3 2015-10-30 Sebastian Dröge <slomo@coaxion.net>
8 2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
10 * gst/rtsp-server/rtsp-stream.c:
11 rtsp-stream: Always unref return value of gst_object_get_parent()
12 Fixes a leak of a GstBin in the udp-mcast case.
13 https://bugzilla.gnome.org/show_bug.cgi?id=756968
15 2015-09-29 13:04:53 +0100 Tim-Philipp Müller <tim@centricular.com>
18 common: update for new suppression
19 Makes check-valgrind pass with glib 2.46
21 2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
23 * gst/rtsp-server/rtsp-media.c:
24 rtsp-media: Take reference to media that will be prepared
25 default_prepare() takes a transfer-none reference GstRTSPMedia object.
26 Later on a g_idle_source_new() is created and a pointer to the media
27 object is passed as user data. If the media is freed before the idle
28 source is dispatched the media object pointer is invalid, but the idle
29 source callback expects it to still be valid. To fix this a reference to
30 the media object is taken when registering the source callback function
31 and a corresponding release of the reference is done when the souce is
33 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
37 2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
43 * gst-rtsp-server.doap:
46 === release 1.5.91 ===
48 2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
54 * gst-rtsp-server.doap:
57 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
59 * docs/libs/gst-rtsp-server-sections.txt:
60 * gst/rtsp-server/rtsp-stream.c:
61 stream: fix docs for recently-added get/set_buffer_size API
62 https://bugzilla.gnome.org/show_bug.cgi?id=749095
64 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
66 * gst/rtsp-server/rtsp-media.c:
67 rtsp-media: Don't crash on encrypted RTX SDP
68 In parse_keymgmt(), don't mutate the input string that's been passed
69 as const, especially since we might need the original value again if
70 the same key info applies to multiple streams (RTX, for example).
71 https://bugzilla.gnome.org/show_bug.cgi?id=754753
73 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
75 * examples/test-mp4.c:
76 test-mp4: Support filenames with spaces in them. Error out on too few arguments
78 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
80 * examples/test-record.c:
81 test-record: Check parameter count and print out help
82 If no launch pipeline was supplied, print out some help
84 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
86 * gst/rtsp-server/rtsp-media.c:
87 * gst/rtsp-server/rtsp-stream.c:
88 * gst/rtsp-server/rtsp-stream.h:
89 rtsp-stream: Implement UDP buffer size setting.
90 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
92 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
93 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
95 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
97 * gst/rtsp-server/rtsp-media.h:
98 rtsp-media: Fix small typo causing gtk-doc to complain
100 === release 1.5.90 ===
102 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
108 * gst-rtsp-server.doap:
111 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
113 * gst/rtsp-server/rtsp-media-factory.c:
114 media-factory: get port number through gst_rtsp_url_get_port
115 https://bugzilla.gnome.org/show_bug.cgi?id=753473
117 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
119 * tests/check/gst/media.c:
120 media-test: Removing unnecessary assertion
121 https://bugzilla.gnome.org/show_bug.cgi?id=753385
123 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
125 * gst/rtsp-server/rtsp-server.c:
126 Document that source keeps a ref on server until it's destroyed
127 https://bugzilla.gnome.org/show_bug.cgi?id=749227
129 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
131 * tests/check/gst/media.c:
132 media-test: Test for multiple dynamic payload
133 https://bugzilla.gnome.org/show_bug.cgi?id=753385
135 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
137 * gst/rtsp-server/rtsp-media.c:
138 media: Only add fakesink once per pipeline
139 The intention is to prevent going PLAYING state before pads are created.
140 If there was mutilple dynamic payload, it would leak few fakesink and
141 actually prevent from ever reaching playing state.
142 https://bugzilla.gnome.org/show_bug.cgi?id=753385
144 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
146 * gst/rtsp-server/rtsp-media.c:
147 Revert "rtsp-media: Only add 1 fakesink per pipeline"
148 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
150 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
152 * gst/rtsp-server/rtsp-media.c:
153 rtsp-media: Only add 1 fakesink per pipeline
154 There should be only one fakesink per pipeline, not per dynpay. This
155 would lead to element naming clash.
157 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
159 * gst/rtsp-server/rtsp-media.c:
160 rtsp-media: assertion error due to wrong condition check
161 In media to caps function, reserved_keys array is being used for variable i,
162 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
163 changed it to variable j
164 https://bugzilla.gnome.org/show_bug.cgi?id=753009
166 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
168 * gst/rtsp-server/rtsp-media.c:
169 rtsp-media: Strip keys from the fmtp that we use internally in our caps
170 Skip keys from the fmtp, which we already use ourselves for the
171 caps. Some software is adding random things like clock-rate into
172 the fmtp, and we would otherwise here set a string-typed clock-rate
173 in the caps... and thus fail to create valid RTP caps
174 https://bugzilla.gnome.org/show_bug.cgi?id=753009
176 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
178 * gst/rtsp-server/rtsp-thread-pool.c:
179 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
180 https://bugzilla.gnome.org/show_bug.cgi?id=752640
182 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
185 Automatic update of common submodule
186 From f74b2df to 9aed1d7
188 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
193 === release 1.5.2 ===
195 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
201 * gst-rtsp-server.doap:
204 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
206 * gst/rtsp-server/rtsp-client.c:
207 * gst/rtsp-server/rtsp-client.h:
208 * tests/check/gst/client.c:
209 rtsp-client: allow application to decide what requirements are supported
210 Add "check-requirements" signal and vfunc to allow application
211 (and subclasses) to check the requirements.
212 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
213 https://bugzilla.gnome.org/show_bug.cgi?id=749417
215 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
218 Automatic update of common submodule
219 From 6015d26 to f74b2df
221 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
223 * gst/rtsp-server/rtsp-media.c:
224 rtsp-media: Always use real payloader when creating streams
225 A bin that contains the real payloader might be used as payloader. In this
226 case we have to get the real payloader for the various properties it provides.
227 Example use cases for this are bins that payload some media and then have
228 additional elements that add metadata or RTP extension headers to the stream.
229 https://bugzilla.gnome.org/show_bug.cgi?id=750800
231 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
233 * examples/test-netclock-client.c:
234 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
236 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
238 * examples/test-netclock-client.c:
239 * examples/test-netclock.c:
240 test-netclock: Use new ntp-time-source property on rtpbin
241 Select the clock time to be used as NTP time source. This allows proper
242 synchronization between receivers, independent of sharing base times, and just
243 requires them to use the same clock.
245 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
247 * examples/test-netclock-client.c:
248 * examples/test-netclock.c:
249 test-netclock: Setting the same base time on sender and receiver is not necessary
250 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
252 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
254 * gst/rtsp-server/rtsp-stream.c:
255 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
256 https://bugzilla.gnome.org/show_bug.cgi?id=750764
258 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
260 * docs/libs/gst-rtsp-server.types:
261 docs: add missing types
262 https://bugzilla.gnome.org/show_bug.cgi?id=750764
264 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
266 * docs/libs/gst-rtsp-server-sections.txt:
267 docs: add missing apis
268 https://bugzilla.gnome.org/show_bug.cgi?id=750764
270 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
272 * examples/test-netclock-client.c:
273 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
275 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
277 * docs/libs/gst-rtsp-server-sections.txt:
278 * gst/rtsp-server/rtsp-auth.c:
279 * gst/rtsp-server/rtsp-auth.h:
280 GstRTSPAuth: Add client certificate authentication support
281 https://bugzilla.gnome.org/show_bug.cgi?id=750471
283 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
285 * examples/test-netclock-client.c:
286 test-netclock-client: Use new GstClock API to wait for clock synchronization
288 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
290 * examples/test-netclock-client.c:
291 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
292 A mainloop is needed to get glimagesink to display something on OSX, and
293 the source-setup signal just makes things a little bit easier.
295 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
298 Automatic update of common submodule
299 From d9a3353 to 6015d26
301 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
304 Automatic update of common submodule
305 From d37af32 to d9a3353
307 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
310 Automatic update of common submodule
311 From 21ba2e5 to d37af32
313 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
316 Automatic update of common submodule
317 From c408583 to 21ba2e5
319 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
321 * docs/libs/Makefile.am:
322 docs: remove variables that we define in the snippet from common
323 This is syncing our Makefile.am with upstream gtkdoc.
325 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
328 Automatic update of common submodule
329 From 44a3517 to c408583
331 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
336 === release 1.5.1 ===
338 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
344 * gst-rtsp-server.doap:
347 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
349 * gst/rtsp-server/rtsp-client.c:
350 rtsp-client: No flush during Teardown.
351 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
352 backlog is empty it can happen that just a part of a message will be
353 sent and rest is in backlog queue. If then flush during teardown
354 just a part of message will be sent.This can lead to client miss
355 teardown response since it expect to get the last part of message.
356 The flushing during teardown was introduced to fix a deadlock that now
357 is fixed more generally in handle_request by temporary setting backlog
359 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
361 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
363 * tests/check/Makefile.am:
364 tests: Use AM_TESTS_ENVIRONMENT
365 Needed by the new automake test runner and the
366 current version of the common submodule.
368 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
370 * gst/rtsp-server/rtsp-media.h:
371 * gst/rtsp-server/rtsp-stream.h:
372 rtsp-server: Use single-include rtsp header to make sure we get all definitions
374 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
376 * gst/rtsp-server/rtsp-media.c:
377 rtsp-media: Mark some more functions static
379 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
381 * gst/rtsp-server/rtsp-media.c:
382 rtsp-media: Only unblock the media in suspend() when actually changing the state
383 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
385 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
387 * examples/test-video-rtx.c:
388 examples: Use AVPF profile for the RTX example
390 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
392 * gst/rtsp-server/rtsp-sdp.c:
393 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
395 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
397 * gst/rtsp-server/rtsp-stream.c:
398 rtsp-stream: get valid clock-rate from last-sample
399 clock-rate in last-sample's caps is integer, not unsigned.
400 To get this value properly, variable needs to be type-casted to int.
401 https://bugzilla.gnome.org/show_bug.cgi?id=747614
403 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
407 autogen.sh: only run autopoint if gettext requested in configure.ac
408 Not just because there happens to be a po directory.
409 https://bugzilla.gnome.org/show_bug.cgi?id=748058
411 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
414 Revert "configure.ac: uncomment gettext version setup"
415 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
416 We don't need a gettext setup here and there's no po
417 directory either, so no reason why autopoint would be
418 run in the first place.
419 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
421 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
423 * examples/test-multicast.c:
424 * examples/test-multicast2.c:
425 * examples/test-sdp.c:
426 * examples/test-video-rtx.c:
427 * examples/test-video.c:
428 * tests/test-cleanup.c:
429 * tests/test-reuse.c:
430 Fix timeout function signatures across tests and examples
432 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
434 * tests/check/Makefile.am:
435 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
436 Make sure the test environment is set up.
437 https://bugzilla.gnome.org//show_bug.cgi?id=747624
439 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
442 configure: bump automake requirement to 1.14 and autoconf to 2.69
443 This is only required for builds from git, people can still
444 build tarballs if they only have older autotools.
445 https://bugzilla.gnome.org//show_bug.cgi?id=747624
447 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
450 configure.ac: uncomment gettext version setup
451 Fixes autogen.sh. It would run autopoint, which would complain
452 that it could not find the gettext version in configure.ac.
453 https://bugzilla.gnome.org/show_bug.cgi?id=748058
455 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
457 * examples/test-video-rtx.c:
458 test-video-rtx: set exact payload type to PCMA payloader
459 Setting wrong payload type causes failure to do retransmission through audio stream
460 https://bugzilla.gnome.org/show_bug.cgi?id=747839
462 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
464 * gst/rtsp-server/rtsp-media.c:
465 * gst/rtsp-server/rtsp-stream.c:
466 * gst/rtsp-server/rtsp-stream.h:
467 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
468 Because of duplicated g_signal_connect for request-aux-sender signal,
469 wrong stream pointer is passed to the signal handler.
470 Instead of passing each stream, pass stream array and get the relevant stream.
471 https://bugzilla.gnome.org/show_bug.cgi?id=747839
473 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
477 Update autogen.sh to latest version from common
478 Fixes build after aclocal_check etc. helpers have been removed.
480 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
483 Automatic update of common submodule
484 From bc76a8b to c8fb372
486 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
488 * gst/rtsp-server/rtsp-stream.c:
489 rtsp-stream: Limit the queues to 1 buffer
490 We only need them to be able to pre-roll, queueing up more data here
491 is only going to harm latency and memory usage.
493 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
495 * gst/rtsp-server/rtsp-stream.c:
496 rtsp-stream: Update comment and ASCII art to the latest code
497 We have a queue in front of the udpsink too to prevent the pipeline from
500 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
502 * gst/rtsp-server/rtsp-stream.c:
503 rtsp-media: Properly return first rtptime
504 Instead we where returning first GstBuffer timestamp. This would result
505 in clock skew and unwanted behaviour in RTSP playback.
506 https://bugzilla.gnome.org/show_bug.cgi?id=746479
508 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
510 * gst/rtsp-server/rtsp-stream.c:
511 rtsp-stream: Don't leave buffer mapped
512 If the seq is NULL, the RTP buffer was left mapped. We should always
515 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
520 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
522 * gst/rtsp-server/rtsp-media-factory.c:
523 * tests/check/gst/client.c:
524 Fix double semicolons
526 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
528 * gst/rtsp-server/rtsp-stream.c:
529 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
530 This gives more accurate values than asking the payloader. There might be
531 queueing happening between the payloader and the sink.
532 https://bugzilla.gnome.org/show_bug.cgi?id=745704
534 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
536 * gst/rtsp-server/rtsp-media.c:
537 rtsp-media: Don't seek for PLAY if the position will not change
538 https://bugzilla.gnome.org/show_bug.cgi?id=745704
540 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
542 * gst/rtsp-server/rtsp-media.c:
543 rtsp-media: Don't include payload type in the caps for framesize
544 When the sdp media attribute framesize are converted to caps
545 the <payload> should not be included.
546 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
547 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
549 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
551 * gst/rtsp-server/rtsp-sdp.c:
552 rtsp-sdp: add payload type to the sdp framesize attribute
553 The sdp framesize attribute is desribed in RFC6064. It is specified
554 for payloading of H263 and has the following form
555 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
556 should be added to the caps in a payloader and the <payload type> should
557 be added by the rtsp-server.
558 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
560 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
562 * examples/test-uri.c:
563 examples: test-uri: fix tainted variable
564 Insignificant but this keeps Coverity happy.
567 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
569 * examples/.gitignore:
570 * examples/Makefile.am:
571 * examples/test-netclock-client.c:
572 * examples/test-netclock.c:
573 examples: Add a simple example of network synch for live streams.
574 An example server and client that works for synchronising live streams
575 only - as it can't support pause/play.
577 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
579 * gst/rtsp-server/rtsp-media-factory.c:
580 * gst/rtsp-server/rtsp-media-factory.h:
581 rtsp-media-factory: Add functions to set/get the media gtype
582 Allow specifying the GType of a GstRtspMedia subclass to create
583 as a simpler way to get the factory to create a custom
584 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
586 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
588 * gst/rtsp-server/rtsp-media.c:
589 rtsp-media: fix double unlock in _get_buffer_size()
590 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
591 because of double g_mutex_unlock () usage.
592 https://bugzilla.gnome.org/show_bug.cgi?id=745434
594 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
596 * gst/rtsp-server/rtsp-session-pool.c:
597 * gst/rtsp-server/rtsp-session.c:
598 * gst/rtsp-server/rtsp-session.h:
599 rtsp-session: Use monotonic time for RTSP session timeout
600 Changed RTSP session timeout handling to monotonic time
601 and deprecating the API for current system time.
602 This fixes timeouts when the system time changes.
603 https://bugzilla.gnome.org/show_bug.cgi?id=743346
605 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
607 * gst/rtsp-server/rtsp-client.c:
608 * gst/rtsp-server/rtsp-media.c:
609 rtsp-client: Only error out in PLAY if seeking actually failed
610 If the media was just not seekable, we continue from whatever position we are
611 and let the client decide if that is what is wanted or not.
612 Only if the actual seek failed, we can't really recover and should error out.
614 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
616 * gst/rtsp-server/rtsp-stream.c:
617 rtsp-stream: Add necessary queues between tee and multiudpsink
618 https://bugzilla.gnome.org/show_bug.cgi?id=744379
620 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
622 * gst/rtsp-server/rtsp-client.c:
623 * gst/rtsp-server/rtsp-media.c:
624 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
625 Instead error out properly the same way as if the SEEKING query already
628 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
630 * gst/rtsp-server/rtsp-stream.h:
631 rtsp-stream: minor code formatting fix
633 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
635 * gst/rtsp-server/rtsp-media.c:
636 rtsp-media: fix logic for collect_streams
637 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
638 all streams it knows if it got any, and can check if the transport mode is OK.
641 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
643 * gst/rtsp-server/rtsp-media.c:
644 rtsp-media: Don't set the transport mode based on what elements we find
645 Just print a warning if the one that was set before disagrees with what
646 elements we found. It must already be set to something before as this
647 function is called after we received the SDP from ANNOUNCE in RECORD mode,
648 and we would reject ANNOUNCE if the RECORD flag was not set.
650 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
652 * tests/check/gst/rtspserver.c:
653 tests: rtspserver: rename shadowed variable
654 We have two different 'sink' variables here,
655 rename one of them for clarity.
657 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
659 * gst/rtsp-server/rtsp-client.c:
660 rtsp-client: fix awkward if clause
662 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
664 * examples/test-uri.c:
665 examples: test-uri: improve uri argument handling and accept file names
666 Print an error if the argument passed is not a URI and can't
667 be converted into one, or no arguments have been provided.
669 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
671 * examples/test-uri.c:
672 examples: test-uri: don't remove mount point after 10 seconds
673 It's very irritating when trying to test stuff repeatedly
674 and serves no real purpose other than showing that it can
677 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
679 * examples/.gitignore:
680 examples: add new test-record to .gitignore
682 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
684 * examples/test-record.c:
685 * gst/rtsp-server/rtsp-client.c:
686 * gst/rtsp-server/rtsp-media-factory.c:
687 * gst/rtsp-server/rtsp-media-factory.h:
688 * gst/rtsp-server/rtsp-media.c:
689 * gst/rtsp-server/rtsp-media.h:
690 * tests/check/gst/rtspserver.c:
691 rtsp-media: Use flags to distinguish between PLAY and RECORD media
693 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
695 * examples/test-record.c:
696 test-record: Set latency for playback-style example to 2s instead of 200ms
698 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
700 * tests/check/gst/rtspserver.c:
701 tests: add some unit tests for ANNOUNCE and RECORD
702 https://bugzilla.gnome.org/show_bug.cgi?id=743175
704 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
706 * gst/rtsp-server/rtsp-client.c:
707 rtsp-client: fix a couple of leaks in handle_announce
709 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
711 * gst/rtsp-server/rtsp-media-factory.c:
712 * gst/rtsp-server/rtsp-media-factory.h:
713 * gst/rtsp-server/rtsp-media.c:
714 * gst/rtsp-server/rtsp-media.h:
715 rtsp-media: Expose latency setting for setting the rtpbin latency
717 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
719 * examples/test-record.c:
720 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
722 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
724 * gst/rtsp-server/rtsp-stream.c:
725 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
727 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
729 * examples/Makefile.am:
730 * examples/test-record.c:
731 * gst/rtsp-server/rtsp-client.c:
732 * gst/rtsp-server/rtsp-client.h:
733 * gst/rtsp-server/rtsp-media-factory.c:
734 * gst/rtsp-server/rtsp-media-factory.h:
735 * gst/rtsp-server/rtsp-media.c:
736 * gst/rtsp-server/rtsp-media.h:
737 * gst/rtsp-server/rtsp-session-media.c:
738 * gst/rtsp-server/rtsp-stream.c:
739 * gst/rtsp-server/rtsp-stream.h:
740 Add initial support for RECORD
741 We currently only support media that is RECORD or PLAY only, not both at once.
742 https://bugzilla.gnome.org/show_bug.cgi?id=743175
744 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
746 * gst/rtsp-server/rtsp-stream.c:
747 rtsp-stream: RTCP and RTP transport cache cookies seperated
748 RTCP packets were not sent because the same tr_cache_cookie was used for
749 both RTP and RTCP. So only one of the tr_cache lists were populated
750 depending on which one was sent first. If the tr_cache list is not
751 populated then no packets can be sent. Most often this happened to be
752 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
753 resulted in both the tr_cache_lists to be populated regardless of which
755 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
757 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
759 * gst/rtsp-server/rtsp-stream.c:
760 rtsp-stream: fix false compiler warning
761 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
763 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
765 * gst/rtsp-server/rtsp-client.c:
766 rtsp-client: log interleaved data received
768 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
770 * gst/rtsp-server/rtsp-client.c:
771 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
773 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
775 * gst/rtsp-server/rtsp-client.c:
776 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
778 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
780 * gst/rtsp-server/rtsp-client.c:
781 rtsp-client: Use a random session ID in the SDP
782 RFC4566 Section 5.2 says that it should make the username, session id,
783 nettype, addrtype and unicast address tuple globally unique. Always using
784 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
785 Instead let's create a 64 bit random number, which at least brings us
786 closer to the goal of global uniqueness.
787 https://tools.ietf.org/html/rfc4566#section-5.2
789 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
791 * examples/test-launch.c:
792 * examples/test-mp4.c:
793 * examples/test-ogg.c:
794 * examples/test-uri.c:
795 examples: Don't call gst_init() and gst_get_option_group()
796 The latter calls the former at the appropriate time.
798 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
800 * gst/rtsp-server/rtsp-client.c:
801 rtsp-client: Drop trailing \0 of RTSP DATA messages
802 We add a trailing \0 in GstRTSPConnection to make parsing of
803 string message bodies easier (e.g. the SDP from DESCRIBE) but
804 for actual data this means we have to drop it or otherwise
807 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
809 * gst/rtsp-server/rtsp-stream.c:
810 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
811 Fixes crash when two threads access handle_new_sample() at the same
812 time, one for RTP, one for RTCP.
813 Otherwise, when iterating over the transports cache, it might be modified by
814 another thread at the same time if the transports cookie has changed.
815 https://bugzilla.gnome.org/show_bug.cgi?id=742954
817 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
819 * gst/rtsp-server/rtsp-stream.c:
820 rtsp-stream: Set format=TIME on our app sources for TCP
822 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
824 * gst/rtsp-server/rtsp-session-pool.c:
825 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
826 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
827 RFC 2326 states that session IDs may consist of alphanumeric as well as
828 the safe characters $-_.+ -- N.B. the percent character is not allowed.
829 Previously the session ID was URI-escaped, this meant that any character
830 which was not alphanumeric or any of the characters +-._~ would be
831 percent encoded. While the RFC (surprisingly) mentions that linear white
832 space in session IDs should be URI-escaped, it does not say anything
833 about other characters. Moreover no white space is allowed in the
834 session ID. Finally the percent character which is the result of
835 URI-escaping is not allowed in a session ID.
836 So there is no reason to do any URI-escaping, and now it is removed.
837 https://bugzilla.gnome.org/show_bug.cgi?id=742869
839 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
842 Automatic update of common submodule
843 From f2c6b95 to bc76a8b
845 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
848 Fix 'make check' from top-level directory
850 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
852 * examples/test-launch.c:
853 * examples/test-mp4.c:
854 * examples/test-ogg.c:
855 * examples/test-uri.c:
856 examples: Add command-line parsing and take a 'port' argument
857 This allows users to run multiple servers on different ports for testing.
858 Only done for examples that actually take arguments and hence are capable of
859 outputting different streams for each instance on each port.
860 https://bugzilla.gnome.org/show_bug.cgi?id=742115
862 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
864 * gst/rtsp-server/rtsp-client.c:
865 * gst/rtsp-server/rtsp-client.h:
866 rtsp-client: Add a send_message default signal handler
867 This allows subclasses to easily hook into the response sending
868 mechanism without doing everything from a signal, which seems
869 awkward from subclasses.
871 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
874 Automatic update of common submodule
875 From ef1ffdc to f2c6b95
877 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
881 configure: add --disable-examples switch
882 https://bugzilla.gnome.org/show_bug.cgi?id=741678
884 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
886 * examples/.gitignore:
887 * examples/Makefile.am:
888 * examples/test-video-rtx.c:
889 examples: add a retransmisison example implementing RFC4588
890 Currently only SSRC-multiplexed rtx streams are supported
892 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
894 * gst/rtsp-server/rtsp-stream.c:
895 rtsp-stream: Fix some minor memory leaks
897 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
899 * gst/rtsp-server/rtsp-media.c:
900 rtsp-media: Some minor cleanup
902 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
904 * gst/rtsp-server/rtsp-stream.c:
905 rtsp-stream: Fix compiler warnings
906 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
907 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
909 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
910 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
913 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
915 * docs/libs/gst-rtsp-server-sections.txt:
916 * gst/rtsp-server/rtsp-media-factory.c:
917 * gst/rtsp-server/rtsp-media-factory.h:
918 * gst/rtsp-server/rtsp-media.c:
919 * gst/rtsp-server/rtsp-media.h:
920 * gst/rtsp-server/rtsp-sdp.c:
921 * gst/rtsp-server/rtsp-stream.c:
922 * gst/rtsp-server/rtsp-stream.h:
923 media: implement ssrc-multiplexed retransmission support
924 based off RFC 4588 and the server-rtpaux example in -good
926 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
928 * gst/rtsp-server/rtsp-client.c:
929 * gst/rtsp-server/rtsp-stream-transport.c:
930 * gst/rtsp-server/rtsp-stream.c:
931 rtsp: Ref transports in hash table.
932 Also ref streams for transports.
933 This solves a crash when reciving a rtcp after teardown but before
935 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
937 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
940 Automatic update of common submodule
941 From 7bb2bce to ef1ffdc
943 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
945 * gst/rtsp-server/rtsp-client.c:
946 client: refactor cleanup of cached media
948 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
950 * tests/check/gst/client.c:
952 The session leak is now fixed, lets remove those FIXME comments.
954 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
956 * tests/check/gst/rtspserver.c:
957 tests: Test to setup two sessions on one connection
958 https://bugzilla.gnome.org/show_bug.cgi?id=739112
960 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
962 * tests/check/gst/rtspserver.c:
963 tests: Test setup with tcp transport
964 https://bugzilla.gnome.org/show_bug.cgi?id=739112
966 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
968 * gst/rtsp-server/rtsp-client.c:
969 client: Configure transport after creating session media
970 The default implementation of configure_client_transport() in
971 rtsp-client uses the session media when it chooses channels for
973 https://bugzilla.gnome.org/show_bug.cgi?id=739112
975 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
977 * gst/rtsp-server/rtsp-client.c:
978 * gst/rtsp-server/rtsp-session-media.c:
979 client: Stop caching media in client when doing setup
980 If the media has been managed by a session media, it should not be
981 cached in the client any longer. The GstRTSPSessionMedia object is now
982 responsible for unpreparing the GstRTSPMedia object using
983 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
985 https://bugzilla.gnome.org/show_bug.cgi?id=739112
987 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
989 * gst/rtsp-server/rtsp-stream.c:
990 rtsp-stream: unref srtp decoder when leaving bin
991 https://bugzilla.gnome.org/show_bug.cgi?id=739481
993 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
995 * gst/rtsp-server/rtsp-client.c:
996 rtsp-client: mikey memory leaks
997 https://bugzilla.gnome.org/show_bug.cgi?id=739383
999 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
1002 Automatic update of common submodule
1003 From 84d06cd to 7bb2bce
1005 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
1008 Parallelise 'make check-valgrind'
1010 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
1013 Automatic update of common submodule
1014 From a8c8939 to 84d06cd
1016 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
1019 Automatic update of common submodule
1020 From 36388a1 to a8c8939
1022 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1024 * gst/rtsp-server/rtsp-media.c:
1025 rtsp-media: deactivate media when shutting down from paused
1026 This was only done when going directly from playing.
1027 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
1029 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1031 * gst/rtsp-server/rtsp-client.c:
1032 * gst/rtsp-server/rtsp-context.h:
1033 rtsp-client: add stream transport to context
1034 We add the stream transport to the context so we can get the configured
1035 client stream transport in the setup request signal.
1036 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
1038 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1040 * gst/rtsp-server/rtsp-stream.c:
1041 stream: release lock even not all transports have been removed
1042 We don't want to keep the lock even we return FALSE because not all the
1043 transports have been removed. This could lead into a deadlock.
1044 https://bugzilla.gnome.org/show_bug.cgi?id=737797
1046 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
1048 * gst/rtsp-server/rtsp-sdp.c:
1049 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
1050 These were renamed in GstRTPBasePayload in 1.0
1052 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1054 * gst/rtsp-server/rtsp-client.c:
1055 client: set session media to NULL without the lock
1056 We need to set session medias to NULL without the client lock otherwise
1057 we can end up in a deadlock if another thread is waiting for the lock
1058 and media unprepare is also waiting for that thread to end.
1059 https://bugzilla.gnome.org/show_bug.cgi?id=737690
1061 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1063 * gst/rtsp-server/rtsp-media.c:
1064 rtsp-media: Set state to UNPREPARING in all cases
1066 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
1068 * gst/rtsp-server/rtsp-media.c:
1069 media: set state to unpreparing when unprepare is initiated
1070 https://bugzilla.gnome.org/show_bug.cgi?id=737675
1072 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
1074 * gst/rtsp-server/rtsp-client.c:
1075 rtsp-client: Remove backlog limit while processings requests
1076 If the backlog limit is kept two cases of deadlocks may be
1077 encountered when streaming over TCP. Without the backlog
1078 limit this deadlocks can not happen, at the expence of
1080 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
1082 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
1084 * gst/rtsp-server/rtsp-client.c:
1085 rtsp-client: do not free main context before rtsp watch
1086 https://bugzilla.gnome.org/show_bug.cgi?id=737110
1088 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
1090 * tests/check/gst/rtspserver.c:
1091 tests: Extend unit test timeout to accomodate for valgrind
1092 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1094 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
1096 * gst/rtsp-server/rtsp-client.c:
1097 * gst/rtsp-server/rtsp-session.c:
1098 * gst/rtsp-server/rtsp-stream-transport.c:
1099 rtsp-*: Treat sending packets to clients as keepalive
1100 As long as gst-rtsp-server can successfully send RTP/RTCP data to
1101 clients then the client must be reading. This change makes the server
1102 timeout the connection if the client stops reading.
1103 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1105 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
1107 * gst/rtsp-server/rtsp-client.c:
1108 rtsp-client: Allow backlog to grow while expiring session
1109 Allow the send backlog in the RTSP watch to grow to unlimited size while
1110 attempting to bring the media pipeline to NULL due to a session
1111 expiring. Without this change the appsink element cannot change state
1112 because it is blocked while rendering data in the new_sample callback.
1113 This callback will block until it has successfully put the data into the
1114 send backlog. There is a chance that the send backlog is full at this
1115 point which means that the callback may block for a long time, possibly
1116 forever. Therefore the media pipeline may also be prevented from
1117 changing state for a long time.
1118 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1120 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
1122 * gst/rtsp-server/rtsp-client.c:
1123 rtsp-client: Make old compilers happy
1124 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
1125 Just in case that guint8 doesn't fit in a pointer. Just in case ...
1127 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
1129 * gst/rtsp-server/rtsp-client.c:
1130 client: raise the backlog limits before pausing
1131 We need to raise the backlog limits before pausing the pipeline or else
1132 the appsink might be blocking in the render method in wait_backlog() and
1133 we would deadlock waiting for paused.
1134 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
1136 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
1138 * gst/rtsp-server/rtsp-client.c:
1139 client: make define for the WATCH_BACKLOG
1140 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
1142 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
1144 * gst/rtsp-server/rtsp-client.c:
1145 client: simplify session transport handling
1146 link/unlink of the transport in a session was done to keep track of all
1147 TCP transports and to send RTP/RTCP data to the streams. We can simplify
1148 that by putting all the TCP transports in a hashtable indexed with the
1150 We also don't need to link/unlink the transports when we pause/resume
1151 the streams. The same effect is already achieved when we pause/play the
1152 media. Indeed, when we pause the media, the transport is removed from
1153 the media and the callbacks will not be called anymore.
1154 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
1156 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
1158 * gst/rtsp-server/rtsp-stream-transport.c:
1159 * gst/rtsp-server/rtsp-stream-transport.h:
1160 stream-transport: make method to handle received data
1161 Make a method to handle the data received on a channel. It sends the
1162 data to the stream of the transport on the RTP or RTCP pads based on
1165 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
1167 * examples/test-mp4.c:
1168 test: add example of dumping RTCP reports
1170 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
1172 * gst/rtsp-server/rtsp-media.c:
1173 * gst/rtsp-server/rtsp-stream.c:
1174 * gst/rtsp-server/rtsp-stream.h:
1175 rtsp-media: Make sure that sequence numbers are monotonic after pause
1176 The sequence number is not monotonic for RTP packets after pause. The
1177 reason is basepayloader generates a randon sequence number when the
1178 pipeline goes from ready to pause. With this fix generation of sequence
1179 number will be monotonic when going from pause to play request.
1180 https://bugzilla.gnome.org/show_bug.cgi?id=736017
1182 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
1184 * gst/rtsp-server/rtsp-client.c:
1185 rtsp-client: Protect saved clients watch with a mutex
1186 Fixes a crash when close() is called while merging clients
1187 in handle_tunnel(). In that case close() would destroy the
1188 watch while it is still being used in handle_tunnel().
1189 https://bugzilla.gnome.org/show_bug.cgi?id=735570
1191 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1193 * gst/rtsp-server/rtsp-stream.c:
1194 rtsp-stream: Remove the multicast group udp sources when removing from the bin
1196 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1198 * gst/rtsp-server/rtsp-media.c:
1199 * gst/rtsp-server/rtsp-stream.c:
1200 * gst/rtsp-server/rtsp-stream.h:
1201 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
1202 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
1203 seeking and will always continue counting the time. This leads to
1204 the NPT after a backwards seek to be something completely different
1205 to the actual seek position.
1206 https://bugzilla.gnome.org/show_bug.cgi?id=732644
1208 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
1210 * examples/test-appsrc.c:
1211 examples: fix another reference leak
1212 gst_rtsp_media_get_element() returns a new ref.
1214 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1216 * examples/test-appsrc.c:
1217 examples: unref element after usage
1218 gst_bin_get_by_name_recurse_up() returns an element
1219 reference that must be unreffed after usage.
1220 https://bugzilla.gnome.org/show_bug.cgi?id=734546
1222 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
1224 * gst/rtsp-server/rtsp-media.c:
1225 signals: Fix copy-pasto in target-state signal offset
1227 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
1231 Makefile: Add usage of build-checks step
1232 Allows building checks without running them
1234 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
1236 * gst/rtsp-server/rtsp-stream.c:
1237 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
1238 When a UDP multicast transport is used it is expected that the server listens
1239 for RTP and RTCP packets on the multicast group with the corresponding port.
1240 Without this we will never get RTCP packets from clients in multicast mode.
1241 https://bugzilla.gnome.org/show_bug.cgi?id=732238
1243 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1248 === release 1.4.0 ===
1250 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1256 * gst-rtsp-server.doap:
1259 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
1261 * gst/rtsp-server/rtsp-media.h:
1262 media: correct misspelled words in description
1263 https://bugzilla.gnome.org/show_bug.cgi?id=733244
1265 === release 1.3.91 ===
1267 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1273 * gst-rtsp-server.doap:
1276 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
1278 * docs/libs/gst-rtsp-server-sections.txt:
1281 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
1283 * gst/rtsp-server/rtsp-server.c:
1284 server: implement client REMOVE filter
1286 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
1288 * gst/rtsp-server/rtsp-client.c:
1289 * gst/rtsp-server/rtsp-client.h:
1290 client: expose _close() method
1291 Expose a previously internal close method to close the client
1294 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
1296 * gst/rtsp-server/rtsp-session-pool.c:
1297 session-pool: signal session-removed outside of the lock
1298 Release the lock before emiting the session-removed signal.
1300 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
1302 * gst/rtsp-server/rtsp-client.c:
1303 * gst/rtsp-server/rtsp-server.c:
1304 * gst/rtsp-server/rtsp-session-pool.c:
1305 * gst/rtsp-server/rtsp-session.c:
1306 * gst/rtsp-server/rtsp-stream.c:
1307 filter: Release lock in filter functions
1308 Release the object lock before calling the filter functions. We need to
1309 keep a cookie to detect when the list changed during the filter
1310 callback. We also keep a hashtable to make sure we only call the filter
1311 function once for each object in case of concurrent modification.
1312 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
1314 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
1316 * gst/rtsp-server/rtsp-client.c:
1317 client: check if watch is set in handle_teardown()
1318 The unit tests run without a watch
1320 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1322 * tests/check/gst/client.c:
1323 client tests: send teardown to cleanup session
1325 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
1327 * tests/check/gst/rtspserver.c:
1328 server tests: send teardown to cleanup session
1330 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1332 * gst/rtsp-server/rtsp-client.c:
1333 client: keep ref to client for the session removed handler
1334 This extra ref will be dropped when all client sessions have been
1335 removed. A session is removed when a client sends teardown, closes its
1336 endpoint of the TCP connection or the sessions expires.
1337 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1339 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
1341 * gst/rtsp-server/rtsp-client.c:
1342 * gst/rtsp-server/rtsp-session.c:
1343 * tests/check/gst/client.c:
1344 client: manage media in session as a last step
1345 Once we manage a media in a session, we can't unmanage it anymore
1346 without destroying it. Therefore, first check everything before we
1347 manage the media, otherwise if something is wrong we have no way to
1349 If we created a new session and something went wrong, remove the session
1350 again. Fixes a leak in the unit test.
1352 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1354 * examples/test-mp4.c:
1355 * examples/test-ogg.c:
1356 examples: print 'stream ready at url' for mp4 and ogg example
1358 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
1360 * gst/rtsp-server/rtsp-client.c:
1361 * gst/rtsp-server/rtsp-sdp.c:
1362 rtsp: fix for MIKEY api change
1364 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
1366 * gst/rtsp-server/rtsp-client.c:
1367 client: free watch context only once
1368 The watch context is freed when the source is destroyed. Avoids
1369 a CRITICAL when we try to unref the context twice.
1371 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
1373 * gst/rtsp-server/rtsp-client.c:
1376 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
1378 * gst/rtsp-server/rtsp-client.c:
1379 client: protect sessions with lock
1380 Protect the list of sessions with the lock.
1381 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1383 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
1385 * gst/rtsp-server/rtsp-client.c:
1386 Client: keep a ref to the session
1387 Don't just keep a weak ref to the session objects but use a hard ref. We
1388 will be notified when a session is removed from the pool (expired) with
1389 the new session-removed signal.
1390 Don't automatically close the RTSP connection when all the sessions of
1391 a client are removed, a client can continue to operate and it can create
1392 a new session if it wants. If you want to remove the client from the
1393 server, you have to use gst_rtsp_server_client_filter() now.
1394 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
1395 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
1397 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
1399 * gst/rtsp-server/rtsp-session-pool.c:
1400 * gst/rtsp-server/rtsp-session-pool.h:
1401 session-pool: add session-removed signal
1402 Add a signal to be notified when a session is removed from the pool.
1404 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
1406 * gst/rtsp-server/Makefile.am:
1407 * gst/rtsp-server/rtsp-server.h:
1408 Make rtsp-server.h a single-include header, use it for G-I
1409 https://bugzilla.gnome.org/show_bug.cgi?id=732411
1411 === release 1.3.90 ===
1413 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1419 * gst-rtsp-server.doap:
1422 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
1424 * gst/rtsp-server/rtsp-stream.c:
1425 stream: crypto can be NULL
1427 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
1429 * gst/rtsp-server/rtsp-client.c:
1430 * gst/rtsp-server/rtsp-media.c:
1431 * gst/rtsp-server/rtsp-mount-points.c:
1432 introspection: add missing allow-none annotations
1433 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1435 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
1437 * gst/rtsp-server/rtsp-address-pool.c:
1438 * gst/rtsp-server/rtsp-media.c:
1439 * gst/rtsp-server/rtsp-session-media.c:
1440 * gst/rtsp-server/rtsp-session-pool.c:
1441 * gst/rtsp-server/rtsp-stream-transport.c:
1442 * gst/rtsp-server/rtsp-stream.c:
1443 * gst/rtsp-server/rtsp-token.c:
1444 introspection: add (nullable) annotations to return values
1445 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1447 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
1449 * gst/rtsp-server/rtsp-client.c:
1450 * gst/rtsp-server/rtsp-stream.c:
1451 gi: improve annotations
1452 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
1454 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
1456 * gst/rtsp-server/rtsp-client.c:
1457 * gst/rtsp-server/rtsp-media-factory.c:
1458 * gst/rtsp-server/rtsp-media.c:
1459 * gst/rtsp-server/rtsp-server.c:
1460 signals: use generic marshal function
1461 Use the generic C marshal function.
1462 Use more explicit type instead of G_TYPE_POINTER
1464 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
1466 * gst/rtsp-server/rtsp-context.h:
1467 context: add type macro
1469 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
1471 * gst/rtsp-server/rtsp-client.c:
1472 * gst/rtsp-server/rtsp-sdp.c:
1473 * gst/rtsp-server/rtsp-sdp.h:
1474 sdp: hide key length defines
1475 They don't have a namespace.
1477 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1482 === release 1.3.3 ===
1484 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
1490 * gst-rtsp-server.doap:
1493 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1495 * gst/rtsp-server/rtsp-client.c:
1496 * gst/rtsp-server/rtsp-sdp.c:
1497 * gst/rtsp-server/rtsp-sdp.h:
1498 mikey: add different key length parameters
1499 Add encryption and authentication key length parameters to MIKEY. For
1500 the encoders, the key lengths are obtained from the cipher and auth
1501 algorithms set in the caps. For the decoders, they are obtained while
1502 parsing the key management from the client.
1503 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
1505 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
1507 * tests/check/gst/stream.c:
1508 stream tests: Make sure we get right multicast address from stream
1509 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
1511 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
1513 * gst/rtsp-server/rtsp-client.c:
1514 client: ref the context until rtsp watch is alive
1515 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
1517 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
1519 * gst/rtsp-server/rtsp-client.c:
1520 client: Destroy the rtsp watch after connection close
1522 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
1524 * gst/rtsp-server/rtsp-media.c:
1525 media: fix confusing comment
1527 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
1529 * gst/rtsp-server/rtsp-session.c:
1530 rtsp-session: Timeout in header.
1531 Adding the possbilty to always have timout in header.
1532 This is configurabe with setting "timeout-always-visible".
1533 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
1535 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
1540 === release 1.3.2 ===
1542 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
1549 * gst-rtsp-server.doap:
1552 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1555 Automatic update of common submodule
1556 From 211fa5f to 1f5d3c3
1558 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
1560 * gst/rtsp-server/rtsp-client.c:
1561 client: store TCP ports in transport
1562 Store the TCP ports in the transport when we are doing RTSP over TCP.
1563 This way, we can easily get to the ports from the transport.
1564 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
1566 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1568 * gst/rtsp-server/rtsp-stream.c:
1569 stream: add signals for new RTP/RTCP encoders
1570 New signals to allow the user to configure the dynamically created
1572 https://bugzilla.gnome.org/show_bug.cgi?id=730228
1574 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1576 * gst/rtsp-server/rtsp-media.c:
1577 * gst/rtsp-server/rtsp-media.h:
1578 media: Make suspend()/unsuspend() virtual
1579 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
1581 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1583 * gst/rtsp-server/rtsp-client.c:
1584 client: fix send-message signal marshaller
1585 Use generic marshalling for the send-message signal. It has
1586 two POINTER arguments, not just one.
1587 https://bugzilla.gnome.org/show_bug.cgi?id=729900
1589 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
1591 * tests/check/gst/media.c:
1592 tests: add and remove pads only once
1593 In this test we simulate a dynamic pad by watching the caps event.
1594 Because of renegotiation in the base payloader now, this caps is sent
1595 multiple times but we can only deal with 1 invocation, use a variable to
1596 only 'add and remove' the pad once.
1598 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1600 * tests/check/gst/rtspserver.c:
1601 tests: add unit test for correct handling of Require headers
1602 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1604 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1606 * gst/rtsp-server/rtsp-client.c:
1607 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
1608 Servers must handle Require headers and must report a failure
1609 if they don't handle any of the Required options, see RFC 2326,
1610 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
1611 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1613 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1618 === release 1.3.1 ===
1620 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1626 * gst-rtsp-server.doap:
1629 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
1632 Automatic update of common submodule
1633 From bcb1518 to 211fa5f
1635 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
1640 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1642 * tests/check/gst/sessionmedia.c:
1643 tests: fix memory leak in sessionmedia unit test
1645 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
1647 * gst/rtsp-server/rtsp-client.c:
1648 client: emit a signal before sending a message
1649 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
1651 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
1653 * gst/rtsp-server/rtsp-client.c:
1654 client: pass context to send_message
1655 Pass the current context to send_message, we will need it later.
1657 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
1659 * gst/rtsp-server/rtsp-client.c:
1660 client: fix typo in comment
1662 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
1664 * gst/rtsp-server/rtsp-media.c:
1665 media: Do not stop thread twice if default_prepare() fails
1667 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
1669 * gst/rtsp-server/rtsp-client.c:
1670 client: set the watch to flushing before going to NULL
1671 First set the watch to flushing so that we unblock any current and
1672 future attempt to send data on the watch, Then set the pipeline to
1674 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
1676 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
1678 * gst/rtsp-server/rtsp-session-pool.c:
1679 * tests/check/gst/sessionpool.c:
1680 rtsp-session-pool: Fixes annotation
1681 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
1682 in the sessionpool test.
1683 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
1685 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
1687 * gst/rtsp-server/rtsp-media.c:
1688 * gst/rtsp-server/rtsp-media.h:
1689 media: make media_prepare virtual
1690 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
1692 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1694 * gst/rtsp-server/rtsp-media.c:
1695 * tests/check/gst/media.c:
1696 media: stop the thread in more error cases
1698 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
1700 * gst/rtsp-server/rtsp-media.c:
1701 * tests/check/gst/media.c:
1702 media: allow NULL as the thread
1703 Use the default context whan passing a NULL thread.
1705 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1707 * gst/rtsp-server/rtsp-client.c:
1708 rtsp-client: indent cleanup
1709 Coverity was moaning about unreachable code, and I think it was just
1710 confused by { being before the label. We'll see if it pops up again.
1713 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
1715 * gst/rtsp-server/rtsp-client.c:
1716 * gst/rtsp-server/rtsp-media.c:
1717 client: Add drop-backlog property
1718 When we have too many messages queued for a client (currently hardcoded
1719 to 100) we overflow and drop the messages. Add a drop-backlog property
1720 to control this behaviour. Setting this property to FALSE will retry
1721 to send the messages to the client by waiting for more room in the
1723 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
1725 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
1727 * gst/rtsp-server/rtsp-client.c:
1728 client: support for POST before GET when setting up a tunnel
1730 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
1732 * gst/rtsp-server/rtsp-client.c:
1733 client: remove watch of the second client after http tunnel setup
1734 The second client will be freed after the HTTP tunnel has been set up.
1735 Make sure it's RTSP watch is never dispatched again.
1736 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
1738 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
1740 * gst/rtsp-server/rtsp-media.c:
1741 * tests/check/gst/media.c:
1742 media: Make media_prepare() fail if port allocation fails
1743 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
1745 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
1747 * tests/check/gst/media.c:
1748 media test: cleanup the thread pool in tests
1750 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
1752 * gst/rtsp-server/rtsp-media.c:
1753 * tests/check/gst/media.c:
1754 rtsp-media: Unblock blocked streams in unprepare
1755 The streams will be blocked when a live media is prepared.
1756 The streams should be unblocked in gst_rtsp_media_unprepare.
1757 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
1759 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
1761 * gst/rtsp-server/rtsp-media.c:
1762 media: release the state lock when going to NULL
1763 Set our state to UNPREPARING and release the state-lock before
1764 setting the pipeline to the NULL state. This way, any pad-added
1765 callback will be able to take the state-lock and check that we are now
1766 unpreparing instead of deadlocking.
1767 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
1769 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
1771 * gst/rtsp-server/rtsp-media.c:
1772 media: protect status with lock
1773 Make sure we only update the status with the lock.
1775 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
1777 * gst/rtsp-server/rtsp-client.c:
1778 * gst/rtsp-server/rtsp-sdp.c:
1779 rtsp: update for MIKEY API changes
1781 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
1783 * gst/rtsp-server/rtsp-client.c:
1784 client: parse the mikey response from the client
1785 Parse the mikey response from the client and update the policy for
1788 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
1790 * gst/rtsp-server/rtsp-stream.c:
1791 * gst/rtsp-server/rtsp-stream.h:
1792 stream: add method to set crypto info
1793 Make a method to configure the crypto information of a stream.
1794 Set udpsrc in READY instead of PAUSED so that we can configure caps
1797 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
1799 * gst/rtsp-server/rtsp-client.c:
1800 client: cleanup error paths
1802 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
1804 * gst/rtsp-server/rtsp-media.c:
1807 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
1809 * examples/test-video.c:
1810 test: enable SRTP only on RTSPS
1811 We only want to enable SRTP when doing rtsp over TLS so that we can
1812 exchange the keys in a secure way.
1814 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
1816 * examples/test-video.c:
1817 test: print an error on failure
1819 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
1822 * examples/test-video.c:
1823 * gst/rtsp-server/rtsp-sdp.c:
1824 * gst/rtsp-server/rtsp-stream.c:
1825 * tests/check/Makefile.am:
1826 stream: add SRTP support
1827 Install srtp encoder and decoder elements in rtpbin
1830 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1832 * tests/check/Makefile.am:
1833 * tests/check/gst/sessionpool.c:
1834 tests: Add unit tests for sessionpool
1835 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
1837 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1839 * tests/check/gst/threadpool.c:
1840 tests: Improve code coverage of rtsp-threadpool tests
1841 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
1843 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1845 * tests/check/gst/sessionmedia.c:
1846 tests: Improve code coverage for rtsp-session-media
1847 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
1849 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1851 gobject-introspection: Add annotations to support language bindings
1852 In addition a few cosmetic changes:
1853 * Adjust the order of arguments
1854 * Fix typo: occured -> occurred
1855 * Fix indentation after Return:-clauses
1856 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
1858 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1860 * gst/rtsp-server/rtsp-stream.c:
1861 rtsp-stream: Don't mix IPv4 and IPv6 addresses
1862 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
1864 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
1866 * gst/rtsp-server/rtsp-stream.c:
1867 stream: take caps after the session manager
1868 Take the caps for the SDP after they leave the rtpbin so that we can
1869 also get the properties added by rtpbin elements.
1871 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
1873 * gst/rtsp-server/rtsp-stream.c:
1874 stream: release lock while pushing out packets
1875 Keep a cache of the transports and use this to iterate the transport
1876 while pushing packets. This allows us to release the lock early.
1877 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
1879 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
1881 * gst/rtsp-server/rtsp-client.c:
1882 * gst/rtsp-server/rtsp-client.h:
1883 rtsp-client: vmethod for modifying tunnel GET response
1884 Add a vmethod tunnel_http_response where the response to the HTTP GET
1885 for tunneled connections can be modified.
1886 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
1888 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
1890 * gst/rtsp-server/rtsp-sdp.c:
1891 sdp: make 1 media line per profile
1892 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
1893 line in the SDP for each profile. The client is then supposed to pick
1894 one of the profiles in the SETUP request. Because the m= lines have the
1895 same pt, the client also knows that only 1 option is possible.
1897 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
1899 * gst/rtsp-server/rtsp-media-factory.c:
1900 * gst/rtsp-server/rtsp-media-factory.h:
1901 * gst/rtsp-server/rtsp-media.c:
1902 factory: add profile property and pass to media and streams
1904 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
1906 * examples/test-multicast.c:
1907 * gst/rtsp-server/rtsp-sdp.c:
1908 sdp: pass multicast connection for multicast-only stream
1909 Pass the multicast address of the stream in the connection info in the
1910 SDP so that clients try a multicast connection first.
1911 Only allow multicast connections in the test-multicast example. Also
1912 increase the TTL a little.
1914 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1917 .gitignore: Ignore gcov intermediate files
1918 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
1920 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
1922 * gst/rtsp-server/rtsp-stream.c:
1923 stream: release some locks in error cases
1925 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1927 docs: Enable and fix gtk-doc warnings
1928 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
1929 * addresspool/mediafactory: Add missing annotation colon
1930 * stream: Annotate return value
1931 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
1933 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1936 Automatic update of common submodule
1937 From fe1672e to bcb1518
1939 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
1942 Automatic update of common submodule
1943 From 1a07da9 to fe1672e
1945 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1947 * examples/Makefile.am:
1948 examples: use LDADD for libs instead of LDFLAGS
1950 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
1953 configure: make sure releases are in .doap file
1955 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
1957 * examples/test-cgroups.c:
1958 examples: test-cgroups: don't put code with side effects into g_assert()
1959 The g_assert() might get compiled out with the right
1960 compiler/preprocessor flags.
1962 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1964 * examples/.gitignore:
1965 examples: add cgroup test binary to .gitignore
1967 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
1969 * examples/test-cgroups.c:
1970 examples: fix cgroup test build
1971 Fixes build failure caused by compiler warning:
1972 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
1974 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
1977 .gitignore: ignore temp files created in the course of 'make check'
1979 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
1981 * gst/rtsp-server/rtsp-media.c:
1982 rtsp-media: don't loose frames handling new PLAY request
1983 If client supplied a range check if the range specifies the start point.
1984 If not, then do an accurate seek to the current position. If a start
1985 point was specified do do a key unit seek to make sure the streaming
1986 starts with decodeable frames.
1987 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
1989 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
1991 * gst/rtsp-server/rtsp-media.c:
1992 Revert "media: only flush when setting a new start position"
1993 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
1994 We need to do the flush in all cases, demuxer block currently for
1997 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
1999 * gst/rtsp-server/rtsp-media.c:
2000 media: only flush when setting a new start position
2001 Only flush the pipeline when we change the start position with
2003 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
2005 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
2007 * gst/rtsp-server/rtsp-stream.c:
2008 stream: set ttl-mc before adding the socket
2009 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
2010 never be set on socket.
2011 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
2013 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2015 * gst/rtsp-server/rtsp-media.c:
2016 media: stop thread if media is already prepared
2017 in gst_rtsp_media_prepare() the thread is not used if media is already
2018 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
2020 https://bugzilla.gnome.org/show_bug.cgi?id=724182
2022 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
2025 build: Ship gst-rtsp-server.doap file
2027 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
2029 * tests/check/gst/rtspserver.c:
2030 tests: Fix another compiler warning with gcc
2032 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
2034 * gst/rtsp-server/rtsp-client.c:
2035 * gst/rtsp-server/rtsp-mount-points.c:
2036 * gst/rtsp-server/rtsp-stream.c:
2037 * tests/check/gst/client.c:
2038 rtsp-server: Fix lots of compiler warnings with clang
2040 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
2043 * gst-rtsp-server.doap:
2044 * tests/Makefile.am:
2045 configure: Synchronise with the configure scripts of the other modules
2047 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2050 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
2052 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2054 * gst/rtsp-server/rtsp-media.c:
2055 * gst/rtsp-server/rtsp-stream.c:
2056 Revert "rtsp-server: support build against last stable release"
2057 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
2058 Let us require 1.2.3 now, which is going to be released in a few
2061 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
2063 * gst/rtsp-server/rtsp-session-media.c:
2064 * gst/rtsp-server/rtsp-stream-transport.c:
2065 session: improve RTP-Info
2066 Ignore streams that can't generate RTP-Info instead of failing.
2067 Don't return the empty string when all streams are unconfigured but
2068 return NULL so that we don't generate and empty RTP-Info header.
2069 Improve docs a little.
2071 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
2073 * gst/rtsp-server/rtsp-session-media.c:
2074 Don't free rtpinfo GString when it is NULL
2075 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2077 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
2079 * gst/rtsp-server/rtsp-media.c:
2080 media: only set keyframe flag when modifying start
2081 Only set the keyframe flag when we modify the start position. The
2082 keyframe flag should probably be ignored when no change is requested but
2083 until we can claim this is all documented properly and all demuxer
2084 implement this, avoid setting the flag.
2085 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
2087 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
2089 * gst/rtsp-server/rtsp-thread-pool.c:
2090 thread-pool: Unref source after mainloop has quit to avoid races in GLib
2091 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
2093 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
2095 * gst/rtsp-server/rtsp-stream.c:
2096 stream: handle NULL seqnum and rtptime arguments
2098 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
2100 * gst/rtsp-server/rtsp-thread-pool.c:
2101 * tests/check/gst/threadpool.c:
2102 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
2103 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2105 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
2107 * gst/rtsp-server/rtsp-stream.c:
2108 stream: add fallback for missing stats property
2109 Use a fallback when the payloader does not have a stats property
2110 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2112 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
2115 Automatic update of common submodule
2116 From f7bc1c3 to 1a07da9
2118 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
2120 * gst/rtsp-server/rtsp-stream.c:
2121 stream: don't leak stats structure
2122 Don't leak the stats structure and deal with NULL stats.
2124 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
2126 * gst/rtsp-server/rtsp-stream.c:
2127 stream: Get rtpinfo properties atomically from payloader
2128 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2130 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
2132 * gst/rtsp-server/rtsp-media.c:
2133 media: refactor state change functions and signals
2134 Make functions to set the target state and the pipeline state and emit
2135 the signals from those functions.
2137 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
2139 * gst/rtsp-server/rtsp-media.c:
2140 * gst/rtsp-server/rtsp-media.h:
2141 media: add signal to notify of pending state changes
2143 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2145 * gst/rtsp-server/rtsp-media.c:
2146 * gst/rtsp-server/rtsp-stream.c:
2147 rtsp-server: support build against last stable release
2148 Until 1.2.3 is out with the new get_type function and we
2151 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
2153 * gst/rtsp-server/rtsp-stream.c:
2154 stream: fix compilation
2156 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
2158 * gst/rtsp-server/rtsp-media.c:
2159 * gst/rtsp-server/rtsp-media.h:
2160 * gst/rtsp-server/rtsp-stream.c:
2161 * gst/rtsp-server/rtsp-stream.h:
2162 stream: add property to configure profiles
2164 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
2166 * gst/rtsp-server/rtsp-client.c:
2167 client: let stream check supported transport
2168 Delegate the check if a transport is allowed to the stream.
2169 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2171 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
2173 * gst/rtsp-server/rtsp-stream.c:
2174 * gst/rtsp-server/rtsp-stream.h:
2175 stream: add method to check supported transport
2176 Add a method to check if a transport is supported
2178 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
2181 configure.ac: Only check for gstreamer-check, not check
2182 We include check in gstreamer-check since quite some time now.
2184 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
2186 * gst/rtsp-server/rtsp-session-media.c:
2187 * gst/rtsp-server/rtsp-stream-transport.c:
2188 * gst/rtsp-server/rtsp-stream.c:
2189 * gst/rtsp-server/rtsp-stream.h:
2190 stream: return clock-rate from get_rtpinfo
2191 And use it to correct the rtptime to the requested start-time.
2192 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2194 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
2196 * gst/rtsp-server/rtsp-session-media.c:
2197 * gst/rtsp-server/rtsp-stream-transport.c:
2198 * gst/rtsp-server/rtsp-stream-transport.h:
2199 session-media: calculate start-time
2201 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
2203 * gst/rtsp-server/rtsp-stream-transport.c:
2204 * gst/rtsp-server/rtsp-stream.c:
2205 * gst/rtsp-server/rtsp-stream.h:
2206 stream: also return the running-time
2207 Return the running-time in the rtpinfo as well.
2209 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
2211 * gst/rtsp-server/rtsp-client.c:
2212 * gst/rtsp-server/rtsp-session-media.c:
2213 * gst/rtsp-server/rtsp-session-media.h:
2214 * gst/rtsp-server/rtsp-stream-transport.c:
2215 * gst/rtsp-server/rtsp-stream-transport.h:
2216 session-media: let the session-media make the RTPInfo
2217 Add method to create the RTPInfo for a stream-transport.
2218 Add method to create the RTPInfo for all stream-transports in a
2220 Use the session-media RTPInfo code in client. This allows us to refactor
2221 another method to link the TCP callbacks.
2223 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2225 mount-points: sort sequence before g_sequence_lookup
2226 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
2227 sort sequence if dirty, otherwise lookup will fail.
2228 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2230 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2233 configure: rename package from gst-rtsp to gst-rtsp-server
2234 To match git module name and avoid confusion with the
2235 rtsp lib in gst-plugins-base and rtsp plugin in -good.
2237 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
2240 configure: bump core/base/good requirement to 1.2.0
2241 Bump to released stable version and make implicit
2242 requirements explicit.
2244 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2249 Fix broken gettext setup which is not used anyway
2251 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
2254 Automatic update of common submodule
2255 From dbedaa0 to d48bed3
2257 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
2259 * gst/rtsp-server/rtsp-client.c:
2260 * gst/rtsp-server/rtsp-media.c:
2261 * gst/rtsp-server/rtsp-media.h:
2262 media: add setup_sdp vmethod
2263 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
2264 gst_rtsp_media_setup_sdp.
2265 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2267 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
2269 * gst/rtsp-server/rtsp-stream.c:
2270 rtsp-stream: Check return value of sscanf
2271 streamid is only valid if sscanf matched something.
2273 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
2275 * gst/rtsp-server/rtsp-client.c:
2276 rtsp-client: Fix iteration
2277 Wouldn't even enter the code block otherwise (i++ was used as the check
2278 and not the postfix).
2280 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
2282 * gst/rtsp-server/rtsp-client.c:
2283 * gst/rtsp-server/rtsp-client.h:
2284 client: add vmethod to configure media and streams
2285 Implement a vmethod that can be used to configure the media and the
2286 streams based on the current context. Handle the blocksize handling in
2287 the default handler.
2288 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
2290 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2293 Make git ignore more unit test binaries
2295 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2297 * gst/rtsp-server/rtsp-address-pool.h:
2298 * gst/rtsp-server/rtsp-auth.h:
2299 * gst/rtsp-server/rtsp-client.h:
2300 * gst/rtsp-server/rtsp-context.h:
2301 * gst/rtsp-server/rtsp-media-factory-uri.h:
2302 * gst/rtsp-server/rtsp-media-factory.h:
2303 * gst/rtsp-server/rtsp-media.h:
2304 * gst/rtsp-server/rtsp-mount-points.h:
2305 * gst/rtsp-server/rtsp-server.h:
2306 * gst/rtsp-server/rtsp-session-media.h:
2307 * gst/rtsp-server/rtsp-session-pool.h:
2308 * gst/rtsp-server/rtsp-session.h:
2309 * gst/rtsp-server/rtsp-stream-transport.h:
2310 * gst/rtsp-server/rtsp-stream.h:
2311 * gst/rtsp-server/rtsp-thread-pool.h:
2312 * gst/rtsp-server/rtsp-token.h:
2313 rtsp-server: add padding to many public structures
2314 Not mini objects though, since they are not subclassable
2315 anyway, nor kept on the stack or inlined in a structure.
2317 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2319 media: add new create_rtpbin vmethod
2320 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
2321 https://bugzilla.gnome.org/show_bug.cgi?id=719734
2323 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
2325 * tests/check/gst/media.c:
2326 tests: fix memory leak, free test's thread pool
2327 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
2329 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
2331 * gst/rtsp-server/rtsp-stream-transport.c:
2332 stream-transport: free url in finalize
2334 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
2336 * gst/rtsp-server/rtsp-media.c:
2337 media: also do state change in suspended state
2339 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
2341 * gst/rtsp-server/rtsp-client.c:
2342 * gst/rtsp-server/rtsp-media.c:
2343 media: also handle prepare and range in suspended state
2344 When we are suspended, we are already prepared.
2345 We can get the range in the suspended state.
2347 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
2349 * tests/check/Makefile.am:
2350 * tests/check/gst/sessionmedia.c:
2351 check: add test for uri in setup
2352 Added unit tests for the new functionality in GstRTSPStreamTransport.
2353 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2355 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
2357 * gst/rtsp-server/rtsp-client.c:
2358 client: store setup uri and use in PLAY response
2359 Store the uri used when doing the setup and use that in the PLAY
2361 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2363 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
2365 * gst/rtsp-server/rtsp-stream-transport.c:
2366 * gst/rtsp-server/rtsp-stream-transport.h:
2367 stream-transport: add method to get/set url
2369 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
2371 * gst/rtsp-server/rtsp-client.c:
2372 client: suspend after SDP and unsuspend before PLAYING
2373 Based on patches by Ognyan Tonchev <ognyan@axis.com>
2374 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
2376 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
2378 * gst/rtsp-server/rtsp-media-factory.c:
2379 * gst/rtsp-server/rtsp-media-factory.h:
2380 * gst/rtsp-server/rtsp-media.c:
2381 * gst/rtsp-server/rtsp-media.h:
2382 * gst/rtsp-server/rtsp-session-media.c:
2383 * gst/rtsp-server/rtsp-session.c:
2384 * tests/check/gst/media.c:
2385 * tests/check/gst/mediafactory.c:
2386 media: add suspend modes
2387 Add support for different suspend modes. The stream is suspended right after
2388 producing the SDP and after PAUSE. Different suspend modes are available that
2389 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
2390 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
2391 state and RESET will bring the pipeline to the NULL state.
2392 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
2393 this means that the pipeline needs to be prerolled again.
2394 Base on patches by Ognyan Tonchev <ognyan@axis.com>
2395 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2397 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
2399 * gst/rtsp-server/rtsp-media.c:
2400 media: start live streams in blocked state
2401 Start live streams in the blocked state and make them preroll using the
2402 messages. This ensure that no data is played by the sink until we explicitly
2403 unblock the stream right before going to PLAYING.
2404 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2406 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
2408 * gst/rtsp-server/rtsp-media.c:
2409 media: refactor starting and waiting for preroll
2410 Based on patches from Ognyan Tonchev <ognyan@axis.com>
2411 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2413 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
2415 * gst/rtsp-server/rtsp-stream.c:
2416 * gst/rtsp-server/rtsp-stream.h:
2417 stream: add API to block streams
2418 Add an API to block on the streams and make it post a message.
2419 Based on patch by Ognyan Tonchev <ognyan@axis.com>
2420 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2422 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
2424 * docs/libs/Makefile.am:
2425 docs: Specify the override file
2426 Even if it's empty (for now) it avoids make distcheck complaining
2428 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
2430 * gst/rtsp-server/rtsp-media.c:
2431 media: move default implementations to where they are used
2433 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
2435 * gst/rtsp-server/rtsp-media.c:
2436 media: take the right lock in gst_rtsp_media_set_pipeline_state()
2437 We need to take the state_lock when calling this method.
2439 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
2441 * gst/rtsp-server/rtsp-media.c:
2442 media: handle add-added on non-bins too
2443 Handle dynamic payloaders that are not bins, as used in the unit-test.
2445 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2447 * gst/rtsp-server/rtsp-media-factory.c:
2448 * gst/rtsp-server/rtsp-media-factory.h:
2449 * gst/rtsp-server/rtsp-media.c:
2450 rtsp-media/-factory: Fix request pad name comments
2451 These must be escaped for gtk-doc to parse the comments without warnings.
2453 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2455 rtsp-media: remove transports if media is in error status
2456 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
2457 trying to change to GST_STATE_NULL and media is in error status, we
2458 remove all transports.
2459 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
2461 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
2463 * gst/rtsp-server/rtsp-media.c:
2464 rtsp-media: use element metadata to find payloader
2465 Use the element metadata to find the payloader instead of checking
2467 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
2469 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2471 rtsp-stream: add getter for payload type
2472 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
2473 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
2474 element and create the stream with this one instead of the dynpay%d
2476 https://bugzilla.gnome.org/show_bug.cgi?id=712396
2478 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2480 * gst/rtsp-server/rtsp-client.c:
2481 * gst/rtsp-server/rtsp-context.h:
2482 * gst/rtsp-server/rtsp-media.c:
2483 * gst/rtsp-server/rtsp-mount-points.c:
2484 * gst/rtsp-server/rtsp-server.c:
2485 * gst/rtsp-server/rtsp-token.c:
2486 rtsp-*: Refer to NULL as a constant in comments
2488 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2490 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2492 rtsp-*: Fix type name typos in comments
2493 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
2494 * rtsp-auth: Refer to part of constant name as text
2495 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
2496 * rtsp-session-media: Fix GstRTSPSessionMedia typo
2497 * rtsp-stream: Fix typo when refering to GstBin
2498 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2500 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2503 * docs/libs/gst-rtsp-server-docs.sgml:
2504 * docs/libs/gst-rtsp-server-sections.txt:
2505 docs: Improve documentation
2506 * Include annotation-glossary to quiet gtk-doc
2507 * Rename remaining ClientState -> Context
2508 * Rename object hierarchy file
2509 * Remove stale chapter references
2510 * Add missing function and object references
2511 * Include missing GstRTSPAddressPoolResult
2512 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2514 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2516 * gst/rtsp-server/rtsp-client.c:
2517 * gst/rtsp-server/rtsp-server.c:
2518 * gst/rtsp-server/rtsp-session-pool.c:
2519 * gst/rtsp-server/rtsp-session.c:
2520 * gst/rtsp-server/rtsp-stream.c:
2521 rtsp-server: sprinkle some allow-none annotations for g-i
2523 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
2525 * gst/rtsp-server/rtsp-stream.c:
2526 * gst/rtsp-server/rtsp-stream.h:
2527 stream: add method to filter transports
2528 Add a method to safely iterate and collect the stream transports
2529 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2531 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
2533 * gst/rtsp-server/rtsp-client.c:
2534 * gst/rtsp-server/rtsp-server.c:
2535 * gst/rtsp-server/rtsp-session-pool.c:
2536 * gst/rtsp-server/rtsp-session.c:
2537 rtsp: allow NULL func in filters
2538 Passing a null function make the filters return a list of
2541 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
2543 * gst/rtsp-server/rtsp-address-pool.c:
2544 * tests/check/gst/addresspool.c:
2545 address-pool: fix address increment
2546 Use a guint instead of guint8 to increment the address. It's still not
2547 completely correct because a guint might not be able to hold the complete
2548 address range, but that's an enhacement for later.
2549 Add unit test to test improved behaviour.
2550 https://bugzilla.gnome.org/show_bug.cgi?id=708237
2552 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
2554 * gst/rtsp-server/rtsp-client.c:
2555 * tests/check/gst/client.c:
2556 client: allow absolute path in requests
2557 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
2559 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
2561 * gst/rtsp-server/rtsp-client.c:
2562 * gst/rtsp-server/rtsp-client.h:
2563 client: make make_path_from_uri a vmethod
2565 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2567 * docs/libs/gst-rtsp-server-sections.txt:
2568 * gst/rtsp-server/rtsp-stream.c:
2569 * gst/rtsp-server/rtsp-stream.h:
2570 * tests/check/Makefile.am:
2571 * tests/check/gst/stream.c:
2572 stream: Add functions to get rtp and rtcp sockets
2573 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2575 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2577 * gst/rtsp-server/rtsp-context.c:
2578 * gst/rtsp-server/rtsp-context.h:
2579 context: defing a GType for the context
2580 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
2582 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
2584 * gst/rtsp-server/Makefile.am:
2585 * gst/rtsp-server/rtsp-auth.c:
2586 * gst/rtsp-server/rtsp-context.c:
2587 * gst/rtsp-server/rtsp-media.c:
2588 * gst/rtsp-server/rtsp-mount-points.c:
2589 * gst/rtsp-server/rtsp-server.h:
2590 * gst/rtsp-server/rtsp-session-media.c:
2591 * gst/rtsp-server/rtsp-session.c:
2592 * gst/rtsp-server/rtsp-stream.c:
2593 Fixed several GIR warnings
2595 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
2597 * gst/rtsp-server/rtsp-auth.c:
2600 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2602 * tests/check/Makefile.am:
2603 * tests/check/gst/token.c:
2604 tests: Add unit tests for token
2605 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2607 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2609 * gst/rtsp-server/rtsp-token.c:
2610 token: Validate args for gst_rtsp_token_is_allowed
2611 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
2613 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2615 * gst/rtsp-server/rtsp-token.c:
2616 token: Fix bug when creating empty token
2617 We always want to have a valid GstStructure in the token.
2618 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2620 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2622 * gst/rtsp-server/rtsp-thread-pool.c:
2623 thread-pool: avoid race in shutdown
2624 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
2625 don't actually stop the mainloop ever. Solve this race by adding an idle source
2626 to the mainloop that calls the _quit. This way we immediately exit the mainloop
2627 if quit was called before we started it.
2629 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2631 * tests/check/Makefile.am:
2632 * tests/check/gst/permissions.c:
2633 tests: Add unit tests for permissions
2634 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
2636 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2638 * tests/check/gst/mediafactory.c:
2639 tests: Test mediafactory permissions
2640 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2642 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2644 * gst/rtsp-server/rtsp-permissions.c:
2645 permissions: Fix refcounting when adding/removing roles
2646 Previously a role that was removed was unreffed twice, and when
2647 replacing an existing role the replaced role was freed while still being
2648 referenced. Both bugs are now fixed.
2649 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2651 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2653 * tests/check/gst/media.c:
2654 * tests/check/gst/mediafactory.c:
2655 * tests/check/gst/rtspserver.c:
2656 tests: Check gst_rtsp_url_parse return value
2657 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2659 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
2662 Automatic update of common submodule
2663 From 865aa20 to dbedaa0
2665 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
2667 * gst/rtsp-server/rtsp-server.c:
2668 rtsp-server: Fix socket leak
2669 https://bugzilla.gnome.org/show_bug.cgi?id=710088
2671 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
2673 * gst/rtsp-server/rtsp-session-pool.c:
2674 rtsp-session-pool: Make sure session IDs are properly URI-escaped
2675 https://bugzilla.gnome.org/show_bug.cgi?id=643812
2677 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2679 * examples/.gitignore:
2680 * examples/test-video.c:
2681 examples: fix compilation when WITH_AUTH is defined
2682 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2684 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
2687 gitignore: Add new test binary
2689 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
2691 * tests/check/Makefile.am:
2692 * tests/check/gst/threadpool.c:
2693 thread-pool: Add unit test for the thread pools
2694 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2696 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2698 * gst/rtsp-server/rtsp-thread-pool.c:
2699 thread-pool: Fix thread leak when reusing threads
2700 https://bugzilla.gnome.org/show_bug.cgi?id=709730
2702 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
2704 * gst/rtsp-server/rtsp-server.c:
2705 * tests/check/gst/rtspserver.c:
2706 tests: fixed racy behavior in rtspserver tests
2707 https://bugzilla.gnome.org/show_bug.cgi?id=710078
2709 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2711 * tests/check/gst/addresspool.c:
2712 tests: Improve address pool unit tests
2713 Add a range with mixed IPV4 and IPV6 addresses to pool.
2714 Get an IPV4 address from an IPV6-only pool.
2715 Get an IPV6 address from an IPV4-only pool.
2716 Reserve a IPV6 address from an IPV4-only pool.
2717 Check for unicast addresses in multicast-only pool.
2718 Check for unicast addresses in uni-/multicast-mixed pool.
2719 https://bugzilla.gnome.org/show_bug.cgi?id=710128
2721 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2723 * gst/rtsp-server/rtsp-client.c:
2724 client: append query string in PAUSE/PLAY/TEARDOWN as well
2726 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
2728 * gst/rtsp-server/rtsp-client.c:
2729 client: Add query to control path
2730 If the SETUP url contains a query it must be appended to the control
2731 path so that it matches any already created stream in the media. The
2732 query will also be appended to the session media path.
2734 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2736 * gst/rtsp-server/rtsp-media.c:
2737 rtsp-media: remove old line
2739 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
2741 * gst/rtsp-server/rtsp-stream.c:
2742 stream: Correct control comparison
2743 https://bugzilla.gnome.org/show_bug.cgi?id=709176
2745 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2747 * gst/rtsp-server/rtsp-media.c:
2748 media: Check dynamically if the pipeline supports seeking
2749 We should not depend on whether or not the pipeline state change
2750 returned NO_PREROLL or not. A media could dynamically change its
2751 element and switch from seekable to non seekable so it's best to test
2752 the seekable nature of the pipeline dynamically when we try to do a seek.
2754 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2756 * gst/rtsp-server/rtsp-media.c:
2757 media: Return FALSE if seeking is not supported
2759 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2761 * gst/rtsp-server/rtsp-media.c:
2762 rtsp-media: don't seek accurate by default
2763 Accurate seeking is perhaps a little overkill in the most common situation and
2764 causes some formats (mp3) over slow media to seek extremely slowly.
2766 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
2768 * tests/check/gst/rtspserver.c:
2769 tests: fix unit test
2770 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
2772 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
2774 * gst/rtsp-server/rtsp-client.c:
2775 client: Reply 400 if media cannot be constructed
2776 Reply 400 Bad Request instead of 503 Service Unavailable if media
2777 cannot be constructed in SETUP.
2778 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
2780 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
2782 * gst/rtsp-server/rtsp-client.c:
2783 client: Send setup reply once only
2784 If find_media() failed in handle_setup_request() two replies was sent.
2785 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
2787 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
2790 Automatic update of common submodule
2791 From 6b03ba7 to 865aa20
2793 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
2795 * gst/rtsp-server/rtsp-server.c:
2796 server: Emit client-connected signal earlier
2797 Emit client-connected before the client ref is given to a GSource,
2798 otherwise client-connected can be emitted after the client object has
2801 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
2803 * gst/rtsp-server/rtsp-address-pool.c:
2804 * gst/rtsp-server/rtsp-address-pool.h:
2805 * gst/rtsp-server/rtsp-stream.c:
2806 * tests/check/gst/addresspool.c:
2807 addresspool: return reason of failure
2808 Let gst_rtsp_address_pool_reserve_address() return the reason why
2809 the address could not be reserved.
2810 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
2812 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
2815 autogen.sh: Sync behaviour with other GStreamer modules
2816 Allows building from outside of tree amongst other things
2818 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
2821 Automatic update of common submodule
2822 From b613661 to 6b03ba7
2824 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
2827 Automatic update of common submodule
2828 From 74a6857 to b613661
2830 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
2833 Automatic update of common submodule
2834 From 01a7a46 to 74a6857
2836 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
2838 * gst/rtsp-server/rtsp-client.c:
2839 client: Do not read beyond end of path string
2840 If the setup was done without a control url, make sure we don't try to read the
2841 non-existing control string and crash.
2843 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2845 * gst/rtsp-server/rtsp-client.c:
2846 client: Fix RTPInfo header
2847 Refactor the method to make the content_base.
2848 Use the content-base and the control url to construct the RTPInfo
2851 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2853 * gst/rtsp-server/rtsp-client.c:
2854 client: map url to path only in describe
2855 Only map the request url to a path in the DESCRIBE method. The SDP then
2856 contains the base and control urls that should be used to SETUP/PAUSE/
2857 PLAY/TEARDOWN the media.
2859 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2861 * gst/rtsp-server/rtsp-client.c:
2862 Revert "client: map URL to path in requests"
2863 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
2864 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
2865 contains the base and control urls which are used in the SETUP, PLAY,
2866 PAUSE and TEARDOWN requests.
2868 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2870 * gst/rtsp-server/rtsp-client.c:
2871 client: map URL to path in requests
2873 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2875 * gst/rtsp-server/rtsp-client.c:
2876 * gst/rtsp-server/rtsp-mount-points.c:
2877 * gst/rtsp-server/rtsp-mount-points.h:
2878 mount-points: make vmethod to make path from uri
2879 Make a vmethod to transform an url into a path. The path is then used to lookup
2880 the factory. This makes it possible to also use other bits of the url, such as
2881 the query parameters, to locate the factory.
2883 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
2885 * gst/rtsp-server/rtsp-thread-pool.c:
2886 * gst/rtsp-server/rtsp-thread-pool.h:
2887 thread-pool: Add cleanup to wait for the threadpool to finish
2888 Also fix race condition if two threads are asking for the first
2889 thread from the thread pool at once. This would case two internal
2890 GThreadPools to be created.
2891 https://bugzilla.gnome.org/show_bug.cgi?id=707753
2893 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
2895 * gst/rtsp-server/rtsp-client.c:
2896 * tests/check/gst/client.c:
2897 client: free threadpool
2898 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2900 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
2902 * tests/check/gst/mountpoints.c:
2903 mountpoints tests: unref matched factories
2904 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2906 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
2908 * tests/check/gst/media.c:
2909 media tests: unref thread pool and caps
2910 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2912 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
2914 * gst/rtsp-server/rtsp-auth.c:
2915 * gst/rtsp-server/rtsp-media-factory.c:
2916 * gst/rtsp-server/rtsp-media.c:
2917 auth, media, media-factory: unref permissions
2918 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2920 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2922 * examples/Makefile.am:
2923 Makefile: add rule for appsrc example
2925 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2927 * examples/test-appsrc.c:
2928 tests: add appsrc example
2929 Add an example on how to use appsrc to feed the server pipeline with data.
2931 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
2933 * gst/rtsp-server/rtsp-client.c:
2934 rtsp-client: remove query part from content-base string
2935 Make sure that after the control url has been resolved, it's
2936 not a part of the query-string.
2937 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
2939 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2941 * gst/rtsp-server/rtsp-client.c:
2942 client: don't check url in response
2943 There is no url or method in the response to check
2945 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2947 * gst/rtsp-server/rtsp-client.c:
2948 * gst/rtsp-server/rtsp-client.h:
2949 Add handle-response signal for when we receive a GET_PARAMETER response
2951 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2953 * gst/rtsp-server/rtsp-server.c:
2954 Fix gst_rtsp_server_client_filter, using wrong variable type
2956 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
2958 * gst/rtsp-server/rtsp-media-factory-uri.c:
2959 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
2960 For AAC we need to check for framed=true instead of parsed=true.
2961 https://bugzilla.gnome.org/show_bug.cgi?id=701384
2963 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2965 * gst/rtsp-server/rtsp-stream.c:
2966 stream: optimize pipeline for protocols
2967 When TCP is not an allowed protocol for the stream, avoid creating the
2968 appsrc/appsink/queue and tee elements.
2970 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2972 * gst/rtsp-server/rtsp-media.c:
2973 media: set protocols on streams
2975 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2977 * gst/rtsp-server/rtsp-client.c:
2978 client: use protocols supported by stream
2980 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2982 * gst/rtsp-server/rtsp-media-factory.c:
2983 * gst/rtsp-server/rtsp-media.c:
2984 * gst/rtsp-server/rtsp-stream.c:
2985 media-factory: allow all protocols
2987 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2989 * gst/rtsp-server/rtsp-media.c:
2990 media: configure protocols in new streams
2992 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2994 * gst/rtsp-server/rtsp-stream.c:
2995 * gst/rtsp-server/rtsp-stream.h:
2996 stream: add protocols property
2998 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3000 * gst/rtsp-server/rtsp-media.c:
3001 rtsp-media: send state in "new-state" signal
3002 https://bugzilla.gnome.org/show_bug.cgi?id=705110
3004 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
3007 build: add subdir-objects to AM_INIT_AUTOMAKE
3008 Fixes warnings with automake 1.14
3009 https://bugzilla.gnome.org/show_bug.cgi?id=705350
3011 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3013 * docs/libs/gst-rtsp-server-sections.txt:
3014 * gst/rtsp-server/rtsp-client.c:
3015 * gst/rtsp-server/rtsp-server.c:
3016 * gst/rtsp-server/rtsp-server.h:
3017 server: add method to iterate clients of server
3019 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3021 * gst/rtsp-server/rtsp-media.c:
3022 * gst/rtsp-server/rtsp-media.h:
3023 Add vmethod for rtsp-media subclass to access rtpbin
3025 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3027 * gst/rtsp-server/rtsp-client.h:
3028 small documentation fix
3030 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3032 * gst/rtsp-server/rtsp-client.c:
3033 Do not take range header if range is invalid
3035 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3037 * docs/libs/gst-rtsp-server-sections.txt:
3038 * gst/rtsp-server/rtsp-media.c:
3039 media: add docs for new method
3041 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3043 * gst/rtsp-server/rtsp-media.c:
3044 * gst/rtsp-server/rtsp-media.h:
3045 Add API to rtsp-media set the pipeline's state
3047 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3049 * gst/rtsp-server/rtsp-media.c:
3050 Update current position/duration when gst_rtsp_media_get_range_string is called
3052 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3054 * examples/test-cgroups.c:
3055 tests: add some more docs
3057 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3059 * examples/test-cgroups.c:
3060 * gst/rtsp-server/Makefile.am:
3061 * gst/rtsp-server/rtsp-auth.c:
3062 * gst/rtsp-server/rtsp-auth.h:
3063 * gst/rtsp-server/rtsp-client.c:
3064 * gst/rtsp-server/rtsp-client.h:
3065 * gst/rtsp-server/rtsp-context.c:
3066 * gst/rtsp-server/rtsp-context.h:
3067 * gst/rtsp-server/rtsp-params.c:
3068 * gst/rtsp-server/rtsp-params.h:
3069 * gst/rtsp-server/rtsp-server.c:
3070 * gst/rtsp-server/rtsp-thread-pool.c:
3071 * gst/rtsp-server/rtsp-thread-pool.h:
3072 * tests/check/gst/client.c:
3073 ClientState -> Context
3074 Rename the clientstate to context and put the code in a separate file.
3076 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3078 * examples/test-auth.c:
3079 * gst/rtsp-server/rtsp-auth.c:
3080 * gst/rtsp-server/rtsp-auth.h:
3081 auth: add support for default token
3082 The default token is used when the user is not authenticated and can be used to
3083 give minimal permissions.
3085 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3087 * examples/test-auth.c:
3088 * gst/rtsp-server/rtsp-auth.c:
3089 auth: use defines when possible
3091 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3093 * gst/rtsp-server/rtsp-address-pool.c:
3094 address-pool: improve docs
3096 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3098 * gst/rtsp-server/rtsp-permissions.c:
3099 permissions: add the role to the copy
3101 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
3103 * gst/rtsp-server/rtsp-permissions.c:
3104 permissions: Also copy the roles
3106 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
3108 * gst/rtsp-server/rtsp-permissions.c:
3109 permissions: Make it build
3111 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3113 * gst/rtsp-server/rtsp-address-pool.h:
3116 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3118 * docs/libs/gst-rtsp-server-sections.txt:
3119 * gst/rtsp-server/rtsp-auth.c:
3120 * gst/rtsp-server/rtsp-auth.h:
3121 * gst/rtsp-server/rtsp-media.c:
3122 * gst/rtsp-server/rtsp-session-media.c:
3123 * gst/rtsp-server/rtsp-stream-transport.c:
3124 * gst/rtsp-server/rtsp-stream-transport.h:
3125 * gst/rtsp-server/rtsp-stream.c:
3126 * tests/check/gst/client.c:
3129 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3131 * docs/libs/gst-rtsp-server-sections.txt:
3132 * gst/rtsp-server/rtsp-address-pool.c:
3133 * gst/rtsp-server/rtsp-address-pool.h:
3134 * tests/check/gst/addresspool.c:
3135 * tests/check/gst/rtspserver.c:
3136 address-pool: cleanups
3137 Remove redundant method, improve docs.
3139 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3141 * docs/libs/gst-rtsp-server-sections.txt:
3142 * gst/rtsp-server/rtsp-auth.h:
3143 * gst/rtsp-server/rtsp-permissions.c:
3144 * gst/rtsp-server/rtsp-permissions.h:
3145 * gst/rtsp-server/rtsp-token.c:
3148 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3150 * gst/rtsp-server/rtsp-permissions.c:
3151 permissions: implement _remove_role
3153 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3155 * gst/rtsp-server/rtsp-permissions.c:
3156 permissions: update docs
3158 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3160 * tests/check/gst/client.c:
3161 tests: simplify tests
3162 Client settings are now disabled by default so we don't need an auth
3163 module to disable them.
3165 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3167 * gst/rtsp-server/rtsp-auth.c:
3168 auth: add default authorizations
3169 When no auth module is specified, use our table of defaults to look up the
3170 default value of the check instead of always allowing everything. This was
3171 we can disallow client settings by default.
3173 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3176 README: update readme
3178 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3180 * gst/rtsp-server/rtsp-thread-pool.c:
3181 * gst/rtsp-server/rtsp-thread-pool.h:
3182 thread-pool: add more docs
3184 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3186 * gst/rtsp-server/rtsp-thread-pool.c:
3187 * gst/rtsp-server/rtsp-thread-pool.h:
3188 thread-pool: fix race in thread reuse
3189 If we try to reuse a thread right after we made it stop, we end up using a
3190 stopped thread. Catch this case and only reuse threads that are not stopping.
3192 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3194 * gst/rtsp-server/rtsp-server.c:
3195 server: add small debug
3197 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3199 * tests/check/gst/client.c:
3201 Add some permissions to media so we can use the auth and enable
3204 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3206 * gst/rtsp-server/rtsp-client.c:
3207 client: support pushed context in handle_request
3208 If we already have a pushed state, reuse it and add our own things. This makes
3209 it easier to write tests.
3211 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3213 * gst/rtsp-server/rtsp-auth.c:
3214 auth: don't auth on methods
3215 Don't authorize on methods anymore but on the resources that we
3216 try to access, this is more flexible.
3217 Move the authorization checks to where they are needed and let the
3218 check return the response on error.
3220 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3222 * gst/rtsp-server/rtsp-mount-points.c:
3223 mount-points: add some debug
3225 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3227 * tests/check/gst/client.c:
3228 tests: almost fix test
3230 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3232 * gst/rtsp-server/rtsp-auth.c:
3233 * gst/rtsp-server/rtsp-auth.h:
3234 * gst/rtsp-server/rtsp-client.c:
3235 * gst/rtsp-server/rtsp-client.h:
3236 * gst/rtsp-server/rtsp-server.c:
3237 * gst/rtsp-server/rtsp-server.h:
3238 auth: let the auth module check client_settings
3239 Let the auth module decide if client settings are allowed for the
3242 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3244 * gst/rtsp-server/rtsp-token.c:
3245 * gst/rtsp-server/rtsp-token.h:
3246 token: add method to check boolean permission
3248 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3250 * examples/test-auth.c:
3251 * examples/test-cgroups.c:
3252 * gst/rtsp-server/rtsp-token.c:
3253 * gst/rtsp-server/rtsp-token.h:
3254 token: simplify token constructor
3255 Use variable arguments to make easier API.
3257 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3259 * examples/test-auth.c:
3260 * examples/test-cgroups.c:
3261 * gst/rtsp-server/rtsp-media-factory.c:
3262 * gst/rtsp-server/rtsp-media-factory.h:
3263 media-factory: add convenience API for factory
3265 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3267 * examples/test-auth.c:
3268 * examples/test-cgroups.c:
3269 * gst/rtsp-server/rtsp-permissions.c:
3270 * gst/rtsp-server/rtsp-permissions.h:
3271 permissions: simplify API a little
3272 Avoid passing GstStructure in the add_role method, use varargs instead
3273 to construct the structure behind the scenes. We can then also use the
3274 structure name as the role and simplify some more logic.
3276 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3278 * gst/rtsp-server/rtsp-auth.c:
3281 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3283 * gst/rtsp-server/rtsp-auth.c:
3284 * gst/rtsp-server/rtsp-auth.h:
3285 * gst/rtsp-server/rtsp-client.c:
3286 auth: handle unauthorized response
3287 Move handling of the unauthorized response to the auth module, it can add
3288 the appropriate headers to request authorization for the required method
3289 much better than the client.
3291 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3293 * gst/rtsp-server/rtsp-client.c:
3294 * gst/rtsp-server/rtsp-client.h:
3295 client: allow for sending any message, not only requests
3296 Change the _send_request() method to _send_message() so that we
3297 can both send requests and replies.
3299 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3301 * docs/libs/gst-rtsp-server-sections.txt:
3302 * gst/rtsp-server/rtsp-server.h:
3305 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3307 * examples/test-video.c:
3308 * gst/rtsp-server/rtsp-auth.c:
3309 * gst/rtsp-server/rtsp-auth.h:
3310 * gst/rtsp-server/rtsp-server.c:
3311 * gst/rtsp-server/rtsp-server.h:
3312 auth: move TLS handling to auth module
3313 Remove the TLS settings on the server and move it to the auth module because
3314 that is where security related bits go.
3316 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3318 * gst/rtsp-server/rtsp-client.c:
3319 * gst/rtsp-server/rtsp-client.h:
3320 client: add state push/pop
3322 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3324 * gst/rtsp-server/rtsp-client.c:
3325 * gst/rtsp-server/rtsp-client.h:
3326 client: add connection to state
3328 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3330 * gst/rtsp-server/rtsp-mount-points.c:
3331 mount-points: fix debug
3333 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3335 * tests/check/gst/media.c:
3336 tests: fix media test
3338 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3340 * gst/rtsp-server/rtsp-thread-pool.c:
3341 thread-pool: we don't require a state
3343 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3345 * gst/rtsp-server/rtsp-server.c:
3346 server: let context ref the server
3347 So that we don't risk losing the server object early anc crash.
3349 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3351 * tests/check/gst/client.c:
3352 tests: fix client test
3354 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3357 * docs/libs/gst-rtsp-server-docs.sgml:
3358 * docs/libs/gst-rtsp-server-sections.txt:
3359 * gst/rtsp-server/rtsp-address-pool.c:
3360 * gst/rtsp-server/rtsp-auth.c:
3361 * gst/rtsp-server/rtsp-client.c:
3362 * gst/rtsp-server/rtsp-client.h:
3363 * gst/rtsp-server/rtsp-media-factory-uri.c:
3364 * gst/rtsp-server/rtsp-media-factory.c:
3365 * gst/rtsp-server/rtsp-media-factory.h:
3366 * gst/rtsp-server/rtsp-media.c:
3367 * gst/rtsp-server/rtsp-mount-points.c:
3368 * gst/rtsp-server/rtsp-params.c:
3369 * gst/rtsp-server/rtsp-permissions.c:
3370 * gst/rtsp-server/rtsp-sdp.c:
3371 * gst/rtsp-server/rtsp-server.c:
3372 * gst/rtsp-server/rtsp-server.h:
3373 * gst/rtsp-server/rtsp-session-media.c:
3374 * gst/rtsp-server/rtsp-session-pool.c:
3375 * gst/rtsp-server/rtsp-session.c:
3376 * gst/rtsp-server/rtsp-stream-transport.c:
3377 * gst/rtsp-server/rtsp-stream.c:
3378 * gst/rtsp-server/rtsp-thread-pool.c:
3379 * gst/rtsp-server/rtsp-token.c:
3382 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3384 * gst/rtsp-server/rtsp-session-pool.c:
3385 * gst/rtsp-server/rtsp-session-pool.h:
3386 session-pool: make vmethod to create a session
3387 Make a vmethod to create a sessions so that subclasses can create
3388 custom session objects
3390 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3392 * gst/rtsp-server/rtsp-auth.c:
3393 * gst/rtsp-server/rtsp-media-factory.h:
3394 * gst/rtsp-server/rtsp-media.h:
3395 * gst/rtsp-server/rtsp-mount-points.h:
3396 * gst/rtsp-server/rtsp-session-pool.h:
3397 * gst/rtsp-server/rtsp-stream.h:
3400 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3402 * docs/libs/gst-rtsp-server-docs.sgml:
3403 * docs/libs/gst-rtsp-server-sections.txt:
3404 * gst/rtsp-server/rtsp-address-pool.c:
3405 * gst/rtsp-server/rtsp-address-pool.h:
3406 * gst/rtsp-server/rtsp-auth.c:
3407 * gst/rtsp-server/rtsp-client.h:
3408 * gst/rtsp-server/rtsp-media-factory.h:
3409 * gst/rtsp-server/rtsp-media.c:
3410 * gst/rtsp-server/rtsp-media.h:
3411 * gst/rtsp-server/rtsp-permissions.c:
3412 * gst/rtsp-server/rtsp-permissions.h:
3413 * gst/rtsp-server/rtsp-server.h:
3414 * gst/rtsp-server/rtsp-session-media.c:
3415 * gst/rtsp-server/rtsp-session-media.h:
3416 * gst/rtsp-server/rtsp-session-pool.h:
3417 * gst/rtsp-server/rtsp-session.h:
3418 * gst/rtsp-server/rtsp-stream-transport.h:
3419 * gst/rtsp-server/rtsp-stream.c:
3420 * gst/rtsp-server/rtsp-thread-pool.h:
3423 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3426 * examples/Makefile.am:
3427 configure: compile cgroup example conditionally
3428 Only compile the cgroup example when we have libcgroup
3430 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3433 * examples/Makefile.am:
3434 * examples/test-cgroups.c:
3435 examples: add cgroups example
3437 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3439 * tests/check/gst/rtspserver.c:
3440 tests: fix compilation
3442 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3444 * gst/rtsp-server/rtsp-thread-pool.c:
3445 thread-pool: fix vmethod invocation
3447 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3449 * gst/rtsp-server/rtsp-thread-pool.c:
3450 * gst/rtsp-server/rtsp-thread-pool.h:
3451 thread-pool: store thread type in thread
3453 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3455 * gst/rtsp-server/rtsp-client.c:
3456 client: pass thread from pool to media _prepare
3457 Get a thread from the configured threadpool and pass it to the prepare method of
3460 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3462 * gst/rtsp-server/rtsp-media.c:
3463 * gst/rtsp-server/rtsp-media.h:
3464 media: Accept a thread in _prepare
3465 Remove out own threadpool handling and use the provided thread and
3466 maincontext for the bus messages and the state changes.
3468 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3470 * gst/rtsp-server/rtsp-server.c:
3471 server: configure client thread pool
3473 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3475 * gst/rtsp-server/rtsp-client.c:
3476 * gst/rtsp-server/rtsp-client.h:
3477 client: add method to configure thread pool
3479 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3481 * gst/rtsp-server/rtsp-client.h:
3482 * gst/rtsp-server/rtsp-server.c:
3483 * gst/rtsp-server/rtsp-server.h:
3484 server: use thread pool
3485 Use the thread pool instead of doing our own thing.
3487 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3489 * gst/rtsp-server/Makefile.am:
3490 * gst/rtsp-server/rtsp-thread-pool.c:
3491 * gst/rtsp-server/rtsp-thread-pool.h:
3492 thread-pool: add object to manage threads
3493 Add an object to manage the client and media threads.
3495 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3497 * gst/rtsp-server/rtsp-auth.c:
3498 auth: debug authorization check
3500 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3502 * gst/rtsp-server/rtsp-media.c:
3503 media: start media pipeline in context
3504 Start the media pipeline in the provided context (or our default one
3505 when NULL). This makes sure that we run the bus thread in this context and that
3506 all media threads are children of this context.
3508 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3510 * gst/rtsp-server/rtsp-media-factory.c:
3511 factory: pass permissions to media by default
3513 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3515 * examples/test-auth.c:
3516 test: add permissions to auth test
3517 Ass some permissions to the media factory in the test.
3519 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3521 * gst/rtsp-server/rtsp-auth.c:
3522 * gst/rtsp-server/rtsp-auth.h:
3523 * gst/rtsp-server/rtsp-client.c:
3524 auth: simplify auth checks
3525 Remove client from methods, it's now in the state
3526 Perform the check specified by the string, use the information from the
3527 thread local context.
3529 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3531 * gst/rtsp-server/rtsp-client.c:
3532 * gst/rtsp-server/rtsp-client.h:
3533 client: add state to current thread
3534 Add the client to the ClientState object.
3535 Place the ClientState on the current thread.
3537 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3539 * gst/rtsp-server/rtsp-media-factory.c:
3540 * gst/rtsp-server/rtsp-media-factory.h:
3541 * gst/rtsp-server/rtsp-media.c:
3542 * gst/rtsp-server/rtsp-media.h:
3543 media: make it possible to set permissions
3544 Make it possible to set permissions on media and media factory objects
3546 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3548 * gst/rtsp-server/Makefile.am:
3549 * gst/rtsp-server/rtsp-permissions.c:
3550 * gst/rtsp-server/rtsp-permissions.h:
3551 permissions: add permissions object
3552 Add a mini object to store permissions based on a role.
3554 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3556 * examples/test-auth.c:
3557 * gst/rtsp-server/rtsp-auth.c:
3558 * gst/rtsp-server/rtsp-auth.h:
3559 * gst/rtsp-server/rtsp-client.c:
3560 auth: add auth checks
3561 Add an enum with auth checks and implement the checks in the auth object.
3562 Perform the checks from the client.
3564 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3566 * examples/test-auth.c:
3567 * gst/rtsp-server/rtsp-auth.c:
3568 * gst/rtsp-server/rtsp-auth.h:
3569 * gst/rtsp-server/rtsp-client.h:
3570 auth: use the token after authentication
3571 After we authenticated a user, keep the Token around in the state.
3573 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3575 * gst/rtsp-server/rtsp-client.c:
3576 * gst/rtsp-server/rtsp-media.c:
3577 * gst/rtsp-server/rtsp-media.h:
3578 * tests/check/gst/media.c:
3579 media: add optional context for bus messages
3580 Add an optional mainloop to _prepare that will handle the bus messages instead
3581 of always using the shared mainloop.
3583 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3585 * gst/rtsp-server/Makefile.am:
3586 * gst/rtsp-server/rtsp-token.c:
3587 * gst/rtsp-server/rtsp-token.h:
3588 token: add authorization token
3589 Add a simply miniobject that contains the authorizations. The object contains a
3590 GstStructure that hold all authorization fields. When a user is authenticated,
3591 the auth module will create a Token for the user. The token is then used to
3592 check what operations the user is allowed to do and various other configuration
3595 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3597 * examples/test-auth.c:
3598 * gst/rtsp-server/rtsp-auth.c:
3599 * gst/rtsp-server/rtsp-auth.h:
3600 * gst/rtsp-server/rtsp-client.c:
3601 * gst/rtsp-server/rtsp-client.h:
3602 * gst/rtsp-server/rtsp-media-factory.c:
3603 * gst/rtsp-server/rtsp-media-factory.h:
3604 * gst/rtsp-server/rtsp-media.c:
3605 * gst/rtsp-server/rtsp-media.h:
3606 auth: remove auth from media and factory
3607 Remove the auth object from media and factory. We want to have the RTSPClient
3608 authenticate and authorize resources, there is no need to place another auth
3609 manager on the media/factory.
3611 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3613 * examples/test-auth.c:
3614 * gst/rtsp-server/rtsp-auth.c:
3615 * gst/rtsp-server/rtsp-auth.h:
3616 * gst/rtsp-server/rtsp-client.h:
3617 auth: add support for multiple basic auth tokens
3618 Make it possible to add multiple basic authorisation tokens to one authorization
3619 object. Associate with each token an authorization group that will define what
3620 capabilities are allowed.
3622 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3624 * gst/rtsp-server/rtsp-client.c:
3625 client: error out on non-aggregate control
3626 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
3628 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3630 * gst/rtsp-server/rtsp-client.c:
3631 client: rework setup request a little
3632 Cache the media in DESCRIBE based on the longest matching path with the uri
3633 that we can find in the mount points.
3634 Rework the setup request a little to get the media from the session or from
3635 the longest matching path, this way we can derive the control string as
3636 everything after the path instead of hardcoding it.
3637 Find the stream based on the control string and only open a session when all
3640 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3642 * gst/rtsp-server/rtsp-media.c:
3643 * gst/rtsp-server/rtsp-media.h:
3644 media: add method to find a stream by control url
3646 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3648 * gst/rtsp-server/rtsp-stream.c:
3649 * gst/rtsp-server/rtsp-stream.h:
3650 stream: add method to check control url of stream
3652 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3654 * gst/rtsp-server/rtsp-client.c:
3655 * gst/rtsp-server/rtsp-session-media.c:
3656 * gst/rtsp-server/rtsp-session-media.h:
3657 * gst/rtsp-server/rtsp-session.c:
3658 * gst/rtsp-server/rtsp-session.h:
3659 session: use path matching for session media
3660 Use a path string instead of a uri to lookup session media in the sessions. Also
3661 use path matching to find the largest possible path that matches.
3663 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3665 * gst/rtsp-server/rtsp-client.c:
3666 * gst/rtsp-server/rtsp-mount-points.c:
3667 * gst/rtsp-server/rtsp-mount-points.h:
3668 * tests/check/gst/mountpoints.c:
3669 mount-points: remove useless vmethod
3670 Making lookups in the mount points should not be done with a URL, if there is a
3671 mapping to be done from URL to mount points, we'll need to do it somewhere
3674 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3676 * gst/rtsp-server/rtsp-mount-points.c:
3677 * gst/rtsp-server/rtsp-mount-points.h:
3678 * tests/check/gst/mountpoints.c:
3679 mount-points: improve mount point searching
3680 Use a GSequence to keep track of the mount points.
3681 Match a URL to the longest matching registered mount point. This should be the
3682 URL to perform aggreagate control and the remainder is the stream specific
3684 Add some unit tests for this.
3686 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
3688 * gst/rtsp-server/Makefile.am:
3689 rtsp-server: Allow building of static library
3691 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3693 * tests/check/gst/mediafactory.c:
3694 tests: fix compilation
3696 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3698 * gst/rtsp-server/rtsp-sdp.c:
3699 sdp: get control string from stream
3700 Use the control string as configured in the stream.
3702 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3704 * gst/rtsp-server/rtsp-stream.c:
3705 * gst/rtsp-server/rtsp-stream.h:
3706 stream: add methods and property to set control string
3708 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3710 * gst/rtsp-server/rtsp-client.c:
3712 Rename variables for clarity
3713 Keep media in state when we can
3715 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3717 * gst/rtsp-server/rtsp-client.c:
3718 * gst/rtsp-server/rtsp-stream.c:
3719 * gst/rtsp-server/rtsp-stream.h:
3720 stream: add more support for IPv6
3721 Rename _get_address to _get_multicast_address in GstRTSPStream to
3722 make it clear that this function only deals with multicast.
3723 Make it possible to have both an IPv4 and IPv6 multicast address on
3724 a stream. Give the client an IPv4 or IPv6 address depending on the
3725 address it used to connect to the server.
3726 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
3728 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3730 * gst/rtsp-server/rtsp-client.c:
3733 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3735 * gst/rtsp-server/rtsp-stream.c:
3736 stream: handle failed port allocation
3737 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
3738 can't allocate any family at all. Also keep track of what port families we
3740 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
3742 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3744 * gst/rtsp-server/rtsp-stream.c:
3745 stream: improve docs
3747 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3749 * gst/rtsp-server/rtsp-stream-transport.c:
3750 stream-transport: remove old if 0 block
3752 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
3754 * tests/check/gst/client.c:
3756 gst_rtsp_client_get_uri() has been removed
3757 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
3759 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3761 * gst/rtsp-server/rtsp-client.c:
3762 * gst/rtsp-server/rtsp-client.h:
3763 client: add method to filter managed sessions
3764 Add a method to filter the sessions managed by this client connection.
3765 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
3767 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3769 * gst/rtsp-server/rtsp-client.c:
3770 * gst/rtsp-server/rtsp-client.h:
3771 client: remove _get_uri() method
3772 Remove the get_uri() method on the client. A client has no uri, the uri
3773 property is an internal property to manage the last cached media for
3776 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3778 * gst/rtsp-server/rtsp-media-factory.h:
3779 media-factory: fix typo
3781 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
3783 * gst/rtsp-server/rtsp-media.c:
3784 rtsp-media: Do not leak the query in default_query_stop
3785 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
3787 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3789 * gst/rtsp-server/rtsp-media.c:
3790 media: don't unlock when conversion fails
3791 Don't unlock the state lock when conversion fails because it was not locked.
3793 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3795 * gst/rtsp-server/rtsp-media.c:
3796 * gst/rtsp-server/rtsp-media.h:
3797 Add query_position and query_stop vmethods to rtsp-media
3799 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3801 * gst/rtsp-server/rtsp-media.c:
3802 Fix typo in property install for rtsp-media's time-provider
3804 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3806 * gst/rtsp-server/rtsp-client.c:
3807 * gst/rtsp-server/rtsp-client.h:
3808 client: clean some variables
3809 Clean some variables and add some guards to _send_request()
3811 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3813 * gst/rtsp-server/rtsp-client.c:
3814 * gst/rtsp-server/rtsp-client.h:
3815 Add gst_rtsp_client_send_request API
3816 This makes it possible to send arbitrary messages to a client, such as
3817 SET_PARAMETER or GET_PARAMETER
3819 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3821 * gst/rtsp-server/rtsp-media.c:
3822 * gst/rtsp-server/rtsp-media.h:
3823 media: add _get_element() method
3824 Add method to get the element used when creating the media.
3825 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
3827 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3829 * gst/rtsp-server/rtsp-media.c:
3832 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3834 * gst/rtsp-server/rtsp-stream.c:
3835 * gst/rtsp-server/rtsp-stream.h:
3836 stream: allow access to the rtp session
3837 https://bugzilla.gnome.org/show_bug.cgi?id=703004
3839 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
3841 * gst/rtsp-server/rtsp-stream.c:
3842 * gst/rtsp-server/rtsp-stream.h:
3843 dscp qos support in gst-rtsp-stream
3844 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
3846 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3848 * tests/check/gst/rtspserver.c:
3850 Actually do what the comment says. Also keep the old code around, not sure what
3851 should happen when you get a 454 from a TEARDOWN, does it close the connection?
3852 it currently doesn't.
3854 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3856 * gst/rtsp-server/rtsp-client.c:
3857 client: also watch newly created session
3858 When we newly created a session, start watching it immediately instead of
3859 on the next request.
3861 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
3863 * tests/check/gst/client.c:
3864 tests: add unit test for new-session
3865 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
3867 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3869 * gst/rtsp-server/rtsp-client.c:
3870 client: emit new-session when new session is created
3871 Only emit new-session when we created a new session for a client, not when a
3872 client picked up a previous session.
3873 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
3875 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
3877 * gst/rtsp-server/rtsp-client.c:
3878 client: handle asterisk as path in requests
3879 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
3881 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3883 * gst/rtsp-server/rtsp-media.c:
3884 media: handle segment query format mismatch
3885 It's possible that the segment query returns with a different format than what
3886 we asked for, handle this case also.
3888 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
3890 * gst/rtsp-server/rtsp-media.c:
3891 media: use segment stop in collect_media_stats
3892 Use segment stop instead of duration as range end point.
3893 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
3895 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3897 * gst/rtsp-server/rtsp-media.c:
3898 * tests/check/gst/media.c:
3899 rtsp-media: Do not leak the element in take_pipeline
3900 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
3902 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
3904 * gst/rtsp-server/rtsp-client.c:
3905 * gst/rtsp-server/rtsp-client.h:
3906 rtsp-client: Make configure_client_transport virtual
3907 This patch makes configure_client_transport virtual. The functionality is
3908 needed to handle some weird clients sending multicast transport settings as url
3910 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
3912 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3914 * gst/rtsp-server/rtsp-client.c:
3915 * gst/rtsp-server/rtsp-client.h:
3916 rtsp-client: Make param_set and param_get virtual
3917 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
3919 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
3921 * gst/rtsp-server/rtsp-client.c:
3922 * gst/rtsp-server/rtsp-media.c:
3923 * gst/rtsp-server/rtsp-media.h:
3924 media: convert_range replaces get_range_times
3925 get_range_times worked for handling UTC ranges for seeks, but we also
3926 need to convert back from NPT to the requested unit in
3927 get_range_string. convert_range is now used for both.
3928 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
3930 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3932 * gst/rtsp-server/rtsp-client.c:
3933 * gst/rtsp-server/rtsp-sdp.c:
3934 * gst/rtsp-server/rtsp-sdp.h:
3935 sdp: cleanup sdp info
3936 We don't need to pass the proto, we can more easily check a boolean.
3937 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
3939 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
3941 * gst/rtsp-server/rtsp-sdp.c:
3942 use 0.0.0.0 or :: for c= line instead of server address
3944 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
3946 * gst/rtsp-server/rtsp-client.c:
3947 use local address, not remote, in SDP
3948 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
3950 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3953 Automatic update of common submodule
3954 From 098c0d7 to 01a7a46
3956 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
3958 * gst/rtsp-server/rtsp-media.c:
3959 * gst/rtsp-server/rtsp-media.h:
3960 media: possibility to override range time conversion
3961 Make it possible to override the conversion from GstRTSPTimeRange to
3962 GstClockTimes, that is done before seeking on the media
3963 pipeline. Overriding can be useful for UTC ranges, where the default
3964 conversion gives nanoseconds since 1900.
3965 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
3967 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
3969 * gst/rtsp-server/rtsp-server.c:
3970 * gst/rtsp-server/rtsp-server.h:
3971 rtsp-server: Expose the use_client_settings API
3972 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
3974 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
3976 * gst/rtsp-server/rtsp-client.c:
3977 * gst/rtsp-server/rtsp-stream.c:
3978 * gst/rtsp-server/rtsp-stream.h:
3979 rtspstream: handle both ipv4 and ipv6 clients
3980 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
3982 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3984 * gst/rtsp-server/rtsp-sdp.c:
3985 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
3986 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
3987 We already have a way to place extra attributes in the SDP by using a string
3988 property with prefix x- or a- in the caps.
3990 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3992 * gst/rtsp-server/rtsp-sdp.c:
3993 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
3994 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
3995 We already have a way to place extra attributes in the SDP, just make a string
3996 property in the payloader with a- or x- prefix.
3998 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4000 * gst/rtsp-server/rtsp-sdp.c:
4001 rtsp: place a- and x- properties as attributes
4002 application/x-rtp has properties with a- and x- prefixes that should be
4003 placed as attributes in the SDP for the media instead of being added to the
4006 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4008 * examples/Makefile.am:
4009 * examples/test-video.c:
4010 example: add TLS example
4012 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4014 * gst/rtsp-server/rtsp-server.c:
4015 * gst/rtsp-server/rtsp-server.h:
4016 server: add support for TLS
4017 Add methods to set and get a TLS certificate.
4018 Add vmethod to configure a new connection. By default, configure the TLS
4019 certificate in a new connection if needed.
4021 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4023 * gst/rtsp-server/rtsp-server.c:
4024 * gst/rtsp-server/rtsp-server.h:
4025 server: remove accept_client vmethod
4026 This vmethod is not very useful so remove it.
4028 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4030 * gst/rtsp-server/rtsp-server.c:
4031 server: don't crash on NULL GError
4033 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
4035 * gst/rtsp-server/rtsp-session-pool.c:
4036 rtsp-session-pool: corrected session timeout detection
4037 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
4039 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4041 * gst/rtsp-server/rtsp-client.c:
4042 client: improve debug
4044 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4046 * gst/rtsp-server/rtsp-client.c:
4047 * gst/rtsp-server/rtsp-client.h:
4048 * gst/rtsp-server/rtsp-server.c:
4049 server: refactor connection setup
4050 Let the server accept the socket connection and construct a GstRTSPConnection
4051 from it. Remove the code from the client and let the client only deal with
4052 a fully configure GstRTSPConnection object.
4053 We will need this later when the server will configure the connection for
4056 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4058 * gst/rtsp-server/rtsp-stream.c:
4059 stream: keep the transport object alive
4060 Keep the transport object alive while we have it as qdata on the
4063 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
4065 * gst/rtsp-server/rtsp-client.c:
4066 * gst/rtsp-server/rtsp-server.c:
4067 rtsp-server: Do not crash on nmapping of server
4068 * generate error when gst_rtsp_connection_accept fails
4069 * do not stop accepting incoming connections because
4070 accepting a client fails
4071 https://bugzilla.gnome.org/show_bug.cgi?id=701072
4073 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
4075 * gst/rtsp-server/rtsp-client.c:
4076 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
4077 https://bugzilla.gnome.org/show_bug.cgi?id=700953
4079 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4081 * gst/rtsp-server/rtsp-sdp.c:
4082 rtsp-sdp: Parse framerate caps field and set SDP attribute
4083 The SDP attribute and its format is described in RFC4566.
4084 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4086 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
4088 * gst/rtsp-server/rtsp-sdp.c:
4089 rtsp-sdp: Parse width/height from caps and set SDP attribute
4090 The SDP attribute and its format is described in RFC6064.
4091 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4093 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
4095 * gst/rtsp-server/rtsp-sdp.c:
4096 * tests/check/gst/client.c:
4097 rtsp-sdp: add bandwidth line
4098 https://bugzilla.gnome.org/show_bug.cgi?id=699220
4100 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4103 Automatic update of common submodule
4104 From 5edcd85 to 098c0d7
4106 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4108 * tests/check/gst/media.c:
4109 tests: add dynamic payloader prepare/unprepare check
4111 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4113 * gst/rtsp-server/rtsp-media.c:
4114 media: release lock when removing fakesink
4116 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4118 * gst/rtsp-server/rtsp-stream.c:
4119 stream: set elements to NULL before removing
4120 When removing a stream, set the elements to NULL first. This avoids
4121 element-is-not-in-NULL-state errors when we dispose the elements.
4123 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4126 Automatic update of common submodule
4127 From 3cb3d3c to 5edcd85
4129 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4131 * gst/rtsp-server/rtsp-media.c:
4132 * gst/rtsp-server/rtsp-media.h:
4133 media: listen to pad-removed signals
4134 Listen to the pad-removed signal and remove the stream associated with the
4136 Add signal to be notified of the removed pad.
4137 Remove the fakesink in unprepare()
4138 Fix signatures of the signal methods
4140 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4142 * examples/test-sdp.c:
4143 tests: add example of reusable pipelines
4145 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4147 * gst/rtsp-server/rtsp-stream.c:
4148 * gst/rtsp-server/rtsp-stream.h:
4149 stream: add method to get the srcpad
4151 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4153 * tests/check/gst/media.c:
4154 check: add media prepare/unprepare test
4155 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4157 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
4159 * gst/rtsp-server/rtsp-media.c:
4160 media: disconnect from signal handlers in unprepare()
4161 We connected to the pad-added and no-more-pads signals in prepare() so
4162 we need to disconnect from them in unprepare().
4163 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4165 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
4167 * gst/rtsp-server/rtsp-media.c:
4168 media: don't free streams array
4169 Don't free the streams array in the unprepare() method, they were not
4171 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4173 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
4175 * gst/rtsp-server/rtsp-media.c:
4176 media: don't unref the pipeline in unprepare
4177 Unprepare() should undo what prepare() does. Because the pipeline is
4178 not created in prepare(), we should not unref it in unprepare()
4180 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
4182 * gst/rtsp-server/rtsp-stream.c:
4183 stream: clear session and caps for reuse
4184 Set the session and caps to NULL after unref otherwise we might unref
4186 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4188 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
4190 * gst/rtsp-server/rtsp-client.c:
4191 client: send out teardown signal before tearing down
4192 The advantage is that in the signal handler you get direct access to
4193 information about what streams are about to get torn down (in the
4194 GstRTSPClientState).
4195 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
4197 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
4199 * gst/rtsp-server/rtsp-client.c:
4200 * gst/rtsp-server/rtsp-client.h:
4201 client: expose connection
4202 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
4204 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
4207 Automatic update of common submodule
4208 From aed87ae to 3cb3d3c
4210 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4212 * gst/rtsp-server/rtsp-media.c:
4213 * gst/rtsp-server/rtsp-media.h:
4214 * gst/rtsp-server/rtsp-session-media.c:
4215 * gst/rtsp-server/rtsp-session-media.h:
4216 media: add method to get the base_time of the pipeline
4217 Together with a shared clock, this base-time could eventually be sent to
4218 the client so that it can reconstruct the exact running-time of the clock
4221 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4223 * gst/rtsp-server/Makefile.am:
4224 * gst/rtsp-server/rtsp-media.c:
4225 * gst/rtsp-server/rtsp-media.h:
4226 * gst/rtsp-server/rtsp-sdp.c:
4227 media: add GstNetTimeProvider support
4228 Add a property to let the media provide a GstNetTimeProvider for its clock.
4229 Make methods to get the clock and nettimeprovider
4230 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
4231 provider and also the current time of the clock. This should make it possible
4232 for (GStreamer) clients to slave their clock to the server clock.
4234 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4237 Automatic update of common submodule
4238 From 04c7a1e to aed87ae
4240 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4242 * gst/rtsp-server/rtsp-media.c:
4243 media: wait for buffering to complete
4244 Wait for buffering to complete before changing the state to the target state.
4246 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4248 * gst/rtsp-server/rtsp-media.c:
4249 media: small cleanup
4251 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
4253 * tests/check/gst/rtspserver.c:
4254 tests: remove extra unref in test_setup_non_existing_stream
4255 The unref is not needed anymore, teardown runs without it.
4256 https://bugzilla.gnome.org/show_bug.cgi?id=696542
4258 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
4260 * tests/check/gst/rtspserver.c:
4261 tests: GSocketService cleanup in test_bind_already_in_use
4262 Use g_socket_service_stop so the rtspserver test stops listening for
4263 incoming connections in test_bind_already_in_use.
4264 https://bugzilla.gnome.org/show_bug.cgi?id=696541
4266 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
4268 * gst/rtsp-server/rtsp-media-factory.c:
4269 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
4270 Instead use a GWeakRef which is safe to use
4271 This is a known GLib bug, see:
4272 https://bugzilla.gnome.org/show_bug.cgi?id=667145
4274 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
4276 * gst/rtsp-server/rtsp-client.c:
4277 * gst/rtsp-server/rtsp-media.c:
4278 * gst/rtsp-server/rtsp-media.h:
4279 * gst/rtsp-server/rtsp-sdp.c:
4280 * tests/check/gst/media.c:
4281 * tests/check/gst/rtspserver.c:
4282 rtsp-media/client: Reply to PLAY request with same type of Range
4283 Remember the type of Range from the PLAY request and use the same type for
4286 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
4288 * gst/rtsp-server/rtsp-client.c:
4289 * gst/rtsp-server/rtsp-client.h:
4290 * tests/check/gst/client.c:
4291 rtsp-client: expose uri
4293 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
4295 * tests/check/gst/mediafactory.c:
4296 tests: Hold ref while creating second media
4297 To test if the media aren't shared, make sure we keep the first one while creating a second
4298 otherwise the same memory address may be reused.
4300 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
4303 configure: remove out-of-date comment
4305 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
4308 .gitignore: ignore more build files
4310 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
4312 * tests/check/Makefile.am:
4313 tests: use right _LIBS variable for gst-plugins-base libs
4315 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4317 * tests/check/Makefile.am:
4318 check: add librtp to libs
4320 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
4322 * tests/check/gst/rtspserver.c:
4323 tests: Add test to check selecting a port the server will send from
4325 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
4327 * tests/check/gst/rtspserver.c:
4328 tests: Make sure packets are actually received
4330 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4332 * gst/rtsp-server/rtsp-stream.c:
4333 stream: Select unicast address from pool if appropriate
4335 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
4337 * gst/rtsp-server/rtsp-stream.c:
4338 stream: Properties are always there in Gst 1.0
4340 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4342 * tests/check/gst/addresspool.c:
4343 tests: Add tests for unicast addresses in pool
4345 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
4347 * gst/rtsp-server/rtsp-address-pool.c:
4348 * tests/check/gst/addresspool.c:
4349 address-pool: Verify that multicast addresses are used for multicast and vice-versa
4351 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
4353 * docs/libs/gst-rtsp-server-sections.txt:
4354 * gst/rtsp-server/rtsp-address-pool.c:
4355 * gst/rtsp-server/rtsp-address-pool.h:
4356 * gst/rtsp-server/rtsp-stream.c:
4357 * tests/check/gst/addresspool.c:
4358 address-pool: Add unicast addresses
4360 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4363 * gst/rtsp-server/rtsp-server.c:
4364 * tests/check/gst/rtspserver.c:
4365 rtsp-server: Limit the number of threads per server instance
4366 If we exceed the maximum, just round robin the clients over the existing
4369 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
4371 * gst/rtsp-server/rtsp-server.c:
4372 rtsp-server: No need to store the GMainContext in the client context
4374 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
4376 * tests/check/gst/rtspserver.c:
4377 tests: Add test for client disconnection
4379 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4381 * tests/check/gst/rtspserver.c:
4382 tests: Test client and session timeouts with multiple threads
4384 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
4386 * gst/rtsp-server/rtsp-address-pool.c:
4387 * gst/rtsp-server/rtsp-auth.c:
4388 * gst/rtsp-server/rtsp-client.c:
4389 * gst/rtsp-server/rtsp-media-factory-uri.c:
4390 * gst/rtsp-server/rtsp-media-factory.c:
4391 * gst/rtsp-server/rtsp-media.c:
4392 * gst/rtsp-server/rtsp-mount-points.c:
4393 * gst/rtsp-server/rtsp-server.c:
4394 * gst/rtsp-server/rtsp-session-media.c:
4395 * gst/rtsp-server/rtsp-session-pool.c:
4396 * gst/rtsp-server/rtsp-session.c:
4397 Document locking and its order
4399 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
4401 * tests/check/gst/rtspserver.c:
4402 tests: Test that slow DESCRIBE don't block other clients
4404 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
4406 * tests/check/gst/client.c:
4407 tests: Add tests for client-requested multicast address
4409 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
4411 * docs/libs/gst-rtsp-server-sections.txt:
4412 docs: Put the various functions in the right sections
4414 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
4416 * docs/libs/gst-rtsp-server-docs.sgml:
4417 * docs/libs/gst-rtsp-server-sections.txt:
4418 * gst/rtsp-server/rtsp-address-pool.c:
4419 * gst/rtsp-server/rtsp-address-pool.h:
4420 docs: Generate docs for GstRTSPAddressPool
4422 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4424 * gst/rtsp-server/rtsp-client.c:
4425 * gst/rtsp-server/rtsp-stream.c:
4426 * gst/rtsp-server/rtsp-stream.h:
4427 client: Check client provided addresses against the address pool
4429 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
4431 * gst/rtsp-server/rtsp-address-pool.c:
4432 * gst/rtsp-server/rtsp-address-pool.h:
4433 * tests/check/gst/addresspool.c:
4434 address-pool: Add API to request a specific address from the pool
4435 Also add relevant unit tests.
4437 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
4439 * tests/check/gst/mediafactory.c:
4440 tests: Check the passing around of a RTSPAddressPool
4441 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
4442 way down to the stream.
4444 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
4446 * tests/check/gst/addresspool.c:
4447 tests: Add more tests for the address pool
4449 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
4451 * gst/rtsp-server/rtsp-address-pool.c:
4452 address-pool: Fix off by one error
4453 When splitting a port range, the port after a skip is not part of range.
4455 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
4458 Automatic update of common submodule
4459 From 2de221c to 04c7a1e
4461 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
4464 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
4465 AM_CONFIG_HEADER was removed in automake 1.13
4466 https://bugzilla.gnome.org/show_bug.cgi?id=693368
4468 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
4471 Automatic update of common submodule
4472 From a942293 to 2de221c
4474 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4476 * gst/rtsp-server/rtsp-client.c:
4477 client: make sure the watch exists while sending data
4478 Protect the send_func with a lock. This allows us to wait for sending
4479 to complete before changing the send_func and user_data. We add an
4480 extra ref to the watch to make sure that it remains valid during
4482 When closing the connection, set the send_func to NULL
4483 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
4485 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4487 * tests/check/Makefile.am:
4488 tests: use GST_*_1_0 environment variables everywhere
4489 The _1_0 suffixed environment variables override the
4490 non-suffixed ones, so if we're in an environment that
4491 sets the _1_0 suffixed ones, such as jhbuild, we need
4492 to set those to make sure ours actually always get
4495 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4498 Automatic update of common submodule
4499 From acb04d9 to a942293
4501 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4503 * gst/rtsp-server/rtsp-client.c:
4504 rtsp-client: set the client backlog
4505 Set the client backlog to a reasonable default
4507 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
4509 * gst/rtsp-server/rtsp-media.c:
4510 rtsp-media: Make the element a constructor parameter
4511 https://bugzilla.gnome.org/show_bug.cgi?id=689594
4513 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4515 * docs/libs/Makefile.am:
4516 docs: Link with gcov library when gcov is enabled
4517 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
4519 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4521 * gst/rtsp-server/rtsp-media.c:
4522 media: match prepare with unprepare
4523 Really unprepare when there were an equal amount of prepare calls.
4525 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4527 * gst/rtsp-server/rtsp-media.c:
4528 media: media has to be unprepared in finalize
4529 Because unprepare takes away the last ref on the media.
4531 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4533 * gst/rtsp-server/rtsp-client.c:
4534 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
4535 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
4536 We can't use the refcount to trigger unprepare because it is the unprepare call
4537 that removes the last refcount after all messages are consumed. What we should
4538 probably do is make a prepared refcount and only unprepare when the refcount
4541 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4543 * gst/rtsp-server/rtsp-media.c:
4544 media: let the source unref the last media ref
4545 the last ref to the media is held by the source so we don't need to add more ref
4546 and unrefs, we simply destroy the media when the source is gone.
4548 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4550 * gst/rtsp-server/rtsp-media.c:
4551 media: improve debug
4553 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4555 * gst/rtsp-server/rtsp-media.c:
4557 Make sure we are in the right state when collecting the position and duration.
4558 Only make ourselves PREPARED when we were previously PREPARING.
4560 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4562 * gst/rtsp-server/rtsp-media.c:
4563 media: use g_object_ref/unref for GObjects
4565 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
4567 * gst/rtsp-server/rtsp-client.c:
4568 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
4569 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
4570 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
4571 isn't being used anymore.
4573 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
4575 * gst/rtsp-server/rtsp-media.c:
4576 Fix compiler warning
4578 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
4580 * gst/rtsp-server/rtsp-media-factory-uri.c:
4581 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
4583 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4585 * gst/rtsp-server/rtsp-session-media.h:
4588 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4590 * gst/rtsp-server/rtsp-media.c:
4591 * tests/check/gst/media.c:
4592 media: avoid element leak
4594 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4596 * gst/rtsp-server/rtsp-media.c:
4597 media: require an element in media constructor
4599 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4601 * gst/rtsp-server/rtsp-client.c:
4602 Revert "client: TEARDOWN brings that state to Init again"
4603 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
4604 The object is already disposed, there is no point in setting the state.
4606 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4608 * gst/rtsp-server/rtsp-client.c:
4609 client: TEARDOWN brings that state to Init again
4611 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4613 * docs/libs/gst-rtsp-server-sections.txt:
4614 * examples/test-auth.c:
4615 * gst/rtsp-server/rtsp-auth.c:
4616 * gst/rtsp-server/rtsp-auth.h:
4617 * gst/rtsp-server/rtsp-client.c:
4618 * gst/rtsp-server/rtsp-client.h:
4619 * gst/rtsp-server/rtsp-media-factory-uri.c:
4620 * gst/rtsp-server/rtsp-media-factory-uri.h:
4621 * gst/rtsp-server/rtsp-media-factory.c:
4622 * gst/rtsp-server/rtsp-media-factory.h:
4623 * gst/rtsp-server/rtsp-media.c:
4624 * gst/rtsp-server/rtsp-media.h:
4625 * gst/rtsp-server/rtsp-mount-points.c:
4626 * gst/rtsp-server/rtsp-mount-points.h:
4627 * gst/rtsp-server/rtsp-sdp.c:
4628 * gst/rtsp-server/rtsp-server.c:
4629 * gst/rtsp-server/rtsp-server.h:
4630 * gst/rtsp-server/rtsp-session-media.c:
4631 * gst/rtsp-server/rtsp-session-media.h:
4632 * gst/rtsp-server/rtsp-session-pool.c:
4633 * gst/rtsp-server/rtsp-session-pool.h:
4634 * gst/rtsp-server/rtsp-session.c:
4635 * gst/rtsp-server/rtsp-session.h:
4636 * gst/rtsp-server/rtsp-stream-transport.c:
4637 * gst/rtsp-server/rtsp-stream-transport.h:
4638 * gst/rtsp-server/rtsp-stream.c:
4639 * gst/rtsp-server/rtsp-stream.h:
4640 * tests/check/gst/media.c:
4641 rtsp: make object details private
4642 Make all object details private
4643 Add methods to access private bits
4645 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4647 * tests/check/Makefile.am:
4648 * tests/check/gst/media.c:
4649 tests: add media tests
4651 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4653 * gst/rtsp-server/rtsp-media.c:
4654 media: check if prepared for some methods
4655 Check that the media object is prepared before doing seek and getting the
4656 current position etc.
4657 Add some g_return checks.
4659 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4661 * tests/check/Makefile.am:
4662 * tests/check/gst/mediafactory.c:
4663 tests: add mediafactory test
4665 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4667 * gst/rtsp-server/rtsp-stream.c:
4668 stream: improve debug
4670 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4672 * gst/rtsp-server/rtsp-media.c:
4673 * gst/rtsp-server/rtsp-media.h:
4674 media: unref pipeline in finalize to avoid leaking it
4676 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4678 * gst/rtsp-server/rtsp-media-factory-uri.c:
4679 * gst/rtsp-server/rtsp-media.c:
4680 rtsp: use gst_object_unref on GstObjects
4682 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4684 * gst/rtsp-server/rtsp-media-factory.c:
4685 media-factory: require an url
4687 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4689 * examples/test-uri.c:
4690 examples: fix include
4692 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4694 * gst/rtsp-server/rtsp-server.h:
4695 server: remove unused include
4697 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4699 * tests/check/Makefile.am:
4700 * tests/check/gst/mountpoints.c:
4701 tests: add test for mountpoints
4703 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4705 * gst/rtsp-server/rtsp-client.c:
4706 client: fix factory leak
4707 Keep the factory in the state object only for authorization checks and make
4708 sure we unref it on failure. Also don't keep invalid objects in the state
4711 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4713 * gst/rtsp-server/rtsp-mount-points.c:
4714 mounts: add g_return_if guards
4716 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4718 * tests/check/gst/client.c:
4719 tests: add more tests
4721 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4723 * gst/rtsp-server/rtsp-client.c:
4724 client: improve debug
4726 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4728 * gst/rtsp-server/rtsp-client.c:
4729 client: improve debug and fix leaks
4730 Cleanup the uri and session when there is a bad request.
4732 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4737 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4739 * tests/check/gst/client.c:
4740 test: add test for session in options request
4742 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4744 * gst/rtsp-server/rtsp-client.c:
4745 client: use 454 when session can't be found
4746 We should use 454 when a session can't be found because there was no session
4747 pool configured in the server. This is not a server configuration problem
4748 because the server on which the request is done might not be the same one that
4749 will keep the sessions for us and so it does not need to support sessions.
4751 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4753 * gst/rtsp-server/rtsp-client.c:
4754 client: only free connection when there is one
4755 It's possible that the client doesn't have a connection when we try to free it.
4757 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4759 * tests/check/Makefile.am:
4760 * tests/check/gst/client.c:
4761 tests: add unit test for the client object
4763 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4765 * gst/rtsp-server/rtsp-client.c:
4766 client: small cleanup
4768 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4770 * gst/rtsp-server/rtsp-client.h:
4771 client: remove unused include
4773 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4775 * gst/rtsp-server/rtsp-client.c:
4776 client: fix compilation
4778 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4780 * gst/rtsp-server/rtsp-client.c:
4781 client: call destroy without the lock
4783 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4785 * gst/rtsp-server/rtsp-client.c:
4786 * gst/rtsp-server/rtsp-client.h:
4787 client: make the client usable without a socket
4788 Make a method to let the client handle a message and a callback when the client
4789 wants us to send a response message back. This makes it possible to also use the
4790 client object without the sockets, which should make it easier to test.
4792 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4794 * gst/rtsp-server/rtsp-client.c:
4795 * gst/rtsp-server/rtsp-client.h:
4796 client: small cleanup
4798 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4800 * docs/libs/gst-rtsp-server-sections.txt:
4801 * gst/rtsp-server/rtsp-client.c:
4802 * gst/rtsp-server/rtsp-client.h:
4803 * gst/rtsp-server/rtsp-server.c:
4804 client: remove reference to server
4805 We don't need to keep a ref to the server
4807 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4809 * gst/rtsp-server/rtsp-client.c:
4810 * gst/rtsp-server/rtsp-client.h:
4812 Also add some g_return_if()
4814 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4816 * gst/rtsp-server/rtsp-client.c:
4817 client: log more errors
4819 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4821 * gst/rtsp-server/rtsp-client.c:
4822 client: fix compilation
4824 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4826 * gst/rtsp-server/rtsp-client.c:
4827 * gst/rtsp-server/rtsp-client.h:
4828 client: add generic close-after-send support
4829 Add a property to send_response() to close the connection after the response has
4830 been sent to the client.
4832 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4835 * docs/libs/gst-rtsp-server-docs.sgml:
4836 * docs/libs/gst-rtsp-server-sections.txt:
4837 * docs/libs/gst-rtsp-server.types:
4838 * examples/test-auth.c:
4839 * examples/test-launch.c:
4840 * examples/test-mp4.c:
4841 * examples/test-multicast.c:
4842 * examples/test-multicast2.c:
4843 * examples/test-ogg.c:
4844 * examples/test-readme.c:
4845 * examples/test-sdp.c:
4846 * examples/test-uri.c:
4847 * examples/test-video.c:
4848 * gst/rtsp-server/Makefile.am:
4849 * gst/rtsp-server/rtsp-auth.h:
4850 * gst/rtsp-server/rtsp-client.c:
4851 * gst/rtsp-server/rtsp-client.h:
4852 * gst/rtsp-server/rtsp-media-mapping.c:
4853 * gst/rtsp-server/rtsp-media-mapping.h:
4854 * gst/rtsp-server/rtsp-mount-points.c:
4855 * gst/rtsp-server/rtsp-mount-points.h:
4856 * gst/rtsp-server/rtsp-server.c:
4857 * gst/rtsp-server/rtsp-server.h:
4858 * gst/rtsp-server/rtsp-session-media.c:
4859 * gst/rtsp-server/rtsp-session-pool.c:
4860 * gst/rtsp-server/rtsp-session-pool.h:
4861 * tests/check/gst/rtspserver.c:
4862 MediaMapping -> MountPoints
4863 Describes better what the object manages.
4865 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4868 configure: bump required version of -base
4870 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4872 * gst/rtsp-server/rtsp-media.c:
4875 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4877 * gst/rtsp-server/rtsp-media.c:
4878 * gst/rtsp-server/rtsp-media.h:
4879 media: support more Range formats
4880 Use the new -base methods to convert the Range string into a seek start and stop
4883 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4885 * examples/test-launch.c:
4886 examples: fix whitespace
4888 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4890 * examples/test-auth.c:
4891 test-auth: add example of how to remove sessions
4892 Add an example of the session filter api.
4894 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4896 * examples/test-uri.c:
4897 test-uri: remove mapping example
4899 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4901 * examples/test-uri.c:
4902 test-uri: fix callback signature
4904 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4906 * gst/rtsp-server/rtsp-media-factory.c:
4907 factory: keep ref to factory while media active
4908 While the media from a factory is alive, keep a ref to the factory.
4909 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
4911 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4913 * gst/rtsp-server/rtsp-media-factory-uri.c:
4914 factory-uri: add some debug
4916 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4918 * gst/rtsp-server/rtsp-stream.c:
4919 stream: set udp sources to PLAYING
4920 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
4921 so that it doesn't cause our pipeline to produce ASYNC-DONE.
4923 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4925 * gst/rtsp-server/rtsp-media-factory-uri.c:
4926 factory-uri: take ref to factory
4927 Take a ref to the factory that we place in our list.
4929 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4931 * tests/Makefile.am:
4932 * tests/test-reuse.c:
4933 test: add test for server reuse
4934 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
4936 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
4938 * gst/rtsp-server/rtsp-server.c:
4939 server: start and stop multiple times
4940 Stop listening on the RTSP port when the GSource is removed, so clients
4941 can't connect and the server can be started again.
4942 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
4944 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4946 * gst/rtsp-server/rtsp-server.c:
4947 server: fix small leak
4949 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4951 * gst/rtsp-server/rtsp-media.c:
4952 media: unref source in finish_unprepare
4953 The source is created in prepare, unref it in finish_unprepare.
4954 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
4956 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
4958 * gst/rtsp-server/rtsp-client.c:
4959 * gst/rtsp-server/rtsp-media.c:
4960 rtsp-media: remove bus watch before finalizing
4961 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
4962 * An extra media ref is added for the bus watch. This extra ref is unreffed by
4963 the GDestroyNotify function.
4964 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
4965 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
4966 gst_rtsp_media_unprepare before unreffing the media.
4967 This way, the bus watch will be removed before the media is finalized.
4968 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
4970 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
4972 * gst/rtsp-server/rtsp-client.c:
4973 * gst/rtsp-server/rtsp-client.h:
4974 client: wait until the TEARDOWN response is sent to close the connection
4975 Responses can be sent async so we need to wait until the TEARDOWN response has
4976 been written before we close the connection to the client. This avoids the risk
4977 of writing/polling closed sockets.
4978 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
4980 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
4982 * gst/rtsp-server/rtsp-stream.c:
4983 rtsp-stream: plug socket leak
4984 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
4986 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
4989 Automatic update of common submodule
4990 From 6bb6951 to a72faea
4992 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
4994 * gst/rtsp-server/rtsp-media-factory-uri.c:
4995 rtsp-server: don't use deprecated API
4997 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
4999 * gst/rtsp-server/rtsp-client.c:
5000 rtsp-client: fix unused-but-set-variable compiler warning
5001 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
5003 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5006 * docs/libs/gst-rtsp-server-sections.txt:
5007 * gst/rtsp-server/rtsp-client.c:
5010 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5012 * examples/Makefile.am:
5013 * examples/test-multicast2.c:
5014 examples: add another multicast example
5015 Add an example for how to configure separate multicast ranges for each media
5018 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5020 * examples/test-multicast.c:
5023 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5025 * gst/rtsp-server/rtsp-client.c:
5026 * gst/rtsp-server/rtsp-media.c:
5027 * gst/rtsp-server/rtsp-session-media.c:
5028 * gst/rtsp-server/rtsp-session-media.h:
5029 * gst/rtsp-server/rtsp-stream-transport.c:
5030 * gst/rtsp-server/rtsp-stream-transport.h:
5031 stream: use the address managed by the stream
5032 Use the address managed by the stream for multicast. This allows us to have 1
5033 multicast address for each stream.
5034 Because the address is now managed by the stream we don't have to pass it around
5036 Set the address pool on the streams.
5038 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5040 * gst/rtsp-server/rtsp-client.c:
5041 * gst/rtsp-server/rtsp-media.c:
5042 * gst/rtsp-server/rtsp-stream.c:
5045 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5047 * gst/rtsp-server/rtsp-media.c:
5048 * gst/rtsp-server/rtsp-media.h:
5049 media: add signal for new streams
5050 This allows applications to listen for new streams and configure properties on
5051 them, like the address pool.
5053 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5055 * gst/rtsp-server/rtsp-media.c:
5056 media: configure address pool in new streams
5058 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5060 * gst/rtsp-server/rtsp-stream.c:
5061 * gst/rtsp-server/rtsp-stream.h:
5062 stream: add methods to deal with address pool
5063 Add methods to get and set the address pool for the stream
5064 Add method to allocate and get the multicast addresses for this stream.
5066 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5068 * docs/libs/gst-rtsp-server-sections.txt:
5069 * gst/rtsp-server/rtsp-media.c:
5070 * gst/rtsp-server/rtsp-media.h:
5071 media: remove MTU property
5072 It is a stream property
5074 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5076 * gst/rtsp-server/rtsp-client.c:
5077 client: set blocksize only on stream
5078 Set the blocksize only on the current stream.
5080 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5082 * gst/rtsp-server/rtsp-stream.c:
5083 stream: share src and sink sockets
5084 the allocated socket is in the used-socket property, not socket.
5086 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5088 * gst/rtsp-server/rtsp-address-pool.c:
5089 * gst/rtsp-server/rtsp-address-pool.h:
5090 * gst/rtsp-server/rtsp-client.c:
5091 * gst/rtsp-server/rtsp-session-media.c:
5092 * gst/rtsp-server/rtsp-session-media.h:
5093 * gst/rtsp-server/rtsp-stream-transport.c:
5094 * gst/rtsp-server/rtsp-stream-transport.h:
5095 * tests/check/gst/addresspool.c:
5096 rtsp: make address-pool return an address object
5097 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
5098 store more info in the structure and allows us to more easily return the address
5099 to the right pool when no longer needed.
5100 Pass the address to the StreamTransport so that we can return it to the pool
5101 when the stream transport is freed or changed.
5103 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5105 * examples/Makefile.am:
5106 * examples/test-multicast.c:
5107 examples: add multicast example
5108 Show how to set up the multicast address pool so that media can be
5109 server with multicast.
5111 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5113 * gst/rtsp-server/rtsp-client.c:
5114 * gst/rtsp-server/rtsp-media-factory.c:
5115 * gst/rtsp-server/rtsp-media-factory.h:
5116 * gst/rtsp-server/rtsp-media.c:
5117 * gst/rtsp-server/rtsp-media.h:
5118 rtsp: use AddressPool
5119 Remove the multicast_group property.
5120 Use the configured addresspool to allocate multicast addresses.
5122 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5124 * gst/rtsp-server/rtsp-address-pool.c:
5125 * gst/rtsp-server/rtsp-address-pool.h:
5126 address-pool: add clear method
5128 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5130 * gst/rtsp-server/rtsp-address-pool.c:
5131 address-pool: small cleanups
5133 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5135 * tests/check/Makefile.am:
5136 * tests/check/gst/addresspool.c:
5137 tests: add addresspool unit test
5139 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5141 * gst/rtsp-server/Makefile.am:
5142 * gst/rtsp-server/rtsp-address-pool.c:
5143 * gst/rtsp-server/rtsp-address-pool.h:
5144 address-pool: add object to manage multicast addresses
5145 Make an object that can manage a rage of multicast addresses and ports.
5147 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5149 * gst/rtsp-server/rtsp-server.c:
5150 server: set default max-threads property
5152 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5154 * gst/rtsp-server/rtsp-media.c:
5155 media: wait for concurrent _prepare
5156 If a prepare is busy, wait for the result.
5158 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5160 * gst/rtsp-server/rtsp-media.c:
5161 media: add lock around message handler
5162 We don't want to dispatch messages while we are still processing the result of
5165 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5167 * gst/rtsp-server/rtsp-media.c:
5168 * gst/rtsp-server/rtsp-media.h:
5169 media: add lock to protect state changes
5171 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5173 * gst/rtsp-server/rtsp-stream.c:
5174 * gst/rtsp-server/rtsp-stream.h:
5177 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5179 * gst/rtsp-server/rtsp-stream-transport.c:
5180 * gst/rtsp-server/rtsp-stream-transport.h:
5181 * gst/rtsp-server/rtsp-stream.c:
5182 stream-transport: add keep-alive method
5184 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5186 * gst/rtsp-server/rtsp-stream-transport.c:
5187 * gst/rtsp-server/rtsp-stream-transport.h:
5188 * gst/rtsp-server/rtsp-stream.c:
5189 stream-transport: add method to handle RTP/RTCP
5190 Call new methods instead of poking into the structures directly.
5192 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5194 * gst/rtsp-server/rtsp-session-media.c:
5195 * gst/rtsp-server/rtsp-session-media.h:
5196 session-media: add locking
5198 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5200 * gst/rtsp-server/rtsp-session.c:
5201 * gst/rtsp-server/rtsp-session.h:
5202 session: add locking
5204 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5206 * gst/rtsp-server/rtsp-server.c:
5207 server: free old socket
5209 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5211 * gst/rtsp-server/rtsp-media-mapping.c:
5212 * gst/rtsp-server/rtsp-media-mapping.h:
5213 mapping: add locking
5215 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5217 * gst/rtsp-server/rtsp-media-factory.c:
5218 media-factory: add locking
5220 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5222 * gst/rtsp-server/rtsp-auth.c:
5223 * gst/rtsp-server/rtsp-auth.h:
5226 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5228 * gst/rtsp-server/rtsp-server.c:
5229 * gst/rtsp-server/rtsp-server.h:
5230 server: add max-thread property
5232 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5234 * gst/rtsp-server/rtsp-server.c:
5235 * gst/rtsp-server/rtsp-server.h:
5236 server: use a threadpool for the mainloops
5238 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5240 * gst/rtsp-server/rtsp-client.c:
5241 * gst/rtsp-server/rtsp-client.h:
5242 client: rename method
5243 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
5244 don't really create the client from the socket, we use the socket for the
5247 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5249 * gst/rtsp-server/rtsp-client.c:
5250 * gst/rtsp-server/rtsp-client.h:
5251 * gst/rtsp-server/rtsp-server.c:
5252 server: rework maincontext handling in clients
5253 Make a separate method to attach a client to a MainContext.
5254 Let the server decide in what GMainContext the client will operate and give this
5255 context to the client in attach. Then the server can later decide to use a
5256 separate thread for each client or just use the mainthread.
5258 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5260 * gst/rtsp-server/rtsp-client.c:
5261 * gst/rtsp-server/rtsp-session.c:
5262 * gst/rtsp-server/rtsp-session.h:
5263 session: move session header code in session object
5265 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
5269 * examples/test-auth.c:
5270 * examples/test-launch.c:
5271 * examples/test-mp4.c:
5272 * examples/test-ogg.c:
5273 * examples/test-readme.c:
5274 * examples/test-sdp.c:
5275 * examples/test-uri.c:
5276 * examples/test-video.c:
5277 * gst/rtsp-server/rtsp-auth.c:
5278 * gst/rtsp-server/rtsp-auth.h:
5279 * gst/rtsp-server/rtsp-client.c:
5280 * gst/rtsp-server/rtsp-client.h:
5281 * gst/rtsp-server/rtsp-media-factory-uri.c:
5282 * gst/rtsp-server/rtsp-media-factory-uri.h:
5283 * gst/rtsp-server/rtsp-media-factory.c:
5284 * gst/rtsp-server/rtsp-media-factory.h:
5285 * gst/rtsp-server/rtsp-media-mapping.c:
5286 * gst/rtsp-server/rtsp-media-mapping.h:
5287 * gst/rtsp-server/rtsp-media.c:
5288 * gst/rtsp-server/rtsp-media.h:
5289 * gst/rtsp-server/rtsp-params.c:
5290 * gst/rtsp-server/rtsp-params.h:
5291 * gst/rtsp-server/rtsp-sdp.c:
5292 * gst/rtsp-server/rtsp-sdp.h:
5293 * gst/rtsp-server/rtsp-server.c:
5294 * gst/rtsp-server/rtsp-server.h:
5295 * gst/rtsp-server/rtsp-session-media.c:
5296 * gst/rtsp-server/rtsp-session-media.h:
5297 * gst/rtsp-server/rtsp-session-pool.c:
5298 * gst/rtsp-server/rtsp-session-pool.h:
5299 * gst/rtsp-server/rtsp-session.c:
5300 * gst/rtsp-server/rtsp-session.h:
5301 * gst/rtsp-server/rtsp-stream-transport.c:
5302 * gst/rtsp-server/rtsp-stream-transport.h:
5303 * gst/rtsp-server/rtsp-stream.c:
5304 * gst/rtsp-server/rtsp-stream.h:
5305 * tests/check/gst/rtspserver.c:
5306 * tests/test-cleanup.c:
5309 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
5311 * gst/rtsp-server/rtsp-media.c:
5312 * gst/rtsp-server/rtsp-session-media.c:
5313 * gst/rtsp-server/rtsp-session.c:
5314 rtsp-server: added annotations to indicate type of ownership transfer of return values
5315 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5317 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5320 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
5322 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
5325 * bindings/Makefile.am:
5326 * bindings/vala/Makefile.am:
5327 * bindings/vala/gst-rtsp-server-0.10.deps:
5328 * bindings/vala/gst-rtsp-server-0.10.vapi:
5329 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
5330 * bindings/vala/packages/gst-rtsp-server-0.10.files:
5331 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5332 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5333 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
5335 bindings: remove vala bindings
5336 They'll be reunited with the other GStreamer bindings
5337 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5339 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5341 * gst/rtsp-server/rtsp-client.c:
5342 * gst/rtsp-server/rtsp-session-media.c:
5343 * gst/rtsp-server/rtsp-session-media.h:
5344 * gst/rtsp-server/rtsp-stream-transport.c:
5345 * gst/rtsp-server/rtsp-stream-transport.h:
5346 rtsp: only create transport when needed
5347 Only create the StreamTransport when configured.
5349 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5351 * gst/rtsp-server/rtsp-client.c:
5352 client: small cleanup
5354 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5356 * gst/rtsp-server/rtsp-client.c:
5357 * gst/rtsp-server/rtsp-client.h:
5358 * gst/rtsp-server/rtsp-stream-transport.c:
5359 * gst/rtsp-server/rtsp-stream-transport.h:
5360 rtsp: refactor configuration of transport
5361 Move the configuration of the transport to a place where it makes
5364 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5366 * gst/rtsp-server/rtsp-client.c:
5367 client: refactor transport parsing
5369 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5371 * gst/rtsp-server/rtsp-client.c:
5372 client: refuse to change the MTU on shared media
5373 If we change the MTU of chared media, it changes for all clients.
5374 We don't want to set the MTU to something large for clients that
5377 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5379 * examples/test-mp4.c:
5380 * gst/rtsp-server/rtsp-media.c:
5381 small fixes to docs and debug
5383 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5385 * gst/rtsp-server/rtsp-stream.c:
5386 stream: transports must already have been removed
5388 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5390 * gst/rtsp-server/rtsp-media.c:
5391 * gst/rtsp-server/rtsp-stream.c:
5392 * gst/rtsp-server/rtsp-stream.h:
5393 stream: improve join and leave of the pipeline
5395 Do the cleanup properly
5398 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5400 * gst/rtsp-server/rtsp-media.c:
5401 media: move unprepare below default implementation
5402 Makes it easier to find the default implementation
5404 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5406 * gst/rtsp-server/rtsp-media.c:
5407 media: signal unprepared when we actually finish
5409 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5411 * gst/rtsp-server/rtsp-media.c:
5412 media: no need to unlock, unprepare does that when needed
5414 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5416 * docs/libs/gst-rtsp-server-sections.txt:
5417 * gst/rtsp-server/rtsp-media-factory.h:
5418 * gst/rtsp-server/rtsp-media-mapping.c:
5419 * gst/rtsp-server/rtsp-media.h:
5420 * gst/rtsp-server/rtsp-params.c:
5421 * gst/rtsp-server/rtsp-server.c:
5422 * gst/rtsp-server/rtsp-session-pool.h:
5423 * gst/rtsp-server/rtsp-session.c:
5424 * gst/rtsp-server/rtsp-session.h:
5425 * gst/rtsp-server/rtsp-stream-transport.h:
5426 * gst/rtsp-server/rtsp-stream.h:
5429 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5431 * gst/rtsp-server/rtsp-client.c:
5432 * gst/rtsp-server/rtsp-media-mapping.h:
5433 * gst/rtsp-server/rtsp-media.c:
5434 * gst/rtsp-server/rtsp-media.h:
5435 * gst/rtsp-server/rtsp-server.h:
5436 * gst/rtsp-server/rtsp-stream.c:
5437 * gst/rtsp-server/rtsp-stream.h:
5438 rtsp: fix MTU setting
5439 Fix setting of the MTU. There is no need for a vmethod.
5441 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5446 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5449 configure: bump version number after refactoring
5451 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5453 * gst/rtsp-server/Makefile.am:
5454 * gst/rtsp-server/rtsp-client.c:
5455 * gst/rtsp-server/rtsp-client.h:
5456 * gst/rtsp-server/rtsp-media-factory-uri.c:
5457 * gst/rtsp-server/rtsp-media-factory.c:
5458 * gst/rtsp-server/rtsp-media-factory.h:
5459 * gst/rtsp-server/rtsp-media.c:
5460 * gst/rtsp-server/rtsp-media.h:
5461 * gst/rtsp-server/rtsp-sdp.c:
5462 * gst/rtsp-server/rtsp-session-media.c:
5463 * gst/rtsp-server/rtsp-session-media.h:
5464 * gst/rtsp-server/rtsp-session.c:
5465 * gst/rtsp-server/rtsp-session.h:
5466 * gst/rtsp-server/rtsp-stream-transport.c:
5467 * gst/rtsp-server/rtsp-stream-transport.h:
5468 * gst/rtsp-server/rtsp-stream.c:
5469 * gst/rtsp-server/rtsp-stream.h:
5470 rtsp: massive refactoring
5471 Make GObjects from the remaining simple structures.
5472 Remove GstRTSPSessionStream, it's not needed.
5473 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
5474 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
5475 a GstRTSPStream should be transported to a client.
5476 Rename GstRTSPMediaFactory::get_element -> create_element because that
5477 more accurately describes what it does.
5478 Make nice methods instead of poking in the structures.
5479 Move some methods inside the relevant object source code.
5480 Use GPtrArray to store objects instead of plain arrays, it is more
5481 natural and allows us to more easily clean up.
5482 Move the allocation of udp ports to the Stream object. The Stream object
5483 contains the elements needed to stream the media to a client.
5484 Improve the prepare and unprepare methods. Unprepare should now undo
5485 everything prepare did. Improve also async unprepare when doing EOS on
5486 shutdown. Make sure we always unprepare correctly.
5488 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
5490 * gst/rtsp-server/rtsp-client.c:
5491 rtsp-client: Unref server address clients connected to
5492 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
5494 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
5496 * gst/rtsp-server/rtsp-server.c:
5497 rtsp-server: don't ref server socket if it is NULL
5498 Fixes test_bind_already_in_use unit test again after commit 6a497440.
5499 https://bugzilla.gnome.org/show_bug.cgi?id=686644
5501 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
5503 * tests/check/Makefile.am:
5504 tests: Add libgio link dependency
5505 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
5507 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5509 * gst/rtsp-server/rtsp-media-mapping.c:
5510 * gst/rtsp-server/rtsp-media-mapping.h:
5511 rtsp-media-mapping: rename find_media vfunc to find_factory
5512 The virtual method and class method should have the same name
5513 so it is correctly represented in GIR file
5514 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5516 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5518 * gst/rtsp-server/rtsp-auth.c:
5519 * gst/rtsp-server/rtsp-client.c:
5520 * gst/rtsp-server/rtsp-media-factory-uri.c:
5521 * gst/rtsp-server/rtsp-media-factory.c:
5522 * gst/rtsp-server/rtsp-media-mapping.c:
5523 * gst/rtsp-server/rtsp-media.c:
5524 * gst/rtsp-server/rtsp-server.c:
5525 * gst/rtsp-server/rtsp-session-pool.c:
5526 * gst/rtsp-server/rtsp-session.c:
5527 rtsp-server: fixed comments and GIR annotations
5528 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5530 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
5532 * gst/rtsp-server/rtsp-media-mapping.c:
5533 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
5535 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
5537 * gst/rtsp-server/rtsp-server.c:
5538 rtsp-server: allow binding on port 0 (binds on a random port)
5540 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
5542 * gst/rtsp-server/rtsp-server.c:
5543 * gst/rtsp-server/rtsp-server.h:
5544 rtsp-server: add bound-port property
5545 bound-port can be used to retrieve the port number when the server is bound on
5546 port 0, which binds on a random port.
5548 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
5550 * gst/rtsp-server/rtsp-media-factory.c:
5551 * gst/rtsp-server/rtsp-media-factory.h:
5552 rtsp-media-factory: make ::get_element overridable by GI bindings
5553 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
5554 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
5555 as the invoker for ::get_element(), making it overridable by GI generated
5558 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5560 * gst/rtsp-server/rtsp-media-factory-uri.c:
5561 rtsp-media-factory-uri: don't autoplug parsers in a loop
5562 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
5565 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5567 * gst/rtsp-server/Makefile.am:
5568 Explicitly link against gio. Fix link error on mac.
5570 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5572 * gst/rtsp-server/rtsp-session.c:
5573 session: add ttl to the transport header in SETUP
5574 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
5576 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5578 * gst/rtsp-server/rtsp-client.c:
5579 * gst/rtsp-server/rtsp-client.h:
5580 * gst/rtsp-server/rtsp-media.c:
5581 client: Use client transport settings for multicast if allowed.
5582 This patch makes it possible for the client to send transport settings for
5583 multicast (destination && ttl). Client settings must be explicitly allowed or
5584 the server will use its own settings.
5585 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
5587 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
5590 Automatic update of common submodule
5591 From 6c0b52c to 6bb6951
5593 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
5595 * gst/rtsp-server/rtsp-client.c:
5596 rtsp-client: do not destroy the rtsp watch
5597 Don't destroy the client watch while dispatching. The rtsp watch is
5598 automatically destroyed after the rtsp watch function closed() has
5600 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
5602 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5605 Automatic update of common submodule
5606 From 4f962f7 to 6c0b52c
5608 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
5610 * gst/rtsp-server/rtsp-media.c:
5611 media: fix check for seekability
5613 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5615 * gst/rtsp-server/rtsp-client.c:
5616 client: use more GIO
5617 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
5619 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5621 * gst/rtsp-server/rtsp-server.c:
5622 server: remove obsolete includes
5624 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5626 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
5627 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
5628 be available in "on_new_ssrc". The transports are added in
5629 gst_rtsp_media_set_state when going to PLAYING state. However,
5630 "on_new_ssrc" might be called before this happens.
5631 https://bugzilla.gnome.org/show_bug.cgi?id=683304
5633 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5635 * gst/rtsp-server/rtsp-client.c:
5636 * gst/rtsp-server/rtsp-client.h:
5637 rtsp-client: add signals for rtsp requests (fixes #683287)
5639 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5641 * gst/rtsp-server/rtsp-client.c:
5642 * gst/rtsp-server/rtsp-client.h:
5643 add new-session signal to rtsp-client (fixes #683058)
5645 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
5648 Automatic update of common submodule
5649 From 668acee to 4f962f7
5651 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
5653 * gst/rtsp-server/rtsp-server.c:
5654 * tests/check/gst/rtspserver.c:
5655 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
5656 Do not assume that *error is set in g_socket_address_enumerator_next.
5657 Added test_bind_already_in_use unit-test.
5658 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
5660 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
5663 Automatic update of common submodule
5664 From 94ccf4c to 668acee
5666 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
5668 * gst/rtsp-server/rtsp-client.c:
5669 * gst/rtsp-server/rtsp-client.h:
5670 rtsp-client: make create_sdp virtual method
5671 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
5673 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5676 Automatic update of common submodule
5677 From 98e386f to 94ccf4c
5679 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5681 * gst/rtsp-server/rtsp-client.c:
5684 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5686 * gst/rtsp-server/rtsp-client.c:
5687 * gst/rtsp-server/rtsp-client.h:
5688 * gst/rtsp-server/rtsp-server.c:
5689 * gst/rtsp-server/rtsp-server.h:
5690 rtsp-server: use an existing socket to establish HTTP tunnel
5691 Make it possible to transfer a socket from an HTTP server to be used as
5692 an RTSP over HTTP tunnel.
5694 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
5696 * gst/rtsp-server/rtsp-client.c:
5697 * gst/rtsp-server/rtsp-media.c:
5698 * gst/rtsp-server/rtsp-media.h:
5699 rtsp: Handle the blocksize parameter
5700 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
5702 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
5704 * tests/check/Makefile.am:
5705 * tests/check/gst/rtspserver.c:
5706 Have unit test get header from source dir, not installed dir
5707 This makes compilation of unit tests work in a build directory other
5708 than the source directory.
5709 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
5711 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
5713 * gst/rtsp-server/rtsp-media.c:
5714 rtsp-media: update for gst_element_make_from_uri() changes
5716 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
5719 * tests/Makefile.am:
5720 * tests/check/Makefile.am:
5721 * tests/check/gst/rtspserver.c:
5723 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
5725 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
5727 * gst/rtsp-server/rtsp-media.c:
5728 rtsp-media: don't collect media stats when going to NULL
5729 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
5731 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5733 * gst/rtsp-server/rtsp-client.c:
5734 client: don't leak transports
5736 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
5738 * gst/rtsp-server/rtsp-client.c:
5739 rtsp-client: free transport on no_stream in SETUP handler
5741 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
5743 * gst/rtsp-server/rtsp-client.c:
5744 rtsp-client: changed session media iteration
5745 In client_unlink_session: now don't iterate in session->medias
5746 list where items are removed by gst_rtsp_session_release_media.
5747 Instead, repeatedly remove the first item.
5749 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
5751 * gst/rtsp-server/rtsp-client.c:
5752 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
5753 GstRTSPSessionMedia is not a GObject type. When the
5754 GstRTSPSession is freed, it will free the media.
5756 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
5758 * gst/rtsp-server/rtsp-media-factory.c:
5759 factory: plug pad leak in collect_streams
5760 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
5761 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
5762 will take one reference, and the other reference will otherwise
5765 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5768 configure: suppress some warnings when debug is disabled
5769 Warnings about unused variables should be suppressed if core has the
5770 debug system disabled.
5771 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5773 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5775 * docs/libs/Makefile.am:
5776 docs: fix build in uninstalled setup
5777 Include gst-plugins-base libs properly.
5779 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
5781 * docs/libs/gst-rtsp-server.types:
5782 docs: include headers defining rtsp-server object types
5783 Fixes compiler warnings during docs build.
5784 https://bugzilla.gnome.org/show_bug.cgi?id=676824
5786 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
5789 configure: Add warning flags for compiler when configuring
5790 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5792 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5795 Automatic update of common submodule
5796 From 03a0e57 to 98e386f
5798 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5801 Automatic update of common submodule
5802 From 1fab359 to 03a0e57
5804 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
5806 * gst/rtsp-server/rtsp-client.c:
5807 client: fix GSocketAddress leak in gst_rtsp_client_accept
5808 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
5810 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5813 Automatic update of common submodule
5814 From f1b5a96 to 1fab359
5816 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5819 Automatic update of common submodule
5820 From 92b7266 to f1b5a96
5822 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5825 Automatic update of common submodule
5826 From ec1c4a8 to 92b7266
5828 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5831 Automatic update of common submodule
5832 From 3429ba6 to ec1c4a8
5834 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
5836 * gst/rtsp-server/rtsp-auth.c:
5837 * gst/rtsp-server/rtsp-client.c:
5838 * gst/rtsp-server/rtsp-media-factory-uri.c:
5839 * gst/rtsp-server/rtsp-server.c:
5840 rtsp: fix compiler warnings
5841 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
5843 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5846 Automatic update of common submodule
5847 From dc70203 to 3429ba6
5849 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5851 * gst/rtsp-server/rtsp-client.c:
5852 * gst/rtsp-server/rtsp-media-factory.c:
5853 * gst/rtsp-server/rtsp-media-factory.h:
5854 * gst/rtsp-server/rtsp-media.c:
5855 * gst/rtsp-server/rtsp-media.h:
5856 * gst/rtsp-server/rtsp-server.c:
5857 * gst/rtsp-server/rtsp-server.h:
5858 * gst/rtsp-server/rtsp-session-pool.c:
5859 * gst/rtsp-server/rtsp-session-pool.h:
5860 rtsp-server: port to new thread API
5862 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5865 Automatic update of common submodule
5866 From 6db25be to dc70203
5868 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5870 * gst/rtsp-server/rtsp-auth.c:
5871 * gst/rtsp-server/rtsp-auth.h:
5872 * gst/rtsp-server/rtsp-client.c:
5873 rtsp-server: Fix compilation and compiler warnings
5875 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5879 * gst/rtsp-server/Makefile.am:
5880 configure: Modernize autotools setup a bit
5881 Also we now only create tar.bz2 and tar.xz tarballs.
5883 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5886 Automatic update of common submodule
5887 From 464fe15 to 6db25be
5889 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5892 Automatic update of common submodule
5893 From 7fda524 to 464fe15
5895 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5898 * docs/libs/Makefile.am:
5899 * docs/version.entities.in:
5901 * gst/rtsp-server/Makefile.am:
5902 * pkgconfig/Makefile.am:
5903 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5904 * pkgconfig/gstreamer-rtsp-server.pc.in:
5905 * tests/Makefile.am:
5906 rtsp-server: Update versioning
5908 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5910 Merge remote-tracking branch 'origin/0.10'
5912 gst/rtsp-server/rtsp-session-pool.c
5914 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5916 * gst/rtsp-server/rtsp-session-pool.c:
5917 rtsp-server: Don't use deprecated GLib API
5919 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5921 Replace master with 0.11
5923 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5925 Merge branch 'master' into 0.11
5927 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5929 Merge branch 'master' into 0.11
5931 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
5934 A couple minor typo fixes
5936 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5938 * gst/rtsp-server/rtsp-media.c:
5939 media: fix state of the appqueue
5941 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5943 * gst/rtsp-server/rtsp-media-factory-uri.c:
5944 factory: use videoconvert
5946 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5948 * gst/rtsp-server/rtsp-media-factory-uri.c:
5949 factory: change to new style caps
5951 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5953 * gst/rtsp-server/rtsp-client.c:
5954 * gst/rtsp-server/rtsp-client.h:
5955 * gst/rtsp-server/rtsp-media-factory-uri.c:
5956 * gst/rtsp-server/rtsp-media.c:
5957 * gst/rtsp-server/rtsp-server.c:
5958 * gst/rtsp-server/rtsp-server.h:
5959 * gst/rtsp-server/rtsp-session-pool.c:
5960 rtsp-server: port to GIO
5963 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5966 configure: fix build
5968 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5971 docs: fix for gst_rtsp_server_set_port() -> _set_service()
5972 https://bugzilla.gnome.org/show_bug.cgi?id=666548
5974 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5977 * examples/Makefile.am:
5978 First rule of gst-rtsp-server club: don't talk about gst-phonon
5980 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5983 * pkgconfig/Makefile.am:
5984 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
5985 * pkgconfig/gst-rtsp-server.pc.in:
5986 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5987 * pkgconfig/gstreamer-rtsp-server.pc.in:
5988 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
5989 For consistency with all other modules.
5991 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5993 * gst/rtsp-server/rtsp-client.c:
5994 rtsp-client: update for new map API
5996 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5999 * bindings/Makefile.am:
6000 * bindings/python/Makefile.am:
6001 * bindings/python/arg-types.py:
6002 * bindings/python/codegen/Makefile.am:
6003 * bindings/python/codegen/__init__.py:
6004 * bindings/python/codegen/argtypes.py:
6005 * bindings/python/codegen/code-coverage.py:
6006 * bindings/python/codegen/codegen.py:
6007 * bindings/python/codegen/definitions.py:
6008 * bindings/python/codegen/defsparser.py:
6009 * bindings/python/codegen/docextract.py:
6010 * bindings/python/codegen/docgen.py:
6011 * bindings/python/codegen/fileprefix.override:
6012 * bindings/python/codegen/fileprefixmodule.c:
6013 * bindings/python/codegen/h2def.py:
6014 * bindings/python/codegen/mergedefs.py:
6015 * bindings/python/codegen/mkskel.py:
6016 * bindings/python/codegen/override.py:
6017 * bindings/python/codegen/reversewrapper.py:
6018 * bindings/python/codegen/scmexpr.py:
6019 * bindings/python/rtspserver-types.defs:
6020 * bindings/python/rtspserver.defs:
6021 * bindings/python/rtspserver.override:
6022 * bindings/python/rtspservermodule.c:
6023 * bindings/python/test.py:
6025 python: remove pygst-based python bindings
6026 pygi is the future, apparently.
6028 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
6031 Automatic update of common submodule
6032 From c463bc0 to 7fda524
6034 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6037 Automatic update of common submodule
6038 From 2a59016 to c463bc0
6040 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6043 Automatic update of common submodule
6044 From 0807187 to 2a59016
6046 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6049 Automatic update of common submodule
6050 From 11f0cd5 to 0807187
6052 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6054 * examples/test-auth.c:
6055 example: update for new caps
6057 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6059 * examples/test-video.c:
6060 * gst/rtsp-server/rtsp-client.c:
6061 * gst/rtsp-server/rtsp-media-factory-uri.c:
6062 * gst/rtsp-server/rtsp-media.c:
6063 * gst/rtsp-server/rtsp-media.h:
6064 * gst/rtsp-server/rtsp-session.c:
6065 * gst/rtsp-server/rtsp-session.h:
6066 rtsp-server: port some more to 0.11
6068 Remove bufferlist stuff
6070 Add queue before appsink now that preroll-queue-len is gone.
6071 Update for request pad changes.
6073 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6075 Merge branch 'master' into 0.11
6077 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6079 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6080 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6081 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6083 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6085 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6086 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6087 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6089 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6091 Merge branch 'master' into 0.11
6093 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6095 * gst/rtsp-server/rtsp-media.c:
6096 * gst/rtsp-server/rtsp-media.h:
6097 media: add a seekable boolean
6098 Maintain the seekable state with a new variable instead of reusing the
6101 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
6103 * gst/rtsp-server/rtsp-media.c:
6104 Disallow seek in live media
6106 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6108 Merge branch 'master' into 0.11
6110 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
6112 * gst/rtsp-server/rtsp-server.c:
6113 #ifdef statements for windows socket creation were missing
6115 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
6118 Automatic update of common submodule
6119 From a39eb83 to 11f0cd5
6121 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
6124 Automatic update of common submodule
6125 From 605cd9a to a39eb83
6127 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6129 Merge branch 'master' into 0.11
6131 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6133 * gst/rtsp-server/rtsp-client.c:
6134 client: use method to access property
6136 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6138 * gst/rtsp-server/rtsp-media-factory.c:
6139 * gst/rtsp-server/rtsp-media-factory.h:
6140 media-factory: add protocols property
6141 Add a property to configure the allowed protocols in the media created from the
6144 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6146 * gst/rtsp-server/rtsp-media-factory.c:
6147 * gst/rtsp-server/rtsp-media-factory.h:
6148 media-factory: add media-configure signal
6149 Add signal to allow the application to configure the media after it was created
6152 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6154 * gst/rtsp-server/rtsp-client.c:
6155 client: use method to access property
6157 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6159 * gst/rtsp-server/rtsp-media-factory.c:
6160 * gst/rtsp-server/rtsp-media-factory.h:
6161 media-factory: add protocols property
6162 Add a property to configure the allowed protocols in the media created from the
6165 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6167 * gst/rtsp-server/rtsp-media-factory.c:
6168 * gst/rtsp-server/rtsp-media-factory.h:
6169 media-factory: add media-configure signal
6170 Add signal to allow the application to configure the media after it was created
6173 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6175 Merge branch 'master' into 0.11
6177 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6179 * gst/rtsp-server/rtsp-client.c:
6180 client: use media multicast group
6182 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6184 * gst/rtsp-server/rtsp-media-factory.h:
6185 * gst/rtsp-server/rtsp-server.h:
6186 * gst/rtsp-server/rtsp-session-pool.h:
6187 * gst/rtsp-server/rtsp-session.h:
6190 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6192 * gst/rtsp-server/rtsp-client.c:
6193 * gst/rtsp-server/rtsp-sdp.h:
6194 sdp: copy and free the server ip address
6195 Copy and free the server ip address to make memory management easier later.
6197 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6199 * gst/rtsp-server/rtsp-media-factory.c:
6200 media-factory: configure multicast in media
6202 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6204 * gst/rtsp-server/rtsp-media.c:
6205 * gst/rtsp-server/rtsp-media.h:
6206 media: add property for multicast group
6207 Add a property to configure the multicast group in the media.
6208 Based on patches from Marc Leeman and Robert Krakora.
6210 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6212 * gst/rtsp-server/rtsp-media-factory.c:
6213 * gst/rtsp-server/rtsp-media-factory.h:
6214 media-factory: add property for multicast group
6215 Add a property to configure the multicast group in the media factory.
6216 Based on patches from Marc Leeman and Robert Krakora.
6218 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6220 * gst/rtsp-server/rtsp-client.c:
6221 client: do configuration of transport in one place
6222 Move the configuration of the transport destination address to where we also
6223 configure the other bits.
6225 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6227 * gst/rtsp-server/rtsp-client.c:
6228 client: use media multicast group
6230 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6232 * gst/rtsp-server/rtsp-media-factory.h:
6233 * gst/rtsp-server/rtsp-server.h:
6234 * gst/rtsp-server/rtsp-session-pool.h:
6235 * gst/rtsp-server/rtsp-session.h:
6238 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6240 * gst/rtsp-server/rtsp-client.c:
6241 * gst/rtsp-server/rtsp-sdp.h:
6242 sdp: copy and free the server ip address
6243 Copy and free the server ip address to make memory management easier later.
6245 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6247 * gst/rtsp-server/rtsp-media-factory.c:
6248 media-factory: configure multicast in media
6250 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6252 * gst/rtsp-server/rtsp-media.c:
6253 * gst/rtsp-server/rtsp-media.h:
6254 media: add property for multicast group
6255 Add a property to configure the multicast group in the media.
6256 Based on patches from Marc Leeman and Robert Krakora.
6258 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6260 * gst/rtsp-server/rtsp-media-factory.c:
6261 * gst/rtsp-server/rtsp-media-factory.h:
6262 media-factory: add property for multicast group
6263 Add a property to configure the multicast group in the media factory.
6264 Based on patches from Marc Leeman and Robert Krakora.
6266 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6268 * gst/rtsp-server/rtsp-client.c:
6269 client: do configuration of transport in one place
6270 Move the configuration of the transport destination address to where we also
6271 configure the other bits.
6273 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6275 Merge branch 'master' into 0.11
6277 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6279 * gst/rtsp-server/rtsp-client.c:
6280 client: destroy pipeline on client disconnect with no prior TEARDOWN.
6281 The problem occurs when the client abruptly closes the connection without
6282 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
6283 server is where the pipeline gets torn down. Since this handler is not called,
6284 the pipeline remains and is up and running. Subsequent clients get their own
6285 pipelines and if the do not issue TEARDOWNs then those pipelines will also
6286 remain up and running. This is a resource leak.
6288 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6290 Merge branch 'master' into 0.11
6292 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
6294 * gst/rtsp-server/rtsp-media-factory.c:
6295 * gst/rtsp-server/rtsp-media-factory.h:
6296 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
6297 For example, it can be used to retrieve source elements like appsrc, in a more
6298 convenient way than subclassing get_element.
6300 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6302 Merge branch 'master' into 0.11
6304 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
6306 * gst/rtsp-server/rtsp-server.c:
6307 rtsp-server: hold on to reference while using object
6309 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6311 * gst/rtsp-server/rtsp-media.c:
6314 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6317 configure: use unstable api
6319 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
6321 * gst/rtsp-server/rtsp-client.c:
6322 client: fix reference counting
6324 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
6326 * gst/rtsp-server/rtsp-client.c:
6327 * gst/rtsp-server/rtsp-media.c:
6328 fix compiler warnings about unused variables
6330 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
6332 * examples/test-launch.c:
6333 * examples/test-readme.c:
6334 * examples/test-uri.c:
6335 * examples/test-video.c:
6336 examples: tell rtsp uri when ready
6338 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
6341 Automatic update of common submodule
6342 From 69b981f to 605cd9a
6344 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6346 * gst/rtsp-server/rtsp-client.c:
6347 client: update for buffer API change
6349 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6351 * gst/rtsp-server/Makefile.am:
6352 Makefile.am: 0.10 => @GST_MAJORMINOR@
6354 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6356 * gst/rtsp-server/rtsp-media-factory-uri.c:
6357 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
6359 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6361 * gst/rtsp-server/.gitignore:
6362 .gitignore: 0.10 => 0.11
6364 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6366 * gst/rtsp-server/Makefile.am:
6367 Makefile.am: 0.10 => @GST_MAJORMINOR@
6369 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6371 Merge branch 'master' into 0.11
6373 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
6376 Automatic update of common submodule
6377 From 9e5bbd5 to 69b981f
6379 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
6382 Automatic update of common submodule
6383 From fd35073 to 9e5bbd5
6385 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
6388 Automatic update of common submodule
6389 From 46dfcea to fd35073
6391 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6393 * gst/rtsp-server/rtsp-media-factory-uri.c:
6394 * gst/rtsp-server/rtsp-media.c:
6395 media: port to new caps API
6397 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6399 Merge branch 'master' into 0.11
6401 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6403 * bindings/vala/gst-rtsp-server-0.10.vapi:
6404 Updated Vala bindings.
6405 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6407 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6409 * gst/rtsp-server/rtsp-server.c:
6410 * gst/rtsp-server/rtsp-server.h:
6411 Add a signal for newly connected clients.
6412 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6414 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6416 * bindings/python/rtspserver.override:
6417 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
6419 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6421 * gst/rtsp-server/Makefile.am:
6422 * gst/rtsp-server/rtsp-client.c:
6423 * gst/rtsp-server/rtsp-funnel.c:
6424 * gst/rtsp-server/rtsp-funnel.h:
6425 * gst/rtsp-server/rtsp-media.c:
6426 rtsp-server: port to 0.11
6428 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6433 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6435 Merge branch 'master' into 0.11
6440 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6443 Automatic update of common submodule
6444 From c3cafe1 to 46dfcea
6446 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
6448 * bindings/python/Makefile.am:
6449 * bindings/python/rtspserver.defs:
6450 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
6452 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
6454 * bindings/python/arg-types.py:
6455 python bindings: add GstRTSPUrlParam
6456 Needed to implement MediaFactory virtual proxies
6458 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
6460 * bindings/python/arg-types.py:
6461 python bindings: fix returning GstRTSPUrl types
6463 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6465 * bindings/python/arg-types.py:
6466 python bindings: add arg type for GstRTSPUrl
6468 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
6470 * bindings/python/rtspserver.defs:
6471 python bindings: fix the definition of MediaFactory.collect_stream
6473 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
6476 Automatic update of common submodule
6477 From 1ccbe09 to c3cafe1
6479 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6482 Automatic update of common submodule
6483 From 193b717 to 1ccbe09
6485 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
6488 Automatic update of common submodule
6489 From b77e2bf to 193b717
6491 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6494 build: Include lcov.mak to allow test coverage report generation
6496 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6499 Automatic update of common submodule
6500 From d8814b6 to b77e2bf
6502 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6505 Automatic update of common submodule
6506 From 6aaa286 to d8814b6
6508 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
6511 Automatic update of common submodule
6512 From 6aec6b9 to 6aaa286
6514 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
6517 autogen: wingo signed comment
6519 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
6521 * gst/rtsp-server/rtsp-session-pool.c:
6522 session: use full charset for RTSP session ID
6523 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
6524 session ID more difficult.
6525 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6527 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6529 * gst/rtsp-server/Makefile.am:
6530 rtsp-server: Don't install the funnel header
6532 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6535 Automatic update of common submodule
6536 From 1de7f6a to 6aec6b9
6538 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6541 configure: require core/base 0.10.31
6542 Needed at least for gst_plugin_feature_rank_compare_func().
6544 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
6547 Automatic update of common submodule
6548 From f94d739 to 1de7f6a
6550 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6552 * gst/rtsp-server/rtsp-media.c:
6553 media: remove more unused code
6555 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6557 * gst/rtsp-server/rtsp-media.c:
6558 * gst/rtsp-server/rtsp-media.h:
6559 media: remove duplicate filtering
6560 Remove the duplicate filtering code now that we have a released -good version.
6561 Give a warning instead.
6563 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6565 * gst/rtsp-server/rtsp-media-factory.c:
6566 * gst/rtsp-server/rtsp-media.c:
6567 media: fix default buffer size
6569 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6571 * gst/rtsp-server/rtsp-media-factory.c:
6572 * gst/rtsp-server/rtsp-media-factory.h:
6573 media-factory: add property to configure the buffer-size
6574 Add a property to configure the kernel UDP buffer size.
6576 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6578 * gst/rtsp-server/rtsp-media.c:
6579 * gst/rtsp-server/rtsp-media.h:
6580 media: add property to configure kernel buffer sizes
6581 Add a property to configure the kernel UDP buffer size.
6583 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6586 configure: set PYGOBJECT_REQ before using it
6587 https://bugzilla.gnome.org/show_bug.cgi?id=640641
6589 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6592 docs: recursive into sub-directories on 'make upload'
6594 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6596 * docs/libs/gst-rtsp-server-docs.sgml:
6597 * docs/version.entities.in:
6598 docs: mention full version these docs are for, not just major-minor
6600 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6605 === release 0.10.8 ===
6607 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6612 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6614 * gst/rtsp-server/rtsp-server.c:
6615 rtsp-server: clarify docs a little
6617 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6619 * gst/rtsp-server/rtsp-media.c:
6620 media: init debug category before starting thread
6622 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6624 * gst/rtsp-server/rtsp-auth.c:
6625 auth: add realm to make it more spec compliant
6627 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6629 * gst/rtsp-server/rtsp-server.c:
6630 * gst/rtsp-server/rtsp-server.h:
6633 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6635 * examples/test-video.c:
6636 example: improve example docs a little
6638 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6640 * gst/rtsp-server/rtsp-server.c:
6641 server: ensure the watch has a ref to the server
6643 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6645 * gst/rtsp-server/rtsp-server.c:
6646 server: simpify channel function
6648 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6650 * gst/rtsp-server/rtsp-server.c:
6651 * gst/rtsp-server/rtsp-server.h:
6652 server: simplify management of channel and source
6653 We don't need to keep around the channel and source objects. Let the mainloop
6654 and the source manage the source and channel respectively.
6656 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6662 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6665 * tests/Makefile.am:
6666 * tests/test-cleanup.c:
6667 tests: add tests directory and cleanup test
6669 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6671 * gst/rtsp-server/rtsp-media-factory-uri.c:
6672 * gst/rtsp-server/rtsp-media-factory.c:
6673 * gst/rtsp-server/rtsp-media-mapping.c:
6674 * gst/rtsp-server/rtsp-media.c:
6675 * gst/rtsp-server/rtsp-session-pool.c:
6676 * gst/rtsp-server/rtsp-session.c:
6677 server: improve debugging in various objects
6679 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6681 * gst/rtsp-server/rtsp-server.c:
6682 server: chain up to the parent finalize
6684 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
6686 * bindings/python/rtspserver-types.defs:
6687 * bindings/python/rtspserver.defs:
6688 * bindings/python/rtspserver.override:
6689 * bindings/python/test.py:
6690 gst-rtsp-server: update python bindings
6692 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6694 * gst/rtsp-server/rtsp-client.c:
6695 client: use the response from the clientstate
6696 Create the response object only once and store in the client state.
6697 Make all methods use the state response,
6699 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6701 * gst/rtsp-server/rtsp-server.c:
6702 server: use signal to keep track of clients
6703 Keep track of all the clients that the server creates and remove them when they
6704 fire the 'closed' signal.
6706 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6708 * gst/rtsp-server/rtsp-client.c:
6709 * gst/rtsp-server/rtsp-client.h:
6710 client: emit signal when closing
6712 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6714 * examples/.gitignore:
6715 * examples/Makefile.am:
6716 * examples/test-auth.c:
6717 * examples/test-video.c:
6718 * gst/rtsp-server/rtsp-auth.c:
6719 * gst/rtsp-server/rtsp-auth.h:
6720 * gst/rtsp-server/rtsp-client.c:
6721 * gst/rtsp-server/rtsp-media-factory.c:
6722 * gst/rtsp-server/rtsp-media.c:
6723 * gst/rtsp-server/rtsp-media.h:
6724 * gst/rtsp-server/rtsp-session-pool.h:
6725 * gst/rtsp-server/rtsp-session.h:
6726 media: enable per factory authorisations
6727 Allow for adding a GstRTSPAuth on the factory and media level and check
6728 permissions when accessing the factory.
6729 Add hints to the auth methods for future more fine grained authorisation.
6730 Add example application for per factory authentication.
6732 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6734 * gst/rtsp-server/rtsp-auth.c:
6735 * gst/rtsp-server/rtsp-auth.h:
6736 * gst/rtsp-server/rtsp-client.c:
6737 * gst/rtsp-server/rtsp-client.h:
6738 * gst/rtsp-server/rtsp-params.c:
6739 * gst/rtsp-server/rtsp-params.h:
6740 rtsp-server: Pass ClientState structure arround
6741 Pass the collected information for the ongoing request in a GstRTSPClientState
6742 structure that we can then pass around to simplify the method arguments. This
6743 will also be handy when we implement logging functionality.
6745 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6747 * gst/rtsp-server/rtsp-media-factory.c:
6748 * gst/rtsp-server/rtsp-media-factory.h:
6749 media-factory: add methods to configure authorisation
6751 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6753 * gst/rtsp-server/rtsp-client.c:
6754 client: unref auth in finalize
6756 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6758 * gst/rtsp-server/rtsp-server.c:
6759 server: unref auth in finalize
6761 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6763 * docs/libs/gst-rtsp-server-docs.sgml:
6764 * docs/libs/gst-rtsp-server-sections.txt:
6765 * docs/libs/gst-rtsp-server.types:
6768 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6770 * gst/rtsp-server/rtsp-server.c:
6771 * gst/rtsp-server/rtsp-server.h:
6772 server: separate create and accept
6773 Create separate create and accept methods so that subclasses can create custom
6775 Configure the server in the client object and prepare for keeping track of
6778 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6780 * gst/rtsp-server/rtsp-client.c:
6781 * gst/rtsp-server/rtsp-client.h:
6782 client: add support for setting the server.
6783 Add support for keeping a ref to the server that started this client
6786 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6788 * gst/rtsp-server/rtsp-auth.c:
6789 auth: fix memleak and add some docs
6790 Fix a memleak of the basic auth token.
6791 Add docs for the helper function
6793 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6795 * gst/rtsp-server/rtsp-auth.c:
6796 * gst/rtsp-server/rtsp-auth.h:
6797 * gst/rtsp-server/rtsp-client.c:
6798 client: delegate setup of auth to the manager
6799 Delegate the configuration of the authentication tokens to the manager object
6802 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6804 * examples/test-video.c:
6805 * gst/rtsp-server/Makefile.am:
6806 * gst/rtsp-server/rtsp-auth.c:
6807 * gst/rtsp-server/rtsp-auth.h:
6808 * gst/rtsp-server/rtsp-client.c:
6809 * gst/rtsp-server/rtsp-client.h:
6810 * gst/rtsp-server/rtsp-server.c:
6811 * gst/rtsp-server/rtsp-server.h:
6812 auth: add authentication object
6813 Add an object that can check the authorization of requests.
6814 Implement basic authentication.
6815 Add example authentication to test-video
6817 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6819 * gst/rtsp-server/rtsp-server.c:
6820 * gst/rtsp-server/rtsp-server.h:
6821 server: move includes back
6822 the includes are needed for sockaddr_in.
6824 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6826 * gst/rtsp-server/rtsp-client.c:
6827 * gst/rtsp-server/rtsp-client.h:
6828 * gst/rtsp-server/rtsp-server.c:
6829 * gst/rtsp-server/rtsp-server.h:
6830 rtsp: move network includes where they are needed
6832 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
6834 * gst/rtsp-server/rtsp-media.h:
6835 rtsp-media.h: Minor corrections in comments.
6838 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
6841 Automatic update of common submodule
6842 From e572c87 to f94d739
6844 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6848 * docs/libs/.gitignore:
6849 * examples/.gitignore:
6850 * gst/rtsp-server/.gitignore:
6853 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6855 * docs/libs/Makefile.am:
6856 docs: We don't build ps/pdf for API reference docs
6858 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6861 Automatic update of common submodule
6862 From ccbaa85 to e572c87
6864 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6867 Automatic update of common submodule
6868 From 46445ad to ccbaa85
6870 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6872 * gst/rtsp-server/Makefile.am:
6873 * gst/rtsp-server/fs-funnel.c:
6874 * gst/rtsp-server/fs-funnel.h:
6875 * gst/rtsp-server/rtsp-funnel.c:
6876 * gst/rtsp-server/rtsp-funnel.h:
6877 * gst/rtsp-server/rtsp-media.c:
6878 funnel: rename fsfunnel to rtspfunnel
6879 Rename the funnel to avoid conflicts with the farsight one.
6881 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6883 * gst/rtsp-server/Makefile.am:
6884 * gst/rtsp-server/fs-funnel.c:
6885 * gst/rtsp-server/fs-funnel.h:
6886 * gst/rtsp-server/rtsp-media.c:
6887 rtsp-media: add and use fsfunnel
6888 Add a copy of fsfunnel to the build because input-selector removed the (broken)
6889 select-all property that we need.
6891 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6893 * gst/rtsp-server/Makefile.am:
6894 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
6895 Use PKG_CONFIG_PATH specified at configure time (if any) as well
6896 for the g-ir-compiler, rather than just assuming the env var has
6899 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6906 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
6908 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6911 * gst/rtsp-server/Makefile.am:
6912 gobject-introspection: fix g-i build for uninstalled setup
6913 Requires gst-plugins-base git (> 0.10.31.2).
6915 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6917 * examples/test-uri.c:
6918 examples: add some more options and comments
6920 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6922 * gst/rtsp-server/rtsp-media-factory-uri.c:
6923 factory-uri: use right property type
6925 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6927 * gst/rtsp-server/rtsp-media-factory-uri.c:
6928 factory-uri: attempt to configure buffer-lists
6929 Attempt to configure buffer lists in the payloader for improved performance.
6931 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6933 * gst/rtsp-server/rtsp-media.c:
6934 media: attempt to configure bigger UDP buffers
6935 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
6936 send buffers with high bitrate streams.
6938 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
6940 * gst/rtsp-server/rtsp-client.c:
6941 client: use the socket length from getsockname
6942 Use the length returned by getsockname to perform the getnameinfo call because
6943 the size can depend on the socket type and platform.
6946 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6948 * docs/libs/gst-rtsp-server-docs.sgml:
6949 * docs/libs/gst-rtsp-server-sections.txt:
6950 docs: add uri factory to the docs
6952 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6954 * gst/rtsp-server/rtsp-client.c:
6955 * gst/rtsp-server/rtsp-media.h:
6958 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6960 * gst/rtsp-server/rtsp-client.c:
6961 * gst/rtsp-server/rtsp-media.c:
6962 * gst/rtsp-server/rtsp-media.h:
6963 * gst/rtsp-server/rtsp-session.c:
6964 * gst/rtsp-server/rtsp-session.h:
6965 rtsp-server: add support for buffer lists
6966 Add support for sending bufferlists received from appsink.
6969 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6971 * gst/rtsp-server/rtsp-client.c:
6972 * gst/rtsp-server/rtsp-media.c:
6973 * gst/rtsp-server/rtsp-media.h:
6974 * gst/rtsp-server/rtsp-sdp.c:
6975 media: make method to retrieve the play range
6976 Make a method to retrieve the playback range so that we can conditionally create
6977 a different range for the SDP and the PLAY requests.
6979 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6981 * gst/rtsp-server/rtsp-media.c:
6982 * gst/rtsp-server/rtsp-media.h:
6983 media: add signal to notify of state changes
6985 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6987 * gst/rtsp-server/rtsp-client.h:
6988 client: cleanup headers
6990 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6992 * gst/rtsp-server/rtsp-client.c:
6995 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6997 * gst/rtsp-server/rtsp-media-factory-uri.c:
6998 * gst/rtsp-server/rtsp-media-factory-uri.h:
6999 factory-uri: add support for gstpay
7000 Add an option to prefer gstpay over decoder + raw payloader.
7002 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7004 * gst/rtsp-server/rtsp-media-factory-uri.c:
7005 * gst/rtsp-server/rtsp-media-factory-uri.h:
7006 factory-uri: rework the autoplugger.
7007 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
7010 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7012 * gst/rtsp-server/rtsp-media-factory-uri.c:
7013 factory-uri: use better factory filter
7014 Make better payloader filter based on autoplug rank and RTP use case.
7016 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7019 Automatic update of common submodule
7020 From 169462a to 46445ad
7022 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7024 * gst/rtsp-server/rtsp-server.c:
7025 server: set SO_REUSEADDR before bind
7026 Set the SO_REUSEADDR _before_ bind() to make it actually work.
7028 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7030 * gst/rtsp-server/rtsp-media.c:
7031 * gst/rtsp-server/rtsp-media.h:
7032 media: emit prepared signal when prepared
7033 Make a 'prepared' signal and emit it when we successfully prepared the element.
7034 This signal can be used to configure the media object after it has been prepared
7037 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
7040 Automatic update of common submodule
7041 From 011bcc8 to 169462a
7043 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
7045 python an optional dependency
7046 * configure.ac: Move up valgrind and g-i checks. Make the python
7047 dependency optional, as it was before.
7049 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7051 Merge branch 'master' into 0.11
7056 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7058 * gst/rtsp-server/rtsp-media.c:
7059 media: update range when active clients changed
7060 When we changed the number of active clients, update the current range
7061 information because we want the second client connecting to a shared resource
7062 continue from where the stream currently.
7064 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7066 * gst/rtsp-server/rtsp-media-factory-uri.c:
7067 * gst/rtsp-server/rtsp-media-factory-uri.h:
7068 factory-uri: add colorspace and fix pt
7069 Rework the way we pass data to the autoplugger.
7070 When we have raw caps, plug a converter element to make pluggin to raw
7071 payloaders more successful.
7072 Make sure all dynamically plugged payloaders have a unique payload types.
7074 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7076 * examples/Makefile.am:
7077 * examples/test-uri.c:
7078 example: add example of the uri factory
7080 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7082 * gst/rtsp-server/Makefile.am:
7083 * gst/rtsp-server/rtsp-media-factory-uri.c:
7084 * gst/rtsp-server/rtsp-media-factory-uri.h:
7085 * gst/rtsp-server/rtsp-server.h:
7086 factory-uri: add a factory to stream any URI
7087 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
7090 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7092 * gst/rtsp-server/rtsp-media.c:
7093 * gst/rtsp-server/rtsp-media.h:
7094 media: ignore spurious ASYNC_DONE messages
7095 When we are dynamically adding pads, the addition of the udpsrc elements will
7096 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
7097 the real ASYNC_DONE when everything is prerolled.
7099 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7101 * gst/rtsp-server/rtsp-media-factory.c:
7102 * gst/rtsp-server/rtsp-media-factory.h:
7103 media-factory: make lock macro
7105 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
7107 * gst/rtsp-server/rtsp-client.c:
7108 rtsp-server: Remove unused variable and dead assignment
7110 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
7112 * examples/test-launch.c:
7113 * examples/test-mp4.c:
7114 * examples/test-ogg.c:
7115 * examples/test-readme.c:
7116 * examples/test-sdp.c:
7117 * examples/test-video.c:
7118 examples: Run gst-indent
7120 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
7122 * gst/rtsp-server/rtsp-client.c:
7123 * gst/rtsp-server/rtsp-media-factory.c:
7124 * gst/rtsp-server/rtsp-media-mapping.c:
7125 * gst/rtsp-server/rtsp-media.c:
7126 * gst/rtsp-server/rtsp-params.c:
7127 * gst/rtsp-server/rtsp-sdp.c:
7128 * gst/rtsp-server/rtsp-server.c:
7129 * gst/rtsp-server/rtsp-session-pool.c:
7130 * gst/rtsp-server/rtsp-session.c:
7131 rtsp-server: Run gst-indent
7132 Since it wasn't using the upstream common previously, there was no
7133 indentation check before commiting.
7135 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
7137 * gst/rtsp-server/rtsp-media-mapping.h:
7138 * gst/rtsp-server/rtsp-media.c:
7139 * gst/rtsp-server/rtsp-media.h:
7140 * gst/rtsp-server/rtsp-sdp.c:
7141 * gst/rtsp-server/rtsp-session-pool.h:
7142 * gst/rtsp-server/rtsp-session.c:
7143 * gst/rtsp-server/rtsp-session.h:
7144 rtsp-server: Some more doc fixups
7146 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7149 Makefile: Add cruft-cleaning support
7151 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7156 * docs/libs/Makefile.am:
7157 * docs/libs/gst-rtsp-server-docs.sgml:
7158 * docs/libs/gst-rtsp-server-sections.txt:
7159 * docs/libs/gst-rtsp-server.types:
7160 * docs/version.entities.in:
7161 docs: Add gtk-doc build system
7163 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7165 * gst/rtsp-server/Makefile.am:
7166 Makefile.am: Use standard GIR make behaviour
7168 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7172 autogen/configure: Bring more in sync to standard gst module behaviour
7174 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7176 * gst/rtsp-server/rtsp-media.c:
7177 media: warn and fail when gstrtpbin is not found
7179 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7182 configure: open 0.11 branch
7184 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
7188 Add common submodule
7190 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
7193 * common/Makefile.am:
7194 * common/c-to-xml.py:
7196 * common/coverage/coverage-report-entry.pl:
7197 * common/coverage/coverage-report.pl:
7198 * common/coverage/coverage-report.xsl:
7199 * common/coverage/lcov.mak:
7200 * common/gettext.patch:
7201 * common/glib-gen.mak:
7202 * common/gst-autogen.sh:
7203 * common/gst-xmlinspect.py:
7205 * common/gstdoc-scangobj:
7206 * common/gtk-doc-plugins.mak:
7207 * common/gtk-doc.mak:
7208 * common/m4/.gitignore:
7209 * common/m4/Makefile.am:
7211 * common/m4/as-ac-expand.m4:
7212 * common/m4/as-auto-alt.m4:
7213 * common/m4/as-compiler-flag.m4:
7214 * common/m4/as-compiler.m4:
7215 * common/m4/as-docbook.m4:
7216 * common/m4/as-libtool-tags.m4:
7217 * common/m4/as-libtool.m4:
7218 * common/m4/as-python.m4:
7219 * common/m4/as-scrub-include.m4:
7220 * common/m4/as-version.m4:
7221 * common/m4/ax_create_stdint_h.m4:
7222 * common/m4/check.m4:
7223 * common/m4/glib-gettext.m4:
7224 * common/m4/gst-arch.m4:
7225 * common/m4/gst-args.m4:
7226 * common/m4/gst-check.m4:
7227 * common/m4/gst-debuginfo.m4:
7228 * common/m4/gst-default.m4:
7229 * common/m4/gst-doc.m4:
7230 * common/m4/gst-error.m4:
7231 * common/m4/gst-feature.m4:
7232 * common/m4/gst-function.m4:
7233 * common/m4/gst-gettext.m4:
7234 * common/m4/gst-glib2.m4:
7235 * common/m4/gst-libxml2.m4:
7236 * common/m4/gst-plugindir.m4:
7237 * common/m4/gst-valgrind.m4:
7238 * common/m4/gtk-doc.m4:
7239 * common/m4/introspection.m4:
7241 * common/mangle-tmpl.py:
7242 * common/plugins.xsl:
7244 * common/release.mak:
7245 * common/scangobj-merge.py:
7246 * common/upload.mak:
7247 common: Remove static version
7249 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
7251 * common/m4/introspection.m4:
7252 Update introspection.m4 to match usage
7254 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7258 Remove old stuff from the README
7260 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7265 === release 0.10.7 ===
7267 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7272 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7274 * examples/test-ogg.c:
7275 test-ogg: remove parsers
7276 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
7277 buffers with timestamps. Using the parsers also seems to break things.
7279 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7281 * bindings/vala/gst-rtsp-server-0.10.vapi:
7282 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7283 Updated Vala bindings
7285 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7287 * common/m4/introspection.m4:
7289 * gst/rtsp-server/Makefile.am:
7290 Added initial gobject-introspection support
7292 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7294 * gst/rtsp-server/rtsp-media-factory.c:
7295 media-factory: don't use host for shared hash key
7296 When we generate the key to share made between connections, don't include the
7297 host used to connect so that we can share media even if between clients that
7298 connected with localhost and ones with the ip address.
7300 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7302 * bindings/vala/Makefile.am:
7303 build: fix distcheck
7305 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7307 * bindings/vala/gst-rtsp-server-0.10.vapi:
7308 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7309 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7310 Update Vala bindings
7312 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7314 * bindings/vala/Makefile.am:
7316 Fix configure checks and installation location for Vala bindings
7319 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7324 === release 0.10.6 ===
7326 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7329 configure: release 0.10.6
7331 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7333 * gst/rtsp-server/rtsp-media.c:
7334 media: help the compiler a little
7336 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7338 * gst/rtsp-server/rtsp-media.c:
7339 * gst/rtsp-server/rtsp-media.h:
7340 * gst/rtsp-server/rtsp-session.c:
7341 media: cleanup media transport before freeing
7342 Cleanup the media transport data before freeing. In particular, remove the qdata
7343 from the rtpsource object.
7345 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7347 * gst/rtsp-server/rtsp-media-factory.c:
7348 * gst/rtsp-server/rtsp-media-factory.h:
7349 * gst/rtsp-server/rtsp-media.c:
7350 * gst/rtsp-server/rtsp-media.h:
7351 media-factory: add eos-shutdown property
7352 Add an eos-shutdown property that will send an EOS to the pipeline before
7353 shutting it down. This allows for nice cleanup in case of a muxer.
7356 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7358 * gst/rtsp-server/rtsp-media.c:
7359 * gst/rtsp-server/rtsp-media.h:
7360 media: use multiudpsink send-duplicates when we can
7361 If we have a new enough multiudpsink with the send-duplicates property, use this
7362 instead of doing our own filtering. Our custom filtering code should eventually
7363 be removed when we can depend on a released -good.
7365 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7367 * gst/rtsp-server/rtsp-media.c:
7368 media: don't leak destinations
7369 Refactor and cleanup the destinations array when the stream is destroyed.
7371 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7373 * gst/rtsp-server/rtsp-media.c:
7374 * gst/rtsp-server/rtsp-media.h:
7375 media: don't add udp addresses multiple times
7376 Keep track of the udp addresses we added to udpsink and never add the same udp
7377 destination twice. This avoids duplicate packets when using multicast.
7379 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7381 * gst/rtsp-server/rtsp-server.c:
7382 server: disable use of SO_LINGER
7383 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
7384 server close()s the connection.
7386 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7388 * gst/rtsp-server/rtsp-server.c:
7389 server: use 5 second linger period in SO_LINGER
7390 Wait 5 seconds before clearing the send buffers and reseting the connection with
7391 the client when we do a close. This should be enough time to get the message to
7395 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7397 * gst/rtsp-server/rtsp-server.c:
7398 server: use SO_LINGER
7399 SO_LINGER on the socket will make sure that any pending data on the socket is
7400 flushed ASAP and that the socket connection is reset. This makes sure that the
7401 socket can be reused immediately.
7404 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7407 README: add blurb about shared media factories
7409 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
7411 * gst/rtsp-server/rtsp-media.c:
7412 Add stdlib.h for atoi()
7414 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7416 * bindings/python/Makefile.am:
7417 * bindings/vala/Makefile.am:
7418 build: distcheck fixes
7419 Fix 'make distcheck', somewhat (it still fails because it tries to
7420 install files into /usr/share/vala/vapi/ irrespective of the
7423 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7426 configure: bump core/base requirements to released version
7427 Makes things less confusing for people.
7429 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7432 configure: fail if GStreamer core/base requirements are not met
7434 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7436 * gst/rtsp-server/rtsp-client.c:
7437 client: improve client cleanups
7438 Make sure the session does not timeout when using TCP. We need to do this
7439 because quicktime player does not send RTCP for some reason in tunneled
7441 Refactor some cleanup code.
7444 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7446 * gst/rtsp-server/rtsp-session.c:
7447 * gst/rtsp-server/rtsp-session.h:
7448 session: add support for prevent session timeouts
7449 Add an atomix counter to prevent session timeouts when we are, for example,
7452 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7454 * gst/rtsp-server/rtsp-client.c:
7455 client: fix unlink on session timeouts
7456 When our session times out, make sure we unlink all streams in this
7458 Remove the tunnelid when closing the connection.
7460 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7462 * gst/rtsp-server/rtsp-session.c:
7463 session: small cleanups
7465 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7467 * gst/rtsp-server/rtsp-client.c:
7468 client: handle lost_tunnel callbacks
7469 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
7470 hashtable so that we can reuse it for when the client reopens the POST
7472 Close the connection after a TEARDOWN.
7473 Make sure or watchid is cleared when the watch is removed.
7476 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7478 * gst/rtsp-server/rtsp-client.c:
7479 * gst/rtsp-server/rtsp-media.c:
7480 * gst/rtsp-server/rtsp-sdp.c:
7481 rtsp-server: add more support for multicast
7483 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7486 * gst/rtsp-server/rtsp-media.c:
7487 * gst/rtsp-server/rtsp-media.h:
7488 media: allow configuration of allowed lower transport
7490 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7492 * gst/rtsp-server/rtsp-client.h:
7493 * gst/rtsp-server/rtsp-media.c:
7494 * gst/rtsp-server/rtsp-media.h:
7495 * gst/rtsp-server/rtsp-sdp.c:
7496 * gst/rtsp-server/rtsp-sdp.h:
7497 * gst/rtsp-server/rtsp-server.c:
7498 rtsp: keep track of server ip and ipv6
7499 Keep track of how the client connected to the server and setup the udp ports
7500 with the same protocol.
7501 Copy the server ip address in the SDP so that clients can send RTCP back to
7504 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7506 * gst/rtsp-server/rtsp-session.c:
7509 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7511 * gst/rtsp-server/rtsp-client.c:
7512 client: use right size for malloc
7514 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7516 * gst/rtsp-server/rtsp-server.c:
7517 server: comment ipv6 server listening address
7519 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7521 * gst/rtsp-server/rtsp-media.c:
7522 media: allow for ipv6 sockets
7524 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7526 * gst/rtsp-server/rtsp-server.c:
7527 * gst/rtsp-server/rtsp-server.h:
7528 server: rework server part
7529 Allow setting a bind address, make sure we can deal with ipv6.
7530 Remove the port property and change with the service property.
7532 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7534 * gst/rtsp-server/rtsp-media.h:
7535 media: update comments a little
7537 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7539 * gst/rtsp-server/rtsp-client.c:
7540 client: make content-base better
7541 Use the URI formatting functions to make a content-base. Also make sure that
7542 there is a trailing / at the end.
7544 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7546 * gst/rtsp-server/rtsp-client.c:
7547 client: guard against invalid paths
7549 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7551 * examples/test-video.c:
7552 test: catch server bind errors
7554 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
7556 * gst/rtsp-server/rtsp-media.c:
7557 rtspmedia: emit "unprepared" if _prepare fails.
7558 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
7559 media object is removed from its factory's cache.
7561 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7563 * gst/rtsp-server/rtsp-media.c:
7564 media: collect media position when seek completes
7566 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
7568 * gst/rtsp-server/rtsp-client.c:
7569 client: call unlink_streams in client finalize
7572 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7574 * gst/rtsp-server/rtsp-media.c:
7575 media: limit the time to wait to something huge
7576 Avoid waiting forever but limit the timeout to 20 seconds.
7578 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7580 * gst/rtsp-server/rtsp-sdp.c:
7581 sdp: reindent and check for prepared status
7583 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7585 * gst/rtsp-server/rtsp-media.c:
7586 * gst/rtsp-server/rtsp-media.h:
7587 * gst/rtsp-server/rtsp-session.c:
7588 media: avoid doing _get_state() for state changes
7589 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
7590 until the media is prerolled or in error. This avoids doing a blocking call of
7591 gst_element_get_state() that can cause lockups when there is an error.
7594 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7596 * gst/rtsp-server/rtsp-media.c:
7599 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7601 * gst/rtsp-server/rtsp-media-factory.c:
7602 media-factory: better error handling
7603 Improve the error handling a bit.
7605 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7607 * gst/rtsp-server/rtsp-client.c:
7608 client: rework transport parsing
7609 Rework the transport parsing code so that we can ignore transports we don't
7610 support instead of just picking the first one we can parse.
7611 Configure a (for now hardcoded) destination for multicast transports.
7613 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7615 * gst/rtsp-server/rtsp-media.c:
7616 media: set multicast sink parameters
7617 Disable loop and automatic multicast join on the udpsink elements.
7618 Add some more debug info.
7619 Reset some state variables in the right place.
7620 Use the right port numbers for multicast.
7622 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7624 * gst/rtsp-server/rtsp-session.c:
7625 session: handle transport setup correctly
7626 Handle UDP, MCAST and TCP transport negotiation more correctly.
7627 Store the server session SSRC in the transport.
7629 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7631 * gst/rtsp-server/rtsp-client.c:
7632 rtsp-client: implement error_full
7633 Implement error_full to avoid some segfaults when the rtspconnection calls it.
7636 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7639 * gst/rtsp-server/rtsp-client.c:
7640 * gst/rtsp-server/rtsp-server.c:
7641 docs: update docs and comments
7643 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
7645 * gst/rtsp-server/rtsp-sdp.c:
7646 sdp: make server work better when behind a proxy
7648 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7650 * gst/rtsp-server/rtsp-client.c:
7651 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
7653 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7655 * gst/rtsp-server/rtsp-client.c:
7656 * gst/rtsp-server/rtsp-media-factory.c:
7657 * gst/rtsp-server/rtsp-media-mapping.c:
7658 * gst/rtsp-server/rtsp-media.c:
7659 * gst/rtsp-server/rtsp-server.c:
7660 * gst/rtsp-server/rtsp-session-pool.c:
7661 * gst/rtsp-server/rtsp-session.c:
7662 Use GStreamer's debugging subsystem
7664 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7666 * gst/rtsp-server/rtsp-media-factory.c:
7667 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
7669 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7674 === release 0.10.5 ===
7676 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7681 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7684 configure: bump required versions
7686 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
7688 * gst/rtsp-server/rtsp-client.c:
7689 client: call weak-unref on client->sessions from finalize
7692 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7694 * gst/rtsp-server/rtsp-media.c:
7695 media: Fixed crasher where caps got unref'ed too often
7697 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7700 * pkgconfig/.gitignore:
7701 * pkgconfig/Makefile.am:
7702 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
7703 Added pkg-config file to use gst-rtsp-server uninstalled
7705 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7707 * gst/rtsp-server/rtsp-media.c:
7708 media: add some docs
7710 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
7712 * gst/rtsp-server/rtsp-client.c:
7713 rtsp: Use gst_rtsp_watch_send_message().
7714 Use gst_rtsp_watch_send_message() since the old API which used
7715 gst_rtsp_watch_queue_message() has been deprecated.
7717 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7722 === release 0.10.4 ===
7724 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7729 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7731 * gst/rtsp-server/rtsp-client.c:
7732 * gst/rtsp-server/rtsp-session.c:
7733 * gst/rtsp-server/rtsp-session.h:
7734 rtsp: allocate channels in TCP mode
7735 When the client does not provide us with channels in TCP mode, allocate channels
7738 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7740 * gst/rtsp-server/rtsp-client.c:
7741 client: don't crash when tunnelid is missing
7742 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
7743 don't crash but return an error response to the client.
7746 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7748 * bindings/vala/gst-rtsp-server-0.10.vapi:
7749 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7750 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7751 bindings: update vala bindings with new method
7753 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7755 * gst/rtsp-server/rtsp-session-pool.c:
7756 * gst/rtsp-server/rtsp-session-pool.h:
7757 sessionpool: add function to filter sessions
7758 Add generic function to retrieve/remove sessions.
7760 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7763 configure: bump core/base requirements to release
7765 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7767 * gst/rtsp-server/rtsp-media.c:
7768 media: fix indentation
7770 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7772 * gst/rtsp-server/rtsp-media.c:
7773 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
7775 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7777 * gst/rtsp-server/rtsp-media.c:
7778 set state and remove elements of media in for loop
7780 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
7782 * bindings/vala/gst-rtsp-server-0.10.vapi:
7783 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7784 Added gst_rtsp_media_remove_elements function to Vala bindings
7786 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
7788 * gst/rtsp-server/rtsp-media.c:
7789 * gst/rtsp-server/rtsp-media.h:
7790 Added gst_rtsp_media_remove_elements function
7792 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
7794 * gst/rtsp-server/rtsp-media.c:
7795 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
7797 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7799 * bindings/vala/gst-rtsp-server-0.10.vapi:
7800 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7801 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7802 Updated Vala bindings
7804 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7806 * gst/rtsp-server/rtsp-media.c:
7807 * gst/rtsp-server/rtsp-media.h:
7808 Added vmethod unprepare to GstRTSPMedia
7809 The default implementation sets the state of the pipeline to GST_STATE_NULL
7811 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7813 * gst/rtsp-server/rtsp-media-factory.c:
7814 * gst/rtsp-server/rtsp-media-factory.h:
7815 Made collect_streams function public
7817 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7819 * gst/rtsp-server/rtsp-media-factory.c:
7820 * gst/rtsp-server/rtsp-media-factory.h:
7821 * gst/rtsp-server/rtsp-media.c:
7822 Added vmethod create_pipeline to GstRTSPMediaFactory
7823 The pipeline is created in this method and the GstRTSPMedia's element is added to it
7825 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7827 * gst/rtsp-server/rtsp-client.c:
7828 client: use g_source_destroy()
7829 We need to use g_source_destroy() because we might have added the source to a
7830 different main context than the default one.
7832 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7834 * gst/rtsp-server/Makefile.am:
7835 * gst/rtsp-server/rtsp-client.c:
7836 * gst/rtsp-server/rtsp-params.c:
7837 * gst/rtsp-server/rtsp-params.h:
7838 rtsp: prepare for handling GET/SET_PARAMETER
7839 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
7841 Fix return codes of handlers.
7843 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7845 * gst/rtsp-server/rtsp-media.c:
7846 media: don't leak session pads
7848 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7850 * gst/rtsp-server/rtsp-media.c:
7851 media: clean up the messages a bit
7853 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7855 * gst/rtsp-server/rtsp-sdp.c:
7856 sdp: warn and skip streams without media
7858 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7860 * bindings/vala/gst-rtsp-server-0.10.vapi:
7861 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7862 vala: Fixed typo in header file of RTSPMediaStream
7864 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7866 * gst/rtsp-server/rtsp-media.c:
7869 Make dumping RTCP stats configurable
7871 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7873 * gst/rtsp-server/rtsp-media.c:
7874 media: be less verbose and leak less
7876 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7878 * gst/rtsp-server/rtsp-media.c:
7879 media: don't leak the destination address
7881 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7883 * gst/rtsp-server/rtsp-client.c:
7884 * gst/rtsp-server/rtsp-media.c:
7885 * gst/rtsp-server/rtsp-media.h:
7886 * gst/rtsp-server/rtsp-session.c:
7887 * gst/rtsp-server/rtsp-session.h:
7888 rtsp: use RTCP to keep the session alive
7889 Use the RTCP rtcp-from stats field to find the associated session and use this
7890 to keep the session alive.
7892 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7894 * gst/rtsp-server/rtsp-session.c:
7895 session: add 5sec to the real session timeout
7896 Allow the session to live 5sec longer before really timing out. This should give
7897 clients some extra time to keep the session active.
7899 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7901 * gst/rtsp-server/rtsp-client.c:
7902 client: replay OK to GET/SET_PARAMETER
7903 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
7904 so that we return OK for those requests.
7906 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7908 * gst/rtsp-server/rtsp-media.c:
7909 * gst/rtsp-server/rtsp-media.h:
7910 media: keep track of active transports
7911 Keep track of which transport is active to avoid closing the connection too
7913 Remove the destination transport also when going to NULL.
7914 Print some stats about the SDES and other RTCP messages we receive from the
7917 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7919 * examples/.gitignore:
7920 * examples/Makefile.am:
7921 * examples/test-sdp.c:
7922 example: add SDP relay example
7924 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7926 * gst/rtsp-server/rtsp-media.c:
7927 media: also count active TCP connections
7929 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7931 * gst/rtsp-server/rtsp-media-factory.c:
7932 * gst/rtsp-server/rtsp-media.c:
7933 * gst/rtsp-server/rtsp-media.h:
7934 rtsp: add support for dynamic elements
7935 Add support for dynamic elements.
7936 Don't set live pipelines back to paused.
7938 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7940 * gst/rtsp-server/rtsp-sdp.c:
7941 sdp: don't add encoding name when absent in caps
7943 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7945 * gst/rtsp-server/rtsp-client.c:
7946 client: warn when we can't do RTP-Info
7948 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7950 * gst/rtsp-server/rtsp-media-factory.c:
7951 factory: factor out the stream construction
7953 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7955 * gst/rtsp-server/rtsp-client.c:
7956 client: only add RTP-Info when we have the info
7957 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
7960 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7965 === release 0.10.3 ===
7967 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7971 - Fixes a bug where it put the wrong verion in pkgconfig
7972 - Link RTP and RTCP sources
7974 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7976 * gst/rtsp-server/rtsp-media.c:
7977 * gst/rtsp-server/rtsp-media.h:
7978 media: link the RTP udpsrc to the session manager
7979 Link the RTP udpsrc and the appsrc to the session manager so that they don't
7980 shut down when the client sends a packet to open firewalls.
7982 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7984 * pkgconfig/gst-rtsp-server.pc.in:
7985 Don't use hard-coded version number in pkg-config file
7987 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7992 === release 0.10.2 ===
7994 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7999 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8002 * common/m4/.gitignore:
8003 * examples/.gitignore:
8004 * pkgconfig/.gitignore:
8005 add some .gitignore files
8007 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8009 * gst/rtsp-server/rtsp-media.c:
8010 media: seek to key frames
8012 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8014 * gst/rtsp-server/rtsp-media.c:
8015 media: emit the unprepared signal by id
8016 Emit the unprepared signal by id instead of name and set the media as
8019 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8021 * gst/rtsp-server/rtsp-media.c:
8022 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
8024 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8026 * gst/rtsp-server/rtsp-server.c:
8027 Added finalize function to GstRTPSPServer to unref session pool and media mapping
8029 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
8031 * bindings/vala/gst-rtsp-server-0.10.vapi:
8032 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8033 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8034 Updated vala bindings
8036 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8038 * gst/rtsp-server/Makefile.am:
8039 * gst/rtsp-server/rtsp-client.c:
8040 * gst/rtsp-server/rtsp-media.c:
8041 server: use appsink and appsrc with the API
8042 Use the appsink/appsrc API instead of the signals for higher
8045 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8047 * examples/test-ogg.c:
8048 tests: set the payload type correctly
8050 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8052 * gst/rtsp-server/rtsp-media-factory.c:
8053 factory: connect to the unprepare signal
8054 Connect to the unprepare signal for non-reusable media so that we can remove
8055 them from the cache.
8057 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8059 * gst/rtsp-server/rtsp-media.c:
8060 * gst/rtsp-server/rtsp-media.h:
8061 media: add signal to notify of unprepare
8063 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8065 * gst/rtsp-server/rtsp-media.c:
8066 * gst/rtsp-server/rtsp-media.h:
8067 media: more work on making the media shared
8068 Add a reusable flag to medias, indicating that they can be reused after a state
8072 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8074 * examples/test-readme.c:
8075 examples: mark the example as shared for testing
8077 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8079 * gst/rtsp-server/rtsp-media.c:
8080 * gst/rtsp-server/rtsp-media.h:
8081 client: support shared media
8082 Always perform the state actions even if the target state of the pipeline is
8083 already correct, we still want to add/remove the transports when we are dealing
8085 Keep a counter of the number of active transports for a media so that we can use
8086 this to perform a state change when needed.
8087 Perform a state change of the pipeline only when the first transport was added
8088 or when there are no active transports.
8090 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8092 * gst/rtsp-server/rtsp-client.c:
8093 client: fix refcounting crasher
8094 Don't need to remove the weak refs in the finalize methods, they are already
8095 removed in the dispose.
8096 Don't register the callback with a DestroyNofity.
8098 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8100 * gst/rtsp-server/rtsp-client.c:
8101 Fix rtsp client refcount management in TCP mode.
8102 Don't unref a client ref we never had. Fixes an unref
8103 of an already-free client object after a client
8104 teardown request for me.
8106 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8108 * gst/rtsp-server/rtsp-session.c:
8109 docs: fix typo in API docs
8111 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8113 * gst/rtsp-server/rtsp-media.c:
8115 Keep the udp sources in playing even if we go to paused. unlock the sources when
8117 Add some more debug info.
8118 Only seek when we need to.
8119 Keep track of the position when we go to paused.
8121 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8123 * gst/rtsp-server/rtsp-client.c:
8124 * gst/rtsp-server/rtsp-media.c:
8125 * gst/rtsp-server/rtsp-media.h:
8126 Add beginnings of seeking.
8127 Parse the Range header and perform a seek on the pipeline for the requested
8128 position. It's disabled currently until I figure out what's going wrong.
8130 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8132 * gst/rtsp-server/rtsp-client.c:
8133 allow pause requests for now.
8136 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8138 * gst/rtsp-server/rtsp-client.c:
8139 Remove weak ref on the session in teardown
8140 We need to remove our weakref from the session when we do a teardown because
8141 else we close the TCP connection prematurely.
8143 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8145 * gst/rtsp-server/rtsp-client.c:
8146 * gst/rtsp-server/rtsp-client.h:
8147 * gst/rtsp-server/rtsp-session-pool.c:
8148 Do some more session cleanup
8149 Make session timeout kill the TCP connection that currently watches the
8151 Remove the client timeout property.
8153 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8155 * gst/rtsp-server/rtsp-client.c:
8156 * gst/rtsp-server/rtsp-client.h:
8157 * gst/rtsp-server/rtsp-media.c:
8158 * gst/rtsp-server/rtsp-media.h:
8159 * gst/rtsp-server/rtsp-server.c:
8160 * gst/rtsp-server/rtsp-session.c:
8161 * gst/rtsp-server/rtsp-session.h:
8163 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
8166 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8168 * examples/Makefile.am:
8169 * examples/test-launch.c:
8170 Add example server that takes launch lines
8171 Add an example server that streams any -launch line.
8173 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8175 * examples/test-readme.c:
8176 * gst/rtsp-server/rtsp-client.c:
8177 * gst/rtsp-server/rtsp-media.c:
8178 * gst/rtsp-server/rtsp-media.h:
8179 Add support for live streams
8180 Add support for live streams and ranges
8181 Start on handling TCP data transfer.
8183 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8185 * gst/rtsp-server/rtsp-media.c:
8186 Free the pipeline before other things
8189 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8191 * gst/rtsp-server/rtsp-client.c:
8192 Only free the pending tunnel if there is one
8195 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8197 * gst/rtsp-server/rtsp-client.c:
8198 * gst/rtsp-server/rtsp-client.h:
8199 * gst/rtsp-server/rtsp-media.c:
8200 rtsp-server: Add support for tunneling
8201 Add support for tunneling over HTTP.
8202 Use new connection methods to retrieve the url.
8203 Dispatch messages based on the message type instead of blindly
8204 assuming it's always a request.
8205 Keep track of the watch id so that we can remove it later.
8206 Set the media pipeline to NULL before unreffing the pipeline.
8208 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8210 * gst/rtsp-server/rtsp-client.c:
8211 * gst/rtsp-server/rtsp-client.h:
8212 Fix for channel -> watch rename in gstreamer
8213 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
8215 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8217 * gst/rtsp-server/rtsp-client.c:
8218 * gst/rtsp-server/rtsp-client.h:
8220 Use the async RTSP channels instead of spawning a new thread for each client.
8221 If a sessionid is specified in a request, fail if we don't have the session.
8223 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8225 * gst/rtsp-server/rtsp-media.c:
8226 Add better debug info
8227 Add some better debug info.
8229 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8231 * examples/test-video.c:
8233 Add support for session timeouts in the example.
8235 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8237 * gst/rtsp-server/rtsp-session-pool.c:
8238 * gst/rtsp-server/rtsp-session-pool.h:
8239 Pass GTimeVal around for performance reasons
8240 Get the current time only once and pass it around so that sessions don't have to
8241 get the current time anymore.
8242 Add experimental support for a GSource that dispatches when the session needs to
8245 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8247 * gst/rtsp-server/rtsp-session.c:
8248 * gst/rtsp-server/rtsp-session.h:
8249 Add better support for session timeouts
8250 Add a method to request the number of milliseconds when a session will timeout.
8252 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8254 * gst/rtsp-server/rtsp-media.c:
8255 * gst/rtsp-server/rtsp-media.h:
8256 Add suport for RTP manager monitoring
8257 Add the first stage in monitoring the rtp manager.
8258 Make sure we don't update the state to something we don't want.
8260 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8262 * gst/rtsp-server/rtsp-client.c:
8263 Add support for session keepalive
8264 Get and update the session timeout for all requests. get the session as early as
8267 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8269 * gst/rtsp-server/rtsp-media-factory.h:
8270 * gst/rtsp-server/rtsp-media.c:
8271 * gst/rtsp-server/rtsp-media.h:
8272 Handle media bus messages
8273 Handle media bus messages in a custom mainloop and dispatch them to the
8274 RTSPMedia objects. Let the default implementation handle some common messages.
8276 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8278 * gst/rtsp-server/rtsp-client.c:
8279 * gst/rtsp-server/rtsp-session-pool.c:
8280 * gst/rtsp-server/rtsp-session.c:
8281 Some more session timeout handling
8282 Move the session header setting code to a central place so that we always add
8283 the timeout parameter too.
8284 Handle timeouts by running the session cleanup code.
8285 Stop media before cleaning up.
8287 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8289 * gst/rtsp-server/rtsp-client.c:
8290 * gst/rtsp-server/rtsp-client.h:
8291 Add timeout property
8292 Add a timeout property ot the client and make the other properties into GObject
8295 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8297 * gst/rtsp-server/rtsp-session-pool.c:
8298 Use getters and setters in property code
8299 Use the getters and setters for the timeout property instead of locking
8302 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8304 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
8306 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8308 * gst/rtsp-server/rtsp-session-pool.c:
8309 * gst/rtsp-server/rtsp-session-pool.h:
8310 * gst/rtsp-server/rtsp-session.c:
8311 * gst/rtsp-server/rtsp-session.h:
8312 Add more timeout stuff
8313 Add method to check if a session is expired.
8314 Add method to perform cleanup on a session pool.
8316 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8318 * gst/rtsp-server/rtsp-client.c:
8319 * gst/rtsp-server/rtsp-session-pool.c:
8320 * gst/rtsp-server/rtsp-session-pool.h:
8321 * gst/rtsp-server/rtsp-session.c:
8322 * gst/rtsp-server/rtsp-session.h:
8323 Add beginnings of session timeouts and limits
8324 Add the timeout value to the Session header for unusual timeout values.
8325 Allow us to configure a limit to the amount of active sessions in a pool. Set a
8326 limit on the amount of retry we do after a sessionid collision.
8327 Add properties to the sessionid and the timeout of a session. Keep track of
8328 creation time and last access time for sessions.
8330 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8332 * gst/rtsp-server/rtsp-client.c:
8333 * gst/rtsp-server/rtsp-media.c:
8334 * gst/rtsp-server/rtsp-media.h:
8335 * gst/rtsp-server/rtsp-sdp.c:
8336 * gst/rtsp-server/rtsp-session-pool.c:
8337 * gst/rtsp-server/rtsp-session.c:
8338 * gst/rtsp-server/rtsp-session.h:
8339 Cleanup of sessions and more
8340 Fix the refcounting of media and sessions in the client. Properly clean up the
8341 session data when the client performs a teardown.
8342 Add Server header to responses.
8343 Allow for multiple uri setups in one session.
8344 Add Range header to the PLAY response and add the range attribute to the SDP
8346 Fix the session pool remove method, it used the wrong key in the hashtable. Also
8347 give the ownership of the sessionid to the session object.
8349 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8351 * gst/rtsp-server/rtsp-server.c:
8352 * gst/rtsp-server/rtsp-server.h:
8354 Rename the 'server_port' variable to simply 'port'.
8356 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8359 * gst/rtsp-server/rtsp-client.c:
8360 * gst/rtsp-server/rtsp-media.c:
8361 * gst/rtsp-server/rtsp-media.h:
8362 * gst/rtsp-server/rtsp-session.c:
8363 * gst/rtsp-server/rtsp-session.h:
8364 Rework the way we handle transports for streams
8365 Make the media accept an array of transports for the streams that we have
8366 configured for the play/pause requests.
8367 Implement server states for a client and its media.
8368 Require 0.10.22.1 (git HEAD) of gstreamer.
8370 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8372 * gst/rtsp-server/rtsp-client.c:
8373 * gst/rtsp-server/rtsp-media-factory.c:
8374 Drop const from functions dealing with urls
8375 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
8376 have the right const in them.
8378 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8380 * gst/rtsp-server/rtsp-client.c:
8381 * gst/rtsp-server/rtsp-media.c:
8382 * gst/rtsp-server/rtsp-sdp.c:
8386 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8388 * gst/rtsp-server/rtsp-client.c:
8389 * gst/rtsp-server/rtsp-media-factory.c:
8390 * gst/rtsp-server/rtsp-media.c:
8391 * gst/rtsp-server/rtsp-media.h:
8393 Don't keep a reference to the GstRTSPMedia in the stream.
8394 Free more things when freeing the GstRTSPMedia.
8396 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8399 * gst/rtsp-server/rtsp-media-factory.c:
8400 * gst/rtsp-server/rtsp-media-factory.h:
8401 * gst/rtsp-server/rtsp-media.c:
8402 * gst/rtsp-server/rtsp-media.h:
8403 * gst/rtsp-server/rtsp-server.c:
8404 * gst/rtsp-server/rtsp-server.h:
8405 More docs and small cleanups
8406 Add some more docs and update the README
8407 Cleanup some method names.
8408 Remove an unneeded idx field in the GstRTSPMediaStream
8410 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8413 * examples/Makefile.am:
8414 * examples/test-readme.c:
8415 Add a README and more example code
8416 Add a README file that contains a small introduction on how to use the server
8417 along with the example code explained in the readme.
8419 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8421 * gst/rtsp-server/rtsp-media.c:
8422 * gst/rtsp-server/rtsp-server.c:
8423 Fix some leaks and change default port
8424 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
8425 we finished the initial preroll. If we keep them locked, setting the pipeline to
8426 NULL will not stop and clean up the sources correctly.
8427 Change the default RTSP port to 8554 aka the official alternative RTSP port.
8429 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8431 * gst/rtsp-server/rtsp-session.c:
8432 * gst/rtsp-server/rtsp-session.h:
8433 Cleanups to the session object
8434 Remove some unneeded variables in the session state of a stream such as the
8435 owner media and the server transport.
8436 Get the configuration of a media stream in a session based on the media_stream
8437 in the original object instead of our cached index.
8438 Free more data in the finalize method.
8440 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8442 * gst/rtsp-server/rtsp-client.c:
8443 * gst/rtsp-server/rtsp-client.h:
8444 Cleanups and reuse media from DESCRIBE
8445 Handle thread create errors.
8446 Rename some internal methods to better match what they actually do.
8447 Handle misconfiguration of session_pool and media_mapping gracefully.
8448 Cache the DESCRIBE media and uri in the client connection and reuse them when
8449 we receive a SETUP request in the same connection for the same uri.
8450 Cleanup the client connection object.
8452 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8454 * gst/rtsp-server/rtsp-media-factory.c:
8455 * gst/rtsp-server/rtsp-media-factory.h:
8456 * gst/rtsp-server/rtsp-media.c:
8457 * gst/rtsp-server/rtsp-media.h:
8458 Add shared properties to media and factory
8459 Add the shared property to media.
8460 Implement some simple caching in the factory depending on if the media is shared
8463 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8465 * gst/rtsp-server/rtsp-client.c:
8466 Add a little comment
8467 Add some comment about the content-base header.
8469 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8471 * examples/Makefile.am:
8473 * examples/test-mp4.c:
8474 * examples/test-ogg.c:
8475 * examples/test-video.c:
8476 * gst/rtsp-server/Makefile.am:
8477 * gst/rtsp-server/rtsp-client.c:
8478 * gst/rtsp-server/rtsp-client.h:
8479 * gst/rtsp-server/rtsp-media-factory.c:
8480 * gst/rtsp-server/rtsp-media-factory.h:
8481 * gst/rtsp-server/rtsp-media.c:
8482 * gst/rtsp-server/rtsp-media.h:
8483 * gst/rtsp-server/rtsp-sdp.c:
8484 * gst/rtsp-server/rtsp-sdp.h:
8485 * gst/rtsp-server/rtsp-server.c:
8486 * gst/rtsp-server/rtsp-server.h:
8487 * gst/rtsp-server/rtsp-session.c:
8488 * gst/rtsp-server/rtsp-session.h:
8489 Reorganize things, prepare for media sharing
8490 Added various other test server examples
8491 Move the SDP message generation to a separate helper.
8492 Refactor common code for finding the session.
8493 Add content-base for realplayer compatibility
8494 Clean up request uris before processing for better vlc compatibility.
8495 Move prerolling and pipeline construction to the RTSPMedia object.
8496 Use multiudpsink for future pipeline reuse.
8498 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8504 === release 0.10.1 ===
8506 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8512 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8514 * bindings/vala/Makefile.am:
8516 Add more directories and files to the dist.
8518 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8520 * bindings/python/Makefile.am:
8521 * bindings/python/rtspserver.override:
8522 Fixed compile error of python bindings
8524 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8526 * bindings/vala/gst-rtsp-server-0.10.vapi:
8527 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8528 Marked values as nullable accordingly
8530 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8532 * bindings/vala/gst-rtsp-server-0.10.vapi:
8533 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8534 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8535 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8536 Updated Vala bindings
8538 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8540 * gst/rtsp-server/rtsp-client.c:
8541 * gst/rtsp-server/rtsp-media-mapping.c:
8542 * gst/rtsp-server/rtsp-media-mapping.h:
8543 * gst/rtsp-server/rtsp-media.h:
8544 * gst/rtsp-server/rtsp-session-pool.h:
8545 Cleanups and doc updates
8546 Add some more documentation and do some minor cleanups here and there.
8548 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8550 * gst/rtsp-server/rtsp-client.c:
8551 * gst/rtsp-server/rtsp-media-factory.c:
8552 * gst/rtsp-server/rtsp-media-factory.h:
8553 * gst/rtsp-server/rtsp-media.c:
8554 * gst/rtsp-server/rtsp-media.h:
8555 * gst/rtsp-server/rtsp-session.c:
8556 * gst/rtsp-server/rtsp-session.h:
8558 Rename GstRTSPMediaBin to GstRTSPMedia
8559 Parse the request url into a GstRTSPUri object and pass this object to the
8560 various handlers and methods that require the uri.
8562 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8566 Add some more docs and remove some old code from the example.
8568 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8570 * gst/rtsp-server/rtsp-client.c:
8571 Handle state change failures better
8572 Handle state change failures better when changing the state of the pipeline to
8575 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8577 * gst/rtsp-server/rtsp-media-factory.c:
8578 * gst/rtsp-server/rtsp-media-factory.h:
8579 Make element creation more extendible
8580 Add get_element vmethod to the default MediaFactory so that subclasses can just
8581 override that method and still use the default logic for making a MediaBin from
8584 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8587 * gst/rtsp-server/Makefile.am:
8588 * gst/rtsp-server/rtsp-client.c:
8589 * gst/rtsp-server/rtsp-client.h:
8590 * gst/rtsp-server/rtsp-media-factory.c:
8591 * gst/rtsp-server/rtsp-media-factory.h:
8592 * gst/rtsp-server/rtsp-media-mapping.c:
8593 * gst/rtsp-server/rtsp-media-mapping.h:
8594 * gst/rtsp-server/rtsp-media.c:
8595 * gst/rtsp-server/rtsp-media.h:
8596 * gst/rtsp-server/rtsp-server.c:
8597 * gst/rtsp-server/rtsp-server.h:
8598 * gst/rtsp-server/rtsp-session.c:
8599 * gst/rtsp-server/rtsp-session.h:
8600 Make the server handle arbitrary pipelines
8601 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
8602 The GstMediaBin object has a handle to a bin with elements and to a list of
8603 GstMediaStream objects that this bin produces.
8604 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
8605 with methods to register and remove those mappings.
8606 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
8607 used by the server instance.
8608 Modify the example application so that it shows how to create custom pipelines
8609 attached to a specific mount point.
8610 Various misc cleanps.
8612 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8614 * gst/rtsp-server/rtsp-server.c:
8615 * gst/rtsp-server/rtsp-server.h:
8616 Allow setting a custom media factory for a server
8618 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8620 * gst/rtsp-server/rtsp-client.c:
8621 * gst/rtsp-server/rtsp-client.h:
8622 Allow setting a custom media factory for a client.
8624 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8626 * gst/rtsp-server/Makefile.am:
8627 Add Makefile entry for the media factory
8629 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8631 * gst/rtsp-server/rtsp-media-factory.c:
8632 * gst/rtsp-server/rtsp-media-factory.h:
8633 Add media factory to map urls to media pipeline objects.
8635 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8637 * gst/rtsp-server/rtsp-media.c:
8638 * gst/rtsp-server/rtsp-media.h:
8639 Add comments. Remove unused field
8641 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8643 * gst/rtsp-server/rtsp-session-pool.c:
8644 * gst/rtsp-server/rtsp-session-pool.h:
8645 Allow custom session pools to override the session id allocation algorithms Add some comments.
8647 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8649 * gst/rtsp-server/rtsp-session.h:
8652 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8654 * gst/rtsp-server/rtsp-client.c:
8655 * gst/rtsp-server/rtsp-client.h:
8656 Move the connection code in one place Add some comments
8658 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8660 * gst/rtsp-server/rtsp-server.c:
8661 * gst/rtsp-server/rtsp-server.h:
8662 Make vmethod to create and accept new clients. Add some docs.
8664 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8666 * gst/rtsp-server/rtsp-server.c:
8667 * gst/rtsp-server/rtsp-server.h:
8668 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
8670 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8672 * gst/rtsp-server/rtsp-client.c:
8673 * gst/rtsp-server/rtsp-client.h:
8674 Name the parameters more appropriately.
8676 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8678 * gst/rtsp-server/rtsp-session-pool.c:
8679 Do some more cleanup of the session pool.
8681 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8683 * gst/rtsp-server/Makefile.am:
8684 * gst/rtsp-server/rtsp-client.c:
8685 Check if return value of gst_rtsp_session_get_media is not NULL
8687 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8689 * gst/rtsp-server/Makefile.am:
8690 Install rtsp-session and rtsp-session-pool headers
8692 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8697 * bindings/python/Makefile.am:
8698 * bindings/python/arg-types.py:
8699 * bindings/python/codegen/Makefile.am:
8700 * bindings/python/codegen/__init__.py:
8701 * bindings/python/codegen/argtypes.py:
8702 * bindings/python/codegen/code-coverage.py:
8703 * bindings/python/codegen/codegen.py:
8704 * bindings/python/codegen/definitions.py:
8705 * bindings/python/codegen/defsparser.py:
8706 * bindings/python/codegen/docextract.py:
8707 * bindings/python/codegen/docgen.py:
8708 * bindings/python/codegen/fileprefix.override:
8709 * bindings/python/codegen/fileprefixmodule.c:
8710 * bindings/python/codegen/h2def.py:
8711 * bindings/python/codegen/mergedefs.py:
8712 * bindings/python/codegen/mkskel.py:
8713 * bindings/python/codegen/override.py:
8714 * bindings/python/codegen/reversewrapper.py:
8715 * bindings/python/codegen/scmexpr.py:
8716 * bindings/python/rtspserver-types.defs:
8717 * bindings/python/rtspserver.defs:
8718 * bindings/python/rtspserver.override:
8719 * bindings/python/rtspservermodule.c:
8721 Add python bindings.
8723 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8725 * bindings/Makefile.am:
8727 Don't go into python dir when requirements for python bindings are missing
8729 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8731 * bindings/Makefile.am:
8732 * bindings/vala/Makefile.am:
8734 Install Vala bindings if vala is available
8736 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8738 * bindings/vala/gst-rtsp-server-0.10.deps:
8739 * bindings/vala/gst-rtsp-server-0.10.vapi:
8740 * bindings/vala/gst-rtsp-server.vapi:
8741 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
8742 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8743 * bindings/vala/packages/gst-rtsp-server-0.10.files:
8744 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8745 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8746 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
8747 * bindings/vala/packages/gst-rtsp-server.deps:
8748 * bindings/vala/packages/gst-rtsp-server.excludes:
8749 * bindings/vala/packages/gst-rtsp-server.files:
8750 * bindings/vala/packages/gst-rtsp-server.gi:
8751 * bindings/vala/packages/gst-rtsp-server.metadata:
8752 * bindings/vala/packages/gst-rtsp-server.namespace:
8753 Regenerated Vala bindings
8755 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8757 * bindings/vala/gst-rtsp-server.vapi:
8758 * bindings/vala/packages/gst-rtsp-server.metadata:
8759 Fixed typo in included headers for vala bindings
8761 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8765 * pkgconfig/Makefile.am:
8766 * pkgconfig/gst-rtsp-server.pc.in:
8767 Added pkgconfig file
8769 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8771 * bindings/vala/gst-rtsp-server.vapi:
8772 * bindings/vala/packages/gst-rtsp-server.excludes:
8773 * bindings/vala/packages/gst-rtsp-server.gi:
8774 * bindings/vala/packages/gst-rtsp-server.metadata:
8775 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
8777 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8779 * bindings/vala/gst-rtsp-server.vapi:
8780 * bindings/vala/packages/gst-rtsp-server.deps:
8781 * bindings/vala/packages/gst-rtsp-server.files:
8782 * bindings/vala/packages/gst-rtsp-server.gi:
8783 * bindings/vala/packages/gst-rtsp-server.metadata:
8784 * bindings/vala/packages/gst-rtsp-server.namespace:
8787 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
8789 * gst/rtsp-server/rtsp-session.c:
8790 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
8792 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8794 * examples/Makefile.am:
8795 * gst/rtsp-server/Makefile.am:
8796 Put GStreamer version in library name
8798 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8800 * examples/Makefile.am:
8801 * gst/rtsp-server/Makefile.am:
8802 Fix some issues to pass distcheck
8804 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8806 * gst/rtsp-server/rtsp-server.c:
8807 Added port property to GstRTSPServer class.
8809 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8814 * examples/Makefile.am:
8817 * gst/rtsp-server/Makefile.am:
8818 * gst/rtsp-server/rtsp-client.c:
8819 * gst/rtsp-server/rtsp-client.h:
8820 * gst/rtsp-server/rtsp-media.c:
8821 * gst/rtsp-server/rtsp-media.h:
8822 * gst/rtsp-server/rtsp-server.c:
8823 * gst/rtsp-server/rtsp-server.h:
8824 * gst/rtsp-server/rtsp-session-pool.c:
8825 * gst/rtsp-server/rtsp-session-pool.h:
8826 * gst/rtsp-server/rtsp-session.c:
8827 * gst/rtsp-server/rtsp-session.h:
8830 * src/rtsp-client.c:
8831 * src/rtsp-client.h:
8834 * src/rtsp-server.c:
8835 * src/rtsp-server.h:
8836 * src/rtsp-session-pool.c:
8837 * src/rtsp-session-pool.h:
8838 * src/rtsp-session.c:
8839 * src/rtsp-session.h:
8840 Split in library and example program
8842 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8844 * src/rtsp-client.h:
8845 Removed obsolete variable
8847 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8849 * src/rtsp-client.c:
8850 * src/rtsp-client.h:
8851 Removed pipeline variable GstRTSPClient, because it's only used in one function
8853 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8856 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
8858 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
8860 * src/rtsp-session.c:
8861 Initialize some more vars.
8863 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
8865 * src/rtsp-session.c:
8866 Initialize variable to avoid compiler warning.
8868 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
8871 Add a reasonable generic .gitignore