1 === release 0.10.24 ===
3 2009-08-05 Jan Schmidt <jan.schmidt@sun.com>
6 releasing 0.10.24, "Counting the days"
8 2009-08-05 00:38:40 +0100 Jan Schmidt <thaytan@noraisin.net>
43 2009-08-01 17:26:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
45 * gst/typefind/gsttypefindfunctions.c:
46 * tests/check/gst/typefindfunctions.c:
47 typefinding: fix detection of fLaC id packet in broken flac-in-ogg
48 There are flac-in-ogg files without the usual flac packet framing
49 and these files just have a 4-byte fLaC ID packet as first packet.
50 We need to recognise the type just from these four bytes if we
51 want oggdemux to recognise these streams correctly.
53 2009-07-30 14:40:50 +0100 Jan Schmidt <thaytan@noraisin.net>
91 2009-07-29 14:15:53 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
93 * gst-libs/gst/audio/gstaudiofilter.c:
94 audiofilter: Don't assert on slightly different caps
95 Plugins should not assert on incompatible caps, caps negotiation will
98 2009-07-30 13:42:21 +0300 Stefan Kost <ensonic@users.sf.net>
100 * gst/adder/gstadder.c:
101 adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146.
103 2009-07-30 09:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
106 configure: bump Gtk+ requirement of GUI examples from 2.12 to 2.14
107 The gio mount example needs GtkMountOperation, which is new in 2.14.
109 2009-07-27 10:29:27 +0100 Balachandran C <balachandran_c@rediffmail.com>
111 * ext/alsa/gstalsasrc.c:
112 alsasrc: set alsasrc->handle back to NULL when closing device
113 Fixes crashes in gst_alsa_find_device_name() when probing or
114 reading the device-name property (e.g. when doing a dot-file
115 dump). Fixes #589797.
117 2009-07-24 19:26:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
119 * gst/playback/gststreamselector.c:
120 playbin: rename GType of stream selector pad to avoid clash with input-selector from -bad
121 Rename the GType of the pads of playbin's internal stream selector
122 element so they don't use the same type name as input-selector's
125 2009-07-24 13:39:55 +0100 Jan Schmidt <thaytan@noraisin.net>
158 0.10.23.4 pre-release
160 2009-07-24 13:46:15 +0100 Jan Schmidt <thaytan@noraisin.net>
162 * tests/examples/v4l/.gitignore:
163 ignores: Ignore v4l probing example binary
165 2009-07-24 09:35:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
167 * gst/typefind/gsttypefindfunctions.c:
168 typefind: recognise Kate spu subtitles as well
169 Recognise spu-subtitles, SUB and K-SPU as valid categories for
170 Kate subtitles as well.
172 2009-07-24 00:42:16 +0300 Stefan Kost <ensonic@users.sf.net>
175 Automatic update of common submodule
176 From fedaaee to 94f95e3
178 2009-07-22 14:21:43 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
180 * gst-plugins-base.spec.in:
181 Update spec file with latest changes
183 2009-07-20 17:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
216 * win32/common/_stdint.h:
217 * win32/common/audio-enumtypes.c:
218 * win32/common/config.h:
219 * win32/common/gstrtsp-enumtypes.c:
220 * win32/common/interfaces-enumtypes.c:
221 * win32/common/video-enumtypes.c:
222 0.10.23.3 pre-release
224 2009-07-20 12:51:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
226 * gst/audiotestsrc/gstaudiotestsrc.c:
227 audiotestsrc: call send_event directly
228 We can't call gst_element_send_event() from a streaming thread as it gets the
229 state lock. Instead call the send_event method directly until we have a nice API
233 2009-07-03 04:42:24 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
235 * gst-libs/gst/audio/gstaudiosink.c:
236 audiosink: Add stream-status messages
239 2009-07-03 04:41:05 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
241 * gst-libs/gst/audio/gstaudiosrc.c:
242 audiosrc: Add stream-status messages
245 2009-07-20 10:53:11 +0200 Edward Hervey <bilboed@bilboed.com>
247 * gst/adder/gstadder.c:
248 gstadder: Don't forget to free pending events on flush/dispose.
251 2009-07-12 10:08:12 +0200 Edward Hervey <bilboed@bilboed.com>
253 * tests/check/elements/adder.c:
254 tests/adder: Add stream consistency checking. Fixes #588748
256 2009-07-12 10:07:34 +0200 Edward Hervey <bilboed@bilboed.com>
258 * gst/audiotestsrc/gstaudiotestsrc.c:
259 audiotestsrc: Make sure tags are properly serialized. Fixes #588746
260 We do this by letting the basesrc base class handle the tags.
262 2009-07-13 09:28:54 +0200 Edward Hervey <bilboed@bilboed.com>
264 * gst/adder/gstadder.c:
265 * gst/adder/gstadder.h:
266 adder: Collect incoming tag events and send them after newsegment. Fixes #588747
268 2009-07-16 09:32:46 +0200 Edward Hervey <bilboed@bilboed.com>
270 * ext/vorbis/vorbisdec.c:
271 vorbisdec: Check for empty tag strings. Fixes #588724
273 2009-07-14 17:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
275 * gst/playback/gstqueue2.c:
276 queue2: fix leak and improve buffering
277 Keep track of the max requested position and compare this to the write position
278 in the temp file to get the current amount of buffered data.
279 Fix memleak of all incomming buffers.
282 2009-07-15 17:40:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
284 * gst/playback/Makefile.am:
285 * gst/playback/gstinputselector.c:
286 * gst/playback/gstinputselector.h:
287 * gst/playback/gstplay-marshal.list:
288 * gst/playback/gstplaybin2.c:
289 playbin2: use private copy of input-selector
290 We shouldn't really depend on elements from -bad for stream
291 selection in playbin2, so use a private copy of input-selector
292 until the selector plugin is ready to be moved to -base or -good.
295 2009-07-15 17:26:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
297 * gst/playback/gstinputselector.c:
298 * gst/playback/gstinputselector.h:
299 playback: add private copy of the input-selector from gst-plugins-bad
300 Not hooked up yet though. See #586356.
302 2009-07-14 19:00:36 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
304 * tests/examples/v4l/Makefile.am:
305 examples: fix v4l probe example build
308 2009-07-14 19:00:10 +0100 Jan Schmidt <thaytan@noraisin.net>
342 0.10.23.2 pre-release
344 2009-07-14 16:24:10 +0100 Jan Schmidt <thaytan@noraisin.net>
348 Add Turkish translations
350 2009-07-14 15:31:13 +0100 Jan Schmidt <thaytan@noraisin.net>
352 * tests/check/elements/adder.c:
353 adder: One more attempt to fix the adder test
354 Give up and discard and recreate the alsasrc after checking it can
355 be opened, due to some strange crash inside alsa when we don't.
357 2009-07-14 15:06:41 +0100 Jan Schmidt <thaytan@noraisin.net>
359 * tests/check/elements/adder.c:
360 adder: Perform get_state() in the unit test
361 Wait for the alsasrc to return to NULL after setting it to PAUSED for
362 testing, otherwise it leads to segfaults later on.
364 2009-07-14 14:39:32 +0100 Jan Schmidt <thaytan@noraisin.net>
366 * tests/check/elements/adder.c:
367 adder: Don't fail when alsasrc is unavailable
368 Make the liveadder test succeed silently when it can't be completed
369 either because alsasrc is unavailable, or because the device is
372 2009-07-13 22:51:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
374 * gst-libs/gst/pbutils/descriptions.c:
375 * gst/typefind/gsttypefindfunctions.c:
376 typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest
377 Differentiate subtitle streams and lyrics/cracktastic/complex streams via
378 the category string in the headers. This seems like a useful distinction
379 to make, and also seems more future-proof. See #525743.
381 2009-02-21 13:18:10 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
383 * ext/ogg/gstoggmux.c:
384 oggmux: add Kate caps to the list of accepted types
387 2009-07-13 21:56:46 +0300 Stefan Kost <ensonic@users.sf.net>
389 * gst/playback/gsturidecodebin.c:
390 uridecodebin: treat uri-schemas incasesensitive
391 Treat uri-schemas incasesensitive. This is mandated in rfc2396 section 3.1.
392 Fixes not showing buffering messages e.g. for HTTP://...
394 2009-07-13 21:54:47 +0300 Stefan Kost <ensonic@users.sf.net>
396 * gst-libs/gst/interfaces/navigation.c:
397 navigation: simplify docs
398 Make short-desc short - its used in the toc. Strip uneeded markup.
400 2009-07-13 18:31:15 +0100 Jan Schmidt <thaytan@noraisin.net>
402 * win32/common/libgstnetbuffer.def:
403 * win32/common/libgstvideo.def:
405 Remove methods from video base classes that have moved to -bad.
406 Add gst_netaddress_to_string
408 2009-07-13 17:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
410 * tests/examples/gio/.gitignore:
411 ignores: ignore the giosrc-mounting example binary
413 2009-07-13 17:54:40 +0100 Jan Schmidt <thaytan@noraisin.net>
415 * gst-libs/gst/interfaces/navigation.c:
416 navigation: Add some partial documentation
417 Add a general documentation blurb for the GstNavigation functionality.
418 Still lacks some example code and detail on how to implement it.
420 2009-07-13 17:52:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
422 * gst-libs/gst/pbutils/descriptions.c:
423 pbutils: add description for Siren codec and make two descriptions non-translatable
425 2009-07-13 12:23:20 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
428 Automatic update of common submodule
429 From 5845b63 to fedaaee
431 2009-07-13 18:21:49 +0200 Elliott Sales de Andrade <quantum.analyst at gmail.com>
433 * gst-libs/gst/riff/riff-ids.h:
434 * gst-libs/gst/riff/riff-media.c:
435 riff: add siren to the RIFF parser
436 Add siren7 caps to the RIFF parser.
438 2009-07-13 14:55:59 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
441 * tests/examples/Makefile.am:
442 * tests/examples/v4l/Makefile.am:
443 * tests/examples/v4l/probe.c:
444 v4lsrc: add a simple test case for device probing
446 2009-07-03 11:38:01 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
449 * sys/v4l/Makefile.am:
450 * sys/v4l/gstv4lelement.c:
451 v4lsrc: optional support for device probing with gudev
452 Enumerate v4l devices using gudev if available.
455 2009-07-10 23:24:36 +0100 Stefan Kost <ensonic@users.sf.net>
457 * gst/adder/gstadder.c:
458 adder: add since tags to docs
460 2009-07-10 21:29:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
462 * tests/examples/seek/seek.c:
463 seek: don't automatically start pipeline in DB
464 Keep the pipeline paused when we detect download buffering. The user has to
465 manually start the pipeline for now because we can't estimate when the buffering
466 will finish or when we have underrun.
468 2009-07-10 21:01:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
470 * gst/playback/gstqueue2.c:
471 queue2: flush differently, avoiding deadlocks
472 Don't flush the file by closing and opening it but instead use g_freopen. This
473 avoids a deadlock in shutdown because we emit the temp-location property change
474 with the wrong lock held.
476 2009-07-10 20:25:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
478 * tests/examples/seek/seek.c:
479 seek: add a checkbox for progressive download
481 2009-07-10 20:24:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
483 * gst/playback/gsturidecodebin.c:
484 uridecodebin: Fix template construction
485 Fix the construction of the temporary filename construction as the application
486 name can be NULL and we don't want a separator between the prgname and the
489 2009-07-10 20:04:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
491 * gst/playback/gstplay-enum.c:
492 * gst/playback/gstplay-enum.h:
493 * gst/playback/gstplaybin2.c:
494 playbin2: add support for progressive download
495 Add a new playbin2 flag (initially disabled) to enable progressive download
496 buffering in uridecodebin.
498 2009-07-10 19:59:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
500 * gst/playback/gsturidecodebin.c:
501 uridecodebin: add download property
502 Add a download property that will attempt to configure queue2 into progressive
504 Make sure we only enable download buffering for quicktime and flv formats.
506 2009-07-10 19:49:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
508 * gst/playback/gstqueue2.c:
509 queue2: add temp-template property
510 Add a new temp-template property so that queue2 can securely allocate a
511 temporary filename. Deprecate the temp-location property for setting the
512 location but still use it to notify the allocated temp file.
514 2009-07-10 20:06:28 +0100 Stefan Kost <ensonic@users.sf.net>
516 * gst/adder/gstadder.c:
517 * gst/adder/gstadder.h:
518 adder: add a caps-property to avoid to need to plug a capsfilter afterwards
519 Adder can only handle one common format accross the pads. Thus one needed to add
520 a capsfilter afterwards and manage the caps. Now one can simply set the caps on
523 2009-07-10 18:59:05 +0100 Stefan Kost <ensonic@users.sf.net>
525 * tests/check/elements/adder.c:
526 adder: skip live-seek text if we have no audiosrc, add new test
527 The seek-test needs a real audiosrc. Also add a test that checks that adder is
528 reusable. Finaly handle warnings as warnings to fix a assertion.
530 2009-07-10 19:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
532 * ext/gio/gstgiosink.c:
533 gio: Also post a "not-mounted" message from giosink
535 2009-07-10 17:15:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
537 * tests/examples/gio/giosrc-mounting.c:
538 gio: Remove workaround for playbin2 bug in the sample application
539 The playbin2 bug was #588078.
541 2009-07-10 17:08:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
543 * gst/playback/gstplaybin2.c:
544 playbin2: Make it possible for READY->PAUSED to succeed after it failed the first time
545 If READY->PAUSED failed in the source element we would've swapped
546 the current and next group already. To allow READY->PAUSED to succeed
547 after the first failure we have to swap the current and next group
548 back again. This also ensure that we're again in the same state
549 as before the failed state change and not at the next group.
550 This was especially a problem for playbin2 pipelines that use the
551 new mounting support in giosrc as the source would fail for READY->PAUSED
552 the first time, the application mounts the location and then tries
553 to go READY->PAUSED again (and this time it would succeed).
556 2009-07-10 11:42:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
559 * tests/examples/Makefile.am:
560 * tests/examples/gio/Makefile.am:
561 * tests/examples/gio/giosrc-mounting.c:
562 gio: Add example application that shows how to handle the "not-mounted" message
564 2009-07-10 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
567 gio: Remove the experimental status from the GIO plugin
570 2009-07-10 11:24:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
572 * ext/gio/gstgiosink.c:
573 * ext/gio/gstgiosrc.c:
574 gio: Add documentation for the new "not-mounted" and "file-exists" messages
576 2009-07-09 13:45:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
578 * ext/gio/gstgiobasesrc.c:
579 gio: Make sure that we have the correct stream position when starting
581 2009-07-08 17:24:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
583 * ext/gio/gstgiobasesink.c:
584 gio: Make sure to flush the output stream if it shouldn't be closed
585 Otherwise there might still be unwritten data after the element
588 2009-07-08 17:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
590 * ext/gio/gstgiobasesink.c:
591 * ext/gio/gstgiobasesink.h:
592 * ext/gio/gstgiobasesrc.c:
593 * ext/gio/gstgiobasesrc.h:
594 * ext/gio/gstgiosink.c:
595 * ext/gio/gstgiosrc.c:
596 gio: Don't close the GIO streams for the giostream{src,sink} elements
597 This makes it possible to do something useful with the streams
598 after the element has stopped. Fixes bug #587896.
600 2009-07-08 17:19:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
602 * tests/check/pipelines/gio.c:
603 gio: Try to reuse the pipeline with the same stream objects
605 2009-07-08 17:02:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
607 * ext/gio/gstgiobasesink.c:
608 * ext/gio/gstgiobasesrc.c:
609 gio: Improve the error message if a stream is already closed before usage
611 2009-07-08 16:55:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
613 * ext/gio/gstgiosink.c:
614 gio: Post a custom file-exists message on the bus if the file already exists
615 An application can handle this message, remove the file in question
616 and restart the pipeline again without showing an error.
617 This fixes bug #529300.
619 2009-07-08 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
621 * ext/gio/gstgiosrc.c:
622 gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted
624 2009-07-08 16:50:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
626 * ext/gio/gstgiosink.c:
627 gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink
629 2009-07-08 15:52:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
631 * ext/gio/gstgiosrc.c:
632 gio: Post a custom "not-mounted" message on the bus
633 This allows applications to mount the GFile if possible and restart
634 the pipeline instead of simply giving an error.
636 2009-07-08 15:08:32 +0200 Philip Jägenstedt <philipj@opera.com>
638 * gst/audioconvert/gstchannelmix.c:
639 audioconvert: Fix compilation when debugging is disabled
642 2009-07-07 20:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
644 * ext/gio/gstgiobasesink.c:
645 * ext/gio/gstgiobasesink.h:
646 * ext/gio/gstgiobasesrc.h:
647 * ext/gio/gstgiosink.c:
648 * ext/gio/gstgiosink.h:
649 * ext/gio/gstgiostreamsink.c:
650 * ext/gio/gstgiostreamsink.h:
651 gio: Add vfunc for requesting the stream for the sinks too
653 2009-07-07 20:21:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
655 * ext/gio/gstgiobasesink.c:
656 * ext/gio/gstgiobasesink.h:
657 * ext/gio/gstgiobasesrc.c:
658 * ext/gio/gstgiosink.c:
659 * ext/gio/gstgiosrc.c:
660 * ext/gio/gstgiostreamsink.c:
661 * ext/gio/gstgiostreamsrc.c:
662 gio: Some more random cleanup
664 2009-07-07 20:20:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
667 * ext/gio/gstgiobasesink.c:
668 * ext/gio/gstgiobasesrc.c:
669 * ext/gio/gstgiobasesrc.h:
670 * ext/gio/gstgiosink.c:
671 * ext/gio/gstgiosrc.c:
672 * ext/gio/gstgiosrc.h:
673 * ext/gio/gstgiostreamsink.c:
674 * ext/gio/gstgiostreamsrc.c:
675 * ext/gio/gstgiostreamsrc.h:
676 gio: Update my mail address and copyright
678 2009-07-07 20:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
680 * ext/gio/gstgiobasesrc.c:
681 * ext/gio/gstgiobasesrc.h:
682 * ext/gio/gstgiosrc.c:
683 * ext/gio/gstgiostreamsrc.c:
684 * ext/gio/gstgiostreamsrc.h:
685 gio: General clean up and simplification
686 The GInputStreams are now requested by a vfunc from
687 the subclasses instead of relying that the subclass
688 sets it until it's needed.
689 This might also fix bug #587896.
691 2009-07-06 22:31:12 +0100 Stefan Kost <ensonic@users.sf.net>
693 * gst/adder/gstadder.c:
694 adder: keep sending newsegments after seeking
695 Adder sends with timestamps from 0 upwards. After seeking we need to send
696 new-segments to get correct positions-queries.
698 2009-07-06 20:44:00 +0100 Stefan Kost <ensonic@users.sf.net>
700 * tests/check/elements/adder.c:
701 adder: make test more robust
702 Add audioconverts to the live-seeking test to make it negotiate.
704 2009-06-30 17:19:50 +0300 Stefan Kost <ensonic@users.sf.net>
706 * sys/xvimage/xvimagesink.c:
707 xvimagesink: use core performance log category
709 2009-07-05 21:29:40 +0200 Edward Hervey <bilboed@bilboed.com>
711 * gst/adder/gstadder.c:
712 adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped.
713 This ensures that collectpads' cookie is properly updated so that when the streaming
714 threads will restart and be checking for the flushing status of all pads there will
715 be no inconsistent state.
717 2009-07-05 18:01:38 +0200 Hans-Peter Nilsson <hp@gcc.gnu.org>
719 * ext/pango/gstclockoverlay.c:
720 pango: Call tzset() before localtime_r()
721 POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't
722 required to set the state variables that define the current timezone. Indeed,
723 glibc (at least 2.9) doesn't do this for subsequent calls. The effect is that
724 if the system timezone is changed for a running program between two calls to
725 gst_clock_overlay_render_time, it won't be noticed. For glibc, changing the
726 timezone equals /etc/localtime being modified.
729 2009-07-01 17:33:14 -0700 David Schleef <ds@schleef.org>
732 build: remove spurious schroedinger reference
734 2009-07-01 10:25:43 -0700 David Schleef <ds@schleef.org>
738 * ext/schroedinger/Makefile.am:
739 * ext/schroedinger/gstschro.c:
740 * ext/schroedinger/gstschrodec.c:
741 * ext/schroedinger/gstschroenc.c:
742 * ext/schroedinger/gstschroparse.c:
743 * ext/schroedinger/gstschroutils.c:
744 * ext/schroedinger/gstschroutils.h:
745 * gst-libs/gst/video/Makefile.am:
746 * gst-libs/gst/video/gstbasevideocodec.c:
747 * gst-libs/gst/video/gstbasevideocodec.h:
748 * gst-libs/gst/video/gstbasevideodecoder.c:
749 * gst-libs/gst/video/gstbasevideodecoder.h:
750 * gst-libs/gst/video/gstbasevideoencoder.c:
751 * gst-libs/gst/video/gstbasevideoencoder.h:
752 * gst-libs/gst/video/gstbasevideoparse.c:
753 * gst-libs/gst/video/gstbasevideoparse.h:
754 * gst-libs/gst/video/gstbasevideoutils.c:
755 * gst-libs/gst/video/gstbasevideoutils.h:
756 basevideo: send basevideo back to remedial school
757 Move basevideo classes and schroedinger plugin to -bad.
759 2009-07-01 12:54:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
761 * docs/libs/gst-plugins-base-libs-sections.txt:
762 * gst-libs/gst/netbuffer/gstnetbuffer.h:
763 netaddress: add constant for max len
765 2009-07-01 12:48:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
767 * docs/libs/gst-plugins-base-libs-sections.txt:
768 * gst-libs/gst/netbuffer/gstnetbuffer.c:
769 * gst-libs/gst/netbuffer/gstnetbuffer.h:
770 netbuffer: add gst_netaddress_to_string
771 Add function to serialize a net address to a string.
772 API: GstNetAddress::gst_netaddress_to_string()
774 2009-06-30 18:44:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
776 * gst/playback/gsturidecodebin.c:
777 uridecodebin: make fd:// uri use buffering too
778 fd:// usually operate in push mode only and are thus suitable for buffering.
780 2009-06-30 14:46:38 +0300 Stefan Kost <ensonic@users.sf.net>
782 * gst/playback/gstplaybin2.c:
783 * gst/volume/gstvolume.c:
784 volume: include "1.0=100%" in property description
786 2009-06-30 14:45:51 +0300 Stefan Kost <ensonic@users.sf.net>
788 * gst/playback/gstplaysink.c:
789 playsink: remove unused property defs
791 2009-06-29 17:11:50 +0300 Stefan Kost <ensonic@users.sf.net>
793 * gst-libs/gst/audio/multichannel.c:
794 multichannel: rewrite the new doc comment a bit
795 Its part of the audio lib.
797 2009-06-29 14:34:02 +0100 Jan Schmidt <thaytan@noraisin.net>
799 * gst/playback/gstplaysink.c:
800 playsink: Avoid a segfault when the video sink fails to start
801 Don't attempt to display the subpictures and segfault when the
802 video sink failed to start (and hence the videochain is NULL).
804 2009-06-29 15:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
806 * gst-libs/gst/audio/gstringbuffer.c:
807 * gst-libs/gst/audio/gstringbuffer.h:
808 ringbuffer: add vmethod to clear the ringbuffer
809 Add a vmethod so that subclasses can be notified when they should clear the data
812 2009-06-29 14:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
814 * gst-libs/gst/riff/riff-media.c:
815 riff-media: Fix the fourcc caps property for VC-1/WMVA
816 The caps property for carrying fourccs is 'format', not 'fourcc'
818 2009-06-29 12:20:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
820 * gst-libs/gst/rtsp/gstrtspconnection.c:
821 rtsp: include in.h for FreeBSD compat
824 2009-06-29 12:20:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
826 * win32/common/libgstapp.def:
827 defs: add defs for new appsink buffer-list method
829 2009-06-29 12:14:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
831 * gst-libs/gst/app/gstappsink.c:
832 * gst-libs/gst/app/gstappsink.h:
833 appsink: add docs and signals
834 Add docs for the new callback.
835 Add signals for the new buffer-list support.
837 2009-06-29 10:24:36 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
839 * tests/check/elements/appsink.c:
840 Added unit tests for buffer list support in appsink.
842 2009-06-17 11:12:08 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
844 * gst-libs/gst/app/gstappsink.c:
845 Added buffer list support.
847 2009-06-17 09:23:11 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
849 * gst-libs/gst/app/gstappsink.h:
850 Added buffer list support.
852 2009-06-29 09:36:27 +0200 Peter Kjellerstedt <pkj@axis.com>
854 * gst-libs/gst/sdp/gstsdpmessage.c:
855 sdp: Include winsock2.h after defining WINVER.
856 Similar to bug #587080.
858 2009-06-29 09:31:40 +0200 Peter Kjellerstedt <pkj@axis.com>
860 * gst-libs/gst/rtsp/gstrtspconnection.c:
861 rtsp: Moved a comment.
863 2009-06-27 23:23:02 +0300 Stefan Kost <ensonic@users.sf.net>
865 * gst-libs/gst/audio/audio.c:
866 * gst-libs/gst/audio/multichannel.c:
867 docs: add basic section docs for multichannel and relocate the ones for audio
868 Add section docs for multichannel, so that it has a short desc in the toc too.
869 Move the section docs in adio up, so that the follow the copyright like
872 2009-06-26 21:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
874 * sys/v4l/gstv4lelement.c:
875 * sys/v4l/gstv4lsrc.c:
876 v4l: open/close device in ready.
877 Simillar change like in v4l2src. This allows probing feature in paused, where
878 streaming is noit yet started.
880 2009-06-10 17:05:22 +0300 René Stadler <rene.stadler@nokia.com>
882 * gst/playback/gstplaysink.c:
883 playbin2: fix initial volume handling also when reusing the element
884 This is a follow-up to commit 452988, making it work correctly when the audio
887 2009-06-26 21:48:58 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
889 * gst-libs/gst/rtsp/gstrtspconnection.c:
890 Define WINVER before including any win headers
893 2009-06-27 00:50:54 +0300 René Stadler <mail@renestadler.de>
895 * gst-libs/gst/riff/riff-read.c:
896 riff: prevent crash if rounded up tag size exceeds data size
897 When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
898 and an invalid read past the buffer data follows.
900 2009-06-26 15:17:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
902 * gst-libs/gst/video/gstbasevideocodec.c:
903 basevideocodec: By default don't allow caps changes on the srcpad
904 This fixed playback of Dirac files with schrodec when upstream wants
905 a different width/height, basevideocodec accepts this and then
906 pushes buffers with new caps but content of the old caps.
907 In the best case this will just result in wrong unit size and a
908 failure in basestransform elements.
910 2009-06-26 14:11:21 +0100 Jan Schmidt <thaytan@noraisin.net>
913 autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
914 Check for more automake command variants. Use printf instead of 'echo -n'
917 2009-06-26 13:41:38 +0100 Jan Schmidt <thaytan@noraisin.net>
920 Automatic update of common submodule
921 From f810030 to 5845b63
923 2009-06-26 13:14:02 +0300 Stefan Kost <ensonic@users.sf.net>
925 * gst/playback/gstscreenshot.c:
926 screenshot: don't leak message
928 2009-06-25 12:04:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
930 * gst/typefind/gsttypefindfunctions.c:
931 typefinding: lower the h264 typefinder's probability
932 A NEARLY_CERTAIN is absolutely not warranted given the kind
933 of things it checks for. Even a LIKELY is probably not entirely
936 2009-06-24 15:13:56 +0100 Jan Schmidt <jan.schmidt@sun.com>
939 Automatic update of common submodule
940 From f3bb51b to f810030
942 2009-06-24 09:48:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
944 * gst-libs/gst/pbutils/descriptions.c:
945 pbutils: add description for multipart
946 So we get slightly nicer error messages when multipartdemux is missing.
948 2009-06-23 18:07:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
950 * gst/adder/gstadder.c:
951 adder: only unflush when we flushed before
952 Ass suggested by Stefan Kost:
953 Keep track of when the sinkpad was set to flushing and unflush the pad when an
954 upstream flushing seek failed.
956 2009-06-23 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
958 * gst/playback/gsturidecodebin.c:
959 uridecodebin: fix leak when the source fails to change state
961 2009-06-23 12:40:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
963 * gst/subparse/gstssaparse.c:
964 ssaparse: avoid leaking all buffers
966 2009-06-22 22:18:03 +0300 Stefan Kost <ensonic@users.sf.net>
968 * tests/check/elements/adder.c:
969 adder: test seek handling in adder
970 This tests seeking on an adder that has a normal and a live source connected.
971 Wheter the current behavior is the desired one needs to be discussed still
974 2009-06-22 16:17:10 +0300 Stefan Kost <ensonic@users.sf.net>
976 * sys/ximage/ximagesink.c:
977 * sys/xvimage/xvimagesink.c:
978 x(v)imagesink: pass the xwindow along to not look at the yet unset var.
979 When we call this from xwindow_new, x(v)imagesink->xwindow is not yet set.
981 2009-06-22 11:40:33 +0300 Stefan Kost <ensonic@users.sf.net>
983 * sys/ximage/ximagesink.c:
984 * sys/ximage/ximagesink.h:
985 * sys/xvimage/xvimagesink.c:
986 * sys/xvimage/xvimagesink.h:
987 x(v)imagesink: catch tags and show title in own window
988 Refactor the code that sets the window title. Catch tag-events and use title
989 metadata for the window title.
991 2009-06-21 19:42:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
993 * gst/audiotestsrc/gstaudiotestsrc.c:
994 audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
995 Also make all the function arrays constant.
997 2009-06-21 12:27:37 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
999 * gst/audiotestsrc/gstaudiotestsrc.c:
1000 * gst/audiotestsrc/gstaudiotestsrc.h:
1001 audiotestsrc: Add support for generating gaussian white noise
1002 This patch adds support for stationary white Gaussian noise.
1003 The Box-Muller algorithm is used to generate pairs of independent
1004 normally-distributed random numbers.
1007 2009-06-20 23:46:28 +0100 Jan Schmidt <thaytan@noraisin.net>
1009 * gst/ffmpegcolorspace/imgconvert.c:
1010 * gst/ffmpegcolorspace/imgconvert_template.h:
1011 ffmpegcolorspace: Fix NV12 and NV21 transformations
1012 Fix some stride problems, fix the nv12 to nv21 direct transformation,
1013 and implement a direct conversion to yuv444 to save CPU.
1015 2009-06-20 22:36:21 +0100 Jan Schmidt <thaytan@noraisin.net>
1017 * gst/videotestsrc/videotestsrc.c:
1018 videotestsrc: Fix NV12 painting for odd strides/heights
1020 2009-06-19 22:16:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1022 * ext/cdparanoia/gstcdparanoiasrc.c:
1023 cdparanoia: run-time license is LGPL now that we require cdparanoia 0.10.2
1024 cdparanoia has an LGPL v2.1 license since 0.10.1 and we now require 0.10.2.
1025 Finally fixes #531035.
1027 2009-06-19 21:25:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1029 * ext/cdparanoia/gstcdparanoiasrc.c:
1030 cdparanoia: try to guess a good cache size if it's set to -1
1031 Try to guess from the paranoia-mode setting whether playback or
1032 ripping is wanted, and use a smaller cache size if we're likely
1033 to be doing playback, to avoid a long startup delay. Since this
1034 was the value used in older cdparanoia versions, it should be
1035 fine in any case. See #586331.
1037 2009-06-19 11:27:40 +1000 Jonathan Matthew <jonathan@d14n.org>
1040 * ext/cdparanoia/gstcdparanoiasrc.c:
1041 * ext/cdparanoia/gstcdparanoiasrc.h:
1042 cdparanoia: expose cache size setting
1043 This setting was added in cdparanoia 10.2. The default value is good
1044 for audio extraction, but lower values (previous versions of cdparanoia
1045 used 150) are better for realtime playback.
1048 2009-06-19 17:43:03 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
1050 * gst-plugins-base.spec.in:
1051 Make build of schro plugin conditional
1053 2009-06-19 15:52:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1055 * docs/libs/gst-plugins-base-libs-sections.txt:
1056 * gst-libs/gst/rtp/gstbasertppayload.c:
1057 * gst-libs/gst/rtp/gstbasertppayload.h:
1058 * win32/common/libgstrtp.def:
1059 basertppayload: add support for bufferlists
1060 Based on patch from Ognyan Tonchev.
1063 2009-06-19 15:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1065 * gst-libs/gst/rtp/gstrtpbuffer.c:
1066 rtpbuffer: use new convenience functions
1067 New core convenience functions makes the list getters and setters trivial.
1068 Maybe even too trivial...
1070 2009-06-18 19:07:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1072 * win32/common/libgstrtp.def:
1073 defs: add new symbol to win32 defs file
1074 Based on patches by Ognyan Tonchev.
1077 2009-06-18 19:04:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1079 * docs/libs/gst-plugins-base-libs-sections.txt:
1080 * gst-libs/gst/rtp/gstrtpbuffer.c:
1081 rtp: cleanups, add _list_get_seq() too
1082 Clean up the docs a little.
1083 Add missing _list_get_seq method.
1084 Add new symbols to the docs
1086 2009-06-18 18:47:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1088 * gst-libs/gst/rtp/gstrtpbuffer.c:
1089 * win32/common/libgstrtp.def:
1091 Add Since tags to docs
1092 Move some code around
1095 2009-06-18 17:46:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1097 * gst-libs/gst/rtp/gstrtpbuffer.c:
1098 * gst-libs/gst/rtp/gstrtpbuffer.h:
1099 * tests/check/libs/rtp.c:
1100 rtp: add bufferlist support
1102 2009-06-18 18:03:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1104 * gst-libs/gst/rtp/gstrtpbuffer.c:
1105 rtp: pass data to macros instead of GstBuffer
1107 2009-06-18 17:42:10 +0100 Jan Schmidt <thaytan@noraisin.net>
1109 * win32/common/libgstrtsp.def:
1110 win32: Add gst_rtsp_watch_queue_data() to the exports
1111 Fix the tests by exporting the new symbol from the win32 dlls
1113 2009-06-18 18:13:22 +0300 Stefan Kost <ensonic@users.sf.net>
1115 * sys/xvimage/xvimagesink.c:
1116 xvimagesink: appname might be NULL
1117 Don't set title if appname is unknown.
1119 2009-06-18 17:58:06 +0300 Stefan Kost <ensonic@users.sf.net>
1121 * sys/xvimage/xvimagesink.c:
1122 xvimagesink: set window title from application name
1124 2009-06-09 19:14:00 +0200 Peter Kjellerstedt <pkj@axis.com>
1126 * gst-libs/gst/rtsp/gstrtspurl.c:
1127 rtsp: Made the parsing of the RTSP URL scheme more generic.
1129 2009-06-15 13:58:26 +0200 Peter Kjellerstedt <pkj@axis.com>
1131 * gst-libs/gst/rtsp/gstrtspconnection.c:
1132 * gst-libs/gst/rtsp/gstrtspconnection.h:
1133 rtsp: Added gst_rtsp_watch_queue_data().
1134 gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
1135 but allows for queuing any data block for writing (much like
1136 gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)
1137 API: gst_rtsp_watch_queue_data()
1139 2009-06-09 16:37:09 +0200 Peter Kjellerstedt <pkj@axis.com>
1141 * gst-libs/gst/rtsp/gstrtspconnection.c:
1142 rtsp: Only extract the session ID from RTSP responses.
1144 2009-06-09 19:06:57 +0200 Peter Kjellerstedt <pkj@axis.com>
1146 * gst-libs/gst/rtsp/gstrtspurl.c:
1147 rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
1149 2009-06-09 14:31:18 +0200 Peter Kjellerstedt <pkj@axis.com>
1151 * gst-libs/gst/rtsp/gstrtspconnection.c:
1152 rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
1154 2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
1156 * gst-libs/gst/rtsp/gstrtspconnection.c:
1157 rtsp: Improved base64 decoding in fill_bytes().
1158 The base64 decoding in fill_bytes() expected the size of the read data to
1159 be evenly divisible by four (which is true for the base64 encoded data
1160 itself). This did not, however, take whitespace (especially line breaks)
1161 into account and would fail the decoding if any whitespace was present.
1163 2009-06-17 14:00:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1165 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1166 audiosrc: fix get_offset
1167 When we need to jump to the most recently captured sample, jump to where the
1168 next sample will be written instead of to some old data.
1171 2009-06-17 13:18:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1173 * gst-libs/gst/audio/gstbaseaudiosink.c:
1174 audiosink: free the ringbuffer when going to NULL
1175 Unparent and free the ringbuffer when going to NULL, like we do with the
1176 audiosrc element. We can do this now because we correctly manage the time
1179 2009-06-17 13:17:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1181 * gst-libs/gst/audio/gstaudiosink.c:
1182 * gst-libs/gst/audio/gstaudiosrc.c:
1183 audio: correctly handle short read/writes
1185 2009-05-05 15:37:54 +0300 René Stadler <rene.stadler@nokia.com>
1187 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1188 baseaudiosrc: add some extra logging for buffer timestamps
1190 2009-06-17 11:22:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1192 * gst/adder/gstadder.c:
1193 adder: more seeking fixes.
1194 When a seek failed upstream, make sure the adder sinkpad is set unflushing again
1195 so that streaming can continue.
1196 We only have a pending segment when we flushed.
1197 Set the flush_stop_pending flag inside the appropriate locks and before we
1198 attempt to perform the upstream seek.
1199 Add some more comments.
1200 Use the right lock to protect the flags in flush_stop.
1203 2009-06-17 07:24:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1205 * gst/playback/gstdecodebin2.c:
1206 decodebin2: Free iterator after removing all groups
1208 2009-06-16 19:38:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1210 * gst-libs/gst/video/gstvideofilter.c:
1211 videofilter: Add a default get_unit_size function
1212 This returns the correct values for all formats that are handled by
1213 GstVideoFormat and makes all the custom get_unit_size functions in
1214 many elements unnecessary.
1216 2009-06-16 18:57:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1218 * gst-libs/gst/rtsp/gstrtspdefs.c:
1219 * gst-libs/gst/rtsp/gstrtspdefs.h:
1220 rtsp: add Timestamp header field
1223 2009-06-16 18:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1225 * gst/playback/gstplaybin2.c:
1226 playbin2: set smarter target state on uridecodebin
1227 Set the target state of the newly added uridecodebins to somthing else that
1228 PAUSED so that we keep their state in sync with the playsink state.
1231 2009-06-16 18:13:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1233 * gst/playback/gstplaysink.c:
1234 playsink: set the sink flag on the element
1236 2009-06-16 18:09:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1238 * gst/playback/gsturidecodebin.c:
1239 uridecodebin: add debug message
1241 2009-06-16 14:05:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1243 * gst-libs/gst/audio/gstaudiosink.c:
1244 * gst-libs/gst/audio/gstaudiosrc.c:
1245 audiosink, audiosrc: do the class_ref()s in the right class_init functions
1246 Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
1248 2009-06-15 15:39:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1250 * gst-libs/gst/audio/gstaudiosink.c:
1251 * gst-libs/gst/audio/gstaudiosrc.c:
1252 audiosink,audiosrc: ref the audio ring buffer class and type in class_init
1253 Hack around thread-safety issues in GObject and our racy _get_type()
1254 functions (we could easily fix the _get_type() functions, but we still
1255 need to hack around the GObject class races until we require a newer
1256 GLib version, I think).
1258 2009-06-15 12:57:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1260 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1261 audiosrc: return FALSE when receiving a SEEK event
1262 When receiving a seek event, return FALSE as we don't implement seeking.
1264 2009-06-15 11:06:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1266 * tests/examples/seek/seek.c:
1267 Don't use deprecated GTK API
1270 2009-06-15 11:40:00 +0300 Stefan Kost <ensonic@users.sf.net>
1272 * gst/adder/gstadder.c:
1273 adder: send flush_stop when seeking failed
1274 At least do the fix to sent the flush_stop when seeking failed to ensure we
1275 keep no pads flushing. before it was send when the seeking worked which is just
1276 plain wrong and was not the intention.
1278 2009-06-12 15:17:14 +0200 Peter Kjellerstedt <pkj@axis.com>
1280 * gst-libs/gst/rtsp/gstrtspconnection.c:
1281 rtsp: Use a more consistent naming of GstRTSPRec variables.
1283 2009-06-12 15:11:05 +0200 Peter Kjellerstedt <pkj@axis.com>
1285 * gst-libs/gst/rtsp/gstrtspconnection.c:
1286 * gst-libs/gst/rtsp/gstrtspconnection.h:
1287 rtsp: Call message_sent() callback for all sent messages.
1288 Previously the messages_sent() callback was only called for messages
1289 which had a CSeq, which excluded all data messages. Instead of using the
1290 CSeq as ID, use a simple index counter.
1292 2009-06-14 22:13:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1294 * ext/ogg/gstoggdemux.c:
1295 * ext/theora/theoradec.c:
1296 * ext/vorbis/vorbisdec.c:
1297 oggdemux: post/send tags with the container-format tag
1298 For this to work properly, theoradec and vorbisdec need to put
1299 tag events received from upstream into the pending_events list
1300 so they get pushed out after any newsegment event, not before.
1302 2009-06-14 20:30:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1304 * tests/examples/seek/scrubby.c:
1305 * tests/examples/seek/seek.c:
1306 * tests/old/examples/seek/cdplayer.c:
1307 Don't use deprecated GTK API
1310 2009-06-12 16:31:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1312 * gst/adder/gstadder.c:
1313 adder: send flush-stop earlier
1314 When no flush-stop has been sent by upstream, we have to send one ourselves to
1315 continue playback. Do this as soon as the collect function is called instead of
1316 after we possibly pushed segment events (that got then flushed out)
1318 2009-06-12 13:55:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1320 * tests/examples/seek/seek.c:
1321 seek: add shuttle controls
1323 2009-06-12 13:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1325 * tests/examples/seek/stepping2.c:
1326 example: fix compile
1328 2009-06-12 13:52:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1330 * tests/examples/seek/Makefile.am:
1331 examples: build the stepping2 example
1333 2009-06-12 13:52:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1335 * gst/playback/gstplaysink.c:
1336 playsink: update for new step API
1338 2009-06-12 13:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1340 * ext/ogg/gstoggdemux.c:
1341 oggdemux: do reverse seeks more accurate
1342 For reverse seeking with the accurate flag set, try to be more precise by
1343 seeking a little bit after the requested position.
1345 2009-06-11 22:32:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1347 * ext/ogg/gstogmparse.c:
1348 * gst/subparse/gstssaparse.c:
1349 * gst/subparse/gstssaparse.h:
1350 * gst/subparse/gstsubparse.c:
1351 * gst/subparse/gstsubparse.h:
1352 subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC
1353 Make subtitle parsers post a taglist with codec tags, so the application
1354 knows what kind of subtitle a subtitle stream is. Fixes #576552.
1356 2009-06-11 19:12:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1358 * gst-libs/gst/audio/gstringbuffer.c:
1359 ringbuffer: handle border cases in resampler
1361 2009-06-11 13:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
1364 * docs/libs/Makefile.am:
1365 * docs/plugins/Makefile.am:
1366 docs: Update common. Use upload-doc.mak instead of upload.mak
1368 2009-06-11 12:39:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1370 * gst-libs/gst/rtp/gstbasertppayload.c:
1373 2009-06-11 12:17:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1375 * gst-libs/gst/audio/gstbaseaudiosink.c:
1376 baseaudiosink: reset accum when dropping samples
1377 When we are resampling and we drop samples because we paused, reset the accum
1378 counter because it's now invalid.
1380 2009-06-11 11:16:15 +0100 Jan Schmidt <thaytan@noraisin.net>
1382 * docs/libs/gst-plugins-base-libs-sections.txt:
1383 * gst-libs/gst/interfaces/mixer.h:
1384 * gst-libs/gst/video/gstbasevideodecoder.h:
1385 docs: Fix a couple of warnings from the docs build.
1387 2009-06-10 21:36:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1389 * gst-libs/gst/audio/testchannels.c:
1390 Don't include config.h multiple times when build audio testchannel app.
1391 Fixes build problem on win32 (#585075).
1393 2009-06-10 16:56:51 +0100 Jan Schmidt <thaytan@noraisin.net>
1395 * gst/playback/gstplaybin2.c:
1396 * gst/playback/gsturidecodebin.c:
1397 playbin2/uridecodebin: Fix connection-speed propagation
1398 uridecodebin expects the passed connection-speed value in kbps, so we
1399 need to divide the value stored in bps by 1000. Also, lower the upper
1400 limit on the properties to the value that we can actually store in our
1401 internal guint (which is plenty high enough)
1403 2009-06-10 14:37:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1405 * gst/subparse/gstsubparse.c:
1406 * tests/check/elements/subparse.c:
1407 subparse: recognise more subrip timestamp variants
1408 Be even less restrictive in what we accept for .srt timestamps when
1409 typefinding and parsing subrip subtitles and add a unit test for
1410 the 'new' format. Fixes #585197.
1412 2009-06-09 22:00:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1414 * gst-libs/gst/rtsp/gstrtsptransport.h:
1415 rtsp: add some more docs
1417 2009-06-09 18:24:55 +0200 Peter Kjellerstedt <pkj@axis.com>
1419 * gst-libs/gst/rtsp/gstrtspmessage.c:
1420 rtsp: Avoid a compiler warning.
1422 2009-06-09 18:23:28 +0200 Peter Kjellerstedt <pkj@axis.com>
1424 * gst-libs/gst/rtsp/gstrtspdefs.h:
1425 rtsp: Updated documentation for GstRTSPResult.
1426 Moved GST_RTSP_ELAST to be last in the documentation to match the actual
1429 2009-05-20 17:30:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1432 autogen: remove -Wno-portability from here
1433 as it is in configure.ac now.
1435 2009-06-09 16:28:20 +0200 Peter Kjellerstedt <pkj@axis.com>
1437 * gst-libs/gst/rtsp/gstrtspconnection.c:
1438 rtsp: Plug a memory leak.
1439 Free memory related to any partially read and/or written RTSP messages.
1441 2009-06-09 12:09:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1443 * gst-libs/gst/audio/gstbaseaudiosink.c:
1444 baseaudiosink: no need to cause discont when clipping
1445 Remove the discont-when-clipping hack now that basesink provides us with
1446 correctly clipped samples when stepping.
1448 2009-06-08 17:26:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1450 * gst-libs/gst/audio/gstbaseaudiosink.c:
1451 audiosink: don't align when we clip
1452 Don't align samples when they were clipped. Not entirely correct but better than
1455 2009-06-08 16:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1457 * tests/examples/seek/.gitignore:
1458 * tests/examples/seek/stepping2.c:
1459 examples: add stepping example in PLAYING
1460 Add stepping example in PLAYING, audio is a bit distorted because basesink does
1461 not provide good clipping info yet.
1463 2009-06-08 10:25:00 +0200 Edward Hervey <bilboed@bilboed.com>
1465 * gst-libs/gst/pbutils/descriptions.c:
1466 pbutils: Add description for hdv/aux-* formats.
1468 2009-06-07 22:20:33 +0400 LRN <lrn1986@gmail.com>
1470 * ext/schroedinger/Makefile.am:
1471 Added libgstbase to schro's LIBADD
1474 2009-06-06 02:15:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1476 * gst-libs/gst/tag/gstid3tag.c:
1477 libgsttag: don't extract genres from empty ID3v1 tags
1478 If we don't have any other info, don't try to interpret the
1479 genre field. In particular we don't want to interpret a genre
1480 of 0 as 'Blues' if no other fields are set and the entire tag
1483 2009-06-05 18:13:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1485 * gst/playback/gstdecodebin2.c:
1486 decodebin2: make sure varargs are of right type
1487 Explicitly cast the variables to g_object_set to their right types.
1489 2009-06-05 16:49:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1491 * gst/playback/gstdecodebin2.c:
1492 decodebin2: increase stream probing queues
1493 When we are probing for streams, we want to set the queue size in such a way
1494 that we can scan a maximum amount of data without consuming too much memory.
1495 Therefore, remove the time limit on the queue and only stop scanning after 2MB
1499 2009-06-05 14:06:17 +0200 Peter Kjellerstedt <pkj@axis.com>
1501 * gst-libs/gst/rtsp/gstrtspconnection.c:
1504 2009-06-05 14:05:54 +0200 Peter Kjellerstedt <pkj@axis.com>
1506 * gst-libs/gst/rtsp/gstrtspconnection.c:
1507 rtsp: Remove an unused variable.
1509 2009-06-05 13:59:14 +0200 Peter Kjellerstedt <pkj@axis.com>
1511 * gst-libs/gst/rtsp/gstrtspconnection.c:
1512 rtsp: Removed duplicate initialization of conn->writefd.
1514 2009-06-05 13:55:08 +0200 Peter Kjellerstedt <pkj@axis.com>
1516 * gst-libs/gst/rtsp/gstrtspconnection.c:
1517 rtsp: Use #defined status codes.
1519 2009-06-05 13:53:29 +0200 Peter Kjellerstedt <pkj@axis.com>
1521 * gst-libs/gst/rtsp/gstrtspconnection.c:
1522 rtsp: Correct gen_tunnel_reply().
1523 Prevent gen_tunnel_reply() from generating an incomplete response
1524 in case an error response code is given.
1526 2009-06-05 10:57:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1529 * win32/common/_stdint.h:
1530 * win32/common/config.h:
1531 * win32/common/video-enumtypes.c:
1532 configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
1533 See #584835. Also update win32 files while we're at it.
1535 2009-06-04 08:57:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1537 * gst/playback/gstplaybin2.c:
1538 playbin2: API: Add {audio,video,text}-tags-changed signals
1541 2009-06-03 20:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1543 * ext/vorbis/vorbisdec.c:
1544 vorbisdec: don't put invalid bitrate values into the taglist
1545 Bitrates are stored as 32-bit signed integers in the vorbis
1546 identification headers, but seem to be read incorrectly,
1547 namely as unsigned 32-bit integers, into the vorbis structure
1548 members which are of type long, which makes our check for
1549 values <= 0 fail with files that put -1 in there for unset
1552 2009-06-03 15:52:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1554 * tests/examples/seek/.gitignore:
1555 ignore: add new stepping app to ignore
1557 2009-06-03 15:31:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1559 * tests/examples/seek/Makefile.am:
1560 * tests/examples/seek/stepping.c:
1561 examples: add stepping example.
1562 Add an example of using playbin2 and frame stepping to simulate variable rate
1563 playback based on a sine wave.
1565 2009-06-03 12:45:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1567 * gst/playback/gstplaybin2.c:
1568 * gst/playback/gstplaysink.h:
1569 playbin2: also set custom text and subp sinks
1570 Set the custom subpicture and text sinks along with the custom audio and video
1572 Fix a little docs blurb too.
1574 2009-06-02 12:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1576 * gst-libs/gst/rtsp/gstrtspconnection.c:
1577 * gst-libs/gst/rtsp/gstrtspconnection.h:
1578 rtsp: add G_LIKELY because we can
1580 2009-06-02 09:53:05 +0200 Edward Hervey <bilboed@bilboed.com>
1582 * gst/typefind/gsttypefindfunctions.c:
1583 typefindfunctions: Fix caps for ogg typefinder.
1585 2009-05-29 11:10:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1587 * docs/libs/gst-plugins-base-libs-sections.txt:
1588 docs: remove some cruft from -sections.txt file
1590 2009-06-01 11:31:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1592 * gst/playback/gstplaysink.c:
1593 * tests/examples/seek/seek.c:
1594 add framestepping to playbin2 and seek
1596 2009-06-01 09:59:22 +0200 Peter Kjellerstedt <pkj@axis.com>
1598 * gst-libs/gst/rtsp/gstrtspconnection.c:
1599 rtsp: Avoid compiler warnings with -Wextra.
1601 2009-06-01 09:58:27 +0200 Peter Kjellerstedt <pkj@axis.com>
1603 * gst-libs/gst/rtsp/gstrtspconnection.h:
1604 rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
1606 2009-06-01 09:43:04 +0200 Peter Kjellerstedt <pkj@axis.com>
1608 * gst-libs/gst/sdp/gstsdpmessage.c:
1609 sdp: Remove an unused variable.
1611 2009-05-30 14:17:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1613 * gst/ffmpegcolorspace/imgconvert.c:
1614 * gst/ffmpegcolorspace/imgconvert_template.h:
1615 ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale
1617 2009-05-29 00:09:15 +0100 Jan Schmidt <thaytan@noraisin.net>
1619 * gst/playback/gstplaybin2.c:
1620 playbin2: Have playbin recognise PGS subpicture streams
1621 Recognise PGS subpicture streams and connect them to the SPU pad
1622 in playsink. Unfortunately this fails badly with negotiation errors
1623 if the SPU is not recent enough to support the stream. I'm not sure
1624 how to add format negotiation in yet.
1626 2009-05-21 23:11:29 +0100 Jan Schmidt <thaytan@noraisin.net>
1628 * gst/playback/gstdecodebin2.c:
1629 * gst/playback/gsturidecodebin.c:
1630 decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.
1632 2009-05-28 20:37:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1634 * gst/playback/gstplaysink.c:
1635 playbin2: fix volume handling for audio sinks without "volume" property
1636 When using an audio sink without a "volume" property, volume control
1637 would only work for the first song. For the next song, we'd try to
1638 re-use the existing audio chain, but inadvertently set chain->volume
1639 to NULL instead of to the existing volume element.
1641 2009-05-28 17:05:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1643 * gst/playback/gstplaysink.c:
1644 playbin2: cosmetic change to avoid unnecessary line breaks
1645 Looks nicer and works around gst-indent silliness.
1647 2009-05-28 17:21:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1649 * gst/playback/gstplaysink.c:
1650 playbin2: don't lose the ref to the volume element
1651 Only release the ref to the volume element when it is controled by a sink. For
1652 software volume we never have to fear that it will change.
1654 2009-05-28 15:21:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1656 * gst/playback/gstplaybin2.c:
1657 * gst/playback/gstplaysink.c:
1658 playbin2: actually use configured audio/video sinks
1659 playbin2 inadvertently used autoaudiosink and autovideosink up to now,
1660 since it would overwrite the sinks configured via the "audio-sink"
1661 and "video-sink" properties with the stream-specific group sinks when
1662 configuring the outputs. Those are usually NULL however, so that would
1663 overwrite the configured sinks with NULL which makes playbin2 then
1664 default to the auto sinks. Fix this by keeping a reference to each
1665 configured sink in playbin2 and setting up the right sinks depending
1666 on whether there is a stream-specific sink or not.
1669 2009-05-27 17:37:38 +0300 Stefan Kost <ensonic@users.sf.net>
1671 * tests/examples/seek/seek.c:
1672 seek: add volume label and sync with sink volume
1673 Look at the volume and have the pulsemixer open at same time. Unfortunately
1674 playbin2 does not emit notify on volume right, so this polls for now.
1676 2009-05-27 18:12:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1678 * gst/playback/gstdecodebin2.c:
1679 decodebin2: remove leftover elements
1680 Remove all of the elements inside decodebin2 when goint to READY and NULL.
1681 Makes decodebin2 reusable.
1684 2009-05-27 15:36:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1686 * gst/playback/gstplaysink.c:
1687 playbin2; release refs to volume/mute properties
1688 Release the refs to the volume and mute property elemens before setting the
1689 child elements to READY or NULL.
1692 2009-05-27 12:10:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1694 * gst/gdp/gstgdppay.c:
1695 gdppay: set caps on outgoing buffers
1696 Set caps on outgoing buffers because NULL caps confuse basetransform.
1699 2009-05-27 11:08:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1701 * gst-libs/gst/netbuffer/gstnetbuffer.c:
1702 netbuffer: also note the order of IP4 addresses
1703 IP4 addresses are also stored in network byte order. Make a note of this in the
1706 2009-05-26 22:43:34 +0200 Alessandro Decina <alessandro.d@gmail.com>
1708 * ext/theora/theoraparse.c:
1709 theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.
1711 2009-05-26 11:13:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1713 * gst-libs/gst/rtsp/gstrtspconnection.c:
1714 Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14"
1715 This reverts commit 418760cf740332c12c3fd9cf3244af134fa9534b.
1716 We now require GLib 2.16.
1718 2009-05-26 15:18:09 +0100 Jan Schmidt <thaytan@noraisin.net>
1723 2009-05-26 15:37:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1725 * gst-libs/gst/netbuffer/gstnetbuffer.c:
1726 netbuffer: document that the port is network order
1727 Document the fact that we store the port number in network order in
1728 GstNetAddress and that the caller should byteswap appropriately.
1730 2009-05-26 15:23:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1732 * gst/videoscale/gstvideoscale.c:
1733 * gst/videoscale/vs_4tap.c:
1734 * gst/videoscale/vs_4tap.h:
1735 * gst/videoscale/vs_image.c:
1736 * gst/videoscale/vs_image.h:
1737 * gst/videoscale/vs_scanline.c:
1738 * gst/videoscale/vs_scanline.h:
1739 videoscale: Add support for 16 bit grayscale in native endianness
1741 2009-05-26 14:58:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1743 * gst/ffmpegcolorspace/avcodec.h:
1744 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
1745 * gst/ffmpegcolorspace/imgconvert.c:
1746 ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian
1748 2009-05-26 14:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1750 * gst/videotestsrc/videotestsrc.c:
1751 * gst/videotestsrc/videotestsrc.h:
1752 videotestsrc: Add support for 16 bit grayscale in native endianness
1754 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
1756 add can-activate-pull property to baseaudiosink
1757 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
1760 2009-05-26 13:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1762 * ext/ogg/gstoggdemux.c:
1763 oggdemux: fix boundary case for seeking.
1764 When we have exactly 0 bytes left to search, make sure we stop instead of going
1765 into an infinite loop.
1767 2009-05-26 11:11:03 +0200 Bastien Nocera <hadess at hadess.net>
1769 * gst-libs/gst/cdda/Makefile.am:
1770 * gst-libs/gst/cdda/gstcddabasesrc.c:
1771 * gst-libs/gst/cdda/sha1.c:
1772 * gst-libs/gst/cdda/sha1.h:
1773 cddabasesrc: Remove copy of sha1 digest
1774 Remove our copy of sha1 digest now that we depend on glib 2.16.
1777 2009-05-25 17:54:01 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
1779 * gst-plugins-base.spec.in:
1782 2009-05-23 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1784 * gst-libs/gst/video/gstbasevideodecoder.c:
1785 * gst-libs/gst/video/gstbasevideoparse.c:
1786 * gst-libs/gst/video/gstbasevideoutils.c:
1787 * gst-libs/gst/video/gstbasevideoutils.h:
1788 * win32/common/libgstvideo.def:
1789 video: don't expose internal gst_adapter_get_buffer() helper function
1790 If it's really needed it should go into GstAdapter in core.
1792 2009-05-22 21:29:51 -0700 David Schleef <ds@schleef.org>
1794 * gst-libs/gst/video/gstbasevideodecoder.c:
1795 basevideo: Fix memleak
1797 2009-05-22 21:27:58 -0700 David Schleef <ds@schleef.org>
1799 * ext/schroedinger/gstschrodec.c:
1800 * ext/schroedinger/gstschroparse.c:
1801 schro: Fix usage of adapter_masked_scan_uint32
1802 Because *somebody* changed the API without telling me.
1804 2009-05-22 21:25:06 -0700 David Schleef <ds@schleef.org>
1806 * ext/schroedinger/gstschro.c:
1807 schro: Change package name to GST_PACKAGE_NAME
1809 2009-05-22 17:34:10 -0700 David Schleef <ds@schleef.org>
1811 * gst-libs/gst/video/gstbasevideoencoder.c:
1812 basevideo: Add preset interface to encoder
1814 2009-05-22 17:31:14 -0700 David Schleef <ds@schleef.org>
1816 * gst/audioresample/gstaudioresample.c:
1817 Run liboil benchmark multiple times
1818 The statistics function requires multiple runs, otherwise
1819 it causes a divide by zero error.
1821 2009-05-22 19:36:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1823 * m4/gst-fionread.m4:
1824 m4: fix 'suspicious cache value' warning for gst-fionread.m4
1825 .. here as well (should really be moved to common, but I'm too lazy).
1827 2009-05-22 17:41:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1829 * ext/vorbis/vorbisdec.c:
1830 vorbisdec: detect and report errors better
1831 Check the return values of a couple more libvorbis functions and post an error
1832 when something is wrong instead of continuing and crashing.
1834 2009-05-22 15:49:14 +0300 Stefan Kost <ensonic@users.sf.net>
1836 * gst/playback/gstplaysink.c:
1837 playbin2: fix initial volume and mute handling
1838 Use two flags to remember volume/mute changes at times when we don't have the
1839 audiochain yet (e.g. construction). Only set values when they were actualy
1840 changed. This makes pulseaudio's stream restore functional.
1842 2009-05-22 10:19:51 +0100 Jan Schmidt <thaytan@noraisin.net>
1845 Automatic update of common submodule
1846 From d3a8fab to 888e0a2
1848 2009-05-22 09:03:22 +0100 Jan Schmidt <thaytan@noraisin.net>
1850 * win32/common/libgstvideo.def:
1851 win32: Remove gst_adapter_masked_scan_uint32 from the exports
1853 2009-05-21 10:48:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1855 * gst-libs/gst/audio/gstbaseaudiosink.c:
1856 audiosink: improve debug message
1858 2009-05-19 18:10:55 -0700 Michael Smith <msmith@songbirdnest.com>
1860 * gst-libs/gst/tag/gstid3tag.c:
1861 gstid3tag: Don't extract a track number unless present.
1862 In ID3v1, a track number is present only if byte 125 is null AND
1863 byte 126 is non-null. If the track number is not present, don't add
1864 a track number tag with value 0.
1866 2009-05-20 00:48:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1868 * gst-libs/gst/video/gstbasevideoutils.c:
1869 * gst-libs/gst/video/gstbasevideoutils.h:
1870 videoutils: remove adapter methods
1871 Remove adapter methods now that they are in core.
1873 2009-05-20 00:42:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1875 * win32/common/libgstvideo.def:
1876 defs: add new symbols
1878 2009-05-19 17:47:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1881 autogen: pass -Wno-portability to automake to suppress warnings
1884 2009-05-19 02:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1886 * docs/libs/.gitignore:
1887 gitignore: remove bogus *.sgml wildcard - these files are tracked in git
1889 2009-05-19 18:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1891 * gst/tcp/gsttcpclientsrc.c:
1892 tcpclientsrc: this is not a live source
1893 Don't mark us as a live source because we are not.
1895 2009-05-19 18:41:02 +0300 Stefan Kost <ensonic@users.sf.net>
1897 * gst/adder/gstadder.c:
1898 adder: only send flush_stop when seek failed
1899 This is still not the ultimate fix. Added some comment to explain the troubles.
1901 2009-05-19 17:17:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1903 * gst-libs/gst/audio/gstbaseaudiosink.c:
1904 audiosink: return the return value of wait_preroll
1905 Return the value that _wait_preroll() returned instead of always WRONG_STATE.
1907 2009-05-19 16:45:56 +0300 Stefan Kost <ensonic@users.sf.net>
1909 * gst/adder/gstadder.c:
1910 * gst/adder/gstadder.h:
1911 adder: send flush_stop to match flush_start
1912 Adder was relying that something else sends a flush stop. When using adder with
1913 a livesource it was not getting a flush_stop and thus all pads downstream where
1914 keept flushing. Mark a pending flush_stop and send it when we are working on
1915 the new segment back in the streaming thread.
1917 2009-05-19 16:02:44 +0300 Stefan Kost <ensonic@users.sf.net>
1919 * tests/examples/seek/seek.c:
1920 seek: ui improvements
1921 Repaint the window black on expose, as this looks nicer when resizing or using
1922 the expander. Also show time after slider, as this saves a whole line (nice on
1925 2009-04-29 18:36:17 +0300 Stefan Kost <ensonic@users.sf.net>
1927 * gst/playback/gstdecodebin.c:
1928 decodebin: use iterators instead of list
1929 The list api is deprecated. Use threadsafe iterators instead.
1931 2009-05-19 15:35:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1933 * gst/playback/gsturidecodebin.c:
1934 uridecodebin: configure caps on decodebin2
1935 Implement the caps property by setting the configured caps on new decodebin2
1939 2009-05-19 15:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1941 * gst/playback/gstdecodebin2.c:
1942 decodebin2: avoid some _caps_ref in some cases
1943 Only mess with the caps refcount when we configure different caps.
1945 2009-05-19 15:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1947 * gst/playback/gsturidecodebin.c:
1948 uridecodebin: fix potential caps leak
1949 Free the user-configured caps in finalize.
1951 2009-05-19 15:20:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1953 * gst/playback/gsturidecodebin.c:
1954 uridecodebin: add queue after cdda://
1955 Add a queue2 after the raw output pads of certain sources such as those for uris
1957 No tuning of the queue is done yet as the defaults seem to work fine for me.
1960 2009-05-19 12:45:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1962 * ext/ogg/gstoggdemux.c:
1963 oggdemux: don't loop when at EOS
1964 When we try to read the last page, don't try to read past the upper boundary, as
1965 this might cause endless loops.
1968 2009-05-19 11:20:19 +0200 Edward Hervey <bilboed@bilboed.com>
1970 * gst/audioresample/gstaudioresample.c:
1971 audioresample: Don't drain remaining buffers after a flush.
1972 If we were resetted (due to a flush), we can not drain the remaining
1973 buffers since they would be pushed before a valid new newsegment event.
1975 2009-05-18 22:29:07 -0700 Michael Smith <msmith@syncword.(none)>
1977 * ext/theora/theoradec.c:
1978 theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.
1980 2009-05-19 01:13:34 +0300 Stefan Kost <ensonic@users.sf.net>
1982 * gst/adder/gstadder.c:
1983 adder: add more logging and return value checking
1985 2009-05-19 01:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
1987 * gst/adder/gstadder.c:
1988 adder: handle the return value from iterator_fold
1990 2009-05-19 01:03:44 +0300 Stefan Kost <ensonic@users.sf.net>
1992 * gst/adder/gstadder.c:
1993 adder: use the pad in logging as objects
1994 Helps to differenciate between source and sinks pads.
1996 2009-04-21 22:54:19 +0300 Stefan Kost <ensonic@users.sf.net>
1998 * tests/examples/seek/seek.c:
1999 seek: use parser for mp3 and rename variable
2001 2009-05-18 11:08:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2003 * tests/examples/seek/seek.c:
2004 seek: add playbin2 options in expander
2005 Add the playbin2 stream selection options inside an expander to preserve some
2008 2009-02-10 15:29:10 -0800 David Schleef <ds@schleef.org>
2010 * gst/videotestsrc/videotestsrc.c:
2011 videotestsrc: Add support for v210 and v216 formats
2013 2009-05-15 16:21:15 -0700 David Schleef <ds@schleef.org>
2015 * gst-libs/gst/video/gstbasevideocodec.c:
2016 * gst-libs/gst/video/gstbasevideodecoder.c:
2017 * gst-libs/gst/video/gstbasevideoencoder.c:
2018 * gst-libs/gst/video/gstbasevideoparse.c:
2019 video: remove // comments
2021 2009-05-15 16:18:18 -0700 David Schleef <ds@schleef.org>
2023 * gst-libs/gst/video/video.c:
2024 * gst-libs/gst/video/video.h:
2025 video: Add Y444, v210, v216 formats
2027 2009-05-15 16:12:37 -0700 David Schleef <ds@schleef.org>
2031 * ext/schroedinger/Makefile.am:
2032 * ext/schroedinger/gstschro.c:
2033 * ext/schroedinger/gstschrodec.c:
2034 * ext/schroedinger/gstschroenc.c:
2035 * ext/schroedinger/gstschroparse.c:
2036 * ext/schroedinger/gstschroutils.c:
2037 * ext/schroedinger/gstschroutils.h:
2038 schro: Move schro plugin from Schroedinger
2039 Previous history is in Schroedinger. Depends on, and is an example
2040 of using, GstBaseVideo* base classes.
2041 Code was reindented, and an #ifdef HAVE_ENCODER removed.
2043 2009-05-15 10:23:08 -0700 David Schleef <ds@schleef.org>
2045 * gst-libs/gst/video/Makefile.am:
2046 * gst-libs/gst/video/gstbasevideocodec.c:
2047 * gst-libs/gst/video/gstbasevideocodec.h:
2048 * gst-libs/gst/video/gstbasevideodecoder.c:
2049 * gst-libs/gst/video/gstbasevideodecoder.h:
2050 * gst-libs/gst/video/gstbasevideoencoder.c:
2051 * gst-libs/gst/video/gstbasevideoencoder.h:
2052 * gst-libs/gst/video/gstbasevideoparse.c:
2053 * gst-libs/gst/video/gstbasevideoparse.h:
2054 * gst-libs/gst/video/gstbasevideoutils.c:
2055 * gst-libs/gst/video/gstbasevideoutils.h:
2056 video: Copy BaseVideo classes from Schroedinger
2058 2009-05-15 23:05:45 +0200 Arnout Vandecappelle <arnout@mind.be>
2060 * gst/tcp/gstmultifdsink.c:
2061 multifdsink: add num-fds property
2062 multifdsink::num-fds
2064 2009-05-15 20:36:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2066 * gst-libs/gst/pbutils/descriptions.c:
2067 pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
2069 2009-05-14 11:44:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2071 * ext/vorbis/vorbisenc.c:
2072 vorbisenc: Implement Preset interface
2074 2009-05-14 11:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2076 * ext/theora/theoraenc.c:
2077 theoraenc: Implement Preset interface
2079 2009-05-14 11:41:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2081 * ext/ogg/gstoggmux.c:
2082 oggmux: Implement Preset interface
2084 2009-05-14 21:37:22 +0100 Jan Schmidt <thaytan@noraisin.net>
2086 * gst/playback/gstplaysink.c:
2087 playbin2: Fix cdda:// playback
2088 Don't send async-start when the playsink has already been configured
2089 before changing state.
2091 2009-05-14 01:31:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2094 configure: require core CVS for gst_adapter_prev_timestamp()
2095 which is used in the libvisual plugin.
2097 2009-04-22 18:34:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2100 AUTHORS: fix my email
2102 2009-04-22 18:35:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2104 * gst-libs/gst/audio/gstaudioclock.c:
2105 audioclock: make our internal time monotonic
2106 Make the internal time increase monotonically.
2108 2009-05-13 19:27:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2110 * ext/libvisual/visual.c:
2111 visual: remove next_ts variable
2112 We can remove the next_ts variable as we don't use it anymore.
2114 2009-05-13 19:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2116 * ext/libvisual/visual.c:
2117 visual: use new adapter timestamp code
2118 Use the new adapter timestamp tracking code to make things easier and produce
2119 vastly better output timestamps.
2121 2009-05-13 01:35:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2153 po: avoid conflicts of local *.po files with files in git
2154 Make it so that filenames and line numbers are only stored in the *.pot file
2155 (which is not in git), but not in the individual *.po files. This information
2156 is hardly useful for translators in our case, and it should avoid the constant
2157 conflicts of local *.po files with the ones in git which are caused by the
2158 source files changing and the line numbers being updated. This commit might
2159 cause one last merge conflict for you, which you can work around with
2160 "git checkout po/*.po" before merging or pulling. After that there should
2161 (hopefully) not be any more local modifications of these files (unless
2162 someone committed additions or changes to translated strings and the
2163 *.po files haven't been updated yet, that is).
2165 2009-05-12 23:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2167 * tests/check/elements/.gitignore:
2168 * tests/check/elements/audioresample.c:
2169 tests: fix audioresample unit test on big endian architectures
2170 Don't hardcode endianness=1234 in the filtercaps, it will cause
2171 pad link failures which will result in the test timing out.
2173 2009-05-12 17:18:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2175 * gst/audiotestsrc/gstaudiotestsrc.c:
2176 audiotestsrc: fix broken enum nick - it should have a hyphen
2177 The enum nick should be 'sine-table', not 'sine table'. Technically this is
2178 an API/ABI change I guess, but anyone who was using this and didn't report
2181 2009-05-01 01:04:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2183 * gst/audiotestsrc/gstaudiotestsrc.c:
2184 audiotestsrc: seek to the requested byte offset, not the expected byte offset
2186 2009-05-01 01:03:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2188 * gst/audiotestsrc/gstaudiotestsrc.c:
2189 * gst/audiotestsrc/gstaudiotestsrc.h:
2190 audiotestsrc: support more than just one channel
2192 2009-05-12 15:52:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2194 * gst-libs/gst/interfaces/propertyprobe.h:
2195 propertyprobe: Fix typo in the docs
2197 2009-05-12 12:17:55 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
2199 * ext/ogg/gstoggmux.c:
2200 * ext/theora/theora.c:
2201 * ext/vorbis/vorbis.c:
2202 Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder
2204 2009-04-30 16:37:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2206 * gst/videorate/gstvideorate.c:
2207 * gst/videorate/gstvideorate.h:
2208 videorate: handle invalid timestamps better
2209 Handle buffers with -1 timestamps better by keeping track of the en time of the
2210 previous buffer and assuming the -1 timestamp buffer goes right after the
2212 when we have two buffers that are equally good, output the oldest buffer once to
2214 don't try to calculate latency when the input framerate is unknown.
2216 2009-04-28 11:37:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2218 * ext/ogg/gstoggmux.c:
2219 oggmux: small debug statement in DISCONT
2221 2009-04-28 11:24:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2223 * ext/ogg/gstoggdemux.c:
2224 * ext/ogg/gstoggdemux.h:
2225 oggdemux: fix abuse of ogg API, handle broken oggs
2226 When we feed the ogg sync layer, we need to feed it contiguous data even if the
2227 sync layer did not consume all of it yet. This makes sure that it always finds
2228 the next page even for more corrupted files. Use a different read_offset for
2229 this purpose. since we now keep track of the sync layer, we don't have to reset
2230 after finding a start of a page.
2231 Add some more debug info for the error paths.
2232 Only reset the sync layer when we perform a seek operation.
2233 Avoid failure when the next chain has no bos pages but instead simply ignore it.
2234 when we receive unknown page serial numbers mid stream, don't fail but post a
2235 warning and hope that we get back on track later.
2238 2009-04-30 16:41:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2240 * gst/playback/gstdecodebin2.c:
2241 decodebin2: make subpictures a raw output format
2242 Subpictures are a raw format, we want those pads exposed so that playbin2 can do
2243 the subpicture mixing.
2245 2009-04-27 10:15:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2247 * gst-libs/gst/rtp/gstbasertppayload.c:
2248 * gst-libs/gst/rtp/gstbasertppayload.h:
2249 rtpdepay: add some more comments
2251 2009-04-17 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2253 * gst-libs/gst/audio/gstaudioclock.c:
2254 audioclock: make sure values are ever increasing
2256 2009-05-05 17:17:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2258 * gst/playback/gstplaysink.c:
2259 playbin2: make fallback identity silent
2260 Set the signal-handoffs to FALSE and silent to TRUE for the fallback identity
2261 element so that it consumes less CPU.
2263 2009-04-17 10:57:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2265 * gst/playback/gstplaybin2.c:
2266 * gst/playback/gstplaysink.c:
2267 playbin2: handle custom audiosinks differently
2268 Keep track of the autoplugged custom sinks and configure them in the playsink
2269 element when we have collected all streams.
2270 Also make sure that we only select one custom sink.
2271 When unreffing the internal sink, we don't need to change the state to NULL.
2273 2009-05-12 10:36:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2275 * gst/playback/gstplaybin2.c:
2276 * gst/playback/gstplaysink.c:
2277 * gst/playback/gstplaysink.h:
2278 playbin2: unify custom sink get/set functions
2279 Use one function to set/get all of the different sink types.
2280 cleanup up the subpicture chain too.
2281 Allow setting a custom subpicture sink.
2283 2009-05-11 18:29:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2285 * gst-libs/gst/interfaces/tunernorm.h:
2286 interfaces: Seperate some more struct definitions from typedefs
2288 2009-05-11 15:48:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2290 * gst-libs/gst/interfaces/navigation.h:
2291 * gst-libs/gst/interfaces/videoorientation.h:
2292 * gst-libs/gst/interfaces/xoverlay.h:
2293 interfaces: Seperate some more struct definitions from typedefs
2295 2009-05-10 17:28:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2297 * win32/common/libgstinterfaces.def:
2298 Add new functions to win32 exports
2300 2009-05-10 17:28:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2302 * docs/libs/gst-plugins-base-libs-sections.txt:
2303 Add new functions to the docs
2305 2009-05-10 17:25:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2307 * gst-libs/gst/interfaces/mixer.c:
2308 * gst-libs/gst/interfaces/mixer.h:
2309 interfaces: API: Add gst_mixer_get_mixer_type()
2310 This is a convenience function that returns the mixer_type
2311 of the interface struct.
2313 2009-05-10 17:25:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2315 * gst-libs/gst/interfaces/colorbalance.c:
2316 interfaces: Add docs for gst_color_balance_get_balance_type()
2318 2009-05-10 11:17:19 +0200 Marc-Andre Lureau <marcandre.lureau@gmail.com>
2321 Run libtoolize before aclocal
2322 This unbreaks the build in some cases. Fixes bug #582021
2324 2009-05-07 17:38:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2326 * ext/pango/gsttextrender.c:
2327 textrender: Correctly initialize the background for ARGB too
2329 2009-05-07 16:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2331 * ext/pango/gsttextrender.c:
2332 * ext/pango/gsttextrender.h:
2333 textrender: Use libgstvideo functions to create caps
2334 Also check if downstream wants ARGB always when we get
2337 2009-05-07 16:52:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2339 * ext/pango/gsttextrender.c:
2340 textrender: Don't always use ARGB if downstream supports it but take it's preference
2342 2009-05-07 16:48:08 +0200 Kapil Agrawal <kapil@mediamagictechnologies.com>
2344 * ext/pango/gsttextrender.c:
2345 * ext/pango/gsttextrender.h:
2346 textrender: Add support for ARGB and alignment properties
2349 2009-05-07 16:42:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2351 * ext/pango/gsttextrender.c:
2352 textrender: Add ; after GST_BOILERPLATE to fix indention
2354 2009-05-07 15:10:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2356 * gst-libs/gst/tag/gstvorbistag.c:
2357 vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
2359 2009-05-07 14:59:36 +0200 Arnout Vandecappelle <arnout@mind.be>
2361 * gst/typefind/gsttypefindfunctions.c:
2362 typefindfunctions: made mp3_type_find less aggressive
2363 mp3_type_find could suggest already when only a single valid header
2364 was found, if it ran out of data before the end of the next frame.
2365 Therefore, ignore the last found frame if it was incomplete.
2368 2009-05-07 14:48:29 +0200 John Millikin <jmillikin@gmail.com>
2370 * gst-libs/gst/tag/gstvorbistag.c:
2371 vorbistag: Store cover art in vorbiscomments
2374 2009-05-07 06:14:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2376 * gst-libs/gst/interfaces/colorbalance.c:
2377 * gst-libs/gst/interfaces/colorbalance.h:
2378 interfaces: API: Add gst_color_balance_get_balance_type()
2379 This is a convenience function that returns the balance_type
2380 of the interface struct.
2382 2009-05-06 17:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2384 * gst-libs/gst/interfaces/colorbalance.h:
2385 * gst-libs/gst/interfaces/colorbalancechannel.h:
2386 * gst-libs/gst/interfaces/tuner.h:
2387 * gst-libs/gst/interfaces/tunerchannel.h:
2388 interfaces: Separate struct definitions from typedefs
2390 2009-05-06 14:03:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2392 * pkgconfig/gstreamer-app-uninstalled.pc.in:
2393 Fix libdir for uninstalled gstreamer-app library
2395 2009-05-12 01:59:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2397 * gst-libs/gst/pbutils/descriptions.c:
2398 pbutils: add description for APE tag caps
2400 2009-05-12 01:35:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2403 configure: bump core requirement to last release
2404 as that's more likely to be true than that we need
2407 2009-05-12 01:21:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2411 configure: rename CVS -> git in a couple of places
2413 2009-05-12 01:17:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2416 configure: bump GLib requirement to GLib >= 2.16
2417 as per the New Regime (see wiki).
2419 2009-05-01 00:09:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2421 * gst-libs/gst/tag/gsttagdemux.c:
2422 tagdemux: cache events from upstream and re-send them once we have a source pad
2423 Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
2426 2009-05-07 14:07:44 -0700 Michael Smith <msmith@songbirdnest.com>
2428 * gst-libs/gst/riff/riff-media.c:
2429 riff: support UYVY raw 4:2:2 in riff.
2431 2009-05-11 21:20:07 +0100 Jan Schmidt <thaytan@noraisin.net>
2434 Back to development -> 0.10.23.1
2436 2009-04-27 22:42:55 -0700 Michael Smith <msmith@syncword.(none)>
2438 * ext/theora/theoradec.c:
2439 theoradec: fix buffer overrun on 422 decode.
2441 2009-04-27 21:39:01 -0700 Michael Smith <msmith@syncword.(none)>
2443 * ext/theora/theoradec.c:
2444 theoradec: 444 support.
2446 2009-04-27 21:30:04 -0700 Michael Smith <msmith@syncword.(none)>
2448 * ext/theora/theoradec.c:
2449 theoradec: handle 422 images (as YUY2).
2451 2009-04-27 21:01:51 -0700 Michael Smith <msmith@syncword.(none)>
2453 * ext/theora/gsttheoradec.h:
2454 * ext/theora/theoradec.c:
2455 theoradec: rearrange code in preparation for 422 and 444 support.
2457 === release 0.10.23 ===
2459 2009-05-10 23:57:01 +0100 Jan Schmidt <thaytan@noraisin.net>
2465 * docs/plugins/gst-plugins-base-plugins.args:
2466 * docs/plugins/gst-plugins-base-plugins.hierarchy:
2467 * docs/plugins/gst-plugins-base-plugins.interfaces:
2468 * docs/plugins/gst-plugins-base-plugins.prerequisites:
2469 * docs/plugins/gst-plugins-base-plugins.signals:
2470 * docs/plugins/inspect/plugin-adder.xml:
2471 * docs/plugins/inspect/plugin-alsa.xml:
2472 * docs/plugins/inspect/plugin-app.xml:
2473 * docs/plugins/inspect/plugin-audioconvert.xml:
2474 * docs/plugins/inspect/plugin-audiorate.xml:
2475 * docs/plugins/inspect/plugin-audioresample.xml:
2476 * docs/plugins/inspect/plugin-audiotestsrc.xml:
2477 * docs/plugins/inspect/plugin-cdparanoia.xml:
2478 * docs/plugins/inspect/plugin-decodebin.xml:
2479 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
2480 * docs/plugins/inspect/plugin-gdp.xml:
2481 * docs/plugins/inspect/plugin-gio.xml:
2482 * docs/plugins/inspect/plugin-gnomevfs.xml:
2483 * docs/plugins/inspect/plugin-libvisual.xml:
2484 * docs/plugins/inspect/plugin-ogg.xml:
2485 * docs/plugins/inspect/plugin-pango.xml:
2486 * docs/plugins/inspect/plugin-playback.xml:
2487 * docs/plugins/inspect/plugin-queue2.xml:
2488 * docs/plugins/inspect/plugin-subparse.xml:
2489 * docs/plugins/inspect/plugin-tcp.xml:
2490 * docs/plugins/inspect/plugin-theora.xml:
2491 * docs/plugins/inspect/plugin-typefindfunctions.xml:
2492 * docs/plugins/inspect/plugin-uridecodebin.xml:
2493 * docs/plugins/inspect/plugin-video4linux.xml:
2494 * docs/plugins/inspect/plugin-videorate.xml:
2495 * docs/plugins/inspect/plugin-videoscale.xml:
2496 * docs/plugins/inspect/plugin-videotestsrc.xml:
2497 * docs/plugins/inspect/plugin-volume.xml:
2498 * docs/plugins/inspect/plugin-vorbis.xml:
2499 * docs/plugins/inspect/plugin-ximagesink.xml:
2500 * docs/plugins/inspect/plugin-xvimagesink.xml:
2501 * gst-plugins-base.doap:
2502 * win32/common/_stdint.h:
2503 * win32/common/config.h:
2506 2009-05-10 23:56:05 +0100 Jan Schmidt <thaytan@noraisin.net>
2539 2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
2571 * win32/common/_stdint.h:
2572 * win32/common/config.h:
2573 0.10.22.6 pre-release
2575 2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2577 * gst/playback/gstplaysink.c:
2578 playbin2: fix resume after pause
2579 Don't ignore the state change of the children, they might be doing an ASYNC
2582 2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
2615 0.10.22.5 pre-release
2617 2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2619 * gst/tcp/gstmultifdsink.c:
2620 * gst/tcp/gsttcp-marshal.list:
2621 multifdsink: fix signature of the add-full signal
2622 The second parameter is a GstSyncMethod enum, not a boolean.
2624 2009-05-07 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2626 * gst/playback/gstplaysink.c:
2627 playsink: initialize variable too
2629 2009-05-07 14:28:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2631 * gst/playback/gstplaysink.c:
2632 playbin2: make playsink go ASYNC to PAUSED
2633 Make playsink go async to the PAUSED state instead of relying on uridecodebin
2634 for async behaviour in playbin. This solves some problems (mainly with DVD)
2635 where the pipeline would go to PLAYING before preroll completed, failing to
2636 select the audiosink clock.
2639 2009-05-06 16:09:52 +0100 Jan Schmidt <thaytan@noraisin.net>
2671 * win32/common/_stdint.h:
2672 * win32/common/config.h:
2673 0.10.22.4 pre-release
2675 2009-05-06 13:19:34 +0100 Zaheer Merali <zaheerabbas@merali.org>
2677 * ext/theora/theoraenc.c:
2678 * ext/vorbis/vorbisenc.c:
2679 vorbisenc, theoraenc: Ensure gp is computed consistently + clip to segment
2680 With vorbisenc, compute the granulepos with running time and clip incoming
2682 With theoraenc, drop out of segment buffers.
2684 2009-05-01 16:47:53 +0100 Jan Schmidt <thaytan@noraisin.net>
2686 * gst/audioresample/gstaudioresample.c:
2687 audioresample: Fix buffer size transformations
2688 When calculating the input/output buffer sizes in the transform_size function,
2689 take the number of channels into account, so we don't end up calculating
2690 a buffer size that only contains a partial number of audio frames.
2691 Also, when going from output size to input size, round down rather than
2692 up, so as to calculate the minimum number of samples that *might* yield
2693 a buffer of the intended destination size.
2694 Fixes: #580470 and #580952
2696 2009-04-29 16:45:27 +0100 Jan Schmidt <thaytan@noraisin.net>
2698 * ext/vorbis/gstvorbisenc.h:
2699 * ext/vorbis/vorbisenc.c:
2700 vorbisenc: Ensure output buffers fall within the segment
2701 Add the start position of the first segment to the running time
2702 used to generate buffer timestamps in vorbisenc. This avoids generating
2703 buffers which fall outside the initial segment. The element segment
2704 handling requires more extensive fixing, but this at least prevents
2705 regressions. Fixes: #580020
2707 2009-04-29 11:18:42 +0200 Andy Wingo <wingo@oblong.net>
2709 * gst-libs/gst/audio/gstbaseaudiosink.c:
2710 Revert "add can-activate-pull property to baseaudiosink"
2711 This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.
2713 2009-04-29 11:18:33 +0200 Andy Wingo <wingo@oblong.net>
2715 * gst-libs/gst/audio/gstbaseaudiosink.c:
2716 Revert "[baseaudiosink] add docs for can-activate-pull"
2717 This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.
2719 2009-04-28 18:48:33 +0200 Andy Wingo <wingo@oblong.net>
2721 [baseaudiosink] add docs for can-activate-pull
2722 * gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
2725 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
2727 add can-activate-pull property to baseaudiosink
2728 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
2731 2009-04-28 11:32:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2733 * gst/videorate/gstvideorate.c:
2734 * gst/videorate/gstvideorate.h:
2735 videorate: clear discont on duplicated buffers
2736 When videorate duplicates a buffer with a DISCONT flag, it copies the discont on
2737 the first pushed buffer but fails to clear it for subsequent buffers. This
2738 causes theoraenc!oggmux and possibly other elements to consider this a discont
2740 Fix videorate to produce discont as the first buffer and after a flushing seek.
2743 2009-04-24 18:13:00 +0100 Jan Schmidt <thaytan@noraisin.net>
2745 * tests/check/Makefile.am:
2746 check: Disable the playbin2 for this release, as it is a bit racy.
2747 Disable the test, as per the discussion in #580120. Needs re-enabling
2748 after the release, when playbin2 is fixed.
2750 2009-04-23 08:41:19 +0200 Edward Hervey <bilboed@bilboed.com>
2752 * gst/playback/gstdecodebin2.c:
2753 decodebin2: Don't reduce max-size-time of exposed groups. Fixes #579912
2754 The 2s limit is way too small for a lot of files (which have an interleave
2755 in time of between 3 and 5s). Instead, leave it to the initial 5s value
2756 and reduce the other limits (allowing us to stay memory-efficient).
2758 2009-04-21 21:06:59 +0100 Jan Schmidt <thaytan@noraisin.net>
2790 * win32/common/_stdint.h:
2791 * win32/common/config.h:
2792 0.10.22.3 pre-release
2794 2009-04-21 20:41:23 +0100 René Stadler <mail@renestadler.de>
2796 * gst/audioresample/gstaudioresample.c:
2797 audioresample: Fix unused variable in compilation with --disable-gst-debug
2800 2009-04-21 22:12:28 +0100 Jan Schmidt <thaytan@noraisin.net>
2803 Automatic update of common submodule
2804 From b3941ea to 6ab11d1
2806 2009-04-21 20:57:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2808 * gst/playback/gstplaybasebin.c:
2809 playbin: only use raw_decoding_mode when it's true
2810 First check the pad caps if they are raw before setting the raw_decoding_mode to
2811 TRUE. Fixes playback of transport streams and other streams that require large
2815 2009-04-19 18:15:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2817 * gst-libs/gst/cdda/gstcddabasesrc.c:
2818 * tests/check/libs/cddabasesrc.c:
2819 cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core
2820 Don't use REPLACE_ALL merge mode when that's not really what we want,
2821 as now that REPLACE_ALL actually does what it's supposed to do in
2822 core, we drop tags we wanted to keep, such as the various disc id
2823 tags. Add unit test for this as well. Fixes #579463.
2825 2009-04-17 10:34:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2827 * gst-libs/gst/rtsp/gstrtspconnection.c:
2828 rtspconnection: don't use GLib-2.16 API, we require only 2.14
2831 2009-04-17 10:55:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2833 * gst-libs/gst/audio/gstbaseaudiosink.c:
2834 baseaudiosink: don't unparent the ringbuffer
2835 when going to NULL, don't unparent the ringbuffer because we don't support going
2836 back to 0 very well yet.
2839 2009-04-17 10:53:10 +0200 Olivier Crete <tester at tester.ca>
2841 * gst-libs/gst/rtp/gstrtcpbuffer.c:
2842 RTCP: don't fail when retrieving invalid PT
2843 We can't meaningfully assert on valid packet types so just return the type as it
2844 is. Update the comments to reflect this.
2847 2009-04-16 12:12:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2849 * docs/libs/gst-plugins-base-libs-sections.txt:
2850 * gst-libs/gst/app/gstappsink.h:
2851 * gst-libs/gst/app/gstappsrc.h:
2852 app: add trivial cast macros
2853 Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
2854 and add the macros to the standard macros in the docs.
2857 2009-04-16 12:09:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2859 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
2860 pkgconfig: add the app/ directory to Libs
2861 Add the appsrc/appsink directory to the Libs in the uninstalled
2862 pkgconfig file so that one can build against it.
2865 2009-04-15 22:59:31 +0100 Jan Schmidt <thaytan@noraisin.net>
2868 0.10.22.2 pre-release
2870 2009-04-15 22:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
2873 ChangeLog: regenerate changelog with the gen-changelog script
2875 2009-04-16 00:41:13 +0100 Jan Schmidt <thaytan@noraisin.net>
2906 po: Update po files from TP
2908 2009-04-16 00:40:59 +0100 Jan Schmidt <thaytan@noraisin.net>
2910 * win32/common/_stdint.h:
2911 * win32/common/config.h:
2912 * win32/common/gstrtsp-enumtypes.c:
2913 * win32/common/interfaces-enumtypes.c:
2914 * win32/common/interfaces-enumtypes.h:
2915 * win32/common/video-enumtypes.c:
2916 win32: Update win32 build files
2918 2009-04-16 00:31:55 +0100 Jan Schmidt <thaytan@noraisin.net>
2920 * tests/check/libs/video.c:
2921 check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.
2923 2009-04-16 00:31:00 +0100 Jan Schmidt <thaytan@noraisin.net>
2925 * tests/check/elements/playbin2.c:
2926 check: Fix the input uri in playbin2 test.
2927 Don't try and use a random file in wim's home directory as a test input
2929 2009-04-15 15:35:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2931 * gst-libs/gst/video/video.h:
2932 video: Fix typo in the docs
2934 2009-04-15 14:53:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2936 * gst-libs/gst/video/video.c:
2937 * gst-libs/gst/video/video.h:
2938 video: Add support for YVYU YUV colorspace
2940 2009-04-15 00:17:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2942 * docs/libs/gst-plugins-base-libs-docs.sgml:
2943 * gst-libs/gst/fft/gstfft.c:
2944 docs: fix hyperlink and move fft attribution to the right place
2946 2009-04-15 00:02:39 +0300 Stefan Kost <ensonic@users.sf.net>
2948 * gst-libs/gst/audio/gstbaseaudiosink.c:
2949 log: use G_GUINT64_FORMAT instead of llu
2951 2009-04-14 18:31:52 +0200 Josep Torra <n770galaxy at gmail.com>
2953 * gst-libs/gst/rtsp/gstrtspdefs.c:
2954 * gst-libs/gst/rtsp/gstrtspdefs.h:
2955 RTSP: add missing headers for WMS RTSP
2956 Add missing headers related to Windows Media RTSP extension.
2959 2009-04-14 18:16:37 +0200 Olivier Crete <tester at tester.ca>
2961 * docs/design/draft-keyframe-force.txt:
2962 * ext/theora/gsttheoraenc.h:
2963 * ext/theora/theoraenc.c:
2964 theoraenc: implement upstream keyframe force
2965 Implement handling of upstream keyframe forcing.
2966 Update the design documents too.
2969 2009-04-14 17:31:31 +0200 Olivier Crete <tester at tester.ca>
2971 * ext/theora/theoraenc.c:
2972 theoraenc: factor out keyframe forcing
2975 2009-04-14 17:01:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2978 * gst-libs/gst/fft/gstfft.c:
2979 Give credit to Mark Borgerding (kissfft author)
2980 and add myself to AUTHORS as well. Fixes #575638.
2982 2009-04-14 17:04:06 +0200 Jan Urbanski <j.urbanski at students.mimuw.edu.pl>
2984 * gst/tcp/gstmultifdsink.c:
2985 * gst/tcp/gstmultifdsink.h:
2986 multifdsink: add property to resend streamheaders
2987 Adds a new property in multifdsink, resend-streamheader.
2988 If this property is false, the multifdsink will not send the streamheader if
2989 there's already one set for a particular client.
2990 There are some formats in which every stream needs to start with a certain
2991 blob, but you can't inject this blob at leisure. If the producer wants to
2992 change the blob in question and sets in as the streamheader on the outgoing
2993 buffers' caps, new clients of multifdsink will get the new streamheader, but
2994 old clients will break, because they'll see the blob in the middle of the
2996 The property is true by default, so existing code will not see any difference.
2999 2009-04-14 16:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3001 * gst/tcp/gstmultifdsink.c:
3002 * gst/tcp/gstmultifdsink.h:
3003 multifdsink: add property to handle client write
3004 Add a property to disable listening to client writes. This property is usefull
3005 when other code will deal with reading from the client socket.
3006 API: GstMultiFdSink::handle-read property
3008 2009-04-14 16:45:20 +0200 Johann Prieur <johann.prieur at gmail.com>
3010 * docs/libs/gst-plugins-base-libs-sections.txt:
3011 * gst-libs/gst/rtp/gstrtcpbuffer.c:
3012 * gst-libs/gst/rtp/gstrtcpbuffer.h:
3013 * win32/common/libgstrtp.def:
3014 RTCP: add beginnings of Feedback messages
3015 Add the beginnings of parsing and constructing Feedback messages.
3018 2009-04-14 13:51:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3020 * gst/playback/gstplaysink.c:
3021 playbin2: clear the target
3022 Clear the target of our ghostpads before we remove the pad from the element.
3023 This to make sure that the internal pad is not left linked to whatever pad we
3024 were ghosted to. This should only be a problem when we leak the ghostpads.
3025 Also release our subpicture pads.
3028 2009-04-14 12:10:30 +0100 Hannes Bistry <hannesb@gmx.net>
3030 * sys/ximage/ximagesink.c:
3031 ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
3034 2009-04-14 13:16:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3036 * gst-libs/gst/audio/gstbaseaudiosrc.c:
3037 baseaudiosrc: adjust the internal timestamp
3038 Adjust the internal timestamp before comparing it against the adjusted clock
3042 2009-04-14 13:12:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3044 * gst-libs/gst/audio/gstbaseaudiosink.c:
3045 baseaudiosink: use new clock time methods
3046 Use the unadjusted internal clock times to calculate the internal/external
3047 offset when calibrating the clock.
3048 When going to NULL, unparent and free the ringbuffer, like we do in the source
3052 2009-04-14 13:08:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3054 * gst-libs/gst/audio/gstaudioclock.c:
3055 * gst-libs/gst/audio/gstaudioclock.h:
3056 * win32/common/libgstaudio.def:
3057 audioclock: add methods for the internal offset
3058 Add two methods for getting the unadjusted time of the clock and one for
3059 adjusting an internal time. We will need these methods for correctly handling
3060 the time after a gst_audio_clock_reset().
3061 Add a debug category and some debug lines to the audio clock.
3062 API: gst_audio_clock_get_time()
3063 API: gst_audio_clock_adjust()
3064 API: GST_AUDIO_CLOCK_CAST()
3066 2009-04-14 11:34:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3068 * gst/playback/gstdecodebin2.c:
3069 decodebin2: fix up the debugs and warnings
3070 Use _OBJECT variants because we can. Go over some log statements and put them in
3074 2009-04-12 22:26:33 +0200 Luca Ognibene <luca.ognibene at gmail.com>
3076 * gst/tcp/gstmultifdsink.c:
3077 multifdsink: fix error in sync-method
3078 Multifdsink did not handle sync-method=latest-keyframe correctly when the
3079 soft-limit is set to -1 (unlimited).
3082 2009-04-10 21:49:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3084 * gst-libs/gst/audio/gstbaseaudiosink.c:
3085 baseaudiosink: use the internal clock time
3086 We can't assume that the internal clock time is the same as the function we
3087 installed on our provided clock because somebody might have changed it.
3089 2009-04-10 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3091 * tests/examples/seek/seek.c:
3092 seek: handle clock-lost messages
3093 When we receive a clock-lost message we need to pause and play to select a new
3096 2009-04-10 13:44:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3098 * tests/check/Makefile.am:
3099 * tests/check/elements/playbin2.c:
3100 check: add a unit test for playbin2
3101 Add unit test for playbin2 and include the refcount test in #577794.
3103 2009-04-10 13:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3105 * gst/playback/gstplaysink.c:
3106 playbin2: fix refcounting of visualisations
3109 2009-04-10 13:27:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3111 * gst/playback/gstplaysink.c:
3112 playsink: fix refcounting of custom elements
3113 Sink the custom sinks, let other elements we create be sunken by the bin we add
3117 2009-04-10 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3119 * tests/check/elements/appsink.c:
3120 check: fix appsink test
3121 Fix the appsink test now that the method signature changed.
3123 2009-04-10 12:26:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3125 * gst/playback/gstplaybin2.c:
3126 playbin2: handle missing input-selector
3127 Gracefully degrade and disable stream selection when input-selector is
3130 2009-04-09 23:46:17 +0200 Martin Samuelsson <martin.samuelsson at axis.com>
3132 * gst-libs/gst/app/gstappsink.c:
3133 * gst-libs/gst/app/gstappsink.h:
3134 appsink: make callbacks return GstFlowReturn
3135 Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
3136 errors can be reported properly.
3139 2009-04-09 18:04:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3141 * gst-libs/gst/audio/gstringbuffer.c:
3142 * gst-libs/gst/audio/gstringbuffer.h:
3143 ringbuffer: allow for custom commit functions
3144 Allow subclasses to override the commit method.
3146 2009-04-08 18:04:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3148 * gst-libs/gst/audio/gstbaseaudiosink.c:
3149 baseaudiosink: fix a small glitch after pause
3150 After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
3151 the amount of output samples we consumed. We can't do this reliably with the
3152 current API when we are doing trick modes but we can do the right thing for
3155 2009-04-08 16:43:27 +0300 Stefan Kost <ensonic@users.sf.net>
3157 * gst/playback/gstplaysink.c:
3158 playbin2: better error message on sink failure
3159 If we could create the sinks, but the don't work, don't send the missing plugin
3160 message and report that the state-changed failed.
3162 2009-04-07 22:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
3164 * gst-libs/gst/audio/gstaudiofilter.c:
3165 audiofilter: don't leak pad-template
3166 gst_element_class_add_pad_template() does not take ownership.
3168 2009-04-04 21:18:38 +0300 Felipe Contreras <felipe.contreras@gmail.com>
3171 Automatic update of common submodule
3172 From d0ea89e to b3941ea
3174 2009-04-04 16:28:14 +0200 Edward Hervey <bilboed@bilboed.com>
3176 * gst-libs/gst/interfaces/navigation.c:
3177 * sys/v4l/v4lsrc_calls.c:
3178 navigation/v4l: Don't use g_return_val_if_fail for computed/used values.
3180 2009-03-22 09:46:37 +0100 Edward Hervey <bilboed@bilboed.com>
3182 * ext/theora/theoradec.c:
3183 theoradec: return GST_CLOCK_TIME_NONE for negative framecounts.
3184 This fixes most seeking issues when used with gnonlin.
3187 2009-04-04 14:53:42 +0200 Edward Hervey <bilboed@bilboed.com>
3190 Automatic update of common submodule
3191 From f8b3d91 to d0ea89e
3193 2009-04-03 10:51:42 -0700 Michael Smith <msmith@songbirdnest.com>
3195 * gst/playback/gstplaybin2.c:
3196 playbin2: don't leak selector when getting current stream numbers.
3198 2009-04-02 22:28:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3200 * gst-libs/gst/rtsp/gstrtspconnection.c:
3201 rtsp: use fully qualified urls when using a proxy
3202 Use a fully qualified url when specifying the url for tunneled requests through
3206 2009-03-31 00:54:30 +0100 Jan Schmidt <thaytan@noraisin.net>
3208 * docs/libs/gst-plugins-base-libs-sections.txt:
3209 * gst-libs/gst/interfaces/navigation.c:
3210 * gst-libs/gst/interfaces/navigation.h:
3211 * tests/check/Makefile.am:
3212 * tests/check/libs/.gitignore:
3213 * tests/check/libs/navigation.c:
3214 * win32/common/libgstinterfaces.def:
3215 navigation: Extend the navigation interface
3216 Add support for a set of standard commands that can be queried and executed to
3217 support applications like DVD. Add query construction and parsing functions.
3218 Add new messages that can be sent on the bus to provide notifications related
3219 to commands, multiangle changes, and button highlight activity.
3220 Add some helper functions to parse the existing GstNavigation events that
3221 elements might receive.
3222 Document it all and add unit tests.
3224 2009-02-04 17:03:07 +0000 Jan Schmidt <thaytan@noraisin.net>
3226 * gst/playback/gstplaybasebin.c:
3227 * gst/playback/gstplaybasebin.h:
3228 playbin: Add simple 'raw decoding mode'.
3229 Raw decoding mode removes almost all buffering in video and audio queues
3230 when a source providing already decoded video/audio is detected, on the
3231 possibly bogus assumption that such a source should provide sufficient
3232 internal queueing. Fixes playback on some DVDs, and improves it
3235 2009-04-02 09:27:07 +0100 Jan Schmidt <thaytan@noraisin.net>
3237 * tests/check/elements/.gitignore:
3238 ignores: Ignore the videoscale check binary
3240 2009-04-02 12:13:57 +0100 Jan Schmidt <thaytan@noraisin.net>
3242 * win32/common/libgstrtsp.def:
3243 win32: Add gst_rtsp_connection_set_proxy to the win32 exports
3245 2009-04-02 10:42:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3247 * ext/alsa/gstalsamixer.c:
3248 alsamixer: don't forget to release locks in a few places
3251 2009-04-02 11:10:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3253 * gst/videoscale/vs_4tap.c:
3254 videoscale: Don't read over line ends when taking the last Cr or Cb
3256 2009-04-02 10:52:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3258 * gst/videoscale/vs_4tap.c:
3259 videoscale: Don't write to few pixels and don't mix Cr and Cb
3262 2009-04-01 15:15:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3264 * gst/audioresample/gstaudioresample.c:
3265 * tests/check/elements/audioresample.c:
3266 audioresample: fix negotiation so that upstream can actually fixate to downstream's rate
3267 If one side has a preference for a particular sample rate or set of sample rates, we
3268 should honour this in the caps we advertise and transform to and from, so that elements
3269 actually know about the other side's sample rate preference and can negotiate to it
3270 if supported. Also add unit test for this.
3272 2009-03-26 19:34:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3274 * gst/playback/gstplaybin2.c:
3275 docs: add a blurb about redirect messages to playbin2 docs
3277 2009-04-01 09:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3279 * gst-libs/gst/rtsp/gstrtspconnection.c:
3280 rtsp: fix little typo in the comments
3282 2009-03-31 17:52:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3284 * gst-libs/gst/rtsp/gstrtspconnection.c:
3285 rtspconnection: make gst_rtsp_watch_queue_message() thread-safe
3286 People might queue messages from a thread other than the thread in which
3287 the main context which this watch is attached is iterated from, so use
3288 a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
3289 over list nodes just freed in the other thread. This just fixes issues
3290 I've had with gst-rtsp-server. We might need more locking in various
3293 2009-03-31 18:13:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3295 * gst-libs/gst/rtsp/gstrtspconnection.c:
3296 * gst-libs/gst/rtsp/gstrtspmessage.c:
3297 rtsp: clear the entire builder structure
3298 And use structure instead of variable with sizeof when
3299 clearing the rtsp message structure, for clarity.
3301 2009-03-31 17:56:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3303 * gst-libs/gst/rtsp/gstrtspmessage.c:
3304 docs: fix typo in gst_rtsp_message_unset() API docs
3306 2009-03-31 19:00:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3308 * gst-libs/gst/rtsp/gstrtspconnection.c:
3309 * gst-libs/gst/rtsp/gstrtspconnection.h:
3310 rtsp: add support for proxies
3311 Add suport for proxy servers. Currently only used for tunneled HTTP
3312 connections without authentication.
3314 2009-03-31 18:57:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3316 * gst-libs/gst/rtsp/gstrtspmessage.c:
3317 Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
3318 This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.
3320 2009-03-26 18:54:56 +0200 Stefan Kost <ensonic@users.sf.net>
3322 * sys/xvimage/xvimagesink.c:
3323 xvimagesink: use xcontext->depth instead of bits in attr.max_value for colorkey
3324 According to the drivers in http://cgit.freedesktop.org/xorg/driver/ we should
3325 format the colorkey depending on xcontext->depth. This is what they will use to
3326 interprete the value. The max_value in turn is usualy a constant regardless of
3329 2009-03-31 12:22:14 +0300 Stefan Kost <ensonic@users.sf.net>
3331 * gst-libs/gst/rtsp/gstrtspmessage.c:
3332 rtsp: reset whole message (was sizeof pointer instead of sizeof type)
3334 2009-03-31 00:56:18 +0100 Jan Schmidt <thaytan@noraisin.net>
3336 * gst-libs/gst/interfaces/mixer.c:
3337 doc: Fix a typo in the GstMixer docs
3339 2009-03-29 12:01:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3341 * gst/videoscale/vs_scanline.c:
3342 videoscale: Fix linear scaling for one byte components
3345 2009-03-29 11:53:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3347 * gst/videoscale/vs_4tap.c:
3348 videoscale: Fix 4tap scaling of YUYV and friends
3350 2009-03-28 16:08:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3352 * gst/videoscale/vs_image.c:
3353 * gst/videoscale/vs_scanline.c:
3354 * gst/videoscale/vs_scanline.h:
3355 videoscale: Rewrite YUYV (and friends) scaling and don't read/write over line ends
3356 Partially fixes bug #577054, there's just one issue left now.
3358 2009-03-28 12:48:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3360 * tests/check/elements/videoscale.c:
3361 videoscale: Add some more unit tests
3363 2009-03-28 11:51:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3365 * gst/videoscale/gstvideoscale.c:
3366 videoscale: Use bilinear instead of 4tap scaling for heights < 4
3367 Partially fixes bug #577054.
3369 2009-03-28 11:45:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3371 * gst/videoscale/vs_scanline.c:
3372 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY/RGB/RGBA
3373 This case is for upscaling a frame with width=1
3374 Partially fixes bug #577054.
3376 2009-03-28 11:27:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3378 * gst/videoscale/vs_scanline.c:
3379 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
3380 Partially fixes bug #577054.
3382 2009-03-28 10:40:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3384 * gst/videotestsrc/gstvideotestsrc.c:
3385 videotestsrc: Initialize buffer memory with zeroes
3386 This prevents valgrind warnings when accessing the "x" parts
3387 of xRGB and friends in other elements that handle (and can handle)
3388 xRGB like ARGB (for example videoscale).
3390 2009-03-28 10:25:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3392 * tests/check/Makefile.am:
3393 * tests/check/elements/videoscale.c:
3394 videoscale: Add a lot of unit tests
3396 2009-03-28 10:06:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3398 * gst/videoscale/gstvideoscale.c:
3399 videocale: Add support for video/x-raw-gray with bpp=depth=8
3401 2009-03-28 10:01:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3403 * gst/videotestsrc/videotestsrc.c:
3404 videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8
3406 2009-03-28 09:43:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3408 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
3409 ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format
3411 2009-03-27 19:12:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3413 * gst/videoscale/vs_4tap.c:
3414 videoscale: Take the next luma value instead of every second next when scaling UYVY and friends
3416 2009-03-27 19:09:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3418 * gst/videoscale/gstvideoscale.c:
3419 videoscale: Add support for v308 YUV colorspace
3421 2009-03-27 13:15:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3423 * gst/videoscale/vs_4tap.c:
3424 videoscale: Add my copyright to the 4tap scalers
3426 2009-03-27 13:14:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3428 * gst/videoscale/gstvideoscale.c:
3429 videoscale: Enable 4-tap scaling for all supported formats
3431 2009-03-27 13:14:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3433 * gst/videoscale/vs_4tap.c:
3434 * gst/videoscale/vs_4tap.h:
3435 videoscale: Implement 4-tap scaling for RGB565 and RGB555
3437 2009-03-27 10:47:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3439 * gst/videoscale/vs_4tap.c:
3440 * gst/videoscale/vs_4tap.h:
3441 videoscale: Implement 4-tap scaling for UYVY
3443 2009-03-27 09:33:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3445 * gst/videoscale/vs_4tap.c:
3446 * gst/videoscale/vs_4tap.h:
3447 videoscale: Implement 4-tap scaling for YUY2 and YVYU
3449 2009-03-26 22:14:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3451 * gst/videoscale/vs_4tap.c:
3452 * gst/videoscale/vs_4tap.h:
3453 videoscale: Implement 4-tap scaling for RGB and BGR
3455 2009-03-26 22:08:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3457 * gst/videoscale/vs_4tap.c:
3458 * gst/videoscale/vs_4tap.h:
3459 videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats
3461 2009-03-26 11:02:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3463 * ext/pango/gsttextoverlay.c:
3464 textoverlay: Fix drawing of UYVY text borders
3466 2009-03-26 10:36:27 +0100 Zeeshan Ali <zeeshan.ali@nokia.com>
3468 * ext/pango/gsttextoverlay.c:
3469 * ext/pango/gsttextoverlay.h:
3470 textoverlay: Add support for UYVY colorspace
3473 2009-03-25 19:01:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3475 * gst/playback/gstdecodebin2.c:
3476 decodebin2: do some more cleanup
3477 Free the groups when we go to READY.
3478 Allow for NO_PREROLL elements.
3480 2009-03-25 16:37:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3482 * gst-libs/gst/rtsp/gstrtspconnection.c:
3483 rtsp: start CSeq counting from 1 instead of 0
3484 Start counting from 1 instead of 0 as this is what most other clients
3487 2009-03-25 16:35:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3489 * gst-libs/gst/rtsp/gstrtspdefs.c:
3490 * gst-libs/gst/rtsp/gstrtspdefs.h:
3491 rtsp: add ETag and If-Match headers
3492 Add new headers, we need them for RealMedia support.
3494 2009-03-25 14:16:25 +0200 Stefan Kost <ensonic@users.sf.net>
3496 * sys/xvimage/xvimagesink.c:
3497 xvimagesink: scale the colorkey components in case of 16bit visuals
3498 Use a default that won't be scales to 0,0,0
3500 2009-03-25 11:27:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3502 * gst-libs/gst/audio/gstbaseaudiosrc.c:
3503 audiosrc: improve 'Dropped n samples' warning message
3505 2009-03-24 19:41:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3507 * tests/examples/app/appsrc-ra.c:
3508 * tests/examples/app/appsrc-seekable.c:
3509 examples: use new method to set flags
3510 Use the new core method for setting object enum properties by name.
3512 2009-03-24 18:29:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3514 * gst/playback/gstplaysink.c:
3515 * gst/playback/gstplaysink.h:
3516 playbin2: add more support for subpictures
3518 2009-03-24 17:12:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3520 * gst/playback/gstplaybin2.c:
3521 * gst/playback/gstplaysink.c:
3522 * gst/playback/gstplaysink.h:
3523 playbin2: first support for subpictures
3524 Add beginnings of subpicture support.
3526 2009-03-24 15:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3528 * tests/examples/seek/seek.c:
3529 seek: print tags from the different tracks
3531 2009-03-24 12:22:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3533 * gst/playback/gstplaybin2.c:
3534 playbin2: blacklist subpictures for now
3535 Blacklist the subpictures until we add support for them.
3536 Add some small debug info.
3539 2009-03-24 12:19:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3541 * gst/playback/gsturidecodebin.c:
3542 uridecodebin: expose more media types
3543 Expose more media types from a raw source, such as the subpicture and various
3545 Small cleanups and add some more debugging.
3548 2009-03-24 10:42:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3550 * gst/playback/gstplaysink.c:
3551 playbin2: rescan audio sinks for volume/mute
3552 Rescan the audio sinks for the mute and volume properties.
3555 2009-03-23 19:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3557 * gst/playback/gstplaysink.c:
3558 playbin2: fix reuse of the video chains
3559 When reusing playbin with visualisations, reset the async property on the video
3560 sink because some sinks might dynamically recreate their sinks.
3563 2009-03-23 17:37:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3565 * gst/playback/gstplaysink.c:
3566 playbin2: allow dynamic swtiching of subtitles
3567 When we have the textpad configured, enable and disable the subtitles by setting
3568 the silent flag on the overlay element instead of trying to remove elements.
3571 2009-03-23 16:59:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3573 * tests/icles/playbin-text.c:
3574 tests: print some more info in the text example
3575 Print both the position and the running_time when the subtitle becomes available
3578 2009-03-23 16:04:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3580 * gst/playback/gstplaysink.c:
3581 playbin2: fix dynamic switching of visualisations
3582 Fix the switching of visualisations by requesting and releasing the tee request
3586 2009-03-23 16:19:11 +0200 Stefan Kost <ensonic@users.sf.net>
3589 * gst/tcp/gsttcpclientsink.c:
3590 * gst/tcp/gsttcpclientsrc.c:
3591 * gst/tcp/gsttcpserversink.c:
3592 * gst/tcp/gsttcpserversrc.c:
3593 docs: add examples for tcp elements, also use correct section name. Fixes #564139
3594 Updated the examples in the README to actually work. Add them to api docs. Tests
3595 the api-docs and fix the section names to make the docs actualy show up.
3596 The example for "tcpserversrc" needs review (might be an element bug).
3598 2009-03-17 09:14:02 +0200 Stefan Kost <ensonic@users.sf.net>
3600 * gst/videoscale/gstvideoscale.c:
3601 indent: fix damange that gst-indent did some time ago
3603 2009-03-23 15:27:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3605 * gst/playback/gstplaysink.c:
3606 playbin2: fix linking order
3607 Link after doing the state change and unlink before shutting down. Makes the
3608 window for causing races in toggling the visualisations smaller.
3611 2009-03-23 12:26:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3613 * gst/playback/gsturidecodebin.c:
3614 uridecodebin: reset counter
3615 reset the number of pending dynamic operations back to 0 when we reuse
3619 2009-03-23 11:38:53 +0100 Edward Hervey <bilboed@bilboed.com>
3621 * ext/theora/theoradec.c:
3622 theoradec: Use GST_CLOCK_TIME_NONE for invalid positions. Fixes #543591
3623 The problem was that previously we didn't check whether _theora_granule_frame
3624 returned a negative framecount or not, resulting in bogus timestamps.
3626 2009-03-21 09:46:28 +0100 René Stadler <mail@renestadler.de>
3628 * ext/vorbis/vorbisenc.c:
3629 vorbisenc: Set caps on non-header ouput buffers.
3632 2009-03-20 16:13:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3634 * tests/examples/seek/seek.c:
3635 seek: Add some more debug
3636 Add some more info about the selected streams.
3638 2009-03-20 15:47:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3640 * gst/playback/gstdecodebin2.c:
3641 decodebin2: a pad starts out being not drained.
3642 Mark a new pad as not drained until we get EOS on it.
3644 2009-03-20 14:17:19 +0100 LRN <lrn1986 at gmail dot com>
3646 * gst/playback/gstqueue2.c:
3647 win32: fix seeking in large files
3648 Fix Seeking in large files by using the 64-bit seek functions.
3651 2009-03-19 20:31:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3653 * gst/playback/gstdecodebin2.c:
3654 decodebin2: recover from failing to add a pad
3655 When we cannot add a pad to the decodebin2 for some reason, print a warning but
3656 continue adding the remaining pads.
3658 2009-03-19 19:35:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3660 * gst/playback/gstdecodebin2.c:
3661 decodebin2: more cleanups and docs.
3662 Add some more comments and use g_list_prepend().
3664 2009-03-19 19:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3666 * gst/playback/gstdecodebin2.c:
3667 decodebin2: refactoring and race fixes
3668 Refactor some code so that we can take the right locks and in the right order.
3669 Fixes quite a bit of races already.
3671 2009-03-19 19:03:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3673 * gst/playback/gstplaybin2.c:
3674 playbin2: remove the group cond + cleanups
3675 Remove the group GCond that we used for waiting for groups to finish because we
3676 use pad blocking on the selectors and counters instead for waiting for the
3678 remove the obsolete about_to_finish variable set while emiting the
3679 about-to-finish signal and fix some old comments.
3680 We don't need to take the playbin lock when querying the uridecodebin.
3682 2009-03-18 10:45:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3684 * tests/icles/playbin-text.c:
3685 icles: print better error and warning messages
3688 2009-03-17 22:53:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3690 * gst-libs/gst/rtsp/gstrtspbase64.c:
3691 * gst-libs/gst/rtsp/gstrtspbase64.h:
3692 rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode
3693 This also fixes another instance of CVE-2008-4316.
3695 2009-03-17 19:53:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3697 * ext/ogg/gstoggdemux.c:
3698 oggdemux: report -1 for duration in push mode
3699 In push mode we must return TRUE from the duration query with a value of -1
3700 meaning that we know that we don't know the duration.
3702 2009-03-17 19:09:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3704 * gst/playback/gstdecodebin2.c:
3705 decodebin2: add extra dynamic ref for demuxers
3706 When we make a group connected to a demuxer, keep an extra dynamic refcount for
3707 the group which is only decremented when no_more_pads or a multiqueue overrun is
3708 detected. This way we avoid a race between exposing the group while more dynamic
3709 refs are added from new pads.
3712 2009-03-17 15:39:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3714 * gst/playback/gstplaysink.c:
3715 playbin2: sync state of the sink correctly
3716 Sync the state of the newly added chains to the state of the parent sink element
3717 to avoid lost async-start messages. Fixes cdda:// async-done message storm.
3719 2009-03-17 11:54:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3721 * gst/playback/gstplaybin2.c:
3722 playbin2: return NOT_LINKED for unselected streams
3723 When streams are not selected in the selector, return NOT_LINKED so that
3724 upstream elements can skip decoding. Only do this for audio and video pads
3725 because for text streams the overhead is smaller and they could come from
3728 2009-03-17 11:51:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3730 * gst/playback/gstplaysink.c:
3731 playbin: set custom text sink properties
3732 Set the custom sink async=FALSE to not make it participate in preroll because we
3733 are dealing with sparse streams.
3734 Try to set sync=TRUE on the custom text sink.
3736 2009-03-17 11:30:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3738 * tests/icles/playbin-text.c:
3739 example: use appsink instead of fakesink
3740 Use appsink instead of fakesink to get the subtitles.
3741 Make things more pretty.
3743 2009-03-17 11:24:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3745 * tests/icles/.gitignore:
3746 * tests/icles/Makefile.am:
3747 * tests/icles/playbin-text.c:
3748 examples: add example of intercepting subtitles
3749 Add an example of how to install a custom sink for receiving subtitles in
3752 2009-03-17 11:03:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3754 * tests/check/elements/appsink.c:
3755 tests: fix include in the appsink test
3756 Fix dist by doing the right include.
3758 2009-03-16 16:42:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3760 * gst/playback/gstplaybin2.c:
3761 playbin2: don't try to set invalid stream numbers
3762 Fix a problem with setting the stream numbers because we check for the wrong
3766 2009-03-16 16:16:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3768 * gst/playback/gstplaybin2.c:
3769 playbin2: release the shutdown lock
3770 Release the shutdown lock when we wait for other groups to complete or else we
3771 have a deadlock when the other group completes and tries to grab the shutdown
3775 2009-03-16 15:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3777 * tests/examples/app/appsrc-ra.c:
3778 * tests/examples/app/appsrc-seekable.c:
3779 * tests/examples/app/appsrc-stream.c:
3780 * tests/examples/app/appsrc-stream2.c:
3781 examples: fix g_object_set() value type.
3782 Make sure we cast the length value as a gint64 to the vararg g_object_set() just
3783 incase sizeof(gsize) != sizeof(gint64).
3785 2009-03-15 19:57:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3787 * gst/typefind/gsttypefindfunctions.c:
3788 typefinding: make flac typefinder return lower probability for frame headers
3789 The flac frame header typefinder overstates the likelihood of a match, leading
3790 to false positives with e.g. aac streams and PDF files. Reduce probabilty
3791 returned from LIKELY to POSSIBLE for the frame header matchin code.
3794 2009-03-11 12:59:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3796 * gst/typefind/gsttypefindfunctions.c:
3797 typefinding: improve image/bmp typefinder
3798 Detect more variations and also bail out in more cases where the values
3799 don't make sense. Furthermore, add width/height and bpp to the caps,
3802 2009-03-13 15:22:42 +0000 Jan Schmidt <thaytan@noraisin.net>
3804 * tests/check/Makefile.am:
3805 check: Ignore alsamixer in the states test too
3807 2009-03-13 15:22:11 +0000 Jan Schmidt <thaytan@noraisin.net>
3809 * sys/v4l/v4l_calls.c:
3810 v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.
3812 2009-03-13 16:19:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3814 * gst-libs/gst/rtsp/gstrtspconnection.c:
3815 rtsp: fix resolving of hostnames
3816 We were returning a pointer to a stack variable with the resolved hostname,
3818 return a copy of the resolved ip address instead.
3821 2009-03-13 15:29:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3823 * ext/vorbis/vorbisparse.c:
3824 vorbisparse: be smarter when queueing headers
3825 Look at the first buffer byte to see if a buffer is a header instead of counting
3828 2009-03-13 15:27:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3830 * ext/theora/gsttheoraparse.h:
3831 * ext/theora/theoraparse.c:
3832 theoraparse: be smarter when queuing headers
3833 Look at the first byte of the buffer data (if we can) to decide if the packet is
3834 a header packet or not instead of counting packets.
3836 2009-03-13 15:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3838 * ext/ogg/gstoggdemux.c:
3839 oggdemux: add some debug info
3840 Add some debug info to log when the seek worked.
3842 2009-03-13 15:14:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3844 * gst-libs/gst/app/gstappsrc.c:
3845 appsrc: release lock in _eos flushing case
3846 Release the mutex when we are flushing in gst_app_src_end_of_stream()
3849 2009-03-13 11:49:10 +0000 Jan Schmidt <thaytan@noraisin.net>
3851 * ext/vorbis/vorbisdec.c:
3852 vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.
3854 2009-03-13 11:48:28 +0000 Jan Schmidt <thaytan@noraisin.net>
3856 * ext/theora/theoradec.c:
3857 theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.
3859 2009-03-12 18:27:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3861 * gst/playback/gsturidecodebin.c:
3862 playbin2: fix raw elements like cdda://
3863 Fix a fixme with a one liner and make cd playback work again.
3865 2009-03-12 17:47:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3867 * gst/playback/gstplaybin2.c:
3868 * gst/playback/gstplaysink.c:
3869 * gst/playback/gstplaysink.h:
3870 playbin2: improve subtitle handling
3871 Add property to playbin2 to configure a custom sink that receives the raw
3872 subtitle buffers instead of using a textoverlay.
3873 Improve the property finding code to make it more usable.
3874 Use property find code to find async properties in custom sinks that are bins.
3875 Improve text overlay code to gracefully handle missing elements.
3877 2009-02-24 15:58:42 +0000 Jan Schmidt <thaytan@noraisin.net>
3879 * gst-libs/gst/tag/gstvorbistag.c:
3880 vorbistag: Protect memory allocation calculation from overflow.
3881 Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
3883 2009-03-12 11:34:20 +0000 Jan Urbanski <jurbanski@flumotion.com>
3885 * gst-plugins-base.spec.in:
3888 2009-03-11 18:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3890 * gst-libs/gst/rtsp/gstrtspconnection.c:
3891 rtsp: fix parsing of the timeout parameter
3894 2009-03-11 16:20:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3896 * gst-libs/gst/rtsp/gstrtspmessage.c:
3897 rtsp: fix g_return condition
3898 when parsing a data message, we require a data message.
3900 2009-03-11 13:33:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3902 * gst/typefind/gsttypefindfunctions.c:
3903 typefinding: flac typefinder fixes
3904 Use scan context for initial peek as well. Peek 6 bytes in the initial
3905 peek rather than 5 bytes, to match the length of the memcmp we're doing
3906 on that data later. Return immediately when we found caps from looking
3907 at the beginning of the data - no point in continuing to scan the next
3908 64kB for something matching a frame header.
3910 2009-03-11 14:08:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3912 * gst-libs/gst/rtsp/gstrtspmessage.c:
3913 rtsp: free the right string.
3914 Free the key value before we remove the header item from the array. The item we
3915 retrieved from the array is only valid until we remove it from the array.
3917 2009-03-11 14:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3919 * gst-libs/gst/rtsp/gstrtspconnection.c:
3920 rtsp: keep track of amount of decoded bytes
3921 Keep track of the actual amount of decoded bytes, which can be less than 3 when
3922 we decode the last bits of a base64 message.
3924 2009-03-10 21:00:26 +0200 Stefan Kost <ensonic@users.sf.net>
3926 * gst/adder/gstadder.c:
3927 adder: log details in getcaps like in setcaps
3929 2009-03-10 13:11:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3932 win32: update MANIFEST, fixing 'make dist'
3934 2009-03-09 23:12:00 +0000 Jan Schmidt <thaytan@noraisin.net>
3937 Automatic update of common submodule
3938 From 7032163 to f8b3d91
3940 2009-03-09 16:19:40 +0100 Jonathan Matthew <notverysmart at gmail dot com>
3942 * gst/typefind/gsttypefindfunctions.c:
3943 typefind: add photoshop typefind functions
3944 Add photoshop typefind functions.
3947 2009-03-09 15:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3949 * gst/playback/gstdecodebin2.c:
3950 decodebin2: only remove pads that were added
3951 Flag pads that were added so that we can see if we need to remove them later or
3954 2009-03-09 13:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3956 * gst-libs/gst/rtsp/gstrtsptransport.c:
3957 rtsp: only add ports when not using TCP
3958 Only add the port numbers in the transport string when we are using udp or
3961 2009-03-09 13:53:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3963 * gst-libs/gst/rtsp/gstrtspmessage.c:
3964 rtsp: use gstreamer dump mem
3967 2009-03-09 13:51:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3969 * gst-libs/gst/rtsp/gstrtspconnection.c:
3970 rtsp: use glib base64 encoder
3973 2009-03-06 19:28:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
3975 * gst/playback/gstdecodebin2.c:
3976 Unblock blocked ghostpads when shutting down. Fixes #574293.
3978 2009-03-09 10:03:13 +0100 Edward Hervey <bilboed@bilboed.com>
3980 * gst-libs/gst/riff/riff-media.c:
3981 Riff: Add mapping for Fraps video codec.
3982 Found through insanity testrun. Confirmed mapping in libavformat.
3984 2009-03-09 09:07:13 +0100 Edward Hervey <bilboed@bilboed.com>
3986 * gst-libs/gst/riff/riff-media.c:
3987 riff: Add the 'DVR ' mapping for mpeg2video.
3988 Found this in 3 files from the insanity suite and mapping is also present
3991 2009-03-09 09:06:40 +0100 Edward Hervey <bilboed@bilboed.com>
3993 * gst/typefind/gsttypefindfunctions.c:
3994 typefind: Use the proper data pointer instead of poking random memory.
3996 2009-03-08 18:17:48 +0100 LRN <lrn1986@gmail.com>
3998 * gst-libs/gst/rtsp/gstrtspconnection.c:
3999 rtsp: fix compilation on windows.
4000 Remove unused variable when building for windows.
4003 2009-03-08 12:03:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4006 Automatic update of common submodule
4007 From ffa738d to 7032163
4009 2009-03-08 11:19:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4012 Automatic update of common submodule
4013 From 3f13e4e to ffa738d
4015 2009-03-07 11:44:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4018 Automatic update of common submodule
4019 From 3c7456b to 3f13e4e
4021 2009-03-07 10:44:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4024 Automatic update of common submodule
4025 From 57c83f2 to 3c7456b
4027 2009-03-06 19:02:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4029 * ext/theora/theoradec.c:
4030 theoradec: parse and use codec_data in the caps
4031 Parse the codec_data in the caps and use this as the headers.
4034 2009-03-06 18:53:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4036 * gst-libs/gst/riff/riff-media.c:
4037 riff: add theora mapping
4038 Add theora mappings. See #574169.
4040 2009-03-06 16:31:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4042 * gst-libs/gst/rtsp/gstrtspconnection.c:
4043 * gst-libs/gst/rtsp/gstrtspconnection.h:
4044 * win32/common/libgstrtsp.def:
4045 rtsp: Add methods for getting the read/write fds
4046 API:gst_rtsp_connection_get_readfd()
4047 API:gst_rtsp_connection_get_writefd()
4049 2009-03-06 10:35:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4052 * win32/common/audio-enumtypes.c:
4053 win32: indent copied *-enumtypes.c files in make win32-update
4055 2009-03-06 10:35:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4058 win32: update MANIFEST
4060 2009-03-06 10:30:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4063 * win32/common/config.h:
4064 win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define
4066 2009-03-06 10:05:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4068 * win32/common/_stdint.h:
4069 * win32/common/config.h:
4070 * win32/common/gstrtsp-enumtypes.c:
4071 * win32/common/interfaces-enumtypes.c:
4072 * win32/common/multichannel-enumtypes.c:
4073 * win32/common/pbutils-enumtypes.c:
4074 * win32/common/video-enumtypes.c:
4075 * win32/common/video-enumtypes.h:
4076 win32: update windows files via make win32-update
4077 Updates win32 files using the new system/hook, and defines HAVE_PROCESS_H,
4078 which fixes the build of pbutils on windows (#574319).
4080 2009-03-06 10:03:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4083 gitignore: ignore more
4085 2009-03-06 10:37:38 +0100 Julien Moutte <julien@fluendo.com>
4087 * gst-libs/gst/rtsp/gstrtspconnection.c:
4088 Fix build on Mac OS X
4090 2009-03-05 15:42:23 -0800 Michael Smith <msmith@songbirdnest.com>
4092 * gst/playback/gstdecodebin2.c:
4093 decodebin2: don't stay connected to notify::caps after negotiation
4094 Disconnect the notify::caps signal in our callback (it'll be re-added
4095 if we're not, in fact, finished getting complete caps). Ensures that
4096 caps changes mid-stream (e.g. from an mp3 that changes from
4097 stereo->mono mid-file) don't cause us to try to add a new pad.
4099 2009-03-05 13:48:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4101 * gst-libs/gst/rtsp/gstrtsprange.c:
4102 rtsp: fix parsing of 'now-' ranges.
4105 2009-03-05 12:43:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4107 * tests/examples/dynamic/.gitignore:
4108 * tests/examples/dynamic/Makefile.am:
4109 * tests/examples/dynamic/sprinkle.c:
4110 * tests/examples/dynamic/sprinkle2.c:
4111 * tests/examples/dynamic/sprinkle3.c:
4112 examples: add some more sprinkle examples
4113 Add some more sprinle examples and add some more comments.
4116 2009-03-05 11:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4118 * docs/plugins/gst-plugins-base-plugins-sections.txt:
4119 docs: add appsrc symbols to standard section
4122 2009-03-05 12:27:16 +0200 Stefan Kost <ensonic@users.sf.net>
4124 * gst/adder/gstadder.c:
4125 adder: add variants for unsigned to fix warnings for unneeded check
4126 For unsigned int out+in can't be < 0.
4128 2009-03-05 10:58:12 +0200 Stefan Kost <ensonic@users.sf.net>
4130 * gst/subparse/gstsubparse.c:
4131 subparse: use the right variable in debug log, encoding is not yet initialized
4133 2009-03-05 10:51:25 +0200 Stefan Kost <ensonic@users.sf.net>
4135 * sys/v4l/v4l_calls.c:
4136 v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix
4138 2009-03-05 10:39:33 +0200 Stefan Kost <ensonic@users.sf.net>
4140 * gst/audioresample/gstaudioresample.c:
4141 audioresample: add missing break in event handling, remove dead code
4143 2009-03-04 16:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4145 * gst-libs/gst/rtsp/gstrtspconnection.c:
4146 rtsp: do some more cleanup in _close
4147 Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
4148 unconnected state as it was allocated.
4150 2009-03-04 16:11:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4152 * gst-libs/gst/rtsp/gstrtspconnection.c:
4153 * gst-libs/gst/rtsp/gstrtspconnection.h:
4154 rtsp: fix the memory management of the url
4155 Constify the url parameter in _create.
4156 Make a copy of the url stored in the connection.
4157 Free the url when the connection is freed.
4159 2009-03-04 12:21:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4161 * docs/libs/gst-plugins-base-libs-sections.txt:
4162 * gst-libs/gst/rtsp/gstrtspconnection.c:
4163 * gst-libs/gst/rtsp/gstrtspconnection.h:
4164 * win32/common/libgstrtsp.def:
4165 RTSP: Add support for server tunneling
4166 Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
4167 that a server can store and match the id against other tunnel requests.
4168 Fix the URI in the tunnel requests so that they contain the absolute uri and the
4169 query string if any instead of just the hostname.
4170 Transparently base64 decode the input stream when tunneling.
4171 Add method to set the connection ip address so that it can be included in the
4173 Add method to connect the two tunnel requests.
4174 Add two callbacks for the async mode to notify a tunnel start and tunnel
4176 Add method to reset the watch after the connection has been tunneled.
4177 Various little refactoring to make more stuff reusable.
4178 API: RTSP::gst_rtsp_connection_set_ip()
4179 API: RTSP::gst_rtsp_connection_get_tunnelid()
4180 API: RTSP::gst_rtsp_connection_do_tunnel()
4181 API: RTSP::gst_rtsp_watch_reset()
4183 2009-03-04 12:18:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4185 * gst-libs/gst/rtsp/gstrtspdefs.c:
4186 * gst-libs/gst/rtsp/gstrtspdefs.h:
4187 rtsp: add new defines for tunneling
4188 Add two more result codes for tunneling support.
4190 2009-03-04 12:12:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4192 * gst-libs/gst/rtsp/gstrtspmessage.h:
4193 rtsp: remove , from last enum member
4194 Remove , from last enum member to improve compatibility with other compilers.
4196 2009-02-28 15:23:20 -0800 LRN <lrn1986@gmail.com>
4198 * gst/subparse/gstsubparse.c:
4199 subparse: Convert regex code to GRegex code
4200 Fixes: #572993. Patch author prefers to use an alias, contact
4201 ds if you actually need a real name.
4202 Signed-off-by: David Schleef <ds@schleef.org>
4204 2009-03-02 16:13:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4206 * gst-libs/gst/rtsp/gstrtspconnection.c:
4207 rtsp: remove debugging g_message
4210 2009-03-02 16:03:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4212 * docs/libs/gst-plugins-base-libs-sections.txt:
4213 * gst-libs/gst/rtsp/gstrtspconnection.c:
4214 * gst-libs/gst/rtsp/gstrtspconnection.h:
4215 * win32/common/libgstrtsp.def:
4216 RTSP: add support for Quicktime tunneled RTSP
4217 Add support for tunneling RTSP over HTTP.
4218 Fix documentation some more.
4220 API: RTSP:gst_rtsp_connection_is_tunneled()
4221 API: RTSP:gst_rtsp_connection_set_tunneled()
4223 2009-03-02 15:48:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4225 * gst-libs/gst/rtsp/gstrtsptransport.h:
4226 * gst-libs/gst/rtsp/gstrtspurl.c:
4227 RTSP: parse rtsph uris as RTSP tunneled over HTTP
4228 Add transport define for RTSP tunneled over HTTP.
4229 Parse rtsph:// uris as tunneled HTTP over TCP.
4230 API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
4233 2009-03-02 12:48:18 +0100 Edward Hervey <bilboed@bilboed.com>
4235 * win32/common/libgstrtsp.def:
4236 win32: Add gst_rtsp_connection_get_url definition
4237 No, I'm not wim's buildslave, seriously.
4239 2009-03-02 10:58:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4241 * gst-libs/gst/rtsp/gstrtspconnection.c:
4242 * gst-libs/gst/rtsp/gstrtspconnection.h:
4243 rtsp: add _get_url method and separate sockets
4244 Add gst_rtsp_connection_get_url() method.
4245 Reserve space for 2 sockets, one for reading and one for writing. Use socket
4246 pointers to select the read and write sockets. This should allow us to implement
4247 tunneling over HTTP soon.
4248 API: RTSP::gst_rtsp_connection_get_url()
4250 2009-03-01 18:31:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4252 * gst-libs/gst/app/gstapp-marshal.list:
4253 app: force automatic rebuild of gstapp-marshal.[ch] after previous change
4254 The previous change to appsrc/appsink requires people to 'make clean'
4255 to get the marshallers rebuilt (causing a build failure otherwise).
4256 Change some lines in the .list file around to force a rebuild of
4257 these files automatically.
4259 2009-02-28 11:07:04 -0800 David Schleef <ds@schleef.org>
4262 Bump glib requirement to 2.14
4264 2009-02-28 19:37:53 +0100 LRN <lrn1986@gmail.com>
4266 * ext/gio/gstgiobasesink.c:
4267 gio: Use correct format modifier for size_t
4270 2009-02-28 19:35:33 +0100 LRN <lrn1986@gmail.com>
4272 * gst-libs/gst/rtsp/gstrtspconnection.c:
4273 rtspconnection: Use correct types for some functions on Win32
4276 2009-02-28 13:11:59 +0100 Edward Hervey <bilboed@bilboed.com>
4278 * gst-libs/gst/rtsp/gstrtspconnection.c:
4279 rtspconnection: Fix warning about using unitialized value.
4281 2009-02-28 12:41:28 +0100 Edward Hervey <bilboed@bilboed.com>
4283 * gst-libs/gst/riff/riff-ids.h:
4284 * gst-libs/gst/riff/riff-media.c:
4285 riff: Add more codec mappings.
4286 This comes mostly from a review of ffmpeg/libavformat/riff.c
4288 2009-02-27 11:14:25 +0200 Stefan Kost <ensonic@users.sf.net>
4290 * ext/alsa/gstalsa.c:
4291 alsa: release pcminfo after the strdup
4293 2009-02-26 17:38:47 +0200 Stefan Kost <ensonic@users.sf.net>
4295 * gst-libs/gst/rtsp/gstrtsprange.c:
4296 rtsprange: don't leak the range in case of parsing error.
4297 Free the gstRTSPTimeRange if we don't return it. Also simplify
4298 gst_rtsp_range_free() as it is valid to pass NULL to g_free().
4300 2009-02-26 16:47:39 +0200 Stefan Kost <ensonic@users.sf.net>
4302 * ext/alsa/gstalsa.c:
4303 alsa: cleanup name lookup.
4304 We can break, once we have a name to make sure, we won't read it ever twice.
4306 2009-02-26 16:09:03 +0200 Stefan Kost <ensonic@users.sf.net>
4308 * gst/subparse/gstsubparse.c:
4309 subparse: don't leak line, if flushing
4311 2009-02-26 16:03:39 +0200 Stefan Kost <ensonic@users.sf.net>
4313 * ext/gio/gstgiosink.c:
4314 giosink: reflow error handling to not leak uri
4316 2009-02-26 15:53:10 +0200 Stefan Kost <ensonic@users.sf.net>
4318 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
4319 * gst/ffmpegcolorspace/imgconvert.c:
4320 ffmpegcolorspace: remove unused code/variables
4322 2009-02-26 12:10:47 +0200 Stefan Kost <ensonic@users.sf.net>
4324 * sys/ximage/ximagesink.c:
4325 ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps
4327 2009-02-26 16:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4329 * docs/libs/gst-plugins-base-libs-sections.txt:
4330 * gst-libs/gst/app/gstappsink.c:
4331 * gst-libs/gst/app/gstappsrc.c:
4332 * gst-libs/gst/app/gstappsrc.h:
4333 * win32/common/libgstapp.def:
4334 app: add callbacks to appsrc, cleanups
4335 Add a uri handler to appsink.
4336 don't emit signals when we have installed callbacks on appsink.
4337 Add callbacks to appsrc to replace the signals.
4338 Add property to disable callbacks in appsrc, default to TRUE for backwards
4339 compatibility but disable when callbacks are installed.
4340 API: GstAppSrc::emit-signals
4341 API: GstAppSrc::gst_app_src_set_emit_signals()
4342 API: GstAppSrc::gst_app_src_get_emit_signals()
4343 API: GstAppSrc::gst_app_src_set_callbacks()
4345 2009-02-26 11:42:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4347 * docs/libs/gst-plugins-base-libs-sections.txt:
4348 * gst-libs/gst/app/gstappsink.h:
4349 * tests/check/elements/appsink.c:
4350 Appsink: add padding for callbacks + docs
4351 Add some padding to the callbacks structure just to be safe.
4352 Remove the now invisible marshaller methods from the docs.
4353 Fix a comment in the unit test.
4355 2009-02-26 09:52:59 +0100 Edward Hervey <bilboed@bilboed.com>
4357 * win32/common/libgstapp.def:
4358 win32: Add new libgstapp symbol
4360 2009-02-26 10:07:21 +0200 Stefan Kost <ensonic@users.sf.net>
4362 * docs/plugins/gst-plugins-base-plugins-sections.txt:
4363 docs: clean section.txt file.
4364 Add appsrc/sink symbols to private, as they are covered in the libs docs.
4366 2009-02-26 10:06:23 +0200 Stefan Kost <ensonic@users.sf.net>
4368 * gst/playback/gstplaybasebin.c:
4369 docs: fix random text after since: tag. Also fix class name to make the docs actual appear.
4371 2009-02-26 09:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
4373 * docs/plugins/gst-plugins-base-plugins.args:
4374 * docs/plugins/gst-plugins-base-plugins.hierarchy:
4375 * docs/plugins/gst-plugins-base-plugins.interfaces:
4376 * docs/plugins/gst-plugins-base-plugins.prerequisites:
4377 * docs/plugins/inspect/plugin-adder.xml:
4378 * docs/plugins/inspect/plugin-alsa.xml:
4379 * docs/plugins/inspect/plugin-app.xml:
4380 * docs/plugins/inspect/plugin-audioconvert.xml:
4381 * docs/plugins/inspect/plugin-audiorate.xml:
4382 * docs/plugins/inspect/plugin-audioresample.xml:
4383 * docs/plugins/inspect/plugin-audiotestsrc.xml:
4384 * docs/plugins/inspect/plugin-cdparanoia.xml:
4385 * docs/plugins/inspect/plugin-decodebin.xml:
4386 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
4387 * docs/plugins/inspect/plugin-gdp.xml:
4388 * docs/plugins/inspect/plugin-gio.xml:
4389 * docs/plugins/inspect/plugin-gnomevfs.xml:
4390 * docs/plugins/inspect/plugin-libvisual.xml:
4391 * docs/plugins/inspect/plugin-ogg.xml:
4392 * docs/plugins/inspect/plugin-pango.xml:
4393 * docs/plugins/inspect/plugin-playback.xml:
4394 * docs/plugins/inspect/plugin-queue2.xml:
4395 * docs/plugins/inspect/plugin-subparse.xml:
4396 * docs/plugins/inspect/plugin-tcp.xml:
4397 * docs/plugins/inspect/plugin-theora.xml:
4398 * docs/plugins/inspect/plugin-typefindfunctions.xml:
4399 * docs/plugins/inspect/plugin-uridecodebin.xml:
4400 * docs/plugins/inspect/plugin-video4linux.xml:
4401 * docs/plugins/inspect/plugin-videorate.xml:
4402 * docs/plugins/inspect/plugin-videoscale.xml:
4403 * docs/plugins/inspect/plugin-videotestsrc.xml:
4404 * docs/plugins/inspect/plugin-volume.xml:
4405 * docs/plugins/inspect/plugin-vorbis.xml:
4406 * docs/plugins/inspect/plugin-ximagesink.xml:
4407 * docs/plugins/inspect/plugin-xvimagesink.xml:
4408 * gst/playback/gstplaybin2.c:
4409 docs: playbin2 has no stream-info
4411 2009-02-26 09:53:03 +0200 Stefan Kost <ensonic@users.sf.net>
4413 * gst-libs/gst/video/video.h:
4414 docs: fix newly added interlace constants and plug holes in video format docs
4416 2009-02-26 09:35:43 +0200 Stefan Kost <ensonic@users.sf.net>
4418 * gst-libs/gst/app/gstappsink.c:
4419 * gst-libs/gst/app/gstappsrc.c:
4420 * gst-libs/gst/audio/gstaudiofilter.c:
4421 * gst-libs/gst/audio/gstringbuffer.c:
4422 * gst-libs/gst/rtp/gstrtcpbuffer.c:
4423 docs: don't put random stuff in tags.
4424 Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
4425 tag to append text again to the documentation body.
4427 2009-02-06 11:10:15 +0200 Stefan Kost <ensonic@users.sf.net>
4429 * sys/ximage/ximagesink.c:
4430 ximagsink: do not access uninitialized height variable.
4431 Exit like in xvimagesink, if we have partial caps.
4433 2009-02-25 20:26:05 -0800 David Schleef <ds@schleef.org>
4437 * win32/common/config.h.in:
4438 Change how win32/common/config.h is updated
4439 Generate win32/common/config.h-new directly from config.h.in,
4440 using shell variables in configure and some hard-coded information.
4441 Change top-level makefile so that 'make win32-update' copies the
4442 generated file to win32/common/config.h, which we keep in source
4443 control. It's kept in source control so that the git tree is
4445 This change is similar to the one recently applied to GStreamer,
4446 except that it adds a few -base specific defines.
4448 2009-02-25 19:40:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4450 * gst-libs/gst/app/Makefile.am:
4451 * gst-libs/gst/app/gstappsink.c:
4452 * gst-libs/gst/app/gstappsrc.c:
4453 * win32/common/libgstapp.def:
4454 app: add win32 .def file and only export functions we want exported
4455 Add a .def file for win32 builds (and make check-exports).
4456 Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes #573165).
4457 Make sure private marshaller functions aren't exported by prefixing them with __gst;
4458 also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
4459 a comment why we're not using glib-genmarshal for this one.
4461 2009-02-25 17:08:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4463 * tests/examples/dynamic/.gitignore:
4464 * tests/examples/dynamic/Makefile.am:
4465 * tests/examples/dynamic/sprinkle.c:
4466 sprinkle: Add another example app
4467 Add an example app that dynamically adds and removes audiotestsrc elements from
4470 2009-02-25 16:25:33 +0100 Peter Kjellerstedt <pkj@axis.com>
4472 * gst-libs/gst/rtsp/gstrtspconnection.c:
4475 2009-02-25 11:31:02 +0100 Peter Kjellerstedt <pkj@axis.com>
4477 * gst-libs/gst/rtsp/gstrtspconnection.c:
4478 * gst/tcp/gstmultifdsink.c:
4479 rtsp, multifdsink: Unify the use of union gst_sockaddr.
4481 2009-02-25 14:22:35 +0000 Jan Schmidt <thaytan@noraisin.net>
4485 build: Update shave init statement for changes in common. Bump common.
4487 2009-02-25 13:16:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4489 * sys/xvimage/xvimagesink.c:
4490 * sys/xvimage/xvimagesink.h:
4491 xvimageink: protect buffer_alloc from shutdown
4492 Use the pool_lock in the buffer_alloc function to detect shutdown. Avoids
4493 crashes when the sink is shutdown.
4495 2009-02-25 12:43:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4497 * gst/playback/gstplaybin2.c:
4498 playbin: use flushing pads instead of fakesink
4499 Use the flushing pads on playsink to terminate on shutdown instead of plugging
4500 fakesinks. this should be a little cheaper.
4502 2009-02-25 12:42:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4504 * gst/playback/gstplaysink.c:
4505 * gst/playback/gstplaysink.h:
4506 playsink: Add FLUSHING pad type
4507 Make it possible to request a flushing pad from the playsink. We can eventually
4508 use these flushing pads to quickly terminate the dataflow when we are shutting
4511 2009-02-25 11:31:52 +0000 Jan Schmidt <thaytan@noraisin.net>
4514 Automatic update of common submodule
4515 From 9cf8c9b to a6ce5c6
4517 2009-02-25 09:52:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4519 * gst-libs/gst/riff/riff-media.c:
4520 riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
4523 2009-02-25 12:07:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4525 * tests/icles/stress-playbin.c:
4526 stress-playbin: print the current uri
4527 Print the current uri so that we can more easily see what uri caused a crash or
4530 2009-02-25 11:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4532 * tests/icles/stress-playbin.c:
4533 Print the errors more clearly
4534 Print some more verbose messages when dealing with errors.
4536 2009-02-25 10:08:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4538 * gst/playback/gstplaybin2.c:
4539 Release the group lock when setting states
4540 Release the group lock while we perform the state changes on the uridecodebins
4541 because that might trigger callbacks that we need to handle with the group lock
4542 taken. Avoids a possible deadly embrace in some id3/flac files.
4545 2009-02-25 10:05:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4547 * gst/playback/gstdecodebin2.c:
4548 Combine finding and creating groups
4549 Combine the search for the current group and optionally creating one into one
4550 function so that we can avoid taking the lock multiple times.
4552 2009-02-25 08:22:00 +0100 Edward Hervey <bilboed@bilboed.com>
4554 * gst/playback/gstplaybin2.c:
4555 Playbin2: Don't leave unused parameters in debug statements.
4556 Fixes build on macosx
4558 2009-02-24 10:33:05 +0100 Edward Hervey <bilboed@bilboed.com>
4560 * gst-libs/gst/riff/riff-media.c:
4561 Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)
4563 2009-02-24 18:43:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4565 * gst/playback/gstplaybin2.c:
4566 Add some G_UNLIKELY because we can
4567 Add a G_UNLIKELY when checking the shutdown variable.
4569 2009-02-24 17:23:58 +0000 Garret D'Amore <garrett.damore@sun.com>
4571 * gst-libs/gst/interfaces/mixer.h:
4572 * gst-libs/gst/interfaces/mixertrack.h:
4573 mixer interface: Add flags to enhance mixer interfaces
4574 This patch adds a few flags to the mixer and mixerctrl interface to
4575 better support OSSv4 (and potentially other backends).
4576 Patch By: Garret D'Amore <garrett.damore@sun.com>
4577 Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>
4578 API: GST_MIXER_FLAG_HAS_WHITELIST, GST_MIXER_FLAG_GROUPING,
4579 API: GST_MIXER_TRACK_NO_RECORD, GST_MIXER_TRACK_NO_MUTE,
4580 API: GST_MIXER_TRACK_WHITELIST
4582 2009-02-24 17:03:08 +0000 Jan Schmidt <thaytan@noraisin.net>
4584 * gst/tcp/gstmultifdsink.c:
4585 multifdsink: Fix strict aliasing error using a union
4587 2009-02-24 16:49:40 +0000 Jan Schmidt <thaytan@noraisin.net>
4589 * gst-libs/gst/rtsp/gstrtspconnection.c:
4590 rtsp: Fix a strict aliasing warning
4591 Fix strict aliasing warnings from casting a sockaddr_storage and
4592 using it as a sockaddr_in6. Use a union instead.
4594 2009-02-24 16:08:49 +0000 Jan Schmidt <thaytan@noraisin.net>
4596 * docs/libs/.gitignore:
4597 * docs/libs/tmpl/.gitignore:
4598 * docs/plugins/.gitignore:
4599 * docs/plugins/tmpl/.gitignore:
4600 Remove .gitignore files from the docs tmpl dirs, that are killed by make clean.
4602 2009-02-24 14:36:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4604 * docs/plugins/Makefile.am:
4605 * ext/vorbis/Makefile.am:
4606 * ext/vorbis/gstvorbisdec.h:
4607 * ext/vorbis/gstvorbisenc.h:
4608 * ext/vorbis/gstvorbisparse.h:
4609 * ext/vorbis/gstvorbistag.h:
4610 * ext/vorbis/vorbis.c:
4611 * ext/vorbis/vorbisdec.c:
4612 * ext/vorbis/vorbisdec.h:
4613 * ext/vorbis/vorbisenc.c:
4614 * ext/vorbis/vorbisenc.h:
4615 * ext/vorbis/vorbisparse.c:
4616 * ext/vorbis/vorbisparse.h:
4617 * ext/vorbis/vorbistag.c:
4618 * ext/vorbis/vorbistag.h:
4619 vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
4621 2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4623 * gst/ffmpegcolorspace/avcodec.h:
4624 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
4625 * gst/ffmpegcolorspace/imgconvert.c:
4626 ffmpegcolorspace: Add conversion from/to YVYU colorspace
4629 2009-02-24 13:42:01 +0100 Jonas Danielsson <jonas.danielsson@axis.com>
4631 * gst/ffmpegcolorspace/imgconvert.c:
4632 ffmpegcolorspace: Add direct UYVY->GRAY8 conversion
4633 The conversion from UYVY to RGB24 and then to GRAY8
4634 is quite slow. Fixes bug #569655.
4636 2009-02-19 17:16:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4638 * gst/playback/gstplaybin2.c:
4639 playbin2: fix deadlock when shutting down. Fixes #572577.
4641 2009-02-19 17:15:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4643 * tests/icles/stress-playbin.c:
4644 stress-playbin: make more flexible, e.g. also useful for playbin2
4646 2009-02-24 12:11:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4648 * gst-libs/gst/rtsp/gstrtspconnection.c:
4649 Match WSAStartup and WSACleanup correctly
4650 Don't randomly call WSAStartup and WSACleanup but instead call the startup when
4651 we create a connection and cleanup when we free it again. Because the internal
4652 datastructure is refcounted, this should not cause any refcounting leaks when
4653 the connection is managed correctly.
4656 2009-02-18 11:59:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
4658 * gst/playback/gstplaysink.c:
4659 playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105.
