27051 (deinterleavers might want to keep the position in the caps,
27052 so that they can be re-interleaved again properly later).
27053 Leave check for unexpected 2-channel layouts intact for now.
27055 2006-05-16 16:28:10 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
27057 gst/tcp/gsttcp.c: Return GST_FLOW_UNEXPECTED when we have an eos on the socket so basesrc can do its job correctly.
27058 Original commit message from CVS:
27059 2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
27060 * gst/tcp/gsttcp.c: (gst_tcp_socket_read):
27061 Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
27062 basesrc can do its job correctly.
27064 2006-05-16 15:52:17 +0000 Tim-Philipp Müller <tim@centricular.net>
27066 ext/alsa/: Refactor and improve caps probing code: probe signedness when we probe the supported formats/widths; set e...
27067 Original commit message from CVS:
27068 * ext/alsa/Makefile.am:
27069 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
27070 (gst_alsa_detect_formats), (get_channel_free_structure),
27071 (caps_add_channel_configuration), (gst_alsa_detect_channels),
27072 (gst_alsa_probe_supported_formats):
27073 * ext/alsa/gstalsa.h:
27074 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
27075 Refactor and improve caps probing code: probe signedness
27076 when we probe the supported formats/widths; set endianness
27077 to the one we actually probed for (ie. cpu endianness).
27078 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
27079 (gst_alsasrc_close):
27080 * ext/alsa/gstalsasrc.h:
27081 Implement caps probing for alsasrc.
27083 2006-05-15 17:42:19 +0000 Wim Taymans <wim.taymans@gmail.com>
27085 ext/theora/theoradec.c: Cleanups, add some G_LIKELY.
27086 Original commit message from CVS:
27087 * ext/theora/theoradec.c: (gst_theora_dec_reset),
27088 (theora_dec_src_query), (theora_dec_src_event),
27089 (theora_dec_sink_event), (theora_handle_comment_packet),
27090 (theora_handle_data_packet), (theora_dec_change_state):
27091 Cleanups, add some G_LIKELY.
27092 Use segment helpers instead of our own wrong code.
27093 Clear queued buffers on seek and READY.
27094 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
27095 (vorbis_dec_convert), (vorbis_dec_src_query),
27096 (vorbis_dec_src_event), (vorbis_dec_sink_event),
27097 (vorbis_handle_comment_packet), (vorbis_dec_push),
27098 (vorbis_handle_data_packet), (vorbis_dec_chain),
27099 (vorbis_dec_change_state):
27100 * ext/vorbis/vorbisdec.h:
27101 Remove old useless packetno variable.
27102 Do position query properly.
27104 Do cleanup of queued buffers in new helper function
27107 2006-05-15 17:17:22 +0000 Tim-Philipp Müller <tim@centricular.net>
27109 ext/alsa/gstalsasink.c: Query supported sample rates. Fixes #341732.
27110 Original commit message from CVS:
27111 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
27112 Query supported sample rates. Fixes #341732.
27114 2006-05-15 17:01:02 +0000 Julien Moutte <julien@moutte.net>
27116 gst/playback/gstdecodebin.c: Make decodebin reusable when going from PAUSE_TO_READY and then back to PAUSED.
27117 Original commit message from CVS:
27118 2006-05-15 Julien MOUTTE <julien@moutte.net>
27119 * gst/playback/gstdecodebin.c: (cleanup_decodebin),
27120 (gst_decode_bin_change_state): Make decodebin reusable
27121 when going from PAUSE_TO_READY and then back to PAUSED.
27124 2006-05-15 16:49:31 +0000 Wim Taymans <wim.taymans@gmail.com>
27126 ext/vorbis/vorbisdec.c: Cleanups. Use refcounting and DEBUG_OBJECT.
27127 Original commit message from CVS:
27128 * ext/vorbis/vorbisdec.c: (vorbis_get_query_types),
27129 (vorbis_dec_convert), (vorbis_dec_src_query),
27130 (vorbis_dec_sink_query), (vorbis_dec_src_event),
27131 (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
27132 (vorbis_dec_clean_queued), (vorbis_dec_push),
27133 (vorbis_handle_data_packet), (vorbis_dec_change_state):
27134 Cleanups. Use refcounting and DEBUG_OBJECT.
27135 Reset segment on flush, use code methods instead of our
27137 Fix potential memleak.
27139 2006-05-15 16:46:44 +0000 Tim-Philipp Müller <tim@centricular.net>
27141 ext/alsa/gstalsasink.*: Don't leak allocated snd_output_t structure if there's more than one alsasink instance at a t...
27142 Original commit message from CVS:
27143 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
27144 (gst_alsasink_init):
27145 * ext/alsa/gstalsasink.h:
27146 Don't leak allocated snd_output_t structure if there's
27147 more than one alsasink instance at a time (#341873).
27148 Also fix GObject macros in header file.
27150 2006-05-15 15:31:30 +0000 Tim-Philipp Müller <tim@centricular.net>
27152 gst/subparse/gstsubparse.c: Don't use libxml functions in the typefinding code.
27153 Original commit message from CVS:
27154 * gst/subparse/gstsubparse.c:
27155 (gst_sub_parse_data_format_autodetect):
27156 Don't use libxml functions in the typefinding code.
27158 2006-05-15 15:01:08 +0000 Wim Taymans <wim.taymans@gmail.com>
27160 ext/ogg/gstoggdemux.c: Fix seeking performance in the case where a non-header packet has a 0 granulepos (busted theor...
27161 Original commit message from CVS:
27162 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
27163 Fix seeking performance in the case where a non-header
27164 packet has a 0 granulepos (busted theora case).
27167 2006-05-15 14:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
27169 gst/subparse/gstsubparse.c: Improve SAMI typefinding: handle case where there are whitespaces or newlines in front of...
27170 Original commit message from CVS:
27171 * gst/subparse/gstsubparse.c:
27172 (gst_sub_parse_data_format_autodetect):
27173 Improve SAMI typefinding: handle case where there are
27174 whitespaces or newlines in front of the first <SAMI>
27177 2006-05-15 12:18:13 +0000 Tim-Philipp Müller <tim@centricular.net>
27179 configure.ac: Build video4linux plugin even if there's no XVIDEO, just without implementing the GstXOverlay interface...
27180 Original commit message from CVS:
27182 Build video4linux plugin even if there's no XVIDEO, just
27183 without implementing the GstXOverlay interface (#334002).
27185 2006-05-15 10:17:04 +0000 Tim-Philipp Müller <tim@centricular.net>
27187 Add tentative support for libvisual-0.4 (#336881).
27188 Original commit message from CVS:
27190 * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
27192 Add tentative support for libvisual-0.4 (#336881).
27194 2006-05-15 09:41:03 +0000 Young-Ho Cha <ganadist@chollian.net>
27196 gst/subparse/samiparse.c: Need to map "silver" colour explicitly (#169936).
27197 Original commit message from CVS:
27198 Patch by: Young-Ho Cha <ganadist at chollian net>
27199 * gst/subparse/samiparse.c: (handle_start_font):
27200 Need to map "silver" colour explicitly (#169936).
27202 2006-05-15 09:14:35 +0000 Young-Ho Cha <ganadist@chollian.net>
27204 gst/subparse/: Add support for SAMI subtitles (#169936).
27205 Original commit message from CVS:
27206 Patch by: Young-Ho Cha <ganadist at chollian net>
27207 * gst/subparse/Makefile.am:
27208 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
27209 (parser_state_dispose), (gst_sub_parse_data_format_autodetect),
27210 (gst_sub_parse_format_autodetect), (feed_textbuf),
27211 (gst_subparse_type_find), (plugin_init):
27212 * gst/subparse/gstsubparse.h:
27213 * gst/subparse/samiparse.c:
27214 * gst/subparse/samiparse.h:
27215 Add support for SAMI subtitles (#169936).
27217 2006-05-14 21:18:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27219 * win32/common/config.h:
27221 Original commit message from CVS:
27224 2006-05-14 21:18:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27227 fix mistakes in README
27228 Original commit message from CVS:
27229 fix mistakes in README
27231 2006-05-14 18:15:17 +0000 Michael Smith <msmith@xiph.org>
27233 gst/audioconvert/gstchannelmix.c: Fix #341696: crash when mixing L+R+C to mono or stereo.
27234 Original commit message from CVS:
27235 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
27236 Fix #341696: crash when mixing L+R+C to mono or stereo.
27237 * tests/check/Makefile.am:
27238 * tests/check/elements/audioconvert.c: (set_channel_positions),
27239 (get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
27240 (audioconvert_suite):
27241 Add test for the above, including some generic framework bits for
27242 testing multichannel things.
27244 2006-05-14 16:05:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27248 Original commit message from CVS:
27251 === release 0.10.7 ===
27253 2006-05-14 16:00:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27255 configure.ac: releasing 0.10.7, "Leave the gun"
27256 Original commit message from CVS:
27257 2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
27259 releasing 0.10.7, "Leave the gun"
27261 2006-05-14 15:55:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27279 Original commit message from CVS:
27282 2006-05-12 22:22:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27285 Original commit message from CVS:
27286 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
27287 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
27290 2006-05-12 21:30:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27292 Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
27293 Original commit message from CVS:
27294 * docs/libs/gst-plugins-base-libs-docs.sgml:
27295 * docs/libs/gst-plugins-base-libs-sections.txt:
27296 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
27297 * gst-libs/gst/video/video.h:
27298 * gst/videoscale/Makefile.am:
27299 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
27300 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
27301 * tests/check/Makefile.am:
27302 * tests/check/libs/video.c: (GST_START_TEST), (video_suite),
27304 Fix integer overflow problem with pixel-aspect-ratio calculations
27305 in videoscale and xvimagesink (#341542)
27307 2006-05-12 16:56:52 +0000 Tim-Philipp Müller <tim@centricular.net>
27309 gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
27310 Original commit message from CVS:
27311 * gst-libs/gst/tag/gstid3tag.c:
27312 Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
27314 2006-05-12 10:39:08 +0000 Sébastien Moutte <sebastien@moutte.net>
27316 win32/MANIFEST: update win32 files listing
27317 Original commit message from CVS:
27319 update win32 files listing
27321 2006-05-11 21:47:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27323 * tests/check/elements/multifdsink.c:
27324 disable failing check on gentoo64
27325 Original commit message from CVS:
27326 disable failing check on gentoo64
27328 2006-05-11 21:35:44 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27330 * tests/check/elements/multifdsink.c:
27331 disable failing check on gentoo64
27332 Original commit message from CVS:
27333 disable failing check on gentoo64
27335 2006-05-11 21:20:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27337 * tests/check/elements/multifdsink.c:
27338 macros show the correct line
27339 Original commit message from CVS:
27340 macros show the correct line
27342 2006-05-11 21:04:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27344 * tests/check/elements/multifdsink.c:
27345 macros show the correct line
27346 Original commit message from CVS:
27347 macros show the correct line
27349 2006-05-11 21:01:05 +0000 Sjoerd Simons <sjoerd@luon.net>
27351 gst/playback/gstplaybasebin.*: API: GstPlayBaseBin::stream-info-value-array property use a more bindings-friendly way...
27352 Original commit message from CVS:
27353 2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
27354 patch by: Sjoerd Simons (sjoerd@luon.net)
27355 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
27356 (group_create), (group_destroy), (add_stream),
27357 (gst_play_base_bin_get_property),
27358 (gst_play_base_bin_get_streaminfo_value_array):
27359 * gst/playback/gstplaybasebin.h:
27360 API: GstPlayBaseBin::stream-info-value-array property
27361 use a more bindings-friendly way of exposing streaminfo
27362 using a GValueArray. Tested in ipython.
27365 2006-05-11 19:44:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27367 * tests/check/elements/multifdsink.c:
27368 fix some type warnings
27369 Original commit message from CVS:
27370 fix some type warnings
27372 2006-05-11 19:38:22 +0000 Wim Taymans <wim.taymans@gmail.com>
27374 gst/playback/gstdecodebin.c: Also catch queue underruns but don't do anything yet.
27375 Original commit message from CVS:
27376 * gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
27377 (queue_underrun_cb), (queue_filled_cb):
27378 Also catch queue underruns but don't do anything yet.
27379 Refactor and comment queue enlarging code a bit.
27380 * gst/playback/gstplaybasebin.c: (queue_overrun),
27381 (queue_threshold_reached), (queue_out_of_data),
27382 (gen_preroll_element):
27383 If a queue over/underruns check that we don't create nasty
27384 deadlocks when the min-threshold is not reached but the
27385 max-bytes is. In those cases disable max-bytes when we
27386 know that the queue is fed timed data.
27389 2006-05-11 18:06:18 +0000 Tim-Philipp Müller <tim@centricular.net>
27391 gst/playback/gstplaybin.c: Make playbin automatically plug an 'audioresample' element before the audio sink as well. ...
27392 Original commit message from CVS:
27393 * gst/playback/gstplaybin.c: (gen_audio_element):
27394 Make playbin automatically plug an 'audioresample'
27395 element before the audio sink as well. This solves
27396 problems with sinks that only accept a very specific
27397 sample rate, like esdsink (e.g. #340379).
27399 2006-05-11 16:04:28 +0000 Tim-Philipp Müller <tim@centricular.net>
27401 gst/playback/gstplaybasebin.c: Make http sources send special headers so that we receive icecast metadata if the http...
27402 Original commit message from CVS:
27403 * gst/playback/gstplaybasebin.c: (gen_source_element):
27404 Make http sources send special headers so that we receive
27405 icecast metadata if the http stream is an icecast stream
27406 (otherwise the server will just ignore them). This also
27407 means that from now on users will need the 'icydemux'
27408 element from gst-plugins-good installed if they want to
27409 listen to icecast radio streams. (#341432, #333657).
27411 2006-05-11 12:34:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27413 * gst/tcp/gstmultifdsink.c:
27415 Original commit message from CVS:
27418 2006-05-11 11:40:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27420 gst/tcp/gstmultifdsink.c: remove stupid example from docs - it should come with a simple
27421 Original commit message from CVS:
27422 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
27423 (gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
27424 remove stupid example from docs - it should come with a simple
27427 * tests/check/elements/multifdsink.c: (wait_bytes_served),
27428 (fail_if_can_read), (GST_START_TEST),
27429 (gst_multifdsink_create_streamheader), (multifdsink_suite):
27430 add a test for changing streamheader which exposes a bug in
27433 2006-05-11 10:33:46 +0000 Michael Smith <msmith@xiph.org>
27435 ext/gnomevfs/gstgnomevfssrc.*: Don't set icy-caps unless we have a sane interval value. Move interval to a local vari...
27436 Original commit message from CVS:
27437 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
27438 (gst_gnome_vfs_src_received_headers_callback):
27439 * ext/gnomevfs/gstgnomevfssrc.h:
27440 Don't set icy-caps unless we have a sane interval value. Move
27441 interval to a local variable; we never use it outside this function.
27443 2006-05-11 10:14:20 +0000 Wim Taymans <wim.taymans@gmail.com>
27445 sys/: Register special buffer types along with the objects so that they are not registered at runtime from N differen...
27446 Original commit message from CVS:
27447 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
27448 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
27449 Register special buffer types along with the objects so
27450 that they are not registered at runtime from N different
27451 streaming threads since they are not threadsafe.
27453 2006-05-10 18:31:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27455 * tests/check/elements/multifdsink.c:
27456 set caps and plug leaks
27457 Original commit message from CVS:
27458 set caps and plug leaks
27460 2006-05-10 18:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27462 tests/check/elements/multifdsink.c: add two more tests, one doing streamheader
27463 Original commit message from CVS:
27464 * tests/check/elements/multifdsink.c: (wait_bytes_served),
27465 (GST_START_TEST), (fail_unless_read), (multifdsink_suite):
27466 add two more tests, one doing streamheader
27468 2006-05-10 16:34:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27470 gst/tcp/gstmultifdsink.c: clean up the bufqueue when shutting down
27471 Original commit message from CVS:
27472 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
27473 clean up the bufqueue when shutting down
27474 * tests/check/Makefile.am:
27475 * tests/check/elements/multifdsink.c: (setup_multifdsink),
27476 (cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
27478 add a test for the leak that was just fixed
27480 2006-05-10 15:16:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27482 * gst/tcp/gstmultifdsink.c:
27484 Original commit message from CVS:
27487 2006-05-10 15:14:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27489 * gst/tcp/gstmultifdsink.c:
27490 * gst/tcp/gstmultifdsink.h:
27492 Original commit message from CVS:
27495 2006-05-10 11:54:36 +0000 Wim Taymans <wim.taymans@gmail.com>
27497 gst/adder/gstadder.*: Updated some docs. Added comments and FIXMEs all over the place.
27498 Original commit message from CVS:
27499 * gst/adder/gstadder.c: (gst_adder_setcaps),
27500 (gst_adder_query_duration), (gst_adder_query), (forward_event),
27501 (gst_adder_src_event), (gst_adder_sink_event),
27502 (gst_adder_class_init), (gst_adder_finalize),
27503 (gst_adder_request_new_pad), (gst_adder_collected):
27504 * gst/adder/gstadder.h:
27505 Updated some docs. Added comments and FIXMEs all over the place.
27506 Improve debugging info.
27507 Fix leak on finalize by not calling the parent.
27508 Implement duration query.
27509 Make event forwarding threadsafe.
27510 Correctly send NEWSEGMENT at start and after flush.
27511 Handle EOS correctly.
27512 Post error when not negotiated.
27513 * tests/check/elements/adder.c: (GST_START_TEST):
27514 Added FIXME in the test.
27516 2006-05-09 19:24:46 +0000 Tim-Philipp Müller <tim@centricular.net>
27518 Const-ify GEnumValue and GFlagsValue arrays. Use
27519 Original commit message from CVS:
27520 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
27521 (gst_text_overlay_halign_get_type),
27522 (gst_text_overlay_wrap_mode_get_type):
27523 * ext/theora/theoradec.c: (theora_handle_type_packet),
27524 (theora_handle_data_packet):
27525 * ext/theora/theoraenc.c: (gst_border_mode_get_type),
27526 (theora_enc_sink_setcaps), (theora_enc_chain):
27527 * gst-libs/gst/cdda/gstcddabasesrc.c:
27528 (gst_cdda_base_src_mode_get_type):
27529 * gst/audiotestsrc/gstaudiotestsrc.c:
27530 (gst_audiostestsrc_wave_get_type):
27531 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
27532 * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
27533 * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
27534 (gst_sync_method_get_type), (gst_unit_type_get_type),
27535 (gst_client_status_get_type):
27536 * gst/videoscale/gstvideoscale.c:
27537 (gst_video_scale_method_get_type):
27538 * gst/videotestsrc/gstvideotestsrc.c:
27539 (gst_video_test_src_pattern_get_type):
27540 * gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
27541 (paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
27542 (paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
27543 (paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
27544 (paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
27545 (paint_setup_RGB565), (paint_setup_xRGB1555):
27546 Const-ify GEnumValue and GFlagsValue arrays. Use
27547 GST_ROUND_UP_* macros instead of home-made ones.
27549 2006-05-09 17:40:41 +0000 Tim-Philipp Müller <tim@centricular.net>
27551 configure.ac: Require core CVS for the new newsegment stuff.
27552 Original commit message from CVS:
27554 Require core CVS for the new newsegment stuff.
27556 2006-05-09 17:30:48 +0000 Sjoerd Simons <sjoerd@luon.net>
27558 gst/tcp/gstmultifdsink.c: Register nick for enum value (#341160).
27559 Original commit message from CVS:
27560 Patch by: Sjoerd Simons <sjoerd at luon net>
27561 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
27562 Register nick for enum value (#341160).
27564 2006-05-09 16:46:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27566 gst/typefind/gsttypefindfunctions.c: backout typefind patch #340375
27567 Original commit message from CVS:
27568 * gst/typefind/gsttypefindfunctions.c: (m4a_type_find),
27570 backout typefind patch #340375
27571 * tests/check/elements/adder.c: (message_received),
27572 (GST_START_TEST), (adder_suite):
27573 redo, signal-handling of test
27575 2006-05-09 16:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
27577 gst/adder/gstadder.*: Remove bogus segment merging and forwarding, we don't care about timestamps anyway and we just ...
27578 Original commit message from CVS:
27579 * gst/adder/gstadder.c: (gst_adder_request_new_pad),
27580 (gst_adder_collected):
27581 * gst/adder/gstadder.h:
27582 Remove bogus segment merging and forwarding, we don't
27583 care about timestamps anyway and we just produce a
27585 Also create a nice NEWSEGMENT event when we start.
27586 Use _scale_int some more.
27588 2006-05-09 11:59:13 +0000 Edward Hervey <bilboed@bilboed.com>
27590 tests/icles/stress-xoverlay.c: Fix if core was built without parsing support.
27591 Original commit message from CVS:
27592 * tests/icles/stress-xoverlay.c:
27593 Fix if core was built without parsing support.
27595 2006-05-09 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
27597 gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
27598 Original commit message from CVS:
27599 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
27600 Add SEDG (Samsung MPEG-4) fourcc.
27602 2006-05-09 11:31:47 +0000 Edward Hervey <bilboed@bilboed.com>
27604 tests/examples/volume/volume.c: Fox if core was built without parsing support.
27605 Original commit message from CVS:
27606 * tests/examples/volume/volume.c:
27607 Fox if core was built without parsing support.
27608 * tests/examples/seek/seek.c:
27609 Disable the parse_launch example if core was built without parsing
27612 2006-05-09 11:21:24 +0000 Edward Hervey <bilboed@bilboed.com>
27614 tests/examples/seek/seek.c: Disable the parse_launch example if core was built without parsing support.
27615 Original commit message from CVS:
27616 * tests/examples/seek/seek.c:
27617 Disable the parse_launch example if core was built without parsing
27620 2006-05-08 15:51:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27622 * docs/libs/tmpl/gstcolorbalance.sgml:
27623 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
27624 * gst/tcp/gstmultifdsink.c:
27625 * gst/videoscale/gstvideoscale.c:
27626 doc reparagraphing and DEBUG_FUNCPTRing
27627 Original commit message from CVS:
27628 doc reparagraphing and DEBUG_FUNCPTRing
27630 2006-05-08 11:51:43 +0000 Edward Hervey <bilboed@bilboed.com>
27632 autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize
27633 Original commit message from CVS:
27634 * autogen.sh: (CONFIGURE_DEF_OPT):
27635 libtoolize on Darwin/MacOSX is called glibtoolize
27637 2006-05-07 17:39:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27639 tests/check/: Disable the adder test, until the build-slaves posses the kindness to either like it or to give valid r...
27640 Original commit message from CVS:
27641 * tests/check/Makefile.am:
27642 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
27643 Disable the adder test, until the build-slaves posses the kindness to
27644 either like it or to give valid reason for not doing so
27646 2006-05-07 17:25:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27648 tests/check/elements/adder.c: Shuffle NULL state change around and raise timeout more
27649 Original commit message from CVS:
27650 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
27652 Shuffle NULL state change around and raise timeout more
27654 2006-05-07 17:07:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27656 gst/typefind/gsttypefindfunctions.c: Add typefind to distinguish between "audio/x-m4a" and new type "video/mp4". Fixe...
27657 Original commit message from CVS:
27658 * gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
27659 (mp4_type_find), (plugin_init):
27660 Add typefind to distinguish between "audio/x-m4a" and new type
27661 "video/mp4". Fixes #340375
27662 * tests/check/elements/adder.c: (adder_suite):
27663 Raise timeout to make buildbot happy
27665 2006-05-07 16:39:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27667 Add sink-event handling to adder. It tries to merge incomming newsegment-events. Added test to check if segment_done ...
27668 Original commit message from CVS:
27669 * gst/adder/gstadder.c: (gst_adder_sink_event),
27670 (gst_adder_request_new_pad), (gst_adder_change_state):
27671 * gst/adder/gstadder.h:
27672 * tests/check/Makefile.am:
27673 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
27674 (adder_suite), (main):
27675 Add sink-event handling to adder. It tries to merge incomming
27676 newsegment-events. Added test to check if segment_done is comming
27679 2006-05-05 16:34:15 +0000 Andy Wingo <wingo@pobox.com>
27682 * ext/theora/theoraparse.c:
27683 * ext/vorbis/vorbisparse.c:
27684 ext/theora/theoraparse.c (gst_theora_parse_init) ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
27685 Original commit message from CVS:
27686 2006-05-05 Andy Wingo <wingo@pobox.com>
27687 * ext/theora/theoraparse.c (gst_theora_parse_init)
27688 (theora_parse_src_convert, theora_parse_src_query):
27689 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
27690 (vorbis_parse_convert, vorbis_parse_src_query): Add convert and
27691 query functions on the source pads of the theora and vorbis parse
27692 elements. Fixes position querying when doing a remux.
27694 2006-05-05 13:46:37 +0000 Michael Smith <msmith@xiph.org>
27696 ext/theora/theoraparse.c: Fix flushing.
27697 Original commit message from CVS:
27698 * ext/theora/theoraparse.c: (parse_granulepos),
27699 (theora_parse_drain_queue_prematurely),
27700 (theora_parse_queue_buffer), (theora_parse_sink_event):
27702 Fix invalid granulepos outputs when starting with a non-keyframe.
27704 2006-05-05 12:37:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27706 gst/typefind/gsttypefindfunctions.c: Rearrange MPEG system stream detection, fixing some memleaks in the process.
27707 Original commit message from CVS:
27708 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
27709 (mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
27710 Rearrange MPEG system stream detection, fixing some memleaks in the
27712 Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
27713 they clean up their data correctly.
27714 Remove unused ogganx caps and move the 'is_annodex' check to inside
27715 the 'is_ogg' if statement.
27717 2006-05-05 11:33:37 +0000 Wim Taymans <wim.taymans@gmail.com>
27719 gst/playback/gstdecodebin.c: Properly remove ghostpads. Fixes #340392
27720 Original commit message from CVS:
27721 * gst/playback/gstdecodebin.c: (cleanup_decodebin):
27722 Properly remove ghostpads. Fixes #340392
27724 2006-05-04 18:43:58 +0000 David Schleef <ds@schleef.org>
27726 gst/typefind/gsttypefindfunctions.c:
27727 Original commit message from CVS:
27728 * gst/typefind/gsttypefindfunctions.c:
27730 2006-05-03 16:32:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27732 gst/typefind/gsttypefindfunctions.c: When typefinding an MP3 in push-based mode, don't penalise the probability down ...
27733 Original commit message from CVS:
27734 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
27735 (mpeg_ts_probe_headers), (mpeg_ts_type_find):
27736 When typefinding an MP3 in push-based mode, don't penalise the
27737 probability down to 74% when we found 5 valid frames just because we
27738 can't peek the end of the file.
27739 Make the probability for detecting MPEG Transport Streams based on the
27740 number of sequential headers we successfully detected.
27742 2006-05-03 15:52:46 +0000 Wim Taymans <wim.taymans@gmail.com>
27744 ext/vorbis/vorbisdec.c: Still produce an error when we receive an empty packet.
27745 Original commit message from CVS:
27746 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
27747 (vorbis_dec_push), (vorbis_dec_chain):
27748 Still produce an error when we receive an empty packet.
27750 2006-05-03 15:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
27752 ext/ogg/gstoggdemux.c: Mark buffers with DISCONT after seek and after activating new chains.
27753 Original commit message from CVS:
27754 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
27755 (gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
27756 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
27757 Mark buffers with DISCONT after seek and after activating new
27759 * ext/theora/gsttheoradec.h:
27760 * ext/theora/theoradec.c: (gst_theora_dec_reset),
27761 (theora_get_query_types), (theora_dec_sink_event),
27762 (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
27763 (theora_dec_change_state):
27765 Detect and mark DISCONT buffers.
27766 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
27767 (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
27768 (vorbis_dec_change_state):
27769 * ext/vorbis/vorbisdec.h:
27771 Detect and mark DISCONT buffers.
27772 Don't crash on 0 sized buffers.
27774 2006-05-03 08:58:13 +0000 Wim Taymans <wim.taymans@gmail.com>
27776 gst/volume/gstvolume.c: Increase "volume" property to 10.0. Fixes #340369.
27777 Original commit message from CVS:
27778 * gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
27779 (volume_transform_ip):
27780 Increase "volume" property to 10.0. Fixes #340369.
27781 Set the process function to NULL when capsnego fails so that
27782 we properly error out.
