ChangeLog 3.63 MB
Newer Older
Sebastian Dröge's avatar
Sebastian Dröge committed
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
=== release 1.7.2 ===

2016-02-19  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 1.7.2

2016-02-19 10:31:05 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  po: Update translations

2016-02-18 14:31:28 +0000  Julien Isorce <j.isorce@samsung.com>

	* pkgconfig/gstreamer-allocators-uninstalled.pc.in:
	* pkgconfig/gstreamer-app-uninstalled.pc.in:
	* pkgconfig/gstreamer-audio-uninstalled.pc.in:
	* pkgconfig/gstreamer-fft-uninstalled.pc.in:
	* pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
	* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
	* pkgconfig/gstreamer-riff-uninstalled.pc.in:
	* pkgconfig/gstreamer-rtp-uninstalled.pc.in:
	* pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
	* pkgconfig/gstreamer-sdp-uninstalled.pc.in:
	* pkgconfig/gstreamer-tag-uninstalled.pc.in:
	* pkgconfig/gstreamer-video-uninstalled.pc.in:
	  uninstalled.pc: add support for non libtool build systems
	  Currently the .la path is provided which requires to use libtool as
	  mentioned in the GStreamer manual section-helloworld-compilerun.html.
	  It is fine as long as the application is built using libtool.
	  So currently it is not possible to compile a GStreamer application
	  within gst-uninstalled with CMake or other build system different
	  than autotools.
	  This patch allows to do the following in gst-uninstalled env:
	  gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
	  gstreamer-video-1.0)
	  Previously it required to prepend libtool --mode=link
	  https://bugzilla.gnome.org/show_bug.cgi?id=720778

2016-01-22 18:26:01 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefind: strengthen check for valid H.263 picture layer
	  Avoids some false positives leading to miss identification:
	  * Prevent picture start code emulation for the first 2 bytes read
	  * Add check for valid "picture coding type" and "PB-frames mode" combination
	  Additionally, change name on confusingly named TR var to what
	  it is, the layer's PTYPE.
	  https://bugzilla.gnome.org/show_bug.cgi?id=693263

2015-11-23 15:06:02 +0900  Vineeth T M <vineeth.tm@samsung.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: return incomplete topology if decode chains' cap could not be obtained
	  When getting caps of the decode chain, in get_topology, the caps are being
	  checked if fixed or not. But get_topology will be called when the decode is
	  chain is being exposed and hence it will always be fixed. Hence removing the
	  check for fixed caps. Removing gst_pad_get_current_caps for the chain->pad, as
	  get_pad_caps will again call the same api.
	  And get_topology can return NULL value if currently shutting down the
	  pipeline, which on being passed to create message will result in assertion
	  error. Check if topology is valid before using it
	  https://bugzilla.gnome.org/show_bug.cgi?id=755918

2016-02-05 10:10:40 +0100  Havard Graff <havard.graff@gmail.com>

	* gst-libs/gst/Makefile.am:
	  rtp: build audio library before rtp
	  Because audio-enumtypes.h needs to be available for
	  gstrtpbaseaudiopayload.c
	  https://bugzilla.gnome.org/show_bug.cgi?id=761949

2016-02-15 21:28:33 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Fix documentation of the autoplug-query signal

2016-01-26 13:54:46 +0100  Stian Selnes <stian@pexip.com>

	* gst-libs/gst/video/gstvideoencoder.c:
	* tests/check/libs/videoencoder.c:
	  videoencoder: Fix leak when pre_push does not return OK
	  https://bugzilla.gnome.org/show_bug.cgi?id=761951

2016-02-11 19:47:04 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioresample/resample.c:
	  resample: avoid overflows
	  Avoid overflow in rate calculation. This can cause the resampler to
	  start on the wrong phase after a rate change.
	  Avoid overflow in cubic fraction calculation. This can cause noise when
	  dealing with higher samplerates.

2016-02-11 18:01:40 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioresample/resample_sse.h:
	  resample: fix double interpolation sse code
	  We were only reading 2 filter taps and we need to read 4 to do cubic
	  interpolation.

