3 2015-09-25 Sebastian Dröge <slomo@coaxion.net>
10 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
16 * gst-rtsp-server.doap:
19 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
21 * docs/libs/gst-rtsp-server-sections.txt:
22 * gst/rtsp-server/rtsp-stream.c:
23 stream: fix docs for recently-added get/set_buffer_size API
24 https://bugzilla.gnome.org/show_bug.cgi?id=749095
26 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
28 * gst/rtsp-server/rtsp-media.c:
29 rtsp-media: Don't crash on encrypted RTX SDP
30 In parse_keymgmt(), don't mutate the input string that's been passed
31 as const, especially since we might need the original value again if
32 the same key info applies to multiple streams (RTX, for example).
33 https://bugzilla.gnome.org/show_bug.cgi?id=754753
35 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
37 * examples/test-mp4.c:
38 test-mp4: Support filenames with spaces in them. Error out on too few arguments
40 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
42 * examples/test-record.c:
43 test-record: Check parameter count and print out help
44 If no launch pipeline was supplied, print out some help
46 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
48 * gst/rtsp-server/rtsp-media.c:
49 * gst/rtsp-server/rtsp-stream.c:
50 * gst/rtsp-server/rtsp-stream.h:
51 rtsp-stream: Implement UDP buffer size setting.
52 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
54 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
55 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
57 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
59 * gst/rtsp-server/rtsp-media.h:
60 rtsp-media: Fix small typo causing gtk-doc to complain
62 === release 1.5.90 ===
64 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
70 * gst-rtsp-server.doap:
73 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
75 * gst/rtsp-server/rtsp-media-factory.c:
76 media-factory: get port number through gst_rtsp_url_get_port
77 https://bugzilla.gnome.org/show_bug.cgi?id=753473
79 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
81 * tests/check/gst/media.c:
82 media-test: Removing unnecessary assertion
83 https://bugzilla.gnome.org/show_bug.cgi?id=753385
85 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
87 * gst/rtsp-server/rtsp-server.c:
88 Document that source keeps a ref on server until it's destroyed
89 https://bugzilla.gnome.org/show_bug.cgi?id=749227
91 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
93 * tests/check/gst/media.c:
94 media-test: Test for multiple dynamic payload
95 https://bugzilla.gnome.org/show_bug.cgi?id=753385
97 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
99 * gst/rtsp-server/rtsp-media.c:
100 media: Only add fakesink once per pipeline
101 The intention is to prevent going PLAYING state before pads are created.
102 If there was mutilple dynamic payload, it would leak few fakesink and
103 actually prevent from ever reaching playing state.
104 https://bugzilla.gnome.org/show_bug.cgi?id=753385
106 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
108 * gst/rtsp-server/rtsp-media.c:
109 Revert "rtsp-media: Only add 1 fakesink per pipeline"
110 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
112 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
114 * gst/rtsp-server/rtsp-media.c:
115 rtsp-media: Only add 1 fakesink per pipeline
116 There should be only one fakesink per pipeline, not per dynpay. This
117 would lead to element naming clash.
119 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
121 * gst/rtsp-server/rtsp-media.c:
122 rtsp-media: assertion error due to wrong condition check
123 In media to caps function, reserved_keys array is being used for variable i,
124 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
125 changed it to variable j
126 https://bugzilla.gnome.org/show_bug.cgi?id=753009
128 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
130 * gst/rtsp-server/rtsp-media.c:
131 rtsp-media: Strip keys from the fmtp that we use internally in our caps
132 Skip keys from the fmtp, which we already use ourselves for the
133 caps. Some software is adding random things like clock-rate into
134 the fmtp, and we would otherwise here set a string-typed clock-rate
135 in the caps... and thus fail to create valid RTP caps
136 https://bugzilla.gnome.org/show_bug.cgi?id=753009
138 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
140 * gst/rtsp-server/rtsp-thread-pool.c:
141 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
142 https://bugzilla.gnome.org/show_bug.cgi?id=752640
144 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
147 Automatic update of common submodule
148 From f74b2df to 9aed1d7
150 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
155 === release 1.5.2 ===
157 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
163 * gst-rtsp-server.doap:
166 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
168 * gst/rtsp-server/rtsp-client.c:
169 * gst/rtsp-server/rtsp-client.h:
170 * tests/check/gst/client.c:
171 rtsp-client: allow application to decide what requirements are supported
172 Add "check-requirements" signal and vfunc to allow application
173 (and subclasses) to check the requirements.
174 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
175 https://bugzilla.gnome.org/show_bug.cgi?id=749417
177 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
180 Automatic update of common submodule
181 From 6015d26 to f74b2df
183 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
185 * gst/rtsp-server/rtsp-media.c:
186 rtsp-media: Always use real payloader when creating streams
187 A bin that contains the real payloader might be used as payloader. In this
188 case we have to get the real payloader for the various properties it provides.
189 Example use cases for this are bins that payload some media and then have
190 additional elements that add metadata or RTP extension headers to the stream.
191 https://bugzilla.gnome.org/show_bug.cgi?id=750800
193 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
195 * examples/test-netclock-client.c:
196 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
198 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
200 * examples/test-netclock-client.c:
201 * examples/test-netclock.c:
202 test-netclock: Use new ntp-time-source property on rtpbin
203 Select the clock time to be used as NTP time source. This allows proper
204 synchronization between receivers, independent of sharing base times, and just
205 requires them to use the same clock.
207 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
209 * examples/test-netclock-client.c:
210 * examples/test-netclock.c:
211 test-netclock: Setting the same base time on sender and receiver is not necessary
212 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
214 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
216 * gst/rtsp-server/rtsp-stream.c:
217 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
218 https://bugzilla.gnome.org/show_bug.cgi?id=750764
220 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
222 * docs/libs/gst-rtsp-server.types:
223 docs: add missing types
224 https://bugzilla.gnome.org/show_bug.cgi?id=750764
226 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
228 * docs/libs/gst-rtsp-server-sections.txt:
229 docs: add missing apis
230 https://bugzilla.gnome.org/show_bug.cgi?id=750764
232 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
234 * examples/test-netclock-client.c:
235 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
237 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
239 * docs/libs/gst-rtsp-server-sections.txt:
240 * gst/rtsp-server/rtsp-auth.c:
241 * gst/rtsp-server/rtsp-auth.h:
242 GstRTSPAuth: Add client certificate authentication support
243 https://bugzilla.gnome.org/show_bug.cgi?id=750471
245 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
247 * examples/test-netclock-client.c:
248 test-netclock-client: Use new GstClock API to wait for clock synchronization
250 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
252 * examples/test-netclock-client.c:
253 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
254 A mainloop is needed to get glimagesink to display something on OSX, and
255 the source-setup signal just makes things a little bit easier.
257 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
260 Automatic update of common submodule
261 From d9a3353 to 6015d26
263 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
266 Automatic update of common submodule
267 From d37af32 to d9a3353
269 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
272 Automatic update of common submodule
273 From 21ba2e5 to d37af32
275 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
278 Automatic update of common submodule
279 From c408583 to 21ba2e5
281 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
283 * docs/libs/Makefile.am:
284 docs: remove variables that we define in the snippet from common
285 This is syncing our Makefile.am with upstream gtkdoc.
287 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
290 Automatic update of common submodule
291 From 44a3517 to c408583
293 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
298 === release 1.5.1 ===
300 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
306 * gst-rtsp-server.doap:
309 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
311 * gst/rtsp-server/rtsp-client.c:
312 rtsp-client: No flush during Teardown.
313 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
314 backlog is empty it can happen that just a part of a message will be
315 sent and rest is in backlog queue. If then flush during teardown
316 just a part of message will be sent.This can lead to client miss
317 teardown response since it expect to get the last part of message.
318 The flushing during teardown was introduced to fix a deadlock that now
319 is fixed more generally in handle_request by temporary setting backlog
321 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
323 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
325 * tests/check/Makefile.am:
326 tests: Use AM_TESTS_ENVIRONMENT
327 Needed by the new automake test runner and the
328 current version of the common submodule.
330 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
332 * gst/rtsp-server/rtsp-media.h:
333 * gst/rtsp-server/rtsp-stream.h:
334 rtsp-server: Use single-include rtsp header to make sure we get all definitions
336 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
338 * gst/rtsp-server/rtsp-media.c:
339 rtsp-media: Mark some more functions static
341 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
343 * gst/rtsp-server/rtsp-media.c:
344 rtsp-media: Only unblock the media in suspend() when actually changing the state
345 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
347 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
349 * examples/test-video-rtx.c:
350 examples: Use AVPF profile for the RTX example
352 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
354 * gst/rtsp-server/rtsp-sdp.c:
355 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
357 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
359 * gst/rtsp-server/rtsp-stream.c:
360 rtsp-stream: get valid clock-rate from last-sample
361 clock-rate in last-sample's caps is integer, not unsigned.
362 To get this value properly, variable needs to be type-casted to int.
363 https://bugzilla.gnome.org/show_bug.cgi?id=747614
365 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
369 autogen.sh: only run autopoint if gettext requested in configure.ac
370 Not just because there happens to be a po directory.
371 https://bugzilla.gnome.org/show_bug.cgi?id=748058
373 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
376 Revert "configure.ac: uncomment gettext version setup"
377 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
378 We don't need a gettext setup here and there's no po
379 directory either, so no reason why autopoint would be
380 run in the first place.
381 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
383 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
385 * examples/test-multicast.c:
386 * examples/test-multicast2.c:
387 * examples/test-sdp.c:
388 * examples/test-video-rtx.c:
389 * examples/test-video.c:
390 * tests/test-cleanup.c:
391 * tests/test-reuse.c:
392 Fix timeout function signatures across tests and examples
394 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
396 * tests/check/Makefile.am:
397 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
398 Make sure the test environment is set up.
399 https://bugzilla.gnome.org//show_bug.cgi?id=747624
401 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
404 configure: bump automake requirement to 1.14 and autoconf to 2.69
405 This is only required for builds from git, people can still
406 build tarballs if they only have older autotools.
407 https://bugzilla.gnome.org//show_bug.cgi?id=747624
409 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
412 configure.ac: uncomment gettext version setup
413 Fixes autogen.sh. It would run autopoint, which would complain
414 that it could not find the gettext version in configure.ac.
415 https://bugzilla.gnome.org/show_bug.cgi?id=748058
417 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
419 * examples/test-video-rtx.c:
420 test-video-rtx: set exact payload type to PCMA payloader
421 Setting wrong payload type causes failure to do retransmission through audio stream
422 https://bugzilla.gnome.org/show_bug.cgi?id=747839
424 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
426 * gst/rtsp-server/rtsp-media.c:
427 * gst/rtsp-server/rtsp-stream.c:
428 * gst/rtsp-server/rtsp-stream.h:
429 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
430 Because of duplicated g_signal_connect for request-aux-sender signal,
431 wrong stream pointer is passed to the signal handler.
432 Instead of passing each stream, pass stream array and get the relevant stream.
433 https://bugzilla.gnome.org/show_bug.cgi?id=747839
435 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
439 Update autogen.sh to latest version from common
440 Fixes build after aclocal_check etc. helpers have been removed.
442 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
445 Automatic update of common submodule
446 From bc76a8b to c8fb372
448 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
450 * gst/rtsp-server/rtsp-stream.c:
451 rtsp-stream: Limit the queues to 1 buffer
452 We only need them to be able to pre-roll, queueing up more data here
453 is only going to harm latency and memory usage.
455 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
457 * gst/rtsp-server/rtsp-stream.c:
458 rtsp-stream: Update comment and ASCII art to the latest code
459 We have a queue in front of the udpsink too to prevent the pipeline from
462 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
464 * gst/rtsp-server/rtsp-stream.c:
465 rtsp-media: Properly return first rtptime
466 Instead we where returning first GstBuffer timestamp. This would result
467 in clock skew and unwanted behaviour in RTSP playback.
468 https://bugzilla.gnome.org/show_bug.cgi?id=746479
470 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
472 * gst/rtsp-server/rtsp-stream.c:
473 rtsp-stream: Don't leave buffer mapped
474 If the seq is NULL, the RTP buffer was left mapped. We should always
477 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
482 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
484 * gst/rtsp-server/rtsp-media-factory.c:
485 * tests/check/gst/client.c:
486 Fix double semicolons
488 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
490 * gst/rtsp-server/rtsp-stream.c:
491 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
492 This gives more accurate values than asking the payloader. There might be
493 queueing happening between the payloader and the sink.
494 https://bugzilla.gnome.org/show_bug.cgi?id=745704
496 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
498 * gst/rtsp-server/rtsp-media.c:
499 rtsp-media: Don't seek for PLAY if the position will not change
500 https://bugzilla.gnome.org/show_bug.cgi?id=745704
502 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
504 * gst/rtsp-server/rtsp-media.c:
505 rtsp-media: Don't include payload type in the caps for framesize
506 When the sdp media attribute framesize are converted to caps
507 the <payload> should not be included.
508 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
509 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
511 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
513 * gst/rtsp-server/rtsp-sdp.c:
514 rtsp-sdp: add payload type to the sdp framesize attribute
515 The sdp framesize attribute is desribed in RFC6064. It is specified
516 for payloading of H263 and has the following form
517 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
518 should be added to the caps in a payloader and the <payload type> should
519 be added by the rtsp-server.
520 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
522 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
524 * examples/test-uri.c:
525 examples: test-uri: fix tainted variable
526 Insignificant but this keeps Coverity happy.
529 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
531 * examples/.gitignore:
532 * examples/Makefile.am:
533 * examples/test-netclock-client.c:
534 * examples/test-netclock.c:
535 examples: Add a simple example of network synch for live streams.
536 An example server and client that works for synchronising live streams
537 only - as it can't support pause/play.
539 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
541 * gst/rtsp-server/rtsp-media-factory.c:
542 * gst/rtsp-server/rtsp-media-factory.h:
543 rtsp-media-factory: Add functions to set/get the media gtype
544 Allow specifying the GType of a GstRtspMedia subclass to create
545 as a simpler way to get the factory to create a custom
546 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
548 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
550 * gst/rtsp-server/rtsp-media.c:
551 rtsp-media: fix double unlock in _get_buffer_size()
552 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
553 because of double g_mutex_unlock () usage.
554 https://bugzilla.gnome.org/show_bug.cgi?id=745434
556 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
558 * gst/rtsp-server/rtsp-session-pool.c:
559 * gst/rtsp-server/rtsp-session.c:
560 * gst/rtsp-server/rtsp-session.h:
561 rtsp-session: Use monotonic time for RTSP session timeout
562 Changed RTSP session timeout handling to monotonic time
563 and deprecating the API for current system time.
564 This fixes timeouts when the system time changes.
565 https://bugzilla.gnome.org/show_bug.cgi?id=743346
567 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
569 * gst/rtsp-server/rtsp-client.c:
570 * gst/rtsp-server/rtsp-media.c:
571 rtsp-client: Only error out in PLAY if seeking actually failed
572 If the media was just not seekable, we continue from whatever position we are
573 and let the client decide if that is what is wanted or not.
574 Only if the actual seek failed, we can't really recover and should error out.
576 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
578 * gst/rtsp-server/rtsp-stream.c:
579 rtsp-stream: Add necessary queues between tee and multiudpsink
580 https://bugzilla.gnome.org/show_bug.cgi?id=744379
582 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
584 * gst/rtsp-server/rtsp-client.c:
585 * gst/rtsp-server/rtsp-media.c:
586 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
587 Instead error out properly the same way as if the SEEKING query already
590 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
592 * gst/rtsp-server/rtsp-stream.h:
593 rtsp-stream: minor code formatting fix
595 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
597 * gst/rtsp-server/rtsp-media.c:
598 rtsp-media: fix logic for collect_streams
599 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
600 all streams it knows if it got any, and can check if the transport mode is OK.
603 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
605 * gst/rtsp-server/rtsp-media.c:
606 rtsp-media: Don't set the transport mode based on what elements we find
607 Just print a warning if the one that was set before disagrees with what
608 elements we found. It must already be set to something before as this
609 function is called after we received the SDP from ANNOUNCE in RECORD mode,
610 and we would reject ANNOUNCE if the RECORD flag was not set.
612 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
614 * tests/check/gst/rtspserver.c:
615 tests: rtspserver: rename shadowed variable
616 We have two different 'sink' variables here,
617 rename one of them for clarity.
619 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
621 * gst/rtsp-server/rtsp-client.c:
622 rtsp-client: fix awkward if clause
624 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
626 * examples/test-uri.c:
627 examples: test-uri: improve uri argument handling and accept file names
628 Print an error if the argument passed is not a URI and can't
629 be converted into one, or no arguments have been provided.
631 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
633 * examples/test-uri.c:
634 examples: test-uri: don't remove mount point after 10 seconds
635 It's very irritating when trying to test stuff repeatedly
636 and serves no real purpose other than showing that it can
639 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
641 * examples/.gitignore:
642 examples: add new test-record to .gitignore
644 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
646 * examples/test-record.c:
647 * gst/rtsp-server/rtsp-client.c:
648 * gst/rtsp-server/rtsp-media-factory.c:
649 * gst/rtsp-server/rtsp-media-factory.h:
650 * gst/rtsp-server/rtsp-media.c:
651 * gst/rtsp-server/rtsp-media.h:
652 * tests/check/gst/rtspserver.c:
653 rtsp-media: Use flags to distinguish between PLAY and RECORD media
655 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
657 * examples/test-record.c:
658 test-record: Set latency for playback-style example to 2s instead of 200ms
660 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
662 * tests/check/gst/rtspserver.c:
663 tests: add some unit tests for ANNOUNCE and RECORD
664 https://bugzilla.gnome.org/show_bug.cgi?id=743175
666 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
668 * gst/rtsp-server/rtsp-client.c:
669 rtsp-client: fix a couple of leaks in handle_announce
671 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
673 * gst/rtsp-server/rtsp-media-factory.c:
674 * gst/rtsp-server/rtsp-media-factory.h:
675 * gst/rtsp-server/rtsp-media.c:
676 * gst/rtsp-server/rtsp-media.h:
677 rtsp-media: Expose latency setting for setting the rtpbin latency
679 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
681 * examples/test-record.c:
682 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
684 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
686 * gst/rtsp-server/rtsp-stream.c:
687 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
689 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
691 * examples/Makefile.am:
692 * examples/test-record.c:
693 * gst/rtsp-server/rtsp-client.c:
694 * gst/rtsp-server/rtsp-client.h:
695 * gst/rtsp-server/rtsp-media-factory.c:
696 * gst/rtsp-server/rtsp-media-factory.h:
697 * gst/rtsp-server/rtsp-media.c:
698 * gst/rtsp-server/rtsp-media.h:
699 * gst/rtsp-server/rtsp-session-media.c:
700 * gst/rtsp-server/rtsp-stream.c:
701 * gst/rtsp-server/rtsp-stream.h:
702 Add initial support for RECORD
703 We currently only support media that is RECORD or PLAY only, not both at once.
704 https://bugzilla.gnome.org/show_bug.cgi?id=743175
706 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
708 * gst/rtsp-server/rtsp-stream.c:
709 rtsp-stream: RTCP and RTP transport cache cookies seperated
710 RTCP packets were not sent because the same tr_cache_cookie was used for
711 both RTP and RTCP. So only one of the tr_cache lists were populated
712 depending on which one was sent first. If the tr_cache list is not
713 populated then no packets can be sent. Most often this happened to be
714 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
715 resulted in both the tr_cache_lists to be populated regardless of which
717 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
719 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
721 * gst/rtsp-server/rtsp-stream.c:
722 rtsp-stream: fix false compiler warning
723 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
725 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
727 * gst/rtsp-server/rtsp-client.c:
728 rtsp-client: log interleaved data received
730 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
732 * gst/rtsp-server/rtsp-client.c:
733 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
735 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
737 * gst/rtsp-server/rtsp-client.c:
738 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
740 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
742 * gst/rtsp-server/rtsp-client.c:
743 rtsp-client: Use a random session ID in the SDP
744 RFC4566 Section 5.2 says that it should make the username, session id,
745 nettype, addrtype and unicast address tuple globally unique. Always using
746 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
747 Instead let's create a 64 bit random number, which at least brings us
748 closer to the goal of global uniqueness.
749 https://tools.ietf.org/html/rfc4566#section-5.2
751 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
753 * examples/test-launch.c:
754 * examples/test-mp4.c:
755 * examples/test-ogg.c:
756 * examples/test-uri.c:
757 examples: Don't call gst_init() and gst_get_option_group()
758 The latter calls the former at the appropriate time.
760 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
762 * gst/rtsp-server/rtsp-client.c:
763 rtsp-client: Drop trailing \0 of RTSP DATA messages
764 We add a trailing \0 in GstRTSPConnection to make parsing of
765 string message bodies easier (e.g. the SDP from DESCRIBE) but
766 for actual data this means we have to drop it or otherwise
769 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
771 * gst/rtsp-server/rtsp-stream.c:
772 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
773 Fixes crash when two threads access handle_new_sample() at the same
774 time, one for RTP, one for RTCP.
775 Otherwise, when iterating over the transports cache, it might be modified by
776 another thread at the same time if the transports cookie has changed.
777 https://bugzilla.gnome.org/show_bug.cgi?id=742954
779 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
781 * gst/rtsp-server/rtsp-stream.c:
782 rtsp-stream: Set format=TIME on our app sources for TCP
784 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
786 * gst/rtsp-server/rtsp-session-pool.c:
787 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
788 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
789 RFC 2326 states that session IDs may consist of alphanumeric as well as
790 the safe characters $-_.+ -- N.B. the percent character is not allowed.
791 Previously the session ID was URI-escaped, this meant that any character
792 which was not alphanumeric or any of the characters +-._~ would be
793 percent encoded. While the RFC (surprisingly) mentions that linear white
794 space in session IDs should be URI-escaped, it does not say anything
795 about other characters. Moreover no white space is allowed in the
796 session ID. Finally the percent character which is the result of
797 URI-escaping is not allowed in a session ID.
798 So there is no reason to do any URI-escaping, and now it is removed.
799 https://bugzilla.gnome.org/show_bug.cgi?id=742869
801 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
804 Automatic update of common submodule
805 From f2c6b95 to bc76a8b
807 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
810 Fix 'make check' from top-level directory
812 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
814 * examples/test-launch.c:
815 * examples/test-mp4.c:
816 * examples/test-ogg.c:
817 * examples/test-uri.c:
818 examples: Add command-line parsing and take a 'port' argument
819 This allows users to run multiple servers on different ports for testing.
820 Only done for examples that actually take arguments and hence are capable of
821 outputting different streams for each instance on each port.
822 https://bugzilla.gnome.org/show_bug.cgi?id=742115
824 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
826 * gst/rtsp-server/rtsp-client.c:
827 * gst/rtsp-server/rtsp-client.h:
828 rtsp-client: Add a send_message default signal handler
829 This allows subclasses to easily hook into the response sending
830 mechanism without doing everything from a signal, which seems
831 awkward from subclasses.
833 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
836 Automatic update of common submodule
837 From ef1ffdc to f2c6b95
839 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
843 configure: add --disable-examples switch
844 https://bugzilla.gnome.org/show_bug.cgi?id=741678
846 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
848 * examples/.gitignore:
849 * examples/Makefile.am:
850 * examples/test-video-rtx.c:
851 examples: add a retransmisison example implementing RFC4588
852 Currently only SSRC-multiplexed rtx streams are supported
854 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
856 * gst/rtsp-server/rtsp-stream.c:
857 rtsp-stream: Fix some minor memory leaks
859 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
861 * gst/rtsp-server/rtsp-media.c:
862 rtsp-media: Some minor cleanup
864 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
866 * gst/rtsp-server/rtsp-stream.c:
867 rtsp-stream: Fix compiler warnings
868 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
869 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
871 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
872 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
875 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
877 * docs/libs/gst-rtsp-server-sections.txt:
878 * gst/rtsp-server/rtsp-media-factory.c:
879 * gst/rtsp-server/rtsp-media-factory.h:
880 * gst/rtsp-server/rtsp-media.c:
881 * gst/rtsp-server/rtsp-media.h:
882 * gst/rtsp-server/rtsp-sdp.c:
883 * gst/rtsp-server/rtsp-stream.c:
884 * gst/rtsp-server/rtsp-stream.h:
885 media: implement ssrc-multiplexed retransmission support
886 based off RFC 4588 and the server-rtpaux example in -good
888 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
890 * gst/rtsp-server/rtsp-client.c:
891 * gst/rtsp-server/rtsp-stream-transport.c:
892 * gst/rtsp-server/rtsp-stream.c:
893 rtsp: Ref transports in hash table.
894 Also ref streams for transports.
895 This solves a crash when reciving a rtcp after teardown but before
897 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
899 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
902 Automatic update of common submodule
903 From 7bb2bce to ef1ffdc
905 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
907 * gst/rtsp-server/rtsp-client.c:
908 client: refactor cleanup of cached media
910 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
912 * tests/check/gst/client.c:
914 The session leak is now fixed, lets remove those FIXME comments.
916 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
918 * tests/check/gst/rtspserver.c:
919 tests: Test to setup two sessions on one connection
920 https://bugzilla.gnome.org/show_bug.cgi?id=739112
922 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
924 * tests/check/gst/rtspserver.c:
925 tests: Test setup with tcp transport
926 https://bugzilla.gnome.org/show_bug.cgi?id=739112
928 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
930 * gst/rtsp-server/rtsp-client.c:
931 client: Configure transport after creating session media
932 The default implementation of configure_client_transport() in
933 rtsp-client uses the session media when it chooses channels for
935 https://bugzilla.gnome.org/show_bug.cgi?id=739112
937 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
939 * gst/rtsp-server/rtsp-client.c:
940 * gst/rtsp-server/rtsp-session-media.c:
941 client: Stop caching media in client when doing setup
942 If the media has been managed by a session media, it should not be
943 cached in the client any longer. The GstRTSPSessionMedia object is now
944 responsible for unpreparing the GstRTSPMedia object using
945 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
947 https://bugzilla.gnome.org/show_bug.cgi?id=739112
949 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
951 * gst/rtsp-server/rtsp-stream.c:
952 rtsp-stream: unref srtp decoder when leaving bin
953 https://bugzilla.gnome.org/show_bug.cgi?id=739481
955 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
957 * gst/rtsp-server/rtsp-client.c:
958 rtsp-client: mikey memory leaks
959 https://bugzilla.gnome.org/show_bug.cgi?id=739383
961 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
964 Automatic update of common submodule
965 From 84d06cd to 7bb2bce
967 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
970 Parallelise 'make check-valgrind'
972 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
975 Automatic update of common submodule
976 From a8c8939 to 84d06cd
978 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
981 Automatic update of common submodule
982 From 36388a1 to a8c8939
984 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
986 * gst/rtsp-server/rtsp-media.c:
987 rtsp-media: deactivate media when shutting down from paused
988 This was only done when going directly from playing.
989 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
991 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
993 * gst/rtsp-server/rtsp-client.c:
994 * gst/rtsp-server/rtsp-context.h:
995 rtsp-client: add stream transport to context
996 We add the stream transport to the context so we can get the configured
997 client stream transport in the setup request signal.
998 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
1000 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1002 * gst/rtsp-server/rtsp-stream.c:
1003 stream: release lock even not all transports have been removed
1004 We don't want to keep the lock even we return FALSE because not all the
1005 transports have been removed. This could lead into a deadlock.
1006 https://bugzilla.gnome.org/show_bug.cgi?id=737797
1008 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
1010 * gst/rtsp-server/rtsp-sdp.c:
1011 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
1012 These were renamed in GstRTPBasePayload in 1.0
1014 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1016 * gst/rtsp-server/rtsp-client.c:
1017 client: set session media to NULL without the lock
1018 We need to set session medias to NULL without the client lock otherwise
1019 we can end up in a deadlock if another thread is waiting for the lock
1020 and media unprepare is also waiting for that thread to end.
1021 https://bugzilla.gnome.org/show_bug.cgi?id=737690
1023 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1025 * gst/rtsp-server/rtsp-media.c:
1026 rtsp-media: Set state to UNPREPARING in all cases
1028 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
1030 * gst/rtsp-server/rtsp-media.c:
1031 media: set state to unpreparing when unprepare is initiated
1032 https://bugzilla.gnome.org/show_bug.cgi?id=737675
1034 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
1036 * gst/rtsp-server/rtsp-client.c:
1037 rtsp-client: Remove backlog limit while processings requests
1038 If the backlog limit is kept two cases of deadlocks may be
1039 encountered when streaming over TCP. Without the backlog
1040 limit this deadlocks can not happen, at the expence of
1042 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
1044 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
1046 * gst/rtsp-server/rtsp-client.c:
1047 rtsp-client: do not free main context before rtsp watch
1048 https://bugzilla.gnome.org/show_bug.cgi?id=737110
1050 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
1052 * tests/check/gst/rtspserver.c:
1053 tests: Extend unit test timeout to accomodate for valgrind
1054 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1056 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
1058 * gst/rtsp-server/rtsp-client.c:
1059 * gst/rtsp-server/rtsp-session.c:
1060 * gst/rtsp-server/rtsp-stream-transport.c:
1061 rtsp-*: Treat sending packets to clients as keepalive
1062 As long as gst-rtsp-server can successfully send RTP/RTCP data to
1063 clients then the client must be reading. This change makes the server
1064 timeout the connection if the client stops reading.
1065 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1067 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
1069 * gst/rtsp-server/rtsp-client.c:
1070 rtsp-client: Allow backlog to grow while expiring session
1071 Allow the send backlog in the RTSP watch to grow to unlimited size while
1072 attempting to bring the media pipeline to NULL due to a session
1073 expiring. Without this change the appsink element cannot change state
1074 because it is blocked while rendering data in the new_sample callback.
1075 This callback will block until it has successfully put the data into the
1076 send backlog. There is a chance that the send backlog is full at this
1077 point which means that the callback may block for a long time, possibly
1078 forever. Therefore the media pipeline may also be prevented from
1079 changing state for a long time.
1080 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1082 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
1084 * gst/rtsp-server/rtsp-client.c:
1085 rtsp-client: Make old compilers happy
1086 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
1087 Just in case that guint8 doesn't fit in a pointer. Just in case ...
1089 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
1091 * gst/rtsp-server/rtsp-client.c:
1092 client: raise the backlog limits before pausing
1093 We need to raise the backlog limits before pausing the pipeline or else
1094 the appsink might be blocking in the render method in wait_backlog() and
1095 we would deadlock waiting for paused.
1096 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
1098 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
1100 * gst/rtsp-server/rtsp-client.c:
1101 client: make define for the WATCH_BACKLOG
1102 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
1104 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
1106 * gst/rtsp-server/rtsp-client.c:
1107 client: simplify session transport handling
1108 link/unlink of the transport in a session was done to keep track of all
1109 TCP transports and to send RTP/RTCP data to the streams. We can simplify
1110 that by putting all the TCP transports in a hashtable indexed with the
1112 We also don't need to link/unlink the transports when we pause/resume
1113 the streams. The same effect is already achieved when we pause/play the
1114 media. Indeed, when we pause the media, the transport is removed from
1115 the media and the callbacks will not be called anymore.
1116 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
1118 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
1120 * gst/rtsp-server/rtsp-stream-transport.c:
1121 * gst/rtsp-server/rtsp-stream-transport.h:
1122 stream-transport: make method to handle received data
1123 Make a method to handle the data received on a channel. It sends the
1124 data to the stream of the transport on the RTP or RTCP pads based on
1127 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
1129 * examples/test-mp4.c:
1130 test: add example of dumping RTCP reports
1132 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
1134 * gst/rtsp-server/rtsp-media.c:
1135 * gst/rtsp-server/rtsp-stream.c:
1136 * gst/rtsp-server/rtsp-stream.h:
1137 rtsp-media: Make sure that sequence numbers are monotonic after pause
1138 The sequence number is not monotonic for RTP packets after pause. The
1139 reason is basepayloader generates a randon sequence number when the
1140 pipeline goes from ready to pause. With this fix generation of sequence
1141 number will be monotonic when going from pause to play request.
1142 https://bugzilla.gnome.org/show_bug.cgi?id=736017
1144 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
1146 * gst/rtsp-server/rtsp-client.c:
1147 rtsp-client: Protect saved clients watch with a mutex
1148 Fixes a crash when close() is called while merging clients
1149 in handle_tunnel(). In that case close() would destroy the
1150 watch while it is still being used in handle_tunnel().
1151 https://bugzilla.gnome.org/show_bug.cgi?id=735570
1153 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1155 * gst/rtsp-server/rtsp-stream.c:
1156 rtsp-stream: Remove the multicast group udp sources when removing from the bin
1158 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1160 * gst/rtsp-server/rtsp-media.c:
1161 * gst/rtsp-server/rtsp-stream.c:
1162 * gst/rtsp-server/rtsp-stream.h:
1163 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
1164 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
1165 seeking and will always continue counting the time. This leads to
1166 the NPT after a backwards seek to be something completely different
1167 to the actual seek position.
1168 https://bugzilla.gnome.org/show_bug.cgi?id=732644
1170 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
1172 * examples/test-appsrc.c:
1173 examples: fix another reference leak
1174 gst_rtsp_media_get_element() returns a new ref.
1176 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1178 * examples/test-appsrc.c:
1179 examples: unref element after usage
1180 gst_bin_get_by_name_recurse_up() returns an element
1181 reference that must be unreffed after usage.
1182 https://bugzilla.gnome.org/show_bug.cgi?id=734546
1184 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
1186 * gst/rtsp-server/rtsp-media.c:
1187 signals: Fix copy-pasto in target-state signal offset
1189 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
1193 Makefile: Add usage of build-checks step
1194 Allows building checks without running them
1196 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
1198 * gst/rtsp-server/rtsp-stream.c:
1199 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
1200 When a UDP multicast transport is used it is expected that the server listens
1201 for RTP and RTCP packets on the multicast group with the corresponding port.
1202 Without this we will never get RTCP packets from clients in multicast mode.
1203 https://bugzilla.gnome.org/show_bug.cgi?id=732238
1205 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1210 === release 1.4.0 ===
1212 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1218 * gst-rtsp-server.doap:
1221 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
1223 * gst/rtsp-server/rtsp-media.h:
1224 media: correct misspelled words in description
1225 https://bugzilla.gnome.org/show_bug.cgi?id=733244
1227 === release 1.3.91 ===
1229 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1235 * gst-rtsp-server.doap:
1238 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
1240 * docs/libs/gst-rtsp-server-sections.txt:
1243 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
1245 * gst/rtsp-server/rtsp-server.c:
1246 server: implement client REMOVE filter
1248 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
1250 * gst/rtsp-server/rtsp-client.c:
1251 * gst/rtsp-server/rtsp-client.h:
1252 client: expose _close() method
1253 Expose a previously internal close method to close the client
1256 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
1258 * gst/rtsp-server/rtsp-session-pool.c:
1259 session-pool: signal session-removed outside of the lock
1260 Release the lock before emiting the session-removed signal.
1262 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
1264 * gst/rtsp-server/rtsp-client.c:
1265 * gst/rtsp-server/rtsp-server.c:
1266 * gst/rtsp-server/rtsp-session-pool.c:
1267 * gst/rtsp-server/rtsp-session.c:
1268 * gst/rtsp-server/rtsp-stream.c:
1269 filter: Release lock in filter functions
1270 Release the object lock before calling the filter functions. We need to
1271 keep a cookie to detect when the list changed during the filter
1272 callback. We also keep a hashtable to make sure we only call the filter
1273 function once for each object in case of concurrent modification.
1274 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
1276 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
1278 * gst/rtsp-server/rtsp-client.c:
1279 client: check if watch is set in handle_teardown()
1280 The unit tests run without a watch
1282 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1284 * tests/check/gst/client.c:
1285 client tests: send teardown to cleanup session
1287 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
1289 * tests/check/gst/rtspserver.c:
1290 server tests: send teardown to cleanup session
1292 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1294 * gst/rtsp-server/rtsp-client.c:
1295 client: keep ref to client for the session removed handler
1296 This extra ref will be dropped when all client sessions have been
1297 removed. A session is removed when a client sends teardown, closes its
1298 endpoint of the TCP connection or the sessions expires.
1299 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1301 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
1303 * gst/rtsp-server/rtsp-client.c:
1304 * gst/rtsp-server/rtsp-session.c:
1305 * tests/check/gst/client.c:
1306 client: manage media in session as a last step
1307 Once we manage a media in a session, we can't unmanage it anymore
1308 without destroying it. Therefore, first check everything before we
1309 manage the media, otherwise if something is wrong we have no way to
1311 If we created a new session and something went wrong, remove the session
1312 again. Fixes a leak in the unit test.
1314 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1316 * examples/test-mp4.c:
1317 * examples/test-ogg.c:
1318 examples: print 'stream ready at url' for mp4 and ogg example
1320 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
1322 * gst/rtsp-server/rtsp-client.c:
1323 * gst/rtsp-server/rtsp-sdp.c:
1324 rtsp: fix for MIKEY api change
1326 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
1328 * gst/rtsp-server/rtsp-client.c:
1329 client: free watch context only once
1330 The watch context is freed when the source is destroyed. Avoids
1331 a CRITICAL when we try to unref the context twice.
1333 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
1335 * gst/rtsp-server/rtsp-client.c:
1338 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
1340 * gst/rtsp-server/rtsp-client.c:
1341 client: protect sessions with lock
1342 Protect the list of sessions with the lock.
1343 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1345 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
1347 * gst/rtsp-server/rtsp-client.c:
1348 Client: keep a ref to the session
1349 Don't just keep a weak ref to the session objects but use a hard ref. We
1350 will be notified when a session is removed from the pool (expired) with
1351 the new session-removed signal.
1352 Don't automatically close the RTSP connection when all the sessions of
1353 a client are removed, a client can continue to operate and it can create
1354 a new session if it wants. If you want to remove the client from the
1355 server, you have to use gst_rtsp_server_client_filter() now.
1356 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
1357 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
1359 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
1361 * gst/rtsp-server/rtsp-session-pool.c:
1362 * gst/rtsp-server/rtsp-session-pool.h:
1363 session-pool: add session-removed signal
1364 Add a signal to be notified when a session is removed from the pool.
1366 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
1368 * gst/rtsp-server/Makefile.am:
1369 * gst/rtsp-server/rtsp-server.h:
1370 Make rtsp-server.h a single-include header, use it for G-I
1371 https://bugzilla.gnome.org/show_bug.cgi?id=732411
1373 === release 1.3.90 ===
1375 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1381 * gst-rtsp-server.doap:
1384 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
1386 * gst/rtsp-server/rtsp-stream.c:
1387 stream: crypto can be NULL
1389 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
1391 * gst/rtsp-server/rtsp-client.c:
1392 * gst/rtsp-server/rtsp-media.c:
1393 * gst/rtsp-server/rtsp-mount-points.c:
1394 introspection: add missing allow-none annotations
1395 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1397 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
1399 * gst/rtsp-server/rtsp-address-pool.c:
1400 * gst/rtsp-server/rtsp-media.c:
1401 * gst/rtsp-server/rtsp-session-media.c:
1402 * gst/rtsp-server/rtsp-session-pool.c:
1403 * gst/rtsp-server/rtsp-stream-transport.c:
1404 * gst/rtsp-server/rtsp-stream.c:
1405 * gst/rtsp-server/rtsp-token.c:
1406 introspection: add (nullable) annotations to return values
1407 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1409 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
1411 * gst/rtsp-server/rtsp-client.c:
1412 * gst/rtsp-server/rtsp-stream.c:
1413 gi: improve annotations
1414 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
1416 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
1418 * gst/rtsp-server/rtsp-client.c:
1419 * gst/rtsp-server/rtsp-media-factory.c:
1420 * gst/rtsp-server/rtsp-media.c:
1421 * gst/rtsp-server/rtsp-server.c:
1422 signals: use generic marshal function
1423 Use the generic C marshal function.
1424 Use more explicit type instead of G_TYPE_POINTER
1426 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
1428 * gst/rtsp-server/rtsp-context.h:
1429 context: add type macro
1431 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
1433 * gst/rtsp-server/rtsp-client.c:
1434 * gst/rtsp-server/rtsp-sdp.c:
1435 * gst/rtsp-server/rtsp-sdp.h:
1436 sdp: hide key length defines
1437 They don't have a namespace.
1439 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1444 === release 1.3.3 ===
1446 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
1452 * gst-rtsp-server.doap:
1455 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1457 * gst/rtsp-server/rtsp-client.c:
1458 * gst/rtsp-server/rtsp-sdp.c:
1459 * gst/rtsp-server/rtsp-sdp.h:
1460 mikey: add different key length parameters
1461 Add encryption and authentication key length parameters to MIKEY. For
1462 the encoders, the key lengths are obtained from the cipher and auth
1463 algorithms set in the caps. For the decoders, they are obtained while
1464 parsing the key management from the client.
1465 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
1467 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
1469 * tests/check/gst/stream.c:
1470 stream tests: Make sure we get right multicast address from stream
1471 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
1473 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
1475 * gst/rtsp-server/rtsp-client.c:
1476 client: ref the context until rtsp watch is alive
1477 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
1479 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
1481 * gst/rtsp-server/rtsp-client.c:
1482 client: Destroy the rtsp watch after connection close
1484 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
1486 * gst/rtsp-server/rtsp-media.c:
1487 media: fix confusing comment
1489 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
1491 * gst/rtsp-server/rtsp-session.c:
1492 rtsp-session: Timeout in header.
1493 Adding the possbilty to always have timout in header.
1494 This is configurabe with setting "timeout-always-visible".
1495 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
1497 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
1502 === release 1.3.2 ===
1504 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
1511 * gst-rtsp-server.doap:
1514 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1517 Automatic update of common submodule
1518 From 211fa5f to 1f5d3c3
1520 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
1522 * gst/rtsp-server/rtsp-client.c:
1523 client: store TCP ports in transport
1524 Store the TCP ports in the transport when we are doing RTSP over TCP.
1525 This way, we can easily get to the ports from the transport.
1526 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
1528 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1530 * gst/rtsp-server/rtsp-stream.c:
1531 stream: add signals for new RTP/RTCP encoders
1532 New signals to allow the user to configure the dynamically created
1534 https://bugzilla.gnome.org/show_bug.cgi?id=730228
1536 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1538 * gst/rtsp-server/rtsp-media.c:
1539 * gst/rtsp-server/rtsp-media.h:
1540 media: Make suspend()/unsuspend() virtual
1541 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
1543 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1545 * gst/rtsp-server/rtsp-client.c:
1546 client: fix send-message signal marshaller
1547 Use generic marshalling for the send-message signal. It has
1548 two POINTER arguments, not just one.
1549 https://bugzilla.gnome.org/show_bug.cgi?id=729900
1551 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
1553 * tests/check/gst/media.c:
1554 tests: add and remove pads only once
1555 In this test we simulate a dynamic pad by watching the caps event.
1556 Because of renegotiation in the base payloader now, this caps is sent
1557 multiple times but we can only deal with 1 invocation, use a variable to
1558 only 'add and remove' the pad once.
1560 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1562 * tests/check/gst/rtspserver.c:
1563 tests: add unit test for correct handling of Require headers
1564 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1566 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1568 * gst/rtsp-server/rtsp-client.c:
1569 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
1570 Servers must handle Require headers and must report a failure
1571 if they don't handle any of the Required options, see RFC 2326,
1572 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
1573 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1575 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1580 === release 1.3.1 ===
1582 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1588 * gst-rtsp-server.doap:
1591 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
1594 Automatic update of common submodule
1595 From bcb1518 to 211fa5f
1597 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
1602 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1604 * tests/check/gst/sessionmedia.c:
1605 tests: fix memory leak in sessionmedia unit test
1607 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
1609 * gst/rtsp-server/rtsp-client.c:
1610 client: emit a signal before sending a message
1611 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
1613 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
1615 * gst/rtsp-server/rtsp-client.c:
1616 client: pass context to send_message
1617 Pass the current context to send_message, we will need it later.
1619 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
1621 * gst/rtsp-server/rtsp-client.c:
1622 client: fix typo in comment
1624 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
1626 * gst/rtsp-server/rtsp-media.c:
1627 media: Do not stop thread twice if default_prepare() fails
1629 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
1631 * gst/rtsp-server/rtsp-client.c:
1632 client: set the watch to flushing before going to NULL
1633 First set the watch to flushing so that we unblock any current and
1634 future attempt to send data on the watch, Then set the pipeline to
1636 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
1638 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
1640 * gst/rtsp-server/rtsp-session-pool.c:
1641 * tests/check/gst/sessionpool.c:
1642 rtsp-session-pool: Fixes annotation
1643 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
1644 in the sessionpool test.
1645 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
1647 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
1649 * gst/rtsp-server/rtsp-media.c:
1650 * gst/rtsp-server/rtsp-media.h:
1651 media: make media_prepare virtual
1652 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
1654 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1656 * gst/rtsp-server/rtsp-media.c:
1657 * tests/check/gst/media.c:
1658 media: stop the thread in more error cases
1660 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
1662 * gst/rtsp-server/rtsp-media.c:
1663 * tests/check/gst/media.c:
1664 media: allow NULL as the thread
1665 Use the default context whan passing a NULL thread.
1667 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1669 * gst/rtsp-server/rtsp-client.c:
1670 rtsp-client: indent cleanup
1671 Coverity was moaning about unreachable code, and I think it was just
1672 confused by { being before the label. We'll see if it pops up again.
1675 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
1677 * gst/rtsp-server/rtsp-client.c:
1678 * gst/rtsp-server/rtsp-media.c:
1679 client: Add drop-backlog property
1680 When we have too many messages queued for a client (currently hardcoded
1681 to 100) we overflow and drop the messages. Add a drop-backlog property
1682 to control this behaviour. Setting this property to FALSE will retry
1683 to send the messages to the client by waiting for more room in the
1685 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
1687 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
1689 * gst/rtsp-server/rtsp-client.c:
1690 client: support for POST before GET when setting up a tunnel
1692 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
1694 * gst/rtsp-server/rtsp-client.c:
1695 client: remove watch of the second client after http tunnel setup
1696 The second client will be freed after the HTTP tunnel has been set up.
1697 Make sure it's RTSP watch is never dispatched again.
1698 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
1700 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
1702 * gst/rtsp-server/rtsp-media.c:
1703 * tests/check/gst/media.c:
1704 media: Make media_prepare() fail if port allocation fails
1705 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
1707 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
1709 * tests/check/gst/media.c:
1710 media test: cleanup the thread pool in tests
1712 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
1714 * gst/rtsp-server/rtsp-media.c:
1715 * tests/check/gst/media.c:
1716 rtsp-media: Unblock blocked streams in unprepare
1717 The streams will be blocked when a live media is prepared.
1718 The streams should be unblocked in gst_rtsp_media_unprepare.
1719 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
1721 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
1723 * gst/rtsp-server/rtsp-media.c:
1724 media: release the state lock when going to NULL
1725 Set our state to UNPREPARING and release the state-lock before
1726 setting the pipeline to the NULL state. This way, any pad-added
1727 callback will be able to take the state-lock and check that we are now
1728 unpreparing instead of deadlocking.
1729 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
1731 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
1733 * gst/rtsp-server/rtsp-media.c:
1734 media: protect status with lock
1735 Make sure we only update the status with the lock.
1737 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
1739 * gst/rtsp-server/rtsp-client.c:
1740 * gst/rtsp-server/rtsp-sdp.c:
1741 rtsp: update for MIKEY API changes
1743 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
1745 * gst/rtsp-server/rtsp-client.c:
1746 client: parse the mikey response from the client
1747 Parse the mikey response from the client and update the policy for
1750 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
1752 * gst/rtsp-server/rtsp-stream.c:
1753 * gst/rtsp-server/rtsp-stream.h:
1754 stream: add method to set crypto info
1755 Make a method to configure the crypto information of a stream.
1756 Set udpsrc in READY instead of PAUSED so that we can configure caps
1759 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
1761 * gst/rtsp-server/rtsp-client.c:
1762 client: cleanup error paths
1764 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
1766 * gst/rtsp-server/rtsp-media.c:
1769 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
1771 * examples/test-video.c:
1772 test: enable SRTP only on RTSPS
1773 We only want to enable SRTP when doing rtsp over TLS so that we can
1774 exchange the keys in a secure way.
1776 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
1778 * examples/test-video.c:
1779 test: print an error on failure
1781 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
1784 * examples/test-video.c:
1785 * gst/rtsp-server/rtsp-sdp.c:
1786 * gst/rtsp-server/rtsp-stream.c:
1787 * tests/check/Makefile.am:
1788 stream: add SRTP support
1789 Install srtp encoder and decoder elements in rtpbin
1792 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1794 * tests/check/Makefile.am:
1795 * tests/check/gst/sessionpool.c:
1796 tests: Add unit tests for sessionpool
1797 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
1799 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1801 * tests/check/gst/threadpool.c:
1802 tests: Improve code coverage of rtsp-threadpool tests
1803 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
1805 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1807 * tests/check/gst/sessionmedia.c:
1808 tests: Improve code coverage for rtsp-session-media
1809 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
1811 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1813 gobject-introspection: Add annotations to support language bindings
1814 In addition a few cosmetic changes:
1815 * Adjust the order of arguments
1816 * Fix typo: occured -> occurred
1817 * Fix indentation after Return:-clauses
1818 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
1820 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1822 * gst/rtsp-server/rtsp-stream.c:
1823 rtsp-stream: Don't mix IPv4 and IPv6 addresses
1824 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
1826 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
1828 * gst/rtsp-server/rtsp-stream.c:
1829 stream: take caps after the session manager
1830 Take the caps for the SDP after they leave the rtpbin so that we can
1831 also get the properties added by rtpbin elements.
1833 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
1835 * gst/rtsp-server/rtsp-stream.c:
1836 stream: release lock while pushing out packets
1837 Keep a cache of the transports and use this to iterate the transport
1838 while pushing packets. This allows us to release the lock early.
1839 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
1841 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
1843 * gst/rtsp-server/rtsp-client.c:
1844 * gst/rtsp-server/rtsp-client.h:
1845 rtsp-client: vmethod for modifying tunnel GET response
1846 Add a vmethod tunnel_http_response where the response to the HTTP GET
1847 for tunneled connections can be modified.
1848 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
1850 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
1852 * gst/rtsp-server/rtsp-sdp.c:
1853 sdp: make 1 media line per profile
1854 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
1855 line in the SDP for each profile. The client is then supposed to pick
1856 one of the profiles in the SETUP request. Because the m= lines have the
1857 same pt, the client also knows that only 1 option is possible.
1859 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
1861 * gst/rtsp-server/rtsp-media-factory.c:
1862 * gst/rtsp-server/rtsp-media-factory.h:
1863 * gst/rtsp-server/rtsp-media.c:
1864 factory: add profile property and pass to media and streams
1866 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
1868 * examples/test-multicast.c:
1869 * gst/rtsp-server/rtsp-sdp.c:
1870 sdp: pass multicast connection for multicast-only stream
1871 Pass the multicast address of the stream in the connection info in the
1872 SDP so that clients try a multicast connection first.
1873 Only allow multicast connections in the test-multicast example. Also
1874 increase the TTL a little.
1876 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1879 .gitignore: Ignore gcov intermediate files
1880 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
1882 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
1884 * gst/rtsp-server/rtsp-stream.c:
1885 stream: release some locks in error cases
1887 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1889 docs: Enable and fix gtk-doc warnings
1890 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
1891 * addresspool/mediafactory: Add missing annotation colon
1892 * stream: Annotate return value
1893 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
1895 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1898 Automatic update of common submodule
1899 From fe1672e to bcb1518
1901 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
1904 Automatic update of common submodule
1905 From 1a07da9 to fe1672e
1907 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1909 * examples/Makefile.am:
1910 examples: use LDADD for libs instead of LDFLAGS
1912 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
1915 configure: make sure releases are in .doap file
1917 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
1919 * examples/test-cgroups.c:
1920 examples: test-cgroups: don't put code with side effects into g_assert()
1921 The g_assert() might get compiled out with the right
1922 compiler/preprocessor flags.
1924 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1926 * examples/.gitignore:
1927 examples: add cgroup test binary to .gitignore
1929 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
1931 * examples/test-cgroups.c:
1932 examples: fix cgroup test build
1933 Fixes build failure caused by compiler warning:
1934 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
1936 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
1939 .gitignore: ignore temp files created in the course of 'make check'
1941 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
1943 * gst/rtsp-server/rtsp-media.c:
1944 rtsp-media: don't loose frames handling new PLAY request
1945 If client supplied a range check if the range specifies the start point.
1946 If not, then do an accurate seek to the current position. If a start
1947 point was specified do do a key unit seek to make sure the streaming
1948 starts with decodeable frames.
1949 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
1951 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
1953 * gst/rtsp-server/rtsp-media.c:
1954 Revert "media: only flush when setting a new start position"
1955 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
1956 We need to do the flush in all cases, demuxer block currently for
1959 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
1961 * gst/rtsp-server/rtsp-media.c:
1962 media: only flush when setting a new start position
1963 Only flush the pipeline when we change the start position with
1965 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
1967 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
1969 * gst/rtsp-server/rtsp-stream.c:
1970 stream: set ttl-mc before adding the socket
1971 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
1972 never be set on socket.
1973 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
1975 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
1977 * gst/rtsp-server/rtsp-media.c:
1978 media: stop thread if media is already prepared
1979 in gst_rtsp_media_prepare() the thread is not used if media is already
1980 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
1982 https://bugzilla.gnome.org/show_bug.cgi?id=724182
1984 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
1987 build: Ship gst-rtsp-server.doap file
1989 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
1991 * tests/check/gst/rtspserver.c:
1992 tests: Fix another compiler warning with gcc
1994 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
1996 * gst/rtsp-server/rtsp-client.c:
1997 * gst/rtsp-server/rtsp-mount-points.c:
1998 * gst/rtsp-server/rtsp-stream.c:
1999 * tests/check/gst/client.c:
2000 rtsp-server: Fix lots of compiler warnings with clang
2002 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
2005 * gst-rtsp-server.doap:
2006 * tests/Makefile.am:
2007 configure: Synchronise with the configure scripts of the other modules
2009 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2012 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
2014 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2016 * gst/rtsp-server/rtsp-media.c:
2017 * gst/rtsp-server/rtsp-stream.c:
2018 Revert "rtsp-server: support build against last stable release"
2019 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
2020 Let us require 1.2.3 now, which is going to be released in a few
2023 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
2025 * gst/rtsp-server/rtsp-session-media.c:
2026 * gst/rtsp-server/rtsp-stream-transport.c:
2027 session: improve RTP-Info
2028 Ignore streams that can't generate RTP-Info instead of failing.
2029 Don't return the empty string when all streams are unconfigured but
2030 return NULL so that we don't generate and empty RTP-Info header.
2031 Improve docs a little.
2033 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
2035 * gst/rtsp-server/rtsp-session-media.c:
2036 Don't free rtpinfo GString when it is NULL
2037 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2039 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
2041 * gst/rtsp-server/rtsp-media.c:
2042 media: only set keyframe flag when modifying start
2043 Only set the keyframe flag when we modify the start position. The
2044 keyframe flag should probably be ignored when no change is requested but
2045 until we can claim this is all documented properly and all demuxer
2046 implement this, avoid setting the flag.
2047 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
2049 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
2051 * gst/rtsp-server/rtsp-thread-pool.c:
2052 thread-pool: Unref source after mainloop has quit to avoid races in GLib
2053 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
2055 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
2057 * gst/rtsp-server/rtsp-stream.c:
2058 stream: handle NULL seqnum and rtptime arguments
2060 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
2062 * gst/rtsp-server/rtsp-thread-pool.c:
2063 * tests/check/gst/threadpool.c:
2064 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
2065 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2067 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
2069 * gst/rtsp-server/rtsp-stream.c:
2070 stream: add fallback for missing stats property
2071 Use a fallback when the payloader does not have a stats property
2072 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2074 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
2077 Automatic update of common submodule
2078 From f7bc1c3 to 1a07da9
2080 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
2082 * gst/rtsp-server/rtsp-stream.c:
2083 stream: don't leak stats structure
2084 Don't leak the stats structure and deal with NULL stats.
2086 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
2088 * gst/rtsp-server/rtsp-stream.c:
2089 stream: Get rtpinfo properties atomically from payloader
2090 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2092 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
2094 * gst/rtsp-server/rtsp-media.c:
2095 media: refactor state change functions and signals
2096 Make functions to set the target state and the pipeline state and emit
2097 the signals from those functions.
2099 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
2101 * gst/rtsp-server/rtsp-media.c:
2102 * gst/rtsp-server/rtsp-media.h:
2103 media: add signal to notify of pending state changes
2105 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2107 * gst/rtsp-server/rtsp-media.c:
2108 * gst/rtsp-server/rtsp-stream.c:
2109 rtsp-server: support build against last stable release
2110 Until 1.2.3 is out with the new get_type function and we
2113 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
2115 * gst/rtsp-server/rtsp-stream.c:
2116 stream: fix compilation
2118 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
2120 * gst/rtsp-server/rtsp-media.c:
2121 * gst/rtsp-server/rtsp-media.h:
2122 * gst/rtsp-server/rtsp-stream.c:
2123 * gst/rtsp-server/rtsp-stream.h:
2124 stream: add property to configure profiles
2126 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
2128 * gst/rtsp-server/rtsp-client.c:
2129 client: let stream check supported transport
2130 Delegate the check if a transport is allowed to the stream.
2131 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2133 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
2135 * gst/rtsp-server/rtsp-stream.c:
2136 * gst/rtsp-server/rtsp-stream.h:
2137 stream: add method to check supported transport
2138 Add a method to check if a transport is supported
2140 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
2143 configure.ac: Only check for gstreamer-check, not check
2144 We include check in gstreamer-check since quite some time now.
2146 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
2148 * gst/rtsp-server/rtsp-session-media.c:
2149 * gst/rtsp-server/rtsp-stream-transport.c:
2150 * gst/rtsp-server/rtsp-stream.c:
2151 * gst/rtsp-server/rtsp-stream.h:
2152 stream: return clock-rate from get_rtpinfo
2153 And use it to correct the rtptime to the requested start-time.
2154 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2156 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
2158 * gst/rtsp-server/rtsp-session-media.c:
2159 * gst/rtsp-server/rtsp-stream-transport.c:
2160 * gst/rtsp-server/rtsp-stream-transport.h:
2161 session-media: calculate start-time
2163 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
2165 * gst/rtsp-server/rtsp-stream-transport.c:
2166 * gst/rtsp-server/rtsp-stream.c:
2167 * gst/rtsp-server/rtsp-stream.h:
2168 stream: also return the running-time
2169 Return the running-time in the rtpinfo as well.
2171 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
2173 * gst/rtsp-server/rtsp-client.c:
2174 * gst/rtsp-server/rtsp-session-media.c:
2175 * gst/rtsp-server/rtsp-session-media.h:
2176 * gst/rtsp-server/rtsp-stream-transport.c:
2177 * gst/rtsp-server/rtsp-stream-transport.h:
2178 session-media: let the session-media make the RTPInfo
2179 Add method to create the RTPInfo for a stream-transport.
2180 Add method to create the RTPInfo for all stream-transports in a
2182 Use the session-media RTPInfo code in client. This allows us to refactor
2183 another method to link the TCP callbacks.
2185 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2187 mount-points: sort sequence before g_sequence_lookup
2188 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
2189 sort sequence if dirty, otherwise lookup will fail.
2190 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2192 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2195 configure: rename package from gst-rtsp to gst-rtsp-server
2196 To match git module name and avoid confusion with the
2197 rtsp lib in gst-plugins-base and rtsp plugin in -good.
2199 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
2202 configure: bump core/base/good requirement to 1.2.0
2203 Bump to released stable version and make implicit
2204 requirements explicit.
2206 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2211 Fix broken gettext setup which is not used anyway
2213 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
2216 Automatic update of common submodule
2217 From dbedaa0 to d48bed3
2219 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
2221 * gst/rtsp-server/rtsp-client.c:
2222 * gst/rtsp-server/rtsp-media.c:
2223 * gst/rtsp-server/rtsp-media.h:
2224 media: add setup_sdp vmethod
2225 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
2226 gst_rtsp_media_setup_sdp.
2227 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2229 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
2231 * gst/rtsp-server/rtsp-stream.c:
2232 rtsp-stream: Check return value of sscanf
2233 streamid is only valid if sscanf matched something.
2235 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
2237 * gst/rtsp-server/rtsp-client.c:
2238 rtsp-client: Fix iteration
2239 Wouldn't even enter the code block otherwise (i++ was used as the check
2240 and not the postfix).
2242 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
2244 * gst/rtsp-server/rtsp-client.c:
2245 * gst/rtsp-server/rtsp-client.h:
2246 client: add vmethod to configure media and streams
2247 Implement a vmethod that can be used to configure the media and the
2248 streams based on the current context. Handle the blocksize handling in
2249 the default handler.
2250 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
2252 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2255 Make git ignore more unit test binaries
2257 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2259 * gst/rtsp-server/rtsp-address-pool.h:
2260 * gst/rtsp-server/rtsp-auth.h:
2261 * gst/rtsp-server/rtsp-client.h:
2262 * gst/rtsp-server/rtsp-context.h:
2263 * gst/rtsp-server/rtsp-media-factory-uri.h:
2264 * gst/rtsp-server/rtsp-media-factory.h:
2265 * gst/rtsp-server/rtsp-media.h:
2266 * gst/rtsp-server/rtsp-mount-points.h:
2267 * gst/rtsp-server/rtsp-server.h:
2268 * gst/rtsp-server/rtsp-session-media.h:
2269 * gst/rtsp-server/rtsp-session-pool.h:
2270 * gst/rtsp-server/rtsp-session.h:
2271 * gst/rtsp-server/rtsp-stream-transport.h:
2272 * gst/rtsp-server/rtsp-stream.h:
2273 * gst/rtsp-server/rtsp-thread-pool.h:
2274 * gst/rtsp-server/rtsp-token.h:
2275 rtsp-server: add padding to many public structures
2276 Not mini objects though, since they are not subclassable
2277 anyway, nor kept on the stack or inlined in a structure.
2279 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2281 media: add new create_rtpbin vmethod
2282 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
2283 https://bugzilla.gnome.org/show_bug.cgi?id=719734
2285 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
2287 * tests/check/gst/media.c:
2288 tests: fix memory leak, free test's thread pool
2289 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
2291 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
2293 * gst/rtsp-server/rtsp-stream-transport.c:
2294 stream-transport: free url in finalize
2296 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
2298 * gst/rtsp-server/rtsp-media.c:
2299 media: also do state change in suspended state
2301 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
2303 * gst/rtsp-server/rtsp-client.c:
2304 * gst/rtsp-server/rtsp-media.c:
2305 media: also handle prepare and range in suspended state
2306 When we are suspended, we are already prepared.
2307 We can get the range in the suspended state.
2309 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
2311 * tests/check/Makefile.am:
2312 * tests/check/gst/sessionmedia.c:
2313 check: add test for uri in setup
2314 Added unit tests for the new functionality in GstRTSPStreamTransport.
2315 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2317 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
2319 * gst/rtsp-server/rtsp-client.c:
2320 client: store setup uri and use in PLAY response
2321 Store the uri used when doing the setup and use that in the PLAY
2323 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2325 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
2327 * gst/rtsp-server/rtsp-stream-transport.c:
2328 * gst/rtsp-server/rtsp-stream-transport.h:
2329 stream-transport: add method to get/set url
2331 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
2333 * gst/rtsp-server/rtsp-client.c:
2334 client: suspend after SDP and unsuspend before PLAYING
2335 Based on patches by Ognyan Tonchev <ognyan@axis.com>
2336 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
2338 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
2340 * gst/rtsp-server/rtsp-media-factory.c:
2341 * gst/rtsp-server/rtsp-media-factory.h:
2342 * gst/rtsp-server/rtsp-media.c:
2343 * gst/rtsp-server/rtsp-media.h:
2344 * gst/rtsp-server/rtsp-session-media.c:
2345 * gst/rtsp-server/rtsp-session.c:
2346 * tests/check/gst/media.c:
2347 * tests/check/gst/mediafactory.c:
2348 media: add suspend modes
2349 Add support for different suspend modes. The stream is suspended right after
2350 producing the SDP and after PAUSE. Different suspend modes are available that
2351 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
2352 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
2353 state and RESET will bring the pipeline to the NULL state.
2354 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
2355 this means that the pipeline needs to be prerolled again.
2356 Base on patches by Ognyan Tonchev <ognyan@axis.com>
2357 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2359 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
2361 * gst/rtsp-server/rtsp-media.c:
2362 media: start live streams in blocked state
2363 Start live streams in the blocked state and make them preroll using the
2364 messages. This ensure that no data is played by the sink until we explicitly
2365 unblock the stream right before going to PLAYING.
2366 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2368 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
2370 * gst/rtsp-server/rtsp-media.c:
2371 media: refactor starting and waiting for preroll
2372 Based on patches from Ognyan Tonchev <ognyan@axis.com>
2373 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2375 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
2377 * gst/rtsp-server/rtsp-stream.c:
2378 * gst/rtsp-server/rtsp-stream.h:
2379 stream: add API to block streams
2380 Add an API to block on the streams and make it post a message.
2381 Based on patch by Ognyan Tonchev <ognyan@axis.com>
2382 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2384 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
2386 * docs/libs/Makefile.am:
2387 docs: Specify the override file
2388 Even if it's empty (for now) it avoids make distcheck complaining
2390 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
2392 * gst/rtsp-server/rtsp-media.c:
2393 media: move default implementations to where they are used
2395 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
2397 * gst/rtsp-server/rtsp-media.c:
2398 media: take the right lock in gst_rtsp_media_set_pipeline_state()
2399 We need to take the state_lock when calling this method.
2401 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
2403 * gst/rtsp-server/rtsp-media.c:
2404 media: handle add-added on non-bins too
2405 Handle dynamic payloaders that are not bins, as used in the unit-test.
2407 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2409 * gst/rtsp-server/rtsp-media-factory.c:
2410 * gst/rtsp-server/rtsp-media-factory.h:
2411 * gst/rtsp-server/rtsp-media.c:
2412 rtsp-media/-factory: Fix request pad name comments
2413 These must be escaped for gtk-doc to parse the comments without warnings.
2415 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2417 rtsp-media: remove transports if media is in error status
2418 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
2419 trying to change to GST_STATE_NULL and media is in error status, we
2420 remove all transports.
2421 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
2423 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
2425 * gst/rtsp-server/rtsp-media.c:
2426 rtsp-media: use element metadata to find payloader
2427 Use the element metadata to find the payloader instead of checking
2429 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
2431 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2433 rtsp-stream: add getter for payload type
2434 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
2435 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
2436 element and create the stream with this one instead of the dynpay%d
2438 https://bugzilla.gnome.org/show_bug.cgi?id=712396
2440 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2442 * gst/rtsp-server/rtsp-client.c:
2443 * gst/rtsp-server/rtsp-context.h:
2444 * gst/rtsp-server/rtsp-media.c:
2445 * gst/rtsp-server/rtsp-mount-points.c:
2446 * gst/rtsp-server/rtsp-server.c:
2447 * gst/rtsp-server/rtsp-token.c:
2448 rtsp-*: Refer to NULL as a constant in comments
2450 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2452 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2454 rtsp-*: Fix type name typos in comments
2455 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
2456 * rtsp-auth: Refer to part of constant name as text
2457 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
2458 * rtsp-session-media: Fix GstRTSPSessionMedia typo
2459 * rtsp-stream: Fix typo when refering to GstBin
2460 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2462 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2465 * docs/libs/gst-rtsp-server-docs.sgml:
2466 * docs/libs/gst-rtsp-server-sections.txt:
2467 docs: Improve documentation
2468 * Include annotation-glossary to quiet gtk-doc
2469 * Rename remaining ClientState -> Context
2470 * Rename object hierarchy file
2471 * Remove stale chapter references
2472 * Add missing function and object references
2473 * Include missing GstRTSPAddressPoolResult
2474 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2476 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2478 * gst/rtsp-server/rtsp-client.c:
2479 * gst/rtsp-server/rtsp-server.c:
2480 * gst/rtsp-server/rtsp-session-pool.c:
2481 * gst/rtsp-server/rtsp-session.c:
2482 * gst/rtsp-server/rtsp-stream.c:
2483 rtsp-server: sprinkle some allow-none annotations for g-i
2485 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
2487 * gst/rtsp-server/rtsp-stream.c:
2488 * gst/rtsp-server/rtsp-stream.h:
2489 stream: add method to filter transports
2490 Add a method to safely iterate and collect the stream transports
2491 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2493 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
2495 * gst/rtsp-server/rtsp-client.c:
2496 * gst/rtsp-server/rtsp-server.c:
2497 * gst/rtsp-server/rtsp-session-pool.c:
2498 * gst/rtsp-server/rtsp-session.c:
2499 rtsp: allow NULL func in filters
2500 Passing a null function make the filters return a list of
2503 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
2505 * gst/rtsp-server/rtsp-address-pool.c:
2506 * tests/check/gst/addresspool.c:
2507 address-pool: fix address increment
2508 Use a guint instead of guint8 to increment the address. It's still not
2509 completely correct because a guint might not be able to hold the complete
2510 address range, but that's an enhacement for later.
2511 Add unit test to test improved behaviour.
2512 https://bugzilla.gnome.org/show_bug.cgi?id=708237
2514 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
2516 * gst/rtsp-server/rtsp-client.c:
2517 * tests/check/gst/client.c:
2518 client: allow absolute path in requests
2519 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
2521 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
2523 * gst/rtsp-server/rtsp-client.c:
2524 * gst/rtsp-server/rtsp-client.h:
2525 client: make make_path_from_uri a vmethod
2527 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2529 * docs/libs/gst-rtsp-server-sections.txt:
2530 * gst/rtsp-server/rtsp-stream.c:
2531 * gst/rtsp-server/rtsp-stream.h:
2532 * tests/check/Makefile.am:
2533 * tests/check/gst/stream.c:
2534 stream: Add functions to get rtp and rtcp sockets
2535 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2537 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2539 * gst/rtsp-server/rtsp-context.c:
2540 * gst/rtsp-server/rtsp-context.h:
2541 context: defing a GType for the context
2542 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
2544 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
2546 * gst/rtsp-server/Makefile.am:
2547 * gst/rtsp-server/rtsp-auth.c:
2548 * gst/rtsp-server/rtsp-context.c:
2549 * gst/rtsp-server/rtsp-media.c:
2550 * gst/rtsp-server/rtsp-mount-points.c:
2551 * gst/rtsp-server/rtsp-server.h:
2552 * gst/rtsp-server/rtsp-session-media.c:
2553 * gst/rtsp-server/rtsp-session.c:
2554 * gst/rtsp-server/rtsp-stream.c:
2555 Fixed several GIR warnings
2557 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
2559 * gst/rtsp-server/rtsp-auth.c:
2562 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2564 * tests/check/Makefile.am:
2565 * tests/check/gst/token.c:
2566 tests: Add unit tests for token
2567 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2569 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2571 * gst/rtsp-server/rtsp-token.c:
2572 token: Validate args for gst_rtsp_token_is_allowed
2573 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
2575 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2577 * gst/rtsp-server/rtsp-token.c:
2578 token: Fix bug when creating empty token
2579 We always want to have a valid GstStructure in the token.
2580 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2582 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2584 * gst/rtsp-server/rtsp-thread-pool.c:
2585 thread-pool: avoid race in shutdown
2586 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
2587 don't actually stop the mainloop ever. Solve this race by adding an idle source
2588 to the mainloop that calls the _quit. This way we immediately exit the mainloop
2589 if quit was called before we started it.
2591 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2593 * tests/check/Makefile.am:
2594 * tests/check/gst/permissions.c:
2595 tests: Add unit tests for permissions
2596 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
2598 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2600 * tests/check/gst/mediafactory.c:
2601 tests: Test mediafactory permissions
2602 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2604 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2606 * gst/rtsp-server/rtsp-permissions.c:
2607 permissions: Fix refcounting when adding/removing roles
2608 Previously a role that was removed was unreffed twice, and when
2609 replacing an existing role the replaced role was freed while still being
2610 referenced. Both bugs are now fixed.
2611 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2613 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2615 * tests/check/gst/media.c:
2616 * tests/check/gst/mediafactory.c:
2617 * tests/check/gst/rtspserver.c:
2618 tests: Check gst_rtsp_url_parse return value
2619 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2621 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
2624 Automatic update of common submodule
2625 From 865aa20 to dbedaa0
2627 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
2629 * gst/rtsp-server/rtsp-server.c:
2630 rtsp-server: Fix socket leak
2631 https://bugzilla.gnome.org/show_bug.cgi?id=710088
2633 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
2635 * gst/rtsp-server/rtsp-session-pool.c:
2636 rtsp-session-pool: Make sure session IDs are properly URI-escaped
2637 https://bugzilla.gnome.org/show_bug.cgi?id=643812
2639 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2641 * examples/.gitignore:
2642 * examples/test-video.c:
2643 examples: fix compilation when WITH_AUTH is defined
2644 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2646 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
2649 gitignore: Add new test binary
2651 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
2653 * tests/check/Makefile.am:
2654 * tests/check/gst/threadpool.c:
2655 thread-pool: Add unit test for the thread pools
2656 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2658 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2660 * gst/rtsp-server/rtsp-thread-pool.c:
2661 thread-pool: Fix thread leak when reusing threads
2662 https://bugzilla.gnome.org/show_bug.cgi?id=709730
2664 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
2666 * gst/rtsp-server/rtsp-server.c:
2667 * tests/check/gst/rtspserver.c:
2668 tests: fixed racy behavior in rtspserver tests
2669 https://bugzilla.gnome.org/show_bug.cgi?id=710078
2671 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2673 * tests/check/gst/addresspool.c:
2674 tests: Improve address pool unit tests
2675 Add a range with mixed IPV4 and IPV6 addresses to pool.
2676 Get an IPV4 address from an IPV6-only pool.
2677 Get an IPV6 address from an IPV4-only pool.
2678 Reserve a IPV6 address from an IPV4-only pool.
2679 Check for unicast addresses in multicast-only pool.
2680 Check for unicast addresses in uni-/multicast-mixed pool.
2681 https://bugzilla.gnome.org/show_bug.cgi?id=710128
2683 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2685 * gst/rtsp-server/rtsp-client.c:
2686 client: append query string in PAUSE/PLAY/TEARDOWN as well
2688 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
2690 * gst/rtsp-server/rtsp-client.c:
2691 client: Add query to control path
2692 If the SETUP url contains a query it must be appended to the control
2693 path so that it matches any already created stream in the media. The
2694 query will also be appended to the session media path.
2696 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2698 * gst/rtsp-server/rtsp-media.c:
2699 rtsp-media: remove old line
2701 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
2703 * gst/rtsp-server/rtsp-stream.c:
2704 stream: Correct control comparison
2705 https://bugzilla.gnome.org/show_bug.cgi?id=709176
2707 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2709 * gst/rtsp-server/rtsp-media.c:
2710 media: Check dynamically if the pipeline supports seeking
2711 We should not depend on whether or not the pipeline state change
2712 returned NO_PREROLL or not. A media could dynamically change its
2713 element and switch from seekable to non seekable so it's best to test
2714 the seekable nature of the pipeline dynamically when we try to do a seek.
2716 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2718 * gst/rtsp-server/rtsp-media.c:
2719 media: Return FALSE if seeking is not supported
2721 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2723 * gst/rtsp-server/rtsp-media.c:
2724 rtsp-media: don't seek accurate by default
2725 Accurate seeking is perhaps a little overkill in the most common situation and
2726 causes some formats (mp3) over slow media to seek extremely slowly.
2728 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
2730 * tests/check/gst/rtspserver.c:
2731 tests: fix unit test
2732 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
2734 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
2736 * gst/rtsp-server/rtsp-client.c:
2737 client: Reply 400 if media cannot be constructed
2738 Reply 400 Bad Request instead of 503 Service Unavailable if media
2739 cannot be constructed in SETUP.
2740 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
2742 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
2744 * gst/rtsp-server/rtsp-client.c:
2745 client: Send setup reply once only
2746 If find_media() failed in handle_setup_request() two replies was sent.
2747 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
2749 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
2752 Automatic update of common submodule
2753 From 6b03ba7 to 865aa20
2755 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
2757 * gst/rtsp-server/rtsp-server.c:
2758 server: Emit client-connected signal earlier
2759 Emit client-connected before the client ref is given to a GSource,
2760 otherwise client-connected can be emitted after the client object has
2763 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
2765 * gst/rtsp-server/rtsp-address-pool.c:
2766 * gst/rtsp-server/rtsp-address-pool.h:
2767 * gst/rtsp-server/rtsp-stream.c:
2768 * tests/check/gst/addresspool.c:
2769 addresspool: return reason of failure
2770 Let gst_rtsp_address_pool_reserve_address() return the reason why
2771 the address could not be reserved.
2772 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
2774 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
2777 autogen.sh: Sync behaviour with other GStreamer modules
2778 Allows building from outside of tree amongst other things
2780 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
2783 Automatic update of common submodule
2784 From b613661 to 6b03ba7
2786 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
2789 Automatic update of common submodule
2790 From 74a6857 to b613661
2792 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
2795 Automatic update of common submodule
2796 From 01a7a46 to 74a6857
2798 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
2800 * gst/rtsp-server/rtsp-client.c:
2801 client: Do not read beyond end of path string
2802 If the setup was done without a control url, make sure we don't try to read the
2803 non-existing control string and crash.
2805 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2807 * gst/rtsp-server/rtsp-client.c:
2808 client: Fix RTPInfo header
2809 Refactor the method to make the content_base.
2810 Use the content-base and the control url to construct the RTPInfo
2813 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2815 * gst/rtsp-server/rtsp-client.c:
2816 client: map url to path only in describe
2817 Only map the request url to a path in the DESCRIBE method. The SDP then
2818 contains the base and control urls that should be used to SETUP/PAUSE/
2819 PLAY/TEARDOWN the media.
2821 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2823 * gst/rtsp-server/rtsp-client.c:
2824 Revert "client: map URL to path in requests"
2825 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
2826 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
2827 contains the base and control urls which are used in the SETUP, PLAY,
2828 PAUSE and TEARDOWN requests.
2830 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2832 * gst/rtsp-server/rtsp-client.c:
2833 client: map URL to path in requests
2835 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2837 * gst/rtsp-server/rtsp-client.c:
2838 * gst/rtsp-server/rtsp-mount-points.c:
2839 * gst/rtsp-server/rtsp-mount-points.h:
2840 mount-points: make vmethod to make path from uri
2841 Make a vmethod to transform an url into a path. The path is then used to lookup
2842 the factory. This makes it possible to also use other bits of the url, such as
2843 the query parameters, to locate the factory.
2845 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
2847 * gst/rtsp-server/rtsp-thread-pool.c:
2848 * gst/rtsp-server/rtsp-thread-pool.h:
2849 thread-pool: Add cleanup to wait for the threadpool to finish
2850 Also fix race condition if two threads are asking for the first
2851 thread from the thread pool at once. This would case two internal
2852 GThreadPools to be created.
2853 https://bugzilla.gnome.org/show_bug.cgi?id=707753
2855 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
2857 * gst/rtsp-server/rtsp-client.c:
2858 * tests/check/gst/client.c:
2859 client: free threadpool
2860 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2862 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
2864 * tests/check/gst/mountpoints.c:
2865 mountpoints tests: unref matched factories
2866 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2868 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
2870 * tests/check/gst/media.c:
2871 media tests: unref thread pool and caps
2872 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2874 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
2876 * gst/rtsp-server/rtsp-auth.c:
2877 * gst/rtsp-server/rtsp-media-factory.c:
2878 * gst/rtsp-server/rtsp-media.c:
2879 auth, media, media-factory: unref permissions
2880 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2882 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2884 * examples/Makefile.am:
2885 Makefile: add rule for appsrc example
2887 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2889 * examples/test-appsrc.c:
2890 tests: add appsrc example
2891 Add an example on how to use appsrc to feed the server pipeline with data.
2893 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
2895 * gst/rtsp-server/rtsp-client.c:
2896 rtsp-client: remove query part from content-base string
2897 Make sure that after the control url has been resolved, it's
2898 not a part of the query-string.
2899 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
2901 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2903 * gst/rtsp-server/rtsp-client.c:
2904 client: don't check url in response
2905 There is no url or method in the response to check
2907 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2909 * gst/rtsp-server/rtsp-client.c:
2910 * gst/rtsp-server/rtsp-client.h:
2911 Add handle-response signal for when we receive a GET_PARAMETER response
2913 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2915 * gst/rtsp-server/rtsp-server.c:
2916 Fix gst_rtsp_server_client_filter, using wrong variable type
2918 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
2920 * gst/rtsp-server/rtsp-media-factory-uri.c:
2921 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
2922 For AAC we need to check for framed=true instead of parsed=true.
2923 https://bugzilla.gnome.org/show_bug.cgi?id=701384
2925 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2927 * gst/rtsp-server/rtsp-stream.c:
2928 stream: optimize pipeline for protocols
2929 When TCP is not an allowed protocol for the stream, avoid creating the
2930 appsrc/appsink/queue and tee elements.
2932 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2934 * gst/rtsp-server/rtsp-media.c:
2935 media: set protocols on streams
2937 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2939 * gst/rtsp-server/rtsp-client.c:
2940 client: use protocols supported by stream
2942 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2944 * gst/rtsp-server/rtsp-media-factory.c:
2945 * gst/rtsp-server/rtsp-media.c:
2946 * gst/rtsp-server/rtsp-stream.c:
2947 media-factory: allow all protocols
2949 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2951 * gst/rtsp-server/rtsp-media.c:
2952 media: configure protocols in new streams
2954 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2956 * gst/rtsp-server/rtsp-stream.c:
2957 * gst/rtsp-server/rtsp-stream.h:
2958 stream: add protocols property
2960 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2962 * gst/rtsp-server/rtsp-media.c:
2963 rtsp-media: send state in "new-state" signal
2964 https://bugzilla.gnome.org/show_bug.cgi?id=705110
2966 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
2969 build: add subdir-objects to AM_INIT_AUTOMAKE
2970 Fixes warnings with automake 1.14
2971 https://bugzilla.gnome.org/show_bug.cgi?id=705350
2973 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2975 * docs/libs/gst-rtsp-server-sections.txt:
2976 * gst/rtsp-server/rtsp-client.c:
2977 * gst/rtsp-server/rtsp-server.c:
2978 * gst/rtsp-server/rtsp-server.h:
2979 server: add method to iterate clients of server
2981 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2983 * gst/rtsp-server/rtsp-media.c:
2984 * gst/rtsp-server/rtsp-media.h:
2985 Add vmethod for rtsp-media subclass to access rtpbin
2987 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2989 * gst/rtsp-server/rtsp-client.h:
2990 small documentation fix
2992 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2994 * gst/rtsp-server/rtsp-client.c:
2995 Do not take range header if range is invalid
2997 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2999 * docs/libs/gst-rtsp-server-sections.txt:
3000 * gst/rtsp-server/rtsp-media.c:
3001 media: add docs for new method
3003 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3005 * gst/rtsp-server/rtsp-media.c:
3006 * gst/rtsp-server/rtsp-media.h:
3007 Add API to rtsp-media set the pipeline's state
3009 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3011 * gst/rtsp-server/rtsp-media.c:
3012 Update current position/duration when gst_rtsp_media_get_range_string is called
3014 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3016 * examples/test-cgroups.c:
3017 tests: add some more docs
3019 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3021 * examples/test-cgroups.c:
3022 * gst/rtsp-server/Makefile.am:
3023 * gst/rtsp-server/rtsp-auth.c:
3024 * gst/rtsp-server/rtsp-auth.h:
3025 * gst/rtsp-server/rtsp-client.c:
3026 * gst/rtsp-server/rtsp-client.h:
3027 * gst/rtsp-server/rtsp-context.c:
3028 * gst/rtsp-server/rtsp-context.h:
3029 * gst/rtsp-server/rtsp-params.c:
3030 * gst/rtsp-server/rtsp-params.h:
3031 * gst/rtsp-server/rtsp-server.c:
3032 * gst/rtsp-server/rtsp-thread-pool.c:
3033 * gst/rtsp-server/rtsp-thread-pool.h:
3034 * tests/check/gst/client.c:
3035 ClientState -> Context
3036 Rename the clientstate to context and put the code in a separate file.
3038 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3040 * examples/test-auth.c:
3041 * gst/rtsp-server/rtsp-auth.c:
3042 * gst/rtsp-server/rtsp-auth.h:
3043 auth: add support for default token
3044 The default token is used when the user is not authenticated and can be used to
3045 give minimal permissions.
3047 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3049 * examples/test-auth.c:
3050 * gst/rtsp-server/rtsp-auth.c:
3051 auth: use defines when possible
3053 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3055 * gst/rtsp-server/rtsp-address-pool.c:
3056 address-pool: improve docs
3058 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3060 * gst/rtsp-server/rtsp-permissions.c:
3061 permissions: add the role to the copy
3063 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
3065 * gst/rtsp-server/rtsp-permissions.c:
3066 permissions: Also copy the roles
3068 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
3070 * gst/rtsp-server/rtsp-permissions.c:
3071 permissions: Make it build
3073 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3075 * gst/rtsp-server/rtsp-address-pool.h:
3078 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3080 * docs/libs/gst-rtsp-server-sections.txt:
3081 * gst/rtsp-server/rtsp-auth.c:
3082 * gst/rtsp-server/rtsp-auth.h:
3083 * gst/rtsp-server/rtsp-media.c:
3084 * gst/rtsp-server/rtsp-session-media.c:
3085 * gst/rtsp-server/rtsp-stream-transport.c:
3086 * gst/rtsp-server/rtsp-stream-transport.h:
3087 * gst/rtsp-server/rtsp-stream.c:
3088 * tests/check/gst/client.c:
3091 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3093 * docs/libs/gst-rtsp-server-sections.txt:
3094 * gst/rtsp-server/rtsp-address-pool.c:
3095 * gst/rtsp-server/rtsp-address-pool.h:
3096 * tests/check/gst/addresspool.c:
3097 * tests/check/gst/rtspserver.c:
3098 address-pool: cleanups
3099 Remove redundant method, improve docs.
3101 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3103 * docs/libs/gst-rtsp-server-sections.txt:
3104 * gst/rtsp-server/rtsp-auth.h:
3105 * gst/rtsp-server/rtsp-permissions.c:
3106 * gst/rtsp-server/rtsp-permissions.h:
3107 * gst/rtsp-server/rtsp-token.c:
3110 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3112 * gst/rtsp-server/rtsp-permissions.c:
3113 permissions: implement _remove_role
3115 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3117 * gst/rtsp-server/rtsp-permissions.c:
3118 permissions: update docs
3120 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3122 * tests/check/gst/client.c:
3123 tests: simplify tests
3124 Client settings are now disabled by default so we don't need an auth
3125 module to disable them.
3127 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3129 * gst/rtsp-server/rtsp-auth.c:
3130 auth: add default authorizations
3131 When no auth module is specified, use our table of defaults to look up the
3132 default value of the check instead of always allowing everything. This was
3133 we can disallow client settings by default.
3135 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3138 README: update readme
3140 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3142 * gst/rtsp-server/rtsp-thread-pool.c:
3143 * gst/rtsp-server/rtsp-thread-pool.h:
3144 thread-pool: add more docs
3146 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3148 * gst/rtsp-server/rtsp-thread-pool.c:
3149 * gst/rtsp-server/rtsp-thread-pool.h:
3150 thread-pool: fix race in thread reuse
3151 If we try to reuse a thread right after we made it stop, we end up using a
3152 stopped thread. Catch this case and only reuse threads that are not stopping.
3154 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3156 * gst/rtsp-server/rtsp-server.c:
3157 server: add small debug
3159 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3161 * tests/check/gst/client.c:
3163 Add some permissions to media so we can use the auth and enable
3166 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3168 * gst/rtsp-server/rtsp-client.c:
3169 client: support pushed context in handle_request
3170 If we already have a pushed state, reuse it and add our own things. This makes
3171 it easier to write tests.
3173 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3175 * gst/rtsp-server/rtsp-auth.c:
3176 auth: don't auth on methods
3177 Don't authorize on methods anymore but on the resources that we
3178 try to access, this is more flexible.
3179 Move the authorization checks to where they are needed and let the
3180 check return the response on error.
3182 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3184 * gst/rtsp-server/rtsp-mount-points.c:
3185 mount-points: add some debug
3187 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3189 * tests/check/gst/client.c:
3190 tests: almost fix test
3192 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3194 * gst/rtsp-server/rtsp-auth.c:
3195 * gst/rtsp-server/rtsp-auth.h:
3196 * gst/rtsp-server/rtsp-client.c:
3197 * gst/rtsp-server/rtsp-client.h:
3198 * gst/rtsp-server/rtsp-server.c:
3199 * gst/rtsp-server/rtsp-server.h:
3200 auth: let the auth module check client_settings
3201 Let the auth module decide if client settings are allowed for the
3204 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3206 * gst/rtsp-server/rtsp-token.c:
3207 * gst/rtsp-server/rtsp-token.h:
3208 token: add method to check boolean permission
3210 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3212 * examples/test-auth.c:
3213 * examples/test-cgroups.c:
3214 * gst/rtsp-server/rtsp-token.c:
3215 * gst/rtsp-server/rtsp-token.h:
3216 token: simplify token constructor
3217 Use variable arguments to make easier API.
3219 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3221 * examples/test-auth.c:
3222 * examples/test-cgroups.c:
3223 * gst/rtsp-server/rtsp-media-factory.c:
3224 * gst/rtsp-server/rtsp-media-factory.h:
3225 media-factory: add convenience API for factory
3227 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3229 * examples/test-auth.c:
3230 * examples/test-cgroups.c:
3231 * gst/rtsp-server/rtsp-permissions.c:
3232 * gst/rtsp-server/rtsp-permissions.h:
3233 permissions: simplify API a little
3234 Avoid passing GstStructure in the add_role method, use varargs instead
3235 to construct the structure behind the scenes. We can then also use the
3236 structure name as the role and simplify some more logic.
3238 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3240 * gst/rtsp-server/rtsp-auth.c:
3243 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3245 * gst/rtsp-server/rtsp-auth.c:
3246 * gst/rtsp-server/rtsp-auth.h:
3247 * gst/rtsp-server/rtsp-client.c:
3248 auth: handle unauthorized response
3249 Move handling of the unauthorized response to the auth module, it can add
3250 the appropriate headers to request authorization for the required method
3251 much better than the client.
3253 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3255 * gst/rtsp-server/rtsp-client.c:
3256 * gst/rtsp-server/rtsp-client.h:
3257 client: allow for sending any message, not only requests
3258 Change the _send_request() method to _send_message() so that we
3259 can both send requests and replies.
3261 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3263 * docs/libs/gst-rtsp-server-sections.txt:
3264 * gst/rtsp-server/rtsp-server.h:
3267 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3269 * examples/test-video.c:
3270 * gst/rtsp-server/rtsp-auth.c:
3271 * gst/rtsp-server/rtsp-auth.h:
3272 * gst/rtsp-server/rtsp-server.c:
3273 * gst/rtsp-server/rtsp-server.h:
3274 auth: move TLS handling to auth module
3275 Remove the TLS settings on the server and move it to the auth module because
3276 that is where security related bits go.
3278 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3280 * gst/rtsp-server/rtsp-client.c:
3281 * gst/rtsp-server/rtsp-client.h:
3282 client: add state push/pop
3284 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3286 * gst/rtsp-server/rtsp-client.c:
3287 * gst/rtsp-server/rtsp-client.h:
3288 client: add connection to state
3290 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3292 * gst/rtsp-server/rtsp-mount-points.c:
3293 mount-points: fix debug
3295 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3297 * tests/check/gst/media.c:
3298 tests: fix media test
3300 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3302 * gst/rtsp-server/rtsp-thread-pool.c:
3303 thread-pool: we don't require a state
3305 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3307 * gst/rtsp-server/rtsp-server.c:
3308 server: let context ref the server
3309 So that we don't risk losing the server object early anc crash.
3311 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3313 * tests/check/gst/client.c:
3314 tests: fix client test
3316 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3319 * docs/libs/gst-rtsp-server-docs.sgml:
3320 * docs/libs/gst-rtsp-server-sections.txt:
3321 * gst/rtsp-server/rtsp-address-pool.c:
3322 * gst/rtsp-server/rtsp-auth.c:
3323 * gst/rtsp-server/rtsp-client.c:
3324 * gst/rtsp-server/rtsp-client.h:
3325 * gst/rtsp-server/rtsp-media-factory-uri.c:
3326 * gst/rtsp-server/rtsp-media-factory.c:
3327 * gst/rtsp-server/rtsp-media-factory.h:
3328 * gst/rtsp-server/rtsp-media.c:
3329 * gst/rtsp-server/rtsp-mount-points.c:
3330 * gst/rtsp-server/rtsp-params.c:
3331 * gst/rtsp-server/rtsp-permissions.c:
3332 * gst/rtsp-server/rtsp-sdp.c:
3333 * gst/rtsp-server/rtsp-server.c:
3334 * gst/rtsp-server/rtsp-server.h:
3335 * gst/rtsp-server/rtsp-session-media.c:
3336 * gst/rtsp-server/rtsp-session-pool.c:
3337 * gst/rtsp-server/rtsp-session.c:
3338 * gst/rtsp-server/rtsp-stream-transport.c:
3339 * gst/rtsp-server/rtsp-stream.c:
3340 * gst/rtsp-server/rtsp-thread-pool.c:
3341 * gst/rtsp-server/rtsp-token.c:
3344 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3346 * gst/rtsp-server/rtsp-session-pool.c:
3347 * gst/rtsp-server/rtsp-session-pool.h:
3348 session-pool: make vmethod to create a session
3349 Make a vmethod to create a sessions so that subclasses can create
3350 custom session objects
3352 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3354 * gst/rtsp-server/rtsp-auth.c:
3355 * gst/rtsp-server/rtsp-media-factory.h:
3356 * gst/rtsp-server/rtsp-media.h:
3357 * gst/rtsp-server/rtsp-mount-points.h:
3358 * gst/rtsp-server/rtsp-session-pool.h:
3359 * gst/rtsp-server/rtsp-stream.h:
3362 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3364 * docs/libs/gst-rtsp-server-docs.sgml:
3365 * docs/libs/gst-rtsp-server-sections.txt:
3366 * gst/rtsp-server/rtsp-address-pool.c:
3367 * gst/rtsp-server/rtsp-address-pool.h:
3368 * gst/rtsp-server/rtsp-auth.c:
3369 * gst/rtsp-server/rtsp-client.h:
3370 * gst/rtsp-server/rtsp-media-factory.h:
3371 * gst/rtsp-server/rtsp-media.c:
3372 * gst/rtsp-server/rtsp-media.h:
3373 * gst/rtsp-server/rtsp-permissions.c:
3374 * gst/rtsp-server/rtsp-permissions.h:
3375 * gst/rtsp-server/rtsp-server.h:
3376 * gst/rtsp-server/rtsp-session-media.c:
3377 * gst/rtsp-server/rtsp-session-media.h:
3378 * gst/rtsp-server/rtsp-session-pool.h:
3379 * gst/rtsp-server/rtsp-session.h:
3380 * gst/rtsp-server/rtsp-stream-transport.h:
3381 * gst/rtsp-server/rtsp-stream.c:
3382 * gst/rtsp-server/rtsp-thread-pool.h:
3385 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3388 * examples/Makefile.am:
3389 configure: compile cgroup example conditionally
3390 Only compile the cgroup example when we have libcgroup
3392 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3395 * examples/Makefile.am:
3396 * examples/test-cgroups.c:
3397 examples: add cgroups example
3399 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3401 * tests/check/gst/rtspserver.c:
3402 tests: fix compilation
3404 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3406 * gst/rtsp-server/rtsp-thread-pool.c:
3407 thread-pool: fix vmethod invocation
3409 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3411 * gst/rtsp-server/rtsp-thread-pool.c:
3412 * gst/rtsp-server/rtsp-thread-pool.h:
3413 thread-pool: store thread type in thread
3415 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3417 * gst/rtsp-server/rtsp-client.c:
3418 client: pass thread from pool to media _prepare
3419 Get a thread from the configured threadpool and pass it to the prepare method of
3422 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3424 * gst/rtsp-server/rtsp-media.c:
3425 * gst/rtsp-server/rtsp-media.h:
3426 media: Accept a thread in _prepare
3427 Remove out own threadpool handling and use the provided thread and
3428 maincontext for the bus messages and the state changes.
3430 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3432 * gst/rtsp-server/rtsp-server.c:
3433 server: configure client thread pool
3435 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3437 * gst/rtsp-server/rtsp-client.c:
3438 * gst/rtsp-server/rtsp-client.h:
3439 client: add method to configure thread pool
3441 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3443 * gst/rtsp-server/rtsp-client.h:
3444 * gst/rtsp-server/rtsp-server.c:
3445 * gst/rtsp-server/rtsp-server.h:
3446 server: use thread pool
3447 Use the thread pool instead of doing our own thing.
3449 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3451 * gst/rtsp-server/Makefile.am:
3452 * gst/rtsp-server/rtsp-thread-pool.c:
3453 * gst/rtsp-server/rtsp-thread-pool.h:
3454 thread-pool: add object to manage threads
3455 Add an object to manage the client and media threads.
3457 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3459 * gst/rtsp-server/rtsp-auth.c:
3460 auth: debug authorization check
3462 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3464 * gst/rtsp-server/rtsp-media.c:
3465 media: start media pipeline in context
3466 Start the media pipeline in the provided context (or our default one
3467 when NULL). This makes sure that we run the bus thread in this context and that
3468 all media threads are children of this context.
3470 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3472 * gst/rtsp-server/rtsp-media-factory.c:
3473 factory: pass permissions to media by default
3475 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3477 * examples/test-auth.c:
3478 test: add permissions to auth test
3479 Ass some permissions to the media factory in the test.
3481 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3483 * gst/rtsp-server/rtsp-auth.c:
3484 * gst/rtsp-server/rtsp-auth.h:
3485 * gst/rtsp-server/rtsp-client.c:
3486 auth: simplify auth checks
3487 Remove client from methods, it's now in the state
3488 Perform the check specified by the string, use the information from the
3489 thread local context.
3491 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3493 * gst/rtsp-server/rtsp-client.c:
3494 * gst/rtsp-server/rtsp-client.h:
3495 client: add state to current thread
3496 Add the client to the ClientState object.
3497 Place the ClientState on the current thread.
3499 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3501 * gst/rtsp-server/rtsp-media-factory.c:
3502 * gst/rtsp-server/rtsp-media-factory.h:
3503 * gst/rtsp-server/rtsp-media.c:
3504 * gst/rtsp-server/rtsp-media.h:
3505 media: make it possible to set permissions
3506 Make it possible to set permissions on media and media factory objects
3508 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3510 * gst/rtsp-server/Makefile.am:
3511 * gst/rtsp-server/rtsp-permissions.c:
3512 * gst/rtsp-server/rtsp-permissions.h:
3513 permissions: add permissions object
3514 Add a mini object to store permissions based on a role.
3516 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3518 * examples/test-auth.c:
3519 * gst/rtsp-server/rtsp-auth.c:
3520 * gst/rtsp-server/rtsp-auth.h:
3521 * gst/rtsp-server/rtsp-client.c:
3522 auth: add auth checks
3523 Add an enum with auth checks and implement the checks in the auth object.
3524 Perform the checks from the client.
3526 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3528 * examples/test-auth.c:
3529 * gst/rtsp-server/rtsp-auth.c:
3530 * gst/rtsp-server/rtsp-auth.h:
3531 * gst/rtsp-server/rtsp-client.h:
3532 auth: use the token after authentication
3533 After we authenticated a user, keep the Token around in the state.
3535 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3537 * gst/rtsp-server/rtsp-client.c:
3538 * gst/rtsp-server/rtsp-media.c:
3539 * gst/rtsp-server/rtsp-media.h:
3540 * tests/check/gst/media.c:
3541 media: add optional context for bus messages
3542 Add an optional mainloop to _prepare that will handle the bus messages instead
3543 of always using the shared mainloop.
3545 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3547 * gst/rtsp-server/Makefile.am:
3548 * gst/rtsp-server/rtsp-token.c:
3549 * gst/rtsp-server/rtsp-token.h:
3550 token: add authorization token
3551 Add a simply miniobject that contains the authorizations. The object contains a
3552 GstStructure that hold all authorization fields. When a user is authenticated,
3553 the auth module will create a Token for the user. The token is then used to
3554 check what operations the user is allowed to do and various other configuration
3557 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3559 * examples/test-auth.c:
3560 * gst/rtsp-server/rtsp-auth.c:
3561 * gst/rtsp-server/rtsp-auth.h:
3562 * gst/rtsp-server/rtsp-client.c:
3563 * gst/rtsp-server/rtsp-client.h:
3564 * gst/rtsp-server/rtsp-media-factory.c:
3565 * gst/rtsp-server/rtsp-media-factory.h:
3566 * gst/rtsp-server/rtsp-media.c:
3567 * gst/rtsp-server/rtsp-media.h:
3568 auth: remove auth from media and factory
3569 Remove the auth object from media and factory. We want to have the RTSPClient
3570 authenticate and authorize resources, there is no need to place another auth
3571 manager on the media/factory.
3573 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3575 * examples/test-auth.c:
3576 * gst/rtsp-server/rtsp-auth.c:
3577 * gst/rtsp-server/rtsp-auth.h:
3578 * gst/rtsp-server/rtsp-client.h:
3579 auth: add support for multiple basic auth tokens
3580 Make it possible to add multiple basic authorisation tokens to one authorization
3581 object. Associate with each token an authorization group that will define what
3582 capabilities are allowed.
3584 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3586 * gst/rtsp-server/rtsp-client.c:
3587 client: error out on non-aggregate control
3588 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
3590 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3592 * gst/rtsp-server/rtsp-client.c:
3593 client: rework setup request a little
3594 Cache the media in DESCRIBE based on the longest matching path with the uri
3595 that we can find in the mount points.
3596 Rework the setup request a little to get the media from the session or from
3597 the longest matching path, this way we can derive the control string as
3598 everything after the path instead of hardcoding it.
3599 Find the stream based on the control string and only open a session when all
3602 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3604 * gst/rtsp-server/rtsp-media.c:
3605 * gst/rtsp-server/rtsp-media.h:
3606 media: add method to find a stream by control url
3608 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3610 * gst/rtsp-server/rtsp-stream.c:
3611 * gst/rtsp-server/rtsp-stream.h:
3612 stream: add method to check control url of stream
3614 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3616 * gst/rtsp-server/rtsp-client.c:
3617 * gst/rtsp-server/rtsp-session-media.c:
3618 * gst/rtsp-server/rtsp-session-media.h:
3619 * gst/rtsp-server/rtsp-session.c:
3620 * gst/rtsp-server/rtsp-session.h:
3621 session: use path matching for session media
3622 Use a path string instead of a uri to lookup session media in the sessions. Also
3623 use path matching to find the largest possible path that matches.
3625 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3627 * gst/rtsp-server/rtsp-client.c:
3628 * gst/rtsp-server/rtsp-mount-points.c:
3629 * gst/rtsp-server/rtsp-mount-points.h:
3630 * tests/check/gst/mountpoints.c:
3631 mount-points: remove useless vmethod
3632 Making lookups in the mount points should not be done with a URL, if there is a
3633 mapping to be done from URL to mount points, we'll need to do it somewhere
3636 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3638 * gst/rtsp-server/rtsp-mount-points.c:
3639 * gst/rtsp-server/rtsp-mount-points.h:
3640 * tests/check/gst/mountpoints.c:
3641 mount-points: improve mount point searching
3642 Use a GSequence to keep track of the mount points.
3643 Match a URL to the longest matching registered mount point. This should be the
3644 URL to perform aggreagate control and the remainder is the stream specific
3646 Add some unit tests for this.
3648 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
3650 * gst/rtsp-server/Makefile.am:
3651 rtsp-server: Allow building of static library
3653 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3655 * tests/check/gst/mediafactory.c:
3656 tests: fix compilation
3658 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3660 * gst/rtsp-server/rtsp-sdp.c:
3661 sdp: get control string from stream
3662 Use the control string as configured in the stream.
3664 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3666 * gst/rtsp-server/rtsp-stream.c:
3667 * gst/rtsp-server/rtsp-stream.h:
3668 stream: add methods and property to set control string
3670 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3672 * gst/rtsp-server/rtsp-client.c:
3674 Rename variables for clarity
3675 Keep media in state when we can
3677 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3679 * gst/rtsp-server/rtsp-client.c:
3680 * gst/rtsp-server/rtsp-stream.c:
3681 * gst/rtsp-server/rtsp-stream.h:
3682 stream: add more support for IPv6
3683 Rename _get_address to _get_multicast_address in GstRTSPStream to
3684 make it clear that this function only deals with multicast.
3685 Make it possible to have both an IPv4 and IPv6 multicast address on
3686 a stream. Give the client an IPv4 or IPv6 address depending on the
3687 address it used to connect to the server.
3688 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
3690 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3692 * gst/rtsp-server/rtsp-client.c:
3695 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3697 * gst/rtsp-server/rtsp-stream.c:
3698 stream: handle failed port allocation
3699 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
3700 can't allocate any family at all. Also keep track of what port families we
3702 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
3704 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3706 * gst/rtsp-server/rtsp-stream.c:
3707 stream: improve docs
3709 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3711 * gst/rtsp-server/rtsp-stream-transport.c:
3712 stream-transport: remove old if 0 block
3714 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
3716 * tests/check/gst/client.c:
3718 gst_rtsp_client_get_uri() has been removed
3719 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
3721 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3723 * gst/rtsp-server/rtsp-client.c:
3724 * gst/rtsp-server/rtsp-client.h:
3725 client: add method to filter managed sessions
3726 Add a method to filter the sessions managed by this client connection.
3727 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
3729 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3731 * gst/rtsp-server/rtsp-client.c:
3732 * gst/rtsp-server/rtsp-client.h:
3733 client: remove _get_uri() method
3734 Remove the get_uri() method on the client. A client has no uri, the uri
3735 property is an internal property to manage the last cached media for
3738 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3740 * gst/rtsp-server/rtsp-media-factory.h:
3741 media-factory: fix typo
3743 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
3745 * gst/rtsp-server/rtsp-media.c:
3746 rtsp-media: Do not leak the query in default_query_stop
3747 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
3749 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3751 * gst/rtsp-server/rtsp-media.c:
3752 media: don't unlock when conversion fails
3753 Don't unlock the state lock when conversion fails because it was not locked.
3755 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3757 * gst/rtsp-server/rtsp-media.c:
3758 * gst/rtsp-server/rtsp-media.h:
3759 Add query_position and query_stop vmethods to rtsp-media
3761 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3763 * gst/rtsp-server/rtsp-media.c:
3764 Fix typo in property install for rtsp-media's time-provider
3766 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3768 * gst/rtsp-server/rtsp-client.c:
3769 * gst/rtsp-server/rtsp-client.h:
3770 client: clean some variables
3771 Clean some variables and add some guards to _send_request()
3773 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3775 * gst/rtsp-server/rtsp-client.c:
3776 * gst/rtsp-server/rtsp-client.h:
3777 Add gst_rtsp_client_send_request API
3778 This makes it possible to send arbitrary messages to a client, such as
3779 SET_PARAMETER or GET_PARAMETER
3781 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3783 * gst/rtsp-server/rtsp-media.c:
3784 * gst/rtsp-server/rtsp-media.h:
3785 media: add _get_element() method
3786 Add method to get the element used when creating the media.
3787 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
3789 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3791 * gst/rtsp-server/rtsp-media.c:
3794 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3796 * gst/rtsp-server/rtsp-stream.c:
3797 * gst/rtsp-server/rtsp-stream.h:
3798 stream: allow access to the rtp session
3799 https://bugzilla.gnome.org/show_bug.cgi?id=703004
3801 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
3803 * gst/rtsp-server/rtsp-stream.c:
3804 * gst/rtsp-server/rtsp-stream.h:
3805 dscp qos support in gst-rtsp-stream
3806 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
3808 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3810 * tests/check/gst/rtspserver.c:
3812 Actually do what the comment says. Also keep the old code around, not sure what
3813 should happen when you get a 454 from a TEARDOWN, does it close the connection?
3814 it currently doesn't.
3816 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3818 * gst/rtsp-server/rtsp-client.c:
3819 client: also watch newly created session
3820 When we newly created a session, start watching it immediately instead of
3821 on the next request.
3823 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
3825 * tests/check/gst/client.c:
3826 tests: add unit test for new-session
3827 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
3829 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3831 * gst/rtsp-server/rtsp-client.c:
3832 client: emit new-session when new session is created
3833 Only emit new-session when we created a new session for a client, not when a
3834 client picked up a previous session.
3835 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
3837 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
3839 * gst/rtsp-server/rtsp-client.c:
3840 client: handle asterisk as path in requests
3841 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
3843 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3845 * gst/rtsp-server/rtsp-media.c:
3846 media: handle segment query format mismatch
3847 It's possible that the segment query returns with a different format than what
3848 we asked for, handle this case also.
3850 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
3852 * gst/rtsp-server/rtsp-media.c:
3853 media: use segment stop in collect_media_stats
3854 Use segment stop instead of duration as range end point.
3855 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
3857 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3859 * gst/rtsp-server/rtsp-media.c:
3860 * tests/check/gst/media.c:
3861 rtsp-media: Do not leak the element in take_pipeline
3862 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
3864 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
3866 * gst/rtsp-server/rtsp-client.c:
3867 * gst/rtsp-server/rtsp-client.h:
3868 rtsp-client: Make configure_client_transport virtual
3869 This patch makes configure_client_transport virtual. The functionality is
3870 needed to handle some weird clients sending multicast transport settings as url
3872 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
3874 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3876 * gst/rtsp-server/rtsp-client.c:
3877 * gst/rtsp-server/rtsp-client.h:
3878 rtsp-client: Make param_set and param_get virtual
3879 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
3881 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
3883 * gst/rtsp-server/rtsp-client.c:
3884 * gst/rtsp-server/rtsp-media.c:
3885 * gst/rtsp-server/rtsp-media.h:
3886 media: convert_range replaces get_range_times
3887 get_range_times worked for handling UTC ranges for seeks, but we also
3888 need to convert back from NPT to the requested unit in
3889 get_range_string. convert_range is now used for both.
3890 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
3892 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3894 * gst/rtsp-server/rtsp-client.c:
3895 * gst/rtsp-server/rtsp-sdp.c:
3896 * gst/rtsp-server/rtsp-sdp.h:
3897 sdp: cleanup sdp info
3898 We don't need to pass the proto, we can more easily check a boolean.
3899 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
3901 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
3903 * gst/rtsp-server/rtsp-sdp.c:
3904 use 0.0.0.0 or :: for c= line instead of server address
3906 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
3908 * gst/rtsp-server/rtsp-client.c:
3909 use local address, not remote, in SDP
3910 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
3912 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3915 Automatic update of common submodule
3916 From 098c0d7 to 01a7a46
3918 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
3920 * gst/rtsp-server/rtsp-media.c:
3921 * gst/rtsp-server/rtsp-media.h:
3922 media: possibility to override range time conversion
3923 Make it possible to override the conversion from GstRTSPTimeRange to
3924 GstClockTimes, that is done before seeking on the media
3925 pipeline. Overriding can be useful for UTC ranges, where the default
3926 conversion gives nanoseconds since 1900.
3927 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
3929 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
3931 * gst/rtsp-server/rtsp-server.c:
3932 * gst/rtsp-server/rtsp-server.h:
3933 rtsp-server: Expose the use_client_settings API
3934 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
3936 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
3938 * gst/rtsp-server/rtsp-client.c:
3939 * gst/rtsp-server/rtsp-stream.c:
3940 * gst/rtsp-server/rtsp-stream.h:
3941 rtspstream: handle both ipv4 and ipv6 clients
3942 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
3944 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3946 * gst/rtsp-server/rtsp-sdp.c:
3947 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
3948 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
3949 We already have a way to place extra attributes in the SDP by using a string
3950 property with prefix x- or a- in the caps.
3952 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3954 * gst/rtsp-server/rtsp-sdp.c:
3955 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
3956 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
3957 We already have a way to place extra attributes in the SDP, just make a string
3958 property in the payloader with a- or x- prefix.
3960 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3962 * gst/rtsp-server/rtsp-sdp.c:
3963 rtsp: place a- and x- properties as attributes
3964 application/x-rtp has properties with a- and x- prefixes that should be
3965 placed as attributes in the SDP for the media instead of being added to the
3968 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3970 * examples/Makefile.am:
3971 * examples/test-video.c:
3972 example: add TLS example
3974 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3976 * gst/rtsp-server/rtsp-server.c:
3977 * gst/rtsp-server/rtsp-server.h:
3978 server: add support for TLS
3979 Add methods to set and get a TLS certificate.
3980 Add vmethod to configure a new connection. By default, configure the TLS
3981 certificate in a new connection if needed.
3983 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3985 * gst/rtsp-server/rtsp-server.c:
3986 * gst/rtsp-server/rtsp-server.h:
3987 server: remove accept_client vmethod
3988 This vmethod is not very useful so remove it.
3990 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3992 * gst/rtsp-server/rtsp-server.c:
3993 server: don't crash on NULL GError
3995 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
3997 * gst/rtsp-server/rtsp-session-pool.c:
3998 rtsp-session-pool: corrected session timeout detection
3999 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
4001 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4003 * gst/rtsp-server/rtsp-client.c:
4004 client: improve debug
4006 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4008 * gst/rtsp-server/rtsp-client.c:
4009 * gst/rtsp-server/rtsp-client.h:
4010 * gst/rtsp-server/rtsp-server.c:
4011 server: refactor connection setup
4012 Let the server accept the socket connection and construct a GstRTSPConnection
4013 from it. Remove the code from the client and let the client only deal with
4014 a fully configure GstRTSPConnection object.
4015 We will need this later when the server will configure the connection for
4018 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4020 * gst/rtsp-server/rtsp-stream.c:
4021 stream: keep the transport object alive
4022 Keep the transport object alive while we have it as qdata on the
4025 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
4027 * gst/rtsp-server/rtsp-client.c:
4028 * gst/rtsp-server/rtsp-server.c:
4029 rtsp-server: Do not crash on nmapping of server
4030 * generate error when gst_rtsp_connection_accept fails
4031 * do not stop accepting incoming connections because
4032 accepting a client fails
4033 https://bugzilla.gnome.org/show_bug.cgi?id=701072
4035 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
4037 * gst/rtsp-server/rtsp-client.c:
4038 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
4039 https://bugzilla.gnome.org/show_bug.cgi?id=700953
4041 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4043 * gst/rtsp-server/rtsp-sdp.c:
4044 rtsp-sdp: Parse framerate caps field and set SDP attribute
4045 The SDP attribute and its format is described in RFC4566.
4046 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4048 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
4050 * gst/rtsp-server/rtsp-sdp.c:
4051 rtsp-sdp: Parse width/height from caps and set SDP attribute
4052 The SDP attribute and its format is described in RFC6064.
4053 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4055 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
4057 * gst/rtsp-server/rtsp-sdp.c:
4058 * tests/check/gst/client.c:
4059 rtsp-sdp: add bandwidth line
4060 https://bugzilla.gnome.org/show_bug.cgi?id=699220
4062 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4065 Automatic update of common submodule
4066 From 5edcd85 to 098c0d7
4068 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4070 * tests/check/gst/media.c:
4071 tests: add dynamic payloader prepare/unprepare check
4073 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4075 * gst/rtsp-server/rtsp-media.c:
4076 media: release lock when removing fakesink
4078 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4080 * gst/rtsp-server/rtsp-stream.c:
4081 stream: set elements to NULL before removing
4082 When removing a stream, set the elements to NULL first. This avoids
4083 element-is-not-in-NULL-state errors when we dispose the elements.
4085 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4088 Automatic update of common submodule
4089 From 3cb3d3c to 5edcd85
4091 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4093 * gst/rtsp-server/rtsp-media.c:
4094 * gst/rtsp-server/rtsp-media.h:
4095 media: listen to pad-removed signals
4096 Listen to the pad-removed signal and remove the stream associated with the
4098 Add signal to be notified of the removed pad.
4099 Remove the fakesink in unprepare()
4100 Fix signatures of the signal methods
4102 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4104 * examples/test-sdp.c:
4105 tests: add example of reusable pipelines
4107 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4109 * gst/rtsp-server/rtsp-stream.c:
4110 * gst/rtsp-server/rtsp-stream.h:
4111 stream: add method to get the srcpad
4113 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4115 * tests/check/gst/media.c:
4116 check: add media prepare/unprepare test
4117 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4119 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
4121 * gst/rtsp-server/rtsp-media.c:
4122 media: disconnect from signal handlers in unprepare()
4123 We connected to the pad-added and no-more-pads signals in prepare() so
4124 we need to disconnect from them in unprepare().
4125 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4127 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
4129 * gst/rtsp-server/rtsp-media.c:
4130 media: don't free streams array
4131 Don't free the streams array in the unprepare() method, they were not
4133 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4135 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
4137 * gst/rtsp-server/rtsp-media.c:
4138 media: don't unref the pipeline in unprepare
4139 Unprepare() should undo what prepare() does. Because the pipeline is
4140 not created in prepare(), we should not unref it in unprepare()
4142 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
4144 * gst/rtsp-server/rtsp-stream.c:
4145 stream: clear session and caps for reuse
4146 Set the session and caps to NULL after unref otherwise we might unref
4148 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4150 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
4152 * gst/rtsp-server/rtsp-client.c:
4153 client: send out teardown signal before tearing down
4154 The advantage is that in the signal handler you get direct access to
4155 information about what streams are about to get torn down (in the
4156 GstRTSPClientState).
4157 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
4159 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
4161 * gst/rtsp-server/rtsp-client.c:
4162 * gst/rtsp-server/rtsp-client.h:
4163 client: expose connection
4164 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
4166 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
4169 Automatic update of common submodule
4170 From aed87ae to 3cb3d3c
4172 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4174 * gst/rtsp-server/rtsp-media.c:
4175 * gst/rtsp-server/rtsp-media.h:
4176 * gst/rtsp-server/rtsp-session-media.c:
4177 * gst/rtsp-server/rtsp-session-media.h:
4178 media: add method to get the base_time of the pipeline
4179 Together with a shared clock, this base-time could eventually be sent to
4180 the client so that it can reconstruct the exact running-time of the clock
4183 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4185 * gst/rtsp-server/Makefile.am:
4186 * gst/rtsp-server/rtsp-media.c:
4187 * gst/rtsp-server/rtsp-media.h:
4188 * gst/rtsp-server/rtsp-sdp.c:
4189 media: add GstNetTimeProvider support
4190 Add a property to let the media provide a GstNetTimeProvider for its clock.
4191 Make methods to get the clock and nettimeprovider
4192 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
4193 provider and also the current time of the clock. This should make it possible
4194 for (GStreamer) clients to slave their clock to the server clock.
4196 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4199 Automatic update of common submodule
4200 From 04c7a1e to aed87ae
4202 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4204 * gst/rtsp-server/rtsp-media.c:
4205 media: wait for buffering to complete
4206 Wait for buffering to complete before changing the state to the target state.
4208 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4210 * gst/rtsp-server/rtsp-media.c:
4211 media: small cleanup
4213 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
4215 * tests/check/gst/rtspserver.c:
4216 tests: remove extra unref in test_setup_non_existing_stream
4217 The unref is not needed anymore, teardown runs without it.
4218 https://bugzilla.gnome.org/show_bug.cgi?id=696542
4220 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
4222 * tests/check/gst/rtspserver.c:
4223 tests: GSocketService cleanup in test_bind_already_in_use
4224 Use g_socket_service_stop so the rtspserver test stops listening for
4225 incoming connections in test_bind_already_in_use.
4226 https://bugzilla.gnome.org/show_bug.cgi?id=696541
4228 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
4230 * gst/rtsp-server/rtsp-media-factory.c:
4231 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
4232 Instead use a GWeakRef which is safe to use
4233 This is a known GLib bug, see:
4234 https://bugzilla.gnome.org/show_bug.cgi?id=667145
4236 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
4238 * gst/rtsp-server/rtsp-client.c:
4239 * gst/rtsp-server/rtsp-media.c:
4240 * gst/rtsp-server/rtsp-media.h:
4241 * gst/rtsp-server/rtsp-sdp.c:
4242 * tests/check/gst/media.c:
4243 * tests/check/gst/rtspserver.c:
4244 rtsp-media/client: Reply to PLAY request with same type of Range
4245 Remember the type of Range from the PLAY request and use the same type for
4248 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
4250 * gst/rtsp-server/rtsp-client.c:
4251 * gst/rtsp-server/rtsp-client.h:
4252 * tests/check/gst/client.c:
4253 rtsp-client: expose uri
4255 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
4257 * tests/check/gst/mediafactory.c:
4258 tests: Hold ref while creating second media
4259 To test if the media aren't shared, make sure we keep the first one while creating a second
4260 otherwise the same memory address may be reused.
4262 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
4265 configure: remove out-of-date comment
4267 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
4270 .gitignore: ignore more build files
4272 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
4274 * tests/check/Makefile.am:
4275 tests: use right _LIBS variable for gst-plugins-base libs
4277 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4279 * tests/check/Makefile.am:
4280 check: add librtp to libs
4282 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
4284 * tests/check/gst/rtspserver.c:
4285 tests: Add test to check selecting a port the server will send from
4287 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
4289 * tests/check/gst/rtspserver.c:
4290 tests: Make sure packets are actually received
4292 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4294 * gst/rtsp-server/rtsp-stream.c:
4295 stream: Select unicast address from pool if appropriate
4297 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
4299 * gst/rtsp-server/rtsp-stream.c:
4300 stream: Properties are always there in Gst 1.0
4302 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4304 * tests/check/gst/addresspool.c:
4305 tests: Add tests for unicast addresses in pool
4307 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
4309 * gst/rtsp-server/rtsp-address-pool.c:
4310 * tests/check/gst/addresspool.c:
4311 address-pool: Verify that multicast addresses are used for multicast and vice-versa
4313 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
4315 * docs/libs/gst-rtsp-server-sections.txt:
4316 * gst/rtsp-server/rtsp-address-pool.c:
4317 * gst/rtsp-server/rtsp-address-pool.h:
4318 * gst/rtsp-server/rtsp-stream.c:
4319 * tests/check/gst/addresspool.c:
4320 address-pool: Add unicast addresses
4322 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4325 * gst/rtsp-server/rtsp-server.c:
4326 * tests/check/gst/rtspserver.c:
4327 rtsp-server: Limit the number of threads per server instance
4328 If we exceed the maximum, just round robin the clients over the existing
4331 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
4333 * gst/rtsp-server/rtsp-server.c:
4334 rtsp-server: No need to store the GMainContext in the client context
4336 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
4338 * tests/check/gst/rtspserver.c:
4339 tests: Add test for client disconnection
4341 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4343 * tests/check/gst/rtspserver.c:
4344 tests: Test client and session timeouts with multiple threads
4346 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
4348 * gst/rtsp-server/rtsp-address-pool.c:
4349 * gst/rtsp-server/rtsp-auth.c:
4350 * gst/rtsp-server/rtsp-client.c:
4351 * gst/rtsp-server/rtsp-media-factory-uri.c:
4352 * gst/rtsp-server/rtsp-media-factory.c:
4353 * gst/rtsp-server/rtsp-media.c:
4354 * gst/rtsp-server/rtsp-mount-points.c:
4355 * gst/rtsp-server/rtsp-server.c:
4356 * gst/rtsp-server/rtsp-session-media.c:
4357 * gst/rtsp-server/rtsp-session-pool.c:
4358 * gst/rtsp-server/rtsp-session.c:
4359 Document locking and its order
4361 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
4363 * tests/check/gst/rtspserver.c:
4364 tests: Test that slow DESCRIBE don't block other clients
4366 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
4368 * tests/check/gst/client.c:
4369 tests: Add tests for client-requested multicast address
4371 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
4373 * docs/libs/gst-rtsp-server-sections.txt:
4374 docs: Put the various functions in the right sections
4376 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
4378 * docs/libs/gst-rtsp-server-docs.sgml:
4379 * docs/libs/gst-rtsp-server-sections.txt:
4380 * gst/rtsp-server/rtsp-address-pool.c:
4381 * gst/rtsp-server/rtsp-address-pool.h:
4382 docs: Generate docs for GstRTSPAddressPool
4384 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4386 * gst/rtsp-server/rtsp-client.c:
4387 * gst/rtsp-server/rtsp-stream.c:
4388 * gst/rtsp-server/rtsp-stream.h:
4389 client: Check client provided addresses against the address pool
4391 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
4393 * gst/rtsp-server/rtsp-address-pool.c:
4394 * gst/rtsp-server/rtsp-address-pool.h:
4395 * tests/check/gst/addresspool.c:
4396 address-pool: Add API to request a specific address from the pool
4397 Also add relevant unit tests.
4399 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
4401 * tests/check/gst/mediafactory.c:
4402 tests: Check the passing around of a RTSPAddressPool
4403 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
4404 way down to the stream.
4406 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
4408 * tests/check/gst/addresspool.c:
4409 tests: Add more tests for the address pool
4411 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
4413 * gst/rtsp-server/rtsp-address-pool.c:
4414 address-pool: Fix off by one error
4415 When splitting a port range, the port after a skip is not part of range.
4417 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
4420 Automatic update of common submodule
4421 From 2de221c to 04c7a1e
4423 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
4426 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
4427 AM_CONFIG_HEADER was removed in automake 1.13
4428 https://bugzilla.gnome.org/show_bug.cgi?id=693368
4430 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
4433 Automatic update of common submodule
4434 From a942293 to 2de221c
4436 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4438 * gst/rtsp-server/rtsp-client.c:
4439 client: make sure the watch exists while sending data
4440 Protect the send_func with a lock. This allows us to wait for sending
4441 to complete before changing the send_func and user_data. We add an
4442 extra ref to the watch to make sure that it remains valid during
4444 When closing the connection, set the send_func to NULL
4445 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
4447 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4449 * tests/check/Makefile.am:
4450 tests: use GST_*_1_0 environment variables everywhere
4451 The _1_0 suffixed environment variables override the
4452 non-suffixed ones, so if we're in an environment that
4453 sets the _1_0 suffixed ones, such as jhbuild, we need
4454 to set those to make sure ours actually always get
4457 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4460 Automatic update of common submodule
4461 From acb04d9 to a942293
4463 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4465 * gst/rtsp-server/rtsp-client.c:
4466 rtsp-client: set the client backlog
4467 Set the client backlog to a reasonable default
4469 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
4471 * gst/rtsp-server/rtsp-media.c:
4472 rtsp-media: Make the element a constructor parameter
4473 https://bugzilla.gnome.org/show_bug.cgi?id=689594
4475 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4477 * docs/libs/Makefile.am:
4478 docs: Link with gcov library when gcov is enabled
4479 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
4481 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4483 * gst/rtsp-server/rtsp-media.c:
4484 media: match prepare with unprepare
4485 Really unprepare when there were an equal amount of prepare calls.
4487 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4489 * gst/rtsp-server/rtsp-media.c:
4490 media: media has to be unprepared in finalize
4491 Because unprepare takes away the last ref on the media.
4493 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4495 * gst/rtsp-server/rtsp-client.c:
4496 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
4497 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
4498 We can't use the refcount to trigger unprepare because it is the unprepare call
4499 that removes the last refcount after all messages are consumed. What we should
4500 probably do is make a prepared refcount and only unprepare when the refcount
4503 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4505 * gst/rtsp-server/rtsp-media.c:
4506 media: let the source unref the last media ref
4507 the last ref to the media is held by the source so we don't need to add more ref
4508 and unrefs, we simply destroy the media when the source is gone.
4510 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4512 * gst/rtsp-server/rtsp-media.c:
4513 media: improve debug
4515 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4517 * gst/rtsp-server/rtsp-media.c:
4519 Make sure we are in the right state when collecting the position and duration.
4520 Only make ourselves PREPARED when we were previously PREPARING.
4522 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4524 * gst/rtsp-server/rtsp-media.c:
4525 media: use g_object_ref/unref for GObjects
4527 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
4529 * gst/rtsp-server/rtsp-client.c:
4530 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
4531 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
4532 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
4533 isn't being used anymore.
4535 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
4537 * gst/rtsp-server/rtsp-media.c:
4538 Fix compiler warning
4540 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
4542 * gst/rtsp-server/rtsp-media-factory-uri.c:
4543 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
4545 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4547 * gst/rtsp-server/rtsp-session-media.h:
4550 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4552 * gst/rtsp-server/rtsp-media.c:
4553 * tests/check/gst/media.c:
4554 media: avoid element leak
4556 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4558 * gst/rtsp-server/rtsp-media.c:
4559 media: require an element in media constructor
4561 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4563 * gst/rtsp-server/rtsp-client.c:
4564 Revert "client: TEARDOWN brings that state to Init again"
4565 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
4566 The object is already disposed, there is no point in setting the state.
4568 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4570 * gst/rtsp-server/rtsp-client.c:
4571 client: TEARDOWN brings that state to Init again
4573 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4575 * docs/libs/gst-rtsp-server-sections.txt:
4576 * examples/test-auth.c:
4577 * gst/rtsp-server/rtsp-auth.c:
4578 * gst/rtsp-server/rtsp-auth.h:
4579 * gst/rtsp-server/rtsp-client.c:
4580 * gst/rtsp-server/rtsp-client.h:
4581 * gst/rtsp-server/rtsp-media-factory-uri.c:
4582 * gst/rtsp-server/rtsp-media-factory-uri.h:
4583 * gst/rtsp-server/rtsp-media-factory.c:
4584 * gst/rtsp-server/rtsp-media-factory.h:
4585 * gst/rtsp-server/rtsp-media.c:
4586 * gst/rtsp-server/rtsp-media.h:
4587 * gst/rtsp-server/rtsp-mount-points.c:
4588 * gst/rtsp-server/rtsp-mount-points.h:
4589 * gst/rtsp-server/rtsp-sdp.c:
4590 * gst/rtsp-server/rtsp-server.c:
4591 * gst/rtsp-server/rtsp-server.h:
4592 * gst/rtsp-server/rtsp-session-media.c:
4593 * gst/rtsp-server/rtsp-session-media.h:
4594 * gst/rtsp-server/rtsp-session-pool.c:
4595 * gst/rtsp-server/rtsp-session-pool.h:
4596 * gst/rtsp-server/rtsp-session.c:
4597 * gst/rtsp-server/rtsp-session.h:
4598 * gst/rtsp-server/rtsp-stream-transport.c:
4599 * gst/rtsp-server/rtsp-stream-transport.h:
4600 * gst/rtsp-server/rtsp-stream.c:
4601 * gst/rtsp-server/rtsp-stream.h:
4602 * tests/check/gst/media.c:
4603 rtsp: make object details private
4604 Make all object details private
4605 Add methods to access private bits
4607 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4609 * tests/check/Makefile.am:
4610 * tests/check/gst/media.c:
4611 tests: add media tests
4613 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4615 * gst/rtsp-server/rtsp-media.c:
4616 media: check if prepared for some methods
4617 Check that the media object is prepared before doing seek and getting the
4618 current position etc.
4619 Add some g_return checks.
4621 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4623 * tests/check/Makefile.am:
4624 * tests/check/gst/mediafactory.c:
4625 tests: add mediafactory test
4627 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4629 * gst/rtsp-server/rtsp-stream.c:
4630 stream: improve debug
4632 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4634 * gst/rtsp-server/rtsp-media.c:
4635 * gst/rtsp-server/rtsp-media.h:
4636 media: unref pipeline in finalize to avoid leaking it
4638 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4640 * gst/rtsp-server/rtsp-media-factory-uri.c:
4641 * gst/rtsp-server/rtsp-media.c:
4642 rtsp: use gst_object_unref on GstObjects
4644 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4646 * gst/rtsp-server/rtsp-media-factory.c:
4647 media-factory: require an url
4649 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4651 * examples/test-uri.c:
4652 examples: fix include
4654 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4656 * gst/rtsp-server/rtsp-server.h:
4657 server: remove unused include
4659 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4661 * tests/check/Makefile.am:
4662 * tests/check/gst/mountpoints.c:
4663 tests: add test for mountpoints
4665 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4667 * gst/rtsp-server/rtsp-client.c:
4668 client: fix factory leak
4669 Keep the factory in the state object only for authorization checks and make
4670 sure we unref it on failure. Also don't keep invalid objects in the state
4673 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4675 * gst/rtsp-server/rtsp-mount-points.c:
4676 mounts: add g_return_if guards
4678 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4680 * tests/check/gst/client.c:
4681 tests: add more tests
4683 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4685 * gst/rtsp-server/rtsp-client.c:
4686 client: improve debug
4688 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4690 * gst/rtsp-server/rtsp-client.c:
4691 client: improve debug and fix leaks
4692 Cleanup the uri and session when there is a bad request.
4694 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4699 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4701 * tests/check/gst/client.c:
4702 test: add test for session in options request
4704 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4706 * gst/rtsp-server/rtsp-client.c:
4707 client: use 454 when session can't be found
4708 We should use 454 when a session can't be found because there was no session
4709 pool configured in the server. This is not a server configuration problem
4710 because the server on which the request is done might not be the same one that
4711 will keep the sessions for us and so it does not need to support sessions.
4713 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4715 * gst/rtsp-server/rtsp-client.c:
4716 client: only free connection when there is one
4717 It's possible that the client doesn't have a connection when we try to free it.
4719 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4721 * tests/check/Makefile.am:
4722 * tests/check/gst/client.c:
4723 tests: add unit test for the client object
4725 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4727 * gst/rtsp-server/rtsp-client.c:
4728 client: small cleanup
4730 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4732 * gst/rtsp-server/rtsp-client.h:
4733 client: remove unused include
4735 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4737 * gst/rtsp-server/rtsp-client.c:
4738 client: fix compilation
4740 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4742 * gst/rtsp-server/rtsp-client.c:
4743 client: call destroy without the lock
4745 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4747 * gst/rtsp-server/rtsp-client.c:
4748 * gst/rtsp-server/rtsp-client.h:
4749 client: make the client usable without a socket
4750 Make a method to let the client handle a message and a callback when the client
4751 wants us to send a response message back. This makes it possible to also use the
4752 client object without the sockets, which should make it easier to test.
4754 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4756 * gst/rtsp-server/rtsp-client.c:
4757 * gst/rtsp-server/rtsp-client.h:
4758 client: small cleanup
4760 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4762 * docs/libs/gst-rtsp-server-sections.txt:
4763 * gst/rtsp-server/rtsp-client.c:
4764 * gst/rtsp-server/rtsp-client.h:
4765 * gst/rtsp-server/rtsp-server.c:
4766 client: remove reference to server
4767 We don't need to keep a ref to the server
4769 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4771 * gst/rtsp-server/rtsp-client.c:
4772 * gst/rtsp-server/rtsp-client.h:
4774 Also add some g_return_if()
4776 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4778 * gst/rtsp-server/rtsp-client.c:
4779 client: log more errors
4781 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4783 * gst/rtsp-server/rtsp-client.c:
4784 client: fix compilation
4786 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4788 * gst/rtsp-server/rtsp-client.c:
4789 * gst/rtsp-server/rtsp-client.h:
4790 client: add generic close-after-send support
4791 Add a property to send_response() to close the connection after the response has
4792 been sent to the client.
4794 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4797 * docs/libs/gst-rtsp-server-docs.sgml:
4798 * docs/libs/gst-rtsp-server-sections.txt:
4799 * docs/libs/gst-rtsp-server.types:
4800 * examples/test-auth.c:
4801 * examples/test-launch.c:
4802 * examples/test-mp4.c:
4803 * examples/test-multicast.c:
4804 * examples/test-multicast2.c:
4805 * examples/test-ogg.c:
4806 * examples/test-readme.c:
4807 * examples/test-sdp.c:
4808 * examples/test-uri.c:
4809 * examples/test-video.c:
4810 * gst/rtsp-server/Makefile.am:
4811 * gst/rtsp-server/rtsp-auth.h:
4812 * gst/rtsp-server/rtsp-client.c:
4813 * gst/rtsp-server/rtsp-client.h:
4814 * gst/rtsp-server/rtsp-media-mapping.c:
4815 * gst/rtsp-server/rtsp-media-mapping.h:
4816 * gst/rtsp-server/rtsp-mount-points.c:
4817 * gst/rtsp-server/rtsp-mount-points.h:
4818 * gst/rtsp-server/rtsp-server.c:
4819 * gst/rtsp-server/rtsp-server.h:
4820 * gst/rtsp-server/rtsp-session-media.c:
4821 * gst/rtsp-server/rtsp-session-pool.c:
4822 * gst/rtsp-server/rtsp-session-pool.h:
4823 * tests/check/gst/rtspserver.c:
4824 MediaMapping -> MountPoints
4825 Describes better what the object manages.
4827 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4830 configure: bump required version of -base
4832 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4834 * gst/rtsp-server/rtsp-media.c:
4837 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4839 * gst/rtsp-server/rtsp-media.c:
4840 * gst/rtsp-server/rtsp-media.h:
4841 media: support more Range formats
4842 Use the new -base methods to convert the Range string into a seek start and stop
4845 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4847 * examples/test-launch.c:
4848 examples: fix whitespace
4850 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4852 * examples/test-auth.c:
4853 test-auth: add example of how to remove sessions
4854 Add an example of the session filter api.
4856 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4858 * examples/test-uri.c:
4859 test-uri: remove mapping example
4861 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4863 * examples/test-uri.c:
4864 test-uri: fix callback signature
4866 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4868 * gst/rtsp-server/rtsp-media-factory.c:
4869 factory: keep ref to factory while media active
4870 While the media from a factory is alive, keep a ref to the factory.
4871 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
4873 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4875 * gst/rtsp-server/rtsp-media-factory-uri.c:
4876 factory-uri: add some debug
4878 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4880 * gst/rtsp-server/rtsp-stream.c:
4881 stream: set udp sources to PLAYING
4882 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
4883 so that it doesn't cause our pipeline to produce ASYNC-DONE.
4885 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4887 * gst/rtsp-server/rtsp-media-factory-uri.c:
4888 factory-uri: take ref to factory
4889 Take a ref to the factory that we place in our list.
4891 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4893 * tests/Makefile.am:
4894 * tests/test-reuse.c:
4895 test: add test for server reuse
4896 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
4898 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
4900 * gst/rtsp-server/rtsp-server.c:
4901 server: start and stop multiple times
4902 Stop listening on the RTSP port when the GSource is removed, so clients
4903 can't connect and the server can be started again.
4904 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
4906 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4908 * gst/rtsp-server/rtsp-server.c:
4909 server: fix small leak
4911 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4913 * gst/rtsp-server/rtsp-media.c:
4914 media: unref source in finish_unprepare
4915 The source is created in prepare, unref it in finish_unprepare.
4916 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
4918 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
4920 * gst/rtsp-server/rtsp-client.c:
4921 * gst/rtsp-server/rtsp-media.c:
4922 rtsp-media: remove bus watch before finalizing
4923 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
4924 * An extra media ref is added for the bus watch. This extra ref is unreffed by
4925 the GDestroyNotify function.
4926 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
4927 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
4928 gst_rtsp_media_unprepare before unreffing the media.
4929 This way, the bus watch will be removed before the media is finalized.
4930 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
4932 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
4934 * gst/rtsp-server/rtsp-client.c:
4935 * gst/rtsp-server/rtsp-client.h:
4936 client: wait until the TEARDOWN response is sent to close the connection
4937 Responses can be sent async so we need to wait until the TEARDOWN response has
4938 been written before we close the connection to the client. This avoids the risk
4939 of writing/polling closed sockets.
4940 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
4942 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
4944 * gst/rtsp-server/rtsp-stream.c:
4945 rtsp-stream: plug socket leak
4946 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
4948 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
4951 Automatic update of common submodule
4952 From 6bb6951 to a72faea
4954 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
4956 * gst/rtsp-server/rtsp-media-factory-uri.c:
4957 rtsp-server: don't use deprecated API
4959 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
4961 * gst/rtsp-server/rtsp-client.c:
4962 rtsp-client: fix unused-but-set-variable compiler warning
4963 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
4965 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4968 * docs/libs/gst-rtsp-server-sections.txt:
4969 * gst/rtsp-server/rtsp-client.c:
4972 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4974 * examples/Makefile.am:
4975 * examples/test-multicast2.c:
4976 examples: add another multicast example
4977 Add an example for how to configure separate multicast ranges for each media
4980 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4982 * examples/test-multicast.c:
4985 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4987 * gst/rtsp-server/rtsp-client.c:
4988 * gst/rtsp-server/rtsp-media.c:
4989 * gst/rtsp-server/rtsp-session-media.c:
4990 * gst/rtsp-server/rtsp-session-media.h:
4991 * gst/rtsp-server/rtsp-stream-transport.c:
4992 * gst/rtsp-server/rtsp-stream-transport.h:
4993 stream: use the address managed by the stream
4994 Use the address managed by the stream for multicast. This allows us to have 1
4995 multicast address for each stream.
4996 Because the address is now managed by the stream we don't have to pass it around
4998 Set the address pool on the streams.
5000 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5002 * gst/rtsp-server/rtsp-client.c:
5003 * gst/rtsp-server/rtsp-media.c:
5004 * gst/rtsp-server/rtsp-stream.c:
5007 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5009 * gst/rtsp-server/rtsp-media.c:
5010 * gst/rtsp-server/rtsp-media.h:
5011 media: add signal for new streams
5012 This allows applications to listen for new streams and configure properties on
5013 them, like the address pool.
5015 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5017 * gst/rtsp-server/rtsp-media.c:
5018 media: configure address pool in new streams
5020 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5022 * gst/rtsp-server/rtsp-stream.c:
5023 * gst/rtsp-server/rtsp-stream.h:
5024 stream: add methods to deal with address pool
5025 Add methods to get and set the address pool for the stream
5026 Add method to allocate and get the multicast addresses for this stream.
5028 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5030 * docs/libs/gst-rtsp-server-sections.txt:
5031 * gst/rtsp-server/rtsp-media.c:
5032 * gst/rtsp-server/rtsp-media.h:
5033 media: remove MTU property
5034 It is a stream property
5036 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5038 * gst/rtsp-server/rtsp-client.c:
5039 client: set blocksize only on stream
5040 Set the blocksize only on the current stream.
5042 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5044 * gst/rtsp-server/rtsp-stream.c:
5045 stream: share src and sink sockets
5046 the allocated socket is in the used-socket property, not socket.
5048 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5050 * gst/rtsp-server/rtsp-address-pool.c:
5051 * gst/rtsp-server/rtsp-address-pool.h:
5052 * gst/rtsp-server/rtsp-client.c:
5053 * gst/rtsp-server/rtsp-session-media.c:
5054 * gst/rtsp-server/rtsp-session-media.h:
5055 * gst/rtsp-server/rtsp-stream-transport.c:
5056 * gst/rtsp-server/rtsp-stream-transport.h:
5057 * tests/check/gst/addresspool.c:
5058 rtsp: make address-pool return an address object
5059 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
5060 store more info in the structure and allows us to more easily return the address
5061 to the right pool when no longer needed.
5062 Pass the address to the StreamTransport so that we can return it to the pool
5063 when the stream transport is freed or changed.
5065 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5067 * examples/Makefile.am:
5068 * examples/test-multicast.c:
5069 examples: add multicast example
5070 Show how to set up the multicast address pool so that media can be
5071 server with multicast.
5073 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5075 * gst/rtsp-server/rtsp-client.c:
5076 * gst/rtsp-server/rtsp-media-factory.c:
5077 * gst/rtsp-server/rtsp-media-factory.h:
5078 * gst/rtsp-server/rtsp-media.c:
5079 * gst/rtsp-server/rtsp-media.h:
5080 rtsp: use AddressPool
5081 Remove the multicast_group property.
5082 Use the configured addresspool to allocate multicast addresses.
5084 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5086 * gst/rtsp-server/rtsp-address-pool.c:
5087 * gst/rtsp-server/rtsp-address-pool.h:
5088 address-pool: add clear method
5090 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5092 * gst/rtsp-server/rtsp-address-pool.c:
5093 address-pool: small cleanups
5095 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5097 * tests/check/Makefile.am:
5098 * tests/check/gst/addresspool.c:
5099 tests: add addresspool unit test
5101 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5103 * gst/rtsp-server/Makefile.am:
5104 * gst/rtsp-server/rtsp-address-pool.c:
5105 * gst/rtsp-server/rtsp-address-pool.h:
5106 address-pool: add object to manage multicast addresses
5107 Make an object that can manage a rage of multicast addresses and ports.
5109 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5111 * gst/rtsp-server/rtsp-server.c:
5112 server: set default max-threads property
5114 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5116 * gst/rtsp-server/rtsp-media.c:
5117 media: wait for concurrent _prepare
5118 If a prepare is busy, wait for the result.
5120 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5122 * gst/rtsp-server/rtsp-media.c:
5123 media: add lock around message handler
5124 We don't want to dispatch messages while we are still processing the result of
5127 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5129 * gst/rtsp-server/rtsp-media.c:
5130 * gst/rtsp-server/rtsp-media.h:
5131 media: add lock to protect state changes
5133 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5135 * gst/rtsp-server/rtsp-stream.c:
5136 * gst/rtsp-server/rtsp-stream.h:
5139 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5141 * gst/rtsp-server/rtsp-stream-transport.c:
5142 * gst/rtsp-server/rtsp-stream-transport.h:
5143 * gst/rtsp-server/rtsp-stream.c:
5144 stream-transport: add keep-alive method
5146 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5148 * gst/rtsp-server/rtsp-stream-transport.c:
5149 * gst/rtsp-server/rtsp-stream-transport.h:
5150 * gst/rtsp-server/rtsp-stream.c:
5151 stream-transport: add method to handle RTP/RTCP
5152 Call new methods instead of poking into the structures directly.
5154 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5156 * gst/rtsp-server/rtsp-session-media.c:
5157 * gst/rtsp-server/rtsp-session-media.h:
5158 session-media: add locking
5160 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5162 * gst/rtsp-server/rtsp-session.c:
5163 * gst/rtsp-server/rtsp-session.h:
5164 session: add locking
5166 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5168 * gst/rtsp-server/rtsp-server.c:
5169 server: free old socket
5171 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5173 * gst/rtsp-server/rtsp-media-mapping.c:
5174 * gst/rtsp-server/rtsp-media-mapping.h:
5175 mapping: add locking
5177 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5179 * gst/rtsp-server/rtsp-media-factory.c:
5180 media-factory: add locking
5182 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5184 * gst/rtsp-server/rtsp-auth.c:
5185 * gst/rtsp-server/rtsp-auth.h:
5188 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5190 * gst/rtsp-server/rtsp-server.c:
5191 * gst/rtsp-server/rtsp-server.h:
5192 server: add max-thread property
5194 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5196 * gst/rtsp-server/rtsp-server.c:
5197 * gst/rtsp-server/rtsp-server.h:
5198 server: use a threadpool for the mainloops
5200 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5202 * gst/rtsp-server/rtsp-client.c:
5203 * gst/rtsp-server/rtsp-client.h:
5204 client: rename method
5205 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
5206 don't really create the client from the socket, we use the socket for the
5209 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5211 * gst/rtsp-server/rtsp-client.c:
5212 * gst/rtsp-server/rtsp-client.h:
5213 * gst/rtsp-server/rtsp-server.c:
5214 server: rework maincontext handling in clients
5215 Make a separate method to attach a client to a MainContext.
5216 Let the server decide in what GMainContext the client will operate and give this
5217 context to the client in attach. Then the server can later decide to use a
5218 separate thread for each client or just use the mainthread.
5220 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5222 * gst/rtsp-server/rtsp-client.c:
5223 * gst/rtsp-server/rtsp-session.c:
5224 * gst/rtsp-server/rtsp-session.h:
5225 session: move session header code in session object
5227 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
5231 * examples/test-auth.c:
5232 * examples/test-launch.c:
5233 * examples/test-mp4.c:
5234 * examples/test-ogg.c:
5235 * examples/test-readme.c:
5236 * examples/test-sdp.c:
5237 * examples/test-uri.c:
5238 * examples/test-video.c:
5239 * gst/rtsp-server/rtsp-auth.c:
5240 * gst/rtsp-server/rtsp-auth.h:
5241 * gst/rtsp-server/rtsp-client.c:
5242 * gst/rtsp-server/rtsp-client.h:
5243 * gst/rtsp-server/rtsp-media-factory-uri.c:
5244 * gst/rtsp-server/rtsp-media-factory-uri.h:
5245 * gst/rtsp-server/rtsp-media-factory.c:
5246 * gst/rtsp-server/rtsp-media-factory.h:
5247 * gst/rtsp-server/rtsp-media-mapping.c:
5248 * gst/rtsp-server/rtsp-media-mapping.h:
5249 * gst/rtsp-server/rtsp-media.c:
5250 * gst/rtsp-server/rtsp-media.h:
5251 * gst/rtsp-server/rtsp-params.c:
5252 * gst/rtsp-server/rtsp-params.h:
5253 * gst/rtsp-server/rtsp-sdp.c:
5254 * gst/rtsp-server/rtsp-sdp.h:
5255 * gst/rtsp-server/rtsp-server.c:
5256 * gst/rtsp-server/rtsp-server.h:
5257 * gst/rtsp-server/rtsp-session-media.c:
5258 * gst/rtsp-server/rtsp-session-media.h:
5259 * gst/rtsp-server/rtsp-session-pool.c:
5260 * gst/rtsp-server/rtsp-session-pool.h:
5261 * gst/rtsp-server/rtsp-session.c:
5262 * gst/rtsp-server/rtsp-session.h:
5263 * gst/rtsp-server/rtsp-stream-transport.c:
5264 * gst/rtsp-server/rtsp-stream-transport.h:
5265 * gst/rtsp-server/rtsp-stream.c:
5266 * gst/rtsp-server/rtsp-stream.h:
5267 * tests/check/gst/rtspserver.c:
5268 * tests/test-cleanup.c:
5271 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
5273 * gst/rtsp-server/rtsp-media.c:
5274 * gst/rtsp-server/rtsp-session-media.c:
5275 * gst/rtsp-server/rtsp-session.c:
5276 rtsp-server: added annotations to indicate type of ownership transfer of return values
5277 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5279 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5282 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
5284 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
5287 * bindings/Makefile.am:
5288 * bindings/vala/Makefile.am:
5289 * bindings/vala/gst-rtsp-server-0.10.deps:
5290 * bindings/vala/gst-rtsp-server-0.10.vapi:
5291 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
5292 * bindings/vala/packages/gst-rtsp-server-0.10.files:
5293 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5294 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5295 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
5297 bindings: remove vala bindings
5298 They'll be reunited with the other GStreamer bindings
5299 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5301 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5303 * gst/rtsp-server/rtsp-client.c:
5304 * gst/rtsp-server/rtsp-session-media.c:
5305 * gst/rtsp-server/rtsp-session-media.h:
5306 * gst/rtsp-server/rtsp-stream-transport.c:
5307 * gst/rtsp-server/rtsp-stream-transport.h:
5308 rtsp: only create transport when needed
5309 Only create the StreamTransport when configured.
5311 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5313 * gst/rtsp-server/rtsp-client.c:
5314 client: small cleanup
5316 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5318 * gst/rtsp-server/rtsp-client.c:
5319 * gst/rtsp-server/rtsp-client.h:
5320 * gst/rtsp-server/rtsp-stream-transport.c:
5321 * gst/rtsp-server/rtsp-stream-transport.h:
5322 rtsp: refactor configuration of transport
5323 Move the configuration of the transport to a place where it makes
5326 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5328 * gst/rtsp-server/rtsp-client.c:
5329 client: refactor transport parsing
5331 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5333 * gst/rtsp-server/rtsp-client.c:
5334 client: refuse to change the MTU on shared media
5335 If we change the MTU of chared media, it changes for all clients.
5336 We don't want to set the MTU to something large for clients that
5339 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5341 * examples/test-mp4.c:
5342 * gst/rtsp-server/rtsp-media.c:
5343 small fixes to docs and debug
5345 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5347 * gst/rtsp-server/rtsp-stream.c:
5348 stream: transports must already have been removed
5350 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5352 * gst/rtsp-server/rtsp-media.c:
5353 * gst/rtsp-server/rtsp-stream.c:
5354 * gst/rtsp-server/rtsp-stream.h:
5355 stream: improve join and leave of the pipeline
5357 Do the cleanup properly
5360 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5362 * gst/rtsp-server/rtsp-media.c:
5363 media: move unprepare below default implementation
5364 Makes it easier to find the default implementation
5366 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5368 * gst/rtsp-server/rtsp-media.c:
5369 media: signal unprepared when we actually finish
5371 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5373 * gst/rtsp-server/rtsp-media.c:
5374 media: no need to unlock, unprepare does that when needed
5376 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5378 * docs/libs/gst-rtsp-server-sections.txt:
5379 * gst/rtsp-server/rtsp-media-factory.h:
5380 * gst/rtsp-server/rtsp-media-mapping.c:
5381 * gst/rtsp-server/rtsp-media.h:
5382 * gst/rtsp-server/rtsp-params.c:
5383 * gst/rtsp-server/rtsp-server.c:
5384 * gst/rtsp-server/rtsp-session-pool.h:
5385 * gst/rtsp-server/rtsp-session.c:
5386 * gst/rtsp-server/rtsp-session.h:
5387 * gst/rtsp-server/rtsp-stream-transport.h:
5388 * gst/rtsp-server/rtsp-stream.h:
5391 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5393 * gst/rtsp-server/rtsp-client.c:
5394 * gst/rtsp-server/rtsp-media-mapping.h:
5395 * gst/rtsp-server/rtsp-media.c:
5396 * gst/rtsp-server/rtsp-media.h:
5397 * gst/rtsp-server/rtsp-server.h:
5398 * gst/rtsp-server/rtsp-stream.c:
5399 * gst/rtsp-server/rtsp-stream.h:
5400 rtsp: fix MTU setting
5401 Fix setting of the MTU. There is no need for a vmethod.
5403 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5408 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5411 configure: bump version number after refactoring
5413 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5415 * gst/rtsp-server/Makefile.am:
5416 * gst/rtsp-server/rtsp-client.c:
5417 * gst/rtsp-server/rtsp-client.h:
5418 * gst/rtsp-server/rtsp-media-factory-uri.c:
5419 * gst/rtsp-server/rtsp-media-factory.c:
5420 * gst/rtsp-server/rtsp-media-factory.h:
5421 * gst/rtsp-server/rtsp-media.c:
5422 * gst/rtsp-server/rtsp-media.h:
5423 * gst/rtsp-server/rtsp-sdp.c:
5424 * gst/rtsp-server/rtsp-session-media.c:
5425 * gst/rtsp-server/rtsp-session-media.h:
5426 * gst/rtsp-server/rtsp-session.c:
5427 * gst/rtsp-server/rtsp-session.h:
5428 * gst/rtsp-server/rtsp-stream-transport.c:
5429 * gst/rtsp-server/rtsp-stream-transport.h:
5430 * gst/rtsp-server/rtsp-stream.c:
5431 * gst/rtsp-server/rtsp-stream.h:
5432 rtsp: massive refactoring
5433 Make GObjects from the remaining simple structures.
5434 Remove GstRTSPSessionStream, it's not needed.
5435 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
5436 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
5437 a GstRTSPStream should be transported to a client.
5438 Rename GstRTSPMediaFactory::get_element -> create_element because that
5439 more accurately describes what it does.
5440 Make nice methods instead of poking in the structures.
5441 Move some methods inside the relevant object source code.
5442 Use GPtrArray to store objects instead of plain arrays, it is more
5443 natural and allows us to more easily clean up.
5444 Move the allocation of udp ports to the Stream object. The Stream object
5445 contains the elements needed to stream the media to a client.
5446 Improve the prepare and unprepare methods. Unprepare should now undo
5447 everything prepare did. Improve also async unprepare when doing EOS on
5448 shutdown. Make sure we always unprepare correctly.
5450 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
5452 * gst/rtsp-server/rtsp-client.c:
5453 rtsp-client: Unref server address clients connected to
5454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
5456 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
5458 * gst/rtsp-server/rtsp-server.c:
5459 rtsp-server: don't ref server socket if it is NULL
5460 Fixes test_bind_already_in_use unit test again after commit 6a497440.
5461 https://bugzilla.gnome.org/show_bug.cgi?id=686644
5463 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
5465 * tests/check/Makefile.am:
5466 tests: Add libgio link dependency
5467 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
5469 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5471 * gst/rtsp-server/rtsp-media-mapping.c:
5472 * gst/rtsp-server/rtsp-media-mapping.h:
5473 rtsp-media-mapping: rename find_media vfunc to find_factory
5474 The virtual method and class method should have the same name
5475 so it is correctly represented in GIR file
5476 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5478 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5480 * gst/rtsp-server/rtsp-auth.c:
5481 * gst/rtsp-server/rtsp-client.c:
5482 * gst/rtsp-server/rtsp-media-factory-uri.c:
5483 * gst/rtsp-server/rtsp-media-factory.c:
5484 * gst/rtsp-server/rtsp-media-mapping.c:
5485 * gst/rtsp-server/rtsp-media.c:
5486 * gst/rtsp-server/rtsp-server.c:
5487 * gst/rtsp-server/rtsp-session-pool.c:
5488 * gst/rtsp-server/rtsp-session.c:
5489 rtsp-server: fixed comments and GIR annotations
5490 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5492 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
5494 * gst/rtsp-server/rtsp-media-mapping.c:
5495 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
5497 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
5499 * gst/rtsp-server/rtsp-server.c:
5500 rtsp-server: allow binding on port 0 (binds on a random port)
5502 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
5504 * gst/rtsp-server/rtsp-server.c:
5505 * gst/rtsp-server/rtsp-server.h:
5506 rtsp-server: add bound-port property
5507 bound-port can be used to retrieve the port number when the server is bound on
5508 port 0, which binds on a random port.
5510 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
5512 * gst/rtsp-server/rtsp-media-factory.c:
5513 * gst/rtsp-server/rtsp-media-factory.h:
5514 rtsp-media-factory: make ::get_element overridable by GI bindings
5515 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
5516 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
5517 as the invoker for ::get_element(), making it overridable by GI generated
5520 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5522 * gst/rtsp-server/rtsp-media-factory-uri.c:
5523 rtsp-media-factory-uri: don't autoplug parsers in a loop
5524 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
5527 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5529 * gst/rtsp-server/Makefile.am:
5530 Explicitly link against gio. Fix link error on mac.
5532 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5534 * gst/rtsp-server/rtsp-session.c:
5535 session: add ttl to the transport header in SETUP
5536 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
5538 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5540 * gst/rtsp-server/rtsp-client.c:
5541 * gst/rtsp-server/rtsp-client.h:
5542 * gst/rtsp-server/rtsp-media.c:
5543 client: Use client transport settings for multicast if allowed.
5544 This patch makes it possible for the client to send transport settings for
5545 multicast (destination && ttl). Client settings must be explicitly allowed or
5546 the server will use its own settings.
5547 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
5549 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
5552 Automatic update of common submodule
5553 From 6c0b52c to 6bb6951
5555 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
5557 * gst/rtsp-server/rtsp-client.c:
5558 rtsp-client: do not destroy the rtsp watch
5559 Don't destroy the client watch while dispatching. The rtsp watch is
5560 automatically destroyed after the rtsp watch function closed() has
5562 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
5564 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5567 Automatic update of common submodule
5568 From 4f962f7 to 6c0b52c
5570 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
5572 * gst/rtsp-server/rtsp-media.c:
5573 media: fix check for seekability
5575 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5577 * gst/rtsp-server/rtsp-client.c:
5578 client: use more GIO
5579 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
5581 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5583 * gst/rtsp-server/rtsp-server.c:
5584 server: remove obsolete includes
5586 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5588 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
5589 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
5590 be available in "on_new_ssrc". The transports are added in
5591 gst_rtsp_media_set_state when going to PLAYING state. However,
5592 "on_new_ssrc" might be called before this happens.
5593 https://bugzilla.gnome.org/show_bug.cgi?id=683304
5595 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5597 * gst/rtsp-server/rtsp-client.c:
5598 * gst/rtsp-server/rtsp-client.h:
5599 rtsp-client: add signals for rtsp requests (fixes #683287)
5601 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5603 * gst/rtsp-server/rtsp-client.c:
5604 * gst/rtsp-server/rtsp-client.h:
5605 add new-session signal to rtsp-client (fixes #683058)
5607 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
5610 Automatic update of common submodule
5611 From 668acee to 4f962f7
5613 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
5615 * gst/rtsp-server/rtsp-server.c:
5616 * tests/check/gst/rtspserver.c:
5617 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
5618 Do not assume that *error is set in g_socket_address_enumerator_next.
5619 Added test_bind_already_in_use unit-test.
5620 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
5622 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
5625 Automatic update of common submodule
5626 From 94ccf4c to 668acee
5628 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
5630 * gst/rtsp-server/rtsp-client.c:
5631 * gst/rtsp-server/rtsp-client.h:
5632 rtsp-client: make create_sdp virtual method
5633 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
5635 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5638 Automatic update of common submodule
5639 From 98e386f to 94ccf4c
5641 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5643 * gst/rtsp-server/rtsp-client.c:
5646 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5648 * gst/rtsp-server/rtsp-client.c:
5649 * gst/rtsp-server/rtsp-client.h:
5650 * gst/rtsp-server/rtsp-server.c:
5651 * gst/rtsp-server/rtsp-server.h:
5652 rtsp-server: use an existing socket to establish HTTP tunnel
5653 Make it possible to transfer a socket from an HTTP server to be used as
5654 an RTSP over HTTP tunnel.
5656 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
5658 * gst/rtsp-server/rtsp-client.c:
5659 * gst/rtsp-server/rtsp-media.c:
5660 * gst/rtsp-server/rtsp-media.h:
5661 rtsp: Handle the blocksize parameter
5662 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
5664 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
5666 * tests/check/Makefile.am:
5667 * tests/check/gst/rtspserver.c:
5668 Have unit test get header from source dir, not installed dir
5669 This makes compilation of unit tests work in a build directory other
5670 than the source directory.
5671 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
5673 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
5675 * gst/rtsp-server/rtsp-media.c:
5676 rtsp-media: update for gst_element_make_from_uri() changes
5678 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
5681 * tests/Makefile.am:
5682 * tests/check/Makefile.am:
5683 * tests/check/gst/rtspserver.c:
5685 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
5687 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
5689 * gst/rtsp-server/rtsp-media.c:
5690 rtsp-media: don't collect media stats when going to NULL
5691 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
5693 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5695 * gst/rtsp-server/rtsp-client.c:
5696 client: don't leak transports
5698 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
5700 * gst/rtsp-server/rtsp-client.c:
5701 rtsp-client: free transport on no_stream in SETUP handler
5703 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
5705 * gst/rtsp-server/rtsp-client.c:
5706 rtsp-client: changed session media iteration
5707 In client_unlink_session: now don't iterate in session->medias
5708 list where items are removed by gst_rtsp_session_release_media.
5709 Instead, repeatedly remove the first item.
5711 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
5713 * gst/rtsp-server/rtsp-client.c:
5714 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
5715 GstRTSPSessionMedia is not a GObject type. When the
5716 GstRTSPSession is freed, it will free the media.
5718 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
5720 * gst/rtsp-server/rtsp-media-factory.c:
5721 factory: plug pad leak in collect_streams
5722 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
5723 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
5724 will take one reference, and the other reference will otherwise
5727 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5730 configure: suppress some warnings when debug is disabled
5731 Warnings about unused variables should be suppressed if core has the
5732 debug system disabled.
5733 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5735 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5737 * docs/libs/Makefile.am:
5738 docs: fix build in uninstalled setup
5739 Include gst-plugins-base libs properly.
5741 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
5743 * docs/libs/gst-rtsp-server.types:
5744 docs: include headers defining rtsp-server object types
5745 Fixes compiler warnings during docs build.
5746 https://bugzilla.gnome.org/show_bug.cgi?id=676824
5748 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
5751 configure: Add warning flags for compiler when configuring
5752 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5754 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5757 Automatic update of common submodule
5758 From 03a0e57 to 98e386f
5760 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5763 Automatic update of common submodule
5764 From 1fab359 to 03a0e57
5766 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
5768 * gst/rtsp-server/rtsp-client.c:
5769 client: fix GSocketAddress leak in gst_rtsp_client_accept
5770 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
5772 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5775 Automatic update of common submodule
5776 From f1b5a96 to 1fab359
5778 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5781 Automatic update of common submodule
5782 From 92b7266 to f1b5a96
5784 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5787 Automatic update of common submodule
5788 From ec1c4a8 to 92b7266
5790 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5793 Automatic update of common submodule
5794 From 3429ba6 to ec1c4a8
5796 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
5798 * gst/rtsp-server/rtsp-auth.c:
5799 * gst/rtsp-server/rtsp-client.c:
5800 * gst/rtsp-server/rtsp-media-factory-uri.c:
5801 * gst/rtsp-server/rtsp-server.c:
5802 rtsp: fix compiler warnings
5803 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
5805 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5808 Automatic update of common submodule
5809 From dc70203 to 3429ba6
5811 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5813 * gst/rtsp-server/rtsp-client.c:
5814 * gst/rtsp-server/rtsp-media-factory.c:
5815 * gst/rtsp-server/rtsp-media-factory.h:
5816 * gst/rtsp-server/rtsp-media.c:
5817 * gst/rtsp-server/rtsp-media.h:
5818 * gst/rtsp-server/rtsp-server.c:
5819 * gst/rtsp-server/rtsp-server.h:
5820 * gst/rtsp-server/rtsp-session-pool.c:
5821 * gst/rtsp-server/rtsp-session-pool.h:
5822 rtsp-server: port to new thread API
5824 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5827 Automatic update of common submodule
5828 From 6db25be to dc70203
5830 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5832 * gst/rtsp-server/rtsp-auth.c:
5833 * gst/rtsp-server/rtsp-auth.h:
5834 * gst/rtsp-server/rtsp-client.c:
5835 rtsp-server: Fix compilation and compiler warnings
5837 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5841 * gst/rtsp-server/Makefile.am:
5842 configure: Modernize autotools setup a bit
5843 Also we now only create tar.bz2 and tar.xz tarballs.
5845 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5848 Automatic update of common submodule
5849 From 464fe15 to 6db25be
5851 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5854 Automatic update of common submodule
5855 From 7fda524 to 464fe15
5857 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5860 * docs/libs/Makefile.am:
5861 * docs/version.entities.in:
5863 * gst/rtsp-server/Makefile.am:
5864 * pkgconfig/Makefile.am:
5865 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5866 * pkgconfig/gstreamer-rtsp-server.pc.in:
5867 * tests/Makefile.am:
5868 rtsp-server: Update versioning
5870 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5872 Merge remote-tracking branch 'origin/0.10'
5874 gst/rtsp-server/rtsp-session-pool.c
5876 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5878 * gst/rtsp-server/rtsp-session-pool.c:
5879 rtsp-server: Don't use deprecated GLib API
5881 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5883 Replace master with 0.11
5885 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5887 Merge branch 'master' into 0.11
5889 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5891 Merge branch 'master' into 0.11
5893 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
5896 A couple minor typo fixes
5898 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5900 * gst/rtsp-server/rtsp-media.c:
5901 media: fix state of the appqueue
5903 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5905 * gst/rtsp-server/rtsp-media-factory-uri.c:
5906 factory: use videoconvert
5908 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5910 * gst/rtsp-server/rtsp-media-factory-uri.c:
5911 factory: change to new style caps
5913 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5915 * gst/rtsp-server/rtsp-client.c:
5916 * gst/rtsp-server/rtsp-client.h:
5917 * gst/rtsp-server/rtsp-media-factory-uri.c:
5918 * gst/rtsp-server/rtsp-media.c:
5919 * gst/rtsp-server/rtsp-server.c:
5920 * gst/rtsp-server/rtsp-server.h:
5921 * gst/rtsp-server/rtsp-session-pool.c:
5922 rtsp-server: port to GIO
5925 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5928 configure: fix build
5930 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5933 docs: fix for gst_rtsp_server_set_port() -> _set_service()
5934 https://bugzilla.gnome.org/show_bug.cgi?id=666548
5936 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5939 * examples/Makefile.am:
5940 First rule of gst-rtsp-server club: don't talk about gst-phonon
5942 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5945 * pkgconfig/Makefile.am:
5946 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
5947 * pkgconfig/gst-rtsp-server.pc.in:
5948 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5949 * pkgconfig/gstreamer-rtsp-server.pc.in:
5950 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
5951 For consistency with all other modules.
5953 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5955 * gst/rtsp-server/rtsp-client.c:
5956 rtsp-client: update for new map API
5958 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5961 * bindings/Makefile.am:
5962 * bindings/python/Makefile.am:
5963 * bindings/python/arg-types.py:
5964 * bindings/python/codegen/Makefile.am:
5965 * bindings/python/codegen/__init__.py:
5966 * bindings/python/codegen/argtypes.py:
5967 * bindings/python/codegen/code-coverage.py:
5968 * bindings/python/codegen/codegen.py:
5969 * bindings/python/codegen/definitions.py:
5970 * bindings/python/codegen/defsparser.py:
5971 * bindings/python/codegen/docextract.py:
5972 * bindings/python/codegen/docgen.py:
5973 * bindings/python/codegen/fileprefix.override:
5974 * bindings/python/codegen/fileprefixmodule.c:
5975 * bindings/python/codegen/h2def.py:
5976 * bindings/python/codegen/mergedefs.py:
5977 * bindings/python/codegen/mkskel.py:
5978 * bindings/python/codegen/override.py:
5979 * bindings/python/codegen/reversewrapper.py:
5980 * bindings/python/codegen/scmexpr.py:
5981 * bindings/python/rtspserver-types.defs:
5982 * bindings/python/rtspserver.defs:
5983 * bindings/python/rtspserver.override:
5984 * bindings/python/rtspservermodule.c:
5985 * bindings/python/test.py:
5987 python: remove pygst-based python bindings
5988 pygi is the future, apparently.
5990 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
5993 Automatic update of common submodule
5994 From c463bc0 to 7fda524
5996 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5999 Automatic update of common submodule
6000 From 2a59016 to c463bc0
6002 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6005 Automatic update of common submodule
6006 From 0807187 to 2a59016
6008 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6011 Automatic update of common submodule
6012 From 11f0cd5 to 0807187
6014 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6016 * examples/test-auth.c:
6017 example: update for new caps
6019 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6021 * examples/test-video.c:
6022 * gst/rtsp-server/rtsp-client.c:
6023 * gst/rtsp-server/rtsp-media-factory-uri.c:
6024 * gst/rtsp-server/rtsp-media.c:
6025 * gst/rtsp-server/rtsp-media.h:
6026 * gst/rtsp-server/rtsp-session.c:
6027 * gst/rtsp-server/rtsp-session.h:
6028 rtsp-server: port some more to 0.11
6030 Remove bufferlist stuff
6032 Add queue before appsink now that preroll-queue-len is gone.
6033 Update for request pad changes.
6035 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6037 Merge branch 'master' into 0.11
6039 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6041 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6042 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6043 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6045 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6047 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6048 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6049 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6051 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6053 Merge branch 'master' into 0.11
6055 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6057 * gst/rtsp-server/rtsp-media.c:
6058 * gst/rtsp-server/rtsp-media.h:
6059 media: add a seekable boolean
6060 Maintain the seekable state with a new variable instead of reusing the
6063 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
6065 * gst/rtsp-server/rtsp-media.c:
6066 Disallow seek in live media
6068 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6070 Merge branch 'master' into 0.11
6072 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
6074 * gst/rtsp-server/rtsp-server.c:
6075 #ifdef statements for windows socket creation were missing
6077 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
6080 Automatic update of common submodule
6081 From a39eb83 to 11f0cd5
6083 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
6086 Automatic update of common submodule
6087 From 605cd9a to a39eb83
6089 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6091 Merge branch 'master' into 0.11
6093 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6095 * gst/rtsp-server/rtsp-client.c:
6096 client: use method to access property
6098 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6100 * gst/rtsp-server/rtsp-media-factory.c:
6101 * gst/rtsp-server/rtsp-media-factory.h:
6102 media-factory: add protocols property
6103 Add a property to configure the allowed protocols in the media created from the
6106 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6108 * gst/rtsp-server/rtsp-media-factory.c:
6109 * gst/rtsp-server/rtsp-media-factory.h:
6110 media-factory: add media-configure signal
6111 Add signal to allow the application to configure the media after it was created
6114 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6116 * gst/rtsp-server/rtsp-client.c:
6117 client: use method to access property
6119 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6121 * gst/rtsp-server/rtsp-media-factory.c:
6122 * gst/rtsp-server/rtsp-media-factory.h:
6123 media-factory: add protocols property
6124 Add a property to configure the allowed protocols in the media created from the
6127 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6129 * gst/rtsp-server/rtsp-media-factory.c:
6130 * gst/rtsp-server/rtsp-media-factory.h:
6131 media-factory: add media-configure signal
6132 Add signal to allow the application to configure the media after it was created
6135 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6137 Merge branch 'master' into 0.11
6139 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6141 * gst/rtsp-server/rtsp-client.c:
6142 client: use media multicast group
6144 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6146 * gst/rtsp-server/rtsp-media-factory.h:
6147 * gst/rtsp-server/rtsp-server.h:
6148 * gst/rtsp-server/rtsp-session-pool.h:
6149 * gst/rtsp-server/rtsp-session.h:
6152 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6154 * gst/rtsp-server/rtsp-client.c:
6155 * gst/rtsp-server/rtsp-sdp.h:
6156 sdp: copy and free the server ip address
6157 Copy and free the server ip address to make memory management easier later.
6159 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6161 * gst/rtsp-server/rtsp-media-factory.c:
6162 media-factory: configure multicast in media
6164 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6166 * gst/rtsp-server/rtsp-media.c:
6167 * gst/rtsp-server/rtsp-media.h:
6168 media: add property for multicast group
6169 Add a property to configure the multicast group in the media.
6170 Based on patches from Marc Leeman and Robert Krakora.
6172 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6174 * gst/rtsp-server/rtsp-media-factory.c:
6175 * gst/rtsp-server/rtsp-media-factory.h:
6176 media-factory: add property for multicast group
6177 Add a property to configure the multicast group in the media factory.
6178 Based on patches from Marc Leeman and Robert Krakora.
6180 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6182 * gst/rtsp-server/rtsp-client.c:
6183 client: do configuration of transport in one place
6184 Move the configuration of the transport destination address to where we also
6185 configure the other bits.
6187 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6189 * gst/rtsp-server/rtsp-client.c:
6190 client: use media multicast group
6192 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6194 * gst/rtsp-server/rtsp-media-factory.h:
6195 * gst/rtsp-server/rtsp-server.h:
6196 * gst/rtsp-server/rtsp-session-pool.h:
6197 * gst/rtsp-server/rtsp-session.h:
6200 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6202 * gst/rtsp-server/rtsp-client.c:
6203 * gst/rtsp-server/rtsp-sdp.h:
6204 sdp: copy and free the server ip address
6205 Copy and free the server ip address to make memory management easier later.
6207 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6209 * gst/rtsp-server/rtsp-media-factory.c:
6210 media-factory: configure multicast in media
6212 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6214 * gst/rtsp-server/rtsp-media.c:
6215 * gst/rtsp-server/rtsp-media.h:
6216 media: add property for multicast group
6217 Add a property to configure the multicast group in the media.
6218 Based on patches from Marc Leeman and Robert Krakora.
6220 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6222 * gst/rtsp-server/rtsp-media-factory.c:
6223 * gst/rtsp-server/rtsp-media-factory.h:
6224 media-factory: add property for multicast group
6225 Add a property to configure the multicast group in the media factory.
6226 Based on patches from Marc Leeman and Robert Krakora.
6228 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6230 * gst/rtsp-server/rtsp-client.c:
6231 client: do configuration of transport in one place
6232 Move the configuration of the transport destination address to where we also
6233 configure the other bits.
6235 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6237 Merge branch 'master' into 0.11
6239 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6241 * gst/rtsp-server/rtsp-client.c:
6242 client: destroy pipeline on client disconnect with no prior TEARDOWN.
6243 The problem occurs when the client abruptly closes the connection without
6244 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
6245 server is where the pipeline gets torn down. Since this handler is not called,
6246 the pipeline remains and is up and running. Subsequent clients get their own
6247 pipelines and if the do not issue TEARDOWNs then those pipelines will also
6248 remain up and running. This is a resource leak.
6250 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6252 Merge branch 'master' into 0.11
6254 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
6256 * gst/rtsp-server/rtsp-media-factory.c:
6257 * gst/rtsp-server/rtsp-media-factory.h:
6258 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
6259 For example, it can be used to retrieve source elements like appsrc, in a more
6260 convenient way than subclassing get_element.
6262 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6264 Merge branch 'master' into 0.11
6266 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
6268 * gst/rtsp-server/rtsp-server.c:
6269 rtsp-server: hold on to reference while using object
6271 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6273 * gst/rtsp-server/rtsp-media.c:
6276 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6279 configure: use unstable api
6281 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
6283 * gst/rtsp-server/rtsp-client.c:
6284 client: fix reference counting
6286 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
6288 * gst/rtsp-server/rtsp-client.c:
6289 * gst/rtsp-server/rtsp-media.c:
6290 fix compiler warnings about unused variables
6292 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
6294 * examples/test-launch.c:
6295 * examples/test-readme.c:
6296 * examples/test-uri.c:
6297 * examples/test-video.c:
6298 examples: tell rtsp uri when ready
6300 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
6303 Automatic update of common submodule
6304 From 69b981f to 605cd9a
6306 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6308 * gst/rtsp-server/rtsp-client.c:
6309 client: update for buffer API change
6311 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6313 * gst/rtsp-server/Makefile.am:
6314 Makefile.am: 0.10 => @GST_MAJORMINOR@
6316 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6318 * gst/rtsp-server/rtsp-media-factory-uri.c:
6319 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
6321 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6323 * gst/rtsp-server/.gitignore:
6324 .gitignore: 0.10 => 0.11
6326 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6328 * gst/rtsp-server/Makefile.am:
6329 Makefile.am: 0.10 => @GST_MAJORMINOR@
6331 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6333 Merge branch 'master' into 0.11
6335 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
6338 Automatic update of common submodule
6339 From 9e5bbd5 to 69b981f
6341 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
6344 Automatic update of common submodule
6345 From fd35073 to 9e5bbd5
6347 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
6350 Automatic update of common submodule
6351 From 46dfcea to fd35073
6353 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6355 * gst/rtsp-server/rtsp-media-factory-uri.c:
6356 * gst/rtsp-server/rtsp-media.c:
6357 media: port to new caps API
6359 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6361 Merge branch 'master' into 0.11
6363 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6365 * bindings/vala/gst-rtsp-server-0.10.vapi:
6366 Updated Vala bindings.
6367 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6369 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6371 * gst/rtsp-server/rtsp-server.c:
6372 * gst/rtsp-server/rtsp-server.h:
6373 Add a signal for newly connected clients.
6374 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6376 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6378 * bindings/python/rtspserver.override:
6379 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
6381 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6383 * gst/rtsp-server/Makefile.am:
6384 * gst/rtsp-server/rtsp-client.c:
6385 * gst/rtsp-server/rtsp-funnel.c:
6386 * gst/rtsp-server/rtsp-funnel.h:
6387 * gst/rtsp-server/rtsp-media.c:
6388 rtsp-server: port to 0.11
6390 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6395 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6397 Merge branch 'master' into 0.11
6402 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6405 Automatic update of common submodule
6406 From c3cafe1 to 46dfcea
6408 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
6410 * bindings/python/Makefile.am:
6411 * bindings/python/rtspserver.defs:
6412 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
6414 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
6416 * bindings/python/arg-types.py:
6417 python bindings: add GstRTSPUrlParam
6418 Needed to implement MediaFactory virtual proxies
6420 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
6422 * bindings/python/arg-types.py:
6423 python bindings: fix returning GstRTSPUrl types
6425 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6427 * bindings/python/arg-types.py:
6428 python bindings: add arg type for GstRTSPUrl
6430 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
6432 * bindings/python/rtspserver.defs:
6433 python bindings: fix the definition of MediaFactory.collect_stream
6435 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
6438 Automatic update of common submodule
6439 From 1ccbe09 to c3cafe1
6441 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6444 Automatic update of common submodule
6445 From 193b717 to 1ccbe09
6447 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
6450 Automatic update of common submodule
6451 From b77e2bf to 193b717
6453 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6456 build: Include lcov.mak to allow test coverage report generation
6458 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6461 Automatic update of common submodule
6462 From d8814b6 to b77e2bf
6464 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6467 Automatic update of common submodule
6468 From 6aaa286 to d8814b6
6470 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
6473 Automatic update of common submodule
6474 From 6aec6b9 to 6aaa286
6476 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
6479 autogen: wingo signed comment
6481 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
6483 * gst/rtsp-server/rtsp-session-pool.c:
6484 session: use full charset for RTSP session ID
6485 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
6486 session ID more difficult.
6487 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6489 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6491 * gst/rtsp-server/Makefile.am:
6492 rtsp-server: Don't install the funnel header
6494 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6497 Automatic update of common submodule
6498 From 1de7f6a to 6aec6b9
6500 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6503 configure: require core/base 0.10.31
6504 Needed at least for gst_plugin_feature_rank_compare_func().
6506 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
6509 Automatic update of common submodule
6510 From f94d739 to 1de7f6a
6512 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6514 * gst/rtsp-server/rtsp-media.c:
6515 media: remove more unused code
6517 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6519 * gst/rtsp-server/rtsp-media.c:
6520 * gst/rtsp-server/rtsp-media.h:
6521 media: remove duplicate filtering
6522 Remove the duplicate filtering code now that we have a released -good version.
6523 Give a warning instead.
6525 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6527 * gst/rtsp-server/rtsp-media-factory.c:
6528 * gst/rtsp-server/rtsp-media.c:
6529 media: fix default buffer size
6531 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6533 * gst/rtsp-server/rtsp-media-factory.c:
6534 * gst/rtsp-server/rtsp-media-factory.h:
6535 media-factory: add property to configure the buffer-size
6536 Add a property to configure the kernel UDP buffer size.
6538 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6540 * gst/rtsp-server/rtsp-media.c:
6541 * gst/rtsp-server/rtsp-media.h:
6542 media: add property to configure kernel buffer sizes
6543 Add a property to configure the kernel UDP buffer size.
6545 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6548 configure: set PYGOBJECT_REQ before using it
6549 https://bugzilla.gnome.org/show_bug.cgi?id=640641
6551 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6554 docs: recursive into sub-directories on 'make upload'
6556 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6558 * docs/libs/gst-rtsp-server-docs.sgml:
6559 * docs/version.entities.in:
6560 docs: mention full version these docs are for, not just major-minor
6562 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6567 === release 0.10.8 ===
6569 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6574 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6576 * gst/rtsp-server/rtsp-server.c:
6577 rtsp-server: clarify docs a little
6579 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6581 * gst/rtsp-server/rtsp-media.c:
6582 media: init debug category before starting thread
6584 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6586 * gst/rtsp-server/rtsp-auth.c:
6587 auth: add realm to make it more spec compliant
6589 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6591 * gst/rtsp-server/rtsp-server.c:
6592 * gst/rtsp-server/rtsp-server.h:
6595 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6597 * examples/test-video.c:
6598 example: improve example docs a little
6600 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6602 * gst/rtsp-server/rtsp-server.c:
6603 server: ensure the watch has a ref to the server
6605 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6607 * gst/rtsp-server/rtsp-server.c:
6608 server: simpify channel function
6610 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6612 * gst/rtsp-server/rtsp-server.c:
6613 * gst/rtsp-server/rtsp-server.h:
6614 server: simplify management of channel and source
6615 We don't need to keep around the channel and source objects. Let the mainloop
6616 and the source manage the source and channel respectively.
6618 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6624 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6627 * tests/Makefile.am:
6628 * tests/test-cleanup.c:
6629 tests: add tests directory and cleanup test
6631 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6633 * gst/rtsp-server/rtsp-media-factory-uri.c:
6634 * gst/rtsp-server/rtsp-media-factory.c:
6635 * gst/rtsp-server/rtsp-media-mapping.c:
6636 * gst/rtsp-server/rtsp-media.c:
6637 * gst/rtsp-server/rtsp-session-pool.c:
6638 * gst/rtsp-server/rtsp-session.c:
6639 server: improve debugging in various objects
6641 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6643 * gst/rtsp-server/rtsp-server.c:
6644 server: chain up to the parent finalize
6646 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
6648 * bindings/python/rtspserver-types.defs:
6649 * bindings/python/rtspserver.defs:
6650 * bindings/python/rtspserver.override:
6651 * bindings/python/test.py:
6652 gst-rtsp-server: update python bindings
6654 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6656 * gst/rtsp-server/rtsp-client.c:
6657 client: use the response from the clientstate
6658 Create the response object only once and store in the client state.
6659 Make all methods use the state response,
6661 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6663 * gst/rtsp-server/rtsp-server.c:
6664 server: use signal to keep track of clients
6665 Keep track of all the clients that the server creates and remove them when they
6666 fire the 'closed' signal.
6668 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6670 * gst/rtsp-server/rtsp-client.c:
6671 * gst/rtsp-server/rtsp-client.h:
6672 client: emit signal when closing
6674 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6676 * examples/.gitignore:
6677 * examples/Makefile.am:
6678 * examples/test-auth.c:
6679 * examples/test-video.c:
6680 * gst/rtsp-server/rtsp-auth.c:
6681 * gst/rtsp-server/rtsp-auth.h:
6682 * gst/rtsp-server/rtsp-client.c:
6683 * gst/rtsp-server/rtsp-media-factory.c:
6684 * gst/rtsp-server/rtsp-media.c:
6685 * gst/rtsp-server/rtsp-media.h:
6686 * gst/rtsp-server/rtsp-session-pool.h:
6687 * gst/rtsp-server/rtsp-session.h:
6688 media: enable per factory authorisations
6689 Allow for adding a GstRTSPAuth on the factory and media level and check
6690 permissions when accessing the factory.
6691 Add hints to the auth methods for future more fine grained authorisation.
6692 Add example application for per factory authentication.
6694 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6696 * gst/rtsp-server/rtsp-auth.c:
6697 * gst/rtsp-server/rtsp-auth.h:
6698 * gst/rtsp-server/rtsp-client.c:
6699 * gst/rtsp-server/rtsp-client.h:
6700 * gst/rtsp-server/rtsp-params.c:
6701 * gst/rtsp-server/rtsp-params.h:
6702 rtsp-server: Pass ClientState structure arround
6703 Pass the collected information for the ongoing request in a GstRTSPClientState
6704 structure that we can then pass around to simplify the method arguments. This
6705 will also be handy when we implement logging functionality.
6707 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6709 * gst/rtsp-server/rtsp-media-factory.c:
6710 * gst/rtsp-server/rtsp-media-factory.h:
6711 media-factory: add methods to configure authorisation
6713 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6715 * gst/rtsp-server/rtsp-client.c:
6716 client: unref auth in finalize
6718 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6720 * gst/rtsp-server/rtsp-server.c:
6721 server: unref auth in finalize
6723 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6725 * docs/libs/gst-rtsp-server-docs.sgml:
6726 * docs/libs/gst-rtsp-server-sections.txt:
6727 * docs/libs/gst-rtsp-server.types:
6730 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6732 * gst/rtsp-server/rtsp-server.c:
6733 * gst/rtsp-server/rtsp-server.h:
6734 server: separate create and accept
6735 Create separate create and accept methods so that subclasses can create custom
6737 Configure the server in the client object and prepare for keeping track of
6740 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6742 * gst/rtsp-server/rtsp-client.c:
6743 * gst/rtsp-server/rtsp-client.h:
6744 client: add support for setting the server.
6745 Add support for keeping a ref to the server that started this client
6748 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6750 * gst/rtsp-server/rtsp-auth.c:
6751 auth: fix memleak and add some docs
6752 Fix a memleak of the basic auth token.
6753 Add docs for the helper function
6755 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6757 * gst/rtsp-server/rtsp-auth.c:
6758 * gst/rtsp-server/rtsp-auth.h:
6759 * gst/rtsp-server/rtsp-client.c:
6760 client: delegate setup of auth to the manager
6761 Delegate the configuration of the authentication tokens to the manager object
6764 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6766 * examples/test-video.c:
6767 * gst/rtsp-server/Makefile.am:
6768 * gst/rtsp-server/rtsp-auth.c:
6769 * gst/rtsp-server/rtsp-auth.h:
6770 * gst/rtsp-server/rtsp-client.c:
6771 * gst/rtsp-server/rtsp-client.h:
6772 * gst/rtsp-server/rtsp-server.c:
6773 * gst/rtsp-server/rtsp-server.h:
6774 auth: add authentication object
6775 Add an object that can check the authorization of requests.
6776 Implement basic authentication.
6777 Add example authentication to test-video
6779 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6781 * gst/rtsp-server/rtsp-server.c:
6782 * gst/rtsp-server/rtsp-server.h:
6783 server: move includes back
6784 the includes are needed for sockaddr_in.
6786 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6788 * gst/rtsp-server/rtsp-client.c:
6789 * gst/rtsp-server/rtsp-client.h:
6790 * gst/rtsp-server/rtsp-server.c:
6791 * gst/rtsp-server/rtsp-server.h:
6792 rtsp: move network includes where they are needed
6794 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
6796 * gst/rtsp-server/rtsp-media.h:
6797 rtsp-media.h: Minor corrections in comments.
6800 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
6803 Automatic update of common submodule
6804 From e572c87 to f94d739
6806 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6810 * docs/libs/.gitignore:
6811 * examples/.gitignore:
6812 * gst/rtsp-server/.gitignore:
6815 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6817 * docs/libs/Makefile.am:
6818 docs: We don't build ps/pdf for API reference docs
6820 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6823 Automatic update of common submodule
6824 From ccbaa85 to e572c87
6826 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6829 Automatic update of common submodule
6830 From 46445ad to ccbaa85
6832 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6834 * gst/rtsp-server/Makefile.am:
6835 * gst/rtsp-server/fs-funnel.c:
6836 * gst/rtsp-server/fs-funnel.h:
6837 * gst/rtsp-server/rtsp-funnel.c:
6838 * gst/rtsp-server/rtsp-funnel.h:
6839 * gst/rtsp-server/rtsp-media.c:
6840 funnel: rename fsfunnel to rtspfunnel
6841 Rename the funnel to avoid conflicts with the farsight one.
6843 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6845 * gst/rtsp-server/Makefile.am:
6846 * gst/rtsp-server/fs-funnel.c:
6847 * gst/rtsp-server/fs-funnel.h:
6848 * gst/rtsp-server/rtsp-media.c:
6849 rtsp-media: add and use fsfunnel
6850 Add a copy of fsfunnel to the build because input-selector removed the (broken)
6851 select-all property that we need.
6853 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6855 * gst/rtsp-server/Makefile.am:
6856 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
6857 Use PKG_CONFIG_PATH specified at configure time (if any) as well
6858 for the g-ir-compiler, rather than just assuming the env var has
6861 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6868 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
6870 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6873 * gst/rtsp-server/Makefile.am:
6874 gobject-introspection: fix g-i build for uninstalled setup
6875 Requires gst-plugins-base git (> 0.10.31.2).
6877 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6879 * examples/test-uri.c:
6880 examples: add some more options and comments
6882 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6884 * gst/rtsp-server/rtsp-media-factory-uri.c:
6885 factory-uri: use right property type
6887 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6889 * gst/rtsp-server/rtsp-media-factory-uri.c:
6890 factory-uri: attempt to configure buffer-lists
6891 Attempt to configure buffer lists in the payloader for improved performance.
6893 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6895 * gst/rtsp-server/rtsp-media.c:
6896 media: attempt to configure bigger UDP buffers
6897 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
6898 send buffers with high bitrate streams.
6900 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
6902 * gst/rtsp-server/rtsp-client.c:
6903 client: use the socket length from getsockname
6904 Use the length returned by getsockname to perform the getnameinfo call because
6905 the size can depend on the socket type and platform.
6908 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6910 * docs/libs/gst-rtsp-server-docs.sgml:
6911 * docs/libs/gst-rtsp-server-sections.txt:
6912 docs: add uri factory to the docs
6914 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6916 * gst/rtsp-server/rtsp-client.c:
6917 * gst/rtsp-server/rtsp-media.h:
6920 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6922 * gst/rtsp-server/rtsp-client.c:
6923 * gst/rtsp-server/rtsp-media.c:
6924 * gst/rtsp-server/rtsp-media.h:
6925 * gst/rtsp-server/rtsp-session.c:
6926 * gst/rtsp-server/rtsp-session.h:
6927 rtsp-server: add support for buffer lists
6928 Add support for sending bufferlists received from appsink.
6931 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6933 * gst/rtsp-server/rtsp-client.c:
6934 * gst/rtsp-server/rtsp-media.c:
6935 * gst/rtsp-server/rtsp-media.h:
6936 * gst/rtsp-server/rtsp-sdp.c:
6937 media: make method to retrieve the play range
6938 Make a method to retrieve the playback range so that we can conditionally create
6939 a different range for the SDP and the PLAY requests.
6941 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6943 * gst/rtsp-server/rtsp-media.c:
6944 * gst/rtsp-server/rtsp-media.h:
6945 media: add signal to notify of state changes
6947 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6949 * gst/rtsp-server/rtsp-client.h:
6950 client: cleanup headers
6952 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6954 * gst/rtsp-server/rtsp-client.c:
6957 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6959 * gst/rtsp-server/rtsp-media-factory-uri.c:
6960 * gst/rtsp-server/rtsp-media-factory-uri.h:
6961 factory-uri: add support for gstpay
6962 Add an option to prefer gstpay over decoder + raw payloader.
6964 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6966 * gst/rtsp-server/rtsp-media-factory-uri.c:
6967 * gst/rtsp-server/rtsp-media-factory-uri.h:
6968 factory-uri: rework the autoplugger.
6969 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
6972 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6974 * gst/rtsp-server/rtsp-media-factory-uri.c:
6975 factory-uri: use better factory filter
6976 Make better payloader filter based on autoplug rank and RTP use case.
6978 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6981 Automatic update of common submodule
6982 From 169462a to 46445ad
6984 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6986 * gst/rtsp-server/rtsp-server.c:
6987 server: set SO_REUSEADDR before bind
6988 Set the SO_REUSEADDR _before_ bind() to make it actually work.
6990 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6992 * gst/rtsp-server/rtsp-media.c:
6993 * gst/rtsp-server/rtsp-media.h:
6994 media: emit prepared signal when prepared
6995 Make a 'prepared' signal and emit it when we successfully prepared the element.
6996 This signal can be used to configure the media object after it has been prepared
6999 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
7002 Automatic update of common submodule
7003 From 011bcc8 to 169462a
7005 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
7007 python an optional dependency
7008 * configure.ac: Move up valgrind and g-i checks. Make the python
7009 dependency optional, as it was before.
7011 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7013 Merge branch 'master' into 0.11
7018 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7020 * gst/rtsp-server/rtsp-media.c:
7021 media: update range when active clients changed
7022 When we changed the number of active clients, update the current range
7023 information because we want the second client connecting to a shared resource
7024 continue from where the stream currently.
7026 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7028 * gst/rtsp-server/rtsp-media-factory-uri.c:
7029 * gst/rtsp-server/rtsp-media-factory-uri.h:
7030 factory-uri: add colorspace and fix pt
7031 Rework the way we pass data to the autoplugger.
7032 When we have raw caps, plug a converter element to make pluggin to raw
7033 payloaders more successful.
7034 Make sure all dynamically plugged payloaders have a unique payload types.
7036 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7038 * examples/Makefile.am:
7039 * examples/test-uri.c:
7040 example: add example of the uri factory
7042 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7044 * gst/rtsp-server/Makefile.am:
7045 * gst/rtsp-server/rtsp-media-factory-uri.c:
7046 * gst/rtsp-server/rtsp-media-factory-uri.h:
7047 * gst/rtsp-server/rtsp-server.h:
7048 factory-uri: add a factory to stream any URI
7049 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
7052 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7054 * gst/rtsp-server/rtsp-media.c:
7055 * gst/rtsp-server/rtsp-media.h:
7056 media: ignore spurious ASYNC_DONE messages
7057 When we are dynamically adding pads, the addition of the udpsrc elements will
7058 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
7059 the real ASYNC_DONE when everything is prerolled.
7061 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7063 * gst/rtsp-server/rtsp-media-factory.c:
7064 * gst/rtsp-server/rtsp-media-factory.h:
7065 media-factory: make lock macro
7067 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
7069 * gst/rtsp-server/rtsp-client.c:
7070 rtsp-server: Remove unused variable and dead assignment
7072 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
7074 * examples/test-launch.c:
7075 * examples/test-mp4.c:
7076 * examples/test-ogg.c:
7077 * examples/test-readme.c:
7078 * examples/test-sdp.c:
7079 * examples/test-video.c:
7080 examples: Run gst-indent
7082 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
7084 * gst/rtsp-server/rtsp-client.c:
7085 * gst/rtsp-server/rtsp-media-factory.c:
7086 * gst/rtsp-server/rtsp-media-mapping.c:
7087 * gst/rtsp-server/rtsp-media.c:
7088 * gst/rtsp-server/rtsp-params.c:
7089 * gst/rtsp-server/rtsp-sdp.c:
7090 * gst/rtsp-server/rtsp-server.c:
7091 * gst/rtsp-server/rtsp-session-pool.c:
7092 * gst/rtsp-server/rtsp-session.c:
7093 rtsp-server: Run gst-indent
7094 Since it wasn't using the upstream common previously, there was no
7095 indentation check before commiting.
7097 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
7099 * gst/rtsp-server/rtsp-media-mapping.h:
7100 * gst/rtsp-server/rtsp-media.c:
7101 * gst/rtsp-server/rtsp-media.h:
7102 * gst/rtsp-server/rtsp-sdp.c:
7103 * gst/rtsp-server/rtsp-session-pool.h:
7104 * gst/rtsp-server/rtsp-session.c:
7105 * gst/rtsp-server/rtsp-session.h:
7106 rtsp-server: Some more doc fixups
7108 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7111 Makefile: Add cruft-cleaning support
7113 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7118 * docs/libs/Makefile.am:
7119 * docs/libs/gst-rtsp-server-docs.sgml:
7120 * docs/libs/gst-rtsp-server-sections.txt:
7121 * docs/libs/gst-rtsp-server.types:
7122 * docs/version.entities.in:
7123 docs: Add gtk-doc build system
7125 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7127 * gst/rtsp-server/Makefile.am:
7128 Makefile.am: Use standard GIR make behaviour
7130 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7134 autogen/configure: Bring more in sync to standard gst module behaviour
7136 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7138 * gst/rtsp-server/rtsp-media.c:
7139 media: warn and fail when gstrtpbin is not found
7141 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7144 configure: open 0.11 branch
7146 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
7150 Add common submodule
7152 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
7155 * common/Makefile.am:
7156 * common/c-to-xml.py:
7158 * common/coverage/coverage-report-entry.pl:
7159 * common/coverage/coverage-report.pl:
7160 * common/coverage/coverage-report.xsl:
7161 * common/coverage/lcov.mak:
7162 * common/gettext.patch:
7163 * common/glib-gen.mak:
7164 * common/gst-autogen.sh:
7165 * common/gst-xmlinspect.py:
7167 * common/gstdoc-scangobj:
7168 * common/gtk-doc-plugins.mak:
7169 * common/gtk-doc.mak:
7170 * common/m4/.gitignore:
7171 * common/m4/Makefile.am:
7173 * common/m4/as-ac-expand.m4:
7174 * common/m4/as-auto-alt.m4:
7175 * common/m4/as-compiler-flag.m4:
7176 * common/m4/as-compiler.m4:
7177 * common/m4/as-docbook.m4:
7178 * common/m4/as-libtool-tags.m4:
7179 * common/m4/as-libtool.m4:
7180 * common/m4/as-python.m4:
7181 * common/m4/as-scrub-include.m4:
7182 * common/m4/as-version.m4:
7183 * common/m4/ax_create_stdint_h.m4:
7184 * common/m4/check.m4:
7185 * common/m4/glib-gettext.m4:
7186 * common/m4/gst-arch.m4:
7187 * common/m4/gst-args.m4:
7188 * common/m4/gst-check.m4:
7189 * common/m4/gst-debuginfo.m4:
7190 * common/m4/gst-default.m4:
7191 * common/m4/gst-doc.m4:
7192 * common/m4/gst-error.m4:
7193 * common/m4/gst-feature.m4:
7194 * common/m4/gst-function.m4:
7195 * common/m4/gst-gettext.m4:
7196 * common/m4/gst-glib2.m4:
7197 * common/m4/gst-libxml2.m4:
7198 * common/m4/gst-plugindir.m4:
7199 * common/m4/gst-valgrind.m4:
7200 * common/m4/gtk-doc.m4:
7201 * common/m4/introspection.m4:
7203 * common/mangle-tmpl.py:
7204 * common/plugins.xsl:
7206 * common/release.mak:
7207 * common/scangobj-merge.py:
7208 * common/upload.mak:
7209 common: Remove static version
7211 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
7213 * common/m4/introspection.m4:
7214 Update introspection.m4 to match usage
7216 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7220 Remove old stuff from the README
7222 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7227 === release 0.10.7 ===
7229 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7234 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7236 * examples/test-ogg.c:
7237 test-ogg: remove parsers
7238 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
7239 buffers with timestamps. Using the parsers also seems to break things.
7241 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7243 * bindings/vala/gst-rtsp-server-0.10.vapi:
7244 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7245 Updated Vala bindings
7247 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7249 * common/m4/introspection.m4:
7251 * gst/rtsp-server/Makefile.am:
7252 Added initial gobject-introspection support
7254 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7256 * gst/rtsp-server/rtsp-media-factory.c:
7257 media-factory: don't use host for shared hash key
7258 When we generate the key to share made between connections, don't include the
7259 host used to connect so that we can share media even if between clients that
7260 connected with localhost and ones with the ip address.
7262 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7264 * bindings/vala/Makefile.am:
7265 build: fix distcheck
7267 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7269 * bindings/vala/gst-rtsp-server-0.10.vapi:
7270 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7271 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7272 Update Vala bindings
7274 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7276 * bindings/vala/Makefile.am:
7278 Fix configure checks and installation location for Vala bindings
7281 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7286 === release 0.10.6 ===
7288 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7291 configure: release 0.10.6
7293 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7295 * gst/rtsp-server/rtsp-media.c:
7296 media: help the compiler a little
7298 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7300 * gst/rtsp-server/rtsp-media.c:
7301 * gst/rtsp-server/rtsp-media.h:
7302 * gst/rtsp-server/rtsp-session.c:
7303 media: cleanup media transport before freeing
7304 Cleanup the media transport data before freeing. In particular, remove the qdata
7305 from the rtpsource object.
7307 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7309 * gst/rtsp-server/rtsp-media-factory.c:
7310 * gst/rtsp-server/rtsp-media-factory.h:
7311 * gst/rtsp-server/rtsp-media.c:
7312 * gst/rtsp-server/rtsp-media.h:
7313 media-factory: add eos-shutdown property
7314 Add an eos-shutdown property that will send an EOS to the pipeline before
7315 shutting it down. This allows for nice cleanup in case of a muxer.
7318 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7320 * gst/rtsp-server/rtsp-media.c:
7321 * gst/rtsp-server/rtsp-media.h:
7322 media: use multiudpsink send-duplicates when we can
7323 If we have a new enough multiudpsink with the send-duplicates property, use this
7324 instead of doing our own filtering. Our custom filtering code should eventually
7325 be removed when we can depend on a released -good.
7327 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7329 * gst/rtsp-server/rtsp-media.c:
7330 media: don't leak destinations
7331 Refactor and cleanup the destinations array when the stream is destroyed.
7333 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7335 * gst/rtsp-server/rtsp-media.c:
7336 * gst/rtsp-server/rtsp-media.h:
7337 media: don't add udp addresses multiple times
7338 Keep track of the udp addresses we added to udpsink and never add the same udp
7339 destination twice. This avoids duplicate packets when using multicast.
7341 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7343 * gst/rtsp-server/rtsp-server.c:
7344 server: disable use of SO_LINGER
7345 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
7346 server close()s the connection.
7348 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7350 * gst/rtsp-server/rtsp-server.c:
7351 server: use 5 second linger period in SO_LINGER
7352 Wait 5 seconds before clearing the send buffers and reseting the connection with
7353 the client when we do a close. This should be enough time to get the message to
7357 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7359 * gst/rtsp-server/rtsp-server.c:
7360 server: use SO_LINGER
7361 SO_LINGER on the socket will make sure that any pending data on the socket is
7362 flushed ASAP and that the socket connection is reset. This makes sure that the
7363 socket can be reused immediately.
7366 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7369 README: add blurb about shared media factories
7371 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
7373 * gst/rtsp-server/rtsp-media.c:
7374 Add stdlib.h for atoi()
7376 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7378 * bindings/python/Makefile.am:
7379 * bindings/vala/Makefile.am:
7380 build: distcheck fixes
7381 Fix 'make distcheck', somewhat (it still fails because it tries to
7382 install files into /usr/share/vala/vapi/ irrespective of the
7385 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7388 configure: bump core/base requirements to released version
7389 Makes things less confusing for people.
7391 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7394 configure: fail if GStreamer core/base requirements are not met
7396 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7398 * gst/rtsp-server/rtsp-client.c:
7399 client: improve client cleanups
7400 Make sure the session does not timeout when using TCP. We need to do this
7401 because quicktime player does not send RTCP for some reason in tunneled
7403 Refactor some cleanup code.
7406 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7408 * gst/rtsp-server/rtsp-session.c:
7409 * gst/rtsp-server/rtsp-session.h:
7410 session: add support for prevent session timeouts
7411 Add an atomix counter to prevent session timeouts when we are, for example,
7414 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7416 * gst/rtsp-server/rtsp-client.c:
7417 client: fix unlink on session timeouts
7418 When our session times out, make sure we unlink all streams in this
7420 Remove the tunnelid when closing the connection.
7422 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7424 * gst/rtsp-server/rtsp-session.c:
7425 session: small cleanups
7427 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7429 * gst/rtsp-server/rtsp-client.c:
7430 client: handle lost_tunnel callbacks
7431 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
7432 hashtable so that we can reuse it for when the client reopens the POST
7434 Close the connection after a TEARDOWN.
7435 Make sure or watchid is cleared when the watch is removed.
7438 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7440 * gst/rtsp-server/rtsp-client.c:
7441 * gst/rtsp-server/rtsp-media.c:
7442 * gst/rtsp-server/rtsp-sdp.c:
7443 rtsp-server: add more support for multicast
7445 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7448 * gst/rtsp-server/rtsp-media.c:
7449 * gst/rtsp-server/rtsp-media.h:
7450 media: allow configuration of allowed lower transport
7452 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7454 * gst/rtsp-server/rtsp-client.h:
7455 * gst/rtsp-server/rtsp-media.c:
7456 * gst/rtsp-server/rtsp-media.h:
7457 * gst/rtsp-server/rtsp-sdp.c:
7458 * gst/rtsp-server/rtsp-sdp.h:
7459 * gst/rtsp-server/rtsp-server.c:
7460 rtsp: keep track of server ip and ipv6
7461 Keep track of how the client connected to the server and setup the udp ports
7462 with the same protocol.
7463 Copy the server ip address in the SDP so that clients can send RTCP back to
7466 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7468 * gst/rtsp-server/rtsp-session.c:
7471 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7473 * gst/rtsp-server/rtsp-client.c:
7474 client: use right size for malloc
7476 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7478 * gst/rtsp-server/rtsp-server.c:
7479 server: comment ipv6 server listening address
7481 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7483 * gst/rtsp-server/rtsp-media.c:
7484 media: allow for ipv6 sockets
7486 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7488 * gst/rtsp-server/rtsp-server.c:
7489 * gst/rtsp-server/rtsp-server.h:
7490 server: rework server part
7491 Allow setting a bind address, make sure we can deal with ipv6.
7492 Remove the port property and change with the service property.
7494 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7496 * gst/rtsp-server/rtsp-media.h:
7497 media: update comments a little
7499 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7501 * gst/rtsp-server/rtsp-client.c:
7502 client: make content-base better
7503 Use the URI formatting functions to make a content-base. Also make sure that
7504 there is a trailing / at the end.
7506 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7508 * gst/rtsp-server/rtsp-client.c:
7509 client: guard against invalid paths
7511 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7513 * examples/test-video.c:
7514 test: catch server bind errors
7516 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
7518 * gst/rtsp-server/rtsp-media.c:
7519 rtspmedia: emit "unprepared" if _prepare fails.
7520 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
7521 media object is removed from its factory's cache.
7523 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7525 * gst/rtsp-server/rtsp-media.c:
7526 media: collect media position when seek completes
7528 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
7530 * gst/rtsp-server/rtsp-client.c:
7531 client: call unlink_streams in client finalize
7534 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7536 * gst/rtsp-server/rtsp-media.c:
7537 media: limit the time to wait to something huge
7538 Avoid waiting forever but limit the timeout to 20 seconds.
7540 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7542 * gst/rtsp-server/rtsp-sdp.c:
7543 sdp: reindent and check for prepared status
7545 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7547 * gst/rtsp-server/rtsp-media.c:
7548 * gst/rtsp-server/rtsp-media.h:
7549 * gst/rtsp-server/rtsp-session.c:
7550 media: avoid doing _get_state() for state changes
7551 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
7552 until the media is prerolled or in error. This avoids doing a blocking call of
7553 gst_element_get_state() that can cause lockups when there is an error.
7556 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7558 * gst/rtsp-server/rtsp-media.c:
7561 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7563 * gst/rtsp-server/rtsp-media-factory.c:
7564 media-factory: better error handling
7565 Improve the error handling a bit.
7567 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7569 * gst/rtsp-server/rtsp-client.c:
7570 client: rework transport parsing
7571 Rework the transport parsing code so that we can ignore transports we don't
7572 support instead of just picking the first one we can parse.
7573 Configure a (for now hardcoded) destination for multicast transports.
7575 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7577 * gst/rtsp-server/rtsp-media.c:
7578 media: set multicast sink parameters
7579 Disable loop and automatic multicast join on the udpsink elements.
7580 Add some more debug info.
7581 Reset some state variables in the right place.
7582 Use the right port numbers for multicast.
7584 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7586 * gst/rtsp-server/rtsp-session.c:
7587 session: handle transport setup correctly
7588 Handle UDP, MCAST and TCP transport negotiation more correctly.
7589 Store the server session SSRC in the transport.
7591 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7593 * gst/rtsp-server/rtsp-client.c:
7594 rtsp-client: implement error_full
7595 Implement error_full to avoid some segfaults when the rtspconnection calls it.
7598 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7601 * gst/rtsp-server/rtsp-client.c:
7602 * gst/rtsp-server/rtsp-server.c:
7603 docs: update docs and comments
7605 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
7607 * gst/rtsp-server/rtsp-sdp.c:
7608 sdp: make server work better when behind a proxy
7610 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7612 * gst/rtsp-server/rtsp-client.c:
7613 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
7615 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7617 * gst/rtsp-server/rtsp-client.c:
7618 * gst/rtsp-server/rtsp-media-factory.c:
7619 * gst/rtsp-server/rtsp-media-mapping.c:
7620 * gst/rtsp-server/rtsp-media.c:
7621 * gst/rtsp-server/rtsp-server.c:
7622 * gst/rtsp-server/rtsp-session-pool.c:
7623 * gst/rtsp-server/rtsp-session.c:
7624 Use GStreamer's debugging subsystem
7626 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7628 * gst/rtsp-server/rtsp-media-factory.c:
7629 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
7631 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7636 === release 0.10.5 ===
7638 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7643 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7646 configure: bump required versions
7648 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
7650 * gst/rtsp-server/rtsp-client.c:
7651 client: call weak-unref on client->sessions from finalize
7654 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7656 * gst/rtsp-server/rtsp-media.c:
7657 media: Fixed crasher where caps got unref'ed too often
7659 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7662 * pkgconfig/.gitignore:
7663 * pkgconfig/Makefile.am:
7664 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
7665 Added pkg-config file to use gst-rtsp-server uninstalled
7667 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7669 * gst/rtsp-server/rtsp-media.c:
7670 media: add some docs
7672 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
7674 * gst/rtsp-server/rtsp-client.c:
7675 rtsp: Use gst_rtsp_watch_send_message().
7676 Use gst_rtsp_watch_send_message() since the old API which used
7677 gst_rtsp_watch_queue_message() has been deprecated.
7679 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7684 === release 0.10.4 ===
7686 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7691 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7693 * gst/rtsp-server/rtsp-client.c:
7694 * gst/rtsp-server/rtsp-session.c:
7695 * gst/rtsp-server/rtsp-session.h:
7696 rtsp: allocate channels in TCP mode
7697 When the client does not provide us with channels in TCP mode, allocate channels
7700 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7702 * gst/rtsp-server/rtsp-client.c:
7703 client: don't crash when tunnelid is missing
7704 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
7705 don't crash but return an error response to the client.
7708 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7710 * bindings/vala/gst-rtsp-server-0.10.vapi:
7711 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7712 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7713 bindings: update vala bindings with new method
7715 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7717 * gst/rtsp-server/rtsp-session-pool.c:
7718 * gst/rtsp-server/rtsp-session-pool.h:
7719 sessionpool: add function to filter sessions
7720 Add generic function to retrieve/remove sessions.
7722 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7725 configure: bump core/base requirements to release
7727 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7729 * gst/rtsp-server/rtsp-media.c:
7730 media: fix indentation
7732 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7734 * gst/rtsp-server/rtsp-media.c:
7735 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
7737 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7739 * gst/rtsp-server/rtsp-media.c:
7740 set state and remove elements of media in for loop
7742 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
7744 * bindings/vala/gst-rtsp-server-0.10.vapi:
7745 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7746 Added gst_rtsp_media_remove_elements function to Vala bindings
7748 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
7750 * gst/rtsp-server/rtsp-media.c:
7751 * gst/rtsp-server/rtsp-media.h:
7752 Added gst_rtsp_media_remove_elements function
7754 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
7756 * gst/rtsp-server/rtsp-media.c:
7757 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
7759 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7761 * bindings/vala/gst-rtsp-server-0.10.vapi:
7762 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7763 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7764 Updated Vala bindings
7766 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7768 * gst/rtsp-server/rtsp-media.c:
7769 * gst/rtsp-server/rtsp-media.h:
7770 Added vmethod unprepare to GstRTSPMedia
7771 The default implementation sets the state of the pipeline to GST_STATE_NULL
7773 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7775 * gst/rtsp-server/rtsp-media-factory.c:
7776 * gst/rtsp-server/rtsp-media-factory.h:
7777 Made collect_streams function public
7779 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7781 * gst/rtsp-server/rtsp-media-factory.c:
7782 * gst/rtsp-server/rtsp-media-factory.h:
7783 * gst/rtsp-server/rtsp-media.c:
7784 Added vmethod create_pipeline to GstRTSPMediaFactory
7785 The pipeline is created in this method and the GstRTSPMedia's element is added to it
7787 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7789 * gst/rtsp-server/rtsp-client.c:
7790 client: use g_source_destroy()
7791 We need to use g_source_destroy() because we might have added the source to a
7792 different main context than the default one.
7794 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7796 * gst/rtsp-server/Makefile.am:
7797 * gst/rtsp-server/rtsp-client.c:
7798 * gst/rtsp-server/rtsp-params.c:
7799 * gst/rtsp-server/rtsp-params.h:
7800 rtsp: prepare for handling GET/SET_PARAMETER
7801 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
7803 Fix return codes of handlers.
7805 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7807 * gst/rtsp-server/rtsp-media.c:
7808 media: don't leak session pads
7810 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7812 * gst/rtsp-server/rtsp-media.c:
7813 media: clean up the messages a bit
7815 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7817 * gst/rtsp-server/rtsp-sdp.c:
7818 sdp: warn and skip streams without media
7820 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7822 * bindings/vala/gst-rtsp-server-0.10.vapi:
7823 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7824 vala: Fixed typo in header file of RTSPMediaStream
7826 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7828 * gst/rtsp-server/rtsp-media.c:
7831 Make dumping RTCP stats configurable
7833 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7835 * gst/rtsp-server/rtsp-media.c:
7836 media: be less verbose and leak less
7838 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7840 * gst/rtsp-server/rtsp-media.c:
7841 media: don't leak the destination address
7843 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7845 * gst/rtsp-server/rtsp-client.c:
7846 * gst/rtsp-server/rtsp-media.c:
7847 * gst/rtsp-server/rtsp-media.h:
7848 * gst/rtsp-server/rtsp-session.c:
7849 * gst/rtsp-server/rtsp-session.h:
7850 rtsp: use RTCP to keep the session alive
7851 Use the RTCP rtcp-from stats field to find the associated session and use this
7852 to keep the session alive.
7854 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7856 * gst/rtsp-server/rtsp-session.c:
7857 session: add 5sec to the real session timeout
7858 Allow the session to live 5sec longer before really timing out. This should give
7859 clients some extra time to keep the session active.
7861 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7863 * gst/rtsp-server/rtsp-client.c:
7864 client: replay OK to GET/SET_PARAMETER
7865 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
7866 so that we return OK for those requests.
7868 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7870 * gst/rtsp-server/rtsp-media.c:
7871 * gst/rtsp-server/rtsp-media.h:
7872 media: keep track of active transports
7873 Keep track of which transport is active to avoid closing the connection too
7875 Remove the destination transport also when going to NULL.
7876 Print some stats about the SDES and other RTCP messages we receive from the
7879 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7881 * examples/.gitignore:
7882 * examples/Makefile.am:
7883 * examples/test-sdp.c:
7884 example: add SDP relay example
7886 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7888 * gst/rtsp-server/rtsp-media.c:
7889 media: also count active TCP connections
7891 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7893 * gst/rtsp-server/rtsp-media-factory.c:
7894 * gst/rtsp-server/rtsp-media.c:
7895 * gst/rtsp-server/rtsp-media.h:
7896 rtsp: add support for dynamic elements
7897 Add support for dynamic elements.
7898 Don't set live pipelines back to paused.
7900 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7902 * gst/rtsp-server/rtsp-sdp.c:
7903 sdp: don't add encoding name when absent in caps
7905 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7907 * gst/rtsp-server/rtsp-client.c:
7908 client: warn when we can't do RTP-Info
7910 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7912 * gst/rtsp-server/rtsp-media-factory.c:
7913 factory: factor out the stream construction
7915 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7917 * gst/rtsp-server/rtsp-client.c:
7918 client: only add RTP-Info when we have the info
7919 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
7922 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7927 === release 0.10.3 ===
7929 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7933 - Fixes a bug where it put the wrong verion in pkgconfig
7934 - Link RTP and RTCP sources
7936 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7938 * gst/rtsp-server/rtsp-media.c:
7939 * gst/rtsp-server/rtsp-media.h:
7940 media: link the RTP udpsrc to the session manager
7941 Link the RTP udpsrc and the appsrc to the session manager so that they don't
7942 shut down when the client sends a packet to open firewalls.
7944 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7946 * pkgconfig/gst-rtsp-server.pc.in:
7947 Don't use hard-coded version number in pkg-config file
7949 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7954 === release 0.10.2 ===
7956 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7961 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7964 * common/m4/.gitignore:
7965 * examples/.gitignore:
7966 * pkgconfig/.gitignore:
7967 add some .gitignore files
7969 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7971 * gst/rtsp-server/rtsp-media.c:
7972 media: seek to key frames
7974 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7976 * gst/rtsp-server/rtsp-media.c:
7977 media: emit the unprepared signal by id
7978 Emit the unprepared signal by id instead of name and set the media as
7981 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7983 * gst/rtsp-server/rtsp-media.c:
7984 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
7986 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7988 * gst/rtsp-server/rtsp-server.c:
7989 Added finalize function to GstRTPSPServer to unref session pool and media mapping
7991 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7993 * bindings/vala/gst-rtsp-server-0.10.vapi:
7994 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7995 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7996 Updated vala bindings
7998 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8000 * gst/rtsp-server/Makefile.am:
8001 * gst/rtsp-server/rtsp-client.c:
8002 * gst/rtsp-server/rtsp-media.c:
8003 server: use appsink and appsrc with the API
8004 Use the appsink/appsrc API instead of the signals for higher
8007 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8009 * examples/test-ogg.c:
8010 tests: set the payload type correctly
8012 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8014 * gst/rtsp-server/rtsp-media-factory.c:
8015 factory: connect to the unprepare signal
8016 Connect to the unprepare signal for non-reusable media so that we can remove
8017 them from the cache.
8019 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8021 * gst/rtsp-server/rtsp-media.c:
8022 * gst/rtsp-server/rtsp-media.h:
8023 media: add signal to notify of unprepare
8025 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8027 * gst/rtsp-server/rtsp-media.c:
8028 * gst/rtsp-server/rtsp-media.h:
8029 media: more work on making the media shared
8030 Add a reusable flag to medias, indicating that they can be reused after a state
8034 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8036 * examples/test-readme.c:
8037 examples: mark the example as shared for testing
8039 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8041 * gst/rtsp-server/rtsp-media.c:
8042 * gst/rtsp-server/rtsp-media.h:
8043 client: support shared media
8044 Always perform the state actions even if the target state of the pipeline is
8045 already correct, we still want to add/remove the transports when we are dealing
8047 Keep a counter of the number of active transports for a media so that we can use
8048 this to perform a state change when needed.
8049 Perform a state change of the pipeline only when the first transport was added
8050 or when there are no active transports.
8052 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8054 * gst/rtsp-server/rtsp-client.c:
8055 client: fix refcounting crasher
8056 Don't need to remove the weak refs in the finalize methods, they are already
8057 removed in the dispose.
8058 Don't register the callback with a DestroyNofity.
8060 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8062 * gst/rtsp-server/rtsp-client.c:
8063 Fix rtsp client refcount management in TCP mode.
8064 Don't unref a client ref we never had. Fixes an unref
8065 of an already-free client object after a client
8066 teardown request for me.
8068 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8070 * gst/rtsp-server/rtsp-session.c:
8071 docs: fix typo in API docs
8073 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8075 * gst/rtsp-server/rtsp-media.c:
8077 Keep the udp sources in playing even if we go to paused. unlock the sources when
8079 Add some more debug info.
8080 Only seek when we need to.
8081 Keep track of the position when we go to paused.
8083 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8085 * gst/rtsp-server/rtsp-client.c:
8086 * gst/rtsp-server/rtsp-media.c:
8087 * gst/rtsp-server/rtsp-media.h:
8088 Add beginnings of seeking.
8089 Parse the Range header and perform a seek on the pipeline for the requested
8090 position. It's disabled currently until I figure out what's going wrong.
8092 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8094 * gst/rtsp-server/rtsp-client.c:
8095 allow pause requests for now.
8098 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8100 * gst/rtsp-server/rtsp-client.c:
8101 Remove weak ref on the session in teardown
8102 We need to remove our weakref from the session when we do a teardown because
8103 else we close the TCP connection prematurely.
8105 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8107 * gst/rtsp-server/rtsp-client.c:
8108 * gst/rtsp-server/rtsp-client.h:
8109 * gst/rtsp-server/rtsp-session-pool.c:
8110 Do some more session cleanup
8111 Make session timeout kill the TCP connection that currently watches the
8113 Remove the client timeout property.
8115 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8117 * gst/rtsp-server/rtsp-client.c:
8118 * gst/rtsp-server/rtsp-client.h:
8119 * gst/rtsp-server/rtsp-media.c:
8120 * gst/rtsp-server/rtsp-media.h:
8121 * gst/rtsp-server/rtsp-server.c:
8122 * gst/rtsp-server/rtsp-session.c:
8123 * gst/rtsp-server/rtsp-session.h:
8125 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
8128 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8130 * examples/Makefile.am:
8131 * examples/test-launch.c:
8132 Add example server that takes launch lines
8133 Add an example server that streams any -launch line.
8135 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8137 * examples/test-readme.c:
8138 * gst/rtsp-server/rtsp-client.c:
8139 * gst/rtsp-server/rtsp-media.c:
8140 * gst/rtsp-server/rtsp-media.h:
8141 Add support for live streams
8142 Add support for live streams and ranges
8143 Start on handling TCP data transfer.
8145 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8147 * gst/rtsp-server/rtsp-media.c:
8148 Free the pipeline before other things
8151 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8153 * gst/rtsp-server/rtsp-client.c:
8154 Only free the pending tunnel if there is one
8157 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8159 * gst/rtsp-server/rtsp-client.c:
8160 * gst/rtsp-server/rtsp-client.h:
8161 * gst/rtsp-server/rtsp-media.c:
8162 rtsp-server: Add support for tunneling
8163 Add support for tunneling over HTTP.
8164 Use new connection methods to retrieve the url.
8165 Dispatch messages based on the message type instead of blindly
8166 assuming it's always a request.
8167 Keep track of the watch id so that we can remove it later.
8168 Set the media pipeline to NULL before unreffing the pipeline.
8170 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8172 * gst/rtsp-server/rtsp-client.c:
8173 * gst/rtsp-server/rtsp-client.h:
8174 Fix for channel -> watch rename in gstreamer
8175 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
8177 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8179 * gst/rtsp-server/rtsp-client.c:
8180 * gst/rtsp-server/rtsp-client.h:
8182 Use the async RTSP channels instead of spawning a new thread for each client.
8183 If a sessionid is specified in a request, fail if we don't have the session.
8185 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8187 * gst/rtsp-server/rtsp-media.c:
8188 Add better debug info
8189 Add some better debug info.
8191 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8193 * examples/test-video.c:
8195 Add support for session timeouts in the example.
8197 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8199 * gst/rtsp-server/rtsp-session-pool.c:
8200 * gst/rtsp-server/rtsp-session-pool.h:
8201 Pass GTimeVal around for performance reasons
8202 Get the current time only once and pass it around so that sessions don't have to
8203 get the current time anymore.
8204 Add experimental support for a GSource that dispatches when the session needs to
8207 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8209 * gst/rtsp-server/rtsp-session.c:
8210 * gst/rtsp-server/rtsp-session.h:
8211 Add better support for session timeouts
8212 Add a method to request the number of milliseconds when a session will timeout.
8214 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8216 * gst/rtsp-server/rtsp-media.c:
8217 * gst/rtsp-server/rtsp-media.h:
8218 Add suport for RTP manager monitoring
8219 Add the first stage in monitoring the rtp manager.
8220 Make sure we don't update the state to something we don't want.
8222 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8224 * gst/rtsp-server/rtsp-client.c:
8225 Add support for session keepalive
8226 Get and update the session timeout for all requests. get the session as early as
8229 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8231 * gst/rtsp-server/rtsp-media-factory.h:
8232 * gst/rtsp-server/rtsp-media.c:
8233 * gst/rtsp-server/rtsp-media.h:
8234 Handle media bus messages
8235 Handle media bus messages in a custom mainloop and dispatch them to the
8236 RTSPMedia objects. Let the default implementation handle some common messages.
8238 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8240 * gst/rtsp-server/rtsp-client.c:
8241 * gst/rtsp-server/rtsp-session-pool.c:
8242 * gst/rtsp-server/rtsp-session.c:
8243 Some more session timeout handling
8244 Move the session header setting code to a central place so that we always add
8245 the timeout parameter too.
8246 Handle timeouts by running the session cleanup code.
8247 Stop media before cleaning up.
8249 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8251 * gst/rtsp-server/rtsp-client.c:
8252 * gst/rtsp-server/rtsp-client.h:
8253 Add timeout property
8254 Add a timeout property ot the client and make the other properties into GObject
8257 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8259 * gst/rtsp-server/rtsp-session-pool.c:
8260 Use getters and setters in property code
8261 Use the getters and setters for the timeout property instead of locking
8264 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8266 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
8268 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8270 * gst/rtsp-server/rtsp-session-pool.c:
8271 * gst/rtsp-server/rtsp-session-pool.h:
8272 * gst/rtsp-server/rtsp-session.c:
8273 * gst/rtsp-server/rtsp-session.h:
8274 Add more timeout stuff
8275 Add method to check if a session is expired.
8276 Add method to perform cleanup on a session pool.
8278 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8280 * gst/rtsp-server/rtsp-client.c:
8281 * gst/rtsp-server/rtsp-session-pool.c:
8282 * gst/rtsp-server/rtsp-session-pool.h:
8283 * gst/rtsp-server/rtsp-session.c:
8284 * gst/rtsp-server/rtsp-session.h:
8285 Add beginnings of session timeouts and limits
8286 Add the timeout value to the Session header for unusual timeout values.
8287 Allow us to configure a limit to the amount of active sessions in a pool. Set a
8288 limit on the amount of retry we do after a sessionid collision.
8289 Add properties to the sessionid and the timeout of a session. Keep track of
8290 creation time and last access time for sessions.
8292 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8294 * gst/rtsp-server/rtsp-client.c:
8295 * gst/rtsp-server/rtsp-media.c:
8296 * gst/rtsp-server/rtsp-media.h:
8297 * gst/rtsp-server/rtsp-sdp.c:
8298 * gst/rtsp-server/rtsp-session-pool.c:
8299 * gst/rtsp-server/rtsp-session.c:
8300 * gst/rtsp-server/rtsp-session.h:
8301 Cleanup of sessions and more
8302 Fix the refcounting of media and sessions in the client. Properly clean up the
8303 session data when the client performs a teardown.
8304 Add Server header to responses.
8305 Allow for multiple uri setups in one session.
8306 Add Range header to the PLAY response and add the range attribute to the SDP
8308 Fix the session pool remove method, it used the wrong key in the hashtable. Also
8309 give the ownership of the sessionid to the session object.
8311 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8313 * gst/rtsp-server/rtsp-server.c:
8314 * gst/rtsp-server/rtsp-server.h:
8316 Rename the 'server_port' variable to simply 'port'.
8318 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8321 * gst/rtsp-server/rtsp-client.c:
8322 * gst/rtsp-server/rtsp-media.c:
8323 * gst/rtsp-server/rtsp-media.h:
8324 * gst/rtsp-server/rtsp-session.c:
8325 * gst/rtsp-server/rtsp-session.h:
8326 Rework the way we handle transports for streams
8327 Make the media accept an array of transports for the streams that we have
8328 configured for the play/pause requests.
8329 Implement server states for a client and its media.
8330 Require 0.10.22.1 (git HEAD) of gstreamer.
8332 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8334 * gst/rtsp-server/rtsp-client.c:
8335 * gst/rtsp-server/rtsp-media-factory.c:
8336 Drop const from functions dealing with urls
8337 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
8338 have the right const in them.
8340 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8342 * gst/rtsp-server/rtsp-client.c:
8343 * gst/rtsp-server/rtsp-media.c:
8344 * gst/rtsp-server/rtsp-sdp.c:
8348 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8350 * gst/rtsp-server/rtsp-client.c:
8351 * gst/rtsp-server/rtsp-media-factory.c:
8352 * gst/rtsp-server/rtsp-media.c:
8353 * gst/rtsp-server/rtsp-media.h:
8355 Don't keep a reference to the GstRTSPMedia in the stream.
8356 Free more things when freeing the GstRTSPMedia.
8358 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8361 * gst/rtsp-server/rtsp-media-factory.c:
8362 * gst/rtsp-server/rtsp-media-factory.h:
8363 * gst/rtsp-server/rtsp-media.c:
8364 * gst/rtsp-server/rtsp-media.h:
8365 * gst/rtsp-server/rtsp-server.c:
8366 * gst/rtsp-server/rtsp-server.h:
8367 More docs and small cleanups
8368 Add some more docs and update the README
8369 Cleanup some method names.
8370 Remove an unneeded idx field in the GstRTSPMediaStream
8372 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8375 * examples/Makefile.am:
8376 * examples/test-readme.c:
8377 Add a README and more example code
8378 Add a README file that contains a small introduction on how to use the server
8379 along with the example code explained in the readme.
8381 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8383 * gst/rtsp-server/rtsp-media.c:
8384 * gst/rtsp-server/rtsp-server.c:
8385 Fix some leaks and change default port
8386 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
8387 we finished the initial preroll. If we keep them locked, setting the pipeline to
8388 NULL will not stop and clean up the sources correctly.
8389 Change the default RTSP port to 8554 aka the official alternative RTSP port.
8391 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8393 * gst/rtsp-server/rtsp-session.c:
8394 * gst/rtsp-server/rtsp-session.h:
8395 Cleanups to the session object
8396 Remove some unneeded variables in the session state of a stream such as the
8397 owner media and the server transport.
8398 Get the configuration of a media stream in a session based on the media_stream
8399 in the original object instead of our cached index.
8400 Free more data in the finalize method.
8402 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8404 * gst/rtsp-server/rtsp-client.c:
8405 * gst/rtsp-server/rtsp-client.h:
8406 Cleanups and reuse media from DESCRIBE
8407 Handle thread create errors.
8408 Rename some internal methods to better match what they actually do.
8409 Handle misconfiguration of session_pool and media_mapping gracefully.
8410 Cache the DESCRIBE media and uri in the client connection and reuse them when
8411 we receive a SETUP request in the same connection for the same uri.
8412 Cleanup the client connection object.
8414 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8416 * gst/rtsp-server/rtsp-media-factory.c:
8417 * gst/rtsp-server/rtsp-media-factory.h:
8418 * gst/rtsp-server/rtsp-media.c:
8419 * gst/rtsp-server/rtsp-media.h:
8420 Add shared properties to media and factory
8421 Add the shared property to media.
8422 Implement some simple caching in the factory depending on if the media is shared
8425 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8427 * gst/rtsp-server/rtsp-client.c:
8428 Add a little comment
8429 Add some comment about the content-base header.
8431 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8433 * examples/Makefile.am:
8435 * examples/test-mp4.c:
8436 * examples/test-ogg.c:
8437 * examples/test-video.c:
8438 * gst/rtsp-server/Makefile.am:
8439 * gst/rtsp-server/rtsp-client.c:
8440 * gst/rtsp-server/rtsp-client.h:
8441 * gst/rtsp-server/rtsp-media-factory.c:
8442 * gst/rtsp-server/rtsp-media-factory.h:
8443 * gst/rtsp-server/rtsp-media.c:
8444 * gst/rtsp-server/rtsp-media.h:
8445 * gst/rtsp-server/rtsp-sdp.c:
8446 * gst/rtsp-server/rtsp-sdp.h:
8447 * gst/rtsp-server/rtsp-server.c:
8448 * gst/rtsp-server/rtsp-server.h:
8449 * gst/rtsp-server/rtsp-session.c:
8450 * gst/rtsp-server/rtsp-session.h:
8451 Reorganize things, prepare for media sharing
8452 Added various other test server examples
8453 Move the SDP message generation to a separate helper.
8454 Refactor common code for finding the session.
8455 Add content-base for realplayer compatibility
8456 Clean up request uris before processing for better vlc compatibility.
8457 Move prerolling and pipeline construction to the RTSPMedia object.
8458 Use multiudpsink for future pipeline reuse.
8460 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8466 === release 0.10.1 ===
8468 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8474 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8476 * bindings/vala/Makefile.am:
8478 Add more directories and files to the dist.
8480 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8482 * bindings/python/Makefile.am:
8483 * bindings/python/rtspserver.override:
8484 Fixed compile error of python bindings
8486 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8488 * bindings/vala/gst-rtsp-server-0.10.vapi:
8489 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8490 Marked values as nullable accordingly
8492 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8494 * bindings/vala/gst-rtsp-server-0.10.vapi:
8495 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8496 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8497 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8498 Updated Vala bindings
8500 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8502 * gst/rtsp-server/rtsp-client.c:
8503 * gst/rtsp-server/rtsp-media-mapping.c:
8504 * gst/rtsp-server/rtsp-media-mapping.h:
8505 * gst/rtsp-server/rtsp-media.h:
8506 * gst/rtsp-server/rtsp-session-pool.h:
8507 Cleanups and doc updates
8508 Add some more documentation and do some minor cleanups here and there.
8510 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8512 * gst/rtsp-server/rtsp-client.c:
8513 * gst/rtsp-server/rtsp-media-factory.c:
8514 * gst/rtsp-server/rtsp-media-factory.h:
8515 * gst/rtsp-server/rtsp-media.c:
8516 * gst/rtsp-server/rtsp-media.h:
8517 * gst/rtsp-server/rtsp-session.c:
8518 * gst/rtsp-server/rtsp-session.h:
8520 Rename GstRTSPMediaBin to GstRTSPMedia
8521 Parse the request url into a GstRTSPUri object and pass this object to the
8522 various handlers and methods that require the uri.
8524 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8528 Add some more docs and remove some old code from the example.
8530 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8532 * gst/rtsp-server/rtsp-client.c:
8533 Handle state change failures better
8534 Handle state change failures better when changing the state of the pipeline to
8537 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8539 * gst/rtsp-server/rtsp-media-factory.c:
8540 * gst/rtsp-server/rtsp-media-factory.h:
8541 Make element creation more extendible
8542 Add get_element vmethod to the default MediaFactory so that subclasses can just
8543 override that method and still use the default logic for making a MediaBin from
8546 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8549 * gst/rtsp-server/Makefile.am:
8550 * gst/rtsp-server/rtsp-client.c:
8551 * gst/rtsp-server/rtsp-client.h:
8552 * gst/rtsp-server/rtsp-media-factory.c:
8553 * gst/rtsp-server/rtsp-media-factory.h:
8554 * gst/rtsp-server/rtsp-media-mapping.c:
8555 * gst/rtsp-server/rtsp-media-mapping.h:
8556 * gst/rtsp-server/rtsp-media.c:
8557 * gst/rtsp-server/rtsp-media.h:
8558 * gst/rtsp-server/rtsp-server.c:
8559 * gst/rtsp-server/rtsp-server.h:
8560 * gst/rtsp-server/rtsp-session.c:
8561 * gst/rtsp-server/rtsp-session.h:
8562 Make the server handle arbitrary pipelines
8563 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
8564 The GstMediaBin object has a handle to a bin with elements and to a list of
8565 GstMediaStream objects that this bin produces.
8566 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
8567 with methods to register and remove those mappings.
8568 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
8569 used by the server instance.
8570 Modify the example application so that it shows how to create custom pipelines
8571 attached to a specific mount point.
8572 Various misc cleanps.
8574 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8576 * gst/rtsp-server/rtsp-server.c:
8577 * gst/rtsp-server/rtsp-server.h:
8578 Allow setting a custom media factory for a server
8580 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8582 * gst/rtsp-server/rtsp-client.c:
8583 * gst/rtsp-server/rtsp-client.h:
8584 Allow setting a custom media factory for a client.
8586 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8588 * gst/rtsp-server/Makefile.am:
8589 Add Makefile entry for the media factory
8591 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8593 * gst/rtsp-server/rtsp-media-factory.c:
8594 * gst/rtsp-server/rtsp-media-factory.h:
8595 Add media factory to map urls to media pipeline objects.
8597 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8599 * gst/rtsp-server/rtsp-media.c:
8600 * gst/rtsp-server/rtsp-media.h:
8601 Add comments. Remove unused field
8603 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8605 * gst/rtsp-server/rtsp-session-pool.c:
8606 * gst/rtsp-server/rtsp-session-pool.h:
8607 Allow custom session pools to override the session id allocation algorithms Add some comments.
8609 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8611 * gst/rtsp-server/rtsp-session.h:
8614 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8616 * gst/rtsp-server/rtsp-client.c:
8617 * gst/rtsp-server/rtsp-client.h:
8618 Move the connection code in one place Add some comments
8620 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8622 * gst/rtsp-server/rtsp-server.c:
8623 * gst/rtsp-server/rtsp-server.h:
8624 Make vmethod to create and accept new clients. Add some docs.
8626 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8628 * gst/rtsp-server/rtsp-server.c:
8629 * gst/rtsp-server/rtsp-server.h:
8630 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
8632 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8634 * gst/rtsp-server/rtsp-client.c:
8635 * gst/rtsp-server/rtsp-client.h:
8636 Name the parameters more appropriately.
8638 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8640 * gst/rtsp-server/rtsp-session-pool.c:
8641 Do some more cleanup of the session pool.
8643 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8645 * gst/rtsp-server/Makefile.am:
8646 * gst/rtsp-server/rtsp-client.c:
8647 Check if return value of gst_rtsp_session_get_media is not NULL
8649 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8651 * gst/rtsp-server/Makefile.am:
8652 Install rtsp-session and rtsp-session-pool headers
8654 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8659 * bindings/python/Makefile.am:
8660 * bindings/python/arg-types.py:
8661 * bindings/python/codegen/Makefile.am:
8662 * bindings/python/codegen/__init__.py:
8663 * bindings/python/codegen/argtypes.py:
8664 * bindings/python/codegen/code-coverage.py:
8665 * bindings/python/codegen/codegen.py:
8666 * bindings/python/codegen/definitions.py:
8667 * bindings/python/codegen/defsparser.py:
8668 * bindings/python/codegen/docextract.py:
8669 * bindings/python/codegen/docgen.py:
8670 * bindings/python/codegen/fileprefix.override:
8671 * bindings/python/codegen/fileprefixmodule.c:
8672 * bindings/python/codegen/h2def.py:
8673 * bindings/python/codegen/mergedefs.py:
8674 * bindings/python/codegen/mkskel.py:
8675 * bindings/python/codegen/override.py:
8676 * bindings/python/codegen/reversewrapper.py:
8677 * bindings/python/codegen/scmexpr.py:
8678 * bindings/python/rtspserver-types.defs:
8679 * bindings/python/rtspserver.defs:
8680 * bindings/python/rtspserver.override:
8681 * bindings/python/rtspservermodule.c:
8683 Add python bindings.
8685 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8687 * bindings/Makefile.am:
8689 Don't go into python dir when requirements for python bindings are missing
8691 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8693 * bindings/Makefile.am:
8694 * bindings/vala/Makefile.am:
8696 Install Vala bindings if vala is available
8698 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8700 * bindings/vala/gst-rtsp-server-0.10.deps:
8701 * bindings/vala/gst-rtsp-server-0.10.vapi:
8702 * bindings/vala/gst-rtsp-server.vapi:
8703 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
8704 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8705 * bindings/vala/packages/gst-rtsp-server-0.10.files:
8706 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8707 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8708 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
8709 * bindings/vala/packages/gst-rtsp-server.deps:
8710 * bindings/vala/packages/gst-rtsp-server.excludes:
8711 * bindings/vala/packages/gst-rtsp-server.files:
8712 * bindings/vala/packages/gst-rtsp-server.gi:
8713 * bindings/vala/packages/gst-rtsp-server.metadata:
8714 * bindings/vala/packages/gst-rtsp-server.namespace:
8715 Regenerated Vala bindings
8717 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8719 * bindings/vala/gst-rtsp-server.vapi:
8720 * bindings/vala/packages/gst-rtsp-server.metadata:
8721 Fixed typo in included headers for vala bindings
8723 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8727 * pkgconfig/Makefile.am:
8728 * pkgconfig/gst-rtsp-server.pc.in:
8729 Added pkgconfig file
8731 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8733 * bindings/vala/gst-rtsp-server.vapi:
8734 * bindings/vala/packages/gst-rtsp-server.excludes:
8735 * bindings/vala/packages/gst-rtsp-server.gi:
8736 * bindings/vala/packages/gst-rtsp-server.metadata:
8737 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
8739 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8741 * bindings/vala/gst-rtsp-server.vapi:
8742 * bindings/vala/packages/gst-rtsp-server.deps:
8743 * bindings/vala/packages/gst-rtsp-server.files:
8744 * bindings/vala/packages/gst-rtsp-server.gi:
8745 * bindings/vala/packages/gst-rtsp-server.metadata:
8746 * bindings/vala/packages/gst-rtsp-server.namespace:
8749 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
8751 * gst/rtsp-server/rtsp-session.c:
8752 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
8754 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8756 * examples/Makefile.am:
8757 * gst/rtsp-server/Makefile.am:
8758 Put GStreamer version in library name
8760 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8762 * examples/Makefile.am:
8763 * gst/rtsp-server/Makefile.am:
8764 Fix some issues to pass distcheck
8766 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8768 * gst/rtsp-server/rtsp-server.c:
8769 Added port property to GstRTSPServer class.
8771 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8776 * examples/Makefile.am:
8779 * gst/rtsp-server/Makefile.am:
8780 * gst/rtsp-server/rtsp-client.c:
8781 * gst/rtsp-server/rtsp-client.h:
8782 * gst/rtsp-server/rtsp-media.c:
8783 * gst/rtsp-server/rtsp-media.h:
8784 * gst/rtsp-server/rtsp-server.c:
8785 * gst/rtsp-server/rtsp-server.h:
8786 * gst/rtsp-server/rtsp-session-pool.c:
8787 * gst/rtsp-server/rtsp-session-pool.h:
8788 * gst/rtsp-server/rtsp-session.c:
8789 * gst/rtsp-server/rtsp-session.h:
8792 * src/rtsp-client.c:
8793 * src/rtsp-client.h:
8796 * src/rtsp-server.c:
8797 * src/rtsp-server.h:
8798 * src/rtsp-session-pool.c:
8799 * src/rtsp-session-pool.h:
8800 * src/rtsp-session.c:
8801 * src/rtsp-session.h:
8802 Split in library and example program
8804 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8806 * src/rtsp-client.h:
8807 Removed obsolete variable
8809 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8811 * src/rtsp-client.c:
8812 * src/rtsp-client.h:
8813 Removed pipeline variable GstRTSPClient, because it's only used in one function
8815 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8818 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
8820 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
8822 * src/rtsp-session.c:
8823 Initialize some more vars.
8825 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
8827 * src/rtsp-session.c:
8828 Initialize variable to avoid compiler warning.
8830 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
8833 Add a reasonable generic .gitignore