27784 2006-05-02 18:15:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27786 gst/playback/: free cpas using gst_caps_unref, don't leak caps-strings
27787 Original commit message from CVS:
27788 * gst/playback/gstplaybin.c: (add_sink):
27789 * gst/playback/test.c: (main):
27790 * gst/playback/test5.c: (dump_element_stats):
27791 * gst/playback/test6.c: (main):
27792 free cpas using gst_caps_unref, don't leak caps-strings
27794 2006-05-02 06:33:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27796 * gst-libs/gst/rtp/gstbasertppayload.c:
27798 Original commit message from CVS:
27801 2006-05-01 19:08:40 +0000 Tim-Philipp Müller <tim@centricular.net>
27803 gst/typefind/gsttypefindfunctions.c: Refine musepack typefinding a bit. Return MAXIMUM probability when we detect str...
27804 Original commit message from CVS:
27805 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
27807 Refine musepack typefinding a bit. Return MAXIMUM
27808 probability when we detect stream version 7 to make
27809 sure the mpeg audio typefinder doesn't trump us.
27811 2006-04-29 16:25:58 +0000 Tim-Philipp Müller <tim@centricular.net>
27813 gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
27814 Original commit message from CVS:
27815 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
27816 Protect against unexpected NULL strf_data buffer.
27818 2006-04-29 13:09:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27820 tests/check/elements/audioconvert.c: interpret the out[] buffer in the order the bytes are actually put in, which is ...
27821 Original commit message from CVS:
27822 * tests/check/elements/audioconvert.c: (verify_convert),
27824 interpret the out[] buffer in the order the bytes are actually
27825 put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
27826 Other tests should use BYTE_ORDER since the array is filled in
27829 2006-04-29 12:10:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27831 * tests/check/elements/audioconvert.c:
27832 dump expected data when audioconvert test fails
27833 Original commit message from CVS:
27834 dump expected data when audioconvert test fails
27836 2006-04-29 11:55:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27838 tests/check/elements/audioconvert.c: when a test fails, give an indication of which it is
27839 Original commit message from CVS:
27840 * tests/check/elements/audioconvert.c: (verify_convert),
27842 when a test fails, give an indication of which it is
27844 2006-04-29 09:48:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27846 * ext/ogg/gstoggmux.c:
27847 * ext/theora/theoraenc.c:
27848 add another include
27849 Original commit message from CVS:
27850 add another include
27852 2006-04-29 01:24:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27854 * gst/subparse/gstssaparse.c:
27855 atoi() needs stdlib.h
27856 Original commit message from CVS:
27857 atoi() needs stdlib.h
27859 2006-04-29 01:18:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27861 * gst/playback/test4.c:
27862 * gst/playback/test5.c:
27863 * gst/playback/test6.c:
27864 exit needs stdlib.h
27865 Original commit message from CVS:
27866 exit needs stdlib.h
27868 2006-04-29 01:10:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27870 gst-libs/gst/cdda/gstcddabasesrc.c: compile fix; strtol() needs <stdlib.h>
27871 Original commit message from CVS:
27872 * gst-libs/gst/cdda/gstcddabasesrc.c:
27873 compile fix; strtol() needs <stdlib.h>
27875 2006-04-29 01:04:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27879 * docs/Makefile.am:
27880 * docs/libs/Makefile.am:
27881 * docs/libs/tmpl/gstcolorbalance.sgml:
27882 * docs/plugins/Makefile.am:
27884 use common upload.mak
27885 Original commit message from CVS:
27886 use common upload.mak
27888 2006-04-28 19:46:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27890 make GstElementDetails const
27891 Original commit message from CVS:
27892 * ext/alsa/gstalsamixerelement.c:
27893 * ext/alsa/gstalsasrc.c:
27894 * ext/cdparanoia/gstcdparanoiasrc.c:
27895 * ext/gnomevfs/gstgnomevfssink.c:
27896 * ext/gnomevfs/gstgnomevfssrc.c:
27897 * ext/ogg/gstoggdemux.c:
27898 * ext/ogg/gstoggmux.c:
27899 * ext/ogg/gstoggparse.c:
27900 * ext/ogg/gstogmparse.c:
27901 * ext/pango/gstclockoverlay.c:
27902 * ext/pango/gsttextoverlay.c:
27903 * ext/pango/gsttextrender.c:
27904 * ext/pango/gsttimeoverlay.c:
27905 * ext/theora/theoradec.c:
27906 * ext/theora/theoraenc.c:
27907 * ext/vorbis/vorbisdec.c:
27908 * ext/vorbis/vorbisenc.c:
27909 * gst-libs/gst/audio/gstaudiofilter.c:
27910 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
27911 * gst/audioconvert/gstaudioconvert.c:
27912 * gst/audiorate/gstaudiorate.c:
27913 * gst/audioresample/gstaudioresample.c:
27914 * gst/audiotestsrc/gstaudiotestsrc.c:
27915 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
27916 * gst/playback/gstdecodebin.c:
27917 * gst/playback/gstplaybin.c:
27918 * gst/playback/gststreamselector.c:
27919 * gst/subparse/gstsubparse.c:
27920 * gst/tcp/gstmultifdsink.c:
27921 * gst/tcp/gsttcpclientsink.c:
27922 * gst/tcp/gsttcpclientsrc.c:
27923 * gst/tcp/gsttcpserversink.c:
27924 * gst/tcp/gsttcpserversrc.c:
27925 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
27926 * gst/videorate/gstvideorate.c:
27927 * gst/videoscale/gstvideoscale.c:
27928 * gst/videotestsrc/gstvideotestsrc.c:
27929 * gst/volume/gstvolume.c:
27930 * sys/v4l/gstv4ljpegsrc.c:
27931 * sys/v4l/gstv4lmjpegsink.c:
27932 * sys/v4l/gstv4lmjpegsrc.c:
27933 * sys/v4l/gstv4lsrc.c:
27934 * sys/ximage/ximagesink.c:
27935 * sys/xvimage/xvimagesink.c:
27936 * tests/check/libs/cddabasesrc.c:
27937 make GstElementDetails const
27939 2006-04-28 19:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27941 gst/adder/gstadder.c: send events from src-pad to all sink-pads fixes #338657
27942 Original commit message from CVS:
27943 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
27945 send events from src-pad to all sink-pads fixes #338657
27947 2006-04-28 19:08:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27949 ext/alsa/gstalsasink.c: query witdh capabilities from alsa, fixes #338919
27950 Original commit message from CVS:
27951 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
27952 (alsasink_parse_spec):
27953 query witdh capabilities from alsa, fixes #338919
27955 2006-04-28 15:31:28 +0000 Wim Taymans <wim.taymans@gmail.com>
27957 gst/tcp/gstmultifdsink.*: Fix race condition in multifdsink that can lead to spurious duplicate clients. this patch a...
27958 Original commit message from CVS:
27959 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
27960 (gst_multi_fd_sink_remove_client_link):
27961 * gst/tcp/gstmultifdsink.h:
27962 Fix race condition in multifdsink that can lead to spurious
27963 duplicate clients. this patch adds a new signal that is fired when
27964 multifdsink has removed all references to the fd.
27966 Updated documentation.
27967 API: client-fd-removed signal added
27969 2006-04-28 15:24:00 +0000 Michael Smith <msmith@xiph.org>
27971 gst/tcp/gstmultifdsink.c: When asking g_value_array_new to prealloc elements, we may as well ask for the right number...
27972 Original commit message from CVS:
27973 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
27974 When asking g_value_array_new to prealloc elements, we may as well
27975 ask for the right number of elements.
27977 2006-04-28 15:08:09 +0000 Wim Taymans <wim.taymans@gmail.com>
27979 gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
27980 Original commit message from CVS:
27981 * gst-libs/gst/audio/gstbaseaudiosink.c:
27982 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
27983 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
27984 patch to make timestamp checking more tollerant to rounding
27985 errors given that real discontinuities are to be marked on
27986 buffers. Fixes some asf files and #338778.
27987 Also avoid some crashers when we receive an event in the
27990 2006-04-28 15:01:58 +0000 Michael Smith <msmith@xiph.org>
27992 ext/gnomevfs/gstgnomevfssrc.*: Remove ICY handling (mostly) from gnomevfssrc, in favour of proper shared support with...
27993 Original commit message from CVS:
27994 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
27995 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
27996 (gst_gnome_vfs_src_get_property),
27997 (gst_gnome_vfs_src_send_additional_headers_callback),
27998 (gst_gnome_vfs_src_received_headers_callback),
27999 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
28000 (gst_gnome_vfs_src_stop):
28001 * ext/gnomevfs/gstgnomevfssrc.h:
28002 Remove ICY handling (mostly) from gnomevfssrc, in favour of
28003 proper shared support within icydemux.
28005 2006-04-28 14:49:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28007 gst/videorate/gstvideorate.c: fix up docs fix a leak when no caps negotiated fix counting of input frames
28008 Original commit message from CVS:
28009 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
28010 (gst_video_rate_swap_prev), (gst_video_rate_chain):
28012 fix a leak when no caps negotiated
28013 fix counting of input frames
28014 * tests/check/elements/.cvsignore:
28015 * tests/check/elements/videorate.c: (assert_videorate_stats),
28016 (GST_START_TEST), (videorate_suite):
28017 add tests for these
28019 2006-04-28 14:48:11 +0000 Wim Taymans <wim.taymans@gmail.com>
28021 gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
28022 Original commit message from CVS:
28023 * gst-libs/gst/audio/gstringbuffer.c:
28024 (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
28025 (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
28026 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
28027 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
28028 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
28029 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
28030 (gst_ring_buffer_commit), (gst_ring_buffer_read),
28031 (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
28032 (gst_ring_buffer_clear), (gst_ring_buffer_may_start):
28033 Check arguments passed to public functions instead of
28036 2006-04-28 14:37:46 +0000 Wim Taymans <wim.taymans@gmail.com>
28038 gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
28039 Original commit message from CVS:
28040 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
28041 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
28042 GstBaseAudioSrc must be live or it does not work.
28043 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
28044 Don't set live to TRUE as this is the default in the parentclass.
28046 2006-04-28 14:37:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28048 * win32/common/config.h:
28050 Original commit message from CVS:
28053 2006-04-28 14:33:45 +0000 Wim Taymans <wim.taymans@gmail.com>
28055 gst/videoscale/gstvideoscale.c: Videoscale doesn't pass on pixel-aspect ratio. Handle all fixation cases better. Fixe...
28056 Original commit message from CVS:
28057 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps),
28058 (gst_video_scale_fixate_caps), (gst_video_scale_src_event):
28059 Videoscale doesn't pass on pixel-aspect ratio. Handle all
28060 fixation cases better. Fixes #338991
28062 2006-04-28 14:24:38 +0000 Wim Taymans <wim.taymans@gmail.com>
28064 gst/videotestsrc/gstvideotestsrc.c: Handle 0/1 framerate correctly Fixes #331901.
28065 Original commit message from CVS:
28066 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
28067 Handle 0/1 framerate correctly Fixes #331901.
28069 2006-04-28 14:22:16 +0000 Wim Taymans <wim.taymans@gmail.com>
28071 tests/check/elements/audioconvert.c: Added check for correct clipping when doing float samples in audioconvert.
28072 Original commit message from CVS:
28073 * tests/check/elements/audioconvert.c: (get_float_caps),
28074 (GST_START_TEST), (audioconvert_suite):
28075 Added check for correct clipping when doing float samples
28078 2006-04-28 14:19:49 +0000 Wim Taymans <wim.taymans@gmail.com>
28080 gst/videorate/gstvideorate.c: Print more debugging info.
28081 Original commit message from CVS:
28082 * gst/videorate/gstvideorate.c: (gst_video_rate_event),
28083 (gst_video_rate_chain):
28084 Print more debugging info.
28086 2006-04-28 14:17:00 +0000 Wim Taymans <wim.taymans@gmail.com>
28088 gst/audioresample/gstaudioresample.c: Add support for other formats audioresample can handle such as 32 bits in and f...
28089 Original commit message from CVS:
28090 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
28091 (resample_set_state_from_caps):
28092 Add support for other formats audioresample can handle such as
28093 32 bits in and float and 64 bits float. Fixes #301759
28095 2006-04-28 14:12:28 +0000 Wim Taymans <wim.taymans@gmail.com>
28097 gst/audioconvert/audioconvert.c: correctly clip float samples > 1.0. Fixes #338718
28098 Original commit message from CVS:
28099 * gst/audioconvert/audioconvert.c: (float):
28100 correctly clip float samples > 1.0. Fixes #338718
28102 2006-04-28 13:35:34 +0000 Young-Ho Cha <ganadist@chollian.net>
28104 ext/pango/gsttextoverlay.c: Don't strip newlines from the text. Also, center lines within multi-line paragraphs (#339...
28105 Original commit message from CVS:
28106 Patch by: Young-Ho Cha <ganadist at chollian net>
28107 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
28108 (gst_text_overlay_render_text):
28109 Don't strip newlines from the text. Also, center lines
28110 within multi-line paragraphs (#339405).
28112 2006-04-28 12:15:33 +0000 Tim-Philipp Müller <tim@centricular.net>
28114 gst/typefind/gsttypefindfunctions.c: Fix wavpack typefinding to work in more cases (don't peek for chunks of multiple...
28115 Original commit message from CVS:
28116 * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
28117 Fix wavpack typefinding to work in more cases (don't peek
28118 for chunks of multiple hundred kBs at once, but process
28119 things step-by-step in smaller units). Fixes #339786.
28121 2006-04-28 10:58:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28126 Original commit message from CVS:
28129 === release 0.10.6 ===
28131 2006-04-28 10:53:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28137 * docs/plugins/gst-plugins-base-plugins.signals:
28138 * docs/plugins/inspect/plugin-adder.xml:
28139 * docs/plugins/inspect/plugin-alsa.xml:
28140 * docs/plugins/inspect/plugin-audioconvert.xml:
28141 * docs/plugins/inspect/plugin-audiorate.xml:
28142 * docs/plugins/inspect/plugin-audioresample.xml:
28143 * docs/plugins/inspect/plugin-audiotestsrc.xml:
28144 * docs/plugins/inspect/plugin-cdparanoia.xml:
28145 * docs/plugins/inspect/plugin-decodebin.xml:
28146 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
28147 * docs/plugins/inspect/plugin-gnomevfs.xml:
28148 * docs/plugins/inspect/plugin-libvisual.xml:
28149 * docs/plugins/inspect/plugin-ogg.xml:
28150 * docs/plugins/inspect/plugin-pango.xml:
28151 * docs/plugins/inspect/plugin-playbin.xml:
28152 * docs/plugins/inspect/plugin-subparse.xml:
28153 * docs/plugins/inspect/plugin-tcp.xml:
28154 * docs/plugins/inspect/plugin-theora.xml:
28155 * docs/plugins/inspect/plugin-typefindfunctions.xml:
28156 * docs/plugins/inspect/plugin-video4linux.xml:
28157 * docs/plugins/inspect/plugin-videorate.xml:
28158 * docs/plugins/inspect/plugin-videoscale.xml:
28159 * docs/plugins/inspect/plugin-videotestsrc.xml:
28160 * docs/plugins/inspect/plugin-volume.xml:
28161 * docs/plugins/inspect/plugin-vorbis.xml:
28162 * docs/plugins/inspect/plugin-ximagesink.xml:
28163 * docs/plugins/inspect/plugin-xvimagesink.xml:
28166 Original commit message from CVS:
28169 2006-04-28 10:42:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28172 * win32/common/config.h:
28173 dist more win32 files
28174 Original commit message from CVS:
28175 dist more win32 files
28177 2006-04-28 10:41:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28194 Original commit message from CVS:
28197 2006-04-27 00:19:29 +0000 David Schleef <ds@schleef.org>
28199 gst/videoscale/gstvideoscale.c: Add call to oil_init().
28200 Original commit message from CVS:
28201 * gst/videoscale/gstvideoscale.c: Add call to oil_init().
28204 2006-04-26 17:20:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28208 * win32/common/config.h:
28210 Original commit message from CVS:
28213 2006-04-26 17:17:39 +0000 Wim Taymans <wim.taymans@gmail.com>
28215 ext/ogg/gstoggdemux.c: make sure correct newsegments are sent, so that the decoder and the demuxer agree on timestamp...
28216 Original commit message from CVS:
28217 2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
28218 patch by: Wim Taymans
28219 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
28220 (gst_ogg_demux_perform_seek):
28221 make sure correct newsegments are sent, so that the decoder
28222 and the demuxer agree on timestamps. Fixes playback of a lot
28223 of Ogg files that do not start from 0. Fixes #339833.
28225 2006-04-26 16:44:20 +0000 Edward Hervey <edward@fluendo.com>
28227 Fix an infinite loop if frames are passed in with wrongly ordered timestamps. Fixes #339013.
28228 Original commit message from CVS:
28229 Patch by: Edward Hervey <edward@fluendo.com>
28230 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
28231 * tests/check/Makefile.am:
28232 * tests/check/elements/videorate.c: (assert_videorate_stats),
28233 (setup_videorate), (cleanup_videorate), (GST_START_TEST),
28234 (videorate_suite), (main):
28235 Fix an infinite loop if frames are passed in with wrongly ordered
28236 timestamps. Fixes #339013.
28238 2006-04-26 13:55:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28241 * win32/common/config.h:
28243 Original commit message from CVS:
28246 2006-04-22 21:25:41 +0000 Tim-Philipp Müller <tim@centricular.net>
28248 gst/typefind/gsttypefindfunctions.c: fix typefinding on some ISO files. Fixes #339212.
28249 Original commit message from CVS:
28250 Patch by: Tim-Philipp Müller <tim at centricular dot net>
28251 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
28252 fix typefinding on some ISO files. Fixes #339212.
28254 2006-04-22 21:19:06 +0000 Tim-Philipp Müller <tim@centricular.net>
28256 gst-libs/gst/riff/riff-media.c: add another H264 fourcc. Fixes #339047.
28257 Original commit message from CVS:
28258 Patch by: Tim-Philipp Müller <tim at centricular dot net>
28259 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
28260 add another H264 fourcc. Fixes #339047.
28262 2006-04-22 21:12:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28264 gst/playback/gststreamselector.c: Restore old StreamSelector behaviour.
28265 Original commit message from CVS:
28266 Patch by: Jan Schmidt
28267 * gst/playback/gststreamselector.c:
28268 (gst_stream_selector_bufferalloc):
28269 Restore old StreamSelector behaviour.
28272 2006-04-13 09:26:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28275 * gst-libs/gst/rtp/Makefile.am:
28276 * gst-libs/gst/rtp/gstrtpbuffer.h:
28277 reverting rtp patches to fix freeze break on -base as explained on the list
28278 Original commit message from CVS:
28279 reverting rtp patches to fix freeze break on -base as explained on the list
28281 2006-04-13 03:55:12 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28283 gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
28284 Original commit message from CVS:
28285 2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28286 * gst-libs/gst/rtp/gstrtpbuffer.h:
28287 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
28288 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
28289 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
28290 New RTP audio base payloader class. Supports frame or sample based codecs
28292 2006-04-12 11:04:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28310 update libtool versioning
28311 Original commit message from CVS:
28312 update libtool versioning
28314 2006-04-12 10:58:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28318 * win32/common/config.h:
28320 Original commit message from CVS:
28323 2006-04-11 17:31:29 +0000 Antoine Tremblay <hexa00@gmail.com>
28325 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
28326 Original commit message from CVS:
28327 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
28328 * gst-libs/gst/rtp/gstbasertpdepayload.c:
28329 (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
28330 Fix some memory leaks: on finalize, free buffers left in the queue
28331 before destroying the queue; in _push(), unref rtp_buf even if
28332 the process vfunc returned a NULL buffer as output buffer (#337548);
28333 demote some recuring debug messages to LOG level.
28335 2006-04-11 15:01:51 +0000 Christian Schaller <uraeus@gnome.org>
28337 * gst-plugins-base.spec.in:
28338 fix version number macro
28339 Original commit message from CVS:
28340 fix version number macro
28342 2006-04-11 14:42:33 +0000 Wim Taymans <wim.taymans@gmail.com>
28344 ext/ogg/gstoggdemux.c: More cleanups.
28345 Original commit message from CVS:
28346 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
28347 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
28348 (gst_ogg_chain_free), (gst_ogg_demux_sink_event),
28349 (gst_ogg_demux_loop):
28351 Respect segment stop when emiting EOS or SEGMENT_DONE.
28354 2006-04-11 10:45:32 +0000 Tim-Philipp Müller <tim@centricular.net>
28356 gst/playback/gststreamselector.c: Don't leak pad name.
28357 Original commit message from CVS:
28358 * gst/playback/gststreamselector.c:
28359 (gst_stream_selector_get_property):
28360 Don't leak pad name.
28362 2006-04-11 09:42:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28365 Mention bug #336617 closed by recent commit
28366 Original commit message from CVS:
28367 Mention bug #336617 closed by recent commit
28369 2006-04-10 20:32:46 +0000 Michael Smith <msmith@xiph.org>
28371 tests/check/: so that FC4 buildslaves can pass.
28372 Original commit message from CVS:
28373 * tests/check/Makefile.am:
28374 * tests/check/gst-plugins-base.supp:
28375 Suppress an old libtheora bug (fixed in more recent versions), so
28376 that FC4 buildslaves can pass.
28378 2006-04-10 19:13:30 +0000 Wim Taymans <wim.taymans@gmail.com>
28380 ext/ogg/gstoggdemux.c: Don't leak events.
28381 Original commit message from CVS:
28382 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
28383 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
28384 (gst_ogg_demux_init), (gst_ogg_demux_finalize),
28385 (gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
28386 (gst_ogg_demux_loop):
28388 Remember what error we got when finding chains, if we
28389 were shutdown, that would not be an error.
28391 2006-04-10 17:05:46 +0000 Wim Taymans <wim.taymans@gmail.com>
28393 gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
28394 Original commit message from CVS:
28395 * gst-libs/gst/audio/gstbaseaudiosink.c:
28396 (gst_base_audio_sink_event):
28397 Starting the ringbuffer when we did not acquire it can cause
28398 a deadlock, is pointless and causes nasty things for
28400 Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
28402 2006-04-10 15:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
28404 ext/ogg/gstoggdemux.c: Add some more debugging.
28405 Original commit message from CVS:
28406 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
28407 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
28408 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
28409 (gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
28410 (gst_ogg_demux_deactivate_current_chain),
28411 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
28412 (gst_ogg_demux_bisect_forward_serialno),
28413 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
28414 Add some more debugging.
28416 2006-04-10 14:52:10 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28419 * ext/theora/theoraenc.c:
28421 Original commit message from CVS:
28424 2006-04-10 10:29:21 +0000 Wim Taymans <wim.taymans@gmail.com>
28426 ext/theora/theoradec.c: Some more debug info.
28427 Original commit message from CVS:
28428 * ext/theora/theoradec.c: (theora_dec_src_event),
28429 (theora_handle_data_packet):
28430 Some more debug info.
28431 * tests/examples/seek/seek.c: (start_seek), (main):
28432 Print element messages too.
28434 2006-04-09 17:14:22 +0000 Sébastien Moutte <sebastien@moutte.net>
28436 gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer sta...
28437 Original commit message from CVS:
28438 * gst/audioresample/debug.h:
28439 replace debug macros with variable number of parameters
28440 by a simple alias to gstreamer standard debug macros
28441 (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
28442 supported by MSVC 6.0 and 7.1)
28443 * gst/audioresample/resample.h:
28444 define M_PI and rint for WIN32
28445 * win32/common/libgstaudio.def:
28446 * win32/common/libgstriff.def:
28447 * win32/common/libgsttag.def:
28448 * win32/common/libgstvideo.def:
28449 add new exported functions
28451 update project files
28453 2006-04-08 21:02:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28455 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
28456 Original commit message from CVS:
28457 * ext/alsa/gstalsamixeroptions.c:
28458 (gst_alsa_mixer_options_class_init):
28459 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
28460 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
28461 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
28462 * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
28463 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
28464 * gst-libs/gst/audio/gstaudiofilter.c:
28465 (gst_audio_filter_class_init):
28466 * gst-libs/gst/audio/gstaudiosink.c:
28467 (gst_audioringbuffer_class_init):
28468 * gst-libs/gst/audio/gstaudiosrc.c:
28469 (gst_audioringbuffer_class_init):
28470 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
28471 * gst-libs/gst/interfaces/colorbalancechannel.c:
28472 (gst_color_balance_channel_class_init):
28473 * gst-libs/gst/interfaces/mixeroptions.c:
28474 (gst_mixer_options_class_init):
28475 * gst-libs/gst/interfaces/mixertrack.c:
28476 (gst_mixer_track_class_init):
28477 * gst-libs/gst/interfaces/tunerchannel.c:
28478 (gst_tuner_channel_class_init):
28479 * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
28480 * gst-libs/gst/netbuffer/gstnetbuffer.c:
28481 (gst_netbuffer_class_init):
28482 * gst-libs/gst/rtp/gstbasertppayload.c:
28483 (gst_basertppayload_class_init):
28484 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
28485 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
28486 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
28487 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
28488 * gst/playback/gststreamselector.c:
28489 (gst_stream_selector_class_init):
28490 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
28491 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
28492 * sys/v4l/gstv4lcolorbalance.c:
28493 (gst_v4l_color_balance_channel_class_init):
28494 * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
28495 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
28496 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
28497 * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
28498 (gst_v4l_tuner_norm_class_init):
28499 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
28500 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
28501 * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
28502 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
28504 2006-04-08 18:09:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28506 Fix broken GObject macros
28507 Original commit message from CVS:
28508 * ext/pango/gsttextrender.h:
28509 * gst-libs/gst/audio/gstaudiosink.h:
28510 * gst-libs/gst/audio/gstaudiosrc.h:
28511 * gst-libs/gst/audio/gstbaseaudiosink.h:
28512 * gst-libs/gst/audio/gstbaseaudiosrc.h:
28513 * gst-libs/gst/audio/gstringbuffer.h:
28514 * gst-libs/gst/rtp/gstbasertpdepayload.h:
28515 * gst-libs/gst/rtp/gstbasertppayload.h:
28516 * gst-libs/gst/video/gstvideofilter.h:
28517 * gst-libs/gst/video/gstvideosink.h:
28518 * gst/playback/gstplaybasebin.h:
28519 * gst/tcp/gstmultifdsink.h:
28520 * sys/v4l/gstv4lelement.h:
28521 Fix broken GObject macros
28523 2006-04-08 16:21:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28525 ext/alsa/gstalsasink.c: More debug to trace why my USB headset is not working with gst
28526 Original commit message from CVS:
28527 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
28528 More debug to trace why my USB headset is not working with gst
28530 2006-04-07 17:18:11 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28532 gst/playback/gstplaybasebin.c: Clean up our group elements properly in the case where it never got committed - it sti...
28533 Original commit message from CVS:
28534 * gst/playback/gstplaybasebin.c: (group_destroy):
28535 Clean up our group elements properly in the case where it never
28536 got committed - it still got added unconditionally to the bin.
28538 2006-04-07 15:14:32 +0000 Wim Taymans <wim.taymans@gmail.com>
28540 ext/theora/theoradec.c: Unref unhandled events.
28541 Original commit message from CVS:
28542 * ext/theora/theoradec.c: (theora_dec_sink_event),
28543 (theora_handle_data_packet), (theora_dec_chain):
28544 Unref unhandled events.
28545 Protect against empty buffers.
28546 Perform QoS on running time.
28548 2006-04-07 13:24:54 +0000 Michael Smith <msmith@xiph.org>
28550 ext/vorbis/vorbisenc.c: Remove leaks from vorbisenc.
28551 Original commit message from CVS:
28552 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
28553 (gst_vorbis_enc_chain):
28554 Remove leaks from vorbisenc.
28555 Mostly minor changes, the only significant one is that now the
28556 buffers we set as 'streamheader' on the caps are copies of the
28557 original buffers, to avoid circular refcounting problems.
28559 2006-04-07 09:51:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28561 gst/playback/gstplaybasebin.c: Don't remove our mute-probe if someone else already did so.
28562 Original commit message from CVS:
28563 * gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
28564 Don't remove our mute-probe if someone else already did so.
28565 Don't set a 2nd one if there is already one pending on the pad.
28566 * gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
28568 When a seek fails, ensure that playbin is still set back to playing.
28569 * gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
28570 (mpeg_ts_type_find), (plugin_init):
28571 Add a typefind function for mpeg-ts streams.
28573 2006-04-06 11:40:45 +0000 Andy Wingo <wingo@pobox.com>
28576 * gst/audiotestsrc/gstaudiotestsrc.c:
28577 * gst/videorate/gstvideorate.c:
28578 gst/videorate/gstvideorate.c (gst_video_rate_reset)
28579 Original commit message from CVS:
28580 2006-04-06 Andy Wingo <wingo@pobox.com>
28581 * gst/videorate/gstvideorate.c (gst_video_rate_reset)
28582 (gst_video_rate_init): Caps-related parameters should not be reset
28583 by a flush -- move their inits to the instance init function.