4661 2009-02-23 10:57:42 -0800 David Flynn <davidf@rd.bbc.co.uk>
4663 * pkgconfig/gstreamer-app-uninstalled.pc.in:
4664 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
4665 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
4666 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
4667 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
4668 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
4669 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
4670 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
4671 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
4672 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
4673 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
4674 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
4675 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
4676 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
4677 * pkgconfig/gstreamer-video-uninstalled.pc.in:
4678 Add srcdir to includes for out-of-source builds
4679 When you use gstreamer uninstalled and build outside
4680 the source tree, the includes need to be specified for
4681 both the source tree and the build tree.
4682 Signed-off-by: David Schleef <ds@schleef.org>
4684 2009-02-22 17:23:52 +0000 Jan Schmidt <thaytan@noraisin.net>
4687 * docs/libs/Makefile.am:
4688 * docs/plugins/Makefile.am:
4689 Use shave for the build output
4691 2009-02-23 12:17:07 +0100 Edward Hervey <bilboed@bilboed.com>
4693 * win32/common/libgstrtsp.def:
4694 win32: Add new symbol to libgstrtsp.def
4696 2009-02-23 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4698 * gst-libs/gst/rtsp/gstrtspextension.c:
4699 * gst-libs/gst/rtsp/gstrtspextension.h:
4700 Add method for handling server requests
4701 Add a receive_request so that extensions can react to server requests.
4703 2009-02-22 19:20:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4705 * tests/check/libs/netbuffer.c:
4706 Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)
4708 2009-02-22 19:19:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4710 * ext/theora/theoraparse.c:
4711 theoraparse: Use the correct unref functions
4713 2009-02-22 19:18:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4715 * sys/ximage/ximagesink.c:
4716 * sys/xvimage/xvimagesink.c:
4717 x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()
4719 2009-02-22 19:12:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4721 * gst-libs/gst/tag/gsttagdemux.c:
4722 tagdemux: Unref the actual buffer instead of the memory address of the buffer
4724 2009-02-22 15:47:53 +0000 Jan Schmidt <thaytan@noraisin.net>
4727 Automatic update of common submodule
4728 From 5d7c9cc to 9cf8c9b
4730 2009-02-22 14:49:29 +0100 Edward Hervey <bilboed@bilboed.com>
4732 * win32/common/libgstrtsp.def:
4733 * win32/common/libgstvideo.def:
4734 win32/common: Update .def files for recent API addition
4736 2009-02-22 13:43:35 +0100 Edward Hervey <bilboed@bilboed.com>
4738 * tests/check/libs/rtp.c:
4739 tests: Fix indentation
4741 2009-02-22 13:42:33 +0100 Edward Hervey <bilboed@bilboed.com>
4743 * gst-libs/gst/video/video.c:
4744 libs/video: Fix gst_video_format_new_caps* functions.
4745 Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
4748 2009-02-21 11:13:36 -0800 David Schleef <ds@schleef.org>
4751 Automatic update of common submodule
4752 From 80c627d to 5d7c9cc
4754 2009-02-20 17:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4756 * gst-libs/gst/rtsp/gstrtspmessage.c:
4757 Improve key/value parsing
4758 Improve header field parsing by keeping a ref to the key/value instead of
4759 copying it into a local variable.
4761 2009-02-20 12:35:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4763 * gst-libs/gst/rtsp/gstrtspconnection.c:
4764 Add trailing \0 to message length
4765 We always put a trailing 0 at the end of the message body. Reflect this fact in
4766 the length of the message.
4768 2009-02-20 09:50:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4770 * gst-libs/gst/rtsp/gstrtspconnection.c:
4771 Don't parse headers for data messages
4772 Don't try to parse the headers on a data message because they don't have
4775 2009-02-19 12:18:29 -0800 Benjamin M. Schwartz <bens@alum.mit.edu>
4777 * ext/theora/gsttheoraenc.h:
4778 * ext/theora/theoraenc.c:
4779 theoraenc: Add property for speed level control
4780 Add property "speed-level" to control the amount of motion searching
4781 the encoder does. This is only available in libtheora >= 1.0 and
4782 will silently fail with earlier libraries. Fixes: #572275.
4783 Signed-off-by: David Schleef <ds@schleef.org>
4785 2009-02-19 17:40:45 +0100 Edward Hervey <bilboed@bilboed.com>
4787 * gst-libs/gst/video/video.c:
4788 * gst-libs/gst/video/video.h:
4789 video: Fix 'Since' tags
4791 2009-01-26 10:30:53 +0100 Edward Hervey <bilboed@bilboed.com>
4793 * docs/libs/gst-plugins-base-libs-sections.txt:
4794 * gst-libs/gst/video/video.c:
4795 * gst-libs/gst/video/video.h:
4796 video: Add flags for interlaced video along with convenience methods for interlaced caps.
4797 These three flags allow all know combinations of interlaced formats. They should
4798 only be used when the caps contain 'interlaced=True'.
4799 Fixes #163577 (yes, it's a 4 year old bug).
4801 2009-02-19 15:51:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4803 * docs/libs/gst-plugins-base-libs-sections.txt:
4804 * gst-libs/gst/rtsp/gstrtspconnection.c:
4805 * gst-libs/gst/rtsp/gstrtspconnection.h:
4806 Make RTSPConnection opaque and rename RTSPChannel
4807 Make the RTSPConnection object opaque so that we can extend it in the future.
4808 Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
4810 2009-01-26 10:31:14 +0100 Edward Hervey <bilboed@bilboed.com>
4812 * gst-libs/gst/riff/riff-media.c:
4813 Add some more mappings for h264 in riff
4815 2009-02-19 10:49:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4817 * win32/common/libgstrtsp.def:
4818 Add new RTSP symbols to def files
4819 Add the new RTSP symbols to the windows def file.
4821 2009-02-19 10:44:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4823 * docs/libs/gst-plugins-base-libs-sections.txt:
4824 * gst-libs/gst/app/gstappsink.c:
4825 * gst-libs/gst/app/gstappsink.h:
4826 * tests/check/Makefile.am:
4827 * tests/check/elements/.gitignore:
4828 * tests/check/elements/appsink.c:
4829 Add method to install callbacks on appsink
4830 Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
4832 Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
4833 performant alternative to connecting to the signals.
4834 Add a unit test for appsink.
4835 Clean up some of the appsink docs.
4836 API: GstAppSink::gst_app_sink_set_callbacks()
4838 2009-02-18 18:46:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4840 * docs/libs/gst-plugins-base-libs-sections.txt:
4841 * gst-libs/gst/rtsp/gstrtspconnection.c:
4842 * gst-libs/gst/rtsp/gstrtspconnection.h:
4843 Add RTSP accept method
4844 Add a method to accept a connection on a socket and create a GstRTSPConnection
4846 API: gst_rtsp_connection_accept()
4848 2009-02-18 17:42:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4850 * docs/libs/gst-plugins-base-libs-sections.txt:
4851 * gst-libs/gst/rtsp/gstrtspconnection.c:
4852 * gst-libs/gst/rtsp/gstrtspconnection.h:
4853 Add RTSP channel object for async io
4854 Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
4855 that the connection can be monitored from a maincontext. This allows us to
4856 operate in ASYNC mode, which is handy when building a server.
4857 Rework the old code to use the async code under the hood.
4858 API: gst_rtsp_channel_new()
4859 API: gst_rtsp_channel_unref()
4860 API: gst_rtsp_channel_attach()
4861 API: gst_rtsp_channel_queue_message()
4863 2009-02-15 07:30:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4865 * gst/audioresample/gstaudioresample.c:
4866 audioresample: Add locking to protect the resampling context
4867 When setting the quality/filter-length while PLAYING the
4868 resampling context will be destroyed and created again in
4869 some cases, which will cause crashes in the transform function
4870 if it's called at that time.
4872 2009-02-13 10:10:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4874 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
4875 * gst/videotestsrc/videotestsrc.c:
4876 ffmpegcolorspace/videotestsrc: Use v308 instead of V308
4878 2009-02-12 19:02:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4880 * gst/ffmpegcolorspace/avcodec.h:
4881 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
4882 * gst/ffmpegcolorspace/imgconvert.c:
4883 * gst/ffmpegcolorspace/imgconvert_template.h:
4884 ffmpegcolorspace: Add support for packed 4:4:4 YUV (format=V308)
4885 Only conversions from/to are implemented, which
4886 gives (indirect) support for all possible conversions.
4887 Partially fixes bug #571147.
4889 2009-02-12 18:17:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4891 * gst/videotestsrc/videotestsrc.c:
4892 videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
4893 Partially fixes bug #571147.
4895 2009-02-12 09:18:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4897 * gst-libs/gst/tag/gsttagdemux.c:
4898 tagdemux: don't abort when downstream pulls a buffer of size 0
4899 Pulling a 0-sized buffer is allowed, and we should handle this correctly instead of
4900 aborting. Fixes #571009 (wma file with ID3v2 tag).
4902 2009-02-11 16:39:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4904 * gst-libs/gst/riff/riff-read.c:
4905 riff: error out on nonsensical chunk sizes instead of aborting
4906 When encountering a nonsensical chunk size such as (guint)-1, error out cleanly instead of
4907 continuing and trying to g_memdup() 4GB of data that doesn't exist, which will either abort
4908 in g_malloc() or crash.
4909 Fixes #553295, crash with fuzzed AVI file.
4911 2009-02-11 16:39:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4914 Make git ignore backup files.
4916 2009-02-10 20:38:58 -0800 Michael Smith <msmith@syncword.(none)>
4918 * gst/playback/gstplaybin2.c:
4919 Revert "Remove pad-removed handlers after setting the decodebins to NULL."
4920 This reverts commit b36d8f3e119f9edc5993c08025614ee32642972e.
4921 This brought back some deadlocks. A small leak is better, for now. Need to
4922 figure out a way to fix the leak properly.
4924 2009-02-10 17:16:07 -0800 Michael Smith <msmith@songbirdnest.com>
4926 * gst/playback/gstplaybin2.c:
4927 playbin2: Fix segfault on notify after group change.
4928 If our group has been switched, then we get a selector active-pad
4929 notification, we don't need to notify.
4931 2009-02-10 17:10:33 -0800 Michael Smith <msmith@songbirdnest.com>
4933 * gst/playback/gstplaysink.c:
4934 playbin2: Look for volume/mute properties recursively in audio element.
4935 Rather than only checking for volume property on the audio sink
4936 directly, recursively look for it on sinks within it (if it's a bin).
4937 Allows use of sink-as-volume-control where the application has supplied
4938 an audio-sink bin that includes a real audio sink internally.
4940 2009-02-10 18:29:22 +0000 Christian Schaller <cschalle@crazyhorse.localdomain>
4942 * gst-plugins-base.spec.in:
4943 Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base
4945 2009-02-10 17:39:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4947 * gst/videotestsrc/videotestsrc.c:
4948 videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
4949 Partially fixes bug #571147.
4951 2009-02-10 17:37:06 +0100 Peter Kjellerstedt <pkj@axis.com>
4953 * gst-libs/gst/rtsp/gstrtspmessage.c:
4954 gstrtspmessage: Minor documentation correction.
4955 Corrected documentation about what needs to be freed after calling
4956 gst_rtsp_message_new(), gst_rtsp_message_new_request(),
4957 gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
4959 2009-02-10 11:00:12 +0100 Antoine Tremblay <hexa00@gmail.com>
4961 * ext/alsa/gstalsamixer.c:
4962 alsamixer: Fix race condition that made alsamixer not working properly
4963 This is due to race conditions between functions that
4964 modified the mixer like set_volume and
4965 snd_mixer_handle_events since the handle_events
4966 can now be called at any time.
4967 Fixed by adding locking around any snd_mixer call
4968 since even read functions can modify the mixer stucture, since
4969 alsa likes to clear it's values before reading new ones.
4970 The favorite race condition seemed to be that set_volume
4971 called read_elem (in alsalib) that reset the volumes to
4972 0 and then read them with read_x_volume. This read looped
4973 on each channel and as the race condition occured the
4974 channels value could be anything , most of the time
4975 it was 0. Thus no value was read or only the value of
4976 one channel was and the volume was reset to 0.
4979 2009-02-09 12:02:21 +0100 Edward Hervey <bilboed@bilboed.com>
4982 Bump revision to use for common submodule.
4984 2009-02-05 15:47:00 +0200 Stefan Kost <ensonic@users.sf.net>
4986 * sys/xvimage/xvimagesink.c:
4987 xvimagesink: do not call _xwindow_clear on ready->paused.
4988 Calling clear at that transition does things like stopping xvideo (which is not
4989 running at that time) and also clearing anything what the application might have drawn.
4990 This breaks handle-expose and autopaint-colorkey features.
4992 2009-02-04 17:03:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4994 * docs/libs/gst-plugins-base-libs-sections.txt:
4995 * gst-libs/gst/rtsp/gstrtsprange.c:
4996 * gst-libs/gst/rtsp/gstrtsprange.h:
4997 RTSPRange: Add method to serialize ranges
4998 Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
4999 be used by a server.
5000 API: GstRTSPRange::gst_rtsp_range_to_string()
5002 2009-02-04 13:16:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5004 * gst-libs/gst/rtsp/gstrtspurl.c:
5005 * gst-libs/gst/rtsp/gstrtspurl.h:
5006 GstRTSPUrl: Add some const to methods
5007 Add const to the methods that do not modify the object.
5009 2009-02-04 13:53:30 +0200 Stefan Kost <ensonic@users.sf.net>
5011 * gst/playback/gstplaysink.c:
5012 playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
5013 The flags where present but actually not been taken into account.
5015 2009-02-04 12:06:38 +0200 Stefan Kost <ensonic@users.sf.net>
5017 * gst/audioresample/gstaudioresample.c:
5018 audioresample: Add a proper deprecation comment and also drop G_PARAM_CONSTRUCT.
5019 The comment will ensure that is is marked properly in the docs and the
5020 GParamSpecflag was causing a duplicated initialisation of the same value.
5022 2009-02-04 11:18:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5024 * gst-libs/gst/rtsp/gstrtspconnection.c:
5025 Add more g_return_if_fail() calls
5026 Check that we have a valid file descriptor before entering certain functions in
5027 order to avoid undesirable situations.
5028 Add some more debugging in the connect method.
5030 2009-02-04 10:31:21 +0200 Stefan Kost <ensonic@users.sf.net>
5033 * gst/audioresample/Makefile.am:
5034 * gst/audioresample/gstaudioresample.c:
5035 audioresample: Only pull in liboil if its actualy used.
5036 Liboil still has quite significant startup overhead especialy on embedded
5037 platforms. In audioresample it was only used for the profiling timer.
5039 2009-02-03 15:26:08 +0200 Stefan Kost <ensonic@users.sf.net>
5041 * gst/typefind/gsttypefindfunctions.c:
5042 typefind: Make the flac check more tight to not mistace some aac files for flac. Fixes #570356.
5043 Add comments about the flac format. Tighten the check to not allow values that
5046 2009-02-03 10:52:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5048 * win32/common/libgstrtsp.def:
5050 Add new methods to the windows def file.
5052 2009-02-02 17:25:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5054 * gst-libs/gst/pbutils/install-plugins.c:
5055 * tests/check/libs/pbutils.c:
5056 pbutils: remove duplicate detail strings when calling the external codec installer
5057 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.
5059 2009-02-02 18:05:42 +0200 Stefan Kost <ensonic@users.sf.net>
5061 * gst-libs/gst/audio/gstaudiosink.c:
5062 * gst-libs/gst/audio/gstaudiosink.h:
5063 Add a FIXME 0.11. Make the log message a bit more detailed and add comments.
5065 2009-02-02 15:43:03 +0200 Stefan Kost <ensonic@users.sf.net>
5068 * gst/audioresample/gstaudioresample.c:
5069 Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark.
5071 2009-02-02 13:30:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5073 * sys/ximage/ximagesink.c:
5074 Fix buffer_alloc in ximagesink
5075 Remove some useless debug info that reported wrong image sizes.
5076 When upstream does not accept out suggested size, fall back to allocating an
5077 image of the requested width/height instead of the currently configured size.
5078 The problem is that an image is reused from the pool because the width/height
5079 match but the caps on the new buffer are the requested caps with possibly
5080 different height/width resulting in errors.
5082 2009-02-02 12:54:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5084 * gst/playback/gstdecodebin2.c:
5085 * gst/playback/gsturidecodebin.c:
5086 Fix documentation for autoplug-select
5087 fix the documentation strings for the autoplug-select signal.
5090 2009-02-02 10:09:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5092 * gst-libs/gst/rtsp/gstrtspmessage.c:
5093 Fix string leak in rtspmessage
5094 when we remove a header field from a message we must free the value associated
5095 with the key to avoid a memory leak.
5097 2009-01-31 18:45:47 +0200 Stefan Kost <ensonic@users.sf.net>
5099 * docs/libs/gst-plugins-base-libs-docs.sgml:
5100 Its "Base Library" and not just "Library".
5102 2009-01-31 18:44:32 +0200 Stefan Kost <ensonic@users.sf.net>
5104 * gst-libs/gst/audio/gstaudiofilter.c:
5105 Link to the class, as we can't link to the members yet.
5107 2009-01-30 17:48:23 -0800 Michael Smith <msmith@songbirdnest.com>
5109 * gst/playback/gstplaybin2.c:
5110 Remove pad-removed handlers after setting the decodebins to NULL.
5111 They do needed cleanup; without this we leak selector requestpads.
5113 2009-01-30 17:47:07 -0800 Michael Smith <msmith@songbirdnest.com>
5115 * gst/playback/gstplaybin2.c:
5116 Unref selector request pad even if we no longer have a selector.
5117 During destruction, we won't have a selector any more, but we still need
5118 to unref the pad to avoid leaking it.
5120 2009-01-30 15:23:23 -0800 Michael Smith <msmith@songbirdnest.com>
5122 * gst/playback/gstplaybin2.c:
5123 Unref source in playbin2's finalize method
5125 2009-01-30 12:04:01 -0800 Michael Smith <msmith@songbirdnest.com>
5127 * gst/playback/gstplaysink.c:
5128 Fix more leaks of pads and elements in gstplaysink.
5129 Don't keep extra references to volume and mute elements; we don't need
5131 Ensure we unref pads that we have references to, and release request
5134 2009-01-30 11:04:37 -0800 Michael Smith <msmith@songbirdnest.com>
5136 * gst/playback/gstplaysink.c:
5137 Avoid leaking all playsinks. Fix some internal leaks.
5138 Playsink was holding references to itself. Don't do that, it's not cool.
5139 Also, free all chains in dispose.
5141 2009-01-30 10:54:12 -0800 Michael Smith <msmith@songbirdnest.com>
5143 * gst/playback/gstplaybin2.c:
5144 Unref peer request pad after releasing it, since we hold a reference.
5146 2009-01-30 10:52:52 -0800 Michael Smith <msmith@songbirdnest.com>
5148 * gst/playback/gstplaybin2.c:
5149 Fix caps leak in playbin2.
5151 2009-01-30 10:51:11 -0800 Michael Smith <msmith@songbirdnest.com>
5153 * gst/playback/gstplaybin2.c:
5154 Unref active pad from selector when finding active stream.
5156 2009-01-30 10:49:55 -0800 Michael Smith <msmith@songbirdnest.com>
5158 * gst/playback/gstplaybin2.c:
5159 Free uris when finalizing playbin2 instance.
5161 2009-01-30 10:38:17 -0800 Michael Smith <msmith@songbirdnest.com>
5163 * gst/playback/gsturidecodebin.c:
5164 Unref pads when iterating over them in analyse_source.
5165 Fixes leak of source's srcpad when using uridecodebin.
5167 2009-01-30 22:22:07 +0200 Stefan Kost <ensonic@users.sf.net>
5169 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
5170 Add releaseinfo with online url.
5172 2009-01-30 17:58:15 +0000 Jan Schmidt <jan.schmidt@sun.com>
5174 * gst/playback/gstplaybasebin.c:
5175 Fix compilation warning on Forte
5177 2009-01-30 17:16:39 +0000 Jan Schmidt <jan.schmidt@sun.com>
5179 * gst/adder/gstadder.c:
5180 Don't do void pointer arithmetic.
5182 2009-01-30 17:25:51 +0000 Jan Schmidt <thaytan@noraisin.net>
5187 2009-01-30 08:50:53 +0100 Edward Hervey <bilboed@bilboed.com>
5191 Use a symbolic link for the pre-commit client-side hook
5193 2009-01-30 08:12:42 +0100 Edward Hervey <bilboed@bilboed.com>
5196 Add more files/directories to ignore
5198 2009-01-29 14:00:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5200 * gst-libs/gst/rtsp/gstrtspdefs.c:
5202 Fix some typos in the doc string of the new
5203 gst_rtsp_options_as_string() method.
5205 2009-01-29 11:55:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5207 * docs/libs/gst-plugins-base-libs-sections.txt:
5208 * gst-libs/gst/rtsp/gstrtspconnection.c:
5209 * gst-libs/gst/rtsp/gstrtspmessage.c:
5210 * gst-libs/gst/rtsp/gstrtspmessage.h:
5211 Add new RTSP message method to set header
5212 Add gst_rtsp_message_take_header() that takes ownership of the passed header
5213 value. This allows us to avoid an allocations and memory copy in some
5215 API: GstRTSPMessage::gst_rtsp_message_take_header()
5217 2009-01-29 11:51:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5219 * docs/libs/gst-plugins-base-libs-sections.txt:
5220 Add new method to docs
5221 Add the new gst_rtsp_options_as_text() method to the docs.
5223 2009-01-28 11:48:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5225 * gst-libs/gst/rtsp/gstrtspdefs.c:
5226 * gst-libs/gst/rtsp/gstrtspdefs.h:
5227 Add method to serialize RTSP options
5228 Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
5230 API: GstRTSP::gst_rtsp_options_as_text()
5232 2009-01-26 17:59:37 -0800 Michael Smith <msmith@songbirdnest.com>
5234 * gst/typefind/gsttypefindfunctions.c:
5235 Ensure we have sufficient data when using data scan contexts.
5236 Fixes crashes typefinding things that look like they might contain AAC
5237 data (but probably aren't actually AAC).
5239 2009-01-26 23:32:09 +0000 Jan Schmidt <thaytan@noraisin.net>
5241 * ext/gio/Makefile.am:
5242 Fix include order for gio plugin
5244 2009-01-23 23:59:48 +0000 Jan Schmidt <thaytan@noraisin.net>
5246 * win32/common/config.h:
5247 Update win32 config.h for 0.10.22.1 dev cycle
5249 2009-01-23 23:16:11 +0000 Jan Schmidt <thaytan@noraisin.net>
5252 * docs/libs/.gitignore:
5253 * gst-libs/gst/audio/.gitignore:
5254 * gst-libs/gst/video/.gitignore:
5256 * tests/examples/dynamic/.gitignore:
5257 Extend and clean up git ignores
5259 2009-01-23 12:31:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5262 * docs/plugins/Makefile.am:
5263 * docs/plugins/gst-plugins-base-plugins-sections.txt:
5264 * docs/plugins/gst-plugins-base-plugins.args:
5265 * docs/plugins/gst-plugins-base-plugins.hierarchy:
5266 * docs/plugins/gst-plugins-base-plugins.interfaces:
5267 * docs/plugins/gst-plugins-base-plugins.prerequisites:
5268 * docs/plugins/inspect/plugin-adder.xml:
5269 * docs/plugins/inspect/plugin-alsa.xml:
5270 * docs/plugins/inspect/plugin-app.xml:
5271 * docs/plugins/inspect/plugin-audioconvert.xml:
5272 * docs/plugins/inspect/plugin-audiorate.xml:
5273 * docs/plugins/inspect/plugin-audioresample.xml:
5274 * docs/plugins/inspect/plugin-audiotestsrc.xml:
5275 * docs/plugins/inspect/plugin-cdparanoia.xml:
5276 * docs/plugins/inspect/plugin-decodebin.xml:
5277 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
5278 * docs/plugins/inspect/plugin-gdp.xml:
5279 * docs/plugins/inspect/plugin-gio.xml:
5280 * docs/plugins/inspect/plugin-gnomevfs.xml:
5281 * docs/plugins/inspect/plugin-libvisual.xml:
5282 * docs/plugins/inspect/plugin-ogg.xml:
5283 * docs/plugins/inspect/plugin-pango.xml:
5284 * docs/plugins/inspect/plugin-playback.xml:
5285 * docs/plugins/inspect/plugin-queue2.xml:
5286 * docs/plugins/inspect/plugin-subparse.xml:
5287 * docs/plugins/inspect/plugin-tcp.xml:
5288 * docs/plugins/inspect/plugin-theora.xml:
5289 * docs/plugins/inspect/plugin-typefindfunctions.xml:
5290 * docs/plugins/inspect/plugin-uridecodebin.xml:
5291 * docs/plugins/inspect/plugin-video4linux.xml:
5292 * docs/plugins/inspect/plugin-videorate.xml:
5293 * docs/plugins/inspect/plugin-videoscale.xml:
5294 * docs/plugins/inspect/plugin-videotestsrc.xml:
5295 * docs/plugins/inspect/plugin-volume.xml:
5296 * docs/plugins/inspect/plugin-vorbis.xml:
5297 * docs/plugins/inspect/plugin-ximagesink.xml:
5298 * docs/plugins/inspect/plugin-xvimagesink.xml:
5299 * gst/audioresample/Makefile.am:
5300 * gst/audioresample/README:
5301 * gst/audioresample/arch.h:
5302 * gst/audioresample/buffer.c:
5303 * gst/audioresample/buffer.h:
5304 * gst/audioresample/debug.c:
5305 * gst/audioresample/debug.h:
5306 * gst/audioresample/fixed_arm4.h:
5307 * gst/audioresample/fixed_arm5e.h:
5308 * gst/audioresample/fixed_bfin.h:
5309 * gst/audioresample/fixed_debug.h:
5310 * gst/audioresample/fixed_generic.h:
5311 * gst/audioresample/functable.c:
5312 * gst/audioresample/functable.h:
5313 * gst/audioresample/gstaudioresample.c:
5314 * gst/audioresample/gstaudioresample.h:
5315 * gst/audioresample/resample.c:
5316 * gst/audioresample/resample.h:
5317 * gst/audioresample/resample_chunk.c:
5318 * gst/audioresample/resample_functable.c:
5319 * gst/audioresample/resample_ref.c:
5320 * gst/audioresample/resample_sse.h:
5321 * gst/audioresample/speex_resampler.h:
5322 * gst/audioresample/speex_resampler_double.c:
5323 * gst/audioresample/speex_resampler_float.c:
5324 * gst/audioresample/speex_resampler_int.c:
5325 * gst/audioresample/speex_resampler_wrapper.h:
5326 * gst/speexresample/Makefile.am:
5327 * gst/speexresample/README:
5328 * gst/speexresample/arch.h:
5329 * gst/speexresample/fixed_arm4.h:
5330 * gst/speexresample/fixed_arm5e.h:
5331 * gst/speexresample/fixed_bfin.h:
5332 * gst/speexresample/fixed_debug.h:
5333 * gst/speexresample/fixed_generic.h:
5334 * gst/speexresample/gstspeexresample.c:
5335 * gst/speexresample/gstspeexresample.h:
5336 * gst/speexresample/resample.c:
5337 * gst/speexresample/resample_sse.h:
5338 * gst/speexresample/speex_resampler.h:
5339 * gst/speexresample/speex_resampler_double.c:
5340 * gst/speexresample/speex_resampler_float.c:
5341 * gst/speexresample/speex_resampler_int.c:
5342 * gst/speexresample/speex_resampler_wrapper.h:
5343 * gst/typefind/gsttypefindfunctions.c:
5344 * tests/check/Makefile.am:
5345 * tests/check/elements/audioresample.c:
5346 * tests/check/elements/speexresample.c:
5347 Rename files and types from speexresample to audioresample
5348 Rename files and types from speexresample to audioresample
5349 to finish the move and to prevent any confusion.
5351 2009-01-23 11:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5353 * sys/xvimage/xvimagesink.c:
5354 Add some more debugging to the Xv strides
5355 Add some more debugging to the strides as they are received from the server and
5356 the expected strides.
5358 2009-01-23 11:40:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5360 * gst/typefind/gsttypefindfunctions.c:
5361 Add typefind function for gsm
5362 Because core now supports typefindfactories without a typefind function we can
5363 register a factory fo GSM that will --if all else fails-- assume the file is a
5364 GSM file based on the registered extension.
5367 2009-01-23 11:37:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5369 * gst/playback/gsturidecodebin.c:
5370 Use more performant link function
5371 We can use gst_element_link_pads() instead of the more generic
5372 gst_element_link() function because we know the pads. This saves some cycles
5373 because the more generic function needs to search for possible compatible caps
5376 2009-01-23 11:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5378 * gst-libs/gst/riff/riff-ids.h:
5379 * gst-libs/gst/riff/riff-media.c:
5380 Add more codec ids for RIFF formats
5381 Handle codec ID for various other AAC formats.
5382 Sync the list of possible codec ids with that of ffmpeg.
5385 2009-01-23 11:27:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5387 * ext/theora/theoradec.c:
5388 Use rounded values for image strides and sizes
5389 Round up the height before calculating the expected size and
5390 strides of the output image.
5392 2009-01-23 11:23:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5394 * ext/alsa/gstalsasink.c:
5395 Improve debug message
5396 Improve the debug message when alsa returns an error.
5398 2009-01-23 11:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5400 * gst-libs/gst/app/gstappsrc.c:
5401 Reset queued_bytes counter when flushing
5402 Set the amount of queued bytes in the internal queue back to 0 when we clear the
5406 2009-01-23 10:19:27 +0100 Benjamin Gaignard <benjamin@gaignard.net>
5408 * gst/typefind/gsttypefindfunctions.c:
5409 Add typefinder for Mobile XMF. Fixes bug #568707.
5411 2009-01-23 10:00:11 +0100 Brian Cameron <brian.cameron@sun.com>
5414 Fix linking on Solaris. Fixes bug #568482.
5415 Check for nsl and socket libraries and add them to
5416 LIBS if they're found. They're needed for socket()
5417 and gethostbyname() on Solaris.
5419 2009-01-22 22:09:47 +0000 Jan Schmidt <thaytan@noraisin.net>
5421 * gst/playback/gstplaybasebin.c:
5422 Fix use-after-unref problem noticed by Josep Torra Valles, and run
5425 2009-01-22 17:46:59 +0200 Stefan Kost <ensonic@users.sf.net>
5428 Update common snapshot.
5430 2009-01-22 13:47:24 +0100 Sebastian Dröge <slomo@circular-chaos.org>
5435 2009-01-22 13:12:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5437 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
5439 2009-01-22 10:14:28 +0100 Sebastian Dröge <slomo@circular-chaos.org>
5441 * gst-libs/gst/fft/gstfftf32.c:
5442 * gst-libs/gst/fft/gstfftf64.c:
5443 * gst-libs/gst/fft/gstffts16.c:
5444 * gst-libs/gst/fft/gstffts32.c:
5445 Reduce the number of allocations for creating FFT contexts
5446 Reduce the number of allocations from 2 to 1 for every FFT
5447 context by allocating enough memory for the FFT context
5448 and passing parts of it to the kissfft allocation functions.
5450 2009-01-22 11:32:56 +0000 Jan Schmidt <thaytan@noraisin.net>
5453 Back to devel -> 0.10.22.1
5455 2009-01-22 05:57:53 +0100 Edward Hervey <bilboed@bilboed.com>
5459 Install and use pre-commit indentation hook from common
5461 2009-01-21 13:09:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5463 * gst-libs/gst/rtp/gstrtpbuffer.c:
5464 * tests/check/libs/rtp.c:
5465 Avoid overflows in the padding checks by doing the check slightly
5467 Add a unit test to check for correct behaviour.
5469 2009-01-21 04:31:32 +0100 Edward Hervey <bilboed@bilboed.com>
5472 autogen.sh : Use git submodule
5474 === release 0.10.22 ===
5476 2009-01-19 23:10:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5482 * docs/plugins/gst-plugins-base-plugins.hierarchy:
5483 * docs/plugins/gst-plugins-base-plugins.interfaces:
5484 * docs/plugins/gst-plugins-base-plugins.prerequisites:
5485 * docs/plugins/inspect/plugin-adder.xml:
5486 * docs/plugins/inspect/plugin-alsa.xml:
5487 * docs/plugins/inspect/plugin-app.xml:
5488 * docs/plugins/inspect/plugin-audioconvert.xml:
5489 * docs/plugins/inspect/plugin-audiorate.xml:
5490 * docs/plugins/inspect/plugin-audioresample.xml:
5491 * docs/plugins/inspect/plugin-audiotestsrc.xml:
5492 * docs/plugins/inspect/plugin-cdparanoia.xml:
5493 * docs/plugins/inspect/plugin-decodebin.xml:
5494 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
5495 * docs/plugins/inspect/plugin-gdp.xml:
5496 * docs/plugins/inspect/plugin-gnomevfs.xml:
5497 * docs/plugins/inspect/plugin-libvisual.xml:
5498 * docs/plugins/inspect/plugin-ogg.xml:
5499 * docs/plugins/inspect/plugin-pango.xml:
5500 * docs/plugins/inspect/plugin-playback.xml:
5501 * docs/plugins/inspect/plugin-queue2.xml:
5502 * docs/plugins/inspect/plugin-subparse.xml:
5503 * docs/plugins/inspect/plugin-tcp.xml:
5504 * docs/plugins/inspect/plugin-theora.xml:
5505 * docs/plugins/inspect/plugin-typefindfunctions.xml:
5506 * docs/plugins/inspect/plugin-uridecodebin.xml:
5507 * docs/plugins/inspect/plugin-video4linux.xml:
5508 * docs/plugins/inspect/plugin-videorate.xml:
5509 * docs/plugins/inspect/plugin-videoscale.xml:
5510 * docs/plugins/inspect/plugin-videotestsrc.xml:
5511 * docs/plugins/inspect/plugin-volume.xml:
5512 * docs/plugins/inspect/plugin-vorbis.xml:
5513 * docs/plugins/inspect/plugin-ximagesink.xml:
5514 * docs/plugins/inspect/plugin-xvimagesink.xml:
5515 * gst-plugins-base.doap:
5545 * win32/common/config.h:
5547 Original commit message from CVS:
5550 2009-01-19 22:01:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5582 Original commit message from CVS:
5585 2009-01-16 11:44:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5587 gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
5588 Original commit message from CVS:
5589 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
5590 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
5591 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
5592 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
5593 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
5594 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
5595 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
5596 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
5597 Use correct struct alignment everywhere to prevent unaligned
5598 memory accesses, resulting in SIGBUS on sparc and probably others.
5601 2009-01-16 11:40:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5603 gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
5604 Original commit message from CVS:
5605 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
5606 Forward unknown events upstream to allow latency configuration.
5609 2009-01-13 14:47:19 +0000 Wim Taymans <wim.taymans@gmail.com>
5611 gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
5612 Original commit message from CVS:
5613 * gst/playback/gstplaybin2.c: (groups_set_locked_state):
5614 Provide the right arguments to a debug line.
5616 2009-01-13 06:51:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
5618 sys/xvimage/xvimagesink.c: Don't reset the colorkey when element is reused. Fixes #567511.
5619 Original commit message from CVS:
5620 * sys/xvimage/xvimagesink.c:
5621 Don't reset the colorkey when element is reused. Fixes #567511.
5623 2009-01-09 23:42:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5625 configure.ac: 0.10.21.3 pre-release
5626 Original commit message from CVS:
5628 0.10.21.3 pre-release
5630 2009-01-09 23:13:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5632 gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
5633 Original commit message from CVS:
5634 * gst-libs/gst/app/gstappsink.c:
5635 Store the returned signal id in the right slot when
5636 registering the pull-buffer signal.
5638 Spotted by: Thomas Vander Stichele <thomas at apestaart dot org>
5640 2009-01-09 17:17:50 +0000 Tim-Philipp Müller <tim@centricular.net>
5642 gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
5643 Original commit message from CVS:
5644 * gst-libs/gst/interfaces/mixer.c:
5645 Small docs addition to clarify that one really mustn't free
5646 the constant GList returned (#566812).
5648 2009-01-08 17:18:24 +0000 Wim Taymans <wim.taymans@gmail.com>
5650 Add GType for GstRTSPUrl and expose a copy function because we can.
5651 Original commit message from CVS:
5652 * docs/libs/gst-plugins-base-libs-sections.txt:
5653 * gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
5654 (gst_rtsp_url_get_type), (gst_rtsp_url_copy):
5655 * gst-libs/gst/rtsp/gstrtspurl.h:
5656 * win32/common/libgstrtsp.def:
5657 Add GType for GstRTSPUrl and expose a copy function because we can.
5658 API: gst_rtsp_url_copy()
5661 2009-01-07 18:36:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5663 Add plugin dependency for the GIO and GVfs modules.
5664 Original commit message from CVS:
5666 * ext/gio/gstgio.c: (plugin_init):
5667 Add plugin dependency for the GIO and GVfs modules.
5670 2009-01-07 18:32:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5672 Add plugin dependency for the gnomevfs modules.
5673 Original commit message from CVS:
5675 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
5676 Add plugin dependency for the gnomevfs modules.
5679 2009-01-07 18:30:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5681 win32/common/libgstcdda.def: Add new symbol to the list of exported symbols.
5682 Original commit message from CVS:
5683 * win32/common/libgstcdda.def:
5684 Add new symbol to the list of exported symbols.
5686 2009-01-07 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
5688 gst/playback/gstplaybin2.c: Fix some comments and docs.
5689 Original commit message from CVS:
5690 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
5691 (gst_play_bin_set_uri), (gst_play_bin_set_suburi),
5692 (no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
5693 (activate_group), (deactivate_group), (groups_set_locked_state),
5694 (gst_play_bin_change_state):
5695 Fix some comments and docs.
5696 Post an error message when we fail to link the selector to the sink.
5697 Remove pushing of EOS, this seems unneeded.
5698 Lock the state of deactivated groups so that they don't accidentally
5699 reactivate when the playbin2 state changes.
5700 Reuse uridecodebins.
5701 Unlock and relock state of groups when playbin goes to NULL.
5704 * gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
5705 Only do something in the pad removed callback when we are dealing with
5706 our sourcepads because the sinkpads don't have a ghostpad.
5708 2009-01-07 10:50:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5710 gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
5711 Original commit message from CVS:
5712 * gst-libs/gst/cdda/gstcddabasesrc.c:
5713 * gst-libs/gst/cdda/gstcddabasesrc.h:
5714 Make the GType of GstCDDABaseSrcMode public for bindings.
5717 2009-01-06 18:03:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5719 Use new core API to make registry re-scan the plugin whenever visualisations are added or removed (see #350477).
5720 Original commit message from CVS:
5722 * ext/libvisual/visual.c: (plugin_init):
5723 Use new core API to make registry re-scan the plugin
5724 whenever visualisations are added or removed (see #350477).
5726 2009-01-06 17:30:31 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
5728 gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
5729 Original commit message from CVS:
5730 Patch by: José Alburquerque <jaalburqu svn gnome org>
5731 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
5732 * gst-libs/gst/audio/gstaudioclock.h:
5733 Make gst_audio_clock_new use const gchar* to ease the wrapping of
5734 C++ bindings. Fixes #566723.
5736 2009-01-06 12:16:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5738 Add pkg-config files for libgstapp. Fixes bug #566761.
5739 Original commit message from CVS:
5741 * pkgconfig/Makefile.am:
5742 * pkgconfig/gstreamer-app-uninstalled.pc.in:
5743 * pkgconfig/gstreamer-app.pc.in:
5744 Add pkg-config files for libgstapp. Fixes bug #566761.
5746 2009-01-06 11:10:29 +0000 Tim-Philipp Müller <tim@centricular.net>
5748 gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
5749 Original commit message from CVS:
5750 * gst-libs/gst/app/gstappsink.c:
5751 * gst-libs/gst/app/gstappsink.h:
5752 * gst-libs/gst/app/gstappsrc.c:
5753 * gst-libs/gst/app/gstappsrc.h:
5754 Make debug categories static. Use _element_class_set_details_simple().
5756 2009-01-06 10:56:45 +0000 Tim-Philipp Müller <tim@centricular.net>
5758 gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
5759 Original commit message from CVS:
5760 * gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
5761 (gst_app_sink_class_init), (gst_app_sink_init),
5762 (gst_app_sink_dispose), (gst_app_sink_finalize),
5763 (gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
5764 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
5765 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
5766 (gst_app_sink_render), (gst_app_sink_getcaps),
5767 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
5768 (gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
5769 (gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
5770 (gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
5771 (gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
5772 (gst_app_sink_pull_buffer)::
5773 * gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
5774 * gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
5775 (gst_app_src_class_init), (gst_app_src_init),
5776 (gst_app_src_flush_queued), (gst_app_src_dispose),
5777 (gst_app_src_finalize), (gst_app_src_set_property),
5778 (gst_app_src_get_property), (gst_app_src_unlock),
5779 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
5780 (gst_app_src_is_seekable), (gst_app_src_check_get_range),
5781 (gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
5782 (gst_app_src_set_caps), (gst_app_src_get_caps),
5783 (gst_app_src_set_size), (gst_app_src_get_size),
5784 (gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
5785 (gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
5786 (gst_app_src_set_latencies), (gst_app_src_set_latency),
5787 (gst_app_src_get_latency), (gst_app_src_push_buffer_full),
5788 (gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
5789 * gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
5790 Move private data into a private instance struct. Add padding to
5791 instance and class structures exposed in public headers. Add
5792 Since markers to the gtk-doc blurbs (#566750).
5794 2009-01-06 10:50:37 +0000 Wim Taymans <wim.taymans@gmail.com>
5796 tests/examples/app/appsrc_ex.c: Some comments.
5797 Original commit message from CVS:
5798 * tests/examples/app/appsrc_ex.c: (main):
5800 When pulling a buffer we can get NULL when the element is EOS, don't try
5801 to unref this NULL buffer.
5803 2009-01-06 10:16:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5805 gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
5806 Original commit message from CVS:
5807 * gst-libs/gst/video/Makefile.am:
5808 * gst-libs/gst/video/video.h:
5809 Fix up build flags and include statement for the new generated
5810 enumtypes files, to fix dist.
5812 2009-01-05 23:04:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5814 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
5815 Original commit message from CVS:
5817 * docs/libs/Makefile.am:
5818 * docs/libs/gst-plugins-base-libs-docs.sgml:
5819 * docs/libs/gst-plugins-base-libs-sections.txt:
5820 * docs/plugins/Makefile.am:
5821 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
5822 * docs/plugins/gst-plugins-base-plugins-sections.txt:
5823 * docs/plugins/gst-plugins-base-plugins.args:
5824 * docs/plugins/gst-plugins-base-plugins.hierarchy:
5825 * docs/plugins/gst-plugins-base-plugins.interfaces:
5826 * docs/plugins/gst-plugins-base-plugins.prerequisites:
5827 * docs/plugins/gst-plugins-base-plugins.signals:
5828 * docs/plugins/inspect/plugin-app.xml:
5829 * gst-libs/gst/Makefile.am:
5830 * gst-libs/gst/app/gstappsink.c:
5831 * gst-libs/gst/app/gstappsrc.c:
5832 * tests/examples/Makefile.am:
5833 * tests/examples/app/Makefile.am:
5834 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
5836 2009-01-05 17:13:13 +0000 Wim Taymans <wim.taymans@gmail.com>
5838 gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
5839 Original commit message from CVS:
5840 * gst-libs/gst/audio/gstbaseaudiosink.c:
5841 (gst_base_audio_sink_change_state):
5842 Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
5843 take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
5844 this because the async_play method is deprecated and usually not called
5847 2009-01-05 12:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
5849 gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
5850 Original commit message from CVS:
5851 * gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
5852 Disconnect signal handlers before destroying a previous decodebin so
5853 that we don't end up causing deadlocks. Fixes #566586.
5855 2009-01-05 10:59:35 +0000 Wim Taymans <wim.taymans@gmail.com>
5857 gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
5858 Original commit message from CVS:
5859 * gst/audiotestsrc/gstaudiotestsrc.c:
5860 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
5861 (gst_audio_test_src_check_get_range),
5862 (gst_audio_test_src_set_property),
5863 (gst_audio_test_src_get_property):
5864 * gst/audiotestsrc/gstaudiotestsrc.h:
5865 Add property to control pull/push based scheduling.
5867 2009-01-02 15:04:13 +0000 Alessandro Decina <alessandro.d@gmail.com>
5869 Make the seek and colorkey examples depend on gtk+-x11 as they use
5870 Original commit message from CVS:
5872 * tests/examples/seek/Makefile.am:
5873 * tests/icles/Makefile.am:
5874 Make the seek and colorkey examples depend on gtk+-x11 as they use
5876 Fixes the build with gtk+-quartz.
5878 2008-12-31 16:04:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
5880 win32/common/: Add new exports to win32 files.
5881 Original commit message from CVS:
5882 * win32/common/libgstaudio.def:
5883 * win32/common/libgsttag.def:
5884 * win32/common/libgstvideo.def:
5885 Add new exports to win32 files.
5887 2008-12-31 13:31:55 +0000 Edward Hervey <bilboed@bilboed.com>
5889 gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
5890 Original commit message from CVS:
5891 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
5892 * gst-libs/gst/tag/gsttagdemux.h:
5893 Add GType for GstTagDemuxResult enum.
5895 2008-12-31 13:01:30 +0000 Edward Hervey <bilboed@bilboed.com>
5897 gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
5898 Original commit message from CVS:
5899 * gst-libs/gst/video/Makefile.am:
5900 * gst-libs/gst/video/video.h:
5901 Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
5902 This will help bindings to use it.
5904 2008-12-31 11:20:26 +0000 Edward Hervey <bilboed@bilboed.com>
5906 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
5907 Original commit message from CVS:
5908 * gst-libs/gst/audio/Makefile.am:
5909 * gst-libs/gst/audio/audio.c:
5910 * gst-libs/gst/audio/multichannel.h:
5911 * gst-libs/gst/audio/testchannels.c:
5913 * win32/common/audio-enumtypes.c:
5914 (gst_audio_channel_position_get_type),
5915 (gst_ring_buffer_state_get_type),
5916 (gst_ring_buffer_seg_state_get_type),
5917 (gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
5918 * win32/common/audio-enumtypes.h:
5919 * win32/common/multichannel-enumtypes.c:
5920 * win32/common/multichannel-enumtypes.h:
5921 * win32/vs6/grammar.dsp:
5922 * win32/vs6/libgstaudio.dsp:
5923 * win32/vs7/libgstaudio.vcproj:
5924 * win32/vs8/libgstaudio.vcproj:
5925 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
5926 audio- in order to wrap all enums declarations of that library.
5927 This modification should not matter since that header file is not a
5928 public header (it will be included by public headers).
5929 Modify win32 crap^Wfiles accordingly.
5931 2008-12-30 17:55:07 +0000 Edward Hervey <bilboed@bilboed.com>
5933 gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
5934 Original commit message from CVS:
5935 * gst-libs/gst/audio/gstbaseaudiosrc.h:
5936 * gst-libs/gst/audio/gstbaseaudiosink.h:
5937 Complete Sebastien's commit from the 13th by exporting the
5938 _slave_method_get_type() methods.
5940 2008-12-29 16:45:20 +0000 Wim Taymans <wim.taymans@gmail.com>
5942 gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
5943 Original commit message from CVS:
5944 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
5945 (gst_app_src_init), (gst_app_src_set_property),
5946 (gst_app_src_get_property), (gst_app_src_query),
5947 (gst_app_src_set_latencies), (gst_app_src_set_latency),
5948 (gst_app_src_get_latency), (gst_app_src_push_buffer_full):
5949 * gst-libs/gst/app/gstappsrc.h:
5950 Add properties and methods to configure and retrieve the min and max
5953 2008-12-20 17:38:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5955 ext/: Implement URI query. Fixes bug #562949.
5956 Original commit message from CVS:
5957 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_query):
5958 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_class_init),
5959 (gst_gio_base_src_query):
5960 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_query):
5961 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
5962 (gst_gnome_vfs_src_query):
5963 Implement URI query. Fixes bug #562949.
5965 2008-12-20 12:48:43 +0000 Wim Taymans <wim.taymans@gmail.com>
5967 gst/playback/gstplaybin2.c: Add some debug info.
5968 Original commit message from CVS:
5969 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
5970 Add some debug info.
5971 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
5972 (gst_play_sink_reconfigure), (gst_play_sink_request_pad),
5973 (gst_play_sink_release_pad):
5974 Add some more debug info.
5975 Reconfigure the audio chain when we switch between raw and encoded audio
5976 in gapless playback.
5978 2008-12-20 12:45:03 +0000 Wim Taymans <wim.taymans@gmail.com>
5980 gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
5981 Original commit message from CVS:
5982 * gst-libs/gst/audio/gstbaseaudiosink.c:
5983 (gst_base_audio_sink_setcaps):
5984 Pause the write thread before deactivating and releasing the ringbuffer
5985 to avoid a deadlock when we do gapless playback with different sample
5986 rates in playbin2. Fixes #564929.
5988 2008-12-19 13:03:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
5990 gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
5991 Original commit message from CVS:
5992 * gst-libs/gst/audio/gstbaseaudiosrc.c:
5993 Make GstAudioSrcSlaveMethod get_type() function non-static
5995 * win32/common/libgstaudio.def:
5996 * win32/common/libgstnetbuffer.def:
5997 Add some missing functions to the list of exported symbols.
5999 2008-12-18 12:37:33 +0000 Andrew Feren <acferen@yahoo.com>
6001 gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
6002 Original commit message from CVS:
6003 Patch by: Andrew Feren <acferen at yahoo dot com>
6004 * gst-libs/gst/netbuffer/gstnetbuffer.c:
6005 (gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
6006 (gst_netaddress_get_address_bytes),
6007 (gst_netaddress_set_address_bytes):
6008 * gst-libs/gst/netbuffer/gstnetbuffer.h:
6009 Make gst_netaddress_get_ip4_address fail for v6 addresses.
6010 Make gst_netaddress_get_ip6_address either fail or return the v4
6011 address as a transitional v6 address.
6012 Add two convenience functions:
6013 API: gst_netaddress_get_address_bytes()
6014 API: gst_netaddress_set_address_bytes()
6017 2008-12-17 13:51:46 +0000 Wim Taymans <wim.taymans@gmail.com>
6019 Add appsrc and appsink documentation.
6020 Original commit message from CVS:
6021 * docs/plugins/Makefile.am:
6022 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
6023 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
6024 * gst-libs/gst/app/gstappsink.c:
6025 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
6026 Add appsrc and appsink documentation.
6028 2008-12-17 08:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6030 gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
6031 Original commit message from CVS:
6032 * gst/adder/Makefile.am:
6033 * gst/adder/gstadder.c:
6034 Cleanup variable names to make the adder-loop easier to understand.
6035 Also try to use liboil to spee it up, but ifdef it out as it does not
6036 make any change for me (Intel pentim M (sse,sse2) please try on other
6039 2008-12-16 20:16:17 +0000 Wim Taymans <wim.taymans@gmail.com>
6041 Add minimal docs to make the remaining tcp elements show up.
6042 Original commit message from CVS:
6043 * docs/plugins/Makefile.am:
6044 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
6045 * docs/plugins/gst-plugins-base-plugins-sections.txt:
6046 * gst/tcp/gsttcpclientsink.c:
6047 * gst/tcp/gsttcpclientsrc.c:
6048 * gst/tcp/gsttcpserversrc.c:
6049 Add minimal docs to make the remaining tcp elements show up.
6052 2008-12-15 12:02:26 +0000 Wim Taymans <wim.taymans@gmail.com>
6054 examples/app/: Fix example to unref after emiting the push-buffer action.
6055 Original commit message from CVS:
6056 * examples/app/appsrc-ra.c: (feed_data):
6057 * examples/app/appsrc-seekable.c: (feed_data):
6058 * examples/app/appsrc-stream.c: (read_data):
6059 * examples/app/appsrc-stream2.c: (feed_data):
6060 Fix example to unref after emiting the push-buffer action.
6061 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
6062 (gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
6063 (gst_app_src_push_buffer_action):
6064 Don't take the ref on the buffer in push-buffer action because it's too
6065 awkward for bindings. Fixes #564482.
6067 2008-12-13 19:32:13 +0000 Tim-Philipp Müller <tim@centricular.net>
6069 win32/common/config.h: Update to CVS version.
6070 Original commit message from CVS:
6071 * win32/common/config.h:
6072 Update to CVS version.
6073 * win32/common/config.h.in:
6074 Hardcode path to plugin install helper exe, just like we hardcode
6075 the paths in core. Removes another source of VCS conflicts for
6076 people hacking gst-plugins-base on systems with autotools.
6078 2008-12-13 16:21:12 +0000 Edward Hervey <bilboed@bilboed.com>
6080 m4/Makefile.am: And a couple more .m4 that don't exist anymore with gettext 0.17
6081 Original commit message from CVS:
6083 And a couple more .m4 that don't exist anymore with gettext 0.17
6085 2008-12-13 12:41:56 +0000 Edward Hervey <bilboed@bilboed.com>
6087 m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we...
6088 Original commit message from CVS:
6090 inttypes.m4 hasn't been available since gettext-0.15, and since we now
6091 require gettext >= 0.17 ... we can remove it from the list of files to
6094 2008-12-13 06:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6096 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
6097 Original commit message from CVS:
6098 * gst-libs/gst/audio/gstbaseaudiosink.c:
6099 (gst_base_audio_sink_slave_method_get_type),
6100 (gst_base_audio_sink_class_init):
6101 * gst-libs/gst/audio/gstbaseaudiosink.h:
6102 * gst-libs/gst/audio/gstbaseaudiosrc.c:
6103 (gst_base_audio_src_slave_method_get_type),
6104 (gst_base_audio_src_class_init):
6105 * gst-libs/gst/audio/gstbaseaudiosrc.h:
6106 API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
6107 public API. This is needed for the C++ bindings to be able
6108 to use this base classes. Fixes bug #564200, #564206.
6110 2008-12-12 19:41:28 +0000 Edward Hervey <bilboed@bilboed.com>
6112 gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
6113 Original commit message from CVS:
6114 * gst-libs/gst/cdda/gstcddabasesrc.c:
6115 (gst_cdda_base_src_handle_event):
6116 Remove erroneous gst_buffer_ref().
6117 * tests/check/libs/rtp.c: (GST_START_TEST):
6118 Don't forget to unref the buffer once you're done with it.
6120 2008-12-12 13:06:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6122 gst/playback/: XRef to GstXOverlay.
6123 Original commit message from CVS:
6124 * gst/playback/gstplaybin.c:
6125 * gst/playback/gstplaybin2.c:
6126 XRef to GstXOverlay.
6128 2008-12-12 10:54:45 +0000 Edward Hervey <bilboed@bilboed.com>
6130 gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
6131 Original commit message from CVS:
6132 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
6133 Free the factory array when finalizing.
6134 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
6135 Use a GstStaticPadTemplate since the src pad caps are fixed.
6137 2008-12-12 07:17:21 +0000 Edward Hervey <bilboed@bilboed.com>
6139 ext/vorbis/vorbisenc.c: Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with pad templates.
6140 Original commit message from CVS:
6141 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_base_init),
6142 (gst_vorbis_enc_init):
6143 Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with
6146 2008-12-12 07:15:22 +0000 Edward Hervey <bilboed@bilboed.com>
6148 gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
6149 Original commit message from CVS:
6150 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
6151 (gst_riff_create_video_template_caps):
6152 Add mapping for VP6 in avi/riff.
6154 2008-12-11 15:49:12 +0000 Edward Hervey <bilboed@bilboed.com>
6156 gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
6157 Original commit message from CVS:
6158 * gst/subparse/samiparse.c: (sami_context_push_state),
6159 (sami_context_pop_state), (start_sami_element), (end_sami_element):
6160 Some versions of libxml seem to be very picky as to strict formatting
6161 of the input and never 'close' the final </body> tag.
6162 In order to fix that bad behaviour, we trigger the flushing of
6163 remaining data on both </body> and </sami>.
6166 2008-12-11 12:32:03 +0000 Guillaume Emont <guillaume@fluendo.com>
6168 gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
6169 Original commit message from CVS:
6170 Patch by: Guillaume Emont <guillaume at fluendo dot com>
6171 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
6172 Add typefinders for MS Word files and OS X .DS_Store files to
6173 prevent them to be recognized as MPEG files. Fixes bug #564098.
6175 2008-12-11 11:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
6177 gst/playback/gstplaysink.c: Add some more debug info.
6178 Original commit message from CVS:
6179 * gst/playback/gstplaysink.c: (gen_audio_chain),
6180 (gst_play_sink_reconfigure):
6181 Add some more debug info.
6182 Fix linking of just an encoded sink.
6183 Handle failure to create a sink chain more gracefully than crashing.
6185 2008-12-11 10:33:48 +0000 Wim Taymans <wim.taymans@gmail.com>
6187 tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
6188 Original commit message from CVS:
6189 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
6190 Pushing 10 buffers is enough to run the test.
6192 2008-12-11 10:28:43 +0000 Wim Taymans <wim.taymans@gmail.com>
6194 tests/examples/seek/seek.c: Hook up the SKIP seek flag.
6195 Original commit message from CVS:
6196 * tests/examples/seek/seek.c: (do_seek), (stop_cb),
6197 (skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
6199 Hook up the SKIP seek flag.
6201 2008-12-10 18:43:32 +0000 Wim Taymans <wim.taymans@gmail.com>
6203 gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
6204 Original commit message from CVS:
6205 * gst/playback/gstplaybin2.c: (pad_added_cb):
6206 Error out with a missing-plugin error when the input-selector was not
6208 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
6211 2008-12-10 17:39:32 +0000 Wim Taymans <wim.taymans@gmail.com>
6213 gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
6214 Original commit message from CVS:
6215 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
6216 (gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
6217 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
6218 (gst_play_sink_send_event), (gst_play_sink_change_state):
6220 Try to set the selected sink to READY before using it. This will allow
6221 for detection of incompatible formats sooner.
6222 Don't cause a fatal error when conversion elements are missing but post
6223 a missing-element message and a warning instead because things might
6224 still link and run fine.
6225 Simplyfy the construction of audio and video sink chains.
6227 2008-12-10 14:55:10 +0000 Wim Taymans <wim.taymans@gmail.com>
6229 ext/ogg/gstoggdemux.c: Use G_DEFINE_TYPE for the OggPad to get some threadsafe type init from glib.
6230 Original commit message from CVS:
6231 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
6232 (gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
6233 Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
6236 2008-12-10 08:19:13 +0000 Luis Menina <liberforce@freeside.fr>
6238 gst/: Include glib.h instead of a specific GLib header. Including single
6239 Original commit message from CVS:
6240 Patch by: Luis Menina <liberforce at freeside dot fr>
6241 * gst-libs/gst/floatcast/floatcast.h:
6242 * gst/typefind/gsttypefindfunctions.c:
6243 Include glib.h instead of a specific GLib header. Including single
6244 GLib headers is deprecated. Fixes bug #563904.
6246 2008-12-09 18:30:10 +0000 Julien Moutte <julien@moutte.net>
6248 gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
6249 Original commit message from CVS:
6250 2008-12-09 Julien Moutte <julien@fluendo.com>
6251 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
6252 Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
6254 2008-12-09 17:21:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6256 gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
6257 Original commit message from CVS:
6258 * gst-libs/gst/riff/riff-read.c:
6259 Fix handling of odd chunks in riff metadata.
6261 2008-12-08 18:44:22 +0000 Wim Taymans <wim.taymans@gmail.com>
6263 gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
6264 Original commit message from CVS:
6265 * gst/volume/gstvolume.c: (gst_volume_class_init),
6266 (volume_before_transform), (volume_transform_ip):
6267 Use new basetransform vmethod to reconfigure the dynamic properties and
6268 any pending volume/mute changes. Fixes #563508.
6270 2008-12-08 18:12:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6272 configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...
6273 Original commit message from CVS:
6275 First check for "theoraenc theoradec" and if that failed check
6276 for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
6277 deprecate the latter. Also linking on Windows fails with just "theora"
6278 and the version check would fail for the release candidates.
6281 2008-12-08 15:25:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6283 gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
6284 Original commit message from CVS:
6285 * gst/playback/gstdecodebin.c:
6286 * gst/playback/gstdecodebin2.c:
6287 Add basic docs to decodebin and link to decodebin from decodebin2.
6289 2008-12-08 12:08:32 +0000 Olivier Crete <tester@tester.ca>
6291 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
6292 Original commit message from CVS:
6293 Patch by: Olivier Crete <tester at tester ca>
6294 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
6295 * gst-libs/gst/rtp/gstrtcpbuffer.h:
6296 Implement gst_rtcp_packet_remove(). Fixes #563174.
6297 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
6298 Add unit test for some RTCP functions.
6300 2008-12-04 20:09:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6302 configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.
6303 Original commit message from CVS:
6305 Apparently AC_CONFIG_MACRO_DIR breaks when using more
6306 than one macro directory, reverting last change.
6308 2008-12-04 19:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6310 configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros.
6311 Original commit message from CVS:
6313 Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
6316 2008-12-03 17:47:44 +0000 Edward Hervey <bilboed@bilboed.com>
6318 sys/: Clear all flags on buffers returned from the image pool.
6319 Original commit message from CVS:
6320 * sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
6321 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
6322 Clear all flags on buffers returned from the image pool.
6325 2008-12-01 19:36:35 +0000 이문형 <iwings@gmail.com>
6327 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...
6328 Original commit message from CVS:
6329 Patch by: 이문형 <iwings at gmail dot com>
6330 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
6331 Don't forget to release the lock again if we bail out because some
6332 pad is flushing or we've reached EOS, otherwise things will lock up
6333 next time _push_buffer() is called (#562802).
6335 2008-11-29 13:31:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6337 Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
6338 Original commit message from CVS:
6339 Patch by: Cygwin Ports maintainer
6340 <yselkowitz at users dot sourceforge dot net>
6343 Require gettext 0.17 because older versions don't mix with libtool
6344 2.2. At build time an older gettext version will still work.
6347 2008-11-28 13:30:36 +0000 Christian Schaller <uraeus@gnome.org>
6350 * gst/speexresample/Makefile.am:
6352 Original commit message from CVS:
6355 2008-11-28 09:44:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6357 Update documentation of speexresample for the new element name.
6358 Original commit message from CVS:
6359 * docs/plugins/gst-plugins-base-plugins.args:
6360 * docs/plugins/gst-plugins-base-plugins.hierarchy:
6361 * docs/plugins/gst-plugins-base-plugins.interfaces:
6362 * docs/plugins/gst-plugins-base-plugins.prerequisites:
6363 * docs/plugins/inspect/plugin-videorate.xml:
6364 * gst/speexresample/gstspeexresample.c:
6365 Update documentation of speexresample for the new element name.
6367 2008-11-28 09:04:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6369 gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
6370 Original commit message from CVS:
6371 * gst/speexresample/README:
6372 Update README with the latest diff between the Speex resampler
6375 2008-11-28 08:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6377 gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
6378 Original commit message from CVS:
6379 * gst/speexresample/gstspeexresample.c: (plugin_init):
6380 Update the debug category from speex_resample to audioresample.
6382 2008-11-27 19:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6384 Remove audioresample files.
6385 Original commit message from CVS:
6386 * gst/audioresample/Makefile.am:
6387 * gst/audioresample/buffer.c:
6388 * gst/audioresample/buffer.h:
6389 * gst/audioresample/debug.c:
6390 * gst/audioresample/debug.h:
6391 * gst/audioresample/functable.c:
6392 * gst/audioresample/functable.h:
6393 * gst/audioresample/gstaudioresample.c:
6394 * gst/audioresample/gstaudioresample.h:
6395 * gst/audioresample/resample.c:
6396 * gst/audioresample/resample.h:
6397 * gst/audioresample/resample_chunk.c:
6398 * gst/audioresample/resample_functable.c:
6399 * gst/audioresample/resample_ref.c:
6400 * tests/check/elements/audioresample.c:
6401 Remove audioresample files.
6403 2008-11-27 17:04:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6405 docs/plugins/inspect/plugin-audioresample.xml: Regenerated for library filename change.
6406 Original commit message from CVS:
6407 * docs/plugins/inspect/plugin-audioresample.xml:
6408 Regenerated for library filename change.
6410 2008-11-27 16:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6412 Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
6413 Original commit message from CVS:
6415 * docs/plugins/Makefile.am:
6416 * docs/plugins/gst-plugins-base-plugins-sections.txt:
6417 * docs/plugins/gst-plugins-base-plugins.args:
6418 * docs/plugins/gst-plugins-base-plugins.hierarchy:
6419 * docs/plugins/gst-plugins-base-plugins.interfaces:
6420 * docs/plugins/gst-plugins-base-plugins.prerequisites:
6421 * docs/plugins/inspect/plugin-adder.xml:
6422 * docs/plugins/inspect/plugin-alsa.xml:
6423 * docs/plugins/inspect/plugin-audioconvert.xml:
6424 * docs/plugins/inspect/plugin-audiorate.xml:
6425 * docs/plugins/inspect/plugin-audioresample.xml:
6426 * docs/plugins/inspect/plugin-audiotestsrc.xml:
6427 * docs/plugins/inspect/plugin-cdparanoia.xml:
6428 * docs/plugins/inspect/plugin-decodebin.xml:
6429 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
6430 * docs/plugins/inspect/plugin-gdp.xml:
6431 * docs/plugins/inspect/plugin-gio.xml:
6432 * docs/plugins/inspect/plugin-gnomevfs.xml:
6433 * docs/plugins/inspect/plugin-libvisual.xml:
6434 * docs/plugins/inspect/plugin-ogg.xml:
6435 * docs/plugins/inspect/plugin-pango.xml:
6436 * docs/plugins/inspect/plugin-playback.xml:
6437 * docs/plugins/inspect/plugin-queue2.xml:
6438 * docs/plugins/inspect/plugin-subparse.xml:
6439 * docs/plugins/inspect/plugin-tcp.xml:
6440 * docs/plugins/inspect/plugin-theora.xml:
6441 * docs/plugins/inspect/plugin-typefindfunctions.xml:
6442 * docs/plugins/inspect/plugin-uridecodebin.xml:
6443 * docs/plugins/inspect/plugin-video4linux.xml:
6444 * docs/plugins/inspect/plugin-videorate.xml:
6445 * docs/plugins/inspect/plugin-videoscale.xml:
6446 * docs/plugins/inspect/plugin-videotestsrc.xml:
6447 * docs/plugins/inspect/plugin-volume.xml:
6448 * docs/plugins/inspect/plugin-vorbis.xml:
6449 * docs/plugins/inspect/plugin-ximagesink.xml:
6450 * docs/plugins/inspect/plugin-xvimagesink.xml:
6451 * gst/speexresample/gstspeexresample.c: (plugin_init):
6452 * gst/speexresample/Makefile.am:
6453 * tests/check/Makefile.am:
6454 * tests/check/elements/speexresample.c: (setup_speexresample),
6455 (GST_START_TEST), (test_pipeline):
6456 Rename the moved speexresample to audioresample, integrate into the
6457 build system and remove the old audioresample from the build system.
6458 Fixes bug #558124, #385061, #346218, #116051.
6460 2008-11-27 16:47:41 +0000 Wim Taymans <wim.taymans@gmail.com>
6462 gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
6463 Original commit message from CVS:
6464 * gst-libs/gst/audio/gstbaseaudiosrc.c:
6465 (gst_base_audio_src_get_offset), (gst_base_audio_src_create):
6466 Avoid nasty int overflows after about 12 hours and 25 minutes when these
6467 code paths are triggered.
6468 A free beer to Håvard Graff for finding this!
6470 2008-11-27 11:16:44 +0000 이문형 <iwings@gmail.com>
6472 gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
6473 Original commit message from CVS:
6474 Patch by: 이문형 <iwings at gmail dot com>
6475 * gst-libs/gst/rtsp/gstrtspconnection.c:
6476 (gst_rtsp_connection_connect):
6477 A successful gst_poll_wait() doesn't always mean successful connect() on
6478 Windows. We should check errors by calling gst_poll_fd_has_error().
6481 2008-11-25 16:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6483 tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.
6484 Original commit message from CVS:
6485 * tests/check/elements/speexresample.c: (test_pipeline):
6486 Make unit test again faster to prevent timeouts with valgrind.
6488 2008-11-25 15:33:30 +0000 Wim Taymans <wim.taymans@gmail.com>
6490 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
6491 Original commit message from CVS:
6492 * gst-libs/gst/rtp/gstrtcpbuffer.c:
6493 Fix typo in the docs.
6495 2008-11-25 15:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
6497 ext/ogg/gstoggdemux.c: If no stream was found before receiving EOS, post an error message.
6498 Original commit message from CVS:
6499 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
6500 If no stream was found before receiving EOS, post an error message.
6503 2008-11-25 15:14:30 +0000 Wim Taymans <wim.taymans@gmail.com>
6505 ext/theora/: Parse segment events.
6506 Original commit message from CVS:
6507 * ext/theora/gsttheoraenc.h:
6508 * ext/theora/theoraenc.c: (gst_theora_enc_init),
6509 (theora_buffer_from_packet), (theora_push_packet),
6510 (theora_enc_sink_event), (theora_enc_is_discontinuous),
6512 Parse segment events.
6513 Pass incomming buffer timestamps to outgoing buffers.
6514 Use the running_time to construct the granulepos.
6517 2008-11-25 11:00:55 +0000 Wim Taymans <wim.taymans@gmail.com>
6519 gst/playback/gstplaybin2.c: Fix buffer-duration property.
6520 Original commit message from CVS:
6521 * gst/playback/gstplaybin2.c: (activate_group):
6522 Fix buffer-duration property.
6524 2008-11-25 10:32:49 +0000 Wim Taymans <wim.taymans@gmail.com>
6526 gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
6527 Original commit message from CVS:
6528 * gst-libs/gst/audio/gstbaseaudiosink.c:
6529 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
6530 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
6531 (gst_base_audio_sink_change_state):
6532 Really fix audiosink drain handling by keeping track of the running_time
6535 2008-11-24 20:25:24 +0000 Michael Smith <msmith@xiph.org>
6537 gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
6538 Original commit message from CVS:
6539 * gst/playback/gstplaybin2.c:
6540 Add notification of current stream. Add ability to configure buffer
6542 * gst/playback/gsturidecodebin.c:
6543 Add ability to configure buffer sizes for streaming mode.
6546 2008-11-24 20:11:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6548 gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
6549 Original commit message from CVS:
6550 * gst-libs/gst/audio/gstbaseaudiosink.c:
6551 Time is already in running_time. Remove base_time handling. Fixes
6552 audiosinks not draining and thus chopping some audio in the end.
6554 2008-11-24 19:18:59 +0000 David Schleef <ds@schleef.org>
6556 ext/ogg/gstoggmux.*: If we're muxing a dirac stream, flush the page after every picture.
6557 Original commit message from CVS:
6558 * ext/ogg/gstoggmux.c:
6559 * ext/ogg/gstoggmux.h:
6560 If we're muxing a dirac stream, flush the page after every picture.
6562 2008-11-24 12:56:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6564 gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
6565 Original commit message from CVS:
6566 * gst-libs/gst/audio/gstbaseaudiosink.c:
6567 Add one log message to check for audio_drained. Sync one log message
6568 with the condition. Send EOS after draining audio in pull mode.
6570 2008-11-24 12:07:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6572 ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr...
6573 Original commit message from CVS:
6574 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
6575 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
6576 Use gst_buffer_try_new_and_alloc() and fail properly if the
6577 allocation failed. This prevents abort() if downstream elements
6578 request an insane amount of memory.
6580 2008-11-24 12:03:11 +0000 Jon Trowbridge <trow@ximian.com>
6582 gst/volume/gstvolume.*: Cleanup volume, define and use default values.
6583 Original commit message from CVS:
6584 * gst/volume/gstvolume.c: (volume_choose_func),
6585 (volume_update_volume), (gst_volume_set_volume),
6586 (gst_volume_get_volume), (gst_volume_set_mute),
6587 (gst_volume_class_init), (gst_volume_init),
6588 (volume_process_double), (volume_process_float),
6589 (volume_process_int32), (volume_process_int32_clamp),
6590 (volume_process_int24), (volume_process_int24_clamp),
6591 (volume_process_int16), (volume_process_int16_clamp),
6592 (volume_process_int8), (volume_process_int8_clamp), (volume_setup),
6593 (volume_transform_ip), (volume_set_property),
6594 (volume_get_property):
6595 * gst/volume/gstvolume.h:
6596 Cleanup volume, define and use default values.
6597 Recalculate new volume and mute setup before processing. Fixes #561789.
6598 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
6599 Add controller unit test. Patch by: Jonathan Matthew
6600 Fix bogus test that messed with basetransform's internal state.
6602 2008-11-22 15:02:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6604 tests/check/elements/speexresample.c: Make the unit test a bit faster to prevent timeouts, especially with valgrind.
6605 Original commit message from CVS:
6606 * tests/check/elements/speexresample.c: (GST_START_TEST):
6607 Make the unit test a bit faster to prevent timeouts, especially
6610 2008-11-22 14:44:26 +0000 Wim Taymans <wim.taymans@gmail.com>
6612 gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
6613 Original commit message from CVS:
6614 * gst/videorate/gstvideorate.c:
6615 Add jpeg and png image media types to the caps. Fixes #561436.
6617 2008-11-22 14:31:43 +0000 Wim Taymans <wim.taymans@gmail.com>
6619 gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ...
6620 Original commit message from CVS:
6621 * gst/playback/gstplaysink.c: (gen_audio_chain):
6622 Don't post an error when we can't configure the volume but post a
6623 warning instead. Fixes #561780.
6625 2008-11-21 20:32:56 +0000 Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
6627 gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video...
6628 Original commit message from CVS:
6629 Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
6630 * gst/videotestsrc/gstvideotestsrc.c:
6631 * gst/videotestsrc/gstvideotestsrc.h:
6632 * gst/videotestsrc/videotestsrc.c:
6633 * gst/videotestsrc/videotestsrc.h:
6634 Add a zone plate pattern generator based on BBC R&D Report
6635 1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate
6636 kx2=20 ky2=20 kt=1'.
6638 2008-11-21 15:45:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
6640 gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
6641 Original commit message from CVS:
6642 * gst/speexresample/gstspeexresample.c:
6643 (gst_speex_resample_class_init), (gst_speex_resample_set_property),
6644 (gst_speex_resample_get_property):
6645 Add a "filter-length" property that maps to the quality values
6646 for compatibilty with audioresample.
6648 2008-11-21 00:04:48 +0000 Michael Smith <msmith@xiph.org>
6650 gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
6651 Original commit message from CVS:
6652 * gst/playback/gstdecodebin2.c:
6653 Fix random fat-fingering making this not compile.
6655 2008-11-20 22:11:38 +0000 Michael Smith <msmith@xiph.org>
6657 gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh...
6658 Original commit message from CVS:
6659 * gst/playback/gstdecodebin2.c:
6660 If the top-level type of the stream is plain text, don't try to decode
6661 it, matching behaviour of decodebin.
6662 * gst/playback/gstplaysink.c:
6663 If we fail to generate a text chain (e.g. due to missing optional
6664 plugins), don't crash.
6666 2008-11-20 22:06:05 +0000 Michael Smith <msmith@xiph.org>
6668 gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
6669 Original commit message from CVS:
6670 * gst-libs/gst/rtsp/gstrtspdefs.c:
6671 Fix win32 build. Oops.
6673 2008-11-20 21:40:49 +0000 Michael Smith <msmith@xiph.org>
6675 gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
6676 Original commit message from CVS:
6677 * gst-libs/gst/rtsp/gstrtspdefs.c:
6678 Use WSAGetLastError() rather than errno/h_errno on win32.
6680 2008-11-20 21:20:27 +0000 Michael Smith <msmith@xiph.org>
6682 gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
6683 Original commit message from CVS:
6684 * gst-libs/gst/riff/riff-media.c:
6685 Support WMA Lossless properly.
6687 2008-11-19 00:24:44 +0000 David Schleef <ds@schleef.org>
6689 gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect...
6690 Original commit message from CVS:
6691 * gst/videotestsrc/gstvideotestsrc.c:
6692 * gst/videotestsrc/gstvideotestsrc.h:
6693 * gst/videotestsrc/videotestsrc.c:
6694 * gst/videotestsrc/videotestsrc.h:
6695 Add "colorspec" property, specifying whether to generate BT.601
6696 or BT.709 video. This only affects YCbCr values, not RGB, since
6697 if you're generating a 709 test pattern, presumably you want
6698 709 RGB primaries, not 601. Also add "smpte75" pattern, which
6699 uses 75% colors instead of 100%, since this is often more useful
6700 for testing (and also follows the SMPTE EG-1 guideline).
6702 2008-11-18 18:08:42 +0000 Alessandro Decina <alessandro.d@gmail.com>
6704 gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2.
6705 Original commit message from CVS:
6706 * gst/playback/gstdecodebin.c:
6707 Add a "sink-caps" property to decodebin like it's done for decodebin2.
6710 2008-11-14 21:44:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
6712 gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
6713 Original commit message from CVS:
6714 * gst/audioresample/gstaudioresample.c:
6715 Guard against a NULL dereference I somehow encountered -
6716 with a FLUSH_STOP arriving either before basetransform _start(),
6718 * gst/typefind/gsttypefindfunctions.c:
6719 Make sure we never jump backwards when typefinding corrupt mov files.
6721 2008-11-14 21:39:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
6723 gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
6724 Original commit message from CVS:
6725 * gst-libs/gst/interfaces/propertyprobe.c:
6726 Fix random type causing a docs warning.
6728 2008-11-14 15:40:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6730 sys/v4l/gstv4l.c: Give it a minimal rank for autovideosrc.
6731 Original commit message from CVS:
6733 Give it a minimal rank for autovideosrc.
6735 2008-11-13 21:11:13 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6737 gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types.
6738 Original commit message from CVS:
6739 * gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
6741 Improve typefinding of ISO JPEG2000 mime types.
6743 2008-11-13 18:18:32 +0000 Wim Taymans <wim.taymans@gmail.com>
6745 sys/xvimage/xvimagesink.*: Avoid typechecking when we do trivial casts.
6746 Original commit message from CVS:
6747 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
6748 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps),
6749 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
6750 * sys/xvimage/xvimagesink.h:
6751 Avoid typechecking when we do trivial casts.
6752 Move error handling out of the main program flow.
6753 Sneak in the display-region caps property, not completely correct yet.
6754 Cache the width/height in buffer_alloc instead of parsing it from the
6757 2008-11-13 17:27:37 +0000 Wim Taymans <wim.taymans@gmail.com>
6759 gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
6760 Original commit message from CVS:
6761 * gst/playback/gstplaybin2.c: (deactivate_group):
6762 don't try to unlink the selector sinkpad when we don't have it yet. This
6763 can happen if an error occured before the group was complete.
6765 2008-11-13 15:37:40 +0000 Wim Taymans <wim.taymans@gmail.com>
6767 gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
6768 Original commit message from CVS:
6769 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
6770 (gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
6771 (gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
6772 (gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
6773 (gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
6774 (gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
6775 (gst_rtp_buffer_get_extension_data),
6776 (gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
6777 (gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
6778 (gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
6779 (gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
6780 (gst_rtp_buffer_get_payload_type),
6781 (gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
6782 (gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
6783 (gst_rtp_buffer_set_timestamp),
6784 (gst_rtp_buffer_get_payload_subbuffer),
6785 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
6786 Avoid expensive type checks we already did as part of the
6787 _validate() function that should be called first.
6789 2008-11-11 16:40:50 +0000 Wim Taymans <wim.taymans@gmail.com>
6791 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some cases where a newsegment event was not sent.
6792 Original commit message from CVS:
6793 * gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
6794 (gst_base_rtp_depayload_push_full),
6795 (gst_base_rtp_depayload_set_gst_timestamp):
6796 Fix some cases where a newsegment event was not sent.
6798 2008-11-11 15:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
6800 gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co...
6801 Original commit message from CVS:
6802 * gst/playback/gstplaybin2.c: (activate_group):
6803 Catch state change errors and stop from the uridecodebin elements
6804 instead of trying to continue in vain.
6806 2008-11-10 14:53:45 +0000 Edward Hervey <bilboed@bilboed.com>
6808 gst/: Wim, you're a bad boy. You don't want people to contact you or what?
6809 Original commit message from CVS:
6810 * gst-libs/gst/app/gstappsink.c:
6811 * gst-libs/gst/app/gstappsrc.c:
6812 * gst/h264parse/gsth264parse.c:
6813 Wim, you're a bad boy. You don't want people to contact you or what?
6815 2008-11-10 14:22:09 +0000 Wim Taymans <wim.taymans@gmail.com>
6817 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
6818 Original commit message from CVS:
6819 * gst-libs/gst/audio/gstbaseaudiosink.c:
6820 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
6821 (gst_base_audio_sink_callback):
6822 Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
6823 for the latency to expire, fixes #559567.
6825 2008-11-10 13:55:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
6827 gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
6828 Original commit message from CVS:
6829 * gst/adder/gstadder.c:
6830 Change author string after seeing output of gst-inspector.
6832 2008-11-10 10:33:26 +0000 Wim Taymans <wim.taymans@gmail.com>
6834 gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559...
6835 Original commit message from CVS:
6836 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
6837 Don't try to do crazy things when we only have a text pad without a
6838 video pad. Fixes #559478.
6840 2008-11-07 17:35:46 +0000 Wim Taymans <wim.taymans@gmail.com>
6842 gst-libs/gst/app/gstappsrc.*: Add is-live property.
6843 Original commit message from CVS:
6844 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
6845 (gst_app_src_init), (gst_app_src_set_property),
6846 (gst_app_src_get_property), (gst_app_src_push_buffer):
6847 * gst-libs/gst/app/gstappsrc.h:
6848 Add is-live property.
6851 2008-11-06 12:14:51 +0000 Wim Taymans <wim.taymans@gmail.com>
6853 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...
6854 Original commit message from CVS:
6855 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
6856 Fix case where we don't have a range for the rates or channels as is the
6857 case with truespeech.
6859 2008-11-05 19:18:25 +0000 Wim Taymans <wim.taymans@gmail.com>
6861 gst/volume/gstvolume.*: Keep negotiated state in a separate variable.
6862 Original commit message from CVS:
6863 * gst/volume/gstvolume.c: (volume_update_real_volume),
6864 (gst_volume_set_volume), (gst_volume_get_volume),
6865 (gst_volume_set_mute), (gst_volume_init), (volume_setup),
6866 (volume_transform_ip), (volume_update_mute),
6867 (volume_update_volume), (volume_get_property):
6868 * gst/volume/gstvolume.h:
6869 Keep negotiated state in a separate variable.
6870 Protect the volume and mute properties with the object lock.
6871 Protect modifying the transform with the transform lock.
6873 2008-11-05 12:20:21 +0000 Wim Taymans <wim.taymans@gmail.com>
6875 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Only convert caps to string when debug is enabled.
6876 Original commit message from CVS:
6877 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
6878 (gst_ffmpeg_pixfmt_to_caps):
6879 Only convert caps to string when debug is enabled.
6881 2008-11-04 18:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
6883 ext/theora/: Copy seqnum.
6884 Original commit message from CVS:
6885 * ext/theora/gsttheoradec.h:
6886 * ext/theora/theoradec.c: (gst_theora_dec_init),
6887 (gst_theora_dec_reset), (theora_dec_src_event),
6888 (theora_dec_sink_event), (theora_handle_type_packet):
6890 Keep events in a pending list, like vorbisdec, instead of trying
6891 to construct a segment event ourselves.
6892 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
6893 (vorbis_dec_src_event), (vorbis_dec_sink_event):
6894 * ext/vorbis/vorbisdec.h:
6897 2008-11-04 17:24:35 +0000 Wim Taymans <wim.taymans@gmail.com>
6899 ext/ogg/gstoggdemux.*: Copy seqnums around to track playback segments and messages.
6900 Original commit message from CVS:
6901 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
6902 (gst_ogg_demux_deactivate_current_chain),
6903 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page),
6904 (gst_ogg_demux_loop):
6905 * ext/ogg/gstoggdemux.h:
6906 Copy seqnums around to track playback segments and messages.
6908 2008-11-04 12:42:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
6910 Don't install static libs for plugins. Fixes #550851 for -bad.
6911 Original commit message from CVS:
6912 * ext/alsaspdif/Makefile.am:
6913 * ext/amrwb/Makefile.am:
6914 * ext/apexsink/Makefile.am:
6915 * ext/arts/Makefile.am:
6916 * ext/artsd/Makefile.am:
6917 * ext/audiofile/Makefile.am:
6918 * ext/audioresample/Makefile.am:
6919 * ext/bz2/Makefile.am:
6920 * ext/cdaudio/Makefile.am:
6921 * ext/celt/Makefile.am:
6922 * ext/dc1394/Makefile.am:
6923 * ext/dirac/Makefile.am:
6924 * ext/directfb/Makefile.am:
6925 * ext/divx/Makefile.am:
6926 * ext/dts/Makefile.am:
6927 * ext/faac/Makefile.am:
6928 * ext/faad/Makefile.am:
6929 * ext/gsm/Makefile.am:
6930 * ext/hermes/Makefile.am:
6931 * ext/ivorbis/Makefile.am:
6932 * ext/jack/Makefile.am:
6933 * ext/jp2k/Makefile.am:
6934 * ext/ladspa/Makefile.am:
6935 * ext/lcs/Makefile.am:
6936 * ext/libfame/Makefile.am:
6937 * ext/libmms/Makefile.am:
6938 * ext/metadata/Makefile.am:
6939 * ext/mpeg2enc/Makefile.am:
6940 * ext/mplex/Makefile.am:
6941 * ext/musepack/Makefile.am:
6942 * ext/musicbrainz/Makefile.am:
6943 * ext/mythtv/Makefile.am:
6944 * ext/nas/Makefile.am:
6945 * ext/neon/Makefile.am:
6946 * ext/ofa/Makefile.am:
6947 * ext/polyp/Makefile.am:
6948 * ext/resindvd/Makefile.am:
6949 * ext/sdl/Makefile.am:
6950 * ext/shout/Makefile.am:
6951 * ext/snapshot/Makefile.am:
6952 * ext/sndfile/Makefile.am:
6953 * ext/soundtouch/Makefile.am:
6954 * ext/spc/Makefile.am:
6955 * ext/swfdec/Makefile.am:
6956 * ext/tarkin/Makefile.am:
6957 * ext/theora/Makefile.am:
6958 * ext/timidity/Makefile.am:
6959 * ext/twolame/Makefile.am:
6960 * ext/x264/Makefile.am:
6961 * ext/xine/Makefile.am:
6962 * ext/xvid/Makefile.am:
6963 * gst-libs/gst/app/Makefile.am:
6964 * gst-libs/gst/dshow/Makefile.am:
6965 * gst/aiffparse/Makefile.am:
6966 * gst/app/Makefile.am:
6967 * gst/audiobuffer/Makefile.am:
6968 * gst/bayer/Makefile.am:
6969 * gst/cdxaparse/Makefile.am:
6970 * gst/chart/Makefile.am:
6971 * gst/colorspace/Makefile.am:
6972 * gst/dccp/Makefile.am:
6973 * gst/deinterlace/Makefile.am:
6974 * gst/deinterlace2/Makefile.am:
6975 * gst/dvdspu/Makefile.am:
6976 * gst/festival/Makefile.am:
6977 * gst/filter/Makefile.am:
6978 * gst/flacparse/Makefile.am:
6979 * gst/flv/Makefile.am:
6980 * gst/games/Makefile.am:
6981 * gst/h264parse/Makefile.am:
6982 * gst/librfb/Makefile.am:
6983 * gst/mixmatrix/Makefile.am:
6984 * gst/modplug/Makefile.am:
6985 * gst/mpeg1sys/Makefile.am:
6986 * gst/mpeg4videoparse/Makefile.am:
6987 * gst/mpegdemux/Makefile.am:
6988 * gst/mpegtsmux/Makefile.am:
6989 * gst/mpegvideoparse/Makefile.am:
6990 * gst/mve/Makefile.am:
6991 * gst/nsf/Makefile.am:
6992 * gst/nuvdemux/Makefile.am:
6993 * gst/overlay/Makefile.am:
6994 * gst/passthrough/Makefile.am:
6995 * gst/pcapparse/Makefile.am:
6996 * gst/playondemand/Makefile.am:
6997 * gst/rawparse/Makefile.am:
6998 * gst/real/Makefile.am:
6999 * gst/rtjpeg/Makefile.am:
7000 * gst/rtpmanager/Makefile.am:
7001 * gst/scaletempo/Makefile.am:
7002 * gst/sdp/Makefile.am:
7003 * gst/selector/Makefile.am:
7004 * gst/smooth/Makefile.am:
7005 * gst/smoothwave/Makefile.am:
7006 * gst/speed/Makefile.am:
7007 * gst/speexresample/Makefile.am:
7008 * gst/stereo/Makefile.am:
7009 * gst/subenc/Makefile.am:
7010 * gst/tta/Makefile.am:
7011 * gst/vbidec/Makefile.am:
7012 * gst/videodrop/Makefile.am:
7013 * gst/videosignal/Makefile.am:
7014 * gst/virtualdub/Makefile.am:
7015 * gst/vmnc/Makefile.am:
7016 * gst/y4m/Makefile.am:
7017 * sys/acmenc/Makefile.am:
7018 * sys/cdrom/Makefile.am:
7019 * sys/dshowdecwrapper/Makefile.am:
7020 * sys/dshowsrcwrapper/Makefile.am:
7021 * sys/dvb/Makefile.am:
7022 * sys/dxr3/Makefile.am:
7023 * sys/fbdev/Makefile.am:
7024 * sys/oss4/Makefile.am:
7025 * sys/qcam/Makefile.am:
7026 * sys/qtwrapper/Makefile.am:
7027 * sys/vcd/Makefile.am:
7028 * sys/wininet/Makefile.am:
7029 * win32/common/config.h:
7030 Don't install static libs for plugins. Fixes #550851 for -bad.
7032 2008-11-03 15:30:14 +0000 Matthias Kretz <kretz@kde.org>
7034 ext/alsa/gstalsasink.c: Make all access non-blocking so that we can better handle unplugging of usb devices. Fixes #5...
7035 Original commit message from CVS:
7036 Based on patch by: Matthias Kretz <kretz at kde dot org>
7037 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
7038 (gst_alsasink_prepare), (gst_alsasink_unprepare),
7039 (gst_alsasink_write):
7040 Make all access non-blocking so that we can better handle unplugging
7041 of usb devices. Fixes #559111
7043 2008-11-03 10:49:24 +0000 Damien Lespiau <damien.lespiau@gmail.com>
7045 gst-libs/gst/rtsp/gstrtspconnection.c: Make the next call to poll not depend on previous calls to poll with or withou...
7046 Original commit message from CVS:
7047 Patch by: Damien Lespiau <damien.lespiau gmail com>
7048 * gst-libs/gst/rtsp/gstrtspconnection.c:
7049 (gst_rtsp_connection_write):
7050 Make the next call to poll not depend on previous calls to poll with or
7051 without reading from the active descriptor. Fixes #544293.
7053 2008-11-03 08:55:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7055 gst/speexresample/gstspeexresample.c: Add TODO at the top of the file for enabling SSE/ARM specific optimizations and...
7056 Original commit message from CVS:
7057 * gst/speexresample/gstspeexresample.c:
7058 (gst_speex_resample_convert_buffer):
7059 Add TODO at the top of the file for enabling SSE/ARM specific
7060 optimizations and choosing the fastest implementation at runtime.
7061 Add g_assert_not_reached() at two places that should really never
7064 2008-11-02 09:19:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7066 gst/speexresample/gstspeexresample.c: Fix format string and arguments.
7067 Original commit message from CVS:
7068 * gst/speexresample/gstspeexresample.c:
7069 (gst_speex_resample_check_discont):
7070 Fix format string and arguments.
7071 * gst/speexresample/resample_sse.h:
7074 2008-11-01 19:38:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7076 gst/speexresample/: Add missing headers to Makefile.am.
7077 Original commit message from CVS:
7078 * gst/speexresample/Makefile.am:
7079 * gst/speexresample/gstspeexresample.c:
7080 (gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
7081 (gst_speex_resample_convert_buffer), (_benchmark_int_float),
7082 (_benchmark_int_int), (_benchmark_integer_resampling),
7084 * gst/speexresample/gstspeexresample.h:
7085 * gst/speexresample/resample.c:
7086 * gst/speexresample/speex_resampler_double.c:
7087 * gst/speexresample/speex_resampler_float.c:
7088 * gst/speexresample/speex_resampler_int.c:
7089 * gst/speexresample/speex_resampler_wrapper.h:
7090 Add missing headers to Makefile.am.
7091 Update copyright, years and my mail address.
7092 Benchmark the integer resampling implementation against the
7093 float implementation and use the faster one for 8/16 bit integer
7094 input. On most recent systems the floating point version is faster.
7096 2008-10-31 09:49:57 +0000 Nick Haddad <nick@haddads.net>
7098 gst-libs/gst/riff/: Add support for other fourcc codes that are commonly used for 'uncompressed RGB', including 'RGB ...
7099 Original commit message from CVS:
7100 Patch by: Nick Haddad <nick at haddads dot net>
7101 * gst-libs/gst/riff/riff-ids.h:
7102 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
7103 Add support for other fourcc codes that are commonly used for
7104 'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
7107 2008-10-30 14:55:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7109 gst/speexresample/gstspeexresample.c: The length for the buffer conversion function is the number of audio frames, i....
7110 Original commit message from CVS:
7111 * gst/speexresample/gstspeexresample.c:
7112 (gst_speex_resample_convert_buffer):
7113 The length for the buffer conversion function is the number of
7114 audio frames, i.e. we need to multiply it by the number of channels
7115 to get the number of values. Also spotted by the unit test after
7116 running in valgrind.
7118 2008-10-30 14:46:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7120 tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
7121 Original commit message from CVS:
7122 * tests/check/elements/speexresample.c: (element_message_cb),
7123 (eos_message_cb), (test_pipeline), (GST_START_TEST),
7124 (speexresample_suite):
7125 Add pipeline unit tests for testing all supported formats with
7126 up/downsampling and different in/outrates.
7127 * gst/speexresample/gstspeexresample.c:
7128 (gst_speex_resample_push_drain), (gst_speex_resample_process):
7129 * gst/speexresample/speex_resampler_wrapper.h:
7130 Fix bugs identified by the testsuite.
7132 2008-10-30 13:44:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7134 gst/speexresample/: Add support for int8, int24 and int32 input by converting internally to/from int16 or double.
7135 Original commit message from CVS:
7136 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
7137 (gst_speex_resample_get_funcs),
7138 (gst_speex_resample_transform_size),
7139 (gst_speex_resample_convert_buffer),
7140 (gst_speex_resample_push_drain), (gst_speex_resample_process):
7141 * gst/speexresample/gstspeexresample.h:
7142 * gst/speexresample/speex_resampler_wrapper.h:
7143 Add support for int8, int24 and int32 input by converting internally
7144 to/from int16 or double.
7146 2008-10-30 12:43:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7148 Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
7149 Original commit message from CVS:
7150 * gst/speexresample/Makefile.am:
7151 * gst/speexresample/arch.h:
7152 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
7153 (gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
7154 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
7155 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
7156 (_gcd), (gst_speex_resample_transform_size),
7157 (gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
7158 (gst_speex_resample_process), (gst_speex_resample_transform),
7159 (gst_speex_resample_query), (gst_speex_resample_set_property):
7160 * gst/speexresample/gstspeexresample.h:
7161 * gst/speexresample/resample.c:
7162 * gst/speexresample/speex_resampler.h:
7163 * gst/speexresample/speex_resampler_double.c:
7164 * gst/speexresample/speex_resampler_wrapper.h:
7165 * tests/check/elements/speexresample.c: (setup_speexresample),
7166 (test_perfect_stream_instance), (GST_START_TEST),
7167 (test_discont_stream_instance):
7168 Add support for double samples as input and refactor the usage
7169 of the different compilation flavors of the speex resampler.
7171 2008-10-30 11:43:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7173 gst/audioresample/gstaudioresample.c: Return the result of parent_class->event().
7174 Original commit message from CVS:
7175 * gst/audioresample/gstaudioresample.c:
7176 Return the result of parent_class->event().
7178 2008-10-29 17:02:55 +0000 Wim Taymans <wim.taymans@gmail.com>
7180 gst-libs/gst/app/gstappsink.c: Fix the docs.
7181 Original commit message from CVS:
7182 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
7185 2008-10-29 12:11:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7187 gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
7188 Original commit message from CVS:
7189 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
7190 (gst_speex_resample_get_unit_size),
7191 (gst_speex_resample_push_drain), (gst_speex_resample_event),
7192 (gst_speex_resample_check_discont), (gst_speex_resample_process),
7193 (gst_speex_resample_transform):
7194 * gst/speexresample/gstspeexresample.h:
7195 Rewrite timestamp tracking to make it more robust and guarantee
7197 * tests/check/Makefile.am:
7198 * tests/check/elements/speexresample.c: (setup_speexresample),
7199 (cleanup_speexresample), (fail_unless_perfect_stream),
7200 (test_perfect_stream_instance), (GST_START_TEST),
7201 (test_discont_stream_instance), (live_switch_alloc_only_48000),
7202 (live_switch_get_sink_caps), (live_switch_push),
7203 (speexresample_suite):
7204 Add unit tests for speexresample based on the audioresample unit tests.
7206 2008-10-28 19:30:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7208 gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
7209 Original commit message from CVS:
7210 * gst/speexresample/gstspeexresample.c:
7211 (gst_speex_resample_get_unit_size),
7212 (gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
7213 (gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
7214 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
7215 (gst_speex_resample_push_drain), (gst_speex_resample_event),
7216 (gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
7217 (gst_speex_resample_process), (gst_speex_resample_transform),
7218 (gst_speex_resample_query), (gst_speex_resample_set_property):
7219 * gst/speexresample/gstspeexresample.h:
7220 Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
7221 instead of GST_DEBUG, ...
7223 2008-10-28 16:28:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7225 gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
7226 Original commit message from CVS:
7227 * gst/speexresample/gstspeexresample.c:
7228 (gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
7229 (gst_speex_resample_process):
7230 Fixate to the nearest supported rate instead of the first one.
7232 2008-10-28 16:25:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7234 gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
7235 Original commit message from CVS:
7236 * gst/audioresample/gstaudioresample.c:
7237 (gst_audioresample_class_init), (audioresample_fixate_caps):
7238 Fixate the rate to the nearest supported rate instead of
7239 the first one. Fixes bug #549510.
7241 2008-10-28 11:46:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7243 gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
7244 Original commit message from CVS:
7245 * gst/speexresample/README:
7246 * gst/speexresample/arch.h:
7247 * gst/speexresample/fixed_arm4.h:
7248 * gst/speexresample/fixed_arm5e.h:
7249 * gst/speexresample/fixed_bfin.h:
7250 * gst/speexresample/fixed_debug.h:
7251 * gst/speexresample/fixed_generic.h:
7252 * gst/speexresample/resample.c: (compute_func), (main), (sinc),
7253 (cubic_coef), (resampler_basic_direct_single),
7254 (resampler_basic_direct_double),
7255 (resampler_basic_interpolate_single),
7256 (resampler_basic_interpolate_double), (update_filter),
7257 (speex_resampler_init_frac), (speex_resampler_process_native),
7258 (speex_resampler_magic), (speex_resampler_process_float),
7259 (speex_resampler_process_int),
7260 (speex_resampler_process_interleaved_float),
7261 (speex_resampler_process_interleaved_int),
7262 (speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
7263 (speex_resampler_reset_mem):
7264 * gst/speexresample/speex_resampler.h:
7265 Update Speex resampler with latest version from Speex GIT.
7267 2008-10-27 14:57:34 +0000 Wim Taymans <wim.taymans@gmail.com>
7269 win32/common/libgstaudio.def: Add new symbols.
7270 Original commit message from CVS:
7271 * win32/common/libgstaudio.def:
7274 2008-10-23 09:57:06 +0000 Wim Taymans <wim.taymans@gmail.com>
7276 ext/vorbis/vorbisdec.c: Attempt to make obfuscated code clearer.
7277 Original commit message from CVS:
7278 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
7279 Attempt to make obfuscated code clearer.
7281 2008-10-23 07:11:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7283 Move float endianness conversion macros to core. Second part of bug ##555196.
7284 Original commit message from CVS:
7285 * docs/libs/gst-plugins-base-libs-sections.txt:
7286 * gst-libs/gst/floatcast/floatcast.h:
7287 Move float endianness conversion macros to core. Second part of
7290 2008-10-22 12:29:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7292 sys/: Don't mark as gtk-doc docs as they aren't public.
7293 Original commit message from CVS:
7294 * sys/ximage/ximagesink.h:
7295 * sys/xvimage/xvimagesink.h:
7296 Don't mark as gtk-doc docs as they aren't public.
7298 2008-10-22 12:25:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7300 Allow setting colorkey if possible. Implement property probe interface for optional X features (autopaint-colorkey, d...
7301 Original commit message from CVS:
7302 * sys/xvimage/xvimagesink.c:
7303 * sys/xvimage/xvimagesink.h:
7304 * tests/icles/Makefile.am:
7305 * tests/icles/test-colorkey.c:
7306 Allow setting colorkey if possible. Implement property probe interface
7307 for optional X features (autopaint-colorkey, double-buffer and
7308 colorkey). Fixes #554533
7310 2008-10-22 12:01:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7312 gst-libs/gst/tag/tags.c: Remove useless buffer size assignment. It already has this value.
7313 Original commit message from CVS:
7314 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
7315 Remove useless buffer size assignment. It already has this value.
7317 2008-10-20 15:35:37 +0000 Wim Taymans <wim.taymans@gmail.com>
7319 gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
7320 Original commit message from CVS:
7321 * gst-libs/gst/audio/gstaudiosink.c:
7322 (gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
7323 (gst_audioringbuffer_activate), (gst_audioringbuffer_release),
7324 (gst_audioringbuffer_stop):
7325 Implement a separate activate functions to start monitoring the segments
7326 or, in pull mode, pulling in data.
7327 * gst-libs/gst/audio/gstbaseaudiosink.c:
7328 (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
7329 (gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
7330 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
7331 (gst_base_audio_sink_activate_pull),
7332 (gst_base_audio_sink_async_play),
7333 (gst_base_audio_sink_change_state):
7334 Implement pad and element convert query function.
7335 Activate the ringbuffer.
7336 Use the segment last_stop value as the offset to pull.
7337 Use new basesink _do_preroll() method to preroll in the pulling thread.
7338 Take appropriate locking in the pulling thread.
7339 * gst-libs/gst/audio/gstringbuffer.h:
7342 2008-10-20 14:08:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7344 gst/typefind/gsttypefindfunctions.c: Improve MXF typefinding a bit by searching for a header partition pack instead o...
7345 Original commit message from CVS:
7346 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
7347 Improve MXF typefinding a bit by searching for a header partition
7348 pack instead of just a general partition pack and checking more
7349 bytes for valid values.
7351 2008-10-20 13:45:55 +0000 Wim Taymans <wim.taymans@gmail.com>
7353 tests/icles/.cvsignore: update ignore file.
7354 Original commit message from CVS:
7355 * tests/icles/.cvsignore:
7357 * tests/icles/Makefile.am:
7358 * tests/icles/test-box.c: (make_pipeline), (main):
7359 Add another interactive command line experimentation suite for
7360 dynamically boxing/cropping/saling an input video.
7362 2008-10-17 13:19:05 +0000 Wim Taymans <wim.taymans@gmail.com>
7364 Add methods to more accuratly control the pulling thread of a ringbuffer.
7365 Original commit message from CVS:
7366 * docs/libs/gst-plugins-base-libs-sections.txt:
7367 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
7368 (gst_ring_buffer_activate), (gst_ring_buffer_is_active):
7369 * gst-libs/gst/audio/gstringbuffer.h:
7370 Add methods to more accuratly control the pulling thread of a
7372 Add format conversion helper code to the ringbuffer.
7373 API: GstRingBuffer:gst_ring_buffer_activate()
7374 API: GstRingBuffer:gst_ring_buffer_is_active()
7375 API: GstRingBuffer:gst_ring_buffer_convert()
7377 2008-10-16 15:44:37 +0000 Wim Taymans <wim.taymans@gmail.com>
7379 gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
7380 Original commit message from CVS:
7381 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
7382 (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
7383 (gst_audioringbuffer_stop):
7384 Signal thread startup earlier so that we can immediatly go into pull
7385 mode when we have to and block on preroll.
7387 2008-10-16 15:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
7389 gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
7390 Original commit message from CVS:
7391 * gst-libs/gst/audio/gstringbuffer.c:
7392 (gst_ring_buffer_prepare_read):
7393 In pull mode we want the callback to prepull a buffer we can preroll on
7394 even when we are not yet playing.
7396 2008-10-16 15:07:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7398 Don't install static libs for plugins. Fixes #550851 for base.
7399 Original commit message from CVS:
7400 * ext/alsa/Makefile.am:
7401 * ext/cdparanoia/Makefile.am:
7402 * ext/gio/Makefile.am:
7403 * ext/gnomevfs/Makefile.am:
7404 * ext/libvisual/Makefile.am:
7405 * ext/ogg/Makefile.am:
7406 * ext/pango/Makefile.am:
7407 * ext/theora/Makefile.am:
7408 * ext/vorbis/Makefile.am:
7409 * gst/adder/Makefile.am:
7410 * gst/audioconvert/Makefile.am:
7411 * gst/audiorate/Makefile.am:
7412 * gst/audioresample/Makefile.am:
7413 * gst/audiotestsrc/Makefile.am:
7414 * gst/ffmpegcolorspace/Makefile.am:
7415 * gst/gdp/Makefile.am:
7416 * gst/playback/Makefile.am:
7417 * gst/subparse/Makefile.am:
7418 * gst/tcp/Makefile.am:
7419 * gst/typefind/Makefile.am:
7420 * gst/videorate/Makefile.am:
7421 * gst/videoscale/Makefile.am:
7422 * gst/videotestsrc/Makefile.am:
7423 * gst/volume/Makefile.am:
7424 * sys/v4l/Makefile.am:
7425 * sys/ximage/Makefile.am:
7426 * sys/xvimage/Makefile.am:
7427 Don't install static libs for plugins. Fixes #550851 for base.
7429 2008-10-16 13:50:00 +0000 Wim Taymans <wim.taymans@gmail.com>
7431 gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe...
7432 Original commit message from CVS:
7433 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
7434 Set the default blocksize to -1 because we will then use the configured
7435 samplesperbuffer to create our output buffer.
7437 2008-10-15 15:28:41 +0000 Edward Hervey <bilboed@bilboed.com>
7439 gst-libs/gst/riff/riff-media.c: Add mappping for the KMVC (Karl Morton's Video) Codec.
7440 Original commit message from CVS:
7441 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
7442 (gst_riff_create_video_template_caps):
7443 Add mappping for the KMVC (Karl Morton's Video) Codec.
7445 2008-10-15 14:25:50 +0000 Edward Hervey <bilboed@bilboed.com>
7447 gst/typefind/gsttypefindfunctions.c: Don't forget to advance the offset of what we're matching against, else we end u...
7448 Original commit message from CVS:
7449 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
7450 Don't forget to advance the offset of what we're matching against, else
7451 we end up in a forever loop.
7453 2008-10-15 11:25:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7455 gst/subparse/gstsubparse.c: Improve typefinding a bit. If we don't have a Unicode charset try GST_SUBTITLE_ENCODING a...
7456 Original commit message from CVS:
7457 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
7458 Improve typefinding a bit. If we don't have a Unicode charset
7459 try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
7461 2008-10-14 11:13:59 +0000 Edward Hervey <bilboed@bilboed.com>
7463 ext/theora/theoradec.c: Fix build on macosx.
7464 Original commit message from CVS:
7465 * ext/theora/theoradec.c: (theora_dec_decode_buffer):
7466 Fix build on macosx.
7468 2008-10-13 11:36:13 +0000 Robin Stocker <robin@nibor.org>
7470 ext/theora/: Parse input caps and make the PAR override the encoded PAR when specified by a container. Fixes #555699.
7471 Original commit message from CVS:
7472 Based on patch by: Robin Stocker <robin at nibor dot org>
7473 * ext/theora/gsttheoradec.h:
7474 * ext/theora/theoradec.c: (gst_theora_dec_init),
7475 (theora_dec_setcaps), (theora_handle_type_packet),
7476 (theora_dec_decode_buffer), (theora_dec_change_state):
7477 Parse input caps and make the PAR override the encoded PAR when
7478 specified by a container. Fixes #555699.
7480 2008-10-13 09:16:59 +0000 Wim Taymans <wim.taymans@gmail.com>
7482 gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY
7483 Original commit message from CVS:
7484 * gst-libs/gst/rtp/gstbasertpdepayload.c:
7485 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
7486 (gst_base_rtp_depayload_set_gst_timestamp),
7487 (gst_base_rtp_depayload_change_state):
7488 * gst-libs/gst/rtp/gstbasertpdepayload.h:
7489 Add some more G_LIKELY
7490 Fail when the setcaps function was not called.
7491 * gst-libs/gst/rtp/gstbasertppayload.c:
7492 (gst_basertppayload_set_outcaps):
7493 Propagate return value of setcaps.
7495 2008-10-13 08:58:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7497 gst/subparse/: Add support for UTF16/UTF32 subtitles as long as the first bytes of the first buffer contain the BOM. ...
7498 Original commit message from CVS:
7499 * gst/subparse/Makefile.am:
7500 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
7501 (gst_sub_parse_class_init), (gst_sub_parse_init),
7502 (gst_convert_to_utf8), (detect_encoding), (convert_encoding),
7503 (get_next_line), (gst_sub_parse_data_format_autodetect),
7504 (feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
7505 (gst_subparse_type_find):
7506 * gst/subparse/gstsubparse.h:
7507 Add support for UTF16/UTF32 subtitles as long as the first bytes of
7508 the first buffer contain the BOM. This also adds support for other
7509 encodings that allow NUL bytes via the encoding property.
7510 Fixes bugs #552237 and #456788.
7512 2008-10-13 08:15:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7514 gst-libs/gst/tag/tags.c: Don't drop the last byte of image tags if they're not an URI list.
7515 Original commit message from CVS:
7516 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
7517 Don't drop the last byte of image tags if they're not an URI list.
7520 2008-10-13 08:00:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7522 gst/typefind/gsttypefindfunctions.c: For looking at the 4th byte we have to get 4 bytes of course and not 3.
7523 Original commit message from CVS:
7524 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
7525 For looking at the 4th byte we have to get 4 bytes of course
7528 2008-10-13 07:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7530 gst/typefind/gsttypefindfunctions.c: Improve FLAC-without-headers typefinding by looking at most of the frame header ...
7531 Original commit message from CVS:
7532 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
7533 Improve FLAC-without-headers typefinding by looking at most of the
7534 frame header and checking if invalid values are used. Should prevent
7535 quite some false positives compared to the old version which only
7536 check if the first 14 bits are set.
7538 2008-10-11 16:27:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7540 sys/xvimage/xvimagesink.c: Don't assert on caps==NULL.
7541 Original commit message from CVS:
7542 * sys/xvimage/xvimagesink.c:
7543 Don't assert on caps==NULL.
7545 2008-10-10 17:13:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7547 Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
7548 Original commit message from CVS:
7549 * gst/subparse/gstsubparse.c:
7550 (gst_sub_parse_data_format_autodetect), (handle_buffer),
7551 (gst_sub_parse_change_state):
7552 * gst/subparse/gstsubparse.h:
7553 * tests/check/elements/subparse.c: (GST_START_TEST):
7554 Add support for subtitle files with UTF-8 BOM at the beginning
7555 by simple stripping it from the first line before passing it
7556 to any parsing code. Fixes bug #555257 and playback of files
7557 created by Gnome Subtitles.
7559 2008-10-10 15:45:15 +0000 Wim Taymans <wim.taymans@gmail.com>
7561 gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place.
7562 Original commit message from CVS:
7563 * gst/audiotestsrc/gstaudiotestsrc.c:
7564 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
7565 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
7566 (gst_audio_test_src_start), (gst_audio_test_src_stop),
7567 (gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
7568 (gst_audio_test_src_create):
7569 * gst/audiotestsrc/gstaudiotestsrc.h:
7570 Define the default property values in the usual place.
7571 Implement start/stop to reset values correctly.
7572 Calculate the sample size only once when we negotiate.
7573 Rename some values to make more sense.
7574 Keep track of our byte range.
7575 Add support for pull based scheduling. Disabled for now until we have
7576 the whole stack working.
7577 Set the BUFFER_OFFSET correctly.
7579 2008-10-10 15:32:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7581 Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
7582 Original commit message from CVS:
7583 Based on a patch by: xavierb at gmail dot com
7584 * gst/subparse/gstsubparse.c:
7585 (gst_sub_parse_data_format_autodetect):
7586 * tests/check/elements/subparse.c: (GST_START_TEST):
7587 Make the detection of the used subtitle a bit less strict
7588 for srt subtitles. Fixes bug #555607.
7590 2008-10-10 15:21:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7592 ext/vorbis/vorbisenc.c: Fix discontinuity detection which was broken by last commit.
7593 Original commit message from CVS:
7594 * ext/vorbis/vorbisenc.c:
7595 (gst_vorbis_enc_buffer_check_discontinuous):
7596 Fix discontinuity detection which was broken by last commit.
7598 2008-10-09 11:18:09 +0000 Tim-Philipp Müller <tim@centricular.net>
7600 configure.ac: Require core CVS for ghostpad API additions used by decodebin2.
7601 Original commit message from CVS:
7603 Require core CVS for ghostpad API additions used by decodebin2.
7605 2008-10-08 15:30:33 +0000 Edward Hervey <bilboed@bilboed.com>
7607 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
7608 Original commit message from CVS:
7609 * gst-libs/gst/audio/gstbaseaudiosrc.c:
7610 (gst_base_audio_src_create):
7611 Fix debug statements (space between '%' and actual format).
7613 2008-10-08 14:44:04 +0000 Wim Taymans <wim.taymans@gmail.com>
7615 gst/playback/gstdecodebin2.c: Remove bogus assert, the decodepad could have been created inside an already existing g...
7616 Original commit message from CVS:
7617 * gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
7618 Remove bogus assert, the decodepad could have been created inside an
7619 already existing group.
7621 2008-10-08 14:01:42 +0000 Andy Wingo <wingo@pobox.com>
7625 Original commit message from CVS:
7628 2008-10-08 14:00:07 +0000 Andy Wingo <wingo@pobox.com>
7630 gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
7631 Original commit message from CVS:
7632 2008-10-08 Andy Wingo <wingo@pobox.com>
7633 * gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
7634 target instead of setting it.
7635 (gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
7636 API for a decode pad. The bugfix is that we set the group in
7637 activate(), not when the pad was created because it might be NULL
7639 (gst_decode_group_control_source_pad, gst_decode_group_expose):
7640 Update to use the API.
7642 2008-10-08 12:49:40 +0000 Andy Wingo <wingo@pobox.com>
7644 gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
7645 Original commit message from CVS:
7646 2008-10-08 Andy Wingo <wingo@pobox.com>
7647 * gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
7648 be a subclass of GstGhostPad.
7649 (analyze_new_pad): So, when emitting the signals that determine
7650 how we do autoplugging, already create the ghost pad and use it as
7651 the pad in the signal arguments. This allows applications to make
7652 a connection between the pad passed in e.g. autoplug-continue, and
7653 the pad passed in new-decoded-pad.
7654 (connect_pad, expose_pad): Update to receive the ghosted decode
7655 pad in the args, retargetting it as necessary if we have to plug
7656 the target pad through a multiqueue.
7657 (gst_decode_group_control_source_pad): Adapt to receive an
7658 already-ghosted pad that just needs activation, blocking, and
7660 (sort_end_pads): Adapt for decode pads actually being pads.
7661 (gst_decode_group_expose): Adapt for decode pads actually being
7662 pads. Rewrite the decode pad names so they appear in order. Adds a
7663 new error case if we couldn't set the name.
7664 (gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
7666 (gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
7667 New API for the decode pad, needed because we shouldn't do these
7668 things inside gst_decode_pad_new(), but after.
7669 (gst_decode_pad_new): Change to actually make the real pad, and
7670 delay the blocking/drainage bits.
7672 2008-10-08 12:12:01 +0000 Daniel Drake <dsd@laptop.org>
7674 ext/ogg/gstoggmux.c: Unref all buffers when clearing collectpads. Fixes bug #546955.
7675 Original commit message from CVS:
7676 Patch by: Daniel Drake <dsd at laptop dot org>
7677 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
7678 Unref all buffers when clearing collectpads. Fixes bug #546955.
7680 2008-10-08 12:08:01 +0000 Klaas <klaas@rivercrew.net>
7682 ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
7683 Original commit message from CVS:
7684 Based on a patch by: Klaas <klaas at rivercrew dot net>
7685 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
7686 (gst_vorbis_enc_buffer_check_discontinuous),
7687 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
7688 * ext/vorbis/vorbisenc.h:
7689 Keep track of the upstream segments and use the running time on that
7690 segment instead of the buffer timestamp everywhere. Fixes bug #525807.
7692 2008-10-08 11:50:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7694 gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
7695 Original commit message from CVS:
7696 * gst/audioconvert/audioconvert.c: (audio_convert_convert):
7697 Prevent overflows with big buffer when calculating the size of
7698 the intermediate buffer by using gst_util_uint64_scale() instead of
7699 plain arithmetics. Fixes bug #552801.
7701 2008-10-08 10:49:15 +0000 Pavel Zeldin <pzeldin@gmail.com>
7703 ext/pango/gstclockoverlay.*: API: Add ability to specify format for date/time display by adding a "time-format" prope...
7704 Original commit message from CVS:
7705 Patch by: Pavel Zeldin <pzeldin at gmail dot com>
7706 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
7707 (gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
7708 (gst_clock_overlay_init), (gst_clock_overlay_set_property),
7709 (gst_clock_overlay_get_property):
7710 * ext/pango/gstclockoverlay.h:
7711 API: Add ability to specify format for date/time display by
7712 adding a "time-format" property.
7715 2008-10-08 09:22:26 +0000 Jan Gerber <j@oil21.org>
7717 gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
7718 Original commit message from CVS:
7719 Patch by: Jan Gerber <j at oil21 dot org>
7720 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
7721 (gst_riff_create_video_template_caps):
7722 Add FFV1 fourcc to support playback of FFMPEG lossless video
7723 in AVI. Fixes bug #555319.
7725 2008-10-08 09:12:36 +0000 Håvard Graff <havard.graff@tandberg.com>
7727 gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
7728 Original commit message from CVS:
7729 Patch by: Håvard Graff <havard dot graff at tandberg dot com>
7730 * gst-libs/gst/audio/gstbaseaudiosrc.c:
7731 (gst_base_audio_src_create):
7732 Implement skew clock slaving. Fixes #552559.
7734 2008-10-08 09:10:23 +0000 Wim Taymans <wim.taymans@gmail.com>
7736 gst-libs/gst/audio/: Fix include of config.h
7737 Original commit message from CVS:
7738 * gst-libs/gst/audio/multichannel.c:
7739 * gst-libs/gst/audio/testchannels.c:
7740 Fix include of config.h
7742 2008-10-06 16:36:20 +0000 Tero Saarni <tero.saarni@gmail.com>
7744 gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
7745 Original commit message from CVS:
7746 Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
7747 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
7748 (print_media), (gst_sdp_message_dump):
7749 Fix parsing of the c= field containing multicast addresses.
7751 Add the connection info to the session or streams.
7752 Fix parsing of the bandwidth.
7753 Add debugging for the connections and bandwidths for a media.
7754 Add debugging for the bandwidth of the session.
7756 2008-10-06 16:31:27 +0000 Wim Taymans <wim.taymans@gmail.com>
7758 gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
7759 Original commit message from CVS:
7760 * gst-libs/gst/rtp/gstbasertppayload.c:
7761 (gst_basertppayload_change_state):
7762 Configure the next seqnum and timestamp in the state change so that they
7763 can be queried soon after.
7765 2008-10-06 16:29:33 +0000 Wim Taymans <wim.taymans@gmail.com>
7767 gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
7768 Original commit message from CVS:
7769 * gst-libs/gst/rtp/gstbasertpdepayload.c:
7770 (gst_base_rtp_depayload_chain):
7771 Improve debugging of the rtptime.
7773 2008-10-05 11:33:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7775 configure.ac: Back to development -> 0.10.21.1
7776 Original commit message from CVS:
7778 Back to development -> 0.10.21.1
7780 2008-10-05 08:18:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7784 Original commit message from CVS:
7787 2008-10-05 08:11:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7789 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
7790 Original commit message from CVS:
7791 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
7793 Add typefinder for MXF.
7795 2008-10-05 08:10:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7797 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
7798 Original commit message from CVS:
7799 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
7801 Add typefinder for MXF.
7803 2008-10-03 15:19:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7805 tests/icles/Makefile.am: Only build test-colorkey if GTK+ is available.
7806 Original commit message from CVS:
7807 * tests/icles/Makefile.am:
7808 Only build test-colorkey if GTK+ is available.
7810 === release 0.10.21 ===
7812 2008-10-03 00:03:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7818 * docs/plugins/gst-plugins-base-plugins.args:
7819 * docs/plugins/gst-plugins-base-plugins.hierarchy:
7820 * docs/plugins/gst-plugins-base-plugins.interfaces:
7821 * docs/plugins/gst-plugins-base-plugins.prerequisites:
7822 * docs/plugins/inspect/plugin-adder.xml:
7823 * docs/plugins/inspect/plugin-alsa.xml:
7824 * docs/plugins/inspect/plugin-audioconvert.xml:
7825 * docs/plugins/inspect/plugin-audiorate.xml:
7826 * docs/plugins/inspect/plugin-audioresample.xml:
7827 * docs/plugins/inspect/plugin-audiotestsrc.xml:
7828 * docs/plugins/inspect/plugin-cdparanoia.xml:
7829 * docs/plugins/inspect/plugin-decodebin.xml:
7830 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
7831 * docs/plugins/inspect/plugin-gdp.xml:
7832 * docs/plugins/inspect/plugin-gio.xml:
7833 * docs/plugins/inspect/plugin-gnomevfs.xml:
7834 * docs/plugins/inspect/plugin-libvisual.xml:
7835 * docs/plugins/inspect/plugin-ogg.xml:
7836 * docs/plugins/inspect/plugin-pango.xml:
7837 * docs/plugins/inspect/plugin-playback.xml:
7838 * docs/plugins/inspect/plugin-queue2.xml:
7839 * docs/plugins/inspect/plugin-subparse.xml:
7840 * docs/plugins/inspect/plugin-tcp.xml:
7841 * docs/plugins/inspect/plugin-theora.xml:
7842 * docs/plugins/inspect/plugin-typefindfunctions.xml:
7843 * docs/plugins/inspect/plugin-uridecodebin.xml:
7844 * docs/plugins/inspect/plugin-video4linux.xml:
7845 * docs/plugins/inspect/plugin-videorate.xml:
7846 * docs/plugins/inspect/plugin-videoscale.xml:
7847 * docs/plugins/inspect/plugin-videotestsrc.xml:
7848 * docs/plugins/inspect/plugin-volume.xml:
7849 * docs/plugins/inspect/plugin-vorbis.xml:
7850 * docs/plugins/inspect/plugin-ximagesink.xml:
7851 * docs/plugins/inspect/plugin-xvimagesink.xml:
7852 * gst-plugins-base.doap:
7853 * win32/common/config.h:
7855 Original commit message from CVS:
7858 2008-10-02 23:44:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7889 Original commit message from CVS:
7892 2008-09-28 22:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7894 configure.ac: 0.10.20.4 pre-release
7895 Original commit message from CVS:
7897 0.10.20.4 pre-release
7899 2008-09-25 10:46:00 +0000 ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>
7901 ext/theora/theoraparse.c: Set the BOS flag on the BOS packet. Fixes #553244.
7902 Original commit message from CVS:
7903 Patch by: ogg.k.ogg.k <ogg dot k dot ogg dot k at googlemail dot com>
7904 * ext/theora/theoraparse.c: (theora_parse_set_streamheader):
7905 Set the BOS flag on the BOS packet. Fixes #553244.
7907 2008-09-23 17:48:14 +0000 Wim Taymans <wim.taymans@gmail.com>
7909 gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
7910 Original commit message from CVS:
7911 * gst-libs/gst/rtsp/gstrtspmessage.c:
7912 (gst_rtsp_message_parse_request),
7913 (gst_rtsp_message_parse_response):
7914 Fix the g_return_val_if_fail() statements.
7916 2008-09-22 17:44:14 +0000 Michael Smith <msmith@xiph.org>
7918 gst-libs/gst/tag/gsttagdemux.c: Fail to activate if there's insufficient data in the file to be usable, preventing an...
7919 Original commit message from CVS:
7920 * gst-libs/gst/tag/gsttagdemux.c:
7921 Fail to activate if there's insufficient data in the file to be usable,
7922 preventing an assertion fail later. Fixes #552960
7924 2008-09-16 15:36:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7926 Commit stuff that should have gone in last week when I made the pre-releases:
7927 Original commit message from CVS:
7928 Commit stuff that should have gone in last week when I made the pre-releases:
7929 2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
7931 0.10.20.2 pre-release
7937 2008-09-15 15:11:18 +0000 Tim-Philipp Müller <tim@centricular.net>
7939 gst/: Recognise Kate subtitle streams (#550582).
7940 Original commit message from CVS:
7941 * gst-libs/gst/pbutils/descriptions.c:
7942 * gst/typefind/gsttypefindfunctions.c:
7943 Recognise Kate subtitle streams (#550582).
7945 2008-09-13 11:04:02 +0000 Tim-Philipp Müller <tim@centricular.net>
7947 gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
7948 Original commit message from CVS:
7949 * gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
7950 Remove trailing comma from enum list, which causes problems
7951 with -pendantic (#550729).
7953 2008-09-05 19:04:47 +0000 Tim-Philipp Müller <tim@centricular.net>
7955 gst-libs/gst/interfaces/propertyprobe.c: More sanity checks for our second-favourite interface.
7956 Original commit message from CVS:
7957 * gst-libs/gst/interfaces/propertyprobe.c:
7958 (gst_property_probe_get_properties),
7959 (gst_property_probe_get_property),
7960 (gst_property_probe_probe_property),
7961 (gst_property_probe_probe_property_name),
7962 (gst_property_probe_needs_probe),
7963 (gst_property_probe_needs_probe_name),
7964 (gst_property_probe_get_values),
7965 (gst_property_probe_get_values_name),
7966 (gst_property_probe_probe_and_get_values),
7967 (gst_property_probe_probe_and_get_values_name):
7968 More sanity checks for our second-favourite interface.
7970 2008-09-05 14:12:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7972 gst-libs/gst/interfaces/propertyprobe.c: Check for NULL pointer, in the hope that this fixes #532864.
7973 Original commit message from CVS:
7974 * gst-libs/gst/interfaces/propertyprobe.c:
7975 Check for NULL pointer, in the hope that this fixes #532864.
7977 2008-09-05 10:24:05 +0000 Tim-Philipp Müller <tim@centricular.net>
7979 sys/xvimage/xvimagesink.c: No really, the next release is 0.10.21 (fix Since: tags in docs).
7980 Original commit message from CVS:
7981 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
7982 No really, the next release is 0.10.21 (fix Since: tags in docs).
7984 2008-09-04 16:25:06 +0000 Wim Taymans <wim.taymans@gmail.com>
7986 gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
7987 Original commit message from CVS:
7988 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
7989 Disable a code path that is now called but causes a deadlock for some
7990 reason and is unneeded.
7992 2008-09-04 13:46:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7994 sys/xvimage/xvimagesink.*: Add a "draw-border" property that can be set to false to disable drawing borders.
7995 Original commit message from CVS:
7996 * sys/xvimage/xvimagesink.c:
7997 * sys/xvimage/xvimagesink.h:
7998 Add a "draw-border" property that can be set to false to disable
8000 * tests/icles/test-colorkey.c:
8001 * tests/icles/Makefile.am:
8002 Add new test application for the colorkey handling.
8004 2008-09-03 14:00:06 +0000 Edward Hervey <bilboed@bilboed.com>
8006 gst-libs/gst/riff/riff-media.c: Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
8007 Original commit message from CVS:
8008 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
8009 Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
8010 This will also be fixed for upcoming gst-ffmpeg release so that once
8011 this release of -base is out, it will work with the latest gst-ffmpeg
8014 2008-09-03 13:27:20 +0000 Edward Hervey <bilboed@bilboed.com>
8016 gst-libs/gst/riff/riff-media.c: Add Truespeech mapping for RIFF formats (AVI/WAV).
8017 Original commit message from CVS:
8018 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
8019 (gst_riff_create_audio_template_caps):
8020 Add Truespeech mapping for RIFF formats (AVI/WAV).
8023 2008-09-03 12:23:44 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
8025 gst/typefind/gsttypefindfunctions.c: Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
8026 Original commit message from CVS:
8027 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
8028 Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
8031 2008-09-03 10:12:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8033 Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
8034 Original commit message from CVS:
8036 * gst/subparse/Makefile.am:
8037 * gst/subparse/gstsubparse.c:
8038 * gst/subparse/samiparse.c:
8039 * tests/check/elements/subparse.c:
8040 Rework last change, so that we build subparse, but just disable the
8041 sami parse functionality, if we're configured to not use xml. In the
8042 tests only the sami test is disabled now.
8044 2008-09-02 15:07:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8046 configure.ac: Disable subparse when xml is disabled. It woundn't work anyway. Fixes test runs.
8047 Original commit message from CVS:
8049 Disable subparse when xml is disabled. It woundn't work anyway. Fixes
8052 2008-09-02 09:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
8054 po/POTFILES.in: Add some more files with strings for translation.
8055 Original commit message from CVS:
8057 Add some more files with strings for translation.
8059 2008-09-02 06:37:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8061 Use new geo location tags from core. Fixes #481169
8062 Original commit message from CVS:
8063 * gst-libs/gst/tag/gstvorbistag.c:
8064 * tests/check/libs/tag.c:
8065 Use new geo location tags from core. Fixes #481169
8067 2008-09-01 16:05:45 +0000 Edward Hervey <bilboed@bilboed.com>
8069 tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
8070 Original commit message from CVS:
8071 * tests/check/elements/audioresample.c: (setup_audioresample),
8072 (fail_unless_perfect_stream), (test_perfect_stream_instance),
8073 (test_discont_stream_instance):
8074 Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
8075 Add debugging for coherence.
8077 2008-08-30 15:55:06 +0000 Jonathan Matthew <notverysmart@gmail.com>
8079 gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
8080 Original commit message from CVS:
8081 Patch by: Jonathan Matthew <notverysmart gmail com>
8082 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
8083 Add typefinder for PDF documents (which is nice to have, since it's a
8084 common format, but also helps prevent false positives). Fixes #549814.
8086 2008-08-27 15:30:16 +0000 Wim Taymans <wim.taymans@gmail.com>
8088 gst/playback/gstplaybin2.c: Fix nasty race where multiple decodebins could start pushing data before we manage to con...
8089 Original commit message from CVS:
8090 * gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
8092 Fix nasty race where multiple decodebins could start pushing data before
8093 we manage to configure the sinks, resulting in not-linked errors in
8094 typical RTSP streaming cases.
8096 2008-08-26 17:24:31 +0000 Wim Taymans <wim.taymans@gmail.com>
8098 gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
8099 Original commit message from CVS:
8100 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
8101 Since we now call stop, we trigger this code path that causes a deadlock
8102 is apparently not needed.
8104 2008-08-26 15:45:36 +0000 Wim Taymans <wim.taymans@gmail.com>
8106 gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
8107 Original commit message from CVS:
8108 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
8109 (gst_ring_buffer_stop):
8110 Also allow the case where the ringbuffer was paused when we try to stop
8111 it so that the basesrc stop function is still called.
8113 2008-08-23 15:25:44 +0000 Mike Ruprecht <cmaiku@gmail.com>
8115 sys/v4l/gstv4lelement.c: Reprobe devices again instead of taking a cached list as new devices could've been plugged i...
8116 Original commit message from CVS:
8117 Patch by: Mike Ruprecht <cmaiku at gmail dot com>
8118 * sys/v4l/gstv4lelement.c: (gst_v4l_class_probe_devices):
8119 Reprobe devices again instead of taking a cached list as new
8120 devices could've been plugged in. Fixes bug #549062.
8122 2008-08-23 15:19:59 +0000 Alessandro Dessina <alessandro@nnva.org>
8124 ext/ogg/gstoggdemux.c: Don't add pads and activate them for skeleton streams. These are already handled inside oggdem...
8125 Original commit message from CVS:
8126 Patch by: Alessandro Dessina <alessandro nnva org>
8127 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
8128 (gst_ogg_demux_activate_chain):
8129 Don't add pads and activate them for skeleton streams. These are already
8130 handled inside oggdemux. Fixes bug #537599.
8132 2008-08-22 15:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
8134 ext/vorbis/vorbisdec.c: Reset variable so that query and convert fail after going back to
8135 Original commit message from CVS:
8136 * ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
8137 Reset variable so that query and convert fail after going back to
8138 READY. Fixes #548898.
8140 2008-08-22 07:24:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8142 ext/vorbis/vorbisenc.c: If a buffer arrives with a timestamp before the timestamp+duration of the previous buffer cli...
8143 Original commit message from CVS:
8144 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
8145 If a buffer arrives with a timestamp before the timestamp+duration
8146 of the previous buffer clip it instead of dropping it completely.
8147 Slight improvement for the unfixable bug #548913.
8149 2008-08-21 14:19:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8151 ext/vorbis/vorbisdec.c: Take the current timestamp instead of timestamp+duration for the offset.
8152 Original commit message from CVS:
8153 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
8154 Take the current timestamp instead of timestamp+duration for the offset.
8155 This offset will later be used for calculating the timestamp and
8156 otherwise vorbisdec will interpolate timestamps wrong if upstream
8157 only sends timestamps and no granulepos.
8159 2008-08-21 11:20:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8161 tests/examples/seek/seek.c: Don't crash when having no visualisations.
8162 Original commit message from CVS:
8163 * tests/examples/seek/seek.c:
8164 Don't crash when having no visualisations.
8166 2008-08-16 20:57:27 +0000 David Schleef <ds@schleef.org>
8168 gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
8169 Original commit message from CVS:
8170 * gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
8171 check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
8174 2008-08-15 07:24:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8176 gst-libs/gst/pbutils/missing-plugins.c: When cleaning up the caps fields also remove "depth" for the same reason we r...
8177 Original commit message from CVS:
8178 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
8179 When cleaning up the caps fields also remove "depth" for the same
8180 reason we remove "width".
8182 2008-08-14 17:14:53 +0000 Tim-Philipp Müller <tim@centricular.net>
8184 gst-libs/gst/pbutils/descriptions.c: Add Lead H.264 here as well.
8185 Original commit message from CVS:
8186 * gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
8187 Add Lead H.264 here as well.
8189 2008-08-14 15:17:31 +0000 Julien Moutte <julien@moutte.net>
8191 gst-libs/gst/riff/riff-media.c: Add Lead H.264 variant.
8192 Original commit message from CVS:
8193 2008-08-14 Julien Moutte <julien@fluendo.com>
8194 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
8195 (gst_riff_create_video_template_caps): Add Lead H.264 variant.
8197 2008-08-13 09:17:38 +0000 Wim Taymans <wim.taymans@gmail.com>
8199 gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
8200 Original commit message from CVS:
8201 * gst-libs/gst/audio/gstbaseaudiosrc.c:
8202 (gst_base_audio_src_create):
8203 When not slaved to another clock also subtract the base_time from our
8204 internal clock time to get the running time.
8206 2008-08-13 00:59:07 +0000 David Schleef <ds@schleef.org>
8208 ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate, since it has no basis in libtheora.
8209 Original commit message from CVS:
8210 * ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
8211 since it has no basis in libtheora.
8213 2008-08-12 06:31:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8215 gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
8216 Original commit message from CVS:
8217 * gst-libs/gst/interfaces/propertyprobe.h:
8218 Remove double "interface" from doc-string.
8219 * gst-libs/gst/interfaces/xoverlay.h:
8221 * gst-libs/gst/riff/riff.c:
8222 Add basic doc blobs.
8224 2008-08-11 15:05:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8226 gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
8227 Original commit message from CVS:
8228 * gst-libs/gst/audio/Makefile.am:
8229 Don't try to build that example anymore.
8231 2008-08-11 14:51:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8233 gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
8234 Original commit message from CVS:
8235 * gst-libs/gst/audio/.cvsignore:
8236 * gst-libs/gst/audio/Makefile.am:
8237 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
8238 * gst-libs/gst/audio/make_filter:
8239 Move audiofiltertemplate to gst-template.
8241 2008-08-11 09:20:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8243 More docs and shuffling. What can we do with the hundreds of #defines.
8244 Original commit message from CVS:
8245 * docs/libs/gst-plugins-base-libs-sections.txt:
8246 * gst-libs/gst/audio/gstaudiosrc.h:
8247 More docs and shuffling. What can we do with the hundreds of #defines.
8249 2008-08-11 08:34:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8251 gst-libs/gst/: Reducing number of dundocumented symbols.
8252 Original commit message from CVS:
8253 * gst-libs/gst/audio/audio.h:
8254 * gst-libs/gst/audio/gstaudiofilter.h:
8255 * gst-libs/gst/audio/gstringbuffer.h:
8256 * gst-libs/gst/interfaces/propertyprobe.h:
8257 * gst-libs/gst/tag/gsttagdemux.h:
8258 Reducing number of dundocumented symbols.
8260 2008-08-11 07:16:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8262 gst-libs/gst/audio/audio.c: Fix doc comment syntax.
8263 Original commit message from CVS:
8264 * gst-libs/gst/audio/audio.c:
8265 Fix doc comment syntax.
8266 * gst-libs/gst/interfaces/propertyprobe.c:
8267 Add more doc-comments and a FIXME: for the signal.
8269 2008-08-07 16:11:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8271 ext/ogg/gstoggmux.*: Don't pretend to support NEWSEGMENT events, instead override the
8272 Original commit message from CVS:
8273 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sink_event),
8274 (gst_ogg_mux_request_new_pad):
8275 * ext/ogg/gstoggmux.h:
8276 Don't pretend to support NEWSEGMENT events, instead override the
8277 GstCollectPads event function to return FALSE on NEWSEGMENT events
8278 and do the normal work for other events.
8279 This prevents elements like flacenc to seek to the start and rewrite
8280 some data which then results in a broken Ogg packet.
8282 2008-08-07 15:58:58 +0000 Frederic Crozat <fcrozat@mandriva.org>
8284 Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
8285 Original commit message from CVS:
8286 Patch by: Frederic Crozat <fcrozat@mandriva.org>
8287 * ext/alsa/gstalsaplugin.c: (plugin_init):
8288 * ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
8289 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
8290 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
8291 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
8292 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
8293 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
8294 * gst/playback/gstdecodebin.c: (plugin_init):
8295 * gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
8296 * gst/playback/gstplayback.c: (plugin_init):
8297 * gst/playback/gstqueue2.c: (plugin_init):
8298 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
8299 * sys/v4l/gstv4l.c: (plugin_init):
8300 Make sure gettext returns translations in UTF-8 encoding rather
8301 than in the current locale encoding (#546822).
8303 2008-08-06 13:12:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8305 gst-libs/gst/pbutils/descriptions.c: Add audio/x-qdm for qtdemux.
8306 Original commit message from CVS:
8307 * gst-libs/gst/pbutils/descriptions.c:
8308 Add audio/x-qdm for qtdemux.
8310 2008-08-05 15:38:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8312 ext/vorbis/vorbisdec.c: Do not leak old taglist.
8313 Original commit message from CVS:
8314 * ext/vorbis/vorbisdec.c:
8315 Do not leak old taglist.
8317 2008-08-04 12:35:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8319 tests/icles/test-scale.c: Include <stdlib.h> for atoi().
8320 Original commit message from CVS:
8321 * tests/icles/test-scale.c:
8322 Include <stdlib.h> for atoi().
8324 2008-08-04 09:11:08 +0000 Andy Wingo <wingo@pobox.com>
8326 gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
8327 Original commit message from CVS:
8328 2008-08-04 Andy Wingo <wingo@pobox.com>
8329 * gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
8332 2008-08-01 13:06:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8334 gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
8335 Original commit message from CVS:
8336 * gst/adder/gstadder.c:
8337 Cleanup lots of empty lines that came from gst-indent going havoc
8338 before I added the INDENT_ON/OFF marker some time agao.
8340 2008-08-01 11:55:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8342 Bump requirement to latest core and use new tag for riff formats.
8343 Original commit message from CVS:
8345 * gst-libs/gst/riff/riff-read.c:
8346 Bump requirement to latest core and use new tag for riff formats.
8349 2008-08-01 11:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
8351 tests/examples/dynamic/: Add example app that dynamically switches between 3 'encoders'.
8352 Original commit message from CVS:
8353 * tests/examples/dynamic/Makefile.am:
8354 * tests/examples/dynamic/codec-select.c: (make_encoder),
8355 (make_pipeline), (do_switch), (my_bus_callback), (main):
8356 Add example app that dynamically switches between 3 'encoders'.
8358 2008-07-31 13:06:13 +0000 Wim Taymans <wim.taymans@gmail.com>
8360 gst/playback/gstplaysink.c: Add some more comments.
8361 Original commit message from CVS:
8362 * gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
8363 Add some more comments.
8365 2008-07-31 12:58:44 +0000 Wim Taymans <wim.taymans@gmail.com>
8367 gst/videotestsrc/gstvideotestsrc.c: Discard buffers of the wrong size after renegotiation, this is perfectly possible...
8368 Original commit message from CVS:
8369 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
8370 (gst_video_test_src_create):
8371 Discard buffers of the wrong size after renegotiation, this is perfectly
8372 possible with things like capsfilter that could suggest caps changes
8373 upstream without knowing the size of the buffer.
8375 2008-07-31 11:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
8377 tests/icles/: Add dynamic rescaling tests for the new basetransform.
8378 Original commit message from CVS:
8379 * tests/icles/.cvsignore:
8380 * tests/icles/Makefile.am:
8381 * tests/icles/test-scale.c: (make_pipeline), (main):
8382 Add dynamic rescaling tests for the new basetransform.
8384 2008-07-30 19:51:36 +0000 Tim-Philipp Müller <tim@centricular.net>
8386 gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
8387 Original commit message from CVS:
8388 * gst/audioconvert/Makefile.am:
8389 Dist recently-added gstfastrandom.h.
8391 2008-07-30 15:29:44 +0000 Edward Hervey <bilboed@bilboed.com>
8393 sys/xvimage/xvimagesink.c: Fix a "may be used uninitialized in this function" which weirdly only appears on macosx (?).
8394 Original commit message from CVS:
8395 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
8396 Fix a "may be used uninitialized in this function" which weirdly only
8397 appears on macosx (?).
8399 2008-07-30 09:02:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8401 gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
8402 Original commit message from CVS:
8403 * gst-libs/gst/riff/riff-ids.h:
8404 Adding acid chunk for tempo and loop information.
8406 2008-07-29 13:01:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8408 sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
8409 Original commit message from CVS:
8410 * sys/xvimage/Makefile.am:
8411 floor() needs linking to $(LIBM).
8413 2008-07-29 12:35:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8415 ext/gnomevfs/gstgnomevfssrc.c: Aggregate short reads and add some comments and debug logging.
8416 Original commit message from CVS:
8417 * ext/gnomevfs/gstgnomevfssrc.c:
8418 Aggregate short reads and add some comments and debug logging.
8421 2008-07-29 10:26:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8423 gst/playback/gstplaybasebin.c: Fix property doc markup (its not a signal).
8424 Original commit message from CVS:
8425 * gst/playback/gstplaybasebin.c:
8426 Fix property doc markup (its not a signal).
8427 * sys/xvimage/xvimagesink.c:
8428 Add since tag for new proeprties (also add sice tags fro the last two
8431 2008-07-29 08:59:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8433 sys/xvimage/xvimagesink.*: Add autofill/colorkey properties. Fixes #538656.
8434 Original commit message from CVS:
8435 * sys/xvimage/xvimagesink.c:
8436 * sys/xvimage/xvimagesink.h:
8437 Add autofill/colorkey properties. Fixes #538656.
8439 2008-07-29 01:58:05 +0000 David Schleef <ds@schleef.org>
8441 sys/xvimage/xvimagesink.c: Fix rounding errors when converting colorbalance values between hardware and object proper...
8442 Original commit message from CVS:
8443 * sys/xvimage/xvimagesink.c:
8444 Fix rounding errors when converting colorbalance values
8445 between hardware and object property ranges. Partial
8446 fix for #537889, however, there still seems to be a small
8447 drift problem that could be totem's fault.
8449 2008-07-28 15:34:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8451 ext/ogg/gstoggdemux.c: Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
8452 Original commit message from CVS:
8453 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
8454 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
8455 Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
8456 This fixes a critical warning.
8458 2008-07-28 13:12:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8460 ext/ogg/gstoggmux.c: Allow muxing of CELT into Ogg streams.
8461 Original commit message from CVS:
8462 * ext/ogg/gstoggmux.c:
8463 Allow muxing of CELT into Ogg streams.
8465 2008-07-28 12:47:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8467 gst/typefind/gsttypefindfunctions.c: Add simple typefinder for the CELT codec (www.celt-codec.org).
8468 Original commit message from CVS:
8469 * gst/typefind/gsttypefindfunctions.c: (celt_type_find),
8471 Add simple typefinder for the CELT codec (www.celt-codec.org).
8473 2008-07-27 11:12:41 +0000 Jan Gerber <j@oil21.org>
8475 ext/ogg/gstoggdemux.c: Fix calculation of the start time from skeleton streams.
8476 Original commit message from CVS:
8477 Patch by: Jan Gerber <j at oil21 dot org>
8478 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
8479 Fix calculation of the start time from skeleton streams.
8482 2008-07-24 13:19:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8484 tests/examples/seek/seek.c: Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
8485 Original commit message from CVS:
8486 * tests/examples/seek/seek.c:
8487 Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
8489 2008-07-23 18:34:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8491 gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
8492 Original commit message from CVS:
8493 * gst/audioconvert/audioconvert.h:
8494 * gst/audioconvert/gstaudioquantize.c:
8495 (gst_audio_quantize_setup_dither),
8496 (gst_audio_quantize_free_dither):
8497 * gst/audioconvert/gstfastrandom.h:
8498 Implement a linear congruential generator as pseudo random number
8499 generator for the dither noise. This is about 2 times faster than
8500 using GLib's mersenne twister. Also this uses only integer math for
8501 generating integers while GLib internally uses floating point math.
8503 2008-07-23 18:27:15 +0000 Michael Smith <msmith@xiph.org>
8505 configure.ac: Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
8506 Original commit message from CVS:
8508 Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
8510 2008-07-23 13:17:31 +0000 Damien Lespiau <damien.lespiau@gmail.com>
8512 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
8513 Original commit message from CVS:
8514 Patch by: Damien Lespiau <damien.lespiau gmail com>
8515 * gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
8516 Use GST_STR_NULL to avoid crashes with libcs that don't
8517 like NULL strings in printf args (such as the win32 one).
8520 2008-07-17 14:21:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8522 sys/xvimage/xvimagesink.c: Oops - set the size of the image used for probing back to 1x1, for consistency with ximage...
8523 Original commit message from CVS:
8524 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
8525 Oops - set the size of the image used for probing back to 1x1, for
8526 consistency with ximagesink
8528 2008-07-17 13:57:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8530 sys/: it's not legal to ask the
8531 Original commit message from CVS:
8532 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
8533 (gst_ximagesink_ximage_new):
8534 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
8535 (gst_xvimagesink_xvimage_new):
8536 Apparently on Solaris and OS/X (at least), it's not legal to ask the
8537 X server to attach to a shared memory segment after we've deleted it,
8538 with the result that MIT-SHM is disabled. Instead, remove it only after
8539 X succeeds in attaching too.
8541 2008-07-17 02:30:24 +0000 David Schleef <ds@schleef.org>
8543 gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second.
8544 Original commit message from CVS:
8545 * gst/audiotestsrc/gstaudiotestsrc.c:
8546 * gst/audiotestsrc/gstaudiotestsrc.h:
8547 Add 'ticks', a 1/30 second sine wave pulse every second.
8549 2008-07-15 22:43:16 +0000 David Schleef <ds@schleef.org>
8551 gst-libs/gst/video/video.c: Revert ABI change.
8552 Original commit message from CVS:
8553 * gst-libs/gst/video/video.c: Revert ABI change.
8555 2008-07-15 13:05:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8557 gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
8558 Original commit message from CVS:
8559 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
8560 Make it impossible to have NULL caps at the point where we set
8561 framerate and other things. Also don't return immediately for "3ivd"
8562 video and let framerate, etc be set. Might fix bug #542508.
8564 2008-07-14 17:06:26 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
8566 gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
8567 Original commit message from CVS:
8568 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
8569 Video format can also be conveniently determined from (many)
8572 2008-07-14 08:18:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8574 gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
8575 Original commit message from CVS:
8576 * gst/playback/gstplaybasebin.c:
8577 * gst/playback/gstplaybasebin.h:
8578 * gst/playback/gstplaybin.c:
8579 * gst/playback/gststreamselector.c:
8580 First stab at integrating DVD subpicture overlay into
8581 playbin. Successfully plugs and plays, but the queues need
8582 shrinking - 3 seconds of video is too much buffering.
8584 2008-07-11 18:06:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8586 gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs.
8587 Original commit message from CVS:
8588 * gst/audioconvert/gstaudioconvert.c:
8589 Remove now obsolete note in the docs.
8591 2008-07-11 06:10:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8593 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
8594 Original commit message from CVS:
8595 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
8596 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
8597 * docs/plugins/gst-plugins-base-plugins-sections.txt:
8598 * docs/plugins/gst-plugins-base-plugins.args:
8599 * docs/plugins/gst-plugins-base-plugins.hierarchy:
8600 * docs/plugins/gst-plugins-base-plugins.interfaces:
8601 * docs/plugins/gst-plugins-base-plugins.prerequisites:
8602 * docs/plugins/gst-plugins-base-plugins.signals:
8603 * docs/plugins/inspect/plugin-adder.xml:
8604 * docs/plugins/inspect/plugin-alsa.xml:
8605 * docs/plugins/inspect/plugin-audioconvert.xml:
8606 * docs/plugins/inspect/plugin-audiorate.xml:
8607 * docs/plugins/inspect/plugin-audioresample.xml:
8608 * docs/plugins/inspect/plugin-audiotestsrc.xml:
8609 * docs/plugins/inspect/plugin-cdparanoia.xml:
8610 * docs/plugins/inspect/plugin-decodebin.xml:
8611 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
8612 * docs/plugins/inspect/plugin-gdp.xml:
8613 * docs/plugins/inspect/plugin-gnomevfs.xml:
8614 * docs/plugins/inspect/plugin-libvisual.xml:
8615 * docs/plugins/inspect/plugin-ogg.xml:
8616 * docs/plugins/inspect/plugin-pango.xml:
8617 * docs/plugins/inspect/plugin-playback.xml:
8618 * docs/plugins/inspect/plugin-queue2.xml:
8619 * docs/plugins/inspect/plugin-subparse.xml:
8620 * docs/plugins/inspect/plugin-tcp.xml:
8621 * docs/plugins/inspect/plugin-theora.xml:
8622 * docs/plugins/inspect/plugin-typefindfunctions.xml:
8623 * docs/plugins/inspect/plugin-uridecodebin.xml:
8624 * docs/plugins/inspect/plugin-video4linux.xml:
8625 * docs/plugins/inspect/plugin-videorate.xml:
8626 * docs/plugins/inspect/plugin-videoscale.xml:
8627 * docs/plugins/inspect/plugin-videotestsrc.xml:
8628 * docs/plugins/inspect/plugin-volume.xml:
8629 * docs/plugins/inspect/plugin-vorbis.xml:
8630 * docs/plugins/inspect/plugin-ximagesink.xml:
8631 * docs/plugins/inspect/plugin-xvimagesink.xml:
8632 * ext/alsa/gstalsamixer.c:
8633 * ext/alsa/gstalsasink.c:
8634 * ext/alsa/gstalsasrc.c:
8635 * ext/gio/gstgiosink.c:
8636 * ext/gio/gstgiosrc.c:
8637 * ext/gio/gstgiostreamsink.c:
8638 * ext/gio/gstgiostreamsrc.c:
8639 * ext/gnomevfs/gstgnomevfssink.c:
8640 * ext/gnomevfs/gstgnomevfssrc.c:
8641 * ext/ogg/gstoggdemux.c:
8642 * ext/ogg/gstoggmux.c:
8643 * ext/pango/gstclockoverlay.c:
8644 * ext/pango/gsttextoverlay.c:
8645 * ext/pango/gsttextrender.c:
8646 * ext/pango/gsttimeoverlay.c:
8647 * ext/theora/theoradec.c:
8648 * ext/theora/theoraenc.c:
8649 * ext/theora/theoraparse.c:
8650 * ext/vorbis/vorbisdec.c:
8651 * ext/vorbis/vorbisenc.c:
8652 * ext/vorbis/vorbisparse.c:
8653 * ext/vorbis/vorbistag.c:
8654 * gst/adder/gstadder.c:
8655 * gst/audioconvert/gstaudioconvert.c:
8656 * gst/audioresample/gstaudioresample.c:
8657 * gst/audiotestsrc/gstaudiotestsrc.c:
8658 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
8659 * gst/gdp/gstgdpdepay.c:
8660 * gst/gdp/gstgdppay.c:
8661 * gst/playback/gstdecodebin2.c:
8662 * gst/playback/gstplaybin.c:
8663 * gst/playback/gstplaybin2.c:
8664 * gst/playback/gstqueue2.c:
8665 * gst/playback/gsturidecodebin.c:
8666 * gst/tcp/gstmultifdsink.c:
8667 * gst/tcp/gsttcpserversink.c:
8668 * gst/videorate/gstvideorate.c:
8669 * gst/videoscale/gstvideoscale.c:
8670 * gst/videotestsrc/gstvideotestsrc.c:
8671 * gst/volume/gstvolume.c:
8672 * sys/ximage/ximagesink.c:
8673 * sys/xvimage/xvimagesink.c:
8674 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
8675 titles. Drop mentining that all our example pipelines are "simple"
8678 2008-07-10 21:06:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8680 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
8681 Original commit message from CVS:
8682 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
8683 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
8684 * docs/plugins/gst-plugins-base-plugins-sections.txt:
8685 * docs/plugins/gst-plugins-base-plugins.args:
8686 * docs/plugins/gst-plugins-base-plugins.hierarchy:
8687 * docs/plugins/gst-plugins-base-plugins.interfaces:
8688 * docs/plugins/gst-plugins-base-plugins.prerequisites:
8689 * docs/plugins/gst-plugins-base-plugins.signals:
8690 * docs/plugins/inspect/plugin-adder.xml:
8691 * docs/plugins/inspect/plugin-alsa.xml:
8692 * docs/plugins/inspect/plugin-audioconvert.xml:
8693 * docs/plugins/inspect/plugin-audiorate.xml:
8694 * docs/plugins/inspect/plugin-audioresample.xml:
8695 * docs/plugins/inspect/plugin-audiotestsrc.xml:
8696 * docs/plugins/inspect/plugin-cdparanoia.xml:
8697 * docs/plugins/inspect/plugin-decodebin.xml:
8698 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
8699 * docs/plugins/inspect/plugin-gdp.xml:
8700 * docs/plugins/inspect/plugin-gnomevfs.xml:
8701 * docs/plugins/inspect/plugin-libvisual.xml:
8702 * docs/plugins/inspect/plugin-ogg.xml:
8703 * docs/plugins/inspect/plugin-pango.xml:
8704 * docs/plugins/inspect/plugin-playback.xml:
8705 * docs/plugins/inspect/plugin-queue2.xml:
8706 * docs/plugins/inspect/plugin-subparse.xml:
8707 * docs/plugins/inspect/plugin-tcp.xml:
8708 * docs/plugins/inspect/plugin-theora.xml:
8709 * docs/plugins/inspect/plugin-typefindfunctions.xml:
8710 * docs/plugins/inspect/plugin-uridecodebin.xml:
8711 * docs/plugins/inspect/plugin-video4linux.xml:
8712 * docs/plugins/inspect/plugin-videorate.xml:
8713 * docs/plugins/inspect/plugin-videoscale.xml:
8714 * docs/plugins/inspect/plugin-videotestsrc.xml:
8715 * docs/plugins/inspect/plugin-volume.xml:
8716 * docs/plugins/inspect/plugin-vorbis.xml:
8717 * docs/plugins/inspect/plugin-ximagesink.xml:
8718 * docs/plugins/inspect/plugin-xvimagesink.xml:
8719 * ext/alsa/gstalsamixer.c:
8720 * ext/alsa/gstalsasink.c:
8721 * ext/alsa/gstalsasrc.c:
8722 * ext/gio/gstgiosink.c:
8723 * ext/gio/gstgiosrc.c:
8724 * ext/gio/gstgiostreamsink.c:
8725 * ext/gio/gstgiostreamsrc.c:
8726 * ext/gnomevfs/gstgnomevfssink.c:
8727 * ext/gnomevfs/gstgnomevfssrc.c:
8728 * ext/ogg/gstoggdemux.c:
8729 * ext/ogg/gstoggmux.c:
8730 * ext/pango/gstclockoverlay.c:
8731 * ext/pango/gsttextoverlay.c:
8732 * ext/pango/gsttextrender.c:
8733 * ext/pango/gsttimeoverlay.c:
8734 * ext/theora/theoradec.c:
8735 * ext/theora/theoraenc.c:
8736 * ext/theora/theoraparse.c:
8737 * ext/vorbis/vorbisdec.c:
8738 * ext/vorbis/vorbisenc.c:
8739 * ext/vorbis/vorbisparse.c:
8740 * ext/vorbis/vorbistag.c:
8741 * gst/adder/gstadder.c:
8742 * gst/audioconvert/gstaudioconvert.c:
8743 * gst/audioresample/gstaudioresample.c:
8744 * gst/audiotestsrc/gstaudiotestsrc.c:
8745 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
8746 * gst/gdp/gstgdpdepay.c:
8747 * gst/gdp/gstgdppay.c:
8748 * gst/playback/gstdecodebin2.c:
8749 * gst/playback/gstplaybin.c:
8750 * gst/playback/gstplaybin2.c:
8751 * gst/playback/gstqueue2.c:
8752 * gst/playback/gsturidecodebin.c:
8753 * gst/tcp/gstmultifdsink.c:
8754 * gst/tcp/gsttcpserversink.c:
8755 * gst/videorate/gstvideorate.c:
8756 * gst/videoscale/gstvideoscale.c:
8757 * gst/videotestsrc/gstvideotestsrc.c:
8758 * gst/volume/gstvolume.c:
8759 * sys/ximage/ximagesink.c:
8760 * sys/xvimage/xvimagesink.c:
8761 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
8762 titles. Drop mentining that all our example pipelines are "simple"
8765 2008-07-07 17:25:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8767 tests/examples/seek/Makefile.am: Fix out of tree build by adding all required CFLAGS.
8768 Original commit message from CVS:
8769 * tests/examples/seek/Makefile.am:
8770 Fix out of tree build by adding all required CFLAGS.
8772 2008-07-07 09:55:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8774 gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
8775 Original commit message from CVS:
8776 * gst/playback/gstdecodebin.c: (add_raw_queue):
8777 And ref the pad before returning it again when linking to the queue
8778 failed. Otherwise we will unref the pad twice later and things break.
8780 2008-07-07 09:48:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8782 gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
8783 Original commit message from CVS:
8784 * gst/playback/gstdecodebin.c: (add_raw_queue):
8785 If linking the raw pad with a queue fails, try it without a queue
8786 instead of failing completely. This should never happen.
8788 2008-07-06 23:22:12 +0000 Evgeniy Stepanov <eugeni.stepanov@gmail.com>
8790 gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
8791 Original commit message from CVS:
8792 Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
8793 * gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
8794 Add a queue after a demuxer if the demuxer outputs raw data. This was
8795 done before only for non-raw data but is required in this case too.
8797 decodebin2 doesn't have this issue because all streams of a group
8798 go through multiqueue.
8800 2008-07-03 09:12:49 +0000 Damien Lespiau <damien.lespiau@gmail.com>
8802 gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
8803 Original commit message from CVS:
8804 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
8805 * gst-libs/gst/sdp/gstsdpmessage.c:
8806 Makes libgstsdp compile with mingw32 by defining the right WINVER so
8807 that getaddrinfo() can be used. Fixes #541358.
8809 2008-07-01 13:22:49 +0000 Wim Taymans <wim.taymans@gmail.com>
8811 gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
8812 Original commit message from CVS:
8813 * gst/videotestsrc/gstvideotestsrc.c:
8814 (gst_video_test_src_class_init), (gst_video_test_src_init),
8815 (gst_video_test_src_set_property),
8816 (gst_video_test_src_get_property), (gst_video_test_src_create):
8817 * gst/videotestsrc/gstvideotestsrc.h:
8818 Cleanups, use default property values as defines.
8819 Add property to enable/disable peer buffer allocation.
8821 2008-06-30 09:46:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8823 tests/check/: Enable unit tests on PPC again as the bugs are now fixed.
8824 Original commit message from CVS:
8825 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
8826 * tests/check/pipelines/streamheader.c: (streamheader_suite):
8827 Enable unit tests on PPC again as the bugs are now fixed.
8829 2008-06-30 09:20:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8831 gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
8832 Original commit message from CVS:
8833 * gst-libs/gst/riff/riff-ids.h:
8834 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
8835 (gst_riff_create_audio_template_caps):
8836 Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
8839 2008-06-30 08:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8841 gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
8842 Original commit message from CVS:
8843 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
8844 (gst_ffmpeg_pixfmt_to_caps):
8845 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
8846 (gst_ffmpegcsp_get_unit_size):
8847 Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
8848 it on other formats. Also adjust the unit size only for that format
8849 to not include the palette. Fixes bug #540497.
8851 2008-06-29 13:45:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8853 gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
8854 Original commit message from CVS:
8855 * gst/adder/gstadder.c:
8856 Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
8858 2008-06-27 07:55:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8860 ChangeLog: ChangeLog surgery.
8861 Original commit message from CVS:
8864 * tests/examples/seek/seek.c:
8865 Move variable into ifdef too.
8867 2008-06-27 07:42:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8869 tests/examples/seek/seek.c: Include config.h and check if we have X. Fixes: #540334.
8870 Original commit message from CVS:
8871 * tests/examples/seek/seek.c:
8872 Include config.h and check if we have X. Fixes: #540334.
8874 2008-06-26 06:03:38 +0000 Sam Morris <sam@robots.org.to.uk>
8876 gst-libs/gst/interfaces/mixertrack.c: API: Add "index" property to GstMixerTrack to differantiate between multiple mi...
8877 Original commit message from CVS:
8878 Patch by: Sam Morris <sam at robots dot org to uk>
8879 * gst-libs/gst/interfaces/mixertrack.c:
8880 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
8881 (gst_mixer_track_set_property):
8882 API: Add "index" property to GstMixerTrack to differantiate between
8883 multiple mixer tracks with the same label.
8884 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
8885 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
8886 Set the "index" property of GstMixerTrack to the index given by ALSA.
8889 2008-06-25 13:15:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8891 tests/examples/seek/: Remove libgstvideo usage. Use gtk_get_option_group instead of gtk_init().
8892 Original commit message from CVS:
8893 * tests/examples/seek/Makefile.am:
8894 * tests/examples/seek/seek.c:
8895 Remove libgstvideo usage. Use gtk_get_option_group instead of
8898 2008-06-24 16:27:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8900 tests/check/Makefile.am: Name the test registry format neutral.
8901 Original commit message from CVS:
8902 * tests/check/Makefile.am:
8903 Name the test registry format neutral.
8905 2008-06-24 16:22:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8907 gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
8908 Original commit message from CVS:
8909 * gst/playback/gstqueue2.c:
8910 Do not double notify. Remove the unsued return value.
8912 2008-06-24 16:15:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8914 ext/alsa/gstalsamixer.c: Also consider "speaker" as a name for master volume. If that doesn't help look for the first...
8915 Original commit message from CVS:
8916 * ext/alsa/gstalsamixer.c:
8917 Also consider "speaker" as a name for master volume. If that doesn't
8918 help look for the first non-mono volume control that also has a
8921 2008-06-24 16:10:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8923 ChangeLog: Forgot to save the ChangeLog :/
8924 Original commit message from CVS:
8926 Forgot to save the ChangeLog :/
8928 2008-06-24 16:05:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8930 tests/examples/seek/: Embedd the xwindow.
8931 Original commit message from CVS:
8932 * tests/examples/seek/Makefile.am:
8933 * tests/examples/seek/seek.c:
8936 2008-06-24 01:14:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8938 sys/ximage/ximagesink.h: When the caps change, make sure to re-draw borders in force-aspect-ratio=true mode.
8939 Original commit message from CVS:
8940 * sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
8941 (gst_ximagesink_setcaps):
8942 * sys/ximage/ximagesink.h:
8943 When the caps change, make sure to re-draw borders in
8944 force-aspect-ratio=true mode.
8945 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
8946 Don't clear the border_draw flag until we actually draw the border.
8947 * tests/check/Makefile.am:
8948 Ignore alsasink/src during the states test too, so it doesn't fail
8949 when running without access to the sound device.
8951 2008-06-22 18:35:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8953 tests/examples/seek/seek.c: Fix crasher when playing a parse-launch line the 2nd time.
8954 Original commit message from CVS:
8955 * tests/examples/seek/seek.c:
8956 Fix crasher when playing a parse-launch line the 2nd time.
8958 2008-06-21 18:56:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
8960 tests/check/pipelines/oggmux.c: Properly ifdef tests to fix compilation.
8961 Original commit message from CVS:
8962 * tests/check/pipelines/oggmux.c:
8963 Properly ifdef tests to fix compilation.
8965 2008-06-21 10:25:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
8969 Original commit message from CVS:
8972 2008-06-20 18:24:24 +0000 Michael Smith <msmith@xiph.org>
8974 gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
8975 Original commit message from CVS:
8976 * gst/playback/gstplay-marshal.list:
8977 * gst/playback/gstplaybin2.c:
8978 Add get-video-pad, get-audio-pad, get-text-pad action signals to
8979 playbin2. This allows the user to get to the selector's sinkpads, and
8980 thus inspect a range of things - caps, tags, etc.
8982 2008-06-20 17:27:03 +0000 Michael Smith <msmith@xiph.org>
8984 gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
8985 Original commit message from CVS:
8986 * gst/playback/gstplaybin2.c:
8987 Use a different constant for the convert-frame signal id.
8990 2008-06-20 17:18:55 +0000 Michael Smith <msmith@xiph.org>
8992 gst/playback/: Fix a whole bunch of typos in comments and log statements.
8993 Original commit message from CVS:
8994 * gst/playback/gstplaybin2.c:
8995 * gst/playback/gstplaysink.c:
8996 Fix a whole bunch of typos in comments and log statements.
8998 2008-06-20 17:02:48 +0000 Michael Smith <msmith@xiph.org>
9000 sys/xvimage/xvimagesink.c: Don't set colour balance values on the Xv port if the user hasn't changed them (via proper...
9001 Original commit message from CVS:
9002 * sys/xvimage/xvimagesink.c:
9003 Don't set colour balance values on the Xv port if the user hasn't
9004 changed them (via properties or the interface). Avoids accumulating
9005 rounding errors for the common case.
9006 Partial fix for bug #537889.
9008 2008-06-20 16:56:18 +0000 Michael Smith <msmith@xiph.org>
9010 gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
9011 Original commit message from CVS:
9012 * gst/playback/gstdecodebin2.c:
9013 Ensure decodebin2 emits 'drained' signal once, and only once, when all
9016 2008-06-20 16:12:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
9019 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...
9020 Original commit message from CVS:
9021 apparently it's an error to specify nc -l -p 3000 - though the short usage
9022 does not make it very clear that you can drop the host arg with -l
9024 2008-06-20 09:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
9026 ext/vorbis/vorbisenc.c: Report the encoder latency. Fixes #538232.
9027 Original commit message from CVS:
9028 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
9029 (gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
9030 Report the encoder latency. Fixes #538232.
9032 2008-06-20 09:19:59 +0000 Wim Taymans <wim.taymans@gmail.com>
9034 gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
9035 Original commit message from CVS:
9036 * gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
9037 (notify_source), (activate_group):
9038 Implement the source property, emit notify when it changes in the
9039 underlying uridecodebin.
9041 2008-06-20 09:14:26 +0000 Wim Taymans <wim.taymans@gmail.com>
9043 tests/examples/seek/seek.c: Free and clear the seek element list so that we don't use invalid references when seeking...
9044 Original commit message from CVS:
9045 * tests/examples/seek/seek.c: (stop_cb):
9046 Free and clear the seek element list so that we don't use invalid
9047 references when seeking after recreating a gst-launch line.
9049 2008-06-20 09:09:37 +0000 Wim Taymans <wim.taymans@gmail.com>
9051 gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
9052 Original commit message from CVS:
9053 * gst-libs/gst/audio/gstbaseaudiosink.c:
9054 (gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
9055 (gst_base_audio_sink_render):
9056 Report latency even if we are not live instead of hiding it.
9057 Take ts-offset and render-delay of the basesink into account when
9059 Rework the clipping code so that we can take the various offsets into
9060 account and still do correct clipping.
9062 2008-06-20 08:52:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9064 configure.ac: Bump verion back to devel -> 0.10.20.1
9065 Original commit message from CVS:
9067 Bump verion back to devel -> 0.10.20.1
9069 2008-06-20 08:47:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9071 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...
9072 Original commit message from CVS:
9073 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
9074 Don't increase the size of non-string image buffers by one as this
9075 might in theory confuse decoders. Still increase it by one for string
9076 image buffers to append '\0'.
9078 2008-06-20 08:45:13 +0000 Antoine Tremblay <hexa00@gmail.com>
9080 gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
9081 Original commit message from CVS:
9082 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
9083 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
9084 Fix a buffer memleak and remove a confusing and wrong debug output.
9087 2008-06-19 11:25:37 +0000 Wim Taymans <wim.taymans@gmail.com>
9089 examples/app/appsink-src.c: Don't use a buffer after unreffing it.
9090 Original commit message from CVS:
9091 * examples/app/appsink-src.c: (on_new_buffer_from_source):
9092 Don't use a buffer after unreffing it.
9094 === release 0.10.20 ===
9096 2008-06-18 14:36:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9102 * docs/plugins/gst-plugins-base-plugins.args:
9103 * docs/plugins/gst-plugins-base-plugins.hierarchy:
9104 * docs/plugins/gst-plugins-base-plugins.interfaces:
9105 * docs/plugins/gst-plugins-base-plugins.prerequisites:
9106 * docs/plugins/inspect/plugin-adder.xml:
9107 * docs/plugins/inspect/plugin-alsa.xml:
9108 * docs/plugins/inspect/plugin-audioconvert.xml:
9109 * docs/plugins/inspect/plugin-audiorate.xml:
9110 * docs/plugins/inspect/plugin-audioresample.xml:
9111 * docs/plugins/inspect/plugin-audiotestsrc.xml:
9112 * docs/plugins/inspect/plugin-cdparanoia.xml:
9113 * docs/plugins/inspect/plugin-decodebin.xml:
9114 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
9115 * docs/plugins/inspect/plugin-gdp.xml:
9116 * docs/plugins/inspect/plugin-gnomevfs.xml:
9117 * docs/plugins/inspect/plugin-libvisual.xml:
9118 * docs/plugins/inspect/plugin-ogg.xml:
9119 * docs/plugins/inspect/plugin-pango.xml:
9120 * docs/plugins/inspect/plugin-playback.xml:
9121 * docs/plugins/inspect/plugin-queue2.xml:
9122 * docs/plugins/inspect/plugin-subparse.xml:
9123 * docs/plugins/inspect/plugin-tcp.xml:
9124 * docs/plugins/inspect/plugin-theora.xml:
9125 * docs/plugins/inspect/plugin-typefindfunctions.xml:
9126 * docs/plugins/inspect/plugin-uridecodebin.xml:
9127 * docs/plugins/inspect/plugin-video4linux.xml:
9128 * docs/plugins/inspect/plugin-videorate.xml:
9129 * docs/plugins/inspect/plugin-videoscale.xml:
9130 * docs/plugins/inspect/plugin-videotestsrc.xml:
9131 * docs/plugins/inspect/plugin-volume.xml:
9132 * docs/plugins/inspect/plugin-vorbis.xml:
9133 * docs/plugins/inspect/plugin-ximagesink.xml:
9134 * docs/plugins/inspect/plugin-xvimagesink.xml:
9135 * gst-plugins-base.doap:
9137 * win32/common/config.h:
9139 Original commit message from CVS:
9142 2008-06-18 14:32:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9171 Original commit message from CVS:
9174 2008-06-18 06:31:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9176 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
9177 Original commit message from CVS:
9178 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
9179 * examples/app/appsrc-ra.c:
9180 * examples/app/appsrc-seekable.c:
9181 * examples/app/appsrc-stream.c:
9182 * examples/app/appsrc-stream2.c:
9183 * ext/directfb/dfbvideosink.h:
9184 * ext/metadata/gstbasemetadata.c:
9185 * ext/metadata/gstbasemetadata.h:
9186 * ext/metadata/metadata.c:
9187 * ext/metadata/metadataexif.c:
9188 * ext/theora/theoradec.h:
9189 * gst/deinterlace2/gstdeinterlace2.h:
9190 * gst/deinterlace2/tvtime/speedy.c:
9191 * gst/deinterlace2/tvtime/speedy.h:
9192 * gst/deinterlace2/tvtime/vfir.c:
9193 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
9196 2008-06-16 14:11:36 +0000 Andy Wingo <wingo@pobox.com>
9198 * gst-libs/gst/app/gstappsrc.c:
9199 gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
9200 Original commit message from CVS:
9201 2008-06-16 Andy Wingo <wingo@pobox.com>
9202 * gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
9203 (gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
9204 G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
9206 2008-06-16 07:30:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9208 Final round of doc updates.
9209 Original commit message from CVS:
9210 * gst/rtpmanager/gstrtpjitterbuffer.c:
9211 * gst/speed/gstspeed.c:
9212 * gst/speexresample/gstspeexresample.c:
9213 * gst/videosignal/gstvideoanalyse.c:
9214 * gst/videosignal/gstvideodetect.c:
9215 * gst/videosignal/gstvideomark.c:
9216 * sys/dvb/gstdvbsrc.c:
9217 * sys/oss4/oss4-mixer.c:
9218 * sys/oss4/oss4-sink.c:
9219 * sys/oss4/oss4-source.c:
9220 * sys/wininet/gstwininetsrc.c:
9221 Final round of doc updates.
9223 2008-06-13 11:59:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9225 docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
9226 Original commit message from CVS:
9227 * docs/plugins/Makefile.am:
9228 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
9229 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
9230 * docs/plugins/gst-plugins-bad-plugins.args:
9231 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
9232 * docs/plugins/gst-plugins-bad-plugins.interfaces:
9233 * docs/plugins/gst-plugins-bad-plugins.prerequisites:
9234 * docs/plugins/gst-plugins-bad-plugins.signals:
9235 * docs/plugins/inspect/plugin-alsaspdif.xml:
9236 * docs/plugins/inspect/plugin-amrwb.xml:
9237 * docs/plugins/inspect/plugin-app.xml:
9238 * docs/plugins/inspect/plugin-bayer.xml:
9239 * docs/plugins/inspect/plugin-bz2.xml:
9240 * docs/plugins/inspect/plugin-cdaudio.xml:
9241 * docs/plugins/inspect/plugin-cdxaparse.xml:
9242 * docs/plugins/inspect/plugin-dtsdec.xml:
9243 * docs/plugins/inspect/plugin-dvb.xml:
9244 * docs/plugins/inspect/plugin-dvdspu.xml:
9245 * docs/plugins/inspect/plugin-faac.xml:
9246 * docs/plugins/inspect/plugin-faad.xml:
9247 * docs/plugins/inspect/plugin-fbdevsink.xml:
9248 * docs/plugins/inspect/plugin-festival.xml:
9249 * docs/plugins/inspect/plugin-filter.xml:
9250 * docs/plugins/inspect/plugin-flvdemux.xml:
9251 * docs/plugins/inspect/plugin-freeze.xml:
9252 * docs/plugins/inspect/plugin-gsm.xml:
9253 * docs/plugins/inspect/plugin-gstinterlace.xml:
9254 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
9255 * docs/plugins/inspect/plugin-h264parse.xml:
9256 * docs/plugins/inspect/plugin-interleave.xml:
9257 * docs/plugins/inspect/plugin-jack.xml:
9258 * docs/plugins/inspect/plugin-ladspa.xml:
9259 * docs/plugins/inspect/plugin-metadata.xml:
9260 * docs/plugins/inspect/plugin-mms.xml:
9261 * docs/plugins/inspect/plugin-modplug.xml:
9262 * docs/plugins/inspect/plugin-mpeg2enc.xml:
9263 * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
9264 * docs/plugins/inspect/plugin-mpegtsparse.xml:
9265 * docs/plugins/inspect/plugin-mpegvideoparse.xml:
9266 * docs/plugins/inspect/plugin-musepack.xml:
9267 * docs/plugins/inspect/plugin-musicbrainz.xml:
9268 * docs/plugins/inspect/plugin-mve.xml:
9269 * docs/plugins/inspect/plugin-mythtv.xml
9270 * docs/plugins/inspect/plugin-nas.xml:
9271 * docs/plugins/inspect/plugin-neon.xml:
9272 * docs/plugins/inspect/plugin-nsfdec.xml:
9273 * docs/plugins/inspect/plugin-nuvdemux.xml:
9274 * docs/plugins/inspect/plugin-oss4.xml
9275 * docs/plugins/inspect/plugin-rawparse.xml:
9276 * docs/plugins/inspect/plugin-real.xml:
9277 * docs/plugins/inspect/plugin-replaygain.xml:
9278 * docs/plugins/inspect/plugin-rfbsrc.xml:
9279 * docs/plugins/inspect/plugin-sdl.xml:
9280 * docs/plugins/inspect/plugin-sdp.xml:
9281 * docs/plugins/inspect/plugin-selector.xml:
9282 * docs/plugins/inspect/plugin-sndfile.xml:
9283 * docs/plugins/inspect/plugin-soundtouch.xml:
9284 * docs/plugins/inspect/plugin-spcdec.xml:
9285 * docs/plugins/inspect/plugin-speed.xml:
9286 * docs/plugins/inspect/plugin-speexresample.xml:
9287 * docs/plugins/inspect/plugin-stereo.xml:
9288 * docs/plugins/inspect/plugin-subenc.xml
9289 * docs/plugins/inspect/plugin-timidity.xml:
9290 * docs/plugins/inspect/plugin-tta.xml:
9291 * docs/plugins/inspect/plugin-vcdsrc.xml:
9292 * docs/plugins/inspect/plugin-videosignal.xml:
9293 * docs/plugins/inspect/plugin-vmnc.xml:
9294 * docs/plugins/inspect/plugin-wildmidi.xml:
9295 * docs/plugins/inspect/plugin-x264.xml:
9296 * docs/plugins/inspect/plugin-xvid.xml:
9297 * docs/plugins/inspect/plugin-y4menc.xml:
9298 * ext/amrwb/gstamrwbdec.c:
9299 * ext/amrwb/gstamrwbenc.c:
9300 * ext/amrwb/gstamrwbparse.c:
9301 * ext/dc1394/gstdc1394.c:
9302 * ext/directfb/dfbvideosink.c:
9303 * ext/ivorbis/vorbisdec.c:
9304 * ext/jack/gstjackaudiosink.c:
9305 * ext/mpeg2enc/gstmpeg2enc.cc:
9306 * ext/mplex/gstmplex.cc:
9307 * ext/musicbrainz/gsttrm.c:
9308 * ext/mythtv/gstmythtvsrc.c:
9309 * ext/theora/theoradec.c:
9310 * ext/timidity/gsttimidity.c:
9311 * ext/timidity/gstwildmidi.c:
9312 * gst-libs/gst/app/gstappsink.c:
9313 * gst/deinterlace/gstdeinterlace.c:
9314 * gst/dvdspu/gstdvdspu.c:
9315 * gst/festival/gstfestival.c:
9316 * gst/freeze/gstfreeze.c:
9317 * gst/interleave/deinterleave.c:
9318 * gst/interleave/interleave.c:
9319 * gst/modplug/gstmodplug.cc:
9320 * gst/nuvdemux/gstnuvdemux.c:
9321 Add missing elements to docs. Fix doc-markup: use convinience syntax
9322 for examples (produces valid docbook), add several refsec2 when we
9323 have several titles. Fix some types.
9325 2008-06-12 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
9327 examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
9328 Original commit message from CVS:
9329 * examples/app/.cvsignore:
9330 * examples/app/Makefile.am:
9331 * examples/app/appsink-src.c: (on_new_buffer_from_source),
9332 (on_source_message), (on_sink_message), (main):
9333 Add beefed up example app from bug #413418. It now also uses appsink
9334 instead of fakesink for more ultimate coolness.
9335 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
9336 (gst_app_src_init), (gst_app_src_set_property),
9337 (gst_app_src_get_property), (gst_app_src_unlock),
9338 (gst_app_src_unlock_stop), (gst_app_src_create),
9339 (gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
9340 (gst_app_src_end_of_stream):
9341 * gst-libs/gst/app/gstappsrc.h:
9342 Add block property to allow push based implementation to block when we
9343 fill up the appsrc queues.
9344 Emit the enough-data signal while releasing our lock.
9346 2008-06-12 14:50:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9348 examples/app/.cvsignore: Ignore more.
9349 Original commit message from CVS:
9350 * examples/app/.cvsignore:
9353 2008-06-12 14:49:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9355 Do not use short_description in section docs for elements. We extract them from element details and there will be war...
9356 Original commit message from CVS:
9357 * ext/dc1394/gstdc1394.c:
9358 * ext/ivorbis/vorbisdec.c:
9359 * ext/jack/gstjackaudiosink.c:
9360 * ext/metadata/gstmetadatademux.c:
9361 * ext/mythtv/gstmythtvsrc.c:
9362 * ext/theora/theoradec.c:
9363 * gst-libs/gst/app/gstappsink.c:
9364 * gst/bayer/gstbayer2rgb.c:
9365 * gst/deinterlace/gstdeinterlace.c:
9366 * gst/rawparse/gstaudioparse.c:
9367 * gst/rawparse/gstvideoparse.c:
9368 * gst/rtpmanager/gstrtpbin.c:
9369 * gst/rtpmanager/gstrtpclient.c:
9370 * gst/rtpmanager/gstrtpjitterbuffer.c:
9371 * gst/rtpmanager/gstrtpptdemux.c:
9372 * gst/rtpmanager/gstrtpsession.c:
9373 * gst/rtpmanager/gstrtpssrcdemux.c:
9374 * gst/selector/gstinputselector.c:
9375 * gst/selector/gstoutputselector.c:
9376 * gst/videosignal/gstvideoanalyse.c:
9377 * gst/videosignal/gstvideodetect.c:
9378 * gst/videosignal/gstvideomark.c:
9379 * sys/oss4/oss4-mixer.c:
9380 * sys/oss4/oss4-sink.c:
9381 * sys/oss4/oss4-source.c:
9382 Do not use short_description in section docs for elements. We extract
9383 them from element details and there will be warnings if they differ.
9384 Also fixing up the ChangeLog order.
9386 2008-06-11 21:17:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9388 configure.ac: 0.10.19.3 pre-release
9389 Original commit message from CVS:
9391 0.10.19.3 pre-release
9393 2008-06-11 20:13:00 +0000 David Schleef <ds@schleef.org>
9395 gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
9396 Original commit message from CVS:
9397 * gst-libs/gst/rtsp/gstrtspconnection.c:
9399 Patch By: David Schleef <ds@schleef.org>
9402 2008-06-11 09:35:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9404 ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
9405 Original commit message from CVS:
9406 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
9407 (gst_gio_base_src_create):
9408 * ext/gio/gstgiobasesrc.h:
9409 Try to read the requested number of bytes, even if the first
9410 read returns less than requested, until nothing is read anymore
9411 or we have the requested amount of bytes. This fixes playback of
9412 files via Samba as Samba only allows to read 64k at once.
9413 Implement a caching algorithm that makes sure that we read at
9414 least 4k of data every time. Some elements will try to read a few
9415 bytes, then seek, read again a few bytes and so on and this is
9416 painfully slow as every operation has to go over DBus if GVfs is
9418 Fixes bug #536849 and #536848.
9419 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
9420 (gst_gio_src_check_get_range):
9421 Override check_get_range() to blacklist http/https URIs
9422 and whitelist file URIs. More to be added on demand.
9424 2008-06-06 16:50:51 +0000 Wim Taymans <wim.taymans@gmail.com>
9426 examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
9427 Original commit message from CVS:
9428 * examples/app/Makefile.am:
9429 * examples/app/appsrc-ra.c: (feed_data), (seek_data),
9430 (found_source), (bus_message), (main):
9431 * examples/app/appsrc-seekable.c: (feed_data), (seek_data),
9432 (found_source), (bus_message), (main):
9433 * examples/app/appsrc-stream2.c: (feed_data), (found_source),
9434 (bus_message), (main):
9435 Added 3 more example application for using appsrc in random-access mode,
9436 pull-mode streaming and pull mode seekable.
9437 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
9438 (gst_app_src_start), (gst_app_src_do_get_size),
9439 (gst_app_src_create):
9440 * gst-libs/gst/app/gstappsrc.h:
9441 Make stream-type property writable.
9442 Unset flushing when starting so that we reuse appsrc.
9443 Inform basesrc about the configured size.
9444 Emit seek-data signal when we are going to a different offset in
9447 2008-06-06 14:19:54 +0000 Wim Taymans <wim.taymans@gmail.com>
9449 examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
9450 Original commit message from CVS:
9451 * examples/app/appsrc-stream.c: (found_source), (main):
9452 Use deep-notify until we can depend on a playbin2 with support for the
9455 2008-06-05 16:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
9457 examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
9458 Original commit message from CVS:
9459 * examples/app/.cvsignore:
9460 * examples/app/Makefile.am:
9461 * examples/app/appsrc-stream.c: (read_data), (start_feed),
9462 (stop_feed), (found_source), (bus_message), (main):
9463 Added an example on how to use appsrc in playbin in streaming mode from
9465 * examples/app/appsrc_ex.c: (main):
9466 Set pipeline to NULL to free queued buffers.
9467 * gst-libs/gst/app/gstapp-marshal.list:
9468 * gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
9469 (gst_app_src_class_init), (gst_app_src_init),
9470 (gst_app_src_flush_queued), (gst_app_src_dispose),
9471 (gst_app_src_set_property), (gst_app_src_get_property),
9472 (gst_app_src_unlock), (gst_app_src_unlock_stop),
9473 (gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
9474 (gst_app_src_check_get_range), (gst_app_src_do_seek),
9475 (gst_app_src_create), (gst_app_src_set_stream_type),
9476 (gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
9477 (gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
9478 (gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
9479 (gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
9480 (gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
9481 * gst-libs/gst/app/gstappsrc.h:
9482 Measure max queue size in bytes instead.
9483 Add support for 3 modes of operation, streaming, seekable and
9484 random-access, making basesrc handle the scheduling modes for each.
9485 Add appsrc:// uri handler so that automatic plugging can be done from
9486 playbin2 or uridecodebin, for example.
9487 Added support for custom segment formats.
9488 Add support for push and pull based operations from the application.
9489 Expand the methods so that errors can be detected.
9490 Flush the queued buffers on seeks and when shutting down.
9491 Add signals to inform the app that a seek must happen.
9493 2008-06-05 09:47:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9495 configure.ac: 0.10.19.2 pre-release
9496 Original commit message from CVS:
9498 0.10.19.2 pre-release
9500 2008-06-04 21:48:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9502 win32/common/: Add new API functions to the dll exports
9503 Original commit message from CVS:
9504 * win32/common/libgstrtsp.def:
9505 * win32/common/libgsttag.def:
9506 Add new API functions to the dll exports
9508 2008-06-04 17:42:38 +0000 Michael Smith <msmith@xiph.org>
9510 gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
9511 Original commit message from CVS:
9512 * gst/playback/gstplaybasebin.c:
9513 Disconnect signals from decodebins we created before we remove it from
9514 playbin, to avoid crashes if the decodebin is eventually disposed after
9515 the playbin itself (possible if the app takes a reference on the
9519 2008-06-04 17:12:40 +0000 Tim-Philipp Müller <tim@centricular.net>
9521 gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
9522 Original commit message from CVS:
9523 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
9524 (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
9525 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
9526 (h264_video_type_find), (mpeg_video_stream_type_find),
9527 (dv_type_find), (mmsh_type_find):
9528 Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
9529 copy caps for no good reason (this may be desirable to make it easier
9530 to detect leaks, but then it should probably be done for all caps
9531 in the typefinder somewhere).
9533 2008-06-04 16:06:49 +0000 Peter Kjellerstedt <pkj@axis.com>
9535 tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
9536 Original commit message from CVS:
9537 * tests/check/Makefile.am:
9538 Do not try to run the check tests for subparse unless it has been
9541 2008-06-04 16:00:26 +0000 Peter Kjellerstedt <pkj@axis.com>
9543 tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
9544 Original commit message from CVS:
9545 * tests/check/pipelines/streamheader.c: (buffer_probe_cb),
9546 (test_multifdsink_gdp_vorbisenc), (streamheader_suite):
9547 Do not try to run a test which requires vorbisenc unless we have
9550 2008-06-04 11:53:53 +0000 Peter Kjellerstedt <pkj@axis.com>
9552 gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
9553 Original commit message from CVS:
9554 * gst-libs/gst/rtsp/gstrtspconnection.c:
9555 (gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
9556 (gst_rtsp_connection_clear_auth_params),
9557 (gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
9558 * gst-libs/gst/rtsp/gstrtspconnection.h:
9559 Add a couple of missing argument guards.
9560 Add a way of setting the DSCP for an RTSP connection.
9561 Add an accessor method for the ip member of GstRTSPConnection as all
9562 members are supposed to be private.
9564 2008-06-04 11:33:23 +0000 Peter Kjellerstedt <pkj@axis.com>
9566 gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
9567 Original commit message from CVS:
9568 * gst/tcp/gstmultifdsink.c: (setup_dscp_client):
9569 Fixed accidental use of IPv4 options for all IPv6 addresses.
9571 2008-06-04 10:18:42 +0000 Tim-Philipp Müller <tim@centricular.net>
9573 gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
9574 Original commit message from CVS:
9575 * gst-libs/gst/interfaces/mixertrack.h:
9576 Document mixer track flags.
9578 2008-06-04 05:58:38 +0000 Antoine Tremblay <hexa00@gmail.com>
9580 gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
9581 Original commit message from CVS:
9582 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
9583 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
9584 Don't set caps on the buffers that contain a copy of the buffer
9585 including the caps of them resulting in an always increasing refcount
9586 of the caps and insanely large caps. Instead include a buffer without
9587 caps in the new caps. Fixes bug #536475.
9589 2008-06-04 05:44:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9591 gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
9592 Original commit message from CVS:
9593 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
9594 Transform a given PAR to a range on the struct with the generic
9595 height/width instead of the struct with the possibly restricted
9598 2008-06-04 04:24:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9600 gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
9601 Original commit message from CVS:
9602 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
9603 Prefer the given format if it contains something stricter than [1,MAX]
9604 for height or width and only put a structure that requires rescaling
9605 as second. This makes it possible to use videoscale in pipelines where
9606 the source can actually produce the wanted height/width but usually
9607 selects a different one from the requested.
9609 2008-06-03 20:01:58 +0000 John Millikin <jmillikin@gmail.com>
9611 gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
9612 Original commit message from CVS:
9613 Based on patch by: John Millikin <jmillikin gmail com>
9614 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
9615 (gst_vorbis_tag_add_coverart):
9616 Retrieve COVERART tags from vorbis comments (#512333)
9618 2008-06-03 19:44:48 +0000 Tim-Philipp Müller <tim@centricular.net>
9620 gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
9621 Original commit message from CVS:
9622 * gst-libs/gst/tag/tag.h:
9623 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
9624 Don't forget to add new enum value here too (should probably use
9625 glib-mkenums here...).
9627 2008-06-03 19:29:06 +0000 Tim-Philipp Müller <tim@centricular.net>
9629 gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
9630 Original commit message from CVS:
9631 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
9632 * gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
9633 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
9634 (gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
9635 (gst_tag_image_data_to_image_buffer):
9636 Add two utility functions to avoid code duplication (#512333):
9637 API: add gst_tag_image_data_to_image_buffer()
9638 API: add gst_tag_list_add_id3_image()
9640 2008-06-03 08:54:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9642 win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
9643 Original commit message from CVS:
9644 * win32/common/libgstaudio.def:
9645 Add gst_audio_check_channel_positions() to the exported symbols.
9647 2008-06-03 08:48:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9649 API: Make gst_audio_check_channel_positions() public.
9650 Original commit message from CVS:
9651 * docs/libs/gst-plugins-base-libs-sections.txt:
9652 * gst-libs/gst/audio/multichannel.c:
9653 (gst_audio_check_channel_positions):
9654 * gst-libs/gst/audio/multichannel.h:
9655 API: Make gst_audio_check_channel_positions() public.
9656 * tests/check/libs/audio.c: (GST_START_TEST):
9657 Add some simple checks for gst_audio_check_channel_positions().
9659 2008-06-02 20:09:14 +0000 Tim-Philipp Müller <tim@centricular.net>
9661 sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.
9662 Original commit message from CVS:
9663 * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
9664 minrange and maxrange are scaled according to the frequency
9667 2008-06-02 18:37:02 +0000 Tim-Philipp Müller <tim@centricular.net>
9669 ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
9670 Original commit message from CVS:
9671 * ext/pango/Makefile.am:
9672 * ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
9673 (gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
9674 Use gstvideo functions to calculate strides and plane offsets. Fixes
9675 rendering issue ('ghost' images of the text on the chroma planes)
9676 with widths or heights that are not multiples of 8 (#506659 and
9677 probably also #485729).
9678 * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
9680 Test with odd height/width too.
9682 2008-06-02 12:20:35 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9684 gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
9685 Original commit message from CVS:
9686 * gst/adder/gstadder.c: (gst_adder_query_duration),
9687 (gst_adder_query_latency):
9688 When using gst_element_iterate_pads() one has to unref every pad
9691 2008-05-31 19:57:57 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9693 gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
9694 Original commit message from CVS:
9695 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9696 (gst_base_audio_src_class_init):
9697 Add a gtk-doc chunk for the new properties to have a Since: indication.
9699 2008-05-31 19:50:59 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9702 ChangeLog surgery, mark API change
9703 Original commit message from CVS:
9704 ChangeLog surgery, mark API change
9706 2008-05-31 18:10:47 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9708 gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
9709 Original commit message from CVS:
9710 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9711 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
9712 (gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
9713 (gst_base_audio_src_change_state):
9714 Provide readable actual-buffer-time and actual-latency-time properties
9715 that reflect the configured ringbuffer values. Fixes #524724.
9717 2008-05-30 15:29:20 +0000 Wim Taymans <wim.taymans@gmail.com>
9719 gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
9720 Original commit message from CVS:
9721 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
9722 (gst_basertppayload_change_state):
9723 Simply converting the running time into an RTP timestamp by scaling it
9724 based on the clock-rate is good enough for making an RTP timestamp. This
9725 has the added benefit that we can later on expose a property with the
9726 RTP timestamp of running time 0, as is needed for RTSP servers to
9727 generate the response of the PLAY request.
9729 2008-05-30 08:42:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9731 gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
9732 Original commit message from CVS:
9733 * gst/audioconvert/gstaudioconvert.c:
9734 (structure_has_fixed_channel_positions),
9735 (gst_audio_convert_transform_caps):
9736 Allow up to 11 positioned channels now that audioconvert can handle
9737 this but add no default positions for > 8 channels.
9738 * tests/check/elements/audioconvert.c: (GST_START_TEST):
9739 Add some unit tests for the above change: Test conversion of
9740 11 positioned channels to stereo and the other way around, test
9741 conversion of 15 unpositioned channels in different ways.
9743 2008-05-29 19:45:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9745 win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
9746 Original commit message from CVS:
9747 * win32/common/libgstaudio.def:
9748 Add gst_audio_clock_reset to the list of exported symbols.
9750 2008-05-29 19:37:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9752 tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
9753 Original commit message from CVS:
9754 * tests/check/elements/vorbisdec.c: (vorbisdec_suite):
9755 Remove wrong_channels_identification_header unit test as we now
9756 support 7 (and more channels).
9758 2008-05-29 12:17:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9760 gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
9761 Original commit message from CVS:
9762 * gst/audioconvert/gstchannelmix.c:
9763 (gst_channel_mix_fill_one_other):
9764 If mixing left or right to center (or the other way around) only take
9765 the complete value if we don't already have the original position in
9768 2008-05-29 11:34:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9770 gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
9771 Original commit message from CVS:
9772 * gst-libs/gst/audio/multichannel.c:
9773 (gst_audio_check_channel_positions),
9774 (gst_audio_set_structure_channel_positions_list),
9775 (gst_audio_fixate_channel_positions):
9776 Allow rear center together with rear left/right and other previously
9777 conflicting channel positions. The reason why they weren't allowed
9778 was the channel mixing implementation in audioconvert.
9779 Also take this into account when fixing channel layouts.
9780 Allow setting channel positions for 1/2 channels when using
9781 gst_audio_set_structure_channel_position().
9782 * gst/audioconvert/gstchannelmix.c:
9783 (gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
9784 (gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
9785 (gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
9786 Major rewrite of the channel mixing.
9787 We now allow previously conflicting channel positions to appear
9788 together (rear center and rear left/right for example).
9790 Rework the way channels are mixed together to take more possible
9791 channel positions into account, properly mix from/to side channels
9792 and don't assume that either center, left&right or nothing of a
9793 specific position is available anymore.
9794 * tests/check/elements/audioconvert.c: (GST_START_TEST):
9795 Adjust unit tests with non-standard 1/2 channel layouts to the more
9796 correct new behaviour.
9797 Add a unit test for 5.1->Stereo downmixing.
9799 2008-05-29 07:02:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9801 ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
9802 Original commit message from CVS:
9803 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
9804 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
9805 Add sane defaults for the 7 and 8 channel layouts as those are
9806 undefined in the Vorbis spec. Use NONE channel layouts when decoding
9807 more than 8 channels instead of erroring out. Fixes bug #535356.
9809 2008-05-28 16:10:20 +0000 Wim Taymans <wim.taymans@gmail.com>
9811 Add theoraparse to the docs and fix some docs.
9812 Original commit message from CVS:
9813 * docs/plugins/Makefile.am:
9814 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
9815 * docs/plugins/gst-plugins-base-plugins-sections.txt:
9816 * ext/theora/theoraparse.c:
9817 Add theoraparse to the docs and fix some docs.
9819 2008-05-28 15:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
9821 gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
9822 Original commit message from CVS:
9823 * gst-libs/gst/cdda/gstcddabasesrc.c:
9824 (gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
9825 Fix EOS condition and track addition check, the track.end sector is
9826 included in the track. Fixes #533265.
9828 2008-05-28 14:49:24 +0000 Mark Nauwelaerts <manauw@skynet.be>
9830 gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
9831 Original commit message from CVS:
9832 Patch by: Mark Nauwelaerts <manauw at skynet be>
9833 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
9834 (gst_video_rate_flush_prev), (gst_video_rate_event),
9835 (gst_video_rate_chain):
9836 * gst/videorate/gstvideorate.h:
9837 React (more) to NEWSEGMENT
9838 Small adjustment in timestamp calculation to prevent mismatches
9841 2008-05-28 11:31:44 +0000 Tim-Philipp Müller <tim@centricular.net>
9843 tests/examples/seek/seek.c: Initialise error to NULL as we should.
9844 Original commit message from CVS:
9845 * tests/examples/seek/seek.c: (make_parselaunch_pipeline):
9846 Initialise error to NULL as we should.
9848 2008-05-28 08:14:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9850 gst/adder/gstadder.c: Implement latency query.
9851 Original commit message from CVS:
9852 * gst/adder/gstadder.c: (gst_adder_query_duration),
9853 (gst_adder_query_latency), (gst_adder_query):
9854 Implement latency query.
9856 2008-05-27 18:10:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9858 gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
9859 Original commit message from CVS:
9860 * gst/adder/gstadder.c: (gst_adder_query_duration):
9861 Correctly resync the iterator if gst_iterator_next() returns
9862 GST_ITERATOR_RESYNC.
9864 2008-05-27 17:14:07 +0000 Tim-Philipp Müller <tim@centricular.net>
9866 win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
9867 Original commit message from CVS:
9868 * win32/vs6/libgstpbutils.dsp:
9869 Add pbutils-enumtypes.c to sources (#518037).
9871 2008-05-27 16:20:17 +0000 Wim Taymans <wim.taymans@gmail.com>
9873 gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
9874 Original commit message from CVS:
9875 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
9876 (gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
9877 * gst-libs/gst/audio/gstaudioclock.h:
9878 Add method to inform the clock that the time starts from 0 again. We use
9879 this info to calculate a clock offset so that the time we report in
9880 internal_time is monotonically increasing, as required by the clock base
9881 class. Fixes #521761.
9882 API: GstAudioClock::gst_audio_clock_reset()
9883 * gst-libs/gst/audio/gstbaseaudiosink.c:
9884 (gst_base_audio_sink_skew_slaving),
9885 (gst_base_audio_sink_change_state):
9886 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9887 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
9888 Reset reported time when we (re)create the ringbuffer.
9890 2008-05-27 16:11:32 +0000 Tim-Philipp Müller <tim@centricular.net>
9892 ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
9893 Original commit message from CVS:
9894 * ext/alsa/gstalsamixertrack.c:
9895 (gst_alsa_mixer_track_update_alsa_capabilities):
9896 Make sure playback volumes aren't accidentally overwritten by
9897 capture volumes if an alsa mixer track has both playback and
9898 capture capabilities: we create two GstMixerTracks in that
9899 case, so make sure we query only the alsa capabilities that
9900 refer to the type of GstMixerTrack we created from the dual
9901 capability alsa element. Should fix issues with Audigy2 sound
9904 2008-05-27 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.net>
9906 tests/check/pipelines/oggmux.c: Don't use deprecated function.
9907 Original commit message from CVS:
9908 * tests/check/pipelines/oggmux.c: (test_pipeline):
9909 Don't use deprecated function.
9911 2008-05-27 10:35:55 +0000 Wim Taymans <wim.taymans@gmail.com>
9913 gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
9914 Original commit message from CVS:
9915 * gst/playback/gstdecodebin2.c:
9916 (gst_decode_group_control_source_pad), (gst_decode_group_expose):
9917 Check for NULL cases and log them, creating ghostpads can, for example,
9918 fail when the pad returns wrong caps.
9919 * gst/playback/gstplaybin2.c: (perform_eos):
9920 When pushing out the EOS event, collect the return value and warn when
9923 2008-05-26 17:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
9925 gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
9926 Original commit message from CVS:
9927 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
9928 (gst_riff_create_video_template_caps):
9929 Add support for DVCPRO.
9931 2008-05-26 10:29:20 +0000 Tim-Philipp Müller <tim@centricular.net>
9933 gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
9934 Original commit message from CVS:
9935 * gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
9936 Change default scaling method from nearest-neighbour to bilinear.
9938 2008-05-26 10:26:00 +0000 Tim-Philipp Müller <tim@centricular.net>
9940 tests/check/libs/video.c: More checks.
9941 Original commit message from CVS:
9942 * tests/check/libs/video.c:
9945 2008-05-25 20:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
9947 Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
9948 Original commit message from CVS:
9949 * gst/subparse/gstsubparse.c: (parser_state_init),
9950 (gst_sub_parse_format_autodetect), (handle_buffer):
9951 * gst/subparse/gstsubparse.h:
9952 * tests/check/elements/subparse.c: (test_tmplayer_style3b):
9953 Limit duration to a maximum of five seconds for tmplayer format where
9954 we can guess the duration only from the timestamp of the next line of
9955 text. We don't want to show a text for eternities just because nothing
9956 else is being said for a while.
9958 2008-05-23 14:14:28 +0000 Wim Taymans <wim.taymans@gmail.com>
9960 gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
9961 Original commit message from CVS:
9962 * gst-libs/gst/rtp/gstbasertpdepayload.c:
9963 (gst_base_rtp_depayload_chain),
9964 (gst_base_rtp_depayload_handle_sink_event),
9965 (gst_base_rtp_depayload_push_full),
9966 (gst_base_rtp_depayload_change_state):
9967 Check sequence numbers, mark input buffers with a discont flag for the
9968 subclass when we detected a gap, drop duplicate buffers. We do this
9969 because one can use the element without a jitterbuffer in front and we
9970 don't want to feed the subclasses invalid or reordered data.
9971 Do an error when the subclass did not provide a process function instead
9973 Some other small cleanups.
9975 2008-05-22 22:35:40 +0000 Tim-Philipp Müller <tim@centricular.net>
9977 gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
9978 Original commit message from CVS:
9979 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
9980 May just as well use the precalculated uvstride here.
9982 2008-05-22 22:09:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9984 Add some documentation comments, and some new headers to be scanned.
9985 Original commit message from CVS:
9986 * docs/plugins/Makefile.am:
9987 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
9988 * docs/plugins/gst-plugins-base-plugins-sections.txt:
9989 * docs/plugins/gst-plugins-base-plugins.args:
9990 * docs/plugins/gst-plugins-base-plugins.hierarchy:
9991 * docs/plugins/gst-plugins-base-plugins.interfaces:
9992 * docs/plugins/gst-plugins-base-plugins.prerequisites:
9993 * docs/plugins/inspect/plugin-adder.xml:
9994 * docs/plugins/inspect/plugin-alsa.xml:
9995 * docs/plugins/inspect/plugin-audioconvert.xml:
9996 * docs/plugins/inspect/plugin-audiorate.xml:
9997 * docs/plugins/inspect/plugin-audioresample.xml:
9998 * docs/plugins/inspect/plugin-audiotestsrc.xml:
9999 * docs/plugins/inspect/plugin-cdparanoia.xml:
10000 * docs/plugins/inspect/plugin-decodebin.xml:
10001 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
10002 * docs/plugins/inspect/plugin-gdp.xml:
10003 * docs/plugins/inspect/plugin-gio.xml:
10004 * docs/plugins/inspect/plugin-gnomevfs.xml:
10005 * docs/plugins/inspect/plugin-libvisual.xml:
10006 * docs/plugins/inspect/plugin-ogg.xml:
10007 * docs/plugins/inspect/plugin-pango.xml:
10008 * docs/plugins/inspect/plugin-playback.xml:
10009 * docs/plugins/inspect/plugin-queue2.xml:
10010 * docs/plugins/inspect/plugin-subparse.xml:
10011 * docs/plugins/inspect/plugin-tcp.xml:
10012 * docs/plugins/inspect/plugin-theora.xml:
10013 * docs/plugins/inspect/plugin-typefindfunctions.xml:
10014 * docs/plugins/inspect/plugin-uridecodebin.xml:
10015 * docs/plugins/inspect/plugin-video4linux.xml:
10016 * docs/plugins/inspect/plugin-videorate.xml:
10017 * docs/plugins/inspect/plugin-videoscale.xml:
10018 * docs/plugins/inspect/plugin-videotestsrc.xml:
10019 * docs/plugins/inspect/plugin-volume.xml:
10020 * docs/plugins/inspect/plugin-vorbis.xml:
10021 * docs/plugins/inspect/plugin-ximagesink.xml:
10022 * docs/plugins/inspect/plugin-xvimagesink.xml:
10023 * ext/cdparanoia/gstcdparanoiasrc.c:
10024 * ext/ogg/gstoggdemux.c:
10025 * ext/ogg/gstoggdemux.h:
10026 * ext/ogg/gstoggmux.c:
10027 * ext/ogg/gstoggmux.h:
10028 * gst/audioconvert/audioconvert.c:
10029 * gst/audioconvert/audioconvert.h:
10030 * gst/audioconvert/gstaudioconvert.h:
10031 * gst/gdp/gstgdpdepay.h:
10032 * gst/gdp/gstgdppay.h:
10033 * gst/playback/gstdecodebin.c:
10034 * gst/playback/gstdecodebin2.c:
10035 * gst/playback/gstplaybin.c:
10036 * gst/playback/gstplaybin2.c:
10037 * gst/playback/gsturidecodebin.c:
10038 * gst/tcp/gstmultifdsink.c:
10039 * gst/tcp/gstmultifdsink.h:
10040 * gst/tcp/gsttcp.h:
10041 Add some documentation comments, and some new headers to be scanned.
10042 Rename some internal enum declarations (audioconvert's DitherType and
10043 NoiseShapingType, GstUnitType from the TCP elements) to match the
10044 documented GObject type names so that the docs pick them up.
10045 Name the playbin2 docs markups properly so they get picked up. They'll
10046 need renaming back when/if playbin2 becomes playbin.
10047 100% symbol coverage for the plugin docs, booya.
10049 2008-05-22 18:30:15 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
10051 gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
10052 Original commit message from CVS:
10053 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
10054 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
10055 Fix generation of NV12/NV21 frames. Fixes bug #532454.
10057 2008-05-22 11:59:33 +0000 Sjoerd Simons <sjoerd@luon.net>
10059 gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
10060 Original commit message from CVS:
10061 Patch by: Sjoerd Simons <sjoerd at luon dot net>
10062 * gst/playback/gstdecodebin.c: (remove_fakesink):
10063 Lock the fakesink before setting the state to NULL and removing it from
10064 the bin so that a concurrent state change cannot interfere.
10067 2008-05-21 17:09:42 +0000 Felipe Contreras <felipe.contreras@nokia.com>
10069 docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
10070 Original commit message from CVS:
10071 * docs/Makefile.am:
10072 Fix installing plugin documentation when gtk-doc is disabled.
10074 2008-05-21 17:01:16 +0000 Felipe Contreras <felipe.contreras@nokia.com>
10076 gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
10077 Original commit message from CVS:
10078 * gst-libs/gst/rtsp/Makefile.am:
10079 Distribute, don't install md5.h
10081 2008-05-21 16:47:58 +0000 Julien Moutte <julien@moutte.net>
10083 gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
10084 Original commit message from CVS:
10085 2008-05-21 Julien Moutte <julien@fluendo.com>
10086 * gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
10087 instead of SOL_IP, works on more platforms.
10088 * gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
10091 2008-05-21 16:44:15 +0000 Wim Taymans <wim.taymans@gmail.com>
10093 Some debug and comment fixes.
10094 Original commit message from CVS:
10095 * ext/vorbis/vorbisdec.c:
10096 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
10097 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
10098 Some debug and comment fixes.
10099 * tests/examples/dynamic/addstream.c: (main):
10102 2008-05-21 16:36:50 +0000 Wim Taymans <wim.taymans@gmail.com>
10104 Don't use bad gst_element_get_pad().
10105 Original commit message from CVS:
10106 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
10107 * gst/playback/decodetest.c: (new_decoded_pad_cb):
10108 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
10109 (try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
10110 (cleanup_decodebin):
10111 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
10112 (connect_element), (gst_decode_group_control_demuxer_pad):
10113 * gst/playback/gstplaybasebin.c: (queue_remove_probe),
10114 (queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
10116 * gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
10117 (gst_play_bin_set_property), (handoff), (gen_video_element),
10118 (gen_text_element), (gen_audio_element), (gen_vis_element),
10119 (remove_sinks), (add_sink), (setup_sinks):
10120 * gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
10121 * gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
10122 (gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
10123 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
10124 (gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
10125 (gen_video_chain), (gen_text_chain), (gen_audio_chain),
10126 (gen_vis_chain), (gst_play_sink_reconfigure),
10127 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
10128 (gst_play_sink_request_pad):
10129 * gst/playback/gsturidecodebin.c: (type_found), (setup_source):
10130 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
10132 * gst/playback/test6.c: (new_decoded_pad_cb):
10133 * tests/check/elements/audioconvert.c: (GST_START_TEST):
10134 * tests/check/elements/audiorate.c: (test_injector_chain),
10135 (do_perfect_stream_test):
10136 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
10137 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
10138 * tests/check/elements/gnomevfssink.c:
10139 * tests/check/elements/textoverlay.c:
10140 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
10141 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
10142 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
10143 * tests/check/pipelines/oggmux.c: (test_pipeline):
10144 * tests/check/pipelines/streamheader.c: (GST_START_TEST):
10145 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
10146 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
10147 * tests/examples/seek/scrubby.c: (make_wav_pipeline):
10148 * tests/examples/seek/seek.c: (make_mod_pipeline),
10149 (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
10150 (make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
10151 (make_theora_pipeline), (make_vorbis_theora_pipeline),
10152 (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
10153 (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
10154 (update_fill), (msg_buffering):
10155 Don't use bad gst_element_get_pad().
10157 2008-05-21 14:35:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10159 gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
10160 Original commit message from CVS:
10161 * gst-libs/gst/riff/riff-media.c:
10162 Fix wrong method name in docs. Fix calculation of strf fields for
10164 * gst-libs/gst/riff/riff-read.c:
10165 Whitespace fix and removing double ';'.
10167 2008-05-21 11:52:30 +0000 Wim Taymans <wim.taymans@gmail.com>
10169 docs/design/part-playbin2.txt: Add some leftover doc.
10170 Original commit message from CVS:
10171 * docs/design/part-playbin2.txt:
10172 Add some leftover doc.
10174 2008-05-21 11:36:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10176 gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
10177 Original commit message from CVS:
10178 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
10179 Fix copy & paste error in last commit.
10181 2008-05-21 11:30:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10183 gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
10184 Original commit message from CVS:
10185 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
10186 Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
10187 other channel positions when source has SIDE channels and dest doesn't
10188 or the other way around.
10190 2008-05-21 11:29:25 +0000 Henrik Eriksson <henriken@axis.com>
10192 gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
10193 Original commit message from CVS:
10194 Patch by: Henrik Eriksson <henriken at axis dot com>
10195 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
10196 (gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
10197 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
10198 (gst_multi_fd_sink_get_property):
10199 * gst/tcp/gstmultifdsink.h:
10200 Add support for DSCP QOS. Fixes #469933.
10202 2008-05-21 07:46:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10204 tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
10205 Original commit message from CVS:
10206 * tests/check/elements/audioconvert.c: (GST_START_TEST):
10207 Add another test that checks if conversion between standard 1 and 2
10208 channel layouts with and without positions set is working.
10210 2008-05-21 07:39:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10212 gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
10213 Original commit message from CVS:
10214 * gst-libs/gst/audio/multichannel.c:
10215 (gst_audio_check_channel_positions):
10216 Allow non-standard 2 channel layouts.
10217 * tests/check/elements/audioconvert.c: (GST_START_TEST):
10218 Add some tests for converting and remapping non-standard 1 and 2
10221 2008-05-21 07:28:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10223 gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
10224 Original commit message from CVS:
10225 * gst/audioconvert/gstchannelmix.c:
10226 (gst_channel_mix_fill_normalize):
10227 Prevent division by zero if the channel mix matrix contains only
10230 2008-05-21 06:45:22 +0000 Antoine Tremblay <hexa00@gmail.com>
10232 gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
10233 Original commit message from CVS:
10234 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
10235 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
10236 Close a buffer memory leak. Fixes bug #534071.
10238 2008-05-21 06:39:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10240 gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
10241 Original commit message from CVS:
10242 * gst-libs/gst/rtsp/gstrtsptransport.h:
10243 Make the GstRTSPTransport struct members public as there are no
10244 setters/getters and it's supposed to be changed directly.
10247 2008-05-21 05:48:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10249 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...
10250 Original commit message from CVS:
10251 * gst/adder/gstadder.c:
10252 Adder also doesn't support audio/x-raw-int with width!=depth so don't
10253 claim this on the pad template caps.
10255 2008-05-20 16:26:53 +0000 Wim Taymans <wim.taymans@gmail.com>
10257 gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
10258 Original commit message from CVS:
10259 * gst-libs/gst/audio/gstbaseaudiosink.c:
10260 (gst_base_audio_sink_sync_latency):
10261 We can only use our optimal calibration if we prerolled before the
10264 2008-05-20 14:35:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10266 configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
10267 Original commit message from CVS:
10269 Require core CVS for GstBaseSrc buffer caps setting magic.
10271 2008-05-20 12:26:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10273 gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
10274 Original commit message from CVS:
10275 * gst/audioconvert/gstaudioconvert.c:
10276 (gst_audio_convert_fixate_channels):
10277 Fix logic in last commit.
10279 2008-05-20 12:15:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10281 gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
10282 Original commit message from CVS:
10283 * gst/audioconvert/gstaudioconvert.c:
10284 (gst_audio_convert_fixate_channels):
10285 Passthrough the channel positions if the number of output channels is
10286 the same as the number of input channels, the input had a channel
10287 layout and downstream requests no special one. We did this already for
10288 > 2 channels but now it's also done for 1 channel. Fixes bug #533617.
10290 2008-05-20 11:13:27 +0000 Wim Taymans <wim.taymans@gmail.com>
10292 ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
10293 Original commit message from CVS:
10294 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
10295 (gst_gnome_vfs_src_finalize),
10296 (gst_gnome_vfs_src_received_headers_callback),
10297 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
10298 * ext/gnomevfs/gstgnomevfssrc.h:
10299 Set the ICY caps on the srcpad from where they get picked up by the base
10300 class now and set on the outgoing buffers.
10301 * gst-libs/gst/audio/gstbaseaudiosrc.c:
10302 (gst_base_audio_src_create):
10303 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
10304 BaseSrc now sets the caps on outgoing buffers automatically.
10306 2008-05-20 11:09:06 +0000 Wim Taymans <wim.taymans@gmail.com>
10308 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
10309 Original commit message from CVS:
10310 * gst-libs/gst/audio/gstbaseaudiosink.c:
10311 (gst_base_audio_sink_resample_slaving),
10312 (gst_base_audio_sink_skew_slaving),
10313 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
10314 (gst_base_audio_sink_async_play),
10315 (gst_base_audio_sink_change_state):
10316 Change the way in which the ringbuffer is started when dealing with a
10317 slaved clock and latency. We now sync to the clock until we reach
10318 upstream latency before starting the ringbuffer. This has the effect
10319 that we can accurately align the master and slave clocks and let the
10320 rate correction code take care of the initial drift or rounding errors
10321 instead of leaving them uncorrected with the old approach.
10323 2008-05-20 08:12:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10325 gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
10326 Original commit message from CVS:
10327 * gst/audioconvert/gstaudioconvert.c:
10328 (gst_audio_convert_fixate_channels):
10329 Correctly set the default channel positions when converting to 8
10332 2008-05-19 16:13:25 +0000 Tim-Philipp Müller <tim@centricular.net>
10334 configure.ac: Error out if we don't have the required version of core.
10335 Original commit message from CVS:
10337 Error out if we don't have the required version of core.
10339 2008-05-19 15:59:40 +0000 Tim-Philipp Müller <tim@centricular.net>
10341 gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
10342 Original commit message from CVS:
10343 * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
10344 Use data scan helper in aac typefinder and stop scanning
10345 for headers when we've found a type. Also fix potential invalid
10346 memory access when calculating the frame length.
10348 2008-05-19 14:09:08 +0000 Tim-Philipp Müller <tim@centricular.net>
10350 gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
10351 Original commit message from CVS:
10352 * gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
10353 (mpeg_sys_is_valid_pack):
10354 Don't modify scan context when we return FALSE in ensure_data, so
10355 it's possible to continue scanning, and we don't end up with a NULL
10356 data pointer and a positive size, which might bite us the next time
10357 we're called. Small constification.
10359 2008-05-16 21:12:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10361 gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
10362 Original commit message from CVS:
10363 * gst/adder/gstadder.c:
10364 Adder doesn't support 24 bit samples so don't claim it supports them
10365 in the pad template caps.
10367 2008-05-14 20:28:02 +0000 Wim Taymans <wim.taymans@gmail.com>
10369 gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
10370 Original commit message from CVS:
10371 * gst-libs/gst/rtp/gstbasertpdepayload.c:
10372 (gst_base_rtp_depayload_chain):
10373 Validate the RTP packet before further processing it. It's just too
10374 dangerous to accept random packets and people are not forced to use a
10375 jitterbuffer or session manager to filter out the bad packets.
10376 * gst-libs/gst/rtp/gstrtpbuffer.c:
10377 (gst_rtp_buffer_set_extension_data),
10378 (gst_rtp_buffer_get_payload_subbuffer):
10380 When setting extension data in a buffer that is too small, we fail and
10381 we should not set the extension bit.
10382 Change GST_WARNINGS into g_warning because they really are
10383 programming errors.
10384 * tests/check/libs/rtp.c: (GST_START_TEST):
10385 Catch the g_warnings now in the unit tests and that fact that failing to
10386 set extension data left the extension bit untouched.
10388 2008-05-14 13:57:41 +0000 Tim-Philipp Müller <tim@centricular.net>
10390 gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
10391 Original commit message from CVS:
10392 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
10393 Revert previous change which made basetransform handle buffer_alloc
10394 and which breaks things badly in the non-passthrough case since it
10395 returned buffers with a different (ie. sometimes smaller) size than
10396 the size requested.
10398 2008-05-14 13:43:12 +0000 Bernard B <b-gnome@largestprime.net>
10400 gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
10401 Original commit message from CVS:
10402 Patch by: Bernard B <b-gnome at largestprime dot net>
10403 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
10404 Fix seqnum compare function for bordercase values and fix the docs
10405 again. Fixes #533075.
10406 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
10407 Add a testcase for seqnum compare function.
10409 2008-05-14 10:58:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10411 gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
10412 Original commit message from CVS:
10413 * gst/adder/gstadder.c: (gst_adder_setcaps),
10414 (gst_adder_class_init):
10415 Correctly declare the supported endianness on the pad templates
10416 and check for correct endianness in the set caps function. Adder
10417 only supports native endianness.
10418 Also use gst_element_class_set_details_simple().
10420 2008-05-14 09:12:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10422 sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.
10423 Original commit message from CVS:
10424 * sys/xvimage/xvimagesink.c:
10425 Better debug logging in port value handling. Merging separate port
10426 value loops into one.
10428 2008-05-13 16:02:19 +0000 Hannes Bistry <hannesb@gmx.de>
10430 gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
10431 Original commit message from CVS:
10432 Patch by: Hannes Bistry <hannesb at gmx dot de>
10433 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
10434 * gst/tcp/gsttcpserversink.c:
10435 (gst_tcp_server_sink_handle_server_read),
10436 (gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
10437 Fix regression in clientsrc because we did not add the fd to the poll
10438 set anymore. Fixes #532364.
10439 Do some cleanups here and there.
10441 2008-05-13 13:04:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10443 gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
10444 Original commit message from CVS:
10445 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
10446 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
10447 * gst/playback/gstplay-marshal.list:
10448 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
10449 Use correct marshallers. GstCaps are a boxed type and no GObject
10452 2008-05-13 11:37:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10454 win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.
10455 Original commit message from CVS:
10456 * win32/common/libgstrtsp.def:
10457 Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
10460 2008-05-13 10:59:49 +0000 Sjoerd Simons <sjoerd@luon.net>
10462 tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
10463 Original commit message from CVS:
10464 Patch by: Sjoerd Simons <sjoerd at luon dot net>
10465 * tests/check/elements/audioresample.c:
10466 (live_switch_alloc_only_48000), (live_switch_get_sink_caps),
10467 (live_switch_push), (GST_START_TEST):
10468 Add unit test for the latest basetransform negotiation changes.
10471 2008-05-13 09:14:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10473 gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
10474 Original commit message from CVS:
10475 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
10476 Fix nv12<->nv21 conversion if stride is larger than width.
10478 2008-05-13 07:28:21 +0000 j^ <j@oil21.org>
10480 ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...
10481 Original commit message from CVS:
10482 Patch by: j^ <j at oil21 dot org>
10483 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
10484 (gst_ogg_pad_parse_skeleton_fisbone):
10485 * ext/ogg/gstoggdemux.h:
10486 Parse presentation time from skeleton streams and use it as offset
10487 for the timestamps. Fixes bug #530068.
10489 2008-05-12 08:45:11 +0000 Wim Taymans <wim.taymans@gmail.com>
10491 gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
10492 Original commit message from CVS:
10493 * gst-libs/gst/audio/gstbaseaudiosink.c:
10494 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
10495 Revert previous patch that attempted to more accurately calculate the
10496 initial offset between master and slave clock. The best thing we can do
10497 in general is take the time of both clocks as the diff since we don't
10498 know when the actual preroll happened.
10500 2008-05-11 19:52:59 +0000 Tim-Philipp Müller <tim@centricular.net>
10502 gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
10503 Original commit message from CVS:
10504 * gst-libs/gst/pbutils/install-plugins.c:
10505 Fix docs: type and missing word.
10507 2008-05-10 20:16:21 +0000 Tim-Philipp Müller <tim@centricular.net>
10509 gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
10510 Original commit message from CVS:
10511 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
10512 Don't do lots of 4-byte peeks, but use the 'new' data scan helper
10513 for this instead; don't check if we've found enough markers after
10514 each and every step, it's enough to do that only if we've actually
10515 found a new marker.
10516 Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
10518 2008-05-10 18:19:17 +0000 Tim-Philipp Müller <tim@centricular.net>
10520 gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
10521 Original commit message from CVS:
10522 * gst/typefind/gsttypefindfunctions.c:
10523 (DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
10524 (data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
10525 (mpeg_video_stream_type_find):
10526 Move scan helper thingy to the beginning of the file so we can use
10527 it in other typefind functions. Rename it to something more
10528 generic. Also improve handling of things towards the end of the
10529 typefind data: peek as much as we can if we know the size of the
10530 data, rather than just min_size.
10532 2008-05-09 21:42:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10534 Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
10535 Original commit message from CVS:
10536 * docs/libs/gst-plugins-base-libs-sections.txt:
10537 * gst-libs/gst/interfaces/colorbalance.c:
10538 * gst-libs/gst/interfaces/colorbalance.h:
10539 * gst-libs/gst/interfaces/colorbalancechannel.c:
10540 * gst-libs/gst/interfaces/colorbalancechannel.h:
10541 * gst-libs/gst/interfaces/tuner.c:
10542 * gst-libs/gst/interfaces/tunerchannel.c:
10543 * gst-libs/gst/interfaces/tunerchannel.h:
10544 * gst-libs/gst/interfaces/tunernorm.c:
10545 * gst-libs/gst/interfaces/tunernorm.h:
10546 * gst-libs/gst/video/video.c:
10547 * gst-libs/gst/video/video.h:
10548 Document the GstTuner and GstColorBalance interfaces, and some
10549 other random API functions that needed it. 70% symbol coverage, woo.
10551 2008-05-09 16:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
10553 gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
10554 Original commit message from CVS:
10555 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
10556 Choose to allocate one less segment but require one additional segment
10558 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
10559 No need to increment the number of segments in the source.
10560 * gst-libs/gst/audio/gstbaseaudiosink.c:
10561 (gst_base_audio_sink_get_time), (clock_convert_external),
10562 (gst_base_audio_sink_resample_slaving),
10563 (gst_base_audio_sink_skew_slaving),
10564 (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
10565 (gst_base_audio_sink_async_play):
10566 Remove adding latency when returning the internal time while subtracting
10567 it again when we use the value a little later.
10568 When calculating the end timestamp, we are making a rounding error
10569 with the current algorithm. Ensure that we don't accumulate these
10570 rounding errors when aligning samples by not resampling at all if we
10571 don't need to. Fixes #419351.
10572 Make the initial calibration of the clock slaving a little more
10573 predictable and accurate. Also handle the case where we don't do
10576 2008-05-09 08:34:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10578 gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
10579 Original commit message from CVS:
10580 Based on a patch by:
10581 Björn Benderius <bjoern dot benderius at axis dot com>
10582 * gst/ffmpegcolorspace/avcodec.h:
10583 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
10584 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
10585 (gst_ffmpegcsp_avpicture_fill):
10586 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
10587 * gst/ffmpegcolorspace/imgconvert_template.h:
10588 Add conversions from/to NV12 and NV21 and conversions between those
10589 two formats. Fixes bug #532166.
10591 2008-05-08 17:35:44 +0000 Edward Hervey <bilboed@bilboed.com>
10593 gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
10594 Original commit message from CVS:
10595 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
10596 Abort the h264 typefinding as soon as _peek() doesn't return anything,
10597 which happens for example with files smaller than 128kb.
10599 2008-05-08 14:46:27 +0000 Wouter Cloetens <zombie@e2big.org>
10601 gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
10602 Original commit message from CVS:
10603 Patch by: Wouter Cloetens <zombie at e2big dot org>
10604 * gst-libs/gst/rtsp/Makefile.am:
10605 * gst-libs/gst/rtsp/gstrtspconnection.c:
10606 (gst_rtsp_connection_create), (md5_digest_to_hex_string),
10607 (auth_digest_compute_hex_urp), (auth_digest_compute_response),
10608 (add_auth_header), (gst_rtsp_connection_free),
10609 (gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
10610 (gst_rtsp_connection_set_auth_param),
10611 (gst_rtsp_connection_clear_auth_params):
10612 * gst-libs/gst/rtsp/gstrtspconnection.h:
10613 Add Digest authorization support for RTSP connections. See #532065.
10614 * gst-libs/gst/rtsp/md5.c:
10615 * gst-libs/gst/rtsp/md5.h:
10616 Yeap, another md5 implementation until we can depend on a glib that has
10619 2008-05-08 06:20:42 +0000 Sjoerd Simons <sjoerd@luon.net>
10621 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
10622 Original commit message from CVS:
10623 Patch by: Sjoerd Simons <sjoerd at luon dot net>
10624 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
10625 Let audioresample use the buffer allocation of basetransform instead
10627 * tests/check/elements/audioresample.c: (alloc_only_48000),
10628 (GST_START_TEST), (audioresample_suite):
10629 Add unit test for the recent basetransform bugfix, where upstream
10630 changes caps to something that can't be passed through anymore.
10632 2008-05-07 19:50:27 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
10634 win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
10635 Original commit message from CVS:
10636 * win32/common/config.h.in:
10637 Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
10638 use the real thing than having "???" unconditionally.
10640 2008-05-07 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
10642 gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
10643 Original commit message from CVS:
10644 * gst-libs/gst/audio/gstbaseaudiosink.c:
10645 (gst_base_audio_sink_query):
10646 Report the latency with the new seglatency parameter.
10647 * gst-libs/gst/audio/gstringbuffer.c:
10648 (gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
10649 (gst_ring_buffer_acquire):
10650 * gst-libs/gst/audio/gstringbuffer.h:
10651 Add new field to the ringbufferspec to specify the expected latency
10652 between the underlying device read/write pointer, this is needed
10653 when writing sinks that sit a little closer to the hardware.
10654 Add some more docs for other fields.
10656 2008-05-07 10:38:23 +0000 Wim Taymans <wim.taymans@gmail.com>
10658 gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
10659 Original commit message from CVS:
10660 * gst-libs/gst/app/.cvsignore:
10661 * gst-libs/gst/app/Makefile.am:
10662 * gst-libs/gst/app/gstapp-marshal.list:
10663 Add marshal.list, make it compile and add to cvsignore.
10664 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
10665 (gst_app_sink_stop):
10667 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
10668 (gst_app_src_init), (gst_app_src_set_property),
10669 (gst_app_src_get_property), (gst_app_src_unlock),
10670 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
10671 (gst_app_src_create), (gst_app_src_set_caps),
10672 (gst_app_src_get_caps), (gst_app_src_set_size),
10673 (gst_app_src_get_size), (gst_app_src_set_seekable),
10674 (gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
10675 (gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
10676 (gst_app_src_end_of_stream):
10677 * gst-libs/gst/app/gstappsrc.h:
10678 Beat appsrc in shape, add signals and actions.
10680 Add properties for caps, size, seekability and max-buffers.
10681 Fix unlock/stop code.
10683 2008-05-06 12:35:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10685 gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
10686 Original commit message from CVS:
10687 * gst/volume/gstvolume.c: (volume_transform_ip):
10688 Return NOT_NEGOTIATED if we didn't set a process function yet for some
10689 reason instead of crashing later. Might fix bug #509125.
10691 2008-05-06 12:12:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
10693 gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
10694 Original commit message from CVS:
10695 Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
10696 * gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
10697 * gst/audioconvert/audioconvert.h:
10698 * gst/audioconvert/gstaudioconvert.c:
10699 (gst_audio_convert_parse_caps),
10700 (structure_has_fixed_channel_positions),
10701 (gst_audio_convert_transform_caps):
10702 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
10703 Add support for more than 8 channels and NONE channel layouts. For
10704 more than 8 channels no channel conversion is supported yet, only
10705 format conversions are supported. Fixes bug #398033.
10706 * tests/check/elements/audioconvert.c: (verify_convert),
10707 (GST_START_TEST), (audioconvert_suite):
10708 Add some unit tests by Tim for checking the NONE channel layouts
10709 and more than 8 channels and add some more unit tests for channel
10712 2008-05-06 10:16:49 +0000 Wim Taymans <wim.taymans@gmail.com>
10714 gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
10715 Original commit message from CVS:
10716 * gst/playback/gstdecodebin2.c: (connect_pad):
10717 When autoplugging fails, set the element back to NULL before
10720 2008-05-06 09:59:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10722 win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols.
10723 Original commit message from CVS:
10724 * win32/common/libgstaudio.def:
10725 Add gst_base_audio_src_[sg]et_slave_method() to the exported
10728 2008-05-05 12:33:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10730 gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
10731 Original commit message from CVS:
10732 * gst/subparse/samiparse.c: (handle_start_sync),
10733 (end_sami_element), (characters_sami):
10734 Remove trailing, leading and double whitespaces.
10735 Correctly timestamp buffers and output the last buffer too.
10736 * tests/check/elements/subparse.c: (GST_START_TEST),
10738 Add a simple unit test for SAMI parsing.
10740 2008-05-05 11:14:48 +0000 Young-Ho Cha <ganadist@chollian.net>
10742 gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
10743 Original commit message from CVS:
10744 Patch by: Young-Ho Cha <ganadist at chollian dot net>
10745 * gst/subparse/samiparse.c: (handle_start_sync),
10746 (start_sami_element), (end_sami_element), (characters_sami),
10747 (sami_context_reset):
10748 Only output characters inside the "sync" elements. There could be
10749 other elements like "style" that have some content but should
10750 not be printed. Fixes bug #467911.
10752 2008-05-05 10:27:45 +0000 Wim Taymans <wim.taymans@gmail.com>
10754 gst-libs/gst/app/gstappsink.*: Start some docs.
10755 Original commit message from CVS:
10756 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
10757 (gst_app_sink_init), (gst_app_sink_set_property),
10758 (gst_app_sink_get_property), (gst_app_sink_unlock_start),
10759 (gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
10760 (gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
10761 (gst_app_sink_preroll), (gst_app_sink_render),
10762 (gst_app_sink_set_caps), (gst_app_sink_set_drop),
10763 (gst_app_sink_get_drop):
10764 * gst-libs/gst/app/gstappsink.h:
10766 Add property to drop buffers when the queue is filled
10767 Fix unlocking and flushing when the queues are filled.
10769 2008-05-05 10:03:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10771 gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
10772 Original commit message from CVS:
10773 * gst/playback/gstplaybasebin.c: (set_audio_mute),
10774 (set_active_source):
10775 * gst/playback/gstplaybasebin.h:
10776 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
10777 (playbin_set_audio_mute):
10778 Allow setting -1 as current-audio to mute the current audio stream,
10779 similar to what is done for subtitles. Fixes bug #342294.
10781 2008-05-05 07:41:03 +0000 Edward Hervey <bilboed@bilboed.com>
10783 gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
10784 Original commit message from CVS:
10785 * gst-libs/gst/pbutils/descriptions.c: (formats):
10786 It's SorensOn and not SorensEn.
10788 2008-05-04 15:23:36 +0000 Tim-Philipp Müller <tim@centricular.net>
10790 gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
10791 Original commit message from CVS:
10792 * gst-libs/gst/pbutils/descriptions.c: (formats):
10793 Fix description of video/x-flash-video.
10795 2008-05-04 15:02:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10797 Remove some unused code.
10798 Original commit message from CVS:
10799 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
10800 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
10801 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
10802 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
10803 Remove some unused code.
10804 * gst/audioconvert/gstaudioquantize.c:
10805 (gst_audio_quantize_free_noise_shaping):
10806 Don't return before freeing the noise shaping history.
10808 2008-05-03 16:00:04 +0000 Tim-Philipp Müller <tim@centricular.net>
10810 tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
10811 Original commit message from CVS:
10812 * tests/check/elements/subparse.c: (do_test),
10813 (test_tmplayer_style3b), (subparse_suite):
10814 Add unit test for the tmplayer variant from bug #530962.
10816 2008-05-03 15:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
10818 gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
10819 Original commit message from CVS:
10820 * gst/subparse/gstsubparse.c: (handle_buffer),
10821 (gst_sub_parse_sink_event):
10822 * gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
10823 (tmplayer_parse_line):
10824 Fix parsing of tmplayer subtitle variant where every single line contains
10825 text and there isn't an empty line after each line to determine the
10826 duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
10827 making sure that we push out the last line of text without a duration if
10828 there's still text left in the buffer at the end.
10830 2008-05-03 15:39:04 +0000 Tim-Philipp Müller <tim@centricular.net>
10832 gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
10833 Original commit message from CVS:
10834 * gst/subparse/gstsubparse.c: (feed_textbuf):
10835 Fix detection of discontinuities based on the buffer offset (doesn't work
10836 so well if no buffer offset is set) and also check for the DISCONT buffer
10837 flag. This keeps the parser state from being reset after each buffer in
10840 2008-05-03 12:09:16 +0000 Tim-Philipp Müller <tim@centricular.net>
10842 gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
10843 Original commit message from CVS:
10844 * gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
10845 Further fine-tuning: don't absolutely require sequence or GOP headers
10846 (as introduced in the previous commit), but adjust the typefind
10847 probabilities returned accordingly if we don't see them. Also make sure
10848 picture header and first slice are somewhat close to each other (which
10849 is not perfect but still better than requiring a fixed offset or having
10852 2008-05-02 12:13:08 +0000 Wim Taymans <wim.taymans@gmail.com>
10854 gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
10855 Original commit message from CVS:
10856 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
10857 (gst_basertppayload_sink_setcaps),
10858 (gst_basertppayload_sink_getcaps):
10859 Rename the setcaps/getcaps function internally to make it clear that
10860 they are called for the sink pad.
10862 2008-05-02 12:11:07 +0000 Wim Taymans <wim.taymans@gmail.com>
10864 gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
10865 Original commit message from CVS:
10866 * gst-libs/gst/rtp/gstbasertpdepayload.c:
10867 (gst_base_rtp_depayload_class_init),
10868 (gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
10869 (gst_base_rtp_depayload_packet_lost),
10870 (gst_base_rtp_depayload_set_gst_timestamp):
10871 * gst-libs/gst/rtp/gstbasertpdepayload.h:
10872 Catch packet-lost events from the jitterbuffer and convert them into a
10873 vmethod call (lost-packet) so that depayloaders can do something smart.
10874 Also add a default packet-lost function that sends out a segment update
10877 2008-05-02 11:13:05 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10879 gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
10880 Original commit message from CVS:
10881 * gst/playback/test4.c:
10882 * gst/playback/test5.c:
10883 * gst/playback/test6.c:
10884 * gst/playback/test7.c:
10885 Also include config.h when relying on defines from it. Fixes the
10886 build. Its been a please to serve :)
10888 2008-05-02 10:54:51 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
10891 * gst/videotestsrc/videotestsrc.c:
10892 Add support for NV12 and NV21 in videotestsrc
10893 Original commit message from CVS:
10894 * gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
10895 (paint_setup_NV21), (paint_hline_NV12_NV21):
10896 Add support for NV12 and NV21 in videotestsrc
10898 2008-05-02 10:02:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10900 gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
10901 Original commit message from CVS:
10902 * gst/videoscale/gstvideoscale.c:
10903 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
10904 * gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
10905 (vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
10906 (vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
10907 (vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
10908 (vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
10909 (vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
10910 (vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
10911 (vs_image_scale_linear_RGB555):
10912 Support 1x1 images as input and output as for example the BBC HQ new
10913 streams have 1x1 GIFs in the playlists for some reason.
10915 2008-05-01 19:11:42 +0000 Tim-Philipp Müller <tim@centricular.net>
10917 gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
10918 Original commit message from CVS:
10919 * gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
10921 If we can't activate one of the decoders we plugged in (such as,
10922 say, musepackdec) for some reason (it might not support push mode,
10923 for example), remove any pad probes that close_pad_link() might
10924 have set up. This makes sure we later don't try to remove a probe
10925 for a pad that doesn't exist any longer, and avoids nast warnings
10926 and probably other things too.
10928 2008-04-30 20:54:56 +0000 Tim-Philipp Müller <tim@centricular.net>
10930 gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
10931 Original commit message from CVS:
10932 * gst/typefind/gsttypefindfunctions.c:
10933 (mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
10935 Rework mpeg video stream typefinding a bit more: make sure sequence,
10936 GOP, picture and slice headers appear in the order they should and
10937 that we've in fact at least had one of each; fix picture header
10938 detection; decouple picture and slice header check - don't assume
10939 they're at a fixed offset, there may be extra data in between. Also,
10940 announce varying degrees of probability depending on what we found
10941 exactly (multiple pictures, at least one picture, just sequence and
10942 GOP headers). Finally, in _ensure_data(), take into account that we
10943 might be typefinding smaller amounts of data, such as the first
10944 buffer of a stream, so fall back to the minimum size needed as long
10945 as that's available, instead of erroring out if there's less than
10946 2kB of data. Fixes #526173. Conveniently also doesn't recognise the
10947 fuzzed file from #399342 as valid.
10949 2008-04-30 17:06:45 +0000 Michael Smith <msmith@xiph.org>
10951 ext/theora/theoradec.c: Cool kids don't divide by zero.
10952 Original commit message from CVS:
10953 * ext/theora/theoradec.c:
10954 Cool kids don't divide by zero.
10955 Treat PAR of x:0 as 1:1.
10958 2008-04-30 14:37:52 +0000 Tim-Philipp Müller <tim@centricular.net>
10960 gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
10961 Original commit message from CVS:
10962 * gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
10963 (mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
10964 (mpeg_video_stream_type_find):
10965 Refactor a bit: use context structure to track parsing offset and size of
10966 available data and make the code a bit clearer. Fixes bad memory access
10969 2008-04-28 22:18:49 +0000 Michael Smith <msmith@xiph.org>
10971 gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
10972 Original commit message from CVS:
10973 * gst/playback/test4.c:
10974 * gst/playback/test5.c:
10975 * gst/playback/test6.c:
10976 * gst/tcp/gstmultifdsink.c:
10977 Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
10980 2008-04-28 08:51:38 +0000 Wim Taymans <wim.taymans@gmail.com>
10982 gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
10983 Original commit message from CVS:
10984 * gst-libs/gst/audio/gstbaseaudiosink.h:
10986 * gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
10987 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
10988 (gst_base_audio_src_set_slave_method),
10989 (gst_base_audio_src_get_slave_method),
10990 (gst_base_audio_src_set_property),
10991 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
10992 * gst-libs/gst/audio/gstbaseaudiosrc.h:
10993 Add property and methods for selecting the clock slave method in the
10994 source, like in the sink.
10995 We only implement "none" and "re-timestamp" for now.
10996 API: gst_base_audio_src_set_slave_method()
10997 API: gst_base_audio_src_get_slave_method()
10999 2008-04-25 18:18:47 +0000 Wim Taymans <wim.taymans@gmail.com>
11001 gst-libs/gst/app/gstappsink.*: Add more docs.
11002 Original commit message from CVS:
11003 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
11004 (gst_app_sink_init), (gst_app_sink_set_property),
11005 (gst_app_sink_get_property), (gst_app_sink_event),
11006 (gst_app_sink_preroll), (gst_app_sink_render),
11007 (gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
11008 (gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
11009 (gst_app_sink_pull_buffer):
11010 * gst-libs/gst/app/gstappsink.h:
11012 Add signals for when preroll and render buffers are available.
11013 Add property to control signal emission.
11014 Add property to control the max queue size.
11016 2008-04-25 07:37:09 +0000 Wim Taymans <wim.taymans@gmail.com>
11018 gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
11019 Original commit message from CVS:
11020 * gst-libs/gst/rtp/gstrtpbuffer.c:
11021 Fix the docs about the seqnum compare function, it returns a difference.
11023 2008-04-24 09:27:35 +0000 Edward Hervey <bilboed@bilboed.com>
11025 ext/alsa/gstalsadeviceprobe.c: Don't return before freeing up the allocated structures.
11026 Original commit message from CVS:
11027 * ext/alsa/gstalsadeviceprobe.c:
11028 (gst_alsa_get_device_list): Don't return before freeing up
11029 the allocated structures.
11031 2008-04-24 08:19:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11033 gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
11034 Original commit message from CVS:
11035 * gst/playback/gstplaybin.c:
11036 Remove obsolete streaminfo code and fix a leak. Fixes #529546
11038 2008-04-23 13:50:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11040 ext/ogg/gstoggdemux.c: Revert the event part, that should not go in.
11041 Original commit message from CVS:
11042 * ext/ogg/gstoggdemux.c:
11043 Revert the event part, that should not go in.
11045 2008-04-23 13:45:29 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11047 ext/ogg/gstoggdemux.c: Don't leak GstPluginFeatures when filtering.
11048 Original commit message from CVS:
11049 * ext/ogg/gstoggdemux.c:
11050 Don't leak GstPluginFeatures when filtering.
11052 2008-04-23 08:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11054 sys/xvimage/xvimagesink.c: Add some logging for cases when grabbing the xv failed.
11055 Original commit message from CVS:
11056 * sys/xvimage/xvimagesink.c:
11057 Add some logging for cases when grabbing the xv failed.
11059 2008-04-22 06:18:04 +0000 David Schleef <ds@schleef.org>
11061 ext/ogg/gstoggmux.c: Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos packet. Should conform to what we cu...
11062 Original commit message from CVS:
11063 * ext/ogg/gstoggmux.c:
11064 Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos
11065 packet. Should conform to what we currently think is the
11066 final Ogg/Dirac muxing spec.
11068 2008-04-22 06:13:43 +0000 David Schleef <ds@schleef.org>
11070 sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display. Dark g...
11071 Original commit message from CVS:
11072 * sys/xvimage/xvimagesink.c:
11073 Fix typo that causes the overlay keying color to bright green
11074 on a 16-bit display. Dark grey good. Bright green bad.
11076 2008-04-21 13:47:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11078 ext/gnomevfs/gstgnomevfsuri.c: Add FIXME comment about using uri-list for source and sink.
11079 Original commit message from CVS:
11080 * ext/gnomevfs/gstgnomevfsuri.c:
11081 Add FIXME comment about using uri-list for source and sink.
11083 2008-04-20 11:42:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11085 ext/ogg/gstogmparse.c: GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to vaargs functions to gin...
11086 Original commit message from CVS:
11087 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
11088 GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to
11089 vaargs functions to gint. Otherwise the fractions will get 0 set
11090 instead of the correct value on big endian systems. Fixes bug #529018.
11092 2008-04-20 10:17:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11094 ext/gnomevfs/: Get the list of supported URI schemes in a threadsafe way and use the same list for the source and sink.
11095 Original commit message from CVS:
11096 * ext/gnomevfs/gstgnomevfssink.c:
11097 (gst_gnome_vfs_sink_uri_get_protocols):
11098 * ext/gnomevfs/gstgnomevfssrc.c:
11099 (gst_gnome_vfs_src_uri_get_protocols):
11100 * ext/gnomevfs/gstgnomevfsuri.c: (_internal_get_supported_uris),
11101 (gst_gnomevfs_get_supported_uris):
11102 Get the list of supported URI schemes in a threadsafe way and use the
11103 same list for the source and sink.
11105 2008-04-20 10:11:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11107 ext/gio/gstgio.c: Don't generate a new supported protocols list on each call but cache it. It's supposed to be static...
11108 Original commit message from CVS:
11109 * ext/gio/gstgio.c: (_internal_get_supported_protocols),
11110 (gst_gio_get_supported_protocols):
11111 Don't generate a new supported protocols list on each call but cache
11112 it. It's supposed to be static anyway, this way we only leak it once
11114 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
11115 (gst_gio_sink_class_init), (gst_gio_sink_finalize),
11116 (gst_gio_sink_set_property), (gst_gio_sink_get_property),
11117 (gst_gio_sink_start):
11118 * ext/gio/gstgiosink.h:
11119 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
11120 (gst_gio_src_class_init), (gst_gio_src_finalize),
11121 (gst_gio_src_set_property), (gst_gio_src_get_property),
11122 (gst_gio_src_start):
11123 * ext/gio/gstgiosrc.h:
11124 API: Add "file" properties where one can set a GFile as source/destination.
11125 Add locking to the properties and use gst_element_class_set_details_simple()
11126 instead of a static GstElementDetails struct.
11128 2008-04-19 20:06:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11130 gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
11131 Original commit message from CVS:
11132 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
11134 Add "mpp" and "mp+" as possible extensions for MusePack files.
11135 Add typefinding for MusePack StreamVersion 8 files and include the
11136 stream version in the caps.
11138 2008-04-19 16:33:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11140 gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
11141 Original commit message from CVS:
11142 * gst-libs/gst/rtp/gstrtppayloads.c:
11143 (gst_rtp_payload_info_for_name):
11144 Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
11146 2008-04-18 17:10:43 +0000 Tim-Philipp Müller <tim@centricular.net>
11148 configure.ac: Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level (NB: this only affects compilation of some...
11149 Original commit message from CVS:
11151 Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level
11152 (NB: this only affects compilation of some of the examples).
11153 Remove some configure.ac cruft that's not needed any longer.
11155 2008-04-18 14:54:01 +0000 Edward Hervey <bilboed@bilboed.com>
11157 gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
11158 Original commit message from CVS:
11159 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
11160 Don't validate the payload if there isn't any.
11163 2008-04-17 07:33:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11165 gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
11166 Original commit message from CVS:
11167 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
11168 Use g_atomic_int_set() instead of gst_atomic_int_set().
11170 2008-04-17 07:29:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11172 ext/gio/gstgio.c: Return NULL instead of a gchar * array with one NULL element if we don't get any supported URI sche...
11173 Original commit message from CVS:
11174 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11175 Return NULL instead of a gchar * array with one NULL element if we
11176 don't get any supported URI schemes from GIO.
11178 2008-04-15 19:06:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11180 gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
11181 Original commit message from CVS:
11182 * gst/audiotestsrc/gstaudiotestsrc.c:
11183 Remove cpp style commented old code.
11185 2008-04-15 19:02:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11187 gst/playback/gstdecodebin2.c: Fix signal docs.
11188 Original commit message from CVS:
11189 * gst/playback/gstdecodebin2.c:
11192 2008-04-14 17:58:19 +0000 Tim-Philipp Müller <tim@centricular.net>
11194 ext/pango/gsttextoverlay.c: Fix textoverlay unit test again by making the supposed default value for the wait-text pr...
11195 Original commit message from CVS:
11196 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
11197 (gst_text_overlay_init):
11198 Fix textoverlay unit test again by making the supposed default
11199 value for the wait-text property the actual default value.
11200 Also fix Since: tag for new property.
11202 2008-04-11 17:13:52 +0000 Tim-Philipp Müller <tim@centricular.net>
11204 gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
11205 Original commit message from CVS:
11206 * gst-libs/gst/video/video.c: (gst_video_format_new_caps),
11207 (gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
11208 (gst_video_format_get_pixel_stride),
11209 (gst_video_format_get_component_width),
11210 (gst_video_format_get_component_height),
11211 (gst_video_format_get_component_offset), (gst_video_format_get_size),
11212 (gst_video_format_convert):
11213 Add guards to these functions to ensure sane input values.
11214 * tests/check/libs/video.c:
11215 Fix unit test not to create caps with width=0 and height=0.
11217 2008-04-11 01:25:01 +0000 Wim Taymans <wim.taymans@gmail.com>
11219 docs/design/draft-keyframe-force.txt: Fix typo.
11220 Original commit message from CVS:
11221 * docs/design/draft-keyframe-force.txt:
11223 * gst/playback/gstqueue2.c: (update_buffering),
11224 (gst_queue_handle_src_query):
11225 Set buffering mode in the messages.
11226 Set buffering percent in the query.
11227 * tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
11228 (do_stream_buffering), (do_download_buffering), (msg_buffering):
11229 Do some more fancy things based on the buffering method in use.
11231 2008-04-09 21:42:24 +0000 Wim Taymans <wim.taymans@gmail.com>
11233 tests/examples/seek/seek.c: Add basic download reports to seek using the new buffering API.
11234 Original commit message from CVS:
11235 * tests/examples/seek/seek.c: (update_fill), (set_update_fill),
11236 (play_cb), (pause_cb), (stop_cb), (msg_state_changed),
11237 (msg_buffering), (main):
11238 Add basic download reports to seek using the new buffering API.
11240 2008-04-09 21:40:17 +0000 Wim Taymans <wim.taymans@gmail.com>
11242 gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
11243 Original commit message from CVS:
11244 * gst/playback/gstqueue2.c: (update_buffering),
11245 (gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
11246 (gst_queue_src_checkgetrange_function):
11247 Include extra buffering stats in the buffering message.
11248 Implement BUFFERING query.
11249 * gst/playback/gsturidecodebin.c: (do_async_start),
11250 (do_async_done), (type_found), (setup_streaming), (setup_source),
11251 (gst_uri_decode_bin_change_state):
11252 Only add decodebin2 when the type is found in streaming mode.
11253 Make uridecodebin async to PAUSED even when we don't have decodebin2
11256 2008-04-09 08:38:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11258 ext/gio/gstgio.c: Filter cdda from the supported URI schemes. We can't support musicbrainz tags and everything else o...
11259 Original commit message from CVS:
11260 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11261 Filter cdda from the supported URI schemes. We can't support
11262 musicbrainz tags and everything else one expects from a cdda source
11263 with GIO. Fixes bug #526794.
11265 2008-04-07 22:37:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11267 * sys/xvimage/xvimagesink.c:
11268 Fix calculation of 'expected size' for YV12 buffers.
11269 Original commit message from CVS:
11270 2008-04-07 Jan Schmidt <jan.schmidt@sun.com>
11271 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
11272 (gst_xvimagesink_buffer_alloc):
11273 Fix calculation of 'expected size' for YV12 buffers.
11274 Be a little more verbose in the debug output for buffer-alloc'ed
11275 buffers which turn out to have the wrong size.
11277 2008-04-07 22:26:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11280 Fix calculation of 'expected size' for YV12 buffers.
11281 Original commit message from CVS:
11282 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
11283 (gst_xvimagesink_buffer_alloc):
11284 Fix calculation of 'expected size' for YV12 buffers.
11285 Be a little more verbose in the debug output for buffer-alloc'ed
11286 buffers which turn out to have the wrong size.
11288 2008-04-07 10:50:11 +0000 Tim-Philipp Müller <tim@centricular.net>
11290 Merge other changes from 0.10.19 release branch.
11291 Original commit message from CVS:
11294 * gst-plugins-base.doap:
11295 Merge other changes from 0.10.19 release branch.
11297 2008-04-06 20:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
11299 gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
11300 Original commit message from CVS:
11301 * gst-libs/gst/audio/gstbaseaudiosink.c:
11302 (gst_base_audio_sink_class_init):
11303 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11304 (gst_base_audio_src_class_init):
11305 * gst/playback/gstplayback.c: (plugin_init):
11306 * gst/volume/gstvolume.c: (plugin_init):
11307 Work around missing bits of thread-safety on older GLibs some
11308 more to avoid assertions when starting up multiple playbin
11309 objects concurrently (see #512382).
11311 2008-04-06 17:19:39 +0000 Tim-Philipp Müller <tim@centricular.net>
11313 gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
11314 Original commit message from CVS:
11315 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
11316 Remove some more fields.
11318 2008-04-06 08:56:07 +0000 Damien Lespiau <damien.lespiau@gmail.com>
11320 configure.ac: Actually build dlls when cross-compiling with mingw32.
11321 Original commit message from CVS:
11322 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
11324 Actually build dlls when cross-compiling with mingw32.
11327 2008-04-03 23:01:11 +0000 Tim-Philipp Müller <tim@centricular.net>
11329 configure.ac: Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
11330 Original commit message from CVS:
11332 Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
11334 2008-04-03 16:10:53 +0000 Wim Taymans <wim.taymans@gmail.com>
11336 tests/examples/seek/seek.c: Add statusbar.
11337 Original commit message from CVS:
11338 * tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
11339 (msg_buffering), (connect_bus_signals), (main):
11341 Add buffering support with feedback in the statusbar.
11343 2008-04-03 15:58:37 +0000 Tim-Philipp Müller <tim@centricular.net>
11345 ext/ogg/gstoggmux.c: Fix sample pipeline description.
11346 Original commit message from CVS:
11347 * ext/ogg/gstoggmux.c:
11348 Fix sample pipeline description.
11350 2008-04-03 14:58:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11352 docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
11353 Original commit message from CVS:
11354 * docs/plugins/Makefile.am:
11355 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
11356 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
11357 * docs/plugins/gst-plugins-base-plugins-sections.txt:
11358 Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
11359 * docs/plugins/gst-plugins-base-plugins.args:
11360 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11361 * docs/plugins/gst-plugins-base-plugins.interfaces:
11362 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11363 * docs/plugins/inspect/plugin-adder.xml:
11364 * docs/plugins/inspect/plugin-alsa.xml:
11365 * docs/plugins/inspect/plugin-audioconvert.xml:
11366 * docs/plugins/inspect/plugin-audiorate.xml:
11367 * docs/plugins/inspect/plugin-audioresample.xml:
11368 * docs/plugins/inspect/plugin-audiotestsrc.xml:
11369 * docs/plugins/inspect/plugin-cdparanoia.xml:
11370 * docs/plugins/inspect/plugin-decodebin.xml:
11371 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
11372 * docs/plugins/inspect/plugin-gdp.xml:
11373 * docs/plugins/inspect/plugin-gnomevfs.xml:
11374 * docs/plugins/inspect/plugin-libvisual.xml:
11375 * docs/plugins/inspect/plugin-ogg.xml:
11376 * docs/plugins/inspect/plugin-pango.xml:
11377 * docs/plugins/inspect/plugin-playback.xml:
11378 * docs/plugins/inspect/plugin-queue2.xml:
11379 * docs/plugins/inspect/plugin-subparse.xml:
11380 * docs/plugins/inspect/plugin-tcp.xml:
11381 * docs/plugins/inspect/plugin-theora.xml:
11382 * docs/plugins/inspect/plugin-typefindfunctions.xml:
11383 * docs/plugins/inspect/plugin-uridecodebin.xml:
11384 * docs/plugins/inspect/plugin-video4linux.xml:
11385 * docs/plugins/inspect/plugin-videorate.xml:
11386 * docs/plugins/inspect/plugin-videoscale.xml:
11387 * docs/plugins/inspect/plugin-videotestsrc.xml:
11388 * docs/plugins/inspect/plugin-volume.xml:
11389 * docs/plugins/inspect/plugin-vorbis.xml:
11390 * docs/plugins/inspect/plugin-ximagesink.xml:
11391 * docs/plugins/inspect/plugin-xvimagesink.xml:
11392 Update introspection data.
11393 * ext/ogg/gstoggmux.c:
11395 * gst/playback/gstdecodebin2.c:
11396 Don't use gtk-doc style comment start for private stuff, but make it
11397 formatted like this for consistency.
11399 2008-04-03 12:16:04 +0000 Wim Taymans <wim.taymans@gmail.com>
11401 gst/playback/gstdecodebin2.c: Remove fakesink hack, we can now implement this more elegantly.
11402 Original commit message from CVS:
11403 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
11404 (gst_decode_bin_init), (gst_decode_bin_dispose),
11405 (gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
11406 (gst_decode_bin_set_property), (gst_decode_bin_get_property),
11407 (analyze_new_pad), (connect_pad), (expose_pad),
11408 (gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
11409 (gst_decode_group_expose), (gst_decode_group_free),
11410 (do_async_start), (do_async_done), (gst_decode_bin_change_state):
11411 Remove fakesink hack, we can now implement this more elegantly.
11412 Added property to bypass typefinding.
11413 Removed underrun callback and demuxer pad probe, we now use the srcpad
11414 probe to expose groups.
11415 API::sink-caps property
11416 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
11417 Guard against multiple emissions of the no_more_pads signal, which
11418 happens when we are dealing with chained oggs.
11419 * gst/playback/gsturidecodebin.c: (remove_decoders),
11420 (make_decoder), (type_found), (setup_streaming), (source_new_pad),
11422 For streams, use our own typefind element and plug our queue after it.
11423 We will need this to determine the type of buffering to use for the
11426 2008-04-03 10:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
11428 gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
11429 Original commit message from CVS:
11430 * gst-libs/gst/audio/gstbaseaudiosink.c:
11431 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
11432 Guard against over and underflows because of clock slaving.
11433 When we are using our own clock, still compensate for any calibrations
11434 that we might have done to our clock.
11436 2008-04-03 10:22:33 +0000 Wim Taymans <wim.taymans@gmail.com>
11438 ext/theora/theoradec.c: Don't try to do anything fancy with the return code from pushing an event, it does not have e...
11439 Original commit message from CVS:
11440 * ext/theora/theoradec.c: (theora_handle_type_packet),
11441 (theora_dec_chain):
11442 Don't try to do anything fancy with the return code from pushing an
11443 event, it does not have enough information to turn it into a
11446 2008-04-03 10:19:43 +0000 Wim Taymans <wim.taymans@gmail.com>
11448 ext/ogg/gstoggdemux.c: Add small debug line.
11449 Original commit message from CVS:
11450 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_reset),
11451 (gst_ogg_demux_chain_elem_pad):
11452 Add small debug line.
11453 Pass return code from the internal decoder instead of the too generic
11456 2008-04-03 06:39:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11458 gst-libs/gst/cdda/: Use GLib's base64 implementation instead of our own.
11459 Original commit message from CVS:
11460 * gst-libs/gst/cdda/Makefile.am:
11461 * gst-libs/gst/cdda/base64.c:
11462 * gst-libs/gst/cdda/base64.h:
11463 * gst-libs/gst/cdda/gstcddabasesrc.c:
11464 (gst_cddabasesrc_calculate_musicbrainz_discid):
11465 Use GLib's base64 implementation instead of our own.
11467 2008-04-02 15:41:50 +0000 Wim Taymans <wim.taymans@gmail.com>
11469 ext/ogg/gstoggdemux.c: Refix oggdemux, we only have a problem if we failed to find a chain and we are not EOF.
11470 Original commit message from CVS:
11471 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
11472 (gst_ogg_demux_read_chain):
11473 Refix oggdemux, we only have a problem if we failed to find a chain and
11476 2008-04-02 15:07:01 +0000 Victor STINNER <victor.stinner@haypocalc.com>
11478 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...
11479 Original commit message from CVS:
11480 Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
11481 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
11482 (gst_ogg_demux_read_chain):
11483 When we fail to find a BOS page and we and up with no chain, error out
11484 properly instead of segfaulting. Fixes #525665.
11486 2008-04-02 14:58:05 +0000 Wim Taymans <wim.taymans@gmail.com>
11488 ext/ogg/gstoggdemux.c: The new-pad-group sequence is add-pads, no-more-pads, add-pads, no-more-pads...
11489 Original commit message from CVS:
11490 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
11491 (gst_ogg_demux_read_chain), (gst_ogg_demux_handle_page):
11492 The new-pad-group sequence is add-pads, no-more-pads, add-pads,
11495 2008-04-02 11:08:05 +0000 Wim Taymans <wim.taymans@gmail.com>
11497 gst/playback/gstqueue2.c: Update the estimated input data when we push out a buffer.
11498 Original commit message from CVS:
11499 * gst/playback/gstqueue2.c: (update_out_rates),
11500 (gst_queue_open_temp_location_file),
11501 (gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
11502 (gst_queue_handle_src_query), (gst_queue_set_property):
11503 Update the estimated input data when we push out a buffer.
11504 Add some debug info about the temp file.
11505 Only forward src events when we are not using a temp file.
11506 Don't block the duration query, we need to find something better.
11507 Don't leak the temp filename.
11509 2008-04-01 14:01:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11511 configure.ac: Require GLib 2.12 and liboil 0.3.14.
11512 Original commit message from CVS:
11514 Require GLib 2.12 and liboil 0.3.14.
11515 * gst/volume/gstvolume.c: (volume_process_double):
11516 Unconditionally use liboil 0.3.14 function.
11518 2008-03-31 16:08:45 +0000 Wim Taymans <wim.taymans@gmail.com>
11520 gst-libs/gst/riff/riff-media.c: ms-gsm can have arbitrarty sample rates. See #481354.
11521 Original commit message from CVS:
11522 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
11523 ms-gsm can have arbitrarty sample rates. See #481354.
11525 2008-03-28 16:22:35 +0000 Wim Taymans <wim.taymans@gmail.com>
11527 gst-libs/gst/riff/riff-media.c: MP4S is generic MPEG-4, not a microsoft variant.
11528 Original commit message from CVS:
11529 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
11530 MP4S is generic MPEG-4, not a microsoft variant.
11532 2008-03-27 15:26:38 +0000 Michael Smith <msmith@xiph.org>
11534 gst/gdp/gstgdpdepay.c: Check the body CRC (if set) when depayloading.
11535 Original commit message from CVS:
11536 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
11537 Check the body CRC (if set) when depayloading.
11540 2008-03-24 17:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
11542 ext/pango/gsttextoverlay.c: Fix Since: version for new property.
11543 Original commit message from CVS:
11544 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
11545 Fix Since: version for new property.
11547 2008-03-24 16:40:08 +0000 Wim Taymans <wim.taymans@gmail.com>
11549 gst-libs/gst/rtsp/gstrtspconnection.c: Don't error when poll_wait returns EAGAIN.
11550 Original commit message from CVS:
11551 * gst-libs/gst/rtsp/gstrtspconnection.c:
11552 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
11553 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
11554 Don't error when poll_wait returns EAGAIN.
11556 2008-03-24 14:08:22 +0000 Wim Taymans <wim.taymans@gmail.com>
11558 gst/playback/gstqueue2.c: The queue is never filled when there are no buffers in the queue at all.
11559 Original commit message from CVS:
11560 * gst/playback/gstqueue2.c: (gst_queue_is_filled):
11561 The queue is never filled when there are no buffers in the queue at all.
11564 2008-03-24 12:26:30 +0000 Wim Taymans <wim.taymans@gmail.com>
11566 gst/playback/gstplaybin2.c: Update some docs.
11567 Original commit message from CVS:
11568 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
11569 (init_group), (free_group), (gst_play_bin_init),
11570 (gst_play_bin_finalize), (gst_play_bin_set_uri),
11571 (gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
11572 (gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
11573 (gst_play_bin_set_current_video_stream),
11574 (gst_play_bin_set_current_audio_stream),
11575 (gst_play_bin_set_current_text_stream),
11576 (gst_play_bin_set_encoding), (gst_play_bin_set_property),
11577 (gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
11578 (no_more_pads_cb), (perform_eos), (autoplug_select_cb),
11579 (activate_group), (deactivate_group), (setup_next_source),
11580 (save_current_group), (gst_play_bin_change_state):
11582 Add new locks and conds to protect pipeline creation and group
11584 Implement the sub-uri property.
11585 Keep track of pending uridecodebin creation and configure the output
11586 pipeline after all streams are configured.
11587 Propagate subtitle encoding to the uridecodebins.
11588 Implement getting the video/audio/visualisation elements.
11589 Use input-selector for stream switching.
11590 If we are asked to do visualisation, prefer to autoplug raw sinks
11591 instead of sinks that accept encoded data.
11593 2008-03-24 12:15:26 +0000 Wim Taymans <wim.taymans@gmail.com>
11595 gst/playback/gstplaysink.*: Add methods to get audio/video/vis elements.
11596 Original commit message from CVS:
11597 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
11598 (gst_play_sink_init), (gst_play_sink_dispose),
11599 (gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
11600 (gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
11601 (gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
11602 (gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
11603 (gst_play_sink_set_volume), (gst_play_sink_get_volume),
11604 (gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
11605 (gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
11606 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
11607 (gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
11608 * gst/playback/gstplaysink.h:
11609 Add methods to get audio/video/vis elements.
11610 Add methods to set the font description for the overlay.
11611 Remove properties, we're using this element with its methods only.
11612 Add support for subtitles.
11613 Rearrange the locking a bit to not use the object lock for protecting
11614 the pipeline construction.
11615 Try to use the volume and mute property on the sink when its available.
11616 Implement the mute option with volume when the sink does not have a mute
11618 Only add volume element when the sink has no volume property.
11619 Only do visualisations with raw audio pads.
11621 2008-03-24 12:03:02 +0000 Wim Taymans <wim.taymans@gmail.com>
11623 ext/pango/gsttextoverlay.*: Add property to configure waiting for text on the textpad or not, with the default behavi...
11624 Original commit message from CVS:
11625 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
11626 (gst_text_overlay_init), (gst_text_overlay_set_property),
11627 (gst_text_overlay_get_property), (gst_text_overlay_src_event),
11628 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
11629 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
11630 (gst_text_overlay_change_state):
11631 * ext/pango/gsttextoverlay.h:
11632 Add property to configure waiting for text on the textpad or not, with
11633 the default behaviour being the old one (always wait for text before
11634 rendering the video). This default behaviour is usually not the best one
11635 because the text stream can very sparse and could require queueing a lot
11637 Fix the flushing and EOS handing so that we don't mix up their meaning.
11639 2008-03-24 11:54:02 +0000 Wim Taymans <wim.taymans@gmail.com>
11641 gst/playback/gsturidecodebin.c: Add a readonly source property and notify.
11642 Original commit message from CVS:
11643 * gst/playback/gsturidecodebin.c:
11644 (gst_uri_decode_bin_autoplug_factories),
11645 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
11646 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
11647 (gst_uri_decode_bin_set_property),
11648 (gst_uri_decode_bin_get_property), (no_more_pads_full),
11649 (new_decoded_pad_cb), (gen_source_element), (remove_decoders),
11650 (proxy_autoplug_factories_signal), (make_decoder),
11651 (source_new_pad), (setup_source):
11652 Add a readonly source property and notify.
11653 Add new lock for protecting the construction of the pipeline.
11654 Keep track of the decodebins we plugged.
11655 Correctly proxy the autoplug signal so that it actually continues.
11656 Proxy subtitle-encoding to the decodebins.
11658 2008-03-24 11:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
11660 tests/examples/seek/seek.c: Rearrange some buttons in playbin2 and make some other boxes insensitive when needed.
11661 Original commit message from CVS:
11662 * tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
11663 (text_toggle_cb), (update_streams), (main):
11664 Rearrange some buttons in playbin2 and make some other boxes insensitive
11666 Add language codes to subtitle selection boxes when we gind the right
11667 tags for the streams.
11669 2008-03-24 11:36:08 +0000 Wim Taymans <wim.taymans@gmail.com>
11671 gst/playback/gstdecodebin2.c: Protect caps property with the object lock.
11672 Original commit message from CVS:
11673 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
11674 (gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
11675 (gst_decode_bin_set_subs_encoding),
11676 (gst_decode_bin_get_subs_encoding),
11677 (gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
11678 (deactivate_free_recursive):
11679 Protect caps property with the object lock.
11680 Protect encoding property with the object lock.
11681 Keep list of elements we added that have the subtitle-encoding property.
11682 Distribute the subtitle-encoding to all of the elements when it
11685 2008-03-24 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
11687 gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
11688 Original commit message from CVS:
11689 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
11690 Small debug improvement.
11691 * gst-libs/gst/audio/gstbaseaudiosink.c:
11692 (gst_base_audio_sink_render):
11693 Fix bug in determining the sample start/stop position, we want to base
11694 this decision on the fact that we are going forwards or backwards, not
11695 slower or faster. This fixes some ugly resync warnings when playing at
11698 2008-03-23 13:41:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11700 ext/gio/gstgio.c: Correctly set the supported URI schemes and don't leave some schemes in the middle or at the start ...
11701 Original commit message from CVS:
11702 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11703 Correctly set the supported URI schemes and don't leave
11704 some schemes in the middle or at the start at NULL.
11706 2008-03-23 13:12:41 +0000 Tim-Philipp Müller <tim@centricular.net>
11708 tests/check/elements/gdpdepay.c: Make test compile without unused function/variable warnings on PPC.
11709 Original commit message from CVS:
11710 * tests/check/elements/gdpdepay.c:
11711 Make test compile without unused function/variable warnings on PPC.
11713 2008-03-22 15:00:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11715 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
11716 Original commit message from CVS:
11718 * ext/alsa/gstalsamixerelement.c:
11719 (gst_alsa_mixer_element_class_init):
11720 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
11721 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
11722 * ext/cdparanoia/gstcdparanoiasrc.c:
11723 (gst_cd_paranoia_src_class_init):
11724 * ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
11725 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
11726 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
11727 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
11728 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
11729 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
11730 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
11731 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
11732 * ext/pango/gsttextrender.c: (gst_text_render_class_init):
11733 * ext/theora/theoradec.c: (gst_theora_dec_class_init):
11734 * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
11735 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
11736 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
11737 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
11738 (gst_audio_filter_template_class_init):
11739 * gst-libs/gst/audio/gstbaseaudiosink.c:
11740 (gst_base_audio_sink_class_init):
11741 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11742 (gst_base_audio_src_class_init):
11743 * gst-libs/gst/cdda/gstcddabasesrc.c:
11744 (gst_cdda_base_src_class_init):
11745 * gst-libs/gst/interfaces/mixertrack.c:
11746 (gst_mixer_track_class_init):
11747 * gst-libs/gst/rtp/gstbasertpdepayload.c:
11748 (gst_base_rtp_depayload_class_init):
11749 * gst-libs/gst/rtp/gstbasertppayload.c:
11750 (gst_basertppayload_class_init):
11751 * gst/audioconvert/gstaudioconvert.c:
11752 (gst_audio_convert_class_init):
11753 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
11754 * gst/audioresample/gstaudioresample.c:
11755 (gst_audioresample_class_init):
11756 * gst/audiotestsrc/gstaudiotestsrc.c:
11757 (gst_audio_test_src_class_init):
11758 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
11759 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
11760 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
11761 (preroll_unlinked):
11762 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
11763 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
11764 * gst/playback/gstplaysink.c: (gst_play_sink_class_init):
11765 * gst/playback/gstqueue2.c: (gst_queue_class_init):
11766 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
11767 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
11768 (gst_stream_selector_class_init):
11769 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
11770 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
11771 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
11772 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
11773 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
11774 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
11775 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
11776 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
11777 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
11778 * gst/videotestsrc/gstvideotestsrc.c:
11779 (gst_video_test_src_class_init):
11780 * gst/volume/gstvolume.c: (gst_volume_class_init):
11781 * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
11782 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
11783 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
11784 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
11785 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
11786 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
11787 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
11788 static strings (i.e. all). This gives us less memory usage,
11789 fewer allocations and thus less memory defragmentation. Depend
11790 on core CVS for this. Fixes bug #523806.
11792 2008-03-22 14:13:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11794 ext/gio/gstgio.c: Filter http and https protocols. GIO/GVfs handles them but it's impossible to implement iradio/icec...
11795 Original commit message from CVS:
11796 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
11797 Filter http and https protocols. GIO/GVfs handles them but it's
11798 impossible to implement iradio/icecast with it. Better use
11799 souphttpsrc or something else for this.
11800 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
11801 If getting the file informations by a query fails try it with the
11802 seek-to-end trick too.
11804 2008-03-21 16:46:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11806 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...
11807 Original commit message from CVS:
11808 * gst/volume/gstvolume.c: (gst_volume_interface_supported),
11809 (gst_volume_base_init), (gst_volume_class_init),
11810 (volume_process_double), (volume_process_float),
11811 (volume_transform_ip), (plugin_init):
11812 memset buffers to zero if we get a GAP buffer. We usually see a
11813 buffer as one unit so let's handle it as one and don't care about
11814 volume changes while processing one buffer.
11815 Also clean up some stuff a bit.
11817 2008-03-21 15:58:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11819 gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
11820 Original commit message from CVS:
11821 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
11822 (gst_audio_convert_create_silence_buffer),
11823 (gst_audio_convert_transform):
11824 Make audioconvert GAP-aware by outputting silence buffers when the
11825 input has the GAP flag set. This is up to 8x faster.
11826 Based on a patch by Stefan Kost. Fixes bug #517813.
11828 2008-03-21 15:54:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11830 gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
11831 Original commit message from CVS:
11832 * gst/volume/gstvolume.c: (volume_process_double):
11833 Use oil_scalarmultiply_f64_ns() for double processing when it's
11834 available at compile time.
11836 2008-03-21 13:27:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11838 configure.ac: Fix lrint/lrintf checks to actually work. These functions are in libm on Linux at least so try to link ...
11839 Original commit message from CVS:
11841 Fix lrint/lrintf checks to actually work. These functions are
11842 in libm on Linux at least so try to link to it.
11844 2008-03-21 00:36:20 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11846 configure.ac: Back to development - 0.10.18.1
11847 Original commit message from CVS:
11849 Back to development - 0.10.18.1
11851 === release 0.10.18 ===
11853 2008-03-21 00:26:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11859 * docs/plugins/gst-plugins-base-plugins.args:
11860 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11861 * docs/plugins/gst-plugins-base-plugins.interfaces:
11862 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11863 * docs/plugins/gst-plugins-base-plugins.signals:
11864 * docs/plugins/inspect/plugin-adder.xml:
11865 * docs/plugins/inspect/plugin-alsa.xml:
11866 * docs/plugins/inspect/plugin-audioconvert.xml:
11867 * docs/plugins/inspect/plugin-audiorate.xml:
11868 * docs/plugins/inspect/plugin-audioresample.xml:
11869 * docs/plugins/inspect/plugin-audiotestsrc.xml:
11870 * docs/plugins/inspect/plugin-cdparanoia.xml:
11871 * docs/plugins/inspect/plugin-decodebin.xml:
11872 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
11873 * docs/plugins/inspect/plugin-gdp.xml:
11874 * docs/plugins/inspect/plugin-gnomevfs.xml:
11875 * docs/plugins/inspect/plugin-libvisual.xml:
11876 * docs/plugins/inspect/plugin-ogg.xml:
11877 * docs/plugins/inspect/plugin-pango.xml:
11878 * docs/plugins/inspect/plugin-playback.xml:
11879 * docs/plugins/inspect/plugin-queue2.xml:
11880 * docs/plugins/inspect/plugin-subparse.xml:
11881 * docs/plugins/inspect/plugin-tcp.xml:
11882 * docs/plugins/inspect/plugin-theora.xml:
11883 * docs/plugins/inspect/plugin-typefindfunctions.xml:
11884 * docs/plugins/inspect/plugin-uridecodebin.xml:
11885 * docs/plugins/inspect/plugin-video4linux.xml:
11886 * docs/plugins/inspect/plugin-videorate.xml:
11887 * docs/plugins/inspect/plugin-videoscale.xml:
11888 * docs/plugins/inspect/plugin-videotestsrc.xml:
11889 * docs/plugins/inspect/plugin-volume.xml:
11890 * docs/plugins/inspect/plugin-vorbis.xml:
11891 * docs/plugins/inspect/plugin-ximagesink.xml:
11892 * docs/plugins/inspect/plugin-xvimagesink.xml:
11893 * gst-plugins-base.doap:
11895 * win32/common/config.h:
11897 Original commit message from CVS:
11900 2008-03-21 00:16:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11927 Original commit message from CVS:
11930 2008-03-18 12:19:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11932 0.10.17.4 pre-release
11933 Original commit message from CVS:
11935 * win32/common/config.h:
11936 0.10.17.4 pre-release
11938 2008-03-18 11:20:05 +0000 Wim Taymans <wim.taymans@gmail.com>
11940 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL when trying to print strings that could be NULL because this might...
11941 Original commit message from CVS:
11942 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
11943 Use GST_STR_NULL when trying to print strings that could be NULL because
11944 this might crash on some platforms. See #520808.
11946 2008-03-18 11:10:12 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
11948 gst-libs/gst/rtsp/gstrtspconnection.c: Generic Windows fixes that makes libgstrtsp work on Windows when coupled with ...
11949 Original commit message from CVS:
11950 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
11951 * gst-libs/gst/rtsp/gstrtspconnection.c:
11952 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
11953 (read_line), (gst_rtsp_connection_read_internal):
11954 Generic Windows fixes that makes libgstrtsp work on Windows when
11955 coupled with the new GstPoll API. See #520808.
11957 2008-03-17 22:06:56 +0000 Milosz Derezynski <internalerror@gmail.com>
11959 ext/gio/gstgiobasesrc.c: If seeking to a new position succeeds don't simply return from create() without creating a b...
11960 Original commit message from CVS:
11961 Patch by: Milosz Derezynski <internalerror at gmail dot com>
11962 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
11963 If seeking to a new position succeeds don't simply return from
11964 create() without creating a buffer. Do this only in the case
11965 seeking to the new position fails. Fixes bug #523054.
11967 2008-03-17 10:32:28 +0000 Tim-Philipp Müller <tim@centricular.net>
11969 gst-libs/gst/video/video.c: Fix gst_video_format_parse_caps() for RGB caps with alpha channel (#522635).
11970 Original commit message from CVS:
11971 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
11972 (gst_video_format_from_rgba32_masks):
11973 Fix gst_video_format_parse_caps() for RGB caps with alpha channel
11975 * tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
11976 Add unit test for the RGB caps parsing and creation, checking for
11977 internal consistency of the new API and consistency of the API with
11978 the old GST_VIDEO_CAPS_* defines.
11980 2008-03-14 18:42:35 +0000 David Schleef <ds@schleef.org>
11982 gst/videotestsrc/videotestsrc.c: Oops, revert last change because -base is in freeze.
11983 Original commit message from CVS:
11984 * gst/videotestsrc/videotestsrc.c: Oops, revert last change
11985 because -base is in freeze.
11987 2008-03-14 17:33:09 +0000 William M. Brack <wbrack@mmm.hk>
11989 gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
11990 Original commit message from CVS:
11991 Patch by: William M. Brack
11992 * gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
11994 2008-03-14 09:54:44 +0000 Wim Taymans <wim.taymans@gmail.com>
11996 gst/playback/gststreamselector.*: Revert change that caused regression until a real fix is found.
11997 Original commit message from CVS:
11998 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
11999 (gst_selector_pad_chain):
12000 * gst/playback/gststreamselector.h:
12001 Revert change that caused regression until a real fix is found.
12004 2008-03-12 12:39:13 +0000 Michael Smith <msmith@xiph.org>
12006 gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
12007 Original commit message from CVS:
12008 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
12009 * gst-libs/gst/audio/gstringbuffer.h:
12010 Rename recently added buffer types to make more sense.
12011 * ext/alsa/gstalsasink.c: (alsasink_parse_spec),
12012 (gst_alsasink_write):
12013 Adapt for above API changes.
12016 2008-03-11 13:23:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12018 win32/common/libgstnetbuffer.def: Add new symbol gst_netaddress_equal. Fixes bug #521743.
12019 Original commit message from CVS:
12020 * win32/common/libgstnetbuffer.def:
12021 Add new symbol gst_netaddress_equal. Fixes bug #521743.
12023 2008-03-11 00:25:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12025 0.10.17.3 pre-release
12026 Original commit message from CVS:
12028 * win32/common/config.h:
12029 0.10.17.3 pre-release
12031 2008-03-10 17:19:56 +0000 Wim Taymans <wim.taymans@gmail.com>
12033 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
12034 Original commit message from CVS:
12035 * gst-libs/gst/audio/gstbaseaudiosrc.c:
12036 (gst_base_audio_src_create):
12037 Fix duration when no clock was provided. Fixes #520300.
12039 2008-03-07 18:17:44 +0000 Olivier Crete <tester@tester.ca>
12041 Add trivial function to compare GstNetAddress. See #520626.
12042 Original commit message from CVS:
12043 Patch by: Olivier Crete <tester at tester ca>
12044 * docs/libs/gst-plugins-base-libs-sections.txt:
12045 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
12046 * gst-libs/gst/netbuffer/gstnetbuffer.h:
12047 Add trivial function to compare GstNetAddress. See #520626.
12048 API: GstNetBuffer::gst_netaddress_equal
12050 2008-03-07 16:10:51 +0000 Wim Taymans <wim.taymans@gmail.com>
12052 gst/tcp/gstmultifdsink.c: Update mode property docs, it's deprecated now.
12053 Original commit message from CVS:
12054 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
12055 Update mode property docs, it's deprecated now.
12057 2008-03-07 15:48:51 +0000 Wim Taymans <wim.taymans@gmail.com>
12059 gst/: Remove GstPollMode from gstpoll constructor.
12060 Original commit message from CVS:
12061 * gst-libs/gst/rtsp/gstrtspconnection.c:
12062 (gst_rtsp_connection_create):
12063 * gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
12064 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
12065 * gst/tcp/gstmultifdsink.h:
12066 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
12067 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
12068 Remove GstPollMode from gstpoll constructor.
12070 2008-03-04 00:26:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12072 0.10.17.2 pre-release
12073 Original commit message from CVS:
12075 * win32/common/config.h:
12076 0.10.17.2 pre-release
12078 2008-03-03 23:59:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12080 gst/Makefile.am: GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean them twice
12081 Original commit message from CVS:
12083 GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
12085 * win32/common/libgstinterfaces.def:
12086 * win32/common/libgstrtp.def:
12087 Add new API to the defs
12089 2008-03-03 16:11:50 +0000 Mersad Jelacic <mersad@axis.com>
12091 gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
12092 Original commit message from CVS:
12093 Patch by: Mersad Jelacic <mersad at axis dot com>
12094 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
12095 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
12096 API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
12097 possible to specify the sample size in bits. (#509637)
12099 2008-03-03 13:59:19 +0000 Tim-Philipp Müller <tim@centricular.net>
12101 tests/check/libs/mixer.c: Add a few simple checks for the new message types.
12102 Original commit message from CVS:
12103 * tests/check/libs/mixer.c:
12104 Add a few simple checks for the new message types.
12106 2008-03-03 13:56:38 +0000 Tim-Philipp Müller <tim@centricular.net>
12108 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
12109 Original commit message from CVS:
12110 * docs/libs/gst-plugins-base-libs-sections.txt:
12111 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
12112 (gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
12113 (gst_mixer_message_get_type),
12114 (gst_mixer_message_parse_option_changed),
12115 (gst_mixer_message_parse_options_list_changed):
12116 * gst-libs/gst/interfaces/mixer.h: (GstMixerType),
12117 (GST_MIXER_MESSAGE_OPTION_CHANGED),
12118 (GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
12119 (GST_MIXER_MESSAGE_MIXER_CHANGED):
12120 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
12121 and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
12123 2008-03-03 13:50:18 +0000 Tim-Philipp Müller <tim@centricular.net>
12125 gst-libs/gst/interfaces/mixeroptions.*: API: add GstMixerOptions::get_values vfunc (#519906)
12126 Original commit message from CVS:
12127 * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
12128 (gst_mixer_options_get_values):
12129 * gst-libs/gst/interfaces/mixeroptions.h:
12130 (GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
12131 (_GstMixerOptions), (_GstMixerOptionsClass):
12132 API: add GstMixerOptions::get_values vfunc (#519906)
12134 2008-03-03 12:01:15 +0000 Peter Kjellerstedt <pkj@axis.com>
12136 configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#4...
12137 Original commit message from CVS:
12139 Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
12140 plug-ins are included/excluded. (#498222)
12142 2008-03-03 06:22:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12144 gst/typefind/gsttypefindfunctions.c: Add typefinder for IMelody files, using audio/x-imelody.
12145 Original commit message from CVS:
12146 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
12147 Add typefinder for IMelody files, using audio/x-imelody.
12150 2008-03-03 06:04:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12152 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
12153 Original commit message from CVS:
12154 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
12155 * ext/alsa/gstalsasink.c: (set_hwparams):
12156 * ext/alsa/gstalsasrc.c: (set_hwparams):
12157 * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
12158 * ext/ogg/gstoggmux.h:
12159 * ext/ogg/gstogmparse.c:
12160 * gst-libs/gst/audio/audio.c:
12161 * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
12162 * gst-libs/gst/pbutils/missing-plugins.c:
12163 (gst_missing_uri_sink_message_new),
12164 (gst_missing_element_message_new),
12165 (gst_missing_decoder_message_new),
12166 (gst_missing_encoder_message_new):
12167 * gst-libs/gst/rtp/gstbasertppayload.c:
12168 * gst-libs/gst/rtp/gstrtcpbuffer.c:
12169 (gst_rtcp_packet_bye_get_reason):
12170 * gst/audioconvert/gstaudioconvert.c:
12171 * gst/audioresample/gstaudioresample.c:
12172 * gst/ffmpegcolorspace/imgconvert.c:
12173 * gst/playback/test.c: (gen_video_element), (gen_audio_element):
12174 * gst/typefind/gsttypefindfunctions.c:
12175 * gst/videoscale/vs_4tap.c:
12176 * gst/videoscale/vs_4tap.h:
12177 * sys/v4l/gstv4lelement.c:
12178 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
12179 * sys/v4l/v4l_calls.c:
12180 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
12181 (gst_v4lsrc_try_capture):
12182 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
12183 (gst_ximagesink_ximage_new):
12184 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
12185 (gst_xvimagesink_xvimage_new):
12186 * tests/check/elements/audioconvert.c:
12187 * tests/check/elements/audioresample.c:
12188 (fail_unless_perfect_stream):
12189 * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
12190 * tests/check/elements/decodebin.c:
12191 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
12192 (setup_gdpdepay_streamheader):
12193 * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
12194 (setup_gdppay_streamheader):
12195 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
12196 * tests/check/elements/multifdsink.c: (setup_multifdsink):
12197 * tests/check/elements/textoverlay.c:
12198 * tests/check/elements/videorate.c: (setup_videorate):
12199 * tests/check/elements/videotestsrc.c: (setup_videotestsrc):
12200 * tests/check/elements/volume.c: (setup_volume):
12201 * tests/check/elements/vorbisdec.c: (setup_vorbisdec):
12202 * tests/check/elements/vorbistag.c:
12203 * tests/check/generic/clock-selection.c:
12204 * tests/check/generic/states.c: (setup), (teardown):
12205 * tests/check/libs/cddabasesrc.c:
12206 * tests/check/libs/video.c:
12207 * tests/check/pipelines/gio.c:
12208 * tests/check/pipelines/oggmux.c:
12209 * tests/check/pipelines/simple-launch-lines.c:
12210 (simple_launch_lines_suite):
12211 * tests/check/pipelines/streamheader.c:
12212 * tests/check/pipelines/theoraenc.c:
12213 * tests/check/pipelines/vorbisdec.c:
12214 * tests/check/pipelines/vorbisenc.c:
12215 * tests/examples/seek/scrubby.c:
12216 * tests/examples/seek/seek.c: (query_positions_elems),
12217 (query_positions_pads):
12218 * tests/icles/stress-xoverlay.c: (myclock):
12219 Correct all relevant warnings found by the sparse semantic code
12220 analyzer. This include marking several symbols static, using
12221 NULL instead of 0 for pointers and using "foo (void)" instead
12222 of "foo ()" for declarations.
12223 * win32/common/libgstrtp.def:
12224 Add gst_rtp_buffer_set_extension_data to the symbol definition file.
12226 2008-03-02 18:43:15 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
12228 gst/playback/gstplaybin2.c: Make the function signature of the _get_*_tags() functions match the signature of the vfu...
12229 Original commit message from CVS:
12230 Patch by: José Alburquerque <jaalburqu svn gnome org>
12231 * gst/playback/gstplaybin2.c:
12232 Make the function signature of the _get_*_tags() functions match
12233 the signature of the vfuncs they implement, ie. return a
12234 GstTagList rather than a GstStructure, which is more correct,
12235 even if one is typedef'ed to the other (#518940).
12237 2008-03-02 18:32:36 +0000 Tim-Philipp Müller <tim@centricular.net>
12239 gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
12240 Original commit message from CVS:
12241 * gst-libs/gst/rtsp/gstrtspconnection.c:
12242 Don't include unix headers unconditionally (fixes #518037).
12244 2008-03-02 18:24:37 +0000 Tim-Philipp Müller <tim@centricular.net>
12246 tests/check/libs/video.c: Add unit test that makes sure that the strides, offsets and sizes returned for the various ...
12247 Original commit message from CVS:
12248 * tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
12249 (fourcc_list_struct), (fourcc_list), (fourcc_get_size),
12250 (paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
12251 (paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
12252 (paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
12253 (paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
12254 (gst_video_format_is_packed), (video_format_is_packed):
12255 Add unit test that makes sure that the strides, offsets and
12256 sizes returned for the various YUV formats by the new video API
12257 match the old reference implementation in videotestsrc.
12259 2008-03-02 18:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
12261 gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
12262 Original commit message from CVS:
12263 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
12264 (gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
12265 (gst_video_format_is_rgb), (gst_video_format_is_yuv),
12266 (gst_video_format_has_alpha), (gst_video_format_get_row_stride),
12267 (gst_video_format_get_pixel_stride),
12268 (gst_video_format_get_component_width),
12269 (gst_video_format_get_component_height),
12270 (gst_video_format_get_component_offset), (gst_video_format_get_size):
12271 * gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
12272 (GST_VIDEO_FORMAT_Y42B):
12273 API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
12275 2008-03-02 18:07:10 +0000 Tim-Philipp Müller <tim@centricular.net>
12277 gst-libs/gst/video/video.c: YV12 is I420 with swapped components 1 and 2, so the offset of component 1 for I420 shoul...
12278 Original commit message from CVS:
12279 * gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
12280 YV12 is I420 with swapped components 1 and 2, so the offset of
12281 component 1 for I420 should be the offset for component 2 for YV12
12284 2008-02-29 21:48:00 +0000 Rene Stadler <mail@renestadler.de>
12286 sys/v4l/gstv4lelement.c: Add missing semicolon to fix indentation.
12287 Original commit message from CVS:
12288 * sys/v4l/gstv4lelement.c:
12289 Add missing semicolon to fix indentation.
12291 2008-02-29 18:44:36 +0000 Julien Moutte <julien@moutte.net>
12293 ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
12294 Original commit message from CVS:
12295 2008-02-29 Julien Moutte <julien@fluendo.com>
12296 * ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
12297 (gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
12299 if we can do SPDIF output.
12300 * ext/alsa/gstalsa.h:
12301 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
12302 (gst_alsasink_prepare), (gst_alsasink_close),
12303 (gst_alsasink_write):
12304 * ext/alsa/gstalsasink.h: Initial support for SPDIF.
12305 * gst-libs/gst/audio/gstringbuffer.c:
12306 (gst_ring_buffer_parse_caps):
12307 * gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
12309 to support AC3, EC3 and IEC958 buffers.
12311 2008-02-29 17:59:16 +0000 Tim-Philipp Müller <tim@centricular.net>
12313 gst-libs/gst/interfaces/mixer.c: De-cruft and fix message type assertions (NULL is not a really valid mixer message t...
12314 Original commit message from CVS:
12315 * gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
12316 (gst_mixer_message_parse_mute_toggled),
12317 (gst_mixer_message_parse_record_toggled),
12318 (gst_mixer_message_parse_volume_changed),
12319 (gst_mixer_message_parse_option_changed):
12320 De-cruft and fix message type assertions (NULL is not a really
12321 valid mixer message type string).
12323 2008-02-29 14:52:02 +0000 Wim Taymans <wim.taymans@gmail.com>
12325 ext/libvisual/visual.c: When negotiating, actually start from a format that we can support instead of from the too ge...
12326 Original commit message from CVS:
12327 * ext/libvisual/visual.c: (gst_vis_src_negotiate):
12328 When negotiating, actually start from a format that we can support
12329 instead of from the too generic template.
12331 2008-02-29 12:26:48 +0000 Wim Taymans <wim.taymans@gmail.com>
12333 gst/playback/gstplaybin2.c: Enable vis setting.
12334 Original commit message from CVS:
12335 * gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
12336 Enable vis setting.
12337 * gst/playback/gstplaysink.c: (gst_play_sink_init),
12338 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
12339 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
12341 Implement vis switching while playing.
12343 2008-02-29 00:04:57 +0000 David Schleef <ds@schleef.org>
12345 gst-libs/gst/riff/riff-media.c: Add Dirac mapping
12346 Original commit message from CVS:
12347 * gst-libs/gst/riff/riff-media.c: Add Dirac mapping
12349 2008-02-28 10:54:14 +0000 Peter Kjellerstedt <pkj@axis.com>
12351 gst/tcp/: Removed fdset and stress test, they are now known as GstPoll in core.
12352 Original commit message from CVS:
12353 Patch by: Peter Kjellerstedt <pkj at axis com>
12354 * gst/tcp/Makefile.am:
12355 * gst/tcp/fdsetstress.c:
12356 * gst/tcp/gstfdset.c:
12357 * gst/tcp/gstfdset.h:
12358 Removed fdset and stress test, they are now known as GstPoll in
12360 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
12361 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
12362 (gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
12363 (gst_multi_fd_sink_handle_client_write),
12364 (gst_multi_fd_sink_queue_buffer),
12365 (gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
12366 (gst_multi_fd_sink_stop):
12367 * gst/tcp/gstmultifdsink.h:
12368 * gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
12369 (gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
12370 (gst_tcp_gdp_read_caps):
12371 * gst/tcp/gsttcp.h:
12372 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
12373 (gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
12374 (gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
12375 * gst/tcp/gsttcpclientsink.h:
12376 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
12377 (gst_tcp_client_src_create), (gst_tcp_client_src_start),
12378 (gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
12379 * gst/tcp/gsttcpclientsrc.h:
12380 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
12381 (gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
12382 * gst/tcp/gsttcpserversink.h:
12383 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
12384 (gst_tcp_server_src_create), (gst_tcp_server_src_start),
12385 (gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
12386 * gst/tcp/gsttcpserversrc.h:
12387 Port to GstPoll. See #505417.
12389 2008-02-28 09:54:14 +0000 Wim Taymans <wim.taymans@gmail.com>
12392 Patch Changelog a bit to give credit and refer to the relevant bug.
12393 Original commit message from CVS:
12394 Patch Changelog a bit to give credit and refer to the
12397 2008-02-28 09:50:52 +0000 Wim Taymans <wim.taymans@gmail.com>
12399 gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection.
12400 Original commit message from CVS:
12401 * gst-libs/gst/rtsp/gstrtspconnection.c:
12402 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
12403 (gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
12404 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
12405 (gst_rtsp_connection_free), (gst_rtsp_connection_poll),
12406 (gst_rtsp_connection_flush):
12407 * gst-libs/gst/rtsp/gstrtspconnection.h:
12408 Use GstPoll for the rtsp connection.
12410 2008-02-27 12:19:31 +0000 Wim Taymans <wim.taymans@gmail.com>
12412 tests/examples/seek/seek.c: Add combo box for visualisations, populate it with a factory list of all visualisation pl...
12413 Original commit message from CVS:
12414 * tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
12415 (init_visualization_features), (vis_combo_cb), (shot_cb), (main):
12416 Add combo box for visualisations, populate it with a factory list
12417 of all visualisation plugins, configure vis plugin instance in
12420 2008-02-27 10:55:03 +0000 Wim Taymans <wim.taymans@gmail.com>
12422 tests/check/libs/rtp.c: Add check for RTP buffer defaults, padding and marker bit API.
12423 Original commit message from CVS:
12424 * tests/check/libs/rtp.c: (GST_START_TEST):
12425 Add check for RTP buffer defaults, padding and marker bit API.
12427 2008-02-27 10:42:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12429 gst-libs/gst/cdda/sha1.c: Use memcpy() instead of upcasting a byte array to long *. This fixes an unaligned memory ac...
12430 Original commit message from CVS:
12431 * gst-libs/gst/cdda/sha1.c: (sha_transform):
12432 Use memcpy() instead of upcasting a byte array to long *. This
12433 fixes an unaligned memory access, resulting in SIGBUS on IA64.
12434 This should be ported to GCheckSum once we can use GLib 2.16.
12435 Partially fixes bug #500833.
12437 2008-02-27 10:23:27 +0000 Tim-Philipp Müller <tim@centricular.net>
12439 gst-libs/gst/tag/gsttagdemux.c: Push tag event after the newsegment event. Log the pointer of the buffer we're actual...
12440 Original commit message from CVS:
12441 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
12442 Push tag event after the newsegment event. Log the pointer of
12443 the buffer we're actually going to push rather than the buffer
12444 we're feeding to _make_metadata_writable().
12446 2008-02-25 07:21:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12448 gst/typefind/gsttypefindfunctions.c: Comment smoke typefinder for now. The smokedec plugin needs one frame per buffer...
12449 Original commit message from CVS:
12450 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
12451 Comment smoke typefinder for now. The smokedec plugin needs one
12452 frame per buffer but we have no parser yet, thus it simply crashes
12453 in most situations.
12455 2008-02-25 06:48:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12457 gst/typefind/gsttypefindfunctions.c: Add typefinder for the smoke video codec. Copied from the jpeg plugin.
12458 Original commit message from CVS:
12459 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
12460 Add typefinder for the smoke video codec. Copied from the jpeg plugin.
12462 2008-02-25 06:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12464 gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
12465 Original commit message from CVS:
12466 * gst/typefind/gsttypefindfunctions.c: (mid_type_find),
12468 Add midi typefinder, copied from the timidity plugin.
12470 2008-02-23 09:51:26 +0000 Tomasz Sałaciński <tsalacinski@gmail.com>
12472 Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
12473 Original commit message from CVS:
12474 Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
12475 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
12476 * tests/check/elements/subparse.c: (test_microdvd_with_italics),
12478 Forward slashes at the beginning and end of a line also signify
12479 italics (Fixes: #518162).
12481 2008-02-22 06:38:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12483 tests/check/gst-plugins-base.supp: Add a suppression for a cached value in GIO that wasn't moved while moving gio fro...
12484 Original commit message from CVS:
12485 * tests/check/gst-plugins-base.supp:
12486 Add a suppression for a cached value in GIO that wasn't moved
12487 while moving gio from -bad to -base.
12489 2008-02-22 05:27:24 +0000 Brian Cameron <brian.cameron@sun.com>
12491 configure.ac: Don't hardcode -Wall and -Werror for configure checks, this fails with non-GCC compilers. Fixes bug #51...
12492 Original commit message from CVS:
12493 Patch by: Brian Cameron <brian dot cameron at sun dot com>
12495 Don't hardcode -Wall and -Werror for configure checks, this fails
12496 with non-GCC compilers. Fixes bug #517991.
12498 2008-02-21 08:05:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12500 gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
12501 Original commit message from CVS:
12502 * gst/audiotestsrc/gstaudiotestsrc.c:
12503 Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
12505 2008-02-20 15:37:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12507 ext/gnomevfs/gstgnomevfssink.c: Return FALSE when seeking for a new segment fails instead of silently ignoring the fa...
12508 Original commit message from CVS:
12509 * ext/gnomevfs/gstgnomevfssink.c:
12510 (gst_gnome_vfs_sink_handle_event):
12511 Return FALSE when seeking for a new segment fails instead
12512 of silently ignoring the failure and appending every buffer
12513 that comes for the new segment.
12515 2008-02-20 11:52:28 +0000 Wim Taymans <wim.taymans@gmail.com>
12517 gst/playback/gstplaysink.c: Recursively search the sink element for a last-frame property so that we can also find th...
12518 Original commit message from CVS:
12519 * gst/playback/gstplaysink.c: (find_property),
12520 (gst_play_sink_find_property), (gen_video_chain),
12521 (gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
12522 Recursively search the sink element for a last-frame property so that we
12523 can also find the property in autovideosink and friends that don't
12524 always proxy the internal sink properties.
12526 2008-02-19 20:42:09 +0000 Tim-Philipp Müller <tim@centricular.net>
12528 gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
12529 Original commit message from CVS:
12530 * gst-libs/gst/audio/multichannel.c:
12531 (GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
12532 (gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
12533 (gst_audio_set_structure_channel_positions_list),
12534 (add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
12535 (gst_audio_fixate_channel_positions):
12536 Fix confusing terminology in docs and code: structure fields are
12537 'fields' and not 'properties'.
12539 2008-02-19 20:36:58 +0000 Tim-Philipp Müller <tim@centricular.net>
12541 gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
12542 Original commit message from CVS:
12543 * gst-libs/gst/audio/multichannel.c:
12544 (gst_audio_check_channel_positions), (add_list_to_struct):
12545 Give more useful warning messages if one of the channel
12546 layout enums passed to us is invalid and if the "channels"
12547 field in the caps has a GType we don't expect.
12549 2008-02-19 20:22:09 +0000 Tim-Philipp Müller <tim@centricular.net>
12551 gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
12552 Original commit message from CVS:
12553 * gst-libs/gst/audio/multichannel.c:
12554 Fix typo in docs blurb.
12556 2008-02-19 16:16:55 +0000 Josep Torra Valles <josep@fluendo.com>
12558 gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS typefind lookup to fix typefinding on HD clips.
12559 Original commit message from CVS:
12560 2008-02-19 Julien Moutte <julien@fluendo.com>
12561 Patch by: Josep Torra Valles <josep@fluendo.com>
12562 * gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
12563 typefind lookup to fix typefinding on HD clips.
12565 2008-02-19 15:50:37 +0000 Tim-Philipp Müller <tim@centricular.net>
12567 gst/playback/gstscreenshot.*: Fix up copyright (I rewrote the GStreamer-0.10 code for this from scratch back in the d...
12568 Original commit message from CVS:
12569 * gst/playback/gstscreenshot.c:
12570 * gst/playback/gstscreenshot.h:
12571 Fix up copyright (I rewrote the GStreamer-0.10 code for
12572 this from scratch back in the days).
12574 2008-02-19 15:02:33 +0000 Wim Taymans <wim.taymans@gmail.com>
12576 gst/playback/: Add screenshot conversion code from totem.
12577 Original commit message from CVS:
12578 * gst/playback/Makefile.am:
12579 * gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
12580 (create_element), (gst_play_frame_conv_convert):
12581 * gst/playback/gstscreenshot.h:
12582 Add screenshot conversion code from totem.
12583 * gst/playback/gstplay-marshal.list:
12584 * gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
12585 (gst_play_bin_class_init), (gst_play_bin_convert_frame),
12586 (gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
12587 Implement frame property to get a color-unconverted snapshot.
12588 Implement convert-frame action signal to get a converted snapshot image.
12589 Configure connection speed in uridecodebin.
12590 Document some more properties.
12591 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
12592 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
12593 (gst_play_sink_get_last_frame):
12594 * gst/playback/gstplaysink.h:
12595 Use last-buffer property of the video sink to get a video snapshot.
12596 * tests/examples/seek/seek.c: (shot_cb), (main):
12597 Add snapshot button for playbin2 and use the frame property to save the
12598 frame as a png in the current directory.
12600 2008-02-19 11:45:56 +0000 Josep Torra Valles <josep@fluendo.com>
12602 gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
12603 Original commit message from CVS:
12604 Patch by: Josep Torra Valles <josep at fluendo dot com>
12605 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
12607 Add typefinding support for h264 elementary streams.
12610 2008-02-18 13:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12612 configure.ac: Require CVS of core for new API in collectpads.
12613 Original commit message from CVS:
12615 Require CVS of core for new API in collectpads.
12616 * gst/adder/gstadder.c:
12617 Use new API to make adder sparse stream aware.
12619 2008-02-18 11:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
12621 gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
12622 Original commit message from CVS:
12623 * gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
12625 Get the object data correct so that we can remove our channels
12627 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
12628 (gen_vis_chain), (gst_play_sink_reconfigure),
12629 (gst_play_sink_request_pad):
12630 Add option to disable async behaviour in the sinks when possible. This
12631 makes it possible to avoid an audio queue when dealing with
12633 Add option to add a queue for the audio path.
12634 * tests/examples/seek/seek.c: (clear_streams), (update_streams),
12636 Disable the vis checkbox to match the defaults of playbin2.
12637 Only get the stream info when we need to.
12639 2008-02-17 05:15:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12641 ext/gio/: Don't use async operations as they require a running main loop.
12642 Original commit message from CVS:
12643 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
12644 (gst_gio_base_sink_set_stream):
12645 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
12646 (gst_gio_base_src_set_stream):
12647 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
12648 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
12649 Don't use async operations as they require a running main loop.
12650 This makes us block again when closing streams and unable
12651 to mount the enclosing volume of an URI if it isn't yet.
12653 2008-02-15 18:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
12655 gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
12656 Original commit message from CVS:
12657 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
12658 (gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
12659 (gen_vis_chain), (gst_play_sink_reconfigure),
12660 (gst_play_sink_request_pad):
12661 Move tee in front of the audio and vis pipelines.
12662 Add queue for audio for now.
12663 Add visualisation support.
12664 * tests/examples/seek/seek.c: (main):
12665 Visualisation is by default disabled.
12667 2008-02-15 11:58:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12669 ext/gio/: Improve debugging a bit.
12670 Original commit message from CVS:
12671 * ext/gio/gstgiobasesink.c: (close_stream_cb):
12672 * ext/gio/gstgiobasesrc.c: (close_stream_cb):
12673 Improve debugging a bit.
12674 * ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
12675 * ext/gio/gstgiosink.h:
12676 * ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
12677 * ext/gio/gstgiosrc.h:
12678 Try to mount the enclosing volume of a GFile if it isn't mounted
12679 yet. This requires us to wait for an async operation to finish, done
12680 with an nested GMainLoop. Authentication is not supported yet, will
12683 2008-02-14 18:24:42 +0000 Wim Taymans <wim.taymans@gmail.com>
12685 gst/playback/: Add mute property.
12686 Original commit message from CVS:
12687 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
12688 (gst_play_bin_set_property), (gst_play_bin_get_property),
12689 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
12690 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
12691 (gst_play_sink_get_mute), (gen_audio_chain):
12692 * gst/playback/gstplaysink.h:
12694 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
12695 (gst_selector_pad_chain):
12696 * gst/playback/gststreamselector.h:
12697 Make sure we forward the event only once.
12698 * tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
12699 Add and implement the mute button for playbin2.
12701 2008-02-13 14:34:55 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
12703 ext/alsa/gstalsasink.c: Add some more debug info.
12704 Original commit message from CVS:
12705 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
12706 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
12707 Add some more debug info.
12708 Make sure we never return a negative delay. Fixes #516246.
12710 2008-02-12 20:09:07 +0000 Tim-Philipp Müller <tim@centricular.net>
12712 ext/alsa/gstalsasink.c: Revert patch that makes the sink hold the object lock when calling snd_pcm_delay(), since it ...
12713 Original commit message from CVS:
12714 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
12715 Revert patch that makes the sink hold the object lock when
12716 calling snd_pcm_delay(), since it breaks playback for me.
12718 2008-02-12 19:50:36 +0000 Julien Moutte <julien@moutte.net>
12720 tests/examples/seek/seek.c: Add some seek flags when changing rate.
12721 Original commit message from CVS:
12722 2008-02-12 Julien Moutte <julien@fluendo.com>
12723 * tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
12724 some seek flags when changing rate.
12726 2008-02-12 14:51:26 +0000 Wim Taymans <wim.taymans@gmail.com>
12728 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
12729 Original commit message from CVS:
12730 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
12731 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
12732 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
12733 Fix potential leaks.
12734 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
12735 Fix leak when there is no function configured.
12737 2008-02-12 11:36:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12739 sys/v4l/v4lsrc_calls.c: Correctly chain up the finalize method.
12740 Original commit message from CVS:
12741 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
12742 (gst_v4lsrc_buffer_finalize):
12743 Correctly chain up the finalize method.
12745 2008-02-12 09:24:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12747 ext/gio/: Add documentation and example code for giostreamsink/giostreamsrc.
12748 Original commit message from CVS:
12749 * ext/gio/gstgiostreamsink.c:
12750 * ext/gio/gstgiostreamsrc.c:
12751 Add documentation and example code for giostreamsink/giostreamsrc.
12752 * tests/check/pipelines/gio.c: (GST_START_TEST):
12753 Ask the GMemoryOutputStream for the data instead of assuming that
12754 the pointer to the data stayed the same. It could've been realloc'ed.
12756 2008-02-12 08:55:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12758 ext/gio/: Make the documentation of giosink/giosrc complete, large parts are based on the gnomevfssink/gnomevfssrc docs.
12759 Original commit message from CVS:
12760 * ext/gio/gstgiosink.c:
12761 * ext/gio/gstgiosrc.c:
12762 Make the documentation of giosink/giosrc complete, large parts
12763 are based on the gnomevfssink/gnomevfssrc docs.
12765 2008-02-12 08:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12767 docs/plugins/: Add the GIO documentation again and while at that run make update.
12768 Original commit message from CVS:
12769 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
12770 * docs/plugins/gst-plugins-base-plugins-sections.txt:
12771 * docs/plugins/gst-plugins-base-plugins.args:
12772 * docs/plugins/gst-plugins-base-plugins.hierarchy:
12773 * docs/plugins/gst-plugins-base-plugins.interfaces:
12774 * docs/plugins/gst-plugins-base-plugins.prerequisites:
12775 * docs/plugins/gst-plugins-base-plugins.signals:
12776 * docs/plugins/inspect/plugin-adder.xml:
12777 * docs/plugins/inspect/plugin-audioconvert.xml:
12778 * docs/plugins/inspect/plugin-audiorate.xml:
12779 * docs/plugins/inspect/plugin-audioresample.xml:
12780 * docs/plugins/inspect/plugin-decodebin.xml:
12781 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
12782 * docs/plugins/inspect/plugin-gdp.xml:
12783 * docs/plugins/inspect/plugin-gio.xml:
12784 * docs/plugins/inspect/plugin-gnomevfs.xml:
12785 * docs/plugins/inspect/plugin-libvisual.xml:
12786 * docs/plugins/inspect/plugin-ogg.xml:
12787 * docs/plugins/inspect/plugin-pango.xml:
12788 * docs/plugins/inspect/plugin-playback.xml:
12789 * docs/plugins/inspect/plugin-queue2.xml:
12790 * docs/plugins/inspect/plugin-subparse.xml:
12791 * docs/plugins/inspect/plugin-theora.xml:
12792 * docs/plugins/inspect/plugin-uridecodebin.xml:
12793 * docs/plugins/inspect/plugin-videorate.xml:
12794 * docs/plugins/inspect/plugin-videoscale.xml:
12795 * docs/plugins/inspect/plugin-volume.xml:
12796 * docs/plugins/inspect/plugin-vorbis.xml:
12797 Add the GIO documentation again and while at that run make update.
12799 2008-02-11 20:23:44 +0000 Tim-Philipp Müller <tim@centricular.net>
12801 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...
12802 Original commit message from CVS:
12803 * ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
12804 * ext/alsa/gstalsasink.c: (set_swparams):
12805 * ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
12806 Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
12807 against libasound >= 1.0.16, since it's been deprecated in
12808 0.10.16, and alignment is always 1 then, apparently. (#512899)
12810 2008-02-11 18:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
12812 gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
12813 Original commit message from CVS:
12814 * gst/playback/gstplaybin.c: (gen_audio_element):
12815 * gst/playback/gstplaysink.c: (gen_audio_chain):
12816 Handle case where we can't create the volume element a bit
12819 2008-02-11 18:02:13 +0000 Tim-Philipp Müller <tim@centricular.net>
12821 ext/gnomevfs/: Add support for https protocol. Fixes #510229.
12822 Original commit message from CVS:
12823 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
12824 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
12825 Add support for https protocol. Fixes #510229.
12827 2008-02-11 17:03:18 +0000 Alan Peevers <peeves@pacbell.net>
12829 ext/alsa/gstalsasink.c: Take appropriate lock when calling alsa methods.
12830 Original commit message from CVS:
12831 2008-02-11 Julien Moutte <julien@fluendo.com>
12832 Patch by: Alan Peevers <peeves@pacbell.net>
12833 * ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
12834 lock when calling alsa methods.
12836 2008-02-11 13:03:13 +0000 Tim-Philipp Müller <tim@centricular.net>
12838 gst/typefind/gsttypefindfunctions.c: Bump rank of jpeg and png typefinders, which will return maximum probability in ...
12839 Original commit message from CVS:
12840 * gst/typefind/gsttypefindfunctions.c:
12841 Bump rank of jpeg and png typefinders, which will return maximum
12842 probability in the most common cases (thus short-circuiting more
12843 expensive typefinders like the mp3 one for these two quite common
12846 2008-02-11 09:48:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12848 ext/theora/theoraparse.c: Fix long description of the theora parser to be more verbose than just the type name.
12849 Original commit message from CVS:
12850 * ext/theora/theoraparse.c:
12851 Fix long description of the theora parser to be more verbose than just
12854 2008-02-11 06:47:50 +0000 Branko Čibej <brane@xbc.nu>
12856 sys/xvimage/xvimagesink.c: Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
12857 Original commit message from CVS:
12858 Patch by: Branko Čibej <brane at xbc dot nu>
12859 * sys/xvimage/xvimagesink.c:
12860 Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
12863 2008-02-09 10:41:36 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
12865 gst/playback/gstplaybasebin.c: Set is_dynamic as True if there are elements with both request and sometimes src pad t...
12866 Original commit message from CVS:
12867 * gst/playback/gstplaybasebin.c:
12868 Set is_dynamic as True if there are elements with both request
12869 and sometimes src pad templates instead of breaking out when it
12870 finds the first pad template that is a src.
12872 2008-02-08 18:17:51 +0000 Wim Taymans <wim.taymans@gmail.com>
12874 tests/examples/seek/seek.c: Add some stream switching and volume gui for playbin2.
12875 Original commit message from CVS:
12876 * tests/examples/seek/seek.c: (stop_cb), (clear_streams),
12877 (update_streams), (video_combo_cb), (audio_combo_cb),
12878 (text_combo_cb), (volume_spinbutton_changed_cb), (main):
12879 Add some stream switching and volume gui for playbin2.
12881 2008-02-08 17:47:37 +0000 Wim Taymans <wim.taymans@gmail.com>
12883 gst/playback/gstplay-marshal.list: Added marshal for streamselector Tags.
12884 Original commit message from CVS:
12885 * gst/playback/gstplay-marshal.list:
12886 Added marshal for streamselector Tags.
12887 * gst/playback/gstplaybasebin.c: (set_active_source):
12888 Streamselector now selects pads based on the pad object instead of its
12890 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
12891 (init_group), (gst_play_bin_init), (get_group), (get_tags),
12892 (gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
12893 (gst_play_bin_get_text_tags),
12894 (gst_play_bin_set_current_video_stream),
12895 (gst_play_bin_set_current_audio_stream),
12896 (gst_play_bin_set_current_text_stream),
12897 (gst_play_bin_set_property), (gst_play_bin_get_property),
12898 (pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
12899 Remove option to mute streams with the current-a/v/t property, we have
12900 this functionality in the flags.
12901 Add signals to notify when the number of A/V/T channels changed.
12902 Add action signals to get tags for the A/V/T streams.
12903 Implement setting the current A/V/T stream.
12904 Rearrange some things to simplify stream selection.
12906 * gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
12907 (gst_play_sink_get_volume), (gst_play_sink_set_property),
12908 (gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
12909 (activate_vis), (gst_play_sink_reconfigure):
12910 * gst/playback/gstplaysink.h:
12911 Add and implement volume setting methods.
12912 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
12913 (gst_selector_pad_finalize), (gst_selector_pad_get_property),
12914 (gst_selector_pad_event), (gst_stream_selector_class_init),
12915 (gst_stream_selector_init), (gst_stream_selector_finalize),
12916 (gst_stream_selector_set_property),
12917 (gst_stream_selector_get_property),
12918 (gst_stream_selector_get_linked_pad),
12919 (gst_stream_selector_request_new_pad):
12920 * gst/playback/gststreamselector.h:
12921 Add pad properties for tags and status of pads.
12923 Make active pad selection based on pad object instead of name.
12925 2008-02-08 16:10:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12927 configure.ac: Revert last change as we now check in gtk-doc.m4 for sed.
12928 Original commit message from CVS:
12930 Revert last change as we now check in gtk-doc.m4 for sed.
12932 2008-02-08 14:54:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12934 configure.ac: Find and subst SED when building the docs.
12935 Original commit message from CVS:
12937 Find and subst SED when building the docs.
12939 2008-02-08 14:34:41 +0000 Julien Moutte <julien@moutte.net>
12941 tests/examples/seek/seek.c: Make sure bus signals are reconnected when pressing STOP and then PLAY again for a parse ...
12942 Original commit message from CVS:
12943 2008-02-08 Julien Moutte <julien@fluendo.com>
12944 * tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
12945 (main): Make sure bus signals are reconnected when pressing STOP
12946 and then PLAY again for a parse launch pipeline. Fix a ref leak
12948 * win32/common/config.h: Updated.
12950 2008-02-08 00:57:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12952 configure.ac: Make DISABLE_DEPRECATED defined *only* during CVS, not during pre-releases or releases.
12953 Original commit message from CVS:
12955 Make DISABLE_DEPRECATED defined *only* during CVS, not during
12956 pre-releases or releases.
12958 2008-02-08 00:45:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12960 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is reporting
12961 Original commit message from CVS:
12963 * ext/gio/Makefile.am:
12964 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
12967 2008-02-07 23:40:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12969 docs/plugins/Makefile.am: Add the headers which need scanning for the GIO plugin. The rest of the docs still need mig...
12970 Original commit message from CVS:
12971 * docs/plugins/Makefile.am:
12972 Add the headers which need scanning for the GIO plugin. The rest of
12973 the docs still need migrating.
12975 2008-02-07 23:22:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12977 Add gio in a few more places.
12978 Original commit message from CVS:
12980 * tests/check/Makefile.am:
12981 * tests/check/pipelines/.cvsignore:
12982 Add gio in a few more places.
12984 2008-02-07 23:18:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12986 Move gio plugin from -bad and mark as experimental.
12987 Original commit message from CVS:
12990 * tests/check/Makefile.am:
12991 Move gio plugin from -bad and mark as experimental.
12993 2008-02-07 22:39:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12995 gst-libs/gst/interfaces/: Comment out a couple of other things which break the build when
12996 Original commit message from CVS:
12997 * gst-libs/gst/interfaces/mixeroptions.c:
12998 * gst-libs/gst/interfaces/mixertrack.c:
12999 Comment out a couple of other things which break the build when
13000 GST_DISABLE_DEPRECATED isn't on but -Werror is.
13002 2008-02-07 18:28:29 +0000 Tim-Philipp Müller <tim@centricular.net>
13004 docs/libs/gst-plugins-base-libs-sections.txt: Fix pbutils header.
13005 Original commit message from CVS:
13006 * docs/libs/gst-plugins-base-libs-sections.txt:
13007 Fix pbutils header.
13009 2008-02-07 18:07:41 +0000 Christian Schaller <uraeus@gnome.org>
13011 * gst-plugins-base.spec.in:
13012 commit spec file update which includes all the split .pc files
13013 Original commit message from CVS:
13014 commit spec file update which includes all the split .pc files
13016 2008-02-07 12:17:49 +0000 Wim Taymans <wim.taymans@gmail.com>
13018 gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
13019 Original commit message from CVS:
13020 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
13021 Fix compiler warning.
13023 2008-02-07 11:00:45 +0000 Peter Kjellerstedt <pkj@axis.com>
13025 gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
13026 Original commit message from CVS:
13027 Patch by: Peter Kjellerstedt <pkj at axis com>
13028 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
13029 Clear the addrinfo struct using memset. Fixes #514937.
13031 2008-02-06 15:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
13033 gst/tcp/gstfdset.h: Remove unused field to same some memory.
13034 Original commit message from CVS:
13035 * gst/tcp/gstfdset.h:
13036 Remove unused field to same some memory.
13037 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
13038 Mark action signals as such.
13040 2008-02-06 13:35:58 +0000 Michael Smith <msmith@xiph.org>
13042 ext/theora/theoradec.c: Increment granulepos for new-bitstream versions appropriately.
13043 Original commit message from CVS:
13044 * ext/theora/theoradec.c: (_theora_granule_frame),
13046 Increment granulepos for new-bitstream versions appropriately.
13049 2008-02-04 11:51:31 +0000 Wim Taymans <wim.taymans@gmail.com>
13051 tests/examples/seek/seek.c: Remove obsolete stream_time reset after flushing seek, core does that automatically now.
13052 Original commit message from CVS:
13053 * tests/examples/seek/seek.c: (do_seek),
13054 (rate_spinbutton_changed_cb), (update_streams), (main):
13055 Remove obsolete stream_time reset after flushing seek, core does that
13057 Improve accuracy of speed spinbutton.
13058 Only do playbin2 stuff when we actually use it.
13060 2008-02-02 17:29:32 +0000 Tim-Philipp Müller <tim@centricular.net>
13062 tests/check/Makefile.am: Revert previous change of the test environment's GST_PLUGIN_PATH.
13063 Original commit message from CVS:
13064 * tests/check/Makefile.am:
13065 Revert previous change of the test environment's GST_PLUGIN_PATH.
13066 The problem is not with the plugins, but with element factories
13067 and only occurs if elements are split out from existing plugins
13068 or if plugins change name (see #512740).
13070 2008-02-02 15:32:23 +0000 Tim-Philipp Müller <tim@centricular.net>
13072 tests/check/Makefile.am: Fix the tests environment's GST_PLUGIN_PATH: we want the directory with the core's plugins f...
13073 Original commit message from CVS:
13074 * tests/check/Makefile.am:
13075 Fix the tests environment's GST_PLUGIN_PATH: we want the directory
13076 with the core's plugins first and our local build directories last,
13077 since we might be building against an installed core, and that
13078 core's plugin directory may contain older or other versions of
13079 our own -base plugins, but we really do want to test our local
13080 ones (if there are multiple plugins or element factories with the
13081 same name, those inspected last will trump those read in earlier).
13082 Fixes #512740 for the most part.
13084 2008-02-02 07:13:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13086 Use gmtime_r if available as gmtime is not MT-safe.
13087 Original commit message from CVS:
13089 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
13090 Use gmtime_r if available as gmtime is not MT-safe.
13093 2008-02-02 06:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13095 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
13096 Original commit message from CVS:
13097 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
13098 Cast glong to time_t as time_t might have a different type on
13099 other platforms, like FreeBSD, and we get a compiler warning
13100 otherwise. Fixes bug #511825.
13102 2008-02-01 16:44:21 +0000 Wim Taymans <wim.taymans@gmail.com>
13104 gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
13105 Original commit message from CVS:
13106 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
13107 (get_group), (get_n_pads), (gst_play_bin_get_property),
13108 (pad_added_cb), (no_more_pads_cb), (perform_eos),
13109 (autoplug_select_cb), (deactivate_group):
13110 Remove stream-info, we going for something easier.
13111 Refactor getting the current group.
13112 Implement getting the number of audio/video/text streams.
13113 * gst/playback/gststreamselector.c:
13114 (gst_stream_selector_class_init), (gst_stream_selector_init),
13115 (gst_stream_selector_get_property),
13116 (gst_stream_selector_request_new_pad),
13117 (gst_stream_selector_release_pad):
13118 * gst/playback/gststreamselector.h:
13119 Add property for number of pads.
13120 * tests/examples/seek/seek.c: (set_scale), (update_flag),
13121 (vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
13122 (text_toggle_cb), (update_streams), (msg_async_done),
13123 (msg_state_changed), (main):
13124 Block slider callback when updating the slider position.
13125 Add gui elements for controlling playbin2.
13126 Add callback for async_done that updates position/duration.
13128 2008-02-01 12:56:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13130 docs/plugins/: First round of plugin docs cleansups.
13131 Original commit message from CVS:
13132 * docs/plugins/Makefile.am:
13133 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
13134 * docs/plugins/gst-plugins-base-plugins-sections.txt:
13135 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13136 * docs/plugins/gst-plugins-base-plugins.interfaces:
13137 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13138 First round of plugin docs cleansups.
13139 * docs/plugins/inspect/plugin-adder.xml:
13140 * docs/plugins/inspect/plugin-alsa.xml:
13141 * docs/plugins/inspect/plugin-audioconvert.xml:
13142 * docs/plugins/inspect/plugin-audiorate.xml:
13143 * docs/plugins/inspect/plugin-audioresample.xml:
13144 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13145 * docs/plugins/inspect/plugin-cdparanoia.xml:
13146 * docs/plugins/inspect/plugin-decodebin.xml:
13147 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13148 * docs/plugins/inspect/plugin-gdp.xml:
13149 * docs/plugins/inspect/plugin-gnomevfs.xml:
13150 * docs/plugins/inspect/plugin-libvisual.xml:
13151 * docs/plugins/inspect/plugin-ogg.xml:
13152 * docs/plugins/inspect/plugin-pango.xml:
13153 * docs/plugins/inspect/plugin-subparse.xml:
13154 * docs/plugins/inspect/plugin-tcp.xml:
13155 * docs/plugins/inspect/plugin-theora.xml:
13156 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13157 * docs/plugins/inspect/plugin-video4linux.xml:
13158 * docs/plugins/inspect/plugin-videorate.xml:
13159 * docs/plugins/inspect/plugin-videoscale.xml:
13160 * docs/plugins/inspect/plugin-videotestsrc.xml:
13161 * docs/plugins/inspect/plugin-volume.xml:
13162 * docs/plugins/inspect/plugin-vorbis.xml:
13163 * docs/plugins/inspect/plugin-ximagesink.xml:
13164 * docs/plugins/inspect/plugin-xvimagesink.xml:
13166 * ext/ogg/Makefile.am:
13167 * ext/ogg/gstoggmux.c:
13168 * ext/ogg/gstoggmux.h:
13169 Add header for oggmux. the c-file needs a doc blob still.
13171 2008-02-01 11:09:16 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13173 Add gst_rtp_buffer_set_extension_data()
13174 Original commit message from CVS:
13175 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
13176 * gst-libs/gst/rtp/gstrtpbuffer.c:
13177 (gst_rtp_buffer_set_extension_data):
13178 * gst-libs/gst/rtp/gstrtpbuffer.h:
13179 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
13180 Add gst_rtp_buffer_set_extension_data()
13181 Add a unit test for this addition. Fixes #511478.
13182 API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
13184 2008-01-31 17:18:46 +0000 Wim Taymans <wim.taymans@gmail.com>
13186 gst-libs/gst/app/gstappsink.c: Really clean up the queue instead of just unreffing all buffers in it.
13187 Original commit message from CVS:
13188 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
13189 Really clean up the queue instead of just unreffing all buffers
13191 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
13192 (gst_app_src_class_init), (gst_app_src_init),
13193 (gst_app_src_dispose), (gst_app_src_finalize):
13194 Fix dispose/finalize.
13196 2008-01-30 15:34:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13198 ext/gio/: Use async variants of the close stream functions to prevent blocking for a long time there and add some mor...
13199 Original commit message from CVS:
13200 * ext/gio/gstgiobasesink.c: (close_stream_cb),
13201 (gst_gio_base_sink_stop), (gst_gio_base_sink_event),
13202 (gst_gio_base_sink_render), (gst_gio_base_sink_set_stream):
13203 * ext/gio/gstgiobasesrc.c: (close_stream_cb),
13204 (gst_gio_base_src_stop), (gst_gio_base_src_create),
13205 (gst_gio_base_src_set_stream):
13206 Use async variants of the close stream functions to prevent blocking
13207 for a long time there and add some more sanity checks for a correct
13210 2008-01-30 14:42:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13212 configure.ac: Back to CVS
13213 Original commit message from CVS:
13217 === release 0.10.17 ===
13219 2008-01-30 14:19:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13225 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13226 * docs/plugins/inspect/plugin-adder.xml:
13227 * docs/plugins/inspect/plugin-alsa.xml:
13228 * docs/plugins/inspect/plugin-audioconvert.xml:
13229 * docs/plugins/inspect/plugin-audiorate.xml:
13230 * docs/plugins/inspect/plugin-audioresample.xml:
13231 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13232 * docs/plugins/inspect/plugin-cdparanoia.xml:
13233 * docs/plugins/inspect/plugin-decodebin.xml:
13234 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13235 * docs/plugins/inspect/plugin-gdp.xml:
13236 * docs/plugins/inspect/plugin-gnomevfs.xml:
13237 * docs/plugins/inspect/plugin-libvisual.xml:
13238 * docs/plugins/inspect/plugin-ogg.xml:
13239 * docs/plugins/inspect/plugin-pango.xml:
13240 * docs/plugins/inspect/plugin-subparse.xml:
13241 * docs/plugins/inspect/plugin-tcp.xml:
13242 * docs/plugins/inspect/plugin-theora.xml:
13243 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13244 * docs/plugins/inspect/plugin-video4linux.xml:
13245 * docs/plugins/inspect/plugin-videorate.xml:
13246 * docs/plugins/inspect/plugin-videoscale.xml:
13247 * docs/plugins/inspect/plugin-videotestsrc.xml:
13248 * docs/plugins/inspect/plugin-volume.xml:
13249 * docs/plugins/inspect/plugin-vorbis.xml:
13250 * docs/plugins/inspect/plugin-ximagesink.xml:
13251 * docs/plugins/inspect/plugin-xvimagesink.xml:
13252 * gst-plugins-base.doap:
13253 * win32/common/config.h:
13255 Original commit message from CVS:
13258 2008-01-30 13:45:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13260 gst-libs/gst/interfaces/: Also remove the conditional registration of the signals that disappeared with the ABI chang...
13261 Original commit message from CVS:
13262 * gst-libs/gst/interfaces/mixeroptions.c:
13263 * gst-libs/gst/interfaces/mixertrack.c:
13264 Also remove the conditional registration of the signals
13265 that disappeared with the ABI change in 0.10.14
13267 2008-01-30 12:28:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13269 gst-libs/gst/rtsp/gstrtspconnection.c: Revert patch to gstrtspconnection.c for brown paper bag release of -base. Re-o...
13270 Original commit message from CVS:
13271 * gst-libs/gst/rtsp/gstrtspconnection.c:
13272 Revert patch to gstrtspconnection.c for brown paper bag
13273 release of -base. Re-opens: #511825
13275 2008-01-30 12:20:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13277 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
13278 Original commit message from CVS:
13279 * gst-libs/gst/interfaces/mixeroptions.h:
13280 * gst-libs/gst/interfaces/mixertrack.h:
13281 Change the way these deprecated function pointers are removed
13282 so that the compiled ABI is unconditionally smaller. This
13283 sets in stone an ABI break that actually occurred when the
13284 things were deprecated in 0.10.14, which seems to be the best
13285 fix as the only known users are oss-mixer and sunaudio-mixer in
13289 2008-01-30 12:19:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13291 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
13292 Original commit message from CVS:
13293 * gst-libs/gst/interfaces/mixeroptions.h:
13294 * gst-libs/gst/interfaces/mixertrack.h:
13295 Change the way these deprecated function pointers are removed
13296 so that the compiled ABI is unconditionally smaller. This
13297 sets in stone an ABI break that actually occurred when the
13298 things were deprecated in 0.10.14, which seems to be the best
13299 fix as the only known users are oss-mixer and sunaudio-mixer in
13302 2008-01-30 11:43:53 +0000 Tim-Philipp Müller <tim@centricular.net>
13304 win32/common/libgstpbutils.def: Export the two new _get_type() functions which are needed by the python bindings.
13305 Original commit message from CVS:
13306 * win32/common/libgstpbutils.def:
13307 Export the two new _get_type() functions which are needed
13308 by the python bindings.
13310 2008-01-29 09:59:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13312 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
13313 Original commit message from CVS:
13314 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
13315 Cast glong to time_t as time_t might have a different type on
13316 other platforms, like FreeBSD, and we get a compiler warning
13317 otherwise. Fixes bug #511825.
13319 2008-01-29 09:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13321 gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
13322 Original commit message from CVS:
13323 * gst-libs/gst/audio/gstaudiofilter.c:
13324 (gst_audio_filter_class_init):
13325 Initialize the GstRingerBuffer class to get it's debug category
13326 initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
13327 category and otherwise we get some g_critical(). Fixes bug #512334.
13329 2008-01-28 23:35:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13331 configure.ac: Back to CVS
13332 Original commit message from CVS:
13336 === release 0.10.16 ===
13338 2008-01-28 23:31:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13344 * docs/plugins/gst-plugins-base-plugins.args:
13345 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13346 * docs/plugins/gst-plugins-base-plugins.interfaces:
13347 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13348 * docs/plugins/gst-plugins-base-plugins.signals:
13349 * docs/plugins/inspect/plugin-adder.xml:
13350 * docs/plugins/inspect/plugin-alsa.xml:
13351 * docs/plugins/inspect/plugin-audioconvert.xml:
13352 * docs/plugins/inspect/plugin-audiorate.xml:
13353 * docs/plugins/inspect/plugin-audioresample.xml:
13354 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13355 * docs/plugins/inspect/plugin-cdparanoia.xml:
13356 * docs/plugins/inspect/plugin-decodebin.xml:
13357 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13358 * docs/plugins/inspect/plugin-gdp.xml:
13359 * docs/plugins/inspect/plugin-gnomevfs.xml:
13360 * docs/plugins/inspect/plugin-libvisual.xml:
13361 * docs/plugins/inspect/plugin-ogg.xml:
13362 * docs/plugins/inspect/plugin-pango.xml:
13363 * docs/plugins/inspect/plugin-subparse.xml:
13364 * docs/plugins/inspect/plugin-tcp.xml:
13365 * docs/plugins/inspect/plugin-theora.xml:
13366 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13367 * docs/plugins/inspect/plugin-video4linux.xml:
13368 * docs/plugins/inspect/plugin-videorate.xml:
13369 * docs/plugins/inspect/plugin-videoscale.xml:
13370 * docs/plugins/inspect/plugin-videotestsrc.xml:
13371 * docs/plugins/inspect/plugin-volume.xml:
13372 * docs/plugins/inspect/plugin-vorbis.xml:
13373 * docs/plugins/inspect/plugin-ximagesink.xml:
13374 * docs/plugins/inspect/plugin-xvimagesink.xml:
13375 * gst-plugins-base.doap:
13376 * win32/common/config.h:
13378 Original commit message from CVS:
13381 2008-01-28 22:15:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13407 Original commit message from CVS:
13410 2008-01-22 15:37:49 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13412 gst-libs/gst/rtp/gstrtpbuffer.c: Fix typos and wrong extension check. Fixes #511274.
13413 Original commit message from CVS:
13414 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
13415 * gst-libs/gst/rtp/gstrtpbuffer.c:
13416 (gst_rtp_buffer_get_extension_data):
13417 Fix typos and wrong extension check. Fixes #511274.
13419 2008-01-18 00:03:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13421 po/sk.po: Oops - add new sk.po mentioned in the LINGUAS I just committed
13422 Original commit message from CVS:
13424 Oops - add new sk.po mentioned in the LINGUAS I just committed
13426 2008-01-17 22:31:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13428 po/LINGUAS: Add ca translation to the disted list.
13429 Original commit message from CVS:
13431 Add ca translation to the disted list.
13432 * win32/vs6/libgstsdp.dsp:
13433 Convert line endings to CRLF
13435 2008-01-17 21:58:53 +0000 Sébastien Moutte <sebastien@moutte.net>
13437 win32/MANIFEST: Add win32/vs6/libgstrtsp.dsp to MANIFEST
13438 Original commit message from CVS:
13440 Add win32/vs6/libgstrtsp.dsp to MANIFEST
13442 2008-01-16 05:40:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13444 Update for API changes in GIO and require GIO 2.15.2 for this.
13445 Original commit message from CVS:
13447 * tests/check/pipelines/gio.c: (GST_START_TEST):
13448 Update for API changes in GIO and require GIO 2.15.2 for this.
13450 2008-01-14 22:20:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13452 win32/common/: Add new API declarations
13453 Original commit message from CVS:
13454 * win32/common/libgstsdp.def:
13455 * win32/common/libgstvideo.def:
13456 Add new API declarations
13458 2008-01-14 17:00:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13460 ext/theora/: Take a 2nd stab at handling libtheora granulepos changes in the decoder and parser by inspecting the bit...
13461 Original commit message from CVS:
13462 * ext/theora/gsttheoradec.h:
13463 * ext/theora/gsttheoraparse.h:
13464 * ext/theora/theoradec.c:
13465 * ext/theora/theoraparse.c:
13466 Take a 2nd stab at handling libtheora granulepos changes in the decoder
13467 and parser by inspecting the bitstream version of the incoming data.
13469 2008-01-14 13:11:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13471 Provide one pkg-config file for every gst-plugins-base library.
13472 Original commit message from CVS:
13474 * pkgconfig/Makefile.am:
13475 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
13476 * pkgconfig/gstreamer-audio.pc.in:
13477 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
13478 * pkgconfig/gstreamer-cdda.pc.in:
13479 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
13480 * pkgconfig/gstreamer-fft.pc.in:
13481 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
13482 * pkgconfig/gstreamer-floatcast.pc.in:
13483 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
13484 * pkgconfig/gstreamer-interfaces.pc.in:
13485 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
13486 * pkgconfig/gstreamer-netbuffer.pc.in:
13487 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
13488 * pkgconfig/gstreamer-pbutils.pc.in:
13489 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
13490 * pkgconfig/gstreamer-riff.pc.in:
13491 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
13492 * pkgconfig/gstreamer-rtp.pc.in:
13493 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
13494 * pkgconfig/gstreamer-rtsp.pc.in:
13495 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
13496 * pkgconfig/gstreamer-sdp.pc.in:
13497 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
13498 * pkgconfig/gstreamer-tag.pc.in:
13499 * pkgconfig/gstreamer-video-uninstalled.pc.in:
13500 * pkgconfig/gstreamer-video.pc.in:
13501 Provide one pkg-config file for every gst-plugins-base library.
13502 This makes linking to those libraries much more intuitive and
13503 provides standard pkg-config behaviour for them. Fixes bug #499697.
13505 2008-01-14 01:19:34 +0000 David Schleef <ds@schleef.org>
13507 gst/videoscale/vs_4tap.c: Fix valgrind error on 4tap scaling method.
13508 Original commit message from CVS:
13509 * gst/videoscale/vs_4tap.c:
13510 Fix valgrind error on 4tap scaling method.
13512 2008-01-13 21:40:45 +0000 Sébastien Moutte <sebastien@moutte.net>
13514 gst-libs/gst/sdp/gstsdpmessage.c: Include Winsock2.h for VS6 and use a different way initialize hints structure so it...
13515 Original commit message from CVS:
13516 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
13517 Include Winsock2.h for VS6 and use a different way initialize
13518 hints structure so it can build with VS6.
13520 * win32/vs6/libgstsdp.dsp:
13521 * win32/common/libgstsdp.def:
13522 Add new files for libgstsdp.
13523 * win32/vs6/grammar.dsp:
13524 Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
13525 * win32/vs6/gst_plugins_base.dsw:
13526 * win32/vs6/libgstdecodebin.dsp:
13527 * win32/vs6/libgstdecodebin2.dsp:
13528 * win32/vs6/libgstplaybin.dsp:
13529 * win32/vs6/libgstvolume.dsp:
13530 Add new dependencies to the link list.
13532 2008-01-13 17:24:49 +0000 Julien Moutte <julien@moutte.net>
13534 win32/common/: Update/Add generated files in the win32 build directory.
13535 Original commit message from CVS:
13536 2008-01-13 Julien Moutte <julien@fluendo.com>
13537 * win32/common/config.h:
13538 * win32/common/gstrtsp-enumtypes.c: (gst_rtsp_result_get_type),
13539 (gst_rtsp_event_get_type), (gst_rtsp_family_get_type),
13540 (gst_rtsp_state_get_type), (gst_rtsp_version_get_type),
13541 (gst_rtsp_method_get_type), (gst_rtsp_auth_method_get_type),
13542 (gst_rtsp_header_field_get_type),
13543 (gst_rtsp_status_code_get_type):
13544 * win32/common/interfaces-enumtypes.c:
13545 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
13546 (gst_mixer_message_type_get_type), (gst_mixer_flags_get_type),
13547 (gst_mixer_track_flags_get_type),
13548 (gst_tuner_channel_flags_get_type):
13549 * win32/common/multichannel-enumtypes.c:
13550 (gst_audio_channel_position_get_type):
13551 * win32/common/pbutils-enumtypes.c:
13552 (gst_install_plugins_return_get_type):
13553 * win32/common/pbutils-enumtypes.h: Update/Add generated files
13554 in the win32 build directory.
13556 2008-01-12 23:24:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13558 tests/check/Makefile.am: Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
13559 Original commit message from CVS:
13560 * tests/check/Makefile.am:
13561 Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
13562 * tests/check/elements/audiorate.c: (do_perfect_stream_test):
13563 * tests/check/elements/playbin.c:
13564 * tests/check/libs/mixer.c: (test_element_interface_supported),
13565 (gst_implements_interface_init):
13566 * tests/check/libs/rtp.c: (GST_START_TEST):
13567 Fix various assignment type mismatches.
13569 2008-01-12 23:08:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13571 Add test to see if hstrerror is available or if we need libresolv (Solaris) for it, then use it in libgstrtsp.
13572 Original commit message from CVS:
13574 * gst-libs/gst/rtsp/Makefile.am:
13575 Add test to see if hstrerror is available or if we need libresolv
13576 (Solaris) for it, then use it in libgstrtsp.
13578 2008-01-12 14:54:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13580 gst-libs/gst/tag/Makefile.am: Fix include path order
13581 Original commit message from CVS:
13582 * gst-libs/gst/tag/Makefile.am:
13583 Fix include path order
13585 2008-01-11 17:15:23 +0000 Tim-Philipp Müller <tim@centricular.net>
13587 * gst-libs/gst/pbutils/.gitignore:
13588 Ignore more and make buildbot happy
13589 Original commit message from CVS:
13590 Ignore more and make buildbot happy
13592 2008-01-11 16:18:10 +0000 Edward Hervey <bilboed@bilboed.com>
13594 gst-libs/gst/pbutils/install-plugins.*: Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping for bi...
13595 Original commit message from CVS:
13596 * gst-libs/gst/pbutils/install-plugins.c:
13597 (gst_install_plugins_context_copy),
13598 (gst_install_plugins_context_get_type):
13599 * gst-libs/gst/pbutils/install-plugins.h:
13600 Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
13603 2008-01-11 15:48:11 +0000 Michael Smith <msmith@xiph.org>
13605 ext/theora/theoradec.c: Adapt for post-alpha meaning of granulepos, when we have a newer version of libtheora.
13606 Original commit message from CVS:
13607 * ext/theora/theoradec.c: (gst_theora_dec_class_init),
13608 (_theora_granule_frame), (_theora_granule_start_time),
13609 (theora_dec_sink_convert), (theora_dec_decode_buffer):
13610 Adapt for post-alpha meaning of granulepos, when we
13611 have a newer version of libtheora.
13612 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
13613 (theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
13614 (theora_enc_is_discontinuous), (theora_enc_chain):
13616 * tests/check/Makefile.am:
13617 Link libtheora into theoraenc test so we can check which version of
13618 libtheora we're testing against.
13619 * tests/check/pipelines/theoraenc.c: (check_libtheora),
13620 (check_buffer_granulepos),
13621 (check_buffer_granulepos_from_starttime), (GST_START_TEST),
13623 Adapt tests to check the values that are now defined for theora; make
13624 the tests backwards-adapt the passed values if we're running against an
13628 2008-01-10 17:55:53 +0000 Tim-Philipp Müller <tim@centricular.net>
13630 gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
13631 Original commit message from CVS:
13632 * gst-libs/gst/audio/gstbaseaudiosink.c:
13633 (gst_base_audio_sink_class_init):
13634 * gst-libs/gst/audio/gstbaseaudiosrc.c:
13635 (gst_base_audio_src_class_init):
13636 Ref audio clock class from a thread-safe context to make sure
13637 we're not bit by GObjects lack of thread-safety here (#349410),
13638 however unlikely that may be in practice.
13640 2008-01-10 12:22:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13642 autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We...
13643 Original commit message from CVS:
13645 Add -Wno-portability to the automake parameters to stop warnings
13646 about GNU make extensions being used. We require GNU make in almost
13647 every Makefile anyway.
13649 Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
13650 at the same time is required for per target flags.
13652 2008-01-08 21:10:02 +0000 Tim-Philipp Müller <tim@centricular.net>
13654 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...
13655 Original commit message from CVS:
13656 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
13657 Post an error message if we can't pull as many bytes as we need
13658 for the tag. This makes sure the user gets to see a proper error
13659 message if a file with a partial ID3 tag is fed to decodebin, and
13660 not a 'no ID3 tag demuxer' error, which would be confusing
13663 2008-01-08 20:59:20 +0000 Tim-Philipp Müller <tim@centricular.net>
13665 gst-libs/gst/pbutils/descriptions.c: Add description strings for ID3, APE, and ICY tags.
13666 Original commit message from CVS:
13667 * gst-libs/gst/pbutils/descriptions.c: (formats):
13668 Add description strings for ID3, APE, and ICY tags.
13670 2008-01-08 20:48:00 +0000 Tim-Philipp Müller <tim@centricular.net>
13672 gst/playback/gstdecodebin.c: Make sure we error out correctly if we can't activate one of the elements we've added. ...
13673 Original commit message from CVS:
13674 * gst/playback/gstdecodebin.c: (try_to_link_1):
13675 Make sure we error out correctly if we can't activate one of
13676 the elements we've added. Fixes #508138.
13678 2008-01-07 13:59:43 +0000 Bastien Nocera <hadess@hadess.net>
13680 ext/alsa/gstalsamixer.c: Use snd_mixer_selem_set_{playback|capture}_volume_all() if the volume is the same for all ch...
13681 Original commit message from CVS:
13682 Patch by: Bastien Nocera <hadess at hadess net>
13683 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
13684 (check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
13685 Use snd_mixer_selem_set_{playback|capture}_volume_all() if
13686 the volume is the same for all channels. This works around
13687 some problem in alsa that leaves us with inconsistent state
13688 for some reason (#486840).
13690 2008-01-07 13:19:50 +0000 Jerone Young <jerone@gmail.com>
13692 ext/alsa/gstalsamixer.c: If there's no mixer track by the name of 'Master' or 'Front', check if there's one called 'P...
13693 Original commit message from CVS:
13694 Patch by: Jerone Young <jerone at gmail com>
13695 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
13696 If there's no mixer track by the name of 'Master' or 'Front',
13697 check if there's one called 'PCM' before trying the generic
13698 fallback logic (fixes #506928, where we pick 'Mic' as master
13699 track for the AD1984 card in a Thinkpad T61/X61 laptop).
13701 2008-01-07 11:40:04 +0000 Wim Taymans <wim.taymans@gmail.com>
13703 gst/playback/gstplay-enum.*: Add enums for configuration flags.
13704 Original commit message from CVS:
13705 * gst/playback/gstplay-enum.c:
13706 (register_gst_autoplug_select_result),
13707 (gst_autoplug_select_result_get_type), (register_gst_play_flags),
13708 (gst_play_flags_get_type):
13709 * gst/playback/gstplay-enum.h:
13710 Add enums for configuration flags.
13711 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
13712 (init_group), (gst_play_bin_init), (gst_play_bin_set_property),
13713 (gst_play_bin_get_property), (no_more_pads_cb),
13714 (autoplug_select_cb), (gst_play_bin_change_state):
13715 Merge mode with flags.
13716 Add more property getters/setters, defaults and docs.
13717 Add properties to get number of audio/video/text streams.
13718 Create sink object in _init so that we can always rely on it being
13720 * gst/playback/gstplaysink.c: (gst_play_sink_init),
13721 (gen_video_chain), (gen_audio_chain), (gen_vis_chain),
13722 (activate_vis), (gst_play_sink_reconfigure),
13723 (gst_play_sink_set_flags), (gst_play_sink_get_flags),
13724 (gst_play_sink_change_state):
13725 * gst/playback/gstplaysink.h:
13726 Use flags to configure the sink pipelines.
13727 Add tee before audio pipeline so that we can use it for visualisations.
13728 Start working on integrating visualisations.
13729 Remove mode, we can do everything with the flags now.
13730 Add method to configue the sink pipeline.
13732 2008-01-06 16:36:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13734 Update to GMemoryInputStream API changes in GLib SVN and require gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
13735 Original commit message from CVS:
13737 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
13738 * tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
13739 Update to GMemoryInputStream API changes in GLib SVN and require
13740 gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
13741 We can also report the duration for every GSeekable, not only
13742 GFileInputStream and GMemoryInputStream.
13744 2008-01-06 14:39:19 +0000 Tim-Philipp Müller <tim@centricular.net>
13746 tests/check/pipelines/theoraenc.c: Turn these functions into macros so we can see right away where the failure occured.
13747 Original commit message from CVS:
13748 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
13749 (check_buffer_timestamp), (check_buffer_duration):
13750 Turn these functions into macros so we can see right away
13751 where the failure occured.
13753 2008-01-05 22:25:05 +0000 Julien Moutte <julien@moutte.net>
13755 sys/xvimage/xvimagesink.c: Add debugging information to understand how X calculates the stride for XvImages.
13756 Original commit message from CVS:
13757 2008-01-05 Julien Moutte <julien@fluendo.com>
13758 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
13759 debugging information to understand how X calculates the stride
13762 2008-01-03 20:33:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13764 gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
13765 Original commit message from CVS:
13766 * gst/volume/Makefile.am:
13767 * gst/volume/gstvolume.c: (volume_choose_func),
13768 (gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
13770 * gst/volume/gstvolume.h:
13771 Use GstAudioFilter as base class for the volume element instead of
13772 plain GstBaseTransform.
13774 2008-01-03 07:17:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13776 gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
13777 Original commit message from CVS:
13778 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
13779 Don't set element details for the abstract GstAudioFilter class.
13781 2008-01-02 12:09:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13783 gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
13784 Original commit message from CVS:
13785 * gst-libs/gst/audio/gstaudiofilter.c:
13786 (gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
13787 Implement get_unit_size() vmethod of GstBaseTransform.
13789 2008-01-01 12:53:48 +0000 Edward Hervey <bilboed@bilboed.com>
13791 gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
13792 Original commit message from CVS:
13793 * gst-libs/gst/pbutils/Makefile.am:
13794 * gst-libs/gst/pbutils/pbutils.h:
13795 Use glib-enum generator to have a proper enum GType for
13796 GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
13798 2008-01-01 01:21:47 +0000 David Schleef <ds@schleef.org>
13800 tests/check/: Reenable theoraenc test, which fails on the buildbot but not locally.
13801 Original commit message from CVS:
13802 * tests/check/Makefile.am:
13803 * tests/check/pipelines/theoraenc.c:
13804 Reenable theoraenc test, which fails on the buildbot but
13807 2007-12-31 21:31:01 +0000 David Schleef <ds@schleef.org>
13809 docs/: Add *-undeclared.txt to fix buildbot.
13810 Original commit message from CVS:
13811 * docs/libs/.cvsignore:
13812 * docs/plugins/.cvsignore:
13813 Add *-undeclared.txt to fix buildbot.
13815 2007-12-31 20:45:28 +0000 David Schleef <ds@schleef.org>
13817 tests/check/Makefile.am: Second attempt at disabling theoraenc test long enough to get buildbot to compile -base.
13818 Original commit message from CVS:
13819 * tests/check/Makefile.am:
13820 Second attempt at disabling theoraenc test long enough to
13821 get buildbot to compile -base.
13823 2007-12-31 20:21:20 +0000 David Schleef <ds@schleef.org>
13825 tests/check/pipelines/theoraenc.c: Disable theoraenc test long enough to get the buildbot to compile a recent -base.
13826 Original commit message from CVS:
13827 * tests/check/pipelines/theoraenc.c:
13828 Disable theoraenc test long enough to get the buildbot to
13829 compile a recent -base.
13831 2007-12-31 13:17:29 +0000 Wim Taymans <wim.taymans@gmail.com>
13833 tests/examples/seek/seek.c: Make sure we reset the slider value to 0.0 without racing against a possible g_idle that ...
13834 Original commit message from CVS:
13835 * tests/examples/seek/seek.c: (stop_cb):
13836 Make sure we reset the slider value to 0.0 without racing against a
13837 possible g_idle that sets it to something else.
13839 2007-12-31 00:32:53 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13841 sys/ximage/ximagesink.c: fix typo
13842 Original commit message from CVS:
13843 * sys/ximage/ximagesink.c:
13846 2007-12-30 19:21:16 +0000 Wim Taymans <wim.taymans@gmail.com>
13848 gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
13849 Original commit message from CVS:
13850 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
13851 * gst-libs/gst/rtsp/gstrtspdefs.h:
13852 Add Location header so that we can start implementing redirects.
13855 2007-12-29 20:55:39 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13857 gst/subparse/gstssaparse.c: combine if's
13858 Original commit message from CVS:
13859 * gst/subparse/gstssaparse.c:
13862 2007-12-29 19:23:59 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13864 gst/subparse/gstssaparse.c: remove duplicate log message
13865 Original commit message from CVS:
13866 * gst/subparse/gstssaparse.c:
13867 remove duplicate log message
13869 2007-12-29 17:29:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13871 Update to latest API changes in GLib/GIO and require at least gio-2.0 2.15.0 for this.
13872 Original commit message from CVS:
13874 * ext/gio/gstgio.c:
13875 * ext/gio/gstgio.h:
13876 * ext/gio/gstgiobasesink.h:
13877 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
13878 * ext/gio/gstgiobasesrc.h:
13879 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
13880 * ext/gio/gstgiosink.h:
13881 * ext/gio/gstgiosrc.h:
13882 * ext/gio/gstgiostreamsink.h:
13883 * ext/gio/gstgiostreamsrc.h:
13884 * tests/check/pipelines/gio.c:
13885 Update to latest API changes in GLib/GIO and require at least
13886 gio-2.0 2.15.0 for this.
13887 * ext/gio/Makefile.am:
13888 Add GST_PLUGIN_LDFLAGS to LDFLAGS.
13890 2007-12-29 16:23:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13892 ext/libvisual/visual.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached()...
13893 Original commit message from CVS:
13894 * ext/libvisual/visual.c: (gst_visual_chain):
13895 Fix 'xyz may be used uninitialized' compiler warnings caused
13896 by broken g_assert_not_reached() macro in GLib-2.15.x and don't
13897 abort() in any case but properly report the error.
13899 2007-12-28 09:00:27 +0000 Wim Taymans <wim.taymans@gmail.com>
13901 gst/playback/gstplaybin2.c: Code cleanups.
13902 Original commit message from CVS:
13903 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
13904 (gst_play_bin_finalize), (gst_play_bin_set_uri),
13905 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
13906 (gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
13907 (autoplug_select_cb), (activate_group), (deactivate_group),
13908 (setup_next_source), (save_current_group),
13909 (gst_play_bin_change_state):
13911 Remove next-uri, we can use the uri property just fine.
13913 Unref uridecodebin when switching.
13914 Fix going to READY.
13915 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
13916 (gst_play_sink_init), (gst_play_sink_dispose),
13917 (gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
13918 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
13919 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
13920 (gst_play_sink_set_property), (gst_play_sink_get_property),
13921 (gen_video_chain), (gen_text_element), (gen_audio_chain),
13922 (gen_vis_element), (gst_play_sink_get_mode),
13923 (gst_play_sink_set_mode), (gst_play_sink_set_flags),
13924 (gst_play_sink_get_flags), (gst_play_sink_request_pad),
13925 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
13926 (gst_play_sink_change_state):
13927 * gst/playback/gstplaysink.h:
13928 Add some locking to make things threadsafe.
13929 * gst/playback/test7.c: (about_to_finish_cb):
13932 2007-12-22 12:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
13934 gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
13935 Original commit message from CVS:
13936 * gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
13937 (gst_video_scale_get_property), (gst_video_scale_transform_caps),
13938 (gst_video_scale_transform):
13939 Don't claim to be able to handle/transform caps that can't really
13940 be handled by the currently selected scaling method (here: RGB or
13941 packed YUV with 4-tap method). Also add locking to method property.
13942 * tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
13943 (test_basetransform_based):
13944 Some test pipelines for the above (not entirely valgrind clean yet
13947 2007-12-22 05:19:00 +0000 David Schleef <ds@schleef.org>
13949 gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
13950 Original commit message from CVS:
13951 * gst-libs/gst/video/video.c:
13952 * gst-libs/gst/video/video.h:
13953 Add additional RGBA and RGB-24 video formats.
13955 2007-12-21 22:46:56 +0000 Tim-Philipp Müller <tim@centricular.net>
13957 tests/check/: Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be deprecated in the future (see #498924).
13958 Original commit message from CVS:
13959 * tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
13960 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
13961 (test_suburi_error_wrongproto), (test_missing_primary_decoder):
13962 * tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
13963 (cddabasesrc_suite):
13964 Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
13965 deprecated in the future (see #498924).
13967 2007-12-21 22:26:47 +0000 Tim-Philipp Müller <tim@centricular.net>
13969 gst/playback/gststreamselector.c: Don't leak event.
13970 Original commit message from CVS:
13971 * gst/playback/gststreamselector.c: (gst_selector_pad_event):
13974 2007-12-20 19:43:25 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
13976 gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
13977 Original commit message from CVS:
13978 * gst-libs/gst/riff/riff-read.c:
13979 Use GST_ROUND_UP_2 macro
13981 2007-12-20 17:13:37 +0000 Tim-Philipp Müller <tim@centricular.net>
13983 gst/playback/.cvsignore: Ignore more.
13984 Original commit message from CVS:
13985 * gst/playback/.cvsignore:
13988 2007-12-20 10:41:29 +0000 Tim-Philipp Müller <tim@centricular.net>
13990 Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
13991 Original commit message from CVS:
13992 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
13993 * gst/playback/gstplaybasebin.c: (set_subtitles_visible),
13994 (set_active_source):
13995 * gst/playback/gstplaybasebin.h:
13996 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
13997 (setup_sinks), (playbin_set_subtitles_visible):
13998 Make switching off of subtitles work. To avoid all kind of
13999 problems with unlinking of the subtitle input, we just keep
14000 the subtitle inputs linked as they are and tell textoverlay
14001 not to render them. Fixes #373011.
14002 Other subtitle switching issues (esp. when there are both
14003 external and in-stream subtitles) remain. They'll be solved
14006 2007-12-18 16:21:35 +0000 Wim Taymans <wim.taymans@gmail.com>
14008 gst/playback/gststreamselector.c: Init the pad segment too.
14009 Original commit message from CVS:
14010 * gst/playback/gststreamselector.c: (gst_selector_pad_init):
14011 Init the pad segment too.
14013 2007-12-18 15:56:51 +0000 Wim Taymans <wim.taymans@gmail.com>
14015 gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
14016 Original commit message from CVS:
14017 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
14018 (gst_audioringbuffer_open_device),
14019 (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
14020 (gst_audioringbuffer_release), (gst_audioringbuffer_start),
14021 (gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
14022 (gst_audio_sink_create_ringbuffer):
14023 Improve debug output.
14024 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
14025 (gst_ring_buffer_pause), (gst_ring_buffer_delay):
14026 Prevent some functions from doing things and failing when the
14027 ringbuffer is not yet acquired.
14029 2007-12-18 15:32:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14031 gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
14032 Original commit message from CVS:
14033 * gst-libs/gst/interfaces/interfaces.h:
14034 Also remove interfaces.h from CVS as it is not needed anymore.
14036 2007-12-18 15:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14038 gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
14039 Original commit message from CVS:
14040 * gst-libs/gst/interfaces/Makefile.am:
14041 interfaces.h is not used anymore so remove it from the build
14044 2007-12-18 01:01:23 +0000 David Schleef <ds@schleef.org>
14046 gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
14047 Original commit message from CVS:
14048 * gst/videotestsrc/gstvideotestsrc.c:
14049 * gst/videotestsrc/gstvideotestsrc.h:
14050 Add a "blink" pattern. Turn on the pain. Apologies. It's useful
14051 for testing vertical refresh synchronization.
14053 2007-12-18 00:13:26 +0000 David Schleef <ds@schleef.org>
14055 Add new GstVideFormat enum and write a bunch of helper functions based around it.
14056 Original commit message from CVS:
14057 * docs/libs/gst-plugins-base-libs-sections.txt:
14058 * gst-libs/gst/video/video.c:
14059 * gst-libs/gst/video/video.h:
14060 Add new GstVideFormat enum and write a bunch of helper functions
14063 2007-12-17 23:41:14 +0000 Tim-Philipp Müller <tim@centricular.net>
14065 Makefile.am: Use new common/win32.mak.
14066 Original commit message from CVS:
14068 Use new common/win32.mak.
14070 2007-12-17 16:44:51 +0000 Wim Taymans <wim.taymans@gmail.com>
14072 gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
14073 Original commit message from CVS:
14074 * gst-libs/gst/audio/gstbaseaudiosrc.c:
14075 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
14077 When going from PLAYING to PAUSED, pause the ringbuffer before calling
14078 the parent state change function, just like the audiosink, because the
14079 parent waits for the element to finish its processing before completing
14080 the state change. This makes going to PAUSED a lot snappier.
14081 When going from READY to PAUSED, don't allow the ringbuffer to start
14084 2007-12-17 00:01:00 +0000 Edward Hervey <bilboed@bilboed.com>
14086 gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
14087 Original commit message from CVS:
14088 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
14089 Yet another fix for broken software that produce files with an empty
14090 blockalign field. Instead of completely failing, make a second attempt
14091 at guessing the width/depth by looking at strf->size.
14093 2007-12-16 23:52:58 +0000 Tim-Philipp Müller <tim@centricular.net>
14095 gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
14096 Original commit message from CVS:
14097 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
14098 (gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
14099 * gst-libs/gst/pbutils/install-plugins.c:
14100 (gst_install_plugins_spawn_child), (gst_install_plugins_supported):
14101 * gst-libs/gst/pbutils/missing-plugins.c:
14102 (gst_missing_plugin_message_get_installer_detail),
14103 (gst_missing_encoder_installer_detail_new):
14104 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
14105 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
14106 Turn a few g_assert_not_reached() into g_return_val_if_reached() to
14107 avoid compiler warnings (#503930).
14109 2007-12-16 23:46:16 +0000 Edward Hervey <bilboed@bilboed.com>
14111 gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
14112 Original commit message from CVS:
14113 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
14114 Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
14115 for jpeg video streams.
14116 Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
14117 for the above modification.
14119 2007-12-15 17:27:48 +0000 Tim-Philipp Müller <tim@centricular.net>
14121 gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
14122 Original commit message from CVS:
14123 * gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
14124 (gst_x_overlay_handle_events):
14125 More guards (we don't want klass to end up being NULL).
14127 2007-12-15 03:40:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14129 Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
14130 Original commit message from CVS:
14132 * gst/volume/gstvolume.c: (gst_volume_init):
14133 Use new gst_base_transform_set_gap_aware() function as volume
14134 correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
14137 2007-12-14 19:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
14139 tests/examples/seek/seek.c: Don't go to READY on EOS as this avoids testing of seeking and restarting after EOS, use ...
14140 Original commit message from CVS:
14141 * tests/examples/seek/seek.c: (msg_segment_done), (main):
14142 Don't go to READY on EOS as this avoids testing of seeking and
14143 restarting after EOS, use the stop button when you want to READY.
14144 Don't try to do a flushing seek in segment-done, it does not make
14145 sense to use this for gapless playback and is not needed.
14147 2007-12-14 18:46:12 +0000 Wim Taymans <wim.taymans@gmail.com>
14149 gst/playback/gstqueue2.c: Use separate timers for input and output rates.
14150 Original commit message from CVS:
14151 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
14152 (reset_rate_timer), (update_in_rates), (update_out_rates),
14153 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
14154 (gst_queue_chain), (gst_queue_loop):
14155 Use separate timers for input and output rates.
14156 Pause measuring the output rate when we block for more data.
14159 2007-12-14 16:23:06 +0000 Christian Schaller <uraeus@gnome.org>
14161 * gst/speexresample/Makefile.am:
14162 update spec file and add two missing files for disting
14163 Original commit message from CVS:
14164 update spec file and add two missing files for disting
14166 2007-12-14 09:24:55 +0000 Wim Taymans <wim.taymans@gmail.com>
14168 gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
14169 Original commit message from CVS:
14170 * gst/playback/gstqueue2.c: (gst_queue_chain):
14171 Pause the timer to measure the input rate when we block because the
14172 queue is filled. See #503262.
14174 2007-12-13 15:54:00 +0000 Peter Kjellerstedt <pkj@axis.com>
14176 gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
14177 Original commit message from CVS:
14178 Patch by: Peter Kjellerstedt <pkj at axis com>
14179 * gst-libs/gst/rtsp/gstrtspconnection.c:
14180 (gst_rtsp_connection_free):
14181 Close control sockets. Fixes #503440.
14183 2007-12-13 12:31:38 +0000 Wim Taymans <wim.taymans@gmail.com>
14185 gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
14186 Original commit message from CVS:
14187 * gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
14188 Expose the right pad in the right place with the right element.
14190 2007-12-13 11:40:10 +0000 Tim-Philipp Müller <tim@centricular.net>
14192 gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
14193 Original commit message from CVS:
14194 * gst-libs/gst/pbutils/descriptions.c: (formats):
14195 Add description for 'private' dts caps (who come up with that name?).
14197 2007-12-13 10:10:35 +0000 Tim-Philipp Müller <tim@centricular.net>
14199 Makefile.am: Add check-exports target and run it with 'make check'.
14200 Original commit message from CVS:
14202 Add check-exports target and run it with 'make check'.
14204 Be stricter about what we export in our libraries: change regexp so that
14205 we only export _gst_foo(), but not __gst_foo().
14206 * gst-libs/gst/cdda/base64.h: (rfc822_binary):
14207 * gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
14208 Change internal functions to __gst_foo so they dont' get exported.
14209 * win32/common/libgstaudio.def:
14210 Add missing symbols.
14212 2007-12-11 21:18:57 +0000 David Schleef <ds@schleef.org>
14215 ChangeLog: remove conflict markers
14216 Original commit message from CVS:
14217 ChangeLog: remove conflict markers
14219 2007-12-11 17:14:13 +0000 Tim-Philipp Müller <tim@centricular.net>
14221 ext/gnomevfs/: Use gst_tag_freeform_string_to_utf8() here, which also takes into account any character sets specified...
14222 Original commit message from CVS:
14223 * ext/gnomevfs/Makefile.am:
14224 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
14225 Use gst_tag_freeform_string_to_utf8() here, which also takes
14226 into account any character sets specified by the user via
14227 environment variables.
14229 2007-12-10 15:21:41 +0000 Wim Taymans <wim.taymans@gmail.com>
14231 gst/audioconvert/Makefile.am: Also link to libm.
14232 Original commit message from CVS:
14233 * gst/audioconvert/Makefile.am:
14236 2007-12-10 15:13:55 +0000 Wim Taymans <wim.taymans@gmail.com>
14238 gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
14239 Original commit message from CVS:
14240 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
14241 No need for floating point operations here. avoids having to link
14242 against the math library too.
14244 2007-12-10 11:16:25 +0000 Tim-Philipp Müller <tim@centricular.net>
14246 Add one or two missing formats. Generate ADPCM description dynamically depending on layout/format.
14247 Original commit message from CVS:
14248 * gst-libs/gst/pbutils/descriptions.c: (formats),
14249 (format_info_get_desc):
14250 * tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
14252 Add one or two missing formats. Generate ADPCM description
14253 dynamically depending on layout/format.
14255 2007-12-09 04:28:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14257 configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
14258 Original commit message from CVS:
14260 Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
14262 2007-12-08 18:38:39 +0000 Robin Stocker <robin.stocker@gmx.ch>
14264 gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
14265 Original commit message from CVS:
14266 Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
14267 * gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
14268 Some .srt files start with chunk number 0 and not chunk number 1,
14269 recognise and accept those as well (fixes #502497).
14270 * tests/check/elements/subparse.c: (srt_input), (srt_input0),
14272 Add unit test for the above.
14274 2007-12-06 12:08:21 +0000 Wim Taymans <wim.taymans@gmail.com>
14276 gst/playback/gstplay-enum.*: Add missing files.
14277 Original commit message from CVS:
14278 * gst/playback/gstplay-enum.c:
14279 (register_gst_autoplug_select_result),
14280 (gst_autoplug_select_result_get_type):
14281 * gst/playback/gstplay-enum.h:
14284 2007-12-05 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
14286 gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
14287 Original commit message from CVS:
14288 * gst/playback/Makefile.am:
14289 Group decodebin2 and uridecodebin into the same plugin so that they
14290 can share the GEnumType.
14291 * gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
14292 (_gst_select_accumulator), (gst_decode_bin_class_init),
14293 (gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
14294 (gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
14295 (analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
14296 Add signal to sort factories instead of the more awkward autoplug-select
14298 Modify autoplug_select so that we can try, skip or expose the
14299 autopluggin of an element on a pad.
14300 * gst/playback/gstfactorylists.c: (compare_ranks),
14301 (decoders_filter), (sinks_filter), (gst_factory_list_is_type),
14302 (element_filter), (gst_factory_list_get_elements),
14303 (gst_factory_list_debug), (gst_factory_list_filter):
14304 * gst/playback/gstfactorylists.h:
14305 Simplify the API, allow getting elements based on mask.
14306 * gst/playback/gstplay-marshal.list:
14307 Add some more marshallers.
14308 * gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
14309 (gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
14310 (autoplug_select_cb), (activate_group):
14311 Add support for managing non-raw sinks by providing a custom element and
14312 sink list to decodebin2.
14313 Try to plug non-raw sinks when decodebin2 using autoplug-select of
14315 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
14316 (gst_play_sink_set_mode), (gst_play_sink_request_pad):
14317 * gst/playback/gstplaysink.h:
14318 Add support for raw and non-raw sinks.
14319 Add support to force sinks selected by playbin2.
14320 Don't plug raw converters for non-raw sinks.
14321 * gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
14322 (_gst_select_accumulator), (gst_uri_decode_bin_class_init),
14323 (proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
14325 Use right accumulators.
14328 2007-12-03 13:47:00 +0000 Wim Taymans <wim.taymans@gmail.com>
14330 gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
14331 Original commit message from CVS:
14332 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
14333 Use runnning time as the base time instead of the timestamp.
14334 Spotted by Saur on IRC.
14336 2007-12-03 11:32:30 +0000 Edward Hervey <bilboed@bilboed.com>
14338 gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
14339 Original commit message from CVS:
14340 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
14341 Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
14343 2007-12-03 10:58:14 +0000 Wim Taymans <wim.taymans@gmail.com>
14345 ext/ogg/gstoggdemux.c: If we find a new serial number but it does not contain a BOS page, make sure we initialize the...
14346 Original commit message from CVS:
14347 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
14348 (gst_ogg_demux_read_chain):
14349 If we find a new serial number but it does not contain a BOS page, make
14350 sure we initialize the chain to NULL because else we will try to scan it
14351 and crash. Fixes #500763
14353 2007-11-30 17:47:15 +0000 Wim Taymans <wim.taymans@gmail.com>
14355 gst/playback/: Refactor some common code to filter factories and check caps compat.
14356 Original commit message from CVS:
14357 * gst/playback/Makefile.am:
14358 * gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
14359 (get_feature_array), (decoders_filter), (sinks_filter),
14360 (gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
14361 (gst_factory_list_filter):
14362 * gst/playback/gstfactorylists.h:
14363 Refactor some common code to filter factories and check caps compat.
14364 * gst/playback/gstdecodebin.c:
14365 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
14366 (gst_decode_bin_init), (gst_decode_bin_dispose),
14367 (gst_decode_bin_autoplug_continue),
14368 (gst_decode_bin_autoplug_factories),
14369 (gst_decode_bin_autoplug_select), (analyze_new_pad),
14370 (find_compatibles):
14371 * gst/playback/gstplaybin.c:
14372 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
14373 (gst_play_bin_init), (gst_play_bin_finalize),
14374 (autoplug_factories_cb), (activate_group):
14375 * gst/playback/gstqueue2.c:
14376 * gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
14377 (proxy_autoplug_continue_signal),
14378 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
14379 (proxy_drained_signal):
14380 Add some more debug info and use factor filtering code.
14382 2007-11-26 13:19:46 +0000 Julien Moutte <julien@moutte.net>
14384 configure.ac: Add QuickTime Wrapper plug-in.
14385 Original commit message from CVS:
14386 2007-11-26 Julien Moutte <julien@fluendo.com>
14387 * configure.ac: Add QuickTime Wrapper plug-in.
14388 * gst/speexresample/gstspeexresample.c:
14389 (gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
14390 build on Mac OS X Leopard. Incorrect printf format arguments.
14392 * sys/qtwrapper/Makefile.am:
14393 * sys/qtwrapper/audiodecoders.c:
14394 (qtwrapper_audio_decoder_base_init),
14395 (qtwrapper_audio_decoder_class_init),
14396 (qtwrapper_audio_decoder_init),
14397 (clear_AudioStreamBasicDescription), (fill_indesc_mp3),
14398 (fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
14399 (make_samr_magic_cookie), (open_decoder),
14400 (qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
14401 (qtwrapper_audio_decoder_chain),
14402 (qtwrapper_audio_decoder_sink_event),
14403 (qtwrapper_audio_decoders_register):
14404 * sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
14406 * sys/qtwrapper/codecmapping.h:
14407 * sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
14408 (image_description_for_mp4v), (image_description_from_stsd_buffer),
14409 (image_description_from_codec_data):
14410 * sys/qtwrapper/imagedescription.h:
14411 * sys/qtwrapper/qtutils.c: (get_name_info_from_component),
14412 (get_output_info_from_component), (dump_avcc_atom),
14413 (dump_image_description), (dump_codec_decompress_params),
14414 (addSInt32ToDictionary), (dump_cvpixel_buffer),
14415 (DestroyAudioBufferList), (AllocateAudioBufferList):
14416 * sys/qtwrapper/qtutils.h:
14417 * sys/qtwrapper/qtwrapper.c: (plugin_init):
14418 * sys/qtwrapper/qtwrapper.h:
14419 * sys/qtwrapper/videodecoders.c:
14420 (qtwrapper_video_decoder_base_init),
14421 (qtwrapper_video_decoder_class_init),
14422 (qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
14423 (fill_image_description), (new_image_description), (close_decoder),
14424 (open_decoder), (qtwrapper_video_decoder_sink_setcaps),
14425 (decompressCb), (qtwrapper_video_decoder_chain),
14426 (qtwrapper_video_decoder_sink_event),
14427 (qtwrapper_video_decoders_register): Initial import of QuickTime
14428 wrapper jointly developped by Songbird authors (Pioneers of the
14429 Inevitable) and Fluendo.
14431 2007-11-26 12:25:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14433 gst/: Add GAP-flag support.
14434 Original commit message from CVS:
14435 * gst/audiotestsrc/gstaudiotestsrc.c:
14436 * gst/volume/gstvolume.c:
14437 * gst/volume/gstvolume.h:
14438 Add GAP-flag support.
14440 2007-11-26 08:43:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14442 gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
14443 Original commit message from CVS:
14444 * gst/speexresample/README:
14445 * gst/speexresample/arch.h:
14446 * gst/speexresample/resample.c: (resampler_basic_direct_single),
14447 (resampler_basic_direct_double),
14448 (resampler_basic_interpolate_single),
14449 (resampler_basic_interpolate_double),
14450 (speex_resampler_process_native), (speex_resampler_process_float),
14451 (speex_resampler_process_int),
14452 (speex_resampler_process_interleaved_float),
14453 (speex_resampler_process_interleaved_int),
14454 (speex_resampler_get_input_latency),
14455 (speex_resampler_get_output_latency):
14456 * gst/speexresample/speex_resampler.h:
14457 Update speex resampler to latest SVN. We're now down to only the
14458 changes noted in README again.
14459 * gst/speexresample/speex_resampler_wrapper.h:
14460 * gst/speexresample/gstspeexresample.c:
14461 (gst_speex_resample_push_drain), (gst_speex_resample_query):
14462 Adjust to API changes.
14464 2007-11-24 15:02:01 +0000 Julien Moutte <julien@moutte.net>
14466 tests/examples/seek/seek.c: Increase the range of the rate selector as I would like to test QOS behavior at higher fo...
14467 Original commit message from CVS:
14468 2007-11-24 Julien MOUTTE <julien@moutte.net>
14469 * tests/examples/seek/seek.c: (main): Increase the range of the
14470 rate selector as I would like to test QOS behavior at higher
14471 forward and reverse playback speed like say 64x.
14473 2007-11-23 10:21:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14475 gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
14476 Original commit message from CVS:
14477 * gst/speexresample/gstspeexresample.c:
14478 (gst_speex_resample_update_state):
14479 Only post the latency message if we have a resampler state already.
14481 2007-11-23 10:21:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14483 gst/audioresample/gstaudioresample.c: Implement latency query.
14484 Original commit message from CVS:
14485 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
14486 (audioresample_query), (audioresample_query_type),
14487 (gst_audioresample_set_property):
14488 Implement latency query.
14490 2007-11-23 10:01:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14492 gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
14493 Original commit message from CVS:
14494 * gst/speexresample/gstspeexresample.c:
14495 (gst_speex_resample_update_state):
14496 Also post GST_MESSAGE_LATENCY if the latency changes.
14498 2007-11-23 08:48:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14500 gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
14501 Original commit message from CVS:
14502 * gst/speexresample/resample.c: (speex_resampler_get_latency),
14503 (speex_resampler_drain_float), (speex_resampler_drain_int),
14504 (speex_resampler_drain_interleaved_float),
14505 (speex_resampler_drain_interleaved_int):
14506 * gst/speexresample/speex_resampler.h:
14507 * gst/speexresample/speex_resampler_wrapper.h:
14508 Add functions to push the remaining samples and to get the latency
14509 of the resampler. These will get added to Speex SVN in this or a
14510 slightly changed form at some point too and should get merged then
14512 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
14513 (gst_speex_resample_init_state),
14514 (gst_speex_resample_transform_size),
14515 (gst_speex_resample_push_drain), (gst_speex_resample_event),
14516 (gst_speex_fix_output_buffer), (gst_speex_resample_process),
14517 (gst_speex_resample_query), (gst_speex_resample_query_type):
14518 Drop the prepending zeroes and output the remaining samples on EOS.
14519 Also properly implement the latency query for this. speexresample
14520 should be completely ready for production use now.
14522 2007-11-21 18:02:21 +0000 Wim Taymans <wim.taymans@gmail.com>
14524 gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
14525 Original commit message from CVS:
14526 * gst-libs/gst/audio/gstbaseaudiosink.c:
14527 (gst_base_audio_sink_drain):
14528 Our EOS time contains the base_time, _wait_eos() expects a running_time
14529 so we have to subtract the base_time again before calling the function.
14530 This fixes an EOS regression where the base_time was added twice and EOS
14531 took longer and longer in certain situations.
14534 2007-11-21 13:04:17 +0000 Wim Taymans <wim.taymans@gmail.com>
14536 Expose methods for some object properties so that subclasses can more easily configure them.
14537 Original commit message from CVS:
14538 * docs/libs/gst-plugins-base-libs-sections.txt:
14539 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
14540 (gst_base_audio_sink_set_provide_clock),
14541 (gst_base_audio_sink_get_provide_clock),
14542 (gst_base_audio_sink_set_slave_method),
14543 (gst_base_audio_sink_get_slave_method),
14544 (gst_base_audio_sink_set_property),
14545 (gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
14546 (gst_base_audio_sink_none_slaving),
14547 (gst_base_audio_sink_handle_slaving):
14548 * gst-libs/gst/audio/gstbaseaudiosink.h:
14549 Expose methods for some object properties so that subclasses can more
14550 easily configure them.
14551 Added slave method none, that completely disables slaving to the
14553 API: gst_base_audio_sink_set_provide_clock()
14554 API: gst_base_audio_sink_get_provide_clock()
14555 API: gst_base_audio_sink_set_slave_method()
14556 API: gst_base_audio_sink_get_slave_method()
14557 * gst-libs/gst/audio/gstbaseaudiosrc.c:
14558 (gst_base_audio_src_set_provide_clock),
14559 (gst_base_audio_src_get_provide_clock),
14560 (gst_base_audio_src_set_property),
14561 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
14562 * gst-libs/gst/audio/gstbaseaudiosrc.h:
14563 Expose methods for some object properties so that subclasses can more
14564 easily configure them.
14565 API: gst_base_audio_src_set_provide_clock()
14566 API: gst_base_audio_src_get_provide_clock()
14568 2007-11-21 10:18:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14570 gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
14571 Original commit message from CVS:
14572 * gst/speexresample/README:
14573 Add README explaining where the resampling code was taken from
14574 and which changes were done.
14575 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
14577 Use g_malloc() and friends instead of malloc() to achieve higher
14578 portability and define the functions inline.
14579 * gst/speexresample/speex_resampler.h:
14580 Add back some useless preprocessor stuff to keep the diff between
14581 our version and the one from the Speex SVN repository lower.
14583 2007-11-20 20:23:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14585 gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
14586 Original commit message from CVS:
14587 * gst/speexresample/gstspeexresample.c:
14588 (gst_speex_fix_output_buffer), (gst_speex_resample_transform):
14589 Some small cleanup and addition of a TODO item.
14591 2007-11-20 12:56:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14593 gst/speexresample/Makefile.am: Add missing file.
14594 Original commit message from CVS:
14595 * gst/speexresample/Makefile.am:
14598 2007-11-20 07:53:56 +0000 Joe Peterson <lavajoe@gentoo.org>
14600 gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
14601 Original commit message from CVS:
14602 Patch by: Joe Peterson <lavajoe at gentoo dot org>
14603 * gst-libs/gst/sdp/gstsdpmessage.c:
14604 Fix compilation on FreeBSD (Gentoo). Fixes #498228.
14606 2007-11-20 07:47:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14608 Add speexresample to the docs and while at that do a make update.
14609 Original commit message from CVS:
14610 * docs/plugins/Makefile.am:
14611 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
14612 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
14613 * docs/plugins/gst-plugins-bad-plugins.args:
14614 * docs/plugins/gst-plugins-bad-plugins.signals:
14615 * docs/plugins/inspect/plugin-bz2.xml:
14616 * docs/plugins/inspect/plugin-cdxaparse.xml:
14617 * docs/plugins/inspect/plugin-dtsdec.xml:
14618 * docs/plugins/inspect/plugin-equalizer.xml:
14619 * docs/plugins/inspect/plugin-faac.xml:
14620 * docs/plugins/inspect/plugin-faad.xml:
14621 * docs/plugins/inspect/plugin-filter.xml:
14622 * docs/plugins/inspect/plugin-freeze.xml:
14623 * docs/plugins/inspect/plugin-gio.xml:
14624 * docs/plugins/inspect/plugin-gsm.xml:
14625 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
14626 * docs/plugins/inspect/plugin-h264parse.xml:
14627 * docs/plugins/inspect/plugin-modplug.xml:
14628 * docs/plugins/inspect/plugin-mpeg2enc.xml:
14629 * docs/plugins/inspect/plugin-musepack.xml:
14630 * docs/plugins/inspect/plugin-musicbrainz.xml:
14631 * docs/plugins/inspect/plugin-nsfdec.xml:
14632 * docs/plugins/inspect/plugin-replaygain.xml:
14633 * docs/plugins/inspect/plugin-soundtouch.xml:
14634 * docs/plugins/inspect/plugin-spcdec.xml:
14635 * docs/plugins/inspect/plugin-spectrum.xml:
14636 * docs/plugins/inspect/plugin-speed.xml:
14637 * docs/plugins/inspect/plugin-tta.xml:
14638 * docs/plugins/inspect/plugin-videosignal.xml:
14639 * docs/plugins/inspect/plugin-xingheader.xml:
14640 * docs/plugins/inspect/plugin-xvid.xml:
14641 * gst/speexresample/gstspeexresample.h:
14642 Add speexresample to the docs and while at that do a make update.
14644 2007-11-20 07:30:30 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14646 gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
14647 Original commit message from CVS:
14648 * gst/speexresample/gstspeexresample.c:
14649 (gst_speex_fix_output_buffer), (gst_speex_resample_process):
14650 If the resampler gives less output samples than expected
14651 adjust the output buffer and print a warning.
14653 2007-11-20 07:02:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14655 Add resample element based on the Speex resampling algorithm.
14656 Original commit message from CVS:
14658 * gst/speexresample/arch.h:
14659 * gst/speexresample/fixed_generic.h:
14660 * gst/speexresample/gstspeexresample.c:
14661 (gst_speex_resample_base_init), (gst_speex_resample_class_init),
14662 (gst_speex_resample_init), (gst_speex_resample_start),
14663 (gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
14664 (gst_speex_resample_transform_caps),
14665 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
14666 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
14667 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
14668 (gst_speex_resample_event), (gst_speex_resample_check_discont),
14669 (gst_speex_resample_process), (gst_speex_resample_transform),
14670 (gst_speex_resample_set_property),
14671 (gst_speex_resample_get_property), (plugin_init):
14672 * gst/speexresample/gstspeexresample.h:
14673 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
14674 (speex_free), (compute_func), (main), (sinc), (cubic_coef),
14675 (resampler_basic_direct_single), (resampler_basic_direct_double),
14676 (resampler_basic_interpolate_single),
14677 (resampler_basic_interpolate_double), (update_filter),
14678 (speex_resampler_init), (speex_resampler_init_frac),
14679 (speex_resampler_destroy), (speex_resampler_process_native),
14680 (speex_resampler_process_float), (speex_resampler_process_int),
14681 (speex_resampler_process_interleaved_float),
14682 (speex_resampler_process_interleaved_int),
14683 (speex_resampler_set_rate), (speex_resampler_get_rate),
14684 (speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
14685 (speex_resampler_set_quality), (speex_resampler_get_quality),
14686 (speex_resampler_set_input_stride),
14687 (speex_resampler_get_input_stride),
14688 (speex_resampler_set_output_stride),
14689 (speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
14690 (speex_resampler_reset_mem), (speex_resampler_strerror):
14691 * gst/speexresample/speex_resampler.h:
14692 * gst/speexresample/speex_resampler_float.c:
14693 * gst/speexresample/speex_resampler_int.c:
14694 * gst/speexresample/speex_resampler_wrapper.h:
14695 Add resample element based on the Speex resampling algorithm.
14697 2007-11-19 12:30:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14699 tests/check/libs/fft.c: Fix scaling to really have dB instead of something else.
14700 Original commit message from CVS:
14701 * tests/check/libs/fft.c: (GST_START_TEST):
14702 Fix scaling to really have dB instead of something else.
14704 2007-11-19 12:08:16 +0000 Julien Moutte <julien@moutte.net>
14706 tests/examples/seek/seek.c: There's a nice macro to check
14707 Original commit message from CVS:
14708 2007-11-19 Julien MOUTTE <julien@moutte.net>
14709 * tests/examples/seek/seek.c: (main): There's a nice macro to
14711 GTK version, use it.
14713 2007-11-19 11:59:20 +0000 Julien Moutte <julien@moutte.net>
14715 tests/examples/seek/seek.c: Try to support stable version of GTK.
14716 Original commit message from CVS:
14717 2007-11-19 Julien MOUTTE <julien@moutte.net>
14718 * tests/examples/seek/seek.c: (main): Try to support stable version
14721 2007-11-17 15:25:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14723 gst/playback/: Fix the build + little README update.
14724 Original commit message from CVS:
14725 * gst/playback/README:
14726 * gst/playback/test7.c:
14727 Fix the build + little README update.
14729 2007-11-16 16:02:45 +0000 Wim Taymans <wim.taymans@gmail.com>
14731 tests/examples/seek/seek.c: Add playbin2 seek pipeline.
14732 Original commit message from CVS:
14733 * tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
14734 Add playbin2 seek pipeline.
14736 2007-11-16 15:44:48 +0000 Wim Taymans <wim.taymans@gmail.com>
14738 gst/playback/: Add playbin2.
14739 Original commit message from CVS:
14740 * gst/playback/Makefile.am:
14741 * gst/playback/gstplayback.c: (plugin_init):
14742 * gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
14743 (eos_cb), (about_to_finish_cb), (main):
14745 Added gapless playback example.
14746 * gst/playback/gstplaybasebin.c:
14747 * gst/playback/gstplaybasebin.h:
14748 * gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
14749 * gst/playback/gstqueue2.c:
14750 * gst/playback/test.c:
14751 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
14753 * gst/playback/gststreaminfo.h:
14755 * gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
14756 (gst_play_bin_class_init), (init_group), (gst_play_bin_init),
14757 (gst_play_bin_dispose), (gst_play_bin_set_uri),
14758 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
14759 (gst_play_bin_get_property), (gst_play_bin_handle_message),
14760 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
14761 (drained_cb), (unlink_group), (activate_group),
14762 (setup_next_source), (gst_play_bin_change_state),
14763 (gst_play_bin2_plugin_init):
14764 Added raw first version of playbin2. Does chained oggs and gapless
14765 playback fine. No support for raw sinks yet. No visualisations or
14767 * gst/playback/gstplaysink.c: (gst_play_sink_get_type),
14768 (gst_play_sink_class_init), (gst_play_sink_init),
14769 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
14770 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
14771 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
14772 (gst_play_sink_set_property), (gst_play_sink_get_property),
14773 (post_missing_element_message), (free_chain), (add_chain),
14774 (activate_chain), (gen_video_chain), (gen_text_element),
14775 (gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
14776 (gst_play_sink_set_mode), (gst_play_sink_request_pad),
14777 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
14778 (gst_play_sink_send_event), (gst_play_sink_change_state):
14779 * gst/playback/gstplaysink.h:
14780 Added Element that abstracts the sinks and their pipelines for playbin2.
14782 2007-11-16 15:05:07 +0000 Wim Taymans <wim.taymans@gmail.com>
14784 gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
14785 Original commit message from CVS:
14786 * gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
14787 (gst_selector_pad_class_init), (gst_selector_pad_init),
14788 (gst_selector_pad_finalize), (gst_selector_pad_reset),
14789 (gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
14790 (gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
14791 (gst_selector_pad_chain), (gst_stream_selector_get_type),
14792 (gst_stream_selector_base_init), (gst_stream_selector_class_init),
14793 (gst_stream_selector_init), (gst_stream_selector_set_property),
14794 (gst_stream_selector_get_linked_pad),
14795 (gst_stream_selector_getcaps),
14796 (gst_stream_selector_is_active_sinkpad),
14797 (gst_stream_selector_activate_sinkpad),
14798 (gst_stream_selector_get_linked_pads),
14799 (gst_stream_selector_request_new_pad),
14800 (gst_stream_selector_release_pad):
14801 * gst/playback/gststreamselector.h:
14802 Improve streamselector, make it select and unselect the current pad more
14804 Subclass GstPad for the sinkpads of the selector.
14805 Handle segments more correctly.
14806 Fix caps negotiation.
14807 Implement release_pad.
14809 2007-11-16 12:51:44 +0000 Wim Taymans <wim.taymans@gmail.com>
14811 gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
14812 Original commit message from CVS:
14813 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
14814 (gst_decode_group_check_if_drained), (source_pad_event_probe),
14816 Add drained signal fired when decodebin finishes decoding the data.
14817 Remove deprecated STATE_DIRTY message.
14818 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
14819 (unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
14820 (analyse_source), (proxy_drained_signal), (make_decoder),
14821 (source_new_pad), (value_list_append_structure_list),
14822 (handle_redirect_message), (handle_message):
14823 Proxy the new drained signal.
14824 Handle pad removed from decodebin.
14825 Handle redirect messages by sorting multiple redirections based on the
14828 2007-11-16 11:22:09 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
14830 gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
14831 Original commit message from CVS:
14832 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
14833 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
14834 Fix leaking headers. Fixes #496761.
14836 2007-11-16 11:16:58 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
14838 sys/: Don't leak the PAR on errors. Fixes #496731.
14839 Original commit message from CVS:
14840 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
14841 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
14842 (gst_ximagesink_change_state):
14843 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
14844 Don't leak the PAR on errors. Fixes #496731.
14846 2007-11-16 10:14:34 +0000 Tim-Philipp Müller <tim@centricular.net>
14848 gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
14849 Original commit message from CVS:
14850 * gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
14851 (gst_tag_from_id3_user_tag):
14852 Add mapping for audio cd discid tags, so we can extract
14853 them from tags as well (see #347848). Also compare identifiers
14854 in ID3v2 TXXX frames in a case-insensitive way to increase
14855 compatibility when reading tags (discid vs. DiscID vs. DiscId).
14857 2007-11-16 01:21:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14859 gst-plugins-base.doap: Oops, fix the release name.
14860 Original commit message from CVS:
14861 * gst-plugins-base.doap:
14862 Oops, fix the release name.
14864 2007-11-16 00:44:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14866 gst-plugins-base.doap: Add 0.10.15 release
14867 Original commit message from CVS:
14868 * gst-plugins-base.doap:
14869 Add 0.10.15 release
14871 2007-11-16 00:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14873 configure.ac: Back to CVS
14874 Original commit message from CVS:
14878 === release 0.10.15 ===
14880 2007-11-16 00:14:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14882 configure.ac: releasing 0.10.15, "No need to argue"
14883 Original commit message from CVS:
14884 === release 0.10.15 ===
14885 2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
14887 releasing 0.10.15, "No need to argue"
14889 2007-11-16 00:04:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14914 Original commit message from CVS:
14917 2007-11-15 21:40:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14919 win32/vs6/libgstfft.dsp: Convert line endings to DOS.
14920 Original commit message from CVS:
14921 * win32/vs6/libgstfft.dsp:
14922 Convert line endings to DOS.
14924 2007-11-15 21:14:04 +0000 Sébastien Moutte <sebastien@moutte.net>
14926 win32/: Add a project file for fft plugin and remove socket based plugin which don't build from the workspace.* win32...
14927 Original commit message from CVS:
14928 * win32/vs6/gst_plugins_base.dsw:
14929 * win32/vs6/libgstfft.dsp:
14931 Add a project file for fft plugin and remove socket
14932 based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
14933 * win32/vs6/libgstrtp.dsp:
14934 * win32/vs6/libgsttag.dsp:
14935 Convert line endings back to DOS.
14938 2007-11-14 12:27:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14940 win32/vs6/: Convert line endings back to DOS
14941 Original commit message from CVS:
14942 * win32/vs6/libgstinterfaces.dsp:
14943 * win32/vs6/libgstrtsp.dsp:
14944 Convert line endings back to DOS
14946 2007-11-14 11:08:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14948 gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
14949 Original commit message from CVS:
14950 * gst-libs/gst/fft/kiss_fft_f32.h:
14951 * gst-libs/gst/fft/kiss_fft_f64.h:
14952 * gst-libs/gst/fft/kiss_fft_s16.h:
14953 * gst-libs/gst/fft/kiss_fft_s32.h:
14954 Don't include malloc.h which doesn't exist on Mac OSX.
14955 Instead, pull in glib.h and use g_malloc/g_free for
14956 consistency. Fixes: #496548
14958 2007-11-09 15:54:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14960 gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
14961 Original commit message from CVS:
14962 * gst/playback/gstdecodebin2.c:
14963 Dont leak ghostpad. Fixes #475451.
14965 2007-11-09 12:21:52 +0000 Wim Taymans <wim.taymans@gmail.com>
14967 Update some more docs and comments.
14968 Original commit message from CVS:
14969 * docs/design/design-decodebin.txt:
14970 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
14971 Update some more docs and comments.
14973 2007-11-07 16:47:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14975 Require GIO >= 0.1.2 and adjust unit test for an API change.
14976 Original commit message from CVS:
14978 * tests/check/pipelines/gio.c: (GST_START_TEST):
14979 Require GIO >= 0.1.2 and adjust unit test for an API change.
14981 2007-11-07 15:18:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14983 ext/gio/gstgio.h: Add macro to check if a stream supports seeking.
14984 Original commit message from CVS:
14985 * ext/gio/gstgio.h:
14986 Add macro to check if a stream supports seeking.
14987 * ext/gio/Makefile.am:
14988 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
14989 (gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
14990 (gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
14991 (gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
14992 (gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
14993 (gst_gio_base_sink_render), (gst_gio_base_sink_query),
14994 (gst_gio_base_sink_set_stream):
14995 * ext/gio/gstgiobasesink.h:
14996 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
14997 (gst_gio_base_src_class_init), (gst_gio_base_src_init),
14998 (gst_gio_base_src_finalize), (gst_gio_base_src_start),
14999 (gst_gio_base_src_stop), (gst_gio_base_src_get_size),
15000 (gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
15001 (gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
15002 (gst_gio_base_src_create), (gst_gio_base_src_set_stream):
15003 * ext/gio/gstgiobasesrc.h:
15004 Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
15005 base classes that only require a GInputStream or GOutputStream to
15007 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
15008 (gst_gio_sink_class_init), (gst_gio_sink_init),
15009 (gst_gio_sink_finalize), (gst_gio_sink_start):
15010 * ext/gio/gstgiosink.h:
15011 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
15012 (gst_gio_src_class_init), (gst_gio_src_init),
15013 (gst_gio_src_finalize), (gst_gio_src_start):
15014 * ext/gio/gstgiosrc.h:
15015 Use the newly created base classes here.
15016 * ext/gio/gstgio.c: (plugin_init):
15017 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
15018 (gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
15019 (gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
15020 (gst_gio_stream_sink_get_property):
15021 * ext/gio/gstgiostreamsink.h:
15022 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
15023 (gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
15024 (gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
15025 (gst_gio_stream_src_get_property):
15026 * ext/gio/gstgiostreamsrc.h:
15027 Implement GstGioStreamSink and GstGioStreamSrc that have a property
15028 to set the GInputStream/GOutputStream that should be used.
15029 * tests/check/Makefile.am:
15030 * tests/check/pipelines/.cvsignore:
15031 * tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
15032 (gio_testsuite), (main):
15033 Add unit test for giostreamsrc and giostreamsink.
15035 2007-11-07 11:48:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15037 ext/gio/gstgio.c: Remove nowadays unnecessary workaround for a crash.
15038 Original commit message from CVS:
15039 * ext/gio/gstgio.c: (plugin_init):
15040 Remove nowadays unnecessary workaround for a crash.
15041 * ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
15042 (gst_gio_sink_start), (gst_gio_sink_stop),
15043 (gst_gio_sink_unlock_stop):
15044 * ext/gio/gstgiosink.h:
15045 * ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
15046 (gst_gio_src_stop), (gst_gio_src_unlock_stop):
15047 * ext/gio/gstgiosrc.h:
15048 Make the finalize function safer, clean up everything that could stay
15050 Reset the cancellable instead of creating a new one after cancelling
15052 Don't store the GFile in the element, it's only necessary for creating
15055 2007-11-06 23:35:39 +0000 Sebastien Moutte <sebastien@moutte.net>
15057 gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
15058 Original commit message from CVS:
15059 Patch by: Sebastien Moutte <sebastien moutte net>
15060 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
15061 (gst_rtcp_unix_to_ntp):
15062 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
15063 Fix some C99-isms and and a missing function that some versions of
15064 MSVC don't like too much (#494346).
15065 * win32/vs6/gst_plugins_base.dsw:
15066 * win32/vs6/libgstaudio.dsp:
15067 * win32/vs6/libgstrtp.dsp:
15068 * win32/vs6/libgsttag.dsp:
15069 Update vs6 projects files (#494346).
15071 2007-11-06 16:38:49 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15073 win32/common/: More missing symbols to export (fixes #493986).
15074 Original commit message from CVS:
15075 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15076 * win32/common/libgstaudio.def:
15077 * win32/common/libgstcdda.def:
15078 * win32/common/libgstinterfaces.def:
15079 * win32/common/libgstnetbuffer.def:
15080 * win32/common/libgstpbutils.def:
15081 * win32/common/libgstrtp.def:
15082 * win32/common/libgstrtsp.def:
15083 * win32/common/libgsttag.def:
15084 * win32/common/libgstvideo.def:
15085 More missing symbols to export (fixes #493986).
15087 2007-11-06 11:58:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15089 Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
15090 Original commit message from CVS:
15091 * docs/libs/gst-plugins-base-libs-sections.txt:
15092 * gst-libs/gst/fft/gstfftf32.c:
15093 * gst-libs/gst/fft/gstfftf32.h:
15094 * gst-libs/gst/fft/gstfftf64.c:
15095 * gst-libs/gst/fft/gstfftf64.h:
15096 * gst-libs/gst/fft/gstffts16.c:
15097 * gst-libs/gst/fft/gstffts16.h:
15098 * gst-libs/gst/fft/gstffts32.c:
15099 * gst-libs/gst/fft/gstffts32.h:
15100 * tests/check/libs/fft.c: (GST_START_TEST):
15101 Remove the magnitude and phase calculation functions as these have
15102 very special use cases and can't even be used for the spectrum
15103 element. Also adjust the docs to mention some properties of the used
15104 FFT implemention, i.e. how the values are scaled. Fixes #492098.
15106 2007-11-06 11:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
15108 gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
15109 Original commit message from CVS:
15110 * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
15112 Avoid crash when there are external subtitles (fixes #491722).
15114 2007-11-03 10:39:21 +0000 Tim-Philipp Müller <tim@centricular.net>
15116 ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
15117 Original commit message from CVS:
15118 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
15119 * ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
15120 'Could not open resource for writing' is not an acceptable
15121 error message when we can't open the audio device (see #492334),
15122 even less so when we're trying to open it to record something.
15124 2007-11-02 21:03:01 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15126 win32/common/libgstrtp.def: Add some more missing symbols (#492813).
15127 Original commit message from CVS:
15128 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15129 * win32/common/libgstrtp.def:
15130 Add some more missing symbols (#492813).
15132 2007-11-02 14:59:06 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
15134 tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
15135 Original commit message from CVS:
15136 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
15137 * tests/check/elements/audioconvert.c: (verify_convert):
15138 Add check to make sure that the out caps have a channel layout
15139 set on them where they should have one.
15141 2007-11-01 13:28:59 +0000 Vincent Torri <vtorri@univ-evry.fr>
15143 gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
15144 Original commit message from CVS:
15145 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
15146 * gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
15147 * gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
15148 Include our own _stdint.h instead of sys/types.h, makes MingW happy
15150 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
15151 Use _pipe directly, GLib doesn't have a pipe() macro any longer
15152 (it disappeared in GLib 2.14.0) (#492306).
15153 * gst-libs/gst/sdp/Makefile.am:
15154 * gst-libs/gst/sdp/gstsdpmessage.c:
15155 Fix includes and LIBS for win32/Mingw (#492306).
15156 * tests/examples/dynamic/addstream.c (pause_play_stream):
15157 Use more portable g_usleep() instead of sleep() (#492306).
15159 2007-11-01 12:51:57 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15161 gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
15162 Original commit message from CVS:
15163 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
15164 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
15165 (gst_ring_buffer_parse_caps):
15166 Return NULL instead of an enum that happens to be 0, fixes warning
15168 * gst-libs/gst/audio/gstringbuffer.h:
15169 No trailing commas in enum list (for gcc-2.9x).
15170 * gst/videotestsrc/videotestsrc.c: (random_char):
15171 Make information loss explicit instead of implicitly truncating to
15172 eight bits via the return value. Fixes runtime error on MSVC when
15173 using the debug CRT (#492114).
15174 * win32/common/config.h.in:
15175 Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
15176 * win32/common/libgstinterfaces.def:
15177 * win32/common/libgstrtp.def:
15178 Export a few more symbols (#492114).
15180 2007-11-01 08:06:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15182 gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
15183 Original commit message from CVS:
15184 * gst-libs/gst/audio/audio.c:
15185 * gst-libs/gst/audio/audio.h:
15186 Readd the deprecation guards, but preserve compilability.
15188 2007-10-31 17:54:48 +0000 Tim-Philipp Müller <tim@centricular.net>
15190 gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
15191 Original commit message from CVS:
15192 * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
15193 (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
15194 Preserve channel layout when fixating the number of channels in the
15195 output caps, or make sure there's a suitable channel position layout
15196 set on the caps if required. Fixes #430677.
15198 2007-10-31 17:32:22 +0000 Tim-Philipp Müller <tim@centricular.net>
15200 tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
15201 Original commit message from CVS:
15202 * tests/check/elements/decodebin.c: (test_text_plain_streams):
15203 Make sure the pipeline really operates in push mode as it should
15206 2007-10-31 15:30:15 +0000 Tim-Philipp Müller <tim@centricular.net>
15208 gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
15209 Original commit message from CVS:
15210 * gst-libs/gst/audio/audio.h:
15211 Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
15212 compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
15213 (ie. normal cvs builds) will fail.
15215 2007-10-31 12:47:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15217 tell gtk-doc about the deprecation guard. Apply more doc fixes.
15218 Original commit message from CVS:
15219 * docs/libs/Makefile.am:
15220 * gst-libs/gst/audio/audio.c:
15221 * gst-libs/gst/audio/audio.h:
15222 * gst-libs/gst/interfaces/mixer.c:
15223 tell gtk-doc about the deprecation guard. Apply more doc fixes.
15225 2007-10-31 12:30:28 +0000 Tim-Philipp Müller <tim@centricular.net>
15227 tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
15228 Original commit message from CVS:
15229 * tests/check/libs/audio.c: (init_value_to_channel_layout),
15230 (test_channel_layout_value_intersect), (audio_suite):
15231 Add simple unit test to make sure GstValue intersection
15232 of channel layouts works the way I think it does.
15234 2007-10-30 20:32:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15236 Fix the docs according to what gtk-doc complained about.
15237 Original commit message from CVS:
15238 * docs/libs/gst-plugins-base-libs-sections.txt:
15239 * gst-libs/gst/audio/gstaudiofilter.h:
15240 * gst-libs/gst/interfaces/mixer.h:
15241 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15242 * gst-libs/gst/rtp/gstbasertpdepayload.h:
15243 * gst-libs/gst/sdp/gstsdpmessage.c:
15244 Fix the docs according to what gtk-doc complained about.
15246 2007-10-30 19:46:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15248 tests/icles/stress-playbin.c: Fix the build.
15249 Original commit message from CVS:
15250 * tests/icles/stress-playbin.c:
15253 2007-10-30 15:54:46 +0000 Tim-Philipp Müller <tim@centricular.net>
15255 gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
15256 Original commit message from CVS:
15257 * gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
15258 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
15259 Post nice/more useful error message if we don't have a decoder for
15262 2007-10-30 15:07:58 +0000 Wim Taymans <wim.taymans@gmail.com>
15264 gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
15265 Original commit message from CVS:
15266 * gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
15267 Be a bit more useful, unblock the pads after we fired the no-more-pads
15268 signal so that we can use the signal to inspect and connect all pads
15269 without having to keep extra state outside of decodebin.
15271 2007-10-30 15:00:06 +0000 Wim Taymans <wim.taymans@gmail.com>
15273 gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
15274 Original commit message from CVS:
15275 * gst/playback/gsturidecodebin.c:
15276 (gst_uri_decode_bin_autoplug_continue),
15277 (gst_uri_decode_bin_class_init), (no_more_pads_full):
15278 Implement default signal handler so that we return TRUE when nothing is
15281 2007-10-28 11:53:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15283 gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
15284 Original commit message from CVS:
15285 * gst-libs/gst/riff/riff-media.c:
15286 (gst_riff_wavext_add_channel_layout),
15287 (gst_riff_wave_add_default_channel_layout),
15288 (gst_riff_wavext_get_default_channel_mask),
15289 (gst_riff_create_audio_caps):
15290 Use the ALSA channel layout as default for wav files without channel
15291 layout information. This fixes playback of chan-id.wav on 5.1 systems
15292 for example. Also refactor the channel layout setting a bit and add
15293 more default channel orders. Fixes #489010.
15295 2007-10-28 11:46:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15298 Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
15299 Original commit message from CVS:
15300 (gst_riff_wavext_add_channel_layout),
15301 (gst_riff_wave_add_default_channel_layout),
15302 (gst_riff_wavext_get_default_channel_mask),
15303 (gst_riff_create_audio_caps):
15304 Use the ALSA channel layout as default for wav files without channel
15305 layout information. This fixes playback of chan-id.wav on 5.1 systems
15306 for example. Also refactor the channel layout setting a bit and add
15307 more default channel orders. Fixes #489010.
15309 2007-10-26 18:57:33 +0000 Tim-Philipp Müller <tim@centricular.net>
15311 tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
15312 Original commit message from CVS:
15313 * tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
15314 GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
15315 -DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
15318 2007-10-26 12:07:14 +0000 Christian Schaller <uraeus@gnome.org>
15320 * gst-plugins-base.spec.in:
15322 Original commit message from CVS:
15325 2007-10-25 17:36:49 +0000 Wim Taymans <wim.taymans@gmail.com>
15327 gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
15328 Original commit message from CVS:
15329 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
15330 (gst_decode_bin_dispose), (gst_decode_bin_set_caps),
15331 (gst_decode_bin_set_subs_encoding),
15332 (gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
15333 (gst_decode_bin_get_property), (analyze_new_pad):
15334 Move subtitle encoding property to decodebin2 so that it can set the
15335 property value on all elements that it autoplugs and that require it.
15336 Make caps refcounting more consistent in get/set.
15337 * gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
15338 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
15339 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
15340 (gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
15341 (proxy_autoplug_continue_signal),
15342 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
15344 Proxy properties and relevant signals from the internal decodebin.
15345 Make properties MT safe.
15347 2007-10-25 15:10:59 +0000 Tim-Philipp Müller <tim@centricular.net>
15349 gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
15350 Original commit message from CVS:
15351 * gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
15352 * gst-libs/gst/tag/tags.c:
15353 Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
15354 GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
15355 * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
15356 Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
15357 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
15358 (gst_tag_to_vorbis_comments):
15359 Map new SORTNAME tags (these tags aren't even semi-official, so I'm
15360 just mapping everything I found in the wild) (#414539).
15362 2007-10-24 11:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
15364 gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
15365 Original commit message from CVS:
15366 Inspired by patch of: René Stadler <mail at renestadler dot de>
15367 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
15368 (gst_decode_bin_autoplug_continue),
15369 (gst_decode_bin_autoplug_factories),
15370 (gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
15371 (find_compatibles):
15372 * gst/playback/gstplay-marshal.list:
15373 Remove the autoplug-sort signal and replace it with a binding friendly
15374 autoplug-select signal.
15375 Add an autoplug-factories signal that can be used to generate a list of
15376 factories to try to autoplug.
15377 Add the GstPad to the autoplugging signal args as it might be needed to
15378 make a good factory selection.
15379 Fix up the marshallers for this. Fixes #407282.
15381 2007-10-23 14:23:14 +0000 Tim-Philipp Müller <tim@centricular.net>
15383 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...
15384 Original commit message from CVS:
15385 * gst-libs/gst/tag/gsttagdemux.c:
15386 Don't abort with an assertion if we receive a seek event with
15387 a start type of NONE (see launchpad bug #155878).
15389 2007-10-22 10:21:46 +0000 Wim Taymans <wim.taymans@gmail.com>
15391 sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
15392 Original commit message from CVS:
15393 * sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
15394 (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
15395 (gst_ximagesink_change_state), (gst_ximagesink_reset):
15396 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
15397 (gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
15398 (gst_xvimagesink_change_state), (gst_xvimagesink_reset):
15399 Make sure that before we clean up the X resources, we shutdown and join
15401 Also make sure the event thread does not shut down immediatly after
15402 startup because the running variable is not yet correctly set.
15405 2007-10-16 16:48:38 +0000 Wim Taymans <wim.taymans@gmail.com>
15407 gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
15408 Original commit message from CVS:
15409 * gst/playback/gstdecodebin.c: (new_pad), (type_found):
15410 Make the window for a race in typefind and shutting down smaller until
15411 we figure out the right locking here. Avoids #485753 usually.
15412 * gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
15413 Remove unneeded lock causing a race in typefind and shutting down.
15415 * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
15416 Also remove sinks when going to NULL because we might not complete the
15417 state change to PAUSED, causing the PAUSED->READY state change not to
15420 2007-10-16 15:33:31 +0000 Wim Taymans <wim.taymans@gmail.com>
15422 gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
15423 Original commit message from CVS:
15424 * gst-libs/gst/audio/gstbaseaudiosink.c:
15425 (gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
15426 Also explicitly release the ringbuffer when going to NULL because it
15427 is required in the setcaps function, before the state change to PAUSED
15430 2007-10-16 14:58:53 +0000 Tim-Philipp Müller <tim@centricular.net>
15432 tests/icles/: Does what it says on the tin.
15433 Original commit message from CVS:
15434 * tests/icles/.cvsignore:
15435 * tests/icles/Makefile.am:
15436 * tests/icles/stress-playbin.c:
15437 Does what it says on the tin.
15439 2007-10-15 11:38:39 +0000 Wim Taymans <wim.taymans@gmail.com>
15441 gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
15442 Original commit message from CVS:
15443 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
15444 Fix queue negotiation. See #486758.
15446 2007-10-12 10:52:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15448 Actual code change to go along with:
15449 Original commit message from CVS:
15450 Actual code change to go along with:
15451 2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
15452 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
15453 (gst_xvimagesink_xwindow_new),
15454 (gst_xvimagesink_update_colorbalance),
15455 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
15456 Fix handling of some of the X atoms. If the last parameter is True,
15457 XInternAtom won't create the atom if it doesn't exist, and therefore
15458 might return None. This causes X errors on Xv implementations that
15459 don't provide the colour balance attributes.
15461 2007-10-12 10:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15464 Remove stray character from the changelog.
15465 Original commit message from CVS:
15466 Remove stray character from the changelog.
15468 2007-10-12 10:33:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15471 I'm too lazy to comment this
15472 Original commit message from CVS:
15473 *** empty log message ***
15475 2007-10-11 18:24:09 +0000 Tim-Philipp Müller <tim@centricular.net>
15477 Extract vorbis comment LICENSE tags correctly.
15478 Original commit message from CVS:
15479 * gst-libs/gst/tag/gstvorbistag.c:
15480 * tests/check/libs/tag.c:
15481 Extract vorbis comment LICENSE tags correctly.
15483 2007-10-11 16:12:21 +0000 Jason Kivlighn <jkivlighn@gmail.com>
15485 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
15486 Original commit message from CVS:
15487 Patch by: Jason Kivlighn <jkivlighn gmail com>
15488 * gst-libs/gst/tag/gstid3tag.c:
15489 * tests/check/libs/tag.c:
15490 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
15492 2007-10-10 17:01:51 +0000 Tim-Philipp Müller <tim@centricular.net>
15494 gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
15495 Original commit message from CVS:
15496 * gst-libs/gst/tag/gsttagdemux.c:
15497 Don't error out when a buggy downstream element doesn't
15498 handle the newsegment event we send properly (especially
15499 not without posting a meaningful error message on the
15500 bus). See bug #471370 and launchpad bug #136264.
15502 2007-10-10 15:36:56 +0000 Wim Taymans <wim.taymans@gmail.com>
15504 gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
15505 Original commit message from CVS:
15506 * gst-libs/gst/audio/gstbaseaudiosink.c:
15507 (gst_base_audio_sink_drain):
15508 Use new basesink method to make our EOS drain interruptable.
15510 2007-10-10 09:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15512 gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
15513 Original commit message from CVS:
15514 * gst-libs/gst/rtp/gstrtppayloads.c:
15515 Fix silly search-replace oversight.
15517 2007-10-09 09:57:17 +0000 Laurent Glayal <spglegle@yahoo.fr>
15519 gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
15520 Original commit message from CVS:
15521 Patch by: Laurent Glayal <spglegle at yahoo dot fr>
15522 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
15523 (gst_basertppayload_set_outcaps):
15524 Fix caps memleak. Fixes #484989.
15526 2007-10-08 18:04:34 +0000 Wim Taymans <wim.taymans@gmail.com>
15528 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
15529 Original commit message from CVS:
15530 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15531 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
15534 2007-10-08 18:02:53 +0000 Wim Taymans <wim.taymans@gmail.com>
15536 gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
15537 Original commit message from CVS:
15538 * gst-libs/gst/audio/gstbaseaudiosrc.c:
15539 (gst_base_audio_src_create):
15540 Also handle the case where there is no clock set on the audio source,
15541 like in the unit tests.
15543 2007-10-08 17:40:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15545 gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
15546 Original commit message from CVS:
15547 * gst-libs/gst/rtp/gstrtppayloads.c:
15548 Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
15549 to avoid compiler warnings
15551 2007-10-08 17:12:32 +0000 Wim Taymans <wim.taymans@gmail.com>
15553 gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
15554 Original commit message from CVS:
15555 * gst/playback/gstdecodebin.c: (type_found),
15556 (gst_decode_bin_change_state):
15557 * gst/playback/gstdecodebin2.c: (type_found),
15558 (gst_decode_bin_change_state):
15559 Don't disconnect the have_type signal because we never reconnect it
15560 later on. Instead keep a variable to see if we already detected a type.
15562 2007-10-08 10:47:26 +0000 Wim Taymans <wim.taymans@gmail.com>
15564 gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
15565 Original commit message from CVS:
15566 * gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
15567 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
15569 Unlink the signal handler when we found the type, we're not going to do
15570 anything sensible with more type_found signals anyway.
15572 2007-10-08 06:07:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15574 ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
15575 Original commit message from CVS:
15576 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
15577 Use GIO function to get a list of supported URI schemes instead of
15578 hard coding something.
15580 2007-10-06 16:49:55 +0000 Tim-Philipp Müller <tim@centricular.net>
15582 gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
15583 Original commit message from CVS:
15584 * gst-libs/gst/tag/gsttagdemux.c:
15587 2007-10-06 15:04:53 +0000 Tim-Philipp Müller <tim@centricular.net>
15589 gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
15590 Original commit message from CVS:
15591 * gst-libs/gst/tag/Makefile.am:
15592 * gst-libs/gst/tag/gsttagdemux.c:
15593 * gst-libs/gst/tag/gsttagdemux.h:
15594 API: add GstTagDemux base class for simple tag demuxers.
15595 * docs/libs/gst-plugins-base-libs-docs.sgml:
15596 * docs/libs/gst-plugins-base-libs-sections.txt:
15597 Add GstTagDemux to docs.
15599 2007-10-05 07:49:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15601 gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
15602 Original commit message from CVS:
15603 * gst-libs/gst/rtp/gstrtpbuffer.c:
15604 (gst_rtp_buffer_get_payload_subbuffer):
15605 Fix bug introduced with last commit which inverted the logic and
15606 caused all buffers to be dropped. Fixes #483620.
15607 Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
15609 2007-10-04 06:50:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15611 gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
15612 Original commit message from CVS:
15613 * gst-libs/gst/rtp/gstrtpbuffer.c:
15614 Replace g_return_if_val (as it could be disabled), with regular return
15617 2007-10-03 14:51:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15619 tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
15620 Original commit message from CVS:
15621 * tests/check/pipelines/simple-launch-lines.c:
15622 Print message name and not just number.
15624 2007-10-02 11:11:13 +0000 Wim Taymans <wim.taymans@gmail.com>
15626 gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
15627 Original commit message from CVS:
15628 * gst-libs/gst/audio/gstbaseaudiosink.c:
15629 (gst_base_audio_sink_async_play):
15630 When slaved to the clock, don't try to align a sample with the previous
15631 one when going to PLAYING again.
15633 2007-10-02 09:04:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15635 tests/examples/snapshot/snapshot.c: Fix the build.
15636 Original commit message from CVS:
15637 * tests/examples/snapshot/snapshot.c:
15640 2007-10-02 07:43:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15642 ext/gio/gstgiosink.c: Update to API changes in GIO.
15643 Original commit message from CVS:
15644 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
15645 Update to API changes in GIO.
15647 2007-10-01 16:33:00 +0000 Wim Taymans <wim.taymans@gmail.com>
15649 gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
15650 Original commit message from CVS:
15651 * gst-libs/gst/sdp/gstsdpmessage.h:
15652 Add RFC 3556 bandwidth modifiers.
15654 2007-10-01 13:37:31 +0000 Wim Taymans <wim.taymans@gmail.com>
15656 Update documentation.
15657 Original commit message from CVS:
15658 * docs/libs/gst-plugins-base-libs-docs.sgml:
15659 * docs/libs/gst-plugins-base-libs-sections.txt:
15660 * gst-libs/gst/rtp/gstrtppayloads.c:
15661 Update documentation.
15663 2007-10-01 13:22:14 +0000 Wim Taymans <wim.taymans@gmail.com>
15665 gst-libs/gst/rtp/: Added new file and header to deal with payload info.
15666 Original commit message from CVS:
15667 * gst-libs/gst/rtp/Makefile.am:
15668 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
15669 (gst_rtp_payload_info_for_name):
15670 * gst-libs/gst/rtp/gstrtppayloads.h:
15671 Added new file and header to deal with payload info.
15672 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
15673 (gst_rtp_buffer_default_clock_rate):
15674 * gst-libs/gst/rtp/gstrtpbuffer.h:
15675 Payload specific stuff is move to new headers.
15676 Implement _default_clock rate using the new payload function.
15677 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
15678 (gst_sdp_parse_line):
15679 * gst-libs/gst/sdp/gstsdpmessage.h:
15680 Add some more comments.
15682 2007-10-01 10:22:46 +0000 Wim Taymans <wim.taymans@gmail.com>
15684 gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
15685 Original commit message from CVS:
15686 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
15687 (sdp_check_header), (sdp_type_find), (plugin_init):
15688 Add typefind function for application/sdp.
15689 Remove some old dirac typefind code that was ifdeffed out.
15691 2007-09-29 12:04:02 +0000 Sébastien Moutte <sebastien@moutte.net>
15693 win32/common/libgstaudio.def: Add new exported functions.
15694 Original commit message from CVS:
15695 * win32/common/libgstaudio.def:
15696 Add new exported functions.
15697 * win32/vs6/grammar.dsp:
15698 Add autogeneration and copy of some autegenerated files from win32/common
15700 * win32/vs6/libgstaudioconvert.dsp:
15701 Add gstaudioquantize.c to the build.
15702 * win32/vs6/libgstinterfaces.dsp:
15703 Add videoorientation.c to the build.
15704 * win32/vs6/libgstriff.dsp:
15705 Add libgsttag to the link libraries list.
15706 * win32/vs6/libgstvolume.dsp:
15707 Add liboil to the link.
15708 * win32/vs6/gst_plugins_base.dsw:
15709 * win32/vs6/libgstrtsp.dsp:
15710 * win32/common/libgstrtsp.def:
15711 Add files to build libgstrtsp library.
15713 2007-09-29 07:01:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15715 ext/gio/: Some minor cleanup and allow setting the location only when the element is not playing or paused.
15716 Original commit message from CVS:
15717 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
15718 (gst_gio_sink_set_property), (gst_gio_sink_render):
15719 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
15720 (gst_gio_src_set_property):
15721 Some minor cleanup and allow setting the location only when the
15722 element is not playing or paused.
15724 2007-09-26 15:14:37 +0000 Wim Taymans <wim.taymans@gmail.com>
15726 tests/examples/snapshot/snapshot.c: Print error when pipeline failed to construct.
15727 Original commit message from CVS:
15728 * tests/examples/snapshot/snapshot.c: (main):
15729 Print error when pipeline failed to construct.
15731 2007-09-25 19:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
15733 Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
15734 Original commit message from CVS:
15736 * gst-libs/gst/tag/gstid3tag.c:
15737 * gst-libs/gst/tag/gstvorbistag.c:
15738 Add mappings for the new GST_TAG_COMPOSER for vorbis comments
15741 2007-09-25 11:54:09 +0000 Tim-Philipp Müller <tim@centricular.net>
15743 gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
15744 Original commit message from CVS:
15745 * gst-libs/gst/floatcast/floatcast.h:
15746 Don't include config.h in an installed public header, this
15747 might break compilation of applications that don't have such
15748 a header and doesn't necessarily do what it's supposed to do
15749 anyway (ie. check for the lrint/lrintf defines) (#442065).
15750 Add docs for the various macros and document how this header
15751 has to be used (link against libm, etc.); add a few FIXMEs;
15752 include math.h for non-c99 code path. Based on patch by
15755 2007-09-25 07:50:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15757 configure.ac: Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead of duplicating these macros in confi...
15758 Original commit message from CVS:
15760 Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
15761 of duplicating these macros in configure.ac.
15763 2007-09-22 17:58:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15765 po/: Updated translations to 0.10.14
15766 Original commit message from CVS:
15770 Updated translations to 0.10.14
15772 2007-09-22 17:57:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15776 Original commit message from CVS:
15779 2007-09-22 17:56:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15781 po/pl.po: Added Polish translation.
15782 Original commit message from CVS:
15783 translated by: Jakub Bogusz <qboosh@pld-linux.org>
15785 Added Polish translation.
15787 2007-09-22 17:55:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15789 po/fi.po: Added Finnish translation.
15790 Original commit message from CVS:
15791 translated by: Ilkka Tuohela <hile@iki.fi>
15793 Added Finnish translation.
15795 2007-09-22 17:54:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15797 po/es.po: Added Spanish translation.
15798 Original commit message from CVS:
15799 translated by: Jorge González González <aloriel@gmail.com>
15801 Added Spanish translation.
15803 2007-09-22 17:53:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15805 po/da.po: Added Danish translation.
15806 Original commit message from CVS:
15807 translated by: Mogens Jaeger <mogens@jaeger.tf>
15809 Added Danish translation.
15811 2007-09-22 17:52:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15813 po/zh_CN.po: Added Chinese (simplified) translation.
15814 Original commit message from CVS:
15815 translated by: Funda Wang <fundawang@linux.net.cn>
15817 Added Chinese (simplified) translation.
15819 2007-09-22 17:51:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
15821 po/bg.po: Added Bulgarian translation.
15822 Original commit message from CVS:
15823 translated by: Alexander Shopov <ash@contact.bg>
15825 Added Bulgarian translation.
15827 2007-09-21 18:00:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15829 docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy.
15830 Original commit message from CVS:
15831 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
15833 * ext/gio/gstgiosink.h:
15834 * ext/gio/gstgiosrc.h:
15835 Mark private fields of the instance structs private.
15837 2007-09-21 17:31:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15839 docs/plugins/: Add the GIO plugin to the docs and do a make update while doing that.
15840 Original commit message from CVS:
15841 * docs/plugins/Makefile.am:
15842 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
15843 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
15844 * docs/plugins/gst-plugins-bad-plugins.args:
15845 * docs/plugins/gst-plugins-bad-plugins.signals:
15846 * docs/plugins/inspect/plugin-bz2.xml:
15847 * docs/plugins/inspect/plugin-cdxaparse.xml:
15848 * docs/plugins/inspect/plugin-dfbvideosink.xml:
15849 * docs/plugins/inspect/plugin-dtsdec.xml:
15850 * docs/plugins/inspect/plugin-equalizer.xml:
15851 * docs/plugins/inspect/plugin-faac.xml:
15852 * docs/plugins/inspect/plugin-faad.xml:
15853 * docs/plugins/inspect/plugin-filter.xml:
15854 * docs/plugins/inspect/plugin-freeze.xml:
15855 * docs/plugins/inspect/plugin-gio.xml:
15856 * docs/plugins/inspect/plugin-gsm.xml:
15857 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
15858 * docs/plugins/inspect/plugin-h264parse.xml:
15859 * docs/plugins/inspect/plugin-modplug.xml:
15860 * docs/plugins/inspect/plugin-mpeg2enc.xml:
15861 * docs/plugins/inspect/plugin-musepack.xml:
15862 * docs/plugins/inspect/plugin-musicbrainz.xml:
15863 * docs/plugins/inspect/plugin-nsfdec.xml:
15864 * docs/plugins/inspect/plugin-replaygain.xml:
15865 * docs/plugins/inspect/plugin-soundtouch.xml:
15866 * docs/plugins/inspect/plugin-spcdec.xml:
15867 * docs/plugins/inspect/plugin-spectrum.xml:
15868 * docs/plugins/inspect/plugin-speed.xml:
15869 * docs/plugins/inspect/plugin-tta.xml:
15870 * docs/plugins/inspect/plugin-videosignal.xml:
15871 * docs/plugins/inspect/plugin-xingheader.xml:
15872 * docs/plugins/inspect/plugin-xvid.xml:
15873 Add the GIO plugin to the docs and do a make update
15875 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
15876 Fix a small memleak.
15878 2007-09-21 17:07:56 +0000 René Stadler <mail@renestadler.de>
15880 Add a GIO/GVFS plugin with source and sink elements. This will only be enabled when --enable-experimental is given to...
15881 Original commit message from CVS:
15882 Patch by: René Stadler <mail at renestadler dot de>
15885 * ext/gio/Makefile.am:
15886 * ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek),
15887 (gst_gio_get_supported_protocols),
15888 (gst_gio_uri_handler_get_type_sink),
15889 (gst_gio_uri_handler_get_type_src),
15890 (gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri),
15891 (gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init),
15892 (gst_gio_uri_handler_do_init), (plugin_init):
15893 * ext/gio/gstgio.h:
15894 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
15895 (gst_gio_sink_class_init), (gst_gio_sink_init),
15896 (gst_gio_sink_finalize), (gst_gio_sink_set_property),
15897 (gst_gio_sink_get_property), (gst_gio_sink_start),
15898 (gst_gio_sink_stop), (gst_gio_sink_unlock),
15899 (gst_gio_sink_unlock_stop), (gst_gio_sink_event),
15900 (gst_gio_sink_render), (gst_gio_sink_query):
15901 * ext/gio/gstgiosink.h:
15902 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
15903 (gst_gio_src_class_init), (gst_gio_src_init),
15904 (gst_gio_src_finalize), (gst_gio_src_set_property),
15905 (gst_gio_src_get_property), (gst_gio_src_start),
15906 (gst_gio_src_stop), (gst_gio_src_get_size),
15907 (gst_gio_src_is_seekable), (gst_gio_src_unlock),
15908 (gst_gio_src_unlock_stop), (gst_gio_src_check_get_range),
15909 (gst_gio_src_create):
15910 * ext/gio/gstgiosrc.h:
15911 Add a GIO/GVFS plugin with source and sink elements. This will
15912 only be enabled when --enable-experimental is given to configure
15913 for now as the GIO API is not stable yet. Fixes #476916.
15915 2007-09-21 14:37:26 +0000 Wim Taymans <wim.taymans@gmail.com>
15917 gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
15918 Original commit message from CVS:
15919 * gst/playback/gstqueue2.c: (gst_queue_push_one):
15920 Fix compilation wrt printf arguments.
15922 2007-09-20 17:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
15924 examples/app/appsrc_ex.c: Fix compilation after changing the name of a method.
15925 Original commit message from CVS:
15926 * examples/app/appsrc_ex.c: (main):
15927 Fix compilation after changing the name of a method.
15929 2007-09-20 14:09:24 +0000 Wim Taymans <wim.taymans@gmail.com>
15931 Add simple snapshot example program using appsink.
15932 Original commit message from CVS:
15934 * tests/examples/Makefile.am:
15935 * tests/examples/snapshot/.cvsignore:
15936 * tests/examples/snapshot/Makefile.am:
15937 * tests/examples/snapshot/snapshot.c: (main):
15938 Add simple snapshot example program using appsink.
15940 2007-09-20 13:59:50 +0000 Wim Taymans <wim.taymans@gmail.com>
15942 gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
15943 Original commit message from CVS:
15944 * gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
15945 (gst_app_sink_class_init), (gst_app_sink_init),
15946 (gst_app_sink_dispose), (gst_app_sink_finalize),
15947 (gst_app_sink_set_property), (gst_app_sink_get_property),
15948 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
15949 (gst_app_sink_event), (gst_app_sink_getcaps),
15950 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
15951 (gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
15952 (gst_app_sink_pull_buffer):
15953 * gst-libs/gst/app/gstappsink.h:
15954 Add properties, signals and actions to access the element even without
15955 linking to the library.
15956 Fix some method names and signatures.
15958 2007-09-20 10:37:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15960 tests/check/generic/states.c: Improved state change unit test.
15961 Original commit message from CVS:
15962 * tests/check/generic/states.c:
15963 Improved state change unit test.
15965 2007-09-19 18:16:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15967 Ignore registries in any format.
15968 Original commit message from CVS:
15969 * docs/plugins/.cvsignore:
15970 * tests/check/.cvsignore:
15971 Ignore registries in any format.
15973 2007-09-19 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
15975 gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
15976 Original commit message from CVS:
15977 * gst-libs/gst/rtp/gstbasertpdepayload.c:
15978 (gst_base_rtp_depayload_chain),
15979 (gst_base_rtp_depayload_set_gst_timestamp):
15980 Only copy timestamp on outgoing packets if the depayloader did not set
15982 Also copy duration on outgoing packets.
15984 2007-09-19 15:55:08 +0000 Wim Taymans <wim.taymans@gmail.com>
15986 gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
15987 Original commit message from CVS:
15988 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
15989 (gst_basertppayload_set_outcaps):
15990 Fix compilation because of missing %d in printf.
15991 When fixating caps, fixate what we can and throw away all remaining
15992 unfixed caps, subclasses should do something smart if they need to.
15994 2007-09-19 12:04:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
15996 ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
15997 Original commit message from CVS:
15998 * ext/gnomevfs/gstgnomevfssrc.c:
15999 Improve debug logs a bit and be more verbose if things go wrong.
16001 2007-09-17 17:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16003 Fix a bunch of compile warnings shown with Forte.
16004 Original commit message from CVS:
16005 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
16006 (gst_text_overlay_set_property):
16007 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
16008 * gst-libs/gst/audio/gstbaseaudiosink.c:
16009 (gst_base_audio_sink_render):
16010 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
16011 (gst_rtcp_unix_to_ntp):
16012 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
16013 * gst/playback/gstqueue2.c:
16014 * tests/examples/seek/seek.c: (set_scale):
16015 Fix a bunch of compile warnings shown with Forte.
16016 * gst/audiorate/gstaudiorate.c:
16017 Always pull in config.h before including any system headers.
16019 2007-09-17 16:22:17 +0000 Wim Taymans <wim.taymans@gmail.com>
16021 gst/playback/gstqueue2.c: Also fix #476514 for queue2.
16022 Original commit message from CVS:
16023 * gst/playback/gstqueue2.c: (update_buffering),
16024 (gst_queue_locked_flush), (gst_queue_locked_enqueue),
16025 (gst_queue_handle_sink_event), (gst_queue_chain),
16026 (gst_queue_push_one), (gst_queue_sink_activate_push),
16027 (gst_queue_src_activate_push), (gst_queue_src_activate_pull):
16028 Also fix #476514 for queue2.
16030 2007-09-16 19:31:06 +0000 Wim Taymans <wim.taymans@gmail.com>
16032 gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
16033 Original commit message from CVS:
16034 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16035 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
16036 (gst_base_rtp_depayload_chain),
16037 (gst_base_rtp_depayload_handle_sink_event),
16038 (gst_base_rtp_depayload_push_full),
16039 (gst_base_rtp_depayload_set_gst_timestamp),
16040 (gst_base_rtp_depayload_change_state):
16041 Remove code to deal with RTP to GST time conversion, we now just copy
16042 the GST timestamp we receive to the outgoing buffers.
16043 Handle segment and flushes correctly.
16044 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
16045 When we have no valid input timestamp, use the previous rtp timestamp on
16046 the outgoing RTP packet instead of the RTP base time.
16048 2007-09-16 01:56:21 +0000 David Schleef <ds@schleef.org>
16050 ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
16051 Original commit message from CVS:
16052 * ext/alsa/gstalsa.c:
16053 * ext/alsa/gstalsadeviceprobe.c:
16054 * ext/alsa/gstalsamixer.c:
16055 * ext/alsa/gstalsasink.c:
16056 * ext/alsa/gstalsasrc.c:
16057 Change alsa alloca's to malloc to fix warnings on gcc-4.2.
16059 2007-09-15 18:41:27 +0000 Wim Taymans <wim.taymans@gmail.com>
16061 gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
16062 Original commit message from CVS:
16063 * gst-libs/gst/rtp/gstbasertppayload.c:
16064 (gst_basertppayload_set_outcaps), (gst_basertppayload_push):
16065 Add some debug info when negotiating caps.
16067 2007-09-15 00:29:11 +0000 Wim Taymans <wim.taymans@gmail.com>
16069 gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
16070 Original commit message from CVS:
16071 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
16072 A buffer with an empty payload is also a valid buffer.
16074 2007-09-14 20:52:00 +0000 Wim Taymans <wim.taymans@gmail.com>
16076 gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
16077 Original commit message from CVS:
16078 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
16079 (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
16080 (gst_basertppayload_change_state):
16081 Make sure we start our RTP timestamp from the random base RTP
16082 timestamp even if the buffer timestamp starts from some random value.
16084 2007-09-14 16:56:16 +0000 Wim Taymans <wim.taymans@gmail.com>
16086 Add simple exmple app to demonstrate starting and pausing live and non-live bins in a PLAYING pipeline.
16087 Original commit message from CVS:
16089 * tests/examples/Makefile.am:
16090 * tests/examples/dynamic/.cvsignore:
16091 * tests/examples/dynamic/Makefile.am:
16092 * tests/examples/dynamic/addstream.c: (create_stream),
16093 (pause_play_stream), (message_received), (eos_message_received),
16094 (perform_step), (main):
16095 Add simple exmple app to demonstrate starting and pausing live and
16096 non-live bins in a PLAYING pipeline.
16098 2007-09-14 10:42:00 +0000 Julien Moutte <julien@moutte.net>
16100 gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
16101 Original commit message from CVS:
16102 2007-09-14 Julien MOUTTE <julien@moutte.net>
16103 * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
16104 typefind for QCP files (RFC #3625)
16106 2007-09-13 22:52:09 +0000 Wim Taymans <wim.taymans@gmail.com>
16108 gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
16109 Original commit message from CVS:
16110 * gst-libs/gst/audio/gstbaseaudiosink.c:
16111 (gst_base_audio_sink_init):
16112 Disable pull mode scheduling, we're not ready for it yet and it subtly
16113 breaks a lot of things.
16115 2007-09-12 17:35:52 +0000 Tim-Philipp Müller <tim@centricular.net>
16117 tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
16118 Original commit message from CVS:
16119 * tests/check/elements/libvisual.c:
16120 Test all libvisual plugins, not just the first one; this reproduces
16121 bug #450336 quite easily. Looks like a problem with the 'jess'
16124 2007-09-12 17:15:12 +0000 Tim-Philipp Müller <tim@centricular.net>
16126 tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
16127 Original commit message from CVS:
16128 * tests/check/Makefile.am:
16129 * tests/check/elements/.cvsignore:
16130 * tests/check/elements/libvisual.c:
16131 Add basic libvisual test case in an attempt to reproduce bug #450336.
16132 Doesn't reproduce that bug, but some other crasher instead (invalid
16133 free), at least with make elements/libvisual.forever and the bumscope
16134 plugin on x86-64/gutsy. Leaving test disabled for now.
16136 2007-09-12 08:38:21 +0000 Peter Kjellerstedt <pkj@axis.com>
16138 gst/: Printf format fixes (#476128).
16139 Original commit message from CVS:
16140 Patch by: Peter Kjellerstedt <pkj at axis com>
16141 * gst-libs/gst/app/gstappsink.c:
16142 * gst/flv/gstflvdemux.c:
16143 * gst/flv/gstflvparse.c:
16144 * gst/interleave/deinterleave.c:
16145 * gst/switch/gstswitch.c:
16146 Printf format fixes (#476128).
16148 2007-09-11 19:07:57 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
16150 gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
16151 Original commit message from CVS:
16152 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
16153 * gst-libs/gst/rtsp/gstrtspconnection.c:
16154 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
16155 (read_body), (gst_rtsp_connection_receive):
16156 Make sure we can not cancel in the middle of receiving a message.
16159 2007-09-11 11:29:12 +0000 Josep Torra Valles <josep@fluendo.com>
16161 gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
16162 Original commit message from CVS:
16163 Patch by: Josep Torra Valles <josep@fluendo.com>
16164 * gst/playback/gstplaybasebin.c:
16165 Increase upper limit for audio queue a bit; fixes preroll problem
16166 with playbin and decodebin2 when playing a quicktime trailer with
16167 multichannel audio via http (#464666).
16169 2007-09-10 22:10:54 +0000 Wim Taymans <wim.taymans@gmail.com>
16171 gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
16172 Original commit message from CVS:
16173 * gst-libs/gst/audio/gstbaseaudiosrc.c:
16174 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
16175 (gst_base_audio_src_provide_clock),
16176 (gst_base_audio_src_set_property),
16177 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
16178 * gst-libs/gst/audio/gstbaseaudiosrc.h:
16179 Allow othe clocks than the internal clock to be used for the pipeline.
16180 Add property to disable clock provide.
16181 API: GstBaseAudioSrc::provide-clock
16183 2007-09-10 12:05:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16185 gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
16186 Original commit message from CVS:
16187 * gst/playback/gstdecodebin2.c:
16188 Don't leak request pads. Fixes #475395.
16190 2007-09-09 10:25:43 +0000 René Stadler <mail@renestadler.de>
16192 sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
16193 Original commit message from CVS:
16194 Patch by: René Stadler <mail at renestadler dot de>
16195 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
16196 (gst_ximage_buffer_class_init):
16197 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
16198 (gst_xvimage_buffer_class_init):
16199 Correctly chain up finalize with the parent class to prevent
16200 memory leaks. Fixes #474880.
16202 2007-09-09 04:08:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16204 Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
16205 Original commit message from CVS:
16206 * gst/volume/gstvolume.c: (volume_choose_func):
16207 * tests/check/elements/volume.c: (GST_START_TEST):
16208 Revert the latest change: floating point samples are allowed to
16209 have any value, not only values in the range [-1,1]. Thanks to Andy
16210 Wingo for noticing.
16211 Also fix processing of int32 samples with volumes > 4 by making the
16212 unity value smaller which prevents overflows.
16214 2007-09-07 17:37:03 +0000 Tim-Philipp Müller <tim@centricular.net>
16216 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
16217 Original commit message from CVS:
16218 * gst-libs/gst/rtp/gstrtpbuffer.c:
16219 * tests/check/libs/rtp.c:
16220 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
16222 2007-09-07 16:46:05 +0000 Haakon Sporsheim <haakon.sporsheim@tandberg.com>
16224 gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
16225 Original commit message from CVS:
16226 Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
16227 * gst-libs/gst/rtp/gstrtpbuffer.c:
16228 Fix up GstRTPHeader helper struct so that compilers will not under
16229 any circumstances add padding in between our fields, as currently
16230 happens with MSVC on win32, because that would lead to us sending
16231 out RTP payloads with broken RTP headers (#471194).
16232 Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
16233 * tests/check/Makefile.am:
16234 * tests/check/libs/.cvsignore:
16235 * tests/check/libs/rtp.c:
16236 Add some simple unit tests for GstRTPBuffer. Some are disabled
16237 because the code tested still needs fixing (set_csrc() does not work).
16239 2007-09-07 15:05:24 +0000 Christian Schaller <uraeus@gnome.org>
16241 * gst-plugins-base.spec.in:
16242 update spec file to include latest RTSP libraries and headers and more
16243 Original commit message from CVS:
16244 update spec file to include latest RTSP libraries and headers and more
16246 2007-09-07 12:41:01 +0000 Tim-Philipp Müller <tim@centricular.net>
16248 win32/: Add rtsp enumtypes (#474384) and update others.
16249 Original commit message from CVS:
16251 * win32/common/gstrtsp-enumtypes.c:
16252 * win32/common/gstrtsp-enumtypes.h:
16253 * win32/common/interfaces-enumtypes.c:
16254 * win32/common/interfaces-enumtypes.h:
16255 * win32/common/multichannel-enumtypes.c:
16256 Add rtsp enumtypes (#474384) and update others.
16258 2007-09-06 20:31:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16260 configure.ac: Fix configure check for HAVE_LIBXML_HTML.
16261 Original commit message from CVS:
16263 Fix configure check for HAVE_LIBXML_HTML.
16265 2007-09-06 12:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
16267 tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
16268 Original commit message from CVS:
16269 * tests/check/libs/.cvsignore:
16270 Ignore more, in case the build bots work again one day.
16272 2007-09-06 07:00:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16274 Add libgstfft, a FFT library based on Kiss FFT which is
16275 Original commit message from CVS:
16276 Reviewed by: Stefan Kost <ensonic@users.sf.net>
16278 * gst-libs/gst/Makefile.am:
16279 * gst-libs/gst/fft/Makefile.am:
16280 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
16281 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
16282 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
16283 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
16284 * gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
16285 * gst-libs/gst/fft/gstfft.h:
16286 * gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
16287 (gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
16288 (gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
16289 * gst-libs/gst/fft/gstfftf32.h:
16290 * gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
16291 (gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
16292 (gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
16293 * gst-libs/gst/fft/gstfftf64.h:
16294 * gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
16295 (gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
16296 (gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
16297 * gst-libs/gst/fft/gstffts16.h:
16298 * gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
16299 (gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
16300 (gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
16301 * gst-libs/gst/fft/gstffts32.h:
16302 * gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
16303 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16304 (kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
16305 (kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
16306 * gst-libs/gst/fft/kiss_fft_f32.h:
16307 * gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
16308 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16309 (kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
16310 (kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
16311 * gst-libs/gst/fft/kiss_fft_f64.h:
16312 * gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
16313 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16314 (kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
16315 (kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
16316 * gst-libs/gst/fft/kiss_fft_s16.h:
16317 * gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
16318 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
16319 (kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
16320 (kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
16321 * gst-libs/gst/fft/kiss_fft_s32.h:
16322 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
16323 (kiss_fftr_f32), (kiss_fftri_f32):
16324 * gst-libs/gst/fft/kiss_fftr_f32.h:
16325 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
16326 (kiss_fftr_f64), (kiss_fftri_f64):
16327 * gst-libs/gst/fft/kiss_fftr_f64.h:
16328 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
16329 (kiss_fftr_s16), (kiss_fftri_s16):
16330 * gst-libs/gst/fft/kiss_fftr_s16.h:
16331 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
16332 (kiss_fftr_s32), (kiss_fftri_s32):
16333 * gst-libs/gst/fft/kiss_fftr_s32.h:
16334 * gst-libs/gst/fft/kiss_version:
16335 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
16336 * pkgconfig/gstreamer-plugins-base.pc.in:
16337 Add libgstfft, a FFT library based on Kiss FFT which is
16338 BSD licensed. Supported sample formats are int16, int32,
16339 float and double. For those formats a real FFT and IFFT
16340 can be done, different windowing functions can be applied
16341 and functions for extracting the magnitude and phase exist.
16343 * docs/libs/Makefile.am:
16344 * docs/libs/gst-plugins-base-libs-docs.sgml:
16345 * docs/libs/gst-plugins-base-libs-sections.txt:
16346 Integrate libgstfft into the docs.
16347 * tests/check/Makefile.am:
16348 * tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
16349 Add unit tests for libgstfft, currently only testing the FFT.
16350 Unit tests for IFFT will follow soon.
16352 2007-09-05 23:07:40 +0000 Peter Kjellerstedt <pkj@axis.com>
16354 gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
16355 Original commit message from CVS:
16356 Patch by: Peter Kjellerstedt <pkj at axis com>
16357 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
16358 (gst_sdp_message_init), (gst_sdp_message_uninit),
16359 (is_multicast_address), (gst_sdp_message_as_text),
16360 (gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
16361 (gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
16362 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
16363 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
16364 (gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
16365 (gst_sdp_media_init), (gst_sdp_media_uninit),
16366 (gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
16367 (gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
16368 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
16369 (gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
16370 (gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
16371 * gst-libs/gst/sdp/gstsdpmessage.h:
16372 Separate INIT_ARRAY() and related macros into two versions, one for
16373 structures and one for pointers (e.g., INIT_ARRAY() and
16374 INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
16375 lists of emails and phone numbers.
16376 Add missing const as appropriate.
16377 Change all gint to guint since they all actually represent unsigned
16379 Do not use time as a variable name as it shadows the global time().
16380 Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
16381 Actually implement gst_sdp_message_add_time().
16382 Make gst_sdp_message_add_time() take repeat times as an argument.
16383 Store repeat times in GstSDPTime as a GArray rather than as gchar**.
16384 Corrected the definition of gst_sdp_media_get_bandwidth() (was
16385 misspelled as badwidth).
16386 gst-indented and a little clean up. Fixes #471067.
16388 2007-09-05 21:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16390 gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
16391 Original commit message from CVS:
16392 * gst/volume/gstvolume.c: (volume_choose_func),
16393 (volume_process_double), (volume_process_double_clamp),
16394 (volume_process_float_clamp):
16395 Correctly clamp float/double samples in the [-1.0,1.0] range to
16396 prevent weird effects.
16397 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
16398 Add unit tests for all samples types that had none before.
16400 2007-09-05 14:09:15 +0000 Tim-Philipp Müller <tim@centricular.net>
16402 gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
16403 Original commit message from CVS:
16404 * gst-libs/gst/rtp/gstrtpbuffer.c:
16405 Need to include stdlib.h for abs() here too.
16407 2007-09-05 14:01:25 +0000 Tim-Philipp Müller <tim@centricular.net>
16409 gst/playback/gststreaminfo.c: Fix build.
16410 Original commit message from CVS:
16411 * gst/playback/gststreaminfo.c:
16414 2007-09-05 10:32:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16416 gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
16417 Original commit message from CVS:
16418 * gst/playback/gststreaminfo.c:
16419 Clean up some half-disabled code and comment.
16421 2007-09-04 16:18:48 +0000 Wim Taymans <wim.taymans@gmail.com>
16423 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
16424 Original commit message from CVS:
16425 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
16426 (gst_base_rtp_payload_audio_handle_event):
16427 Return FALSE from the event handler to let the parent class handle the
16429 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16430 (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
16431 Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
16432 * gst-libs/gst/rtp/gstbasertppayload.c:
16433 Bump the MTU to 1400.
16435 2007-09-04 01:50:55 +0000 Johan Dahlin <johan@gnome.org>
16437 gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
16438 Original commit message from CVS:
16439 2007-09-03 Johan Dahlin <jdahlin@async.com.br>
16440 * gst/typefind/gsttypefindfunctions.c (plugin_init):
16441 Add an audio/x-nsf typefind function for the nsfdec element.
16443 2007-09-03 20:46:38 +0000 Renato Filho <renato.filho@indt.org.br>
16445 gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
16446 Original commit message from CVS:
16447 * gst/playback/gstplaybasebin.c:
16448 Included "myth://" on stream_uris list for enable buffering to mythtv files
16450 2007-09-03 19:31:11 +0000 Wim Taymans <wim.taymans@gmail.com>
16452 Fix parsing of RB blocks.
16453 Original commit message from CVS:
16454 * docs/libs/gst-plugins-base-libs-sections.txt:
16455 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
16456 (gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
16457 (gst_rtcp_unix_to_ntp):
16458 * gst-libs/gst/rtp/gstrtcpbuffer.h:
16459 Fix parsing of RB blocks.
16461 Added helper functions to convert to/from UNIX and NTP time.
16462 API: gst_rtcp_ntp_to_unix()
16463 API: gst_rtcp_unix_to_ntp()
16464 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
16465 (gst_rtp_buffer_get_header_len),
16466 (gst_rtp_buffer_get_extension_data),
16467 (gst_rtp_buffer_get_payload_subbuffer),
16468 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
16469 (gst_rtp_buffer_ext_timestamp):
16470 * gst-libs/gst/rtp/gstrtpbuffer.h:
16471 Fix some more docs.
16472 Implement handling of packets with extensions.
16473 Fix padding check in _validate().
16474 Added function to get extension data.
16475 API: gst_rtp_buffer_get_header_len()
16476 API: gst_rtp_buffer_get_extension_data()
16478 2007-09-03 19:19:35 +0000 Wim Taymans <wim.taymans@gmail.com>
16480 gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
16481 Original commit message from CVS:
16482 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16483 (gst_base_rtp_depayload_class_init),
16484 (gst_base_rtp_depayload_set_gst_timestamp):
16485 Add some more docs for the queue-delay property and fix a typo in a
16487 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
16490 2007-09-03 19:17:33 +0000 Wim Taymans <wim.taymans@gmail.com>
16492 gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
16493 Original commit message from CVS:
16494 * gst-libs/gst/audio/gstbaseaudiosink.c:
16495 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
16496 (gst_base_audio_sink_change_state):
16497 When skew slaving, try to hover around the middle of a segment so that
16498 we at most drift by half a segment.
16499 If we are aligning in the oposite direction of the clock skew, we don't
16502 2007-08-31 21:07:20 +0000 Wim Taymans <wim.taymans@gmail.com>
16504 gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
16505 Original commit message from CVS:
16506 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16507 (gst_base_rtp_depayload_setcaps),
16508 (gst_base_rtp_depayload_set_gst_timestamp):
16509 Be less silly with the segment start, just apply the clock-base to the
16512 2007-08-31 15:58:30 +0000 Wim Taymans <wim.taymans@gmail.com>
16514 gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
16515 Original commit message from CVS:
16516 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16517 (gst_base_rtp_depayload_class_init),
16518 (gst_base_rtp_depayload_finalize),
16519 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
16520 (gst_base_rtp_depayload_handle_sink_event),
16521 (gst_base_rtp_depayload_set_gst_timestamp),
16522 (gst_base_rtp_depayload_change_state):
16523 * gst-libs/gst/rtp/gstbasertpdepayload.h:
16524 Deprecate the queue handling thread thing and remove the code.
16525 Use new method to calculate the extended timestamp.
16527 2007-08-31 15:21:13 +0000 Wim Taymans <wim.taymans@gmail.com>
16529 gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
16530 Original commit message from CVS:
16531 * gst-libs/gst/rtp/gstrtcpbuffer.c:
16532 (gst_rtcp_packet_sdes_copy_entry):
16533 Use g_strndup which does exactly what we want.
16534 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
16535 (gst_rtp_buffer_ext_timestamp):
16536 * gst-libs/gst/rtp/gstrtpbuffer.h:
16537 Add helper function to compare seqnums.
16538 Add helper function to calculate extended timestamps.
16539 API: gst_rtp_buffer_compare_seqnum()
16540 API: gst_rtp_buffer_ext_timestamp()
16542 2007-08-30 21:59:23 +0000 Wim Taymans <wim.taymans@gmail.com>
16544 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
16545 Original commit message from CVS:
16546 * gst-libs/gst/rtp/gstrtcpbuffer.c:
16547 (gst_rtcp_packet_sdes_get_entry),
16548 (gst_rtcp_packet_sdes_copy_entry):
16549 * gst-libs/gst/rtp/gstrtcpbuffer.h:
16550 Fix and document SDES item data function.
16551 Add new function that makes a proper copy of SDES item data.
16552 API: gst_rtcp_packet_sdes_copy_entry()
16554 2007-08-30 07:29:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16556 The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
16557 Original commit message from CVS:
16560 The tcp and subparse plugins are under gst, but not totaly free of
16561 dependencies. Handle selection inconfigure.ac, so that they show up
16562 on the final list of what is build and what is not. Maybe they should
16563 better be moved to ext.
16565 2007-08-30 06:58:46 +0000 Daniel Díaz <yosoy@danieldiaz.org>
16567 Check if libxml provides HTML parser which subparse needs.
16568 Original commit message from CVS:
16569 Patch by: Daniel Díaz <yosoy@danieldiaz.org>
16572 Check if libxml provides HTML parser which subparse needs.
16575 2007-08-29 14:22:04 +0000 Tim-Philipp Müller <tim@centricular.net>
16577 ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
16578 Original commit message from CVS:
16579 * ext/alsa/gstalsa.c:
16580 Fix typo and compilation on big endian systems.
16582 2007-08-29 12:16:46 +0000 Tim-Philipp Müller <tim@centricular.net>
16584 gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
16585 Original commit message from CVS:
16586 * gst/subparse/gstssaparse.c:
16587 Convert SSA newline codes into actual newline characters (#470766).
16589 2007-08-28 14:58:17 +0000 Tim-Philipp Müller <tim@centricular.net>
16591 API: also add gst_install_plugins_supported() while we're at it (see #470456).
16592 Original commit message from CVS:
16593 * docs/libs/gst-plugins-base-libs-sections.txt:
16594 * gst-libs/gst/pbutils/install-plugins.c:
16595 * gst-libs/gst/pbutils/install-plugins.h:
16596 * tests/check/libs/pbutils.c:
16597 API: also add gst_install_plugins_supported() while we're at it
16600 2007-08-28 14:23:55 +0000 Tim-Philipp Müller <tim@centricular.net>
16602 API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
16603 Original commit message from CVS:
16604 * docs/libs/gst-plugins-base-libs-sections.txt:
16605 * gst-libs/gst/pbutils/missing-plugins.c:
16606 * gst-libs/gst/pbutils/missing-plugins.h:
16607 * tests/check/libs/pbutils.c:
16608 API: add gst_missing_*_installer_detail_new() convenience API so
16609 that applications that know exactly what they're missing can request
16610 installer detail strings for those items directly instead of having
16611 to first create a dummy missing-plugin message and then get the
16612 installer detail string from that. Fixes #470456.
16614 2007-08-27 11:59:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16616 gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
16617 Original commit message from CVS:
16618 * gst/playback/gstdecodebin.c: (close_pad_link):
16619 We need to set up delayed-linking whenever the caps are non-fixed,
16620 not just when there are multiple types - use gst_pad_is_fixed()
16623 2007-08-26 14:14:33 +0000 Tim-Philipp Müller <tim@centricular.net>
16625 gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
16626 Original commit message from CVS:
16627 * gst-libs/gst/pbutils/missing-plugins.c:
16628 (gst_missing_plugin_message_get_installer_detail):
16629 Add missing separator in PID fallback case.
16631 2007-08-24 15:28:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16633 ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
16634 Original commit message from CVS:
16635 * ext/alsa/Makefile.am:
16636 There is no GST_PLUGINS_BASE_LIBS defined.
16637 * ext/alsa/gstalsa.c:
16638 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
16639 * ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
16640 Add support for ALSA 24-bit formats.
16641 snd_pcm_delay can return an error code, especially
16642 during XRUNS. In that case, the best we can do is assume
16644 * gst/audioconvert/Makefile.am:
16645 Add flags from -base before any more-remote dependencies.
16647 2007-08-23 20:45:45 +0000 Davyd <davyd@madeley.id.au>
16649 gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
16650 Original commit message from CVS:
16651 Based on a patch by: Davyd <davyd at madeley dot id dot au>
16652 * gst/volume/gstvolume.c: (volume_choose_func),
16653 (volume_update_real_volume), (gst_volume_set_volume),
16654 (gst_volume_init), (volume_process_int32),
16655 (volume_process_int32_clamp), (volume_process_int24),
16656 (volume_process_int24_clamp), (volume_process_int16),
16657 (volume_process_int16_clamp), (volume_process_int8),
16658 (volume_process_int8_clamp), (volume_update_volume), (plugin_init):
16659 * gst/volume/gstvolume.h:
16660 Add support for int32, int24 and int8 to the volume element.
16663 2007-08-23 12:37:42 +0000 Tim-Philipp Müller <tim@centricular.net>
16665 tests/examples/Makefile.am: Fix even more.
16666 Original commit message from CVS:
16667 * tests/examples/Makefile.am:
16670 2007-08-23 10:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16672 Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
16673 Original commit message from CVS:
16675 * docs/libs/Makefile.am:
16676 * docs/libs/gst-plugins-base-libs-docs.sgml:
16677 * docs/libs/gst-plugins-base-libs-sections.txt:
16678 * ext/gnomevfs/gstgnomevfssrc.c:
16679 * ext/gnomevfs/gstgnomevfssrc.h:
16680 * gst-libs/gst/Makefile.am:
16681 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
16682 * pkgconfig/gstreamer-plugins-base.pc.in:
16683 * sys/v4l/v4lsrc_calls.c:
16684 * tests/examples/Makefile.am:
16685 * win32/common/config.h:
16686 Revert unwanted commit. many thanks to moap. I want a fix for
16687 https://thomas.apestaart.org/moap/trac/ticket/239
16689 2007-08-23 08:33:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16691 Original commit message from CVS:
16692 reviewed by: <delete if not using a buddy>
16693 patch by: <delete if not someone else's patch>
16695 * docs/libs/Makefile.am:
16696 * docs/libs/gst-plugins-base-libs-docs.sgml:
16697 * docs/libs/gst-plugins-base-libs-sections.txt:
16698 * ext/gnomevfs/gstgnomevfssrc.c:
16699 * ext/gnomevfs/gstgnomevfssrc.h:
16700 * gst-libs/gst/Makefile.am:
16701 * gst-libs/gst/audio/gstaudiofilter.h:
16702 * gst/typefind/gsttypefindfunctions.c:
16703 * gst/volume/gstvolume.c:
16704 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
16705 * pkgconfig/gstreamer-plugins-base.pc.in:
16706 * sys/v4l/v4lsrc_calls.c:
16707 * tests/examples/Makefile.am:
16708 * win32/common/config.h:
16710 2007-08-22 15:29:04 +0000 Wim Taymans <wim.taymans@gmail.com>
16712 gst-libs/gst/audio/audio.c: Clarify the docs a little.
16713 Original commit message from CVS:
16714 * gst-libs/gst/audio/audio.c:
16715 Clarify the docs a little.
16717 2007-08-22 11:20:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16719 gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
16720 Original commit message from CVS:
16721 * gst/volume/gstvolume.c:
16722 Enable liboil for float and add more details about problems with
16725 2007-08-21 15:43:24 +0000 Wim Taymans <wim.taymans@gmail.com>
16727 sys/v4l/gstv4lsrc.c: Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
16728 Original commit message from CVS:
16729 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
16730 Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
16732 2007-08-21 12:08:43 +0000 Wim Taymans <wim.taymans@gmail.com>
16734 ext/vorbis/vorbisdec.c: When calculating the first timestamp of the buffers, don't go below 0 and clip the samples be...
16735 Original commit message from CVS:
16736 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
16737 When calculating the first timestamp of the buffers, don't go below 0
16738 and clip the samples because the offset was on the eos page.
16741 2007-08-21 11:42:39 +0000 Wim Taymans <wim.taymans@gmail.com>
16743 ext/ogg/gstoggdemux.c: Also submit the eos page when trying to find the first timestamp.
16744 Original commit message from CVS:
16745 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
16746 (gst_ogg_demux_collect_chain_info):
16747 Also submit the eos page when trying to find the first timestamp.
16750 2007-08-17 15:24:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16752 gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
16753 Original commit message from CVS:
16754 * gst-libs/gst/audio/audio.h:
16755 Use gst_util_uint64_scale() instead of doing the math
16756 with double for GST_FRAMES_TO_CLOCK_TIME() and
16757 GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
16758 prevents rounding errors. Fixes #467667.
16760 2007-08-17 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
16762 gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
16763 Original commit message from CVS:
16764 * gst-libs/gst/rtsp/gstrtspconnection.c:
16765 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
16766 (gst_rtsp_connection_read), (gst_rtsp_connection_poll):
16767 * gst-libs/gst/rtsp/gstrtspconnection.h:
16769 On shutdown, don't read the control socket yet.
16770 Set timeout value correctly in all cases.
16771 Add function to check if the server accepts reads or writes.
16772 API: gst_rtsp_connection_poll()
16773 * gst-libs/gst/rtsp/gstrtspdefs.h:
16774 Fix compilation with -pedantic.
16775 Add enum for _poll.
16777 2007-08-16 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
16779 gst-libs/gst/app/gstappsink.c: Override the preroll vmethod instead of overriding the render method twice.
16780 Original commit message from CVS:
16781 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
16782 Override the preroll vmethod instead of overriding the render method
16785 2007-08-16 16:06:21 +0000 Olivier Crete <tester@tester.ca>
16787 gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
16788 Original commit message from CVS:
16789 Patch by: Olivier Crete <tester at tester ca>
16790 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
16791 (gst_basertppayload_getcaps):
16792 * gst-libs/gst/rtp/gstbasertppayload.h:
16793 Add getcaps vfunc to basertppayload. See #465146.
16795 2007-08-16 11:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
16797 gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
16798 Original commit message from CVS:
16799 * gst/playback/gstplaybasebin.c: (queue_threshold_reached):
16800 Only post buffering messages when we are a stream.
16802 2007-08-15 17:05:45 +0000 Tim-Philipp Müller <tim@centricular.net>
16804 gst-libs/gst/pbutils/: Small docs fix and addition.
16805 Original commit message from CVS:
16806 * gst-libs/gst/pbutils/install-plugins.c:
16807 * gst-libs/gst/pbutils/missing-plugins.c:
16808 Small docs fix and addition.
16810 2007-08-14 17:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
16812 gst-libs/gst/app/gstappsink.c: Don't use new API.
16813 Original commit message from CVS:
16814 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
16817 2007-08-14 17:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
16819 gst-libs/gst/app/gstappsink.*: Make love to appsink.
16820 Original commit message from CVS:
16821 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
16822 (gst_app_sink_class_init), (gst_app_sink_dispose),
16823 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
16824 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
16825 (gst_app_sink_render), (gst_app_sink_get_caps),
16826 (gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
16827 (gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
16828 * gst-libs/gst/app/gstappsink.h:
16829 Make love to appsink.
16830 Make it support pulling of the preroll buffer.
16831 Add docs and debug statements.
16832 Fix some races wrt to EOS handling and stopping.
16834 Implement FLUSHING.
16835 API: gst_app_sink_pull_preroll()
16837 2007-08-13 15:37:29 +0000 Tim-Philipp Müller <tim@centricular.net>
16839 tests/icles/: Add a dumb little test for textoverlay alignments.
16840 Original commit message from CVS:
16841 * tests/icles/.cvsignore:
16842 * tests/icles/Makefile.am:
16843 * tests/icles/test-textoverlay.c:
16844 Add a dumb little test for textoverlay alignments.
16846 2007-08-13 15:26:54 +0000 Dan Williams <dcbw@redhat.com>
16848 ext/pango/gsttextoverlay.*: API: add "line-alignment" property (#459334). Add gtk-doc blurb for "silent" property so ...
16849 Original commit message from CVS:
16850 Patch by: Dan Williams <dcbw redhat com>
16851 * ext/pango/gsttextoverlay.c:
16852 * ext/pango/gsttextoverlay.h:
16853 API: add "line-alignment" property (#459334). Add gtk-doc blurb for
16854 "silent" property so there's a Since tag in the API reference.
16856 2007-08-13 11:21:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
16860 Original commit message from CVS:
16863 2007-08-12 16:30:36 +0000 Wim Taymans <wim.taymans@gmail.com>
16865 gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
16866 Original commit message from CVS:
16867 * gst-libs/gst/rtp/gstbasertppayload.c:
16868 (gst_basertppayload_set_outcaps):
16869 * gst-libs/gst/rtp/gstbasertppayload.h:
16870 Improve caps negotiation so that downstream elements can confiure
16871 certain RTP properties by fixing them on the caps. See #465146.
16874 2007-08-11 12:39:51 +0000 Tim-Philipp Müller <tim@centricular.net>
16876 Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
16877 Original commit message from CVS:
16878 * docs/libs/gst-plugins-base-libs-sections.txt:
16879 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16880 * gst-libs/gst/rtp/gstbasertpdepayload.h:
16881 Mark as deprecated some macros which were presumably meant to be
16882 private API and accidentally exposed in the public header file.
16883 Also actually _init() lock (only works at the moment because the
16884 struct is zeroed out when created and the initial values in the
16885 mutex struct are zeroes too). (#459585)
16887 2007-08-10 17:35:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16889 docs/libs/Makefile.am: Remove cruft and do some cleanups.
16890 Original commit message from CVS:
16891 * docs/libs/Makefile.am:
16892 Remove cruft and do some cleanups.
16893 * docs/libs/gst-plugins-base-libs-docs.sgml:
16894 Prepare for comming gtkdoc features (rebase against online docs).
16896 2007-08-10 13:55:44 +0000 Michael Smith <msmith@xiph.org>
16898 gst/audiorate/gstaudiorate.c: Debug output fixes.
16899 Original commit message from CVS:
16900 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
16901 Debug output fixes.
16902 * tests/check/elements/audiorate.c: (do_perfect_stream_test),
16904 Change the number of buffers used; 500 is too many and leads to
16907 2007-08-10 10:08:05 +0000 Tim-Philipp Müller <tim@centricular.net>
16909 gst/: Printf format fixes (#465028).
16910 Original commit message from CVS:
16911 * gst/playback/gstqueue2.c:
16912 * gst/videorate/gstvideorate.c:
16913 Printf format fixes (#465028).
16915 2007-08-09 15:44:02 +0000 Michael Smith <msmith@xiph.org>
16917 gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
16918 Original commit message from CVS:
16919 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
16920 If we have a large (> 1 second) discontinuity, push a series of
16921 smaller buffers rather than a single very large buffer. Avoids
16922 unreasonably large single buffer allocations when encountering a
16924 * tests/check/elements/audiorate.c: (GST_START_TEST),
16926 Add a test for this.
16928 2007-08-09 12:06:43 +0000 Josep Torra Valles <josep@fluendo.com>
16930 gst/playback/gstplaybasebin.c: Fixes: #465015
16931 Original commit message from CVS:
16932 * gst/playback/gstplaybasebin.c: (group_commit),
16933 (queue_remove_probe), (queue_threshold_reached):
16934 Patch by: Josep Torra Valles <josep@fluendo.com>
16936 Make sure we remove the check_queues buffer probe from the
16937 correct queue to avoid racily going back to "buffering 99%" when
16938 buffering is actually complete.
16939 Also, fix the spelling of Josep's surname in the ChangeLog.
16941 2007-08-09 11:37:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16943 ext/ogg/gstoggmux.c: Do not leak oggmux instance.
16944 Original commit message from CVS:
16945 * ext/ogg/gstoggmux.c:
16946 Do not leak oggmux instance.
16947 * ext/vorbis/vorbisenc.c:
16950 2007-08-09 10:51:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
16952 po/: Updated translations.
16953 Original commit message from CVS:
16959 Updated translations.
16961 2007-08-08 16:07:21 +0000 Yang Hong <hongyang@redflag-linux.com>
16963 ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
16964 Original commit message from CVS:
16965 patch by: Yang Hong <hongyang@redflag-linux.com>
16966 * ext/pango/gsttextoverlay.c:
16967 * ext/pango/gsttextoverlay.h:
16968 Add 'silent' property to GstTimeOverlay. Fixes #462979
16970 2007-08-08 15:05:22 +0000 Josep Torre Valles <josep@fluendo.com>
16972 Add connection-speed property. Fixes #464690.
16973 Original commit message from CVS:
16974 Patch by: Josep Torre Valles <josep@fluendo.com>
16975 * docs/plugins/gst-plugins-base-plugins.args:
16976 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
16977 (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
16978 (gst_uri_decode_bin_get_property), (gen_source_element):
16979 Add connection-speed property. Fixes #464690.
16981 2007-08-07 15:13:46 +0000 Damien Lespiau <damien.lespiau@gmail.com>
16983 Fix compilation on windows. Fixes #464320.
16984 Original commit message from CVS:
16985 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
16987 * gst-libs/gst/rtsp/Makefile.am:
16988 * gst-libs/gst/rtsp/gstrtspconnection.c:
16989 (gst_rtsp_connection_connect):
16990 Fix compilation on windows. Fixes #464320.
16992 2007-08-07 14:14:54 +0000 Josep Torre Valles <josep@fluendo.com>
16994 gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
16995 Original commit message from CVS:
16996 Patch by: Josep Torre Valles <josep@fluendo.com>
16997 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
16998 (gst_play_base_bin_init), (queue_threshold_reached),
16999 (gen_source_element), (setup_substreams),
17000 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
17001 (gst_play_base_bin_get_streaminfo_value_array):
17002 * gst/playback/gstplaybasebin.h:
17003 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
17004 (gst_play_bin_set_property), (gst_play_bin_get_property),
17005 (gst_play_bin_handle_redirect_message):
17006 Move connection-speed property from playbin to playbasebin so that we
17007 can also configure it in source elements that have the connection-speed
17008 property. Fixes #464028.
17009 Add some debug info here and there.
17011 2007-08-06 16:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17013 gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
17014 Original commit message from CVS:
17015 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
17016 Properly respond to conversion queries. Fixes #464079.
17018 2007-08-03 19:53:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17020 gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
17021 Original commit message from CVS:
17022 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
17023 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
17024 (gst_audio_test_src_init_sine_table),
17025 (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
17026 * gst/audiotestsrc/gstaudiotestsrc.h:
17027 Add float/double and int32 support to audiotestsrc. Fixes #460422.
17028 Also set the default volume to the default value specified in the
17031 2007-08-03 19:40:14 +0000 Jens Granseuer <jensgr@gmx.net>
17033 gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
17034 Original commit message from CVS:
17035 Patch by: Jens Granseuer <jensgr at gmx dot net>
17036 * gst/audioconvert/gstaudioquantize.c:
17037 Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
17039 2007-08-03 15:44:01 +0000 Wim Taymans <wim.taymans@gmail.com>
17041 gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
17042 Original commit message from CVS:
17043 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
17044 Add rdt manager for rdt transport.
17045 Fix parsing of RDT transport.
17047 2007-08-03 14:43:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17049 configure.ac: Back to CVS
17050 Original commit message from CVS:
17054 === release 0.10.14 ===
17056 2007-08-03 14:41:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17062 * docs/plugins/gst-plugins-base-plugins.args:
17063 * docs/plugins/inspect/plugin-adder.xml:
17064 * docs/plugins/inspect/plugin-alsa.xml:
17065 * docs/plugins/inspect/plugin-audioconvert.xml:
17066 * docs/plugins/inspect/plugin-audiorate.xml:
17067 * docs/plugins/inspect/plugin-audioresample.xml:
17068 * docs/plugins/inspect/plugin-audiotestsrc.xml:
17069 * docs/plugins/inspect/plugin-cdparanoia.xml:
17070 * docs/plugins/inspect/plugin-decodebin.xml:
17071 * docs/plugins/inspect/plugin-decodebin2.xml:
17072 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
17073 * docs/plugins/inspect/plugin-gdp.xml:
17074 * docs/plugins/inspect/plugin-gnomevfs.xml:
17075 * docs/plugins/inspect/plugin-libvisual.xml:
17076 * docs/plugins/inspect/plugin-ogg.xml:
17077 * docs/plugins/inspect/plugin-pango.xml:
17078 * docs/plugins/inspect/plugin-playbin.xml:
17079 * docs/plugins/inspect/plugin-subparse.xml:
17080 * docs/plugins/inspect/plugin-tcp.xml:
17081 * docs/plugins/inspect/plugin-theora.xml:
17082 * docs/plugins/inspect/plugin-typefindfunctions.xml:
17083 * docs/plugins/inspect/plugin-video4linux.xml:
17084 * docs/plugins/inspect/plugin-videorate.xml:
17085 * docs/plugins/inspect/plugin-videoscale.xml:
17086 * docs/plugins/inspect/plugin-videotestsrc.xml:
17087 * docs/plugins/inspect/plugin-volume.xml:
17088 * docs/plugins/inspect/plugin-vorbis.xml:
17089 * docs/plugins/inspect/plugin-ximagesink.xml:
17090 * docs/plugins/inspect/plugin-xvimagesink.xml:
17091 * gst-plugins-base.doap:
17092 * win32/common/config.h:
17094 Original commit message from CVS:
17097 2007-08-03 14:24:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17115 Original commit message from CVS:
17118 2007-07-27 17:37:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17120 tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
17121 Original commit message from CVS:
17122 * tests/check/libs/audio.c: (GST_START_TEST):
17123 Fix the test to reflect the behaviour of gst_audio_clip_buffer.
17125 2007-07-27 17:10:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17127 gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
17128 Original commit message from CVS:
17129 * gst-libs/gst/audio/audio.c:
17130 When clipping a buffer with no timestamp, assume it is
17131 within the segment without warnings.
17134 2007-07-27 11:16:23 +0000 Wim Taymans <wim.taymans@gmail.com>
17136 gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
17137 Original commit message from CVS:
17138 * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
17139 Fire the signal on the object, not the interface.
17141 2007-07-27 09:17:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17143 gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
17144 Original commit message from CVS:
17145 * gst-libs/gst/rtsp/.cvsignore:
17146 Ber. Don't include the full path, idiot.
17148 2007-07-27 08:29:29 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17150 gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
17151 Original commit message from CVS:
17152 * gst-libs/gst/rtsp/.cvsignore:
17153 Ignore generated files.
17155 2007-07-26 19:57:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17157 gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
17158 Original commit message from CVS:
17159 * gst-libs/gst/interfaces/Makefile.am:
17160 * gst-libs/gst/interfaces/interfaces-marshal.list:
17161 * gst-libs/gst/interfaces/rtspextension.c:
17162 * gst-libs/gst/interfaces/rtspextension.h:
17163 * gst-libs/gst/rtsp/Makefile.am:
17164 * gst-libs/gst/rtsp/gstrtsp.h:
17165 * gst-libs/gst/rtsp/gstrtspextension.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/rtsp/gstrtspextension.h:
17174 * gst-libs/gst/rtsp/rtsp-marshal.list:
17175 Move the rtspextension.h interface into gstrtspextension.h
17176 as part of libgstrtsp instead of libgstinterfaces, because it's
17177 only for use within plugins, not applications.
17178 Add stuff to do the enum & marshal generation needed in libgstrtsp now.
17179 Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
17180 signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
17183 2007-07-26 15:48:01 +0000 Wim Taymans <wim.taymans@gmail.com>
17185 gst-libs/gst/interfaces/: Fix marshaller for the send signal.
17186 Original commit message from CVS:
17187 * gst-libs/gst/interfaces/Makefile.am:
17188 * gst-libs/gst/interfaces/interfaces-marshal.list:
17189 * gst-libs/gst/interfaces/rtspextension.c:
17190 (gst_rtsp_extension_iface_init),
17191 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
17192 * gst-libs/gst/interfaces/rtspextension.h:
17193 Fix marshaller for the send signal.
17194 Add URL to stream selection interface method.
17196 2007-07-26 15:35:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17198 gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
17199 Original commit message from CVS:
17200 * gst-libs/gst/riff/Makefile.am:
17201 Pull in our dependencies from -base before those from outside.
17203 2007-07-26 14:33:01 +0000 Wim Taymans <wim.taymans@gmail.com>
17205 API: gst_rtsp_base64_decode_ip()
17206 Original commit message from CVS:
17207 * docs/libs/gst-plugins-base-libs-sections.txt:
17208 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
17209 * gst-libs/gst/rtsp/gstrtspbase64.h:
17210 API: gst_rtsp_base64_decode_ip()
17211 Added function to decode Base64 in-place.
17213 2007-07-26 14:08:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17215 tests/check/libs/.cvsignore: Ignore the mixer test binary.
17216 Original commit message from CVS:
17217 * tests/check/libs/.cvsignore:
17218 Ignore the mixer test binary.
17220 2007-07-26 10:00:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17222 ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
17223 Original commit message from CVS:
17224 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
17225 Gratuitous comment change to trigger a rebuild on the buildbots.
17227 2007-07-25 18:20:36 +0000 Wim Taymans <wim.taymans@gmail.com>
17229 gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
17230 Original commit message from CVS:
17231 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
17232 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
17233 (gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
17234 (gst_sdp_media_get_format), (gst_sdp_media_get_information),
17235 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
17236 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
17237 (gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
17238 (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
17239 (gst_sdp_media_get_attribute_val):
17240 * gst-libs/gst/sdp/gstsdpmessage.h:
17241 Constify args where we can.
17243 2007-07-25 18:18:49 +0000 Wim Taymans <wim.taymans@gmail.com>
17245 gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
17246 Original commit message from CVS:
17247 * gst-libs/gst/interfaces/Makefile.am:
17248 * gst-libs/gst/interfaces/rtspextension.c:
17249 (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
17250 (gst_rtsp_extension_detect_server),
17251 (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
17252 (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
17253 (gst_rtsp_extension_configure_stream),
17254 (gst_rtsp_extension_get_transports),
17255 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
17256 * gst-libs/gst/interfaces/rtspextension.h:
17257 Move interface for RTSP extensions from -good to here.
17258 Added helper methods to invoke interface methods.
17260 2007-07-25 11:22:30 +0000 Wim Taymans <wim.taymans@gmail.com>
17262 Fix some more RTSP docs.
17263 Original commit message from CVS:
17264 * docs/libs/gst-plugins-base-libs-sections.txt:
17265 * gst-libs/gst/rtsp/gstrtspdefs.h:
17266 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
17267 (gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
17268 (gst_rtsp_message_init_response),
17269 (gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
17270 (gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
17271 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
17272 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
17273 (gst_rtsp_message_get_body), (dump_key_value):
17274 * gst-libs/gst/rtsp/gstrtspmessage.h:
17275 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
17276 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
17277 (gst_rtsp_range_parse):
17278 * gst-libs/gst/rtsp/gstrtsprange.h:
17279 * gst-libs/gst/rtsp/gstrtsptransport.c:
17280 * gst-libs/gst/rtsp/gstrtspurl.c:
17281 Fix some more RTSP docs.
17282 Add some missing methods for dealing with messages.
17284 2007-07-24 19:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
17286 Added beginnings of RTSP documentation.
17287 Original commit message from CVS:
17288 * docs/libs/gst-plugins-base-libs-docs.sgml:
17289 * docs/libs/gst-plugins-base-libs-sections.txt:
17290 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
17291 * gst-libs/gst/rtsp/gstrtspbase64.h:
17292 * gst-libs/gst/rtsp/gstrtspconnection.c:
17293 (gst_rtsp_connection_connect), (add_auth_header),
17294 (gst_rtsp_connection_write), (gst_rtsp_connection_send),
17295 (read_body), (gst_rtsp_connection_receive),
17296 (gst_rtsp_connection_next_timeout),
17297 (gst_rtsp_connection_reset_timeout),
17298 (gst_rtsp_connection_set_auth):
17299 * gst-libs/gst/rtsp/gstrtspconnection.h:
17300 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
17301 * gst-libs/gst/rtsp/gstrtspdefs.h:
17302 * gst-libs/gst/rtsp/gstrtspmessage.h:
17303 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
17304 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
17305 (gst_rtsp_range_parse):
17306 * gst-libs/gst/rtsp/gstrtspurl.h:
17307 Added beginnings of RTSP documentation.
17309 2007-07-24 17:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
17311 Document the SDP library.
17312 Original commit message from CVS:
17313 * docs/libs/Makefile.am:
17314 * docs/libs/gst-plugins-base-libs-docs.sgml:
17315 * docs/libs/gst-plugins-base-libs-sections.txt:
17316 * gst-libs/gst/sdp/gstsdp.h:
17317 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
17318 (gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
17319 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
17320 (gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
17321 (gst_sdp_message_get_attribute_val),
17322 (gst_sdp_message_add_attribute), (gst_sdp_media_new),
17323 (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
17324 (gst_sdp_media_get_media), (gst_sdp_media_set_media),
17325 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
17326 (gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
17327 (gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
17328 (gst_sdp_media_get_format), (gst_sdp_media_add_format),
17329 (gst_sdp_media_get_information), (gst_sdp_media_set_information),
17330 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
17331 (gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
17332 (gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
17333 (gst_sdp_media_set_key), (gst_sdp_media_get_key),
17334 (gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
17335 (gst_sdp_media_get_attribute_val_n),
17336 (gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
17337 (print_media), (gst_sdp_message_dump):
17338 * gst-libs/gst/sdp/gstsdpmessage.h:
17339 Document the SDP library.
17340 Add some of the missing SDPMedia methods.
17342 2007-07-24 11:52:56 +0000 Wim Taymans <wim.taymans@gmail.com>
17344 Move SDP and RTSP from helper objects in -good to a reusable library.
17345 Original commit message from CVS:
17347 * gst-libs/gst/Makefile.am:
17348 * gst-libs/gst/rtsp/Makefile.am:
17349 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
17350 * gst-libs/gst/rtsp/gstrtspbase64.h:
17351 * gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
17352 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
17353 (add_auth_header), (add_date_header), (gst_rtsp_connection_write),
17354 (gst_rtsp_connection_send), (read_line), (read_string), (read_key),
17355 (parse_response_status), (parse_request_line), (parse_line),
17356 (gst_rtsp_connection_read), (read_body),
17357 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
17358 (gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
17359 (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
17360 (gst_rtsp_connection_set_auth):
17361 * gst-libs/gst/rtsp/gstrtspconnection.h:
17362 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
17363 (gst_rtsp_strresult), (gst_rtsp_method_as_text),
17364 (gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
17365 (gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
17366 (gst_rtsp_find_method):
17367 * gst-libs/gst/rtsp/gstrtspdefs.h:
17368 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
17369 (gst_rtsp_message_new), (gst_rtsp_message_init),
17370 (gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
17371 (gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
17372 (gst_rtsp_message_init_data), (gst_rtsp_message_unset),
17373 (gst_rtsp_message_free), (gst_rtsp_message_add_header),
17374 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
17375 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
17376 (gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
17377 (gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
17378 (gst_rtsp_message_dump):
17379 * gst-libs/gst/rtsp/gstrtspmessage.h:
17380 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
17381 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
17382 (gst_rtsp_range_parse), (gst_rtsp_range_free):
17383 * gst-libs/gst/rtsp/gstrtsprange.h:
17384 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
17385 (gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
17386 (gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
17387 (range_as_text), (rtsp_transport_mode_as_text),
17388 (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
17389 (gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
17390 (gst_rtsp_transport_free):
17391 * gst-libs/gst/rtsp/gstrtsptransport.h:
17392 * gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
17393 (gst_rtsp_url_free), (gst_rtsp_url_set_port),
17394 (gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
17395 * gst-libs/gst/rtsp/gstrtspurl.h:
17396 * gst-libs/gst/sdp/Makefile.am:
17397 * gst-libs/gst/sdp/gstsdp.h:
17398 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
17399 (gst_sdp_connection_init), (gst_sdp_bandwidth_init),
17400 (gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
17401 (gst_sdp_attribute_init), (gst_sdp_message_new),
17402 (gst_sdp_message_init), (gst_sdp_message_uninit),
17403 (gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
17404 (gst_sdp_media_uninit), (gst_sdp_media_free),
17405 (gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
17406 (gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
17407 (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
17408 (gst_sdp_message_add_zone), (gst_sdp_message_set_key),
17409 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
17410 (gst_sdp_message_get_attribute_val),
17411 (gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
17412 (gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
17413 (gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
17414 (gst_sdp_media_get_attribute_val_n),
17415 (gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
17416 (read_string), (read_string_del), (gst_sdp_parse_line),
17417 (gst_sdp_message_parse_buffer), (print_media),
17418 (gst_sdp_message_dump):
17419 * gst-libs/gst/sdp/gstsdpmessage.h:
17420 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
17421 Move SDP and RTSP from helper objects in -good to a reusable library.
17422 Use a proper gst_ namespace.
17424 2007-07-23 18:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17426 ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
17427 Original commit message from CVS:
17428 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
17429 (vorbis_dec_flush_decode):
17430 Use the new buffer clipping function from gstaudio here.
17432 2007-07-23 18:26:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17434 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
17435 Original commit message from CVS:
17436 * docs/libs/gst-plugins-base-libs-sections.txt:
17437 * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
17438 * gst-libs/gst/audio/audio.h:
17439 * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
17440 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
17441 Also add deprecation guards for gst_audio_structure_set_int() to the
17444 2007-07-23 14:45:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17446 docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
17447 Original commit message from CVS:
17448 * docs/libs/gst-plugins-base-libs-sections.txt:
17451 2007-07-23 11:18:35 +0000 Dan Williams <dcbw@redhat.com>
17453 gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
17454 Original commit message from CVS:
17455 Patch by: Dan Williams <dcbw at redhat dot com>
17456 * gst/playback/gstplaybasebin.c:
17457 (gst_play_base_bin_get_streaminfo_value_array):
17458 Don't return NULL when querying the stream info value array but instead
17459 return an empty array. Fixes #459204.
17461 2007-07-23 10:41:18 +0000 Tim-Philipp Müller <tim@centricular.net>
17463 gst/playback/gsturidecodebin.c: Init debug category before using it.
17464 Original commit message from CVS:
17465 * gst/playback/gsturidecodebin.c:
17466 Init debug category before using it.
17468 2007-07-21 09:56:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17470 gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
17471 Original commit message from CVS:
17472 * gst-libs/gst/interfaces/mixer.h:
17473 Add padding vars in place of the signal pointers
17474 when building with DISABLE_DEPRECATED so that the
17475 interface structure doesn't change size.
17477 2007-07-21 09:21:12 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
17480 Original commit message from CVS:
17481 * docs/libs/gst-plugins-base-libs-sections.txt:
17482 * ext/alsa/gstalsamixer.c:
17483 * ext/alsa/gstalsamixer.h:
17484 * ext/alsa/gstalsamixerelement.c:
17485 * ext/alsa/gstalsamixertrack.c:
17486 * gst-libs/gst/interfaces/mixer.c:
17487 * gst-libs/gst/interfaces/mixer.h:
17488 * gst-libs/gst/interfaces/mixeroptions.c:
17489 * gst-libs/gst/interfaces/mixeroptions.h:
17490 * gst-libs/gst/interfaces/mixertrack.c:
17491 * gst-libs/gst/interfaces/mixertrack.h:
17492 * tests/check/Makefile.am:
17493 * tests/check/libs/mixer.c:
17494 Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
17496 Add support for notifying mixer changes on the message bus, and
17497 implement it in alsamixer.
17498 API: gst_mixer_get_mixer_flags
17499 API: gst_mixer_message_parse_mute_toggled
17500 API: gst_mixer_message_parse_record_toggled
17501 API: gst_mixer_message_parse_volume_changed
17502 API: gst_mixer_message_parse_option_changed
17503 API: GstMixerMessageType
17506 2007-07-20 16:09:03 +0000 Michael Smith <msmith@xiph.org>
17508 sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
17509 Original commit message from CVS:
17510 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
17511 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
17512 xcontext->im_format is only for testing XShm support (as the header
17513 file comments document). Use xvimage->im_format for everything else.
17514 Avoids spurious warnings on buffer allocation before setcaps.
17516 2007-07-20 07:22:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17518 tests/: We should use $(LIBM).
17519 Original commit message from CVS:
17520 * tests/examples/volume/Makefile.am:
17521 * tests/icles/Makefile.am:
17522 We should use $(LIBM).
17524 2007-07-20 06:13:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17526 tests/icles/Makefile.am: This needs -lm.
17527 Original commit message from CVS:
17528 * tests/icles/Makefile.am:
17531 2007-07-18 07:35:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17533 Add stdlib include (free, atoi, exit).
17534 Original commit message from CVS:
17535 * examples/app/appsrc_ex.c:
17536 * examples/switch/switcher.c:
17537 * ext/neon/gstneonhttpsrc.c:
17538 * ext/timidity/gstwildmidi.c:
17539 * ext/x264/gstx264enc.c:
17540 * gst/mve/mveaudioenc.c: (mve_compress_audio):
17541 * gst/rtpmanager/gstrtpclient.c:
17542 * gst/rtpmanager/gstrtpjitterbuffer.c:
17543 * gst/spectrum/demo-audiotest.c:
17544 * gst/spectrum/demo-osssrc.c:
17545 * sys/dvb/gstdvbsrc.c:
17546 Add stdlib include (free, atoi, exit).
17548 2007-07-16 10:10:28 +0000 Wim Taymans <wim.taymans@gmail.com>
17550 gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
17551 Original commit message from CVS:
17552 * gst-libs/gst/rtp/gstbasertppayload.c:
17553 (gst_basertppayload_class_init), (gst_basertppayload_init),
17554 (gst_basertppayload_set_property),
17555 (gst_basertppayload_get_property):
17556 Don't break ABI, restore previous ranges. Keep the default random
17557 selection of timestamp and seqnum offset but as soon as the app sets a
17558 specific value, use that one.
17560 2007-07-14 18:33:15 +0000 Bastien Nocera <hadess@hadess.net>
17562 sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
17563 Original commit message from CVS:
17564 Patch by: Bastien Nocera <hadess at hadess dot net>
17565 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
17566 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
17567 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
17568 * sys/xvimage/xvimagesink.h:
17569 Add option to turn off double-buffering for debugging purposes.
17572 2007-07-14 18:20:41 +0000 Jorn Baayen <jorn@openedhand.com>
17574 sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
17575 Original commit message from CVS:
17576 Patch by: Jorn Baayen <jorn at openedhand dot com>
17577 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
17578 (gst_ximagesink_set_property), (gst_ximagesink_get_property),
17579 (gst_ximagesink_init), (gst_ximagesink_class_init):
17580 * sys/ximage/ximagesink.h:
17581 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
17582 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
17583 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
17584 * sys/xvimage/xvimagesink.h:
17585 add 'handle-expose' property. Useful for video widgets which may want to
17586 be in control of Expose behaviour. Fixes #380625
17588 2007-07-14 17:23:42 +0000 Wim Taymans <wim.taymans@gmail.com>
17590 gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
17591 Original commit message from CVS:
17592 * gst-libs/gst/rtp/gstbasertppayload.c:
17593 (gst_basertppayload_class_init), (gst_basertppayload_init),
17594 (gst_basertppayload_event), (gst_basertppayload_push),
17595 (gst_basertppayload_set_property),
17596 (gst_basertppayload_get_property),
17597 (gst_basertppayload_change_state):
17598 * gst-libs/gst/rtp/gstbasertppayload.h:
17599 Fix ranges of rtp payloader properties so that the full range can be
17600 used in addition to -1 (random).
17601 Fix wrong seqnum reporting in caps.
17604 2007-07-13 18:12:19 +0000 Wim Taymans <wim.taymans@gmail.com>
17606 gst/videorate/gstvideorate.c: Use boilerplate.
17607 Original commit message from CVS:
17608 * gst/videorate/gstvideorate.c: (gst_video_rate_init),
17609 (gst_video_rate_query):
17611 Add latency query, might not be perfect yet but already works a lot
17612 better. Fixes #442557.
17614 2007-07-13 16:05:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17616 sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
17617 Original commit message from CVS:
17618 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
17619 (gst_xvimagesink_setcaps):
17620 * sys/xvimage/xvimagesink.h:
17621 After a caps change, redraw our borders to avoid garbage left there
17622 when the image format changes to a smaller size, like 16:9 -> 4:3
17623 Also, hold the flow_lock a bit longer in the set_caps while we're
17624 fiddling with the xcontext.
17626 2007-07-13 16:02:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17628 Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
17629 Original commit message from CVS:
17632 * tests/Makefile.am:
17633 Remove bogus check for libcheck, since we check for
17634 gstreamer-check and it pulls in the required info from there, and we
17635 weren't actually _using_ the information for libcheck ourselves
17638 2007-07-13 15:52:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17640 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
17641 Original commit message from CVS:
17642 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
17643 (gst_ffmpeg_caps_to_pixfmt):
17644 Fix the r_mask test for RGBA32 on little-endian.
17645 Fix a stupid typo that would have obviously broken
17646 compilation on big-endian, if anyone was testing.
17648 2007-07-12 15:02:43 +0000 Wim Taymans <wim.taymans@gmail.com>
17650 gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
17651 Original commit message from CVS:
17652 * gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
17653 (paint_hline_str4):
17654 * gst/videotestsrc/videotestsrc.h:
17655 Add alpha to the color struct.
17656 Use a default alpha value of 255 instead of 128.
17658 2007-07-12 12:01:20 +0000 Wim Taymans <wim.taymans@gmail.com>
17660 gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
17661 Original commit message from CVS:
17662 * gst/playback/gstplaybasebin.c: (no_more_pads_full),
17664 Clear the dynamic pads counter when starting a new uri. This makes
17665 reusing playbin work again.
17668 2007-07-12 11:13:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17670 configure.ac: Use pkg-config to locate check.
17671 Original commit message from CVS:
17673 Use pkg-config to locate check.
17675 2007-07-11 23:12:12 +0000 Tim-Philipp Müller <tim@centricular.net>
17677 Fix 'make check' build against core CVS.
17678 Original commit message from CVS:
17680 * tests/check/elements/volume.c: (GST_START_TEST):
17681 Fix 'make check' build against core CVS.
17683 2007-07-10 20:46:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17685 gst-libs/gst/: Make gtk-doc happy.
17686 Original commit message from CVS:
17687 * gst-libs/gst/interfaces/propertyprobe.c:
17688 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
17689 * gst-libs/gst/tag/gstvorbistag.c:
17690 Make gtk-doc happy.
17692 2007-07-08 13:07:38 +0000 Tim-Philipp Müller <tim@centricular.net>
17694 gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
17695 Original commit message from CVS:
17696 * gst-libs/gst/audio/gstbaseaudiosink.c:
17697 (gst_base_audio_sink_callback):
17698 Quick hack to make audiosinks stop at EOS when operating in
17699 pull-mode; needs to be fixed properly some day.
17701 2007-07-06 18:19:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17703 docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
17704 Original commit message from CVS:
17705 * docs/libs/gst-plugins-base-libs-sections.txt:
17706 Fix location of includes in the docs.
17708 2007-07-06 11:40:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17710 gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
17711 Original commit message from CVS:
17712 * gst/ffmpegcolorspace/avcodec.h:
17713 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
17714 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
17715 (gst_ffmpegcsp_avpicture_fill):
17716 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
17717 (img_get_alpha_info):
17718 Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
17719 of the existing BGRA32 and RGBA32 formats with the alpha at the other
17720 end of the word. Partially fixes #451908
17722 2007-07-05 08:43:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17724 docs/: Simplify --extra-dir as gtkdoc scans recursively.
17725 Original commit message from CVS:
17726 * docs/libs/Makefile.am:
17727 * docs/plugins/Makefile.am:
17728 Simplify --extra-dir as gtkdoc scans recursively.
17730 2007-07-03 11:52:47 +0000 Wim Taymans <wim.taymans@gmail.com>
17732 gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
17733 Original commit message from CVS:
17734 * gst/adder/gstadder.c: (gst_adder_sink_getcaps),
17735 (gst_adder_request_new_pad):
17736 Make getcaps more robust by not using the proxycaps function. This makes
17737 sure that we don't end up recursively calling getcaps upstream.
17740 2007-06-29 17:21:18 +0000 Wim Taymans <wim.taymans@gmail.com>
17742 gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
17743 Original commit message from CVS:
17744 * gst/audioconvert/audioconvert.c:
17745 Include math.h to fix compilation.
17747 2007-06-29 14:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17749 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
17750 Original commit message from CVS:
17751 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
17752 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
17753 Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
17754 format, as produced by some dc1394 cameras like the iSight.
17755 See http://www.fourcc.org/yuv.php#IYU1
17757 2007-06-28 20:37:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17759 gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
17760 Original commit message from CVS:
17761 * gst/audioconvert/Makefile.am:
17762 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
17763 (check_default), (audio_convert_prepare_context),
17764 (audio_convert_clean_context), (audio_convert_convert):
17765 * gst/audioconvert/audioconvert.h:
17766 * gst/audioconvert/gstaudioconvert.c:
17767 (gst_audio_convert_dithering_get_type),
17768 (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
17769 (gst_audio_convert_init), (gst_audio_convert_set_caps),
17770 (gst_audio_convert_set_property), (gst_audio_convert_get_property):
17771 * gst/audioconvert/gstaudioconvert.h:
17772 * gst/audioconvert/gstaudioquantize.c:
17773 (gst_audio_quantize_setup_noise_shaping),
17774 (gst_audio_quantize_free_noise_shaping),
17775 (gst_audio_quantize_setup_dither),
17776 (gst_audio_quantize_free_dither),
17777 (gst_audio_quantize_setup_quantize_func),
17778 (gst_audio_quantize_setup), (gst_audio_quantize_free):
17779 * gst/audioconvert/gstaudioquantize.h:
17780 Implement dithering and noise shaping in audioconvert. By default now
17781 TPDF dithering (and no noise shaping) will be used when converting
17782 from a higher bit depth to 20 bit depth or smaller, otherwise
17783 everything will be as it is now.
17784 For the last audioconvert in a pipeline it would make sense to
17785 use some kind of noise shaping, enabling it by default for all
17786 conversions would give undesired results though. Fixes #360246.
17787 * tests/check/elements/audioconvert.c: (setup_audioconvert),
17789 Adjust unit test for the new audioconvert.
17791 2007-06-28 11:06:56 +0000 Wim Taymans <wim.taymans@gmail.com>
17793 gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
17794 Original commit message from CVS:
17795 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
17796 Use other metrics as well when estimating the buffer level.
17798 2007-06-28 10:21:19 +0000 Wim Taymans <wim.taymans@gmail.com>
17800 gst/playback/gstplaybasebin.c: Small debug improvement.
17801 Original commit message from CVS:
17802 * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
17803 Small debug improvement.
17804 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
17806 Tweak the rate estimation period.
17807 When calculating the buffer filledness in rate estimation mode, don't
17808 mix it with other metrics.
17810 2007-06-28 09:46:11 +0000 Wim Taymans <wim.taymans@gmail.com>
17812 gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
17813 Original commit message from CVS:
17814 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
17815 (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
17816 When creating the groups, allow for a 5 second, unlimited buffers
17817 preroll phase after which we expose the group.
17818 When the group is exposed, use a small number of buffers up to a 2
17819 second limit. Also disconnect the overrun signal from multiqueue when we
17820 exposed the group because it is not needed anymore.
17822 2007-06-27 22:30:19 +0000 Tim-Philipp Müller <tim@centricular.net>
17824 gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
17825 Original commit message from CVS:
17826 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
17827 Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
17828 to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
17829 (#451707); also, output some debugging info when dealing with
17831 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
17832 Add unit test for the above.
17834 2007-06-27 12:55:20 +0000 Tim-Philipp Müller <tim@centricular.net>
17836 gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
17837 Original commit message from CVS:
17838 * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
17839 Add description for Windows Media RTP caps.
17840 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
17841 Remove RTP fields that don't define the format from caps.
17843 2007-06-27 10:14:03 +0000 Tim-Philipp Müller <tim@centricular.net>
17845 ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
17846 Original commit message from CVS:
17847 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
17848 Skip empty buffers, but not empty header buffers. That way the original
17849 vorbisdec unit test still passes (#451145); also, take into account
17850 that those empty packets might carry a granulepos.
17851 * tests/check/Makefile.am:
17852 * tests/check/elements/vorbisdec.c:
17853 (_create_codebook_header_buffer), (_create_audio_buffer),
17854 (GST_START_TEST), (vorbisdec_suite):
17855 Add unit test that sends an empty packet.
17857 2007-06-27 09:49:51 +0000 Wim Taymans <wim.taymans@gmail.com>
17859 ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
17860 Original commit message from CVS:
17861 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
17862 Don't error out on 0-sized packets, just emit a warning because this is
17863 not a fatal error. Fixes #451145.
17865 2007-06-25 12:43:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17867 docs/plugins/: Update docs with caps info.
17868 Original commit message from CVS:
17869 * docs/plugins/gst-plugins-base-plugins.args:
17870 * docs/plugins/gst-plugins-base-plugins.signals:
17871 * docs/plugins/inspect/plugin-adder.xml:
17872 * docs/plugins/inspect/plugin-alsa.xml:
17873 * docs/plugins/inspect/plugin-audioconvert.xml:
17874 * docs/plugins/inspect/plugin-audiorate.xml:
17875 * docs/plugins/inspect/plugin-audioresample.xml:
17876 * docs/plugins/inspect/plugin-audiotestsrc.xml:
17877 * docs/plugins/inspect/plugin-cdparanoia.xml:
17878 * docs/plugins/inspect/plugin-decodebin.xml:
17879 * docs/plugins/inspect/plugin-decodebin2.xml:
17880 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
17881 * docs/plugins/inspect/plugin-gdp.xml:
17882 * docs/plugins/inspect/plugin-gnomevfs.xml:
17883 * docs/plugins/inspect/plugin-libvisual.xml:
17884 * docs/plugins/inspect/plugin-ogg.xml:
17885 * docs/plugins/inspect/plugin-pango.xml:
17886 * docs/plugins/inspect/plugin-playbin.xml:
17887 * docs/plugins/inspect/plugin-subparse.xml:
17888 * docs/plugins/inspect/plugin-tcp.xml:
17889 * docs/plugins/inspect/plugin-theora.xml:
17890 * docs/plugins/inspect/plugin-typefindfunctions.xml:
17891 * docs/plugins/inspect/plugin-video4linux.xml:
17892 * docs/plugins/inspect/plugin-videorate.xml:
17893 * docs/plugins/inspect/plugin-videoscale.xml:
17894 * docs/plugins/inspect/plugin-videotestsrc.xml:
17895 * docs/plugins/inspect/plugin-volume.xml:
17896 * docs/plugins/inspect/plugin-vorbis.xml:
17897 * docs/plugins/inspect/plugin-ximagesink.xml:
17898 * docs/plugins/inspect/plugin-xvimagesink.xml:
17899 Update docs with caps info.
17901 2007-06-25 12:04:15 +0000 Tim-Philipp Müller <tim@centricular.net>
17903 po/POTFILES.in: Add more files with translatable strings (#450875).
17904 Original commit message from CVS:
17906 Add more files with translatable strings (#450875).
17908 2007-06-23 14:44:07 +0000 Edward Hervey <bilboed@bilboed.com>
17910 ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
17911 Original commit message from CVS:
17912 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
17913 The chain should be freed if we error out here, else it will leak.
17914 * gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
17915 (cleanup_decodebin):
17916 Don't forget to *properly* remove the signals, else it will leak.
17918 2007-06-22 14:25:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17920 MAINTAINERS: Updating all the maintainers files
17921 Original commit message from CVS:
17923 Updating all the maintainers files
17925 2007-06-21 08:34:46 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17927 tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
17928 Original commit message from CVS:
17929 * tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
17931 Destroy and recreate parse-launch based pipeline after stop to be able
17932 to play again. Reorder some code and add more comments.
17934 2007-06-20 11:09:03 +0000 Wim Taymans <wim@fluendo.com>
17936 gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
17937 Original commit message from CVS:
17938 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
17939 When handling a delayed-caps notification case, mark
17940 the group as dynamic so that the nbdynamic count is
17941 incremented and decremented correctly. Fixes: #449156
17942 Patch by: Wim Taymans <wim@fluendo.com>
17944 2007-06-19 19:13:04 +0000 Andy Wingo <wingo@pobox.com>
17947 * gst-libs/gst/audio/gstbaseaudiosink.c:
17948 * win32/common/config.h:
17949 gst-libs/gst/audio/gstbaseaudiosink.c
17950 Original commit message from CVS:
17951 2007-06-19 Andy Wingo <wingo@pobox.com>
17952 * gst-libs/gst/audio/gstbaseaudiosink.c
17953 (gst_base_audio_sink_init): Enable pull-mode operation.
17955 2007-06-19 09:34:35 +0000 Michael Smith <msmith@xiph.org>
17957 gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
17958 Original commit message from CVS:
17959 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
17960 Change minimum rate back to 1000 to allow low-sample-rate wav files
17963 2007-06-17 17:27:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
17965 po/vi.po: Update translations.
17966 Original commit message from CVS:
17968 Update translations.
17970 2007-06-16 03:42:14 +0000 David Schleef <ds@schleef.org>
17972 gst/playback/gstqueue2.c: Fix compile error from ignored return value.
17973 Original commit message from CVS:
17974 * gst/playback/gstqueue2.c:
17975 Fix compile error from ignored return value.
17977 2007-06-15 15:23:36 +0000 Michael Smith <msmith@xiph.org>
17979 gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
17980 Original commit message from CVS:
17981 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
17982 Update tmpbuf for all neccesary rows, not just one, as is required
17986 2007-06-15 11:15:28 +0000 Michael Smith <msmith@xiph.org>
17988 tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
17989 Original commit message from CVS:
17990 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
17991 (eos_buffer_probe):
17992 Add a test that ensures we set DELTA_UNIT on all non-header,
17993 non-video buffers, if we have a video stream.
17994 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
17995 (gst_ogg_mux_process_best_pad):
17996 Move setting delta_pad to earlier, where we inspect all pads, so
17997 that leading audio pages don't get DELTA_UNIT unset if they come
17998 before the first DELTA_UNIT from video pages. Fixes the newly-added
17999 test. Fixes #385527.
18001 2007-06-14 19:53:27 +0000 Tim-Philipp Müller <tim@centricular.net>
18003 tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
18004 Original commit message from CVS:
18005 * tests/check/pipelines/streamheader.c: (streamheader_suite):
18006 Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
18007 fails on the p5-ppc64 build bot and the failure looks like it is due
18008 to the same issue as #348114, ie. a compiler bug.
18010 2007-06-13 18:20:57 +0000 Edward Hervey <bilboed@bilboed.com>
18012 gst/playback/gstqueue2.c: Fix build on MacOSX.
18013 Original commit message from CVS:
18014 * gst/playback/gstqueue2.c: (gst_queue_create_read):
18015 Fix build on MacOSX.
18017 2007-06-13 09:01:32 +0000 Wim Taymans <wim.taymans@gmail.com>
18019 ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
18020 Original commit message from CVS:
18021 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
18022 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
18023 Fix compilation on mingw. Fixes #446972.
18025 2007-06-12 08:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
18027 gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
18028 Original commit message from CVS:
18029 Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
18030 * gst/playback/gstqueue2.c: (update_buffering),
18031 (gst_queue_locked_enqueue):
18032 Fix a division by zero when the max percent is <= 0. Fixes #446572.
18033 also update the buffering status when receiving events. Fixes #446551.
18035 2007-06-11 11:32:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
18037 gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
18038 Original commit message from CVS:
18039 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
18040 * gst/playback/gstqueue2.c: (gst_queue_peer_query),
18041 (gst_queue_handle_src_query):
18042 Wait for preroll before attempting to forward a duration query upstream.
18045 2007-06-07 21:08:38 +0000 Sébastien Moutte <sebastien@moutte.net>
18047 gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
18048 Original commit message from CVS:
18049 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18050 (gst_base_rtp_depayload_set_gst_timestamp):
18051 Use G_GINT64_CONSTANT macro for int64 constant.
18052 * win32/common/libgstinterfaces.def:
18053 * win32/common/libgsttag.def:
18054 Add new exported functions.
18056 2007-06-07 14:25:32 +0000 Tim-Philipp Müller <tim@centricular.net>
18058 ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
18059 Original commit message from CVS:
18060 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
18061 The BOS page of the first Dirac video stream needs to come before
18062 the BOS page of any Vorbis streams or other audio streams, just like
18065 2007-06-07 09:11:27 +0000 Wim Taymans <wim.taymans@gmail.com>
18067 gst/playback/gstqueue2.c: Fix compilation.
18068 Original commit message from CVS:
18069 * gst/playback/gstqueue2.c: (gst_queue_get_range):
18072 2007-06-06 13:36:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
18074 gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
18075 Original commit message from CVS:
18076 Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
18077 * gst/playback/gstqueue2.c: (gst_queue_init),
18078 (gst_queue_handle_sink_event), (gst_queue_chain),
18079 (gst_queue_get_range), (gst_queue_src_checkgetrange_function),
18080 (gst_queue_sink_activate_push), (gst_queue_src_activate_push),
18081 (gst_queue_src_activate_pull):
18082 Add pull based scheduling and fix some deadlocks. Fixes #444523.
18083 Does not yet completely work because duration queries upstream won't
18086 2007-06-06 09:08:50 +0000 Wim Taymans <wim.taymans@gmail.com>
18088 Some more fseeko checks.
18089 Original commit message from CVS:
18091 * gst/playback/gstqueue2.c: (gst_queue_create_read):
18092 Some more fseeko checks.
18094 2007-06-06 08:01:42 +0000 Wim Taymans <wim.taymans@gmail.com>
18096 configure.ac: check for large file support.
18097 Original commit message from CVS:
18099 check for large file support.
18101 2007-06-05 21:36:11 +0000 Sven Arvidsson <sa@whiz.se>
18103 gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
18104 Original commit message from CVS:
18105 Based on a patch by Sven Arvidsson <sa at whiz dot se>:
18106 * gst/subparse/gstsubparse.c: (parse_subrip),
18107 (subviewer_unescape_newlines), (parse_subviewer),
18108 (gst_sub_parse_data_format_autodetect),
18109 (gst_sub_parse_format_autodetect), (gst_subparse_type_find):
18110 * gst/subparse/gstsubparse.h:
18111 Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
18112 * tests/check/elements/subparse.c: (GST_START_TEST),
18114 Add a unit test for both SubViewer formats.
18116 2007-06-05 17:08:04 +0000 Michael Smith <msmith@xiph.org>
18118 gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
18119 Original commit message from CVS:
18120 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
18121 Don't overflow intermediate values when seeking to large time values
18124 2007-06-05 17:02:13 +0000 Wim Taymans <wim.taymans@gmail.com>
18126 gst/playback/gstqueue2.c: Include stdio to define fseeko.
18127 Original commit message from CVS:
18128 * gst/playback/gstqueue2.c: (gst_queue_have_data),
18129 (gst_queue_create_read), (gst_queue_read_item_from_file),
18130 (gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
18131 Include stdio to define fseeko.
18133 2007-06-05 16:37:09 +0000 Edward Hervey <edward@fluendo.com>
18135 sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
18136 Original commit message from CVS:
18137 Patch by: Edward Hervey <edward@fluendo.com>
18138 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
18139 (gst_v4lsrc_query):
18140 Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
18142 2007-06-05 16:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
18144 gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
18145 Original commit message from CVS:
18146 * gst-libs/gst/riff/Makefile.am:
18147 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
18148 Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
18149 our own implementation.
18151 2007-06-05 16:19:30 +0000 Wim Taymans <wim.taymans@gmail.com>
18153 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
18154 Original commit message from CVS:
18155 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18156 (gst_base_rtp_depayload_setcaps),
18157 (gst_base_rtp_depayload_set_gst_timestamp),
18158 (gst_base_rtp_depayload_change_state):
18159 Handle timestamp wraparound.
18161 2007-06-05 16:17:30 +0000 Wim Taymans <wim.taymans@gmail.com>
18163 gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
18164 Original commit message from CVS:
18165 * gst/playback/gsturidecodebin.c: (no_more_pads_full),
18166 (new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
18167 (gst_uri_decode_bin_change_state):
18168 Make sure we name srcpads uniquely even when using different internal
18170 Signal no-more-pads when no more dynamic elements exist.
18171 Remove pads on cleanup.
18173 2007-06-05 16:14:23 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
18175 gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
18176 Original commit message from CVS:
18177 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
18178 * gst/playback/gstqueue2.c: (gst_queue_class_init),
18179 (gst_queue_init), (gst_queue_finalize),
18180 (gst_queue_write_buffer_to_file), (gst_queue_have_data),
18181 (gst_queue_create_read), (gst_queue_read_item_from_file),
18182 (gst_queue_open_temp_location_file),
18183 (gst_queue_close_temp_location_file), (gst_queue_locked_flush),
18184 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
18185 (gst_queue_is_empty), (gst_queue_is_filled),
18186 (gst_queue_change_state), (gst_queue_set_temp_location),
18187 (gst_queue_set_property):
18188 Add support for filebased buffering. Fixes #441264.
18190 2007-06-05 16:05:19 +0000 Wim Taymans <wim.taymans@gmail.com>
18192 gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
18193 Original commit message from CVS:
18194 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
18195 (analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
18196 (caps_notify_group_cb), (gst_decode_group_new),
18197 (gst_decode_group_free):
18198 Add support for delayed caps fixation when autoplugging.
18199 Optimize cases where a multiqueue is not needed/wanted, like right after
18200 anything that is not a demuxer.
18202 2007-06-05 16:02:57 +0000 Wim Taymans <wim.taymans@gmail.com>
18204 ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
18205 Original commit message from CVS:
18206 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
18207 (gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
18208 (gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
18209 consideratly speedup ogg chain detection by not trying to find a base
18210 timestamp for skeleton streams.
18212 2007-06-05 16:00:33 +0000 Wim Taymans <wim.taymans@gmail.com>
18214 gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
18215 Original commit message from CVS:
18216 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
18217 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
18218 (gst_multi_fd_sink_remove_flush),
18219 (gst_multi_fd_sink_remove_client_link),
18220 (gst_multi_fd_sink_handle_client_write),
18221 (gst_multi_fd_sink_handle_clients):
18222 * gst/tcp/gstmultifdsink.h:
18223 Add support for remuve_flush.
18225 2007-06-05 15:59:00 +0000 Wim Taymans <wim.taymans@gmail.com>
18227 Add draft design for forcing keyframes in encoders and implement in theoraenc.
18228 Original commit message from CVS:
18229 * docs/design/draft-keyframe-force.txt:
18230 * ext/theora/theoraenc.c: (theora_enc_sink_event),
18231 (theora_enc_chain):
18232 Add draft design for forcing keyframes in encoders and implement in
18235 2007-06-05 13:22:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18237 configure.ac: Back to CVS
18238 Original commit message from CVS:
18242 === release 0.10.13 ===
18244 2007-06-05 12:50:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18250 * docs/plugins/gst-plugins-base-plugins.args:
18251 * docs/plugins/inspect/plugin-adder.xml:
18252 * docs/plugins/inspect/plugin-alsa.xml:
18253 * docs/plugins/inspect/plugin-audioconvert.xml:
18254 * docs/plugins/inspect/plugin-audiorate.xml:
18255 * docs/plugins/inspect/plugin-audioresample.xml:
18256 * docs/plugins/inspect/plugin-audiotestsrc.xml:
18257 * docs/plugins/inspect/plugin-cdparanoia.xml:
18258 * docs/plugins/inspect/plugin-decodebin.xml:
18259 * docs/plugins/inspect/plugin-decodebin2.xml:
18260 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
18261 * docs/plugins/inspect/plugin-gdp.xml:
18262 * docs/plugins/inspect/plugin-gnomevfs.xml:
18263 * docs/plugins/inspect/plugin-libvisual.xml:
18264 * docs/plugins/inspect/plugin-ogg.xml:
18265 * docs/plugins/inspect/plugin-pango.xml:
18266 * docs/plugins/inspect/plugin-playbin.xml:
18267 * docs/plugins/inspect/plugin-subparse.xml:
18268 * docs/plugins/inspect/plugin-tcp.xml:
18269 * docs/plugins/inspect/plugin-theora.xml:
18270 * docs/plugins/inspect/plugin-typefindfunctions.xml:
18271 * docs/plugins/inspect/plugin-video4linux.xml:
18272 * docs/plugins/inspect/plugin-videorate.xml:
18273 * docs/plugins/inspect/plugin-videoscale.xml:
18274 * docs/plugins/inspect/plugin-videotestsrc.xml:
18275 * docs/plugins/inspect/plugin-volume.xml:
18276 * docs/plugins/inspect/plugin-vorbis.xml:
18277 * docs/plugins/inspect/plugin-ximagesink.xml:
18278 * docs/plugins/inspect/plugin-xvimagesink.xml:
18279 * gst-plugins-base.doap:
18280 * win32/common/config.h:
18281 * win32/vs6/grammar.dsp:
18282 * win32/vs6/gst_plugins_base.dsw:
18283 * win32/vs6/libgstadder.dsp:
18284 * win32/vs6/libgstaudio.dsp:
18285 * win32/vs6/libgstaudioconvert.dsp:
18286 * win32/vs6/libgstaudiorate.dsp:
18287 * win32/vs6/libgstaudioresample.dsp:
18288 * win32/vs6/libgstaudioscale.dsp:
18289 * win32/vs6/libgstaudiotestsrc.dsp:
18290 * win32/vs6/libgstcdda.dsp:
18291 * win32/vs6/libgstdecodebin.dsp:
18292 * win32/vs6/libgstdecodebin2.dsp:
18293 * win32/vs6/libgstdirectsound.dsp:
18294 * win32/vs6/libgstffmpegcolorspace.dsp:
18295 * win32/vs6/libgstgdp.dsp:
18296 * win32/vs6/libgstinterfaces.dsp:
18297 * win32/vs6/libgstnetbuffer.dsp:
18298 * win32/vs6/libgstogg.dsp:
18299 * win32/vs6/libgstpbutils.dsp:
18300 * win32/vs6/libgstplaybin.dsp:
18301 * win32/vs6/libgstriff.dsp:
18302 * win32/vs6/libgstrtp.dsp:
18303 * win32/vs6/libgstsinesrc.dsp:
18304 * win32/vs6/libgstsubparse.dsp:
18305 * win32/vs6/libgsttag.dsp:
18306 * win32/vs6/libgsttheora.dsp:
18307 * win32/vs6/libgsttypefindfunctions.dsp:
18308 * win32/vs6/libgstutils.dsp:
18309 * win32/vs6/libgstvideo.dsp:
18310 * win32/vs6/libgstvideorate.dsp:
18311 * win32/vs6/libgstvideoscale.dsp:
18312 * win32/vs6/libgstvideotestsrc.dsp:
18313 * win32/vs6/libgstvolume.dsp:
18314 * win32/vs6/libgstvorbis.dsp:
18315 Release 0.10.13 "What's going on?"
18316 Original commit message from CVS:
18317 Release 0.10.13 "What's going on?"
18319 2007-06-05 12:32:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18337 Original commit message from CVS:
18340 2007-05-31 17:08:58 +0000 Wim Taymans <wim@fluendo.com>
18342 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...
18343 Original commit message from CVS:
18344 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
18345 In riff, the depth is stored in the size field but it just means that
18346 the least significant bits are cleared. We can therefore just play
18347 the sample as if it had a depth == width. Fixes: #440997
18348 Patch by: Wim Taymans <wim@fluendo.com>
18349 Patch by: Sebastian Dröge <slomo@circular-chaos.org>
18351 2007-05-31 16:36:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18353 gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
18354 Original commit message from CVS:
18355 * gst-libs/gst/floatcast/floatcast.h:
18356 Define inline when needed on win32 builds. Fixes: #441295
18358 2007-05-29 13:38:35 +0000 Wim Taymans <wim.taymans@gmail.com>
18360 gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
18361 Original commit message from CVS:
18362 * gst/playback/gstplaybasebin.c: (queue_overrun),
18363 (no_more_pads_full):
18364 Stop buffering when the group is commited because the queues filled up.
18367 2007-05-25 10:07:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18369 Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
18370 Original commit message from CVS:
18371 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
18372 (gst_alsa_mixer_free), (gst_alsa_mixer_update),
18373 (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
18374 (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
18375 (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
18376 * ext/alsa/gstalsamixer.h:
18377 * ext/alsa/gstalsamixerelement.c:
18378 (gst_alsa_mixer_element_interface_supported),
18379 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
18380 (gst_alsa_mixer_element_set_property),
18381 (gst_alsa_mixer_element_get_property),
18382 (gst_alsa_mixer_element_change_state):
18383 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
18384 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
18385 (gst_mixer_option_changed):
18386 * gst-libs/gst/interfaces/mixer.h:
18387 Revert commits towards #152864 made so far. We'll pick it up again
18388 after the 0.10.13 release.
18390 2007-05-24 16:22:23 +0000 Wim Taymans <wim.taymans@gmail.com>
18392 gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
18393 Original commit message from CVS:
18394 * gst-libs/gst/audio/gstbaseaudiosink.c:
18395 (gst_base_audio_sink_render):
18396 After an interrupt (PAUSED/flush) assume that the next sample should not
18397 be aligned to the previous sample. Fixes #417992.
18399 2007-05-24 15:16:59 +0000 Tim-Philipp Müller <tim@centricular.net>
18401 gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
18402 Original commit message from CVS:
18403 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
18404 Don't add channels and rate fields to the template caps for
18405 audio/x-dts, as wavparse might not always be able to set them,
18406 which would then lead to 'caps are not a real subset of the
18407 template caps' warnings.
18409 2007-05-24 11:15:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18411 gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
18412 Original commit message from CVS:
18413 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
18414 Handle unknown or invalid pads without crashing, as might occur if
18415 a media file like an mp3 is specified as a subtitle file.
18418 2007-05-24 10:19:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18420 gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
18421 Original commit message from CVS:
18422 * gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
18424 Block the subtitle bin output queue before ghosting it and linking,
18425 then unblock after. This avoids spurious not-linked errors caused
18426 by the queue starting up (because it gets linked when it is ghosted).
18429 2007-05-23 15:54:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18431 tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
18432 Original commit message from CVS:
18433 * tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
18434 Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
18435 file. Avoids flukes where the input gets typefound to some valid but
18438 2007-05-22 15:45:19 +0000 Tim-Philipp Müller <tim@centricular.net>
18440 tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
18441 Original commit message from CVS:
18442 * tests/check/Makefile.am:
18443 * tests/check/elements/.cvsignore:
18444 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
18445 (cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
18446 Add unit test for gnomevfssink seeking and position reporting for
18449 2007-05-22 15:30:26 +0000 Mark Nauwelaerts <manauw@skynet.be>
18451 ext/gnomevfs/gstgnomevfssink.*: see #412648.
18452 Original commit message from CVS:
18453 Patch by: Mark Nauwelaerts <manauw at skynet be>
18454 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
18455 (gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
18456 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
18457 * ext/gnomevfs/gstgnomevfssink.h:
18458 Fix position reporting, especially after a seek (from upstream),
18461 2007-05-22 15:04:41 +0000 Tim-Philipp Müller <tim@centricular.net>
18463 ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
18464 Original commit message from CVS:
18465 * ext/cdparanoia/gstcdparanoiasrc.c:
18468 2007-05-22 11:40:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18470 gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
18471 Original commit message from CVS:
18472 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
18473 Specify the full valid range for MP3 samplerates. Fixes a regression
18474 caused by extra header checks since the last release.
18476 2007-05-21 15:32:42 +0000 Mike Smith <msmith@xiph.org>
18478 sys/: Fix a locking-order bug I introduced with my changes the other day.
18479 Original commit message from CVS:
18480 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
18481 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
18482 Fix a locking-order bug I introduced with my changes the other day.
18483 Patch by Mike Smith.
18485 2007-05-21 15:24:21 +0000 Michael Smith <msmith@xiph.org>
18487 ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
18488 Original commit message from CVS:
18489 * ext/theora/theoradec.c: (theora_handle_data_packet):
18490 Don't look inside 0-length packets (which indicate duplicated
18493 2007-05-21 10:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
18496 Original commit message from CVS:
18497 * ext/cdparanoia/gstcdparanoiasrc.c:
18498 (gst_cd_paranoia_src_read_sector):
18499 * gst-libs/gst/audio/gstbaseaudiosrc.c:
18500 (gst_base_audio_src_create):
18502 * ext/theora/theoradec.c: (theora_dec_sink_event):
18504 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18505 (gst_base_rtp_depayload_set_gst_timestamp):
18507 * gst/playback/gstdecodebin.c: (queue_underrun_cb):
18508 And some debug info when a FIXME path is hit.
18510 2007-05-21 09:45:28 +0000 Wim Taymans <wim.taymans@gmail.com>
18512 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
18513 Original commit message from CVS:
18514 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18515 (gst_base_rtp_audio_payload_class_init),
18516 (gst_base_rtp_audio_payload_init),
18517 (gst_base_rtp_audio_payload_finalize),
18518 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
18519 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
18520 (gst_base_rtp_payload_audio_handle_event):
18521 Some cleanups, remove minptime property as it is now in the parent
18523 Override parent class event function.
18524 * gst-libs/gst/rtp/gstbasertppayload.c:
18525 (gst_basertppayload_class_init), (gst_basertppayload_init),
18526 (gst_basertppayload_event), (gst_basertppayload_set_property),
18527 (gst_basertppayload_get_property):
18528 * gst-libs/gst/rtp/gstbasertppayload.h:
18529 Add min-ptime property.
18530 Add handle-event vmethod. Fixes #415001.
18532 2007-05-18 17:10:03 +0000 Christian Schaller <uraeus@gnome.org>
18534 * gst-plugins-base.spec.in:
18536 Original commit message from CVS:
18539 2007-05-18 15:23:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18541 gst-libs/gst/audio/gstbaseaudiosink.c
18542 Original commit message from CVS:
18543 * gst-libs/gst/audio/gstbaseaudiosink.c
18544 (gst_base_audio_sink_change_state):
18545 Fix typo in comment.
18546 * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
18547 free_dynamics, pad_probe, close_pad_link, try_to_link_1,
18548 get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
18550 * gst/playback/gstplaybin.c (gst_play_bin_set_property,
18551 gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
18552 Remove trailing whitespaces in comments.
18553 * gst/volume/Makefile.am:
18556 2007-05-18 15:10:08 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
18559 * gst-libs/gst/interfaces/mixer.h:
18560 gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
18561 Original commit message from CVS:
18562 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
18563 * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
18564 set_option, get_option, _gst_reserved):
18565 Revert reordering functions (keep ABI).
18567 2007-05-17 17:35:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18569 sys/: When we create our own window, indicate that we handle the
18570 Original commit message from CVS:
18571 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
18572 (gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
18573 (gst_ximagesink_show_frame):
18574 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
18575 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
18576 (gst_xvimagesink_show_frame):
18577 When we create our own window, indicate that we handle the
18578 WM_DELETE client message from the window manager, so that it won't
18579 kill our window (and our app) along with it. Handle ClientMessage,
18580 post an error on the bus, and close the window. Further buffers
18581 arriving will result in a FlowError because the window has been
18584 Clean up the X event handling loop and make them the same for
18585 both xvimagesink and ximagesink while I'm at it.
18587 2007-05-17 16:27:32 +0000 Wim Taymans <wim.taymans@gmail.com>
18589 gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
18590 Original commit message from CVS:
18591 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
18592 Make decodebin2 autoplug depayloaders too.
18593 * gst/playback/gsturidecodebin.c: (source_new_pad):
18594 Set the newly created decoder in a usable state when autoplugging a
18595 dynamic source such as RTSP.
18597 2007-05-17 16:11:03 +0000 Tim-Philipp Müller <tim@centricular.net>
18599 gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
18600 Original commit message from CVS:
18601 * gst/playback/gststreaminfo.c: (cb_probe):
18602 Ignore video-codec tag for audio streams and ignore audio-codec tags
18603 for video streams. Should make codec name collection a bit more
18604 robust against sloppy demuxers that send tag events containing both
18605 tags down each pad.
18607 2007-05-17 15:22:44 +0000 Wim Taymans <wim.taymans@gmail.com>
18609 gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
18610 Original commit message from CVS:
18611 * gst/playback/gstqueue2.c: (update_rates):
18612 Tweak the buffering thresholds a little.
18613 Update the buffer size with the previously calculate rate instead of
18614 only when we calculate a new rate so that we get smoother buffering
18616 * gst/playback/Makefile.am:
18617 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
18618 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
18619 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
18620 (gst_uri_decode_bin_get_property), (unknown_type),
18621 (add_element_stream), (no_more_pads_full), (no_more_pads),
18622 (source_no_more_pads), (new_decoded_pad), (array_has_value),
18623 (gen_source_element), (has_all_raw_caps), (analyse_source),
18624 (remove_decoders), (make_decoder), (remove_source),
18625 (source_new_pad), (setup_source), (decoder_query_init),
18626 (decoder_query_duration_fold), (decoder_query_duration_done),
18627 (decoder_query_position_fold), (decoder_query_position_done),
18628 (decoder_query_latency_fold), (decoder_query_latency_done),
18629 (decoder_query_seeking_fold), (decoder_query_seeking_done),
18630 (decoder_query_generic_fold), (gst_uri_decode_bin_query),
18631 (gst_uri_decode_bin_change_state), (plugin_init):
18632 New element that intergrates a source, optional buffering element and
18635 2007-05-17 14:17:17 +0000 Tim-Philipp Müller <tim@centricular.net>
18637 configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
18638 Original commit message from CVS:
18640 Bump libtheora requirement to 1.0alpha5 for the pixformat check
18641 (also has a .pc file, so we don't need the fallback check any
18642 longer). Fixes #438840.
18644 2007-05-17 13:36:11 +0000 Wim Taymans <wim.taymans@gmail.com>
18646 gst/playback/gstqueue2.c: fix build.
18647 Original commit message from CVS:
18648 * gst/playback/gstqueue2.c: (gst_queue_get_type),
18649 (gst_queue_class_init), (gst_queue_finalize), (update_time_level),
18650 (apply_segment), (apply_buffer), (update_buffering),
18651 (reset_rate_timer), (update_rates), (gst_queue_locked_flush),
18652 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
18653 (gst_queue_handle_sink_event), (gst_queue_is_filled),
18654 (gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
18658 2007-05-17 11:57:44 +0000 Wim Taymans <wim.taymans@gmail.com>
18660 gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
18661 Original commit message from CVS:
18662 * gst/playback/Makefile.am:
18663 * gst/playback/gstqueue2.c: (gst_queue_get_type),
18664 (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
18665 (gst_queue_getcaps), (gst_queue_bufferalloc),
18666 (gst_queue_acceptcaps), (update_time_level), (apply_segment),
18667 (apply_buffer), (update_buffering), (reset_rate_timer),
18668 (update_rates), (gst_queue_locked_flush),
18669 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
18670 (gst_queue_handle_sink_event), (gst_queue_is_empty),
18671 (gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
18672 (gst_queue_loop), (gst_queue_handle_src_event),
18673 (gst_queue_handle_src_query), (gst_queue_sink_activate_push),
18674 (gst_queue_src_activate_push), (gst_queue_change_state),
18675 (gst_queue_set_property), (gst_queue_get_property), (plugin_init):
18676 On our way to playbin2 this is the new network queue that does buffering
18677 all by itself using high and low watermarks. It can also measure up and
18678 downstream bandwidth to optimally size the queue.
18680 2007-05-17 11:16:14 +0000 Michael Smith <msmith@xiph.org>
18682 gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
18683 Original commit message from CVS:
18684 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
18685 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
18686 Use the segment->last_stop value to calculate the next timestamp to
18687 generate after a seek; not the segment->start value.
18689 2007-05-15 20:14:06 +0000 David Schleef <ds@schleef.org>
18691 docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
18692 Original commit message from CVS:
18693 * docs/Makefile.am: Install docs even when --disable-gtk-doc
18694 is disabled. This matches the behavior of gtk+. Fixes #349099.
18696 2007-05-15 17:11:09 +0000 Wim Taymans <wim.taymans@gmail.com>
18698 ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
18699 Original commit message from CVS:
18700 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
18701 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
18702 Some more chained streaming ogg timestamp fixes.
18704 2007-05-15 16:46:10 +0000 Wim Taymans <wim.taymans@gmail.com>
18706 ext/ogg/gstoggdemux.c: Add some FIXMEs.
18707 Original commit message from CVS:
18708 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
18709 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
18710 (gst_ogg_demux_handle_page):
18712 Fix chain start/stop segment handling based on patch by
18713 <ahalda at cs dot mcgill dot ca> see #320984.
18715 2007-05-15 15:33:54 +0000 Michael Smith <msmith@xiph.org>
18717 configure.ac: We don't require a C++ compiler. So don't require one.
18718 Original commit message from CVS:
18720 We don't require a C++ compiler. So don't require one.
18722 2007-05-15 15:29:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18725 * ext/alsa/gstalsamixer.c:
18726 ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
18727 Original commit message from CVS:
18728 * ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
18729 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
18730 gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
18731 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
18732 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
18733 gst_alsa_mixer_update_track):
18734 Apply some of the cleanup Tim suggested in #152864 afterwards.
18736 2007-05-15 14:01:26 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
18738 ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
18739 Original commit message from CVS:
18740 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
18741 * ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
18742 _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
18743 gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
18744 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
18745 gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
18746 gst_alsa_mixer_handle_source_callback,
18747 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
18748 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
18749 gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
18750 gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
18751 gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
18752 gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
18753 * ext/alsa/gstalsamixer.h (handle_source, interface, dir):
18754 * ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
18755 gst_alsa_mixer_element_interface_supported,
18756 gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
18757 gst_alsa_mixer_element_set_property,
18758 gst_alsa_mixer_element_get_property,
18759 gst_alsa_mixer_element_change_state):
18760 * ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
18761 * gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
18762 gst_mixer_option_changed):
18763 * gst-libs/gst/interfaces/mixer.h (set_option, get_option,
18764 volume_changed, option_changed, _gst_reserved):
18765 Implement notification for alsamixer. Fixes #152864
18767 2007-05-15 03:53:11 +0000 David Schleef <ds@schleef.org>
18769 gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
18770 Original commit message from CVS:
18771 * gst/videotestsrc/videotestsrc.c:
18772 * gst/videotestsrc/videotestsrc.h:
18773 Add support for video/x-raw-bayer.
18775 2007-05-13 01:06:19 +0000 David Schleef <ds@schleef.org>
18777 sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
18778 Original commit message from CVS:
18779 * sys/xvimage/xvimagesink.c:
18780 Add some sanity checking for the XVImage size returned by X.
18781 Related to #377400.
18783 2007-05-12 16:18:39 +0000 Wim Taymans <wim.taymans@gmail.com>
18785 gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
18786 Original commit message from CVS:
18787 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18788 (gst_base_rtp_depayload_setcaps),
18789 (gst_base_rtp_depayload_set_gst_timestamp):
18790 Parse and use additional caps fields as described in updated
18791 application/x-rtp caps spec.
18793 2007-05-12 16:16:22 +0000 Wim Taymans <wim.taymans@gmail.com>
18795 ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
18796 Original commit message from CVS:
18797 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
18798 (gst_ogg_demux_collect_chain_info):
18799 If there is a stream in a chain without any data packets, ignore the
18800 stream in the total length calculations. Might be related to #436820.
18802 2007-05-11 17:33:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18804 gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
18805 Original commit message from CVS:
18806 * gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
18807 (mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
18808 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
18809 (mpeg_video_type_find), (mpeg_video_stream_type_find),
18811 Consolidate and re-work our mpeg system stream detection to probe
18812 more packets and produce a higher confidence result. Fixes a
18813 regression caused by lowering the typefind probability last year
18814 - related to bug #397810. Remove the redundant MPEG-1 specific
18815 typefind function, as the new one detects both MPEG-1 & MPEG-2
18817 Also cleanup the MPEG elementary and MPEG-TS detection functions a
18819 Tested against my media test directory, with some improvements and
18822 2007-05-10 15:28:13 +0000 Wim Taymans <wim.taymans@gmail.com>
18824 gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
18825 Original commit message from CVS:
18826 * gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
18827 (queue_out_of_data):
18828 Connect to the new queue "pushing" signal instead of the broken
18831 2007-05-09 21:17:40 +0000 Sébastien Moutte <sebastien@moutte.net>
18833 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
18834 Original commit message from CVS:
18835 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18836 (gst_base_rtp_audio_payload_handle_frame_based_buffer):
18837 Move variable declaration before the first instruction.
18838 * gst/videotestsrc/videotestsrc.c:
18839 Define M_PI if it's not defined yet.
18840 * win32/common/libgstrtp.def:
18841 Add new exported functions.
18843 2007-05-09 11:54:32 +0000 Michael Smith <msmith@xiph.org>
18845 ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
18846 Original commit message from CVS:
18847 * ext/theora/theoradec.c: (theora_handle_type_packet):
18848 gst_pad_push_event() does not return a GstFlowReturn!
18850 2007-05-09 11:25:34 +0000 Wim Taymans <wim.taymans@gmail.com>
18852 tests/examples/seek/: Some small cosmetic changes.
18853 Original commit message from CVS:
18854 * tests/examples/seek/scrubby.c: (stop_cb), (main):
18855 * tests/examples/seek/seek.c: (do_seek):
18856 Some small cosmetic changes.
18858 2007-05-08 19:24:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18861 * gst/adder/gstadder.c:
18862 * gst/adder/gstadder.h:
18863 gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
18864 Original commit message from CVS:
18865 * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
18866 gst_adder_change_state):
18867 * gst/adder/gstadder.h (bps, offset, collect_event, segment,
18868 segment_pending, segment_position, segment_rate):
18869 Handle playback-rate on adder.
18871 2007-05-07 11:43:31 +0000 Michael Smith <msmith@xiph.org>
18873 ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
18874 Original commit message from CVS:
18875 * ext/theora/gsttheoradec.h:
18876 * ext/theora/theoradec.c: (gst_theora_dec_reset),
18877 (theora_dec_sink_event), (theora_handle_comment_packet),
18878 (theora_handle_type_packet), (theora_dec_change_state):
18879 Don't push events (newsegment, tags) before initialising the
18881 This is neccesary for seeking to work correctly in gnonlin.
18883 2007-05-04 13:10:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18885 gst/: gst/audiotestsrc/gstaudiotestsrc.c
18886 Original commit message from CVS:
18887 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18888 * gst/adder/gstadder.c:
18889 * gst/audiotestsrc/gstaudiotestsrc.c
18890 (gst_audio_test_src_create_white_noise):
18891 * gst/videotestsrc/gstvideotestsrc.c:
18892 * gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
18893 VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
18894 volume_sink_template, volume_src_template, gst_volume_init,
18895 volume_process_double, volume_process_int16,
18896 volume_process_int16_clamp):
18897 Doc fixes and formatting.
18899 2007-05-04 12:41:21 +0000 Tim-Philipp Müller <tim@centricular.net>
18901 tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
18902 Original commit message from CVS:
18903 * tests/check/Makefile.am:
18904 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
18905 Minimal check for volume's GstController usability; also another
18908 2007-05-04 09:06:38 +0000 Tim-Philipp Müller <tim@centricular.net>
18910 gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
18911 Original commit message from CVS:
18912 * gst-libs/gst/cdda/gstcddabasesrc.c:
18913 (gst_cdda_base_src_add_track):
18914 Fix it so that it (a) makes sense and (b) doesn't break
18915 everything cdda-related including the unit test.
18917 2007-05-04 08:46:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18919 gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
18920 Original commit message from CVS:
18921 * gst-libs/gst/cdda/gstcddabasesrc.c:
18922 (gst_cdda_base_src_add_track):
18923 Fix build when disabling asserts.
18925 2007-05-03 16:29:10 +0000 Tim-Philipp Müller <tim@centricular.net>
18927 sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
18928 Original commit message from CVS:
18929 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
18930 When XShm is not available, we might get row strides that are not
18931 rounded up to multiples of four; this is bad, because virtually
18932 every RGB-processing element in GStreamer assumes rowstrides are
18933 rounded up to multiples of four, so let's allocate at least enough
18934 memory to avoid crashes in this case. The image will still be
18935 displayed distorted though if this happens, so that still needs
18936 fixing (maybe by allocating a bigger image with an 'even' width
18937 and then clipping it appropriately when rendering - something for
18938 Xlib aficionados in any case).
18940 2007-05-03 13:16:21 +0000 Michael Smith <msmith@xiph.org>
18942 gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
18943 Original commit message from CVS:
18944 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
18945 If a buffer doesn't have a timestamp, assume it's contiguous with
18946 the previous buffer, and synthesise timestamps appropriately.
18948 2007-05-03 11:24:00 +0000 Edward Hervey <bilboed@bilboed.com>
18950 tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
18951 Original commit message from CVS:
18952 * tests/check/elements/videorate.c: (GST_START_TEST):
18953 Set buffer timestamp to a valid value in order to test the buffer
18954 really does stay in videorate.
18956 2007-05-03 10:47:22 +0000 Edward Hervey <bilboed@bilboed.com>
18958 gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
18959 Original commit message from CVS:
18960 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
18961 There is no sensible way to handle incoming buffers which don't have a
18962 valid timestamp. We therefore discard them and wait for the next one.
18964 2007-05-01 18:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
18966 gst/playback/: Better error message for text files.
18967 Original commit message from CVS:
18968 * gst/playback/gstdecodebin.c: (type_found), (plugin_init):
18969 * gst/playback/gstdecodebin2.c: (plugin_init):
18970 Better error message for text files.
18972 2007-04-29 14:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
18974 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
18975 Original commit message from CVS:
18976 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
18977 Fix offset bug in generation RR packets.
18979 2007-04-27 15:33:46 +0000 Julien Moutte <julien@moutte.net>
18981 ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
18982 Original commit message from CVS:
18983 2007-04-27 Julien MOUTTE <julien@moutte.net>
18984 * ext/theora/theoradec.c: (_theora_granule_time),
18985 (theora_dec_push_forward), (theora_handle_data_packet),
18986 (theora_dec_decode_buffer): Calculate buffer duration correctly
18987 to generate a perfect stream (#433888).
18988 * gst/audioresample/gstaudioresample.c:
18989 (audioresample_check_discont): Glib provides ABS.
18991 2007-04-27 15:01:40 +0000 Wim Taymans <wim.taymans@gmail.com>
18993 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
18994 Original commit message from CVS:
18995 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
18996 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
18997 (gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
18998 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
18999 (gst_rtcp_packet_bye_set_reason):
19000 * gst-libs/gst/rtp/gstrtcpbuffer.h:
19001 Fix RB block parsing and writing.
19002 Add support for constructing BYE packets.
19004 2007-04-25 08:54:34 +0000 Tim-Philipp Müller <tim@centricular.net>
19006 When posting a warning message because samples were dropped, post something more intelligible than he default error m...
19007 Original commit message from CVS:
19008 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
19009 (gst_base_audio_src_create):
19011 When posting a warning message because samples were dropped, post
19012 something more intelligible than he default error message for clock
19013 errors which is just confusing in this context (#432984).
19015 2007-04-25 08:10:26 +0000 Wim Taymans <wim.taymans@gmail.com>
19017 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
19018 Original commit message from CVS:
19019 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
19020 (gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
19021 (read_packet_header), (gst_rtcp_packet_move_to_next),
19022 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
19023 (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
19024 (gst_rtcp_packet_sdes_get_item_count),
19025 (gst_rtcp_packet_sdes_first_item),
19026 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
19027 (gst_rtcp_packet_sdes_first_entry),
19028 (gst_rtcp_packet_sdes_next_entry),
19029 (gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
19030 (gst_rtcp_packet_sdes_add_entry):
19031 * gst-libs/gst/rtp/gstrtcpbuffer.h:
19032 Implement code to write SR, RR and SDES packets.
19034 2007-04-24 20:45:24 +0000 Christian Kirbach <Christian.Kirbach@googlemail.com>
19036 sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
19037 Original commit message from CVS:
19038 Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
19039 * sys/ximage/ximagesink.c:
19040 Fix build if XShm is not available (#432362).
19042 2007-04-24 18:58:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19044 gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
19045 Original commit message from CVS:
19046 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
19047 Initalize the AudioConvertCtx with zeroes, otherwise it will contain
19048 pointers to random memory which are passed to g_free() when
19049 audio_convert_prepare_context() is called the first time.
19051 2007-04-24 15:00:07 +0000 Dan Williams <dcbw@redhat.com>
19053 gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
19054 Original commit message from CVS:
19055 Patch by: Dan Williams <dcbw redhat com>
19056 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
19057 Don't leak incoming buffer if gst_pad_push() returns a
19058 non-OK flow. Fixes #432755.
19059 * tests/check/elements/videorate.c: (GST_START_TEST),
19061 Unit test for the above by Yours Truly.
19063 2007-04-23 20:04:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19065 gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
19066 Original commit message from CVS:
19067 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
19068 (gst_adder_sink_event), (gst_adder_collected):
19069 Fix non-flushing segmented seeks, Fixes #340060 for me
19071 2007-04-21 15:29:27 +0000 Tim-Philipp Müller <tim@centricular.net>
19074 ChangeLog surgery: add API keyword
19075 Original commit message from CVS:
19076 ChangeLog surgery: add API keyword
19078 2007-04-21 15:25:22 +0000 Olivier Crete <tester@tester.ca>
19080 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
19081 Original commit message from CVS:
19082 Patch by: Olivier Crete <tester at tester ca>
19083 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19084 (gst_base_rtp_audio_payload_class_init),
19085 (gst_base_rtp_audio_payload_init),
19086 (gst_base_rtp_audio_payload_dispose):
19087 Chain up to parent class in dispose function; get rid of
19088 unnecessary 'diposed' flag in private structure (#415001).
19090 2007-04-21 15:10:25 +0000 Tim-Philipp Müller <tim@centricular.net>
19092 Some minor docs fixes and additions; also add missing 'Since' bits.
19093 Original commit message from CVS:
19094 * docs/libs/gst-plugins-base-libs.types:
19095 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19096 (gst_base_rtp_audio_payload_class_init):
19097 * gst-libs/gst/rtp/gstbasertpdepayload.c:
19098 * gst-libs/gst/rtp/gstbasertppayload.c:
19099 Some minor docs fixes and additions; also add missing 'Since' bits.
19101 2007-04-21 14:40:45 +0000 Zeeshan Ali <zeenix@gmail.com>
19103 gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
19104 Original commit message from CVS:
19105 Patch by: Zeeshan Ali <zeenix gmail com>
19106 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19107 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
19108 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
19109 (gst_base_rtp_audio_payload_push):
19110 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
19111 The recently-added gst_base_rtp_audio_payload_push() should take an
19112 object of type GstBaseRTPAudioPayload as first argument (#431672).
19114 2007-04-21 14:14:24 +0000 Tim-Philipp Müller <tim@centricular.net>
19116 gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
19117 Original commit message from CVS:
19118 * gst/audioresample/gstaudioresample.c:
19119 Make more functions static, just because we can.
19121 2007-04-21 13:54:39 +0000 Tim-Philipp Müller <tim@centricular.net>
19123 tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
19124 Original commit message from CVS:
19125 * tests/check/elements/audioresample.c:
19126 Add unit test for audioresample shutdown crasher (#420106).
19128 2007-04-20 10:42:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19130 gst/subparse/: Use GST_DISABLE_XML here
19131 Original commit message from CVS:
19132 * gst/subparse/gstsubparse.c:
19133 * gst/subparse/samiparse.c:
19134 Use GST_DISABLE_XML here
19135 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
19136 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
19137 (gst_xvimagesink_buffer_alloc),
19138 (gst_xvimagesink_navigation_send_event):
19139 * sys/xvimage/xvimagesink.h:
19140 Include stdlib.h when using atoi.
19141 * tests/check/elements/playbin.c: (playbin_suite):
19142 Use GST_DISABLE_REGISTRY here
19144 2007-04-19 16:58:53 +0000 Michael Smith <msmith@xiph.org>
19146 ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
19147 Original commit message from CVS:
19148 * ext/theora/gsttheoraenc.h:
19149 * ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
19150 (theora_enc_sink_event), (theora_enc_change_state):
19151 Track initialisation state; don't try to use encoder state if we're
19152 not initialised (it'll segfault).
19154 2007-04-18 11:06:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19156 tests/check/pipelines/.cvsignore: Fix build.
19157 Original commit message from CVS:
19158 * tests/check/pipelines/.cvsignore:
19161 2007-04-17 10:56:37 +0000 Tim-Philipp Müller <tim@centricular.net>
19163 gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
19164 Original commit message from CVS:
19165 * gst/app/Makefile.am:
19166 Fix CFLAGS and hopefully #430594.
19168 2007-04-17 02:53:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19170 gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
19171 Original commit message from CVS:
19172 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19173 Allow random depths between 1 and 32 instead of only multiplies of 8.
19175 2007-04-17 02:04:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19177 gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
19178 Original commit message from CVS:
19179 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19180 Set the maximum number of channels for PCM and float in the correct
19181 place to have it also used when creating the template caps.
19183 2007-04-17 01:56:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19185 gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
19186 Original commit message from CVS:
19187 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19188 Correctly support 4, 6 and 8 channels with normal PCM and float
19190 Fix the depth and signedness calculation in extensible wav files and
19191 also handle 1, 2, 4, 6, 8 channels here when a file without channel
19193 Add support for float, alaw and mulaw in extensible wav files.
19194 This allows correct playback of all but 5 files from
19195 http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
19196 (gst_riff_create_audio_template_caps):
19197 Add voxware and float formats to the template caps.
19199 2007-04-16 22:20:03 +0000 Vincent Torri <vtorri@univ-evry.fr>
19201 ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
19202 Original commit message from CVS:
19203 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
19204 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
19205 Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
19206 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19207 * gst/audioresample/gstaudioresample.c: (audioresample_do_output):
19208 Use the correct format strings for integer formats.
19210 2007-04-16 21:44:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19212 * gst-plugins-base.doap:
19214 Original commit message from CVS:
19217 2007-04-16 21:42:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19219 * gst-plugins-base.doap:
19221 Original commit message from CVS:
19224 2007-04-15 14:35:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19226 ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
19227 Original commit message from CVS:
19228 * ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
19229 Don't use pad_alloc_buffer_and_set_caps to create a small header
19230 packet, or, worse, to create a big temporary video buffer using the
19233 2007-04-14 12:34:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19235 gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19236 Original commit message from CVS:
19237 * gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
19238 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19239 GST_START_TEST, buffer_probe_cb, GST_START_TEST):
19240 Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
19242 2007-04-13 22:10:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19244 * gst/tcp/gstmultifdsink.c:
19246 Original commit message from CVS:
19249 2007-04-13 21:55:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19252 * tests/check/pipelines/streamheader.c:
19253 tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19254 Original commit message from CVS:
19255 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
19256 GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
19257 streamheader_suite):
19258 Add another test set up for failure
19260 2007-04-13 21:09:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19262 * ext/ogg/gstoggmux.c:
19263 * gst/gdp/gstgdpdepay.c:
19265 Original commit message from CVS:
19268 2007-04-13 21:08:11 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19270 tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
19271 Original commit message from CVS:
19272 * tests/check/Makefile.am:
19273 * tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
19274 GST_START_TEST, streamheader_suite, main):
19275 Add a test for the streamheader bug Wim fixed.
19277 2007-04-13 11:42:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19279 ext/theora/theoradec.c: Fix misleading comment.
19280 Original commit message from CVS:
19281 * ext/theora/theoradec.c: (theora_dec_sink_event):
19282 Fix misleading comment.
19284 2007-04-13 06:17:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19286 gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
19287 Original commit message from CVS:
19288 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19289 More sanity checks for the header fields.
19291 2007-04-12 16:36:36 +0000 Tim-Philipp Müller <tim@centricular.net>
19293 gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
19294 Original commit message from CVS:
19295 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
19296 Try encodings from all environment variables, not just those in the
19297 first environment variable that is set.
19299 2007-04-12 15:00:03 +0000 Wim Taymans <wim.taymans@gmail.com>
19301 gst/videorate/gstvideorate.c: Add some debug.
19302 Original commit message from CVS:
19303 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
19304 (gst_video_rate_chain):
19306 * tests/check/elements/videorate.c: (GST_START_TEST),
19308 Added check for videorate changing caps handling. Closes #421834.
19310 2007-04-12 12:57:33 +0000 Michael Smith <msmith@xiph.org>
19312 ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
19313 Original commit message from CVS:
19314 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
19315 Use scale functions to avoid overflow when calculating duration of
19318 2007-04-12 12:19:20 +0000 Tim-Philipp Müller <tim@centricular.net>
19320 API: add gst_tag_freeform_string_to_utf8() (#405072).
19321 Original commit message from CVS:
19322 * docs/libs/gst-plugins-base-libs-sections.txt:
19323 * gst-libs/gst/tag/tag.h:
19324 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
19325 API: add gst_tag_freeform_string_to_utf8() (#405072).
19326 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
19327 Use gst_tag_freeform_string_to_utf8() here.
19329 2007-04-12 10:38:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19331 * gst/tcp/gstmultifdsink.c:
19333 Original commit message from CVS:
19336 2007-04-12 10:03:22 +0000 Wim Taymans <wim.taymans@gmail.com>
19338 gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
19339 Original commit message from CVS:
19340 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
19341 (gst_gdp_pay_sink_event):
19342 Make sure we set the IN_CAPS flag correctly.
19343 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
19344 Get the IN_CAPS flag before we call functions that mess with the flags.
19346 2007-04-10 20:37:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19349 * gst/gdp/gstgdppay.c:
19350 gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
19351 Original commit message from CVS:
19352 * gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
19353 gst_gdp_pay_chain, gst_gdp_pay_sink_event):
19354 Only stamp buffers with offset/offset_end right before they get
19355 pushed. This ensures offset continuity, which was not the case
19357 gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
19359 2007-04-10 20:25:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19361 * gst/gdp/gstgdpdepay.c:
19362 * gst/gdp/gstgdppay.c:
19364 Original commit message from CVS:
19367 2007-04-10 11:23:18 +0000 Christian Schaller <uraeus@gnome.org>
19370 * gst-plugins-base.spec.in:
19371 update spec file for RTP changes
19372 Original commit message from CVS:
19373 update spec file for RTP changes
19375 2007-04-06 12:58:06 +0000 Wim Taymans <wim.taymans@gmail.com>
19377 gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
19378 Original commit message from CVS:
19379 * gst/playback/gstplaybin.c: (add_sink),
19380 (gst_play_bin_change_state):
19381 Activate sync in playbin, we are ready to handle it for live streams.
19383 2007-04-06 09:56:18 +0000 Tim-Philipp Müller <tim@centricular.net>
19385 tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
19386 Original commit message from CVS:
19387 * tests/check/elements/playbin.c:
19388 (test_sink_usage_video_only_stream), (playbin_suite):
19389 Add small test for stream-info-value-array code paths.
19391 2007-04-05 15:44:40 +0000 Wim Taymans <wim.taymans@gmail.com>
19393 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
19394 Original commit message from CVS:
19395 * gst-libs/gst/audio/gstbaseaudiosink.c:
19396 (gst_base_audio_sink_skew_slaving):
19397 Don't try to create invalid calibration parameters by making the
19398 internal time go backwards, instead make external time go forward.
19400 2007-04-05 10:27:06 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
19402 gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
19403 Original commit message from CVS:
19404 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
19405 * gst/playback/gstplaybasebin.c: (add_stream):
19406 Fix leak in add_stream(), when g_value_set_object() increases the
19407 refcount of streaminfo object. Fixes #426250.
19409 2007-04-04 02:45:03 +0000 David Schleef <ds@schleef.org>
19411 gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
19412 Original commit message from CVS:
19413 * gst/videotestsrc/gstvideotestsrc.c:
19414 * gst/videotestsrc/gstvideotestsrc.h:
19415 * gst/videotestsrc/videotestsrc.c:
19416 * gst/videotestsrc/videotestsrc.h:
19417 Add a test pattern called "circular", which has concentric
19418 rings with varying radial frequency. The main purpose of this
19419 pattern is to test fidelity loss in a filter or scaler element.
19420 Notably, this pattern is scale invariant, and is optimally viewed
19421 with a width (and height) of 400.
19423 2007-04-03 11:10:52 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
19425 gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
19426 Original commit message from CVS:
19427 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
19428 * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
19429 (deactivate_free_recursive):
19430 Decodebin2 doesn't unref pads it obtains in some occasions:
19431 - multiqueue src pads, when either connecting further or exposing
19432 - sink pads of new autoplugged elements
19433 - peer pads when recursively freeing elements
19436 2007-03-30 17:05:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19438 gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
19439 Original commit message from CVS:
19440 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19441 Add audio/x-raw-float support, now that audioconvert support
19442 non-native endianness floats.
19444 2007-03-30 15:00:49 +0000 Tim-Philipp Müller <tim@centricular.net>
19446 docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
19447 Original commit message from CVS:
19448 * docs/libs/gst-plugins-base-libs-docs.sgml:
19449 gstreamer-plugins-base.pc doesn't exist, it's
19450 gstreamer-plugins-base-0.10.pc.
19452 2007-03-29 18:42:34 +0000 René Stadler <mail@renestadler.de>
19454 with some minor changes
19455 Original commit message from CVS:
19456 Patch by: René Stadler <mail at renestadler dot de>
19457 with some minor changes
19458 * gst-libs/gst/floatcast/floatcast.h:
19459 Use more efficient float endianness conversion functions that don't
19460 involve 2 function calls per value.
19461 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
19462 (check_default), (audio_convert_prepare_context):
19463 * gst/audioconvert/gstaudioconvert.c:
19464 (gst_audio_convert_parse_caps), (make_lossless_changes):
19465 Support non-native endianness floats as input and output.
19467 * tests/check/elements/audioconvert.c: (verify_convert),
19469 Add unit tests for the non-native endianness float conversions.
19471 2007-03-29 16:23:53 +0000 Wim Taymans <wim.taymans@gmail.com>
19473 gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
19474 Original commit message from CVS:
19475 * gst-libs/gst/rtp/gstbasertpdepayload.c:
19476 (gst_base_rtp_depayload_base_init),
19477 (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
19478 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
19479 (gst_base_rtp_depayload_set_gst_timestamp),
19480 (gst_base_rtp_depayload_change_state),
19481 (gst_base_rtp_depayload_set_property),
19482 (gst_base_rtp_depayload_get_property):
19483 * gst-libs/gst/rtp/gstbasertpdepayload.h:
19484 Add Private structure.
19485 Bring element code to 2007.
19486 Parse clock-base caps param and use it when generating the
19488 Reset variables before going to PAUSED.
19491 2007-03-29 16:20:31 +0000 Wim Taymans <wim.taymans@gmail.com>
19494 Original commit message from CVS:
19495 * docs/libs/gst-plugins-base-libs-docs.sgml:
19496 * docs/libs/gst-plugins-base-libs-sections.txt:
19497 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19498 (gst_base_rtp_audio_payload_get_adapter):
19500 Fix some more docs.
19501 * gst-libs/gst/rtp/Makefile.am:
19502 * gst-libs/gst/rtp/gstrtcpbuffer.c:
19503 (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
19504 (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
19505 (gst_rtcp_buffer_get_packet_count), (read_packet_header),
19506 (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
19507 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
19508 (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
19509 (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
19510 (gst_rtcp_packet_sr_get_sender_info),
19511 (gst_rtcp_packet_sr_set_sender_info),
19512 (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
19513 (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
19514 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
19515 (gst_rtcp_packet_sdes_get_chunk_count),
19516 (gst_rtcp_packet_sdes_first_chunk),
19517 (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
19518 (gst_rtcp_packet_sdes_first_item),
19519 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
19520 (gst_rtcp_packet_bye_get_ssrc_count),
19521 (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
19522 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
19523 (gst_rtcp_packet_bye_get_reason_len),
19524 (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
19525 * gst-libs/gst/rtp/gstrtcpbuffer.h:
19526 Add new helper object for parsing and creating RTCP messages.
19528 2007-03-29 12:07:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19530 gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
19531 Original commit message from CVS:
19532 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19533 PCM samples with width=8 must be always unsigned, no matter what
19536 2007-03-29 11:24:47 +0000 Andy Wingo <wingo@pobox.com>
19538 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
19539 Original commit message from CVS:
19540 2007-03-29 Andy Wingo <wingo@pobox.com>
19541 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
19542 perfect offsets also, not just timestamps.
19543 * tests/check/elements/videorate.c (test_more): Test that given
19544 any incoming offsets, that videorate produces perfect offsets.
19546 2007-03-29 10:19:45 +0000 Wim Taymans <wim.taymans@gmail.com>
19548 gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
19549 Original commit message from CVS:
19550 * gst-libs/gst/riff/riff-ids.h:
19551 Add some more RIFF formats.
19553 2007-03-29 10:17:52 +0000 Wim Taymans <wim.taymans@gmail.com>
19555 gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
19556 Original commit message from CVS:
19557 * gst-libs/gst/rtp/gstrtpbuffer.c:
19558 (gst_rtp_buffer_default_clock_rate):
19559 * gst-libs/gst/rtp/gstrtpbuffer.h:
19560 Fix fixed payload names and docs.
19561 Added method to get the default clock rates of fixed payload types.
19562 API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
19564 2007-03-28 15:24:40 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
19566 tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
19567 Original commit message from CVS:
19568 * tests/check/pipelines/.cvsignore:
19569 Add new vorbisdec test to cvsignore.
19571 2007-03-28 14:50:47 +0000 Wim Taymans <wim.taymans@gmail.com>
19573 gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
19574 Original commit message from CVS:
19575 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
19576 (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
19577 (gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
19578 (gst_base_audio_sink_set_property),
19579 (gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
19580 (clock_convert_external), (gst_base_audio_sink_resample_slaving),
19581 (gst_base_audio_sink_skew_slaving),
19582 (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
19583 (gst_base_audio_sink_async_play):
19584 * gst-libs/gst/audio/gstbaseaudiosink.h:
19585 Store private stuff in GstBaseAudioSinkPrivate.
19586 Add configurable clock slaving modes property.
19587 API:: GstBaseAudioSink::slave-method property
19588 Some more latency reporting tweaks.
19589 Added skew based clock slaving correction and make it the default until
19590 the resampling method is more robust.
19592 2007-03-27 12:44:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19594 gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
19595 Original commit message from CVS:
19596 * gst/audioconvert/audioconvert.c:
19597 Add docs to the integer pack functions and implement proper
19598 rounding. Before we had rounding towards negative infinity, i.e.
19599 always the smaller number was taken. Now we use natural rounding,
19600 i.e. rounding to the nearest integer and to the one with the largest
19601 absolute value for X.5. The old rounding introduced some minor
19602 distortions. Fixes #420079
19603 * tests/check/elements/audioconvert.c: (GST_START_TEST):
19604 Fix one unit test that assumed the old rounding and added unit tests
19605 for checking signed/unsigned int16 <-> signed/unsigned int16 with
19606 depth 8, one for signed int16 <-> unsigned int16 and one for the new
19607 rounding from signed int32 to signed/unsigned int16.
19609 2007-03-27 11:31:17 +0000 Michael Smith <msmith@xiph.org>
19611 gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
19612 Original commit message from CVS:
19613 * gst/audioconvert/gstaudioconvert.c: (strip_width_64),
19614 (gst_audio_convert_transform_caps):
19615 Fix typo in debug line introduced recently, as pointed out on irc.
19617 2007-03-27 10:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
19619 Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
19620 Original commit message from CVS:
19621 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
19622 * tests/check/libs/tag.c: (GST_START_TEST):
19623 Make sure we parse floating-point numbers in vorbis comments
19624 correctly with either '.' or ',' as separator, no matter what
19625 the current locale is. Add unit test for this too.
19627 2007-03-27 09:37:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19629 * tests/check/pipelines/vorbisdec.c:
19631 Original commit message from CVS:
19634 2007-03-26 22:38:19 +0000 René Stadler <mail@renestadler.de>
19636 gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
19637 Original commit message from CVS:
19638 Patch by: René Stadler <mail at renestadler de>
19639 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
19640 When writing out floating-point numbers to vorbis comment tags, always
19641 use the same character as separator no matter what the current locale is
19643 * tests/check/libs/tag.c: (GST_START_TEST):
19644 Add unit tests for replaygain tags in vorbis comments (closes #423055).
19646 2007-03-26 20:56:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19648 ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
19649 Original commit message from CVS:
19650 * ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
19651 vorbis_handle_data_packet):
19652 Correctly set DURATION to generate a timestamp-continuous stream.
19653 One bug left at the end; see
19654 ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
19655 * tests/check/Makefile.am:
19656 * tests/check/pipelines/vorbisenc.c (GST_START_TEST):
19657 Add a test to check this. Without the above patch this test fails.
19659 2007-03-26 11:44:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19661 gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
19662 Original commit message from CVS:
19663 * gst-libs/gst/rtp/Makefile.am:
19664 The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
19666 2007-03-23 15:43:24 +0000 Christian Schaller <uraeus@gnome.org>
19668 * gst-plugins-base.spec.in:
19670 Original commit message from CVS:
19673 2007-03-23 12:32:33 +0000 Michael Smith <msmith@xiph.org>
19675 gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
19676 Original commit message from CVS:
19677 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
19678 (gst_video_rate_reset), (gst_video_rate_chain):
19679 If videorate changes caps, we can no longer use the old buffer
19680 (which may have a different size, incompatible with our caps).
19681 So don't do that; just duplicate the new frame more times.
19683 2007-03-22 17:43:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19685 gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
19686 Original commit message from CVS:
19687 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
19688 Remove playbin's override of the set_clock vmethod. It's irrelevant
19689 after Wim's commit on the 19th.
19691 2007-03-22 14:37:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19693 gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
19694 Original commit message from CVS:
19695 * gst-libs/gst/app/Makefile.am:
19696 Use GST_ALL_LDFLAGS, which actually exists, but maybe David
19697 can confirm that was what he wanted.
19699 2007-03-22 09:26:02 +0000 Wim Taymans <wim.taymans@gmail.com>
19701 ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
19702 Original commit message from CVS:
19703 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
19704 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
19705 * ext/gnomevfs/gstgnomevfssrc.h:
19706 Don't cache file sizes. Fixes #341078.
19708 2007-03-21 11:03:23 +0000 Tim-Philipp Müller <tim@centricular.net>
19710 gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
19711 Original commit message from CVS:
19712 * gst/playback/gstplaybin.c: (add_sink):
19713 Use GST_PTR_FORMAT to log caps.
19715 2007-03-21 10:23:11 +0000 Young-Ho Cha <ganadist@chollian.net>
19717 gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
19718 Original commit message from CVS:
19719 Patch by: Young-Ho Cha <ganadist at chollian net>
19720 * gst/subparse/samiparse.c: (handle_start_font):
19721 Special-case some more colour names that pango doesn't handle by
19722 default. Fixes #420578.
19724 2007-03-20 11:49:55 +0000 Michael Smith <msmith@xiph.org>
19726 ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
19727 Original commit message from CVS:
19728 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
19729 If we get a zero-sized input buffer, don't pass it to libvorbis, as
19730 that marks EOS internally. After that, libvorbis will buffer all
19731 input data, and encode none of it, eventually leading to memory
19734 2007-03-19 10:52:50 +0000 Wim Taymans <wim.taymans@gmail.com>
19736 gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
19737 Original commit message from CVS:
19738 * gst/playback/gstdecodebin.c: (remove_fakesink):
19739 Don't post STATE_DIRTY anymore.
19740 * gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
19741 (gst_play_bin_change_state):
19742 Remove stream_time reset in seek handling, core does that now.
19743 Disable clocking for live pipelines by forcing a NULL clock to the
19744 complete pipeline, core is too smart now for our previous hack.
19745 We can always autoplug in PAUSED now.
19747 2007-03-18 03:14:01 +0000 David Schleef <ds@schleef.org>
19749 REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
19750 Original commit message from CVS:
19751 * REQUIREMENTS: Update this file, change the formatting to make
19752 it more consistent, plus more machine readable.
19754 2007-03-16 17:29:09 +0000 Michael Smith <msmith@xiph.org>
19756 gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
19757 Original commit message from CVS:
19758 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
19759 (strip_width_64), (append_with_other_format):
19760 Previous fix was too simplistic, and broke the tests. Use a better
19761 approach; only strip 64 from widths for integer audio.
19763 2007-03-16 16:42:23 +0000 Michael Smith <msmith@xiph.org>
19765 gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
19766 Original commit message from CVS:
19767 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
19768 (gst_audio_convert_transform_caps):
19769 We don't support 64 bit integer audio, so don't try to claim we can.
19770 Stops us producing caps don't match our template caps.
19773 2007-03-15 10:52:21 +0000 Michael Smith <msmith@xiph.org>
19775 gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
19776 Original commit message from CVS:
19777 * gst/audioresample/gstaudioresample.c:
19778 (audioresample_check_discont), (audioresample_transform):
19779 Don't trigger discontinuities for very small imperfections; a filter
19780 flush will sound bad, and many plugins have rounding errors leading
19783 2007-03-14 21:11:18 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
19785 gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
19786 Original commit message from CVS:
19787 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19788 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
19789 Add min-ptime property to RTP base audio payloader. Patch by
19790 olivier.crete@collabora.co.uk.
19792 Indentation/whitespace/documentation fixes.
19794 2007-03-14 17:16:30 +0000 Julien Moutte <julien@moutte.net>
19796 gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
19797 Original commit message from CVS:
19798 2007-03-14 Julien MOUTTE <julien@moutte.net>
19799 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
19800 (audioresample_transform_size), (audioresample_do_output),
19801 (audioresample_transform), (audioresample_pushthrough): Handle
19802 discontinuous streams.
19803 * gst/audioresample/gstaudioresample.h:
19804 * tests/check/elements/audioresample.c:
19805 (test_discont_stream_instance), (GST_START_TEST),
19806 (audioresample_suite): Add a test for discontinuous streams.
19807 * win32/common/config.h: Updated.
19809 2007-03-14 15:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19811 po/: Update translations from translation project.
19812 Original commit message from CVS:
19826 Update translations from translation project.
19828 2007-03-14 15:05:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19830 * gst/gdp/gstgdpdepay.c:
19832 Original commit message from CVS:
19835 2007-03-14 14:48:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19837 gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
19838 Original commit message from CVS:
19839 * gst/audioresample/debug.h:
19840 * gst/audioresample/resample.c: (resample_init):
19841 Since I really am not interested in a debug line for each sample
19842 being processed, move the library's debugging to its own category,
19845 2007-03-14 14:09:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19847 * gst/audioresample/gstaudioresample.c:
19848 add debugging and reformat docs
19849 Original commit message from CVS:
19850 add debugging and reformat docs
19852 2007-03-12 23:29:07 +0000 Michael Smith <msmith@xiph.org>
19854 ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
19855 Original commit message from CVS:
19856 * ext/theora/theoradec.c: (theora_handle_type_packet):
19857 Since the plugin doesn't support anything other than 4:2:0 right
19858 now, post an error and fail if we get something else. Won't matter
19859 until libtheora supports the other pixel formats, but hopefully
19862 2007-03-12 15:50:35 +0000 Alex Lancaster <alexlan@fedoraproject.org>
19865 I'm too lazy to comment this
19866 Original commit message from CVS:
19867 Mention Patch by: Alex Lancaster in a recent commit.
19869 2007-03-12 11:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19871 examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.
19872 Original commit message from CVS:
19873 * examples/app/.cvsignore:
19874 The buildbot demands .cvsignore files, and I comply.
19876 2007-03-11 00:48:26 +0000 David Schleef <ds@schleef.org>
19878 Add appsrc/appsink example.
19879 Original commit message from CVS:
19881 * examples/Makefile.am:
19882 * examples/app/Makefile.am:
19883 * examples/app/appsrc_ex.c:
19884 Add appsrc/appsink example.
19885 * gst-libs/gst/app/Makefile.am:
19886 * gst-libs/gst/app/gstapp.c:
19887 * gst-libs/gst/app/gstappsink.c:
19888 * gst-libs/gst/app/gstappsink.h:
19889 * gst/app/gstapp.c:
19892 2007-03-10 15:59:33 +0000 Sébastien Moutte <sebastien@moutte.net>
19894 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
19895 Original commit message from CVS:
19896 * gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
19897 Use gst_guint64_to_gdouble for conversion.
19899 Add new files to the win32 MANIFEST.
19900 * win32/common/libgstaudio.def:
19901 * win32/common/libgstpbutils.def:
19902 Add new exported functions.
19903 * win32/vs6/gst_plugins_base.dsw:
19904 * win32/vs6/libgstdecodebin.dsp:
19905 * win32/vs6/libgstplaybin.dsp:
19906 Change the link to libgstpbutils.lib.
19907 * win32/vs6/libgstdecodebin2.dsp:
19908 Add a new project for decodebin2.
19909 * win32/vs6/libgstpbutils.dsp:
19910 Add a new project for pbutils.
19912 2007-03-10 12:18:58 +0000 Tim-Philipp Müller <tim@centricular.net>
19914 gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
19915 Original commit message from CVS:
19916 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
19917 Also accept partial dates with only year and month,
19918 like 1999-12-00 (fixes #410396 even more).
19919 * tests/check/libs/tag.c: (GST_START_TEST):
19920 Add unit test for the above.
19922 2007-03-10 11:21:08 +0000 Tim-Philipp Müller <tim@centricular.net>
19924 tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
19925 Original commit message from CVS:
19926 * tests/check/elements/subparse.c: (GST_START_TEST),
19928 Add unit test for MPL2 subtitle format (#413799).
19930 2007-03-10 11:17:52 +0000 Kamil Pawlowski <kamilpe@gmail.com>
19932 gst/subparse/: Add support for MPL2 subtitle format (#413799).
19933 Original commit message from CVS:
19934 Patch by: Kamil Pawlowski <kamilpe gmail com>
19935 * gst/subparse/Makefile.am:
19936 * gst/subparse/gstsubparse.c:
19937 (gst_sub_parse_data_format_autodetect),
19938 (gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
19939 (gst_subparse_type_find):
19940 * gst/subparse/gstsubparse.h:
19941 * gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
19942 * gst/subparse/mpl2parse.h:
19943 Add support for MPL2 subtitle format (#413799).
19945 2007-03-09 17:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
19947 configure.ac: We require core CVS for the new buffer metadata copy functions.
19948 Original commit message from CVS:
19950 We require core CVS for the new buffer metadata copy functions.
19952 2007-03-09 16:51:13 +0000 Wim Taymans <wim.taymans@gmail.com>
19954 gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
19955 Original commit message from CVS:
19956 * gst-libs/gst/tag/gstid3tag.c:
19957 Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
19960 2007-03-09 16:46:35 +0000 Wim Taymans <wim.taymans@gmail.com>
19962 ext/libvisual/visual.c: Improve adapter usage and comments.
19963 Original commit message from CVS:
19964 * ext/libvisual/visual.c: (gst_visual_sink_setcaps),
19965 (gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
19966 Improve adapter usage and comments.
19968 2007-03-09 16:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
19970 Use new metadata copy function.
19971 Original commit message from CVS:
19972 * ext/pango/gsttextrender.c: (gst_text_render_chain):
19973 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
19974 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
19975 Use new metadata copy function.
19976 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
19977 (gst_ffmpegcsp_transform):
19978 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
19979 Basetransform copied the metadata for us.
19981 2007-03-09 16:28:04 +0000 Tim-Philipp Müller <tim@centricular.net>
19983 ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
19984 Original commit message from CVS:
19985 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
19986 (gst_text_overlay_video_event):
19987 Some more logging. Only accept newsegment events in TIME format and
19988 send a WARNING message if they are not in TIME format.
19989 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
19990 (gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
19991 (gst_sub_parse_chain), (gst_sub_parse_sink_event):
19992 * gst/subparse/gstsubparse.h:
19993 No need to allocate GstSegment structure dynamically, just put it
19994 into the instance structure; ignore newsegment events in BYTE
19995 format and in particular don't let it overwrite our saved TIME
19996 segment from the last seek.
19998 2007-03-09 13:05:04 +0000 Michael Smith <msmith@xiph.org>
20000 gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
20001 Original commit message from CVS:
20002 * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
20003 Replace AC3 typefinder with one that isn't terrible, and actually
20006 2007-03-09 12:22:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20008 gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
20009 Original commit message from CVS:
20010 * gst/audioconvert/gstaudioconvert.c:
20011 (gst_audio_convert_transform):
20012 fix error category and translatable string
20014 2007-03-09 11:23:32 +0000 Tim-Philipp Müller <tim@centricular.net>
20016 pkgconfig/: Fix up utils => pbutils here too.
20017 Original commit message from CVS:
20018 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
20019 * pkgconfig/gstreamer-plugins-base.pc.in:
20020 Fix up utils => pbutils here too.
20022 2007-03-09 10:49:53 +0000 Tim-Philipp Müller <tim@centricular.net>
20024 gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
20025 Original commit message from CVS:
20026 * gst/subparse/gstsubparse.c: (handle_buffer):
20027 Break out of loop in chain function as soon as possible if we get
20028 a non-OK flow return.
20030 2007-03-08 18:26:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20032 tests/check/elements/alsa.c: Unref the mixer if the state change fails too (if the alsa devices are inaccessible, for...
20033 Original commit message from CVS:
20034 * tests/check/elements/alsa.c: (GST_START_TEST):
20035 Unref the mixer if the state change fails too (if the
20036 alsa devices are inaccessible, for example)
20038 2007-03-08 17:49:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20040 tests/check/Makefile.am: Don't test libvisual elements in the states check, because libvisual seems to leak internally.
20041 Original commit message from CVS:
20042 * tests/check/Makefile.am:
20043 Don't test libvisual elements in the states check, because libvisual
20044 seems to leak internally.
20045 Re-enable the alsa and states tests now that there's new suppressions
20047 * tests/check/elements/alsa.c: (GST_START_TEST):
20048 Don't leak the alsamixer we instantiated.
20050 2007-03-08 15:22:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20052 sys/: Move some cleanup stuff from the state change handler into a _reset() function that can be called from _finaliz...
20053 Original commit message from CVS:
20054 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
20055 (gst_ximagesink_change_state), (gst_ximagesink_reset),
20056 (gst_ximagesink_finalize):
20057 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
20058 (gst_xvimagesink_reset), (gst_xvimagesink_finalize):
20059 Move some cleanup stuff from the state change handler into a _reset()
20060 function that can be called from _finalize(). This ensures that things
20061 get freed even if (for some reason) the NULL->READY state transition
20062 fails in the parent class.
20063 Even if a parent state change fails, process our downward state change
20064 logic instead of bailing out early.
20065 Free the correct xcontext pointer in ximagesink's xcontext_clear.
20067 2007-03-08 12:53:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20069 ext/alsa/gstalsasink.c: Extra log line.
20070 Original commit message from CVS:
20071 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
20073 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
20074 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
20075 Use pango_font_description_set_family_static instead of
20076 pango_font_description_set_family to save a string copy (it was
20077 leaking due to the strdup anyway)
20078 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
20079 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
20080 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
20081 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
20082 Chain up in finalize.
20084 2007-03-07 18:50:10 +0000 Tim-Philipp Müller <tim@centricular.net>
20086 gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
20087 Original commit message from CVS:
20088 * gst-libs/gst/interfaces/mixertrack.c:
20089 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
20090 (gst_mixer_track_set_property):
20091 API: add "untranslated-label" property which should be set by
20092 implementations at construct time (#414645).
20093 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
20094 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
20095 Set "untranslated-label" when constructing mixer track objects.
20096 * tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
20097 Unit test to check the above.
20099 2007-03-07 17:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
20101 ext/ogg/gstoggdemux.c: Fix confusing debug message.
20102 Original commit message from CVS:
20103 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
20104 Fix confusing debug message.
20106 2007-03-07 17:12:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20108 gst-plugins-base.doap: update doap file with new version
20109 Original commit message from CVS:
20110 * gst-plugins-base.doap:
20111 update doap file with new version
20113 2007-03-07 17:05:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20115 * gst/tcp/gstmultifdsink.c:
20117 Original commit message from CVS:
20120 2007-03-07 16:56:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20122 configure.ac: Back to CVS
20123 Original commit message from CVS:
20127 === release 0.10.12 ===
20129 2007-03-07 16:46:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20135 * docs/plugins/gst-plugins-base-plugins.args:
20136 * docs/plugins/inspect/plugin-adder.xml:
20137 * docs/plugins/inspect/plugin-alsa.xml:
20138 * docs/plugins/inspect/plugin-audioconvert.xml:
20139 * docs/plugins/inspect/plugin-audiorate.xml:
20140 * docs/plugins/inspect/plugin-audioresample.xml:
20141 * docs/plugins/inspect/plugin-audiotestsrc.xml:
20142 * docs/plugins/inspect/plugin-cdparanoia.xml:
20143 * docs/plugins/inspect/plugin-decodebin.xml:
20144 * docs/plugins/inspect/plugin-decodebin2.xml:
20145 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
20146 * docs/plugins/inspect/plugin-gdp.xml:
20147 * docs/plugins/inspect/plugin-gnomevfs.xml:
20148 * docs/plugins/inspect/plugin-libvisual.xml:
20149 * docs/plugins/inspect/plugin-ogg.xml:
20150 * docs/plugins/inspect/plugin-pango.xml:
20151 * docs/plugins/inspect/plugin-playbin.xml:
20152 * docs/plugins/inspect/plugin-subparse.xml:
20153 * docs/plugins/inspect/plugin-tcp.xml:
20154 * docs/plugins/inspect/plugin-theora.xml:
20155 * docs/plugins/inspect/plugin-typefindfunctions.xml:
20156 * docs/plugins/inspect/plugin-video4linux.xml:
20157 * docs/plugins/inspect/plugin-videorate.xml:
20158 * docs/plugins/inspect/plugin-videoscale.xml:
20159 * docs/plugins/inspect/plugin-videotestsrc.xml:
20160 * docs/plugins/inspect/plugin-volume.xml:
20161 * docs/plugins/inspect/plugin-vorbis.xml:
20162 * docs/plugins/inspect/plugin-ximagesink.xml:
20163 * docs/plugins/inspect/plugin-xvimagesink.xml:
20164 * win32/common/config.h:
20166 Original commit message from CVS:
20169 2007-03-07 15:35:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20188 Original commit message from CVS:
20191 2007-03-06 12:31:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20193 configure.ac: Bump version to 0.10.11.4 pre-release
20194 Original commit message from CVS:
20196 Bump version to 0.10.11.4 pre-release
20198 2007-03-06 12:10:08 +0000 Wim Taymans <wim.taymans@gmail.com>
20200 gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
20201 Original commit message from CVS:
20202 * gst-libs/gst/audio/gstbaseaudiosink.c:
20203 (gst_base_audio_sink_async_play):
20204 Fix regression that made GStreamer skip the first samples of audio.
20207 2007-03-05 11:21:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20209 configure.ac: Bump version to 0.10.11.3 pre-release
20210 Original commit message from CVS:
20212 Bump version to 0.10.11.3 pre-release
20214 2007-03-05 09:35:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20216 po/POTFILES.in: Update paths for the rename from utils to pbutils to fix the build.
20217 Original commit message from CVS:
20219 Update paths for the rename from utils to pbutils to fix the build.
20221 2007-03-05 09:27:55 +0000 Tim-Philipp Müller <tim@centricular.net>
20223 gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
20224 Original commit message from CVS:
20225 * gst-libs/gst/pbutils/Makefile.am:
20226 Change directory to install headers in from gst/utils to gst/pbutils
20229 2007-03-04 23:41:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20231 * tests/check/libs/.gitignore:
20233 Original commit message from CVS:
20236 2007-03-04 23:41:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20238 * win32/common/config.h:
20239 * win32/common/libgstutils.def:
20241 Original commit message from CVS:
20244 2007-03-04 23:39:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20246 rename utils to pbutils
20247 Original commit message from CVS:
20249 * docs/libs/gst-plugins-base-libs-docs.sgml:
20250 * docs/libs/gst-plugins-base-libs-sections.txt:
20251 * gst-libs/gst/Makefile.am:
20252 * gst-libs/gst/interfaces/mixer.c:
20253 * gst-libs/gst/pbutils/Makefile.am:
20254 * gst-libs/gst/pbutils/descriptions.c:
20255 (gst_pb_utils_get_source_description),
20256 (gst_pb_utils_get_sink_description),
20257 (gst_pb_utils_get_decoder_description),
20258 (gst_pb_utils_get_encoder_description),
20259 (gst_pb_utils_get_element_description),
20260 (gst_pb_utils_add_codec_description_to_tag_list),
20261 (gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
20262 * gst-libs/gst/pbutils/descriptions.h:
20263 * gst-libs/gst/pbutils/install-plugins.c:
20264 * gst-libs/gst/pbutils/install-plugins.h:
20265 * gst-libs/gst/pbutils/missing-plugins.c:
20266 (gst_missing_uri_source_message_new),
20267 (gst_missing_uri_sink_message_new),
20268 (gst_missing_element_message_new),
20269 (gst_missing_decoder_message_new),
20270 (gst_missing_encoder_message_new),
20271 (gst_missing_plugin_message_get_description):
20272 * gst-libs/gst/pbutils/missing-plugins.h:
20273 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
20274 * gst-libs/gst/pbutils/pbutils.h:
20275 * gst-libs/gst/utils/Makefile.am:
20276 * gst-libs/gst/utils/base-utils.c:
20277 * gst-libs/gst/utils/base-utils.h:
20278 * gst-libs/gst/utils/descriptions.c:
20279 * gst-libs/gst/utils/descriptions.h:
20280 * gst-libs/gst/utils/install-plugins.c:
20281 * gst-libs/gst/utils/install-plugins.h:
20282 * gst-libs/gst/utils/missing-plugins.c:
20283 * gst-libs/gst/utils/missing-plugins.h:
20284 * gst-plugins-base.spec.in:
20285 * gst/playback/Makefile.am:
20286 * gst/playback/gstdecodebin.c:
20287 * gst/playback/gstdecodebin2.c:
20288 * gst/playback/gstplaybasebin.c: (setup_subtitle),
20289 (gen_source_element):
20290 * gst/playback/gstplaybin.c: (plugin_init):
20291 * tests/check/Makefile.am:
20292 * tests/check/libs/pbutils.c: (GST_START_TEST),
20293 (test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
20294 * tests/check/libs/utils.c:
20295 rename utils to pbutils
20297 2007-03-03 10:23:03 +0000 David Schleef <ds@schleef.org>
20299 gst-libs/gst/app/Makefile.am: Install the headers.
20300 Original commit message from CVS:
20301 * gst-libs/gst/app/Makefile.am:
20302 Install the headers.
20304 2007-03-03 10:10:30 +0000 David Schleef <ds@schleef.org>
20306 gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.
20307 Original commit message from CVS:
20308 * gst-libs/gst/app/Makefile.am:
20309 * gst-libs/gst/app/gstappbuffer.c:
20310 * gst-libs/gst/app/gstappbuffer.h:
20311 * gst-libs/gst/app/gstappsrc.c:
20312 Add GstAppBuffer that includes a callback and closure for
20313 proper handling of data chunks.
20315 2007-03-03 09:06:06 +0000 David Schleef <ds@schleef.org>
20317 gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.
20318 Original commit message from CVS:
20319 * gst-libs/gst/app/gstappsrc.c:
20320 * gst-libs/gst/app/gstappsrc.h:
20321 Hacking to address issues in 413418.
20323 2007-03-03 08:16:57 +0000 David Schleef <ds@schleef.org>
20325 Move the app library to gst-libs/gst/app (duh!)
20326 Original commit message from CVS:
20330 * gst-libs/gst/Makefile.am:
20331 * gst-libs/gst/app/Makefile.am:
20332 * gst-libs/gst/app/gstapp.c:
20333 * gst-libs/gst/app/gstappsrc.c:
20334 * gst-libs/gst/app/gstappsrc.h:
20335 * gst/app/Makefile.am:
20336 * gst/app/gstapp.c:
20337 * gst/app/gstappsrc.c:
20338 * gst/app/gstappsrc.h:
20339 Move the app library to gst-libs/gst/app (duh!)
20341 2007-03-02 12:59:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20343 Add documentation for decodebin2 that indicates that the API is still unstable.
20344 Original commit message from CVS:
20345 * docs/plugins/Makefile.am:
20346 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
20347 * docs/plugins/gst-plugins-base-plugins-sections.txt:
20348 * docs/plugins/inspect/plugin-decodebin2.xml:
20349 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
20350 Add documentation for decodebin2 that indicates that the API
20353 2007-03-01 18:50:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20355 configure.ac: Update to 0.10.11.2 (0.10.12 pre-release)
20356 Original commit message from CVS:
20358 Update to 0.10.11.2 (0.10.12 pre-release)
20360 2007-03-01 17:29:55 +0000 Wim Taymans <wim.taymans@gmail.com>
20362 gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
20363 Original commit message from CVS:
20364 * gst-libs/gst/audio/gstbaseaudiosink.c:
20365 (gst_base_audio_sink_async_play):
20366 base time is irrelevant here.
20368 2007-03-01 17:01:43 +0000 Wim Taymans <wim.taymans@gmail.com>
20370 gst-libs/gst/audio/: Improve debugging.
20371 Original commit message from CVS:
20372 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
20373 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
20375 * gst-libs/gst/audio/gstbaseaudiosink.c:
20376 (gst_base_audio_sink_query), (gst_base_audio_sink_event),
20377 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
20378 Improve latency and clock slaving calculations.
20379 Improve slave clock calibration.
20380 * gst-libs/gst/audio/gstringbuffer.c:
20381 (gst_ring_buffer_commit_full):
20382 When we are asked to render N sample to 0 bytes, return N.
20384 2007-03-01 16:48:45 +0000 Wim Taymans <wim.taymans@gmail.com>
20386 ext/alsa/gstalsasink.*: Remove unused dispose function.
20387 Original commit message from CVS:
20388 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
20389 (gst_alsasink_write), (gst_alsasink_reset):
20390 * ext/alsa/gstalsasink.h:
20391 Remove unused dispose function.
20392 Rename lock to not interfere with alsasrc lock.
20393 * ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
20394 (gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
20395 (gst_alsasrc_read), (gst_alsasrc_reset):
20396 * ext/alsa/gstalsasrc.h:
20397 Implement finalize function.
20398 Use lock to protect alsa access.
20400 Fine tune sw params.
20402 2007-03-01 10:20:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20407 Original commit message from CVS:
20410 2007-02-28 19:27:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20412 configure.ac: Convert to new AG_GST style.
20413 Original commit message from CVS:
20415 Convert to new AG_GST style.
20417 2007-02-28 15:17:20 +0000 Ed Catmur <ed@catmur.co.uk>
20419 gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
20420 Original commit message from CVS:
20421 Patch by: Ed Catmur <ed at catmur dot co dot uk>
20422 * gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
20423 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
20424 Fix race condition when rapidly switching visualisations in playbin.
20427 2007-02-28 15:11:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20429 tests/check/Makefile.am: Include local stuff before system installed things in LDFLAGS and
20430 Original commit message from CVS:
20431 * tests/check/Makefile.am:
20432 Include local stuff before system installed things in LDFLAGS and
20435 2007-02-28 15:10:06 +0000 Wim Taymans <wim.taymans@gmail.com>
20437 ext/ogg/gstoggdemux.c: Improve debugging.
20438 Original commit message from CVS:
20439 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
20442 2007-02-28 15:05:03 +0000 Wim Taymans <wim.taymans@gmail.com>
20444 sys/v4l/: Fix duration and timestamping, taking latency into account.
20445 Original commit message from CVS:
20446 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
20447 (gst_v4lsrc_fixate), (gst_v4lsrc_query):
20448 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
20449 Fix duration and timestamping, taking latency into account.
20450 Implement latency query.
20452 2007-02-28 15:02:25 +0000 Wim Taymans <wim.taymans@gmail.com>
20454 gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
20455 Original commit message from CVS:
20456 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
20457 (gst_audio_clock_new):
20459 * gst-libs/gst/audio/gstbaseaudiosink.c:
20460 (gst_base_audio_sink_init), (gst_base_audio_sink_query):
20461 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
20462 (gst_base_audio_src_query), (gst_base_audio_src_get_offset),
20463 (gst_base_audio_src_create):
20464 Improve latency query code.
20465 Use proper clock names.
20467 2007-02-28 12:57:46 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20469 * tests/check/generic/states.c:
20471 Original commit message from CVS:
20474 2007-02-28 12:44:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20476 tests/check/generic/states.c: Copy the states.c test from core again
20477 Original commit message from CVS:
20478 * tests/check/generic/states.c: (GST_START_TEST):
20479 Copy the states.c test from core again
20480 * tests/check/Makefile.am:
20481 ignore cdio and cdparanoiasrc
20483 2007-02-28 12:08:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20485 gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
20486 Original commit message from CVS:
20487 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
20488 (double_hq), (audio_convert_get_func_index), (check_default),
20489 (audio_convert_prepare_context), (audio_convert_convert):
20490 Also make valgrind happy and avoid copying data in some cases.
20492 2007-02-28 11:58:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20494 * tests/check/generic/states.c:
20496 Original commit message from CVS:
20499 2007-02-28 11:47:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20501 Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
20502 Original commit message from CVS:
20503 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
20504 (double_hq), (audio_convert_get_func_index),
20505 (audio_convert_prepare_context), (audio_convert_convert):
20506 * gst/audioconvert/gstaudioconvert.c:
20507 (gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
20508 (gst_audio_convert_transform_caps):
20509 * tests/check/elements/audioconvert.c: (GST_START_TEST),
20510 (audioconvert_suite):
20511 Don't run inplace if that overwrites source data as we go. Add more
20512 tests. Fixes #339837 even more.
20514 2007-02-27 18:45:37 +0000 Julien Moutte <julien@moutte.net>
20516 tests/examples/seek/seek.c: Fix various seeking bugs (Slider was not updating when doing a non flushing seek, Reverse...
20517 Original commit message from CVS:
20518 2007-02-27 Julien MOUTTE <julien@moutte.net>
20519 * tests/examples/seek/seek.c: (do_seek), (set_update_scale),
20520 (msg_segment_done): Fix various seeking bugs (Slider was not
20521 updating when doing a non flushing seek, Reverse playback
20522 on segment seek was wrong).
20524 2007-02-26 21:01:03 +0000 David Schleef <ds@schleef.org>
20526 Add a new plugin/library to make it easy for apps to shove data into a pipeline.
20527 Original commit message from CVS:
20529 * gst/app/Makefile.am:
20530 * gst/app/gstapp.c:
20531 * gst/app/gstappsrc.c:
20532 * gst/app/gstappsrc.h:
20533 Add a new plugin/library to make it easy for apps to shove
20534 data into a pipeline.
20536 2007-02-26 11:48:49 +0000 Wim Taymans <wim.taymans@gmail.com>
20538 tests/examples/seek/seek.c: When we stop scrubbing, don't leave the pipeline PLAYING when we requested a PAUSED state.
20539 Original commit message from CVS:
20540 * tests/examples/seek/seek.c: (stop_seek):
20541 When we stop scrubbing, don't leave the pipeline PLAYING when we
20542 requested a PAUSED state.
20544 2007-02-25 23:51:03 +0000 René Stadler <mail@renestadler.de>
20546 gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
20547 Original commit message from CVS:
20548 Patch by: René Stadler <mail at renestadler de>
20549 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
20550 Parse date strings in vorbis comments that have an invalid (zero)
20551 month or day (#410396).
20552 * tests/check/libs/tag.c: (GST_START_TEST):
20553 Test case for the above.
20555 2007-02-24 20:12:49 +0000 Loïc Minier <lool+gnome@via.ecp.fr>
20557 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
20558 Original commit message from CVS:
20559 Patch by: Loïc Minier <lool+gnome at via ecp fr>
20561 * ext/alsa/Makefile.am:
20562 * gst/audiotestsrc/Makefile.am:
20563 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
20565 2007-02-23 18:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
20567 gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
20568 Original commit message from CVS:
20569 * gst/playback/gstplaybin.c:
20570 Improve docs: point out that the application needs to assist playbin
20573 2007-02-23 13:10:50 +0000 Tim-Philipp Müller <tim@centricular.net>
20575 Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
20576 Original commit message from CVS:
20577 * gst-libs/gst/utils/install-plugins.c:
20578 * gst-libs/gst/utils/missing-plugins.c:
20579 * tests/check/libs/utils.c: (missing_msg_check_getters):
20580 Change GStreamer marker prefix in detail string from 'gstreamer.net'
20581 to just 'gstreamer'. Document the caps string component of the
20582 decoder/encoder detail a bit better, since not everyone will be
20583 familiar with the GStreamer media type/caps system (but they better
20584 enjoy nested itemized lists).
20586 2007-02-22 12:57:47 +0000 Tim-Philipp Müller <tim@centricular.net>
20588 gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
20589 Original commit message from CVS:
20590 * gst-libs/gst/netbuffer/gstnetbuffer.c:
20591 (notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
20592 Fix copying of GstNetBuffer (would crash before, or at least lead to
20593 invalid memory access, #410772), for now by copying the GstBuffer copy
20594 code from the core over here so we can copy the GstBuffer fields on a
20595 provided buffer instance (of type GstNetBuffer in this case). Would be
20596 better to fix this with some support by the core though (and in the long
20597 run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
20598 * tests/check/Makefile.am:
20599 Enable unit test for GstNetBuffer.
20601 2007-02-22 11:04:10 +0000 Andy Wingo <wingo@pobox.com>
20604 * gst-libs/gst/audio/gstbaseaudiosink.c:
20605 gst-libs/gst/audio/gstbaseaudiosink.c
20606 Original commit message from CVS:
20607 2007-02-22 Andy Wingo <wingo@pobox.com>
20608 * gst-libs/gst/audio/gstbaseaudiosink.c
20609 (gst_base_audio_sink_init): Disable pull-mode activation until we
20610 figure out how to make audio sinks go to PLAYING.
20612 2007-02-22 09:04:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20614 Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
20615 Original commit message from CVS:
20616 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
20617 (double_hq), (audio_convert_get_func_index),
20618 (audio_convert_prepare_context), (audio_convert_convert):
20619 * gst/audioconvert/audioconvert.h:
20620 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
20621 (gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
20622 * gst/audioconvert/gstchannelmix.h:
20623 * tests/check/elements/audioconvert.c: (GST_START_TEST):
20624 Add float as an intermediate format, as well as float mixing. Enable
20625 test that was failing before. Fixes #339837
20627 2007-02-21 16:12:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20629 tests/examples/seek/seek.c: Undo the previous commit: -1 as a stop time implies that the stop time is the end of file...
20630 Original commit message from CVS:
20631 * tests/examples/seek/seek.c: (do_seek):
20632 Undo the previous commit: -1 as a stop time implies that the stop
20633 time is the end of file, clearing any previously configured segment.
20635 2007-02-21 15:36:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20637 tests/examples/seek/seek.c: Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
20638 Original commit message from CVS:
20639 * tests/examples/seek/seek.c: (do_seek):
20640 Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
20642 2007-02-21 13:55:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20644 gst/volume/gstvolume.c: Unbreak volume, value remains gint.
20645 Original commit message from CVS:
20646 * gst/volume/gstvolume.c: (volume_process_int16),
20647 (volume_process_int16_clamp), (volume_set_caps):
20648 Unbreak volume, value remains gint.
20650 2007-02-21 13:08:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20652 gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
20653 Original commit message from CVS:
20654 * gst/volume/gstvolume.c: (volume_choose_func),
20655 (volume_update_real_volume), (gst_volume_set_volume),
20656 (gst_volume_init), (volume_process_double), (volume_process_float),
20657 (volume_process_int16), (volume_process_int16_clamp),
20658 (volume_set_caps), (volume_transform_ip), (volume_update_volume):
20659 * gst/volume/gstvolume.h:
20660 Extend float audio support (double) and some int->uint cleanups.
20662 2007-02-20 15:44:32 +0000 Edward Hervey <bilboed@bilboed.com>
20664 gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
20665 Original commit message from CVS:
20666 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
20667 (multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
20668 (sort_end_pads), (gst_decode_group_expose),
20669 (gst_decode_group_hide):
20670 Don't free groups from the streaming threads. Just put them aside and
20671 free them in dispose.
20673 2007-02-20 11:20:52 +0000 Edward Hervey <bilboed@bilboed.com>
20675 gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
20676 Original commit message from CVS:
20677 * gst/playback/gstdecodebin2.c: (connect_element),
20678 (pad_added_group_cb), (gst_decode_group_check_if_blocked),
20679 (sort_end_pads), (gst_decode_group_expose):
20680 Handle dynamic pads within groups.
20681 Sort pads before exposing them in order to make playbin happy.
20682 There still is a race with the multiqueue filling up. This should be
20686 2007-02-18 21:02:36 +0000 Tim-Philipp Müller <tim@centricular.net>
20688 gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
20689 Original commit message from CVS:
20690 * gst-libs/gst/utils/base-utils.c:
20691 * gst-libs/gst/utils/descriptions.c:
20692 * gst-libs/gst/utils/install-plugins.c:
20693 * gst-libs/gst/utils/missing-plugins.c:
20694 Some more docs (and descriptions for two subtitle formats).
20696 2007-02-16 10:19:45 +0000 Tim-Philipp Müller <tim@centricular.net>
20698 gst-libs/gst/audio/audio.c: Fix documentation.
20699 Original commit message from CVS:
20700 * gst-libs/gst/audio/audio.c:
20703 2007-02-16 10:15:46 +0000 Yves Lefebvre <ivanohe@abacom.com>
20705 gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
20706 Original commit message from CVS:
20707 Patch by: Yves Lefebvre <ivanohe abacom com>
20708 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
20709 Don't leak caps. Fixes #408278.
20711 2007-02-15 15:17:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20713 More docs coverage and some ChangeLog surgery (add missing names)
20714 Original commit message from CVS:
20715 * ext/cdparanoia/gstcdparanoiasrc.h:
20716 * ext/ogg/gstoggdemux.h:
20717 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
20718 (gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
20719 (gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
20720 * gst-libs/gst/audio/audio.h:
20721 * gst-libs/gst/audio/gstaudiofilter.h:
20722 * gst-libs/gst/interfaces/videoorientation.h:
20723 * gst/adder/gstadder.h:
20724 More docs coverage and some ChangeLog surgery (add missing names)
20726 2007-02-15 12:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
20728 sys/: Small constifications.
20729 Original commit message from CVS:
20730 * sys/ximage/ximagesink.c:
20731 (gst_ximagesink_calculate_pixel_aspect_ratio):
20732 * sys/xvimage/xvimagesink.c:
20733 (gst_xvimagesink_calculate_pixel_aspect_ratio):
20734 Small constifications.
20736 2007-02-15 12:06:25 +0000 Wim Taymans <wim.taymans@gmail.com>
20738 gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
20739 Original commit message from CVS:
20740 * gst-libs/gst/audio/gstbaseaudiosink.c:
20741 (gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
20742 (gst_base_audio_sink_render), (gst_base_audio_sink_callback),
20743 (gst_base_audio_sink_async_play),
20744 (gst_base_audio_sink_change_state):
20745 Answer latency query.
20746 Use configured latency when syncing.
20748 * gst-libs/gst/audio/gstbaseaudiosrc.c:
20749 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
20750 (gst_base_audio_src_query), (gst_base_audio_src_change_state):
20751 Fix possible memleak.
20752 Implement latency query.
20755 2007-02-15 11:59:41 +0000 Wim Taymans <wim.taymans@gmail.com>
20757 ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
20758 Original commit message from CVS:
20759 * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
20760 Ignore errors in reset, these are not fatal. They also grab the element
20761 lock which is already taking when this function is called. Fixes
20764 2007-02-13 13:50:56 +0000 Christian Schaller <uraeus@gnome.org>
20766 * gst-plugins-base.spec.in:
20767 add header file for easy codec install
20768 Original commit message from CVS:
20769 add header file for easy codec install
20771 2007-02-13 10:24:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20773 configure.ac: Remove 'tests/examples/xerror/Makefile' from output files again.
20774 Original commit message from CVS:
20776 Remove 'tests/examples/xerror/Makefile' from output files again.
20778 2007-02-13 09:12:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20780 Also crossref against gst-plugins-base-libs.
20781 Original commit message from CVS:
20783 * docs/plugins/Makefile.am:
20784 Also crossref against gst-plugins-base-libs.
20786 2007-02-12 20:42:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20788 Add crossreferences to glib/gobject/gstream docs.
20789 Original commit message from CVS:
20791 * docs/libs/Makefile.am:
20792 * docs/plugins/Makefile.am:
20793 Add crossreferences to glib/gobject/gstream docs.
20794 * gst-libs/gst/audio/audio.h:
20796 * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
20797 Add own debug category.
20799 2007-02-12 11:01:04 +0000 René Stadler <mail@renestadler.de>
20801 gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
20802 Original commit message from CVS:
20803 Patch by: René Stadler <mail at renestadler de>
20804 * gst-libs/gst/tag/gstvorbistag.c:
20805 Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
20808 2007-02-12 10:33:40 +0000 Tim-Philipp Müller <tim@centricular.net>
20810 gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
20811 Original commit message from CVS:
20812 * gst/playback/gstplaybasebin.c: (setup_source):
20813 When we have external subtitles and wait for the subtitle decodebin
20814 to get up and running, we set up a (sync) bus handler for the
20815 subtitle decodebin, so we can stop waiting when it posts an error
20816 message. However, we should do that before we set the subtitle
20817 decodebin's state to playing, otherwise things are racy and we might
20818 miss error messages posted before we had a chance to set up the bus.
20819 This should finally fix totem hanging on .txt pseudo-subtitle files.
20821 2007-02-10 19:27:48 +0000 Sébastien Moutte <sebastien@moutte.net>
20823 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
20824 Original commit message from CVS:
20825 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
20826 Use gst_gdouble_to_guint64 for conversions.
20827 * win32/common/config.h.in:
20828 Add a define for GST_INSTALL_PLUGINS_HELPER
20829 * win32/common/libgstaudio.def:
20830 * win32/common/libgstcdda.def:
20831 * win32/common/libgstnetbuffer.def:
20832 * win32/common/libgstrtp.def:
20833 * win32/common/libgutils.def:
20834 Add new exported functions.
20835 * win32/vs6/gst_plugins_base.dsw:
20836 * win32/vs6/libgstdecodebin.dsp:
20837 * win32/vs6/libgstnetbuffer.dsp:
20838 * win32/vs6/libgstplaybin.dsp:
20839 * win32/vs6/libgstrtp.dsp:
20840 * win32/vs6/libgstvorbis.dsp:
20841 * win32/vs6/libgstcdda.dsp:
20842 * win32/vs6/libgstgdp.dsp:
20843 * win32/vs6/libgstutils.dsp:
20844 Update and add new project files.
20846 2007-02-10 18:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
20848 gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
20849 Original commit message from CVS:
20850 * gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
20851 (subrip_remove_unhandled_tags), (parse_subrip):
20852 For SubRip (.srt) subtitles, ignore all markup tags we don't
20853 handle (like font tags, for example).
20854 * tests/check/elements/subparse.c:
20857 2007-02-09 13:28:01 +0000 Tim-Philipp Müller <tim@centricular.net>
20861 Original commit message from CVS:
20864 2007-02-09 13:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
20866 gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
20867 Original commit message from CVS:
20868 * gst/playback/gstdecodebin.c: (add_fakesink),
20869 (gst_decode_bin_change_state):
20870 * gst/playback/gstdecodebin2.c: (add_fakesink),
20871 (gst_decode_bin_change_state):
20872 Don't error out if there is no fakesink in the READY to NULL state
20873 change, since when decodebin is re-used, we're only adding the
20874 fakesink element in READY to PAUSED.
20875 * tests/check/elements/decodebin.c:
20876 (new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
20878 Minimal unit test to make sure we can use the same decodebin
20879 instance twice (at least with audiotestsrc input).
20881 2007-02-09 09:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
20883 ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
20884 Original commit message from CVS:
20885 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
20886 Try to get devic-name from device string first, and from handle only
20887 as fallback (seems to yield better results and is more robust
20888 against buggy probing code on the application side).
20890 2007-02-08 15:43:26 +0000 Julien Puydt <julien.puydt@laposte.net>
20892 ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
20893 Original commit message from CVS:
20894 Based on patch by: Julien Puydt <julien.puydt at laposte net>
20895 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
20896 (gst_alsa_find_device_name):
20897 * ext/alsa/gstalsa.h:
20898 * ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
20899 * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
20900 Improve device-name detection a bit, especially in the case where
20901 the device is not actually open (#405020, #405024). Move common code
20902 into gstalsa.c instead of duplicating it.
20904 2007-02-07 13:05:01 +0000 Tim-Philipp Müller <tim@centricular.net>
20906 gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
20907 Original commit message from CVS:
20908 * gst/audioconvert/gstaudioconvert.c:
20909 Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
20911 2007-02-06 17:47:32 +0000 Julien Moutte <julien@moutte.net>
20913 sys/xvimage/xvimagesink.*: Implement PropertyProbe Interface for XVAdaptors so that one can choose the adaptor to use...
20914 Original commit message from CVS:
20915 2007-02-06 Julien MOUTTE <julien@moutte.net>
20916 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
20917 (gst_xvimagesink_get_xv_support),
20918 (gst_xvimagesink_xcontext_clear),
20919 (gst_xvimagesink_interface_supported),
20920 (gst_xvimagesink_probe_get_properties),
20921 (gst_xvimagesink_probe_probe_property),
20922 (gst_xvimagesink_probe_needs_probe),
20923 (gst_xvimagesink_probe_get_values),
20924 (gst_xvimagesink_property_probe_interface_init),
20925 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
20926 (gst_xvimagesink_init), (gst_xvimagesink_class_init),
20927 (gst_xvimagesink_get_type):
20928 * sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
20929 for XVAdaptors so that one can choose the adaptor to use with
20930 gstreamer-properties.
20932 2007-02-06 14:00:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20934 gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
20935 Original commit message from CVS:
20936 * gst/audioconvert/gstaudioconvert.c:
20937 Also mention that a conversion from double to float is suboptimal still.
20939 2007-02-06 09:42:05 +0000 Tim-Philipp Müller <tim@centricular.net>
20941 gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
20942 Original commit message from CVS:
20943 * gst-libs/gst/audio/gstaudiofilter.c:
20944 (gst_audio_filter_class_init), (gst_audio_filter_change_state):
20945 Clear our formats structure and free the caps contained in it when
20948 2007-02-05 18:39:51 +0000 Andy Wingo <wingo@pobox.com>
20951 * gst-libs/gst/audio/gstbaseaudiosink.c:
20952 gst-libs/gst/audio/gstbaseaudiosink.c
20953 Original commit message from CVS:
20954 2007-02-05 Andy Wingo <wingo@pobox.com>
20955 * gst-libs/gst/audio/gstbaseaudiosink.c
20956 (gst_base_audio_sink_callback): Update basesink->offset so that we
20957 pull monotonically increasing offsets instead of, um, seeking back
20958 to 0 each time. Fixes alsasrc ! alsasink!
20960 2007-02-05 11:44:52 +0000 Tim-Philipp Müller <tim@centricular.net>
20962 gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
20963 Original commit message from CVS:
20964 * gst/videoscale/gstvideoscale.c:
20965 A width and height of 1 makes us crash, so increase minimum size to
20966 2x2 pixels until someone feels like fixing this (#404512).
20968 2007-02-04 16:23:37 +0000 Tim-Philipp Müller <tim@centricular.net>
20970 tests/check/pipelines/oggmux.c: Add small test to make sure request pads are cleaned up properly even if oggmux never...
20971 Original commit message from CVS:
20972 * tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
20973 Add small test to make sure request pads are cleaned up properly
20974 even if oggmux never changes state out of NULL.
20976 2007-02-04 14:11:51 +0000 Tim-Philipp Müller <tim@centricular.net>
20978 tests/check/libs/utils.c: Fix unit test. Turns out things work much better when you
20979 Original commit message from CVS:
20980 * tests/check/libs/utils.c: (GST_START_TEST):
20981 Fix unit test. Turns out things work much better when you
20982 NULL-terminate string arrays. Should make p5 build bot happy again.
20984 2007-02-03 23:28:45 +0000 Tim-Philipp Müller <tim@centricular.net>
20986 gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
20987 Original commit message from CVS:
20988 * gst-libs/gst/audio/Makefile.am:
20989 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
20990 (gst_audio_filter_template_base_init),
20991 (gst_audio_filter_template_class_init),
20992 (gst_audio_filter_template_init),
20993 (gst_audio_filter_template_set_property),
20994 (gst_audio_filter_template_get_property),
20995 (gst_audio_filter_template_setup),
20996 (gst_audio_filter_template_filter),
20997 (gst_audio_filter_template_filter_inplace), (plugin_init):
20998 Oops, forgot to commit fixed-up example.
21000 2007-02-03 20:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
21002 Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
21003 Original commit message from CVS:
21004 * docs/libs/gst-plugins-base-libs-sections.txt:
21005 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
21006 (gst_audio_filter_class_init), (gst_audio_filter_init),
21007 (gst_audio_filter_set_caps),
21008 (gst_audio_filter_class_add_pad_templates):
21009 * gst-libs/gst/audio/gstaudiofilter.h:
21010 Port GstAudioFilter to 0.10. This change technically breaks
21011 API and ABI (and thus also every library developer's heart),
21012 but seems justifiable on the grounds that the base class was
21013 completely unusable before (ie. would crash immediately when
21014 actually used). Fixes #403963 (and eventually also #403572).
21015 Also document all of this a bit.
21017 2007-02-03 14:26:54 +0000 Tim-Philipp Müller <tim@centricular.net>
21019 Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
21020 Original commit message from CVS:
21021 * gst-libs/gst/utils/install-plugins.c:
21022 (gst_install_plugins_spawn_child):
21023 * tests/check/libs/utils.c:
21024 (test_base_utils_install_plugins_do_callout):
21025 Lowering log level to see why things fail on the p5 build bot;
21026 fix some typos in unit test messages.
21028 2007-02-03 13:59:27 +0000 Tim-Philipp Müller <tim@centricular.net>
21030 tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
21031 Original commit message from CVS:
21032 * tests/check/libs/utils.c:
21033 (test_base_utils_install_plugins_do_callout):
21034 Don't hard-code temp directory for test helper; use GLib functions
21035 to write out file and do error checking etc.
21037 2007-02-02 20:42:08 +0000 Tim-Philipp Müller <tim@centricular.net>
21039 gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
21040 Original commit message from CVS:
21041 * gst-libs/gst/utils/Makefile.am:
21042 * gst-libs/gst/utils/base-utils.h:
21043 * gst-libs/gst/utils/install-plugins.c:
21044 (gst_install_plugins_context_set_xid),
21045 (gst_install_plugins_context_new),
21046 (gst_install_plugins_context_free),
21047 (gst_install_plugins_get_helper),
21048 (gst_install_plugins_spawn_child),
21049 (gst_install_plugins_return_from_status),
21050 (gst_install_plugins_installer_exited),
21051 (gst_install_plugins_async), (gst_install_plugins_sync),
21052 (gst_install_plugins_return_get_name),
21053 (gst_install_plugins_installation_in_progress):
21054 * gst-libs/gst/utils/install-plugins.h:
21055 API: add API for applications to initiate installation of missing
21056 plugins, ie. gst_install_plugins_async() primarily.
21057 Based on libgimme-codec by Ryan Lortie.
21059 Add --with-install-plugins-helper configure option so distros can specify
21060 the path of the helper script or program to call when plugin installation
21061 is requested (distros: please do any argument munging in this helper
21062 script instead of patching GStreamer to pass arguments differently
21063 to another program directly).
21064 * docs/libs/gst-plugins-base-libs-docs.sgml:
21065 * docs/libs/gst-plugins-base-libs-sections.txt:
21066 Build and document new API.
21067 * tests/check/libs/utils.c: (result_cb),
21068 (test_base_utils_install_plugins_do_callout), (GST_START_TEST),
21069 (libgstbaseutils_suite):
21070 Some simple checks for the new API.
21072 2007-02-02 14:44:29 +0000 Tim-Philipp Müller <tim@centricular.net>
21074 tests/check/elements/audioconvert.c: Add small test for 32bit float <=> 64bit float conversion (works only one way so...
21075 Original commit message from CVS:
21076 * tests/check/elements/audioconvert.c: (test_float_conversion):
21077 Add small test for 32bit float <=> 64bit float conversion (works
21078 only one way so far, 32=>64 produces structured noise).
21080 2007-02-02 11:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
21082 gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
21083 Original commit message from CVS:
21084 * gst/audioconvert/gstaudioconvert.c:
21085 (set_structure_widths_32_and_64), (make_lossless_changes):
21086 We don't support floats with a width of 40, 48 or 56 bits.
21088 2007-02-02 09:48:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21090 gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
21091 Original commit message from CVS:
21092 * gst/audioconvert/audioconvert.c: (float), (double),
21093 (audio_convert_get_func_index):
21094 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
21095 (make_lossless_changes):
21096 Support for 64-bit float audio in audioconvert (#339837)
21098 2007-02-01 18:50:08 +0000 Holger Wansing <linux@wansing-online.de>
21100 po/: Add German translation (#352069).
21101 Original commit message from CVS:
21102 Patch by: Holger Wansing <linux wansing-online de>
21105 Add German translation (#352069).
21107 2007-02-01 17:52:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21109 ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
21110 Original commit message from CVS:
21111 reviewed by: Wim Taymans <wim@fluendo.com>
21112 * ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
21113 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
21114 Use newly added GstCollectPads API to free the allocated resources in
21115 the GstOggPad structures (#402393).
21117 2007-01-31 15:58:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21119 gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
21120 Original commit message from CVS:
21121 * gst/playback/gstplaybin.c: (gen_vis_element):
21122 Add audioresample+audioconvert in front of the visualisation
21123 element, so that elements like libvisual 0.4 that don't support all
21124 samplerates can work.
21127 2007-01-30 19:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
21129 gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
21130 Original commit message from CVS:
21131 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
21132 (gst_play_base_bin_get_streaminfo_value_array):
21133 Take some locks and make a copy of the streaminfo value array we
21134 maintain while holding the lock, so that the application can
21135 retrieve the stream-info as a value array in a thread-safe way.
21137 2007-01-30 11:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
21139 gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
21140 Original commit message from CVS:
21141 * gst/audioconvert/gstaudioconvert.c:
21142 Don't fail on 0 sized buffers. Fixes #396835.
21144 2007-01-29 21:13:07 +0000 David Schleef <ds@schleef.org>
21146 gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
21147 Original commit message from CVS:
21148 * gst/typefind/gsttypefindfunctions.c:
21149 Detect BBCD as video/x-dirac, so we can play raw dirac
21152 2007-01-29 18:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
21154 ext/theora/theoraenc.c: Check return value of theora_encode_header(), or we might try to allocate a random number of ...
21155 Original commit message from CVS:
21156 * ext/theora/theoraenc.c: (theora_enc_chain):
21157 Check return value of theora_encode_header(), or we might try to
21158 allocate a random number of bytes. theora_encode_header() can fail
21159 if libtheora has been compiled with encoding support disabled.
21162 2007-01-29 10:53:06 +0000 Wim Taymans <wim.taymans@gmail.com>
21164 tests/check/gst/.cvsignore: Do as buildbot says.
21165 Original commit message from CVS:
21166 * tests/check/gst/.cvsignore:
21167 Do as buildbot says.
21169 2007-01-29 10:25:11 +0000 Wim Taymans <wim.taymans@gmail.com>
21171 ext/libvisual/visual.c: Fix strides in libvisual. Gst uses X strides.
21172 Original commit message from CVS:
21173 * ext/libvisual/visual.c: (gst_visual_src_setcaps):
21174 Fix strides in libvisual. Gst uses X strides.
21175 Inspired by: <ed at catmur dot co dot uk> and
21176 <tim at centricular dot net>
21179 2007-01-27 13:32:24 +0000 Wim Taymans <wim.taymans@gmail.com>
21181 ext/ogg/gstoggdemux.*: Properly propagate streaming errors when we are scanning the file for chains so that we don't ...
21182 Original commit message from CVS:
21183 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
21184 (gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
21185 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
21186 (gst_ogg_demux_perform_seek),
21187 (gst_ogg_demux_bisect_forward_serialno),
21188 (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
21189 (gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
21190 (gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
21191 (gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
21192 * ext/ogg/gstoggdemux.h:
21193 Properly propagate streaming errors when we are scanning the file for
21194 chains so that we don't crash when shut down. Might fix some crashers
21195 when quickly switching oggs in RB such as #332503 and #378436.
21197 2007-01-26 12:44:46 +0000 Tim-Philipp Müller <tim@centricular.net>
21199 ext/gnomevfs/gstgnomevfssrc.c: Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND error code as well.
21200 Original commit message from CVS:
21201 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
21202 Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
21203 error code as well.
21205 2007-01-25 16:02:41 +0000 Wim Taymans <wim.taymans@gmail.com>
21207 gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
21208 Original commit message from CVS:
21209 * gst/playback/gstplaybasebin.c: (remove_source):
21210 Don't try to disconnect a signal from a finalized object.
21212 2007-01-25 14:29:21 +0000 Tim-Philipp Müller <tim@centricular.net>
21214 gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
21215 Original commit message from CVS:
21216 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
21217 Cast lock macro parameters to make sure we're actually accessing the
21218 lock member at the right class level. Free list itself in _dispose()
21219 as well and NULL it in case dispose gets called multiple times.
21221 2007-01-25 14:02:37 +0000 Edward Hervey <bilboed@bilboed.com>
21223 gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
21224 Original commit message from CVS:
21225 * gst/playback/gstdecodebin2.c:
21226 (gst_decode_bin_dispose),(gst_decode_bin_finalize):
21227 Free GstDecodeGroups no longer used.
21228 (gst_decode_group_expose):
21229 Don't unlock too many times !
21230 (deactivate_free_recursive):
21231 Free iterator once we're done with it.
21232 Fix for recursively deactivating elements (stop at ghostpads).
21234 2007-01-25 12:24:18 +0000 Tim-Philipp Müller <tim@centricular.net>
21236 gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
21237 Original commit message from CVS:
21238 * gst/playback/gstplaybin.c: (handoff):
21239 Fix up caps on the frame buffer before we save it and potentially
21240 make it accessible to other threads via g_object_get; also use
21241 gst_buffer_replace() instead of gst_mini_object_replace().
21243 2007-01-25 12:06:59 +0000 Tim-Philipp Müller <tim@centricular.net>
21245 gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
21246 Original commit message from CVS:
21247 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
21248 Make getting the current frame thread-safe.
21250 2007-01-25 11:48:10 +0000 Edward Hervey <bilboed@bilboed.com>
21252 gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
21253 Original commit message from CVS:
21254 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
21255 (gst_decode_group_new), (gst_decode_group_free):
21256 Set queues to bigger sizes to cope with HD contents.
21257 Fix some mutex freeing and add comment about MT safe methods.
21259 2007-01-24 12:51:20 +0000 Tim-Philipp Müller <tim@centricular.net>
21261 ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
21262 Original commit message from CVS:
21263 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
21264 (gst_text_overlay_text_event):
21265 Don't unnecessarily ref (and then leak) upstream events if the text
21266 pad is not linked. Fixes #399948.
21267 * tests/check/gst-plugins-base.supp:
21268 Add suppression for pango on edgy/x86 for textoverlay test.
21270 2007-01-24 12:10:56 +0000 Wim Taymans <wim.taymans@gmail.com>
21272 gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
21273 Original commit message from CVS:
21274 * gst-libs/gst/rtp/gstrtpbuffer.h:
21275 Add some more fixed payloads.
21277 2007-01-23 18:39:45 +0000 Tim-Philipp Müller <tim@centricular.net>
21279 ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
21280 Original commit message from CVS:
21281 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
21282 Error out properly if we get an error from libogg while reading the
21283 BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).
21285 2007-01-23 17:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
21287 gst/playback/gstdecodebin2.c: Don't leak mutex.
21288 Original commit message from CVS:
21289 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
21291 * tests/check/elements/playbin.c:
21292 (test_sink_usage_video_only_stream),
21293 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
21294 (test_suburi_error_wrongproto), (test_missing_urisource_handler),
21295 (test_missing_suburisource_handler),
21296 (test_missing_primary_decoder), (playbin_suite):
21297 Run all tests once with decodebin and once with decodebin2.
21298 One test does not pass yet with decodebin2.
21300 2007-01-23 14:30:28 +0000 Edward Hervey <bilboed@bilboed.com>
21302 ext/ogg/gstoggmux.c: Fix the cases where oggmux doesn't properly figure out that all sinkpads have gone EOS, and ther...
21303 Original commit message from CVS:
21304 * ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
21305 Fix the cases where oggmux doesn't properly figure out that all
21306 sinkpads have gone EOS, and therefore doesn't push out the remaining
21307 buffers and the final EOS event.
21310 2007-01-23 13:19:19 +0000 Julien Moutte <julien@moutte.net>
21312 sys/: Don't lock on navigation event push, just on keysym to string.
21313 Original commit message from CVS:
21314 2007-01-23 Julien MOUTTE <julien@moutte.net>
21315 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
21316 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
21317 Don't lock on navigation event push, just on keysym to string.
21318 Fixes #397673 again.
21320 2007-01-22 17:37:38 +0000 Edward Hervey <bilboed@bilboed.com>
21322 gst/playback/gstdecodebin2.c: Cleanups.
21323 Original commit message from CVS:
21324 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
21325 (get_current_group), (group_demuxer_event_probe),
21326 (gst_decode_group_expose), (deactivate_free_recursive),
21327 (gst_decode_group_free):
21329 Don't forget to emit 'no-more-pads' once a group is exposed.
21330 Cleanup elements from a DecodeGroup once we remove it.
21331 Protect call to gst_decode_group_expose() with the decodebin lock.
21333 2007-01-22 13:16:42 +0000 Julien Moutte <julien@moutte.net>
21335 sys/: Looking at Xorg code i can't figure out if that XKeysymToString function is thread sensible or not. Lock it jus...
21336 Original commit message from CVS:
21337 2007-01-22 Julien MOUTTE <julien@moutte.net>
21338 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
21339 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
21340 Looking at Xorg code i can't figure out if that XKeysymToString
21341 function is thread sensible or not. Lock it just in case as
21342 recommended by Radek Doulik <rodo at ximian dot com>.
21344 2007-01-22 13:10:13 +0000 Julien Moutte <julien@moutte.net>
21346 sys/: Lock that X Call as well. Fixes #397673.
21347 Original commit message from CVS:
21348 2007-01-22 Julien MOUTTE <julien@moutte.net>
21349 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
21350 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
21351 Lock that X Call as well. Fixes #397673.
21353 2007-01-22 12:03:27 +0000 Tim-Philipp Müller <tim@centricular.net>
21355 gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
21356 Original commit message from CVS:
21357 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
21358 Don't go into an endless loop if the file starts with 00 00 01 2X,
21359 like quicktime redirect files might. Fixes #396042.
21360 * tests/check/Makefile.am:
21361 * tests/check/gst/.cvsignore:
21362 * tests/check/gst/typefindfunctions.c: (GST_START_TEST),
21363 (typefindfunctions_suite):
21364 Add unit test for the above.
21366 2007-01-22 10:27:26 +0000 Tim-Philipp Müller <tim@centricular.net>
21368 gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
21369 Original commit message from CVS:
21370 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
21371 On second thought, use "depth" field rather than "bpp" field.
21373 2007-01-22 09:23:01 +0000 Tim-Philipp Müller <tim@centricular.net>
21375 gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
21376 Original commit message from CVS:
21377 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
21378 Camtasia caps apparently need a bpp field (#398875).
21380 2007-01-19 19:09:05 +0000 Tim-Philipp Müller <tim@centricular.net>
21382 gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
21383 Original commit message from CVS:
21384 * gst/playback/gstplaybasebin.c: (setup_subtitle),
21385 (gen_source_element), (gst_play_base_bin_change_state):
21386 Attempt at a better error message in case we don't have the required
21387 URI handler installed; post missing-plugin message also when we're
21388 missing an URI handler for the subtitle URI; clean up properly also
21389 when an error occurs and we never made it to PAUSED state.
21390 * tests/check/elements/playbin.c: (GST_START_TEST),
21392 Check that we're also getting a missing-plugin messsage for a
21393 missing subtitle URI handler (and clean up properly).
21395 2007-01-19 18:47:30 +0000 Tim-Philipp Müller <tim@centricular.net>
21397 gst/playback/gstplaybasebin.c: Plug a few reference leaks.
21398 Original commit message from CVS:
21399 * gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
21400 Plug a few reference leaks.
21402 2007-01-19 12:23:06 +0000 Tim-Philipp Müller <tim@centricular.net>
21404 gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
21405 Original commit message from CVS:
21406 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
21407 Lower probability a bit if the marker isn't right at the start,
21408 to decrease the chance of false positives.
21410 2007-01-19 11:31:50 +0000 Tim-Philipp Müller <tim@centricular.net>
21412 gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
21413 Original commit message from CVS:
21414 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
21415 Small mpeg2 system stream typefinding improvement: make typefinder
21416 probe a bit into the stream instead of just looking for a marker
21417 at the beginning. Fixes #397810.
21419 2007-01-18 16:23:35 +0000 Tim-Philipp Müller <tim@centricular.net>
21421 gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
21422 Original commit message from CVS:
21423 * gst/audioconvert/gstchannelmix.c:
21424 Remove compatibility cruft for prehistoric GLib versions.
21426 2007-01-17 16:11:14 +0000 Tim-Philipp Müller <tim@centricular.net>
21428 gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
21429 Original commit message from CVS:
21430 * gst/playback/Makefile.am:
21431 * gst/playback/gstdecodebin.c: (close_pad_link):
21432 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
21433 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
21434 (gst_play_base_bin_handle_message_func), (unknown_type):
21435 Let decodebin be the element to post missing-plugin messages for
21436 missing decoders (rather than playbin); make playbin implement
21437 GstBin::handle_message so we can suppress missing-plugin messages
21438 for types we're not handling on purpose (don't want to bring up an
21439 installer in those cases).
21441 2007-01-16 19:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21443 gst/: Fix potentially unaligned access (#397207).
21444 Original commit message from CVS:
21445 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
21446 * gst-libs/gst/tag/gstvorbistag.c:
21447 (gst_tag_list_to_vorbiscomment_buffer):
21448 * gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
21449 Fix potentially unaligned access (#397207).
21451 2007-01-16 12:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21453 tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
21454 Original commit message from CVS:
21455 * tests/examples/seek/seek.c: (set_scale), (update_scale),
21456 (do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
21457 (rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
21459 Allow to toggle looping while it plays. Fix callback prototype. Clean
21460 up code a bit more. Add copyright header.
21462 2007-01-16 11:41:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21464 sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
21465 Original commit message from CVS:
21466 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
21467 Red and blue mask was swapped (spotted by Dan Williams).
21469 2007-01-15 13:58:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21471 gst-libs/gst/tag/: Use new beats-per-minute tag from core.
21472 Original commit message from CVS:
21473 * gst-libs/gst/tag/gstid3tag.c:
21474 * gst-libs/gst/tag/gstvorbistag.c:
21475 Use new beats-per-minute tag from core.
21477 2007-01-15 11:30:53 +0000 Tim-Philipp Müller <tim@centricular.net>
21479 po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
21480 Original commit message from CVS:
21482 Add new files with translatable strings, so they actually make it
21483 into the template file one day.
21485 2007-01-12 21:19:35 +0000 Andy Wingo <wingo@pobox.com>
21488 * gst-libs/gst/audio/gstbaseaudiosink.c:
21489 * gst-libs/gst/audio/gstbaseaudiosrc.c:
21490 gst-libs/gst/audio/gstbaseaudiosink.c
21491 Original commit message from CVS:
21492 2007-01-12 Andy Wingo <wingo@pobox.com>
21493 * gst-libs/gst/audio/gstbaseaudiosink.c
21494 (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
21495 (gst_base_audio_sink_activate_pull): Remove the handwavey nego
21496 stuff, as the base class handles this now. Actually tell the ring
21498 (gst_base_audio_sink_callback): Cast the ring buffer correctly.
21499 How did this work before? Maybe I'm not as awesome a programmer as
21501 * gst-libs/gst/audio/gstbaseaudiosrc.c
21502 (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
21505 2007-01-12 18:08:23 +0000 Tim-Philipp Müller <tim@centricular.net>
21507 gst-libs/gst/utils/missing-plugins.c: Remove more fields so that the application can better blacklist formats that ha...
21508 Original commit message from CVS:
21509 * gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
21510 Remove more fields so that the application can better blacklist
21511 formats that have been tried before.
21513 2007-01-12 17:43:40 +0000 Christian Schaller <uraeus@gnome.org>
21515 * gst-plugins-base.spec.in:
21517 Original commit message from CVS:
21520 2007-01-12 12:47:29 +0000 Tim-Philipp Müller <tim@centricular.net>
21522 gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
21523 Original commit message from CVS:
21524 * gst-libs/gst/audio/mixerutils.h:
21525 Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
21526 used when compiling with c++ compilers as well.
21528 2007-01-12 09:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
21530 gst/typefind/gsttypefindfunctions.c: Fix comment.
21531 Original commit message from CVS:
21532 * gst/typefind/gsttypefindfunctions.c:
21535 2007-01-11 13:12:17 +0000 Tim-Philipp Müller <tim@centricular.net>
21537 gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
21538 Original commit message from CVS:
21539 * gst/playback/gstplaybin.c: (post_missing_element_message),
21540 (gen_video_element), (gen_text_element), (gen_audio_element),
21542 Post missing-plugin messages also when we error out because
21543 converters, textoverlay or auto*sinks are missing (#161922).
21545 2007-01-10 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
21547 gst/playback/: Fix the case where we try to ref a NULL element when we delay a link because of unfixed caps.
21548 Original commit message from CVS:
21549 * gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
21550 (is_demuxer_element), (new_caps):
21551 * gst/playback/gstplaybasebin.c: (source_new_pad):
21552 Fix the case where we try to ref a NULL element when we delay a link
21553 because of unfixed caps.
21554 Set the state of autoplugged decodebins to PAUSED.
21555 RTSP now works in playbin, we can remove it from the blacklist.
21557 2007-01-09 14:33:24 +0000 Tim-Philipp Müller <tim@centricular.net>
21559 gst/playback/: Post missing-plugin messages on the bus for missing sources and missing decoders/demuxers/depayloaders...
21560 Original commit message from CVS:
21561 * gst/playback/Makefile.am:
21562 * gst/playback/gstplaybasebin.c: (string_arr_has_str),
21563 (unknown_type), (setup_subtitle), (gen_source_element):
21564 * gst/playback/gstplaybin.c: (plugin_init):
21565 Post missing-plugin messages on the bus for missing sources and
21566 missing decoders/demuxers/depayloaders; fix error code used when
21567 we're missing an URI handler source; for media types that we are not
21568 handling on purpose at the moment, don't print "don't know how to
21569 handle xyz" messages to the terminal or post missing-plugin
21570 messages on the bus.
21571 * tests/check/elements/playbin.c: (create_playbin),
21572 (GST_START_TEST), (gst_codec_src_uri_get_type),
21573 (gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
21574 (gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
21575 (gst_codec_src_init_type), (gst_codec_src_base_init),
21576 (gst_codec_src_create), (gst_codec_src_class_init),
21577 (gst_codec_src_init), (plugin_init), (playbin_suite):
21578 Add some tests for the missing-plugin stuff.
21580 2007-01-09 14:20:08 +0000 Tim-Philipp Müller <tim@centricular.net>
21582 API: add new libgstbaseutils library with functions
21583 Original commit message from CVS:
21585 * gst-libs/gst/Makefile.am:
21586 * gst-libs/gst/utils/Makefile.am:
21587 * gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
21588 * gst-libs/gst/utils/base-utils.h:
21589 * gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
21590 (find_format_info), (caps_are_rtp_caps),
21591 (gst_base_utils_get_source_description),
21592 (gst_base_utils_get_sink_description),
21593 (gst_base_utils_get_decoder_description),
21594 (gst_base_utils_get_encoder_description),
21595 (gst_base_utils_get_element_description),
21596 (gst_base_utils_add_codec_description_to_tag_list),
21597 (gst_base_utils_get_codec_description), (gst_base_utils_list_all):
21598 * gst-libs/gst/utils/descriptions.h:
21599 * gst-libs/gst/utils/missing-plugins.c:
21600 (missing_structure_get_type), (copy_and_clean_caps),
21601 (gst_missing_uri_source_message_new),
21602 (gst_missing_uri_sink_message_new),
21603 (gst_missing_element_message_new),
21604 (gst_missing_decoder_message_new),
21605 (gst_missing_encoder_message_new),
21606 (missing_structure_get_string_detail),
21607 (missing_structure_get_caps_detail),
21608 (gst_missing_plugin_message_get_installer_detail),
21609 (gst_missing_plugin_message_get_description),
21610 (gst_is_missing_plugin_message):
21611 * gst-libs/gst/utils/missing-plugins.h:
21612 API: add new libgstbaseutils library with functions
21613 - to create and parse missing-plugins messages
21614 - that provide (translated) descriptions for caps/decoders/sources/etc.
21616 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
21617 * pkgconfig/gstreamer-plugins-base.pc.in:
21619 * docs/libs/gst-plugins-base-libs-docs.sgml:
21620 * docs/libs/gst-plugins-base-libs-sections.txt:
21621 Generate docs for new lib and API.
21622 * tests/check/Makefile.am:
21623 * tests/check/libs/.cvsignore:
21624 * tests/check/libs/utils.c: (missing_msg_check_getters),
21625 (GST_START_TEST), (libgstbaseutils_suite):
21626 Add some basic unit tests.
21628 2007-01-09 13:35:08 +0000 Tim-Philipp Müller <tim@centricular.net>
21630 ext/ogg/Makefile.am: Dist gstoggdemux.h to fix 'make distcheck'.
21631 Original commit message from CVS:
21632 * ext/ogg/Makefile.am:
21633 Dist gstoggdemux.h to fix 'make distcheck'.
21634 * sys/v4l/Makefile.am:
21635 Fix 'make distcheck' even more.
21637 2007-01-09 12:30:46 +0000 Wim Taymans <wim.taymans@gmail.com>
21640 Original commit message from CVS:
21641 * docs/plugins/Makefile.am:
21642 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
21643 * docs/plugins/gst-plugins-base-plugins-sections.txt:
21644 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
21645 (gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
21646 (gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
21647 (gst_ogg_demux_perform_seek):
21648 * ext/ogg/gstoggdemux.h:
21650 Add some more comments.
21653 2007-01-09 11:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
21655 Small documentation updates/fixes
21656 Original commit message from CVS:
21657 * ext/theora/theoradec.c:
21658 * ext/vorbis/vorbisdec.c:
21659 * gst-libs/gst/audio/gstringbuffer.c:
21660 (gst_ring_buffer_commit_full):
21661 * gst-libs/gst/audio/gstringbuffer.h:
21662 * gst-libs/gst/rtp/gstrtpbuffer.c:
21663 * gst-libs/gst/tag/gstvorbistag.c:
21664 Small documentation updates/fixes
21666 2007-01-09 10:37:01 +0000 Tim-Philipp Müller <tim@centricular.net>
21668 configure.ac: Require core CVS HEAD for Andy's basesrc/sink API additions.
21669 Original commit message from CVS:
21671 Require core CVS HEAD for Andy's basesrc/sink API additions.
21673 2007-01-08 14:01:23 +0000 Günter Thelen <daedalus.inc@gmx.net>
21675 gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
21676 Original commit message from CVS:
21677 Patch by: Günter Thelen <daedalus dot inc at gmx net>
21678 * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
21680 Add typefinder for flac-in-ogg in conformance with the ogg-mapping
21681 on flac.sf.net (there appear to be other versions of the first
21682 ogg page in the wild) (#391365).
21684 2007-01-08 13:32:32 +0000 Tim-Philipp Müller <tim@centricular.net>
21686 configure.ac: Check if localtime_r() is available.
21687 Original commit message from CVS:
21689 Check if localtime_r() is available.
21690 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
21691 If localtime_r() is not available, fall back to localtime(). Should
21692 fix build on MingW (#393310).
21694 2007-01-08 12:30:03 +0000 Tim-Philipp Müller <tim@centricular.net>
21696 gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
21697 Original commit message from CVS:
21698 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
21699 * gst/subparse/gstsubparse.h:
21700 Remove spurious 1000 subtrahend when calculating the timestamp from
21701 the frame number and the frame rate . Also, use the frames/second
21702 value specified in the first line of the file, if one is specified
21703 there. Should fix #357503.
21704 * tests/check/elements/subparse.c: (do_test),
21705 (test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
21707 Add some basic unit tests for the microdvd subtitle format.
21709 2007-01-07 21:53:38 +0000 Young-Ho Cha <ganadist@chollian.net>
21711 sys/xvimage/xvimagesink.c: Fixes : #390076.
21712 Original commit message from CVS:
21713 2007-01-07 Julien MOUTTE <julien@moutte.net>
21714 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
21715 (gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
21716 (gst_xvimagesink_xvimage_put),
21717 (gst_lookup_xv_port_from_adaptor),
21718 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
21719 (gst_xvimagesink_set_xwindow_id),
21720 (gst_xvimagesink_set_event_handling),
21721 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
21722 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
21723 Patch by : Young-Ho Cha <ganadist at chollian dot net>
21725 Add an adaptor property to select a specific XV adaptor.
21726 * sys/xvimage/xvimagesink.h:
21728 2007-01-07 18:50:13 +0000 Julien Moutte <julien@moutte.net>
21730 sys/: Use flow_lock much more to protect every access to xwindow.
21731 Original commit message from CVS:
21732 2007-01-07 Julien MOUTTE <julien@moutte.net>
21733 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
21734 (gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
21735 (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
21736 (gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
21737 (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
21738 (gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
21739 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
21740 (gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
21741 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
21742 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
21743 (gst_xvimagesink_change_state),
21744 (gst_xvimagesink_set_xwindow_id),
21745 (gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
21746 Use flow_lock much more to protect every access to xwindow.
21747 Try to catch erros while creating images in case some drivers
21749 just generating an XError when the requested image is too big.
21750 Should fix : #354698, #384008, #384060.
21751 * tests/icles/stress-xoverlay.c: (cycle_window),
21753 Implement some stress testing of setting window xid.
21755 2007-01-07 10:33:55 +0000 Sébastien Moutte <sebastien@moutte.net>
21757 win32/common/libgsaudio.def: Add new exported function.
21758 Original commit message from CVS:
21759 * win32/common/libgsaudio.def:
21760 Add new exported function.
21761 * win32/common/libgstogg.dsp:
21762 Add gstoggaviparse.c to the build.
21763 * win32/common/libgstvideoscale.dsp:
21764 Add vs_4tap.c to the build.
21765 * win32/common/libgstvorbis.dsp:
21766 Add vorbistag.c to the build.
21768 2007-01-06 17:28:40 +0000 Andy Wingo <wingo@pobox.com>
21771 * gst-libs/gst/audio/gstbaseaudiosink.c:
21772 gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
21773 Original commit message from CVS:
21774 2007-01-06 Andy Wingo <wingo@pobox.com>
21775 * gst-libs/gst/audio/gstbaseaudiosink.c
21776 (gst_base_audio_sink_class_init)
21777 (gst_base_audio_sink_init):
21778 (gst_base_audio_sink_activate_pull): Add an activate_pull function
21779 to baseaudiosink, and tell basesink that we can work in pull mode.
21780 This way the ring buffer thread drives the pipeline directly, if
21781 pull mode is possible. There is some lingering nastiness regarding
21783 (gst_base_audio_sink_callback): Implement the callback to pull
21784 data. This interface is a bit light, though -- it should get a
21785 GstFlowReturn return value at least.
21787 2007-01-05 19:43:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21789 Printf format and missing argument fixes.
21790 Original commit message from CVS:
21791 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
21792 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
21793 * gst/playback/gstdecodebin2.c:
21794 (gst_decode_group_check_if_blocked):
21795 Printf format and missing argument fixes.
21797 2007-01-05 18:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21799 ext/ogg/gstogmparse.c: Activate pads before adding them to the element.
21800 Original commit message from CVS:
21801 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
21802 (gst_ogm_parse_change_state):
21803 Activate pads before adding them to the element.
21805 2007-01-05 16:02:50 +0000 Tim-Philipp Müller <tim@centricular.net>
21807 tests/examples/seek/: Call g_thread_init() first thing in main() (see #391278).
21808 Original commit message from CVS:
21809 * tests/examples/seek/scrubby.c: (main):
21810 * tests/examples/seek/seek.c: (main):
21811 Call g_thread_init() first thing in main() (see #391278).
21813 2007-01-05 12:19:34 +0000 Tim-Philipp Müller <tim@centricular.net>
21815 tests/check/: Add test for GstNetBuffer + gst_buffer_copy(). Disabled for the time being, since it's broken, see #393...
21816 Original commit message from CVS:
21817 * tests/check/Makefile.am:
21818 * tests/check/libs/.cvsignore:
21819 * tests/check/libs/netbuffer.c: (GST_START_TEST),
21821 Add test for GstNetBuffer + gst_buffer_copy(). Disabled
21822 for the time being, since it's broken, see #393099.
21824 2007-01-05 12:13:24 +0000 Tim-Philipp Müller <tim@centricular.net>
21826 tests/check/Makefile.am: Update to use GST_PLUGINS_BASE_CFLAGS as well.
21827 Original commit message from CVS:
21828 * tests/check/Makefile.am:
21829 Update to use GST_PLUGINS_BASE_CFLAGS as well.
21831 2007-01-04 12:49:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21833 configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
21834 Original commit message from CVS:
21836 split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
21837 so that GST_BASE_CFLAGS can go inbetween them, making sure
21838 we use uninstalled gst-libs headers
21839 * docs/libs/Makefile.am:
21840 * ext/alsa/Makefile.am:
21841 * ext/cdparanoia/Makefile.am:
21842 * ext/gnomevfs/Makefile.am:
21843 * ext/libvisual/Makefile.am:
21844 * ext/ogg/Makefile.am:
21845 * ext/theora/Makefile.am:
21846 * ext/vorbis/Makefile.am:
21847 * gst-libs/gst/audio/Makefile.am:
21848 * gst-libs/gst/cdda/Makefile.am:
21849 * gst-libs/gst/interfaces/Makefile.am:
21850 * gst-libs/gst/riff/Makefile.am:
21851 * gst-libs/gst/rtp/Makefile.am:
21852 * gst-libs/gst/tag/Makefile.am:
21853 * gst/adder/Makefile.am:
21854 * gst/audioconvert/Makefile.am:
21855 * gst/audiorate/Makefile.am:
21856 * gst/audioresample/Makefile.am:
21857 * gst/playback/Makefile.am:
21858 * gst/tcp/Makefile.am:
21859 * gst/videoscale/Makefile.am:
21860 * gst/volume/Makefile.am:
21861 * sys/ximage/Makefile.am:
21862 * sys/xvimage/Makefile.am:
21863 * tests/icles/Makefile.am:
21866 2007-01-04 11:30:53 +0000 Julien Moutte <julien@moutte.net>
21868 Add a method to the XOverlay interface to allow disabling of event handling in x[v]imagesink elements. This will let ...
21869 Original commit message from CVS:
21870 2007-01-04 Julien MOUTTE <julien@moutte.net>
21871 * gst-libs/gst/interfaces/xoverlay.c:
21872 (gst_x_overlay_handle_events):
21873 * gst-libs/gst/interfaces/xoverlay.h:
21874 * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
21875 (gst_ximagesink_set_xwindow_id),
21876 (gst_ximagesink_set_event_handling),
21877 (gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
21878 (gst_ximagesink_get_property), (gst_ximagesink_init),
21879 (gst_ximagesink_class_init):
21880 * sys/ximage/ximagesink.h:
21881 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
21882 (gst_xvimagesink_set_xwindow_id),
21883 (gst_xvimagesink_set_event_handling),
21884 (gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
21885 (gst_xvimagesink_get_property), (gst_xvimagesink_init),
21886 (gst_xvimagesink_class_init):
21887 * sys/xvimage/xvimagesink.h:
21888 * tests/icles/stress-xoverlay.c: (toggle_events),
21890 Add a method to the XOverlay interface to allow disabling of
21891 event handling in x[v]imagesink elements. This will let X events
21892 propagate to parent windows which can be usefull in some cases.
21893 Be carefull that the application is then responsible of pushing
21894 navigation events and expose events to the video sink.
21897 2007-01-03 15:45:06 +0000 Tim-Philipp Müller <tim@centricular.net>
21899 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION (fixes #392070).
21900 Original commit message from CVS:
21901 * gst-libs/gst/tag/gstvorbistag.c:
21902 * tests/check/libs/tag.c: (GST_START_TEST):
21903 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
21906 2007-01-01 18:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
21909 Original commit message from CVS:
21911 * docs/Makefile.am:
21912 * docs/design/Makefile.am:
21915 2006-12-27 17:15:35 +0000 Julien Moutte <julien@moutte.net>
21917 docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation typo. Fixes: #390063.
21918 Original commit message from CVS:
21919 2006-12-27 Julien MOUTTE <julien@moutte.net>
21920 * docs/libs/gst-plugins-base-libs-sections.txt: Fix a
21922 typo. Fixes: #390063.
21924 2006-12-27 12:08:13 +0000 Julien Moutte <julien@moutte.net>
21926 sys/: Plug a caps leak.
21927 Original commit message from CVS:
21928 2006-12-27 Julien MOUTTE <julien@moutte.net>
21929 * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
21930 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
21932 * win32/common/config.h: Updated.
21934 2006-12-22 12:10:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21936 tests/check/elements/: Fix the dp tests, but activating the pads for the streamheader tests too and cleaning up condi...
21937 Original commit message from CVS:
21938 * tests/check/elements/gdpdepay.c: (cleanup_gdpdepay),
21939 (setup_gdpdepay_streamheader):
21940 * tests/check/elements/gdppay.c: (cleanup_gdppay),
21941 (setup_gdppay_streamheader):
21942 Fix the dp tests, but activating the pads for the streamheader tests
21943 too and cleaning up conditionaly
21945 2006-12-22 11:09:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21947 gst/ffmpegcolorspace/: Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the other end of the wo...
21948 Original commit message from CVS:
21949 * gst/ffmpegcolorspace/avcodec.h:
21950 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
21951 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
21952 (gst_ffmpegcsp_avpicture_fill):
21953 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
21954 (img_get_alpha_info):
21955 Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
21956 other end of the word. Fixes: #387073.
21957 Add some inconsequential branch hints in a couple of places.
21959 2006-12-21 12:30:11 +0000 Tim-Philipp Müller <tim@centricular.net>
21961 gst/ffmpegcolorspace/gstffmpegcodecmap.c: The "signed" field in raw audio caps is of boolean type, trying to extract ...
21962 Original commit message from CVS:
21963 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
21964 (gst_ffmpeg_caps_to_smpfmt):
21965 The "signed" field in raw audio caps is of boolean type, trying to
21966 extract the value with _get_int() will fail (fix to keep in sync with
21967 the copy in gst-ffmpeg)
21969 2006-12-21 08:12:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
21971 tests/check/elements/: consistent pad (de)activation
21972 Original commit message from CVS:
21973 * tests/check/elements/audioresample.c: (cleanup_audioresample):
21974 * tests/check/elements/audiotestsrc.c: (cleanup_audiotestsrc):
21975 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
21976 (cleanup_gdpdepay):
21977 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay):
21978 * tests/check/elements/subparse.c: (teardown_subparse):
21979 * tests/check/elements/textoverlay.c: (cleanup_textoverlay):
21980 * tests/check/elements/videorate.c: (cleanup_videorate):
21981 * tests/check/elements/videotestsrc.c: (cleanup_videotestsrc):
21982 * tests/check/elements/volume.c: (cleanup_volume):
21983 * tests/check/elements/vorbisdec.c: (setup_vorbisdec),
21984 (cleanup_vorbisdec):
21985 * tests/check/elements/vorbistag.c: (setup_vorbistag),
21986 (cleanup_vorbistag):
21987 consistent pad (de)activation
21989 2006-12-20 10:29:58 +0000 Tim-Philipp Müller <tim@centricular.net>
21991 gst/typefind/gsttypefindfunctions.c: Forgot to register the extensions.
21992 Original commit message from CVS:
21993 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
21994 Forgot to register the extensions.
21996 2006-12-20 09:25:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21998 gst/typefind/gsttypefindfunctions.c: Add typefinder for VIVO files (my christmas present to the 90s).
21999 Original commit message from CVS:
22000 * gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
22002 Add typefinder for VIVO files (my christmas present to the 90s).
22004 2006-12-16 13:59:09 +0000 Tim-Philipp Müller <tim@centricular.net>
22006 gst/playback/gstdecodebin.c: Special-case the text/plain media type: we only want to recognise it as a 'raw' decoded ...
22007 Original commit message from CVS:
22008 * gst/playback/gstdecodebin.c: (type_found):
22009 Special-case the text/plain media type: we only want to recognise it
22010 as a 'raw' decoded media type if it comes from a demuxer or subtitle
22011 parser, but not if the entire stream is of text/plain type. If the
22012 entire stream is text/plain, we should just error out.
22013 This fixes playback of audio files with lyrics in totem. Totem can't
22014 distinguish between text files and subtitle files and passes any
22015 .txt file with the same basename as the main file to playbin as
22016 suburi, and playbin will then throw a 'subtitle found, but no video
22017 stream' error, which isn't entirely helpful. See #380342.
22018 Also, with this change we'll show a slightly more correct error
22019 message in case totem passes a playlist file to us (although a
22020 custom error message wording instead of the default text would
22021 probably not be a bad idea either).
22022 Same problem also needs to be fixed for playbin+decodebin2.
22023 * tests/check/Makefile.am:
22024 * tests/check/elements/decodebin.c: (src_handoff_cb),
22025 (decodebin_new_decoded_pad_cb), (GST_START_TEST),
22027 Add simple unit test for decodebin for the above.
22029 2006-12-16 12:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
22031 gst/playback/: Refuse to change state to READY when we failed to create any of the required elements in our instance ...
22032 Original commit message from CVS:
22033 * gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
22034 * gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
22035 Refuse to change state to READY when we failed to create any of the
22036 required elements in our instance init function.
22038 2006-12-15 10:52:23 +0000 Tim-Philipp Müller <tim@centricular.net>
22040 docs/libs/gst-plugins-base-libs-sections.txt: Small docs fixes/updates.
22041 Original commit message from CVS:
22042 * docs/libs/gst-plugins-base-libs-sections.txt:
22043 Small docs fixes/updates.
22044 * gst-libs/gst/video/gstvideosink.h:
22045 Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
22046 from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
22047 removed from the base sink API between 0.9.6 and 0.9.7).
22048 API: add GST_VIDEO_SINK_CAST and use it for the height/width
22049 accessor macros, so we don't do a runtime GObject type check every
22052 2006-12-15 00:20:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
22055 Original commit message from CVS:
22057 * gst-plugins-base.doap:
22058 * gst-plugins-base.spec.in:
22061 2006-12-09 15:12:38 +0000 Jens Granseuer <jensgr@gmx.net>
22063 Declare variables at the beginning of a block. Fixes #383195.
22064 Original commit message from CVS:
22065 Patch by: Jens Granseuer <jensgr at gmx net>
22066 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
22067 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
22068 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
22069 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
22070 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
22071 Declare variables at the beginning of a block. Fixes #383195.
22073 2006-12-07 02:38:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22075 configure.ac: Bump version nano - back to CVS.
22076 Original commit message from CVS:
22078 Bump version nano - back to CVS.
22080 === release 0.10.11 ===
22082 2006-12-07 02:30:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22084 configure.ac: releasing 0.10.11, "Dumb things"
22085 Original commit message from CVS:
22086 === release 0.10.11 ===
22087 2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
22089 releasing 0.10.11, "Dumb things"
22091 2006-12-05 12:44:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22093 gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
22094 Original commit message from CVS:
22095 * gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
22096 (close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
22097 Handle the case where an element has multiple pads with
22098 unfixed caps as well as still possibly producing more dynamic
22099 pads by storing each case as a distinct entry in the dynamic list.
22100 Fixes #38223 again.
22102 2006-12-04 13:02:37 +0000 Wim Taymans <wim.taymans@gmail.com>
22104 gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
22105 Original commit message from CVS:
22106 * gst/playback/gstdecodebin.c: (close_pad_link):
22107 Fix #382223, add more dynamic caps handling.
22109 2006-12-01 11:35:57 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
22112 Ignore all pot files
22113 Original commit message from CVS:
22114 Ignore all pot files
22116 2006-12-01 10:36:50 +0000 Michael Smith <msmith@xiph.org>
22118 gst/audiorate/gstaudiorate.c: Delete bad debug code.
22119 Original commit message from CVS:
22120 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
22121 Delete bad debug code.
22124 2006-12-01 10:27:54 +0000 Sergey Scobich <sergey.scobich@gmail.com>
22126 Fix compilation on win32 under VS8
22127 Original commit message from CVS:
22128 * gst/videoscale/vs_4tap.c:
22130 * win32/common/config.h:
22131 * win32/vs8/libgstvideoscale.vcproj:
22132 Fix compilation on win32 under VS8
22133 Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
22134 Partially fixes #381175
22136 2006-11-30 23:46:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22153 Original commit message from CVS:
22156 2006-11-30 12:50:42 +0000 Michael Smith <msmith@xiph.org>
22158 tests/check/pipelines/theoraenc.c: It would be very bad if, after a discont buffer, we thought every single following...
22159 Original commit message from CVS:
22160 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
22162 It would be very bad if, after a discont buffer, we thought every
22163 single following buffer was also discont. So, add to the test to
22164 ensure that this isn't the case.
22165 * ext/theora/theoraenc.c: (theora_enc_is_discontinuous):
22166 ... it was the case. So fix it.
22168 2006-11-28 16:43:18 +0000 Wim Taymans <wim.taymans@gmail.com>
22170 gst/playback/gstplaybasebin.c: Improve debug.
22171 Original commit message from CVS:
22172 * gst/playback/gstplaybasebin.c: (check_queue_event):
22174 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
22175 Fix width and height range from 16 - 4096 to 1 - MAXINT, just like the
22176 padtemplate caps. Refixes #357577.
22178 2006-11-28 16:21:27 +0000 Wim Taymans <wim.taymans@gmail.com>
22180 gst/playback/gstplaybasebin.c: Add event probe to see when EOS is in a queue and we can disable the underrun signals....
22181 Original commit message from CVS:
22182 * gst/playback/gstplaybasebin.c: (check_queue_event),
22183 (queue_threshold_reached), (queue_out_of_data),
22184 (gen_preroll_element):
22185 Add event probe to see when EOS is in a queue and we can disable the
22186 underrun signals. Fixes #357577.
22188 2006-11-28 14:40:39 +0000 Edward Hervey <bilboed@bilboed.com>
22190 gst/playback/: New decodebin2 element.
22191 Original commit message from CVS:
22192 * gst/playback/Makefile.am:
22193 * gst/playback/gstdecodebin2.c: (gst_decode_bin_get_type),
22194 (_gst_boolean_accumulator), (gst_decode_bin_class_init),
22195 (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
22196 (gst_decode_bin_init), (gst_decode_bin_dispose),
22197 (gst_decode_bin_finalize), (gst_decode_bin_set_property),
22198 (gst_decode_bin_get_property), (gst_decode_bin_set_caps),
22199 (gst_decode_bin_get_caps), (gst_decode_bin_autoplug_continue),
22200 (gst_decode_bin_autoplug_sort), (analyze_new_pad), (connect_pad),
22201 (connect_element), (expose_pad), (type_found),
22202 (pad_added_group_cb), (pad_removed_group_cb),
22203 (no_more_pads_group_cb), (pad_added_cb), (pad_removed_cb),
22204 (no_more_pads_cb), (find_compatibles), (is_demuxer_element),
22205 (are_raw_caps), (multi_queue_overrun_cb),
22206 (multi_queue_underrun_cb), (gst_decode_group_new),
22207 (get_current_group), (group_demuxer_event_probe),
22208 (gst_decode_group_control_demuxer_pad),
22209 (gst_decode_group_control_source_pad),
22210 (gst_decode_group_check_if_blocked),
22211 (gst_decode_group_check_if_drained), (gst_decode_group_expose),
22212 (gst_decode_group_hide), (gst_decode_group_free),
22213 (gst_decode_group_set_complete), (source_pad_blocked_cb),
22214 (source_pad_event_probe), (gst_decode_pad_new), (add_fakesink),
22215 (remove_fakesink), (find_sink_pad), (gst_decode_bin_change_state),
22217 New decodebin2 element.
22219 * gst/playback/gstplay-marshal.list:
22220 Added marshallers for new signals in decodebin2
22221 * gst/playback/gstplaybasebin.c: (setup_subtitle), (make_decoder):
22222 Use decodebin2 if *and only if* the USE_DECODEBIN2 environment variable
22225 2006-11-28 10:45:40 +0000 Wim Taymans <wim.taymans@gmail.com>
22227 gst/playback/gstplaybasebin.c: Disable rtsp:// uris for the release, it's not good enough yet.
22228 Original commit message from CVS:
22229 * gst/playback/gstplaybasebin.c: (setup_source),
22230 (gst_play_base_bin_change_state):
22231 Disable rtsp:// uris for the release, it's not good enough yet.
22234 2006-11-26 16:39:41 +0000 Wim Taymans <wim.taymans@gmail.com>
22236 ext/theora/theoradec.c: Implement reverse playback.
22237 Original commit message from CVS:
22238 * ext/theora/theoradec.c: (gst_theora_dec_reset),
22239 (theora_dec_push_forward), (theora_dec_push_reverse),
22240 (theora_handle_data_packet), (theora_dec_decode_buffer),
22241 (theora_dec_flush_decode), (theora_dec_chain_reverse),
22242 (theora_dec_chain_forward), (theora_dec_chain):
22243 Implement reverse playback.
22244 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
22245 (vorbis_dec_decode_buffer), (vorbis_dec_flush_decode),
22246 (vorbis_dec_chain_forward):
22247 Clear buffers used for reverse playback in _reset.
22248 No need to set the eos flag, we clip samples using the segment.
22250 2006-11-24 15:40:58 +0000 Wim Taymans <wim.taymans@gmail.com>
22252 ext/ogg/gstoggdemux.c: Some cleanups.
22253 Original commit message from CVS:
22254 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
22255 (gst_ogg_pad_init), (gst_ogg_pad_dispose), (gst_ogg_pad_reset),
22256 (gst_ogg_pad_stream_out), (gst_ogg_pad_submit_page),
22257 (gst_ogg_chain_reset), (gst_ogg_demux_perform_seek):
22259 Handle continued pages in reverse mode.
22261 2006-11-24 15:39:03 +0000 Wim Taymans <wim.taymans@gmail.com>
22263 ext/vorbis/vorbisdec.c: Small cleanups.
22264 Original commit message from CVS:
22265 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
22266 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
22267 (vorbis_dec_flush_decode):
22269 Don't try to add invalid timestamps.
22270 Clipping will unref the buffer.
22272 2006-11-24 08:56:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22274 gst/: remove obsolete _factory_init protos
22275 Original commit message from CVS:
22276 * gst/adder/gstadder.h:
22277 * gst/audiotestsrc/gstaudiotestsrc.h:
22278 remove obsolete _factory_init protos
22280 2006-11-24 08:35:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22282 sys/xvimage/xvimagesink.c: Fix spacing in debug message.
22283 Original commit message from CVS:
22284 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
22285 Fix spacing in debug message.
22287 2006-11-23 11:07:23 +0000 Wim Taymans <wim.taymans@gmail.com>
22289 ext/ogg/gstoggdemux.c: Don't just ignore return values from _pad_push().
22290 Original commit message from CVS:
22291 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
22292 (gst_ogg_demux_chain):
22293 Don't just ignore return values from _pad_push().
22294 Small debug improvements.
22296 2006-11-23 11:02:11 +0000 Michael Smith <msmith@xiph.org>
22298 ext/ogg/gstoggmux.c: If our incoming buffer is marked as DISCONT, then increment the page number (so that the discont...
22299 Original commit message from CVS:
22300 * ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
22301 If our incoming buffer is marked as DISCONT, then increment the page
22302 number (so that the discontinuity is marked in the final ogg
22303 bitstream) and flush the previous page.
22305 2006-11-22 14:34:03 +0000 Michael Smith <msmith@xiph.org>
22307 ext/theora/: Mark discontinuities of > 3/4 of a frame, reinit encoder.
22308 Original commit message from CVS:
22309 * ext/theora/gsttheoraenc.h:
22310 * ext/theora/theoraenc.c: (gst_theora_enc_init),
22311 (theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
22312 (theora_buffer_from_packet), (theora_enc_is_discontinuous),
22313 (theora_enc_chain), (theora_enc_change_state):
22314 Mark discontinuities of > 3/4 of a frame, reinit encoder.
22315 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
22316 (GST_START_TEST), (theoraenc_suite):
22317 Enable discontinuity test, fix it.
22319 2006-11-21 18:39:34 +0000 Tim-Philipp Müller <tim@centricular.net>
22321 ext/pango/gsttextoverlay.*: Some textoverlay fixes: for one, in the video chain function, actually wait for a text bu...
22322 Original commit message from CVS:
22323 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
22324 (gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
22325 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
22326 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
22327 (gst_text_overlay_change_state):
22328 * ext/pango/gsttextoverlay.h:
22329 Some textoverlay fixes: for one, in the video chain function,
22330 actually wait for a text buffer to come in if there is none at the
22331 moment and there should be one; also, deal more gracefully with
22332 incoming buffers that do not have a timestamp or duration; discard
22333 text buffer when not needed any longer. Fixes #341681.
22334 * tests/check/Makefile.am:
22335 * tests/check/elements/.cvsignore:
22336 * tests/check/elements/textoverlay.c:
22337 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
22338 (setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
22339 (create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
22340 (test_video_waits_for_text_send_text_newsegment_thread),
22341 (test_video_waits_for_text_shutdown_element),
22342 (test_render_continuity_push_video_buffers_thread),
22343 (textoverlay_suite):
22344 Add some unit tests for textoverlay.
22346 2006-11-21 09:29:56 +0000 Tim-Philipp Müller <tim@centricular.net>
22348 gst/typefind/gsttypefindfunctions.c: Avoid integer underflow when the found probability for mp3 is smaller than the '...
22349 Original commit message from CVS:
22350 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
22351 Avoid integer underflow when the found probability for mp3 is
22352 smaller than the 'penalty' we subtract if there's not a clean
22353 mp3 header sync at offset 0.
22355 2006-11-21 08:17:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22357 docs/libs/gst-plugins-base-libs-sections.txt: Add some new symbols to the docs
22358 Original commit message from CVS:
22359 * docs/libs/gst-plugins-base-libs-sections.txt:
22360 Add some new symbols to the docs
22362 2006-11-20 16:44:28 +0000 Tim-Philipp Müller <tim@centricular.net>
22364 tests/check/: Enable ffmpegcolorspace test now that the RGBA32 issue is fixed (for now not for valgrinding though, si...
22365 Original commit message from CVS:
22366 * tests/check/Makefile.am:
22367 * tests/check/elements/ffmpegcolorspace.c:
22368 (ffmpegcolorspace_suite):
22369 Enable ffmpegcolorspace test now that the RGBA32 issue is fixed
22370 (for now not for valgrinding though, since it takes too long).
22372 2006-11-20 15:01:09 +0000 Wim Taymans <wim.taymans@gmail.com>
22374 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix RGBA32 caps. Fixes #357038.
22375 Original commit message from CVS:
22376 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
22377 (gst_ffmpeg_pixfmt_to_caps):
22378 Fix RGBA32 caps. Fixes #357038.
22380 2006-11-20 12:20:39 +0000 Tim-Philipp Müller <tim@centricular.net>
22382 gst-libs/gst/interfaces/mixertrack.h: Add FIXME so we can add some padding here in 0.11
22383 Original commit message from CVS:
22384 * gst-libs/gst/interfaces/mixertrack.h:
22385 Add FIXME so we can add some padding here in 0.11
22387 2006-11-19 17:07:34 +0000 Tim-Philipp Müller <tim@centricular.net>
22389 gst-libs/gst/rtp/gstbasertpaudiopayload.h: Fix GstBaseRTPAudioPayload structure so the whole GObject inheritance busi...
22390 Original commit message from CVS:
22391 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
22392 Fix GstBaseRTPAudioPayload structure so the whole GObject
22393 inheritance business actually works (parent class instance structure
22394 must always come first in the derived class instance structure).
22396 2006-11-16 14:35:30 +0000 Tim-Philipp Müller <tim@centricular.net>
22398 Make sure our checks and the videotestsrc plugin link against the local uninstalled gst libs and not any installed gs...
22399 Original commit message from CVS:
22400 * gst/videotestsrc/Makefile.am:
22401 * tests/check/Makefile.am:
22402 Make sure our checks and the videotestsrc plugin link against the
22403 local uninstalled gst libs and not any installed gst libs that
22404 might happen to exist as well.
22405 * tests/check/elements/adder.c: (message_received),
22406 (test_event_message_received), (test_play_twice_message_received):
22407 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
22408 Fix compiler warnings when compiling against core with disabled
22411 2006-11-16 12:55:08 +0000 Michael Smith <msmith@xiph.org>
22413 gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
22414 Original commit message from CVS:
22415 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
22416 (gst_audio_rate_sink_event), (gst_audio_rate_chain):
22417 Fix audiorate, so that it accurately sets offsets and timestamps.
22418 Doesn't change the fundamental algorithmic decisions; so should be
22420 * tests/check/Makefile.am:
22421 Enable audiorate test now that it passes.
22423 2006-11-15 10:05:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22425 sys/xvimage/xvimagesink.c: clear xv when going to NULL, remove // commented non-existant proto
22426 Original commit message from CVS:
22427 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
22428 clear xv when going to NULL, remove // commented non-existant proto
22429 * tests/examples/seek/seek.c: (main):
22430 add missing tooltip description for scrub and play_scrub
22432 2006-11-14 23:34:19 +0000 David Schleef <ds@schleef.org>
22434 configure.ac: Bump liboil requirement to 0.3.8.
22435 Original commit message from CVS:
22437 Bump liboil requirement to 0.3.8.
22438 * gst-libs/gst/riff/riff-media.c:
22440 * gst/videoscale/vs_image.h:
22441 * gst/videoscale/vs_scanline.h:
22442 Use liboil's stdint.h.
22443 * gst/videotestsrc/videotestsrc.c:
22444 Remove liboil related ifdef's, since they aren't needed now, and
22445 won't work with future versions.
22447 2006-11-14 23:08:38 +0000 David Schleef <ds@schleef.org>
22449 gst/videoscale/: Add a 4-tap image scaler. Theoretically looks much prettier.
22450 Original commit message from CVS:
22451 * gst/videoscale/Makefile.am:
22452 * gst/videoscale/gstvideoscale.c:
22453 * gst/videoscale/gstvideoscale.h:
22454 * gst/videoscale/vs_4tap.c:
22455 * gst/videoscale/vs_4tap.h:
22456 * gst/videoscale/vs_image.c:
22457 * gst/videoscale/vs_image.h:
22458 * gst/videoscale/vs_scanline.c:
22459 * gst/videoscale/vs_scanline.h:
22460 Add a 4-tap image scaler. Theoretically looks much prettier.
22461 The tap calculation could use some improvement.
22463 2006-11-14 11:54:14 +0000 Jan David Mol <j.j.d.mol@tudelft.nl>
22465 Various gsize and gssize printf fixes. Fixes #372507.
22466 Original commit message from CVS:
22467 Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
22468 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
22469 (gst_riff_parse_strf_iavs):
22470 * gst/subparse/gstsubparse.c: (convert_encoding):
22471 * gst/tcp/gstmultifdsink.c:
22472 (gst_multi_fd_sink_handle_client_write):
22473 * gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
22474 (gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
22475 (gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
22476 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
22477 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
22478 (gst_ximagesink_ximage_new):
22479 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
22480 Various gsize and gssize printf fixes. Fixes #372507.
22482 2006-11-13 18:14:48 +0000 Wim Taymans <wim.taymans@gmail.com>
22484 ext/vorbis/vorbisdec.*: First stab at vorbis reverse playback.
22485 Original commit message from CVS:
22486 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
22487 (vorbis_dec_push_forward), (vorbis_dec_push_reverse),
22488 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
22489 (vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
22490 (vorbis_dec_chain_forward), (vorbis_dec_chain):
22491 * ext/vorbis/vorbisdec.h:
22492 First stab at vorbis reverse playback.
22494 2006-11-13 17:30:17 +0000 Wim Taymans <wim.taymans@gmail.com>
22496 gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
22497 Original commit message from CVS:
22498 * gst-libs/gst/audio/gstbaseaudiosink.c:
22499 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
22500 * gst-libs/gst/audio/gstbaseaudiosink.h:
22501 Make the clock sync code more accurate wrt resampling and playback
22502 at different rates.
22503 * gst-libs/gst/audio/gstringbuffer.c:
22504 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
22505 * gst-libs/gst/audio/gstringbuffer.h:
22506 Use better algorithm to interpolate sample rates.
22508 2006-11-13 15:31:01 +0000 Michael Smith <msmith@xiph.org>
22510 ext/ogg/gstoggdemux.c: Improve a debug line slightly.
22511 Original commit message from CVS:
22512 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
22513 Improve a debug line slightly.
22514 * ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
22515 Call gst_riff_init() in plugin_init, to avoid getting errors from
22516 the debug system (unrelated changes to another plugin made this turn
22519 2006-11-10 19:20:21 +0000 Sergey Scobich <sergery.scobich@gmail.com>
22521 win32/common/libgsttag.def: Add missing symbol (#366492).
22522 Original commit message from CVS:
22523 Patch by: Sergey Scobich <sergery.scobich at gmail com>
22524 * win32/common/libgsttag.def:
22525 Add missing symbol (#366492).
22527 2006-11-10 00:52:55 +0000 Tim-Philipp Müller <tim@centricular.net>
22529 gst/playback/gststreamselector.c: Don't unref a NULL pad.
22530 Original commit message from CVS:
22531 * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
22532 Don't unref a NULL pad.
22534 2006-11-09 00:50:00 +0000 Christian Schaller <uraeus@gnome.org>
22536 ext/ogg/gstoggdemux.c: Implement first stab at reverse playback.
22537 Original commit message from CVS:
22538 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
22539 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_perform_seek),
22540 (gst_ogg_demux_handle_page), (gst_ogg_demux_chain),
22541 (gst_ogg_demux_loop_forward), (gst_ogg_demux_loop_reverse),
22542 (gst_ogg_demux_loop):
22543 Implement first stab at reverse playback.
22545 2006-11-07 07:22:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22547 gst-libs/gst/riff/riff-media.c: add h263/h264 variants to the caps, Fixes #363118
22548 Original commit message from CVS:
22549 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
22550 (gst_riff_create_video_template_caps):
22551 add h263/h264 variants to the caps, Fixes #363118
22553 2006-11-06 18:24:59 +0000 Tim-Philipp Müller <tim@centricular.net>
22555 gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
22556 Original commit message from CVS:
22557 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
22558 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
22559 Use g_strerror instead of strerror so we get UTF-8.
22561 2006-11-04 07:25:58 +0000 David Schleef <ds@schleef.org>
22563 ext/ogg/: Add/remove KW-DIRAC header here, since it is ogg-specific.
22564 Original commit message from CVS:
22565 * ext/ogg/gstoggdemux.c:
22566 * ext/ogg/gstoggmux.c:
22567 Add/remove KW-DIRAC header here, since it is ogg-specific.
22569 2006-11-03 15:44:31 +0000 Michael Smith <msmith@xiph.org>
22571 gst/typefind/gsttypefindfunctions.c: Recognise more mpeg4 elementary video streams.
22572 Original commit message from CVS:
22573 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
22574 Recognise more mpeg4 elementary video streams.
22576 2006-11-02 17:26:03 +0000 Edward Hervey <bilboed@bilboed.com>
22578 gst/typefind/gsttypefindfunctions.c: Lower the probability of mp3 typefinding functions if we don't find a valid mp3 ...
22579 Original commit message from CVS:
22580 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
22581 Lower the probability of mp3 typefinding functions if we don't find a
22582 valid mp3 header at the start of the file.
22585 2006-11-02 15:06:36 +0000 Wim Taymans <wim.taymans@gmail.com>
22587 ext/theora/: Document and partially implement an algorithm for doing reverse playback of theora video.
22588 Original commit message from CVS:
22589 * ext/theora/gsttheoradec.h:
22590 * ext/theora/theoradec.c: (gst_theora_dec_init),
22591 (theora_dec_sink_event), (theora_dec_chain_forward),
22592 (theora_dec_flush_decode), (theora_dec_chain_reverse),
22593 (theora_dec_chain):
22594 Document and partially implement an algorithm for doing reverse playback
22597 2006-11-02 14:18:45 +0000 Sergey Scobich <sergey.scobich@gmail.com>
22599 win32/: Misc. VS8 build fixes: fix syntax in config.h, add missing entries to libgsttag.def; add missing dependencies...
22600 Original commit message from CVS:
22601 Patch by: Sergey Scobich <sergey.scobich at gmail com>
22602 * win32/common/config.h:
22603 * win32/common/interfaces-enumtypes.c:
22604 * win32/common/libgsttag.def:
22605 * win32/vs8/gst-plugins-base.sln:
22606 * win32/vs8/libgstaudioresample.vcproj:
22607 * win32/vs8/libgstinterfaces.vcproj:
22608 * win32/vs8/libgstogg.vcproj:
22609 * win32/vs8/libgstriff.vcproj:
22610 * win32/vs8/libgsttag.vcproj:
22611 * win32/vs8/libgsttheora.vcproj:
22612 * win32/vs8/libgstvideoscale.vcproj:
22613 * win32/vs8/libgstvorbis.vcproj:
22614 Misc. VS8 build fixes: fix syntax in config.h, add missing entries
22615 to libgsttag.def; add missing dependencies for some vs8 projects;
22616 re-arrange placement of .def files in vs8 projects (#366334).
22618 2006-11-01 14:08:31 +0000 Tim-Philipp Müller <tim@centricular.net>
22620 ext/ogg/gstogg.c: Remove unused variable.
22621 Original commit message from CVS:
22622 * ext/ogg/gstogg.c:
22623 Remove unused variable.
22624 * ext/ogg/gstoggdemux.c:
22625 Fix Wim's surname in plugin description.
22627 2006-10-31 15:05:33 +0000 Wim Taymans <wim.taymans@gmail.com>
22629 gst-plugins-base.spec.in: spec new .h file. Fixes #368310.
22630 Original commit message from CVS:
22631 * gst-plugins-base.spec.in:
22632 spec new .h file. Fixes #368310.
22634 2006-10-31 14:19:07 +0000 Michael Smith <msmith@xiph.org>
22636 gst/tcp/gstmultifdsink.*: Make using the remove or clear signals threadsafe.
22637 Original commit message from CVS:
22638 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
22639 (gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
22640 (gst_multi_fd_sink_get_stats),
22641 (gst_multi_fd_sink_remove_client_link),
22642 (gst_multi_fd_sink_queue_buffer),
22643 (gst_multi_fd_sink_handle_clients):
22644 * gst/tcp/gstmultifdsink.h:
22645 Make using the remove or clear signals threadsafe.
22646 Make calling get-stats with an invalid fd not segfault.
22649 2006-10-31 10:49:19 +0000 Wim Taymans <wim.taymans@gmail.com>
22651 gst-libs/gst/rtp/: Fix and activate base audio payloader.
22652 Original commit message from CVS:
22653 * gst-libs/gst/rtp/Makefile.am:
22654 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
22655 (gst_base_rtp_audio_payload_init):
22656 Fix and activate base audio payloader.
22658 2006-10-28 17:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
22660 gst/typefind/gsttypefindfunctions.c: Add typefinder for QuickTime Image Files (see #366156).
22661 Original commit message from CVS:
22662 * gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
22664 Add typefinder for QuickTime Image Files (see #366156).
22666 2006-10-28 16:00:51 +0000 Tim-Philipp Müller <tim@centricular.net>
22668 gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
22669 Original commit message from CVS:
22670 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
22671 Another typo fix (#366212).
22673 2006-10-27 17:13:35 +0000 Wim Taymans <wim.taymans@gmail.com>
22675 gst/volume/gstvolume.c: Use stream time to synchronize volume property instead of rather random timestamps. This is n...
22676 Original commit message from CVS:
22677 * gst/volume/gstvolume.c: (volume_transform_ip):
22678 Use stream time to synchronize volume property instead of rather random
22679 timestamps. This is needed when gnonlin does its time shifting.
22681 2006-10-27 16:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
22684 I'm too lazy to comment this
22685 Original commit message from CVS:
22686 *** empty log message ***
22688 2006-10-27 16:45:30 +0000 Mark Nauwelaerts <manauw@skynet.be>
22690 ext/ogg/gstoggmux.c: Remove the pad from the element in release_pad.
22691 Original commit message from CVS:
22692 Patch by: Mark Nauwelaerts <manauw at skynet dot be>
22693 * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
22694 Remove the pad from the element in release_pad.
22696 2006-10-27 11:57:18 +0000 Tim-Philipp Müller <tim@centricular.net>
22698 sys/: Explicitly create our custom buffer classes at a thread-safe location as well, since g_type_class_ref() doesn't...
22699 Original commit message from CVS:
22700 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
22701 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
22702 Explicitly create our custom buffer classes at a thread-safe
22703 location as well, since g_type_class_ref() doesn't seem to be
22704 entirely thread-safe either (#365501; also see #349410).
22706 2006-10-26 10:49:00 +0000 Tim-Philipp Müller <tim@centricular.net>
22708 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...
22709 Original commit message from CVS:
22710 * gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
22711 (gst_riff_parse_info):
22712 If strings in INFO chunk are not UTF-8, do something similar to
22713 what we do for ID3v1 tags: check a number of environment variables
22714 (GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
22715 character sets to try, otherwise try the current locale and/or fall
22716 back on ISO-8859-1. Fixes #360552.
22718 2006-10-23 12:46:41 +0000 Tim-Philipp Müller <tim@centricular.net>
22720 gst/videotestsrc/: Add a bunch of exciting new checkers patterns.
22721 Original commit message from CVS:
22722 * gst/videotestsrc/gstvideotestsrc.c:
22723 (gst_video_test_src_pattern_get_type),
22724 (gst_video_test_src_set_pattern):
22725 * gst/videotestsrc/gstvideotestsrc.h:
22726 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_checkers1),
22727 (gst_video_test_src_checkers2), (gst_video_test_src_checkers4),
22728 (gst_video_test_src_checkers8):
22729 * gst/videotestsrc/videotestsrc.h:
22730 Add a bunch of exciting new checkers patterns.
22732 2006-10-23 12:06:44 +0000 Tim-Philipp Müller <tim@centricular.net>
22734 gst/subparse/: Add support for TMPlayer-type subtitles (#362845).
22735 Original commit message from CVS:
22736 * gst/subparse/Makefile.am:
22737 * gst/subparse/gstsubparse.c:
22738 (gst_sub_parse_data_format_autodetect),
22739 (gst_sub_parse_format_autodetect), (handle_buffer),
22740 (gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
22741 * gst/subparse/gstsubparse.h:
22742 * gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
22744 * gst/subparse/tmplayerparse.h:
22745 Add support for TMPlayer-type subtitles (#362845).
22746 * tests/check/elements/subparse.c: (test_tmplayer_do_test),
22747 (GST_START_TEST), (subparse_suite):
22748 Add some basic unit tests for the above.
22750 2006-10-23 11:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
22752 tests/check/elements/audiorate.c: More tests for audiorate: inject buffers to check behaviour when buffers overlap.
22753 Original commit message from CVS:
22754 * tests/check/elements/audiorate.c: (test_injector_base_init),
22755 (test_injector_class_init), (test_injector_chain),
22756 (test_injector_init), (probe_cb), (do_perfect_stream_test),
22757 (GST_START_TEST), (audiorate_suite):
22758 More tests for audiorate: inject buffers to check behaviour when
22761 2006-10-21 16:39:54 +0000 Tim-Philipp Müller <tim@centricular.net>
22763 tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...
22764 Original commit message from CVS:
22765 * tests/check/Makefile.am:
22766 * tests/check/elements/.cvsignore:
22767 * tests/check/elements/audiorate.c: (probe_cb), (got_buf),
22768 (do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
22769 Add some basic unit tests for audiorate. Disabled at the moment
22770 since it doesn't pass yet (see bug #363119).
22772 2006-10-20 17:02:19 +0000 Tim-Philipp Müller <tim@centricular.net>
22774 gst/subparse/gstsubparse.c: Add missing closing tags for markup and fix broken markup, otherwise pango won't render a...
22775 Original commit message from CVS:
22776 * gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
22777 (parse_subrip), (handle_buffer):
22778 Add missing closing tags for markup and fix broken markup,
22779 otherwise pango won't render anything (fixes #357531). Also,
22780 make sure the text we send out is always NUL-terminated
22781 (better safe than sorry etc.).
22782 * tests/check/elements/subparse.c: (test_srt_do_test),
22784 Some more tests for .srt incl. tests for the above stuff.
22786 2006-10-20 13:56:55 +0000 Stefan Kost <ensonic@users.sf.net>
22788 sys/: Try to redraw borders only when needed. Apparently this consumes resources on small devices... :-O (#363607)
22789 Original commit message from CVS:
22790 2006-10-20 Julien MOUTTE <julien@moutte.net>
22791 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
22792 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
22793 Patch by: Stefan Kost <ensonic@users.sf.net>
22794 Try to redraw borders only when needed. Apparently this consumes
22795 resources on small devices... :-O (#363607)
22797 2006-10-20 13:54:19 +0000 Michael Smith <msmith@xiph.org>
22799 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...
22800 Original commit message from CVS:
22801 * gst/tcp/gstmultifdsink.c:
22802 (gst_multi_fd_sink_client_queue_buffer):
22803 If caps change, then update the client's idea of the caps so that we
22804 don't end up re-sending streamheaders for every single buffer after
22807 2006-10-20 12:31:02 +0000 Michael Smith <msmith@xiph.org>
22809 ext/ogg/gstoggparse.c: Set caps on pushed buffers; fix up refcounting of caps objects.
22810 Original commit message from CVS:
22811 * ext/ogg/gstoggparse.c: (gst_ogg_parse_dispose),
22812 (gst_ogg_parse_append_header), (gst_ogg_parse_chain):
22813 Set caps on pushed buffers; fix up refcounting of caps objects.
22815 2006-10-19 14:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
22817 gst/typefind/gsttypefindfunctions.c: Typefind mmsh header data packet to application/x-mmsh (#362625).
22818 Original commit message from CVS:
22819 * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
22821 Typefind mmsh header data packet to application/x-mmsh (#362625).
22823 2006-10-19 09:17:48 +0000 Tim-Philipp Müller <tim@centricular.net>
22825 tests/check/: Add very simple unit test for subparse.
22826 Original commit message from CVS:
22827 * tests/check/Makefile.am:
22828 * tests/check/elements/.cvsignore:
22829 * tests/check/elements/subparse.c: (buffer_from_static_string),
22830 (setup_subparse), (teardown_subparse), (test_srt_do_test),
22831 (GST_START_TEST), (subparse_suite):
22832 Add very simple unit test for subparse.
22834 2006-10-19 09:00:21 +0000 Tim-Philipp Müller <tim@centricular.net>
22836 gst/subparse/gstsubparse.c: Strip trailing newlines from subtitle text output.
22837 Original commit message from CVS:
22838 * gst/subparse/gstsubparse.c: (strip_trailing_newlines),
22840 Strip trailing newlines from subtitle text output.
22842 2006-10-18 18:40:12 +0000 Tim-Philipp Müller <tim@centricular.net>
22844 gst/subparse/gstsubparse.c: Fix memleak; clear subparse->textbuf n state change function.
22845 Original commit message from CVS:
22846 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
22847 (gst_sub_parse_change_state):
22848 Fix memleak; clear subparse->textbuf n state change function.
22850 2006-10-18 15:13:59 +0000 Tim-Philipp Müller <tim@centricular.net>
22852 gst/subparse/gstsubparse.c: Don't require subrip (.srt) files to start with a chunk number of 1.
22853 Original commit message from CVS:
22854 * gst/subparse/gstsubparse.c:
22855 (gst_sub_parse_data_format_autodetect):
22856 Don't require subrip (.srt) files to start with a chunk number of 1.
22858 2006-10-18 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
22860 gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
22861 Original commit message from CVS:
22862 * gst-libs/gst/audio/gstbaseaudiosink.c:
22863 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
22864 * gst-libs/gst/audio/gstbaseaudiosink.h:
22865 Extract rate from the NEWSEGMENT event.
22866 Use commit_full to also take rate adjustment into account when writing
22867 samples to the ringbuffer.
22868 * gst-libs/gst/audio/gstringbuffer.c:
22869 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
22870 (gst_ring_buffer_read):
22871 * gst-libs/gst/audio/gstringbuffer.h:
22872 Added _commit_full() to also take rate into account.
22873 Use simple interpolation algorithm to resample audio.
22874 API: gst_ring_buffer_commit_full()
22875 * tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
22876 * tests/examples/seek/seek.c: (segment_done):
22877 Don't try to seek with 0.0 rate, just pause instead.
22878 Remove bogus debug line.
22880 2006-10-18 12:57:54 +0000 Tim-Philipp Müller <tim@centricular.net>
22882 gst/playback/gstplaybasebin.c: Catch async errors when starting up the subtitle bin, so we can stop waiting and conti...
22883 Original commit message from CVS:
22884 * gst/playback/gstplaybasebin.c: (subbin_startup_sync_msg),
22886 Catch async errors when starting up the subtitle bin, so we can
22887 stop waiting and continue with the main film instead of hanging
22888 forever. Fixes #339366.
22889 * tests/check/elements/playbin.c: (playbin_suite):
22890 Enable unit test for the above.
22892 2006-10-18 09:53:03 +0000 Tim-Philipp Müller <tim@centricular.net>
22894 tests/check/: Some small and basic unit tests for playbin; not very useful yet, but at least a start.
22895 Original commit message from CVS:
22896 * tests/check/Makefile.am:
22897 * tests/check/elements/.cvsignore:
22898 * tests/check/elements/playbin.c: (GST_START_TEST),
22899 (gst_red_video_src_uri_get_type),
22900 (gst_red_video_src_uri_get_protocols),
22901 (gst_red_video_src_uri_get_uri), (gst_red_video_src_uri_set_uri),
22902 (gst_red_video_src_uri_handler_init),
22903 (gst_red_video_src_init_type), (gst_red_video_src_base_init),
22904 (gst_red_video_src_create), (gst_red_video_src_class_init),
22905 (gst_red_video_src_init), (plugin_init), (playbin_suite):
22906 Some small and basic unit tests for playbin; not very useful yet,
22907 but at least a start.
22909 2006-10-18 09:46:35 +0000 Tim-Philipp Müller <tim@centricular.net>
22911 gst/playback/gstplaybin.c: The old pad activation spiel.
22912 Original commit message from CVS:
22913 * gst/playback/gstplaybin.c: (setup_sinks):
22914 The old pad activation spiel.
22916 2006-10-18 09:31:49 +0000 Tim-Philipp Müller <tim@centricular.net>
22918 gst/playback/gstplaybasebin.c: Don't hang forever if the subbin already fails to start up in the state change to PAUS...
22919 Original commit message from CVS:
22920 * gst/playback/gstplaybasebin.c: (setup_source):
22921 Don't hang forever if the subbin already fails to start up in
22922 the state change to PAUSED (#339366).
22924 2006-10-17 17:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
22926 gst-libs/gst/interfaces/tuner.c: Fix some function guards, add some more function guards.
22927 Original commit message from CVS:
22928 * gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
22929 (gst_tuner_set_channel), (gst_tuner_get_channel),
22930 (gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
22931 (gst_tuner_set_frequency), (gst_tuner_get_frequency),
22932 (gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
22933 (gst_tuner_find_channel_by_name):
22934 Fix some function guards, add some more function guards.
22936 2006-10-17 11:34:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22938 gst/playback/gstdecodebin.c: Don't return a pad from get_our_ghost_pad unless it is actually the one we want.
22939 Original commit message from CVS:
22940 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
22941 (remove_element_chain):
22942 Don't return a pad from get_our_ghost_pad unless it is actually the
22944 Change a cast in remove_element_chain slightly.
22946 2006-10-13 15:20:29 +0000 Julien Moutte <julien@moutte.net>
22948 tests/examples/seek/seek.c: Segment seeking needs to use the rate and set stop to -1.
22949 Original commit message from CVS:
22950 2006-10-13 Julien MOUTTE <julien@moutte.net>
22951 * tests/examples/seek/seek.c: (do_seek), (start_seek),
22952 (rate_spinbutton_changed_cb), (segment_done),
22953 (msg_state_changed):
22954 Segment seeking needs to use the rate and set stop to -1.
22956 2006-10-13 14:15:42 +0000 Ville Syrjala <ville.syrjala@movial.fi>
22958 gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
22959 Original commit message from CVS:
22960 * gst-libs/gst/audio/gstbaseaudiosink.c:
22961 (gst_base_audio_sink_setcaps):
22962 Don't crash when ringbuffer is not yet created.
22963 Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
22965 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
22966 * gst/playback/gststreamselector.c:
22967 (gst_stream_selector_request_new_pad):
22968 Activate pads befre adding them to running elements.
22970 2006-10-13 11:25:10 +0000 Julien Moutte <julien@moutte.net>
22972 tests/examples/seek/seek.c: Stop the scale updater when we start grabing the slider. Don't wait for the pipeline to b...
22973 Original commit message from CVS:
22974 2006-10-13 Julien MOUTTE <julien@moutte.net>
22975 * tests/examples/seek/seek.c: (do_seek), (start_seek),
22976 (rate_spinbutton_changed_cb), (msg_state_changed): Stop the
22978 updater when we start grabing the slider. Don't wait for the
22979 pipeline to be PAUSED.
22981 2006-10-13 08:57:52 +0000 Tim-Philipp Müller <tim@centricular.net>
22983 gst-libs/gst/interfaces/mixer.c: Guard mixer interface functions against bogus arguments.
22984 Original commit message from CVS:
22985 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
22986 (gst_mixer_set_volume), (gst_mixer_get_volume),
22987 (gst_mixer_set_mute), (gst_mixer_set_option),
22988 (gst_mixer_get_option), (gst_mixer_mute_toggled),
22989 (gst_mixer_record_toggled), (gst_mixer_volume_changed),
22990 (gst_mixer_option_changed):
22991 Guard mixer interface functions against bogus arguments.
22993 2006-10-12 19:39:07 +0000 Julien Moutte <julien@moutte.net>
22995 tests/examples/seek/seek.c: Use state-changed messages to trigger start/stop of scale update timer. Indeed the scale ...
22996 Original commit message from CVS:
22997 2006-10-12 Julien MOUTTE <julien@moutte.net>
22998 * tests/examples/seek/seek.c: (do_seek), (start_seek),
23000 (play_cb), (pause_cb), (stop_cb),
23001 (rate_spinbutton_changed_cb),
23002 (msg_state_changed), (main): Use state-changed messages to
23004 start/stop of scale update timer. Indeed the scale slider was
23005 jumping here and there because the update timer was activated
23006 before seek completed. This fixes instant applying of rate
23008 by pressing the spinbutton like a crazy man !
23010 2006-10-12 19:09:06 +0000 Sebastien Cote <sebas642@yahoo.ca>
23012 gst-libs/gst/rtp/gstbasertppayload.c: Fix two small memory leaks (#361456).
23013 Original commit message from CVS:
23014 Patch by: Sebastien Cote <sebas642 at yahoo.ca>
23015 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
23016 (gst_basertppayload_finalize):
23017 Fix two small memory leaks (#361456).
23019 2006-10-10 18:56:01 +0000 Julien Moutte <julien@moutte.net>
23021 tests/examples/seek/seek.c: When changing spinbutton we try to change the rate on the fly.
23022 Original commit message from CVS:
23023 2006-10-10 Julien MOUTTE <julien@moutte.net>
23024 * tests/examples/seek/seek.c: (do_seek),
23025 (rate_spinbutton_changed_cb): When changing spinbutton we try
23026 to change the rate on the fly.
23028 2006-10-10 16:50:06 +0000 Wim Taymans <wim.taymans@gmail.com>
23030 gst-libs/gst/riff/: Add WMS caps.
23031 Original commit message from CVS:
23032 * gst-libs/gst/riff/riff-ids.h:
23033 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
23034 (gst_riff_create_audio_template_caps):
23037 2006-10-10 12:49:03 +0000 Josep Torre Valles <josep@fluendo.com>
23039 ext/gnomevfs/: Fix URI interface implementation return type.
23040 Original commit message from CVS:
23041 2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
23042 Patch by: Josep Torre Valles <josep@fluendo.com>
23043 * ext/gnomevfs/gstgnomevfssink.c:
23044 * ext/gnomevfs/gstgnomevfssrc.c:
23045 Fix URI interface implementation return type.
23046 * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
23047 Fix what looks like a copy/paste issue when assigning values.
23048 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
23049 (gst_audio_filter_template_get_type):
23050 Cast to prevent Forte warnings.
23051 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
23052 Fix URI interface implementation return type.
23053 gst_pad_query_position requires a signed integer pointer as
23054 3rd parameter, GstClockTime is unsigned.
23055 * gst/audioconvert/audioconvert.c:
23056 Fix integer overflow when treated as signed.
23057 * gst/audioresample/resample.c: (resample_add_input_data):
23058 Cast to prevent warnings on Forte.
23059 * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
23060 Fix integer overflow when treated as signed.
23061 * gst/ffmpegcolorspace/imgconvert_template.h:
23062 Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
23063 * gst/playback/gstdecodebin.c: (queue_filled_cb),
23064 (cleanup_decodebin):
23065 Who initialises a guint to -1!
23066 Cast function pointers to prevent warnings on Forte.
23067 * gst/playback/gstplaybasebin.c: (queue_deadlock_check),
23068 (queue_threshold_reached):
23069 Cast function pointers correctly to prevent warnings on Forte.
23070 * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
23071 Cast function pointers correctly to prevent warnings on Forte.
23072 * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
23073 Obvious change to unsigned, 0xEF > max signed char.
23074 * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
23075 GstClockTime is unsigned, initialise correctly.
23076 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
23077 Cast so pointer arithemetic doesn't cause warnings on Forte.
23078 * gst/videorate/gstvideorate.c:
23079 Use correct return value.
23080 * tests/examples/seek/scrubby.c:
23081 GstClockTime is unsigned, initialise correctly.
23083 2006-10-10 11:20:03 +0000 Ferenc Gerlits <fgerlits@gmail.com>
23085 gst/typefind/gsttypefindfunctions.c: Recognise XML files and XML-like files shorter than 256 bytes as well (fixes #35...
23086 Original commit message from CVS:
23087 Patch by: Ferenc Gerlits <fgerlits at gmail com>
23088 * gst/typefind/gsttypefindfunctions.c:
23089 Recognise XML files and XML-like files shorter than 256 bytes as
23090 well (fixes #359237).
23092 2006-10-09 15:01:30 +0000 Edgard Lima <edgard.lima@indt.org.br>
23096 * gst/typefind/gsttypefindfunctions.c:
23097 Added typefind functions to video/x-nuv media.
23098 Original commit message from CVS:
23099 Added typefind functions to video/x-nuv media.
23101 2006-10-08 16:59:31 +0000 Tim-Philipp Müller <tim@centricular.net>
23103 gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
23104 Original commit message from CVS:
23105 * gst-libs/gst/interfaces/xoverlay.c:
23106 (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
23107 Some more guards against invalid input.
23109 2006-10-07 18:35:39 +0000 Julien Moutte <julien@moutte.net>
23111 ext/pango/gsttextoverlay.c: Useless goto.
23112 Original commit message from CVS:
23113 2006-10-07 Julien MOUTTE <julien@moutte.net>
23114 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
23116 * tests/examples/seek/seek.c: (do_seek),
23117 (rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
23118 seek example to experiment with rates != 1.0 (reverse playback
23121 2006-10-06 19:20:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23123 gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
23124 Original commit message from CVS:
23125 * gst-libs/gst/interfaces/xoverlay.c:
23126 Unref message in doc-example (spotted by Robert McQueen)
23128 2006-10-06 17:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
23130 gst/typefind/gsttypefindfunctions.c: printf fix.
23131 Original commit message from CVS:
23132 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
23133 (mpeg1_parse_header), (mpeg1_sys_type_find):
23136 2006-10-06 14:37:43 +0000 Wim Taymans <wim.taymans@gmail.com>
23138 gst/playback/: Activate dynamic pads before adding them to the element.
23139 Original commit message from CVS:
23140 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
23142 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
23143 Activate dynamic pads before adding them to the element.
23145 2006-10-06 14:04:53 +0000 Michael Smith <msmith@xiph.org>
23147 gst-libs/gst/floatcast/floatcast.h: Fix obviously-bogus macros; use the correct types.
23148 Original commit message from CVS:
23149 * gst-libs/gst/floatcast/floatcast.h:
23150 Fix obviously-bogus macros; use the correct types.
23152 2006-10-06 13:34:46 +0000 Wim Taymans <wim.taymans@gmail.com>
23154 gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
23155 Original commit message from CVS:
23156 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23157 (gst_base_rtp_depayload_change_state):
23158 Also call parent state change function to activate pads.
23159 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
23160 (mpeg1_parse_header), (mpeg1_sys_type_find):
23161 Add some more debug info in mpeg typefinding.
23163 2006-10-06 12:57:10 +0000 Michael Smith <msmith@xiph.org>
23165 ext/theora/theoradec.c: Zero byte theora packets are valid and well-defined; don't warn on them.
23166 Original commit message from CVS:
23167 * ext/theora/theoradec.c: (theora_dec_chain):
23168 Zero byte theora packets are valid and well-defined; don't warn on
23171 2006-10-06 10:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23173 gst/tcp/gstmultifdsink.c: API: add dropped_buffers to the get-stats GValueArray
23174 Original commit message from CVS:
23175 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
23176 (gst_multi_fd_sink_get_stats), (find_limits),
23177 (gst_multi_fd_sink_queue_buffer):
23178 API: add dropped_buffers to the get-stats GValueArray
23180 2006-10-05 15:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
23182 Printf format fixes.
23183 Original commit message from CVS:
23184 * ext/alsa/gstalsadeviceprobe.c:
23185 (gst_alsa_device_property_probe_get_values):
23186 * ext/alsa/gstalsasink.c: (set_hwparams):
23187 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
23188 (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
23189 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
23190 (gst_ogg_mux_process_best_pad):
23191 * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
23192 (gst_ogg_parse_chain):
23193 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
23194 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
23195 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
23196 (gst_vorbis_enc_buffer_check_discontinuous):
23197 * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
23198 * gst-libs/gst/audio/gstbaseaudiosink.c:
23199 (gst_base_audio_sink_render):
23200 * gst-libs/gst/cdda/gstcddabasesrc.c:
23201 (gst_cdda_base_src_handle_track_seek):
23202 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23203 (gst_base_rtp_depayload_push_full):
23204 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
23205 * gst/audioresample/resample.c: (resample_input_pushthrough):
23206 * gst/playback/gstplaybasebin.c: (queue_out_of_data):
23207 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
23208 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
23209 (wavpack_type_find):
23210 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
23211 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
23212 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
23213 * tests/check/elements/volume.c: (GST_START_TEST):
23214 Printf format fixes.
23216 2006-10-04 13:18:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23218 gst/tcp/gsttcp.c: Fix a simple mistake (see the docs)
23219 Original commit message from CVS:
23220 * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
23221 Fix a simple mistake (see the docs)
23224 2006-10-04 13:15:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23226 * win32/common/config.h:
23228 Original commit message from CVS:
23231 2006-10-03 12:11:45 +0000 Tim-Philipp Müller <tim@centricular.net>
23233 docs/plugins/: Add vorbistag element to docs; update version numbers to 0.10.10.1.
23234 Original commit message from CVS:
23235 * docs/plugins/Makefile.am:
23236 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
23237 * docs/plugins/gst-plugins-base-plugins-sections.txt:
23238 * docs/plugins/gst-plugins-base-plugins.args:
23239 * docs/plugins/gst-plugins-base-plugins.hierarchy:
23240 * docs/plugins/inspect/plugin-adder.xml:
23241 * docs/plugins/inspect/plugin-alsa.xml:
23242 * docs/plugins/inspect/plugin-audioconvert.xml:
23243 * docs/plugins/inspect/plugin-audiorate.xml:
23244 * docs/plugins/inspect/plugin-audioresample.xml:
23245 * docs/plugins/inspect/plugin-audiotestsrc.xml:
23246 * docs/plugins/inspect/plugin-cdparanoia.xml:
23247 * docs/plugins/inspect/plugin-decodebin.xml:
23248 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
23249 * docs/plugins/inspect/plugin-gdp.xml:
23250 * docs/plugins/inspect/plugin-gnomevfs.xml:
23251 * docs/plugins/inspect/plugin-libvisual.xml:
23252 * docs/plugins/inspect/plugin-ogg.xml:
23253 * docs/plugins/inspect/plugin-pango.xml:
23254 * docs/plugins/inspect/plugin-playbin.xml:
23255 * docs/plugins/inspect/plugin-subparse.xml:
23256 * docs/plugins/inspect/plugin-tcp.xml:
23257 * docs/plugins/inspect/plugin-theora.xml:
23258 * docs/plugins/inspect/plugin-typefindfunctions.xml:
23259 * docs/plugins/inspect/plugin-video4linux.xml:
23260 * docs/plugins/inspect/plugin-videorate.xml:
23261 * docs/plugins/inspect/plugin-videoscale.xml:
23262 * docs/plugins/inspect/plugin-videotestsrc.xml:
23263 * docs/plugins/inspect/plugin-volume.xml:
23264 * docs/plugins/inspect/plugin-vorbis.xml:
23265 * docs/plugins/inspect/plugin-ximagesink.xml:
23266 * docs/plugins/inspect/plugin-xvimagesink.xml:
23267 Add vorbistag element to docs; update version numbers to 0.10.10.1.
23269 2006-10-03 11:51:48 +0000 James Doc Livingston <doclivingston@gmail.com>
23271 ext/vorbis/: Add new vorbistag element which derives from vorbisparse and is essentially the same as well, only that ...
23272 Original commit message from CVS:
23273 Patch by: James "Doc" Livingston <doclivingston at gmail com>
23274 * ext/vorbis/Makefile.am:
23275 * ext/vorbis/vorbis.c: (plugin_init):
23276 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_class_init),
23277 (vorbis_parse_parse_packet), (vorbis_parse_chain):
23278 * ext/vorbis/vorbisparse.h:
23279 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_base_init),
23280 (gst_vorbis_tag_class_init), (gst_vorbis_tag_init),
23281 (gst_vorbis_tag_parse_packet):
23282 * ext/vorbis/vorbistag.h:
23283 Add new vorbistag element which derives from vorbisparse
23284 and is essentially the same as well, only that it implements
23285 the GstTagSetter interface and can modify the stream's
23286 vorbiscomment on the fly (#335635).
23287 * tests/check/Makefile.am:
23288 * tests/check/elements/.cvsignore:
23289 * tests/check/elements/vorbistag.c: (setup_vorbistag),
23290 (cleanup_vorbistag), (buffer_probe), (start_pipeline),
23291 (get_buffer), (stop_pipeline), (_create_codebook_header_buffer),
23292 (_create_audio_buffer), (GST_START_TEST), (vorbistag_suite):
23293 Add unit test for new vorbistag element.
23295 2006-10-03 10:36:38 +0000 Tim-Philipp Müller <tim@centricular.net>
23297 ext/vorbis/vorbisparse.c: Set BOS flag in packet structure to fix 'jump depends on unitialized value' errors in valgr...
23298 Original commit message from CVS:
23299 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init),
23300 (vorbis_parse_push_headers), (vorbis_parse_chain):
23301 Set BOS flag in packet structure to fix 'jump depends
23302 on unitialized value' errors in valgrind; various minor
23305 2006-09-30 15:30:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23307 gst/playback/gstdecodebin.c: Fix typo in a debug statement.
23308 Original commit message from CVS:
23309 * gst/playback/gstdecodebin.c: (close_pad_link):
23310 Fix typo in a debug statement.
23311 * gst/playback/gstplaybasebin.c: (probe_triggered),
23312 (new_decoded_pad_full), (new_decoded_pad), (subs_new_decoded_pad),
23313 (gen_source_element), (source_new_pad), (analyse_source),
23315 When handling no_more_pads in new_decoded_pad, make sure to treat
23316 subtitle pads correctly. Fixes playback with subtitle files.
23317 Move a recurring message to LOG level.
23318 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
23319 The maximum value for the Xv colorkey on this Radeon is 0xFFFFFFFF,
23320 which ends up as -1 when cast to an int. Make the logic handle the
23321 max value as an unsigned mask and only change the colorkey when it's
23322 a value we recognise.
23324 2006-09-30 00:14:20 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
23326 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
23327 Original commit message from CVS:
23328 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23329 Removed empty * between paragraphs
23331 2006-09-29 23:50:53 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
23333 gst-libs/gst/rtp/: Moved some documentation into .c file
23334 Original commit message from CVS:
23335 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23336 * gst-libs/gst/rtp/README:
23337 Moved some documentation into .c file
23339 2006-09-29 17:35:01 +0000 Wim Taymans <wim.taymans@gmail.com>
23341 gst/playback/gstdecodebin.c: Fix compilation.
23342 Original commit message from CVS:
23343 * gst/playback/gstdecodebin.c: (no_more_pads):
23346 2006-09-29 16:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
23348 gst/playback/gstdecodebin.c: Remove g_print
23349 Original commit message from CVS:
23350 * gst/playback/gstdecodebin.c: (new_caps):
23352 * gst/playback/gstplaybin.c:
23355 2006-09-29 15:16:32 +0000 Tim-Philipp Müller <tim@centricular.net>
23357 tests/check/Makefile.am: Re-enable cddabasesrc test to see if it works again now.
23358 Original commit message from CVS:
23359 * tests/check/Makefile.am:
23360 Re-enable cddabasesrc test to see if it works again
23363 2006-09-29 13:46:45 +0000 Wim Taymans <wim.taymans@gmail.com>
23365 gst/playback/gstplaybasebin.c: Handle invalid URIs a bit more gracefully.
23366 Original commit message from CVS:
23367 * gst/playback/gstplaybasebin.c: (setup_subtitle),
23368 (gen_source_element):
23369 Handle invalid URIs a bit more gracefully.
23371 2006-09-29 12:54:28 +0000 Tim-Philipp Müller <tim@centricular.net>
23373 tests/check/pipelines/oggmux.c: Remove obsolete comment.
23374 Original commit message from CVS:
23375 * tests/check/pipelines/oggmux.c:
23376 Remove obsolete comment.
23378 2006-09-29 10:43:05 +0000 James Doc Livingston <doclivingston@gmail.com>
23380 ext/ogg/gstoggmux.c: Commit patch from James "Doc" Livingston, adds proper EOS handling in oggmux. GStreamer can, for...
23381 Original commit message from CVS:
23382 * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
23383 (gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
23384 (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
23385 (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
23386 (gst_ogg_mux_collected):
23387 Commit patch from James "Doc" Livingston, adds proper EOS handling
23388 in oggmux. GStreamer can, for the first time ever, create a valid
23390 * tests/check/pipelines/oggmux.c: (check_chain_final_state),
23392 Reenable tests now that they pass.
23394 2006-09-29 08:20:03 +0000 Wim Taymans <wim.taymans@gmail.com>
23396 gst/tcp/gstmultifdsink.c: Stop reading commands when EOF (we read 0) as well.
23397 Original commit message from CVS:
23398 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
23399 Stop reading commands when EOF (we read 0) as well.
23401 2006-09-28 15:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
23403 gst/playback/gstdecodebin.c: Implement delayed caps linking needed for element with a lot of different caps on the sr...
23404 Original commit message from CVS:
23405 * gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
23406 (close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
23407 (find_dynamic), (unlinked), (close_link):
23408 Implement delayed caps linking needed for element with a lot of
23409 different caps on the src pads that get fixed at runtime.
23410 Improve management of dynamic elements.
23411 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
23412 (group_destroy), (group_commit), (check_queue), (queue_overrun),
23413 (gen_preroll_element), (remove_groups), (unknown_type),
23414 (add_element_stream), (no_more_pads_full), (no_more_pads),
23415 (sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
23416 (new_decoded_pad), (setup_subtitle), (array_has_value),
23417 (gen_source_element), (source_new_pad), (has_all_raw_caps),
23418 (analyse_source), (remove_decoders), (make_decoder),
23419 (remove_source), (setup_source), (finish_source), (prepare_output),
23420 (gst_play_base_bin_change_state):
23421 * gst/playback/gstplaybasebin.h:
23422 Use more _CAST instead of full type checking casts.
23423 Small cleanups, plug some leaks.
23424 Handle dynamic sources.
23425 Add some helper functions to create lists of strings used for
23426 blacklisting and other stuff.
23427 Refactor some code dealing with analysing the source.
23428 Re-enable sources without pads (like cd:// or other selfcontained
23431 2006-09-28 15:08:15 +0000 Wim Taymans <wim.taymans@gmail.com>
23433 gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
23434 Original commit message from CVS:
23435 * gst-libs/gst/audio/gstbaseaudiosink.c:
23436 (gst_base_audio_sink_render):
23437 When we have a timestamp, we can still perform clipping.
23438 When we have no clock, we must play the sample ASAP.
23440 2006-09-28 11:46:26 +0000 Wim Taymans <wim.taymans@gmail.com>
23442 gst/audiorate/gstaudiorate.c: Set caps on outgoing buffers.
23443 Original commit message from CVS:
23444 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
23445 Set caps on outgoing buffers.
23446 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
23447 (gst_video_rate_event), (gst_video_rate_chain):
23448 * gst/videorate/gstvideorate.h:
23449 Fix videorate some more. Fixes #357977
23451 2006-09-28 11:34:05 +0000 Tim-Philipp Müller <tim@centricular.net>
23453 tests/check/elements/adder.c: Don't set timeout to 6 seconds when we're running in valgrind ... (and how is 6 seconds...
23454 Original commit message from CVS:
23455 * tests/check/elements/adder.c: (adder_suite):
23456 Don't set timeout to 6 seconds when we're running
23457 in valgrind ... (and how is 6 seconds longer than
23458 the default anyway?)
23460 2006-09-28 10:49:56 +0000 Wim Taymans <wim.taymans@gmail.com>
23462 gst/audiorate/gstaudiorate.c: Keep sink and src segment to keep track of time and support more input formats.
23463 Original commit message from CVS:
23464 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
23465 (gst_audio_rate_sink_event), (gst_audio_rate_convert),
23466 (gst_audio_rate_convert_segments), (gst_audio_rate_chain):
23467 Keep sink and src segment to keep track of time and support more
23469 Fix bogus next_offset and run_time calculation, don't understand how
23470 this could have worked before. Fixes #357976.
23471 Remove some unneeded vars.
23473 2006-09-28 09:41:20 +0000 Tim-Philipp Müller <tim@centricular.net>
23475 gst/playback/gstplaybin.c: Only remove visualisation from visbin if there is a visbin (or: don't throw warnings when ...
23476 Original commit message from CVS:
23477 * gst/playback/gstplaybin.c: (remove_sinks):
23478 Only remove visualisation from visbin if there is a visbin (or:
23479 don't throw warnings when closing totem without playing a file).
23481 2006-09-27 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
23483 gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
23484 Original commit message from CVS:
23485 * gst-libs/gst/audio/gstbaseaudiosink.c:
23486 (gst_base_audio_sink_render):
23487 Add some more info in a WARNING.
23488 * gst-libs/gst/audio/gstbaseaudiosrc.c:
23489 (gst_base_audio_src_create):
23490 Handle PAUSE in create function, use new -core addition to
23491 wait for playing. Fixes pausing and resuming capture from an
23493 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
23494 (gst_ring_buffer_read):
23495 Constify some more.
23496 Caller supports interrupted reads now.
23498 2006-09-27 13:29:49 +0000 Christian Schaller <uraeus@gnome.org>
23500 * gst-plugins-base.spec.in:
23501 add new header file to spec
23502 Original commit message from CVS:
23503 add new header file to spec
23505 2006-09-27 12:55:45 +0000 Tim-Philipp Müller <tim@centricular.net>
23507 tests/check/Makefile.am: Another attempt to make the gen64 buildbot happy.
23508 Original commit message from CVS:
23509 * tests/check/Makefile.am:
23510 Another attempt to make the gen64 buildbot happy.
23512 2006-09-27 11:58:17 +0000 Jonathan Matthew <jonathan@kaolin.wh9.net>
23514 ext/libvisual/visual.c: Libvisual plugin was not passing audio data to libvisual 0.4.0 correctly. Fixes #357800
23515 Original commit message from CVS:
23516 Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
23517 * ext/libvisual/visual.c: (gst_visual_clear_actors),
23518 (gst_visual_chain), (gst_visual_change_state):
23519 Libvisual plugin was not passing audio data to libvisual 0.4.0
23520 correctly. Fixes #357800
23522 2006-09-27 11:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
23524 tests/check/pipelines/simple-launch-lines.c: Add timeout to _get_state() so we see which pipeline it is that causes t...
23525 Original commit message from CVS:
23526 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
23527 Add timeout to _get_state() so we see which pipeline it is
23528 that causes trouble on the gen64 build bot.
23530 2006-09-27 11:06:54 +0000 Wim Taymans <wim.taymans@gmail.com>
23532 gst-libs/gst/rtp/gstbasertpdepayload.c: the source pad always uses fixed caps.
23533 Original commit message from CVS:
23534 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23535 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
23536 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
23537 (gst_base_rtp_depayload_set_gst_timestamp):
23538 the source pad always uses fixed caps.
23540 2006-09-27 11:05:08 +0000 Wim Taymans <wim.taymans@gmail.com>
23542 Added docs for the audio libs.
23543 Original commit message from CVS:
23544 * docs/libs/gst-plugins-base-libs-docs.sgml:
23545 * docs/libs/gst-plugins-base-libs-sections.txt:
23546 * gst-libs/gst/audio/gstaudioclock.c:
23547 * gst-libs/gst/audio/gstaudioclock.h:
23548 * gst-libs/gst/audio/gstaudiosink.c:
23549 * gst-libs/gst/audio/gstaudiosink.h:
23550 * gst-libs/gst/audio/gstaudiosrc.c:
23551 * gst-libs/gst/audio/gstbaseaudiosink.c:
23552 (gst_base_audio_sink_render):
23553 * gst-libs/gst/audio/gstbaseaudiosink.h:
23554 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
23555 * gst-libs/gst/audio/gstbaseaudiosrc.h:
23556 * gst-libs/gst/audio/gstringbuffer.h:
23557 Added docs for the audio libs.
23559 2006-09-27 10:59:24 +0000 Tim-Philipp Müller <tim@centricular.net>
23561 tests/check/Makefile.am: Temporarily disable test that fails on the bots for unknown reasons.
23562 Original commit message from CVS:
23563 * tests/check/Makefile.am:
23564 Temporarily disable test that fails on the bots for unknown reasons.
23566 2006-09-27 00:13:29 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
23568 gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
23569 Original commit message from CVS:
23570 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23571 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
23572 Moved AudioCodecType into priv
23573 Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
23575 2006-09-25 15:47:25 +0000 Wim Taymans <wim.taymans@gmail.com>
23577 gst/playback/gstdecodebin.c: Cleanups and small leak fixes.
23578 Original commit message from CVS:
23579 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
23580 (add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
23581 (is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
23583 Cleanups and small leak fixes.
23584 Added Depayloaders to valid list of autopluggable elements.
23586 2006-09-25 13:24:59 +0000 Wim Taymans <wim.taymans@gmail.com>
23588 gst/playback/gstplaybin.c: Detect NO_PREROLL state change returns and disable clock distribution to the sinks so that...
23589 Original commit message from CVS:
23590 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
23591 (gst_play_bin_vis_blocked), (gst_play_bin_set_property),
23592 (gen_video_element), (gen_text_element), (gen_audio_element),
23593 (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
23594 (gst_play_bin_set_clock_func), (gst_play_bin_change_state):
23595 Detect NO_PREROLL state change returns and disable clock distribution to
23596 the sinks so that sync is disabled.
23597 Avoid some type checking and do simple casts instead.
23598 Small cleanups, fix some FIXMEs.
23599 Be more robust when linking user specified elements, catch an report
23600 errors. Fixes #357404.
23601 Fix some leaks in the error paths.
23603 2006-09-25 12:55:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23606 ChangeLog surgery for missing bug-number
23607 Original commit message from CVS:
23608 ChangeLog surgery for missing bug-number
23610 2006-09-25 11:28:15 +0000 Peter Kjellerstedt <pkj@axis.com>
23612 gst/playback/test.c: Fix compilation with uClibc and -Werror (#357591).
23613 Original commit message from CVS:
23614 Patch by: Peter Kjellerstedt <pkj at axis com>
23615 * gst/playback/test.c:
23616 Fix compilation with uClibc and -Werror (#357591).
23618 2006-09-25 10:21:31 +0000 Tim-Philipp Müller <tim@centricular.net>
23620 gst-libs/gst/tag/gstvorbistag.c: Parse dates that are followed by a time as well (#357532).
23621 Original commit message from CVS:
23622 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
23623 Parse dates that are followed by a time as well (#357532).
23624 * tests/check/libs/tag.c: (test_vorbis_tags):
23625 Add unit test for this.
23627 2006-09-23 15:24:55 +0000 Tim-Philipp Müller <tim@centricular.net>
23629 gst/: A few array const-ifications.
23630 Original commit message from CVS:
23631 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
23632 (gst_audio_convert_transform_caps):
23633 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
23634 * gst/videotestsrc/videotestsrc.h:
23635 A few array const-ifications.
23637 2006-09-23 15:02:51 +0000 Tim-Philipp Müller <tim@centricular.net>
23639 tests/check/Makefile.am: See if this makes the build bots happy.
23640 Original commit message from CVS:
23641 * tests/check/Makefile.am:
23642 See if this makes the build bots happy.
23643 * tests/check/libs/cddabasesrc.c:
23646 2006-09-23 14:30:53 +0000 Young-Ho Cha <ganadist@chollian.net>
23648 gst/subparse/samiparse.c: More case-insensitivity for certain tags; recognise entities with decimal codes as special ...
23649 Original commit message from CVS:
23650 Patch by: Young-Ho Cha <ganadist at chollian dot net>
23651 * gst/subparse/samiparse.c: (handle_start_font),
23652 (fix_invalid_entities):
23653 More case-insensitivity for certain tags; recognise entities with
23654 decimal codes as special entities as well (#357330).
23656 2006-09-23 13:32:07 +0000 Tim-Philipp Müller <tim@centricular.net>
23658 gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
23659 Original commit message from CVS:
23660 * gst-libs/gst/Makefile.am:
23661 Need to build tag directory before cdda.
23663 2006-09-23 13:21:07 +0000 Tim-Philipp Müller <tim@centricular.net>
23665 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
23666 Original commit message from CVS:
23667 * docs/libs/gst-plugins-base-libs-sections.txt:
23668 * gst-libs/gst/cdda/Makefile.am:
23669 * gst-libs/gst/cdda/gstcddabasesrc.c:
23670 (gst_cdda_base_src_base_init):
23671 * gst-libs/gst/cdda/gstcddabasesrc.h:
23672 * gst-libs/gst/tag/tag.h:
23673 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
23674 (gst_tag_register_musicbrainz_tags):
23675 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
23676 depend on libgsttag. This is required so we can extract/read tags like
23677 DISCID without depending on libgstcddabasesrc (which used to register
23679 * gst-libs/gst/tag/gstvorbistag.c:
23680 Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
23681 tags (also see #347848).
23682 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
23683 Log vorbis comments we are actually writing. Const-ify array.
23685 2006-09-23 08:53:30 +0000 Wim Taymans <wim.taymans@gmail.com>
23687 gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
23688 Original commit message from CVS:
23689 * gst/playback/gstplaybasebin.c: (gen_preroll_element):
23690 Improve buffering a bit by avoiding a deadlock because we cannot assume
23691 the underrun is always called.
23693 2006-09-23 08:51:14 +0000 Young-Ho Cha <ganadist@chollian.net>
23695 gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
23696 Original commit message from CVS:
23697 Patch by: Young-Ho Cha <ganadist at chollian dot net>
23698 * gst-libs/gst/riff/riff-ids.h:
23699 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
23700 (gst_riff_create_audio_template_caps):
23701 Added MPEG-4 AAC and id and caps. Fixes #357289
23702 Added WMA9 Lossless id.
23704 2006-09-22 14:50:01 +0000 Tim-Philipp Müller <tim@centricular.net>
23706 ext/gnomevfs/gstgnomevfssrc.c: Fix misleading docs addition.
23707 Original commit message from CVS:
23708 * ext/gnomevfs/gstgnomevfssrc.c:
23709 Fix misleading docs addition.
23710 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
23711 Get rid of compiler warning the right way.
23713 2006-09-22 14:13:34 +0000 Wim Taymans <wim.taymans@gmail.com>
23715 gst-libs/gst/rtp/gstbasertpdepayload.*: Small cleanups.
23716 Original commit message from CVS:
23717 * gst-libs/gst/rtp/gstbasertpdepayload.c:
23718 (gst_base_rtp_depayload_finalize),
23719 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
23720 (gst_base_rtp_depayload_push_full),
23721 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
23722 (gst_base_rtp_depayload_process),
23723 (gst_base_rtp_depayload_set_gst_timestamp),
23724 (gst_base_rtp_depayload_queue_release):
23725 * gst-libs/gst/rtp/gstbasertpdepayload.h:
23728 Refactored the process method and added methods to push from the process
23730 Use _scale functions.
23731 API: gst_base_rtp_depayload_push_ts
23732 API: gst_base_rtp_depayload_push
23733 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
23734 timestamps are uint.
23736 2006-09-22 11:59:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23738 gst-libs/gst/interfaces/xoverlay.c: Remove unused statement from doc example.
23739 Original commit message from CVS:
23740 * gst-libs/gst/interfaces/xoverlay.c:
23741 Remove unused statement from doc example.
23743 2006-09-22 09:52:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23745 * gst/videorate/gstvideorate.c:
23747 Original commit message from CVS:
23750 2006-09-21 13:49:47 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23752 gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
23753 Original commit message from CVS:
23754 * gst-libs/gst/interfaces/videoorientation.c:
23755 (gst_video_orientation_iface_init),
23756 (gst_video_orientation_get_hflip),
23757 (gst_video_orientation_get_vflip),
23758 (gst_video_orientation_get_hcenter),
23759 (gst_video_orientation_get_vcenter),
23760 (gst_video_orientation_set_hflip),
23761 (gst_video_orientation_set_vflip),
23762 (gst_video_orientation_set_hcenter),
23763 (gst_video_orientation_set_vcenter):
23764 Add since tags to new API docs, ChangeLog surgery (forgot API keyword
23767 2006-09-21 09:27:47 +0000 Tim-Philipp Müller <tim@centricular.net>
23769 tests/check/: but disable for now since it doesn't pass (something wrong with
23770 Original commit message from CVS:
23771 * tests/check/Makefile.am:
23772 * tests/check/elements/.cvsignore:
23773 * tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
23774 (create_rgb_conversions), (rgb_conversion_free),
23775 (right_shift_colour), (fix_expected_colour), (check_rgb_buf),
23776 (got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
23777 Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
23778 but disable for now since it doesn't pass (something wrong with
23781 2006-09-21 07:01:48 +0000 Wim Taymans <wim.taymans@gmail.com>
23783 gst/playback/gstplaybasebin.c: Refactor handling of overrun detection.
23784 Original commit message from CVS:
23785 * gst/playback/gstplaybasebin.c: (group_commit),
23786 (queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
23787 (queue_out_of_data), (gen_preroll_element),
23788 (preroll_remove_overrun), (probe_triggered):
23789 Refactor handling of overrun detection.
23790 Separate handling of group completion and deadlock detection when doing
23791 network buffering. This should fix some deadlocks that were not detected
23792 because the group was completed.
23793 Add more comments, improve debugging.
23795 2006-09-21 05:31:00 +0000 Wim Taymans <wim.taymans@gmail.com>
23797 tests/check/: Some more compilation fixes.
23798 Original commit message from CVS:
23799 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
23800 * tests/check/libs/audio.c:
23801 Some more compilation fixes.
23803 2006-09-21 05:12:18 +0000 Wim Taymans <wim.taymans@gmail.com>
23805 gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
23806 Original commit message from CVS:
23807 * gst-libs/gst/audio/gstringbuffer.c:
23808 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
23809 (gst_ring_buffer_read):
23810 Early morning compilation fix.
23812 2006-09-20 18:09:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23816 Original commit message from CVS:
23819 2006-09-20 17:04:57 +0000 Wim Taymans <wim.taymans@gmail.com>
23821 tests/check/: Fix some warnings.
23822 Original commit message from CVS:
23823 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
23824 * tests/check/elements/multifdsink.c: (GST_START_TEST):
23825 * tests/check/elements/videorate.c: (GST_START_TEST):
23826 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
23827 * tests/check/pipelines/oggmux.c: (eos_buffer_probe):
23830 2006-09-20 10:59:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23832 sys/xvimage/xvimagesink.c: change colorkey behaviour back according to #354773 comment 6/7
23833 Original commit message from CVS:
23834 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
23835 (gst_xvimagesink_get_times):
23836 change colorkey behaviour back according to #354773 comment 6/7
23838 2006-09-20 10:42:34 +0000 Tim-Philipp Müller <tim@centricular.net>
23841 ChangeLog surgery: remove junk
23842 Original commit message from CVS:
23843 ChangeLog surgery: remove junk
23845 2006-09-19 11:31:06 +0000 Michael Smith <msmith@xiph.org>
23847 gst/tcp/gstmultifdsink.*: Implement stubbed out properties unit-type, units-soft-max, units-max, to allow specifying ...
23848 Original commit message from CVS:
23849 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
23850 (gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
23851 (gst_multi_fd_sink_recover_client),
23852 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
23853 (gst_multi_fd_sink_get_property):
23854 * gst/tcp/gstmultifdsink.h:
23855 Implement stubbed out properties unit-type, units-soft-max,
23856 units-max, to allow specifying maximum sizes in units other than
23860 2006-09-19 10:23:03 +0000 Wim Taymans <wim.taymans@gmail.com>
23862 gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
23863 Original commit message from CVS:
23864 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
23865 (gst_riff_create_audio_template_caps):
23866 Reorder the audio formats a bit for clarity.
23867 Detect and create caps for MSGSM and MSN (WAV49).
23869 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
23870 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
23871 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
23872 Small cleanups, move error handling out of normal flow for clarity.
23874 2006-09-18 15:59:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23876 Add new interface to control video orientation (fixes #354908)
23877 Original commit message from CVS:
23878 * docs/libs/gst-plugins-base-libs-docs.sgml:
23879 * docs/libs/gst-plugins-base-libs.types:
23880 * gst-libs/gst/interfaces/Makefile.am:
23881 * gst-libs/gst/interfaces/videoorientation.c:
23882 (gst_video_orientation_get_type),
23883 (gst_video_orientation_iface_init),
23884 (gst_video_orientation_get_hflip),
23885 (gst_video_orientation_get_vflip),
23886 (gst_video_orientation_get_hcenter),
23887 (gst_video_orientation_get_vcenter),
23888 (gst_video_orientation_set_hflip),
23889 (gst_video_orientation_set_vflip),
23890 (gst_video_orientation_set_hcenter),
23891 (gst_video_orientation_set_vcenter):
23892 * gst-libs/gst/interfaces/videoorientation.h:
23893 Add new interface to control video orientation (fixes #354908)
23895 2006-09-18 15:48:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23897 gst/videotestsrc/gstvideotestsrc.c: Use G_UNLIKELY in _create and log one more detail.
23898 Original commit message from CVS:
23899 * gst/videotestsrc/gstvideotestsrc.c:
23900 Use G_UNLIKELY in _create and log one more detail.
23901 (gst_video_test_src_get_times), (gst_video_test_src_create):
23902 * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
23903 Use gst_util_uint64_scale_int in _get_times().
23905 2006-09-18 15:00:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23907 sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
23908 Original commit message from CVS:
23909 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
23910 Give better warning message (add object and detail).
23912 2006-09-18 14:42:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23914 sys/xvimage/xvimagesink.c: xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes #354773), use gst_util...
23915 Original commit message from CVS:
23916 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
23917 (gst_xvimagesink_get_times):
23918 xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
23919 #354773), use gst_util_uint64_scale_int in _get_times()
23921 2006-09-18 14:21:45 +0000 Michael Smith <msmith@xiph.org>
23923 ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
23924 Original commit message from CVS:
23925 * ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
23926 Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
23927 always true, leading to dropping all timestamps.
23929 2006-09-18 11:40:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23931 ext/libvisual/visual.c: update to work also with libvisual 0.4 API
23932 Original commit message from CVS:
23933 * ext/libvisual/visual.c: (gst_vis_src_negotiate),
23934 (gst_visual_chain), (gst_visual_change_state):
23935 update to work also with libvisual 0.4 API
23936 * tools/gst-launch-ext.1.in:
23937 * tools/gst-visualise.1.in:
23938 remove references to old man-pages
23939 * tests/examples/seek/seek.c: (main):
23940 add real meadi-buttons, add tool-tips for the seek-options, arrange
23941 seek options in a table
23943 2006-09-18 10:57:28 +0000 Michael Smith <msmith@xiph.org>
23945 ext/ogg/gstoggmux.c: Don't generate out-of-order timestamps from oggmux, instead clamp output timestamps to be >= the...
23946 Original commit message from CVS:
23947 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
23948 (gst_ogg_mux_push_buffer):
23949 Don't generate out-of-order timestamps from oggmux, instead clamp
23950 output timestamps to be >= the previously output ts.
23953 2006-09-18 10:18:22 +0000 Michael Smith <msmith@xiph.org>
23955 gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
23956 Original commit message from CVS:
23957 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
23958 (gst_multi_fd_sink_class_init):
23959 Updates, fixes, and typo corrections for multifdsink. No functional
23962 2006-09-17 21:58:06 +0000 Michael Smith <msmith@xiph.org>
23964 gst/typefind/gsttypefindfunctions.c: Don't crash on truncated files - check that we got an 8 byte buffer before tryin...
23965 Original commit message from CVS:
23966 * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
23967 Don't crash on truncated files - check that we got an 8 byte buffer
23968 before trying to memcmp it.
23970 2006-09-17 20:32:09 +0000 Tim-Philipp Müller <tim@centricular.net>
23972 gst/playback/gstplaybasebin.c: Make stream-switching appear instant to the application (ie. make sure that a g_object...
23973 Original commit message from CVS:
23974 * gst/playback/gstplaybasebin.c: (get_active_source):
23975 Make stream-switching appear instant to the application
23976 (ie. make sure that a g_object_get on 'current-foo' returns
23977 the stream previously set with g_object_set(). Totem needs
23978 this to update stream-related meta-info (like audio-codec)
23979 correctly when switching streams.
23981 2006-09-17 20:14:43 +0000 Tim-Philipp Müller <tim@centricular.net>
23983 ext/alsa/gstalsamixer.c: Try harder to guess which mixer track is the master mixer track (instead of just taking the ...
23984 Original commit message from CVS:
23985 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
23986 (gst_alsa_mixer_ensure_track_list):
23987 Try harder to guess which mixer track is the master mixer
23988 track (instead of just taking the first one that has a pvolume).
23991 2006-09-17 11:24:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23993 gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
23994 Original commit message from CVS:
23995 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
23996 (gst_audio_convert_transform_caps):
23997 Get structure-name just once.
23999 2006-09-16 22:30:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24001 tests/check/: Fix big batch of compiler warnings.
24002 Original commit message from CVS:
24003 * tests/check/elements/audioresample.c: (GST_START_TEST):
24004 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
24005 * tests/check/elements/volume.c: (GST_START_TEST):
24006 * tests/check/elements/vorbisdec.c: (GST_START_TEST):
24007 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
24008 (test_pipeline), (GST_START_TEST):
24009 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
24010 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
24011 Fix big batch of compiler warnings.
24013 2006-09-16 21:54:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24015 ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
24016 Original commit message from CVS:
24017 * ext/gnomevfs/gstgnomevfssrc.c:
24018 Add docs about icydemux usage in connection with gnomevfssrc
24019 * ext/libvisual/visual.c:
24020 * ext/ogg/gstoggaviparse.c:
24021 * ext/ogg/gstoggdemux.c:
24022 * ext/ogg/gstoggmux.c:
24023 * ext/ogg/gstoggparse.c:
24024 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
24025 * gst-libs/gst/audio/gstaudiosink.c:
24026 * gst-libs/gst/audio/gstaudiosrc.c:
24027 * gst/audiorate/gstaudiorate.c:
24028 More G_OBJECT macro fixing.
24029 * gst/audiotestsrc/gstaudiotestsrc.h:
24030 Fix wrong info in header due to copy & paste
24032 2006-09-15 14:53:44 +0000 Wim Taymans <wim.taymans@gmail.com>
24034 gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
24035 Original commit message from CVS:
24036 * gst-libs/gst/audio/gstbaseaudiosink.c:
24037 (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
24038 * gst-libs/gst/audio/gstbaseaudiosrc.c:
24039 (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
24040 (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
24041 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
24042 Do the delay calculation in the source/sink base classes as this is
24043 specific for the capture/playback mode.
24044 Try to fixate a bit better, like round depth up to a multiple of 8
24046 Handle underruns correctly by marking DISCONT on buffers and adjusting
24047 timestamps to handle the gap.
24048 Set offset/offset_end correctly on buffers.
24049 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
24050 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
24051 (gst_ring_buffer_read):
24052 Remove resync and underrun recovery from the ringbuffer.
24053 Fix ringbuffer read code on under/overrun.
24055 2006-09-15 11:17:02 +0000 Wim Taymans <wim.taymans@gmail.com>
24057 gst/playback/gstplaybasebin.*: Don't use a 0 low watermark when buffering, it is catching starvation way too late. In...
24058 Original commit message from CVS:
24059 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
24060 (gst_play_base_bin_init), (fill_buffer), (check_queue),
24061 (queue_threshold_reached), (gst_play_base_bin_set_property),
24062 (gst_play_base_bin_get_property):
24063 * gst/playback/gstplaybasebin.h:
24064 Don't use a 0 low watermark when buffering, it is catching starvation
24065 way too late. Instead, use a 3 second queue with 30 and 95
24066 percent low/high watermarks.
24067 Added queue-min-threshold property to configure low watermark.
24068 Use new _buffering message API.
24069 Make queue_threshold variable big enough to store a uint64 time value.
24070 API: playbin::queue-min-threshold property.
24072 2006-09-15 09:13:50 +0000 Wim Taymans <wim.taymans@gmail.com>
24074 configure.ac: We require 0.10.10.1 now because of _wait_preroll().
24075 Original commit message from CVS:
24077 We require 0.10.10.1 now because of _wait_preroll().
24078 * gst-libs/gst/audio/gstbaseaudiosink.c:
24079 (gst_base_audio_sink_render):
24080 Use gst_base_sink_wait_preroll().
24082 2006-09-15 09:09:00 +0000 Wim Taymans <wim.taymans@gmail.com>
24084 ext/alsa/: Use DEBUG_OBJECT more.
24085 Original commit message from CVS:
24086 * ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
24087 * ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
24088 Use DEBUG_OBJECT more.
24090 === release 0.10.10 ===
24092 2006-09-14 20:09:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24099 * docs/plugins/gst-plugins-base-plugins.args:
24100 * docs/plugins/inspect/plugin-adder.xml:
24101 * docs/plugins/inspect/plugin-alsa.xml:
24102 * docs/plugins/inspect/plugin-audioconvert.xml:
24103 * docs/plugins/inspect/plugin-audiorate.xml:
24104 * docs/plugins/inspect/plugin-audioresample.xml:
24105 * docs/plugins/inspect/plugin-audiotestsrc.xml:
24106 * docs/plugins/inspect/plugin-cdparanoia.xml:
24107 * docs/plugins/inspect/plugin-decodebin.xml:
24108 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
24109 * docs/plugins/inspect/plugin-gdp.xml:
24110 * docs/plugins/inspect/plugin-gnomevfs.xml:
24111 * docs/plugins/inspect/plugin-libvisual.xml:
24112 * docs/plugins/inspect/plugin-ogg.xml:
24113 * docs/plugins/inspect/plugin-pango.xml:
24114 * docs/plugins/inspect/plugin-playbin.xml:
24115 * docs/plugins/inspect/plugin-subparse.xml:
24116 * docs/plugins/inspect/plugin-tcp.xml:
24117 * docs/plugins/inspect/plugin-theora.xml:
24118 * docs/plugins/inspect/plugin-typefindfunctions.xml:
24119 * docs/plugins/inspect/plugin-video4linux.xml:
24120 * docs/plugins/inspect/plugin-videorate.xml:
24121 * docs/plugins/inspect/plugin-videoscale.xml:
24122 * docs/plugins/inspect/plugin-videotestsrc.xml:
24123 * docs/plugins/inspect/plugin-volume.xml:
24124 * docs/plugins/inspect/plugin-vorbis.xml:
24125 * docs/plugins/inspect/plugin-ximagesink.xml:
24126 * docs/plugins/inspect/plugin-xvimagesink.xml:
24127 * ext/theora/theoraparse.c:
24128 * gst-libs/gst/rtp/gstrtpbuffer.c:
24129 * gst/playback/gstplaybin.c:
24130 * tests/check/Makefile.am:
24131 * win32/common/config.h:
24133 Original commit message from CVS:
24136 2006-09-09 16:08:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24139 * win32/common/config.h:
24141 Original commit message from CVS:
24144 2006-09-07 19:01:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24147 update bug in changelog
24148 Original commit message from CVS:
24149 update bug in changelog
24151 2006-09-07 19:00:33 +0000 Michael Smith <msmith@fluendo.com>
24153 Fix implementation of sync-method 'next-keyframe'
24154 Original commit message from CVS:
24155 patch by: Michael Smith <msmith at fluendo dot com>
24156 * gst/tcp/gstmultifdsink.c: (is_sync_frame),
24157 (gst_multi_fd_sink_client_queue_buffer),
24158 (gst_multi_fd_sink_new_client):
24159 * tests/check/elements/multifdsink.c: (GST_START_TEST),
24160 (multifdsink_suite):
24161 Fix implementation of sync-method 'next-keyframe'
24163 2006-09-07 15:00:08 +0000 Wim Taymans <wim@fluendo.com>
24165 ext/gnomevfs/gstgnomevfssrc.c: This patch removes the RANDOM flag that was incorrectly introduced with revision 1.91....
24166 Original commit message from CVS:
24167 patch by: Wim Taymans <wim at fluendo dot com>
24168 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
24169 This patch removes the RANDOM flag that was incorrectly introduced with
24170 revision 1.91. Fixes #354590
24172 2006-09-07 14:56:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24175 * win32/common/config.h:
24177 Original commit message from CVS:
24180 2006-09-07 14:56:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24197 Original commit message from CVS:
24200 2006-09-05 09:12:25 +0000 Tim-Philipp Müller <tim@centricular.net>
24202 tests/check/Makefile.am: Random variation in Makefile line to see if it makes the gen64-base-full bot any happier.
24203 Original commit message from CVS:
24204 * tests/check/Makefile.am:
24205 Random variation in Makefile line to see if it makes the
24206 gen64-base-full bot any happier.
24208 2006-09-04 19:04:35 +0000 Tim-Philipp Müller <tim@centricular.net>
24210 tests/check/pipelines/oggmux.c: Disable test that fails at the moment (killed after timeout).
24211 Original commit message from CVS:
24212 * tests/check/pipelines/oggmux.c: (oggmux_suite):
24213 Disable test that fails at the moment (killed after timeout).
24215 2006-09-04 18:19:06 +0000 James Livingston <doclivingston@gmail.com>
24217 tests/check/: Add simple unit test for oggmux from #337026 with checking for the
24218 Original commit message from CVS:
24219 Patch by: James Livingston <doclivingston at gmail.com>
24220 * tests/check/Makefile.am:
24221 * tests/check/pipelines/.cvsignore:
24222 * tests/check/pipelines/oggmux.c: (get_page_codec),
24223 (check_chain_final_state), (fail_if_audio), (validate_ogg_page),
24224 (eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
24225 (test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
24226 (test_theora_vorbis), (oggmux_suite):
24227 Add simple unit test for oggmux from #337026 with checking for the
24228 EOS flags disabled for the time being.
24230 2006-09-04 09:13:01 +0000 Alessandro Dessina <alessandro@nnva.org>
24232 ext/ogg/gstoggmux.c: Add cmml caps to oggmux. Fixes #353912
24233 Original commit message from CVS:
24234 patch by: Alessandro Dessina <alessandro nnva org>
24235 * ext/ogg/gstoggmux.c:
24236 Add cmml caps to oggmux. Fixes #353912
24238 2006-09-02 13:20:59 +0000 Tim-Philipp Müller <tim@centricular.net>
24240 tests/check/elements/videotestsrc.c: Returning a return value often helps. In this case, we don't need the return val...
24241 Original commit message from CVS:
24242 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
24243 Returning a return value often helps. In this case, we
24244 don't need the return value anyway, so just get rid of it.
24245 Should make build bots much happier.
24247 2006-09-02 12:59:48 +0000 Tim-Philipp Müller <tim@centricular.net>
24249 gst/videotestsrc/videotestsrc.*: Add support for AYUV and the various RGBA formats. Initialise fields of paintinfo st...
24250 Original commit message from CVS:
24251 * gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
24252 (paint_get_structure), (gst_video_test_src_get_size),
24253 (gst_video_test_src_smpte), (gst_video_test_src_snow),
24254 (gst_video_test_src_unicolor), (paint_setup_AYUV),
24255 (paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
24256 (paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
24257 * gst/videotestsrc/videotestsrc.h:
24258 Add support for AYUV and the various RGBA formats. Initialise
24259 fields of paintinfo structs allocated on the stack.
24260 * tests/check/elements/videotestsrc.c: (right_shift_colour),
24261 (fix_expected_colour), (check_rgb_buf), (got_buf_cb),
24262 (GST_START_TEST), (videotestsrc_suite):
24263 Add unit tests for videotestsrc's RGB output.
24265 2006-09-01 16:12:35 +0000 Tim-Philipp Müller <tim@centricular.net>
24267 gst/videotestsrc/: Add more uni-colour patterns ("white", "red", "green", and "blue").
24268 Original commit message from CVS:
24269 * gst/videotestsrc/gstvideotestsrc.c:
24270 (gst_video_test_src_pattern_get_type),
24271 (gst_video_test_src_set_pattern):
24272 * gst/videotestsrc/gstvideotestsrc.h:
24273 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
24274 (gst_video_test_src_black), (gst_video_test_src_white),
24275 (gst_video_test_src_red), (gst_video_test_src_green),
24276 (gst_video_test_src_blue):
24277 * gst/videotestsrc/videotestsrc.h:
24278 Add more uni-colour patterns ("white", "red", "green", and "blue").
24280 2006-09-01 10:07:05 +0000 Tim-Philipp Müller <tim@centricular.net>
24282 gst/videotestsrc/videotestsrc.c: Fix stride for YVYU, should be word-aligned (#353658).
24283 Original commit message from CVS:
24284 * gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
24285 Fix stride for YVYU, should be word-aligned (#353658).
24287 2006-08-31 14:37:33 +0000 Tim-Philipp Müller <tim@centricular.net>
24289 gst/adder/gstadder.c: Fix build.
24290 Original commit message from CVS:
24291 * gst/adder/gstadder.c: (gst_adder_src_event):
24294 2006-08-31 12:39:17 +0000 Edward Hervey <bilboed@bilboed.com>
24296 gst/adder/gstadder.*: Remember the start position asked in the incoming seeks, so we can output GST_EVENT_NEW_SEGMENT...
24297 Original commit message from CVS:
24298 * gst/adder/gstadder.c: (forward_event_func),
24299 (gst_adder_src_event), (gst_adder_collected),
24300 (gst_adder_change_state):
24301 * gst/adder/gstadder.h:
24302 Remember the start position asked in the incoming seeks, so we can
24303 output GST_EVENT_NEW_SEGMENT with a correct position value (instead
24304 of assuming it will always be 0).
24306 2006-08-31 12:31:00 +0000 Edward Hervey <bilboed@bilboed.com>
24308 ext/ogg/gstoggdemux.c: Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
24309 Original commit message from CVS:
24310 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
24311 (gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
24312 (gst_ogg_demux_loop):
24313 Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
24315 2006-08-30 17:22:27 +0000 Tim-Philipp Müller <tim@centricular.net>
24317 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Return FALSE instead of returning a random false unit size when the forma...
24318 Original commit message from CVS:
24319 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
24320 (gst_ffmpegcsp_get_unit_size):
24321 Return FALSE instead of returning a random false unit
24322 size when the format isn't known/supported (even if
24323 this shouldn't happen under normal circumstances).
24325 2006-08-29 15:23:46 +0000 Tim-Philipp Müller <tim@centricular.net>
24327 ext/gnomevfs/gstgnomevfssrc.c: Try harder to get the size from a uri by using _info_uri() when _info_from_handle() do...
24328 Original commit message from CVS:
24329 Patch by: Tim-Philipp Müller <tim at centricular dot net>
24330 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
24331 (gst_gnome_vfs_src_start):
24332 Try harder to get the size from a uri by using _info_uri() when
24333 _info_from_handle() does not give us enough info.
24334 Also follow symlinks when getting the size.
24335 Partially Fixes #332864.
24337 2006-08-29 11:50:51 +0000 Viktor Peters <viktor.peters@gmail.com>
24339 ext/alsa/: Improve and fix mixer track handling, in particular better handling of alsa's pvolume/pswitch/cvolume/cswi...
24340 Original commit message from CVS:
24341 Patch by: Viktor Peters <viktor dot peters at gmail dot com>
24342 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
24343 (gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
24344 (gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
24345 (gst_alsa_mixer_set_record):
24346 * ext/alsa/gstalsamixertrack.c:
24347 (gst_alsa_mixer_track_update_alsa_capabilities),
24348 (alsa_track_has_cap), (gst_alsa_mixer_track_new),
24349 (gst_alsa_mixer_track_update):
24350 * ext/alsa/gstalsamixertrack.h:
24351 Improve and fix mixer track handling, in particular better handling
24352 of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
24353 track objects for tracks that have both capture and playback volume
24354 (and label them differently as well so they're not mistakenly
24355 assumed to be duplicates); classify mixer tracks that only affect
24356 the audible volume of something (rather than the capture volume)
24357 as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
24358 for capture tracks to correspond to alsa-pswitch alsa-cswitch
24359 (following the meaning documented in the mixer interface header
24360 file); add support for alsa's exclusive cswitch groups; update/sync
24361 state/flags better if mixer settings are changed by another
24362 application. Fixes #336075.
24364 2006-08-29 10:58:43 +0000 Tim-Philipp Müller <tim@centricular.net>
24366 gst/playback/gstplaybin.c: Improve docs: add section about BUFFERING messages sent by playbin.
24367 Original commit message from CVS:
24368 * gst/playback/gstplaybin.c:
24369 Improve docs: add section about BUFFERING messages sent by playbin.
24371 2006-08-29 10:51:12 +0000 Michael Smith <msmith@xiph.org>
24373 ext/vorbis/vorbisenc.c: Ignore explicit DISCONT marked on buffers (which is often spurious, particularly when using m...
24374 Original commit message from CVS:
24375 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
24376 (gst_vorbis_enc_buffer_check_discontinuous),
24377 (gst_vorbis_enc_chain):
24378 Ignore explicit DISCONT marked on buffers (which is often spurious,
24379 particularly when using multiple segments), in favour of solely
24380 using the timestamps/durations.
24382 2006-08-29 10:32:34 +0000 Edward Hervey <bilboed@bilboed.com>
24384 gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
24385 Original commit message from CVS:
24386 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
24387 Don't rely on incoming buffers offset anymore, since it is completely
24388 broken when using multiple segments.
24389 Instead convert the incoming buffers timestamp to running time, and
24390 then convert that value to the offsets.
24391 Also inform GstSegment of the last outputted stop position, which is
24392 needed if we received several segments with an unknown stop value.
24394 2006-08-29 08:03:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24396 ext/vorbis/vorbisenc.c: fix buffer unreffing on a header push failure
24397 Original commit message from CVS:
24398 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
24399 fix buffer unreffing on a header push failure
24401 2006-08-28 16:17:13 +0000 Wim Taymans <wim.taymans@gmail.com>
24403 gst/audiorate/gstaudiorate.c: Make the metadata of the buffer writable before changing its flags.
24404 Original commit message from CVS:
24405 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
24406 (gst_audio_rate_chain):
24407 Make the metadata of the buffer writable before changing its
24410 2006-08-28 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
24413 Fix changelog with bugzilla bug it fixed.
24414 Original commit message from CVS:
24415 Fix changelog with bugzilla bug it fixed.
24417 2006-08-28 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
24419 gst/audiorate/gstaudiorate.c: Fix audiorate some more.
24420 Original commit message from CVS:
24421 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
24422 (gst_audio_rate_setcaps), (gst_audio_rate_init),
24423 (gst_audio_rate_sink_event), (gst_audio_rate_src_event),
24424 (gst_audio_rate_chain), (gst_audio_rate_change_state):
24425 Fix audiorate some more.
24426 Reset and resync counters on flush and READY.
24427 Handle the DISCONT flag correctly.
24428 Use GstSegment to track position.
24429 Fail when not negotiated.
24431 2006-08-25 16:48:28 +0000 Michael Smith <msmith@xiph.org>
24433 gst/tcp/gstmultifdsink.c: Fix spelling.
24434 Original commit message from CVS:
24435 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
24437 Remove accidently included debug line.
24439 2006-08-25 16:39:38 +0000 Wim Taymans <wim.taymans@gmail.com>
24441 gst/tcp/gstmultifdsink.c: Small cleanups.
24442 Original commit message from CVS:
24443 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
24445 If a buffer is received with no caps, make the buffer metadata
24446 writable and set the caps, making sure that we don't screw up the
24449 2006-08-25 16:19:55 +0000 Michael Smith <msmith@xiph.org>
24451 gst/gdp/gstgdppay.c: Fix memory leaks and misleading debug messages, add a couple of comments.
24452 Original commit message from CVS:
24453 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
24454 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
24455 Fix memory leaks and misleading debug messages, add a couple of
24457 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
24458 (gst_multi_fd_sink_render):
24459 Do not use gst_buffer_make_writable() in a basesink render method,
24460 as it may incorrectly unref the buffer. Instead, use convoluted
24461 dance to avoid copying the buffer except when we need to.
24463 2006-08-25 09:54:56 +0000 Michael Smith <msmith@xiph.org>
24465 ext/vorbis/vorbisenc.c: Allow very small discontinuities in the timestamps. These we can't do anything useful with an...
24466 Original commit message from CVS:
24467 * ext/vorbis/vorbisenc.c:
24468 (gst_vorbis_enc_buffer_check_discontinuous):
24469 Allow very small discontinuities in the timestamps. These we can't
24470 do anything useful with anyway (because vorbis's timestamps have
24471 only sample granularity), and are commonly produced by elements with
24472 minor bugs. Allow up to 1/2 a sample out.
24475 2006-08-24 11:18:56 +0000 Wim Taymans <wim.taymans@gmail.com>
24477 tests/examples/seek/seek.c: Add a checkbox to enable play scrubbing. Makes it possible to disable normal scrubbing.
24478 Original commit message from CVS:
24479 * tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
24480 (play_scrub_toggle_cb), (main):
24481 Add a checkbox to enable play scrubbing. Makes it possible to disable
24484 2006-08-23 19:37:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24486 tests/check/elements/.cvsignore: make buildbot happy
24487 Original commit message from CVS:
24488 * tests/check/elements/.cvsignore:
24489 make buildbot happy
24491 2006-08-23 16:43:03 +0000 Tim-Philipp Müller <tim@centricular.net>
24493 ext/ogg/gstogmparse.c: Refactor ogm parse, do better input checking, misc. clean-ups.
24494 Original commit message from CVS:
24495 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
24496 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
24497 (gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
24498 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
24499 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
24500 (gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
24501 (gst_ogm_text_parse_strip_trailing_zeroes),
24502 (gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
24503 (gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
24504 Refactor ogm parse, do better input checking, misc. clean-ups.
24505 Cache incoming events and push them once the source pad has
24506 been created. Don't pass unterminated strings to sscanf().
24507 Strip trailing zeroes from subtitle text output, since they
24508 are not valid UTF-8. Don't push vorbiscomment packets on
24509 the subtitle text pad. Output perfect streams if possible.
24511 2006-08-23 15:27:38 +0000 Wim Taymans <wim.taymans@gmail.com>
24513 tests/check/libs/cddabasesrc.c: Waits for tasks to settle down so that we clean up correctly for valgrind.
24514 Original commit message from CVS:
24515 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
24516 Waits for tasks to settle down so that we clean up correctly for
24519 2006-08-23 15:11:56 +0000 Tim-Philipp Müller <tim@centricular.net>
24521 tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
24522 Original commit message from CVS:
24523 * tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
24524 Unit test fixes: \377 is more likely to fit into 8 bits than \777;
24525 actually return return value in taglists_are_equal.
24527 2006-08-23 12:14:20 +0000 Tim-Philipp Müller <tim@centricular.net>
24529 ext/ogg/gstogmparse.c: Fix crash due to broken bitstream parsing on x86-64: can't make any assumptions about sizeof(s...
24530 Original commit message from CVS:
24531 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
24532 Fix crash due to broken bitstream parsing on x86-64: can't make
24533 any assumptions about sizeof(struct) due to alignment/packing
24534 differences on different architectures. Fixes #351790.
24536 2006-08-22 16:31:47 +0000 Wim Taymans <wim.taymans@gmail.com>
24538 gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
24539 Original commit message from CVS:
24540 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
24541 (gst_riff_parse_chunk), (gst_riff_parse_file_header),
24542 (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
24543 (gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
24544 (gst_riff_parse_info):
24545 Protect public functions against bad input.
24549 2006-08-22 15:50:36 +0000 Tim-Philipp Müller <tim@centricular.net>
24551 gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
24552 Original commit message from CVS:
24553 * gst-libs/gst/riff/riff-ids.h:
24554 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
24555 Add voxware audio IDs (even if we can't play it) (#351795).
24557 2006-08-22 15:11:52 +0000 Tim-Philipp Müller <tim@centricular.net>
24559 gst-libs/gst/riff/riff-media.c: Const-ify some arrays and use G_N_ELEMENTS instead of wasting oodles of RAM on termin...
24560 Original commit message from CVS:
24561 * gst-libs/gst/riff/riff-media.c:
24562 (gst_riff_create_video_template_caps),
24563 (gst_riff_create_audio_template_caps),
24564 (gst_riff_create_iavs_template_caps):
24565 Const-ify some arrays and use G_N_ELEMENTS instead
24566 of wasting oodles of RAM on terminator bits.
24568 2006-08-22 08:27:07 +0000 Tim-Philipp Müller <tim@centricular.net>
24570 And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
24571 Original commit message from CVS:
24572 * gst-libs/gst/tag/gstvorbistag.c:
24573 (gst_tag_list_to_vorbiscomment_buffer):
24574 * tests/check/libs/tag.c: (GST_START_TEST):
24575 And the same for _to_vorbiscomment_buffer(): allow
24576 id_data_len == 0 for speex.
24578 2006-08-21 19:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24582 Original commit message from CVS:
24585 2006-08-21 19:01:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24587 Move GDP plugin to -base from -bad. Closes #347783.
24588 Original commit message from CVS:
24590 * docs/plugins/Makefile.am:
24591 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
24592 * docs/plugins/gst-plugins-base-plugins-sections.txt:
24593 * docs/plugins/inspect/plugin-gdp.xml:
24594 * gst/gdp/Makefile.am:
24595 * tests/check/Makefile.am:
24596 Move GDP plugin to -base from -bad. Closes #347783.
24598 2006-08-21 18:34:46 +0000 Tim-Philipp Müller <tim@centricular.net>
24600 gst-libs/gst/tag/gstvorbistag.c: Allow id_data_len == 0 (needed for vorbis comments in Speex files).
24601 Original commit message from CVS:
24602 * gst-libs/gst/tag/gstvorbistag.c:
24603 (gst_tag_list_from_vorbiscomment_buffer):
24604 Allow id_data_len == 0 (needed for vorbis comments in Speex files).
24605 Also add some checks to make sure we don't memcmp() beyond the end of
24606 vorbiscomment buffer if the ID to check for is larger than the buffer.
24607 * tests/check/libs/tag.c: (GST_START_TEST):
24608 Some more tests for gst_tag_list_from_vorbiscomment_buffer().
24610 2006-08-21 16:39:25 +0000 Tim-Philipp Müller <tim@centricular.net>
24612 ext/vorbis/vorbisenc.c: Use vorbis comment utility functions from libgsttag instead of re-inventing the wheel (partia...
24613 Original commit message from CVS:
24614 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
24615 (gst_vorbis_enc_set_metadata):
24616 Use vorbis comment utility functions from libgsttag
24617 instead of re-inventing the wheel (partially fixes #347091).
24619 2006-08-21 11:42:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24621 tests/check/elements/audioconvert.c: Fix leaks. Wait for state transitions that might happen ASYNC, as well as some t...
24622 Original commit message from CVS:
24623 * tests/check/elements/audioconvert.c: (GST_START_TEST):
24624 Fix leaks. Wait for state transitions that might happen ASYNC, as well
24625 as some that won't.
24627 2006-08-21 10:32:51 +0000 Wim Taymans <wim.taymans@gmail.com>
24629 docs/libs/: Don't try to GObject scan the netbuffer as it's not a GObject.
24630 Original commit message from CVS:
24631 * docs/libs/Makefile.am:
24632 * docs/libs/gst-plugins-base-libs-sections.txt:
24633 * docs/libs/gst-plugins-base-libs.types:
24634 Don't try to GObject scan the netbuffer as it's not a GObject.
24636 * gst-libs/gst/netbuffer/gstnetbuffer.c:
24637 * gst-libs/gst/netbuffer/gstnetbuffer.h:
24638 Document GstNetBuffer.
24640 2006-08-21 08:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24642 tests/check/elements/audioconvert.c: Add testcase for caps-size-explosion
24643 Original commit message from CVS:
24644 * tests/check/elements/audioconvert.c: (GST_START_TEST),
24645 (audioconvert_suite):
24646 Add testcase for caps-size-explosion
24648 2006-08-20 13:05:44 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24650 gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size
24651 Original commit message from CVS:
24652 * gst/audioconvert/gstaudioconvert.c:
24653 (gst_audio_convert_get_unit_size), (set_structure_widths):
24654 Lower debug, use g_assert in _get_unit_size
24655 * gst/audioresample/gstaudioresample.c:
24656 (audioresample_get_unit_size):
24657 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
24658 (gst_ffmpegcsp_get_unit_size):
24659 * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
24660 use g_assert in _get_unit_size
24662 2006-08-18 21:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
24665 ChangeLog surgery: fix bug number
24666 Original commit message from CVS:
24667 ChangeLog surgery: fix bug number
24669 2006-08-18 16:43:26 +0000 Wim Taymans <wim.taymans@gmail.com>
24671 Document GstRTPBuffer.
24672 Original commit message from CVS:
24673 * docs/libs/gst-plugins-base-libs-sections.txt:
24674 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
24675 (gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
24676 (gst_rtp_buffer_get_payload_buffer):
24677 * gst-libs/gst/rtp/gstrtpbuffer.h:
24678 Document GstRTPBuffer.
24679 Added function to efficiently strip payload headers.
24680 API: gst_rtp_buffer_get_payload_subbuffer()
24682 2006-08-17 16:52:06 +0000 Tim-Philipp Müller <tim@centricular.net>
24684 gst-libs/gst/tag/gstvorbistag.c: Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT tags and deserialise...
24685 Original commit message from CVS:
24686 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
24687 (gst_tag_to_vorbis_comments):
24688 Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
24689 tags and deserialise them properly as well (#351768).
24690 Add some more gtk-doc blurbs and also some g_return_if_fail().
24691 * tests/check/libs/tag.c: (GST_START_TEST),
24692 (back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
24695 2006-08-17 15:43:40 +0000 Wim Taymans <wim.taymans@gmail.com>
24697 ext/ogg/: Added ogg-in-avi parser element. Fixes #140139.
24698 Original commit message from CVS:
24699 * ext/ogg/Makefile.am:
24700 * ext/ogg/gstogg.c: (plugin_init):
24701 * ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
24702 (gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
24703 (gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
24704 (gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
24705 (gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
24706 (gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
24707 Added ogg-in-avi parser element. Fixes #140139.
24708 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
24709 Fixed a bug in oggdemux debug code.
24710 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
24711 (gst_riff_create_audio_template_caps):
24712 Recognise Ogg in the AVI extensible wave format.
24714 2006-08-17 10:00:00 +0000 Tim-Philipp Müller <tim@centricular.net>
24716 gst-libs/gst/cdda/gstcddabasesrc.c: Make buffer durations add up (duration should be next_ts-ts for perfect streams)....
24717 Original commit message from CVS:
24718 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
24719 Make buffer durations add up (duration should be next_ts-ts for
24720 perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
24722 * tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
24723 (test_buffer_timestamps), (cddabasesrc_suite):
24724 Add unit test for the above.
24725 * tests/check/Makefile.am:
24726 Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
24727 to see what happens.
24729 2006-08-16 11:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
24731 ext/alsa/: Avoid setting and using a NULL device name.
24732 Original commit message from CVS:
24733 * ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
24734 (gst_alsasink_open):
24735 * ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
24736 (gst_alsasrc_open):
24737 Avoid setting and using a NULL device name.
24738 Print more info when we fail to open a device.
24740 2006-08-16 11:28:57 +0000 Tim-Philipp Müller <tim@centricular.net>
24742 API: add gst_tag_parse_extended_comment() (#351426).
24743 Original commit message from CVS:
24744 * docs/libs/gst-plugins-base-libs-sections.txt:
24745 * gst-libs/gst/tag/tag.h:
24746 * gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
24747 API: add gst_tag_parse_extended_comment() (#351426).
24748 * tests/check/Makefile.am:
24749 * tests/check/libs/.cvsignore:
24750 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
24751 Add unit test for gst_tag_parse_extended_comment().
24753 2006-08-15 19:20:16 +0000 Tim-Philipp Müller <tim@centricular.net>
24755 sys/: Fix leak (#351502).
24756 Original commit message from CVS:
24757 * sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
24758 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
24759 Fix leak (#351502).
24761 2006-08-15 17:21:33 +0000 Tim-Philipp Müller <tim@centricular.net>
24764 Original commit message from CVS:
24765 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
24766 * docs/plugins/gst-plugins-base-plugins-sections.txt:
24767 * docs/plugins/gst-plugins-base-plugins.args:
24768 * gst/playback/gstplaybin.c:
24770 * docs/plugins/inspect/plugin-adder.xml:
24771 * docs/plugins/inspect/plugin-alsa.xml:
24772 * docs/plugins/inspect/plugin-audioconvert.xml:
24773 * docs/plugins/inspect/plugin-audiorate.xml:
24774 * docs/plugins/inspect/plugin-audioresample.xml:
24775 * docs/plugins/inspect/plugin-audiotestsrc.xml:
24776 * docs/plugins/inspect/plugin-cdparanoia.xml:
24777 * docs/plugins/inspect/plugin-decodebin.xml:
24778 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
24779 * docs/plugins/inspect/plugin-gnomevfs.xml:
24780 * docs/plugins/inspect/plugin-ogg.xml:
24781 * docs/plugins/inspect/plugin-pango.xml:
24782 * docs/plugins/inspect/plugin-playbin.xml:
24783 * docs/plugins/inspect/plugin-subparse.xml:
24784 * docs/plugins/inspect/plugin-tcp.xml:
24785 * docs/plugins/inspect/plugin-theora.xml:
24786 * docs/plugins/inspect/plugin-typefindfunctions.xml:
24787 * docs/plugins/inspect/plugin-video4linux.xml:
24788 * docs/plugins/inspect/plugin-videorate.xml:
24789 * docs/plugins/inspect/plugin-videoscale.xml:
24790 * docs/plugins/inspect/plugin-videotestsrc.xml:
24791 * docs/plugins/inspect/plugin-volume.xml:
24792 * docs/plugins/inspect/plugin-vorbis.xml:
24793 * docs/plugins/inspect/plugin-ximagesink.xml:
24794 * docs/plugins/inspect/plugin-xvimagesink.xml:
24795 Update to CVS version.
24797 2006-08-14 17:54:01 +0000 Tim-Philipp Müller <tim@centricular.net>
24799 gst/playback/gstplaybin.c: Add "connection-speed" property; re-order redirect messages with multiple redirect locatio...
24800 Original commit message from CVS:
24801 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
24802 (gst_play_bin_set_property), (gst_play_bin_get_property),
24803 (value_list_append_structure_list),
24804 (gst_play_bin_handle_redirect_message),
24805 (gst_play_bin_handle_message):
24806 Add "connection-speed" property; re-order redirect messages with
24807 multiple redirect locations depending on the minimum bitrate if
24808 that information is available and a connection speed is set
24811 2006-08-14 11:41:04 +0000 Tim-Philipp Müller <tim@centricular.net>
24813 gst/playback/gstplaybin.c: Update max volume to the same value that the volume element uses.
24814 Original commit message from CVS:
24815 * gst/playback/gstplaybin.c:
24816 Update max volume to the same value that the volume element uses.
24818 2006-08-14 10:50:15 +0000 Wim Taymans <wim.taymans@gmail.com>
24820 ext/alsa/gstalsamixer.c: Less uglyness..
24821 Original commit message from CVS:
24822 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
24825 2006-08-14 10:49:10 +0000 Wim Taymans <wim.taymans@gmail.com>
24827 ext/ogg/gstoggdemux.c: Add some more debug info.
24828 Original commit message from CVS:
24829 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
24830 (gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
24831 (gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
24832 Add some more debug info.
24833 Don't crash when a seek failed.
24834 Actually return the result of the seek instead of TRUE.
24835 Ignore multiple BOS pages with the same serial so that we don't create
24836 the same stream multiple times.
24837 Post an error when we fail to do the initial seek.
24839 2006-08-13 14:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
24841 ext/alsa/gstalsa.c: Small code cleanup.
24842 Original commit message from CVS:
24843 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
24844 (gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
24845 Small code cleanup.
24846 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
24847 (gst_alsa_mixer_new):
24848 Remove hack that always set the device to hw:0*.
24849 Properly find the card name for whatever device was configured.
24850 Do some better debugging.
24852 * ext/alsa/gstalsamixerelement.c:
24853 (gst_alsa_mixer_element_set_property),
24854 (gst_alsa_mixer_element_change_state):
24856 Handle setting of a NULL device name better.
24858 2006-08-11 15:53:43 +0000 Wim Taymans <wim.taymans@gmail.com>
24860 gst/adder/gstadder.c: Don't clip float values. Fixes #350900.
24861 Original commit message from CVS:
24862 * gst/adder/gstadder.c:
24863 Don't clip float values. Fixes #350900.
24865 2006-08-11 15:33:17 +0000 Andy Wingo <wingo@pobox.com>
24867 gst/tcp/gsttcp.c: Really fix the build?
24868 Original commit message from CVS:
24869 2006-08-11 Andy Wingo <wingo@pobox.com>
24870 * gst/tcp/gsttcp.c: Really fix the build?
24872 2006-08-11 15:29:56 +0000 Andy Wingo <wingo@pobox.com>
24874 gst/tcp/gsttcp.h: For now, always disable deprecation here -- fixes the build.
24875 Original commit message from CVS:
24876 2006-08-11 Andy Wingo <wingo@pobox.com>
24877 * gst/tcp/gsttcp.h: For now, always disable deprecation here --
24880 2006-08-10 13:01:31 +0000 Tim-Philipp Müller <tim@centricular.net>
24882 gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field.
24883 Original commit message from CVS:
24884 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
24885 Float caps shouldn't have a "signed" field.
24887 2006-08-10 08:56:22 +0000 Tim-Philipp Müller <tim@centricular.net>
24889 ext/ogg/gstoggdemux.c: Implement SEEKING query in its most basic form, so that we can at least check if we're seekabl...
24890 Original commit message from CVS:
24891 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
24892 Implement SEEKING query in its most basic form, so that we can
24893 at least check if we're seekable or not (#350655).
24895 2006-08-09 14:42:58 +0000 Tim-Philipp Müller <tim@centricular.net>
24897 gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
24898 Original commit message from CVS:
24899 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
24900 The checks here are not even close to anything that would
24901 justify MAXIMUM probability, lowering to POSSIBLE until someone
24902 fixes the checks (case at hand: quicktime redirection files
24903 might start with 00 00 01 XX and pass the checks here just
24904 fine, see #350399).
24906 2006-08-08 13:57:29 +0000 Edward Hervey <bilboed@bilboed.com>
24908 tests/check/elements/gdpdepay.c: I forgot to include the file containing the #define :)
24909 Original commit message from CVS:
24910 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
24911 I forgot to include the file containing the #define :)
24912 Now includes "config.h"
24914 2006-08-08 13:45:44 +0000 Edward Hervey <bilboed@bilboed.com>
24916 tests/check/elements/gdpdepay.c: Ignore test known to fail on PPC64. See #348114.
24917 Original commit message from CVS:
24918 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
24919 Ignore test known to fail on PPC64. See #348114.
24921 2006-08-08 08:41:13 +0000 Sjoerd Simons <sjoerd@luon.net>
24923 gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
24924 Original commit message from CVS:
24925 Patch by: Sjoerd Simons <sjoerd at luon net>
24926 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
24927 Better detection for multipart/x-mixed-replace: accept leading
24928 whitespaces before the boundary marker as well (as our very own
24929 multipartmux used to produce) (#349068).
24931 2006-08-07 08:26:03 +0000 Young-Ho Cha <ganadist@chollian.net>
24933 gst-libs/gst/riff/: Detect DTS audio streams (#350157).
24934 Original commit message from CVS:
24935 Patch by: Young-Ho Cha <ganadist at chollian net>
24936 * gst-libs/gst/riff/riff-ids.h:
24937 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
24938 (gst_riff_create_audio_template_caps):
24939 Detect DTS audio streams (#350157).
24941 2006-08-05 17:08:05 +0000 Andy Wingo <wingo@pobox.com>
24943 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (gst_theora_parse_class_init) (theora_parse_dispose, theora_par...
24944 Original commit message from CVS:
24945 2006-08-05 Andy Wingo <wingo@pobox.com>
24946 * ext/theora/gsttheoraparse.h:
24947 * ext/theora/theoraparse.c (gst_theora_parse_class_init)
24948 (theora_parse_dispose, theora_parse_set_property)
24949 (theora_parse_get_property, theora_parse_munge_granulepos)
24950 (theora_parse_push_buffer, theora_parse_change_state): Add a
24951 property 'synchronization-points' to fix badly synchronized oggs.
24953 2006-08-04 13:20:23 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
24955 gst/gdp/gstgdpdepay.c: Fix event parsing by gdpdepay. Fixes #349916.
24956 Original commit message from CVS:
24957 2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
24958 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
24959 Fix event parsing by gdpdepay. Fixes #349916.
24961 2006-08-03 15:04:42 +0000 Tim-Philipp Müller <tim@centricular.net>
24963 tests/check/: Add a few tests for the channel position stuff in libgstaudio.
24964 Original commit message from CVS:
24965 * tests/check/Makefile.am:
24966 * tests/check/libs/.cvsignore:
24967 * tests/check/libs/audio.c: (structure_contains_channel_positions),
24968 (fixed_caps_have_channel_positions), (GST_START_TEST),
24969 (audio_suite), (main):
24970 Add a few tests for the channel position stuff in libgstaudio.
24972 2006-08-03 14:16:06 +0000 Tim-Philipp Müller <tim@centricular.net>
24974 ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
24975 Original commit message from CVS:
24976 * ext/alsa/gstalsa.c: (caps_add_channel_configuration),
24977 (gst_alsa_detect_channels):
24978 * ext/alsa/gstalsasink.c:
24979 Add support for cards that (only) do more than 8 channels,
24980 like the Delta 44 (#345188).
24981 * gst-libs/gst/audio/multichannel.c:
24982 (gst_audio_check_channel_positions):
24983 * gst-libs/gst/audio/multichannel.h:
24984 API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
24985 unspecified channel position and cannot be combined with any
24986 of the other audio channel positions; adjust position layout
24987 checks accordingly (#345188).
24989 2006-08-03 11:15:53 +0000 Tim-Philipp Müller <tim@centricular.net>
24991 gst/typefind/gsttypefindfunctions.c: Recognise ancient RealAudio files (see #349779).
24992 Original commit message from CVS:
24993 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
24994 Recognise ancient RealAudio files (see #349779).
24996 2006-08-03 09:01:25 +0000 Jens Granseuer <jensgr@gmx.net>
24998 gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
24999 Original commit message from CVS:
25000 Patch by: Jens Granseuer <jensgr at gmx net>
25001 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
25002 Add typefinder for Interplay's MVE format (#348973).
25004 2006-08-02 17:03:29 +0000 Marcel Moreaux <marcelm@luon.net>
25006 gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
25007 Original commit message from CVS:
25008 Patch by: Marcel Moreaux <marcelm at luon dot net>
25009 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25010 (gst_base_rtp_depayload_add_to_queue):
25011 * gst-libs/gst/rtp/gstbasertpdepayload.h:
25012 Handle RTP sequence number rollover.
25013 Disable jitterbuffer by default.
25015 2006-08-02 16:56:19 +0000 Wim Taymans <wim.taymans@gmail.com>
25017 gst/gdp/gstgdpdepay.c: Disable seeking.
25018 Original commit message from CVS:
25019 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
25020 (gst_gdp_depay_finalize), (gst_gdp_depay_sink_event),
25021 (gst_gdp_depay_src_event), (gst_gdp_depay_chain),
25022 (gst_gdp_depay_change_state):
25025 Clear adapter on disconts.
25026 Clear caps when going to READY instead of NULL
25027 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
25028 (gst_gdp_pay_init), (gst_gdp_pay_finalize), (gst_gdp_pay_reset),
25029 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
25030 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
25031 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
25032 (gst_gdp_pay_sink_event), (gst_gdp_pay_src_event),
25033 (gst_gdp_pay_change_state):
25034 * gst/gdp/gstgdppay.h:
25035 Reset payloader when going to READY.
25036 Fix leaked buffers in ->queue on push errors.
25039 Create packetizer in _init, free in _finalize.
25041 2006-07-31 08:48:36 +0000 Wim Taymans <wim.taymans@gmail.com>
25043 gst/gdp/gstgdpdepay.c: Consume all events except EOS because we generate events from the gdp payload instead. Fixes #...
25044 Original commit message from CVS:
25045 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
25046 (gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
25047 Consume all events except EOS because we generate events from
25048 the gdp payload instead. Fixes #349204
25050 2006-07-28 17:17:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25052 gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
25053 Original commit message from CVS:
25054 * gst/audioresample/gstaudioresample.c: (audioresample_stop),
25055 (audioresample_set_caps):
25056 Don't leak references to the incoming caps. Clean them up when
25058 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
25059 (gst_video_scale_finalize):
25060 Don't leak our temporary pixel buffer.
25061 * tests/check/Makefile.am:
25062 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
25063 (GST_START_TEST), (simple_launch_lines_suite):
25064 Fix leaks and re-enable the test for valgrind checking.
25066 2006-07-28 16:39:31 +0000 Sjoerd Simons <sjoerd@luon.net>
25068 gst/typefind/gsttypefindfunctions.c: Add typefind function for multipart/x-mixed-replace (#348916).
25069 Original commit message from CVS:
25070 Patch by: Sjoerd Simons <sjoerd at luon net>
25071 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
25073 Add typefind function for multipart/x-mixed-replace (#348916).
25075 2006-07-28 14:14:58 +0000 Wim Taymans <wim.taymans@gmail.com>
25077 gst/adder/gstadder.c: Fix leak in duration query.
25078 Original commit message from CVS:
25079 * gst/adder/gstadder.c: (gst_adder_setcaps),
25080 (gst_adder_query_duration):
25081 Fix leak in duration query.
25082 Reflow some docs and notes.
25084 2006-07-28 13:42:48 +0000 Michael Smith <msmith@xiph.org>
25086 tests/check/pipelines/vorbisenc.c: Enable Andy's extra vorbisenc test, now that it passes. Also fix one aspect of it.
25087 Original commit message from CVS:
25088 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
25090 Enable Andy's extra vorbisenc test, now that it passes. Also fix one
25093 2006-07-28 12:48:21 +0000 Michael Smith <msmith@xiph.org>
25095 ext/vorbis/vorbisenc.*: Handle discontinuities in the input vorbis stream correctly, so that the output is properly t...
25096 Original commit message from CVS:
25097 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
25098 (gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
25099 (gst_vorbis_enc_push_buffer),
25100 (gst_vorbis_enc_buffer_check_discontinuous),
25101 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
25102 * ext/vorbis/vorbisenc.h:
25103 Handle discontinuities in the input vorbis stream correctly,
25104 so that the output is properly timestamped (and has good granulepos
25105 values). Needs some oggmux fixes too.
25107 2006-07-27 10:52:52 +0000 Kai Vehmanen <kv2004@eca.cx>
25109 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
25110 Original commit message from CVS:
25111 patch by: Kai Vehmanen <kv2004 eca cx>
25112 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25113 (gst_base_rtp_depayload_chain),
25114 (gst_base_rtp_depayload_handle_sink_event),
25115 (gst_base_rtp_depayload_change_state):
25116 Don't send multiple newsegments with different formats.
25119 2006-07-26 15:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
25121 ext/ogg/gstoggdemux.c: Make seeking in ogg more accurate again by doing the more correct granuletime to stream time c...
25122 Original commit message from CVS:
25123 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
25124 (gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
25125 Make seeking in ogg more accurate again by doing the more correct
25126 granuletime to stream time conversion.
25128 2006-07-26 10:59:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25130 gst/tcp/gstmultifdsink.c: debug a little more understandably do not use goto as a substitute for break, especially if...
25131 Original commit message from CVS:
25132 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
25133 (gst_multi_fd_sink_new_client):
25134 debug a little more understandably
25135 do not use goto as a substitute for break, especially if
25136 break is also being used
25138 2006-07-26 10:55:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25140 * gst/tcp/gsttcp.c:
25141 move a recurring normal event to LOG, where it should be
25142 Original commit message from CVS:
25143 move a recurring normal event to LOG, where it should be
25145 2006-07-26 10:54:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25147 * ext/vorbis/vorbisdec.c:
25149 Original commit message from CVS:
25152 2006-07-26 10:52:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25154 gst/gdp/gstgdpdepay.c: proxying get/set caps is the wrong thing to do, since we really do change caps quite fundament...
25155 Original commit message from CVS:
25156 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
25157 proxying get/set caps is the wrong thing to do, since we really
25158 do change caps quite fundamentally
25159 * tests/check/elements/gdpdepay.c:
25160 * tests/check/elements/gdppay.c:
25161 remove declaration of buffers, it's already done in gstcheck.h
25163 2006-07-26 10:31:06 +0000 Tim-Philipp Müller <tim@centricular.net>
25165 gst/playback/: Remove GLib-2.6 compatibility cruft.
25166 Original commit message from CVS:
25167 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
25168 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
25169 Remove GLib-2.6 compatibility cruft.
25171 2006-07-24 16:47:10 +0000 Wim Taymans <wim.taymans@gmail.com>
25173 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
25174 Original commit message from CVS:
25175 * gst-libs/gst/audio/gstbaseaudiosink.c:
25176 (gst_base_audio_sink_render):
25177 Don't try to align a sample to an unknown value.
25179 2006-07-24 15:14:17 +0000 Wim Taymans <wim.taymans@gmail.com>
25181 gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
25182 Original commit message from CVS:
25183 * gst-libs/gst/audio/gstbaseaudiosink.c:
25184 (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
25185 When the audio clock is slaved to another clock, never try to align
25186 samples but trust the rate interpolation algorithm.
25188 2006-07-24 14:34:42 +0000 Wim Taymans <wim.taymans@gmail.com>
25190 ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
25191 Original commit message from CVS:
25192 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
25193 Don't try to calculate silence samples, base class does this much
25195 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
25196 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
25197 (gst_ring_buffer_acquire):
25198 Calculate silence samples correctly.
25199 * gst-libs/gst/audio/gstringbuffer.h:
25202 2006-07-22 17:01:12 +0000 Tim-Philipp Müller <tim@centricular.net>
25204 gst/typefind/gsttypefindfunctions.c: Limit search for the first markup tag to the first few kB of the file. If we don...
25205 Original commit message from CVS:
25206 * gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
25207 Limit search for the first markup tag to the first few kB of
25208 the file. If we don't find one there, it's highly unlikely that
25209 this is an XML(-ish) file.
25211 2006-07-21 17:04:06 +0000 Andy Wingo <wingo@pobox.com>
25213 tests/check/pipelines/theoraenc.c (test_discontinuity): Similar test to the one in vorbisenc. Also commented out.
25214 Original commit message from CVS:
25215 2006-07-21 Andy Wingo <wingo@pobox.com>
25216 * tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
25217 test to the one in vorbisenc. Also commented out.
25219 2006-07-21 16:54:19 +0000 Andy Wingo <wingo@pobox.com>
25221 tests/check/pipelines/vorbisenc.c: New test, commented out until Mike lands some elite vorbisenc patches.
25222 Original commit message from CVS:
25223 2006-07-21 Andy Wingo <wingo@pobox.com>
25224 * tests/check/pipelines/vorbisenc.c:
25225 (test_discontinuity): New test, commented out until Mike lands
25226 some elite vorbisenc patches.
25228 2006-07-21 15:59:24 +0000 Andy Wingo <wingo@pobox.com>
25230 tests/check/pipelines/: Port to bufferstraw.
25231 Original commit message from CVS:
25232 2006-07-21 Andy Wingo <wingo@pobox.com>
25233 * tests/check/pipelines/vorbisenc.c:
25234 * tests/check/pipelines/theoraenc.c: Port to bufferstraw.
25235 Bufferstraw was actually factored out of these tests. Now we share
25238 2006-07-21 11:03:28 +0000 Wim Taymans <wim.taymans@gmail.com>
25240 ext/theora/theoradec.c: Better clipping.
25241 Original commit message from CVS:
25242 * ext/theora/theoradec.c: (clip_buffer):
25245 2006-07-21 10:43:54 +0000 Wim Taymans <wim.taymans@gmail.com>
25247 gst-libs/gst/audio/gstaudiosink.c: Fix leak.
25248 Original commit message from CVS:
25249 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
25250 (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
25251 (gst_audioringbuffer_release), (gst_audioringbuffer_stop):
25253 Avoid type casting when we can.
25254 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
25257 2006-07-20 16:57:29 +0000 Tim-Philipp Müller <tim@centricular.net>
25259 ext/alsa/gstalsamixerelement.c: Make state change fail if the specified device can't be opened for some reason.
25260 Original commit message from CVS:
25261 * ext/alsa/gstalsamixerelement.c:
25262 (gst_alsa_mixer_element_change_state):
25263 Make state change fail if the specified device can't be opened
25266 2006-07-20 10:42:21 +0000 Wim Taymans <wim.taymans@gmail.com>
25268 gst/playback/test.c: Example of a small audio/video player using decodebin.
25269 Original commit message from CVS:
25270 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
25271 (cb_newpad), (main):
25272 Example of a small audio/video player using decodebin.
25274 2006-07-20 05:56:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25276 gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
25277 Original commit message from CVS:
25278 * gst-libs/gst/riff/riff-ids.h:
25279 Add 'fact' chunk id
25281 2006-07-19 18:20:43 +0000 Wim Taymans <wim.taymans@gmail.com>
25283 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't assert when not negotiated but post a meaningfull error message. Fixes ...
25284 Original commit message from CVS:
25285 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25286 (gst_base_rtp_depayload_chain),
25287 (gst_base_rtp_depayload_change_state):
25288 Don't assert when not negotiated but post a meaningfull
25289 error message. Fixes #347918.
25290 * gst-libs/gst/rtp/gstbasertppayload.c:
25291 Add comment about better default MTU size.
25292 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
25293 Small cleanups, start docs.
25295 2006-07-19 14:46:36 +0000 Martin Szulecki <compiz@sukimashita.com>
25297 sys/v4l/gstv4lelement.c: If "device-name" is requested and the device is not open, try to temporarily open it to obta...
25298 Original commit message from CVS:
25299 Patch by: Martin Szulecki
25300 * sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
25301 If "device-name" is requested and the device is not
25302 open, try to temporarily open it to obtain this
25303 information (#342494).
25305 2006-07-19 12:25:00 +0000 Tim-Philipp Müller <tim@centricular.net>
25307 gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
25308 Original commit message from CVS:
25309 * gst-libs/gst/tag/gstid3tag.c:
25310 Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
25311 * gst-libs/gst/tag/gsttageditingprivate.h:
25312 * gst-libs/gst/tag/gstvorbistag.c:
25313 Some more random const-ifications.
25315 2006-07-18 19:48:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25317 gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
25318 Original commit message from CVS:
25319 * gst-libs/gst/riff/riff-ids.h:
25320 * gst-libs/gst/riff/riff-media.c:
25321 (gst_riff_create_video_template_caps):
25322 Add more FOURCCs (sort list to make stuff easier to find),
25323 add comment what those 16 bytes in struct _gst_riff_strh according to
25326 2006-07-17 14:17:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25328 gst/gdp/: remove parent_class setting, BOILERPLATE does this fix typo in comment
25329 Original commit message from CVS:
25330 2006-07-17 Thomas Vander Stichele <thomas at apestaart dot org>
25331 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
25332 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
25333 remove parent_class setting, BOILERPLATE does this
25334 (gst_gdp_pay_reset_streamheader):
25335 fix typo in comment
25337 2006-07-17 13:48:10 +0000 Tim-Philipp Müller <tim@centricular.net>
25339 gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
25340 Original commit message from CVS:
25341 * gst-libs/gst/audio/multichannel.c:
25342 (gst_audio_check_channel_positions),
25343 (gst_audio_fixate_channel_positions):
25344 Const-ify two arrays.
25346 2006-07-17 12:33:42 +0000 Tim-Philipp Müller <tim@centricular.net>
25348 ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
25349 Original commit message from CVS:
25350 * ext/alsa/gstalsa.c: (caps_add_channel_configuration):
25351 Fix typo, so that alsasink also advertises 8 channels
25352 if that's supported (tags: can, worms, open, alsa, ph34r).
25354 2006-07-17 12:01:04 +0000 Wim Taymans <wim.taymans@gmail.com>
25356 ext/ogg/gstoggdemux.c: *sigh*, when is the compiler going to warn when the comments are out-of-sync with the code.. R...
25357 Original commit message from CVS:
25358 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
25359 (gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
25360 *sigh*, when is the compiler going to warn when the comments
25361 are out-of-sync with the code.. Refix case of busted theora
25362 headers with 0 granule pos.
25364 2006-07-14 17:56:59 +0000 Wim Taymans <wim.taymans@gmail.com>
25366 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix 99% cpu load by waiting for absolute times on the clock. Fixes #347300.
25367 Original commit message from CVS:
25368 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25369 (gst_base_rtp_depayload_wait),
25370 (gst_base_rtp_depayload_change_state),
25371 (gst_base_rtp_depayload_set_property),
25372 (gst_base_rtp_depayload_get_property):
25373 Fix 99% cpu load by waiting for absolute times on the
25374 clock. Fixes #347300.
25376 2006-07-14 17:07:08 +0000 Andy Wingo <wingo@pobox.com>
25378 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (theora_parse_drain_event_queue) (theora_parse_push_headers, th...
25379 Original commit message from CVS:
25380 2006-07-14 Andy Wingo <wingo@pobox.com>
25381 * ext/theora/gsttheoraparse.h:
25382 * ext/theora/theoraparse.c (theora_parse_drain_event_queue)
25383 (theora_parse_push_headers, theora_parse_clear_queue)
25384 (theora_parse_drain_queue_prematurely, )
25385 (theora_parse_sink_event, theora_parse_change_state): Queue events
25386 until we initialized our state, like in vorbisparse.
25388 2006-07-14 16:45:17 +0000 Iain * <iaingnome@gmail.com>
25390 ext/vorbis/vorbisparse.h: ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue) (vorbis_parse_push_headers, vorbi...
25391 Original commit message from CVS:
25392 2006-07-14 Andy Wingo <wingo@pobox.com>
25393 * ext/vorbis/vorbisparse.h:
25394 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
25395 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
25396 (vorbis_parse_drain_queue_prematurely, )
25397 (vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
25398 until we have initialized our state. Fixes seeking after an
25400 2006-07-14 Andy Wingo <wingo@pobox.com>
25401 Patch by: Iain * <iaingnome@gmail.com>
25402 * ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.
25404 2006-07-14 15:52:39 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25406 configure.ac: Bump nano back to CVS
25407 Original commit message from CVS:
25409 Bump nano back to CVS
25411 === release 0.10.9 ===
25413 2006-07-14 15:51:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25415 configure.ac: releasing 0.10.9, "I walk the line"
25416 Original commit message from CVS:
25417 2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
25419 releasing 0.10.9, "I walk the line"
25421 2006-07-14 14:12:40 +0000 Michael Smith <msmith@xiph.org>
25423 tests/check/pipelines/vorbisenc.c: Move a g_cond_signal to earlier to avoid sometimes deadlocking (commonly happens w...
25424 Original commit message from CVS:
25425 * tests/check/pipelines/vorbisenc.c: (stop_pipeline):
25426 Move a g_cond_signal to earlier to avoid sometimes deadlocking
25427 (commonly happens when running this test under valgrind) when trying
25428 to remove the buffer probe.
25430 2006-07-14 10:34:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25432 * gst/gdp/Makefile.am:
25433 build as a plugin, not a lib
25434 Original commit message from CVS:
25435 build as a plugin, not a lib
25437 2006-07-13 16:43:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25439 sys/xvimage/xvimagesink.c: Fix missing g_unlock from the previous commit
25440 Original commit message from CVS:
25441 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
25442 Fix missing g_unlock from the previous commit
25444 2006-07-13 16:34:04 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25446 sys/: Implement a locking order to ensure we always take the object lock before the x_lock and never vice-versa.
25447 Original commit message from CVS:
25448 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
25449 (gst_ximagesink_change_state):
25450 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
25451 (gst_xvimagesink_change_state):
25452 Implement a locking order to ensure we always take the object lock
25453 before the x_lock and never vice-versa.
25455 2006-07-13 15:25:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25457 docs/plugins/: add more plugins and elements to docs
25458 Original commit message from CVS:
25459 * docs/plugins/Makefile.am:
25460 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
25461 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
25462 add more plugins and elements to docs
25463 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
25464 fix segfaults due to wrong g_free
25466 * gst/gdp/gstgdppay.c:
25469 2006-07-13 14:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25471 gst/playback/gstdecodebin.c: Fix a caps leak when linking (#347304)
25472 Original commit message from CVS:
25473 * gst/playback/gstdecodebin.c: (find_compatibles):
25474 Fix a caps leak when linking (#347304)
25475 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
25476 (gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
25477 (gst_ximagesink_change_state):
25478 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
25479 (gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
25480 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
25481 (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
25482 Don't leak shared memory resources. Use the object lock to protect
25483 against the xcontext disappearing while returning a buffer from the
25484 pipeline. (#347304)
25486 2006-07-12 14:20:43 +0000 Edward Hervey <bilboed@bilboed.com>
25488 ext/vorbis/vorbisdec.c: gst_tag_list_merge() returns a new object. Take that into account when using it. This avoids ...
25489 Original commit message from CVS:
25490 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
25491 (vorbis_handle_comment_packet):
25492 gst_tag_list_merge() returns a new object. Take that into account when
25493 using it. This avoids memleak.
25494 Revert previous commit which is not needed.
25496 2006-07-12 13:30:20 +0000 Edward Hervey <bilboed@bilboed.com>
25498 ext/vorbis/vorbisdec.c: Reset the decoder in finalize so that all fields get cleared.
25499 Original commit message from CVS:
25500 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
25501 Reset the decoder in finalize so that all fields get cleared.
25503 2006-07-12 13:24:19 +0000 Wim Taymans <wim.taymans@gmail.com>
25505 gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
25506 Original commit message from CVS:
25507 * gst-libs/gst/audio/gstbaseaudiosrc.c:
25508 (gst_base_audio_src_set_clock),
25509 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
25510 Don't try to post an error message when setting the clock fails
25511 as this can happen when adding an element to a bin which will then
25512 deadlock. Fixes #347296.
25514 2006-07-12 13:04:15 +0000 Edward Hervey <bilboed@bilboed.com>
25516 ext/vorbis/vorbisdec.c: Post tag messages on the bus even if we're not initialized.
25517 Original commit message from CVS:
25518 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
25519 (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
25520 (vorbis_handle_type_packet):
25521 Post tag messages on the bus even if we're not initialized.
25522 If we're not initialized, we still postpone the event pushing of tags.
25524 2006-07-12 11:28:37 +0000 Wim Taymans <wim.taymans@gmail.com>
25526 Revert last two changes that broke the freeze.
25527 Original commit message from CVS:
25528 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
25529 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
25530 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
25531 Revert last two changes that broke the freeze.
25533 2006-07-12 10:59:55 +0000 Wim Taymans <wim.taymans@gmail.com>
25535 ext/alsa/gstalsasink.c: basesink calculates silence sample correctly for us.
25536 Original commit message from CVS:
25537 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
25538 basesink calculates silence sample correctly for us.
25540 2006-07-12 10:58:42 +0000 Wim Taymans <wim.taymans@gmail.com>
25542 gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
25543 Original commit message from CVS:
25544 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
25545 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
25546 Calculate correct silence samples so we don't fill our ringbuffer
25549 2006-07-12 10:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
25551 ext/vorbis/vorbisdec.*: Delay sending events (newsegment, tags) until the decoder is properly initialized.
25552 Original commit message from CVS:
25553 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
25554 (gst_vorbis_dec_reset), (vorbis_dec_sink_event),
25555 (vorbis_handle_comment_packet), (vorbis_handle_type_packet):
25556 * ext/vorbis/vorbisdec.h:
25557 Delay sending events (newsegment, tags) until the decoder is properly
25561 2006-07-11 22:40:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25578 Original commit message from CVS:
25581 2006-07-11 21:04:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25583 tests/check/elements/audioconvert.c: Patch from #347221 adding a test for audioconvert channel remappings.
25584 Original commit message from CVS:
25585 * tests/check/elements/audioconvert.c: (get_float_mc_caps),
25586 (get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
25587 Patch from #347221 adding a test for audioconvert
25588 channel remappings.
25590 2006-07-11 12:03:25 +0000 Tim-Philipp Müller <tim@centricular.net>
25592 gst/subparse/gstssaparse.c: Don't include the terminating NUL in the buffer size, it's only there for extra paranoia ...
25593 Original commit message from CVS:
25594 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
25595 (gst_ssa_parse_parse_line):
25596 Don't include the terminating NUL in the buffer size,
25597 it's only there for extra paranoia (would add random
25598 '*' characters at the end of each subtitle since the
25599 terminator itself is not valid UTF-8 technically).
25600 Also fix indenting after boilerplate macro.
25602 2006-07-10 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.net>
25604 gst/playback/gstdecodebin.c: Also emit 'unknown-type' signal (which should really be called unhandled-type) if we fou...
25605 Original commit message from CVS:
25606 * gst/playback/gstdecodebin.c: (close_pad_link):
25607 Also emit 'unknown-type' signal (which should really be
25608 called unhandled-type) if we found potential decoders/demuxers
25609 in the registry but none of them worked in the end (as in the
25610 case where the plugins don't exist any longer but are still
25611 listed in the registry). Fixes #329798.
25613 2006-07-08 13:48:58 +0000 Andy Wingo <wingo@pobox.com>
25616 * ext/theora/theoraparse.c:
25617 theoraparse.c (theora_parse_push_buffer)
25618 Original commit message from CVS:
25619 2006-07-08 Andy Wingo <wingo@pobox.com>
25620 * theoraparse.c (theora_parse_push_buffer)
25621 (theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
25622 Add some more debugging. Fix granulepos reconstruction in the face
25623 of discontinuities.
25625 2006-07-06 15:54:50 +0000 Wim Taymans <wim.taymans@gmail.com>
25627 gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
25628 Original commit message from CVS:
25629 * gst-libs/gst/audio/gstbaseaudiosink.c:
25630 (gst_base_audio_sink_class_init),
25631 (gst_base_audio_sink_provide_clock):
25632 Use gobject_class instead of G_OBJECT_CLASS (klass)
25633 * gst-libs/gst/audio/gstbaseaudiosrc.c:
25634 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
25635 (gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
25636 (gst_base_audio_src_get_time),
25637 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
25638 (gst_base_audio_src_create_ringbuffer):
25639 Fix latency and buffer-time constants and properties ala basesink.
25640 Implement pull based scheduling. Fixes #346527.
25641 Set default blocksize in GstBaseSrc to 0, we default to pushing out
25643 Refuse slaving to another clock instead of silently not working.
25644 Only provide a clock when we are actually able to do so.
25645 Various small cleanups and compiler hints.
25647 2006-07-06 13:23:07 +0000 Lutz Mueller <lutz@topfrose.de>
25649 gst/typefind/gsttypefindfunctions.c: Add typefinding for text/html (#346581).
25650 Original commit message from CVS:
25651 Patch by: Lutz Mueller <lutz at topfrose de>
25652 * gst/typefind/gsttypefindfunctions.c: (html_type_find),
25654 Add typefinding for text/html (#346581).
25656 2006-07-06 13:12:02 +0000 Tim-Philipp Müller <tim@centricular.net>
25658 gst/typefind/gsttypefindfunctions.c: Fix SMIL typefinding, make xml_check_first_element() more useful.
25659 Original commit message from CVS:
25660 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
25661 (xml_check_first_element), (xml_type_find), (smil_type_find):
25662 Fix SMIL typefinding, make xml_check_first_element() more
25665 2006-07-06 13:04:24 +0000 Tim-Philipp Müller <tim@centricular.net>
25667 gst/playback/gstplaybasebin.*: Protect list of elements with a subtitle-encoding property and the subtitle encoding m...
25668 Original commit message from CVS:
25669 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
25670 (gst_play_base_bin_finalize), (decodebin_element_added_cb),
25671 (decodebin_element_removed_cb), (gst_play_base_bin_set_property):
25672 * gst/playback/gstplaybasebin.h:
25673 Protect list of elements with a subtitle-encoding property and
25674 the subtitle encoding member itself with a lock of their own
25675 instead of using the object lock. This prevents a dead-lock in
25676 the element-remove callback in some circumstances when shutting
25679 2006-07-05 20:11:13 +0000 Sébastien Moutte <sebastien@moutte.net>
25681 win32/common/libgsttag.def: Export some new functions.
25682 Original commit message from CVS:
25683 * win32/common/libgsttag.def:
25684 Export some new functions.
25685 * win32/vs6/libgstogg.dsp:
25686 Add a link to libgsttag-0.10.lib.
25688 2006-07-04 16:50:21 +0000 Tim-Philipp Müller <tim@centricular.net>
25690 ext/alsa/gstalsamixertrack.c: Some const-ification.
25691 Original commit message from CVS:
25692 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
25693 Some const-ification.
25695 2006-07-04 14:06:03 +0000 Wim Taymans <wim.taymans@gmail.com>
25697 gst/playback/gstplaybasebin.c: Improve checking if we are dealing with a stream. Added some more uris that need buffe...
25698 Original commit message from CVS:
25699 * gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
25700 Improve checking if we are dealing with a stream. Added some
25701 more uris that need buffering.
25703 2006-07-03 10:43:31 +0000 Edward Hervey <bilboed@bilboed.com>
25705 ext/vorbis/vorbisdec.c: Remove unused variable.
25706 Original commit message from CVS:
25707 * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
25708 Remove unused variable.
25710 2006-07-02 21:48:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25712 Makefile.am: include lcov.mak
25713 Original commit message from CVS:
25717 add GCOV_LIBS to GST_LIBS
25719 2006-07-02 11:08:58 +0000 Michael Sheldon <webmaster@mikeasoft.com>
25721 ext/alsa/gstalsasrc.c: Add 32 bps to template caps and increase channels range from [1,2] to [1,MAX]. See #346326.
25722 Original commit message from CVS:
25723 Patch by: Michael Sheldon <webmaster at mikeasoft com>
25724 * ext/alsa/gstalsasrc.c:
25725 Add 32 bps to template caps and increase channels range
25726 from [1,2] to [1,MAX]. See #346326.
25728 2006-06-30 12:04:51 +0000 Tim-Philipp Müller <tim@centricular.net>
25730 gst-libs/gst/riff/riff-media.c: Recognise 'WMVA' video codec fourcc (#345879).
25731 Original commit message from CVS:
25732 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
25733 Recognise 'WMVA' video codec fourcc (#345879).
25735 2006-06-29 12:21:06 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
25737 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
25738 Original commit message from CVS:
25739 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
25740 Fixed nasty memory leak
25742 2006-06-26 13:19:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25744 gst/tcp/gsttcp.c: fix logging
25745 Original commit message from CVS:
25746 * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
25747 (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
25750 2006-06-23 16:45:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25752 gst/playback/gstdecodebin.c: Protect remove_fakesink using a mutex, so that we don't try and remove the fakesink simu...
25753 Original commit message from CVS:
25754 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
25755 (gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
25756 (remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
25757 Protect remove_fakesink using a mutex, so that we don't try and
25758 remove the fakesink simultaneously from multiple threads.
25759 When going from READY to PAUSED, restore the fakesink, so that
25760 it is there when decodebin gets reused.
25762 2006-06-23 09:53:09 +0000 Tim-Philipp Müller <tim@centricular.net>
25764 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
25765 Original commit message from CVS:
25766 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
25767 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25768 * gst-libs/gst/rtp/gstbasertppayload.c:
25769 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
25770 * gst/tcp/gstmultifdsink.c:
25771 * gst/tcp/gsttcpclientsink.c:
25772 * gst/tcp/gsttcpclientsrc.c:
25773 * gst/tcp/gsttcpserversink.c:
25774 * gst/tcp/gsttcpserversrc.c:
25775 * gst/videorate/gstvideorate.c:
25776 * gst/videotestsrc/gstvideotestsrc.c:
25777 * sys/v4l/gstv4ljpegsrc.c:
25778 * sys/v4l/gstv4lmjpegsink.c:
25779 * sys/v4l/gstv4lsrc.c:
25780 * tests/examples/seek/scrubby.c:
25781 * tests/examples/seek/seek.c:
25782 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
25784 2006-06-23 09:09:44 +0000 Tim-Philipp Müller <tim@centricular.net>
25786 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) and fix one GObject boilerplate macro.
25787 Original commit message from CVS:
25788 * ext/directfb/dfbvideosink.c:
25789 * ext/gsm/gstgsmdec.c:
25790 * ext/gsm/gstgsmenc.c:
25791 * ext/libmms/gstmms.c:
25792 * ext/neon/gstneonhttpsrc.c:
25793 * ext/theora/theoradec.c:
25794 * gst/freeze/gstfreeze.c:
25795 * gst/gdp/gstgdpdepay.c:
25796 * gst/gdp/gstgdppay.c:
25797 * sys/glsink/glimagesink.c:
25798 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
25799 and fix one GObject boilerplate macro.
25801 2006-06-22 12:13:31 +0000 Tim-Philipp Müller <tim@centricular.net>
25803 gst-libs/gst/tag/tags.c: Second field in GEnumValue shouldn't be a description, but a stringified version of the enum...
25804 Original commit message from CVS:
25805 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
25806 Second field in GEnumValue shouldn't be a description,
25807 but a stringified version of the enum value.
25809 2006-06-22 12:03:14 +0000 Wim Taymans <wim.taymans@gmail.com>
25811 sys/ximage/ximagesink.c: Avoid type checking in buffer casts.
25812 Original commit message from CVS:
25813 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
25814 (gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
25815 (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
25816 Avoid type checking in buffer casts.
25817 Avoid caps copy in buffer_alloc when we can.
25818 Use pad_peer_accept.
25820 2006-06-22 11:01:41 +0000 Tim-Philipp Müller <tim@centricular.net>
25822 gst-libs/gst/tag/tag.h: Oops, make that 'Since: 0.10.9'.
25823 Original commit message from CVS:
25824 * gst-libs/gst/tag/tag.h:
25825 Oops, make that 'Since: 0.10.9'.
25827 2006-06-22 10:55:05 +0000 Tim-Philipp Müller <tim@centricular.net>
25829 API: add GstTagImageType enum to describe images contained in image tags (#345641).
25830 Original commit message from CVS:
25831 * docs/libs/gst-plugins-base-libs-sections.txt:
25832 * gst-libs/gst/tag/tag.h:
25833 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
25834 (gst_tag_image_type_get_type):
25835 API: add GstTagImageType enum to describe images contained
25836 in image tags (#345641).
25838 2006-06-22 10:31:22 +0000 Tim-Philipp Müller <tim@centricular.net>
25840 gst/tcp/gstmultifdsink.c: Fix warnings with gst-inspect: "buffers-min" property should be of G_TYPE_INT and not G_TYP...
25841 Original commit message from CVS:
25842 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
25843 Fix warnings with gst-inspect: "buffers-min" property
25844 should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
25845 typo in property description.
25847 2006-06-22 10:10:51 +0000 Cody Russell <bratsche@gnome.org>
25849 gst/: Avoid unnecessary class cast check in class_init functions (#337747).
25850 Original commit message from CVS:
25851 Patch by: Cody Russell <bratsche at gnome org>
25852 * gst/audioresample/gstaudioresample.c:
25853 (gst_audioresample_class_init):
25854 * gst/playback/gststreamselector.c:
25855 (gst_stream_selector_class_init):
25856 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
25857 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
25858 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
25859 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
25860 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
25861 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
25862 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
25863 * gst/videotestsrc/gstvideotestsrc.c:
25864 (gst_video_test_src_class_init):
25865 * gst/volume/gstvolume.c: (gst_volume_class_init):
25866 Avoid unnecessary class cast check in class_init
25867 functions (#337747).
25869 2006-06-21 18:39:07 +0000 Tim-Philipp Müller <tim@centricular.net>
25871 ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
25872 Original commit message from CVS:
25873 * ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
25874 (gst_text_overlay_video_chain):
25875 g_markup_escape_text() REALLY doesn't like non-UTF8 input
25876 and doesn't validate its input either (and neither did
25877 textoverlay it seems). Let's do that then and fix #345206.
25879 2006-06-19 17:12:57 +0000 Wim Taymans <wim.taymans@gmail.com>
25881 gst/tcp/gstmultifdsink.*: Added shiny new burst-on-connect methods.
25882 Original commit message from CVS:
25883 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
25884 (gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
25885 (gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
25886 (gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
25887 (find_syncframe), (find_limits), (assign_value),
25888 (count_burst_unit), (gst_multi_fd_sink_new_client),
25889 (gst_multi_fd_sink_handle_client_write),
25890 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
25891 (gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
25892 (gst_multi_fd_sink_change_state):
25893 * gst/tcp/gstmultifdsink.h:
25894 Added shiny new burst-on-connect methods.
25895 Add properties to control the minimal amount of data queued.
25897 API: bytes-min property
25898 API: time-min property
25899 API: buffers-min property
25900 API: burst-unit property
25901 API: burst-value property
25902 API: add-full signal
25903 * gst/tcp/gsttcp-marshal.list:
25904 Added new marshaller code for the new signal.
25905 * tests/check/elements/multifdsink.c: (GST_START_TEST),
25906 (multifdsink_suite):
25907 Added testcases for new burst methods.
25909 2006-06-19 11:35:47 +0000 Christian Schaller <uraeus@gnome.org>
25911 * gst-plugins-base.spec.in:
25912 update for latest changes
25913 Original commit message from CVS:
25914 update for latest changes
25916 2006-06-19 09:57:50 +0000 Edward Hervey <bilboed@bilboed.com>
25918 ext/theora/theoradec.c: Implement clipping for accurate seeking.
25919 Original commit message from CVS:
25920 * ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
25921 Implement clipping for accurate seeking.
25924 2006-06-19 09:08:05 +0000 Philip Jaegenstedt <philip@lysator.liu.se>
25926 gst/videoscale/gstvideoscale.c: Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
25927 Original commit message from CVS:
25928 Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
25929 * gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
25930 (gst_video_scale_transform):
25931 Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
25933 2006-06-17 14:18:41 +0000 Tim-Philipp Müller <tim@centricular.net>
25937 Original commit message from CVS:
25940 2006-06-17 14:13:03 +0000 Tim-Philipp Müller <tim@centricular.net>
25942 configure.ac: Fix --disable-extern (can't set conditionals conditionally, #343602).
25943 Original commit message from CVS:
25945 Fix --disable-extern (can't set conditionals conditionally,
25948 2006-06-16 15:43:23 +0000 Tim-Philipp Müller <tim@centricular.net>
25950 tests/check/elements/audioresample.c: Add test case for bug #342789 fixed below.
25951 Original commit message from CVS:
25952 * tests/check/elements/audioresample.c: (test_reuse),
25953 (audioresample_suite):
25954 Add test case for bug #342789 fixed below.
25956 2006-06-16 15:17:44 +0000 Tim-Philipp Müller <tim@centricular.net>
25958 gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
25959 Original commit message from CVS:
25960 * gst/audioresample/gstaudioresample.c:
25961 (gst_audioresample_class_init), (gst_audioresample_init),
25962 (audioresample_start), (audioresample_stop),
25963 (gst_audioresample_set_property), (gst_audioresample_get_property):
25964 Implement GstBaseTransform::start and ::stop so that audioresample
25965 can clear its internal state properly and be reused insted of
25966 causing non-negotiated errors with playbin under some circumstances
25968 * tests/check/elements/audioresample.c: (setup_audioresample),
25969 (cleanup_audioresample):
25970 Need to set element state here so that ::start and ::stop are
25973 2006-06-16 13:59:29 +0000 Young-Ho Cha <ganadist@chollian.net>
25975 gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
25976 Original commit message from CVS:
25977 Patch by: Young-Ho Cha <ganadist at chollian dot net>
25978 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
25979 Parse extra data better, apparently it's right behind
25980 the normal strf header size. Fixes #343500.
25982 2006-06-16 11:04:21 +0000 Wim Taymans <wim.taymans@gmail.com>
25984 ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
25985 Original commit message from CVS:
25986 * ext/alsa/gstalsasink.c: (set_hwparams):
25987 If we fail to set the buffer_time and period_time alsa
25988 parameters, post a warning and leave alsa select a
25989 default instead of failing. Fixes #342085
25991 2006-06-16 10:30:25 +0000 Tim-Philipp Müller <tim@centricular.net>
25994 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
25995 Original commit message from CVS:
25996 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
25998 2006-06-16 10:20:10 +0000 Tim-Philipp Müller <tim@centricular.net>
26000 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.
26001 Original commit message from CVS:
26002 * docs/libs/gst-plugins-base-libs-sections.txt:
26003 * gst-libs/gst/cdda/gstcddabasesrc.h:
26004 Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
26005 out in the header file and shouldn't be listed in the docs.
26006 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
26007 Fix it so that it doesn't crash in the debug statement.
26009 2006-06-16 10:02:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26011 docs/libs/: add remaining symbols into correct setions
26012 Original commit message from CVS:
26013 * docs/libs/Makefile.am:
26014 * docs/libs/gst-plugins-base-libs-docs.sgml:
26015 * docs/libs/gst-plugins-base-libs-sections.txt:
26016 * docs/libs/gst-plugins-base-libs.types:
26017 add remaining symbols into correct setions
26018 * gst-libs/gst/audio/gstringbuffer.c:
26019 fix incomplete docs
26020 * gst-libs/gst/audio/gstringbuffer.h:
26021 comment out not yet implemented function
26022 * gst-libs/gst/floatcast/floatcast.h:
26023 * gst-libs/gst/netbuffer/gstnetbuffer.c:
26024 add short descriptions
26025 * gst-libs/gst/interfaces/propertyprobe.c:
26026 fix return value docs
26027 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
26028 simplify debug logging
26029 * gst-libs/gst/riff/riff-read.h:
26030 sync function prototype and docs
26031 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
26032 remove left over symbol
26034 2006-06-16 09:56:41 +0000 Tim-Philipp Müller <tim@centricular.net>
26036 Use GST_PLUGIN_DOCS macro in configure.ac, add
26037 Original commit message from CVS:
26040 * docs/Makefile.am:
26041 Use GST_PLUGIN_DOCS macro in configure.ac, add
26042 --enable-plugin-docs default to autogen.sh and use
26043 ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
26045 2006-06-15 15:27:49 +0000 Wim Taymans <wim.taymans@gmail.com>
26047 ext/ogg/gstoggdemux.c: Combine GstFlowReturn from the source pads to give a meaningfull result to the upstream peer o...
26048 Original commit message from CVS:
26049 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
26050 (gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
26051 (gst_ogg_demux_loop):
26052 Combine GstFlowReturn from the source pads to give a
26053 meaningfull result to the upstream peer or to stop the
26054 processing task in case of errors.
26056 2006-06-14 14:49:33 +0000 Tim-Philipp Müller <tim@centricular.net>
26058 gst/playback/gststreaminfo.c: Try GST_TAG_CODEC as fallback when extracting the codec name; more debug info.
26059 Original commit message from CVS:
26060 * gst/playback/gststreaminfo.c: (cb_probe):
26061 Try GST_TAG_CODEC as fallback when extracting the
26062 codec name; more debug info.
26064 2006-06-14 14:34:28 +0000 Tim-Philipp Müller <tim@centricular.net>
26066 ext/ogg/: Extract language tags from ogm subtitle streams, so that the subtitle menu choices are labelled correctly in
26067 Original commit message from CVS:
26068 * ext/ogg/Makefile.am:
26069 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
26070 Extract language tags from ogm subtitle streams, so that
26071 the subtitle menu choices are labelled correctly in
26072 Totem (fixes #344708).
26074 2006-06-14 09:13:54 +0000 Alessandro Decina <alessandro@nnva.org>
26076 ext/ogg/gstoggmux.c: Fix various leaks. Fixes #343699.
26077 Original commit message from CVS:
26078 Patch by: Alessandro Decina <alessandro at nnva dot org>
26079 * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
26080 (gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
26081 (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
26082 (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
26083 Fix various leaks. Fixes #343699.
26084 Add x-smoke mime type.
26086 2006-06-14 08:17:45 +0000 Tim-Philipp Müller <tim@centricular.net>
26088 gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
26089 Original commit message from CVS:
26090 * gst-libs/gst/riff/riff-ids.h:
26091 Add IDs for 'bext' chunks (see #343837).
26093 2006-06-12 12:44:38 +0000 Young-Ho Cha <ganadist@chollian.net>
26095 gst/subparse/samiparse.c: Honour font face tags in SAMI subtitles (#344503).
26096 Original commit message from CVS:
26097 Patch by: Young-Ho Cha <ganadist at chollian net>
26098 * gst/subparse/samiparse.c: (sami_context_pop_state),
26099 (handle_start_font), (end_sami_element):
26100 Honour font face tags in SAMI subtitles (#344503).
26102 2006-06-11 20:41:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26104 po/POTFILES.in: add missing files containing translatable strings
26105 Original commit message from CVS:
26107 add missing files containing translatable strings
26109 2006-06-11 19:55:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26111 docs/libs/tmpl/.cvsignore: we don't want those *.sgml files in CVS either
26112 Original commit message from CVS:
26113 * docs/libs/tmpl/.cvsignore:
26114 we don't want those *.sgml files in CVS either
26116 2006-06-11 19:44:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26119 Original commit message from CVS:
26120 * docs/libs/.cvsignore:
26121 * tests/check/elements/.cvsignore:
26122 * tests/check/libs/.cvsignore:
26125 2006-06-11 18:33:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26127 docs/libs/Makefile.am: also commiting the changed Makefile.am (added more libs to the doc-build)
26128 Original commit message from CVS:
26129 * docs/libs/Makefile.am:
26130 also commiting the changed Makefile.am (added more libs to the
26133 2006-06-11 17:08:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26135 docs/libs/: first batch of reordering things, add index & hierarchy
26136 Original commit message from CVS:
26137 * docs/libs/gst-plugins-base-libs-docs.sgml:
26138 * docs/libs/gst-plugins-base-libs-sections.txt:
26139 * docs/libs/gst-plugins-base-libs.types:
26140 first batch of reordering things, add index & hierarchy
26142 2006-06-11 14:08:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26145 * ext/alsa/Makefile.am:
26146 * ext/cdparanoia/Makefile.am:
26147 * ext/gnomevfs/Makefile.am:
26148 * ext/libvisual/Makefile.am:
26149 * ext/ogg/Makefile.am:
26150 * ext/pango/Makefile.am:
26151 * ext/theora/Makefile.am:
26152 * ext/vorbis/Makefile.am:
26153 * sys/v4l/Makefile.am:
26154 * sys/ximage/Makefile.am:
26155 * sys/xvimage/Makefile.am:
26156 further clean up build
26157 Original commit message from CVS:
26158 further clean up build
26160 2006-06-11 12:14:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26162 configure.ac: use GST_PKG_CHECK_MODULES, cleans up output
26163 Original commit message from CVS:
26165 use GST_PKG_CHECK_MODULES, cleans up output
26167 2006-06-11 12:10:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26170 * win32/common/config.h:
26172 Original commit message from CVS:
26175 2006-06-10 18:52:03 +0000 Tim-Philipp Müller <tim@centricular.net>
26177 ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
26178 Original commit message from CVS:
26179 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
26180 Add support for burn:// URIs (#343385); const-ify things a bit,
26181 use G_N_ELEMENTS instead of hard-coded array size.
26183 2006-06-10 18:25:07 +0000 Young-Ho Cha <ganadist@chollian.net>
26185 gst/subparse/samiparse.c: Fix up broken entities before passing them to libxml *sigh*. (#343303).
26186 Original commit message from CVS:
26187 Patch by: Young-Ho Cha <ganadist at chollian net>
26188 * gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
26189 Fix up broken entities before passing them to libxml *sigh*.
26192 2006-06-09 18:52:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26197 Original commit message from CVS:
26200 === release 0.10.8 ===
26202 2006-06-09 18:49:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26208 * docs/plugins/gst-plugins-base-plugins.args:
26209 * docs/plugins/inspect/plugin-adder.xml:
26210 * docs/plugins/inspect/plugin-alsa.xml:
26211 * docs/plugins/inspect/plugin-audioconvert.xml:
26212 * docs/plugins/inspect/plugin-audiorate.xml:
26213 * docs/plugins/inspect/plugin-audioresample.xml:
26214 * docs/plugins/inspect/plugin-audiotestsrc.xml:
26215 * docs/plugins/inspect/plugin-cdparanoia.xml:
26216 * docs/plugins/inspect/plugin-decodebin.xml:
26217 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
26218 * docs/plugins/inspect/plugin-gnomevfs.xml:
26219 * docs/plugins/inspect/plugin-libvisual.xml:
26220 * docs/plugins/inspect/plugin-ogg.xml:
26221 * docs/plugins/inspect/plugin-pango.xml:
26222 * docs/plugins/inspect/plugin-playbin.xml:
26223 * docs/plugins/inspect/plugin-subparse.xml:
26224 * docs/plugins/inspect/plugin-tcp.xml:
26225 * docs/plugins/inspect/plugin-theora.xml:
26226 * docs/plugins/inspect/plugin-typefindfunctions.xml:
26227 * docs/plugins/inspect/plugin-video4linux.xml:
26228 * docs/plugins/inspect/plugin-videorate.xml:
26229 * docs/plugins/inspect/plugin-videoscale.xml:
26230 * docs/plugins/inspect/plugin-videotestsrc.xml:
26231 * docs/plugins/inspect/plugin-volume.xml:
26232 * docs/plugins/inspect/plugin-vorbis.xml:
26233 * docs/plugins/inspect/plugin-ximagesink.xml:
26234 * docs/plugins/inspect/plugin-xvimagesink.xml:
26235 * win32/common/config.h:
26237 Original commit message from CVS:
26240 2006-06-07 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26242 0.10.7.2 prerelease
26243 Original commit message from CVS:
26259 * win32/common/config.h:
26260 0.10.7.2 prerelease
26262 2006-06-07 11:03:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26264 move last template doc snippets to source code and delete them
26265 Original commit message from CVS:
26266 * docs/libs/tmpl/gstaudio.sgml:
26267 * docs/libs/tmpl/gstcolorbalance.sgml:
26268 * docs/libs/tmpl/gstmixer.sgml:
26269 * docs/libs/tmpl/gstringbuffer.sgml:
26270 * docs/libs/tmpl/gsttuner.sgml:
26271 * docs/libs/tmpl/gstxoverlay.sgml:
26272 * gst-libs/gst/audio/audio.c:
26273 * gst-libs/gst/audio/gstringbuffer.c:
26274 * gst-libs/gst/interfaces/colorbalance.c:
26275 * gst-libs/gst/interfaces/mixer.c:
26276 * gst-libs/gst/interfaces/tuner.c:
26277 * gst-libs/gst/interfaces/xoverlay.c:
26278 move last template doc snippets to source code and delete them
26280 2006-06-06 16:26:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26282 * gst/gdp/gstgdppay.c:
26284 Original commit message from CVS:
26287 2006-06-06 14:39:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26289 configure.ac: enable building of GDP elements
26290 Original commit message from CVS:
26292 enable building of GDP elements
26293 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
26294 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
26295 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
26296 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
26297 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
26298 (gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
26299 (gst_gdp_pay_change_state):
26300 * gst/gdp/gstgdppay.h:
26303 2006-06-06 11:13:18 +0000 Michael Smith <msmith@xiph.org>
26305 ext/theora/theoraparse.c: Mark DELTA_UNIT on non-keyframes.
26306 Original commit message from CVS:
26307 * ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
26308 (theora_parse_drain_queue):
26309 Mark DELTA_UNIT on non-keyframes.
26311 2006-06-03 21:06:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26313 gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
26314 Original commit message from CVS:
26315 * gst-libs/gst/audio/gstbaseaudiosink.c:
26316 (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
26317 * gst-libs/gst/audio/gstbaseaudiosink.h:
26318 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
26319 (gst_ring_buffer_samples_done):
26320 * gst-libs/gst/audio/gstringbuffer.h:
26321 Document better the fact that latency_time and buffer_time are values
26322 stored in microseconds, and not the usual GStreamer nanoseconds.
26323 Change the variables (compatibly) that store them from GstClockTime
26324 to guint64 to make it more clear that they're not storing clock times.
26325 Also, remove the bogus property description that says the user can
26326 specify -1 to get the default value, since that's never been the case.
26327 When computing the default segment size for the ring buffer, make it
26328 an integer number of samples.
26329 When the sub-class indicates a delay greater than the number of
26330 samples we've written return 0 from the audio sink get_time method.
26332 2006-06-02 17:01:02 +0000 Michael Smith <msmith@xiph.org>
26334 tests/check/: Don't busy-wait in tests; this was causing test timeouts very frequently when running under valgrind.
26335 Original commit message from CVS:
26336 * tests/check/elements/audioconvert.c: (set_channel_positions),
26337 (get_float_mc_caps), (get_int_mc_caps):
26338 * tests/check/elements/audioresample.c:
26339 * tests/check/elements/audiotestsrc.c: (GST_START_TEST):
26340 * tests/check/elements/videorate.c:
26341 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
26342 * tests/check/elements/volume.c:
26343 * tests/check/elements/vorbisdec.c:
26344 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
26345 Don't busy-wait in tests; this was causing test timeouts very
26346 frequently when running under valgrind.
26348 2006-06-02 16:45:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26350 * gst/gdp/gstgdpdepay.c:
26351 * gst/gdp/gstgdppay.h:
26353 Original commit message from CVS:
26356 2006-06-02 16:35:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26358 * tests/check/elements/multifdsink.c:
26359 fail_if_can_read is racy
26360 Original commit message from CVS:
26361 fail_if_can_read is racy
26363 2006-06-02 16:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26365 gst/tcp/: make multifdsink properly deal with streamheader:
26366 Original commit message from CVS:
26368 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
26369 (gst_multi_fd_sink_remove_client_link),
26370 (gst_multi_fd_sink_client_queue_caps),
26371 (gst_multi_fd_sink_client_queue_buffer),
26372 (gst_multi_fd_sink_handle_client_write),
26373 (gst_multi_fd_sink_render):
26374 * gst/tcp/gstmultifdsink.h:
26375 make multifdsink properly deal with streamheader:
26376 - streamheader is taken from caps
26377 - buffers marked with IN_CAPS are not sent
26378 - streamheaders are sent, on connection, from the caps of the
26379 buffer where the client gets positioned to
26380 - further streamheader changes are done every time the client
26381 will receive a buffer with different caps
26382 * tests/check/elements/multifdsink.c: (GST_START_TEST),
26383 (gst_multifdsink_create_streamheader):
26386 2006-06-02 15:06:59 +0000 Michael Smith <msmith@xiph.org>
26388 ext/vorbis/vorbisdec.c: Reinstate limit on channel count. Vorbis does not define the meaning of > 6 channels, so they...
26389 Original commit message from CVS:
26390 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
26391 Reinstate limit on channel count. Vorbis does not define the meaning
26392 of > 6 channels, so they're just independent channels. Gstreamer
26393 currently has no mechanism to represent N independent channels.
26395 2006-06-02 14:23:34 +0000 Michael Smith <msmith@xiph.org>
26397 ext/vorbis/vorbisdec.c: Don't arbitrarily restrict channel counts and rate in vorbis.
26398 Original commit message from CVS:
26399 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
26400 Don't arbitrarily restrict channel counts and rate in vorbis.
26401 In terms of effects likely on real-world files, this fixes 96kHz
26402 playback of vorbis.
26404 2006-06-02 14:19:18 +0000 Michael Smith <msmith@xiph.org>
26406 gst/audioconvert/audioconvert.c: More correct float->int conversion.
26407 Original commit message from CVS:
26408 * gst/audioconvert/audioconvert.c: (float):
26409 More correct float->int conversion.
26411 2006-06-02 14:07:42 +0000 Michael Smith <msmith@xiph.org>
26413 ext/ogg/gstoggdemux.c: Don't accidently send GST_CLOCK_TIME_NONE as a new segment start value. Fixes g-critical on tr...
26414 Original commit message from CVS:
26415 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
26416 Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
26417 value. Fixes g-critical on trying to play back ogg containing
26420 2006-06-02 10:34:12 +0000 Wim Taymans <wim.taymans@gmail.com>
26422 gst/playback/gstplaybasebin.*: Make the subtitle detection work from any thread so we don't deadlock. Fixes #343397.
26423 Original commit message from CVS:
26424 * gst/playback/gstplaybasebin.c: (group_create), (group_commit),
26426 * gst/playback/gstplaybasebin.h:
26427 Make the subtitle detection work from any thread so we don't
26428 deadlock. Fixes #343397.
26430 2006-06-02 10:28:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26432 gst/gdp/gstgdppay.c: add crc-header and crc-payload properties don't error out on some things that are recoverable
26433 Original commit message from CVS:
26434 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
26435 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
26436 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
26437 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
26438 (gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
26439 (gst_gdp_pay_get_property):
26440 add crc-header and crc-payload properties
26441 don't error out on some things that are recoverable
26442 * tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
26445 2006-06-02 09:17:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26447 * gst/tcp/gsttcp.c:
26448 show type number when packet is of the wrong type
26449 Original commit message from CVS:
26450 show type number when packet is of the wrong type
26452 2006-06-01 23:04:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26454 gst/volume/Makefile.am: Seriously, it's not *that* hard to get compilation right. Even a drunk can do it ! Add LIBOI...
26455 Original commit message from CVS:
26456 * gst/volume/Makefile.am:
26457 Seriously, it's not *that* hard to get compilation right. Even
26458 a drunk can do it ! Add LIBOIL CFLAGS and LIBS
26460 2006-06-01 22:00:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26462 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26463 Original commit message from CVS:
26464 * ext/alsaspdif/alsaspdifsink.h:
26465 * ext/amrwb/gstamrwbdec.h:
26466 * ext/amrwb/gstamrwbenc.h:
26467 * ext/amrwb/gstamrwbparse.h:
26468 * ext/arts/gst_arts.h:
26469 * ext/artsd/gstartsdsink.h:
26470 * ext/audiofile/gstafparse.h:
26471 * ext/audiofile/gstafsink.h:
26472 * ext/audiofile/gstafsrc.h:
26473 * ext/audioresample/gstaudioresample.h:
26474 * ext/bz2/gstbz2dec.h:
26475 * ext/bz2/gstbz2enc.h:
26476 * ext/dirac/gstdiracdec.h:
26477 * ext/directfb/dfbvideosink.h:
26478 * ext/divx/gstdivxdec.h:
26479 * ext/divx/gstdivxenc.h:
26480 * ext/dts/gstdtsdec.h:
26481 * ext/faac/gstfaac.h:
26482 * ext/gsm/gstgsmdec.h:
26483 * ext/gsm/gstgsmenc.h:
26484 * ext/ivorbis/vorbisenc.h:
26485 * ext/libfame/gstlibfame.h:
26486 * ext/nas/nassink.h:
26487 * ext/neon/gstneonhttpsrc.h:
26488 * ext/polyp/polypsink.h:
26489 * ext/sdl/sdlaudiosink.h:
26490 * ext/sdl/sdlvideosink.h:
26491 * ext/shout/gstshout.h:
26492 * ext/snapshot/gstsnapshot.h:
26493 * ext/sndfile/gstsf.h:
26494 * ext/swfdec/gstswfdec.h:
26495 * ext/tarkin/gsttarkindec.h:
26496 * ext/tarkin/gsttarkinenc.h:
26497 * ext/theora/theoradec.h:
26498 * ext/wavpack/gstwavpackdec.h:
26499 * ext/wavpack/gstwavpackparse.h:
26500 * ext/xine/gstxine.h:
26501 * ext/xvid/gstxviddec.h:
26502 * ext/xvid/gstxvidenc.h:
26503 * gst/cdxaparse/gstcdxaparse.h:
26504 * gst/cdxaparse/gstcdxastrip.h:
26505 * gst/colorspace/gstcolorspace.h:
26506 * gst/festival/gstfestival.h:
26507 * gst/freeze/gstfreeze.h:
26508 * gst/gdp/gstgdpdepay.h:
26509 * gst/gdp/gstgdppay.h:
26510 * gst/modplug/gstmodplug.h:
26511 * gst/mpeg1sys/gstmpeg1systemencode.h:
26512 * gst/mpeg1videoparse/gstmp1videoparse.h:
26513 * gst/mpeg2sub/gstmpeg2subt.h:
26514 * gst/mpegaudioparse/gstmpegaudioparse.h:
26515 * gst/multifilesink/gstmultifilesink.h:
26516 * gst/overlay/gstoverlay.h:
26517 * gst/playondemand/gstplayondemand.h:
26518 * gst/qtdemux/qtdemux.h:
26519 * gst/rtjpeg/gstrtjpegdec.h:
26520 * gst/rtjpeg/gstrtjpegenc.h:
26521 * gst/smooth/gstsmooth.h:
26522 * gst/smoothwave/gstsmoothwave.h:
26523 * gst/spectrum/gstspectrum.h:
26524 * gst/speed/gstspeed.h:
26525 * gst/stereo/gststereo.h:
26526 * gst/switch/gstswitch.h:
26527 * gst/tta/gstttadec.h:
26528 * gst/tta/gstttaparse.h:
26529 * gst/videodrop/gstvideodrop.h:
26530 * gst/xingheader/gstxingmux.h:
26531 * sys/directdraw/gstdirectdrawsink.h:
26532 * sys/directsound/gstdirectsoundsink.h:
26533 * sys/dxr3/dxr3audiosink.h:
26534 * sys/dxr3/dxr3spusink.h:
26535 * sys/dxr3/dxr3videosink.h:
26536 * sys/qcam/gstqcamsrc.h:
26537 * sys/vcd/vcdsrc.h:
26538 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26540 2006-06-01 20:39:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26542 gst/volume/gstvolume.*: rewrite the passthrough check, split _int16 and _int16_clamp, fix another property desc., rem...
26543 Original commit message from CVS:
26544 * gst/volume/gstvolume.c: (volume_choose_func),
26545 (volume_update_real_volume), (gst_volume_class_init),
26546 (gst_volume_init), (volume_process_float), (volume_process_int16),
26547 (volume_process_int16_clamp), (volume_set_caps),
26548 (volume_transform_ip), (plugin_init):
26549 * gst/volume/gstvolume.h:
26550 rewrite the passthrough check, split _int16 and _int16_clamp, fix
26551 another property desc., remove unused param from process function
26552 * tests/check/elements/volume.c: (volume_suite):
26553 reactivate the passthrough test
26555 2006-06-01 19:19:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26557 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26558 Original commit message from CVS:
26559 * ext/alsa/gstalsamixerelement.h:
26560 * ext/alsa/gstalsamixeroptions.h:
26561 * ext/alsa/gstalsamixertrack.h:
26562 * ext/gnomevfs/gstgnomevfssink.h:
26563 * ext/gnomevfs/gstgnomevfssrc.h:
26564 * ext/theora/gsttheoradec.h:
26565 * ext/theora/gsttheoraenc.h:
26566 * ext/theora/gsttheoraparse.h:
26567 * ext/vorbis/vorbisparse.h:
26568 * gst-libs/gst/audio/gstaudioclock.h:
26569 * gst-libs/gst/audio/gstaudiofilter.h:
26570 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
26571 * gst/audioconvert/gstaudioconvert.h:
26572 * gst/audioresample/gstaudioresample.h:
26573 * gst/audiotestsrc/gstaudiotestsrc.h:
26574 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
26575 * gst/playback/gststreamselector.h:
26576 * gst/tcp/gstmultifdsink.h:
26577 * gst/tcp/gsttcpclientsink.h:
26578 * gst/tcp/gsttcpclientsrc.h:
26579 * gst/tcp/gsttcpserversink.h:
26580 * gst/tcp/gsttcpserversrc.h:
26581 * gst/videorate/gstvideorate.h:
26582 * gst/videoscale/gstvideoscale.h:
26583 * gst/videotestsrc/gstvideotestsrc.h:
26584 * gst/volume/gstvolume.h:
26585 * sys/v4l/gstv4ljpegsrc.h:
26586 * sys/v4l/gstv4lmjpegsink.h:
26587 * sys/v4l/gstv4lmjpegsrc.h:
26588 * sys/v4l/gstv4lsrc.h:
26589 * sys/ximage/ximagesink.h:
26590 * sys/xvimage/xvimagesink.h:
26591 * tests/old/testsuite/alsa/sinesrc.h:
26592 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
26594 2006-05-31 16:56:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26596 * tests/check/elements/multifdsink.c:
26597 remove wrong commit
26598 Original commit message from CVS:
26599 remove wrong commit
26601 2006-05-31 16:21:48 +0000 Wim Taymans <wim.taymans@gmail.com>
26603 ext/libvisual/visual.c: Handle DISCONT.
26604 Original commit message from CVS:
26605 * ext/libvisual/visual.c: (gst_visual_reset),
26606 (gst_visual_sink_setcaps), (gst_visual_sink_event),
26607 (gst_visual_src_event), (get_buffer), (gst_visual_chain):
26609 Use running time before doing QoS.
26612 2006-05-31 14:17:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26614 docs/libs/Makefile.am: set a magic variable to indicate we know the docs are incomplete
26615 Original commit message from CVS:
26616 * docs/libs/Makefile.am:
26617 set a magic variable to indicate we know the docs are incomplete
26619 2006-05-30 20:33:59 +0000 Sébastien Moutte <sebastien@moutte.net>
26621 win32/common/libgstvideo.def: export gst_video_calculate_display_ratio
26622 Original commit message from CVS:
26623 * win32/common/libgstvideo.def:
26624 export gst_video_calculate_display_ratio
26625 * win32/vs6/libgstvideoscale.dsp:
26626 add link to libgstvideo-0.10.lib
26628 2006-05-30 19:00:39 +0000 Tim-Philipp Müller <tim@centricular.net>
26630 gst/playback/gstplaybasebin.c: Throw a more comprehensible error for rtsp:// URIs (rather than erroring out with a ne...
26631 Original commit message from CVS:
26632 * gst/playback/gstplaybasebin.c: (gen_source_element):
26633 Throw a more comprehensible error for rtsp:// URIs (rather
26634 than erroring out with a negotiation error later on) until
26635 we fix playbin to handle rtspsrc etc.
26637 2006-05-30 16:09:36 +0000 Wim Taymans <wim.taymans@gmail.com>
26639 ext/pango/gsttextoverlay.c: Added some FIXMEs.
26640 Original commit message from CVS:
26641 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
26642 (gst_text_overlay_text_event):
26645 2006-05-30 16:07:50 +0000 Wim Taymans <wim.taymans@gmail.com>
26647 gst/adder/gstadder.*: Implement release_request_pad.
26648 Original commit message from CVS:
26649 * gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
26650 (gst_adder_request_new_pad), (gst_adder_release_pad):
26651 * gst/adder/gstadder.h:
26652 Implement release_request_pad.
26653 Make padcounter atomic.
26654 * tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
26655 Added check for release_pad in adder.
26657 2006-05-30 16:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
26659 ext/ogg/gstoggdemux.c: Fix build again.
26660 Original commit message from CVS:
26661 * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
26664 2006-05-30 14:59:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26666 ext/ogg/gstoggdemux.c: add more debugging clean up printf formats for granulepos and serialno
26667 Original commit message from CVS:
26668 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
26669 (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
26670 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
26671 (gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
26672 (gst_ogg_demux_seek), (gst_ogg_demux_get_data),
26673 (gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
26674 (gst_ogg_demux_bisect_forward_serialno),
26675 (gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
26676 (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
26678 clean up printf formats for granulepos and serialno
26680 2006-05-30 14:31:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26682 * tests/check/elements/multifdsink.c:
26683 * tests/check/generic/states.c:
26684 properly fail if we can't make an element
26685 Original commit message from CVS:
26686 properly fail if we can't make an element
26688 2006-05-30 13:22:58 +0000 Michael Smith <msmith@xiph.org>
26690 ext/vorbis/vorbisenc.*: Multi-channel caps negotiation, so we can do proper multichannel vorbis encoding, negotiated ...
26691 Original commit message from CVS:
26692 * ext/vorbis/vorbisenc.c: (raw_caps_factory),
26693 (gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
26694 (gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
26695 (gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
26696 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
26697 * ext/vorbis/vorbisenc.h:
26698 Multi-channel caps negotiation, so we can do proper multichannel
26699 vorbis encoding, negotiated through audioconvert.
26701 2006-05-30 11:45:52 +0000 Wim Taymans <wim.taymans@gmail.com>
26703 tests/check/elements/adder.c: Added check to show that #339935 is fixed with ongoing adder and collectpads fixes.
26704 Original commit message from CVS:
26705 * tests/check/elements/adder.c: (test_event_message_received),
26706 (test_play_twice_message_received), (GST_START_TEST),
26708 Added check to show that #339935 is fixed with ongoing
26709 adder and collectpads fixes.
26711 2006-05-29 17:19:48 +0000 Wim Taymans <wim.taymans@gmail.com>
26713 gst/adder/gstadder.c: Don't leak pad name.
26714 Original commit message from CVS:
26715 * gst/adder/gstadder.c: (gst_adder_request_new_pad):
26716 Don't leak pad name.
26718 2006-05-29 15:49:53 +0000 Wim Taymans <wim.taymans@gmail.com>
26720 gst/adder/gstadder.c: Fix adder seeking.
26721 Original commit message from CVS:
26722 * gst/adder/gstadder.c: (gst_adder_query_duration),
26723 (forward_event_func), (forward_event), (gst_adder_src_event):
26725 Make query/seeking code threadsafe.
26726 * tests/check/Makefile.am:
26727 * tests/check/elements/adder.c: (test_event_message_received),
26728 (GST_START_TEST), (test_play_twice_message_received):
26729 Fix adder test case.
26731 2006-05-29 13:21:00 +0000 Young-Ho Cha <ganadist@chollian.net>
26733 gst/playback/gstplaybasebin.*: Add 'subtitle-encoding' property to playbin, so applications can force a subtitle enco...
26734 Original commit message from CVS:
26735 Patch by: Young-Ho Cha <ganadist at chollian net>
26736 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
26737 (gst_play_base_bin_init), (gst_play_base_bin_dispose),
26738 (set_encoding_element), (decodebin_element_added_cb),
26739 (decodebin_element_removed_cb), (setup_subtitle), (setup_source),
26740 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
26741 * gst/playback/gstplaybasebin.h:
26742 Add 'subtitle-encoding' property to playbin, so applications can
26743 force a subtitle encoding for non-UTF8 subtitles (#342268).
26744 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
26745 (gst_sub_parse_set_property):
26746 Rename recently-added 'encoding' property to 'subtitle-encoding'
26747 (so it can be proxied by playbin/decodebin in a generic way
26748 with less danger of false positives).
26750 2006-05-29 11:04:48 +0000 Michael Smith <msmith@xiph.org>
26752 gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf...
26753 Original commit message from CVS:
26754 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
26755 (append_with_other_format), (set_structure_widths),
26756 (gst_audio_convert_transform_caps):
26757 Patch from #341562: give more specific audio caps in get_caps, so
26758 that basetransform can make better decisions on what caps to
26761 2006-05-28 20:04:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26763 tests/check/elements/volume.c: make it compile again
26764 Original commit message from CVS:
26765 * tests/check/elements/volume.c:
26766 make it compile again
26768 2006-05-28 19:56:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26770 tests/check/elements/volume.c: disable test until #343196 gets resolved
26771 Original commit message from CVS:
26772 * tests/check/elements/volume.c: (volume_suite):
26773 disable test until #343196 gets resolved
26775 2006-05-28 19:42:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26777 gst/adder/gstadder.c: Make it easier to copy&paste
26778 Original commit message from CVS:
26779 * gst/adder/gstadder.c: (gst_adder_get_type):
26780 Make it easier to copy&paste
26781 * gst/volume/Makefile.am:
26782 * gst/volume/gstvolume.c: (volume_update_real_volume),
26783 (gst_volume_set_volume), (gst_volume_set_mute),
26784 (gst_volume_class_init), (volume_process_int16), (volume_set_caps),
26785 (volume_transform_ip), (volume_update_mute),
26786 (volume_update_volume):
26787 * gst/volume/gstvolume.h:
26788 Add own debug category, move duplicate code to helper function, fix
26789 property texts, add more comments and prepare ffor liboil-goodness
26790 * tests/check/Makefile.am:
26791 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
26792 add test for mute and passtrough case, be a bit more verbose to track
26794 * tests/check/generic/states.c: (GST_START_TEST):
26795 catch elements that fail to instantiate
26797 2006-05-28 09:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
26799 tests/check/pipelines/: Comment out tests using parse_launch() if core was built without parsing capabilities.
26800 Original commit message from CVS:
26801 * tests/check/pipelines/simple-launch-lines.c:
26802 * tests/check/pipelines/theoraenc.c:
26803 * tests/check/pipelines/vorbisenc.c:
26804 Comment out tests using parse_launch() if core was built without
26805 parsing capabilities.
26807 2006-05-27 13:34:03 +0000 Edward Hervey <bilboed@bilboed.com>
26809 tests/check/Makefile.am: Extra bonus points for whoever explains to ensonic that you are meant to test unit tests tho...
26810 Original commit message from CVS:
26811 * tests/check/Makefile.am:
26812 Extra bonus points for whoever explains to ensonic that you are meant
26813 to test unit tests thoroughly before commiting them, especially if
26814 you know it's going to break.
26815 De-activated element/adder tests.
26817 2006-05-27 13:09:16 +0000 Edward Hervey <bilboed@bilboed.com>
26819 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Marking caps conversion issues as GST_WARNING is way too verbose,
26820 Original commit message from CVS:
26821 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
26822 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
26823 Marking caps conversion issues as GST_WARNING is way too verbose,
26824 Moving them to GST_LOG.
26826 2006-05-27 11:26:18 +0000 Tim-Philipp Müller <tim@centricular.net>
26828 README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from...
26829 Original commit message from CVS:
26831 Replace current README (containing the release notes from
26832 some 0.9.x version) with a proper README taken from the core.
26834 2006-05-26 15:52:23 +0000 Wim Taymans <wim.taymans@gmail.com>
26836 ext/vorbis/vorbisdec.c: Small cleanups.
26837 Original commit message from CVS:
26838 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
26839 (vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
26840 (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
26841 (vorbis_dec_change_state):
26844 Clip output samples to segment boundaries.
26846 2006-05-26 11:17:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26848 sys/ximage/ximagesink.c: Improve the errors produced on bad output, including some human readable description strings.
26849 Original commit message from CVS:
26850 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
26851 (gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
26852 Improve the errors produced on bad output, including some human
26853 readable description strings.
26854 Handle the (theoretical for ximagesink) case where the XServer
26855 has a different idea about the size required for a particular
26856 frame and gives us too small a memory allocation.
26858 2006-05-26 10:18:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26861 Mention bugs fixed by previous commit
26862 Original commit message from CVS:
26863 Mention bugs fixed by previous commit
26865 2006-05-26 09:40:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26867 sys/xvimage/xvimagesink.c: Improve the errors produced on bad output, including some human readable description strings.
26868 Original commit message from CVS:
26869 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
26870 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
26871 (gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
26872 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
26873 Improve the errors produced on bad output, including some human
26874 readable description strings.
26875 Handle RGB Xv formats properly by transforming them into our
26876 big-endian caps description.
26877 Use gst_caps_truncate to ensure that we never try and choose a
26878 non-fixed caps in buffer_alloc.
26879 Handle the case where the XServer has a different idea about the size
26880 required for a particular frame and gives us too small a memory
26882 Use -1 to indicate 'no image format', because 0 is a valid XServer
26883 image format number.
26884 Put RGB Xv formats at the end of the caps, so that we always prefer
26886 Iterate the available Xv Encodings to determine the maximum width and
26887 height, and then return that in our caps.
26889 2006-05-25 16:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26891 gst/playback/gstdecodebin.c: When there is only one unfinished pad and it receives an event that doesn't match our re...
26892 Original commit message from CVS:
26893 * gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
26894 When there is only one unfinished pad and it receives an event that
26895 doesn't match our requirements, we need to set alldone=FALSE so that
26896 the fakesink is not removed yet.
26898 2006-05-25 09:32:31 +0000 Tim-Philipp Müller <tim@centricular.net>
26900 ext/ogg/gstoggdemux.c: Use gst_type_find_helper_for_buffer() to find the type of stream from the first packet.
26901 Original commit message from CVS:
26902 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
26903 Use gst_type_find_helper_for_buffer() to find the type
26904 of stream from the first packet.
26906 Bump requirements to core CVS (needed for vorbis
26907 typefinding to work).
26909 2006-05-24 08:34:53 +0000 Edward Hervey <bilboed@bilboed.com>
26911 gst/typefind/gsttypefindfunctions.c: Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
26912 Original commit message from CVS:
26913 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
26914 Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
26915 Else they play perfectly fine with qtdemux.
26917 2006-05-23 20:38:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26919 make more debug catagories static
26920 Original commit message from CVS:
26921 * ext/theora/theoradec.c:
26922 * ext/theora/theoraenc.c:
26923 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
26924 * gst/audiorate/gstaudiorate.c:
26925 make more debug catagories static
26926 * tests/check/Makefile.am:
26927 * tests/check/elements/adder.c: (message_received),
26928 (test_event_message_received), (GST_START_TEST),
26929 (test_play_twice_message_received), (adder_suite):
26930 added test case for using element twice, extra bonus points for anyone
26931 who can make these test run reliably
26933 2006-05-23 15:18:40 +0000 Tim-Philipp Müller <tim@centricular.net>
26935 ext/theora/theoradec.c: Make work with time-stamped input buffers that do not have a granulepos in BUFFER_OFFSET_END ...
26936 Original commit message from CVS:
26937 * ext/theora/theoradec.c: (theora_dec_chain):
26938 Make work with time-stamped input buffers that do not
26939 have a granulepos in BUFFER_OFFSET_END (like theora
26940 buffers coming from matroskademux). Fixes #342448.
26942 2006-05-22 15:53:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26944 gst/gdp/: Handle error cases when calling functions do downwards state change after parent's change_state
26945 Original commit message from CVS:
26946 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain),
26947 (gst_gdp_depay_change_state):
26948 * gst/gdp/gstgdpdepay.h:
26949 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader),
26950 (gst_gdp_pay_chain), (gst_gdp_pay_sink_event),
26951 (gst_gdp_pay_change_state):
26952 * gst/gdp/gstgdppay.h:
26953 Handle error cases when calling functions
26954 do downwards state change after parent's change_state
26955 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
26956 * tests/check/elements/gdppay.c: (GST_START_TEST):
26959 2006-05-22 13:25:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26961 adding GDP payloader and depayloader. Build integration will follow later when the GDP issues for core are sorted out.
26962 Original commit message from CVS:
26963 * gst/gdp/Makefile.am:
26964 * gst/gdp/gstgdp.c: (plugin_init):
26965 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_base_init),
26966 (gst_gdp_depay_class_init), (gst_gdp_depay_init),
26967 (gst_gdp_depay_finalize), (gst_gdp_depay_chain),
26968 (gst_gdp_depay_change_state), (gst_gdp_depay_plugin_init):
26969 * gst/gdp/gstgdpdepay.h:
26970 * gst/gdp/gstgdppay.c: (gst_gdp_pay_base_init),
26971 (gst_gdp_pay_class_init), (gst_gdp_pay_init),
26972 (gst_gdp_pay_dispose), (gst_gdp_stamp_buffer),
26973 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
26974 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
26975 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
26976 (gst_gdp_pay_sink_event), (gst_gdp_pay_change_state),
26977 (gst_gdp_pay_plugin_init):
26978 * gst/gdp/gstgdppay.h:
26979 * tests/check/Makefile.am:
26980 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
26981 (cleanup_gdpdepay), (gdpdepay_push_per_byte), (GST_START_TEST),
26982 (setup_gdpdepay_streamheader), (gdpdepay_suite), (main):
26983 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay),
26984 (GST_START_TEST), (setup_gdppay_streamheader), (gdppay_suite),
26986 adding GDP payloader and depayloader. Build integration will
26987 follow later when the GDP issues for core are sorted out.
26989 2006-05-22 11:42:03 +0000 Peter Kjellerstedt <pkj@axis.com>
26991 gst/tcp/Makefile.am: fdstresstest doesn't need Gtk+, fix compilation if gtk is not available (#342566).
26992 Original commit message from CVS:
26993 Patch by: Peter Kjellerstedt <pkj at axis com>
26994 * gst/tcp/Makefile.am:
26995 fdstresstest doesn't need Gtk+, fix compilation if
26996 gtk is not available (#342566).
26998 2006-05-19 17:57:56 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
27000 gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
27001 Original commit message from CVS:
27002 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
27004 Removed redundant floor()
27006 2006-05-19 15:00:43 +0000 Tim-Philipp Müller <tim@centricular.net>
27008 gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
27009 Original commit message from CVS:
27010 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
27011 On second thought, just skip JUNK chunks automatically, so
27012 the caller doesn't have to handle this. Fixes #342345.
27013 Also, return GST_FLOW_UNEXPECTED if we get a short read,
27014 not GST_FLOW_ERROR.
27016 2006-05-19 13:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
27018 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...
27019 Original commit message from CVS:
27020 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
27021 Don't bail out on JUNK chunks with a size of 0 (would try to
27022 pull_range 0 bytes before, which sources don't like too much).
27025 2006-05-19 13:02:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27027 Use the gstutil scaling function to preserve 64 bits while calculating output width and height from the display-aspec...
27028 Original commit message from CVS:
27029 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
27030 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
27031 Use the gstutil scaling function to preserve 64 bits while calculating
27032 output width and height from the display-aspect-ratio. (A continuation
27035 2006-05-19 11:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27037 sys/xvimage/xvimagesink.*: When performing buffer allocations, remember the caps and image format we return so that i...
27038 Original commit message from CVS:
27039 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
27040 (gst_xvimagesink_buffer_alloc):
27041 * sys/xvimage/xvimagesink.h:
27042 When performing buffer allocations, remember the caps and image format
27043 we return so that if the same caps are asked for next time we can
27044 return them immediately without doing any caps intersections.
27046 2006-05-18 23:00:02 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
27048 gst-libs/gst/rtp/README: Some new documentation
27049 Original commit message from CVS:
27050 2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
27051 * gst-libs/gst/rtp/README:
27052 Some new documentation
27053 * gst-libs/gst/rtp/gstrtpbuffer.h:
27054 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
27055 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
27056 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
27057 New RTP audio base payloader class. Supports frame or sample based codecs.
27058 Not enabled in Makefile.am until approved.
27060 2006-05-18 20:30:26 +0000 Tim-Philipp Müller <tim@centricular.net>
27062 tests/check/elements/alsa.c: Fix test case: don't try to free NULL GValueArray when there are no devices.
27063 Original commit message from CVS:
27064 * tests/check/elements/alsa.c: (test_device_property_probe):
27065 Fix test case: don't try to free NULL GValueArray when there
27068 2006-05-18 19:21:53 +0000 Tim-Philipp Müller <tim@centricular.net>
27070 tests/check/: Add simple test that runs a device property probe on alsasrc, alsasink and alsamixer. Disable valgrind ...
27071 Original commit message from CVS:
27072 * tests/check/Makefile.am:
27073 * tests/check/elements/alsa.c: (test_device_property_probe),
27074 (alsa_suite), (main):
27075 Add simple test that runs a device property probe on alsasrc,
27076 alsasink and alsamixer. Disable valgrind check for now (too
27077 many leaks in libasound, and valgrind ignored my suppressions
27080 2006-05-18 17:19:39 +0000 Martin Szulecki <gnomebugzilla@sukimashita.com>
27082 ext/alsa/: Clean up and simplify alsa device probing. Make it actually work for multiple classes. Don't cache results...
27083 Original commit message from CVS:
27084 * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
27085 (gst_alsa_device_property_probe_probe_property),
27086 (gst_alsa_device_property_probe_needs_probe),
27087 (gst_alsa_device_property_probe_get_values),
27088 (gst_alsa_type_add_device_property_probe_interface):
27089 * ext/alsa/gstalsadeviceprobe.h:
27090 * ext/alsa/gstalsamixerelement.c:
27091 (gst_alsa_mixer_element_init_interfaces):
27092 * ext/alsa/gstalsamixerelement.h:
27093 Clean up and simplify alsa device probing. Make it actually work
27094 for multiple classes. Don't cache results any longer.
27095 * ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
27096 (gst_alsasink_init):
27097 * ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
27098 (gst_alsasrc_interface_supported), (gst_implements_interface_init),
27099 (gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
27100 Make alsasink and alsasrc implement the GstPropertyProbe interface
27101 for device probing (#342181).
27102 Patch by: Martin Szulecki <gnomebugzilla at sukimashita com>
27104 2006-05-18 10:05:23 +0000 Tim-Philipp Müller <tim@centricular.net>
27106 gst/subparse/samiparse.c: Don't ignore return value of strtol (++compiler_happiness).
27107 Original commit message from CVS:
27108 * gst/subparse/samiparse.c: (handle_start_font):
27109 Don't ignore return value of strtol (++compiler_happiness).
27111 2006-05-17 17:49:10 +0000 Young-Ho Cha <ganadist@chollian.net>
27113 gst/subparse/gstsubparse.*: Add 'encoding' property (#341681).
27114 Original commit message from CVS:
27115 Patch by: Young-Ho Cha <ganadist chollian net>
27116 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
27117 (gst_sub_parse_class_init), (gst_sub_parse_init),
27118 (gst_sub_parse_set_property), (gst_sub_parse_get_property),
27119 (convert_encoding):
27120 * gst/subparse/gstsubparse.h:
27121 Add 'encoding' property (#341681).
27122 * gst/subparse/samiparse.c: (characters_sami):
27123 Output is pango markup, so we need to escape text
27124 between tags (#342143).
27126 2006-05-16 17:34:14 +0000 Tim-Philipp Müller <tim@centricular.net>
27128 gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
27129 Original commit message from CVS:
27130 * gst-libs/gst/audio/multichannel.c:
27131 (gst_audio_check_channel_positions):
27132 It's okay to have caps with channels=1 and a channel position
27133 different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
27134 (deinterleavers might want to keep the position in the caps,
27135 so that they can be re-interleaved again properly later).
27136 Leave check for unexpected 2-channel layouts intact for now.
27138 2006-05-16 16:28:10 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
27140 gst/tcp/gsttcp.c: Return GST_FLOW_UNEXPECTED when we have an eos on the socket so basesrc can do its job correctly.
27141 Original commit message from CVS:
27142 2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
27143 * gst/tcp/gsttcp.c: (gst_tcp_socket_read):
27144 Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
27145 basesrc can do its job correctly.
27147 2006-05-16 15:52:17 +0000 Tim-Philipp Müller <tim@centricular.net>
27149 ext/alsa/: Refactor and improve caps probing code: probe signedness when we probe the supported formats/widths; set e...
27150 Original commit message from CVS:
27151 * ext/alsa/Makefile.am:
27152 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
27153 (gst_alsa_detect_formats), (get_channel_free_structure),
27154 (caps_add_channel_configuration), (gst_alsa_detect_channels),
27155 (gst_alsa_probe_supported_formats):
27156 * ext/alsa/gstalsa.h:
27157 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
27158 Refactor and improve caps probing code: probe signedness
27159 when we probe the supported formats/widths; set endianness
27160 to the one we actually probed for (ie. cpu endianness).
27161 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
27162 (gst_alsasrc_close):
27163 * ext/alsa/gstalsasrc.h:
27164 Implement caps probing for alsasrc.
27166 2006-05-15 17:42:19 +0000 Wim Taymans <wim.taymans@gmail.com>
27168 ext/theora/theoradec.c: Cleanups, add some G_LIKELY.
27169 Original commit message from CVS:
27170 * ext/theora/theoradec.c: (gst_theora_dec_reset),
27171 (theora_dec_src_query), (theora_dec_src_event),
27172 (theora_dec_sink_event), (theora_handle_comment_packet),
27173 (theora_handle_data_packet), (theora_dec_change_state):
27174 Cleanups, add some G_LIKELY.
27175 Use segment helpers instead of our own wrong code.
27176 Clear queued buffers on seek and READY.
27177 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
27178 (vorbis_dec_convert), (vorbis_dec_src_query),
27179 (vorbis_dec_src_event), (vorbis_dec_sink_event),
27180 (vorbis_handle_comment_packet), (vorbis_dec_push),
27181 (vorbis_handle_data_packet), (vorbis_dec_chain),
27182 (vorbis_dec_change_state):
27183 * ext/vorbis/vorbisdec.h:
27184 Remove old useless packetno variable.
27185 Do position query properly.
27187 Do cleanup of queued buffers in new helper function
27190 2006-05-15 17:17:22 +0000 Tim-Philipp Müller <tim@centricular.net>
27192 ext/alsa/gstalsasink.c: Query supported sample rates. Fixes #341732.
27193 Original commit message from CVS:
27194 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
27195 Query supported sample rates. Fixes #341732.
27197 2006-05-15 17:01:02 +0000 Julien Moutte <julien@moutte.net>
27199 gst/playback/gstdecodebin.c: Make decodebin reusable when going from PAUSE_TO_READY and then back to PAUSED.
27200 Original commit message from CVS:
27201 2006-05-15 Julien MOUTTE <julien@moutte.net>
27202 * gst/playback/gstdecodebin.c: (cleanup_decodebin),
27203 (gst_decode_bin_change_state): Make decodebin reusable
27204 when going from PAUSE_TO_READY and then back to PAUSED.
27207 2006-05-15 16:49:31 +0000 Wim Taymans <wim.taymans@gmail.com>
27209 ext/vorbis/vorbisdec.c: Cleanups. Use refcounting and DEBUG_OBJECT.
27210 Original commit message from CVS:
27211 * ext/vorbis/vorbisdec.c: (vorbis_get_query_types),
27212 (vorbis_dec_convert), (vorbis_dec_src_query),
27213 (vorbis_dec_sink_query), (vorbis_dec_src_event),
27214 (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
27215 (vorbis_dec_clean_queued), (vorbis_dec_push),
27216 (vorbis_handle_data_packet), (vorbis_dec_change_state):
27217 Cleanups. Use refcounting and DEBUG_OBJECT.
27218 Reset segment on flush, use code methods instead of our
27220 Fix potential memleak.
27222 2006-05-15 16:46:44 +0000 Tim-Philipp Müller <tim@centricular.net>
27224 ext/alsa/gstalsasink.*: Don't leak allocated snd_output_t structure if there's more than one alsasink instance at a t...
27225 Original commit message from CVS:
27226 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
27227 (gst_alsasink_init):
27228 * ext/alsa/gstalsasink.h:
27229 Don't leak allocated snd_output_t structure if there's
27230 more than one alsasink instance at a time (#341873).
27231 Also fix GObject macros in header file.
27233 2006-05-15 15:31:30 +0000 Tim-Philipp Müller <tim@centricular.net>
27235 gst/subparse/gstsubparse.c: Don't use libxml functions in the typefinding code.
27236 Original commit message from CVS:
27237 * gst/subparse/gstsubparse.c:
27238 (gst_sub_parse_data_format_autodetect):
27239 Don't use libxml functions in the typefinding code.
27241 2006-05-15 15:01:08 +0000 Wim Taymans <wim.taymans@gmail.com>
27243 ext/ogg/gstoggdemux.c: Fix seeking performance in the case where a non-header packet has a 0 granulepos (busted theor...
27244 Original commit message from CVS:
27245 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
27246 Fix seeking performance in the case where a non-header
27247 packet has a 0 granulepos (busted theora case).
27250 2006-05-15 14:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
27252 gst/subparse/gstsubparse.c: Improve SAMI typefinding: handle case where there are whitespaces or newlines in front of...
27253 Original commit message from CVS:
27254 * gst/subparse/gstsubparse.c:
27255 (gst_sub_parse_data_format_autodetect):
27256 Improve SAMI typefinding: handle case where there are
27257 whitespaces or newlines in front of the first <SAMI>
27260 2006-05-15 12:18:13 +0000 Tim-Philipp Müller <tim@centricular.net>
27262 configure.ac: Build video4linux plugin even if there's no XVIDEO, just without implementing the GstXOverlay interface...
27263 Original commit message from CVS:
27265 Build video4linux plugin even if there's no XVIDEO, just
27266 without implementing the GstXOverlay interface (#334002).
27268 2006-05-15 10:17:04 +0000 Tim-Philipp Müller <tim@centricular.net>
27270 Add tentative support for libvisual-0.4 (#336881).
27271 Original commit message from CVS:
27273 * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
27275 Add tentative support for libvisual-0.4 (#336881).
27277 2006-05-15 09:41:03 +0000 Young-Ho Cha <ganadist@chollian.net>
27279 gst/subparse/samiparse.c: Need to map "silver" colour explicitly (#169936).
27280 Original commit message from CVS:
27281 Patch by: Young-Ho Cha <ganadist at chollian net>
27282 * gst/subparse/samiparse.c: (handle_start_font):
27283 Need to map "silver" colour explicitly (#169936).
27285 2006-05-15 09:14:35 +0000 Young-Ho Cha <ganadist@chollian.net>
27287 gst/subparse/: Add support for SAMI subtitles (#169936).
27288 Original commit message from CVS:
27289 Patch by: Young-Ho Cha <ganadist at chollian net>
27290 * gst/subparse/Makefile.am:
27291 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
27292 (parser_state_dispose), (gst_sub_parse_data_format_autodetect),
27293 (gst_sub_parse_format_autodetect), (feed_textbuf),
27294 (gst_subparse_type_find), (plugin_init):
27295 * gst/subparse/gstsubparse.h:
27296 * gst/subparse/samiparse.c:
27297 * gst/subparse/samiparse.h:
27298 Add support for SAMI subtitles (#169936).
27300 2006-05-14 21:18:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27302 * win32/common/config.h:
27304 Original commit message from CVS:
27307 2006-05-14 21:18:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27310 fix mistakes in README
27311 Original commit message from CVS:
27312 fix mistakes in README
27314 2006-05-14 18:15:17 +0000 Michael Smith <msmith@xiph.org>
27316 gst/audioconvert/gstchannelmix.c: Fix #341696: crash when mixing L+R+C to mono or stereo.
27317 Original commit message from CVS:
27318 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
27319 Fix #341696: crash when mixing L+R+C to mono or stereo.
27320 * tests/check/Makefile.am:
27321 * tests/check/elements/audioconvert.c: (set_channel_positions),
27322 (get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
27323 (audioconvert_suite):
27324 Add test for the above, including some generic framework bits for
27325 testing multichannel things.
27327 2006-05-14 16:05:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27331 Original commit message from CVS:
27334 === release 0.10.7 ===
27336 2006-05-14 16:00:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27338 configure.ac: releasing 0.10.7, "Leave the gun"
27339 Original commit message from CVS:
27340 2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
27342 releasing 0.10.7, "Leave the gun"
27344 2006-05-14 15:55:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27362 Original commit message from CVS:
27365 2006-05-12 22:22:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27368 Original commit message from CVS:
27369 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
27370 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
27373 2006-05-12 21:30:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27375 Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
27376 Original commit message from CVS:
27377 * docs/libs/gst-plugins-base-libs-docs.sgml:
27378 * docs/libs/gst-plugins-base-libs-sections.txt:
27379 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
27380 * gst-libs/gst/video/video.h:
27381 * gst/videoscale/Makefile.am:
27382 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
27383 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
27384 * tests/check/Makefile.am:
27385 * tests/check/libs/video.c: (GST_START_TEST), (video_suite),
27387 Fix integer overflow problem with pixel-aspect-ratio calculations
27388 in videoscale and xvimagesink (#341542)
27390 2006-05-12 16:56:52 +0000 Tim-Philipp Müller <tim@centricular.net>
27392 gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
27393 Original commit message from CVS:
27394 * gst-libs/gst/tag/gstid3tag.c:
27395 Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
27397 2006-05-12 10:39:08 +0000 Sébastien Moutte <sebastien@moutte.net>
27399 win32/MANIFEST: update win32 files listing
27400 Original commit message from CVS:
27402 update win32 files listing
27404 2006-05-11 21:47:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27406 * tests/check/elements/multifdsink.c:
27407 disable failing check on gentoo64
27408 Original commit message from CVS:
27409 disable failing check on gentoo64
27411 2006-05-11 21:35:44 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27413 * tests/check/elements/multifdsink.c:
27414 disable failing check on gentoo64
27415 Original commit message from CVS:
27416 disable failing check on gentoo64
27418 2006-05-11 21:20:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27420 * tests/check/elements/multifdsink.c:
27421 macros show the correct line
27422 Original commit message from CVS:
27423 macros show the correct line
27425 2006-05-11 21:04:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27427 * tests/check/elements/multifdsink.c:
27428 macros show the correct line
27429 Original commit message from CVS:
27430 macros show the correct line
27432 2006-05-11 21:01:05 +0000 Sjoerd Simons <sjoerd@luon.net>
27434 gst/playback/gstplaybasebin.*: API: GstPlayBaseBin::stream-info-value-array property use a more bindings-friendly way...
27435 Original commit message from CVS:
27436 2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
27437 patch by: Sjoerd Simons (sjoerd@luon.net)
27438 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
27439 (group_create), (group_destroy), (add_stream),
27440 (gst_play_base_bin_get_property),
27441 (gst_play_base_bin_get_streaminfo_value_array):
27442 * gst/playback/gstplaybasebin.h:
27443 API: GstPlayBaseBin::stream-info-value-array property
27444 use a more bindings-friendly way of exposing streaminfo
27445 using a GValueArray. Tested in ipython.
27448 2006-05-11 19:44:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27450 * tests/check/elements/multifdsink.c:
27451 fix some type warnings
27452 Original commit message from CVS:
27453 fix some type warnings
27455 2006-05-11 19:38:22 +0000 Wim Taymans <wim.taymans@gmail.com>
27457 gst/playback/gstdecodebin.c: Also catch queue underruns but don't do anything yet.
27458 Original commit message from CVS:
27459 * gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
27460 (queue_underrun_cb), (queue_filled_cb):
27461 Also catch queue underruns but don't do anything yet.
27462 Refactor and comment queue enlarging code a bit.
27463 * gst/playback/gstplaybasebin.c: (queue_overrun),
27464 (queue_threshold_reached), (queue_out_of_data),
27465 (gen_preroll_element):
27466 If a queue over/underruns check that we don't create nasty
27467 deadlocks when the min-threshold is not reached but the
27468 max-bytes is. In those cases disable max-bytes when we
27469 know that the queue is fed timed data.
27472 2006-05-11 18:06:18 +0000 Tim-Philipp Müller <tim@centricular.net>
27474 gst/playback/gstplaybin.c: Make playbin automatically plug an 'audioresample' element before the audio sink as well. ...
27475 Original commit message from CVS:
27476 * gst/playback/gstplaybin.c: (gen_audio_element):
27477 Make playbin automatically plug an 'audioresample'
27478 element before the audio sink as well. This solves
27479 problems with sinks that only accept a very specific
27480 sample rate, like esdsink (e.g. #340379).
27482 2006-05-11 16:04:28 +0000 Tim-Philipp Müller <tim@centricular.net>
27484 gst/playback/gstplaybasebin.c: Make http sources send special headers so that we receive icecast metadata if the http...
27485 Original commit message from CVS:
27486 * gst/playback/gstplaybasebin.c: (gen_source_element):
27487 Make http sources send special headers so that we receive
27488 icecast metadata if the http stream is an icecast stream
27489 (otherwise the server will just ignore them). This also
27490 means that from now on users will need the 'icydemux'
27491 element from gst-plugins-good installed if they want to
27492 listen to icecast radio streams. (#341432, #333657).
27494 2006-05-11 12:34:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27496 * gst/tcp/gstmultifdsink.c:
27498 Original commit message from CVS:
27501 2006-05-11 11:40:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27503 gst/tcp/gstmultifdsink.c: remove stupid example from docs - it should come with a simple
27504 Original commit message from CVS:
27505 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
27506 (gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
27507 remove stupid example from docs - it should come with a simple
27510 * tests/check/elements/multifdsink.c: (wait_bytes_served),
27511 (fail_if_can_read), (GST_START_TEST),
27512 (gst_multifdsink_create_streamheader), (multifdsink_suite):
27513 add a test for changing streamheader which exposes a bug in
27516 2006-05-11 10:33:46 +0000 Michael Smith <msmith@xiph.org>
27518 ext/gnomevfs/gstgnomevfssrc.*: Don't set icy-caps unless we have a sane interval value. Move interval to a local vari...
27519 Original commit message from CVS:
27520 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
27521 (gst_gnome_vfs_src_received_headers_callback):
27522 * ext/gnomevfs/gstgnomevfssrc.h:
27523 Don't set icy-caps unless we have a sane interval value. Move
27524 interval to a local variable; we never use it outside this function.
27526 2006-05-11 10:14:20 +0000 Wim Taymans <wim.taymans@gmail.com>
27528 sys/: Register special buffer types along with the objects so that they are not registered at runtime from N differen...
27529 Original commit message from CVS:
27530 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
27531 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
27532 Register special buffer types along with the objects so
27533 that they are not registered at runtime from N different
27534 streaming threads since they are not threadsafe.
27536 2006-05-10 18:31:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27538 * tests/check/elements/multifdsink.c:
27539 set caps and plug leaks
27540 Original commit message from CVS:
27541 set caps and plug leaks
27543 2006-05-10 18:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27545 tests/check/elements/multifdsink.c: add two more tests, one doing streamheader
27546 Original commit message from CVS:
27547 * tests/check/elements/multifdsink.c: (wait_bytes_served),
27548 (GST_START_TEST), (fail_unless_read), (multifdsink_suite):
27549 add two more tests, one doing streamheader
27551 2006-05-10 16:34:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27553 gst/tcp/gstmultifdsink.c: clean up the bufqueue when shutting down
27554 Original commit message from CVS:
27555 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
27556 clean up the bufqueue when shutting down
27557 * tests/check/Makefile.am:
27558 * tests/check/elements/multifdsink.c: (setup_multifdsink),
27559 (cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
27561 add a test for the leak that was just fixed
27563 2006-05-10 15:16:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27565 * gst/tcp/gstmultifdsink.c:
27567 Original commit message from CVS:
27570 2006-05-10 15:14:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27572 * gst/tcp/gstmultifdsink.c:
27573 * gst/tcp/gstmultifdsink.h:
27575 Original commit message from CVS:
27578 2006-05-10 11:54:36 +0000 Wim Taymans <wim.taymans@gmail.com>
27580 gst/adder/gstadder.*: Updated some docs. Added comments and FIXMEs all over the place.
27581 Original commit message from CVS:
27582 * gst/adder/gstadder.c: (gst_adder_setcaps),
27583 (gst_adder_query_duration), (gst_adder_query), (forward_event),
27584 (gst_adder_src_event), (gst_adder_sink_event),
27585 (gst_adder_class_init), (gst_adder_finalize),
27586 (gst_adder_request_new_pad), (gst_adder_collected):
27587 * gst/adder/gstadder.h:
27588 Updated some docs. Added comments and FIXMEs all over the place.
27589 Improve debugging info.
27590 Fix leak on finalize by not calling the parent.
27591 Implement duration query.
27592 Make event forwarding threadsafe.
27593 Correctly send NEWSEGMENT at start and after flush.
27594 Handle EOS correctly.
27595 Post error when not negotiated.
27596 * tests/check/elements/adder.c: (GST_START_TEST):
27597 Added FIXME in the test.
27599 2006-05-09 19:24:46 +0000 Tim-Philipp Müller <tim@centricular.net>
27601 Const-ify GEnumValue and GFlagsValue arrays. Use
27602 Original commit message from CVS:
27603 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
27604 (gst_text_overlay_halign_get_type),
27605 (gst_text_overlay_wrap_mode_get_type):
27606 * ext/theora/theoradec.c: (theora_handle_type_packet),
27607 (theora_handle_data_packet):
27608 * ext/theora/theoraenc.c: (gst_border_mode_get_type),
27609 (theora_enc_sink_setcaps), (theora_enc_chain):
27610 * gst-libs/gst/cdda/gstcddabasesrc.c:
27611 (gst_cdda_base_src_mode_get_type):
27612 * gst/audiotestsrc/gstaudiotestsrc.c:
27613 (gst_audiostestsrc_wave_get_type):
27614 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
27615 * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
27616 * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
27617 (gst_sync_method_get_type), (gst_unit_type_get_type),
27618 (gst_client_status_get_type):
27619 * gst/videoscale/gstvideoscale.c:
27620 (gst_video_scale_method_get_type):
27621 * gst/videotestsrc/gstvideotestsrc.c:
27622 (gst_video_test_src_pattern_get_type):
27623 * gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
27624 (paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
27625 (paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
27626 (paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
27627 (paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
27628 (paint_setup_RGB565), (paint_setup_xRGB1555):
27629 Const-ify GEnumValue and GFlagsValue arrays. Use
27630 GST_ROUND_UP_* macros instead of home-made ones.
27632 2006-05-09 17:40:41 +0000 Tim-Philipp Müller <tim@centricular.net>
27634 configure.ac: Require core CVS for the new newsegment stuff.
27635 Original commit message from CVS:
27637 Require core CVS for the new newsegment stuff.
27639 2006-05-09 17:30:48 +0000 Sjoerd Simons <sjoerd@luon.net>
27641 gst/tcp/gstmultifdsink.c: Register nick for enum value (#341160).
27642 Original commit message from CVS:
27643 Patch by: Sjoerd Simons <sjoerd at luon net>
27644 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
27645 Register nick for enum value (#341160).
27647 2006-05-09 16:46:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27649 gst/typefind/gsttypefindfunctions.c: backout typefind patch #340375
27650 Original commit message from CVS:
27651 * gst/typefind/gsttypefindfunctions.c: (m4a_type_find),
27653 backout typefind patch #340375
27654 * tests/check/elements/adder.c: (message_received),
27655 (GST_START_TEST), (adder_suite):
27656 redo, signal-handling of test
27658 2006-05-09 16:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
27660 gst/adder/gstadder.*: Remove bogus segment merging and forwarding, we don't care about timestamps anyway and we just ...
27661 Original commit message from CVS:
27662 * gst/adder/gstadder.c: (gst_adder_request_new_pad),
27663 (gst_adder_collected):
27664 * gst/adder/gstadder.h:
27665 Remove bogus segment merging and forwarding, we don't
27666 care about timestamps anyway and we just produce a
27668 Also create a nice NEWSEGMENT event when we start.
27669 Use _scale_int some more.
27671 2006-05-09 11:59:13 +0000 Edward Hervey <bilboed@bilboed.com>
27673 tests/icles/stress-xoverlay.c: Fix if core was built without parsing support.
27674 Original commit message from CVS:
27675 * tests/icles/stress-xoverlay.c:
27676 Fix if core was built without parsing support.
27678 2006-05-09 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
27680 gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
27681 Original commit message from CVS:
27682 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
27683 Add SEDG (Samsung MPEG-4) fourcc.
27685 2006-05-09 11:31:47 +0000 Edward Hervey <bilboed@bilboed.com>
27687 tests/examples/volume/volume.c: Fox if core was built without parsing support.
27688 Original commit message from CVS:
27689 * tests/examples/volume/volume.c:
27690 Fox if core was built without parsing support.
27691 * tests/examples/seek/seek.c:
27692 Disable the parse_launch example if core was built without parsing
27695 2006-05-09 11:21:24 +0000 Edward Hervey <bilboed@bilboed.com>
27697 tests/examples/seek/seek.c: Disable the parse_launch example if core was built without parsing support.
27698 Original commit message from CVS:
27699 * tests/examples/seek/seek.c:
27700 Disable the parse_launch example if core was built without parsing
27703 2006-05-08 15:51:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27705 * docs/libs/tmpl/gstcolorbalance.sgml:
27706 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
27707 * gst/tcp/gstmultifdsink.c:
27708 * gst/videoscale/gstvideoscale.c:
27709 doc reparagraphing and DEBUG_FUNCPTRing
27710 Original commit message from CVS:
27711 doc reparagraphing and DEBUG_FUNCPTRing
27713 2006-05-08 11:51:43 +0000 Edward Hervey <bilboed@bilboed.com>
27715 autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize
27716 Original commit message from CVS:
27717 * autogen.sh: (CONFIGURE_DEF_OPT):
27718 libtoolize on Darwin/MacOSX is called glibtoolize
27720 2006-05-07 17:39:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27722 tests/check/: Disable the adder test, until the build-slaves posses the kindness to either like it or to give valid r...
27723 Original commit message from CVS:
27724 * tests/check/Makefile.am:
27725 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
27726 Disable the adder test, until the build-slaves posses the kindness to
27727 either like it or to give valid reason for not doing so
27729 2006-05-07 17:25:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27731 tests/check/elements/adder.c: Shuffle NULL state change around and raise timeout more
27732 Original commit message from CVS:
27733 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
27735 Shuffle NULL state change around and raise timeout more
27737 2006-05-07 17:07:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27739 gst/typefind/gsttypefindfunctions.c: Add typefind to distinguish between "audio/x-m4a" and new type "video/mp4". Fixe...
27740 Original commit message from CVS:
27741 * gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
27742 (mp4_type_find), (plugin_init):
27743 Add typefind to distinguish between "audio/x-m4a" and new type
27744 "video/mp4". Fixes #340375
27745 * tests/check/elements/adder.c: (adder_suite):
27746 Raise timeout to make buildbot happy
27748 2006-05-07 16:39:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27750 Add sink-event handling to adder. It tries to merge incomming newsegment-events. Added test to check if segment_done ...
27751 Original commit message from CVS:
27752 * gst/adder/gstadder.c: (gst_adder_sink_event),
27753 (gst_adder_request_new_pad), (gst_adder_change_state):
27754 * gst/adder/gstadder.h:
27755 * tests/check/Makefile.am:
27756 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
27757 (adder_suite), (main):
27758 Add sink-event handling to adder. It tries to merge incomming
27759 newsegment-events. Added test to check if segment_done is comming
27762 2006-05-05 16:34:15 +0000 Andy Wingo <wingo@pobox.com>
27765 * ext/theora/theoraparse.c:
27766 * ext/vorbis/vorbisparse.c:
27767 ext/theora/theoraparse.c (gst_theora_parse_init) ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
27768 Original commit message from CVS:
27769 2006-05-05 Andy Wingo <wingo@pobox.com>
27770 * ext/theora/theoraparse.c (gst_theora_parse_init)
27771 (theora_parse_src_convert, theora_parse_src_query):
27772 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
27773 (vorbis_parse_convert, vorbis_parse_src_query): Add convert and
27774 query functions on the source pads of the theora and vorbis parse
27775 elements. Fixes position querying when doing a remux.
27777 2006-05-05 13:46:37 +0000 Michael Smith <msmith@xiph.org>
27779 ext/theora/theoraparse.c: Fix flushing.
27780 Original commit message from CVS:
27781 * ext/theora/theoraparse.c: (parse_granulepos),
27782 (theora_parse_drain_queue_prematurely),
27783 (theora_parse_queue_buffer), (theora_parse_sink_event):
27785 Fix invalid granulepos outputs when starting with a non-keyframe.
27787 2006-05-05 12:37:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27789 gst/typefind/gsttypefindfunctions.c: Rearrange MPEG system stream detection, fixing some memleaks in the process.
27790 Original commit message from CVS:
27791 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
27792 (mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
27793 Rearrange MPEG system stream detection, fixing some memleaks in the
27795 Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
27796 they clean up their data correctly.
27797 Remove unused ogganx caps and move the 'is_annodex' check to inside
27798 the 'is_ogg' if statement.
27800 2006-05-05 11:33:37 +0000 Wim Taymans <wim.taymans@gmail.com>
27802 gst/playback/gstdecodebin.c: Properly remove ghostpads. Fixes #340392
27803 Original commit message from CVS:
27804 * gst/playback/gstdecodebin.c: (cleanup_decodebin):
27805 Properly remove ghostpads. Fixes #340392
27807 2006-05-04 18:43:58 +0000 David Schleef <ds@schleef.org>
27809 gst/typefind/gsttypefindfunctions.c:
27810 Original commit message from CVS:
27811 * gst/typefind/gsttypefindfunctions.c:
27813 2006-05-03 16:32:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27815 gst/typefind/gsttypefindfunctions.c: When typefinding an MP3 in push-based mode, don't penalise the probability down ...
27816 Original commit message from CVS:
27817 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
27818 (mpeg_ts_probe_headers), (mpeg_ts_type_find):
27819 When typefinding an MP3 in push-based mode, don't penalise the
27820 probability down to 74% when we found 5 valid frames just because we
27821 can't peek the end of the file.
27822 Make the probability for detecting MPEG Transport Streams based on the
27823 number of sequential headers we successfully detected.
27825 2006-05-03 15:52:46 +0000 Wim Taymans <wim.taymans@gmail.com>
27827 ext/vorbis/vorbisdec.c: Still produce an error when we receive an empty packet.
27828 Original commit message from CVS:
27829 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
27830 (vorbis_dec_push), (vorbis_dec_chain):
27831 Still produce an error when we receive an empty packet.
27833 2006-05-03 15:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
27835 ext/ogg/gstoggdemux.c: Mark buffers with DISCONT after seek and after activating new chains.
27836 Original commit message from CVS:
27837 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
27838 (gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
27839 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
27840 Mark buffers with DISCONT after seek and after activating new
27842 * ext/theora/gsttheoradec.h:
27843 * ext/theora/theoradec.c: (gst_theora_dec_reset),
27844 (theora_get_query_types), (theora_dec_sink_event),
27845 (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
27846 (theora_dec_change_state):
27848 Detect and mark DISCONT buffers.
27849 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
27850 (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
27851 (vorbis_dec_change_state):
27852 * ext/vorbis/vorbisdec.h:
27854 Detect and mark DISCONT buffers.
27855 Don't crash on 0 sized buffers.
27857 2006-05-03 08:58:13 +0000 Wim Taymans <wim.taymans@gmail.com>
27859 gst/volume/gstvolume.c: Increase "volume" property to 10.0. Fixes #340369.
27860 Original commit message from CVS:
27861 * gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
27862 (volume_transform_ip):
27863 Increase "volume" property to 10.0. Fixes #340369.
27864 Set the process function to NULL when capsnego fails so that
27865 we properly error out.
27867 2006-05-02 18:15:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27869 gst/playback/: free cpas using gst_caps_unref, don't leak caps-strings
27870 Original commit message from CVS:
27871 * gst/playback/gstplaybin.c: (add_sink):
27872 * gst/playback/test.c: (main):
27873 * gst/playback/test5.c: (dump_element_stats):
27874 * gst/playback/test6.c: (main):
27875 free cpas using gst_caps_unref, don't leak caps-strings
27877 2006-05-02 06:33:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27879 * gst-libs/gst/rtp/gstbasertppayload.c:
27881 Original commit message from CVS:
27884 2006-05-01 19:08:40 +0000 Tim-Philipp Müller <tim@centricular.net>
27886 gst/typefind/gsttypefindfunctions.c: Refine musepack typefinding a bit. Return MAXIMUM probability when we detect str...
27887 Original commit message from CVS:
27888 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
27890 Refine musepack typefinding a bit. Return MAXIMUM
27891 probability when we detect stream version 7 to make
27892 sure the mpeg audio typefinder doesn't trump us.
27894 2006-04-29 16:25:58 +0000 Tim-Philipp Müller <tim@centricular.net>
27896 gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
27897 Original commit message from CVS:
27898 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
27899 Protect against unexpected NULL strf_data buffer.
27901 2006-04-29 13:09:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27903 tests/check/elements/audioconvert.c: interpret the out[] buffer in the order the bytes are actually put in, which is ...
27904 Original commit message from CVS:
27905 * tests/check/elements/audioconvert.c: (verify_convert),
27907 interpret the out[] buffer in the order the bytes are actually
27908 put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
27909 Other tests should use BYTE_ORDER since the array is filled in
27912 2006-04-29 12:10:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27914 * tests/check/elements/audioconvert.c:
27915 dump expected data when audioconvert test fails
27916 Original commit message from CVS:
27917 dump expected data when audioconvert test fails
27919 2006-04-29 11:55:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27921 tests/check/elements/audioconvert.c: when a test fails, give an indication of which it is
27922 Original commit message from CVS:
27923 * tests/check/elements/audioconvert.c: (verify_convert),
27925 when a test fails, give an indication of which it is
27927 2006-04-29 09:48:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27929 * ext/ogg/gstoggmux.c:
27930 * ext/theora/theoraenc.c:
27931 add another include
27932 Original commit message from CVS:
27933 add another include
27935 2006-04-29 01:24:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27937 * gst/subparse/gstssaparse.c:
27938 atoi() needs stdlib.h
27939 Original commit message from CVS:
27940 atoi() needs stdlib.h
27942 2006-04-29 01:18:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27944 * gst/playback/test4.c:
27945 * gst/playback/test5.c:
27946 * gst/playback/test6.c:
27947 exit needs stdlib.h
27948 Original commit message from CVS:
27949 exit needs stdlib.h
27951 2006-04-29 01:10:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27953 gst-libs/gst/cdda/gstcddabasesrc.c: compile fix; strtol() needs <stdlib.h>
27954 Original commit message from CVS:
27955 * gst-libs/gst/cdda/gstcddabasesrc.c:
27956 compile fix; strtol() needs <stdlib.h>
27958 2006-04-29 01:04:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27962 * docs/Makefile.am:
27963 * docs/libs/Makefile.am:
27964 * docs/libs/tmpl/gstcolorbalance.sgml:
27965 * docs/plugins/Makefile.am:
27967 use common upload.mak
27968 Original commit message from CVS:
27969 use common upload.mak
27971 2006-04-28 19:46:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27973 make GstElementDetails const
27974 Original commit message from CVS:
27975 * ext/alsa/gstalsamixerelement.c:
27976 * ext/alsa/gstalsasrc.c:
27977 * ext/cdparanoia/gstcdparanoiasrc.c:
27978 * ext/gnomevfs/gstgnomevfssink.c:
27979 * ext/gnomevfs/gstgnomevfssrc.c:
27980 * ext/ogg/gstoggdemux.c:
27981 * ext/ogg/gstoggmux.c:
27982 * ext/ogg/gstoggparse.c:
27983 * ext/ogg/gstogmparse.c:
27984 * ext/pango/gstclockoverlay.c:
27985 * ext/pango/gsttextoverlay.c:
27986 * ext/pango/gsttextrender.c:
27987 * ext/pango/gsttimeoverlay.c:
27988 * ext/theora/theoradec.c:
27989 * ext/theora/theoraenc.c:
27990 * ext/vorbis/vorbisdec.c:
27991 * ext/vorbis/vorbisenc.c:
27992 * gst-libs/gst/audio/gstaudiofilter.c:
27993 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
27994 * gst/audioconvert/gstaudioconvert.c:
27995 * gst/audiorate/gstaudiorate.c:
27996 * gst/audioresample/gstaudioresample.c:
27997 * gst/audiotestsrc/gstaudiotestsrc.c:
27998 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
27999 * gst/playback/gstdecodebin.c:
28000 * gst/playback/gstplaybin.c:
28001 * gst/playback/gststreamselector.c:
28002 * gst/subparse/gstsubparse.c:
28003 * gst/tcp/gstmultifdsink.c:
28004 * gst/tcp/gsttcpclientsink.c:
28005 * gst/tcp/gsttcpclientsrc.c:
28006 * gst/tcp/gsttcpserversink.c:
28007 * gst/tcp/gsttcpserversrc.c:
28008 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
28009 * gst/videorate/gstvideorate.c:
28010 * gst/videoscale/gstvideoscale.c:
28011 * gst/videotestsrc/gstvideotestsrc.c:
28012 * gst/volume/gstvolume.c:
28013 * sys/v4l/gstv4ljpegsrc.c:
28014 * sys/v4l/gstv4lmjpegsink.c:
28015 * sys/v4l/gstv4lmjpegsrc.c:
28016 * sys/v4l/gstv4lsrc.c:
28017 * sys/ximage/ximagesink.c:
28018 * sys/xvimage/xvimagesink.c:
28019 * tests/check/libs/cddabasesrc.c:
28020 make GstElementDetails const
28022 2006-04-28 19:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28024 gst/adder/gstadder.c: send events from src-pad to all sink-pads fixes #338657
28025 Original commit message from CVS:
28026 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
28028 send events from src-pad to all sink-pads fixes #338657
28030 2006-04-28 19:08:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28032 ext/alsa/gstalsasink.c: query witdh capabilities from alsa, fixes #338919
28033 Original commit message from CVS:
28034 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
28035 (alsasink_parse_spec):
28036 query witdh capabilities from alsa, fixes #338919
28038 2006-04-28 15:31:28 +0000 Wim Taymans <wim.taymans@gmail.com>
28040 gst/tcp/gstmultifdsink.*: Fix race condition in multifdsink that can lead to spurious duplicate clients. this patch a...
28041 Original commit message from CVS:
28042 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
28043 (gst_multi_fd_sink_remove_client_link):
28044 * gst/tcp/gstmultifdsink.h:
28045 Fix race condition in multifdsink that can lead to spurious
28046 duplicate clients. this patch adds a new signal that is fired when
28047 multifdsink has removed all references to the fd.
28049 Updated documentation.
28050 API: client-fd-removed signal added
28052 2006-04-28 15:24:00 +0000 Michael Smith <msmith@xiph.org>
28054 gst/tcp/gstmultifdsink.c: When asking g_value_array_new to prealloc elements, we may as well ask for the right number...
28055 Original commit message from CVS:
28056 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
28057 When asking g_value_array_new to prealloc elements, we may as well
28058 ask for the right number of elements.
28060 2006-04-28 15:08:09 +0000 Wim Taymans <wim.taymans@gmail.com>
28062 gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
28063 Original commit message from CVS:
28064 * gst-libs/gst/audio/gstbaseaudiosink.c:
28065 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
28066 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
28067 patch to make timestamp checking more tollerant to rounding
28068 errors given that real discontinuities are to be marked on
28069 buffers. Fixes some asf files and #338778.
28070 Also avoid some crashers when we receive an event in the
28073 2006-04-28 15:01:58 +0000 Michael Smith <msmith@xiph.org>
28075 ext/gnomevfs/gstgnomevfssrc.*: Remove ICY handling (mostly) from gnomevfssrc, in favour of proper shared support with...
28076 Original commit message from CVS:
28077 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
28078 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
28079 (gst_gnome_vfs_src_get_property),
28080 (gst_gnome_vfs_src_send_additional_headers_callback),
28081 (gst_gnome_vfs_src_received_headers_callback),
28082 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
28083 (gst_gnome_vfs_src_stop):
28084 * ext/gnomevfs/gstgnomevfssrc.h:
28085 Remove ICY handling (mostly) from gnomevfssrc, in favour of
28086 proper shared support within icydemux.
28088 2006-04-28 14:49:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28090 gst/videorate/gstvideorate.c: fix up docs fix a leak when no caps negotiated fix counting of input frames
28091 Original commit message from CVS:
28092 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
28093 (gst_video_rate_swap_prev), (gst_video_rate_chain):
28095 fix a leak when no caps negotiated
28096 fix counting of input frames
28097 * tests/check/elements/.cvsignore:
28098 * tests/check/elements/videorate.c: (assert_videorate_stats),
28099 (GST_START_TEST), (videorate_suite):
28100 add tests for these
28102 2006-04-28 14:48:11 +0000 Wim Taymans <wim.taymans@gmail.com>
28104 gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
28105 Original commit message from CVS:
28106 * gst-libs/gst/audio/gstringbuffer.c:
28107 (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
28108 (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
28109 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
28110 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
28111 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
28112 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
28113 (gst_ring_buffer_commit), (gst_ring_buffer_read),
28114 (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
28115 (gst_ring_buffer_clear), (gst_ring_buffer_may_start):
28116 Check arguments passed to public functions instead of
28119 2006-04-28 14:37:46 +0000 Wim Taymans <wim.taymans@gmail.com>
28121 gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
28122 Original commit message from CVS:
28123 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
28124 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
28125 GstBaseAudioSrc must be live or it does not work.
28126 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
28127 Don't set live to TRUE as this is the default in the parentclass.
28129 2006-04-28 14:37:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28131 * win32/common/config.h:
28133 Original commit message from CVS:
28136 2006-04-28 14:33:45 +0000 Wim Taymans <wim.taymans@gmail.com>
28138 gst/videoscale/gstvideoscale.c: Videoscale doesn't pass on pixel-aspect ratio. Handle all fixation cases better. Fixe...
28139 Original commit message from CVS:
28140 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps),
28141 (gst_video_scale_fixate_caps), (gst_video_scale_src_event):
28142 Videoscale doesn't pass on pixel-aspect ratio. Handle all
28143 fixation cases better. Fixes #338991
28145 2006-04-28 14:24:38 +0000 Wim Taymans <wim.taymans@gmail.com>
28147 gst/videotestsrc/gstvideotestsrc.c: Handle 0/1 framerate correctly Fixes #331901.
28148 Original commit message from CVS:
28149 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
28150 Handle 0/1 framerate correctly Fixes #331901.
28152 2006-04-28 14:22:16 +0000 Wim Taymans <wim.taymans@gmail.com>
28154 tests/check/elements/audioconvert.c: Added check for correct clipping when doing float samples in audioconvert.
28155 Original commit message from CVS:
28156 * tests/check/elements/audioconvert.c: (get_float_caps),
28157 (GST_START_TEST), (audioconvert_suite):
28158 Added check for correct clipping when doing float samples
28161 2006-04-28 14:19:49 +0000 Wim Taymans <wim.taymans@gmail.com>
28163 gst/videorate/gstvideorate.c: Print more debugging info.
28164 Original commit message from CVS:
28165 * gst/videorate/gstvideorate.c: (gst_video_rate_event),
28166 (gst_video_rate_chain):
28167 Print more debugging info.
28169 2006-04-28 14:17:00 +0000 Wim Taymans <wim.taymans@gmail.com>
28171 gst/audioresample/gstaudioresample.c: Add support for other formats audioresample can handle such as 32 bits in and f...
28172 Original commit message from CVS:
28173 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
28174 (resample_set_state_from_caps):
28175 Add support for other formats audioresample can handle such as
28176 32 bits in and float and 64 bits float. Fixes #301759
28178 2006-04-28 14:12:28 +0000 Wim Taymans <wim.taymans@gmail.com>
28180 gst/audioconvert/audioconvert.c: correctly clip float samples > 1.0. Fixes #338718
28181 Original commit message from CVS:
28182 * gst/audioconvert/audioconvert.c: (float):
28183 correctly clip float samples > 1.0. Fixes #338718
28185 2006-04-28 13:35:34 +0000 Young-Ho Cha <ganadist@chollian.net>
28187 ext/pango/gsttextoverlay.c: Don't strip newlines from the text. Also, center lines within multi-line paragraphs (#339...
28188 Original commit message from CVS:
28189 Patch by: Young-Ho Cha <ganadist at chollian net>
28190 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
28191 (gst_text_overlay_render_text):
28192 Don't strip newlines from the text. Also, center lines
28193 within multi-line paragraphs (#339405).
28195 2006-04-28 12:15:33 +0000 Tim-Philipp Müller <tim@centricular.net>
28197 gst/typefind/gsttypefindfunctions.c: Fix wavpack typefinding to work in more cases (don't peek for chunks of multiple...
28198 Original commit message from CVS:
28199 * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
28200 Fix wavpack typefinding to work in more cases (don't peek
28201 for chunks of multiple hundred kBs at once, but process
28202 things step-by-step in smaller units). Fixes #339786.
28204 2006-04-28 10:58:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28209 Original commit message from CVS:
28212 === release 0.10.6 ===
28214 2006-04-28 10:53:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28220 * docs/plugins/gst-plugins-base-plugins.signals:
28221 * docs/plugins/inspect/plugin-adder.xml:
28222 * docs/plugins/inspect/plugin-alsa.xml:
28223 * docs/plugins/inspect/plugin-audioconvert.xml:
28224 * docs/plugins/inspect/plugin-audiorate.xml:
28225 * docs/plugins/inspect/plugin-audioresample.xml:
28226 * docs/plugins/inspect/plugin-audiotestsrc.xml:
28227 * docs/plugins/inspect/plugin-cdparanoia.xml:
28228 * docs/plugins/inspect/plugin-decodebin.xml:
28229 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
28230 * docs/plugins/inspect/plugin-gnomevfs.xml:
28231 * docs/plugins/inspect/plugin-libvisual.xml:
28232 * docs/plugins/inspect/plugin-ogg.xml:
28233 * docs/plugins/inspect/plugin-pango.xml:
28234 * docs/plugins/inspect/plugin-playbin.xml:
28235 * docs/plugins/inspect/plugin-subparse.xml:
28236 * docs/plugins/inspect/plugin-tcp.xml:
28237 * docs/plugins/inspect/plugin-theora.xml:
28238 * docs/plugins/inspect/plugin-typefindfunctions.xml:
28239 * docs/plugins/inspect/plugin-video4linux.xml:
28240 * docs/plugins/inspect/plugin-videorate.xml:
28241 * docs/plugins/inspect/plugin-videoscale.xml:
28242 * docs/plugins/inspect/plugin-videotestsrc.xml:
28243 * docs/plugins/inspect/plugin-volume.xml:
28244 * docs/plugins/inspect/plugin-vorbis.xml:
28245 * docs/plugins/inspect/plugin-ximagesink.xml:
28246 * docs/plugins/inspect/plugin-xvimagesink.xml:
28249 Original commit message from CVS:
28252 2006-04-28 10:42:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28255 * win32/common/config.h:
28256 dist more win32 files
28257 Original commit message from CVS:
28258 dist more win32 files
28260 2006-04-28 10:41:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28277 Original commit message from CVS:
28280 2006-04-27 00:19:29 +0000 David Schleef <ds@schleef.org>
28282 gst/videoscale/gstvideoscale.c: Add call to oil_init().
28283 Original commit message from CVS:
28284 * gst/videoscale/gstvideoscale.c: Add call to oil_init().
28287 2006-04-26 17:20:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28291 * win32/common/config.h:
28293 Original commit message from CVS:
28296 2006-04-26 17:17:39 +0000 Wim Taymans <wim.taymans@gmail.com>
28298 ext/ogg/gstoggdemux.c: make sure correct newsegments are sent, so that the decoder and the demuxer agree on timestamp...
28299 Original commit message from CVS:
28300 2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
28301 patch by: Wim Taymans
28302 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
28303 (gst_ogg_demux_perform_seek):
28304 make sure correct newsegments are sent, so that the decoder
28305 and the demuxer agree on timestamps. Fixes playback of a lot
28306 of Ogg files that do not start from 0. Fixes #339833.
28308 2006-04-26 16:44:20 +0000 Edward Hervey <edward@fluendo.com>
28310 Fix an infinite loop if frames are passed in with wrongly ordered timestamps. Fixes #339013.
28311 Original commit message from CVS:
28312 Patch by: Edward Hervey <edward@fluendo.com>
28313 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
28314 * tests/check/Makefile.am:
28315 * tests/check/elements/videorate.c: (assert_videorate_stats),
28316 (setup_videorate), (cleanup_videorate), (GST_START_TEST),
28317 (videorate_suite), (main):
28318 Fix an infinite loop if frames are passed in with wrongly ordered
28319 timestamps. Fixes #339013.
28321 2006-04-26 13:55:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28324 * win32/common/config.h:
28326 Original commit message from CVS:
28329 2006-04-22 21:25:41 +0000 Tim-Philipp Müller <tim@centricular.net>
28331 gst/typefind/gsttypefindfunctions.c: fix typefinding on some ISO files. Fixes #339212.
28332 Original commit message from CVS:
28333 Patch by: Tim-Philipp Müller <tim at centricular dot net>
28334 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
28335 fix typefinding on some ISO files. Fixes #339212.
28337 2006-04-22 21:19:06 +0000 Tim-Philipp Müller <tim@centricular.net>
28339 gst-libs/gst/riff/riff-media.c: add another H264 fourcc. Fixes #339047.
28340 Original commit message from CVS:
28341 Patch by: Tim-Philipp Müller <tim at centricular dot net>
28342 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
28343 add another H264 fourcc. Fixes #339047.
28345 2006-04-22 21:12:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28347 gst/playback/gststreamselector.c: Restore old StreamSelector behaviour.
28348 Original commit message from CVS:
28349 Patch by: Jan Schmidt
28350 * gst/playback/gststreamselector.c:
28351 (gst_stream_selector_bufferalloc):
28352 Restore old StreamSelector behaviour.
28355 2006-04-13 09:26:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28358 * gst-libs/gst/rtp/Makefile.am:
28359 * gst-libs/gst/rtp/gstrtpbuffer.h:
28360 reverting rtp patches to fix freeze break on -base as explained on the list
28361 Original commit message from CVS:
28362 reverting rtp patches to fix freeze break on -base as explained on the list
28364 2006-04-13 03:55:12 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28366 gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
28367 Original commit message from CVS:
28368 2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28369 * gst-libs/gst/rtp/gstrtpbuffer.h:
28370 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
28371 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
28372 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
28373 New RTP audio base payloader class. Supports frame or sample based codecs
28375 2006-04-12 11:04:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28393 update libtool versioning
28394 Original commit message from CVS:
28395 update libtool versioning
28397 2006-04-12 10:58:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28401 * win32/common/config.h:
28403 Original commit message from CVS:
28406 2006-04-11 17:31:29 +0000 Antoine Tremblay <hexa00@gmail.com>
28408 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
28409 Original commit message from CVS:
28410 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
28411 * gst-libs/gst/rtp/gstbasertpdepayload.c:
28412 (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
28413 Fix some memory leaks: on finalize, free buffers left in the queue
28414 before destroying the queue; in _push(), unref rtp_buf even if
28415 the process vfunc returned a NULL buffer as output buffer (#337548);
28416 demote some recuring debug messages to LOG level.
28418 2006-04-11 15:01:51 +0000 Christian Schaller <uraeus@gnome.org>
28420 * gst-plugins-base.spec.in:
28421 fix version number macro
28422 Original commit message from CVS:
28423 fix version number macro
28425 2006-04-11 14:42:33 +0000 Wim Taymans <wim.taymans@gmail.com>
28427 ext/ogg/gstoggdemux.c: More cleanups.
28428 Original commit message from CVS:
28429 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
28430 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
28431 (gst_ogg_chain_free), (gst_ogg_demux_sink_event),
28432 (gst_ogg_demux_loop):
28434 Respect segment stop when emiting EOS or SEGMENT_DONE.
28437 2006-04-11 10:45:32 +0000 Tim-Philipp Müller <tim@centricular.net>
28439 gst/playback/gststreamselector.c: Don't leak pad name.
28440 Original commit message from CVS:
28441 * gst/playback/gststreamselector.c:
28442 (gst_stream_selector_get_property):
28443 Don't leak pad name.
28445 2006-04-11 09:42:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28448 Mention bug #336617 closed by recent commit
28449 Original commit message from CVS:
28450 Mention bug #336617 closed by recent commit
28452 2006-04-10 20:32:46 +0000 Michael Smith <msmith@xiph.org>
28454 tests/check/: so that FC4 buildslaves can pass.
28455 Original commit message from CVS:
28456 * tests/check/Makefile.am:
28457 * tests/check/gst-plugins-base.supp:
28458 Suppress an old libtheora bug (fixed in more recent versions), so
28459 that FC4 buildslaves can pass.
28461 2006-04-10 19:13:30 +0000 Wim Taymans <wim.taymans@gmail.com>
28463 ext/ogg/gstoggdemux.c: Don't leak events.
28464 Original commit message from CVS:
28465 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
28466 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
28467 (gst_ogg_demux_init), (gst_ogg_demux_finalize),
28468 (gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
28469 (gst_ogg_demux_loop):
28471 Remember what error we got when finding chains, if we
28472 were shutdown, that would not be an error.
28474 2006-04-10 17:05:46 +0000 Wim Taymans <wim.taymans@gmail.com>
28476 gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
28477 Original commit message from CVS:
28478 * gst-libs/gst/audio/gstbaseaudiosink.c:
28479 (gst_base_audio_sink_event):
28480 Starting the ringbuffer when we did not acquire it can cause
28481 a deadlock, is pointless and causes nasty things for
28483 Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
28485 2006-04-10 15:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
28487 ext/ogg/gstoggdemux.c: Add some more debugging.
28488 Original commit message from CVS:
28489 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
28490 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
28491 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
28492 (gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
28493 (gst_ogg_demux_deactivate_current_chain),
28494 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
28495 (gst_ogg_demux_bisect_forward_serialno),
28496 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
28497 Add some more debugging.
28499 2006-04-10 14:52:10 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28502 * ext/theora/theoraenc.c:
28504 Original commit message from CVS:
28507 2006-04-10 10:29:21 +0000 Wim Taymans <wim.taymans@gmail.com>
28509 ext/theora/theoradec.c: Some more debug info.
28510 Original commit message from CVS:
28511 * ext/theora/theoradec.c: (theora_dec_src_event),
28512 (theora_handle_data_packet):
28513 Some more debug info.
28514 * tests/examples/seek/seek.c: (start_seek), (main):
28515 Print element messages too.
28517 2006-04-09 17:14:22 +0000 Sébastien Moutte <sebastien@moutte.net>
28519 gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer sta...
28520 Original commit message from CVS:
28521 * gst/audioresample/debug.h:
28522 replace debug macros with variable number of parameters
28523 by a simple alias to gstreamer standard debug macros
28524 (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
28525 supported by MSVC 6.0 and 7.1)
28526 * gst/audioresample/resample.h:
28527 define M_PI and rint for WIN32
28528 * win32/common/libgstaudio.def:
28529 * win32/common/libgstriff.def:
28530 * win32/common/libgsttag.def:
28531 * win32/common/libgstvideo.def:
28532 add new exported functions
28534 update project files
28536 2006-04-08 21:02:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28538 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
28539 Original commit message from CVS:
28540 * ext/alsa/gstalsamixeroptions.c:
28541 (gst_alsa_mixer_options_class_init):
28542 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
28543 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
28544 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
28545 * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
28546 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
28547 * gst-libs/gst/audio/gstaudiofilter.c:
28548 (gst_audio_filter_class_init):
28549 * gst-libs/gst/audio/gstaudiosink.c:
28550 (gst_audioringbuffer_class_init):
28551 * gst-libs/gst/audio/gstaudiosrc.c:
28552 (gst_audioringbuffer_class_init):
28553 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
28554 * gst-libs/gst/interfaces/colorbalancechannel.c:
28555 (gst_color_balance_channel_class_init):
28556 * gst-libs/gst/interfaces/mixeroptions.c:
28557 (gst_mixer_options_class_init):
28558 * gst-libs/gst/interfaces/mixertrack.c:
28559 (gst_mixer_track_class_init):
28560 * gst-libs/gst/interfaces/tunerchannel.c:
28561 (gst_tuner_channel_class_init):
28562 * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
28563 * gst-libs/gst/netbuffer/gstnetbuffer.c:
28564 (gst_netbuffer_class_init):
28565 * gst-libs/gst/rtp/gstbasertppayload.c:
28566 (gst_basertppayload_class_init):
28567 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
28568 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
28569 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
28570 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
28571 * gst/playback/gststreamselector.c:
28572 (gst_stream_selector_class_init):
28573 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
28574 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
28575 * sys/v4l/gstv4lcolorbalance.c:
28576 (gst_v4l_color_balance_channel_class_init):
28577 * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
28578 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
28579 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
28580 * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
28581 (gst_v4l_tuner_norm_class_init):
28582 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
28583 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
28584 * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
28585 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
28587 2006-04-08 18:09:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28589 Fix broken GObject macros
28590 Original commit message from CVS:
28591 * ext/pango/gsttextrender.h:
28592 * gst-libs/gst/audio/gstaudiosink.h:
28593 * gst-libs/gst/audio/gstaudiosrc.h:
28594 * gst-libs/gst/audio/gstbaseaudiosink.h:
28595 * gst-libs/gst/audio/gstbaseaudiosrc.h:
28596 * gst-libs/gst/audio/gstringbuffer.h:
28597 * gst-libs/gst/rtp/gstbasertpdepayload.h:
28598 * gst-libs/gst/rtp/gstbasertppayload.h:
28599 * gst-libs/gst/video/gstvideofilter.h:
28600 * gst-libs/gst/video/gstvideosink.h:
28601 * gst/playback/gstplaybasebin.h:
28602 * gst/tcp/gstmultifdsink.h:
28603 * sys/v4l/gstv4lelement.h:
28604 Fix broken GObject macros
28606 2006-04-08 16:21:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28608 ext/alsa/gstalsasink.c: More debug to trace why my USB headset is not working with gst
28609 Original commit message from CVS:
28610 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
28611 More debug to trace why my USB headset is not working with gst
28613 2006-04-07 17:18:11 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28615 gst/playback/gstplaybasebin.c: Clean up our group elements properly in the case where it never got committed - it sti...
28616 Original commit message from CVS:
28617 * gst/playback/gstplaybasebin.c: (group_destroy):
28618 Clean up our group elements properly in the case where it never
28619 got committed - it still got added unconditionally to the bin.
28621 2006-04-07 15:14:32 +0000 Wim Taymans <wim.taymans@gmail.com>
28623 ext/theora/theoradec.c: Unref unhandled events.
28624 Original commit message from CVS:
28625 * ext/theora/theoradec.c: (theora_dec_sink_event),
28626 (theora_handle_data_packet), (theora_dec_chain):
28627 Unref unhandled events.
28628 Protect against empty buffers.
28629 Perform QoS on running time.
28631 2006-04-07 13:24:54 +0000 Michael Smith <msmith@xiph.org>
28633 ext/vorbis/vorbisenc.c: Remove leaks from vorbisenc.
28634 Original commit message from CVS:
28635 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
28636 (gst_vorbis_enc_chain):
28637 Remove leaks from vorbisenc.
28638 Mostly minor changes, the only significant one is that now the
28639 buffers we set as 'streamheader' on the caps are copies of the
28640 original buffers, to avoid circular refcounting problems.
28642 2006-04-07 09:51:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28644 gst/playback/gstplaybasebin.c: Don't remove our mute-probe if someone else already did so.
28645 Original commit message from CVS:
28646 * gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
28647 Don't remove our mute-probe if someone else already did so.
28648 Don't set a 2nd one if there is already one pending on the pad.
28649 * gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
28651 When a seek fails, ensure that playbin is still set back to playing.
28652 * gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
28653 (mpeg_ts_type_find), (plugin_init):
28654 Add a typefind function for mpeg-ts streams.
28656 2006-04-06 11:40:45 +0000 Andy Wingo <wingo@pobox.com>
28659 * gst/audiotestsrc/gstaudiotestsrc.c:
28660 * gst/videorate/gstvideorate.c:
28661 gst/videorate/gstvideorate.c (gst_video_rate_reset)
28662 Original commit message from CVS:
28663 2006-04-06 Andy Wingo <wingo@pobox.com>
28664 * gst/videorate/gstvideorate.c (gst_video_rate_reset)
28665 (gst_video_rate_init): Caps-related parameters should not be reset
28666 by a flush -- move their inits to the instance init function.
28667 (gst_video_rate_flush_prev): Don't complain if gst_pad_push
28668 is not OK, just return the result.
28669 * gst/audiotestsrc/gstaudiotestsrc.c
28670 (gst_audio_test_src_class_init)
28671 (gst_audio_test_src_get_times): Re-enable is-live=true, as was
28672 broken by Stefan's commit on 24 March.
28674 2006-04-06 10:50:14 +0000 Andy Wingo <wingo@pobox.com>
28676 ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on buffers being pushed out. Fixes oggmux ! multifdsink.
28677 Original commit message from CVS:
28678 2006-04-06 Andy Wingo <wingo@pobox.com>
28679 * ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
28680 buffers being pushed out. Fixes oggmux ! multifdsink.
28682 2006-04-05 13:05:25 +0000 Tim-Philipp Müller <tim@centricular.net>
28684 ext/vorbis/: Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make vorbisenc adhere to the official nomenclature; u...
28685 Original commit message from CVS:
28686 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
28687 (gst_vorbis_dec_init), (vorbis_dec_finalize):
28688 * ext/vorbis/vorbisdec.h:
28689 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_add_interfaces),
28690 (gst_vorbis_enc_base_init), (gst_vorbis_enc_class_init),
28691 (gst_vorbis_enc_sink_setcaps), (gst_vorbis_enc_convert_src),
28692 (gst_vorbis_enc_convert_sink), (gst_vorbis_enc_get_query_types),
28693 (gst_vorbis_enc_src_query), (gst_vorbis_enc_sink_query),
28694 (gst_vorbis_enc_init), (gst_vorbis_enc_get_tag_value),
28695 (gst_vorbis_enc_metadata_set1), (gst_vorbis_enc_set_metadata),
28696 (gst_vorbis_enc_setup), (gst_vorbis_enc_clear),
28697 (gst_vorbis_enc_buffer_from_packet),
28698 (gst_vorbis_enc_buffer_from_header_packet),
28699 (gst_vorbis_enc_push_buffer), (gst_vorbis_enc_push_packet),
28700 (gst_vorbis_enc_set_header_on_caps), (gst_vorbis_enc_sink_event),
28701 (gst_vorbis_enc_chain), (gst_vorbis_enc_output_buffers),
28702 (gst_vorbis_enc_get_property), (gst_vorbis_enc_set_property),
28703 (gst_vorbis_enc_change_state):
28704 * ext/vorbis/vorbisenc.h:
28705 Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make
28706 vorbisenc adhere to the official nomenclature; use boilerplate
28709 2006-04-04 11:20:24 +0000 Andy Wingo <wingo@pobox.com>
28711 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Whoops, fix bug introduced. Bad hacker!
28712 Original commit message from CVS:
28713 2006-04-04 Andy Wingo <wingo@pobox.com>
28714 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
28715 Whoops, fix bug introduced. Bad hacker!
28717 2006-04-04 11:15:00 +0000 Andy Wingo <wingo@pobox.com>
28719 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Properly handle the case where you get EOS before any buffe...
28720 Original commit message from CVS:
28721 2006-04-04 Andy Wingo <wingo@pobox.com>
28722 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
28723 Properly handle the case where you get EOS before any buffers are
28724 received. Use gst_buffer_make_metadata_writable where appropriate.
28726 2006-04-04 10:16:46 +0000 Andy Wingo <wingo@pobox.com>
28728 ext/theora/theoradec.c (theora_handle_data_packet): This value is often negative -- make it signed so as not to wrap ...
28729 Original commit message from CVS:
28730 2006-04-04 Andy Wingo <wingo@pobox.com>
28731 * ext/theora/theoradec.c (theora_handle_data_packet): This value
28732 is often negative -- make it signed so as not to wrap around.
28733 Fixes segfaults introduced on 9 March.
28735 2006-04-03 16:43:10 +0000 Wim Taymans <wim.taymans@gmail.com>
28737 ext/theora/: Don't try to store a gdouble in a gboolean.
28738 Original commit message from CVS:
28739 * ext/theora/gsttheoradec.h:
28740 * ext/theora/theoradec.c: (theora_dec_src_event):
28741 Don't try to store a gdouble in a gboolean.
28744 2006-04-03 12:55:18 +0000 Michael Smith <msmith@xiph.org>
28746 ext/ogg/gstoggmux.c: Oggmux sucks.
28747 Original commit message from CVS:
28748 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
28750 Make it suck slightly less by writing out the final page.
28751 Still can't encode a vorbis-in-ogg file correctly, though.
28753 2006-04-03 08:49:06 +0000 Andy Wingo <wingo@pobox.com>
28755 ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove a g_print.
28756 Original commit message from CVS:
28757 2006-04-03 Andy Wingo <wingo@pobox.com>
28758 * ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
28761 2006-04-03 08:32:21 +0000 Andy Wingo <wingo@pobox.com>
28763 ext/theora/theora.c (plugin_init): Register theoraparse.
28764 Original commit message from CVS:
28765 2006-04-03 Andy Wingo <wingo@pobox.com>
28766 * ext/theora/theora.c (plugin_init): Register theoraparse.
28767 * ext/theora/gsttheoraparse.h:
28768 * ext/theora/theoraparse.c: New files implementing a theora
28769 parser. Now we can properly remux ogg/theora+vorbis, yay.
28771 2006-04-03 08:28:58 +0000 Andy Wingo <wingo@pobox.com>
28773 ext/vorbis/vorbisparse.c: Add some docs and a copyright.
28774 Original commit message from CVS:
28775 2006-04-03 Andy Wingo <wingo@pobox.com>
28776 * ext/vorbis/vorbisparse.c: Add some docs and a copyright.
28778 2006-04-01 15:34:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28782 don't use AS_LIBTOOL_TAGS, it doesn't work
28783 Original commit message from CVS:
28784 don't use AS_LIBTOOL_TAGS, it doesn't work
28786 2006-04-01 11:41:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28789 * ext/pango/gsttextoverlay.c:
28790 * sys/v4l/gstv4lsrc.c:
28791 remove BT8x8 from description, works for more devices
28792 Original commit message from CVS:
28793 remove BT8x8 from description, works for more devices
28795 2006-04-01 11:21:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28797 gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798)
28798 Original commit message from CVS:
28799 * gst/audiotestsrc/gstaudiotestsrc.c:
28800 Fixed the sample pipeline (see #323798)
28802 2006-04-01 09:50:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28804 use AS_VERSION and AS_NANO more cleanups
28805 Original commit message from CVS:
28807 * win32/common/config.h:
28808 * win32/common/config.h.in:
28809 use AS_VERSION and AS_NANO
28812 2006-03-31 17:08:41 +0000 Andy Wingo <wingo@pobox.com>
28814 ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix uninitialized variable return that would happen.
28815 Original commit message from CVS:
28816 2006-03-31 Andy Wingo <wingo@pobox.com>
28817 * ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
28818 uninitialized variable return that would happen.
28820 2006-03-31 16:57:47 +0000 Andy Wingo <wingo@pobox.com>
28822 ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix uninitialized variable return that would never happen.
28823 Original commit message from CVS:
28824 2006-03-31 Andy Wingo <wingo@pobox.com>
28825 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
28826 uninitialized variable return that would never happen.
28828 2006-03-31 16:43:43 +0000 Andy Wingo <wingo@pobox.com>
28830 ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
28831 Original commit message from CVS:
28832 2006-03-31 Andy Wingo <wingo@pobox.com>
28833 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
28834 (vorbis_parse_sink_event): Add an event function to flush our
28835 state on a seek, and to drain buffers on a premature EOS.
28836 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
28837 (vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
28838 (vorbis_parse_chain, vorbis_parse_queue_buffer)
28839 (vorbis_parse_drain_queue): Queue up buffers until we can set
28840 their timestamps and granulepos values.
28841 * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
28842 and keep track of data needed for deriving granulepos and
28843 timestamps for buffers.
28845 2006-03-30 11:05:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28847 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
28848 * pkgconfig/gstreamer-plugins-base.pc.in:
28849 expose pluginsdir so gonlin can use it for tests
28850 Original commit message from CVS:
28851 expose pluginsdir so gonlin can use it for tests
28853 2006-03-30 10:03:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28855 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
28856 * pkgconfig/gstreamer-plugins-base.pc.in:
28857 add ccda to libraries
28858 Original commit message from CVS:
28859 add ccda to libraries
28861 2006-03-29 14:00:08 +0000 j^ <j@bootlab.org>
28863 better/unified long descriptions
28864 Original commit message from CVS:
28865 Patch by: j^ <j at bootlab dot org>
28866 * ext/alsa/gstalsamixerelement.c:
28867 (gst_alsa_mixer_element_class_init):
28868 * ext/alsa/gstalsasink.c:
28869 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
28870 * ext/ogg/gstoggdemux.c:
28871 * ext/ogg/gstoggmux.c:
28872 * ext/ogg/gstoggparse.c:
28873 * ext/pango/gstclockoverlay.c:
28874 * ext/pango/gsttextoverlay.c:
28875 * ext/pango/gsttextrender.c:
28876 * ext/pango/gsttimeoverlay.c:
28877 * ext/theora/theoradec.c:
28878 * ext/theora/theoraenc.c:
28879 * ext/vorbis/vorbisdec.c:
28880 * ext/vorbis/vorbisenc.c:
28881 * gst/audioconvert/gstaudioconvert.c:
28882 * gst/subparse/gstsubparse.c:
28883 * gst/tcp/gstmultifdsink.c:
28884 * gst/tcp/gsttcpclientsink.c:
28885 * gst/tcp/gsttcpclientsrc.c:
28886 * gst/tcp/gsttcpserversink.c:
28887 * gst/tcp/gsttcpserversrc.c:
28888 better/unified long descriptions
28891 2006-03-29 13:54:24 +0000 Wim Taymans <wim.taymans@gmail.com>
28893 tests/examples/seek/seek.c: Don't let double and tripple clicks mess up our state.
28894 Original commit message from CVS:
28895 * tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
28897 Don't let double and tripple clicks mess up our state.
28899 2006-03-28 13:13:43 +0000 Tim-Philipp Müller <tim@centricular.net>
28901 gst/playback/gstplaybin.c: Error out gracefully when we can't create any of the usual conversion elements for some re...
28902 Original commit message from CVS:
28903 * gst/playback/gstplaybin.c: (gen_video_element),
28904 (gen_text_element), (gen_audio_element), (gen_vis_element):
28905 Error out gracefully when we can't create any of the usual
28906 conversion elements for some reason. Also, don't try to
28907 create an audioscale (sic) element that's not used anyway.
28909 2006-03-28 10:21:52 +0000 Tim-Philipp Müller <tim@centricular.net>
28911 gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
28912 Original commit message from CVS:
28913 * gst/playback/gstplaybasebin.c: (setup_source):
28914 Don't post RESOURCE_NOT_FOUND error when we can't find a source
28915 element for a particular protocol, that's confusing for users.
28916 Instead, post a RESOURCE_FAILED error, so that our own error
28917 message is actually shown in totem etc. (#336303).
28919 2006-03-27 16:36:46 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
28921 ext/gnomevfs/gstgnomevfssrc.c: Fix some minor memory leaks (#336194).
28922 Original commit message from CVS:
28923 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
28924 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
28925 (gst_gnome_vfs_src_get_icy_metadata):
28926 Fix some minor memory leaks (#336194).
28928 2006-03-27 16:15:00 +0000 Tim-Philipp Müller <tim@centricular.net>
28930 ext/gnomevfs/: Make gnomevfssink accept filenames as well as URIs for the "location" property, just like gnomevfssrc ...
28931 Original commit message from CVS:
28932 * ext/gnomevfs/gstgnomevfs.c:
28933 (gst_gnome_vfs_location_to_uri_string):
28934 * ext/gnomevfs/gstgnomevfs.h:
28935 * ext/gnomevfs/gstgnomevfssink.c:
28936 (gst_gnome_vfs_sink_set_property):
28937 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
28938 Make gnomevfssink accept filenames as well as URIs for the
28939 "location" property, just like gnomevfssrc does (and
28940 filesrc/filesink do) (#336190).
28942 2006-03-24 20:35:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28944 tests/check/generic/clock-selection.c: set to NULL before unreffing, fixes a valgrind leak.
28945 Original commit message from CVS:
28946 * tests/check/generic/clock-selection.c: (GST_START_TEST):
28947 set to NULL before unreffing, fixes a valgrind leak.
28948 Why was this not triggering the error that an object needs to
28949 be NULL before unreffing ?
28950 * win32/common/config.h:
28953 2006-03-24 17:57:39 +0000 Tim-Philipp Müller <tim@centricular.net>
28955 gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
28956 Original commit message from CVS:
28957 * gst/subparse/gstsubparse.c: (convert_encoding),
28958 (gst_sub_parse_change_state):
28959 * gst/subparse/gstsubparse.h:
28960 Text subtitle files may or may not be UTF-8. If it's not, we
28961 don't really want to see '?' characters in place of non-ASCII
28962 characters like accented characters. So let's assume the input
28963 is UTF-8 until we come across text that is clearly not. If it's
28964 not UTF-8, we don't really know what it is, so try the following:
28965 (a) see whether the GST_SUBTITLE_ENCODING environment variable
28966 is set; if not, check (b) if the current locale encoding is
28967 non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
28968 the current locale encoding is UTF-8 and the environment variable
28969 was not set to any particular encoding. Not perfect, but better
28970 than nothing (and better than before, I think) (fixes #172848).
28972 2006-03-24 17:39:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28974 * docs/plugins/tmpl/.gitignore:
28975 * tests/check/libs/.gitignore:
28976 * tests/check/pipelines/.gitignore:
28977 * tests/examples/volume/.gitignore:
28979 Original commit message from CVS:
28982 2006-03-24 17:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28984 configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
28985 Original commit message from CVS:
28986 2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
28988 update core requirement to 0.10.4.1 because of async_playback
28989 vmethod on GstBaseSink
28991 2006-03-24 17:11:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28993 use DEBUG_FUNCPTR for collectpads
28994 Original commit message from CVS:
28995 * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
28996 * gst/adder/gstadder.c: (gst_adder_init):
28997 use DEBUG_FUNCPTR for collectpads
28999 2006-03-24 14:11:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29002 don't go through check-torture if no check installed
29003 Original commit message from CVS:
29004 don't go through check-torture if no check installed
29006 2006-03-24 10:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29008 Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
29009 Original commit message from CVS:
29010 * docs/plugins/Makefile.am:
29011 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29012 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29013 * ext/cdparanoia/gstcdparanoiasrc.c:
29014 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
29015 (gst_gnome_vfs_sink_class_init):
29016 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
29017 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
29018 * ext/ogg/gstoggmux.c:
29019 * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
29020 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
29021 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
29022 * ext/pango/gsttextoverlay.c:
29023 * ext/pango/gsttextrender.c:
29024 * ext/theora/theoradec.c:
29025 * ext/theora/theoraenc.c:
29026 * ext/vorbis/vorbisdec.c:
29027 * ext/vorbis/vorbisenc.c:
29028 * gst-libs/gst/audio/gstaudiofilter.c:
29029 (gst_audio_filter_base_init):
29030 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
29031 (gst_audio_filter_template_base_init):
29032 * gst/adder/gstadder.c: (gst_adder_get_type):
29033 * gst/adder/gstadder.h:
29034 * gst/audioconvert/gstaudioconvert.c:
29035 * gst/audiotestsrc/gstaudiotestsrc.c:
29036 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
29037 (gst_audio_test_src_create):
29038 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
29039 * gst/playback/gstdecodebin.c:
29040 * gst/playback/gstplaybin.c:
29041 * gst/playback/gststreamselector.c:
29042 (gst_stream_selector_base_init):
29043 * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
29044 * gst/volume/gstvolume.c:
29045 * sys/v4l/gstv4lmjpegsink.c:
29046 * sys/v4l/gstv4lmjpegsrc.c:
29047 * tests/check/libs/cddabasesrc.c:
29048 * tests/old/examples/gob/gst-identity2.gob:
29049 Add docs for adder, use GST_ELEMENT_DETAILS macro,
29050 define GstElementDetails at the top
29052 2006-03-23 21:48:18 +0000 Sébastien Moutte <sebastien@moutte.net>
29054 win32/common/libgstinterfaces.def: Add a lot of export functions for gst-python
29055 Original commit message from CVS:
29056 * win32/common/libgstinterfaces.def:
29057 Add a lot of export functions for gst-python
29058 * win32/common/libgstinterfaces.dsp:
29059 Add a missing include folder in the project configuration
29061 2006-03-23 16:58:03 +0000 Wim Taymans <wim.taymans@gmail.com>
29063 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
29064 Original commit message from CVS:
29065 * gst-libs/gst/audio/gstbaseaudiosrc.c:
29066 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
29067 (gst_base_audio_src_change_state):
29068 Fix audio sources, forgot to make the ringbuffer
29071 2006-03-23 16:29:58 +0000 Wim Taymans <wim.taymans@gmail.com>
29073 gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
29074 Original commit message from CVS:
29075 * gst-libs/gst/audio/gstbaseaudiosrc.c:
29076 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
29077 (gst_base_audio_src_change_state):
29078 unparent instead of unref the ringbuffer.
29080 2006-03-23 16:24:23 +0000 Wim Taymans <wim.taymans@gmail.com>
29082 gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
29083 Original commit message from CVS:
29084 * gst-libs/gst/audio/gstbaseaudiosink.c:
29085 (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
29086 (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
29087 Implement new async_play vmethod to start slaving and allow
29088 playback start in case of async PLAY state changes.
29089 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
29090 Enable QoS with new method in base class.
29092 2006-03-23 11:18:19 +0000 Julien MOUTTE <julien@moutte.net>
29094 gst/videotestsrc/gstvideotestsrc.c: Partially handle 0 framerate, only EOS after the first frame is missing.
29095 Original commit message from CVS:
29096 Patch by: Julien MOUTTE <julien at moutte dot net>
29097 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
29098 (gst_video_test_src_do_seek), (gst_video_test_src_create):
29099 Partially handle 0 framerate, only EOS after the first frame
29102 2006-03-23 09:38:59 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
29104 gst/: Patch for support of YVU9 AVI files (#334822)
29105 Original commit message from CVS:
29106 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
29107 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
29108 (gst_riff_create_video_template_caps):
29109 * gst/ffmpegcolorspace/avcodec.h:
29110 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
29111 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
29112 (gst_ffmpegcsp_avpicture_fill):
29113 * gst/ffmpegcolorspace/imgconvert.c:
29114 Patch for support of YVU9 AVI files (#334822)
29116 2006-03-22 15:29:25 +0000 Edward Hervey <bilboed@bilboed.com>
29118 docs/design/design-decodebin.txt: Added design document for new decodebin text/x-pango-markup is also a default targe...
29119 Original commit message from CVS:
29120 * docs/design/design-decodebin.txt:
29121 Added design document for new decodebin
29122 (Target Caps): text/x-pango-markup is also a default target caps.
29124 2006-03-22 15:11:47 +0000 Edward Hervey <bilboed@bilboed.com>
29126 docs/design/design-decodebin.txt: Added design document for new decodebin
29127 Original commit message from CVS:
29128 * docs/design/design-decodebin.txt:
29129 Added design document for new decodebin
29131 2006-03-22 12:33:09 +0000 Wim Taymans <wim.taymans@gmail.com>
29133 gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
29134 Original commit message from CVS:
29135 * gst-libs/gst/audio/gstbaseaudiosink.c:
29136 (gst_base_audio_sink_dispose):
29137 Since we _parent the ringbuffer, we also need to
29138 _unparent instead of a plain _unref.
29140 2006-03-22 12:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
29142 tests/examples/seek/seek.c: Add scrub checkbox.
29143 Original commit message from CVS:
29144 * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
29145 (stop_seek), (scrub_toggle_cb), (main):
29146 Add scrub checkbox.
29148 2006-03-21 17:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
29150 ext/ogg/gstoggparse.c: Fix very inefficient usage of linked lists (#335365).
29151 Original commit message from CVS:
29152 * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
29153 (gst_ogg_parse_chain):
29154 Fix very inefficient usage of linked lists (#335365).
29156 2006-03-21 14:26:01 +0000 Edward Hervey <bilboed@bilboed.com>
29158 gcc 4.1 unreferenced pointer fixes.
29159 Original commit message from CVS:
29160 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
29161 * gst/playback/gstplaybin.c: (handoff):
29162 * gst/playback/gststreamselector.c:
29163 (gst_stream_selector_set_property):
29164 gcc 4.1 unreferenced pointer fixes.
29165 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
29166 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
29167 gst_buffer_ref() now takes a GstBuffer*.
29169 2006-03-20 18:09:41 +0000 Julien Moutte <julien@moutte.net>
29171 sys/xvimage/xvimagesink.c: Fix a memleak reported by Jan Schmidt.
29172 Original commit message from CVS:
29173 2006-03-20 Julien MOUTTE <julien@moutte.net>
29174 * sys/xvimage/xvimagesink.c:
29175 (gst_xvimagesink_get_format_from_caps): Fix a memleak reported
29178 2006-03-19 11:37:46 +0000 Tim-Philipp Müller <tim@centricular.net>
29180 gst/typefind/gsttypefindfunctions.c: Can't do tag preferences via probability, as tags would then lose against types ...
29181 Original commit message from CVS:
29182 * gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
29183 (id3v1_type_find), (apetag_type_find), (plugin_init):
29184 Can't do tag preferences via probability, as tags would then
29185 lose against types that are recognised with MAXIMUM probability
29186 (like .wav); so let all tag typefinders return MAXIMUM themselves
29187 and order them via the rank. Split ID3v1 and ID3v2 typefinders so
29188 that we can prefer APE to ID3v1 (fixes #335028).
29190 2006-03-17 17:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
29192 gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
29193 Original commit message from CVS:
29194 * gst-libs/gst/audio/gstbaseaudiosink.c:
29195 (gst_base_audio_sink_change_state):
29196 * gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
29197 (gst_ring_buffer_may_start):
29198 * gst-libs/gst/audio/gstringbuffer.h:
29199 Only start playback if we are playing.
29200 should fix #330748.
29202 2006-03-17 13:11:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29204 Revert accidental commits to these files.
29205 Original commit message from CVS:
29206 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
29207 * win32/common/config.h:
29208 Revert accidental commits to these files.
29210 2006-03-16 20:01:03 +0000 Michal Benes <michal.benes@xeris.cz>
29212 tests/Makefile.am: Don't try to build tests in tests/icles if we don't have X (#323852)
29213 Original commit message from CVS:
29214 Patch by: Michal Benes <michal dot benes at xeris dot cz>
29215 * tests/Makefile.am:
29216 Don't try to build tests in tests/icles if we
29217 don't have X (#323852)
29219 2006-03-16 13:08:01 +0000 Tim-Philipp Müller <tim@centricular.net>
29221 gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
29222 Original commit message from CVS:
29223 * gst-libs/gst/tag/gstid3tag.c:
29224 Add TXXX frame identifiers for replaygain stuff as used
29225 by some taggers (see #323721).
29227 2006-03-16 10:22:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29229 gst/playback/gststreamselector.c: Preserve the existing buggy streamselector behaviour by performing a fallback buffe...
29230 Original commit message from CVS:
29231 * gst/playback/gststreamselector.c:
29232 (gst_stream_selector_set_property),
29233 (gst_stream_selector_bufferalloc):
29234 Preserve the existing buggy streamselector behaviour by performing
29235 a fallback buffer allocation when downstream isn't linked yet.
29236 This should really be fixed in playbin by blocking pads until it's
29238 Also, use gst_pad_alloc_buffer instead of
29239 gst_pad_alloc_buffer_and_set.
29241 2006-03-15 22:40:08 +0000 Tim-Philipp Müller <tim@centricular.net>
29243 gst-libs/gst/tag/gstid3tag.c: Don't crash on unknown ID3v2 TXXX frames.
29244 Original commit message from CVS:
29245 * gst-libs/gst/tag/gstid3tag.c:
29246 Don't crash on unknown ID3v2 TXXX frames.
29248 2006-03-15 17:59:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29250 ext/alsa/gstalsasink.c: Chain up to the parent finalize method.
29251 Original commit message from CVS:
29252 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
29253 Chain up to the parent finalize method.
29254 Add 32-bit sample size to the template caps.
29255 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
29256 (gst_riff_create_video_template_caps):
29257 Add the fourcc that the VMWare codec uses.
29258 * gst/playback/gststreamselector.c:
29259 (gst_stream_selector_set_property),
29260 (gst_stream_selector_bufferalloc),
29261 (gst_stream_selector_request_new_pad):
29262 For the active pad, forward buffer-alloc requests, otherwise
29263 return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
29264 having to memcpy every frame when used by playbin.
29265 * gst/tcp/gstmultifdsink.c:
29266 (gst_multi_fd_sink_handle_client_write):
29267 Get negotiated caps from the sink pad, rather than the sink
29270 2006-03-15 17:11:34 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
29272 ext/gnomevfs/gstgnomevfssrc.c: Don't forget to set src->callbacks_pushed to FALSE again when popping them, otherwise ...
29273 Original commit message from CVS:
29274 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
29275 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_pop_callbacks):
29276 Don't forget to set src->callbacks_pushed to FALSE again when
29277 popping them, otherwise re-activation in a different mode won't
29280 2006-03-15 11:30:29 +0000 Sebastien Moutte <sebastien@moutte.net>
29282 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
29283 Original commit message from CVS:
29284 Patch by: Sebastien Moutte <sebastien moutte net>
29285 * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
29286 (gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
29287 (gst_ffmpeg_smpfmt_to_caps):
29288 Replace __VA_ARGS__ caps creation macros with varargs functions.
29289 Makes things compile on MSVC (#320765), looks nicer, and we can
29290 tell the compiler to check for the NULL terminator.
29292 2006-03-14 15:13:04 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
29294 gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
29295 Original commit message from CVS:
29296 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
29297 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29298 Make sure the buffer we copy into is really always big
29299 enough, this time for real (#333488).
29301 2006-03-14 13:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
29303 gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
29304 Original commit message from CVS:
29305 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29306 Add support for 24bpp DIB (#305279).
29308 2006-03-14 11:11:59 +0000 Wim Taymans <wim.taymans@gmail.com>
29310 gst/: Re-enable QoS after the release.
29311 Original commit message from CVS:
29312 * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
29313 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
29314 * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
29315 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
29316 (gst_video_scale_init), (gst_video_scale_src_event):
29317 Re-enable QoS after the release.
29318 Rework videoscale to use the base class src_event handler.
29320 2006-03-14 09:51:01 +0000 Tim-Philipp Müller <tim@centricular.net>
29322 configure.ac: back to CVS.
29323 Original commit message from CVS:
29327 === release 0.10.5 ===
29329 2006-03-13 19:50:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29335 * docs/plugins/inspect/plugin-adder.xml:
29336 * docs/plugins/inspect/plugin-alsa.xml:
29337 * docs/plugins/inspect/plugin-audioconvert.xml:
29338 * docs/plugins/inspect/plugin-audiorate.xml:
29339 * docs/plugins/inspect/plugin-audioresample.xml:
29340 * docs/plugins/inspect/plugin-audiotestsrc.xml:
29341 * docs/plugins/inspect/plugin-cdparanoia.xml:
29342 * docs/plugins/inspect/plugin-decodebin.xml:
29343 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
29344 * docs/plugins/inspect/plugin-gnomevfs.xml:
29345 * docs/plugins/inspect/plugin-libvisual.xml:
29346 * docs/plugins/inspect/plugin-ogg.xml:
29347 * docs/plugins/inspect/plugin-pango.xml:
29348 * docs/plugins/inspect/plugin-playbin.xml:
29349 * docs/plugins/inspect/plugin-subparse.xml:
29350 * docs/plugins/inspect/plugin-tcp.xml:
29351 * docs/plugins/inspect/plugin-theora.xml:
29352 * docs/plugins/inspect/plugin-typefindfunctions.xml:
29353 * docs/plugins/inspect/plugin-video4linux.xml:
29354 * docs/plugins/inspect/plugin-videorate.xml:
29355 * docs/plugins/inspect/plugin-videoscale.xml:
29356 * docs/plugins/inspect/plugin-videotestsrc.xml:
29357 * docs/plugins/inspect/plugin-volume.xml:
29358 * docs/plugins/inspect/plugin-vorbis.xml:
29359 * docs/plugins/inspect/plugin-ximagesink.xml:
29360 * docs/plugins/inspect/plugin-xvimagesink.xml:
29361 * win32/common/config.h:
29363 Original commit message from CVS:
29366 2006-03-13 17:28:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29383 Original commit message from CVS:
29386 2006-03-13 11:17:19 +0000 Tim-Philipp Müller <tim@centricular.net>
29388 docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
29389 Original commit message from CVS:
29390 * docs/plugins/Makefile.am:
29391 Part of previous cdparanoiasrc docs fixes, forgot to commit.
29393 2006-03-12 14:56:31 +0000 Tim-Philipp Müller <tim@centricular.net>
29395 docs/plugins/: Add cdparanoiasrc to docs.
29396 Original commit message from CVS:
29397 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29398 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29399 * docs/plugins/gst-plugins-base-plugins.hierarchy:
29400 Add cdparanoiasrc to docs.
29401 * gst-libs/gst/cdda/gstcddabasesrc.c:
29402 More GstCddaBaseSrc docs.
29404 2006-03-12 13:47:22 +0000 Tim-Philipp Müller <tim@centricular.net>
29406 Add new API to libgsttag: gst_tag_from_id3_user_tag().
29407 Original commit message from CVS:
29408 * docs/libs/gst-plugins-base-libs-sections.txt:
29409 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
29410 * gst-libs/gst/tag/tag.h:
29411 Add new API to libgsttag: gst_tag_from_id3_user_tag().
29413 2006-03-11 19:47:16 +0000 Tim-Philipp Müller <tim@centricular.net>
29415 gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
29416 Original commit message from CVS:
29417 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29418 NULL-terminate array of mpeg4 video file extensions.
29419 Fixes crash on PPC (#334226).
29421 2006-03-11 16:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
29423 ext/gnomevfs/gstgnomevfssrc.c: gnome_vfs_uri_is_local() alone is not a good indicator whether we can operate in pull-...
29424 Original commit message from CVS:
29425 * ext/gnomevfs/gstgnomevfssrc.c:
29426 (gst_gnome_vfs_src_check_get_range):
29427 gnome_vfs_uri_is_local() alone is not a good indicator
29428 whether we can operate in pull-mode with a specific URI,
29429 as it returns FALSE for file:// URIs that point to an
29430 NFS-mounted path. Be more conservative here: whitelist
29431 local files, blacklist http URIs and use the old
29432 mechanism for anything else (fixes #334216).
29434 2006-03-10 19:15:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29436 configure.ac: back to trunk
29437 Original commit message from CVS:
29441 === release 0.10.4 ===
29443 2006-03-10 19:05:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29449 * docs/plugins/gst-plugins-base-plugins.args:
29450 * docs/plugins/inspect/plugin-adder.xml:
29451 * docs/plugins/inspect/plugin-alsa.xml:
29452 * docs/plugins/inspect/plugin-audioconvert.xml:
29453 * docs/plugins/inspect/plugin-audiorate.xml:
29454 * docs/plugins/inspect/plugin-audioresample.xml:
29455 * docs/plugins/inspect/plugin-audiotestsrc.xml:
29456 * docs/plugins/inspect/plugin-cdparanoia.xml:
29457 * docs/plugins/inspect/plugin-decodebin.xml:
29458 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
29459 * docs/plugins/inspect/plugin-gnomevfs.xml:
29460 * docs/plugins/inspect/plugin-libvisual.xml:
29461 * docs/plugins/inspect/plugin-ogg.xml:
29462 * docs/plugins/inspect/plugin-pango.xml:
29463 * docs/plugins/inspect/plugin-playbin.xml:
29464 * docs/plugins/inspect/plugin-subparse.xml:
29465 * docs/plugins/inspect/plugin-tcp.xml:
29466 * docs/plugins/inspect/plugin-theora.xml:
29467 * docs/plugins/inspect/plugin-typefindfunctions.xml:
29468 * docs/plugins/inspect/plugin-video4linux.xml:
29469 * docs/plugins/inspect/plugin-videorate.xml:
29470 * docs/plugins/inspect/plugin-videoscale.xml:
29471 * docs/plugins/inspect/plugin-videotestsrc.xml:
29472 * docs/plugins/inspect/plugin-volume.xml:
29473 * docs/plugins/inspect/plugin-vorbis.xml:
29474 * docs/plugins/inspect/plugin-ximagesink.xml:
29475 * docs/plugins/inspect/plugin-xvimagesink.xml:
29477 * win32/common/config.h:
29479 Original commit message from CVS:
29482 2006-03-10 12:37:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29484 gst-libs/gst/video/gstvideosink.c: Disable max-lateness by setting it to -1 for now, so that we can bed QoS stuff in ...
29485 Original commit message from CVS:
29486 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
29487 Disable max-lateness by setting it to -1 for now, so that
29488 we can bed QoS stuff in thoroughly between now and the next
29491 2006-03-10 11:09:23 +0000 Fabrizio <fabrizio.ge@tiscali.it>
29493 gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
29494 Original commit message from CVS:
29495 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29496 Make sure we don't read beyond the palette buffer in case of
29497 broken or manipulated files (#333488, patch by: Fabrizio
29500 2006-03-10 10:44:02 +0000 Edward Hervey <bilboed@bilboed.com>
29502 gst/typefind/gsttypefindfunctions.c: Fix for variable not initialized.
29503 Original commit message from CVS:
29504 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
29505 Fix for variable not initialized.
29507 2006-03-09 19:02:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29510 * docs/libs/tmpl/gstringbuffer.sgml:
29525 * win32/common/config.h:
29527 Original commit message from CVS:
29530 2006-03-09 17:58:00 +0000 Wim Taymans <wim.taymans@gmail.com>
29532 ext/libvisual/visual.c: Small cleanups.
29533 Original commit message from CVS:
29534 * ext/libvisual/visual.c: (gst_visual_get_type),
29535 (gst_visual_src_setcaps), (gst_vis_src_negotiate),
29536 (gst_visual_chain):
29538 * ext/theora/gsttheoradec.h:
29539 * ext/theora/theoradec.c: (gst_theora_dec_init),
29540 (gst_theora_dec_reset), (_theora_granule_time),
29541 (theora_dec_src_convert), (theora_dec_sink_convert),
29542 (theora_dec_src_query), (theora_dec_src_event),
29543 (theora_dec_sink_event), (theora_handle_comment_packet),
29544 (theora_handle_header_packet), (theora_dec_push),
29545 (theora_handle_data_packet), (theora_dec_chain),
29546 (theora_dec_change_state):
29549 2006-03-09 17:50:59 +0000 Wim Taymans <wim.taymans@gmail.com>
29551 ext/gnomevfs/gstgnomevfssrc.c: Some cleanups.
29552 Original commit message from CVS:
29553 * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
29554 (audiocast_register_listener), (gst_gnome_vfs_src_start):
29557 2006-03-09 17:45:39 +0000 Wim Taymans <wim.taymans@gmail.com>
29559 ext/ogg/gstoggdemux.c: Don't try to activate NULL chains.
29560 Original commit message from CVS:
29561 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
29562 Don't try to activate NULL chains.
29564 2006-03-09 16:30:41 +0000 Tim-Philipp Müller <tim@centricular.net>
29566 gst/typefind/gsttypefindfunctions.c: Fix invalid memory access to region before peek'd data (#332964).
29567 Original commit message from CVS:
29568 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
29569 Fix invalid memory access to region before peek'd data (#332964).
29571 2006-03-09 15:05:03 +0000 Christophe Fergeau <teuf@gnome.org>
29574 Original commit message from CVS:
29575 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init):
29576 * ext/pango/gsttextrender.c: (gst_text_render_init):
29577 * gst/adder/gstadder.c: (gst_adder_init):
29578 Don't leak padtemplates, patch by Christophe Fergeau,
29581 2006-03-09 12:56:35 +0000 Tim-Philipp Müller <tim@centricular.net>
29583 gst/subparse/gstsubparse.c: Fix invalid memory access: make sure string passed to regexec() is NUL-termianted.
29584 Original commit message from CVS:
29585 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
29586 Fix invalid memory access: make sure string passed to
29587 regexec() is NUL-termianted.
29589 2006-03-09 12:37:59 +0000 Tim-Philipp Müller <tim@centricular.net>
29591 gst/typefind/gsttypefindfunctions.c: Refactor mpeg/audio typefinding to make it more maintainable and easier to fine-...
29592 Original commit message from CVS:
29593 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
29595 Refactor mpeg/audio typefinding to make it more maintainable
29596 and easier to fine-tune. Make probing into middle of the file
29597 work properly (fixes #333900, also see #152688).
29599 2006-03-09 11:10:03 +0000 Tim-Philipp Müller <tim@centricular.net>
29601 gst/typefind/gsttypefindfunctions.c: Remove part from previous commit that was bogus: g_utf8_validate() does in fact ...
29602 Original commit message from CVS:
29603 * gst/typefind/gsttypefindfunctions.c:
29604 (utf8_type_find_have_valid_utf8_at_offset):
29605 Remove part from previous commit that was bogus:
29606 g_utf8_validate() does in fact not accept embedded
29607 zeroes, so we don't need to check for those (thanks
29608 to Mike for the hint).
29610 2006-03-08 17:11:29 +0000 Tim-Philipp Müller <tim@centricular.net>
29612 gst/typefind/gsttypefindfunctions.c: Make plain/text typefinder more conservative: firstly, check for embedded zeroes...
29613 Original commit message from CVS:
29614 * gst/typefind/gsttypefindfunctions.c:
29615 (utf8_type_find_count_embedded_zeroes),
29616 (utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
29617 Make plain/text typefinder more conservative: firstly, check
29618 for embedded zeroes, which are perfectly valid UTF-8 characters,
29619 but also a fairly good sign that something is not a plain text
29620 file; secondly, probe into the middle of the file if possible.
29621 If we can't probe into the middle, limit the probability value
29622 to be returned to TYPE_FIND_POSSIBLE (see #333900).
29624 2006-03-08 11:34:45 +0000 Michael Smith <msmith@xiph.org>
29626 gst/typefind/gsttypefindfunctions.c: Make typefind function name for mpeg4 video unique.
29627 Original commit message from CVS:
29628 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29629 Make typefind function name for mpeg4 video unique.
29631 2006-03-08 09:53:31 +0000 Wim Taymans <wim.taymans@gmail.com>
29633 ext/libvisual/visual.c: Cleanups, post nice errors.
29634 Original commit message from CVS:
29635 * ext/libvisual/visual.c: (gst_visual_init),
29636 (gst_visual_clear_actors), (gst_visual_dispose),
29637 (gst_visual_reset), (gst_visual_src_setcaps),
29638 (gst_visual_sink_setcaps), (gst_vis_src_negotiate),
29639 (gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
29640 (gst_visual_chain), (gst_visual_change_state):
29641 Cleanups, post nice errors.
29642 Handle sink and src events.
29643 Implement simple QoS.
29644 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
29645 Use new basesink methods to configure max-lateness.
29647 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
29648 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
29649 Debug statement cleanups.
29650 * gst/volume/gstvolume.c: (gst_volume_class_init):
29653 2006-03-08 09:50:23 +0000 Tim-Philipp Müller <tim@centricular.net>
29655 ext/pango/gsttextoverlay.c: Revert API/ABI break from March 1. Keep 'halign' and 'valign' as string type properties, ...
29656 Original commit message from CVS:
29657 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
29658 (gst_text_overlay_init), (gst_text_overlay_set_property),
29659 (gst_text_overlay_get_property):
29660 Revert API/ABI break from March 1. Keep 'halign' and 'valign'
29661 as string type properties, but mark them deprecated. Add
29662 'halignment' and 'valignment' properties that use enums
29663 instead of strings.
29665 2006-03-08 09:37:12 +0000 Fabrizio <fabrizio.ge@tiscali.it>
29667 gst-libs/gst/riff/riff-media.c: Allow palettes with less than 256 colours in AVI files
29668 Original commit message from CVS:
29669 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29670 Allow palettes with less than 256 colours in AVI files
29671 (#333488, patch by: Fabrizio Gennari).
29673 2006-03-07 21:56:09 +0000 Julien Moutte <julien@moutte.net>
29675 ext/pango/gsttextoverlay.c: Fix wrong EOS handling on text pad. We were releasing the queued text buffer when we shou...
29676 Original commit message from CVS:
29677 2006-03-07 Julien MOUTTE <julien@moutte.net>
29678 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
29679 (gst_text_overlay_video_event): Fix wrong EOS handling on text
29680 pad. We were releasing the queued text buffer when we should keep
29681 it until video pad gets EOS or discard the text buffer because it's
29682 too old. That was eating the last subtitle buffer. Add some more
29685 2006-03-07 17:28:36 +0000 Tim-Philipp Müller <tim@centricular.net>
29687 ext/pango/gsttextoverlay.c: Fix invalid memory access (we can't access a buffer after it's been pushed downstream wit...
29688 Original commit message from CVS:
29689 * ext/pango/gsttextoverlay.c: (gst_text_overlay_render_text),
29690 (gst_text_overlay_video_chain):
29691 Fix invalid memory access (we can't access a buffer after it's been
29692 pushed downstream without taking a reference); fix memory leak (if
29693 there's no text to render, bail out before allocating stuff).
29695 2006-03-07 15:08:15 +0000 Tim-Philipp Müller <tim@centricular.net>
29697 ext/pango/gsttextoverlay.*: If input is plain text, escape it before passing it to pango_layout_set_markup().
29698 Original commit message from CVS:
29699 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
29700 (gst_text_overlay_setcaps_txt), (gst_text_overlay_video_chain):
29701 * ext/pango/gsttextoverlay.h:
29702 If input is plain text, escape it before passing it to
29703 pango_layout_set_markup().
29705 2006-03-07 13:01:21 +0000 Tim-Philipp Müller <tim@centricular.net>
29707 gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
29708 Original commit message from CVS:
29709 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
29710 Don't ignore flow return from gst_pad_push().
29712 2006-03-07 12:49:03 +0000 Christophe Fergeau <teuf@gnome.org>
29714 Don't leak references returned by gst_pad_get_parent()
29715 Original commit message from CVS:
29716 * ext/libvisual/visual.c: (gst_visual_getcaps),
29717 (gst_visual_src_setcaps), (gst_visual_sink_setcaps):
29718 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
29719 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
29720 (gst_vorbisenc_convert_sink):
29721 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
29722 (gst_audio_duration_from_pad_buffer):
29723 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
29724 (gst_audio_filter_chain):
29725 * gst-libs/gst/rtp/gstbasertpdepayload.c:
29726 (gst_base_rtp_depayload_setcaps):
29727 * gst-libs/gst/video/video.c: (gst_video_frame_rate),
29728 (gst_video_get_size):
29729 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
29730 Don't leak references returned by gst_pad_get_parent()
29731 (#333663, based on patch by: Christophe Fergeau).
29733 2006-03-06 20:52:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29735 ext/gnomevfs/gstgnomevfssink.c: change location param details
29736 Original commit message from CVS:
29737 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
29738 change location param details
29739 * gst/volume/gstvolume.c: (plugin_init):
29740 correct plugin description
29742 2006-03-06 20:07:55 +0000 Tim-Philipp Müller <tim@centricular.net>
29744 ext/gnomevfs/gstgnomevfssrc.c: Override GstBaseSrc::check_get_range() in order to avoid opening the resource just to ...
29745 Original commit message from CVS:
29746 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
29747 (gst_gnome_vfs_src_check_get_range):
29748 Override GstBaseSrc::check_get_range() in order to avoid opening
29749 the resource just to check whether we can operate in pull-mode or
29750 not - we can predict that pretty well from the URI alone. Should
29751 fix problems with last.fm (#331690). (Requires latest core CVS).
29753 2006-03-06 16:18:51 +0000 Wim Taymans <wim.taymans@gmail.com>
29755 gst-libs/gst/video/gstvideosink.c: Throw away frames that are later than 20 ms.
29756 Original commit message from CVS:
29757 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
29758 (gst_video_sink_class_init):
29759 Throw away frames that are later than 20 ms.
29761 2006-03-06 14:14:47 +0000 Fabrizio <fabrizio.ge@tiscali.it>
29763 gst-libs/gst/riff/riff-media.c:
29764 Original commit message from CVS:
29765 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
29766 Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).
29768 2006-03-05 23:39:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29770 ext/ogg/gstoggmux.c: put Theora BOS pages before others. This hardcodes the Ogg/Theora I profile, but hey.
29771 Original commit message from CVS:
29772 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
29773 (gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
29774 put Theora BOS pages before others. This hardcodes
29775 the Ogg/Theora I profile, but hey.
29777 2006-03-05 23:06:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29779 * ext/ogg/gstoggmux.c:
29780 changed more than 5 lines
29781 Original commit message from CVS:
29782 changed more than 5 lines
29784 2006-03-05 22:57:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29786 ogg muxing of vorbis and theora now has pages ordered correctly again, even with delays.
29787 Original commit message from CVS:
29788 ogg muxing of vorbis and theora now has pages ordered correctly again,
29791 updated with some examples
29792 * ext/theora/theoraenc.c: (granulepos_to_timestamp),
29793 (granulepos_add), (theora_buffer_from_packet):
29794 * ext/vorbis/vorbisenc.c: (granulepos_to_timestamp_offset),
29795 (granulepos_to_timestamp), (gst_vorbisenc_buffer_from_packet),
29796 (gst_vorbisenc_chain):
29797 implement strategy from ext/ogg/README
29798 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
29799 (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
29800 (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_compare_pads),
29801 (gst_ogg_mux_queue_pads), (gst_ogg_mux_collected):
29802 Fix muxer so that oggz-validate is happy with all streams;
29803 except for no eos mark, and the BOS page ordering
29804 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
29805 (check_buffer_granulepos):
29806 * tests/check/pipelines/vorbisenc.c: (check_buffer_granulepos):
29807 update tests to check for OFFSET being set as requested
29808 fixed type of granulepos, it's not a ClockTime
29810 2006-03-05 21:34:23 +0000 Julien Moutte <julien@moutte.net>
29812 sys/xvimage/xvimagesink.c: Check that the xvimage we are creating has a correct size before returning it. (#3...
29813 Original commit message from CVS:
29814 2006-03-05 Julien MOUTTE <julien@moutte.net>
29815 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
29816 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
29817 Check that the xvimage we are creating has a correct size before returning it. (#314897)
29819 2006-03-05 13:44:05 +0000 Tim-Philipp Müller <tim@centricular.net>
29821 gst/typefind/gsttypefindfunctions.c: Give id3 and ape tag typefinders a rank slightly higher than PRIMARY to ensure t...
29822 Original commit message from CVS:
29823 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29824 Give id3 and ape tag typefinders a rank slightly higher
29825 than PRIMARY to ensure they're always run before any of
29826 the other typefinders (in particular wav and mp3) (#324186).
29828 2006-03-05 13:08:37 +0000 Tim-Philipp Müller <tim@centricular.net>
29830 gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
29831 Original commit message from CVS:
29832 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29833 Add support for '3IVD' fourcc (#333403).
29835 2006-03-04 14:35:10 +0000 Tim-Philipp Müller <tim@centricular.net>
29837 configure.ac: Bump requirements to GStreamer CVS for the new error enum.
29838 Original commit message from CVS:
29840 Bump requirements to GStreamer CVS for the new error enum.
29841 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
29842 Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
29843 space left on the device (fixes #333352).
29845 2006-03-03 23:53:50 +0000 Sébastien Moutte <sebastien@moutte.net>
29847 win32/vs6: add a project file for libgstvolume update the workspace
29848 Original commit message from CVS:
29850 add a project file for libgstvolume
29851 update the workspace
29853 2006-03-03 15:26:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29856 * ext/ogg/gstoggmux.c:
29858 Original commit message from CVS:
29861 2006-03-03 15:22:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29863 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
29864 Original commit message from CVS:
29865 2006-03-03 Thomas Vander Stichele <thomas at apestaart dot org>
29866 * ext/theora/theoraenc.c: (theora_set_header_on_caps):
29867 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
29869 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
29870 Set IN_CAPS on header buffers
29872 2006-03-02 18:23:55 +0000 Wim Taymans <wim.taymans@gmail.com>
29874 docs/plugins/: Add audioresample to docs.
29875 Original commit message from CVS:
29876 * docs/plugins/Makefile.am:
29877 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29878 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29879 Add audioresample to docs.
29880 * gst/audioconvert/gstaudioconvert.c:
29882 * gst/audioresample/gstaudioresample.c:
29883 (gst_audioresample_base_init), (gst_audioresample_class_init),
29884 (gst_audioresample_init), (gst_audioresample_dispose),
29885 (audioresample_get_unit_size), (audioresample_transform_caps),
29886 (resample_set_state_from_caps), (audioresample_transform_size),
29887 (audioresample_set_caps), (audioresample_event),
29888 (audioresample_do_output), (audioresample_transform),
29889 (audioresample_pushthrough), (gst_audioresample_set_property),
29890 (gst_audioresample_get_property), (plugin_init):
29891 * gst/audioresample/gstaudioresample.h:
29893 Small code cleanups.
29895 2006-03-02 18:12:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29897 * gst/videorate/Makefile.am:
29899 Original commit message from CVS:
29902 2006-03-02 17:48:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29904 * ext/ogg/gstoggmux.c:
29905 debug using the actual GstPad, that allows us to see the serialno in the padname
29906 Original commit message from CVS:
29907 debug using the actual GstPad, that allows us to see the serialno in the padname
29909 2006-03-02 17:46:36 +0000 Wim Taymans <wim.taymans@gmail.com>
29911 docs/plugins/: Added videoscale to docs.
29912 Original commit message from CVS:
29913 * docs/plugins/Makefile.am:
29914 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29915 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29916 Added videoscale to docs.
29917 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
29918 (gst_video_rate_swap_prev), (gst_video_rate_event),
29919 (gst_video_rate_chain):
29921 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
29922 (gst_video_scale_init), (gst_video_scale_prepare_size),
29923 (gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
29924 (gst_video_scale_fixate_caps), (gst_video_scale_transform):
29925 * gst/videoscale/gstvideoscale.h:
29926 Added docs, examples.
29927 Some code cleanups.
29928 Post errors instead of g_warning.
29930 2006-03-02 17:30:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29932 * ext/ogg/gstoggmux.c:
29933 clean up debug messages
29934 Original commit message from CVS:
29935 clean up debug messages
29937 2006-03-02 17:15:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29939 * ext/ogg/gstoggmux.c:
29940 extra debugging from older version, makes it easier to compare
29941 Original commit message from CVS:
29942 extra debugging from older version, makes it easier to compare
29944 2006-03-02 17:04:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29946 * ext/ogg/gstoggmux.c:
29947 some space cleanup and debug fixes
29948 Original commit message from CVS:
29949 some space cleanup and debug fixes
29951 2006-03-02 16:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
29953 docs/: Added some more docs to libs and plugins.
29954 Original commit message from CVS:
29955 * docs/libs/gst-plugins-base-libs-docs.sgml:
29956 * docs/libs/gst-plugins-base-libs-sections.txt:
29957 * docs/libs/gst-plugins-base-libs.types:
29958 * docs/plugins/Makefile.am:
29959 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
29960 * docs/plugins/gst-plugins-base-plugins-sections.txt:
29961 Added some more docs to libs and plugins.
29962 * gst-libs/gst/audio/gstringbuffer.c:
29963 (gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
29964 * gst-libs/gst/audio/gstringbuffer.h:
29965 Document ringbuffer some more.
29966 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
29967 (gst_video_rate_setcaps), (gst_video_rate_reset),
29968 (gst_video_rate_init), (gst_video_rate_flush_prev),
29969 (gst_video_rate_swap_prev), (gst_video_rate_event),
29970 (gst_video_rate_chain), (gst_video_rate_change_state):
29971 * gst/videorate/gstvideorate.h:
29972 Fix videorate to use segments.
29973 Make it work with 0/1 framerates (closes #331903)
29974 Handle EOS correctly.
29977 2006-03-02 13:13:00 +0000 Tim-Philipp Müller <tim@centricular.net>
29979 ext/ogg/gstogmparse.c: In state change function, first chain up to parent class, then handle downwards state change s...
29980 Original commit message from CVS:
29981 * ext/ogg/gstogmparse.c: (gst_ogm_parse_class_init),
29982 (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
29983 (gst_ogm_text_parse_init), (gst_ogm_parse_change_state):
29984 In state change function, first chain up to parent class,
29985 then handle downwards state change stuff. Remove some
29986 commented out cruft from 0.8 code.
29988 2006-03-02 12:35:59 +0000 Tim-Philipp Müller <tim@centricular.net>
29990 ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
29991 Original commit message from CVS:
29992 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
29993 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
29994 (gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
29995 (gst_ogm_parse_chain):
29996 Don't remove/re-add source pad if the new caps are the same as
29997 the old caps anyway (#333042). When removing source pad, don't
29998 unref it afterwards - we didn't ref it when adding. Sprinkle some
29999 GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
30000 after using gst_pad_get_parent(). Return downstream flow return
30001 value in chain function.
30003 2006-03-02 11:28:23 +0000 Wim Taymans <wim.taymans@gmail.com>
30005 docs/plugins/: Fix hierarchy, added some more elements to the docs.
30006 Original commit message from CVS:
30007 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
30008 * docs/plugins/gst-plugins-base-plugins.args:
30009 * docs/plugins/gst-plugins-base-plugins.hierarchy:
30010 * docs/plugins/gst-plugins-base-plugins.interfaces:
30011 * docs/plugins/gst-plugins-base-plugins.signals:
30012 Fix hierarchy, added some more elements to the docs.
30013 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30014 (gst_ffmpegcsp_get_type):
30015 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
30016 Fix docs for ffmpegcolorspace.
30018 2006-03-01 19:24:44 +0000 Tim-Philipp Müller <tim@centricular.net>
30020 gst/typefind/gsttypefindfunctions.c: Some typefinding fine-tuning:
30021 Original commit message from CVS:
30022 * gst/typefind/gsttypefindfunctions.c: (id3_type_find),
30023 (apetag_type_find), (ape_type_find), (plugin_init):
30024 Some typefinding fine-tuning:
30025 - rank ID3/APE tags in order of preference via probabilities, so that
30026 ID3v2 > APEv2 > APEv1 > ID3v1.
30027 - three or four bytes don't really justify MAXIMUM probability,
30028 change those to 'very likely' (musepack and monkeysaudio).
30030 2006-03-01 18:25:18 +0000 Wim Taymans <wim.taymans@gmail.com>
30033 Original commit message from CVS:
30034 * docs/plugins/Makefile.am:
30035 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
30036 * docs/plugins/gst-plugins-base-plugins-sections.txt:
30037 * ext/alsa/gstalsamixer.c:
30038 * ext/alsa/gstalsamixer.h:
30039 * ext/alsa/gstalsamixerelement.c:
30040 (gst_alsa_mixer_element_class_init), (gst_alsa_mixer_element_init):
30041 * ext/alsa/gstalsamixerelement.h:
30042 * ext/alsa/gstalsasink.c:
30043 * ext/alsa/gstalsasink.h:
30044 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
30045 (gst_alsasrc_init):
30046 * ext/alsa/gstalsasrc.h:
30048 Small code cleanups.
30050 2006-03-01 17:52:45 +0000 Wim Taymans <wim.taymans@gmail.com>
30052 ext/theora/Makefile.am: Dist new header too,
30053 Original commit message from CVS:
30054 * ext/theora/Makefile.am:
30055 Dist new header too,
30057 2006-03-01 17:39:28 +0000 Wim Taymans <wim.taymans@gmail.com>
30059 Fix some more docs.
30060 Original commit message from CVS:
30061 * docs/plugins/Makefile.am:
30062 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
30063 * docs/plugins/gst-plugins-base-plugins-sections.txt:
30064 * ext/gnomevfs/gstgnomevfssink.h:
30065 * ext/gnomevfs/gstgnomevfssrc.h:
30066 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
30067 * ext/vorbis/vorbisdec.h:
30068 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
30069 * ext/vorbis/vorbisenc.h:
30070 * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
30071 (vorbis_parse_chain), (vorbis_parse_change_state):
30072 * ext/vorbis/vorbisparse.h:
30073 * gst/audioconvert/gstaudioconvert.h:
30074 * gst/tcp/gsttcpserversink.h:
30075 * gst/videotestsrc/gstvideotestsrc.c:
30076 * gst/videotestsrc/gstvideotestsrc.h:
30077 * gst/volume/gstvolume.c:
30078 * gst/volume/gstvolume.h:
30079 Fix some more docs.
30080 Added docs for vorbisdec and vorbisparse.
30083 2006-03-01 16:24:37 +0000 Wim Taymans <wim.taymans@gmail.com>
30085 Updated/added documentation.
30086 Original commit message from CVS:
30087 * docs/plugins/Makefile.am:
30088 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
30089 * docs/plugins/gst-plugins-base-plugins-sections.txt:
30090 * ext/pango/gstclockoverlay.h:
30091 * ext/pango/gsttextoverlay.h:
30092 * ext/pango/gsttextrender.h:
30093 * ext/pango/gsttimeoverlay.h:
30094 * ext/theora/gsttheoradec.h:
30095 * ext/theora/gsttheoraenc.h:
30096 * ext/theora/theoradec.c:
30097 * ext/theora/theoraenc.c:
30098 * gst/audioconvert/gstaudioconvert.h:
30099 * gst/audiotestsrc/gstaudiotestsrc.h:
30100 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
30101 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
30102 * gst/tcp/gstmultifdsink.h:
30103 Updated/added documentation.
30104 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
30105 (gst_text_overlay_halign_get_type),
30106 (gst_text_overlay_wrap_mode_get_type),
30107 (gst_text_overlay_base_init), (gst_text_overlay_class_init),
30108 (gst_text_overlay_init), (gst_text_overlay_set_property),
30109 (gst_text_overlay_get_property):
30110 Fix up properties to be enums instead of string to make bindings,
30111 introspection and automatic GUI creation possible.
30112 Add getters for the properties.
30114 2006-02-28 21:21:07 +0000 Sébastien Moutte <sebastien@moutte.net>
30116 gst/audiotestsrc/gstaudiotestsrc.c: added defines of M_PI and M_PI_2
30117 Original commit message from CVS:
30118 * gst/audiotestsrc/gstaudiotestsrc.c:
30119 added defines of M_PI and M_PI_2
30120 * gst/ffmpegcolorspace/avcodec.h:
30121 removed #include "stdint.h" for win32 as _stdint.h is
30122 autogenerated to win32/common
30123 * win32/common/libgstaudio.def:
30124 * win32/common/libgsttag.def:
30127 some project files bugs corrected
30129 project files are reset to the default vs7 configuration
30130 (they link to msvcr71.dll using default optimizations)
30132 2006-02-28 19:08:12 +0000 Wim Taymans <wim.taymans@gmail.com>
30134 ext/gnomevfs/gstgnomevfssink.c: Fix some docs.
30135 Original commit message from CVS:
30136 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
30139 2006-02-28 13:52:04 +0000 Edward Hervey <bilboed@bilboed.com>
30141 ext/alsa/gstalsasrc.c: Set proper class on the ElementDetails:
30142 Original commit message from CVS:
30143 * ext/alsa/gstalsasrc.c:
30144 Set proper class on the ElementDetails:
30145 Source/Audio instead of Src/Audio
30147 2006-02-28 12:19:11 +0000 Edward Hervey <bilboed@bilboed.com>
30149 gst/videoscale/vs_scanline.c: Revert optimization in videoscale. It should go in liboil and have an appropriate liboi...
30150 Original commit message from CVS:
30151 * gst/videoscale/vs_scanline.c:
30152 (vs_scanline_resample_nearest_RGBA):
30153 Revert optimization in videoscale. It should go in liboil and have
30154 an appropriate liboil function.
30156 2006-02-28 11:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
30158 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
30159 Original commit message from CVS:
30160 * gst-libs/gst/audio/gstbaseaudiosink.c:
30161 (gst_base_audio_sink_provide_clock):
30162 Don't try to provide a clock in the NULL state.
30164 2006-02-28 11:04:47 +0000 Wim Taymans <wim.taymans@gmail.com>
30166 ext/ogg/gstoggdemux.c: Use GstSegment infrastructure to remove duplicated code and handle more seek cases correctly.
30167 Original commit message from CVS:
30168 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
30169 (gst_ogg_pad_event), (gst_ogg_pad_internal_chain),
30170 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
30171 (gst_ogg_demux_deactivate_current_chain),
30172 (gst_ogg_demux_activate_chain), (gst_ogg_demux_do_seek),
30173 (gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_info),
30174 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
30175 (gst_ogg_demux_loop), (gst_ogg_demux_change_state):
30176 Use GstSegment infrastructure to remove duplicated code
30177 and handle more seek cases correctly.
30179 2006-02-28 10:39:19 +0000 Wim Taymans <wim.taymans@gmail.com>
30181 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Don't ignore return code from ffmpeg convert function.
30182 Original commit message from CVS:
30183 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30184 (gst_ffmpegcsp_transform):
30185 Don't ignore return code from ffmpeg convert function.
30186 * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
30187 Split out some long statements to ease debugging.
30189 2006-02-27 12:08:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30191 ext/libvisual/visual.c: Don't use gst_pad_use_fixed_caps, because it prevents downstream from being able to renegotia...
30192 Original commit message from CVS:
30193 * ext/libvisual/visual.c: (gst_visual_init),
30194 (gst_vis_src_negotiate), (get_buffer), (plugin_init):
30195 Don't use gst_pad_use_fixed_caps, because it prevents downstream from
30196 being able to renegotiate the size. Instead, use the negotiation
30197 algorithm from the goom plugin to pick an initial output caps.
30198 Also, allow theoretical libvisual plugins that might support non-GL
30199 output even if they also do GL.
30201 2006-02-26 21:05:46 +0000 Julien Moutte <julien@moutte.net>
30203 ext/libvisual/visual.c: Load only non GL plugins. Fix some memleaks and possible negotiation issues.
30204 Original commit message from CVS:
30205 2006-02-26 Julien MOUTTE <julien@moutte.net>
30206 * ext/libvisual/visual.c: (gst_visual_init),
30207 (gst_visual_src_setcaps), (get_buffer), (gst_visual_chain),
30208 (plugin_init): Load only non GL plugins. Fix some memleaks and
30209 possible negotiation issues.
30211 2006-02-24 23:19:44 +0000 Julien Moutte <julien@moutte.net>
30213 gst-libs/gst/tag/tag.h: Adding Annodex tags here.
30214 Original commit message from CVS:
30215 2006-02-25 Julien MOUTTE <julien@moutte.net>
30216 * gst-libs/gst/tag/tag.h: Adding Annodex tags here.
30218 2006-02-24 18:55:27 +0000 Michael Smith <msmith@xiph.org>
30220 gst/typefind/gsttypefindfunctions.c: Fix CMML type find function to not require a specific minor version of the CMML ...
30221 Original commit message from CVS:
30222 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
30223 (cmml_type_find), (plugin_init):
30224 Fix CMML type find function to not require a specific minor version
30225 of the CMML header.
30226 Add an MPEG4 video elementary stream typefind function.
30228 2006-02-24 17:31:53 +0000 Michael Smith <msmith@xiph.org>
30230 ext/ogg/gstoggdemux.c: Annodex support in ogg demuxer. Doesn't do very much without the other annodex patches (to come).
30231 Original commit message from CVS:
30232 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
30233 (gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
30234 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
30235 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
30236 (gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
30237 (gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
30238 Annodex support in ogg demuxer. Doesn't do very much without the
30239 other annodex patches (to come).
30241 2006-02-24 16:21:34 +0000 Tim-Philipp Müller <tim@centricular.net>
30243 gst-libs/gst/riff/riff-media.c:
30244 Original commit message from CVS:
30245 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
30246 Pick up palette for MS video v1 (#327028, patch by:
30247 Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
30249 2006-02-24 13:54:04 +0000 Tim-Philipp Müller <tim@centricular.net>
30251 gst/ffmpegcolorspace/gstffmpegcolorspace.c: The 'palette_data' field from incoming RGB caps shouldn't be proxied on o...
30252 Original commit message from CVS:
30253 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30254 (gst_ffmpegcsp_caps_remove_format_info),
30255 (gst_ffmpegcsp_get_unit_size):
30256 The 'palette_data' field from incoming RGB caps shouldn't be
30257 proxied on outgoing YUV caps; also, restrict unit size
30258 adjustment in case of paletted data only to the unit that
30259 actually has a palette. Fixes #330711.
30261 2006-02-24 12:18:14 +0000 Tim-Philipp Müller <tim@centricular.net>
30263 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Plug some memory leaks.
30264 Original commit message from CVS:
30265 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30266 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
30267 (gst_ffmpegcsp_finalize), (gst_ffmpegcsp_class_init),
30268 (gst_ffmpegcsp_get_unit_size):
30269 Plug some memory leaks.
30271 2006-02-24 10:18:52 +0000 Tim-Philipp Müller <tim@centricular.net>
30273 sys/: Add some _CFLAGS and _LIBS that seem to be missing and/or required for Cygwin (see #317048).
30274 Original commit message from CVS:
30275 * sys/ximage/Makefile.am:
30276 * sys/xvimage/Makefile.am:
30277 Add some _CFLAGS and _LIBS that seem to be missing
30278 and/or required for Cygwin (see #317048).
30280 2006-02-24 00:07:18 +0000 Tim-Philipp Müller <tim@centricular.net>
30283 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
30284 Original commit message from CVS:
30285 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
30287 2006-02-22 18:46:46 +0000 Wim Taymans <wim.taymans@gmail.com>
30289 ext/alsa/gstalsasrc.c: Fix description as pointed out by caugier.
30290 Original commit message from CVS:
30291 * ext/alsa/gstalsasrc.c:
30292 Fix description as pointed out by caugier.
30294 2006-02-22 10:29:22 +0000 Edward Hervey <bilboed@bilboed.com>
30296 gst/typefind/gsttypefindfunctions.c: Better 3gp typefinding.
30297 Original commit message from CVS:
30298 Reviewed by : Edward Hervey <edward@fluendo.com>
30299 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
30301 Better 3gp typefinding.
30303 2006-02-21 12:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
30305 ext/gnomevfs/gstgnomevfssrc.c: Don't send EOS event here, the base class will send one for us.
30306 Original commit message from CVS:
30307 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
30308 Don't send EOS event here, the base class will send one for us.
30309 * gst/playback/gstplaybasebin.c: (prepare_output):
30310 Subpictures without video stream aren't allowed either.
30311 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
30312 Fix debug statement copy'n'paste-o.
30314 2006-02-21 12:05:18 +0000 Tim-Philipp Müller <tim@centricular.net>
30316 ext/alsa/gstalsamixer.c: Fix issues with mixer keeping state when muting/unmuting and when changing the volume whilst...
30317 Original commit message from CVS:
30318 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
30319 Fix issues with mixer keeping state when muting/unmuting
30320 and when changing the volume whilst muted (see #331763
30323 2006-02-20 18:27:06 +0000 Tim-Philipp Müller <tim@centricular.net>
30325 gst/subparse/gstsubparse.c: Set right caps given that we send escaped text. Also, honour <i></i>, <b></b> and <u></u>...
30326 Original commit message from CVS:
30327 * gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
30328 (parse_subrip), (gst_sub_parse_format_autodetect):
30329 Set right caps given that we send escaped text. Also,
30330 honour <i></i>, <b></b> and <u></u> markers that can be found
30331 in .srt files (fixes #310202).
30333 2006-02-20 16:21:14 +0000 Tim-Philipp Müller <tim@centricular.net>
30335 gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
30336 Original commit message from CVS:
30337 * gst-libs/gst/audio/mixerutils.c:
30338 (element_factory_rank_compare_func):
30339 Make order in which elements are tried more determinable.
30341 2006-02-20 15:57:51 +0000 Julien Moutte <julien@moutte.net>
30343 gst/playback/gstdecodebin.c: Make decodebin reusable by fixing remove_element_chain first and then introduce a cleane...
30344 Original commit message from CVS:
30345 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
30346 (remove_element_chain), (cleanup_decodebin),
30347 (gst_decode_bin_change_state): Make decodebin reusable by
30348 fixing remove_element_chain first and then introduce a
30349 cleaner in state change to ->NULL. (Closes #331678)
30350 ------------------------------------------------------
30352 2006-02-19 14:32:35 +0000 Wim Taymans <wim.taymans@gmail.com>
30354 ext/gnomevfs/gstgnomevfssink.c: use 0666 mask when creating files so umask gets applied correctly. Fixes #331295.
30355 Original commit message from CVS:
30356 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
30357 use 0666 mask when creating files so umask gets applied
30358 correctly. Fixes #331295.
30360 2006-02-19 14:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
30362 gst/subparse/: Add very basic parser for SSA subtitle streams (as often found in matroska files).
30363 Original commit message from CVS:
30364 * gst/subparse/Makefile.am:
30365 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
30366 (gst_ssa_parse_dispose), (gst_ssa_parse_init),
30367 (gst_ssa_parse_class_init), (gst_ssa_parse_src_event),
30368 (gst_ssa_parse_sink_event), (gst_ssa_parse_setcaps),
30369 (gst_ssa_parse_remove_override_codes), (gst_ssa_parse_parse_line),
30370 (gst_ssa_parse_chain), (gst_ssa_parse_change_state):
30371 * gst/subparse/gstssaparse.h:
30372 * gst/subparse/gstsubparse.c: (plugin_init):
30373 Add very basic parser for SSA subtitle streams (as often
30374 found in matroska files).
30376 2006-02-19 14:09:40 +0000 Tim-Philipp Müller <tim@centricular.net>
30378 gst/playback/gstdecodebin.c: That should be text/x-pango-markup, not text/x-pango-layout.
30379 Original commit message from CVS:
30380 * gst/playback/gstdecodebin.c: (mimetype_is_raw):
30381 That should be text/x-pango-markup, not text/x-pango-layout.
30383 2006-02-19 12:41:03 +0000 Julien Moutte <julien@moutte.net>
30385 ext/pango/gsttextoverlay.c: Polishing.
30386 Original commit message from CVS:
30387 2006-02-19 Julien MOUTTE <julien@moutte.net>
30388 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize):
30391 2006-02-19 12:05:23 +0000 Julien Moutte <julien@moutte.net>
30393 ext/pango/gsttextoverlay.c: Fix state change deadlock.
30394 Original commit message from CVS:
30395 2006-02-19 Julien MOUTTE <julien@moutte.net>
30396 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
30397 (gst_text_overlay_finalize), (gst_text_overlay_init),
30398 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
30399 (gst_text_overlay_render_text),
30400 (gst_text_overlay_text_pad_link),
30401 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
30402 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
30403 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
30404 Fix state change deadlock.
30406 2006-02-19 11:56:28 +0000 Julien Moutte <julien@moutte.net>
30408 ext/pango/gsttextoverlay.*: Fix seeking both for muxed formats and subtitles files.
30409 Original commit message from CVS:
30410 2006-02-19 Julien MOUTTE <julien@moutte.net>
30411 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
30412 (gst_text_overlay_finalize), (gst_text_overlay_init),
30413 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
30414 (gst_text_overlay_render_text),
30415 (gst_text_overlay_text_pad_link),
30416 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
30417 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
30418 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
30419 * ext/pango/gsttextoverlay.h: Fix seeking both for muxed formats
30420 and subtitles files.
30422 2006-02-19 00:40:38 +0000 Julien Moutte <julien@moutte.net>
30424 gst/playback/gstdecodebin.c: pango layout should be considered as row.
30425 Original commit message from CVS:
30426 2006-02-19 Julien MOUTTE <julien@moutte.net>
30427 * gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
30428 should be considered as row.
30430 2006-02-19 00:25:16 +0000 Julien Moutte <julien@moutte.net>
30432 gst/playback/gststreaminfo.*: Introduce language informations.
30433 Original commit message from CVS:
30434 2006-02-19 Julien MOUTTE <julien@moutte.net>
30435 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
30437 * gst/playback/gststreaminfo.h: Introduce language informations.
30439 2006-02-18 22:41:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30441 sys/: Set shared memory segments to be deleted as soon as we have attached, that way they get cleaned up automaticall...
30442 Original commit message from CVS:
30443 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
30444 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
30445 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
30446 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
30447 Set shared memory segments to be deleted as soon as we have attached,
30448 that way they get cleaned up automatically if we crash.
30450 2006-02-18 19:53:48 +0000 Julien Moutte <julien@moutte.net>
30452 ext/pango/: Those functions are called with lock held.
30453 Original commit message from CVS:
30454 2006-02-18 Julien MOUTTE <julien@moutte.net>
30455 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_get_text):
30456 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_get_text): Those
30457 functions are called with lock held.
30459 2006-02-18 19:51:47 +0000 Julien Moutte <julien@moutte.net>
30463 Original commit message from CVS:
30466 2006-02-18 19:10:35 +0000 Julien Moutte <julien@moutte.net>
30468 ext/pango/gsttextoverlay.c: Refactoring of textoverlay without collectpads. This now supports sparse subtitles coming...
30469 Original commit message from CVS:
30470 2006-02-18 Julien MOUTTE <julien@moutte.net>
30471 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
30472 (gst_text_overlay_finalize), (gst_text_overlay_init),
30473 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
30474 (gst_text_overlay_render_text),
30475 (gst_text_overlay_text_pad_link),
30476 (gst_text_overlay_text_pad_unlink),
30477 (gst_text_overlay_text_event),
30478 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
30479 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
30480 (gst_text_overlay_change_state): Refactoring of textoverlay
30481 without collectpads. This now supports sparse subtitles coming
30482 from a demuxer instead of a sub file. Seeking is still broken
30483 though. Need to discuss with wtay some more on how to handle
30485 * ext/pango/gsttextoverlay.h:
30486 * gst/playback/gstplaybin.c: (setup_sinks): Support linking with
30487 subtitles coming from the demuxer.
30489 2006-02-17 19:31:12 +0000 Wim Taymans <wim.taymans@gmail.com>
30491 ext/vorbis/vorbisenc.c: Use some more scaling functions.
30492 Original commit message from CVS:
30493 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
30494 (gst_vorbisenc_convert_sink):
30495 Use some more scaling functions.
30497 2006-02-17 16:12:11 +0000 Tim-Philipp Müller <tim@centricular.net>
30499 ext/cdparanoia/gstcdparanoiasrc.*: Add back 'transport-error' and 'uncorrected-error' signals and make them actually ...
30500 Original commit message from CVS:
30501 * ext/cdparanoia/gstcdparanoiasrc.c:
30502 (gst_cd_paranoia_src_class_init), (gst_cd_paranoia_dummy_callback),
30503 (gst_cd_paranoia_paranoia_callback),
30504 (gst_cd_paranoia_src_signal_is_being_watched),
30505 (gst_cd_paranoia_src_read_sector):
30506 * ext/cdparanoia/gstcdparanoiasrc.h:
30507 Add back 'transport-error' and 'uncorrected-error' signals and
30508 make them actually be fired when bad stuff happens (#319340).
30510 2006-02-17 14:07:01 +0000 Wim Taymans <wim.taymans@gmail.com>
30512 gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
30513 Original commit message from CVS:
30514 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
30515 (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
30516 (gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
30517 (gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
30518 (gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
30519 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
30520 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
30521 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
30522 (gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
30523 (gst_ring_buffer_clear):
30525 Added some G_LIKELY.
30527 2006-02-17 10:15:52 +0000 Wim Taymans <wim.taymans@gmail.com>
30529 gst-libs/gst/audio/TODO: Update TODO
30530 Original commit message from CVS:
30531 * gst-libs/gst/audio/TODO:
30533 * gst-libs/gst/audio/gstbaseaudiosink.c:
30534 (gst_base_audio_sink_get_offset):
30535 When trying to play samples ASAP and we don't have a
30536 previous sample, try to play at position 0 instead of
30537 an invalid position.
30539 2006-02-17 09:24:56 +0000 Wim Taymans <wim.taymans@gmail.com>
30541 ext/alsa/gstalsasink.c: Also release lock when we get an error in _reset(); fix an error message.
30542 Original commit message from CVS:
30543 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
30544 (gst_alsasink_reset):
30545 Also release lock when we get an error in _reset();
30546 fix an error message.
30548 2006-02-16 21:01:23 +0000 Tim-Philipp Müller <tim@centricular.net>
30550 ext/alsa/gstalsasink.*: Add support for more than 2 channels (#326720).
30551 Original commit message from CVS:
30552 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
30553 (gst_alsasink_init), (get_channel_free_structure),
30554 (caps_add_channel_configuration), (gst_alsasink_getcaps),
30555 (gst_alsasink_close):
30556 * ext/alsa/gstalsasink.h:
30557 Add support for more than 2 channels (#326720).
30559 2006-02-16 20:19:51 +0000 Tim-Philipp Müller <tim@centricular.net>
30561 gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
30562 Original commit message from CVS:
30563 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
30564 Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
30565 with 4 or 6 channels, assume a default channel layout to make things
30566 work (not sure there's anything else we can do in those cases).
30568 2006-02-16 19:18:46 +0000 Tim-Philipp Müller <tim@centricular.net>
30570 gst-libs/gst/audio/multichannel.c: Minor docs fix.
30571 Original commit message from CVS:
30572 * gst-libs/gst/audio/multichannel.c:
30574 * gst-libs/gst/riff/Makefile.am:
30575 * gst-libs/gst/riff/riff-ids.h:
30576 * gst-libs/gst/riff/riff-media.c:
30577 (gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
30578 Add support for WAVEFORMATEX, eg. PCM audio with more than two
30579 channels and a channel layout map.
30581 2006-02-16 17:06:46 +0000 Edward Hervey <bilboed@bilboed.com>
30583 gst/videoscale/vs_scanline.c: C-level optimization of the RGBA nearest neighbour function.
30584 Original commit message from CVS:
30585 Reviewed by Edward Hervey <edward@fluendo.com>
30586 * gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
30587 C-level optimization of the RGBA nearest neighbour function.
30588 Eventually this might end up in liboil with vectorized versions.
30590 2006-02-16 11:44:43 +0000 Tim-Philipp Müller <tim@centricular.net>
30592 gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
30593 Original commit message from CVS:
30594 * gst-libs/gst/audio/multichannel.c:
30595 (gst_audio_get_channel_positions):
30596 When we have more than 2 channels, but no channel layout is
30597 specified in the caps, return some default channel layout
30598 to the caller and warn about about a possibly buggy element
30599 (could be buggy filtercaps as well of course) (#317038).
30601 2006-02-16 09:29:38 +0000 Tim-Philipp Müller <tim@centricular.net>
30603 pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: Add gst-libs/gst/cdda to list of lib search paths.
30604 Original commit message from CVS:
30605 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
30606 Add gst-libs/gst/cdda to list of lib search paths.
30608 2006-02-15 12:20:47 +0000 Andy Wingo <wingo@pobox.com>
30610 ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating timestamp, update timestamp_end as well. Fixes a bugaboo. ...
30611 Original commit message from CVS:
30612 2006-02-15 Andy Wingo <wingo@pobox.com>
30613 * ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating
30614 timestamp, update timestamp_end as well. Fixes a bugaboo. I hope
30615 to the Lord Jesus that I do not have to touch the ogg muxer ever
30618 2006-02-15 12:07:57 +0000 Edward Hervey <bilboed@bilboed.com>
30620 gst/typefind/gsttypefindfunctions.c: quicktime movie files can also contain 'uuid' atoms.
30621 Original commit message from CVS:
30622 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
30623 quicktime movie files can also contain 'uuid' atoms.
30625 2006-02-14 18:52:52 +0000 Tim-Philipp Müller <tim@centricular.net>
30627 gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun...
30628 Original commit message from CVS:
30629 * gst/audioconvert/plugin.c: (plugin_init):
30630 Register the GstAudioChannelPosition enum type with the type
30631 system in the plugin_init function, so that it is known before
30632 any element actually makes use of multi-channel stuff. This is
30633 required for example if one wants to be able to deserialise/use
30634 a caps string with channel positions before any pipeline has
30635 been setup and started, like with gst-launch.
30637 2006-02-14 13:45:35 +0000 Wim Taymans <wim.taymans@gmail.com>
30639 gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
30640 Original commit message from CVS:
30641 * gst-libs/gst/audio/gstringbuffer.c:
30642 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
30643 (gst_ring_buffer_samples_done), (wait_segment),
30644 (gst_ring_buffer_commit), (gst_ring_buffer_clear):
30645 Add some compiler G_(UN_)LIKELY help.
30646 SIGNAL the ringbuffer waiters when going to PAUSED as well to
30647 make sure they can exit their functions. Should fix #330748
30649 2006-02-13 20:49:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30651 Windows does not have long long; copy the generated _stdint.h
30652 Original commit message from CVS:
30656 * win32/common/_stdint.h:
30657 Windows does not have long long; copy the generated _stdint.h
30658 * win32/common/interfaces-enumtypes.c:
30659 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
30660 (gst_mixer_track_flags_get_type),
30661 (gst_tuner_channel_flags_get_type):
30662 * win32/common/multichannel-enumtypes.c:
30663 (gst_audio_channel_position_get_type):
30666 2006-02-13 18:49:02 +0000 Wim Taymans <wim.taymans@gmail.com>
30668 gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
30669 Original commit message from CVS:
30670 * gst-libs/gst/audio/gstbaseaudiosink.c:
30671 (gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
30672 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
30673 Always sync on first sample we receive when starting.
30675 2006-02-13 15:59:48 +0000 Wim Taymans <wim.taymans@gmail.com>
30677 gst/playback/gstplaybin.c: Update vis bin docs.
30678 Original commit message from CVS:
30679 * gst/playback/gstplaybin.c: (gen_vis_element):
30680 Update vis bin docs.
30681 Move queue after tee so we don't queue video buffers but
30682 audio samples instead. Fixes problems where the video queue
30683 is filled and the audio queue empty.
30685 2006-02-13 15:17:34 +0000 Tim-Philipp Müller <tim@centricular.net>
30687 gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
30688 Original commit message from CVS:
30689 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
30690 No need to push an EOS event here, GstBaseSrc will do that for us
30691 when we return FLOW_UNEXPECTED.
30693 2006-02-12 14:54:55 +0000 Wim Taymans <wim.taymans@gmail.com>
30695 gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
30696 Original commit message from CVS:
30697 * gst-libs/gst/audio/gstbaseaudiosink.c:
30698 (gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
30699 (gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
30700 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
30701 Use scale functions when possible.
30702 Fix error messages.
30703 Free clockid when after waiting for EOS.
30704 Use G_(UN_)LIKLY when it makes sense.
30705 Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
30707 2006-02-12 14:26:55 +0000 Edward Hervey <bilboed@bilboed.com>
30709 gst/playback/gstplaybasebin.c: Remove stray semi-colon (fixes #330888).
30710 Original commit message from CVS:
30711 * gst/playback/gstplaybasebin.c: (prepare_output):
30712 Remove stray semi-colon (fixes #330888).
30714 2006-02-11 23:35:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30716 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...
30717 Original commit message from CVS:
30718 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
30719 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
30720 Fix up the XShm call testing so that we catch errors, and don't
30721 cause new ones by attempting to detach from a segment we failed
30722 to attach to. Fixes #312439.
30724 2006-02-10 11:29:55 +0000 Edward Hervey <bilboed@bilboed.com>
30726 gst/typefind/gsttypefindfunctions.c: Added flv file typefind (video/x-flv).
30727 Original commit message from CVS:
30728 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
30729 Added flv file typefind (video/x-flv).
30731 2006-02-10 10:53:33 +0000 Edward Hervey <bilboed@bilboed.com>
30733 gst-libs/gst/riff/riff-media.c: Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
30734 Original commit message from CVS:
30735 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
30736 (gst_riff_create_video_template_caps):
30737 Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
30738 Also added the caps to the default set of riff video caps.
30740 2006-02-09 19:05:23 +0000 Andy Wingo <wingo@pobox.com>
30742 ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start time and the end time of the last packet in the page.
30743 Original commit message from CVS:
30744 2006-02-09 Andy Wingo <wingo@pobox.com>
30745 * ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start
30746 time and the end time of the last packet in the page.
30747 (gst_ogg_mux_pad_queue_page): In addition to setting the timestamp
30748 on the pages in our queue, set the duration as well. Reflow a
30750 (gst_ogg_mux_collected): Keep track of GstOggPad->timestamp_end.
30751 Fixes bad muxing order.
30753 2006-02-09 17:04:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30755 gst-libs/gst/rtp/gstbasertppayload.c: update seqnum before setting it on the packet; this makes sure that the timesta...
30756 Original commit message from CVS:
30757 * gst-libs/gst/rtp/gstbasertppayload.c:
30758 (gst_basertppayload_setcaps), (gst_basertppayload_push):
30759 update seqnum before setting it on the packet; this makes sure
30760 that the timestamp and seqnum properties match after pushing
30763 2006-02-09 12:16:35 +0000 Andy Wingo <wingo@pobox.com>
30767 Original commit message from CVS:
30770 2006-02-09 11:46:03 +0000 Andy Wingo <wingo@pobox.com>
30772 * gst-libs/gst/audio/gstringbuffer.c:
30773 * win32/common/config.h:
30775 Original commit message from CVS:
30778 2006-02-09 11:36:18 +0000 Andy Wingo <wingo@pobox.com>
30780 gst-libs/gst/audio/gstringbuffer.c
30781 Original commit message from CVS:
30782 2006-02-09 Andy Wingo <wingo@pobox.com>
30783 * gst-libs/gst/audio/gstringbuffer.c
30784 (gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
30785 overflow after 13.5 hours of recording. Kapow!
30786 * ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
30787 the buffer size -- we don't care about underrun/overrun reporting
30788 right now, just need to return a useful value.
30790 2006-02-09 11:21:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30792 configure.ac: Back to CVS
30793 Original commit message from CVS:
30797 === release 0.10.3 ===
30799 2006-02-09 11:18:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30805 * docs/plugins/inspect/plugin-adder.xml:
30806 * docs/plugins/inspect/plugin-alsa.xml:
30807 * docs/plugins/inspect/plugin-audioconvert.xml:
30808 * docs/plugins/inspect/plugin-audiorate.xml:
30809 * docs/plugins/inspect/plugin-audioresample.xml:
30810 * docs/plugins/inspect/plugin-audiotestsrc.xml:
30811 * docs/plugins/inspect/plugin-cdparanoia.xml:
30812 * docs/plugins/inspect/plugin-decodebin.xml:
30813 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
30814 * docs/plugins/inspect/plugin-gnomevfs.xml:
30815 * docs/plugins/inspect/plugin-libvisual.xml:
30816 * docs/plugins/inspect/plugin-ogg.xml:
30817 * docs/plugins/inspect/plugin-pango.xml:
30818 * docs/plugins/inspect/plugin-playbin.xml:
30819 * docs/plugins/inspect/plugin-subparse.xml:
30820 * docs/plugins/inspect/plugin-tcp.xml:
30821 * docs/plugins/inspect/plugin-theora.xml:
30822 * docs/plugins/inspect/plugin-typefindfunctions.xml:
30823 * docs/plugins/inspect/plugin-video4linux.xml:
30824 * docs/plugins/inspect/plugin-videorate.xml:
30825 * docs/plugins/inspect/plugin-videoscale.xml:
30826 * docs/plugins/inspect/plugin-videotestsrc.xml:
30827 * docs/plugins/inspect/plugin-volume.xml:
30828 * docs/plugins/inspect/plugin-vorbis.xml:
30829 * docs/plugins/inspect/plugin-ximagesink.xml:
30830 * docs/plugins/inspect/plugin-xvimagesink.xml:
30831 * win32/common/config.h:
30833 Original commit message from CVS:
30836 2006-02-08 18:37:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30838 configure.ac: Drat. Bump libtool version number for new API.
30839 Original commit message from CVS:
30841 Drat. Bump libtool version number for new API.
30842 Prelease 0.10.2.3 (of 0.10.3)
30844 2006-02-08 15:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30846 0.10.2.2 prerelease (of 0.10.3).
30847 Original commit message from CVS:
30849 * win32/common/config.h:
30850 0.10.2.2 prerelease (of 0.10.3).
30852 2006-02-08 15:50:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30854 gst/tcp/gsttcpclientsrc.c: Revert Andy's newsegment change pending a more correct fix.
30855 Original commit message from CVS:
30856 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
30857 Revert Andy's newsegment change pending a more correct
30860 2006-02-08 12:46:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30877 Original commit message from CVS:
30880 2006-02-08 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30882 * gst/tcp/gstmultifdsink.c:
30884 Original commit message from CVS:
30887 2006-02-08 09:20:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30889 gst/typefind/gsttypefindfunctions.c: detect more files as 3gp group and reorder the iso file formats
30890 Original commit message from CVS:
30892 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
30893 (qt_type_find), (plugin_init):
30894 detect more files as 3gp
30895 group and reorder the iso file formats
30897 2006-02-07 18:32:00 +0000 Tim-Philipp Müller <tim@centricular.net>
30899 ext/vorbis/vorbis.c: Register musicbrainz tags, so apps don't have to.
30900 Original commit message from CVS:
30901 * ext/vorbis/vorbis.c: (plugin_init):
30902 Register musicbrainz tags, so apps don't have to.
30904 2006-02-07 17:44:53 +0000 Tim-Philipp Müller <tim@centricular.net>
30906 gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
30907 Original commit message from CVS:
30908 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
30909 (gst_tag_to_vorbis_tag):
30910 Make sure we called gst_tag_register_musicbrainz_tags()
30911 before possibly mapping a vorbiscomment string from/to a
30914 2006-02-07 16:16:41 +0000 Tim-Philipp Müller <tim@centricular.net>
30916 gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
30917 Original commit message from CVS:
30918 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
30919 In case we can't find the required number of consecutive
30920 mpeg audio frames to positively identify an MPEG audio
30921 stream, check if there's at least a valid mpeg audio
30922 frame right at offset 0 and if so suggest mpeg/audio
30923 caps with a very low probability (#153004).
30925 2006-02-07 15:52:26 +0000 Andy Wingo <wingo@pobox.com>
30927 gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to a TIME segment if we get timestamped buffers. Requir...
30928 Original commit message from CVS:
30929 2006-02-07 Andy Wingo <wingo@pobox.com>
30930 * gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
30931 a TIME segment if we get timestamped buffers. Requires recent
30932 fixes in core to work properly.
30934 2006-02-07 14:57:46 +0000 Tim-Philipp Müller <tim@centricular.net>
30936 gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
30937 Original commit message from CVS:
30938 * gst/playback/gstplaybasebin.c: (prepare_output):
30939 Don't print the URI as part of the error message, it
30940 makes error dialogs look rather ugly, especially if
30941 the URI is very long or has characters in it that
30944 2006-02-07 13:11:31 +0000 Tim-Philipp Müller <tim@centricular.net>
30946 gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
30947 Original commit message from CVS:
30948 * gst/playback/gstplaybasebin.c: (prepare_output):
30949 Error out if we have only text or subtitles, but nothing
30950 else. Also error out if we have subtitles but no video
30953 2006-02-07 11:44:39 +0000 Tim-Philipp Müller <tim@centricular.net>
30955 ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
30956 Original commit message from CVS:
30957 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
30958 Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
30959 Post an error message on the bus when we encounter an
30960 error, which will hopefully be more meaningful than the
30961 'Internal Flow Error' message users get to see if we
30962 just return GST_FLOW_ERROR.
30964 2006-02-07 11:28:04 +0000 Andy Wingo <wingo@pobox.com>
30966 configure.ac (GST_MAJORMINOR): Update core version req to 0.10.2.2, for the collectpads API addition (#330244).
30967 Original commit message from CVS:
30968 2006-02-07 Andy Wingo <wingo@pobox.com>
30969 * configure.ac (GST_MAJORMINOR): Update core version req to
30970 0.10.2.2, for the collectpads API addition (#330244).
30972 2006-02-06 19:09:26 +0000 Tim-Philipp Müller <tim@centricular.net>
30974 ext/gnomevfs/gstgnomevfs.c: Return FALSE from plugin_init() when GnomeVFS can't be initialised for some reason (#3284...
30975 Original commit message from CVS:
30976 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
30977 Return FALSE from plugin_init() when GnomeVFS can't
30978 be initialised for some reason (#328423).
30980 2006-02-06 13:26:54 +0000 Julien Moutte <julien@moutte.net>
30982 ext/pango/gsttextoverlay.c: Stick to seeking theory until i find the bug.
30983 Original commit message from CVS:
30984 2006-02-06 Julien MOUTTE <julien@moutte.net>
30985 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
30986 Stick to seeking theory until i find the bug.
30987 * gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
30989 2006-02-06 12:38:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30991 Make theoraenc and the tests leak free. Like, really.
30992 Original commit message from CVS:
30993 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
30994 (theora_enc_finalize), (theora_enc_sink_setcaps),
30995 (theora_set_header_on_caps), (theora_enc_chain),
30996 (theora_enc_change_state):
30997 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
30998 Make theoraenc and the tests leak free. Like, really.
31000 2006-02-05 23:31:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31002 Add a finalize method to ensure we clean up state even if someone omitted the state change back to NULL.
31003 Original commit message from CVS:
31004 (theora_enc_finalize), (theora_enc_sink_setcaps):
31005 Add a finalize method to ensure we clean up state even if
31006 someone omitted the state change back to NULL.
31007 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
31008 (gst_vorbisenc_chain):
31009 Free some more leaked bits.
31010 * tests/check/pipelines/theoraenc.c: (start_pipeline),
31012 Wait for state changes to happen if they're ASYNC.
31013 This ought to teach those fancy pants buildbots a lesson.
31015 2006-02-05 22:47:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31017 gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
31018 Original commit message from CVS:
31019 * gst-libs/gst/tag/gstid3tag.c:
31020 Add mapping for ID3 International Standard Recording Code
31023 2006-02-05 22:44:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31025 ext/vorbis/vorbisenc.c: Don't leak tag names.
31026 Original commit message from CVS:
31027 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
31028 Don't leak tag names.
31030 2006-02-05 18:22:01 +0000 Tim-Philipp Müller <tim@centricular.net>
31032 Split libgsttag docs into multiple sections.
31033 Original commit message from CVS:
31034 * docs/libs/gst-plugins-base-libs-docs.sgml:
31035 * docs/libs/gst-plugins-base-libs-sections.txt:
31036 * gst-libs/gst/tag/gstid3tag.c:
31037 * gst-libs/gst/tag/gstvorbistag.c:
31038 * gst-libs/gst/tag/tags.c:
31039 Split libgsttag docs into multiple sections.
31041 2006-02-05 18:01:33 +0000 Tim-Philipp Müller <tim@centricular.net>
31043 Add libgsttag to the docs.
31044 Original commit message from CVS:
31045 * docs/libs/Makefile.am:
31046 * docs/libs/gst-plugins-base-libs-docs.sgml:
31047 * docs/libs/gst-plugins-base-libs-sections.txt:
31048 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
31049 * gst-libs/gst/tag/gstvorbistag.c:
31050 * gst-libs/gst/tag/tag.h:
31051 * gst-libs/gst/tag/tags.c:
31052 Add libgsttag to the docs.
31054 2006-02-05 17:21:23 +0000 Julien Moutte <julien@moutte.net>
31056 ext/pango/gsttextoverlay.c: Fix clockoverlay.
31057 Original commit message from CVS:
31058 2006-02-05 Julien MOUTTE <julien@moutte.net>
31059 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
31060 (gst_text_overlay_init), (gst_text_overlay_src_event),
31061 (gst_text_overlay_collected): Fix clockoverlay.
31063 2006-02-05 17:15:17 +0000 Tim-Philipp Müller <tim@centricular.net>
31065 docs/libs/compiling.sgml: Fix typo: it's pkg-config, not pkg-gconfig
31066 Original commit message from CVS:
31067 * docs/libs/compiling.sgml:
31068 Fix typo: it's pkg-config, not pkg-gconfig
31069 * docs/libs/gst-plugins-base-libs-docs.sgml:
31070 * docs/libs/gst-plugins-base-libs-sections.txt:
31071 * docs/libs/tmpl/gstgconf.sgml:
31072 There is no libgstgconf in 0.10, remove it
31075 2006-02-05 16:03:48 +0000 Julien Moutte <julien@moutte.net>
31077 docs/libs/tmpl/gstcolorbalance.sgml: Updated.
31078 Original commit message from CVS:
31079 2006-02-05 Julien MOUTTE <julien@moutte.net>
31080 * docs/libs/tmpl/gstcolorbalance.sgml: Updated.
31081 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
31082 (gst_text_overlay_src_event), (gst_text_overlay_collected):
31083 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
31084 (gst_sub_parse_class_init), (gst_sub_parse_init),
31085 (gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
31086 (parse_mpsub), (parser_state_init), (handle_buffer),
31087 (gst_sub_parse_chain), (gst_sub_parse_sink_event),
31089 * gst/subparse/gstsubparse.h: Introduce seeking code.
31091 2006-02-05 15:14:06 +0000 Tim-Philipp Müller <tim@centricular.net>
31093 gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
31094 Original commit message from CVS:
31095 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
31096 Add comment about LANGUAGE tag inconsistency (we want
31097 ISO-639-1, but extract three-letter identifiers?)
31099 Add two translatable files.
31101 2006-02-05 14:59:28 +0000 Tim-Philipp Müller <tim@centricular.net>
31103 gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
31104 Original commit message from CVS:
31105 * gst-libs/gst/tag/Makefile.am:
31106 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
31107 * gst-libs/gst/tag/tag.h:
31108 * gst-libs/gst/tag/tags.c:
31109 (gst_tag_register_musicbrainz_tags_internal),
31110 (gst_tag_register_musicbrainz_tags):
31111 Forward-port some tags stuff from the 0.8 branch. This is
31112 mostly the addition of musicbrainz tags and their mapping
31113 to vorbistags, and a vorbistag mapping of the language tag.
31115 2006-02-05 12:06:25 +0000 Julien Moutte <julien@moutte.net>
31117 gst/playback/gstplaybin.c: Fix broken code refactoring.
31118 Original commit message from CVS:
31119 2006-02-05 Julien MOUTTE <julien@moutte.net>
31120 * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
31123 2006-02-05 03:05:41 +0000 David Schleef <ds@schleef.org>
31125 Add Dirac typefinding and add dirac format to oggmux.
31126 Original commit message from CVS:
31127 * ext/ogg/gstoggmux.c:
31128 * gst/typefind/gsttypefindfunctions.c:
31129 Add Dirac typefinding and add dirac format to oggmux.
31131 2006-02-04 07:49:03 +0000 Michael Smith <msmith@xiph.org>
31134 Improve error message for liboil missingness.
31135 Original commit message from CVS:
31136 Improve error message for liboil missingness.
31138 2006-02-03 19:23:41 +0000 Tim-Philipp Müller <tim@centricular.net>
31140 gst/playback/gstdecodebin.c: Don't put essential function call into g_return_*() macro, otherwise it'll all be replac...
31141 Original commit message from CVS:
31142 * gst/playback/gstdecodebin.c: (try_to_link_1):
31143 Don't put essential function call into
31144 g_return_*() macro, otherwise it'll all be
31145 replaced by NOOPs when compiling with
31146 G_DISABLE_CHECKS defined.
31148 2006-02-03 17:45:44 +0000 Edgard Lima <edgard.lima@indt.org.br>
31151 * ext/ogg/gstoggdemux.c:
31152 * ext/ogg/gstoggparse.c:
31153 * gst/tcp/gsttcpserversink.c:
31154 * sys/v4l/v4lsrc_calls.c:
31155 * sys/v4l/v4lsrc_calls.h:
31156 Just make it compile with --disable-gst-debug.
31157 Original commit message from CVS:
31158 Just make it compile with --disable-gst-debug.
31160 2006-02-03 12:51:47 +0000 Wim Taymans <wim.taymans@gmail.com>
31162 ext/alsa/gstalsasink.*: Add lock to protect alsa calls.
31163 Original commit message from CVS:
31164 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
31165 (gst_alsasink_class_init), (gst_alsasink_init),
31166 (gst_alsasink_write), (gst_alsasink_reset):
31167 * ext/alsa/gstalsasink.h:
31168 Add lock to protect alsa calls.
31169 Implement reset to flush samples ASAP, does not work
31172 2006-02-02 18:18:31 +0000 Wim Taymans <wim.taymans@gmail.com>
31174 gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
31175 Original commit message from CVS:
31176 * gst-libs/gst/audio/gstbaseaudiosink.c:
31177 (gst_base_audio_sink_provide_clock):
31178 Ugh.. getting late I guess...
31180 2006-02-02 18:13:26 +0000 Wim Taymans <wim.taymans@gmail.com>
31182 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
31183 Original commit message from CVS:
31184 * gst-libs/gst/audio/gstbaseaudiosink.c:
31185 (gst_base_audio_sink_provide_clock),
31186 (gst_base_audio_sink_set_property),
31187 (gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
31188 Don't try to provide a clock when we are not negotiated since
31189 we might not be able to make it run.
31191 2006-02-02 17:51:48 +0000 Tim-Philipp Müller <tim@centricular.net>
31193 gst/playback/gstdecodebin.c: Unlinking two source pads is ... hard.
31194 Original commit message from CVS:
31195 * gst/playback/gstdecodebin.c: (try_to_link_1):
31196 Unlinking two source pads is ... hard.
31198 2006-02-02 12:14:35 +0000 Wim Taymans <wim.taymans@gmail.com>
31200 gst-libs/gst/audio/TODO: Updated.
31201 Original commit message from CVS:
31202 * gst-libs/gst/audio/TODO:
31204 * gst-libs/gst/audio/gstbaseaudiosink.c:
31205 (gst_base_audio_sink_drain), (gst_base_audio_sink_event):
31206 On EOS, wait till the last sample is played before posting EOS.
31208 2006-02-02 08:53:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31210 * tests/check/pipelines/theoraenc.c:
31211 comment on my understanding
31212 Original commit message from CVS:
31213 comment on my understanding
31215 2006-02-02 08:47:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31218 * tests/check/pipelines/theoraenc.c:
31219 reformat to fit 80 chars
31220 Original commit message from CVS:
31221 reformat to fit 80 chars
31223 2006-02-02 00:04:37 +0000 Kai Vehmanen <kv2004@eca.cx>
31225 gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
31226 Original commit message from CVS:
31227 2006-02-01 Philippe Kalaf <burger at speedy dot org>
31228 * gst-libs/gst/rtp/gstbasertpdepayload.c:
31229 Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
31230 setting queue_delay to zero. Also avoid thread being started if
31231 queue_delay is zero.
31233 2006-02-01 14:51:29 +0000 Tim-Philipp Müller <tim@centricular.net>
31235 gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
31236 Original commit message from CVS:
31237 * gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
31238 Make test work again by connecting fakesinks to each decoded pad,
31239 which makes the pipeline wait until each fakesink has a buffer
31240 queued before going to PAUSED state. At that point we know the
31241 decodebin pads are negotiated.
31243 2006-02-01 11:59:47 +0000 Tim-Philipp Müller <tim@centricular.net>
31245 gst/: Pass unhandled queries to the parent class's query function.
31246 Original commit message from CVS:
31247 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
31248 (gst_cdda_base_src_handle_event):
31249 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
31250 Pass unhandled queries to the parent class's query function.
31252 2006-02-01 11:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
31254 Pass unhandled queries upstream instead of just dropping them (#326447). Also, fix supported query types list for som...
31255 Original commit message from CVS:
31256 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
31257 (gst_ogg_pad_src_query):
31258 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
31259 * ext/theora/theoradec.c: (theora_dec_src_query),
31260 (theora_dec_sink_query):
31261 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
31262 (vorbis_dec_sink_query):
31263 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
31264 (gst_vorbisenc_sink_query):
31265 * gst/adder/gstadder.c: (gst_adder_query):
31266 Pass unhandled queries upstream instead of just
31267 dropping them (#326447). Also, fix supported
31268 query types list for some elements.
31270 2006-02-01 09:58:15 +0000 Tim-Philipp Müller <tim@centricular.net>
31272 gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
31273 Original commit message from CVS:
31274 * gst/typefind/gsttypefindfunctions.c: (au_type_find),
31275 (paris_type_find), (ilbc_type_find), (plugin_init):
31276 Fix typefinding for audio/x-au, audio/x-paris and
31277 audio/iLBC-sh. We cannot use the START_WITH macros
31278 here, because there can only be one typefind factory
31279 with the same name (caps), so the second one would
31280 replace the first one and the first one would never
31281 be called when doing typefinding (see #161712).
31283 2006-01-31 19:25:10 +0000 Wim Taymans <wim.taymans@gmail.com>
31285 ext/vorbis/vorbisdec.c: Use scale_int when we can, add some more scaling.
31286 Original commit message from CVS:
31287 * ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
31288 (vorbis_handle_header_packet), (vorbis_dec_push),
31289 (vorbis_handle_data_packet):
31290 Use scale_int when we can, add some more scaling.
31291 Check packettype before parsing it.
31293 2006-01-31 17:44:35 +0000 Wim Taymans <wim.taymans@gmail.com>
31295 ext/theora/theoradec.c: Call right _scale functions.
31296 Original commit message from CVS:
31297 * ext/theora/theoradec.c: (_theora_granule_time),
31298 (theora_dec_src_convert), (theora_dec_sink_convert):
31299 Call right _scale functions.
31300 Use parameter instead of some other random value.
31302 2006-01-31 17:27:00 +0000 Wim Taymans <wim.taymans@gmail.com>
31304 ext/theora/theoradec.c: Use higher precision timestamps calculation.
31305 Original commit message from CVS:
31306 * ext/theora/theoradec.c: (_theora_granule_frame),
31307 (_theora_granule_time), (_inc_granulepos),
31308 (theora_dec_src_convert), (theora_dec_sink_convert),
31309 (theora_handle_type_packet), (theora_handle_data_packet),
31310 (theora_dec_chain):
31311 Use higher precision timestamps calculation.
31312 Convert some other conversions to _scale.
31314 2006-01-31 17:19:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31316 gst/: initialize gst_controller before using
31317 Original commit message from CVS:
31318 * gst/audiotestsrc/gstaudiotestsrc.c:
31319 (gst_audio_test_src_create_sine_table), (plugin_init):
31320 * gst/volume/gstvolume.c: (plugin_init):
31321 initialize gst_controller before using
31323 2006-01-31 16:26:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31325 tests/check/pipelines/: Define constant using G_GINT64_CONSTANT to avoid errors when passing it around - otherwise it...
31326 Original commit message from CVS:
31327 * tests/check/pipelines/theoraenc.c:
31328 * tests/check/pipelines/vorbisenc.c:
31329 Define constant using G_GINT64_CONSTANT to avoid errors when
31330 passing it around - otherwise it gets truncated to 32 bits.
31331 Fixes failing tests.
31333 2006-01-31 15:36:13 +0000 Andy Wingo <wingo@pobox.com>
31335 sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the caps being set doesn't have a framerate value. Basic...
31336 Original commit message from CVS:
31337 2006-01-31 Andy Wingo <wingo@pobox.com>
31338 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
31339 caps being set doesn't have a framerate value. Basically a stopgap
31341 * ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
31342 technically correct enough to put into core though.
31343 (gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
31344 DURATION. Fixes theoraenc ! oggmux.
31345 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
31346 fraction, not double.
31348 2006-01-31 12:23:35 +0000 Christian Schaller <uraeus@gnome.org>
31350 * gst-plugins-base.spec.in:
31351 update with latest files
31352 Original commit message from CVS:
31353 update with latest files
31355 2006-01-30 23:42:54 +0000 Sébastien Moutte <sebastien@moutte.net>
31357 win32/vs7: add vs7 project files created by Sergey Scobich
31358 Original commit message from CVS:
31360 add vs7 project files created by Sergey Scobich
31362 2006-01-30 22:18:53 +0000 Sébastien Moutte <sebastien@moutte.net>
31364 win32/vs8: add vs8 project files created by Sergey Scobich
31365 Original commit message from CVS:
31367 add vs8 project files created by Sergey Scobich
31369 2006-01-30 19:22:22 +0000 Andy Wingo <wingo@pobox.com>
31371 ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare timestamp + duration, not just timestamp -- ogg pages should ...
31372 Original commit message from CVS:
31373 2006-01-30 Andy Wingo <wingo@pobox.com>
31374 * ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
31375 timestamp + duration, not just timestamp -- ogg pages should be
31376 ordered by stop time. Necessary fix given the change in vorbis
31379 2006-01-30 19:21:07 +0000 Andy Wingo <wingo@pobox.com>
31382 * ext/theora/gsttheoraenc.h:
31383 * ext/theora/theoraenc.c:
31384 * tests/check/pipelines/theoraenc.c:
31385 ext/theora/theoraenc.c (theora_enc_sink_setcaps)
31386 Original commit message from CVS:
31387 2006-01-30 Andy Wingo <wingo@pobox.com>
31388 * ext/theora/theoraenc.c (theora_enc_sink_setcaps)
31389 (gst_theora_enc_init): Pull the granule shift out of the encoder.
31390 (granulepos_add): New function, handles the messiness of adjusting
31392 (theora_buffer_from_packet):
31393 (theora_enc_chain):
31394 (theora_enc_sink_event): Use granulepos_add, not +.
31395 * tests/check/pipelines/theoraenc.c
31396 (check_buffer_granulepos_from_starttime): Just check the frame
31397 count, not the actual granulepos -- we can't dictate to the
31398 encoder when it should be placing keyframes.
31400 2006-01-30 18:17:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31402 ext/gnomevfs/gstgnomevfssrc.c: SERVICE_NOT_AVAILABLE happens for example when you're trying to play an http:// stream...
31403 Original commit message from CVS:
31404 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
31405 SERVICE_NOT_AVAILABLE happens for example when you're trying to
31406 play an http:// stream from a server that's not serving
31408 2006-01-30 17:08:11 +0000 Andy Wingo <wingo@pobox.com>
31410 tests/check/pipelines/: Totally remove the UINT64_CONSTANT macro, doesn't appear to be needed or available.
31411 Original commit message from CVS:
31412 2006-01-30 Andy Wingo <wingo@pobox.com>
31413 * tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
31414 * tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
31415 remove the UINT64_CONSTANT macro, doesn't appear to be needed or
31418 2006-01-30 17:01:54 +0000 Andy Wingo <wingo@pobox.com>
31420 ext/theora/: Same changes as were done to vorbisenc, although theoraenc was timestamping correctly. Added handling of...
31421 Original commit message from CVS:
31422 2006-01-30 Andy Wingo <wingo@pobox.com>
31423 * ext/theora/gsttheoraenc.h:
31424 * ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
31425 although theoraenc was timestamping correctly. Added handling of
31426 streams that start with nonzero timestamps.
31427 * tests/check/Makefile.am:
31428 * tests/check/pipelines/theoraenc.c: New file, basically does same
31429 tests as vorbisenc.
31430 * tests/check/pipelines/vorbisenc.c: I claim these bugs.
31432 2006-01-30 16:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
31434 gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
31435 Original commit message from CVS:
31436 * gst-libs/gst/audio/gstaudiosink.c:
31437 (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
31438 (gst_audioringbuffer_pause):
31439 Implement pause that does not wait for completion.
31440 * gst-libs/gst/audio/gstbaseaudiosink.c:
31441 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
31442 Don't drop buffers when going to PAUSED but perform preroll on
31443 remaining samples now that core base class supports this.
31444 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
31445 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
31446 (gst_ring_buffer_commit):
31447 Pause should not signal waiters.
31448 Implement return value of _commit correctly.
31450 2006-01-30 15:01:28 +0000 Andy Wingo <wingo@pobox.com>
31452 tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
31453 Original commit message from CVS:
31454 2006-01-30 Andy Wingo <wingo@pobox.com>
31455 * tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
31456 * ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
31457 updated to timestamp from the first sample, not the last.
31458 (gst_vorbisenc_buffer_from_header_packet): New function, takes
31459 special care of granulepos and timestamp for header packets.
31460 (gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
31461 when the first buffer has a nonzero timestamp.
31462 * ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
31463 (GstVorbisEnc.subgranule_offset): New members. Take care of the
31464 case when the first audio buffer we get has a nonzero timestamp.
31465 (GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
31466 properly timestamp vorbis buffers with the time of the first
31467 sample, not the last.
31468 * ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
31469 vorbis_granule_time_copy -- now it takes the granule/subgranule
31470 offset into account.
31471 * tests/check/pipelines/vorbisenc.c: New test for correctness of
31472 timestamps, durations, and granulepos on buffers produced by
31475 2006-01-30 14:42:28 +0000 Eric Jonas <jonas@mit.edu>
31477 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Patch from Eric Jonas to support conversions to/from UYVY (Fixes: #324626)
31478 Original commit message from CVS:
31479 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
31480 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
31481 Patch from Eric Jonas to support conversions to/from UYVY
31484 2006-01-30 08:11:14 +0000 Julien Moutte <julien@moutte.net>
31486 gst/playback/: Implement subtitles.
31487 Original commit message from CVS:
31488 2006-01-30 Julien MOUTTE <julien@moutte.net>
31489 * gst/playback/gstplaybasebin.c: (group_commit),
31491 (setup_subtitle), (setup_source), (set_active_source):
31492 * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
31493 (gen_text_element), (gen_audio_element), (gen_vis_element),
31494 (remove_sinks), (add_sink), (setup_sinks): Implement subtitles.
31496 2006-01-29 19:13:39 +0000 Sébastien Moutte <sebastien@moutte.net>
31498 gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
31499 Original commit message from CVS:
31500 * gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
31501 * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
31502 use of gst_guint64_to_gdouble to be compliant with vs6
31503 * gst/playback/gstdecodebin.c: (try_to_link_1)
31504 * gst/videorate/videorate.c: (gst_video_rate_blank_data)
31505 use of G_GINT64_CONSTANT for int64 constants
31506 * win32/common/libgstinterfaces.def:
31507 export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
31509 update and add new project files
31511 2006-01-29 18:21:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31513 add a win32-update rule like in core, and copy over enumtypes files
31514 Original commit message from CVS:
31517 * win32/common/interfaces-enumtypes.c:
31518 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
31519 (gst_mixer_track_flags_get_type),
31520 (gst_tuner_channel_flags_get_type):
31521 * win32/common/interfaces-enumtypes.h:
31522 * win32/common/multichannel-enumtypes.c:
31523 (gst_audio_channel_position_get_type):
31524 * win32/common/multichannel-enumtypes.h:
31525 add a win32-update rule like in core, and copy over enumtypes files
31527 2006-01-29 18:07:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31530 generate win32/common/config.h
31531 Original commit message from CVS:
31532 generate win32/common/config.h
31534 2006-01-29 18:05:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31536 win32/: add config files just like in core
31537 Original commit message from CVS:
31539 * win32/common/config.h:
31540 * win32/common/config.h.in:
31541 add config files just like in core
31543 2006-01-28 18:22:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31545 ext/alsa/: Update all error messages. All of them should either use the default translated message, or actually prov...
31546 Original commit message from CVS:
31547 * ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
31548 (set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
31549 (gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
31550 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
31551 (set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
31552 (gst_alsasrc_unprepare), (gst_alsasrc_read):
31553 Update all error messages. All of them should either use
31554 the default translated message, or actually provide a
31555 translatable string.
31556 Make the string for channel count problems meaningful.
31558 2006-01-28 18:19:18 +0000 Tim-Philipp Müller <tim@centricular.net>
31560 gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
31561 Original commit message from CVS:
31562 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
31563 Make gcc-4.1 happy (part of #327357).
31565 2006-01-28 16:35:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31567 sys/v4l/v4l_calls.c: check for and throw RESOURCE_BUSY
31568 Original commit message from CVS:
31569 * sys/v4l/v4l_calls.c: (gst_v4l_open):
31570 check for and throw RESOURCE_BUSY
31572 2006-01-28 02:13:14 +0000 David Schleef <ds@schleef.org>
31574 gst/videoscale/vs_scanline.c: Oops, *that's* why I never checked in this change -- it requires liboil features not in...
31575 Original commit message from CVS:
31576 * gst/videoscale/vs_scanline.c: Oops, *that's* why I never
31577 checked in this change -- it requires liboil features not
31578 in 0.3.6. Revert parts.
31580 2006-01-27 23:40:19 +0000 David Schleef <ds@schleef.org>
31582 update liboil requirement to 0.3.6
31583 Original commit message from CVS:
31585 * configure.ac: update liboil requirement to 0.3.6
31586 * gst/videoscale/Makefile.am:
31587 * gst/videoscale/vs_scanline.c: liboilify
31589 2006-01-27 17:00:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31591 ext/libvisual/visual.c: When pad_alloc returns a GstFlowReturn other than GST_FLOW_OK, make sure it is passed upstream.
31592 Original commit message from CVS:
31593 * ext/libvisual/visual.c: (get_buffer):
31594 When pad_alloc returns a GstFlowReturn other
31595 than GST_FLOW_OK, make sure it is passed upstream.
31597 2006-01-27 01:36:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31599 ext/alsa/gstalsasink.c: Free the device name string.
31600 Original commit message from CVS:
31601 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
31602 (gst_alsasink_class_init):
31603 Free the device name string.
31604 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
31605 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
31606 (gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
31607 Don't remove a pad from the collectpads structure until it
31608 is released - it's a request pad, and may receive data again
31609 if the element gets moved back to PLAYING state.
31610 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
31611 Ensure we turn on double buffering on the Xv port, and
31612 set the colour key to something dark and mysterious that
31615 2006-01-27 01:06:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31617 ext/: - a library should not call setlocale. see Libraries node in gettext manual
31618 Original commit message from CVS:
31619 * ext/alsa/gstalsaplugin.c: (plugin_init):
31620 * ext/cdparanoia/gstcdparanoiasrc.c:
31621 (gst_cd_paranoia_src_base_init), (plugin_init):
31622 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
31623 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
31624 - a library should not call setlocale. see Libraries node in
31626 - make sure all plugins that use translation do bindtextdomain
31627 to point to the localedir
31628 * gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
31629 (setup_sinks), (plugin_init):
31630 all this, and check for NULL when creating sinks
31632 2006-01-26 23:21:31 +0000 Julien Moutte <julien@moutte.net>
31634 gst/subparse/gstsubparse.c: Make typefinding of subtitles work again.
31635 Original commit message from CVS:
31636 2006-01-27 Julien MOUTTE <julien@moutte.net>
31637 * gst/subparse/gstsubparse.c: (gst_subparse_type_find),
31638 (plugin_init): Make typefinding of subtitles work again.
31640 2006-01-26 20:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
31642 gst/typefind/gsttypefindfunctions.c: Backport a bunch of typefinding fixes from the 0.8 branch.
31643 Original commit message from CVS:
31644 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
31645 (mp3_type_frame_length_from_header), (mp3_type_find),
31646 (wavpack_type_find), (m4a_type_find), (ircam_type_find),
31648 Backport a bunch of typefinding fixes from the 0.8 branch.
31649 Also, improve wavpack typefinding: if we can't peek the
31650 entire wavpack block, try to parse the bits we can get and
31651 see if we find what we're looking for in those.
31653 2006-01-26 19:17:38 +0000 Julien Moutte <julien@moutte.net>
31655 sys/: Handle some more cases of pixel aspect ratio.
31656 Original commit message from CVS:
31657 2006-01-26 Julien MOUTTE <julien@moutte.net>
31658 * sys/ximage/ximagesink.c:
31659 (gst_ximagesink_calculate_pixel_aspect_ratio):
31660 * sys/xvimage/xvimagesink.c:
31661 (gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
31662 more cases of pixel aspect ratio.
31664 2006-01-26 13:09:24 +0000 Edward Hervey <bilboed@bilboed.com>
31666 gst/playback/gstdecodebin.c: Also consider the flush-start and tag events as unblockers for the pad probes.
31667 Original commit message from CVS:
31668 * gst/playback/gstdecodebin.c: (pad_probe):
31669 Also consider the flush-start and tag events as unblockers
31670 for the pad probes.
31672 2006-01-26 12:32:58 +0000 Julien Moutte <julien@moutte.net>
31674 gst/playback/gstplaybin.c: On the fly visualisation switch, works disabling, enabling as well but it won't be able to...
31675 Original commit message from CVS:
31676 2006-01-26 Julien MOUTTE <julien@moutte.net>
31677 * gst/playback/gstplaybin.c: (gst_play_bin_init),
31678 (gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
31679 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
31680 On the fly visualisation switch, works disabling, enabling as
31681 well but it won't be able to enable vis in a playbin that was
31682 created with no visualisation.
31684 2006-01-25 10:50:32 +0000 Wim Taymans <wim.taymans@gmail.com>
31686 gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
31687 Original commit message from CVS:
31688 * gst-libs/gst/audio/gstbaseaudiosink.c:
31689 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
31690 Undo previous commit, it breaks resume after pause.
31692 2006-01-25 09:27:01 +0000 Wim Taymans <wim.taymans@gmail.com>
31694 gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
31695 Original commit message from CVS:
31696 * gst-libs/gst/audio/gstbaseaudiosink.c:
31697 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
31698 (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
31700 Post error when caps cannot be parsed.
31701 Resync on discontinuity in the stream.
31702 Clip samples to segment boundaries.
31703 return WRONG_STATE sooner when we are flushing.
31704 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
31705 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
31706 Make audiosrc operate in TIME.
31707 Set TIMESTAMP and DURATION on buffers.
31709 2006-01-24 21:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
31711 tests/examples/seek/seek.c: Output tag messages as well.
31712 Original commit message from CVS:
31713 * tests/examples/seek/seek.c: (main):
31714 Output tag messages as well.
31716 2006-01-23 15:05:24 +0000 Edward Hervey <bilboed@bilboed.com>
31718 gst/playback/gstdecodebin.c: Replace GstPadBlockCallback with pad probes that detect first buffer AND eos before remo...
31719 Original commit message from CVS:
31720 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
31721 (free_pad_probes), (remove_fakesink), (pad_probe),
31722 (close_pad_link), (gst_decode_bin_change_state):
31723 Replace GstPadBlockCallback with pad probes that detect
31724 first buffer AND eos before removing fakesink.
31725 Fixes hang with demuxers doing EOS while pre-rolling.
31728 2006-01-23 10:10:36 +0000 Jens Granseuer <jensgr@gmx.net>
31730 GCC 2.95 fixes (#328263).
31731 Original commit message from CVS:
31732 2006-01-23 Andy Wingo <wingo@pobox.com>
31733 * ext/alsa/gstalsasink.c:
31734 * gst-libs/gst/rtp/gstbasertpdepayload.c:
31735 (gst_base_rtp_depayload_setcaps),
31736 (gst_base_rtp_depayload_add_to_queue),
31737 (gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
31738 Patch by: Jens Granseuer <jensgr at gmx dot net>
31740 2006-01-22 17:24:02 +0000 Julien Moutte <julien@moutte.net>
31742 sys/: Playbin keeps some ref to some frames. We might get a frame destroyed after changing state to
31743 Original commit message from CVS:
31744 2006-01-22 Julien MOUTTE <julien@moutte.net>
31745 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
31746 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
31747 (gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
31748 frames. We might get a frame destroyed after changing state to
31749 NULL, adding a safety check on xcontext.
31751 2006-01-22 14:50:53 +0000 Tim-Philipp Müller <tim@centricular.net>
31753 gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
31754 Original commit message from CVS:
31755 * gst-libs/gst/interfaces/xoverlay.c:
31756 Fix prepare-xwindow-id code example in the docs - we need to
31757 ignore all messages that aren't element messages as well.
31759 2006-01-21 22:40:03 +0000 Julien Moutte <julien@moutte.net>
31761 sys/xvimage/xvimagesink.c: I think one day i'll completely undestand how caps negotiation is supposed to work. This r...
31762 Original commit message from CVS:
31763 2006-01-21 Julien MOUTTE <julien@moutte.net>
31764 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
31765 I think one day i'll completely undestand how caps negotiation
31766 is supposed to work. This refactoring handles buffer_alloc
31767 called with caps we can't handle. We definitely don't want a
31768 set_caps with those caps, so we define and allocate a buffer
31769 we would like to receive.
31771 2006-01-20 19:10:26 +0000 Christian Schaller <uraeus@gnome.org>
31775 up automake requirement to 1.7
31776 Original commit message from CVS:
31777 up automake requirement to 1.7
31779 2006-01-19 10:59:51 +0000 Tim-Philipp Müller <tim@centricular.net>
31781 gst/playback/gstplaybasebin.c: Free iterator when done.
31782 Original commit message from CVS:
31783 * gst/playback/gstplaybasebin.c: (setup_source):
31784 Free iterator when done.
31786 2006-01-17 11:43:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31788 gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
31789 Original commit message from CVS:
31790 * gst-libs/gst/audio/gstbaseaudiosink.c:
31791 (gst_base_audio_sink_render):
31792 Fix playback of non-synchronised streams by assuming a rate
31793 of 1.0 instead of a random one.
31794 Makes this work again:
31795 gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
31796 endianness=(int)4321, signed=(boolean)true, width=(int)16,
31797 depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
31798 audioresample ! alsasink
31800 2006-01-16 21:01:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31804 Original commit message from CVS:
31807 === release 0.10.2 ===
31809 2006-01-16 20:59:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31815 * docs/plugins/gst-plugins-base-plugins.args:
31816 * docs/plugins/inspect/plugin-adder.xml:
31817 * docs/plugins/inspect/plugin-alsa.xml:
31818 * docs/plugins/inspect/plugin-audioconvert.xml:
31819 * docs/plugins/inspect/plugin-audiorate.xml:
31820 * docs/plugins/inspect/plugin-audioresample.xml:
31821 * docs/plugins/inspect/plugin-audiotestsrc.xml:
31822 * docs/plugins/inspect/plugin-cdparanoia.xml:
31823 * docs/plugins/inspect/plugin-decodebin.xml:
31824 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
31825 * docs/plugins/inspect/plugin-gnomevfs.xml:
31826 * docs/plugins/inspect/plugin-libvisual.xml:
31827 * docs/plugins/inspect/plugin-ogg.xml:
31828 * docs/plugins/inspect/plugin-pango.xml:
31829 * docs/plugins/inspect/plugin-playbin.xml:
31830 * docs/plugins/inspect/plugin-subparse.xml:
31831 * docs/plugins/inspect/plugin-tcp.xml:
31832 * docs/plugins/inspect/plugin-theora.xml:
31833 * docs/plugins/inspect/plugin-typefindfunctions.xml:
31834 * docs/plugins/inspect/plugin-video4linux.xml:
31835 * docs/plugins/inspect/plugin-videorate.xml:
31836 * docs/plugins/inspect/plugin-videoscale.xml:
31837 * docs/plugins/inspect/plugin-videotestsrc.xml:
31838 * docs/plugins/inspect/plugin-volume.xml:
31839 * docs/plugins/inspect/plugin-vorbis.xml:
31840 * docs/plugins/inspect/plugin-ximagesink.xml:
31841 * docs/plugins/inspect/plugin-xvimagesink.xml:
31843 Original commit message from CVS:
31846 2006-01-16 16:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31848 gst/playback/: Comment out broken code that connects to the state-changed signal.
31849 Original commit message from CVS:
31850 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
31851 * gst/playback/gststreamselector.c:
31852 (gst_stream_selector_set_property):
31853 Comment out broken code that connects to the state-changed signal.
31854 At this point, changing current stream selection is broken, but
31855 stuff like gst-launch playbin current-audio=1 works and filters
31856 to the chosen stream.
31858 2006-01-16 15:31:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31860 ext/vorbis/vorbisdec.c: Fix #327216 (null dereference in vorbisdec)
31861 Original commit message from CVS:
31862 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
31863 Fix #327216 (null dereference in vorbisdec)
31865 2006-01-16 15:19:55 +0000 Tim-Philipp Müller <tim@centricular.net>
31867 ext/theora/theoradec.c: Post taglist actually on bus instead of just freeing it (fixes #327114 and totem bug #327080).
31868 Original commit message from CVS:
31869 * ext/theora/theoradec.c: (theora_handle_comment_packet):
31870 Post taglist actually on bus instead of just freeing it
31871 (fixes #327114 and totem bug #327080).
31872 * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
31873 Use gst_element_found_tags_for_pad(), so that the tags
31874 are sent downstream as an event as well.
31876 2006-01-15 10:06:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31878 sys/: move all regularly occurring messages to GST_LOG level add some more object logs
31879 Original commit message from CVS:
31880 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
31881 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
31882 (gst_ximagesink_buffer_alloc):
31883 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
31884 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
31885 (gst_xvimagesink_buffer_alloc):
31886 move all regularly occurring messages to GST_LOG level
31887 add some more object logs
31889 2006-01-14 22:59:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31907 Original commit message from CVS:
31910 2006-01-14 20:46:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31912 ext/ogg/gstoggmux.c: fix a silly segfault
31913 Original commit message from CVS:
31914 2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
31915 * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
31916 fix a silly segfault
31918 2006-01-14 12:52:22 +0000 Tim-Philipp Müller <tim@centricular.net>
31920 Add docs for mixerutils stuff.
31921 Original commit message from CVS:
31922 * docs/libs/gst-plugins-base-libs-docs.sgml:
31923 * docs/libs/gst-plugins-base-libs-sections.txt:
31924 * gst-libs/gst/audio/mixerutils.c:
31925 * gst-libs/gst/audio/mixerutils.h:
31926 Add docs for mixerutils stuff.
31928 2006-01-13 17:17:07 +0000 Tim-Philipp Müller <tim@centricular.net>
31930 gst/playback/gstplaybasebin.c: Fix playback for sources that emit raw audio or raw video streams (e.g.: cd audio sour...
31931 Original commit message from CVS:
31932 * gst/playback/gstplaybasebin.c: (setup_source):
31933 Fix playback for sources that emit raw audio or
31934 raw video streams (e.g.: cd audio sources) (#325984).
31936 2006-01-13 16:45:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31938 gst-libs/gst/audio/mixerutils.c: actually save the element we create
31939 Original commit message from CVS:
31940 * gst-libs/gst/audio/mixerutils.c:
31941 (gst_audio_mixer_filter_do_filter):
31942 actually save the element we create
31944 2006-01-13 16:17:50 +0000 Christian Schaller <uraeus@gnome.org>
31946 * gst-plugins-base.spec.in:
31947 remove version suffix
31948 Original commit message from CVS:
31949 remove version suffix
31951 2006-01-12 14:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
31953 gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
31954 Original commit message from CVS:
31955 * gst-libs/gst/cdda/gstcddabasesrc.c:
31956 (gst_cdda_base_src_handle_track_seek):
31957 No need to post a tag message on the bus when seeking
31958 within the same track, only post it when the current
31961 2006-01-11 18:30:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31963 gst/playback/: Reenable stream selection. These mechanisms need a complete overhaul in the face of 0.8->0.10 changes ...
31964 Original commit message from CVS:
31965 * gst/playback/gstplaybasebin.c: (group_destroy),
31966 (probe_triggered), (new_decoded_pad), (mute_group_type),
31967 (set_active_source):
31968 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
31969 * gst/playback/gststreamselector.c:
31970 (gst_stream_selector_base_init),
31971 (gst_stream_selector_set_property),
31972 (gst_stream_selector_request_new_pad):
31973 Reenable stream selection. These mechanisms need a complete overhaul
31974 in the face of 0.8->0.10 changes though.
31976 2006-01-11 18:03:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31978 ext/ogg/gstoggdemux.c: Change the pad template to src_%d to match the pads that are created from it. decodebin needs ...
31979 Original commit message from CVS:
31980 * ext/ogg/gstoggdemux.c:
31981 Change the pad template to src_%d to match the pads that
31982 are created from it. decodebin needs this information in order
31983 to decide that oggdemux is capable of producing multiple pads
31984 (and hence needs queues inserted).
31985 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
31986 (gst_ogg_mux_collected):
31987 Make debug output more useful by using GST_PTR_FORMAT.
31989 2006-01-11 17:38:35 +0000 Christian Schaller <uraeus@gnome.org>
31991 * gst-plugins-base.spec.in:
31992 update spec.in file
31993 Original commit message from CVS:
31994 update spec.in file
31996 2006-01-11 15:11:20 +0000 Tim-Philipp Müller <tim@centricular.net>
31998 gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
31999 Original commit message from CVS:
32000 Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
32001 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
32002 Set depth and width for alaw/mulaw (fixes #326601).
32004 2006-01-10 23:58:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32006 tests/icles/Makefile.am: don't build the tests if we don't have the libs
32007 Original commit message from CVS:
32008 * tests/icles/Makefile.am:
32009 don't build the tests if we don't have the libs
32011 2006-01-10 18:06:56 +0000 Tim-Philipp Müller <tim@centricular.net>
32013 ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
32014 Original commit message from CVS:
32015 * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
32016 (gst_cd_paranoia_paranoia_callback):
32017 Don't try to free NULL pointers.
32019 2006-01-10 15:47:48 +0000 Edward Hervey <bilboed@bilboed.com>
32021 gst/audiorate/gstaudiorate.c: Add debugging category.
32022 Original commit message from CVS:
32023 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
32024 (gst_audio_rate_change_state), (plugin_init):
32025 Add debugging category.
32027 Add case for incoming buffers without valid offset/offset_end.
32029 2006-01-10 12:25:59 +0000 Michael Smith <msmith@xiph.org>
32031 gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
32032 Original commit message from CVS:
32033 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
32034 Don't leak GCond in audio sources.
32036 2006-01-10 11:49:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32038 gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
32039 Original commit message from CVS:
32040 * gst/playback/gstplaybin.c: (gen_audio_element):
32041 Don't leak an autoaudiosink/alsasink when we generate
32042 a new audio element. (old code, I guess)
32044 2006-01-10 11:04:21 +0000 Michael Smith <msmith@xiph.org>
32046 gst/audiorate/gstaudiorate.c: Support float audio in audiorate.
32047 Original commit message from CVS:
32048 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
32049 Support float audio in audiorate.
32050 Use width rather than depth for selecting sample width.
32052 2006-01-10 10:06:53 +0000 Tim-Philipp Müller <tim@centricular.net>
32054 gst/videotestsrc/videotestsrc.h: Use GLib types here (that way we don't have to include the generated _stdint.h heade...
32055 Original commit message from CVS:
32056 * gst/videotestsrc/videotestsrc.h:
32057 Use GLib types here (that way we don't have to include the
32058 generated _stdint.h header, which makes life easier for win32
32059 folks that don't use autotools for the build) (#325990, patch
32060 by: Sergey Scobich).
32062 2006-01-10 09:38:44 +0000 Tim-Philipp Müller <tim@centricular.net>
32064 gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
32065 Original commit message from CVS:
32066 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
32067 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
32068 (gst_ring_buffer_pause), (wait_segment):
32069 * gst-libs/gst/audio/gstringbuffer.h:
32070 Name (private) union, makes Forte compiler happy (this time
32071 for real) (#324900).
32073 2006-01-09 10:52:33 +0000 Tim-Philipp Müller <tim@centricular.net>
32075 gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
32076 Original commit message from CVS:
32077 * gst-libs/gst/audio/Makefile.am:
32078 Link against libgstinterfaces, needed for mixer
32079 and property probe stuff.
32081 2006-01-09 10:46:52 +0000 Edward Hervey <bilboed@bilboed.com>
32083 gst-libs/gst/Makefile.am:
32084 Original commit message from CVS:
32085 * gst-libs/gst/Makefile.am:
32087 2006-01-09 09:38:34 +0000 Tim-Philipp Müller <tim@centricular.net>
32089 gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
32090 Original commit message from CVS:
32091 * gst-libs/gst/audio/Makefile.am:
32092 * gst-libs/gst/audio/mixerutils.c:
32093 (gst_audio_mixer_filter_do_filter),
32094 (gst_audio_mixer_filter_check_element),
32095 (gst_audio_mixer_filter_probe_feature),
32096 (element_factory_rank_compare_func),
32097 (gst_audio_default_registry_mixer_filter):
32098 * gst-libs/gst/audio/mixerutils.h:
32099 Add gst_audio_default_registry_mixer_filter() utility
32102 2006-01-03 17:33:16 +0000 Michael Smith <msmith@xiph.org>
32104 gst/audioresample/resample.h: As before, but for o_buf
32105 Original commit message from CVS:
32106 * gst/audioresample/resample.h:
32107 As before, but for o_buf
32109 2006-01-03 17:27:13 +0000 Michael Smith <msmith@xiph.org>
32111 gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
32112 Original commit message from CVS:
32113 * gst/audioresample/resample.h:
32114 Declare struct _ResampleState.buffer as unsigned char *, not void *,
32115 since we do arithmetic on it.
32117 2006-01-02 23:37:38 +0000 Tim-Philipp Müller <tim@centricular.net>
32119 gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
32120 Original commit message from CVS:
32121 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
32122 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
32123 (gst_ring_buffer_pause), (wait_segment):
32124 * gst-libs/gst/audio/gstringbuffer.h:
32125 Sun's Forte compiler doesn't seem to like anonymous structs,
32126 so use same setup as in GstBaseSrc (fixes #324900).
32128 2005-12-30 14:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32130 move old example to tests/examples/volume/volune.c
32131 Original commit message from CVS:
32133 * gst/volume/Makefile.am:
32134 * gst/volume/demo.c:
32135 move old example to tests/examples/volume/volune.c
32136 * tests/examples/Makefile.am:
32137 * tests/examples/seek/seek.c: (main):
32138 change window-close event from "delete-event" to "destroy"
32139 * tests/examples/volume/Makefile.am:
32140 * tests/examples/volume/volume.c: (value_changed_callback),
32141 (setup_gui), (message_received), (eos_message_received), (main):
32142 fix event handling and bus usage
32144 2005-12-29 20:37:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32146 gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...
32147 Original commit message from CVS:
32148 * gst/audiotestsrc/gstaudiotestsrc.c:
32149 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
32150 (gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
32151 (gst_audio_test_src_query), (gst_audio_test_src_create_sine),
32152 (gst_audio_test_src_create_square),
32153 (gst_audio_test_src_create_saw),
32154 (gst_audio_test_src_create_triangle),
32155 (gst_audio_test_src_create_silence),
32156 (gst_audio_test_src_create_white_noise),
32157 (gst_audio_test_src_create_pink_noise),
32158 (gst_audio_test_src_init_sine_table),
32159 (gst_audio_test_src_create_sine_table),
32160 (gst_audio_test_src_change_wave),
32161 (gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
32162 (gst_audio_test_src_create), (gst_audio_test_src_set_property):
32163 * gst/audiotestsrc/gstaudiotestsrc.h:
32164 update to basesrc changes, implement segmented seeking and eos handling,
32165 add a 'sine-tab' waveform for performance critical playback
32167 2005-12-29 16:17:55 +0000 Tim-Philipp Müller <tim@centricular.net>
32169 po/POTFILES.in: ... and this time the other modified file that I missed last time.
32170 Original commit message from CVS:
32172 ... and this time the other modified file that I missed last time.
32174 2005-12-29 14:31:49 +0000 Michael Smith <msmith@xiph.org>
32176 gst/playback/gstdecodebin.c: Fix non-C89 variable declaration not at the start of a block. Should help some compilers.
32177 Original commit message from CVS:
32178 * gst/playback/gstdecodebin.c: (new_pad):
32179 Fix non-C89 variable declaration not at the start of a block. Should
32180 help some compilers.
32182 2005-12-29 12:43:22 +0000 Tim-Philipp Müller <tim@centricular.net>
32184 tests/check/Makefile.am: And now fix 'make distcheck' (builddir != srcdir)
32185 Original commit message from CVS:
32186 * tests/check/Makefile.am:
32187 And now fix 'make distcheck' (builddir != srcdir)
32189 2005-12-29 12:22:24 +0000 Tim-Philipp Müller <tim@centricular.net>
32191 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia plugin again (there are still fixes required to pla...
32192 Original commit message from CVS:
32194 * ext/cdparanoia/Makefile.am:
32195 * ext/cdparanoia/gstcdparanoia.c:
32196 * ext/cdparanoia/gstcdparanoia.h:
32197 * ext/cdparanoia/gstcdparanoiasrc.c:
32198 (gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
32199 (gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
32200 (gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
32201 (gst_cd_paranoia_paranoia_callback),
32202 (gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
32203 (gst_cd_paranoia_src_set_property),
32204 (gst_cd_paranoia_src_get_property), (plugin_init):
32205 * ext/cdparanoia/gstcdparanoiasrc.h:
32206 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
32207 plugin again (there are still fixes required to playbin to make
32208 cdda:// uris work there).
32210 2005-12-29 12:13:57 +0000 Tim-Philipp Müller <tim@centricular.net>
32212 tests/check/Makefile.am: Fix test case compilation.
32213 Original commit message from CVS:
32214 * tests/check/Makefile.am:
32215 Fix test case compilation.
32217 2005-12-29 11:49:11 +0000 Tim-Philipp Müller <tim@centricular.net>
32219 gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
32220 Original commit message from CVS:
32221 * gst-libs/gst/cdda/gstcddabasesrc.c:
32222 (gst_cdda_base_src_update_duration),
32223 (gst_cdda_base_src_calculate_cddb_id):
32224 An integer is not a string. Fix access to uninitialised variable.
32225 * tests/check/Makefile.am:
32226 Add cddabasesrc unit test; also actually enable the vorbis test.
32227 * tests/check/generic/states.c:
32228 Blacklist new cd audio elements as well.
32229 * tests/check/libs/cddabasesrc.c:
32230 Unit test for GstCddaBaseSrc (discid calculation mostly).
32232 2005-12-28 18:19:25 +0000 Tim-Philipp Müller <tim@centricular.net>
32234 docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
32235 Original commit message from CVS:
32236 * docs/libs/Makefile.am:
32237 * docs/libs/gst-plugins-base-libs-docs.sgml:
32238 * docs/libs/gst-plugins-base-libs-sections.txt:
32239 * docs/libs/gst-plugins-base-libs.types:
32240 Add docs for libgstcdda/GstCddaBaseSrc.
32241 * gst-libs/gst/interfaces/mixertrack.h:
32242 Do one struct member per line with a semicolon at the end, that way
32243 even gtk-doc might parse it without complaining.
32245 2005-12-28 18:06:50 +0000 Tim-Philipp Müller <tim@centricular.net>
32247 Add new libgstcdda with GstCddaBaseSrc class.
32248 Original commit message from CVS:
32250 * gst-libs/gst/Makefile.am:
32251 * gst-libs/gst/cdda/Makefile.am:
32252 * gst-libs/gst/cdda/base64.c:
32253 * gst-libs/gst/cdda/base64.h:
32254 * gst-libs/gst/cdda/gstcddabasesrc.c:
32255 (gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
32256 (gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
32257 (gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
32258 (gst_cdda_base_src_get_property),
32259 (gst_cdda_base_src_get_track_from_sector),
32260 (gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
32261 (gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
32262 (gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
32263 (gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
32264 (gst_cdda_base_src_uri_get_protocols),
32265 (gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
32266 (gst_cdda_base_src_uri_handler_init),
32267 (gst_cdda_base_src_setup_interfaces),
32268 (gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
32269 (cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
32270 (lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
32271 (gst_cdda_base_src_add_tags),
32272 (gst_cdda_base_src_add_index_associations),
32273 (gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
32274 (gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
32275 (gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
32276 (gst_cdda_base_src_create):
32277 * gst-libs/gst/cdda/gstcddabasesrc.h:
32278 * gst-libs/gst/cdda/sha1.c:
32279 * gst-libs/gst/cdda/sha1.h:
32280 Add new libgstcdda with GstCddaBaseSrc class.
32282 2005-12-28 14:59:41 +0000 Tim-Philipp Müller <tim@centricular.net>
32284 ext/gnomevfs/gstgnomevfssink.h: Use GstBaseSinkClass as parent_class member for class struct, not
32285 Original commit message from CVS:
32286 * ext/gnomevfs/gstgnomevfssink.h:
32287 Use GstBaseSinkClass as parent_class member for class struct, not
32290 2005-12-27 22:29:43 +0000 Tim-Philipp Müller <tim@centricular.net>
32292 gst/videotestsrc/gstvideotestsrc.c: Add start method to reset running time and number of frames sent
32293 Original commit message from CVS:
32294 * gst/videotestsrc/gstvideotestsrc.c:
32295 (gst_video_test_src_class_init), (gst_video_test_src_start):
32296 Add start method to reset running time and number of frames sent
32297 when starting up (fixes #324696; patch by: Michal Benes).
32299 2005-12-27 21:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
32301 docs/plugins/: Add docs stuff for gnomevfssrc and gnomevfssink.
32302 Original commit message from CVS:
32303 * docs/plugins/Makefile.am:
32304 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
32305 * docs/plugins/gst-plugins-base-plugins-sections.txt:
32306 * docs/plugins/gst-plugins-base-plugins.args:
32307 * docs/plugins/gst-plugins-base-plugins.hierarchy:
32308 * docs/plugins/gst-plugins-base-plugins.signals:
32309 Add docs stuff for gnomevfssrc and gnomevfssink.
32310 * ext/gnomevfs/gstgnomevfssrc.c:
32311 Fix example pipeline in gtk-doc blurb.
32313 2005-12-27 21:42:23 +0000 Tim-Philipp Müller <tim@centricular.net>
32315 ext/gnomevfs/: Port gnomevfssink; add gtk-doc blurb.
32316 Original commit message from CVS:
32317 * ext/gnomevfs/Makefile.am:
32318 * ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
32319 (gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
32320 (gst_gnome_vfs_handle_get_type), (plugin_init):
32321 * ext/gnomevfs/gstgnomevfs.h:
32322 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
32323 (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
32324 (gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
32325 (gst_gnome_vfs_sink_set_property),
32326 (gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
32327 (gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
32328 (gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
32329 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
32330 (gst_gnome_vfs_sink_uri_get_type),
32331 (gst_gnome_vfs_sink_uri_get_protocols),
32332 (gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
32333 (gst_gnome_vfs_sink_uri_handler_init):
32334 * ext/gnomevfs/gstgnomevfssink.h:
32335 Port gnomevfssink; add gtk-doc blurb.
32336 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
32337 (gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
32338 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
32339 (gst_gnome_vfs_src_uri_get_type),
32340 (gst_gnome_vfs_src_uri_get_protocols),
32341 (gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
32342 (gst_gnome_vfs_src_uri_handler_init),
32343 (gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
32344 (gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
32345 (gst_gnome_vfs_src_send_additional_headers_callback),
32346 (gst_gnome_vfs_src_received_headers_callback),
32347 (gst_gnome_vfs_src_push_callbacks),
32348 (gst_gnome_vfs_src_pop_callbacks),
32349 (gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
32350 (gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
32351 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
32352 * ext/gnomevfs/gstgnomevfssrc.h:
32353 s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
32354 file; add gtk-doc blurb with example pipelines.
32356 2005-12-23 18:16:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32360 Original commit message from CVS:
32363 === release 0.10.1 ===
32365 2005-12-23 18:08:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32371 * docs/libs/tmpl/gstcolorbalance.sgml:
32372 * docs/plugins/gst-plugins-base-plugins.args:
32373 * docs/plugins/gst-plugins-base-plugins.signals:
32374 * docs/plugins/inspect/plugin-adder.xml:
32375 * docs/plugins/inspect/plugin-alsa.xml:
32376 * docs/plugins/inspect/plugin-audioconvert.xml:
32377 * docs/plugins/inspect/plugin-audiorate.xml:
32378 * docs/plugins/inspect/plugin-audioresample.xml:
32379 * docs/plugins/inspect/plugin-audiotestsrc.xml:
32380 * docs/plugins/inspect/plugin-decodebin.xml:
32381 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
32382 * docs/plugins/inspect/plugin-gnomevfs.xml:
32383 * docs/plugins/inspect/plugin-libvisual.xml:
32384 * docs/plugins/inspect/plugin-ogg.xml:
32385 * docs/plugins/inspect/plugin-pango.xml:
32386 * docs/plugins/inspect/plugin-playbin.xml:
32387 * docs/plugins/inspect/plugin-subparse.xml:
32388 * docs/plugins/inspect/plugin-tcp.xml:
32389 * docs/plugins/inspect/plugin-theora.xml:
32390 * docs/plugins/inspect/plugin-typefindfunctions.xml:
32391 * docs/plugins/inspect/plugin-video4linux.xml:
32392 * docs/plugins/inspect/plugin-videorate.xml:
32393 * docs/plugins/inspect/plugin-videoscale.xml:
32394 * docs/plugins/inspect/plugin-videotestsrc.xml:
32395 * docs/plugins/inspect/plugin-volume.xml:
32396 * docs/plugins/inspect/plugin-vorbis.xml:
32397 * docs/plugins/inspect/plugin-ximagesink.xml:
32398 * docs/plugins/inspect/plugin-xvimagesink.xml:
32400 Original commit message from CVS:
32403 2005-12-21 20:59:52 +0000 Edgard Lima <edgard.lima@indt.org.br>
32406 * gst/typefind/gsttypefindfunctions.c:
32407 iLBC30 and iLBC20 added to typefind.
32408 Original commit message from CVS:
32409 iLBC30 and iLBC20 added to typefind.
32411 2005-12-20 15:57:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32415 * docs/libs/tmpl/gstcolorbalance.sgml:
32431 Original commit message from CVS:
32434 2005-12-20 12:24:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32436 * gst-libs/gst/audio/gstbaseaudiosink.c:
32437 * gst-libs/gst/audio/gstbaseaudiosrc.c:
32438 stop making fun of older compilers
32439 Original commit message from CVS:
32440 stop making fun of older compilers
32442 2005-12-20 12:00:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32444 gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
32445 Original commit message from CVS:
32446 * gst-libs/gst/audio/gstbaseaudiosink.c:
32447 (gst_base_audio_sink_class_init):
32448 * gst-libs/gst/audio/gstbaseaudiosrc.c:
32449 (gst_base_audio_src_class_init):
32450 update strings, values are in microseconds
32451 change the default sink buffer time to something that is smaller
32452 (to help software volume mixing have a slightly lower delay) but
32453 still be acceptable on Wim's laptop
32455 2005-12-20 10:13:05 +0000 Edward Hervey <bilboed@bilboed.com>
32457 gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
32458 Original commit message from CVS:
32459 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
32460 Made a quack, forgot to add DUCK to the riff video template.
32462 2005-12-19 15:00:38 +0000 Edward Hervey <bilboed@bilboed.com>
32464 ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
32465 Original commit message from CVS:
32466 * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
32467 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
32468 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
32469 (gst_ogm_parse_chain):
32470 Make sure pads are initialized correctly.
32471 * gst-libs/gst/riff/riff-ids.h:
32472 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
32473 (gst_riff_create_video_template_caps):
32474 Add a whole bunch of FOURCC <=> MimeType.
32475 Extend the riff video pad template to support the newly added fourcc.
32477 2005-12-18 15:04:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32479 ext/ogg/gstoggdemux.c: Extra debug output when activating/deactivating chains.
32480 Original commit message from CVS:
32481 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
32482 (gst_ogg_demux_activate_chain):
32483 Extra debug output when activating/deactivating chains.
32484 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
32485 (is_demuxer_element), (try_to_link_1), (remove_element_chain),
32487 Remove a queue from our list when it becomes unlinked.
32488 Don't add queues to elements in class 'Demux' if they
32489 can only produce one pad
32491 2005-12-18 10:54:33 +0000 Julien Moutte <julien@moutte.net>
32493 gst-libs/gst/video/gstvideosink.c: Add a debug category.
32494 Original commit message from CVS:
32495 2005-12-18 Julien MOUTTE <julien@moutte.net>
32496 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
32497 (gst_video_sink_get_type): Add a debug category.
32499 2005-12-18 00:56:07 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
32501 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
32502 Original commit message from CVS:
32503 2005-12-17 Philippe Khalaf <burger@speedy.org>
32504 * gst-libs/gst/rtp/gstbasertpdepayload.c:
32505 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
32506 Handle downstream newsegment by sending our own newsegment before the
32507 next buffer to be released. (#323900)
32509 2005-12-18 00:41:10 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
32511 gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
32512 Original commit message from CVS:
32513 2005-12-17 Philippe Khalaf <burger@speedy.org>
32514 * gst-libs/gst/rtp/gstbasertpdepayload.c:
32515 (gst_base_rtp_depayload_set_gst_timestamp):
32516 add queue delay to new segment as well (as opposed to just the first
32517 buffer). (bug #322347)
32519 2005-12-16 22:00:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32521 ext/libvisual/visual.c: change some char* into char[]
32522 Original commit message from CVS:
32523 * ext/libvisual/visual.c: (make_valid_name):
32524 change some char* into char[]
32525 * gst/audiotestsrc/gstaudiotestsrc.c:
32526 (gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
32527 (gst_audio_test_src_create):
32528 * gst/audiotestsrc/gstaudiotestsrc.h:
32529 prepare to handle EOS and SEGMENT_DONE
32531 2005-12-16 12:32:37 +0000 Tim-Philipp Müller <tim@centricular.net>
32533 tests/check/generic/states.c: Blacklist cdparanoia element in state test.
32534 Original commit message from CVS:
32535 * tests/check/generic/states.c: (GST_START_TEST):
32536 Blacklist cdparanoia element in state test.
32538 2005-12-16 11:25:51 +0000 Benjamin Pineau <ben.pineau@gmail.com>
32540 gst/tcp/: Add <string.h> includes for memset and FD_ZERO (fixes #323878;
32541 Original commit message from CVS:
32542 * gst/tcp/gsttcp.c:
32543 * gst/tcp/gsttcpclientsink.c:
32544 * gst/tcp/gsttcpserversink.c:
32545 * gst/tcp/gsttcpserversrc.c:
32546 Add <string.h> includes for memset and FD_ZERO (fixes #323878;
32547 patch by: Benjamin Pineau).
32549 2005-12-15 14:43:38 +0000 Michael Smith <msmith@xiph.org>
32551 gst/videorate/gstvideorate.c: Fix timestamping for videorate when the first buffer it sees has a non-zero timestamp. ...
32552 Original commit message from CVS:
32553 * gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
32554 (gst_video_rate_chain):
32555 Fix timestamping for videorate when the first buffer it sees has a
32556 non-zero timestamp. Fix some misleading debug output.
32558 2005-12-15 10:30:14 +0000 Michael Smith <msmith@xiph.org>
32560 gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
32561 Original commit message from CVS:
32562 * gst/audioresample/gstaudioresample.c:
32563 Don't leak all input buffers to audioresample.
32565 2005-12-15 10:15:10 +0000 Tim-Philipp Müller <tim@centricular.net>
32567 ext/pango/gsttextoverlay.c: Don't operate on empty text buffers. Strip newlines and tabs only from the end of the tex...
32568 Original commit message from CVS:
32569 * ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
32570 Don't operate on empty text buffers. Strip newlines and
32571 tabs only from the end of the text, but leave them intact
32572 in the middle. Fix typo in gtk-doc description.
32574 2005-12-15 09:48:19 +0000 Tim-Philipp Müller <tim@centricular.net>
32576 gst/playback/: Make sure the video frame buffer we return to apps via the "frame" property always has caps set on it....
32577 Original commit message from CVS:
32578 * gst/playback/gstplaybasebin.c:
32579 * gst/playback/gstplaybin.c: (handoff):
32580 Make sure the video frame buffer we return to apps via the
32581 "frame" property always has caps set on it. Modify
32582 _gst_gvalue_set_object() macro to handle NULL objects
32585 2005-12-14 20:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
32587 gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
32588 Original commit message from CVS:
32589 * gst/audiotestsrc/gstaudiotestsrc.c:
32590 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
32591 (gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
32592 (gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
32593 (gst_audio_test_src_create):
32594 * gst/audiotestsrc/gstaudiotestsrc.h:
32595 Adjust to some recent api changes and add wtays new cool seeking
32598 2005-12-14 17:58:48 +0000 Tim-Philipp Müller <tim@centricular.net>
32600 ext/alsa/: Helper functions to add device probing via the GstPropertyProbe interface to a class.
32601 Original commit message from CVS:
32602 * ext/alsa/Makefile.am:
32603 * ext/alsa/gstalsadeviceprobe.c:
32604 * ext/alsa/gstalsadeviceprobe.h:
32605 Helper functions to add device probing via the GstPropertyProbe
32606 interface to a class.
32607 * ext/alsa/gstalsamixer.h:
32608 Comment out GST_ALSA_MIXER, it returns a struct that's not
32610 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
32611 Add some debug info.
32612 * ext/alsa/gstalsamixerelement.c:
32613 (gst_alsa_mixer_element_interface_supported),
32614 (gst_implements_interface_init),
32615 (gst_alsa_mixer_element_init_interfaces),
32616 (gst_alsa_mixer_element_class_init),
32617 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
32618 (gst_alsa_mixer_element_set_property),
32619 (gst_alsa_mixer_element_get_property),
32620 (gst_alsa_mixer_element_change_state):
32621 * ext/alsa/gstalsamixerelement.h:
32622 Add 'device' and 'device-name' properties. Add GstPropertyProbe
32623 for device handling (gnome-volume-control will need that).
32625 2005-12-12 20:31:24 +0000 Christian Schaller <uraeus@gnome.org>
32629 * gst-plugins-base.spec.in:
32630 updates to activate cdparanoia plugin
32631 Original commit message from CVS:
32632 updates to activate cdparanoia plugin
32634 2005-12-12 19:13:09 +0000 Michael Smith <msmith@xiph.org>
32636 ext/ogg/gstoggdemux.c: Use the correct function to free list of typefind factories.
32637 Original commit message from CVS:
32638 * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
32639 Use the correct function to free list of typefind factories.
32641 2005-12-12 15:09:55 +0000 Wim Taymans <wim.taymans@gmail.com>
32643 gst/videotestsrc/gstvideotestsrc.*: Implement seeking in videotestsrc.
32644 Original commit message from CVS:
32645 * gst/videotestsrc/gstvideotestsrc.c:
32646 (gst_video_test_src_class_init), (gst_video_test_src_init),
32647 (gst_video_test_src_parse_caps), (gst_video_test_src_query),
32648 (gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
32649 (gst_video_test_src_create):
32650 * gst/videotestsrc/gstvideotestsrc.h:
32651 Implement seeking in videotestsrc.
32654 2005-12-12 15:06:46 +0000 Wim Taymans <wim.taymans@gmail.com>
32656 ext/cdparanoia/: Partially ported cdparanoia now that basesrc can support a plugin like this..
32657 Original commit message from CVS:
32658 * ext/cdparanoia/Makefile.am:
32659 * ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
32660 (gst_paranoia_endian_get_type), (_do_init),
32661 (cdparanoia_class_init), (cdparanoia_init),
32662 (cdparanoia_set_property), (cdparanoia_get_property),
32663 (cdparanoia_do_seek), (cdparanoia_is_seekable),
32664 (cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
32665 (cdparanoia_convert), (cdparanoia_get_query_types),
32666 (cdparanoia_query), (cdparanoia_set_index),
32667 (cdparanoia_uri_set_uri):
32668 * ext/cdparanoia/gstcdparanoia.h:
32669 Partially ported cdparanoia now that basesrc can support a
32672 2005-12-12 13:03:50 +0000 Wim Taymans <wim.taymans@gmail.com>
32674 tests/examples/seek/scrubby.c: Set higher priority for bus events so they don't get reordered with gtk gui events.
32675 Original commit message from CVS:
32676 * tests/examples/seek/scrubby.c: (main):
32677 Set higher priority for bus events so they don't get reordered with
32679 * tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
32680 (flush_toggle_cb), (main):
32681 Added checkbox do disable flushing seeks.
32682 Disable scrubbing when doing non flushing seeks.
32684 2005-12-12 09:52:37 +0000 Tim-Philipp Müller <tim@centricular.net>
32686 gst/subparse/gstsubparse.c: Implement some sort of event handling that doesn't rely on g_return_if_fail; make sure we...
32687 Original commit message from CVS:
32688 * gst/subparse/gstsubparse.c: (gst_sub_parse_init),
32689 (gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
32690 (parser_state_init), (handle_buffer), (gst_sub_parse_chain),
32691 (gst_sub_parse_sink_event), (gst_sub_parse_change_state):
32692 Implement some sort of event handling that doesn't rely on
32693 g_return_if_fail; make sure we always push the last chunk of an
32694 .srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
32695 state change function; remove some old cruft. Seeking is still
32696 rather unlikely to work though.
32697 * tools/.cvsignore:
32700 2005-12-11 22:26:08 +0000 Julien Moutte <julien@moutte.net>
32702 sys/xvimage/xvimagesink.c: Fixed a leak of the current image reference when cleaning up.
32703 Original commit message from CVS:
32704 2005-12-11 Julien MOUTTE <julien@moutte.net>
32705 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
32706 Fixed a leak of the current image reference when cleaning up.
32707 Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
32709 2005-12-09 10:23:42 +0000 Michael Smith <msmith@xiph.org>
32711 tools/: Remove gst-launch-ext. It doesn't work, and is no longer particularly useful.
32712 Original commit message from CVS:
32713 * tools/Makefile.am:
32714 * tools/gst-launch-ext-m.m:
32715 Remove gst-launch-ext. It doesn't work, and is no longer
32716 particularly useful.
32718 2005-12-08 18:53:57 +0000 Luca Ognibene <luogni@tin.it>
32720 ext/ogg/gstogmparse.c: don't pass random values to ogmparse convert function.
32721 Original commit message from CVS:
32722 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
32723 don't pass random values to ogmparse convert function.
32724 Make seeking possible in the exile1.ogm file.
32726 2005-12-07 18:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
32728 gst/playback/: Work around refcount problem with g_value_set_object() that occur if the core has been compiled agains...
32729 Original commit message from CVS:
32730 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
32731 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
32732 Work around refcount problem with g_value_set_object() that occur
32733 if the core has been compiled against GLib-2.6 (g_value_set_object()
32734 will only g_object_ref() the element, but the caller will
32735 gst_object_unref() it and bad things will happen due to the way
32736 GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
32737 totem for people on FC4 using Thomas's 0.10 RPMs.
32739 2005-12-07 11:34:37 +0000 Edward Hervey <bilboed@bilboed.com>
32741 Time to welcome ogm to 0.10 :)
32742 Original commit message from CVS:
32743 Time to welcome ogm to 0.10 :)
32744 * ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
32745 (gst_ogg_pad_typefind):
32746 Oggdemux can now properly typefind elements with dynamic pads.
32747 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
32748 Properly set caps on src pad, and set caps on outgoing buffers.
32750 2005-12-06 19:42:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32753 * ext/alsa/gstalsamixer.h:
32754 * ext/alsa/gstalsamixerelement.h:
32755 * ext/alsa/gstalsamixeroptions.h:
32756 * ext/alsa/gstalsamixertrack.h:
32757 * ext/alsa/gstalsasink.c:
32758 * ext/alsa/gstalsasink.h:
32759 * ext/alsa/gstalsasrc.c:
32760 * ext/alsa/gstalsasrc.h:
32761 * ext/cdparanoia/gstcdparanoia.h:
32762 * ext/gnomevfs/gstgnomevfsuri.h:
32763 * ext/ogg/gstoggdemux.c:
32764 * ext/ogg/gstoggmux.c:
32765 * ext/pango/gsttextoverlay.h:
32766 * ext/theora/theoradec.c:
32767 * ext/theora/theoraenc.c:
32768 * ext/vorbis/vorbisdec.h:
32769 * ext/vorbis/vorbisenc.c:
32770 * ext/vorbis/vorbisenc.h:
32771 * ext/vorbis/vorbisparse.h:
32772 * gst-libs/gst/audio/gstaudioclock.h:
32773 * gst-libs/gst/audio/gstaudiosink.c:
32774 * gst-libs/gst/audio/gstaudiosink.h:
32775 * gst-libs/gst/audio/gstaudiosrc.c:
32776 * gst-libs/gst/audio/gstaudiosrc.h:
32777 * gst-libs/gst/audio/gstbaseaudiosink.c:
32778 * gst-libs/gst/audio/gstbaseaudiosink.h:
32779 * gst-libs/gst/audio/gstbaseaudiosrc.c:
32780 * gst-libs/gst/audio/gstbaseaudiosrc.h:
32781 * gst-libs/gst/audio/gstringbuffer.h:
32782 * gst-libs/gst/audio/multichannel.h:
32783 * gst-libs/gst/floatcast/floatcast.h:
32784 * gst-libs/gst/interfaces/colorbalance.c:
32785 * gst-libs/gst/interfaces/colorbalance.h:
32786 * gst-libs/gst/interfaces/colorbalancechannel.h:
32787 * gst-libs/gst/interfaces/mixer.h:
32788 * gst-libs/gst/interfaces/mixeroptions.h:
32789 * gst-libs/gst/interfaces/mixertrack.h:
32790 * gst-libs/gst/interfaces/navigation.h:
32791 * gst-libs/gst/interfaces/propertyprobe.h:
32792 * gst-libs/gst/interfaces/tuner.h:
32793 * gst-libs/gst/interfaces/tunerchannel.h:
32794 * gst-libs/gst/interfaces/tunernorm.h:
32795 * gst-libs/gst/interfaces/xoverlay.h:
32796 * gst-libs/gst/netbuffer/gstnetbuffer.h:
32797 * gst-libs/gst/riff/riff-ids.h:
32798 * gst-libs/gst/riff/riff-media.h:
32799 * gst-libs/gst/riff/riff-read.h:
32800 * gst-libs/gst/rtp/gstbasertpdepayload.h:
32801 * gst-libs/gst/rtp/gstbasertppayload.c:
32802 * gst-libs/gst/rtp/gstbasertppayload.h:
32803 * gst-libs/gst/rtp/gstrtpbuffer.c:
32804 * gst-libs/gst/rtp/gstrtpbuffer.h:
32805 * gst-libs/gst/tag/gsttageditingprivate.h:
32806 * gst-libs/gst/tag/gstvorbistag.c:
32807 * gst-libs/gst/tag/tag.h:
32808 * gst-libs/gst/video/video.h:
32809 * gst/adder/gstadder.c:
32810 * gst/adder/gstadder.h:
32811 * gst/audioconvert/audioconvert.c:
32812 * gst/audioconvert/audioconvert.h:
32813 * gst/audioconvert/gstaudioconvert.c:
32814 * gst/audioconvert/gstchannelmix.c:
32815 * gst/audioconvert/gstchannelmix.h:
32816 * gst/audiorate/gstaudiorate.c:
32817 * gst/audioresample/buffer.h:
32818 * gst/audioresample/functable.h:
32819 * gst/audioresample/gstaudioresample.c:
32820 * gst/audioresample/resample.h:
32821 * gst/ffmpegcolorspace/avcodec.h:
32822 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
32823 * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
32824 * gst/ffmpegcolorspace/imgconvert.c:
32825 * gst/ffmpegcolorspace/imgconvert_template.h:
32826 * gst/playback/gstdecodebin.c:
32827 * gst/playback/gstplaybasebin.h:
32828 * gst/playback/gstplaybin.c:
32829 * gst/playback/gststreaminfo.h:
32830 * gst/tcp/gstfdset.c:
32831 * gst/tcp/gstfdset.h:
32832 * gst/tcp/gstmultifdsink.c:
32833 * gst/tcp/gstmultifdsink.h:
32834 * gst/tcp/gsttcp.h:
32835 * gst/tcp/gsttcpclientsrc.c:
32836 * gst/tcp/gsttcpclientsrc.h:
32837 * gst/tcp/gsttcpplugin.h:
32838 * gst/tcp/gsttcpserversink.c:
32839 * gst/tcp/gsttcpserversrc.c:
32840 * gst/typefind/gsttypefindfunctions.c:
32841 * gst/videorate/gstvideorate.c:
32842 * gst/videotestsrc/gstvideotestsrc.h:
32843 * gst/videotestsrc/videotestsrc.h:
32844 * sys/v4l/gstv4lcolorbalance.h:
32845 * sys/v4l/gstv4ltuner.h:
32846 * sys/v4l/gstv4lxoverlay.h:
32847 * sys/v4l/v4l_calls.h:
32848 * sys/v4l/videodev_mjpeg.h:
32849 * tests/check/elements/audioconvert.c:
32850 * tests/check/elements/audioresample.c:
32851 * tests/check/elements/audiotestsrc.c:
32852 * tests/check/elements/videotestsrc.c:
32853 * tests/check/elements/volume.c:
32854 * tests/examples/seek/scrubby.c:
32855 * tests/examples/seek/seek.c:
32857 Original commit message from CVS:
32860 2005-12-06 19:28:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32862 * docs/libs/tmpl/gstaudio.sgml:
32863 * docs/libs/tmpl/gstcolorbalance.sgml:
32864 * docs/libs/tmpl/gstgconf.sgml:
32865 * docs/libs/tmpl/gstmixer.sgml:
32866 * docs/libs/tmpl/gstringbuffer.sgml:
32867 * docs/libs/tmpl/gsttuner.sgml:
32868 * docs/libs/tmpl/gstxoverlay.sgml:
32869 put back stability level
32870 Original commit message from CVS:
32871 put back stability level
32873 2005-12-05 18:11:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32877 Original commit message from CVS:
32880 === release 0.10.0 ===
32882 2005-12-05 18:02:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32888 * docs/libs/tmpl/gstcolorbalance.sgml:
32889 * docs/plugins/inspect/plugin-adder.xml:
32890 * docs/plugins/inspect/plugin-alsa.xml:
32891 * docs/plugins/inspect/plugin-audioconvert.xml:
32892 * docs/plugins/inspect/plugin-audiorate.xml:
32893 * docs/plugins/inspect/plugin-audioresample.xml:
32894 * docs/plugins/inspect/plugin-audiotestsrc.xml:
32895 * docs/plugins/inspect/plugin-decodebin.xml:
32896 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
32897 * docs/plugins/inspect/plugin-gnomevfs.xml:
32898 * docs/plugins/inspect/plugin-libvisual.xml:
32899 * docs/plugins/inspect/plugin-ogg.xml:
32900 * docs/plugins/inspect/plugin-pango.xml:
32901 * docs/plugins/inspect/plugin-playbin.xml:
32902 * docs/plugins/inspect/plugin-subparse.xml:
32903 * docs/plugins/inspect/plugin-tcp.xml:
32904 * docs/plugins/inspect/plugin-theora.xml:
32905 * docs/plugins/inspect/plugin-typefindfunctions.xml:
32906 * docs/plugins/inspect/plugin-video4linux.xml:
32907 * docs/plugins/inspect/plugin-videorate.xml:
32908 * docs/plugins/inspect/plugin-videoscale.xml:
32909 * docs/plugins/inspect/plugin-videotestsrc.xml:
32910 * docs/plugins/inspect/plugin-volume.xml:
32911 * docs/plugins/inspect/plugin-vorbis.xml:
32912 * docs/plugins/inspect/plugin-ximagesink.xml:
32913 * docs/plugins/inspect/plugin-xvimagesink.xml:
32915 Original commit message from CVS: