-
Notifications
You must be signed in to change notification settings - Fork 0
/
ChangeLog
8834 lines (6437 loc) · 299 KB
/
ChangeLog
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
802
803
804
805
806
807
808
809
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
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
=== release 1.6.0 ===
2015-09-25 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.6.0
=== release 1.5.91 ===
2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.5.91
2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-stream.c:
stream: fix docs for recently-added get/set_buffer_size API
https://bugzilla.gnome.org/show_bug.cgi?id=749095
2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't crash on encrypted RTX SDP
In parse_keymgmt(), don't mutate the input string that's been passed
as const, especially since we might need the original value again if
the same key info applies to multiple streams (RTX, for example).
https://bugzilla.gnome.org/show_bug.cgi?id=754753
2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
* examples/test-mp4.c:
test-mp4: Support filenames with spaces in them. Error out on too few arguments
2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
* examples/test-record.c:
test-record: Check parameter count and print out help
If no launch pipeline was supplied, print out some help
2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: Implement UDP buffer size setting.
Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
UDP TX buffer size.
Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-media.h:
rtsp-media: Fix small typo causing gtk-doc to complain
=== release 1.5.90 ===
2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.5.90
2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: get port number through gst_rtsp_url_get_port
https://bugzilla.gnome.org/show_bug.cgi?id=753473
2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
* tests/check/gst/media.c:
media-test: Removing unnecessary assertion
https://bugzilla.gnome.org/show_bug.cgi?id=753385
2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-server.c:
Document that source keeps a ref on server until it's destroyed
https://bugzilla.gnome.org/show_bug.cgi?id=749227
2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* tests/check/gst/media.c:
media-test: Test for multiple dynamic payload
https://bugzilla.gnome.org/show_bug.cgi?id=753385
2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: Only add fakesink once per pipeline
The intention is to prevent going PLAYING state before pads are created.
If there was mutilple dynamic payload, it would leak few fakesink and
actually prevent from ever reaching playing state.
https://bugzilla.gnome.org/show_bug.cgi?id=753385
2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
Revert "rtsp-media: Only add 1 fakesink per pipeline"
This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Only add 1 fakesink per pipeline
There should be only one fakesink per pipeline, not per dynpay. This
would lead to element naming clash.
2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: assertion error due to wrong condition check
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j
https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Strip keys from the fmtp that we use internally in our caps
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps
https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-thread-pool.c:
threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
https://bugzilla.gnome.org/show_bug.cgi?id=752640
2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From f74b2df to 9aed1d7
2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.5.2 ===
2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.5.2
2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* tests/check/gst/client.c:
rtsp-client: allow application to decide what requirements are supported
Add "check-requirements" signal and vfunc to allow application
(and subclasses) to check the requirements.
Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
https://bugzilla.gnome.org/show_bug.cgi?id=749417
2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* common:
Automatic update of common submodule
From 6015d26 to f74b2df
2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Always use real payloader when creating streams
A bin that contains the real payloader might be used as payloader. In this
case we have to get the real payloader for the various properties it provides.
Example use cases for this are bins that payload some media and then have
additional elements that add metadata or RTP extension headers to the stream.
https://bugzilla.gnome.org/show_bug.cgi?id=750800
2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
* examples/test-netclock.c:
test-netclock: Use new ntp-time-source property on rtpbin
Select the clock time to be used as NTP time source. This allows proper
synchronization between receivers, independent of sharing base times, and just
requires them to use the same clock.
2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
* examples/test-netclock.c:
test-netclock: Setting the same base time on sender and receiver is not necessary
It's going to be fixed up by rtpbin when using ntp-sync=TRUE
2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
https://bugzilla.gnome.org/show_bug.cgi?id=750764
2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* docs/libs/gst-rtsp-server.types:
docs: add missing types
https://bugzilla.gnome.org/show_bug.cgi?id=750764
2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* docs/libs/gst-rtsp-server-sections.txt:
docs: add missing apis
https://bugzilla.gnome.org/show_bug.cgi?id=750764
2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
GstRTSPAuth: Add client certificate authentication support
https://bugzilla.gnome.org/show_bug.cgi?id=750471
2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
test-netclock-client: Use new GstClock API to wait for clock synchronization
2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
test-netclock-client: Use a GMainLoop and playbin's source-setup signal
A mainloop is needed to get glimagesink to display something on OSX, and
the source-setup signal just makes things a little bit easier.