2016-02-10 12:48:15 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: make a copy if we can't write in unpack
	  If we don't have writable memory, make sure to make a copy of the input
	  samples into a temporary (writable) buffer, even if we are dealing with
	  a native intermediate format that we don't need to call the unpack
	  function for.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=761655

2016-02-05 19:15:16 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/check/Makefile.am:
	  tests: extend the AM_TESTS_ENVIRONMENT from check.mak
	  To get the CK_DEFAULT_TIMEOUT defined for all tests.
	  Also replaces a 120 timeout that was set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761472

2016-02-05 18:03:07 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* autogen.sh:
	* common:
	  Automatic update of common submodule
	  From 86e4663 to b64f03f

2016-01-21 09:43:35 +0100  Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>

	* ext/pango/gstbasetextoverlay.c:
	* ext/pango/gstbasetextoverlay.h:
	  textoverlay: Expose rendering dimensions as properties.
	  In order to detect graphical user input on the
	  textoverlay, the resulting rendering properties
	  need to be exposed to applications.
	  Fixes delayx property declaration.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761251

2016-01-20 15:37:44 +0100  Lubosz Sarnecki <lubosz.sarnecki@collabora.co.uk>

	* ext/pango/gstbasetextoverlay.c:
	  textoverlay: Do not limit positioning to video area.
	  The current position property is limited to X,Y positions
	  in the range of [0, 1]. This patch allows full control
	  over the overlay position, including partially outside
	  of the video area.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761251

2016-01-28 13:29:39 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiorate/gstaudiorate.c:
	  audiorate: Use gst_audio_format_fill_silence() instead of memset with 0 for generating silence
	  For unsigned formats, silence is not all bits 0.

2016-01-28 13:21:33 +0100  HoonHee Lee <hoonhee.lee@lge.com>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/video/gstvideodecoder.c:
	  audio/videodecoder: Minor cleanup of last commit
	  https://bugzilla.gnome.org/show_bug.cgi?id=761218

2016-01-28 18:06:44 +0900  HoonHee Lee <hoonhee.lee@lge.com>

	* gst-libs/gst/audio/gstaudiodecoder.c:
	* gst-libs/gst/video/gstvideodecoder.c:
	  audio/videodecoder: use gst_pad_peer_query_caps to make output caps
	  gst_pad_get_allowed_caps() will return NULL if the srcpad has no peer.
	  In that case, use gst_pad_peer_query_caps() with template caps as filter
	  to have negotiated output caps properly before forwarding GAP event.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761218

2016-01-26 19:23:04 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/encoding/gstencodebin.c:
	  encodebin: Allow streamheader update when profile.allow_dynamic_output == FALSE
	  Some encoders can update the stream header through time (for example
	  vp8 might do that) but it does not strictly changes the output format.

2016-01-26 14:09:42 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* gst-libs/gst/video/video-format.h:
	  video-format: fix GstVideoFormatInfo documentation warnings
	  Add missing ':' to tile_ws and tile_hs fields documentation to avoid
	  bad render of these two fields, mark reserved bytes as private to hide
	  field and avoid gtkdoc warning and add parameters description to
	  documented macro to avoid gtkdoc warnings.
	  https://bugzilla.gnome.org/show_bug.cgi?id=761132

2016-01-26 16:56:57 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* win32/common/libgstaudio.def:
	  audio-converter: add reset function

2016-01-26 16:36:41 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: handle NULL input
	  Allow NULL as input to mean silence samples.

2016-01-26 17:16:52 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: improve _update_config
	  Allow NULL config to keep the existing parameters.
	  Fix the docs.

2016-01-26 17:14:20 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	  audio-converter: audio-converter: make some optimized functions
	  Make optimized functions for generic and passthrough conversion.

2016-01-26 16:34:35 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-quantize.c:
	* gst-libs/gst/audio/audio-quantize.h:
	  audio-quantize: add _reset function
	  Add a reset function that clears any history.

2016-01-25 17:40:23 +0000  Tim-Philipp Müller <tim@centricular.com>

	* configure.ac:
	* m4/Makefile.am:
	* m4/freetype2.m4:
	* tests/examples/Makefile.am:
	  build: remove nonsensical check for freetype
	  The examples need Gtk+, nothing uses freetype directly.

2016-01-25 16:22:17 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/elements/libvisual.c:
	  tests: libvisual: make run faster
	  Reduce resolution, which shouldn't make any difference
	  to what's tested here. Makes test finish in less than
	  half the time it took before (8s vs. 21s).

2016-01-25 18:30:30 +0530  Arun Raghavan <git@arunraghavan.net>

	* ext/alsa/gstalsasink.c:
	  alsa: Trivial doc update
	  alsasink now does more than just raw audio.

2016-01-21 18:30:40 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  decodebin: Correctly expose pads from elements that have directly exposable pads
	  analyze_new_pad() can return a new decode chain, which might have a new
	  GstDecodePad in the end. We should use those two for expose_pad() and not the
	  original ones that were passed to analyze_new_pad().
	  This fails when having a demuxer element that has raw pads immediately or
	  if a decoder with raw caps is after an adaptive demuxer.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760949

2016-01-21 16:08:46 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: ensure correct alignment of samples
	  Make sure that the data we allocate for our temporary buffers is
	  properly aligned.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=760938

2016-01-21 10:45:40 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/video/video-color.c:
	* gst-libs/gst/video/video-color.h:
	  video-color: add Adobe RGB primaries and transfer function

2016-01-20 10:19:34 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/video/video-info.c:
	  video-info: enfore RGB matrix for RGB formats
	  In gst_video_info_to_caps(), make sure we end up with an RGB matrix for
	  RGB formats and warn when the GstVideoInfo colorimetry is wrong.
	  In gst_video_info_from_caps(), fix the GstVideoInfo with an RGB matrix
	  for RGB formats and warn about inconsistent caps.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-20 10:02:20 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/video/video-converter.c:
	  video-converter: ignore matrix for RGB formats
	  For RGB formats, the matrix in the colorimetry (conversion from YUV to
	  RGB) is irrelevant and we should ignore it and assume the identity
	  transform for everything we do.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-19 23:26:57 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/video/gstvideoencoder.h:
	  videoencoder: Deprecate GST_VIDEO_ENCODER_FLOW_DROPPED
	  It was never actually supported or used
	  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-19 23:22:35 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/video/gstvideoencoder.c:
	  Revert "videoencoder: Release video frame when ->handle return ERROR or DROPPED"
	  This reverts commit 63517d0ed348784cce4ab4b295c2c0f1b78baa81.
	  It was wrong ref counting wise and we decided to deprecated DROPPED
	  return value
	  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-18 11:40:36 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* tests/check/elements/audioconvert.c:
	  tests:audioconvert: Fix integer overflow build error
	  value of 32768L << 16 and 1L << 31 is 2147483648
	  but it exceeds the positive range of int which is 2147483647
	  resulting in integer overflow error. Use G_GINT64_CONSTANT instead of L.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760769

2016-01-19 12:39:22 +0530  Arun Raghavan <git@arunraghavan.net>

	* gst-libs/gst/app/gstappsrc.c:
	  appsrc: Minor documentation cleanup

2016-01-14 23:14:27 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-play.c:
	  tools: gst-play: allow setting of flags in serialized foo+bar format
	  https://bugzilla.gnome.org/show_bug.cgi?id=751901

2015-07-02 17:58:00 +0200  Hugues Fruchet <hugues.fruchet@st.com>

	* tools/gst-play.c:
	  tools: gst-play: add command line options for verbose output and playbin flags
	  https://bugzilla.gnome.org/show_bug.cgi?id=751901

2016-01-18 15:51:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* win32/common/libgstapp.def:
	  win32: Update exports

2015-10-15 10:38:16 -0400  Evan Callaway <evan.callaway@ipconfigure.com>

	* gst-libs/gst/app/gstappsink.c:
	* gst-libs/gst/app/gstappsink.h:
	  Add WAIT_ON_EOS flag to gstappsink.
	  If set, an appsink that receives an EOS will wait until all of its buffers have been processed before continuing.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756187

2016-01-16 10:17:50 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudioencoder.c:
	  audioencoder: Add note to the documentation about various settings being reset before set_format()
	  It's quite unexpected behaviour that various subclass settings are just
	  reset before set_format(). Unfortunately changing this now has the risk
	  of breaking existing code but we should reconsider this for 2.0.

2016-01-09 04:35:23 +0100  Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>

	* gst/playback/gststreamsynchronizer.c:
	  streamsynchronizer: Ignore flushing streams [..]
	  [..] when resetting group start time. In GES, we are usually connected
	  to the streamsynchronizer on one audio and one video pad.
	  When seeking the timeline, both nlecompositions often output their flush_start
	  before any of them has output its flush_stop.
	  The current code, when receiving the first flush stop was using the
	  running time of the start of the second composition, which could
	  be pretty much anything, and means nothing at that point.
	  This patch is thread-safe, as STREAM_SYNCHRONIZER_LOCK is taken
	  both when setting flushing and when checking it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750013

2016-01-08 18:53:52 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaybin2.c:
	  playbin: Only append non-raw and sysmem pad template caps to the autoplug-query result
	  Otherwise a decoder supporting GL memory will think that all downstream can
	  support GL memory because of seeing its own template caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758212

2016-01-08 18:37:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaybin2.c:
	  Revert "playbin: only add the template caps when the result is empty"
	  This reverts commit 023af2d3b192f8ebf1bd4fe75a22a4adaedc1e05.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758212

2016-01-15 13:35:22 +0000  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/video/gstvideoencoder.c:
	  videoencoder: Release video frame when ->handle return ERROR or DROPPED
	  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-15 09:50:29 +0100  Edward Hervey <edward@centricular.com>

	* gst/playback/gstplaysink.c:
	  playsink: Properly mark pending blocked pads
	  When blocking input pads, we also need to properly set the appropriate
	  pending flag.
	  Without this, when switching stream types after initial configuration
	  (like going from Audio+Video to Audio+Video+Sub) playsink would never
	  wait for *all* input streams to be blocked (it would just wait for the
	  new input pad (text in this case) to be blocked).
	  Since the reconfiguration might introduce unlinking/relinking of elements,
	  we need to ensure that *ALL* input streams are blocked.
	  Failure to do so would result in having some input streams pushing data
	  to inactive elements (returning GST_FLOW_FLUSHING) or unlinked pads
	  (returning GST_FLOW_NOT_LINKED).
	  A later optimization could involve only blocking the input pads that
	  might be involved in reconfiguration. But better be safe than sorry for
	  now :)

2016-01-06 10:12:43 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* tools/gst-device-monitor.c:
	  gst-device-monitor: Use g_printerr instead of g_error
	  g_error is meant to be used for programmer errors (causes an abort),
	  not for expected runtime errors.

2016-01-13 16:32:25 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: replace gst_caps_can_intersect() with is_subset()
	  Subset check verifies also that all required fields are present
	  and is mostly commonly used when checking if an element accepts
	  a certain caps

2016-01-12 11:31:50 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/playback/gstplaybin2.c:
	  playbin: use subset check instead of intersect
	  Elements usually require that all fields on their caps are present
	  on the fixed caps they receive. Using intersection won't verify it,
	  resort to using is_subset() checks.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760477

2016-01-12 15:56:36 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channel-mixer.c:
	  audio-channel-mixer: round before truncating
	  Round the result before truncating for int channel mixing.

2016-01-12 15:27:16 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: Avoid conversion when possible
	  When the input and output formats are the same and in a possible
	  intermediate format, avoid unpack and pack.
	  Never do passthrough channel mixing.
	  Only do dithering and noise shaping in S32 format

2016-01-12 11:43:20 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-channel-mixer.c:
	  audio-channel-mixer: add more formats
	  Add support for float and int16 mixing
	  Remove in-place processing, this simplifies things as we won't be using it.
	  Don't do clipping for float audio formats

2016-01-12 11:37:17 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: improve processing loop
	  Process as many samples as we can from the input and return the number
	  of processed samples from the chain. This simplifies some code.
	  Fix the IN_WRITABLE handling, don't overwrite the flags.

2016-01-11 18:24:48 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/playback/gstsubtitleoverlay.c:
	  subtitleoverlay: replace accept-caps with caps query
	  Those accept caps are actually checking if downstream supports
	  some particular caps to check if it need to negotiate a different
	  format. Checking only the next element with accept-caps is not enough
	  to guarantee that it is supported.
	  Using a caps query makes it obtain the supported caps for downstream
	  as a whole instead of only the next element.

2016-01-08 21:27:16 +0200  Sebastian Dröge <sebastian@centricular.com>

	* win32/common/libgstaudio.def:
	  audio: Update exported symbols list

2016-01-08 15:05:38 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/videorate/gstvideorate.c:
	  videorate: replace accept-caps with a caps query
	  accept-caps is only a shallow check, it needs to know
	  whether downstream as a whole accepts the framerate

2016-01-08 16:08:47 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	  docs: fix up for GstAudioChannelMix rename as well

2016-01-08 17:34:50 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* gst/audioconvert/gstaudioconvert.c:
	  audio-converter: small API tweaks
	  Pass flags in _converter_new() so that we can configure ourselves
	  differently depending on some options.
	  SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'

2016-01-08 17:28:31 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	  audio-converter: prepare API for rate changes
	  Use the update function to update the sample rates along with the config
	  once we implement resampling.

2016-01-08 17:17:44 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* gst/audioconvert/gstaudioconvert.c:
	  audio-convert: simplify API
	  Simplify the API, we don't need the consumed and produced output
	  arguments. The caller needs to use the _get_in_frames/get_out_frames API
	  to check how much input is needed and how much output will be produced.

2016-01-08 17:50:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/gstaudioutilsprivate.h:
	* gst-libs/gst/video/gstvideoutilsprivate.h:
	  audio/video: Use G_GNUC_INTERNAL for internal functions

2016-01-08 16:22:25 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/Makefile.am:
	* gst-libs/gst/audio/audio-channel-mix.c:
	* gst-libs/gst/audio/audio-channel-mix.h:
	* gst-libs/gst/audio/audio-channel-mixer.c:
	* gst-libs/gst/audio/audio-channel-mixer.h:
	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio.h:
	* win32/common/libgstaudio.def:
	  audio: GstAudioChannelMix -> GstAudioChannelMixer
	  Rename the GstAudioChannelMix object to GstAudioChannelMixer because it
	  looks better and to avoid a conflict with a library in -bad.

2016-01-07 15:24:25 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaybin2.c:
	  playbin: Use the caps query instead of accept-caps to detect if a sink accepts caps
	  accept-caps is only for one element, caps query is recursive. Fixes playback
	  with totem and other situations.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760234

2016-01-06 15:49:59 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* gst-libs/gst/video/gstvideopool.c:
	  videopool: store videoinfo after choosing the biggest buffer size
	  Otherwise, pool could be negotiated with a size which will be different
	  from the one used in allocation which is the GstVideoInfo.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760222

2016-01-06 12:14:39 +0100  Aurélien Zanelli <aurelien.zanelli@parrot.com>

	* gst/videotestsrc/gstvideotestsrc.c:
	  videotestsrc: add missing break in set_property switch case
	  To avoid future issue when adding new properties.
	  https://bugzilla.gnome.org/show_bug.cgi?id=760204

2016-01-06 01:04:31 +0000  Koop Mast <kwm@FreeBSD.org>

	* tests/check/elements/audioconvert.c:
	  tests: audioconvert: fix test compilation with clang
	  With clang 3.7.1 on FreeBSD:
	  elements/audioconvert.c:650:12: error: shifting a negative signed value is
	  undefined [-Werror,-Wshift-negative-value]
	  (-32 << 16) + (1 << 15), (-32 << 16) - (1 << 15),
	  ~~~ ^
	  https://bugzilla.gnome.org/show_bug.cgi?id=760134

2016-01-06 01:06:10 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tests/check/libs/audiodecoder.c:
	* tests/check/libs/audioencoder.c:
	* tests/check/libs/rtp.c:
	* tests/check/libs/rtpbasepayload.c:
	  tests: fix indentation of various unit tests

2016-01-05 22:52:34 +0000  Tim-Philipp Müller <tim@centricular.com>

	* docs/libs/gst-plugins-base-libs-docs.sgml:
	* docs/libs/gst-plugins-base-libs-sections.txt:
	  docs: add new audio API

2016-01-03 17:21:18 +0000  Tim-Philipp Müller <tim@centricular.com>

	* gst-libs/gst/sdp/gstmikey.h:
	* gst-libs/gst/video/video-overlay-composition.h:
	  docs: remove dummy function declarations with G_INLINE_FUNCTION for gtk-doc
	  gtk-doc can handle static inline functions just fine these days,
	  there's no need for this stuff any more.

2016-01-03 10:33:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/riff/riff-ids.h:
	  riff: Add missing closing parenthesis to GST_RIFF_WAVE_FORMAT_ANTEX_ADPCME
	  Apparently this #define is unused.

2016-01-02 23:29:22 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-ids.h:
	  riff-ids: remove trailing whitespace

2016-01-02 23:27:44 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst-libs/gst/riff/riff-ids.h:
	  riff-ids: fix two swapped ids
	  For these fourcc ids the name and value is swapped. This was causing a warning
	  when registering the avi ids.

2015-12-31 20:43:28 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/Makefile.am:
	  sdp: Also reorder SUBDIRS to try even harder to build the RTP library first

2015-12-31 20:41:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/Makefile.am:
	  sdp: The SDP library depends on the RTP library now and is not independent anymore
	  Fix up the build dependencies.

2015-10-07 18:50:18 +0900  Hyunjun Ko <zzoon.ko@samsung.com>

	* docs/libs/gst-plugins-base-libs-sections.txt:
	* gst-libs/gst/sdp/Makefile.am:
	* gst-libs/gst/sdp/gstmikey.c:
	* gst-libs/gst/sdp/gstmikey.h:
	* gst-libs/gst/sdp/gstsdpmessage.c:
	* gst-libs/gst/sdp/gstsdpmessage.h:
	* tests/check/libs/sdp.c:
	* win32/common/libgstsdp.def:
	  sdp: add helper fuctions from/to sdp from/to caps
	  <gstsdpmessage.h>
	  GstCaps*       gst_sdp_media_get_caps_from_media   (const GstSDPMedia *media, gint pt);
	  GstSDPResult   gst_sdp_media_set_media_from_caps   (const GstCaps* caps, GstSDPMedia *media);
	  gchar *        gst_sdp_make_keymgmt                (const gchar *uri, const gchar *base64);
	  GstSDPResult   gst_sdp_message_attributes_to_caps  (GstSDPMessage *msg, GstCaps *caps);
	  GstSDPResult   gst_sdp_media_attributes_to_caps    (GstSDPMedia *media, GstCaps *caps);
	  <gstmikey.h>
	  GstMIKEYMessage * gst_mikey_message_new_from_caps  (GstCaps *caps);
	  gchar *           gst_mikey_message_base64_encode  (GstMIKEYMessage* msg);
	  https://bugzilla.gnome.org/show_bug.cgi?id=745880

2015-12-29 18:14:54 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioconvert/gstaudioconvert.c:
	  audioconvert: Pass pointer arrays instead of singleton pointers to gst_audio_converter_samples()
	  In this specific case it wouldn't cause problems as we only ever access the
	  first array element, but let's make explicit what is happening here.
	  CID 1346530 and 1346529

2015-12-29 17:56:21 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Check for FALSE'ness directly, not by comparing with FALSE

2015-12-29 17:54:44 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encoding-profile: Don't use preset_name string after free
	  When we run the loop for another time and do not have a preset name, we would
	  try to print the preset name of a previous iteration that is already freed.
	  Also move some other variables into the block where they are actually used
	  to prevent similar mistakes in the future.
	  CID 1346536

2015-12-29 14:40:04 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/audioconvert.c:
	  audioconvert: add a test for gap handling

2015-12-29 14:23:59 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst-libs/gst/audio/audio-converter.c:
	* tests/check/elements/audioconvert.c:
	  audioconvert: fix passthrough operation
	  We did not take the sample size into account. Rearrange the tests to have more
	  conversion test and an extra test case for passthrough operations.
	  Fixes #759890

2015-12-29 11:29:31 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-device-monitor.c:
	  tools: gst-device-monitor: print uint properties in both decimal and hex
	  Some values are easier to read and make sense of in hex.
	  https://bugzilla.gnome.org//show_bug.cgi?id=759780

2015-11-12 14:01:03 -0800  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* gst-libs/gst/video/video-blend.c:
	  videoblend: special case 1x1 src dims on increment computation
	  Fix crash with 1x1 overlay pixmap
	  https://bugzilla.gnome.org/show_bug.cgi?id=757290

2015-12-28 12:28:26 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/typefind/gsttypefindfunctions.c:
	  typefindfunctions: Make sure that enough data is available in AAC/ADTS typefinder
	  We would otherwise read beyond the array bounds and crash every now and then.
	  This was introduced with 5640ba17c8db80976b7718904e4024dcfe9ee1a0.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759910

2015-12-27 19:41:43 +0100  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/audioconvert.c:
	  tests: remove commented code from audioconvert test
	  This is just what we have in gst_check_buffer_data().

2015-12-27 19:25:20 +0100  Stefan Sauer <ensonic@users.sf.net>

	* gst-libs/gst/audio/audio-converter.c:
	  audio-converter: code cleanup
	  Rename samples to num_samples, since we also have samples in chain, but that is
	  the data pointer. Always use gzize for num_samples. Make the log output a bit
	  more homogenous.

2015-12-26 11:34:47 +0000  Tim-Philipp Müller <tim@centricular.com>

	* tools/gst-device-monitor.c:
	  tools: gst-device-monitor: print non-string device properties too

2015-12-26 09:43:56 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/audio/audio-channel-mix.c:
	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-quantize.c:
	  audio: Fix some documentation warnings
	  Remove/rename function parameters and skip some functions that can't
	  be used by bindings as they are now.

2015-12-26 09:43:51 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/video/gstvideoaffinetransformationmeta.c:
	  videoaffinetransformmeta: Add (transfer none) annotation for return value

2015-12-25 11:34:10 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaysink.c:
	  playsink: Don't leak audio/video filters due to floating references weirdness
	  The filters' floating references are sinked during set_property() already,
	  which means that GstBin takes a new reference when adding the filter to it.
	  Get rid of the additional reference after adding the filter to the bin.

2015-12-25 10:36:44 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaysink.c:
	  playsink: Allow reuse of audio/video filters by unparenting them from their bins
	  And also recreate the chains if the filter is changing.

2015-12-25 10:28:02 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstplaysink.c:
	  playsink: Don't leak audio/video filters when using non-raw media

2015-12-24 15:27:43 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  Back to development

2015-12-24 13:59:52 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst-libs/gst/pbutils/Makefile.am:
	  pbutils: Link to libgstbase for bytewriter and adapter

Sebastian Dröge's avatar
Sebastian Dröge committed
802
803
=== release 1.7.1 ===

Sebastian Dröge's avatar
Sebastian Dröge committed
804
2015-12-24 13:59:15 +0100  Sebastian Dröge <sebastian@centricular.com>
Sebastian Dröge's avatar
Sebastian Dröge committed
805

Sebastian Dröge's avatar
Sebastian Dröge committed
806
807
808
	* ChangeLog:
	* NEWS:
	* RELEASE:
Sebastian Dröge's avatar
Sebastian Dröge committed
809
	* configure.ac:
Sebastian Dröge's avatar
Sebastian Dröge committed
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-alsa.xml:
	* docs/plugins/inspect/plugin-app.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
	* docs/plugins/inspect/plugin-audiorate.xml:
	* docs/plugins/inspect/plugin-audioresample.xml:
	* docs/plugins/inspect/plugin-audiotestsrc.xml:
	* docs/plugins/inspect/plugin-cdparanoia.xml:
	* docs/plugins/inspect/plugin-encoding.xml:
	* docs/plugins/inspect/plugin-gio.xml:
	* docs/plugins/inspect/plugin-libvisual.xml:
	* docs/plugins/inspect/plugin-ogg.xml:
	* docs/plugins/inspect/plugin-pango.xml:
	* docs/plugins/inspect/plugin-playback.xml:
	* docs/plugins/inspect/plugin-subparse.xml:
	* docs/plugins/inspect/plugin-tcp.xml:
	* docs/plugins/inspect/plugin-theora.xml:
	* docs/plugins/inspect/plugin-typefindfunctions.xml:
	* docs/plugins/inspect/plugin-videoconvert.xml:
	* docs/plugins/inspect/plugin-videorate.xml:
	* docs/plugins/inspect/plugin-videoscale.xml:
	* docs/plugins/inspect/plugin-videotestsrc.xml:
	* docs/plugins/inspect/plugin-volume.xml:
	* docs/plugins/inspect/plugin-vorbis.xml:
	* docs/plugins/inspect/plugin-ximagesink.xml:
	* docs/plugins/inspect/plugin-xvimagesink.xml:
	* gst-plugins-base.doap:
	* win32/common/_stdint.h:
	* win32/common/audio-enumtypes.c:
	* win32/common/audio-enumtypes.h:
	* win32/common/config.h:
	* win32/common/pbutils-enumtypes.c:
	* win32/common/pbutils-enumtypes.h:
	  Release 1.7.1

2015-12-24 13:10:08 +0100  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/id.po:
	* po/it.po:
	* po/ja.po:
	* po/lt.po:
	* po/lv.po:
	* po/nb.po:
	* po/nl.po:
	* po/or.po:
	* po/pl.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  Update .po files
Sebastian Dröge's avatar
Sebastian Dröge committed
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000

2015-12-24 12:22:04 +0100  Sebastian Dröge <sebastian@centricular.com>

	* po/nl.po:
	* po/sv.po:
	* po/zh_CN.po:
	  po: Update translations

2015-12-11 15:38:00 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst-libs/gst/pbutils/encoding-profile.c:
	  encodebin: Implement an encoding profile serialization format
	  https://bugzilla.gnome.org/show_bug.cgi?id=759356

2015-12-21 00:43:49 +0100  Koop Mast <kwm@rainbow-runner.nl>

	* configure.ac:
	  configure: Make -Bsymbolic check work with clang.
	  Update the -Bsymbolic check with the version glib has. This version
	  works with clang.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759713

2015-12-03 11:53:05 +0900  Kazunori Kobayashi <kkobayas@igel.co.jp>

	* gst-libs/gst/app/gstappsrc.c:
	  appsrc: Clear is_eos flag when receiving the flush-stop event
	  The EOS event can be propagated to the downstream elements when
	  is_eos flag remains set even after leaving the flushing state.
	  This fix allows this element to normally restart the streaming
	  after receiving the flush event by clearing the is_eos flag.
	  https://bugzilla.gnome.org/show_bug.cgi?id=759110

2015-12-16 18:11:05 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* tests/examples/playback/playback-test.c:
	  examples: playback-test: remove unused variables
	  audiosink and videosink string variables are unused

2015-11-30 10:28:55 +1100  Matthew Waters <matthew@centricular.com>

	* gst/playback/gstplaybin2.c:
	  playbin: only add the template caps when the result is empty
	  Unconditionally adding the template caps when proxying the caps query will play
	  havoc with decoders that attempt to choose an output format based on some caps
	  features.  Creating a sink that does not include those caps features and a
	  decoder/parser/etc that preferentially chooses some specific caps feature when
	  available, will always return the decoder/parser/etc template caps and choose a
	  feature that downstream will be unable to support.
	  Fix by limiting the addition of the template caps to when the result is actually
	  empty.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758212

2015-12-17 13:39:01 +0100  Sebastian Dröge <sebastian@centricular.com>

	* configure.ac:
	  configure: Don't use AG_GST_CHECK_FEATURE for checking for gio-unix-2.0
	  It's meant to be used for external plugins that can then all be disabled via
	  --disable-external. gio-unix-2.0 however is just an optional dependency for
	  the TCP unit test.
	  Also when using AG_GST_CHECK_FEATURE like this, in the --disable-external part
	  there needs to be an AM_CONDITIONAL for the feature with FALSE.

2015-12-16 17:07:54 +0100  Sebastian Dröge <sebastian@centricular.com>

	* gst/playback/gstdecodebin2.c:
	  Revert "decodebin2: fix deadlock on chain shutdown"
	  This reverts commit 77dc09c3a9a5e5e371e189f39b5557db440a8dc9.
	  It can cause the FLUSH_START/STOP events to go to the sink elements, which
	  then causes state changes and various other problems. We shouldn't really
	  flush downstream here, the idea is to do *draining*.
	  Apart from that the testcase for the original bug here works without this
	  commit now.

2015-12-16 11:12:00 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>

	* gst/tcp/gstmultifdsink.c:
	  multifdsink: fix typo in GST_WARNING_OBJECT
	  This should make easier to parse the debug logs.
	  s/fnctl/fcntl

2014-04-10 15:36:15 +0100  Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>

	* gst/videorate/gstvideorate.c:
	  videorate: remove dead code
	  Since the loops increasing count from 0 are always run at least
	  once (if count < 1), count will always be at least one when
	  compared to the drop/dup conditions.
	  Coverity 1139674

2015-12-16 10:45:48 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst-libs/gst/audio/audio-converter.c:
	* gst-libs/gst/audio/audio-converter.h:
	* win32/common/libgstaudio.def:
	  audio-converter: rework the main processing loop
	  Rework the main processing loop. We now create an audio processing
	  chain from small core functions. This is very similar to how the
	  video-converter core works and allows us to statically calculate an
	  optimal allocation strategy for all possible combinations of operations.
	  Make sure we support non-interleaved data everywhere.
	  Add functions to calculate in and out frames and latency.

2015-12-16 10:44:16 +0100  Wim Taymans <wtaymans@redhat.com>

	* gst/audioconvert/gstaudioconvert.c:
	  audioconvert: clear convert object

2015-12-16 09:35:38 +0100  Sebastian Dröge <sebastian@centricular.com>

	* docs/plugins/gst-plugins-base-plugins.args:
	* docs/plugins/gst-plugins-base-plugins.hierarchy:
	* docs/plugins/gst-plugins-base-plugins.signals:
	* docs/plugins/inspect/plugin-adder.xml:
	* docs/plugins/inspect/plugin-app.xml:
	* docs/plugins/inspect/plugin-audioconvert.xml:
For faster browsing, not all history is shown. View entire blame