28584 (gst_video_rate_flush_prev): Don't complain if gst_pad_push
28585 is not OK, just return the result.
28586 * gst/audiotestsrc/gstaudiotestsrc.c
28587 (gst_audio_test_src_class_init)
28588 (gst_audio_test_src_get_times): Re-enable is-live=true, as was
28589 broken by Stefan's commit on 24 March.
28591 2006-04-06 10:50:14 +0000 Andy Wingo <wingo@pobox.com>
28593 ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on buffers being pushed out. Fixes oggmux ! multifdsink.
28594 Original commit message from CVS:
28595 2006-04-06 Andy Wingo <wingo@pobox.com>
28596 * ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
28597 buffers being pushed out. Fixes oggmux ! multifdsink.
28599 2006-04-05 13:05:25 +0000 Tim-Philipp Müller <tim@centricular.net>
28601 ext/vorbis/: Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make vorbisenc adhere to the official nomenclature; u...
28602 Original commit message from CVS:
28603 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
28604 (gst_vorbis_dec_init), (vorbis_dec_finalize):
28605 * ext/vorbis/vorbisdec.h:
28606 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_add_interfaces),
28607 (gst_vorbis_enc_base_init), (gst_vorbis_enc_class_init),
28608 (gst_vorbis_enc_sink_setcaps), (gst_vorbis_enc_convert_src),
28609 (gst_vorbis_enc_convert_sink), (gst_vorbis_enc_get_query_types),
28610 (gst_vorbis_enc_src_query), (gst_vorbis_enc_sink_query),
28611 (gst_vorbis_enc_init), (gst_vorbis_enc_get_tag_value),
28612 (gst_vorbis_enc_metadata_set1), (gst_vorbis_enc_set_metadata),
28613 (gst_vorbis_enc_setup), (gst_vorbis_enc_clear),
28614 (gst_vorbis_enc_buffer_from_packet),
28615 (gst_vorbis_enc_buffer_from_header_packet),
28616 (gst_vorbis_enc_push_buffer), (gst_vorbis_enc_push_packet),
28617 (gst_vorbis_enc_set_header_on_caps), (gst_vorbis_enc_sink_event),
28618 (gst_vorbis_enc_chain), (gst_vorbis_enc_output_buffers),
28619 (gst_vorbis_enc_get_property), (gst_vorbis_enc_set_property),
28620 (gst_vorbis_enc_change_state):
28621 * ext/vorbis/vorbisenc.h:
28622 Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make
28623 vorbisenc adhere to the official nomenclature; use boilerplate
28626 2006-04-04 11:20:24 +0000 Andy Wingo <wingo@pobox.com>
28628 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Whoops, fix bug introduced. Bad hacker!
28629 Original commit message from CVS:
28630 2006-04-04 Andy Wingo <wingo@pobox.com>
28631 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
28632 Whoops, fix bug introduced. Bad hacker!
28634 2006-04-04 11:15:00 +0000 Andy Wingo <wingo@pobox.com>
28636 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Properly handle the case where you get EOS before any buffe...
28637 Original commit message from CVS:
28638 2006-04-04 Andy Wingo <wingo@pobox.com>
28639 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
28640 Properly handle the case where you get EOS before any buffers are
28641 received. Use gst_buffer_make_metadata_writable where appropriate.
28643 2006-04-04 10:16:46 +0000 Andy Wingo <wingo@pobox.com>
28645 ext/theora/theoradec.c (theora_handle_data_packet): This value is often negative -- make it signed so as not to wrap ...
28646 Original commit message from CVS:
28647 2006-04-04 Andy Wingo <wingo@pobox.com>
28648 * ext/theora/theoradec.c (theora_handle_data_packet): This value
28649 is often negative -- make it signed so as not to wrap around.
28650 Fixes segfaults introduced on 9 March.
28652 2006-04-03 16:43:10 +0000 Wim Taymans <wim.taymans@gmail.com>
28654 ext/theora/: Don't try to store a gdouble in a gboolean.
28655 Original commit message from CVS:
28656 * ext/theora/gsttheoradec.h:
28657 * ext/theora/theoradec.c: (theora_dec_src_event):
28658 Don't try to store a gdouble in a gboolean.
28661 2006-04-03 12:55:18 +0000 Michael Smith <msmith@xiph.org>
28663 ext/ogg/gstoggmux.c: Oggmux sucks.
28664 Original commit message from CVS:
28665 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
28667 Make it suck slightly less by writing out the final page.
28668 Still can't encode a vorbis-in-ogg file correctly, though.
28670 2006-04-03 08:49:06 +0000 Andy Wingo <wingo@pobox.com>
28672 ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove a g_print.
28673 Original commit message from CVS:
28674 2006-04-03 Andy Wingo <wingo@pobox.com>
28675 * ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
28678 2006-04-03 08:32:21 +0000 Andy Wingo <wingo@pobox.com>
28680 ext/theora/theora.c (plugin_init): Register theoraparse.
28681 Original commit message from CVS:
28682 2006-04-03 Andy Wingo <wingo@pobox.com>
28683 * ext/theora/theora.c (plugin_init): Register theoraparse.
28684 * ext/theora/gsttheoraparse.h:
28685 * ext/theora/theoraparse.c: New files implementing a theora
28686 parser. Now we can properly remux ogg/theora+vorbis, yay.
28688 2006-04-03 08:28:58 +0000 Andy Wingo <wingo@pobox.com>
28690 ext/vorbis/vorbisparse.c: Add some docs and a copyright.
28691 Original commit message from CVS:
28692 2006-04-03 Andy Wingo <wingo@pobox.com>
28693 * ext/vorbis/vorbisparse.c: Add some docs and a copyright.
28695 2006-04-01 15:34:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28699 don't use AS_LIBTOOL_TAGS, it doesn't work
28700 Original commit message from CVS:
28701 don't use AS_LIBTOOL_TAGS, it doesn't work
28703 2006-04-01 11:41:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28706 * ext/pango/gsttextoverlay.c:
28707 * sys/v4l/gstv4lsrc.c:
28708 remove BT8x8 from description, works for more devices
28709 Original commit message from CVS:
28710 remove BT8x8 from description, works for more devices
28712 2006-04-01 11:21:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28714 gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798)
28715 Original commit message from CVS:
28716 * gst/audiotestsrc/gstaudiotestsrc.c:
28717 Fixed the sample pipeline (see #323798)
28719 2006-04-01 09:50:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28721 use AS_VERSION and AS_NANO more cleanups
28722 Original commit message from CVS:
28724 * win32/common/config.h:
28725 * win32/common/config.h.in:
28726 use AS_VERSION and AS_NANO
28729 2006-03-31 17:08:41 +0000 Andy Wingo <wingo@pobox.com>
28731 ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix uninitialized variable return that would happen.
28732 Original commit message from CVS:
28733 2006-03-31 Andy Wingo <wingo@pobox.com>
28734 * ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
28735 uninitialized variable return that would happen.
28737 2006-03-31 16:57:47 +0000 Andy Wingo <wingo@pobox.com>
28739 ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix uninitialized variable return that would never happen.
28740 Original commit message from CVS:
28741 2006-03-31 Andy Wingo <wingo@pobox.com>
28742 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
28743 uninitialized variable return that would never happen.
28745 2006-03-31 16:43:43 +0000 Andy Wingo <wingo@pobox.com>
28747 ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
28748 Original commit message from CVS:
28749 2006-03-31 Andy Wingo <wingo@pobox.com>
28750 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
28751 (vorbis_parse_sink_event): Add an event function to flush our
28752 state on a seek, and to drain buffers on a premature EOS.
28753 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
28754 (vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
28755 (vorbis_parse_chain, vorbis_parse_queue_buffer)
28756 (vorbis_parse_drain_queue): Queue up buffers until we can set
28757 their timestamps and granulepos values.
28758 * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
28759 and keep track of data needed for deriving granulepos and
28760 timestamps for buffers.
28762 2006-03-30 11:05:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28764 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
28765 * pkgconfig/gstreamer-plugins-base.pc.in:
28766 expose pluginsdir so gonlin can use it for tests
28767 Original commit message from CVS:
28768 expose pluginsdir so gonlin can use it for tests
28770 2006-03-30 10:03:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28772 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
28773 * pkgconfig/gstreamer-plugins-base.pc.in:
28774 add ccda to libraries
28775 Original commit message from CVS:
28776 add ccda to libraries
28778 2006-03-29 14:00:08 +0000 j^ <j@bootlab.org>
28780 better/unified long descriptions
28781 Original commit message from CVS:
28782 Patch by: j^ <j at bootlab dot org>
28783 * ext/alsa/gstalsamixerelement.c:
28784 (gst_alsa_mixer_element_class_init):
28785 * ext/alsa/gstalsasink.c:
28786 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
28787 * ext/ogg/gstoggdemux.c:
28788 * ext/ogg/gstoggmux.c:
28789 * ext/ogg/gstoggparse.c:
28790 * ext/pango/gstclockoverlay.c:
28791 * ext/pango/gsttextoverlay.c:
28792 * ext/pango/gsttextrender.c:
28793 * ext/pango/gsttimeoverlay.c:
28794 * ext/theora/theoradec.c:
28795 * ext/theora/theoraenc.c:
28796 * ext/vorbis/vorbisdec.c:
28797 * ext/vorbis/vorbisenc.c:
28798 * gst/audioconvert/gstaudioconvert.c:
28799 * gst/subparse/gstsubparse.c:
28800 * gst/tcp/gstmultifdsink.c:
28801 * gst/tcp/gsttcpclientsink.c:
28802 * gst/tcp/gsttcpclientsrc.c:
28803 * gst/tcp/gsttcpserversink.c:
28804 * gst/tcp/gsttcpserversrc.c:
28805 better/unified long descriptions
28808 2006-03-29 13:54:24 +0000 Wim Taymans <wim.taymans@gmail.com>
28810 tests/examples/seek/seek.c: Don't let double and tripple clicks mess up our state.
28811 Original commit message from CVS:
28812 * tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
28814 Don't let double and tripple clicks mess up our state.
28816 2006-03-28 13:13:43 +0000 Tim-Philipp Müller <tim@centricular.net>
28818 gst/playback/gstplaybin.c: Error out gracefully when we can't create any of the usual conversion elements for some re...
28819 Original commit message from CVS:
28820 * gst/playback/gstplaybin.c: (gen_video_element),
28821 (gen_text_element), (gen_audio_element), (gen_vis_element):
28822 Error out gracefully when we can't create any of the usual
28823 conversion elements for some reason. Also, don't try to
28824 create an audioscale (sic) element that's not used anyway.
28826 2006-03-28 10:21:52 +0000 Tim-Philipp Müller <tim@centricular.net>
28828 gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
28829 Original commit message from CVS:
28830 * gst/playback/gstplaybasebin.c: (setup_source):
28831 Don't post RESOURCE_NOT_FOUND error when we can't find a source
28832 element for a particular protocol, that's confusing for users.
28833 Instead, post a RESOURCE_FAILED error, so that our own error
28834 message is actually shown in totem etc. (#336303).
28836 2006-03-27 16:36:46 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
28838 ext/gnomevfs/gstgnomevfssrc.c: Fix some minor memory leaks (#336194).
28839 Original commit message from CVS:
28840 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
28841 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
28842 (gst_gnome_vfs_src_get_icy_metadata):
28843 Fix some minor memory leaks (#336194).
28845 2006-03-27 16:15:00 +0000 Tim-Philipp Müller <tim@centricular.net>
28847 ext/gnomevfs/: Make gnomevfssink accept filenames as well as URIs for the "location" property, just like gnomevfssrc ...
28848 Original commit message from CVS:
28849 * ext/gnomevfs/gstgnomevfs.c:
28850 (gst_gnome_vfs_location_to_uri_string):
28851 * ext/gnomevfs/gstgnomevfs.h:
28852 * ext/gnomevfs/gstgnomevfssink.c:
28853 (gst_gnome_vfs_sink_set_property):
28854 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
28855 Make gnomevfssink accept filenames as well as URIs for the
28856 "location" property, just like gnomevfssrc does (and
28857 filesrc/filesink do) (#336190).
28859 2006-03-24 20:35:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28861 tests/check/generic/clock-selection.c: set to NULL before unreffing, fixes a valgrind leak.
28862 Original commit message from CVS:
28863 * tests/check/generic/clock-selection.c: (GST_START_TEST):
28864 set to NULL before unreffing, fixes a valgrind leak.
28865 Why was this not triggering the error that an object needs to
28866 be NULL before unreffing ?
28867 * win32/common/config.h:
28870 2006-03-24 17:57:39 +0000 Tim-Philipp Müller <tim@centricular.net>
28872 gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
28873 Original commit message from CVS:
28874 * gst/subparse/gstsubparse.c: (convert_encoding),
28875 (gst_sub_parse_change_state):
28876 * gst/subparse/gstsubparse.h:
28877 Text subtitle files may or may not be UTF-8. If it's not, we
28878 don't really want to see '?' characters in place of non-ASCII
28879 characters like accented characters. So let's assume the input
28880 is UTF-8 until we come across text that is clearly not. If it's
28881 not UTF-8, we don't really know what it is, so try the following:
28882 (a) see whether the GST_SUBTITLE_ENCODING environment variable
28883 is set; if not, check (b) if the current locale encoding is
28884 non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
28885 the current locale encoding is UTF-8 and the environment variable
28886 was not set to any particular encoding. Not perfect, but better
28887 than nothing (and better than before, I think) (fixes #172848).
28889 2006-03-24 17:39:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28891 * docs/plugins/tmpl/.gitignore:
28892 * tests/check/libs/.gitignore:
28893 * tests/check/pipelines/.gitignore:
28894 * tests/examples/volume/.gitignore:
28896 Original commit message from CVS:
28899 2006-03-24 17:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28901 configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
28902 Original commit message from CVS:
28903 2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
28905 update core requirement to 0.10.4.1 because of async_playback
28906 vmethod on GstBaseSink
28908 2006-03-24 17:11:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28910 use DEBUG_FUNCPTR for collectpads
28911 Original commit message from CVS:
28912 * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
28913 * gst/adder/gstadder.c: (gst_adder_init):
28914 use DEBUG_FUNCPTR for collectpads
28916 2006-03-24 14:11:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28919 don't go through check-torture if no check installed
28920 Original commit message from CVS:
28921 don't go through check-torture if no check installed
28923 2006-03-24 10:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28925 Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
28926 Original commit message from CVS:
28927 * docs/plugins/Makefile.am:
28928 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
28929 * docs/plugins/gst-plugins-base-plugins-sections.txt:
28930 * ext/cdparanoia/gstcdparanoiasrc.c:
28931 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
28932 (gst_gnome_vfs_sink_class_init):
28933 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
28934 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
28935 * ext/ogg/gstoggmux.c:
28936 * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
28937 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
28938 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
28939 * ext/pango/gsttextoverlay.c:
28940 * ext/pango/gsttextrender.c:
28941 * ext/theora/theoradec.c:
28942 * ext/theora/theoraenc.c:
28943 * ext/vorbis/vorbisdec.c:
28944 * ext/vorbis/vorbisenc.c:
28945 * gst-libs/gst/audio/gstaudiofilter.c:
28946 (gst_audio_filter_base_init):
28947 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
28948 (gst_audio_filter_template_base_init):
28949 * gst/adder/gstadder.c: (gst_adder_get_type):
28950 * gst/adder/gstadder.h:
28951 * gst/audioconvert/gstaudioconvert.c:
28952 * gst/audiotestsrc/gstaudiotestsrc.c:
28953 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
28954 (gst_audio_test_src_create):
28955 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
28956 * gst/playback/gstdecodebin.c:
28957 * gst/playback/gstplaybin.c:
28958 * gst/playback/gststreamselector.c:
28959 (gst_stream_selector_base_init):
28960 * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
28961 * gst/volume/gstvolume.c:
28962 * sys/v4l/gstv4lmjpegsink.c:
28963 * sys/v4l/gstv4lmjpegsrc.c:
28964 * tests/check/libs/cddabasesrc.c:
28965 * tests/old/examples/gob/gst-identity2.gob:
28966 Add docs for adder, use GST_ELEMENT_DETAILS macro,
28967 define GstElementDetails at the top
28969 2006-03-23 21:48:18 +0000 Sébastien Moutte <sebastien@moutte.net>
28971 win32/common/libgstinterfaces.def: Add a lot of export functions for gst-python
28972 Original commit message from CVS:
28973 * win32/common/libgstinterfaces.def:
28974 Add a lot of export functions for gst-python
28975 * win32/common/libgstinterfaces.dsp:
28976 Add a missing include folder in the project configuration
28978 2006-03-23 16:58:03 +0000 Wim Taymans <wim.taymans@gmail.com>
28980 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
28981 Original commit message from CVS:
28982 * gst-libs/gst/audio/gstbaseaudiosrc.c:
28983 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
28984 (gst_base_audio_src_change_state):
28985 Fix audio sources, forgot to make the ringbuffer
28988 2006-03-23 16:29:58 +0000 Wim Taymans <wim.taymans@gmail.com>
28990 gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
28991 Original commit message from CVS:
28992 * gst-libs/gst/audio/gstbaseaudiosrc.c:
28993 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
28994 (gst_base_audio_src_change_state):
28995 unparent instead of unref the ringbuffer.
28997 2006-03-23 16:24:23 +0000 Wim Taymans <wim.taymans@gmail.com>
28999 gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
29000 Original commit message from CVS:
29001 * gst-libs/gst/audio/gstbaseaudiosink.c:
29002 (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
29003 (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
29004 Implement new async_play vmethod to start slaving and allow
29005 playback start in case of async PLAY state changes.
29006 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
29007 Enable QoS with new method in base class.
29009 2006-03-23 11:18:19 +0000 Julien MOUTTE <julien@moutte.net>
29011 gst/videotestsrc/gstvideotestsrc.c: Partially handle 0 framerate, only EOS after the first frame is missing.
29012 Original commit message from CVS:
29013 Patch by: Julien MOUTTE <julien at moutte dot net>
29014 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
29015 (gst_video_test_src_do_seek), (gst_video_test_src_create):
29016 Partially handle 0 framerate, only EOS after the first frame
29019 2006-03-23 09:38:59 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
29021 gst/: Patch for support of YVU9 AVI files (#334822)
29022 Original commit message from CVS:
29023 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
29024 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
29025 (gst_riff_create_video_template_caps):
29026 * gst/ffmpegcolorspace/avcodec.h:
29027 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
29028 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
29029 (gst_ffmpegcsp_avpicture_fill):
29030 * gst/ffmpegcolorspace/imgconvert.c:
29031 Patch for support of YVU9 AVI files (#334822)
29033 2006-03-22 15:29:25 +0000 Edward Hervey <bilboed@bilboed.com>
29035 docs/design/design-decodebin.txt: Added design document for new decodebin text/x-pango-markup is also a default targe...
29036 Original commit message from CVS:
29037 * docs/design/design-decodebin.txt:
29038 Added design document for new decodebin
29039 (Target Caps): text/x-pango-markup is also a default target caps.
29041 2006-03-22 15:11:47 +0000 Edward Hervey <bilboed@bilboed.com>
29043 docs/design/design-decodebin.txt: Added design document for new decodebin
29044 Original commit message from CVS:
29045 * docs/design/design-decodebin.txt:
29046 Added design document for new decodebin
29048 2006-03-22 12:33:09 +0000 Wim Taymans <wim.taymans@gmail.com>
29050 gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
29051 Original commit message from CVS:
29052 * gst-libs/gst/audio/gstbaseaudiosink.c:
29053 (gst_base_audio_sink_dispose):
29054 Since we _parent the ringbuffer, we also need to
29055 _unparent instead of a plain _unref.
29057 2006-03-22 12:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
29059 tests/examples/seek/seek.c: Add scrub checkbox.
29060 Original commit message from CVS:
29061 * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
29062 (stop_seek), (scrub_toggle_cb), (main):
29063 Add scrub checkbox.
29065 2006-03-21 17:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
29067 ext/ogg/gstoggparse.c: Fix very inefficient usage of linked lists (#335365).
29068 Original commit message from CVS:
29069 * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
29070 (gst_ogg_parse_chain):
29071 Fix very inefficient usage of linked lists (#335365).
29073 2006-03-21 14:26:01 +0000 Edward Hervey <bilboed@bilboed.com>
29075 gcc 4.1 unreferenced pointer fixes.
29076 Original commit message from CVS:
29077 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
29078 * gst/playback/gstplaybin.c: (handoff):
29079 * gst/playback/gststreamselector.c:
29080 (gst_stream_selector_set_property):
29081 gcc 4.1 unreferenced pointer fixes.
29082 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
29083 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
29084 gst_buffer_ref() now takes a GstBuffer*.
29086 2006-03-20 18:09:41 +0000 Julien Moutte <julien@moutte.net>
29088 sys/xvimage/xvimagesink.c: Fix a memleak reported by Jan Schmidt.
29089 Original commit message from CVS:
29090 2006-03-20 Julien MOUTTE <julien@moutte.net>
29091 * sys/xvimage/xvimagesink.c:
29092 (gst_xvimagesink_get_format_from_caps): Fix a memleak reported
29095 2006-03-19 11:37:46 +0000 Tim-Philipp Müller <tim@centricular.net>
29097 gst/typefind/gsttypefindfunctions.c: Can't do tag preferences via probability, as tags would then lose against types ...
29098 Original commit message from CVS:
29099 * gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
29100 (id3v1_type_find), (apetag_type_find), (plugin_init):
29101 Can't do tag preferences via probability, as tags would then
29102 lose against types that are recognised with MAXIMUM probability
29103 (like .wav); so let all tag typefinders return MAXIMUM themselves
29104 and order them via the rank. Split ID3v1 and ID3v2 typefinders so
29105 that we can prefer APE to ID3v1 (fixes #335028).
29107 2006-03-17 17:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
29109 gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
29110 Original commit message from CVS:
29111 * gst-libs/gst/audio/gstbaseaudiosink.c:
29112 (gst_base_audio_sink_change_state):
29113 * gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
29114 (gst_ring_buffer_may_start):
29115 * gst-libs/gst/audio/gstringbuffer.h:
29116 Only start playback if we are playing.
29117 should fix #330748.
29119 2006-03-17 13:11:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29121 Revert accidental commits to these files.
29122 Original commit message from CVS:
29123 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
29124 * win32/common/config.h:
29125 Revert accidental commits to these files.
29127 2006-03-16 20:01:03 +0000 Michal Benes <michal.benes@xeris.cz>
29129 tests/Makefile.am: Don't try to build tests in tests/icles if we don't have X (#323852)
29130 Original commit message from CVS:
29131 Patch by: Michal Benes <michal dot benes at xeris dot cz>
29132 * tests/Makefile.am:
29133 Don't try to build tests in tests/icles if we
29134 don't have X (#323852)
29136 2006-03-16 13:08:01 +0000 Tim-Philipp Müller <tim@centricular.net>
29138 gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
29139 Original commit message from CVS:
29140 * gst-libs/gst/tag/gstid3tag.c:
29141 Add TXXX frame identifiers for replaygain stuff as used
29142 by some taggers (see #323721).
29144 2006-03-16 10:22:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29146 gst/playback/gststreamselector.c: Preserve the existing buggy streamselector behaviour by performing a fallback buffe...
29147 Original commit message from CVS:
29148 * gst/playback/gststreamselector.c:
29149 (gst_stream_selector_set_property),
29150 (gst_stream_selector_bufferalloc):
29151 Preserve the existing buggy streamselector behaviour by performing
29152 a fallback buffer allocation when downstream isn't linked yet.
29153 This should really be fixed in playbin by blocking pads until it's
29155 Also, use gst_pad_alloc_buffer instead of
29156 gst_pad_alloc_buffer_and_set.
29158 2006-03-15 22:40:08 +0000 Tim-Philipp Müller <tim@centricular.net>
29160 gst-libs/gst/tag/gstid3tag.c: Don't crash on unknown ID3v2 TXXX frames.
29161 Original commit message from CVS:
29162 * gst-libs/gst/tag/gstid3tag.c:
29163 Don't crash on unknown ID3v2 TXXX frames.
29165 2006-03-15 17:59:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29167 ext/alsa/gstalsasink.c: Chain up to the parent finalize method.
29168 Original commit message from CVS:
29169 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
29170 Chain up to the parent finalize method.
29171 Add 32-bit sample size to the template caps.
29172 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
29173 (gst_riff_create_video_template_caps):
29174 Add the fourcc that the VMWare codec uses.
29175 * gst/playback/gststreamselector.c:
29176 (gst_stream_selector_set_property),
29177 (gst_stream_selector_bufferalloc),
29178 (gst_stream_selector_request_new_pad):
29179 For the active pad, forward buffer-alloc requests, otherwise
29180 return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
29181 having to memcpy every frame when used by playbin.
29182 * gst/tcp/gstmultifdsink.c:
29183 (gst_multi_fd_sink_handle_client_write):
29184 Get negotiated caps from the sink pad, rather than the sink
29187 2006-03-15 17:11:34 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
29189 ext/gnomevfs/gstgnomevfssrc.c: Don't forget to set src->callbacks_pushed to FALSE again when popping them, otherwise ...
29190 Original commit message from CVS:
29191 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
29192 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_pop_callbacks):
29193 Don't forget to set src->callbacks_pushed to FALSE again when
29194 popping them, otherwise re-activation in a different mode won't
29197 2006-03-15 11:30:29 +0000 Sebastien Moutte <sebastien@moutte.net>
29199 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
29200 Original commit message from CVS:
29201 Patch by: Sebastien Moutte <sebastien moutte net>
29202 * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
29203 (gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
29204 (gst_ffmpeg_smpfmt_to_caps):
29205 Replace __VA_ARGS__ caps creation macros with varargs functions.
29206 Makes things compile on MSVC (#320765), looks nicer, and we can
29207 tell the compiler to check for the NULL terminator.
29209 2006-03-14 15:13:04 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
29211 gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
29212 Original commit message from CVS:
29213 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
29214 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29215 Make sure the buffer we copy into is really always big
29216 enough, this time for real (#333488).
29218 2006-03-14 13:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
29220 gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
29221 Original commit message from CVS:
29222 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29223 Add support for 24bpp DIB (#305279).
29225 2006-03-14 11:11:59 +0000 Wim Taymans <wim.taymans@gmail.com>
29227 gst/: Re-enable QoS after the release.
29228 Original commit message from CVS:
29229 * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
29230 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
29231 * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
29232 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
29233 (gst_video_scale_init), (gst_video_scale_src_event):
29234 Re-enable QoS after the release.
29235 Rework videoscale to use the base class src_event handler.
29237 2006-03-14 09:51:01 +0000 Tim-Philipp Müller <tim@centricular.net>
29239 configure.ac: back to CVS.
29240 Original commit message from CVS:
29244 === release 0.10.5 ===
29246 2006-03-13 19:50:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29252 * docs/plugins/inspect/plugin-adder.xml:
29253 * docs/plugins/inspect/plugin-alsa.xml:
29254 * docs/plugins/inspect/plugin-audioconvert.xml:
29255 * docs/plugins/inspect/plugin-audiorate.xml:
29256 * docs/plugins/inspect/plugin-audioresample.xml:
29257 * docs/plugins/inspect/plugin-audiotestsrc.xml:
29258 * docs/plugins/inspect/plugin-cdparanoia.xml:
29259 * docs/plugins/inspect/plugin-decodebin.xml:
29260 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
29261 * docs/plugins/inspect/plugin-gnomevfs.xml:
29262 * docs/plugins/inspect/plugin-libvisual.xml:
29263 * docs/plugins/inspect/plugin-ogg.xml:
29264 * docs/plugins/inspect/plugin-pango.xml:
29265 * docs/plugins/inspect/plugin-playbin.xml:
29266 * docs/plugins/inspect/plugin-subparse.xml:
29267 * docs/plugins/inspect/plugin-tcp.xml:
29268 * docs/plugins/inspect/plugin-theora.xml:
29269 * docs/plugins/inspect/plugin-typefindfunctions.xml:
29270 * docs/plugins/inspect/plugin-video4linux.xml:
29271 * docs/plugins/inspect/plugin-videorate.xml:
29272 * docs/plugins/inspect/plugin-videoscale.xml:
29273 * docs/plugins/inspect/plugin-videotestsrc.xml:
29274 * docs/plugins/inspect/plugin-volume.xml:
29275 * docs/plugins/inspect/plugin-vorbis.xml:
29276 * docs/plugins/inspect/plugin-ximagesink.xml:
29277 * docs/plugins/inspect/plugin-xvimagesink.xml:
29278 * win32/common/config.h:
29280 Original commit message from CVS:
29283 2006-03-13 17:28:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29300 Original commit message from CVS:
29303 2006-03-13 11:17:19 +0000 Tim-Philipp Müller <tim@centricular.net>
29305 docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
29306 Original commit message from CVS:
29307 * docs/plugins/Makefile.am:
29308 Part of previous cdparanoiasrc docs fixes, forgot to commit.
29310 2006-03-12 14:56:31 +0000 Tim-Philipp Müller <tim@centricular.net>
29312 docs/plugins/: Add cdparanoiasrc to docs.