2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From d9a3353 to 6015d26
2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From d37af32 to d9a3353
2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 21ba2e5 to d37af32
2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From c408583 to 21ba2e5
2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
* docs/libs/Makefile.am:
docs: remove variables that we define in the snippet from common
This is syncing our Makefile.am with upstream gtkdoc.
2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 44a3517 to c408583
2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.5.1 ===
2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.5.1
2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: No flush during Teardown.
When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
backlog is empty it can happen that just a part of a message will be
sent and rest is in backlog queue. If then flush during teardown
just a part of message will be sent.This can lead to client miss
teardown response since it expect to get the last part of message.
The flushing during teardown was introduced to fix a deadlock that now
is fixed more generally in handle_request by temporary setting backlog
size to unlimited.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/Makefile.am:
tests: Use AM_TESTS_ENVIRONMENT
Needed by the new automake test runner and the
current version of the common submodule.
2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.h:
rtsp-server: Use single-include rtsp header to make sure we get all definitions
2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Mark some more functions static
2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Only unblock the media in suspend() when actually changing the state
Otherwise we're going to lose a few packets for live streams during DESCRIBE.
2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-video-rtx.c:
examples: Use AVPF profile for the RTX example
2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Only add RTX to the SDP when using a feedback profile
2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: get valid clock-rate from last-sample
clock-rate in last-sample's caps is integer, not unsigned.
To get this value properly, variable needs to be type-casted to int.
https://bugzilla.gnome.org/show_bug.cgi?id=747614
2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
* autogen.sh:
* common:
autogen.sh: only run autopoint if gettext requested in configure.ac
Not just because there happens to be a po directory.
https://bugzilla.gnome.org/show_bug.cgi?id=748058
2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
Revert "configure.ac: uncomment gettext version setup"
This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
We don't need a gettext setup here and there's no po
directory either, so no reason why autopoint would be
run in the first place.
See https://bugzilla.gnome.org/show_bug.cgi?id=748058
2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
* examples/test-multicast.c:
* examples/test-multicast2.c:
* examples/test-sdp.c:
* examples/test-video-rtx.c:
* examples/test-video.c:
* tests/test-cleanup.c:
* tests/test-reuse.c:
Fix timeout function signatures across tests and examples
2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/Makefile.am:
tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
Make sure the test environment is set up.
https://bugzilla.gnome.org//show_bug.cgi?id=747624
2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
configure: bump automake requirement to 1.14 and autoconf to 2.69
This is only required for builds from git, people can still
build tarballs if they only have older autotools.
https://bugzilla.gnome.org//show_bug.cgi?id=747624
2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* configure.ac:
configure.ac: uncomment gettext version setup
Fixes autogen.sh. It would run autopoint, which would complain
that it could not find the gettext version in configure.ac.
https://bugzilla.gnome.org/show_bug.cgi?id=748058
2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* examples/test-video-rtx.c:
test-video-rtx: set exact payload type to PCMA payloader
Setting wrong payload type causes failure to do retransmission through audio stream
https://bugzilla.gnome.org/show_bug.cgi?id=747839
2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: fix to get valid each stream data for request-aux-sender signal
Because of duplicated g_signal_connect for request-aux-sender signal,
wrong stream pointer is passed to the signal handler.
Instead of passing each stream, pass stream array and get the relevant stream.
https://bugzilla.gnome.org/show_bug.cgi?id=747839
2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
* acinclude.m4:
* autogen.sh:
Update autogen.sh to latest version from common
Fixes build after aclocal_check etc. helpers have been removed.
2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From bc76a8b to c8fb372
2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Limit the queues to 1 buffer
We only need them to be able to pre-roll, queueing up more data here
is only going to harm latency and memory usage.
2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Update comment and ASCII art to the latest code
We have a queue in front of the udpsink too to prevent the pipeline from
locking up.
2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-media: Properly return first rtptime
Instead we where returning first GstBuffer timestamp. This would result
in clock skew and unwanted behaviour in RTSP playback.
https://bugzilla.gnome.org/show_bug.cgi?id=746479
2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Don't leave buffer mapped
If the seq is NULL, the RTP buffer was left mapped. We should always
unmap the buffer.
2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
* README:
Fix typo in README
2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* tests/check/gst/client.c:
Fix double semicolons
2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
This gives more accurate values than asking the payloader. There might be
queueing happening between the payloader and the sink.
https://bugzilla.gnome.org/show_bug.cgi?id=745704
2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't seek for PLAY if the position will not change
https://bugzilla.gnome.org/show_bug.cgi?id=745704
2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't include payload type in the caps for framesize
When the sdp media attribute framesize are converted to caps
the <payload> should not be included.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: add payload type to the sdp framesize attribute
The sdp framesize attribute is desribed in RFC6064. It is specified
for payloading of H263 and has the following form
a=framesize:<payload type> <width>-<height>. The <width>-<height> part
should be added to the caps in a payloader and the <payload type> should
be added by the rtsp-server.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
* examples/test-uri.c:
examples: test-uri: fix tainted variable
Insignificant but this keeps Coverity happy.
CID #1268404
2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-netclock-client.c:
* examples/test-netclock.c:
examples: Add a simple example of network synch for live streams.
An example server and client that works for synchronising live streams
only - as it can't support pause/play.
2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
rtsp-media-factory: Add functions to set/get the media gtype
Allow specifying the GType of a GstRtspMedia subclass to create
as a simpler way to get the factory to create a custom
GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: fix double unlock in _get_buffer_size()
Fixes an abort when calling gst_rtsp_media_get_buffer_size()
because of double g_mutex_unlock () usage.
https://bugzilla.gnome.org/show_bug.cgi?id=745434
2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
rtsp-session: Use monotonic time for RTSP session timeout
Changed RTSP session timeout handling to monotonic time
and deprecating the API for current system time.
This fixes timeouts when the system time changes.
https://bugzilla.gnome.org/show_bug.cgi?id=743346
2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-client: Only error out in PLAY if seeking actually failed
If the media was just not seekable, we continue from whatever position we are
and let the client decide if that is what is wanted or not.
Only if the actual seek failed, we can't really recover and should error out.
2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Add necessary queues between tee and multiudpsink
https://bugzilla.gnome.org/show_bug.cgi?id=744379
2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-media: If seeking fails, don't wait forever for the media to preroll again
Instead error out properly the same way as if the SEEKING query already
failed.
2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: minor code formatting fix
2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: fix logic for collect_streams
Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
all streams it knows if it got any, and can check if the transport mode is OK.
CID #1268400
2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't set the transport mode based on what elements we find
Just print a warning if the one that was set before disagrees with what
elements we found. It must already be set to something before as this
function is called after we received the SDP from ANNOUNCE in RECORD mode,
and we would reject ANNOUNCE if the RECORD flag was not set.
2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspserver.c:
tests: rtspserver: rename shadowed variable
We have two different 'sink' variables here,
rename one of them for clarity.
2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix awkward if clause
2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-uri.c:
examples: test-uri: improve uri argument handling and accept file names
Print an error if the argument passed is not a URI and can't
be converted into one, or no arguments have been provided.
2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-uri.c:
examples: test-uri: don't remove mount point after 10 seconds
It's very irritating when trying to test stuff repeatedly
and serves no real purpose other than showing that it can
be done.
2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/.gitignore:
examples: add new test-record to .gitignore
2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-record.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* tests/check/gst/rtspserver.c:
rtsp-media: Use flags to distinguish between PLAY and RECORD media
2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-record.c:
test-record: Set latency for playback-style example to 2s instead of 200ms
2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspserver.c:
tests: add some unit tests for ANNOUNCE and RECORD
https://bugzilla.gnome.org/show_bug.cgi?id=743175
2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix a couple of leaks in handle_announce
2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp-media: Expose latency setting for setting the rtpbin latency
2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-record.c:
test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/Makefile.am:
* examples/test-record.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
Add initial support for RECORD
We currently only support media that is RECORD or PLAY only, not both at once.
https://bugzilla.gnome.org/show_bug.cgi?id=743175
2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: RTCP and RTP transport cache cookies seperated
RTCP packets were not sent because the same tr_cache_cookie was used for
both RTP and RTCP. So only one of the tr_cache lists were populated
depending on which one was sent first. If the tr_cache list is not
populated then no packets can be sent. Most often this happened to be
RTCP. Now seperate RTCP and RTP transport cache cookies are added which
resulted in both the tr_cache_lists to be populated regardless of which
one was sent first.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fix false compiler warning
rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: log interleaved data received
2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Use a random session ID in the SDP
RFC4566 Section 5.2 says that it should make the username, session id,
nettype, addrtype and unicast address tuple globally unique. Always using
1188340656180883 is not going to guarantee that: https://xkcd.com/221/
Instead let's create a 64 bit random number, which at least brings us
closer to the goal of global uniqueness.
https://tools.ietf.org/html/rfc4566#section-5.2
2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-launch.c:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-uri.c:
examples: Don't call gst_init() and gst_get_option_group()
The latter calls the former at the appropriate time.
2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Drop trailing \0 of RTSP DATA messages
We add a trailing \0 in GstRTSPConnection to make parsing of
string message bodies easier (e.g. the SDP from DESCRIBE) but
for actual data this means we have to drop it or otherwise
create invalid data.
2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
Fixes crash when two threads access handle_new_sample() at the same
time, one for RTP, one for RTCP.
Otherwise, when iterating over the transports cache, it might be modified by
another thread at the same time if the transports cookie has changed.
https://bugzilla.gnome.org/show_bug.cgi?id=742954
2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Set format=TIME on our app sources for TCP
2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
* gst/rtsp-server/rtsp-session-pool.c:
Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
RFC 2326 states that session IDs may consist of alphanumeric as well as
the safe characters $-_.+ -- N.B. the percent character is not allowed.
Previously the session ID was URI-escaped, this meant that any character
which was not alphanumeric or any of the characters +-._~ would be
percent encoded. While the RFC (surprisingly) mentions that linear white
space in session IDs should be URI-escaped, it does not say anything
about other characters. Moreover no white space is allowed in the
session ID. Finally the percent character which is the result of
URI-escaping is not allowed in a session ID.
So there is no reason to do any URI-escaping, and now it is removed.
https://bugzilla.gnome.org/show_bug.cgi?id=742869
2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From f2c6b95 to bc76a8b
2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
* Makefile.am:
Fix 'make check' from top-level directory
2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* examples/test-launch.c:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-uri.c:
examples: Add command-line parsing and take a 'port' argument
This allows users to run multiple servers on different ports for testing.
Only done for examples that actually take arguments and hence are capable of
outputting different streams for each instance on each port.
https://bugzilla.gnome.org/show_bug.cgi?id=742115
2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-client: Add a send_message default signal handler
This allows subclasses to easily hook into the response sending
mechanism without doing everything from a signal, which seems
awkward from subclasses.
2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From ef1ffdc to f2c6b95
2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* Makefile.am:
* configure.ac:
configure: add --disable-examples switch
https://bugzilla.gnome.org/show_bug.cgi?id=741678
2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-video-rtx.c:
examples: add a retransmisison example implementing RFC4588
Currently only SSRC-multiplexed rtx streams are supported
2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix some minor memory leaks
2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Some minor cleanup
2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix compiler warnings
rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
^
rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
^
2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
media: implement ssrc-multiplexed retransmission support
based off RFC 4588 and the server-rtpaux example in -good
2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp: Ref transports in hash table.
Also ref streams for transports.
This solves a crash when reciving a rtcp after teardown but before
client finalize.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From 7bb2bce to ef1ffdc
2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: refactor cleanup of cached media
2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
* tests/check/gst/client.c:
tests: Remove FIXME
The session leak is now fixed, lets remove those FIXME comments.
2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
* tests/check/gst/rtspserver.c:
tests: Test to setup two sessions on one connection
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
* tests/check/gst/rtspserver.c:
tests: Test setup with tcp transport
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: Configure transport after creating session media
The default implementation of configure_client_transport() in
rtsp-client uses the session media when it chooses channels for
interleaved traffic.
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-media.c:
client: Stop caching media in client when doing setup
If the media has been managed by a session media, it should not be
cached in the client any longer. The GstRTSPSessionMedia object is now
responsible for unpreparing the GstRTSPMedia object using
gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
session media.
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: unref srtp decoder when leaving bin
https://bugzilla.gnome.org/show_bug.cgi?id=739481
2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: mikey memory leaks
https://bugzilla.gnome.org/show_bug.cgi?id=739383
2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From 84d06cd to 7bb2bce
2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
* Makefile.am:
Parallelise 'make check-valgrind'
2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From a8c8939 to 84d06cd
2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 36388a1 to a8c8939
2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: deactivate media when shutting down from paused
This was only done when going directly from playing.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-context.h:
rtsp-client: add stream transport to context
We add the stream transport to the context so we can get the configured
client stream transport in the setup request signal.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>