29313 Original commit message from CVS:
29314 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29315 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29316 * docs/plugins/gst-plugins-base-plugins.hierarchy:
29317 Add cdparanoiasrc to docs.
29318 * gst-libs/gst/cdda/gstcddabasesrc.c:
29319 More GstCddaBaseSrc docs.
29321 2006-03-12 13:47:22 +0000 Tim-Philipp Müller <tim@centricular.net>
29323 Add new API to libgsttag: gst_tag_from_id3_user_tag().
29324 Original commit message from CVS:
29325 * docs/libs/gst-plugins-base-libs-sections.txt:
29326 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
29327 * gst-libs/gst/tag/tag.h:
29328 Add new API to libgsttag: gst_tag_from_id3_user_tag().
29330 2006-03-11 19:47:16 +0000 Tim-Philipp Müller <tim@centricular.net>
29332 gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
29333 Original commit message from CVS:
29334 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29335 NULL-terminate array of mpeg4 video file extensions.
29336 Fixes crash on PPC (#334226).
29338 2006-03-11 16:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
29340 ext/gnomevfs/gstgnomevfssrc.c: gnome_vfs_uri_is_local() alone is not a good indicator whether we can operate in pull-...
29341 Original commit message from CVS:
29342 * ext/gnomevfs/gstgnomevfssrc.c:
29343 (gst_gnome_vfs_src_check_get_range):
29344 gnome_vfs_uri_is_local() alone is not a good indicator
29345 whether we can operate in pull-mode with a specific URI,
29346 as it returns FALSE for file:// URIs that point to an
29347 NFS-mounted path. Be more conservative here: whitelist
29348 local files, blacklist http URIs and use the old
29349 mechanism for anything else (fixes #334216).
29351 2006-03-10 19:15:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29353 configure.ac: back to trunk
29354 Original commit message from CVS:
29358 === release 0.10.4 ===
29360 2006-03-10 19:05:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29366 * docs/plugins/gst-plugins-base-plugins.args:
29367 * docs/plugins/inspect/plugin-adder.xml:
29368 * docs/plugins/inspect/plugin-alsa.xml:
29369 * docs/plugins/inspect/plugin-audioconvert.xml:
29370 * docs/plugins/inspect/plugin-audiorate.xml:
29371 * docs/plugins/inspect/plugin-audioresample.xml:
29372 * docs/plugins/inspect/plugin-audiotestsrc.xml:
29373 * docs/plugins/inspect/plugin-cdparanoia.xml:
29374 * docs/plugins/inspect/plugin-decodebin.xml:
29375 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
29376 * docs/plugins/inspect/plugin-gnomevfs.xml:
29377 * docs/plugins/inspect/plugin-libvisual.xml:
29378 * docs/plugins/inspect/plugin-ogg.xml:
29379 * docs/plugins/inspect/plugin-pango.xml:
29380 * docs/plugins/inspect/plugin-playbin.xml:
29381 * docs/plugins/inspect/plugin-subparse.xml:
29382 * docs/plugins/inspect/plugin-tcp.xml:
29383 * docs/plugins/inspect/plugin-theora.xml:
29384 * docs/plugins/inspect/plugin-typefindfunctions.xml:
29385 * docs/plugins/inspect/plugin-video4linux.xml:
29386 * docs/plugins/inspect/plugin-videorate.xml:
29387 * docs/plugins/inspect/plugin-videoscale.xml:
29388 * docs/plugins/inspect/plugin-videotestsrc.xml:
29389 * docs/plugins/inspect/plugin-volume.xml:
29390 * docs/plugins/inspect/plugin-vorbis.xml:
29391 * docs/plugins/inspect/plugin-ximagesink.xml:
29392 * docs/plugins/inspect/plugin-xvimagesink.xml:
29394 * win32/common/config.h:
29396 Original commit message from CVS:
29399 2006-03-10 12:37:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29401 gst-libs/gst/video/gstvideosink.c: Disable max-lateness by setting it to -1 for now, so that we can bed QoS stuff in ...
29402 Original commit message from CVS:
29403 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
29404 Disable max-lateness by setting it to -1 for now, so that
29405 we can bed QoS stuff in thoroughly between now and the next
29408 2006-03-10 11:09:23 +0000 Fabrizio <fabrizio.ge@tiscali.it>
29410 gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
29411 Original commit message from CVS:
29412 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29413 Make sure we don't read beyond the palette buffer in case of
29414 broken or manipulated files (#333488, patch by: Fabrizio
29417 2006-03-10 10:44:02 +0000 Edward Hervey <bilboed@bilboed.com>
29419 gst/typefind/gsttypefindfunctions.c: Fix for variable not initialized.
29420 Original commit message from CVS:
29421 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
29422 Fix for variable not initialized.
29424 2006-03-09 19:02:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29427 * docs/libs/tmpl/gstringbuffer.sgml:
29442 * win32/common/config.h:
29444 Original commit message from CVS:
29447 2006-03-09 17:58:00 +0000 Wim Taymans <wim.taymans@gmail.com>
29449 ext/libvisual/visual.c: Small cleanups.
29450 Original commit message from CVS:
29451 * ext/libvisual/visual.c: (gst_visual_get_type),
29452 (gst_visual_src_setcaps), (gst_vis_src_negotiate),
29453 (gst_visual_chain):
29455 * ext/theora/gsttheoradec.h:
29456 * ext/theora/theoradec.c: (gst_theora_dec_init),
29457 (gst_theora_dec_reset), (_theora_granule_time),
29458 (theora_dec_src_convert), (theora_dec_sink_convert),
29459 (theora_dec_src_query), (theora_dec_src_event),
29460 (theora_dec_sink_event), (theora_handle_comment_packet),
29461 (theora_handle_header_packet), (theora_dec_push),
29462 (theora_handle_data_packet), (theora_dec_chain),
29463 (theora_dec_change_state):
29466 2006-03-09 17:50:59 +0000 Wim Taymans <wim.taymans@gmail.com>
29468 ext/gnomevfs/gstgnomevfssrc.c: Some cleanups.
29469 Original commit message from CVS:
29470 * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
29471 (audiocast_register_listener), (gst_gnome_vfs_src_start):
29474 2006-03-09 17:45:39 +0000 Wim Taymans <wim.taymans@gmail.com>
29476 ext/ogg/gstoggdemux.c: Don't try to activate NULL chains.
29477 Original commit message from CVS:
29478 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
29479 Don't try to activate NULL chains.
29481 2006-03-09 16:30:41 +0000 Tim-Philipp Müller <tim@centricular.net>
29483 gst/typefind/gsttypefindfunctions.c: Fix invalid memory access to region before peek'd data (#332964).
29484 Original commit message from CVS:
29485 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
29486 Fix invalid memory access to region before peek'd data (#332964).
29488 2006-03-09 15:05:03 +0000 Christophe Fergeau <teuf@gnome.org>
29491 Original commit message from CVS:
29492 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init):
29493 * ext/pango/gsttextrender.c: (gst_text_render_init):
29494 * gst/adder/gstadder.c: (gst_adder_init):
29495 Don't leak padtemplates, patch by Christophe Fergeau,
29498 2006-03-09 12:56:35 +0000 Tim-Philipp Müller <tim@centricular.net>
29500 gst/subparse/gstsubparse.c: Fix invalid memory access: make sure string passed to regexec() is NUL-termianted.
29501 Original commit message from CVS:
29502 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
29503 Fix invalid memory access: make sure string passed to
29504 regexec() is NUL-termianted.
29506 2006-03-09 12:37:59 +0000 Tim-Philipp Müller <tim@centricular.net>
29508 gst/typefind/gsttypefindfunctions.c: Refactor mpeg/audio typefinding to make it more maintainable and easier to fine-...
29509 Original commit message from CVS:
29510 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
29512 Refactor mpeg/audio typefinding to make it more maintainable
29513 and easier to fine-tune. Make probing into middle of the file
29514 work properly (fixes #333900, also see #152688).
29516 2006-03-09 11:10:03 +0000 Tim-Philipp Müller <tim@centricular.net>
29518 gst/typefind/gsttypefindfunctions.c: Remove part from previous commit that was bogus: g_utf8_validate() does in fact ...
29519 Original commit message from CVS:
29520 * gst/typefind/gsttypefindfunctions.c:
29521 (utf8_type_find_have_valid_utf8_at_offset):
29522 Remove part from previous commit that was bogus:
29523 g_utf8_validate() does in fact not accept embedded
29524 zeroes, so we don't need to check for those (thanks
29525 to Mike for the hint).
29527 2006-03-08 17:11:29 +0000 Tim-Philipp Müller <tim@centricular.net>
29529 gst/typefind/gsttypefindfunctions.c: Make plain/text typefinder more conservative: firstly, check for embedded zeroes...
29530 Original commit message from CVS:
29531 * gst/typefind/gsttypefindfunctions.c:
29532 (utf8_type_find_count_embedded_zeroes),
29533 (utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
29534 Make plain/text typefinder more conservative: firstly, check
29535 for embedded zeroes, which are perfectly valid UTF-8 characters,
29536 but also a fairly good sign that something is not a plain text
29537 file; secondly, probe into the middle of the file if possible.
29538 If we can't probe into the middle, limit the probability value
29539 to be returned to TYPE_FIND_POSSIBLE (see #333900).
29541 2006-03-08 11:34:45 +0000 Michael Smith <msmith@xiph.org>
29543 gst/typefind/gsttypefindfunctions.c: Make typefind function name for mpeg4 video unique.
29544 Original commit message from CVS:
29545 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29546 Make typefind function name for mpeg4 video unique.
29548 2006-03-08 09:53:31 +0000 Wim Taymans <wim.taymans@gmail.com>
29550 ext/libvisual/visual.c: Cleanups, post nice errors.
29551 Original commit message from CVS:
29552 * ext/libvisual/visual.c: (gst_visual_init),
29553 (gst_visual_clear_actors), (gst_visual_dispose),
29554 (gst_visual_reset), (gst_visual_src_setcaps),
29555 (gst_visual_sink_setcaps), (gst_vis_src_negotiate),
29556 (gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
29557 (gst_visual_chain), (gst_visual_change_state):
29558 Cleanups, post nice errors.
29559 Handle sink and src events.
29560 Implement simple QoS.
29561 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
29562 Use new basesink methods to configure max-lateness.
29564 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
29565 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
29566 Debug statement cleanups.
29567 * gst/volume/gstvolume.c: (gst_volume_class_init):
29570 2006-03-08 09:50:23 +0000 Tim-Philipp Müller <tim@centricular.net>
29572 ext/pango/gsttextoverlay.c: Revert API/ABI break from March 1. Keep 'halign' and 'valign' as string type properties, ...
29573 Original commit message from CVS:
29574 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
29575 (gst_text_overlay_init), (gst_text_overlay_set_property),
29576 (gst_text_overlay_get_property):
29577 Revert API/ABI break from March 1. Keep 'halign' and 'valign'
29578 as string type properties, but mark them deprecated. Add
29579 'halignment' and 'valignment' properties that use enums
29580 instead of strings.
29582 2006-03-08 09:37:12 +0000 Fabrizio <fabrizio.ge@tiscali.it>
29584 gst-libs/gst/riff/riff-media.c: Allow palettes with less than 256 colours in AVI files
29585 Original commit message from CVS:
29586 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29587 Allow palettes with less than 256 colours in AVI files
29588 (#333488, patch by: Fabrizio Gennari).
29590 2006-03-07 21:56:09 +0000 Julien Moutte <julien@moutte.net>
29592 ext/pango/gsttextoverlay.c: Fix wrong EOS handling on text pad. We were releasing the queued text buffer when we shou...
29593 Original commit message from CVS:
29594 2006-03-07 Julien MOUTTE <julien@moutte.net>
29595 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
29596 (gst_text_overlay_video_event): Fix wrong EOS handling on text
29597 pad. We were releasing the queued text buffer when we should keep
29598 it until video pad gets EOS or discard the text buffer because it's
29599 too old. That was eating the last subtitle buffer. Add some more
29602 2006-03-07 17:28:36 +0000 Tim-Philipp Müller <tim@centricular.net>
29604 ext/pango/gsttextoverlay.c: Fix invalid memory access (we can't access a buffer after it's been pushed downstream wit...
29605 Original commit message from CVS:
29606 * ext/pango/gsttextoverlay.c: (gst_text_overlay_render_text),
29607 (gst_text_overlay_video_chain):
29608 Fix invalid memory access (we can't access a buffer after it's been
29609 pushed downstream without taking a reference); fix memory leak (if
29610 there's no text to render, bail out before allocating stuff).
29612 2006-03-07 15:08:15 +0000 Tim-Philipp Müller <tim@centricular.net>
29614 ext/pango/gsttextoverlay.*: If input is plain text, escape it before passing it to pango_layout_set_markup().
29615 Original commit message from CVS:
29616 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
29617 (gst_text_overlay_setcaps_txt), (gst_text_overlay_video_chain):
29618 * ext/pango/gsttextoverlay.h:
29619 If input is plain text, escape it before passing it to
29620 pango_layout_set_markup().
29622 2006-03-07 13:01:21 +0000 Tim-Philipp Müller <tim@centricular.net>
29624 gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
29625 Original commit message from CVS:
29626 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
29627 Don't ignore flow return from gst_pad_push().
29629 2006-03-07 12:49:03 +0000 Christophe Fergeau <teuf@gnome.org>
29631 Don't leak references returned by gst_pad_get_parent()
29632 Original commit message from CVS:
29633 * ext/libvisual/visual.c: (gst_visual_getcaps),
29634 (gst_visual_src_setcaps), (gst_visual_sink_setcaps):
29635 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
29636 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
29637 (gst_vorbisenc_convert_sink):
29638 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
29639 (gst_audio_duration_from_pad_buffer):
29640 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
29641 (gst_audio_filter_chain):
29642 * gst-libs/gst/rtp/gstbasertpdepayload.c:
29643 (gst_base_rtp_depayload_setcaps):
29644 * gst-libs/gst/video/video.c: (gst_video_frame_rate),
29645 (gst_video_get_size):
29646 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
29647 Don't leak references returned by gst_pad_get_parent()
29648 (#333663, based on patch by: Christophe Fergeau).
29650 2006-03-06 20:52:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29652 ext/gnomevfs/gstgnomevfssink.c: change location param details
29653 Original commit message from CVS:
29654 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
29655 change location param details
29656 * gst/volume/gstvolume.c: (plugin_init):
29657 correct plugin description
29659 2006-03-06 20:07:55 +0000 Tim-Philipp Müller <tim@centricular.net>
29661 ext/gnomevfs/gstgnomevfssrc.c: Override GstBaseSrc::check_get_range() in order to avoid opening the resource just to ...
29662 Original commit message from CVS:
29663 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
29664 (gst_gnome_vfs_src_check_get_range):
29665 Override GstBaseSrc::check_get_range() in order to avoid opening
29666 the resource just to check whether we can operate in pull-mode or
29667 not - we can predict that pretty well from the URI alone. Should
29668 fix problems with last.fm (#331690). (Requires latest core CVS).
29670 2006-03-06 16:18:51 +0000 Wim Taymans <wim.taymans@gmail.com>
29672 gst-libs/gst/video/gstvideosink.c: Throw away frames that are later than 20 ms.
29673 Original commit message from CVS:
29674 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
29675 (gst_video_sink_class_init):
29676 Throw away frames that are later than 20 ms.
29678 2006-03-06 14:14:47 +0000 Fabrizio <fabrizio.ge@tiscali.it>
29680 gst-libs/gst/riff/riff-media.c:
29681 Original commit message from CVS:
29682 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
29683 Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).
29685 2006-03-05 23:39:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29687 ext/ogg/gstoggmux.c: put Theora BOS pages before others. This hardcodes the Ogg/Theora I profile, but hey.
29688 Original commit message from CVS:
29689 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
29690 (gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
29691 put Theora BOS pages before others. This hardcodes
29692 the Ogg/Theora I profile, but hey.
29694 2006-03-05 23:06:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29696 * ext/ogg/gstoggmux.c:
29697 changed more than 5 lines
29698 Original commit message from CVS:
29699 changed more than 5 lines
29701 2006-03-05 22:57:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29703 ogg muxing of vorbis and theora now has pages ordered correctly again, even with delays.
29704 Original commit message from CVS:
29705 ogg muxing of vorbis and theora now has pages ordered correctly again,
29708 updated with some examples
29709 * ext/theora/theoraenc.c: (granulepos_to_timestamp),
29710 (granulepos_add), (theora_buffer_from_packet):
29711 * ext/vorbis/vorbisenc.c: (granulepos_to_timestamp_offset),
29712 (granulepos_to_timestamp), (gst_vorbisenc_buffer_from_packet),
29713 (gst_vorbisenc_chain):
29714 implement strategy from ext/ogg/README
29715 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
29716 (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
29717 (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_compare_pads),
29718 (gst_ogg_mux_queue_pads), (gst_ogg_mux_collected):
29719 Fix muxer so that oggz-validate is happy with all streams;
29720 except for no eos mark, and the BOS page ordering
29721 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
29722 (check_buffer_granulepos):
29723 * tests/check/pipelines/vorbisenc.c: (check_buffer_granulepos):
29724 update tests to check for OFFSET being set as requested
29725 fixed type of granulepos, it's not a ClockTime
29727 2006-03-05 21:34:23 +0000 Julien Moutte <julien@moutte.net>
29729 sys/xvimage/xvimagesink.c: Check that the xvimage we are creating has a correct size before returning it. (#3...
29730 Original commit message from CVS:
29731 2006-03-05 Julien MOUTTE <julien@moutte.net>
29732 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
29733 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
29734 Check that the xvimage we are creating has a correct size before returning it. (#314897)
29736 2006-03-05 13:44:05 +0000 Tim-Philipp Müller <tim@centricular.net>
29738 gst/typefind/gsttypefindfunctions.c: Give id3 and ape tag typefinders a rank slightly higher than PRIMARY to ensure t...
29739 Original commit message from CVS:
29740 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29741 Give id3 and ape tag typefinders a rank slightly higher
29742 than PRIMARY to ensure they're always run before any of
29743 the other typefinders (in particular wav and mp3) (#324186).
29745 2006-03-05 13:08:37 +0000 Tim-Philipp Müller <tim@centricular.net>
29747 gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
29748 Original commit message from CVS:
29749 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29750 Add support for '3IVD' fourcc (#333403).
29752 2006-03-04 14:35:10 +0000 Tim-Philipp Müller <tim@centricular.net>
29754 configure.ac: Bump requirements to GStreamer CVS for the new error enum.
29755 Original commit message from CVS:
29757 Bump requirements to GStreamer CVS for the new error enum.
29758 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
29759 Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
29760 space left on the device (fixes #333352).
29762 2006-03-03 23:53:50 +0000 Sébastien Moutte <sebastien@moutte.net>
29764 win32/vs6: add a project file for libgstvolume update the workspace
29765 Original commit message from CVS:
29767 add a project file for libgstvolume
29768 update the workspace
29770 2006-03-03 15:26:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29773 * ext/ogg/gstoggmux.c:
29775 Original commit message from CVS:
29778 2006-03-03 15:22:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29780 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
29781 Original commit message from CVS:
29782 2006-03-03 Thomas Vander Stichele <thomas at apestaart dot org>
29783 * ext/theora/theoraenc.c: (theora_set_header_on_caps):
29784 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
29786 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
29787 Set IN_CAPS on header buffers
29789 2006-03-02 18:23:55 +0000 Wim Taymans <wim.taymans@gmail.com>
29791 docs/plugins/: Add audioresample to docs.
29792 Original commit message from CVS:
29793 * docs/plugins/Makefile.am:
29794 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29795 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29796 Add audioresample to docs.
29797 * gst/audioconvert/gstaudioconvert.c:
29799 * gst/audioresample/gstaudioresample.c:
29800 (gst_audioresample_base_init), (gst_audioresample_class_init),
29801 (gst_audioresample_init), (gst_audioresample_dispose),
29802 (audioresample_get_unit_size), (audioresample_transform_caps),
29803 (resample_set_state_from_caps), (audioresample_transform_size),
29804 (audioresample_set_caps), (audioresample_event),
29805 (audioresample_do_output), (audioresample_transform),
29806 (audioresample_pushthrough), (gst_audioresample_set_property),
29807 (gst_audioresample_get_property), (plugin_init):
29808 * gst/audioresample/gstaudioresample.h:
29810 Small code cleanups.
29812 2006-03-02 18:12:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29814 * gst/videorate/Makefile.am:
29816 Original commit message from CVS:
29819 2006-03-02 17:48:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29821 * ext/ogg/gstoggmux.c:
29822 debug using the actual GstPad, that allows us to see the serialno in the padname
29823 Original commit message from CVS:
29824 debug using the actual GstPad, that allows us to see the serialno in the padname
29826 2006-03-02 17:46:36 +0000 Wim Taymans <wim.taymans@gmail.com>
29828 docs/plugins/: Added videoscale to docs.
29829 Original commit message from CVS:
29830 * docs/plugins/Makefile.am:
29831 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29832 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29833 Added videoscale to docs.
29834 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
29835 (gst_video_rate_swap_prev), (gst_video_rate_event),
29836 (gst_video_rate_chain):
29838 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
29839 (gst_video_scale_init), (gst_video_scale_prepare_size),
29840 (gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
29841 (gst_video_scale_fixate_caps), (gst_video_scale_transform):
29842 * gst/videoscale/gstvideoscale.h:
29843 Added docs, examples.
29844 Some code cleanups.
29845 Post errors instead of g_warning.
29847 2006-03-02 17:30:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29849 * ext/ogg/gstoggmux.c:
29850 clean up debug messages
29851 Original commit message from CVS:
29852 clean up debug messages
29854 2006-03-02 17:15:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29856 * ext/ogg/gstoggmux.c:
29857 extra debugging from older version, makes it easier to compare
29858 Original commit message from CVS:
29859 extra debugging from older version, makes it easier to compare
29861 2006-03-02 17:04:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29863 * ext/ogg/gstoggmux.c:
29864 some space cleanup and debug fixes
29865 Original commit message from CVS:
29866 some space cleanup and debug fixes
29868 2006-03-02 16:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
29870 docs/: Added some more docs to libs and plugins.
29871 Original commit message from CVS:
29872 * docs/libs/gst-plugins-base-libs-docs.sgml:
29873 * docs/libs/gst-plugins-base-libs-sections.txt:
29874 * docs/libs/gst-plugins-base-libs.types:
29875 * docs/plugins/Makefile.am:
29876 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29877 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29878 Added some more docs to libs and plugins.
29879 * gst-libs/gst/audio/gstringbuffer.c:
29880 (gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
29881 * gst-libs/gst/audio/gstringbuffer.h:
29882 Document ringbuffer some more.
29883 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
29884 (gst_video_rate_setcaps), (gst_video_rate_reset),
29885 (gst_video_rate_init), (gst_video_rate_flush_prev),
29886 (gst_video_rate_swap_prev), (gst_video_rate_event),
29887 (gst_video_rate_chain), (gst_video_rate_change_state):
29888 * gst/videorate/gstvideorate.h:
29889 Fix videorate to use segments.
29890 Make it work with 0/1 framerates (closes #331903)
29891 Handle EOS correctly.
29894 2006-03-02 13:13:00 +0000 Tim-Philipp Müller <tim@centricular.net>
29896 ext/ogg/gstogmparse.c: In state change function, first chain up to parent class, then handle downwards state change s...
29897 Original commit message from CVS:
29898 * ext/ogg/gstogmparse.c: (gst_ogm_parse_class_init),
29899 (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
29900 (gst_ogm_text_parse_init), (gst_ogm_parse_change_state):
29901 In state change function, first chain up to parent class,
29902 then handle downwards state change stuff. Remove some
29903 commented out cruft from 0.8 code.
29905 2006-03-02 12:35:59 +0000 Tim-Philipp Müller <tim@centricular.net>
29907 ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
29908 Original commit message from CVS:
29909 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
29910 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
29911 (gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
29912 (gst_ogm_parse_chain):
29913 Don't remove/re-add source pad if the new caps are the same as
29914 the old caps anyway (#333042). When removing source pad, don't
29915 unref it afterwards - we didn't ref it when adding. Sprinkle some
29916 GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
29917 after using gst_pad_get_parent(). Return downstream flow return
29918 value in chain function.
29920 2006-03-02 11:28:23 +0000 Wim Taymans <wim.taymans@gmail.com>
29922 docs/plugins/: Fix hierarchy, added some more elements to the docs.
29923 Original commit message from CVS:
29924 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29925 * docs/plugins/gst-plugins-base-plugins.args:
29926 * docs/plugins/gst-plugins-base-plugins.hierarchy:
29927 * docs/plugins/gst-plugins-base-plugins.interfaces:
29928 * docs/plugins/gst-plugins-base-plugins.signals:
29929 Fix hierarchy, added some more elements to the docs.
29930 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
29931 (gst_ffmpegcsp_get_type):
29932 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
29933 Fix docs for ffmpegcolorspace.
29935 2006-03-01 19:24:44 +0000 Tim-Philipp Müller <tim@centricular.net>
29937 gst/typefind/gsttypefindfunctions.c: Some typefinding fine-tuning:
29938 Original commit message from CVS:
29939 * gst/typefind/gsttypefindfunctions.c: (id3_type_find),
29940 (apetag_type_find), (ape_type_find), (plugin_init):
29941 Some typefinding fine-tuning:
29942 - rank ID3/APE tags in order of preference via probabilities, so that
29943 ID3v2 > APEv2 > APEv1 > ID3v1.
29944 - three or four bytes don't really justify MAXIMUM probability,
29945 change those to 'very likely' (musepack and monkeysaudio).
29947 2006-03-01 18:25:18 +0000 Wim Taymans <wim.taymans@gmail.com>
29950 Original commit message from CVS:
29951 * docs/plugins/Makefile.am:
29952 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29953 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29954 * ext/alsa/gstalsamixer.c:
29955 * ext/alsa/gstalsamixer.h:
29956 * ext/alsa/gstalsamixerelement.c:
29957 (gst_alsa_mixer_element_class_init), (gst_alsa_mixer_element_init):
29958 * ext/alsa/gstalsamixerelement.h:
29959 * ext/alsa/gstalsasink.c:
29960 * ext/alsa/gstalsasink.h:
29961 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
29962 (gst_alsasrc_init):
29963 * ext/alsa/gstalsasrc.h:
29965 Small code cleanups.
29967 2006-03-01 17:52:45 +0000 Wim Taymans <wim.taymans@gmail.com>
29969 ext/theora/Makefile.am: Dist new header too,
29970 Original commit message from CVS:
29971 * ext/theora/Makefile.am:
29972 Dist new header too,
29974 2006-03-01 17:39:28 +0000 Wim Taymans <wim.taymans@gmail.com>
29976 Fix some more docs.
29977 Original commit message from CVS:
29978 * docs/plugins/Makefile.am:
29979 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29980 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29981 * ext/gnomevfs/gstgnomevfssink.h:
29982 * ext/gnomevfs/gstgnomevfssrc.h:
29983 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
29984 * ext/vorbis/vorbisdec.h:
29985 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
29986 * ext/vorbis/vorbisenc.h:
29987 * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
29988 (vorbis_parse_chain), (vorbis_parse_change_state):
29989 * ext/vorbis/vorbisparse.h:
29990 * gst/audioconvert/gstaudioconvert.h:
29991 * gst/tcp/gsttcpserversink.h:
29992 * gst/videotestsrc/gstvideotestsrc.c:
29993 * gst/videotestsrc/gstvideotestsrc.h:
29994 * gst/volume/gstvolume.c:
29995 * gst/volume/gstvolume.h:
29996 Fix some more docs.
29997 Added docs for vorbisdec and vorbisparse.
30000 2006-03-01 16:24:37 +0000 Wim Taymans <wim.taymans@gmail.com>
30002 Updated/added documentation.
30003 Original commit message from CVS:
30004 * docs/plugins/Makefile.am:
30005 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
30006 * docs/plugins/gst-plugins-base-plugins-sections.txt:
30007 * ext/pango/gstclockoverlay.h:
30008 * ext/pango/gsttextoverlay.h:
30009 * ext/pango/gsttextrender.h:
30010 * ext/pango/gsttimeoverlay.h:
30011 * ext/theora/gsttheoradec.h:
30012 * ext/theora/gsttheoraenc.h:
30013 * ext/theora/theoradec.c:
30014 * ext/theora/theoraenc.c:
30015 * gst/audioconvert/gstaudioconvert.h:
30016 * gst/audiotestsrc/gstaudiotestsrc.h:
30017 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
30018 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
30019 * gst/tcp/gstmultifdsink.h:
30020 Updated/added documentation.
30021 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
30022 (gst_text_overlay_halign_get_type),
30023 (gst_text_overlay_wrap_mode_get_type),
30024 (gst_text_overlay_base_init), (gst_text_overlay_class_init),
30025 (gst_text_overlay_init), (gst_text_overlay_set_property),
30026 (gst_text_overlay_get_property):
30027 Fix up properties to be enums instead of string to make bindings,
30028 introspection and automatic GUI creation possible.
30029 Add getters for the properties.
30031 2006-02-28 21:21:07 +0000 Sébastien Moutte <sebastien@moutte.net>
30033 gst/audiotestsrc/gstaudiotestsrc.c: added defines of M_PI and M_PI_2
30034 Original commit message from CVS:
30035 * gst/audiotestsrc/gstaudiotestsrc.c:
30036 added defines of M_PI and M_PI_2
30037 * gst/ffmpegcolorspace/avcodec.h:
30038 removed #include "stdint.h" for win32 as _stdint.h is
30039 autogenerated to win32/common
30040 * win32/common/libgstaudio.def:
30041 * win32/common/libgsttag.def:
30044 some project files bugs corrected
30046 project files are reset to the default vs7 configuration
30047 (they link to msvcr71.dll using default optimizations)
30049 2006-02-28 19:08:12 +0000 Wim Taymans <wim.taymans@gmail.com>
30051 ext/gnomevfs/gstgnomevfssink.c: Fix some docs.
30052 Original commit message from CVS:
30053 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
30056 2006-02-28 13:52:04 +0000 Edward Hervey <bilboed@bilboed.com>
30058 ext/alsa/gstalsasrc.c: Set proper class on the ElementDetails:
30059 Original commit message from CVS:
30060 * ext/alsa/gstalsasrc.c:
30061 Set proper class on the ElementDetails:
30062 Source/Audio instead of Src/Audio
30064 2006-02-28 12:19:11 +0000 Edward Hervey <bilboed@bilboed.com>
30066 gst/videoscale/vs_scanline.c: Revert optimization in videoscale. It should go in liboil and have an appropriate liboi...
30067 Original commit message from CVS:
30068 * gst/videoscale/vs_scanline.c:
30069 (vs_scanline_resample_nearest_RGBA):
30070 Revert optimization in videoscale. It should go in liboil and have
30071 an appropriate liboil function.
30073 2006-02-28 11:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
30075 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
30076 Original commit message from CVS:
30077 * gst-libs/gst/audio/gstbaseaudiosink.c:
30078 (gst_base_audio_sink_provide_clock):
30079 Don't try to provide a clock in the NULL state.
30081 2006-02-28 11:04:47 +0000 Wim Taymans <wim.taymans@gmail.com>
30083 ext/ogg/gstoggdemux.c: Use GstSegment infrastructure to remove duplicated code and handle more seek cases correctly.
30084 Original commit message from CVS:
30085 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
30086 (gst_ogg_pad_event), (gst_ogg_pad_internal_chain),
30087 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
30088 (gst_ogg_demux_deactivate_current_chain),
30089 (gst_ogg_demux_activate_chain), (gst_ogg_demux_do_seek),
30090 (gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_info),
30091 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
30092 (gst_ogg_demux_loop), (gst_ogg_demux_change_state):
30093 Use GstSegment infrastructure to remove duplicated code
30094 and handle more seek cases correctly.
30096 2006-02-28 10:39:19 +0000 Wim Taymans <wim.taymans@gmail.com>
30098 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Don't ignore return code from ffmpeg convert function.
30099 Original commit message from CVS:
30100 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30101 (gst_ffmpegcsp_transform):
30102 Don't ignore return code from ffmpeg convert function.
30103 * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
30104 Split out some long statements to ease debugging.
30106 2006-02-27 12:08:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30108 ext/libvisual/visual.c: Don't use gst_pad_use_fixed_caps, because it prevents downstream from being able to renegotia...
30109 Original commit message from CVS:
30110 * ext/libvisual/visual.c: (gst_visual_init),
30111 (gst_vis_src_negotiate), (get_buffer), (plugin_init):
30112 Don't use gst_pad_use_fixed_caps, because it prevents downstream from
30113 being able to renegotiate the size. Instead, use the negotiation
30114 algorithm from the goom plugin to pick an initial output caps.
30115 Also, allow theoretical libvisual plugins that might support non-GL
30116 output even if they also do GL.
30118 2006-02-26 21:05:46 +0000 Julien Moutte <julien@moutte.net>
30120 ext/libvisual/visual.c: Load only non GL plugins. Fix some memleaks and possible negotiation issues.
30121 Original commit message from CVS:
30122 2006-02-26 Julien MOUTTE <julien@moutte.net>
30123 * ext/libvisual/visual.c: (gst_visual_init),
30124 (gst_visual_src_setcaps), (get_buffer), (gst_visual_chain),
30125 (plugin_init): Load only non GL plugins. Fix some memleaks and
30126 possible negotiation issues.
30128 2006-02-24 23:19:44 +0000 Julien Moutte <julien@moutte.net>
30130 gst-libs/gst/tag/tag.h: Adding Annodex tags here.
30131 Original commit message from CVS:
30132 2006-02-25 Julien MOUTTE <julien@moutte.net>
30133 * gst-libs/gst/tag/tag.h: Adding Annodex tags here.
30135 2006-02-24 18:55:27 +0000 Michael Smith <msmith@xiph.org>
30137 gst/typefind/gsttypefindfunctions.c: Fix CMML type find function to not require a specific minor version of the CMML ...
30138 Original commit message from CVS:
30139 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
30140 (cmml_type_find), (plugin_init):
30141 Fix CMML type find function to not require a specific minor version
30142 of the CMML header.
30143 Add an MPEG4 video elementary stream typefind function.
30145 2006-02-24 17:31:53 +0000 Michael Smith <msmith@xiph.org>
30147 ext/ogg/gstoggdemux.c: Annodex support in ogg demuxer. Doesn't do very much without the other annodex patches (to come).
30148 Original commit message from CVS:
30149 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
30150 (gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
30151 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
30152 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
30153 (gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
30154 (gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
30155 Annodex support in ogg demuxer. Doesn't do very much without the
30156 other annodex patches (to come).
30158 2006-02-24 16:21:34 +0000 Tim-Philipp Müller <tim@centricular.net>
30160 gst-libs/gst/riff/riff-media.c:
30161 Original commit message from CVS:
30162 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
30163 Pick up palette for MS video v1 (#327028, patch by:
30164 Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
30166 2006-02-24 13:54:04 +0000 Tim-Philipp Müller <tim@centricular.net>
30168 gst/ffmpegcolorspace/gstffmpegcolorspace.c: The 'palette_data' field from incoming RGB caps shouldn't be proxied on o...
30169 Original commit message from CVS:
30170 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30171 (gst_ffmpegcsp_caps_remove_format_info),
30172 (gst_ffmpegcsp_get_unit_size):
30173 The 'palette_data' field from incoming RGB caps shouldn't be
30174 proxied on outgoing YUV caps; also, restrict unit size
30175 adjustment in case of paletted data only to the unit that
30176 actually has a palette. Fixes #330711.
30178 2006-02-24 12:18:14 +0000 Tim-Philipp Müller <tim@centricular.net>
30180 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Plug some memory leaks.
30181 Original commit message from CVS:
30182 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30183 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
30184 (gst_ffmpegcsp_finalize), (gst_ffmpegcsp_class_init),
30185 (gst_ffmpegcsp_get_unit_size):
30186 Plug some memory leaks.
30188 2006-02-24 10:18:52 +0000 Tim-Philipp Müller <tim@centricular.net>
30190 sys/: Add some _CFLAGS and _LIBS that seem to be missing and/or required for Cygwin (see #317048).
30191 Original commit message from CVS:
30192 * sys/ximage/Makefile.am:
30193 * sys/xvimage/Makefile.am:
30194 Add some _CFLAGS and _LIBS that seem to be missing
30195 and/or required for Cygwin (see #317048).
30197 2006-02-24 00:07:18 +0000 Tim-Philipp Müller <tim@centricular.net>
30200 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
30201 Original commit message from CVS:
30202 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
30204 2006-02-22 18:46:46 +0000 Wim Taymans <wim.taymans@gmail.com>
30206 ext/alsa/gstalsasrc.c: Fix description as pointed out by caugier.
30207 Original commit message from CVS:
30208 * ext/alsa/gstalsasrc.c:
30209 Fix description as pointed out by caugier.
30211 2006-02-22 10:29:22 +0000 Edward Hervey <bilboed@bilboed.com>
30213 gst/typefind/gsttypefindfunctions.c: Better 3gp typefinding.
30214 Original commit message from CVS:
30215 Reviewed by : Edward Hervey <edward@fluendo.com>
30216 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
30218 Better 3gp typefinding.
30220 2006-02-21 12:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
30222 ext/gnomevfs/gstgnomevfssrc.c: Don't send EOS event here, the base class will send one for us.
30223 Original commit message from CVS:
30224 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
30225 Don't send EOS event here, the base class will send one for us.
30226 * gst/playback/gstplaybasebin.c: (prepare_output):
30227 Subpictures without video stream aren't allowed either.
30228 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
30229 Fix debug statement copy'n'paste-o.
30231 2006-02-21 12:05:18 +0000 Tim-Philipp Müller <tim@centricular.net>
30233 ext/alsa/gstalsamixer.c: Fix issues with mixer keeping state when muting/unmuting and when changing the volume whilst...
30234 Original commit message from CVS:
30235 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
30236 Fix issues with mixer keeping state when muting/unmuting
30237 and when changing the volume whilst muted (see #331763
30240 2006-02-20 18:27:06 +0000 Tim-Philipp Müller <tim@centricular.net>
30242 gst/subparse/gstsubparse.c: Set right caps given that we send escaped text. Also, honour <i></i>, <b></b> and <u></u>...
30243 Original commit message from CVS:
30244 * gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
30245 (parse_subrip), (gst_sub_parse_format_autodetect):
30246 Set right caps given that we send escaped text. Also,
30247 honour <i></i>, <b></b> and <u></u> markers that can be found
30248 in .srt files (fixes #310202).
30250 2006-02-20 16:21:14 +0000 Tim-Philipp Müller <tim@centricular.net>
30252 gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
30253 Original commit message from CVS:
30254 * gst-libs/gst/audio/mixerutils.c:
30255 (element_factory_rank_compare_func):
30256 Make order in which elements are tried more determinable.
30258 2006-02-20 15:57:51 +0000 Julien Moutte <julien@moutte.net>
30260 gst/playback/gstdecodebin.c: Make decodebin reusable by fixing remove_element_chain first and then introduce a cleane...
30261 Original commit message from CVS:
30262 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
30263 (remove_element_chain), (cleanup_decodebin),
30264 (gst_decode_bin_change_state): Make decodebin reusable by
30265 fixing remove_element_chain first and then introduce a
30266 cleaner in state change to ->NULL. (Closes #331678)
30267 ------------------------------------------------------
30269 2006-02-19 14:32:35 +0000 Wim Taymans <wim.taymans@gmail.com>
30271 ext/gnomevfs/gstgnomevfssink.c: use 0666 mask when creating files so umask gets applied correctly. Fixes #331295.
30272 Original commit message from CVS:
30273 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
30274 use 0666 mask when creating files so umask gets applied
30275 correctly. Fixes #331295.
30277 2006-02-19 14:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
30279 gst/subparse/: Add very basic parser for SSA subtitle streams (as often found in matroska files).
30280 Original commit message from CVS:
30281 * gst/subparse/Makefile.am:
30282 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
30283 (gst_ssa_parse_dispose), (gst_ssa_parse_init),
30284 (gst_ssa_parse_class_init), (gst_ssa_parse_src_event),
30285 (gst_ssa_parse_sink_event), (gst_ssa_parse_setcaps),
30286 (gst_ssa_parse_remove_override_codes), (gst_ssa_parse_parse_line),
30287 (gst_ssa_parse_chain), (gst_ssa_parse_change_state):
30288 * gst/subparse/gstssaparse.h:
30289 * gst/subparse/gstsubparse.c: (plugin_init):
30290 Add very basic parser for SSA subtitle streams (as often
30291 found in matroska files).
30293 2006-02-19 14:09:40 +0000 Tim-Philipp Müller <tim@centricular.net>
30295 gst/playback/gstdecodebin.c: That should be text/x-pango-markup, not text/x-pango-layout.
30296 Original commit message from CVS:
30297 * gst/playback/gstdecodebin.c: (mimetype_is_raw):
30298 That should be text/x-pango-markup, not text/x-pango-layout.
30300 2006-02-19 12:41:03 +0000 Julien Moutte <julien@moutte.net>
30302 ext/pango/gsttextoverlay.c: Polishing.
30303 Original commit message from CVS:
30304 2006-02-19 Julien MOUTTE <julien@moutte.net>
30305 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize):
30308 2006-02-19 12:05:23 +0000 Julien Moutte <julien@moutte.net>
30310 ext/pango/gsttextoverlay.c: Fix state change deadlock.
30311 Original commit message from CVS:
30312 2006-02-19 Julien MOUTTE <julien@moutte.net>
30313 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
30314 (gst_text_overlay_finalize), (gst_text_overlay_init),
30315 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
30316 (gst_text_overlay_render_text),
30317 (gst_text_overlay_text_pad_link),
30318 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
30319 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
30320 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
30321 Fix state change deadlock.
30323 2006-02-19 11:56:28 +0000 Julien Moutte <julien@moutte.net>
30325 ext/pango/gsttextoverlay.*: Fix seeking both for muxed formats and subtitles files.
30326 Original commit message from CVS:
30327 2006-02-19 Julien MOUTTE <julien@moutte.net>
30328 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
30329 (gst_text_overlay_finalize), (gst_text_overlay_init),
30330 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
30331 (gst_text_overlay_render_text),
30332 (gst_text_overlay_text_pad_link),
30333 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
30334 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
30335 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
30336 * ext/pango/gsttextoverlay.h: Fix seeking both for muxed formats
30337 and subtitles files.
30339 2006-02-19 00:40:38 +0000 Julien Moutte <julien@moutte.net>
30341 gst/playback/gstdecodebin.c: pango layout should be considered as row.
30342 Original commit message from CVS:
30343 2006-02-19 Julien MOUTTE <julien@moutte.net>
30344 * gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
30345 should be considered as row.
30347 2006-02-19 00:25:16 +0000 Julien Moutte <julien@moutte.net>
30349 gst/playback/gststreaminfo.*: Introduce language informations.
30350 Original commit message from CVS:
30351 2006-02-19 Julien MOUTTE <julien@moutte.net>
30352 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
30354 * gst/playback/gststreaminfo.h: Introduce language informations.
30356 2006-02-18 22:41:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30358 sys/: Set shared memory segments to be deleted as soon as we have attached, that way they get cleaned up automaticall...
30359 Original commit message from CVS:
30360 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
30361 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
30362 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
30363 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
30364 Set shared memory segments to be deleted as soon as we have attached,
30365 that way they get cleaned up automatically if we crash.
30367 2006-02-18 19:53:48 +0000 Julien Moutte <julien@moutte.net>
30369 ext/pango/: Those functions are called with lock held.
30370 Original commit message from CVS:
30371 2006-02-18 Julien MOUTTE <julien@moutte.net>
30372 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_get_text):
30373 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_get_text): Those
30374 functions are called with lock held.
30376 2006-02-18 19:51:47 +0000 Julien Moutte <julien@moutte.net>
30380 Original commit message from CVS:
30383 2006-02-18 19:10:35 +0000 Julien Moutte <julien@moutte.net>
30385 ext/pango/gsttextoverlay.c: Refactoring of textoverlay without collectpads. This now supports sparse subtitles coming...
30386 Original commit message from CVS:
30387 2006-02-18 Julien MOUTTE <julien@moutte.net>
30388 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
30389 (gst_text_overlay_finalize), (gst_text_overlay_init),
30390 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
30391 (gst_text_overlay_render_text),
30392 (gst_text_overlay_text_pad_link),
30393 (gst_text_overlay_text_pad_unlink),
30394 (gst_text_overlay_text_event),
30395 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
30396 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
30397 (gst_text_overlay_change_state): Refactoring of textoverlay
30398 without collectpads. This now supports sparse subtitles coming
30399 from a demuxer instead of a sub file. Seeking is still broken
30400 though. Need to discuss with wtay some more on how to handle
30402 * ext/pango/gsttextoverlay.h:
30403 * gst/playback/gstplaybin.c: (setup_sinks): Support linking with
30404 subtitles coming from the demuxer.
30406 2006-02-17 19:31:12 +0000 Wim Taymans <wim.taymans@gmail.com>
30408 ext/vorbis/vorbisenc.c: Use some more scaling functions.
30409 Original commit message from CVS:
30410 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
30411 (gst_vorbisenc_convert_sink):
30412 Use some more scaling functions.
30414 2006-02-17 16:12:11 +0000 Tim-Philipp Müller <tim@centricular.net>
30416 ext/cdparanoia/gstcdparanoiasrc.*: Add back 'transport-error' and 'uncorrected-error' signals and make them actually ...
30417 Original commit message from CVS:
30418 * ext/cdparanoia/gstcdparanoiasrc.c:
30419 (gst_cd_paranoia_src_class_init), (gst_cd_paranoia_dummy_callback),
30420 (gst_cd_paranoia_paranoia_callback),
30421 (gst_cd_paranoia_src_signal_is_being_watched),
30422 (gst_cd_paranoia_src_read_sector):
30423 * ext/cdparanoia/gstcdparanoiasrc.h:
30424 Add back 'transport-error' and 'uncorrected-error' signals and
30425 make them actually be fired when bad stuff happens (#319340).
30427 2006-02-17 14:07:01 +0000 Wim Taymans <wim.taymans@gmail.com>
30429 gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
30430 Original commit message from CVS:
30431 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
30432 (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
30433 (gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
30434 (gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
30435 (gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
30436 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
30437 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
30438 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
30439 (gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
30440 (gst_ring_buffer_clear):
30442 Added some G_LIKELY.
30444 2006-02-17 10:15:52 +0000 Wim Taymans <wim.taymans@gmail.com>
30446 gst-libs/gst/audio/TODO: Update TODO
30447 Original commit message from CVS:
30448 * gst-libs/gst/audio/TODO:
30450 * gst-libs/gst/audio/gstbaseaudiosink.c:
30451 (gst_base_audio_sink_get_offset):
30452 When trying to play samples ASAP and we don't have a
30453 previous sample, try to play at position 0 instead of
30454 an invalid position.
30456 2006-02-17 09:24:56 +0000 Wim Taymans <wim.taymans@gmail.com>
30458 ext/alsa/gstalsasink.c: Also release lock when we get an error in _reset(); fix an error message.
30459 Original commit message from CVS:
30460 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
30461 (gst_alsasink_reset):
30462 Also release lock when we get an error in _reset();
30463 fix an error message.
30465 2006-02-16 21:01:23 +0000 Tim-Philipp Müller <tim@centricular.net>
30467 ext/alsa/gstalsasink.*: Add support for more than 2 channels (#326720).
30468 Original commit message from CVS:
30469 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
30470 (gst_alsasink_init), (get_channel_free_structure),
30471 (caps_add_channel_configuration), (gst_alsasink_getcaps),
30472 (gst_alsasink_close):
30473 * ext/alsa/gstalsasink.h:
30474 Add support for more than 2 channels (#326720).
30476 2006-02-16 20:19:51 +0000 Tim-Philipp Müller <tim@centricular.net>
30478 gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
30479 Original commit message from CVS:
30480 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
30481 Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
30482 with 4 or 6 channels, assume a default channel layout to make things
30483 work (not sure there's anything else we can do in those cases).
30485 2006-02-16 19:18:46 +0000 Tim-Philipp Müller <tim@centricular.net>
30487 gst-libs/gst/audio/multichannel.c: Minor docs fix.
30488 Original commit message from CVS:
30489 * gst-libs/gst/audio/multichannel.c:
30491 * gst-libs/gst/riff/Makefile.am:
30492 * gst-libs/gst/riff/riff-ids.h:
30493 * gst-libs/gst/riff/riff-media.c:
30494 (gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
30495 Add support for WAVEFORMATEX, eg. PCM audio with more than two
30496 channels and a channel layout map.
30498 2006-02-16 17:06:46 +0000 Edward Hervey <bilboed@bilboed.com>
30500 gst/videoscale/vs_scanline.c: C-level optimization of the RGBA nearest neighbour function.
30501 Original commit message from CVS:
30502 Reviewed by Edward Hervey <edward@fluendo.com>
30503 * gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
30504 C-level optimization of the RGBA nearest neighbour function.
30505 Eventually this might end up in liboil with vectorized versions.
30507 2006-02-16 11:44:43 +0000 Tim-Philipp Müller <tim@centricular.net>
30509 gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
30510 Original commit message from CVS:
30511 * gst-libs/gst/audio/multichannel.c:
30512 (gst_audio_get_channel_positions):
30513 When we have more than 2 channels, but no channel layout is
30514 specified in the caps, return some default channel layout
30515 to the caller and warn about about a possibly buggy element
30516 (could be buggy filtercaps as well of course) (#317038).
30518 2006-02-16 09:29:38 +0000 Tim-Philipp Müller <tim@centricular.net>
30520 pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: Add gst-libs/gst/cdda to list of lib search paths.
30521 Original commit message from CVS:
30522 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
30523 Add gst-libs/gst/cdda to list of lib search paths.
30525 2006-02-15 12:20:47 +0000 Andy Wingo <wingo@pobox.com>
30527 ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating timestamp, update timestamp_end as well. Fixes a bugaboo. ...
30528 Original commit message from CVS:
30529 2006-02-15 Andy Wingo <wingo@pobox.com>
30530 * ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating
30531 timestamp, update timestamp_end as well. Fixes a bugaboo. I hope
30532 to the Lord Jesus that I do not have to touch the ogg muxer ever
30535 2006-02-15 12:07:57 +0000 Edward Hervey <bilboed@bilboed.com>
30537 gst/typefind/gsttypefindfunctions.c: quicktime movie files can also contain 'uuid' atoms.
30538 Original commit message from CVS:
30539 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
30540 quicktime movie files can also contain 'uuid' atoms.
30542 2006-02-14 18:52:52 +0000 Tim-Philipp Müller <tim@centricular.net>
30544 gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun...
30545 Original commit message from CVS:
30546 * gst/audioconvert/plugin.c: (plugin_init):
30547 Register the GstAudioChannelPosition enum type with the type
30548 system in the plugin_init function, so that it is known before
30549 any element actually makes use of multi-channel stuff. This is
30550 required for example if one wants to be able to deserialise/use
30551 a caps string with channel positions before any pipeline has
30552 been setup and started, like with gst-launch.
30554 2006-02-14 13:45:35 +0000 Wim Taymans <wim.taymans@gmail.com>
30556 gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
30557 Original commit message from CVS:
30558 * gst-libs/gst/audio/gstringbuffer.c:
30559 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
30560 (gst_ring_buffer_samples_done), (wait_segment),
30561 (gst_ring_buffer_commit), (gst_ring_buffer_clear):
30562 Add some compiler G_(UN_)LIKELY help.
30563 SIGNAL the ringbuffer waiters when going to PAUSED as well to
30564 make sure they can exit their functions. Should fix #330748
30566 2006-02-13 20:49:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30568 Windows does not have long long; copy the generated _stdint.h
30569 Original commit message from CVS:
30573 * win32/common/_stdint.h:
30574 Windows does not have long long; copy the generated _stdint.h
30575 * win32/common/interfaces-enumtypes.c:
30576 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
30577 (gst_mixer_track_flags_get_type),
30578 (gst_tuner_channel_flags_get_type):
30579 * win32/common/multichannel-enumtypes.c:
30580 (gst_audio_channel_position_get_type):
30583 2006-02-13 18:49:02 +0000 Wim Taymans <wim.taymans@gmail.com>
30585 gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
30586 Original commit message from CVS:
30587 * gst-libs/gst/audio/gstbaseaudiosink.c:
30588 (gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
30589 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
30590 Always sync on first sample we receive when starting.
30592 2006-02-13 15:59:48 +0000 Wim Taymans <wim.taymans@gmail.com>
30594 gst/playback/gstplaybin.c: Update vis bin docs.
30595 Original commit message from CVS:
30596 * gst/playback/gstplaybin.c: (gen_vis_element):
30597 Update vis bin docs.
30598 Move queue after tee so we don't queue video buffers but
30599 audio samples instead. Fixes problems where the video queue
30600 is filled and the audio queue empty.
30602 2006-02-13 15:17:34 +0000 Tim-Philipp Müller <tim@centricular.net>
30604 gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
30605 Original commit message from CVS:
30606 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
30607 No need to push an EOS event here, GstBaseSrc will do that for us
30608 when we return FLOW_UNEXPECTED.
30610 2006-02-12 14:54:55 +0000 Wim Taymans <wim.taymans@gmail.com>
30612 gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
30613 Original commit message from CVS:
30614 * gst-libs/gst/audio/gstbaseaudiosink.c:
30615 (gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
30616 (gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
30617 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
30618 Use scale functions when possible.
30619 Fix error messages.
30620 Free clockid when after waiting for EOS.
30621 Use G_(UN_)LIKLY when it makes sense.
30622 Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
30624 2006-02-12 14:26:55 +0000 Edward Hervey <bilboed@bilboed.com>
30626 gst/playback/gstplaybasebin.c: Remove stray semi-colon (fixes #330888).
30627 Original commit message from CVS:
30628 * gst/playback/gstplaybasebin.c: (prepare_output):
30629 Remove stray semi-colon (fixes #330888).
30631 2006-02-11 23:35:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30633 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...
30634 Original commit message from CVS:
30635 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
30636 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
30637 Fix up the XShm call testing so that we catch errors, and don't
30638 cause new ones by attempting to detach from a segment we failed
30639 to attach to. Fixes #312439.
30641 2006-02-10 11:29:55 +0000 Edward Hervey <bilboed@bilboed.com>
30643 gst/typefind/gsttypefindfunctions.c: Added flv file typefind (video/x-flv).
30644 Original commit message from CVS:
30645 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
30646 Added flv file typefind (video/x-flv).
30648 2006-02-10 10:53:33 +0000 Edward Hervey <bilboed@bilboed.com>
30650 gst-libs/gst/riff/riff-media.c: Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
30651 Original commit message from CVS:
30652 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
30653 (gst_riff_create_video_template_caps):
30654 Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
30655 Also added the caps to the default set of riff video caps.
30657 2006-02-09 19:05:23 +0000 Andy Wingo <wingo@pobox.com>
30659 ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start time and the end time of the last packet in the page.
30660 Original commit message from CVS:
30661 2006-02-09 Andy Wingo <wingo@pobox.com>
30662 * ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start
30663 time and the end time of the last packet in the page.
30664 (gst_ogg_mux_pad_queue_page): In addition to setting the timestamp
30665 on the pages in our queue, set the duration as well. Reflow a
30667 (gst_ogg_mux_collected): Keep track of GstOggPad->timestamp_end.
30668 Fixes bad muxing order.
30670 2006-02-09 17:04:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30672 gst-libs/gst/rtp/gstbasertppayload.c: update seqnum before setting it on the packet; this makes sure that the timesta...
30673 Original commit message from CVS:
30674 * gst-libs/gst/rtp/gstbasertppayload.c:
30675 (gst_basertppayload_setcaps), (gst_basertppayload_push):
30676 update seqnum before setting it on the packet; this makes sure
30677 that the timestamp and seqnum properties match after pushing
30680 2006-02-09 12:16:35 +0000 Andy Wingo <wingo@pobox.com>
30684 Original commit message from CVS:
30687 2006-02-09 11:46:03 +0000 Andy Wingo <wingo@pobox.com>
30689 * gst-libs/gst/audio/gstringbuffer.c:
30690 * win32/common/config.h:
30692 Original commit message from CVS:
30695 2006-02-09 11:36:18 +0000 Andy Wingo <wingo@pobox.com>
30697 gst-libs/gst/audio/gstringbuffer.c
30698 Original commit message from CVS:
30699 2006-02-09 Andy Wingo <wingo@pobox.com>
30700 * gst-libs/gst/audio/gstringbuffer.c
30701 (gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
30702 overflow after 13.5 hours of recording. Kapow!
30703 * ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
30704 the buffer size -- we don't care about underrun/overrun reporting
30705 right now, just need to return a useful value.
30707 2006-02-09 11:21:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30709 configure.ac: Back to CVS
30710 Original commit message from CVS:
30714 === release 0.10.3 ===
30716 2006-02-09 11:18:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30722 * docs/plugins/inspect/plugin-adder.xml:
30723 * docs/plugins/inspect/plugin-alsa.xml:
30724 * docs/plugins/inspect/plugin-audioconvert.xml:
30725 * docs/plugins/inspect/plugin-audiorate.xml:
30726 * docs/plugins/inspect/plugin-audioresample.xml:
30727 * docs/plugins/inspect/plugin-audiotestsrc.xml:
30728 * docs/plugins/inspect/plugin-cdparanoia.xml:
30729 * docs/plugins/inspect/plugin-decodebin.xml:
30730 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
30731 * docs/plugins/inspect/plugin-gnomevfs.xml:
30732 * docs/plugins/inspect/plugin-libvisual.xml:
30733 * docs/plugins/inspect/plugin-ogg.xml:
30734 * docs/plugins/inspect/plugin-pango.xml:
30735 * docs/plugins/inspect/plugin-playbin.xml:
30736 * docs/plugins/inspect/plugin-subparse.xml:
30737 * docs/plugins/inspect/plugin-tcp.xml:
30738 * docs/plugins/inspect/plugin-theora.xml:
30739 * docs/plugins/inspect/plugin-typefindfunctions.xml:
30740 * docs/plugins/inspect/plugin-video4linux.xml:
30741 * docs/plugins/inspect/plugin-videorate.xml:
30742 * docs/plugins/inspect/plugin-videoscale.xml:
30743 * docs/plugins/inspect/plugin-videotestsrc.xml:
30744 * docs/plugins/inspect/plugin-volume.xml:
30745 * docs/plugins/inspect/plugin-vorbis.xml:
30746 * docs/plugins/inspect/plugin-ximagesink.xml:
30747 * docs/plugins/inspect/plugin-xvimagesink.xml:
30748 * win32/common/config.h:
30750 Original commit message from CVS:
30753 2006-02-08 18:37:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30755 configure.ac: Drat. Bump libtool version number for new API.
30756 Original commit message from CVS:
30758 Drat. Bump libtool version number for new API.
30759 Prelease 0.10.2.3 (of 0.10.3)
30761 2006-02-08 15:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30763 0.10.2.2 prerelease (of 0.10.3).
30764 Original commit message from CVS:
30766 * win32/common/config.h:
30767 0.10.2.2 prerelease (of 0.10.3).
30769 2006-02-08 15:50:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30771 gst/tcp/gsttcpclientsrc.c: Revert Andy's newsegment change pending a more correct fix.
30772 Original commit message from CVS:
30773 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
30774 Revert Andy's newsegment change pending a more correct
30777 2006-02-08 12:46:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30794 Original commit message from CVS:
30797 2006-02-08 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30799 * gst/tcp/gstmultifdsink.c:
30801 Original commit message from CVS:
30804 2006-02-08 09:20:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30806 gst/typefind/gsttypefindfunctions.c: detect more files as 3gp group and reorder the iso file formats
30807 Original commit message from CVS:
30809 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
30810 (qt_type_find), (plugin_init):
30811 detect more files as 3gp
30812 group and reorder the iso file formats
30814 2006-02-07 18:32:00 +0000 Tim-Philipp Müller <tim@centricular.net>
30816 ext/vorbis/vorbis.c: Register musicbrainz tags, so apps don't have to.
30817 Original commit message from CVS:
30818 * ext/vorbis/vorbis.c: (plugin_init):
30819 Register musicbrainz tags, so apps don't have to.
30821 2006-02-07 17:44:53 +0000 Tim-Philipp Müller <tim@centricular.net>
30823 gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
30824 Original commit message from CVS:
30825 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
30826 (gst_tag_to_vorbis_tag):
30827 Make sure we called gst_tag_register_musicbrainz_tags()
30828 before possibly mapping a vorbiscomment string from/to a
30831 2006-02-07 16:16:41 +0000 Tim-Philipp Müller <tim@centricular.net>
30833 gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
30834 Original commit message from CVS:
30835 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
30836 In case we can't find the required number of consecutive
30837 mpeg audio frames to positively identify an MPEG audio
30838 stream, check if there's at least a valid mpeg audio
30839 frame right at offset 0 and if so suggest mpeg/audio
30840 caps with a very low probability (#153004).
30842 2006-02-07 15:52:26 +0000 Andy Wingo <wingo@pobox.com>
30844 gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to a TIME segment if we get timestamped buffers. Requir...
30845 Original commit message from CVS:
30846 2006-02-07 Andy Wingo <wingo@pobox.com>
30847 * gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
30848 a TIME segment if we get timestamped buffers. Requires recent
30849 fixes in core to work properly.
30851 2006-02-07 14:57:46 +0000 Tim-Philipp Müller <tim@centricular.net>
30853 gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
30854 Original commit message from CVS:
30855 * gst/playback/gstplaybasebin.c: (prepare_output):
30856 Don't print the URI as part of the error message, it
30857 makes error dialogs look rather ugly, especially if
30858 the URI is very long or has characters in it that
30861 2006-02-07 13:11:31 +0000 Tim-Philipp Müller <tim@centricular.net>
30863 gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
30864 Original commit message from CVS:
30865 * gst/playback/gstplaybasebin.c: (prepare_output):
30866 Error out if we have only text or subtitles, but nothing
30867 else. Also error out if we have subtitles but no video
30870 2006-02-07 11:44:39 +0000 Tim-Philipp Müller <tim@centricular.net>
30872 ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
30873 Original commit message from CVS:
30874 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
30875 Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
30876 Post an error message on the bus when we encounter an
30877 error, which will hopefully be more meaningful than the
30878 'Internal Flow Error' message users get to see if we
30879 just return GST_FLOW_ERROR.
30881 2006-02-07 11:28:04 +0000 Andy Wingo <wingo@pobox.com>
30883 configure.ac (GST_MAJORMINOR): Update core version req to 0.10.2.2, for the collectpads API addition (#330244).
30884 Original commit message from CVS:
30885 2006-02-07 Andy Wingo <wingo@pobox.com>
30886 * configure.ac (GST_MAJORMINOR): Update core version req to
30887 0.10.2.2, for the collectpads API addition (#330244).
30889 2006-02-06 19:09:26 +0000 Tim-Philipp Müller <tim@centricular.net>
30891 ext/gnomevfs/gstgnomevfs.c: Return FALSE from plugin_init() when GnomeVFS can't be initialised for some reason (#3284...
30892 Original commit message from CVS:
30893 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
30894 Return FALSE from plugin_init() when GnomeVFS can't
30895 be initialised for some reason (#328423).
30897 2006-02-06 13:26:54 +0000 Julien Moutte <julien@moutte.net>
30899 ext/pango/gsttextoverlay.c: Stick to seeking theory until i find the bug.
30900 Original commit message from CVS:
30901 2006-02-06 Julien MOUTTE <julien@moutte.net>
30902 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
30903 Stick to seeking theory until i find the bug.
30904 * gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
30906 2006-02-06 12:38:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30908 Make theoraenc and the tests leak free. Like, really.
30909 Original commit message from CVS:
30910 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
30911 (theora_enc_finalize), (theora_enc_sink_setcaps),
30912 (theora_set_header_on_caps), (theora_enc_chain),
30913 (theora_enc_change_state):
30914 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
30915 Make theoraenc and the tests leak free. Like, really.
30917 2006-02-05 23:31:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30919 Add a finalize method to ensure we clean up state even if someone omitted the state change back to NULL.
30920 Original commit message from CVS:
30921 (theora_enc_finalize), (theora_enc_sink_setcaps):
30922 Add a finalize method to ensure we clean up state even if
30923 someone omitted the state change back to NULL.
30924 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
30925 (gst_vorbisenc_chain):
30926 Free some more leaked bits.
30927 * tests/check/pipelines/theoraenc.c: (start_pipeline),
30929 Wait for state changes to happen if they're ASYNC.
30930 This ought to teach those fancy pants buildbots a lesson.
30932 2006-02-05 22:47:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30934 gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
30935 Original commit message from CVS:
30936 * gst-libs/gst/tag/gstid3tag.c:
30937 Add mapping for ID3 International Standard Recording Code
30940 2006-02-05 22:44:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30942 ext/vorbis/vorbisenc.c: Don't leak tag names.
30943 Original commit message from CVS:
30944 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
30945 Don't leak tag names.
30947 2006-02-05 18:22:01 +0000 Tim-Philipp Müller <tim@centricular.net>
30949 Split libgsttag docs into multiple sections.
30950 Original commit message from CVS:
30951 * docs/libs/gst-plugins-base-libs-docs.sgml:
30952 * docs/libs/gst-plugins-base-libs-sections.txt:
30953 * gst-libs/gst/tag/gstid3tag.c:
30954 * gst-libs/gst/tag/gstvorbistag.c:
30955 * gst-libs/gst/tag/tags.c:
30956 Split libgsttag docs into multiple sections.
30958 2006-02-05 18:01:33 +0000 Tim-Philipp Müller <tim@centricular.net>
30960 Add libgsttag to the docs.
30961 Original commit message from CVS:
30962 * docs/libs/Makefile.am:
30963 * docs/libs/gst-plugins-base-libs-docs.sgml:
30964 * docs/libs/gst-plugins-base-libs-sections.txt:
30965 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
30966 * gst-libs/gst/tag/gstvorbistag.c:
30967 * gst-libs/gst/tag/tag.h:
30968 * gst-libs/gst/tag/tags.c:
30969 Add libgsttag to the docs.
30971 2006-02-05 17:21:23 +0000 Julien Moutte <julien@moutte.net>
30973 ext/pango/gsttextoverlay.c: Fix clockoverlay.
30974 Original commit message from CVS:
30975 2006-02-05 Julien MOUTTE <julien@moutte.net>
30976 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
30977 (gst_text_overlay_init), (gst_text_overlay_src_event),
30978 (gst_text_overlay_collected): Fix clockoverlay.
30980 2006-02-05 17:15:17 +0000 Tim-Philipp Müller <tim@centricular.net>
30982 docs/libs/compiling.sgml: Fix typo: it's pkg-config, not pkg-gconfig
30983 Original commit message from CVS:
30984 * docs/libs/compiling.sgml:
30985 Fix typo: it's pkg-config, not pkg-gconfig
30986 * docs/libs/gst-plugins-base-libs-docs.sgml:
30987 * docs/libs/gst-plugins-base-libs-sections.txt:
30988 * docs/libs/tmpl/gstgconf.sgml:
30989 There is no libgstgconf in 0.10, remove it
30992 2006-02-05 16:03:48 +0000 Julien Moutte <julien@moutte.net>
30994 docs/libs/tmpl/gstcolorbalance.sgml: Updated.
30995 Original commit message from CVS:
30996 2006-02-05 Julien MOUTTE <julien@moutte.net>
30997 * docs/libs/tmpl/gstcolorbalance.sgml: Updated.
30998 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
30999 (gst_text_overlay_src_event), (gst_text_overlay_collected):
31000 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
31001 (gst_sub_parse_class_init), (gst_sub_parse_init),
31002 (gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
31003 (parse_mpsub), (parser_state_init), (handle_buffer),
31004 (gst_sub_parse_chain), (gst_sub_parse_sink_event),
31006 * gst/subparse/gstsubparse.h: Introduce seeking code.
31008 2006-02-05 15:14:06 +0000 Tim-Philipp Müller <tim@centricular.net>
31010 gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
31011 Original commit message from CVS:
31012 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
31013 Add comment about LANGUAGE tag inconsistency (we want
31014 ISO-639-1, but extract three-letter identifiers?)
31016 Add two translatable files.
31018 2006-02-05 14:59:28 +0000 Tim-Philipp Müller <tim@centricular.net>
31020 gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
31021 Original commit message from CVS:
31022 * gst-libs/gst/tag/Makefile.am:
31023 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
31024 * gst-libs/gst/tag/tag.h:
31025 * gst-libs/gst/tag/tags.c:
31026 (gst_tag_register_musicbrainz_tags_internal),
31027 (gst_tag_register_musicbrainz_tags):
31028 Forward-port some tags stuff from the 0.8 branch. This is
31029 mostly the addition of musicbrainz tags and their mapping
31030 to vorbistags, and a vorbistag mapping of the language tag.
31032 2006-02-05 12:06:25 +0000 Julien Moutte <julien@moutte.net>
31034 gst/playback/gstplaybin.c: Fix broken code refactoring.
31035 Original commit message from CVS:
31036 2006-02-05 Julien MOUTTE <julien@moutte.net>
31037 * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
31040 2006-02-05 03:05:41 +0000 David Schleef <ds@schleef.org>
31042 Add Dirac typefinding and add dirac format to oggmux.
31043 Original commit message from CVS:
31044 * ext/ogg/gstoggmux.c:
31045 * gst/typefind/gsttypefindfunctions.c:
31046 Add Dirac typefinding and add dirac format to oggmux.
31048 2006-02-04 07:49:03 +0000 Michael Smith <msmith@xiph.org>
31051 Improve error message for liboil missingness.
31052 Original commit message from CVS:
31053 Improve error message for liboil missingness.
31055 2006-02-03 19:23:41 +0000 Tim-Philipp Müller <tim@centricular.net>
31057 gst/playback/gstdecodebin.c: Don't put essential function call into g_return_*() macro, otherwise it'll all be replac...
31058 Original commit message from CVS:
31059 * gst/playback/gstdecodebin.c: (try_to_link_1):
31060 Don't put essential function call into
31061 g_return_*() macro, otherwise it'll all be
31062 replaced by NOOPs when compiling with
31063 G_DISABLE_CHECKS defined.
31065 2006-02-03 17:45:44 +0000 Edgard Lima <edgard.lima@indt.org.br>
31068 * ext/ogg/gstoggdemux.c:
31069 * ext/ogg/gstoggparse.c:
31070 * gst/tcp/gsttcpserversink.c:
31071 * sys/v4l/v4lsrc_calls.c:
31072 * sys/v4l/v4lsrc_calls.h:
31073 Just make it compile with --disable-gst-debug.
31074 Original commit message from CVS:
31075 Just make it compile with --disable-gst-debug.
31077 2006-02-03 12:51:47 +0000 Wim Taymans <wim.taymans@gmail.com>
31079 ext/alsa/gstalsasink.*: Add lock to protect alsa calls.
31080 Original commit message from CVS:
31081 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
31082 (gst_alsasink_class_init), (gst_alsasink_init),
31083 (gst_alsasink_write), (gst_alsasink_reset):
31084 * ext/alsa/gstalsasink.h:
31085 Add lock to protect alsa calls.
31086 Implement reset to flush samples ASAP, does not work
31089 2006-02-02 18:18:31 +0000 Wim Taymans <wim.taymans@gmail.com>
31091 gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
31092 Original commit message from CVS:
31093 * gst-libs/gst/audio/gstbaseaudiosink.c:
31094 (gst_base_audio_sink_provide_clock):
31095 Ugh.. getting late I guess...
31097 2006-02-02 18:13:26 +0000 Wim Taymans <wim.taymans@gmail.com>
31099 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
31100 Original commit message from CVS:
31101 * gst-libs/gst/audio/gstbaseaudiosink.c:
31102 (gst_base_audio_sink_provide_clock),
31103 (gst_base_audio_sink_set_property),
31104 (gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
31105 Don't try to provide a clock when we are not negotiated since
31106 we might not be able to make it run.
31108 2006-02-02 17:51:48 +0000 Tim-Philipp Müller <tim@centricular.net>
31110 gst/playback/gstdecodebin.c: Unlinking two source pads is ... hard.
31111 Original commit message from CVS:
31112 * gst/playback/gstdecodebin.c: (try_to_link_1):
31113 Unlinking two source pads is ... hard.
31115 2006-02-02 12:14:35 +0000 Wim Taymans <wim.taymans@gmail.com>
31117 gst-libs/gst/audio/TODO: Updated.
31118 Original commit message from CVS:
31119 * gst-libs/gst/audio/TODO:
31121 * gst-libs/gst/audio/gstbaseaudiosink.c:
31122 (gst_base_audio_sink_drain), (gst_base_audio_sink_event):
31123 On EOS, wait till the last sample is played before posting EOS.
31125 2006-02-02 08:53:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31127 * tests/check/pipelines/theoraenc.c:
31128 comment on my understanding
31129 Original commit message from CVS:
31130 comment on my understanding
31132 2006-02-02 08:47:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31135 * tests/check/pipelines/theoraenc.c:
31136 reformat to fit 80 chars
31137 Original commit message from CVS:
31138 reformat to fit 80 chars
31140 2006-02-02 00:04:37 +0000 Kai Vehmanen <kv2004@eca.cx>
31142 gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
31143 Original commit message from CVS:
31144 2006-02-01 Philippe Kalaf <burger at speedy dot org>
31145 * gst-libs/gst/rtp/gstbasertpdepayload.c:
31146 Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
31147 setting queue_delay to zero. Also avoid thread being started if
31148 queue_delay is zero.
31150 2006-02-01 14:51:29 +0000 Tim-Philipp Müller <tim@centricular.net>
31152 gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
31153 Original commit message from CVS:
31154 * gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
31155 Make test work again by connecting fakesinks to each decoded pad,
31156 which makes the pipeline wait until each fakesink has a buffer
31157 queued before going to PAUSED state. At that point we know the
31158 decodebin pads are negotiated.
31160 2006-02-01 11:59:47 +0000 Tim-Philipp Müller <tim@centricular.net>
31162 gst/: Pass unhandled queries to the parent class's query function.
31163 Original commit message from CVS:
31164 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
31165 (gst_cdda_base_src_handle_event):
31166 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
31167 Pass unhandled queries to the parent class's query function.
31169 2006-02-01 11:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
31171 Pass unhandled queries upstream instead of just dropping them (#326447). Also, fix supported query types list for som...
31172 Original commit message from CVS:
31173 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
31174 (gst_ogg_pad_src_query):
31175 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
31176 * ext/theora/theoradec.c: (theora_dec_src_query),
31177 (theora_dec_sink_query):
31178 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
31179 (vorbis_dec_sink_query):
31180 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
31181 (gst_vorbisenc_sink_query):
31182 * gst/adder/gstadder.c: (gst_adder_query):
31183 Pass unhandled queries upstream instead of just
31184 dropping them (#326447). Also, fix supported
31185 query types list for some elements.
31187 2006-02-01 09:58:15 +0000 Tim-Philipp Müller <tim@centricular.net>
31189 gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
31190 Original commit message from CVS:
31191 * gst/typefind/gsttypefindfunctions.c: (au_type_find),
31192 (paris_type_find), (ilbc_type_find), (plugin_init):
31193 Fix typefinding for audio/x-au, audio/x-paris and
31194 audio/iLBC-sh. We cannot use the START_WITH macros
31195 here, because there can only be one typefind factory
31196 with the same name (caps), so the second one would
31197 replace the first one and the first one would never
31198 be called when doing typefinding (see #161712).
31200 2006-01-31 19:25:10 +0000 Wim Taymans <wim.taymans@gmail.com>
31202 ext/vorbis/vorbisdec.c: Use scale_int when we can, add some more scaling.
31203 Original commit message from CVS:
31204 * ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
31205 (vorbis_handle_header_packet), (vorbis_dec_push),
31206 (vorbis_handle_data_packet):
31207 Use scale_int when we can, add some more scaling.
31208 Check packettype before parsing it.
31210 2006-01-31 17:44:35 +0000 Wim Taymans <wim.taymans@gmail.com>
31212 ext/theora/theoradec.c: Call right _scale functions.
31213 Original commit message from CVS:
31214 * ext/theora/theoradec.c: (_theora_granule_time),
31215 (theora_dec_src_convert), (theora_dec_sink_convert):
31216 Call right _scale functions.
31217 Use parameter instead of some other random value.
31219 2006-01-31 17:27:00 +0000 Wim Taymans <wim.taymans@gmail.com>
31221 ext/theora/theoradec.c: Use higher precision timestamps calculation.
31222 Original commit message from CVS:
31223 * ext/theora/theoradec.c: (_theora_granule_frame),
31224 (_theora_granule_time), (_inc_granulepos),
31225 (theora_dec_src_convert), (theora_dec_sink_convert),
31226 (theora_handle_type_packet), (theora_handle_data_packet),
31227 (theora_dec_chain):
31228 Use higher precision timestamps calculation.
31229 Convert some other conversions to _scale.
31231 2006-01-31 17:19:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31233 gst/: initialize gst_controller before using
31234 Original commit message from CVS:
31235 * gst/audiotestsrc/gstaudiotestsrc.c:
31236 (gst_audio_test_src_create_sine_table), (plugin_init):
31237 * gst/volume/gstvolume.c: (plugin_init):
31238 initialize gst_controller before using
31240 2006-01-31 16:26:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31242 tests/check/pipelines/: Define constant using G_GINT64_CONSTANT to avoid errors when passing it around - otherwise it...
31243 Original commit message from CVS:
31244 * tests/check/pipelines/theoraenc.c:
31245 * tests/check/pipelines/vorbisenc.c:
31246 Define constant using G_GINT64_CONSTANT to avoid errors when
31247 passing it around - otherwise it gets truncated to 32 bits.
31248 Fixes failing tests.
31250 2006-01-31 15:36:13 +0000 Andy Wingo <wingo@pobox.com>
31252 sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the caps being set doesn't have a framerate value. Basic...
31253 Original commit message from CVS:
31254 2006-01-31 Andy Wingo <wingo@pobox.com>
31255 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
31256 caps being set doesn't have a framerate value. Basically a stopgap
31258 * ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
31259 technically correct enough to put into core though.
31260 (gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
31261 DURATION. Fixes theoraenc ! oggmux.
31262 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
31263 fraction, not double.
31265 2006-01-31 12:23:35 +0000 Christian Schaller <uraeus@gnome.org>
31267 * gst-plugins-base.spec.in:
31268 update with latest files
31269 Original commit message from CVS:
31270 update with latest files
31272 2006-01-30 23:42:54 +0000 Sébastien Moutte <sebastien@moutte.net>
31274 win32/vs7: add vs7 project files created by Sergey Scobich
31275 Original commit message from CVS:
31277 add vs7 project files created by Sergey Scobich
31279 2006-01-30 22:18:53 +0000 Sébastien Moutte <sebastien@moutte.net>
31281 win32/vs8: add vs8 project files created by Sergey Scobich
31282 Original commit message from CVS:
31284 add vs8 project files created by Sergey Scobich
31286 2006-01-30 19:22:22 +0000 Andy Wingo <wingo@pobox.com>
31288 ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare timestamp + duration, not just timestamp -- ogg pages should ...
31289 Original commit message from CVS:
31290 2006-01-30 Andy Wingo <wingo@pobox.com>
31291 * ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
31292 timestamp + duration, not just timestamp -- ogg pages should be
31293 ordered by stop time. Necessary fix given the change in vorbis
31296 2006-01-30 19:21:07 +0000 Andy Wingo <wingo@pobox.com>
31299 * ext/theora/gsttheoraenc.h:
31300 * ext/theora/theoraenc.c:
31301 * tests/check/pipelines/theoraenc.c:
31302 ext/theora/theoraenc.c (theora_enc_sink_setcaps)
31303 Original commit message from CVS:
31304 2006-01-30 Andy Wingo <wingo@pobox.com>
31305 * ext/theora/theoraenc.c (theora_enc_sink_setcaps)
31306 (gst_theora_enc_init): Pull the granule shift out of the encoder.
31307 (granulepos_add): New function, handles the messiness of adjusting
31309 (theora_buffer_from_packet):
31310 (theora_enc_chain):
31311 (theora_enc_sink_event): Use granulepos_add, not +.
31312 * tests/check/pipelines/theoraenc.c
31313 (check_buffer_granulepos_from_starttime): Just check the frame
31314 count, not the actual granulepos -- we can't dictate to the
31315 encoder when it should be placing keyframes.
31317 2006-01-30 18:17:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31319 ext/gnomevfs/gstgnomevfssrc.c: SERVICE_NOT_AVAILABLE happens for example when you're trying to play an http:// stream...
31320 Original commit message from CVS:
31321 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
31322 SERVICE_NOT_AVAILABLE happens for example when you're trying to
31323 play an http:// stream from a server that's not serving
31325 2006-01-30 17:08:11 +0000 Andy Wingo <wingo@pobox.com>
31327 tests/check/pipelines/: Totally remove the UINT64_CONSTANT macro, doesn't appear to be needed or available.
31328 Original commit message from CVS:
31329 2006-01-30 Andy Wingo <wingo@pobox.com>
31330 * tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
31331 * tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
31332 remove the UINT64_CONSTANT macro, doesn't appear to be needed or
31335 2006-01-30 17:01:54 +0000 Andy Wingo <wingo@pobox.com>
31337 ext/theora/: Same changes as were done to vorbisenc, although theoraenc was timestamping correctly. Added handling of...
31338 Original commit message from CVS:
31339 2006-01-30 Andy Wingo <wingo@pobox.com>
31340 * ext/theora/gsttheoraenc.h:
31341 * ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
31342 although theoraenc was timestamping correctly. Added handling of
31343 streams that start with nonzero timestamps.
31344 * tests/check/Makefile.am:
31345 * tests/check/pipelines/theoraenc.c: New file, basically does same
31346 tests as vorbisenc.
31347 * tests/check/pipelines/vorbisenc.c: I claim these bugs.
31349 2006-01-30 16:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
31351 gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
31352 Original commit message from CVS:
31353 * gst-libs/gst/audio/gstaudiosink.c:
31354 (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
31355 (gst_audioringbuffer_pause):
31356 Implement pause that does not wait for completion.
31357 * gst-libs/gst/audio/gstbaseaudiosink.c:
31358 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
31359 Don't drop buffers when going to PAUSED but perform preroll on
31360 remaining samples now that core base class supports this.
31361 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
31362 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
31363 (gst_ring_buffer_commit):
31364 Pause should not signal waiters.
31365 Implement return value of _commit correctly.
31367 2006-01-30 15:01:28 +0000 Andy Wingo <wingo@pobox.com>
31369 tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
31370 Original commit message from CVS:
31371 2006-01-30 Andy Wingo <wingo@pobox.com>
31372 * tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
31373 * ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
31374 updated to timestamp from the first sample, not the last.
31375 (gst_vorbisenc_buffer_from_header_packet): New function, takes
31376 special care of granulepos and timestamp for header packets.
31377 (gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
31378 when the first buffer has a nonzero timestamp.
31379 * ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
31380 (GstVorbisEnc.subgranule_offset): New members. Take care of the
31381 case when the first audio buffer we get has a nonzero timestamp.
31382 (GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
31383 properly timestamp vorbis buffers with the time of the first
31384 sample, not the last.
31385 * ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
31386 vorbis_granule_time_copy -- now it takes the granule/subgranule
31387 offset into account.
31388 * tests/check/pipelines/vorbisenc.c: New test for correctness of
31389 timestamps, durations, and granulepos on buffers produced by
31392 2006-01-30 14:42:28 +0000 Eric Jonas <jonas@mit.edu>
31394 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Patch from Eric Jonas to support conversions to/from UYVY (Fixes: #324626)
31395 Original commit message from CVS:
31396 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
31397 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
31398 Patch from Eric Jonas to support conversions to/from UYVY
31401 2006-01-30 08:11:14 +0000 Julien Moutte <julien@moutte.net>
31403 gst/playback/: Implement subtitles.
31404 Original commit message from CVS:
31405 2006-01-30 Julien MOUTTE <julien@moutte.net>
31406 * gst/playback/gstplaybasebin.c: (group_commit),
31408 (setup_subtitle), (setup_source), (set_active_source):
31409 * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
31410 (gen_text_element), (gen_audio_element), (gen_vis_element),
31411 (remove_sinks), (add_sink), (setup_sinks): Implement subtitles.
31413 2006-01-29 19:13:39 +0000 Sébastien Moutte <sebastien@moutte.net>
31415 gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
31416 Original commit message from CVS:
31417 * gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
31418 * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
31419 use of gst_guint64_to_gdouble to be compliant with vs6
31420 * gst/playback/gstdecodebin.c: (try_to_link_1)
31421 * gst/videorate/videorate.c: (gst_video_rate_blank_data)
31422 use of G_GINT64_CONSTANT for int64 constants
31423 * win32/common/libgstinterfaces.def:
31424 export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
31426 update and add new project files
31428 2006-01-29 18:21:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31430 add a win32-update rule like in core, and copy over enumtypes files
31431 Original commit message from CVS:
31434 * win32/common/interfaces-enumtypes.c:
31435 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
31436 (gst_mixer_track_flags_get_type),
31437 (gst_tuner_channel_flags_get_type):
31438 * win32/common/interfaces-enumtypes.h:
31439 * win32/common/multichannel-enumtypes.c:
31440 (gst_audio_channel_position_get_type):
31441 * win32/common/multichannel-enumtypes.h:
31442 add a win32-update rule like in core, and copy over enumtypes files
31444 2006-01-29 18:07:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31447 generate win32/common/config.h
31448 Original commit message from CVS:
31449 generate win32/common/config.h
31451 2006-01-29 18:05:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31453 win32/: add config files just like in core
31454 Original commit message from CVS:
31456 * win32/common/config.h:
31457 * win32/common/config.h.in:
31458 add config files just like in core
31460 2006-01-28 18:22:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31462 ext/alsa/: Update all error messages. All of them should either use the default translated message, or actually prov...
31463 Original commit message from CVS:
31464 * ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
31465 (set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
31466 (gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
31467 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
31468 (set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
31469 (gst_alsasrc_unprepare), (gst_alsasrc_read):
31470 Update all error messages. All of them should either use
31471 the default translated message, or actually provide a
31472 translatable string.
31473 Make the string for channel count problems meaningful.
31475 2006-01-28 18:19:18 +0000 Tim-Philipp Müller <tim@centricular.net>
31477 gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
31478 Original commit message from CVS:
31479 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
31480 Make gcc-4.1 happy (part of #327357).
31482 2006-01-28 16:35:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31484 sys/v4l/v4l_calls.c: check for and throw RESOURCE_BUSY
31485 Original commit message from CVS:
31486 * sys/v4l/v4l_calls.c: (gst_v4l_open):
31487 check for and throw RESOURCE_BUSY
31489 2006-01-28 02:13:14 +0000 David Schleef <ds@schleef.org>
31491 gst/videoscale/vs_scanline.c: Oops, *that's* why I never checked in this change -- it requires liboil features not in...
31492 Original commit message from CVS:
31493 * gst/videoscale/vs_scanline.c: Oops, *that's* why I never
31494 checked in this change -- it requires liboil features not
31495 in 0.3.6. Revert parts.
31497 2006-01-27 23:40:19 +0000 David Schleef <ds@schleef.org>
31499 update liboil requirement to 0.3.6
31500 Original commit message from CVS:
31502 * configure.ac: update liboil requirement to 0.3.6
31503 * gst/videoscale/Makefile.am:
31504 * gst/videoscale/vs_scanline.c: liboilify
31506 2006-01-27 17:00:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31508 ext/libvisual/visual.c: When pad_alloc returns a GstFlowReturn other than GST_FLOW_OK, make sure it is passed upstream.
31509 Original commit message from CVS:
31510 * ext/libvisual/visual.c: (get_buffer):
31511 When pad_alloc returns a GstFlowReturn other
31512 than GST_FLOW_OK, make sure it is passed upstream.
31514 2006-01-27 01:36:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31516 ext/alsa/gstalsasink.c: Free the device name string.
31517 Original commit message from CVS:
31518 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
31519 (gst_alsasink_class_init):
31520 Free the device name string.
31521 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
31522 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
31523 (gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
31524 Don't remove a pad from the collectpads structure until it
31525 is released - it's a request pad, and may receive data again
31526 if the element gets moved back to PLAYING state.
31527 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
31528 Ensure we turn on double buffering on the Xv port, and
31529 set the colour key to something dark and mysterious that
31532 2006-01-27 01:06:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31534 ext/: - a library should not call setlocale. see Libraries node in gettext manual
31535 Original commit message from CVS:
31536 * ext/alsa/gstalsaplugin.c: (plugin_init):
31537 * ext/cdparanoia/gstcdparanoiasrc.c:
31538 (gst_cd_paranoia_src_base_init), (plugin_init):
31539 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
31540 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
31541 - a library should not call setlocale. see Libraries node in
31543 - make sure all plugins that use translation do bindtextdomain
31544 to point to the localedir
31545 * gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
31546 (setup_sinks), (plugin_init):
31547 all this, and check for NULL when creating sinks
31549 2006-01-26 23:21:31 +0000 Julien Moutte <julien@moutte.net>
31551 gst/subparse/gstsubparse.c: Make typefinding of subtitles work again.
31552 Original commit message from CVS:
31553 2006-01-27 Julien MOUTTE <julien@moutte.net>
31554 * gst/subparse/gstsubparse.c: (gst_subparse_type_find),
31555 (plugin_init): Make typefinding of subtitles work again.
31557 2006-01-26 20:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
31559 gst/typefind/gsttypefindfunctions.c: Backport a bunch of typefinding fixes from the 0.8 branch.
31560 Original commit message from CVS:
31561 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
31562 (mp3_type_frame_length_from_header), (mp3_type_find),
31563 (wavpack_type_find), (m4a_type_find), (ircam_type_find),
31565 Backport a bunch of typefinding fixes from the 0.8 branch.
31566 Also, improve wavpack typefinding: if we can't peek the
31567 entire wavpack block, try to parse the bits we can get and
31568 see if we find what we're looking for in those.
31570 2006-01-26 19:17:38 +0000 Julien Moutte <julien@moutte.net>
31572 sys/: Handle some more cases of pixel aspect ratio.
31573 Original commit message from CVS:
31574 2006-01-26 Julien MOUTTE <julien@moutte.net>
31575 * sys/ximage/ximagesink.c:
31576 (gst_ximagesink_calculate_pixel_aspect_ratio):
31577 * sys/xvimage/xvimagesink.c:
31578 (gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
31579 more cases of pixel aspect ratio.
31581 2006-01-26 13:09:24 +0000 Edward Hervey <bilboed@bilboed.com>
31583 gst/playback/gstdecodebin.c: Also consider the flush-start and tag events as unblockers for the pad probes.
31584 Original commit message from CVS:
31585 * gst/playback/gstdecodebin.c: (pad_probe):
31586 Also consider the flush-start and tag events as unblockers
31587 for the pad probes.
31589 2006-01-26 12:32:58 +0000 Julien Moutte <julien@moutte.net>
31591 gst/playback/gstplaybin.c: On the fly visualisation switch, works disabling, enabling as well but it won't be able to...
31592 Original commit message from CVS:
31593 2006-01-26 Julien MOUTTE <julien@moutte.net>
31594 * gst/playback/gstplaybin.c: (gst_play_bin_init),
31595 (gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
31596 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
31597 On the fly visualisation switch, works disabling, enabling as
31598 well but it won't be able to enable vis in a playbin that was
31599 created with no visualisation.
31601 2006-01-25 10:50:32 +0000 Wim Taymans <wim.taymans@gmail.com>
31603 gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
31604 Original commit message from CVS:
31605 * gst-libs/gst/audio/gstbaseaudiosink.c:
31606 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
31607 Undo previous commit, it breaks resume after pause.
31609 2006-01-25 09:27:01 +0000 Wim Taymans <wim.taymans@gmail.com>
31611 gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
31612 Original commit message from CVS:
31613 * gst-libs/gst/audio/gstbaseaudiosink.c:
31614 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
31615 (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
31617 Post error when caps cannot be parsed.
31618 Resync on discontinuity in the stream.
31619 Clip samples to segment boundaries.
31620 return WRONG_STATE sooner when we are flushing.
31621 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
31622 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
31623 Make audiosrc operate in TIME.
31624 Set TIMESTAMP and DURATION on buffers.
31626 2006-01-24 21:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
31628 tests/examples/seek/seek.c: Output tag messages as well.
31629 Original commit message from CVS:
31630 * tests/examples/seek/seek.c: (main):
31631 Output tag messages as well.
31633 2006-01-23 15:05:24 +0000 Edward Hervey <bilboed@bilboed.com>
31635 gst/playback/gstdecodebin.c: Replace GstPadBlockCallback with pad probes that detect first buffer AND eos before remo...
31636 Original commit message from CVS:
31637 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
31638 (free_pad_probes), (remove_fakesink), (pad_probe),
31639 (close_pad_link), (gst_decode_bin_change_state):
31640 Replace GstPadBlockCallback with pad probes that detect
31641 first buffer AND eos before removing fakesink.
31642 Fixes hang with demuxers doing EOS while pre-rolling.
31645 2006-01-23 10:10:36 +0000 Jens Granseuer <jensgr@gmx.net>
31647 GCC 2.95 fixes (#328263).
31648 Original commit message from CVS:
31649 2006-01-23 Andy Wingo <wingo@pobox.com>
31650 * ext/alsa/gstalsasink.c:
31651 * gst-libs/gst/rtp/gstbasertpdepayload.c:
31652 (gst_base_rtp_depayload_setcaps),
31653 (gst_base_rtp_depayload_add_to_queue),
31654 (gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
31655 Patch by: Jens Granseuer <jensgr at gmx dot net>
31657 2006-01-22 17:24:02 +0000 Julien Moutte <julien@moutte.net>
31659 sys/: Playbin keeps some ref to some frames. We might get a frame destroyed after changing state to
31660 Original commit message from CVS:
31661 2006-01-22 Julien MOUTTE <julien@moutte.net>
31662 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
31663 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
31664 (gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
31665 frames. We might get a frame destroyed after changing state to
31666 NULL, adding a safety check on xcontext.
31668 2006-01-22 14:50:53 +0000 Tim-Philipp Müller <tim@centricular.net>
31670 gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
31671 Original commit message from CVS:
31672 * gst-libs/gst/interfaces/xoverlay.c:
31673 Fix prepare-xwindow-id code example in the docs - we need to
31674 ignore all messages that aren't element messages as well.
31676 2006-01-21 22:40:03 +0000 Julien Moutte <julien@moutte.net>
31678 sys/xvimage/xvimagesink.c: I think one day i'll completely undestand how caps negotiation is supposed to work. This r...
31679 Original commit message from CVS:
31680 2006-01-21 Julien MOUTTE <julien@moutte.net>
31681 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
31682 I think one day i'll completely undestand how caps negotiation
31683 is supposed to work. This refactoring handles buffer_alloc
31684 called with caps we can't handle. We definitely don't want a
31685 set_caps with those caps, so we define and allocate a buffer
31686 we would like to receive.
31688 2006-01-20 19:10:26 +0000 Christian Schaller <uraeus@gnome.org>
31692 up automake requirement to 1.7
31693 Original commit message from CVS:
31694 up automake requirement to 1.7
31696 2006-01-19 10:59:51 +0000 Tim-Philipp Müller <tim@centricular.net>
31698 gst/playback/gstplaybasebin.c: Free iterator when done.
31699 Original commit message from CVS:
31700 * gst/playback/gstplaybasebin.c: (setup_source):
31701 Free iterator when done.
31703 2006-01-17 11:43:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31705 gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
31706 Original commit message from CVS:
31707 * gst-libs/gst/audio/gstbaseaudiosink.c:
31708 (gst_base_audio_sink_render):
31709 Fix playback of non-synchronised streams by assuming a rate
31710 of 1.0 instead of a random one.
31711 Makes this work again:
31712 gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
31713 endianness=(int)4321, signed=(boolean)true, width=(int)16,
31714 depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
31715 audioresample ! alsasink
31717 2006-01-16 21:01:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31721 Original commit message from CVS:
31724 === release 0.10.2 ===
31726 2006-01-16 20:59:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31732 * docs/plugins/gst-plugins-base-plugins.args:
31733 * docs/plugins/inspect/plugin-adder.xml:
31734 * docs/plugins/inspect/plugin-alsa.xml:
31735 * docs/plugins/inspect/plugin-audioconvert.xml:
31736 * docs/plugins/inspect/plugin-audiorate.xml:
31737 * docs/plugins/inspect/plugin-audioresample.xml:
31738 * docs/plugins/inspect/plugin-audiotestsrc.xml:
31739 * docs/plugins/inspect/plugin-cdparanoia.xml:
31740 * docs/plugins/inspect/plugin-decodebin.xml:
31741 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
31742 * docs/plugins/inspect/plugin-gnomevfs.xml:
31743 * docs/plugins/inspect/plugin-libvisual.xml:
31744 * docs/plugins/inspect/plugin-ogg.xml:
31745 * docs/plugins/inspect/plugin-pango.xml:
31746 * docs/plugins/inspect/plugin-playbin.xml:
31747 * docs/plugins/inspect/plugin-subparse.xml:
31748 * docs/plugins/inspect/plugin-tcp.xml:
31749 * docs/plugins/inspect/plugin-theora.xml:
31750 * docs/plugins/inspect/plugin-typefindfunctions.xml:
31751 * docs/plugins/inspect/plugin-video4linux.xml:
31752 * docs/plugins/inspect/plugin-videorate.xml:
31753 * docs/plugins/inspect/plugin-videoscale.xml:
31754 * docs/plugins/inspect/plugin-videotestsrc.xml:
31755 * docs/plugins/inspect/plugin-volume.xml:
31756 * docs/plugins/inspect/plugin-vorbis.xml:
31757 * docs/plugins/inspect/plugin-ximagesink.xml:
31758 * docs/plugins/inspect/plugin-xvimagesink.xml:
31760 Original commit message from CVS:
31763 2006-01-16 16:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31765 gst/playback/: Comment out broken code that connects to the state-changed signal.
31766 Original commit message from CVS:
31767 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
31768 * gst/playback/gststreamselector.c:
31769 (gst_stream_selector_set_property):
31770 Comment out broken code that connects to the state-changed signal.
31771 At this point, changing current stream selection is broken, but
31772 stuff like gst-launch playbin current-audio=1 works and filters
31773 to the chosen stream.
31775 2006-01-16 15:31:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31777 ext/vorbis/vorbisdec.c: Fix #327216 (null dereference in vorbisdec)
31778 Original commit message from CVS:
31779 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
31780 Fix #327216 (null dereference in vorbisdec)
31782 2006-01-16 15:19:55 +0000 Tim-Philipp Müller <tim@centricular.net>
31784 ext/theora/theoradec.c: Post taglist actually on bus instead of just freeing it (fixes #327114 and totem bug #327080).
31785 Original commit message from CVS:
31786 * ext/theora/theoradec.c: (theora_handle_comment_packet):
31787 Post taglist actually on bus instead of just freeing it
31788 (fixes #327114 and totem bug #327080).
31789 * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
31790 Use gst_element_found_tags_for_pad(), so that the tags
31791 are sent downstream as an event as well.
31793 2006-01-15 10:06:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31795 sys/: move all regularly occurring messages to GST_LOG level add some more object logs
31796 Original commit message from CVS:
31797 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
31798 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
31799 (gst_ximagesink_buffer_alloc):
31800 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
31801 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
31802 (gst_xvimagesink_buffer_alloc):
31803 move all regularly occurring messages to GST_LOG level
31804 add some more object logs
31806 2006-01-14 22:59:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31824 Original commit message from CVS:
31827 2006-01-14 20:46:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31829 ext/ogg/gstoggmux.c: fix a silly segfault
31830 Original commit message from CVS:
31831 2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
31832 * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
31833 fix a silly segfault
31835 2006-01-14 12:52:22 +0000 Tim-Philipp Müller <tim@centricular.net>
31837 Add docs for mixerutils stuff.
31838 Original commit message from CVS:
31839 * docs/libs/gst-plugins-base-libs-docs.sgml:
31840 * docs/libs/gst-plugins-base-libs-sections.txt:
31841 * gst-libs/gst/audio/mixerutils.c:
31842 * gst-libs/gst/audio/mixerutils.h:
31843 Add docs for mixerutils stuff.
31845 2006-01-13 17:17:07 +0000 Tim-Philipp Müller <tim@centricular.net>
31847 gst/playback/gstplaybasebin.c: Fix playback for sources that emit raw audio or raw video streams (e.g.: cd audio sour...
31848 Original commit message from CVS:
31849 * gst/playback/gstplaybasebin.c: (setup_source):
31850 Fix playback for sources that emit raw audio or
31851 raw video streams (e.g.: cd audio sources) (#325984).
31853 2006-01-13 16:45:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31855 gst-libs/gst/audio/mixerutils.c: actually save the element we create
31856 Original commit message from CVS:
31857 * gst-libs/gst/audio/mixerutils.c:
31858 (gst_audio_mixer_filter_do_filter):
31859 actually save the element we create
31861 2006-01-13 16:17:50 +0000 Christian Schaller <uraeus@gnome.org>
31863 * gst-plugins-base.spec.in:
31864 remove version suffix
31865 Original commit message from CVS:
31866 remove version suffix
31868 2006-01-12 14:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
31870 gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
31871 Original commit message from CVS:
31872 * gst-libs/gst/cdda/gstcddabasesrc.c:
31873 (gst_cdda_base_src_handle_track_seek):
31874 No need to post a tag message on the bus when seeking
31875 within the same track, only post it when the current
31878 2006-01-11 18:30:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31880 gst/playback/: Reenable stream selection. These mechanisms need a complete overhaul in the face of 0.8->0.10 changes ...
31881 Original commit message from CVS:
31882 * gst/playback/gstplaybasebin.c: (group_destroy),
31883 (probe_triggered), (new_decoded_pad), (mute_group_type),
31884 (set_active_source):
31885 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
31886 * gst/playback/gststreamselector.c:
31887 (gst_stream_selector_base_init),
31888 (gst_stream_selector_set_property),
31889 (gst_stream_selector_request_new_pad):
31890 Reenable stream selection. These mechanisms need a complete overhaul
31891 in the face of 0.8->0.10 changes though.
31893 2006-01-11 18:03:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31895 ext/ogg/gstoggdemux.c: Change the pad template to src_%d to match the pads that are created from it. decodebin needs ...
31896 Original commit message from CVS:
31897 * ext/ogg/gstoggdemux.c:
31898 Change the pad template to src_%d to match the pads that
31899 are created from it. decodebin needs this information in order
31900 to decide that oggdemux is capable of producing multiple pads
31901 (and hence needs queues inserted).
31902 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
31903 (gst_ogg_mux_collected):
31904 Make debug output more useful by using GST_PTR_FORMAT.
31906 2006-01-11 17:38:35 +0000 Christian Schaller <uraeus@gnome.org>
31908 * gst-plugins-base.spec.in:
31909 update spec.in file
31910 Original commit message from CVS:
31911 update spec.in file
31913 2006-01-11 15:11:20 +0000 Tim-Philipp Müller <tim@centricular.net>
31915 gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
31916 Original commit message from CVS:
31917 Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
31918 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
31919 Set depth and width for alaw/mulaw (fixes #326601).
31921 2006-01-10 23:58:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31923 tests/icles/Makefile.am: don't build the tests if we don't have the libs
31924 Original commit message from CVS:
31925 * tests/icles/Makefile.am:
31926 don't build the tests if we don't have the libs
31928 2006-01-10 18:06:56 +0000 Tim-Philipp Müller <tim@centricular.net>
31930 ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
31931 Original commit message from CVS:
31932 * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
31933 (gst_cd_paranoia_paranoia_callback):
31934 Don't try to free NULL pointers.
31936 2006-01-10 15:47:48 +0000 Edward Hervey <bilboed@bilboed.com>
31938 gst/audiorate/gstaudiorate.c: Add debugging category.
31939 Original commit message from CVS:
31940 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
31941 (gst_audio_rate_change_state), (plugin_init):
31942 Add debugging category.
31944 Add case for incoming buffers without valid offset/offset_end.
31946 2006-01-10 12:25:59 +0000 Michael Smith <msmith@xiph.org>
31948 gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
31949 Original commit message from CVS:
31950 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
31951 Don't leak GCond in audio sources.
31953 2006-01-10 11:49:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31955 gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
31956 Original commit message from CVS:
31957 * gst/playback/gstplaybin.c: (gen_audio_element):
31958 Don't leak an autoaudiosink/alsasink when we generate
31959 a new audio element. (old code, I guess)
31961 2006-01-10 11:04:21 +0000 Michael Smith <msmith@xiph.org>
31963 gst/audiorate/gstaudiorate.c: Support float audio in audiorate.
31964 Original commit message from CVS:
31965 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
31966 Support float audio in audiorate.
31967 Use width rather than depth for selecting sample width.
31969 2006-01-10 10:06:53 +0000 Tim-Philipp Müller <tim@centricular.net>
31971 gst/videotestsrc/videotestsrc.h: Use GLib types here (that way we don't have to include the generated _stdint.h heade...
31972 Original commit message from CVS:
31973 * gst/videotestsrc/videotestsrc.h:
31974 Use GLib types here (that way we don't have to include the
31975 generated _stdint.h header, which makes life easier for win32
31976 folks that don't use autotools for the build) (#325990, patch
31977 by: Sergey Scobich).
31979 2006-01-10 09:38:44 +0000 Tim-Philipp Müller <tim@centricular.net>
31981 gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
31982 Original commit message from CVS:
31983 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
31984 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
31985 (gst_ring_buffer_pause), (wait_segment):
31986 * gst-libs/gst/audio/gstringbuffer.h:
31987 Name (private) union, makes Forte compiler happy (this time
31988 for real) (#324900).
31990 2006-01-09 10:52:33 +0000 Tim-Philipp Müller <tim@centricular.net>
31992 gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
31993 Original commit message from CVS:
31994 * gst-libs/gst/audio/Makefile.am:
31995 Link against libgstinterfaces, needed for mixer
31996 and property probe stuff.
31998 2006-01-09 10:46:52 +0000 Edward Hervey <bilboed@bilboed.com>
32000 gst-libs/gst/Makefile.am:
32001 Original commit message from CVS:
32002 * gst-libs/gst/Makefile.am:
32004 2006-01-09 09:38:34 +0000 Tim-Philipp Müller <tim@centricular.net>
32006 gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
32007 Original commit message from CVS:
32008 * gst-libs/gst/audio/Makefile.am:
32009 * gst-libs/gst/audio/mixerutils.c:
32010 (gst_audio_mixer_filter_do_filter),
32011 (gst_audio_mixer_filter_check_element),
32012 (gst_audio_mixer_filter_probe_feature),
32013 (element_factory_rank_compare_func),
32014 (gst_audio_default_registry_mixer_filter):
32015 * gst-libs/gst/audio/mixerutils.h:
32016 Add gst_audio_default_registry_mixer_filter() utility
32019 2006-01-03 17:33:16 +0000 Michael Smith <msmith@xiph.org>
32021 gst/audioresample/resample.h: As before, but for o_buf
32022 Original commit message from CVS:
32023 * gst/audioresample/resample.h:
32024 As before, but for o_buf
32026 2006-01-03 17:27:13 +0000 Michael Smith <msmith@xiph.org>
32028 gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
32029 Original commit message from CVS:
32030 * gst/audioresample/resample.h:
32031 Declare struct _ResampleState.buffer as unsigned char *, not void *,
32032 since we do arithmetic on it.
32034 2006-01-02 23:37:38 +0000 Tim-Philipp Müller <tim@centricular.net>
32036 gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
32037 Original commit message from CVS:
32038 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
32039 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
32040 (gst_ring_buffer_pause), (wait_segment):
32041 * gst-libs/gst/audio/gstringbuffer.h:
32042 Sun's Forte compiler doesn't seem to like anonymous structs,
32043 so use same setup as in GstBaseSrc (fixes #324900).
32045 2005-12-30 14:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32047 move old example to tests/examples/volume/volune.c
32048 Original commit message from CVS:
32050 * gst/volume/Makefile.am:
32051 * gst/volume/demo.c:
32052 move old example to tests/examples/volume/volune.c
32053 * tests/examples/Makefile.am:
32054 * tests/examples/seek/seek.c: (main):
32055 change window-close event from "delete-event" to "destroy"
32056 * tests/examples/volume/Makefile.am:
32057 * tests/examples/volume/volume.c: (value_changed_callback),
32058 (setup_gui), (message_received), (eos_message_received), (main):
32059 fix event handling and bus usage
32061 2005-12-29 20:37:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32063 gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...
32064 Original commit message from CVS:
32065 * gst/audiotestsrc/gstaudiotestsrc.c:
32066 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
32067 (gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
32068 (gst_audio_test_src_query), (gst_audio_test_src_create_sine),
32069 (gst_audio_test_src_create_square),
32070 (gst_audio_test_src_create_saw),
32071 (gst_audio_test_src_create_triangle),
32072 (gst_audio_test_src_create_silence),
32073 (gst_audio_test_src_create_white_noise),
32074 (gst_audio_test_src_create_pink_noise),
32075 (gst_audio_test_src_init_sine_table),
32076 (gst_audio_test_src_create_sine_table),
32077 (gst_audio_test_src_change_wave),
32078 (gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
32079 (gst_audio_test_src_create), (gst_audio_test_src_set_property):
32080 * gst/audiotestsrc/gstaudiotestsrc.h:
32081 update to basesrc changes, implement segmented seeking and eos handling,
32082 add a 'sine-tab' waveform for performance critical playback
32084 2005-12-29 16:17:55 +0000 Tim-Philipp Müller <tim@centricular.net>
32086 po/POTFILES.in: ... and this time the other modified file that I missed last time.
32087 Original commit message from CVS:
32089 ... and this time the other modified file that I missed last time.
32091 2005-12-29 14:31:49 +0000 Michael Smith <msmith@xiph.org>
32093 gst/playback/gstdecodebin.c: Fix non-C89 variable declaration not at the start of a block. Should help some compilers.
32094 Original commit message from CVS:
32095 * gst/playback/gstdecodebin.c: (new_pad):
32096 Fix non-C89 variable declaration not at the start of a block. Should
32097 help some compilers.
32099 2005-12-29 12:43:22 +0000 Tim-Philipp Müller <tim@centricular.net>
32101 tests/check/Makefile.am: And now fix 'make distcheck' (builddir != srcdir)
32102 Original commit message from CVS:
32103 * tests/check/Makefile.am:
32104 And now fix 'make distcheck' (builddir != srcdir)
32106 2005-12-29 12:22:24 +0000 Tim-Philipp Müller <tim@centricular.net>
32108 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia plugin again (there are still fixes required to pla...
32109 Original commit message from CVS:
32111 * ext/cdparanoia/Makefile.am:
32112 * ext/cdparanoia/gstcdparanoia.c:
32113 * ext/cdparanoia/gstcdparanoia.h:
32114 * ext/cdparanoia/gstcdparanoiasrc.c:
32115 (gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
32116 (gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
32117 (gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
32118 (gst_cd_paranoia_paranoia_callback),
32119 (gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
32120 (gst_cd_paranoia_src_set_property),
32121 (gst_cd_paranoia_src_get_property), (plugin_init):
32122 * ext/cdparanoia/gstcdparanoiasrc.h:
32123 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
32124 plugin again (there are still fixes required to playbin to make
32125 cdda:// uris work there).
32127 2005-12-29 12:13:57 +0000 Tim-Philipp Müller <tim@centricular.net>
32129 tests/check/Makefile.am: Fix test case compilation.
32130 Original commit message from CVS:
32131 * tests/check/Makefile.am:
32132 Fix test case compilation.
32134 2005-12-29 11:49:11 +0000 Tim-Philipp Müller <tim@centricular.net>
32136 gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
32137 Original commit message from CVS:
32138 * gst-libs/gst/cdda/gstcddabasesrc.c:
32139 (gst_cdda_base_src_update_duration),
32140 (gst_cdda_base_src_calculate_cddb_id):
32141 An integer is not a string. Fix access to uninitialised variable.
32142 * tests/check/Makefile.am:
32143 Add cddabasesrc unit test; also actually enable the vorbis test.
32144 * tests/check/generic/states.c:
32145 Blacklist new cd audio elements as well.
32146 * tests/check/libs/cddabasesrc.c:
32147 Unit test for GstCddaBaseSrc (discid calculation mostly).
32149 2005-12-28 18:19:25 +0000 Tim-Philipp Müller <tim@centricular.net>
32151 docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
32152 Original commit message from CVS:
32153 * docs/libs/Makefile.am:
32154 * docs/libs/gst-plugins-base-libs-docs.sgml:
32155 * docs/libs/gst-plugins-base-libs-sections.txt:
32156 * docs/libs/gst-plugins-base-libs.types:
32157 Add docs for libgstcdda/GstCddaBaseSrc.
32158 * gst-libs/gst/interfaces/mixertrack.h:
32159 Do one struct member per line with a semicolon at the end, that way
32160 even gtk-doc might parse it without complaining.
32162 2005-12-28 18:06:50 +0000 Tim-Philipp Müller <tim@centricular.net>
32164 Add new libgstcdda with GstCddaBaseSrc class.
32165 Original commit message from CVS:
32167 * gst-libs/gst/Makefile.am:
32168 * gst-libs/gst/cdda/Makefile.am:
32169 * gst-libs/gst/cdda/base64.c:
32170 * gst-libs/gst/cdda/base64.h:
32171 * gst-libs/gst/cdda/gstcddabasesrc.c:
32172 (gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
32173 (gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
32174 (gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
32175 (gst_cdda_base_src_get_property),
32176 (gst_cdda_base_src_get_track_from_sector),
32177 (gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
32178 (gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
32179 (gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
32180 (gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
32181 (gst_cdda_base_src_uri_get_protocols),
32182 (gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
32183 (gst_cdda_base_src_uri_handler_init),
32184 (gst_cdda_base_src_setup_interfaces),
32185 (gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
32186 (cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
32187 (lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
32188 (gst_cdda_base_src_add_tags),
32189 (gst_cdda_base_src_add_index_associations),
32190 (gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
32191 (gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
32192 (gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
32193 (gst_cdda_base_src_create):
32194 * gst-libs/gst/cdda/gstcddabasesrc.h:
32195 * gst-libs/gst/cdda/sha1.c:
32196 * gst-libs/gst/cdda/sha1.h:
32197 Add new libgstcdda with GstCddaBaseSrc class.
32199 2005-12-28 14:59:41 +0000 Tim-Philipp Müller <tim@centricular.net>
32201 ext/gnomevfs/gstgnomevfssink.h: Use GstBaseSinkClass as parent_class member for class struct, not
32202 Original commit message from CVS:
32203 * ext/gnomevfs/gstgnomevfssink.h:
32204 Use GstBaseSinkClass as parent_class member for class struct, not
32207 2005-12-27 22:29:43 +0000 Tim-Philipp Müller <tim@centricular.net>
32209 gst/videotestsrc/gstvideotestsrc.c: Add start method to reset running time and number of frames sent
32210 Original commit message from CVS:
32211 * gst/videotestsrc/gstvideotestsrc.c:
32212 (gst_video_test_src_class_init), (gst_video_test_src_start):
32213 Add start method to reset running time and number of frames sent
32214 when starting up (fixes #324696; patch by: Michal Benes).
32216 2005-12-27 21:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
32218 docs/plugins/: Add docs stuff for gnomevfssrc and gnomevfssink.
32219 Original commit message from CVS:
32220 * docs/plugins/Makefile.am:
32221 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
32222 * docs/plugins/gst-plugins-base-plugins-sections.txt:
32223 * docs/plugins/gst-plugins-base-plugins.args:
32224 * docs/plugins/gst-plugins-base-plugins.hierarchy:
32225 * docs/plugins/gst-plugins-base-plugins.signals:
32226 Add docs stuff for gnomevfssrc and gnomevfssink.
32227 * ext/gnomevfs/gstgnomevfssrc.c:
32228 Fix example pipeline in gtk-doc blurb.
32230 2005-12-27 21:42:23 +0000 Tim-Philipp Müller <tim@centricular.net>
32232 ext/gnomevfs/: Port gnomevfssink; add gtk-doc blurb.
32233 Original commit message from CVS:
32234 * ext/gnomevfs/Makefile.am:
32235 * ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
32236 (gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
32237 (gst_gnome_vfs_handle_get_type), (plugin_init):
32238 * ext/gnomevfs/gstgnomevfs.h:
32239 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
32240 (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
32241 (gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
32242 (gst_gnome_vfs_sink_set_property),
32243 (gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
32244 (gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
32245 (gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
32246 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
32247 (gst_gnome_vfs_sink_uri_get_type),
32248 (gst_gnome_vfs_sink_uri_get_protocols),
32249 (gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
32250 (gst_gnome_vfs_sink_uri_handler_init):
32251 * ext/gnomevfs/gstgnomevfssink.h:
32252 Port gnomevfssink; add gtk-doc blurb.
32253 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
32254 (gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
32255 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
32256 (gst_gnome_vfs_src_uri_get_type),
32257 (gst_gnome_vfs_src_uri_get_protocols),
32258 (gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
32259 (gst_gnome_vfs_src_uri_handler_init),
32260 (gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
32261 (gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
32262 (gst_gnome_vfs_src_send_additional_headers_callback),
32263 (gst_gnome_vfs_src_received_headers_callback),
32264 (gst_gnome_vfs_src_push_callbacks),
32265 (gst_gnome_vfs_src_pop_callbacks),
32266 (gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
32267 (gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
32268 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
32269 * ext/gnomevfs/gstgnomevfssrc.h:
32270 s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
32271 file; add gtk-doc blurb with example pipelines.
32273 2005-12-23 18:16:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32277 Original commit message from CVS:
32280 === release 0.10.1 ===
32282 2005-12-23 18:08:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32288 * docs/libs/tmpl/gstcolorbalance.sgml:
32289 * docs/plugins/gst-plugins-base-plugins.args:
32290 * docs/plugins/gst-plugins-base-plugins.signals:
32291 * docs/plugins/inspect/plugin-adder.xml:
32292 * docs/plugins/inspect/plugin-alsa.xml:
32293 * docs/plugins/inspect/plugin-audioconvert.xml:
32294 * docs/plugins/inspect/plugin-audiorate.xml:
32295 * docs/plugins/inspect/plugin-audioresample.xml:
32296 * docs/plugins/inspect/plugin-audiotestsrc.xml:
32297 * docs/plugins/inspect/plugin-decodebin.xml:
32298 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
32299 * docs/plugins/inspect/plugin-gnomevfs.xml:
32300 * docs/plugins/inspect/plugin-libvisual.xml:
32301 * docs/plugins/inspect/plugin-ogg.xml:
32302 * docs/plugins/inspect/plugin-pango.xml:
32303 * docs/plugins/inspect/plugin-playbin.xml:
32304 * docs/plugins/inspect/plugin-subparse.xml:
32305 * docs/plugins/inspect/plugin-tcp.xml:
32306 * docs/plugins/inspect/plugin-theora.xml:
32307 * docs/plugins/inspect/plugin-typefindfunctions.xml:
32308 * docs/plugins/inspect/plugin-video4linux.xml:
32309 * docs/plugins/inspect/plugin-videorate.xml:
32310 * docs/plugins/inspect/plugin-videoscale.xml:
32311 * docs/plugins/inspect/plugin-videotestsrc.xml:
32312 * docs/plugins/inspect/plugin-volume.xml:
32313 * docs/plugins/inspect/plugin-vorbis.xml:
32314 * docs/plugins/inspect/plugin-ximagesink.xml:
32315 * docs/plugins/inspect/plugin-xvimagesink.xml:
32317 Original commit message from CVS:
32320 2005-12-21 20:59:52 +0000 Edgard Lima <edgard.lima@indt.org.br>
32323 * gst/typefind/gsttypefindfunctions.c:
32324 iLBC30 and iLBC20 added to typefind.
32325 Original commit message from CVS:
32326 iLBC30 and iLBC20 added to typefind.
32328 2005-12-20 15:57:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32332 * docs/libs/tmpl/gstcolorbalance.sgml:
32348 Original commit message from CVS:
32351 2005-12-20 12:24:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32353 * gst-libs/gst/audio/gstbaseaudiosink.c:
32354 * gst-libs/gst/audio/gstbaseaudiosrc.c:
32355 stop making fun of older compilers
32356 Original commit message from CVS:
32357 stop making fun of older compilers
32359 2005-12-20 12:00:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32361 gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
32362 Original commit message from CVS:
32363 * gst-libs/gst/audio/gstbaseaudiosink.c:
32364 (gst_base_audio_sink_class_init):
32365 * gst-libs/gst/audio/gstbaseaudiosrc.c:
32366 (gst_base_audio_src_class_init):
32367 update strings, values are in microseconds
32368 change the default sink buffer time to something that is smaller
32369 (to help software volume mixing have a slightly lower delay) but
32370 still be acceptable on Wim's laptop
32372 2005-12-20 10:13:05 +0000 Edward Hervey <bilboed@bilboed.com>
32374 gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
32375 Original commit message from CVS:
32376 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
32377 Made a quack, forgot to add DUCK to the riff video template.
32379 2005-12-19 15:00:38 +0000 Edward Hervey <bilboed@bilboed.com>
32381 ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
32382 Original commit message from CVS:
32383 * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
32384 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
32385 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
32386 (gst_ogm_parse_chain):
32387 Make sure pads are initialized correctly.
32388 * gst-libs/gst/riff/riff-ids.h:
32389 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
32390 (gst_riff_create_video_template_caps):
32391 Add a whole bunch of FOURCC <=> MimeType.
32392 Extend the riff video pad template to support the newly added fourcc.
32394 2005-12-18 15:04:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32396 ext/ogg/gstoggdemux.c: Extra debug output when activating/deactivating chains.
32397 Original commit message from CVS:
32398 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
32399 (gst_ogg_demux_activate_chain):
32400 Extra debug output when activating/deactivating chains.
32401 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
32402 (is_demuxer_element), (try_to_link_1), (remove_element_chain),
32404 Remove a queue from our list when it becomes unlinked.
32405 Don't add queues to elements in class 'Demux' if they
32406 can only produce one pad
32408 2005-12-18 10:54:33 +0000 Julien Moutte <julien@moutte.net>
32410 gst-libs/gst/video/gstvideosink.c: Add a debug category.
32411 Original commit message from CVS:
32412 2005-12-18 Julien MOUTTE <julien@moutte.net>
32413 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
32414 (gst_video_sink_get_type): Add a debug category.
32416 2005-12-18 00:56:07 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
32418 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
32419 Original commit message from CVS:
32420 2005-12-17 Philippe Khalaf <burger@speedy.org>
32421 * gst-libs/gst/rtp/gstbasertpdepayload.c:
32422 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
32423 Handle downstream newsegment by sending our own newsegment before the
32424 next buffer to be released. (#323900)
32426 2005-12-18 00:41:10 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
32428 gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
32429 Original commit message from CVS:
32430 2005-12-17 Philippe Khalaf <burger@speedy.org>
32431 * gst-libs/gst/rtp/gstbasertpdepayload.c:
32432 (gst_base_rtp_depayload_set_gst_timestamp):
32433 add queue delay to new segment as well (as opposed to just the first
32434 buffer). (bug #322347)
32436 2005-12-16 22:00:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32438 ext/libvisual/visual.c: change some char* into char[]
32439 Original commit message from CVS:
32440 * ext/libvisual/visual.c: (make_valid_name):
32441 change some char* into char[]
32442 * gst/audiotestsrc/gstaudiotestsrc.c:
32443 (gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
32444 (gst_audio_test_src_create):
32445 * gst/audiotestsrc/gstaudiotestsrc.h:
32446 prepare to handle EOS and SEGMENT_DONE
32448 2005-12-16 12:32:37 +0000 Tim-Philipp Müller <tim@centricular.net>
32450 tests/check/generic/states.c: Blacklist cdparanoia element in state test.
32451 Original commit message from CVS:
32452 * tests/check/generic/states.c: (GST_START_TEST):
32453 Blacklist cdparanoia element in state test.
32455 2005-12-16 11:25:51 +0000 Benjamin Pineau <ben.pineau@gmail.com>
32457 gst/tcp/: Add <string.h> includes for memset and FD_ZERO (fixes #323878;
32458 Original commit message from CVS:
32459 * gst/tcp/gsttcp.c:
32460 * gst/tcp/gsttcpclientsink.c:
32461 * gst/tcp/gsttcpserversink.c:
32462 * gst/tcp/gsttcpserversrc.c:
32463 Add <string.h> includes for memset and FD_ZERO (fixes #323878;
32464 patch by: Benjamin Pineau).
32466 2005-12-15 14:43:38 +0000 Michael Smith <msmith@xiph.org>
32468 gst/videorate/gstvideorate.c: Fix timestamping for videorate when the first buffer it sees has a non-zero timestamp. ...
32469 Original commit message from CVS:
32470 * gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
32471 (gst_video_rate_chain):
32472 Fix timestamping for videorate when the first buffer it sees has a
32473 non-zero timestamp. Fix some misleading debug output.
32475 2005-12-15 10:30:14 +0000 Michael Smith <msmith@xiph.org>
32477 gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
32478 Original commit message from CVS:
32479 * gst/audioresample/gstaudioresample.c:
32480 Don't leak all input buffers to audioresample.
32482 2005-12-15 10:15:10 +0000 Tim-Philipp Müller <tim@centricular.net>
32484 ext/pango/gsttextoverlay.c: Don't operate on empty text buffers. Strip newlines and tabs only from the end of the tex...
32485 Original commit message from CVS:
32486 * ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
32487 Don't operate on empty text buffers. Strip newlines and
32488 tabs only from the end of the text, but leave them intact
32489 in the middle. Fix typo in gtk-doc description.
32491 2005-12-15 09:48:19 +0000 Tim-Philipp Müller <tim@centricular.net>
32493 gst/playback/: Make sure the video frame buffer we return to apps via the "frame" property always has caps set on it....
32494 Original commit message from CVS:
32495 * gst/playback/gstplaybasebin.c:
32496 * gst/playback/gstplaybin.c: (handoff):
32497 Make sure the video frame buffer we return to apps via the
32498 "frame" property always has caps set on it. Modify
32499 _gst_gvalue_set_object() macro to handle NULL objects
32502 2005-12-14 20:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32504 gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
32505 Original commit message from CVS:
32506 * gst/audiotestsrc/gstaudiotestsrc.c:
32507 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
32508 (gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
32509 (gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
32510 (gst_audio_test_src_create):
32511 * gst/audiotestsrc/gstaudiotestsrc.h:
32512 Adjust to some recent api changes and add wtays new cool seeking
32515 2005-12-14 17:58:48 +0000 Tim-Philipp Müller <tim@centricular.net>
32517 ext/alsa/: Helper functions to add device probing via the GstPropertyProbe interface to a class.
32518 Original commit message from CVS:
32519 * ext/alsa/Makefile.am:
32520 * ext/alsa/gstalsadeviceprobe.c:
32521 * ext/alsa/gstalsadeviceprobe.h:
32522 Helper functions to add device probing via the GstPropertyProbe
32523 interface to a class.
32524 * ext/alsa/gstalsamixer.h:
32525 Comment out GST_ALSA_MIXER, it returns a struct that's not
32527 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
32528 Add some debug info.
32529 * ext/alsa/gstalsamixerelement.c:
32530 (gst_alsa_mixer_element_interface_supported),
32531 (gst_implements_interface_init),
32532 (gst_alsa_mixer_element_init_interfaces),
32533 (gst_alsa_mixer_element_class_init),
32534 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
32535 (gst_alsa_mixer_element_set_property),
32536 (gst_alsa_mixer_element_get_property),
32537 (gst_alsa_mixer_element_change_state):
32538 * ext/alsa/gstalsamixerelement.h:
32539 Add 'device' and 'device-name' properties. Add GstPropertyProbe
32540 for device handling (gnome-volume-control will need that).
32542 2005-12-12 20:31:24 +0000 Christian Schaller <uraeus@gnome.org>
32546 * gst-plugins-base.spec.in:
32547 updates to activate cdparanoia plugin
32548 Original commit message from CVS:
32549 updates to activate cdparanoia plugin
32551 2005-12-12 19:13:09 +0000 Michael Smith <msmith@xiph.org>
32553 ext/ogg/gstoggdemux.c: Use the correct function to free list of typefind factories.
32554 Original commit message from CVS:
32555 * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
32556 Use the correct function to free list of typefind factories.
32558 2005-12-12 15:09:55 +0000 Wim Taymans <wim.taymans@gmail.com>
32560 gst/videotestsrc/gstvideotestsrc.*: Implement seeking in videotestsrc.
32561 Original commit message from CVS:
32562 * gst/videotestsrc/gstvideotestsrc.c:
32563 (gst_video_test_src_class_init), (gst_video_test_src_init),
32564 (gst_video_test_src_parse_caps), (gst_video_test_src_query),
32565 (gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
32566 (gst_video_test_src_create):
32567 * gst/videotestsrc/gstvideotestsrc.h:
32568 Implement seeking in videotestsrc.
32571 2005-12-12 15:06:46 +0000 Wim Taymans <wim.taymans@gmail.com>
32573 ext/cdparanoia/: Partially ported cdparanoia now that basesrc can support a plugin like this..
32574 Original commit message from CVS:
32575 * ext/cdparanoia/Makefile.am:
32576 * ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
32577 (gst_paranoia_endian_get_type), (_do_init),
32578 (cdparanoia_class_init), (cdparanoia_init),
32579 (cdparanoia_set_property), (cdparanoia_get_property),
32580 (cdparanoia_do_seek), (cdparanoia_is_seekable),
32581 (cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
32582 (cdparanoia_convert), (cdparanoia_get_query_types),
32583 (cdparanoia_query), (cdparanoia_set_index),
32584 (cdparanoia_uri_set_uri):
32585 * ext/cdparanoia/gstcdparanoia.h:
32586 Partially ported cdparanoia now that basesrc can support a
32589 2005-12-12 13:03:50 +0000 Wim Taymans <wim.taymans@gmail.com>
32591 tests/examples/seek/scrubby.c: Set higher priority for bus events so they don't get reordered with gtk gui events.
32592 Original commit message from CVS:
32593 * tests/examples/seek/scrubby.c: (main):
32594 Set higher priority for bus events so they don't get reordered with
32596 * tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
32597 (flush_toggle_cb), (main):
32598 Added checkbox do disable flushing seeks.
32599 Disable scrubbing when doing non flushing seeks.
32601 2005-12-12 09:52:37 +0000 Tim-Philipp Müller <tim@centricular.net>
32603 gst/subparse/gstsubparse.c: Implement some sort of event handling that doesn't rely on g_return_if_fail; make sure we...
32604 Original commit message from CVS:
32605 * gst/subparse/gstsubparse.c: (gst_sub_parse_init),
32606 (gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
32607 (parser_state_init), (handle_buffer), (gst_sub_parse_chain),
32608 (gst_sub_parse_sink_event), (gst_sub_parse_change_state):
32609 Implement some sort of event handling that doesn't rely on
32610 g_return_if_fail; make sure we always push the last chunk of an
32611 .srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
32612 state change function; remove some old cruft. Seeking is still
32613 rather unlikely to work though.
32614 * tools/.cvsignore:
32617 2005-12-11 22:26:08 +0000 Julien Moutte <julien@moutte.net>
32619 sys/xvimage/xvimagesink.c: Fixed a leak of the current image reference when cleaning up.
32620 Original commit message from CVS:
32621 2005-12-11 Julien MOUTTE <julien@moutte.net>
32622 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
32623 Fixed a leak of the current image reference when cleaning up.
32624 Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
32626 2005-12-09 10:23:42 +0000 Michael Smith <msmith@xiph.org>
32628 tools/: Remove gst-launch-ext. It doesn't work, and is no longer particularly useful.
32629 Original commit message from CVS:
32630 * tools/Makefile.am:
32631 * tools/gst-launch-ext-m.m:
32632 Remove gst-launch-ext. It doesn't work, and is no longer
32633 particularly useful.
32635 2005-12-08 18:53:57 +0000 Luca Ognibene <luogni@tin.it>
32637 ext/ogg/gstogmparse.c: don't pass random values to ogmparse convert function.
32638 Original commit message from CVS:
32639 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
32640 don't pass random values to ogmparse convert function.
32641 Make seeking possible in the exile1.ogm file.
32643 2005-12-07 18:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
32645 gst/playback/: Work around refcount problem with g_value_set_object() that occur if the core has been compiled agains...
32646 Original commit message from CVS:
32647 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
32648 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
32649 Work around refcount problem with g_value_set_object() that occur
32650 if the core has been compiled against GLib-2.6 (g_value_set_object()
32651 will only g_object_ref() the element, but the caller will
32652 gst_object_unref() it and bad things will happen due to the way
32653 GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
32654 totem for people on FC4 using Thomas's 0.10 RPMs.
32656 2005-12-07 11:34:37 +0000 Edward Hervey <bilboed@bilboed.com>
32658 Time to welcome ogm to 0.10 :)
32659 Original commit message from CVS:
32660 Time to welcome ogm to 0.10 :)
32661 * ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
32662 (gst_ogg_pad_typefind):
32663 Oggdemux can now properly typefind elements with dynamic pads.
32664 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
32665 Properly set caps on src pad, and set caps on outgoing buffers.
32667 2005-12-06 19:42:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32670 * ext/alsa/gstalsamixer.h:
32671 * ext/alsa/gstalsamixerelement.h:
32672 * ext/alsa/gstalsamixeroptions.h:
32673 * ext/alsa/gstalsamixertrack.h:
32674 * ext/alsa/gstalsasink.c:
32675 * ext/alsa/gstalsasink.h:
32676 * ext/alsa/gstalsasrc.c:
32677 * ext/alsa/gstalsasrc.h:
32678 * ext/cdparanoia/gstcdparanoia.h:
32679 * ext/gnomevfs/gstgnomevfsuri.h:
32680 * ext/ogg/gstoggdemux.c:
32681 * ext/ogg/gstoggmux.c:
32682 * ext/pango/gsttextoverlay.h:
32683 * ext/theora/theoradec.c:
32684 * ext/theora/theoraenc.c:
32685 * ext/vorbis/vorbisdec.h:
32686 * ext/vorbis/vorbisenc.c:
32687 * ext/vorbis/vorbisenc.h:
32688 * ext/vorbis/vorbisparse.h:
32689 * gst-libs/gst/audio/gstaudioclock.h:
32690 * gst-libs/gst/audio/gstaudiosink.c:
32691 * gst-libs/gst/audio/gstaudiosink.h:
32692 * gst-libs/gst/audio/gstaudiosrc.c:
32693 * gst-libs/gst/audio/gstaudiosrc.h:
32694 * gst-libs/gst/audio/gstbaseaudiosink.c:
32695 * gst-libs/gst/audio/gstbaseaudiosink.h:
32696 * gst-libs/gst/audio/gstbaseaudiosrc.c:
32697 * gst-libs/gst/audio/gstbaseaudiosrc.h:
32698 * gst-libs/gst/audio/gstringbuffer.h:
32699 * gst-libs/gst/audio/multichannel.h:
32700 * gst-libs/gst/floatcast/floatcast.h:
32701 * gst-libs/gst/interfaces/colorbalance.c:
32702 * gst-libs/gst/interfaces/colorbalance.h:
32703 * gst-libs/gst/interfaces/colorbalancechannel.h:
32704 * gst-libs/gst/interfaces/mixer.h:
32705 * gst-libs/gst/interfaces/mixeroptions.h:
32706 * gst-libs/gst/interfaces/mixertrack.h:
32707 * gst-libs/gst/interfaces/navigation.h:
32708 * gst-libs/gst/interfaces/propertyprobe.h:
32709 * gst-libs/gst/interfaces/tuner.h:
32710 * gst-libs/gst/interfaces/tunerchannel.h:
32711 * gst-libs/gst/interfaces/tunernorm.h:
32712 * gst-libs/gst/interfaces/xoverlay.h:
32713 * gst-libs/gst/netbuffer/gstnetbuffer.h:
32714 * gst-libs/gst/riff/riff-ids.h:
32715 * gst-libs/gst/riff/riff-media.h:
32716 * gst-libs/gst/riff/riff-read.h:
32717 * gst-libs/gst/rtp/gstbasertpdepayload.h:
32718 * gst-libs/gst/rtp/gstbasertppayload.c:
32719 * gst-libs/gst/rtp/gstbasertppayload.h:
32720 * gst-libs/gst/rtp/gstrtpbuffer.c:
32721 * gst-libs/gst/rtp/gstrtpbuffer.h:
32722 * gst-libs/gst/tag/gsttageditingprivate.h:
32723 * gst-libs/gst/tag/gstvorbistag.c:
32724 * gst-libs/gst/tag/tag.h:
32725 * gst-libs/gst/video/video.h:
32726 * gst/adder/gstadder.c:
32727 * gst/adder/gstadder.h:
32728 * gst/audioconvert/audioconvert.c:
32729 * gst/audioconvert/audioconvert.h:
32730 * gst/audioconvert/gstaudioconvert.c:
32731 * gst/audioconvert/gstchannelmix.c:
32732 * gst/audioconvert/gstchannelmix.h:
32733 * gst/audiorate/gstaudiorate.c:
32734 * gst/audioresample/buffer.h:
32735 * gst/audioresample/functable.h:
32736 * gst/audioresample/gstaudioresample.c:
32737 * gst/audioresample/resample.h:
32738 * gst/ffmpegcolorspace/avcodec.h:
32739 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
32740 * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
32741 * gst/ffmpegcolorspace/imgconvert.c:
32742 * gst/ffmpegcolorspace/imgconvert_template.h:
32743 * gst/playback/gstdecodebin.c:
32744 * gst/playback/gstplaybasebin.h:
32745 * gst/playback/gstplaybin.c:
32746 * gst/playback/gststreaminfo.h:
32747 * gst/tcp/gstfdset.c:
32748 * gst/tcp/gstfdset.h:
32749 * gst/tcp/gstmultifdsink.c:
32750 * gst/tcp/gstmultifdsink.h:
32751 * gst/tcp/gsttcp.h:
32752 * gst/tcp/gsttcpclientsrc.c:
32753 * gst/tcp/gsttcpclientsrc.h:
32754 * gst/tcp/gsttcpplugin.h:
32755 * gst/tcp/gsttcpserversink.c:
32756 * gst/tcp/gsttcpserversrc.c:
32757 * gst/typefind/gsttypefindfunctions.c:
32758 * gst/videorate/gstvideorate.c:
32759 * gst/videotestsrc/gstvideotestsrc.h:
32760 * gst/videotestsrc/videotestsrc.h:
32761 * sys/v4l/gstv4lcolorbalance.h:
32762 * sys/v4l/gstv4ltuner.h:
32763 * sys/v4l/gstv4lxoverlay.h:
32764 * sys/v4l/v4l_calls.h:
32765 * sys/v4l/videodev_mjpeg.h:
32766 * tests/check/elements/audioconvert.c:
32767 * tests/check/elements/audioresample.c:
32768 * tests/check/elements/audiotestsrc.c:
32769 * tests/check/elements/videotestsrc.c:
32770 * tests/check/elements/volume.c:
32771 * tests/examples/seek/scrubby.c:
32772 * tests/examples/seek/seek.c:
32774 Original commit message from CVS:
32777 2005-12-06 19:28:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32779 * docs/libs/tmpl/gstaudio.sgml:
32780 * docs/libs/tmpl/gstcolorbalance.sgml:
32781 * docs/libs/tmpl/gstgconf.sgml:
32782 * docs/libs/tmpl/gstmixer.sgml:
32783 * docs/libs/tmpl/gstringbuffer.sgml:
32784 * docs/libs/tmpl/gsttuner.sgml:
32785 * docs/libs/tmpl/gstxoverlay.sgml:
32786 put back stability level
32787 Original commit message from CVS:
32788 put back stability level
32790 2005-12-05 18:11:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32794 Original commit message from CVS:
32797 === release 0.10.0 ===
32799 2005-12-05 18:02:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32805 * docs/libs/tmpl/gstcolorbalance.sgml:
32806 * docs/plugins/inspect/plugin-adder.xml:
32807 * docs/plugins/inspect/plugin-alsa.xml:
32808 * docs/plugins/inspect/plugin-audioconvert.xml:
32809 * docs/plugins/inspect/plugin-audiorate.xml:
32810 * docs/plugins/inspect/plugin-audioresample.xml:
32811 * docs/plugins/inspect/plugin-audiotestsrc.xml:
32812 * docs/plugins/inspect/plugin-decodebin.xml:
32813 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
32814 * docs/plugins/inspect/plugin-gnomevfs.xml:
32815 * docs/plugins/inspect/plugin-libvisual.xml:
32816 * docs/plugins/inspect/plugin-ogg.xml:
32817 * docs/plugins/inspect/plugin-pango.xml:
32818 * docs/plugins/inspect/plugin-playbin.xml:
32819 * docs/plugins/inspect/plugin-subparse.xml:
32820 * docs/plugins/inspect/plugin-tcp.xml:
32821 * docs/plugins/inspect/plugin-theora.xml:
32822 * docs/plugins/inspect/plugin-typefindfunctions.xml:
32823 * docs/plugins/inspect/plugin-video4linux.xml:
32824 * docs/plugins/inspect/plugin-videorate.xml:
32825 * docs/plugins/inspect/plugin-videoscale.xml:
32826 * docs/plugins/inspect/plugin-videotestsrc.xml:
32827 * docs/plugins/inspect/plugin-volume.xml:
32828 * docs/plugins/inspect/plugin-vorbis.xml:
32829 * docs/plugins/inspect/plugin-ximagesink.xml:
32830 * docs/plugins/inspect/plugin-xvimagesink.xml:
32832 Original commit message from CVS: