3 2015-06-24 Sebastian Dröge <slomo@coaxion.net>
8 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
10 * gst/rtsp-server/rtsp-client.c:
11 * gst/rtsp-server/rtsp-client.h:
12 * tests/check/gst/client.c:
13 rtsp-client: allow application to decide what requirements are supported
14 Add "check-requirements" signal and vfunc to allow application
15 (and subclasses) to check the requirements.
16 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
17 https://bugzilla.gnome.org/show_bug.cgi?id=749417
19 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
22 Automatic update of common submodule
23 From 6015d26 to f74b2df
25 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
27 * gst/rtsp-server/rtsp-media.c:
28 rtsp-media: Always use real payloader when creating streams
29 A bin that contains the real payloader might be used as payloader. In this
30 case we have to get the real payloader for the various properties it provides.
31 Example use cases for this are bins that payload some media and then have
32 additional elements that add metadata or RTP extension headers to the stream.
33 https://bugzilla.gnome.org/show_bug.cgi?id=750800
35 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
37 * examples/test-netclock-client.c:
38 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
40 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
42 * examples/test-netclock-client.c:
43 * examples/test-netclock.c:
44 test-netclock: Use new ntp-time-source property on rtpbin
45 Select the clock time to be used as NTP time source. This allows proper
46 synchronization between receivers, independent of sharing base times, and just
47 requires them to use the same clock.
49 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
51 * examples/test-netclock-client.c:
52 * examples/test-netclock.c:
53 test-netclock: Setting the same base time on sender and receiver is not necessary
54 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
56 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
58 * gst/rtsp-server/rtsp-stream.c:
59 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
60 https://bugzilla.gnome.org/show_bug.cgi?id=750764
62 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
64 * docs/libs/gst-rtsp-server.types:
65 docs: add missing types
66 https://bugzilla.gnome.org/show_bug.cgi?id=750764
68 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
70 * docs/libs/gst-rtsp-server-sections.txt:
71 docs: add missing apis
72 https://bugzilla.gnome.org/show_bug.cgi?id=750764
74 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
76 * examples/test-netclock-client.c:
77 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
79 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
81 * docs/libs/gst-rtsp-server-sections.txt:
82 * gst/rtsp-server/rtsp-auth.c:
83 * gst/rtsp-server/rtsp-auth.h:
84 GstRTSPAuth: Add client certificate authentication support
85 https://bugzilla.gnome.org/show_bug.cgi?id=750471
87 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
89 * examples/test-netclock-client.c:
90 test-netclock-client: Use new GstClock API to wait for clock synchronization
92 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
94 * examples/test-netclock-client.c:
95 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
96 A mainloop is needed to get glimagesink to display something on OSX, and
97 the source-setup signal just makes things a little bit easier.
99 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
102 Automatic update of common submodule
103 From d9a3353 to 6015d26
105 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
108 Automatic update of common submodule
109 From d37af32 to d9a3353
111 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
114 Automatic update of common submodule
115 From 21ba2e5 to d37af32
117 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
120 Automatic update of common submodule
121 From c408583 to 21ba2e5
123 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
125 * docs/libs/Makefile.am:
126 docs: remove variables that we define in the snippet from common
127 This is syncing our Makefile.am with upstream gtkdoc.
129 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
132 Automatic update of common submodule
133 From 44a3517 to c408583
135 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
140 === release 1.5.1 ===
142 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
148 * gst-rtsp-server.doap:
151 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
153 * gst/rtsp-server/rtsp-client.c:
154 rtsp-client: No flush during Teardown.
155 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
156 backlog is empty it can happen that just a part of a message will be
157 sent and rest is in backlog queue. If then flush during teardown
158 just a part of message will be sent.This can lead to client miss
159 teardown response since it expect to get the last part of message.
160 The flushing during teardown was introduced to fix a deadlock that now
161 is fixed more generally in handle_request by temporary setting backlog
163 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
165 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
167 * tests/check/Makefile.am:
168 tests: Use AM_TESTS_ENVIRONMENT
169 Needed by the new automake test runner and the
170 current version of the common submodule.
172 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
174 * gst/rtsp-server/rtsp-media.h:
175 * gst/rtsp-server/rtsp-stream.h:
176 rtsp-server: Use single-include rtsp header to make sure we get all definitions
178 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
180 * gst/rtsp-server/rtsp-media.c:
181 rtsp-media: Mark some more functions static
183 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
185 * gst/rtsp-server/rtsp-media.c:
186 rtsp-media: Only unblock the media in suspend() when actually changing the state
187 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
189 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
191 * examples/test-video-rtx.c:
192 examples: Use AVPF profile for the RTX example
194 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
196 * gst/rtsp-server/rtsp-sdp.c:
197 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
199 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
201 * gst/rtsp-server/rtsp-stream.c:
202 rtsp-stream: get valid clock-rate from last-sample
203 clock-rate in last-sample's caps is integer, not unsigned.
204 To get this value properly, variable needs to be type-casted to int.
205 https://bugzilla.gnome.org/show_bug.cgi?id=747614
207 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
211 autogen.sh: only run autopoint if gettext requested in configure.ac
212 Not just because there happens to be a po directory.
213 https://bugzilla.gnome.org/show_bug.cgi?id=748058
215 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
218 Revert "configure.ac: uncomment gettext version setup"
219 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
220 We don't need a gettext setup here and there's no po
221 directory either, so no reason why autopoint would be
222 run in the first place.
223 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
225 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
227 * examples/test-multicast.c:
228 * examples/test-multicast2.c:
229 * examples/test-sdp.c:
230 * examples/test-video-rtx.c:
231 * examples/test-video.c:
232 * tests/test-cleanup.c:
233 * tests/test-reuse.c:
234 Fix timeout function signatures across tests and examples
236 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
238 * tests/check/Makefile.am:
239 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
240 Make sure the test environment is set up.
241 https://bugzilla.gnome.org//show_bug.cgi?id=747624
243 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
246 configure: bump automake requirement to 1.14 and autoconf to 2.69
247 This is only required for builds from git, people can still
248 build tarballs if they only have older autotools.
249 https://bugzilla.gnome.org//show_bug.cgi?id=747624
251 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
254 configure.ac: uncomment gettext version setup
255 Fixes autogen.sh. It would run autopoint, which would complain
256 that it could not find the gettext version in configure.ac.
257 https://bugzilla.gnome.org/show_bug.cgi?id=748058
259 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
261 * examples/test-video-rtx.c:
262 test-video-rtx: set exact payload type to PCMA payloader
263 Setting wrong payload type causes failure to do retransmission through audio stream
264 https://bugzilla.gnome.org/show_bug.cgi?id=747839
266 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
268 * gst/rtsp-server/rtsp-media.c:
269 * gst/rtsp-server/rtsp-stream.c:
270 * gst/rtsp-server/rtsp-stream.h:
271 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
272 Because of duplicated g_signal_connect for request-aux-sender signal,
273 wrong stream pointer is passed to the signal handler.
274 Instead of passing each stream, pass stream array and get the relevant stream.
275 https://bugzilla.gnome.org/show_bug.cgi?id=747839
277 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
281 Update autogen.sh to latest version from common
282 Fixes build after aclocal_check etc. helpers have been removed.
284 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
287 Automatic update of common submodule
288 From bc76a8b to c8fb372
290 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
292 * gst/rtsp-server/rtsp-stream.c:
293 rtsp-stream: Limit the queues to 1 buffer
294 We only need them to be able to pre-roll, queueing up more data here
295 is only going to harm latency and memory usage.
297 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
299 * gst/rtsp-server/rtsp-stream.c:
300 rtsp-stream: Update comment and ASCII art to the latest code
301 We have a queue in front of the udpsink too to prevent the pipeline from
304 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
306 * gst/rtsp-server/rtsp-stream.c:
307 rtsp-media: Properly return first rtptime
308 Instead we where returning first GstBuffer timestamp. This would result
309 in clock skew and unwanted behaviour in RTSP playback.
310 https://bugzilla.gnome.org/show_bug.cgi?id=746479
312 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
314 * gst/rtsp-server/rtsp-stream.c:
315 rtsp-stream: Don't leave buffer mapped
316 If the seq is NULL, the RTP buffer was left mapped. We should always
319 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
324 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
326 * gst/rtsp-server/rtsp-media-factory.c:
327 * tests/check/gst/client.c:
328 Fix double semicolons
330 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
332 * gst/rtsp-server/rtsp-stream.c:
333 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
334 This gives more accurate values than asking the payloader. There might be
335 queueing happening between the payloader and the sink.
336 https://bugzilla.gnome.org/show_bug.cgi?id=745704
338 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
340 * gst/rtsp-server/rtsp-media.c:
341 rtsp-media: Don't seek for PLAY if the position will not change
342 https://bugzilla.gnome.org/show_bug.cgi?id=745704
344 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
346 * gst/rtsp-server/rtsp-media.c:
347 rtsp-media: Don't include payload type in the caps for framesize
348 When the sdp media attribute framesize are converted to caps
349 the <payload> should not be included.
350 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
351 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
353 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
355 * gst/rtsp-server/rtsp-sdp.c:
356 rtsp-sdp: add payload type to the sdp framesize attribute
357 The sdp framesize attribute is desribed in RFC6064. It is specified
358 for payloading of H263 and has the following form
359 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
360 should be added to the caps in a payloader and the <payload type> should
361 be added by the rtsp-server.
362 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
364 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
366 * examples/test-uri.c:
367 examples: test-uri: fix tainted variable
368 Insignificant but this keeps Coverity happy.
371 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
373 * examples/.gitignore:
374 * examples/Makefile.am:
375 * examples/test-netclock-client.c:
376 * examples/test-netclock.c:
377 examples: Add a simple example of network synch for live streams.
378 An example server and client that works for synchronising live streams
379 only - as it can't support pause/play.
381 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
383 * gst/rtsp-server/rtsp-media-factory.c:
384 * gst/rtsp-server/rtsp-media-factory.h:
385 rtsp-media-factory: Add functions to set/get the media gtype
386 Allow specifying the GType of a GstRtspMedia subclass to create
387 as a simpler way to get the factory to create a custom
388 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
390 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
392 * gst/rtsp-server/rtsp-media.c:
393 rtsp-media: fix double unlock in _get_buffer_size()
394 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
395 because of double g_mutex_unlock () usage.
396 https://bugzilla.gnome.org/show_bug.cgi?id=745434
398 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
400 * gst/rtsp-server/rtsp-session-pool.c:
401 * gst/rtsp-server/rtsp-session.c:
402 * gst/rtsp-server/rtsp-session.h:
403 rtsp-session: Use monotonic time for RTSP session timeout
404 Changed RTSP session timeout handling to monotonic time
405 and deprecating the API for current system time.
406 This fixes timeouts when the system time changes.
407 https://bugzilla.gnome.org/show_bug.cgi?id=743346
409 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
411 * gst/rtsp-server/rtsp-client.c:
412 * gst/rtsp-server/rtsp-media.c:
413 rtsp-client: Only error out in PLAY if seeking actually failed
414 If the media was just not seekable, we continue from whatever position we are
415 and let the client decide if that is what is wanted or not.
416 Only if the actual seek failed, we can't really recover and should error out.
418 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
420 * gst/rtsp-server/rtsp-stream.c:
421 rtsp-stream: Add necessary queues between tee and multiudpsink
422 https://bugzilla.gnome.org/show_bug.cgi?id=744379
424 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
426 * gst/rtsp-server/rtsp-client.c:
427 * gst/rtsp-server/rtsp-media.c:
428 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
429 Instead error out properly the same way as if the SEEKING query already
432 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
434 * gst/rtsp-server/rtsp-stream.h:
435 rtsp-stream: minor code formatting fix
437 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
439 * gst/rtsp-server/rtsp-media.c:
440 rtsp-media: fix logic for collect_streams
441 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
442 all streams it knows if it got any, and can check if the transport mode is OK.
445 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
447 * gst/rtsp-server/rtsp-media.c:
448 rtsp-media: Don't set the transport mode based on what elements we find
449 Just print a warning if the one that was set before disagrees with what
450 elements we found. It must already be set to something before as this
451 function is called after we received the SDP from ANNOUNCE in RECORD mode,
452 and we would reject ANNOUNCE if the RECORD flag was not set.
454 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
456 * tests/check/gst/rtspserver.c:
457 tests: rtspserver: rename shadowed variable
458 We have two different 'sink' variables here,
459 rename one of them for clarity.
461 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
463 * gst/rtsp-server/rtsp-client.c:
464 rtsp-client: fix awkward if clause
466 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
468 * examples/test-uri.c:
469 examples: test-uri: improve uri argument handling and accept file names
470 Print an error if the argument passed is not a URI and can't
471 be converted into one, or no arguments have been provided.
473 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
475 * examples/test-uri.c:
476 examples: test-uri: don't remove mount point after 10 seconds
477 It's very irritating when trying to test stuff repeatedly
478 and serves no real purpose other than showing that it can
481 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
483 * examples/.gitignore:
484 examples: add new test-record to .gitignore
486 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
488 * examples/test-record.c:
489 * gst/rtsp-server/rtsp-client.c:
490 * gst/rtsp-server/rtsp-media-factory.c:
491 * gst/rtsp-server/rtsp-media-factory.h:
492 * gst/rtsp-server/rtsp-media.c:
493 * gst/rtsp-server/rtsp-media.h:
494 * tests/check/gst/rtspserver.c:
495 rtsp-media: Use flags to distinguish between PLAY and RECORD media
497 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
499 * examples/test-record.c:
500 test-record: Set latency for playback-style example to 2s instead of 200ms
502 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
504 * tests/check/gst/rtspserver.c:
505 tests: add some unit tests for ANNOUNCE and RECORD
506 https://bugzilla.gnome.org/show_bug.cgi?id=743175
508 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
510 * gst/rtsp-server/rtsp-client.c:
511 rtsp-client: fix a couple of leaks in handle_announce
513 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
515 * gst/rtsp-server/rtsp-media-factory.c:
516 * gst/rtsp-server/rtsp-media-factory.h:
517 * gst/rtsp-server/rtsp-media.c:
518 * gst/rtsp-server/rtsp-media.h:
519 rtsp-media: Expose latency setting for setting the rtpbin latency
521 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
523 * examples/test-record.c:
524 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
526 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
528 * gst/rtsp-server/rtsp-stream.c:
529 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
531 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
533 * examples/Makefile.am:
534 * examples/test-record.c:
535 * gst/rtsp-server/rtsp-client.c:
536 * gst/rtsp-server/rtsp-client.h:
537 * gst/rtsp-server/rtsp-media-factory.c:
538 * gst/rtsp-server/rtsp-media-factory.h:
539 * gst/rtsp-server/rtsp-media.c:
540 * gst/rtsp-server/rtsp-media.h:
541 * gst/rtsp-server/rtsp-session-media.c:
542 * gst/rtsp-server/rtsp-stream.c:
543 * gst/rtsp-server/rtsp-stream.h:
544 Add initial support for RECORD
545 We currently only support media that is RECORD or PLAY only, not both at once.
546 https://bugzilla.gnome.org/show_bug.cgi?id=743175
548 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
550 * gst/rtsp-server/rtsp-stream.c:
551 rtsp-stream: RTCP and RTP transport cache cookies seperated
552 RTCP packets were not sent because the same tr_cache_cookie was used for
553 both RTP and RTCP. So only one of the tr_cache lists were populated
554 depending on which one was sent first. If the tr_cache list is not
555 populated then no packets can be sent. Most often this happened to be
556 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
557 resulted in both the tr_cache_lists to be populated regardless of which
559 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
561 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
563 * gst/rtsp-server/rtsp-stream.c:
564 rtsp-stream: fix false compiler warning
565 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
567 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
569 * gst/rtsp-server/rtsp-client.c:
570 rtsp-client: log interleaved data received
572 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
574 * gst/rtsp-server/rtsp-client.c:
575 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
577 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
579 * gst/rtsp-server/rtsp-client.c:
580 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
582 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
584 * gst/rtsp-server/rtsp-client.c:
585 rtsp-client: Use a random session ID in the SDP
586 RFC4566 Section 5.2 says that it should make the username, session id,
587 nettype, addrtype and unicast address tuple globally unique. Always using
588 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
589 Instead let's create a 64 bit random number, which at least brings us
590 closer to the goal of global uniqueness.
591 https://tools.ietf.org/html/rfc4566#section-5.2
593 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
595 * examples/test-launch.c:
596 * examples/test-mp4.c:
597 * examples/test-ogg.c:
598 * examples/test-uri.c:
599 examples: Don't call gst_init() and gst_get_option_group()
600 The latter calls the former at the appropriate time.
602 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
604 * gst/rtsp-server/rtsp-client.c:
605 rtsp-client: Drop trailing \0 of RTSP DATA messages
606 We add a trailing \0 in GstRTSPConnection to make parsing of
607 string message bodies easier (e.g. the SDP from DESCRIBE) but
608 for actual data this means we have to drop it or otherwise
611 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
613 * gst/rtsp-server/rtsp-stream.c:
614 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
615 Fixes crash when two threads access handle_new_sample() at the same
616 time, one for RTP, one for RTCP.
617 Otherwise, when iterating over the transports cache, it might be modified by
618 another thread at the same time if the transports cookie has changed.
619 https://bugzilla.gnome.org/show_bug.cgi?id=742954
621 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
623 * gst/rtsp-server/rtsp-stream.c:
624 rtsp-stream: Set format=TIME on our app sources for TCP
626 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
628 * gst/rtsp-server/rtsp-session-pool.c:
629 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
630 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
631 RFC 2326 states that session IDs may consist of alphanumeric as well as
632 the safe characters $-_.+ -- N.B. the percent character is not allowed.
633 Previously the session ID was URI-escaped, this meant that any character
634 which was not alphanumeric or any of the characters +-._~ would be
635 percent encoded. While the RFC (surprisingly) mentions that linear white
636 space in session IDs should be URI-escaped, it does not say anything
637 about other characters. Moreover no white space is allowed in the
638 session ID. Finally the percent character which is the result of
639 URI-escaping is not allowed in a session ID.
640 So there is no reason to do any URI-escaping, and now it is removed.
641 https://bugzilla.gnome.org/show_bug.cgi?id=742869
643 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
646 Automatic update of common submodule
647 From f2c6b95 to bc76a8b
649 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
652 Fix 'make check' from top-level directory
654 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
656 * examples/test-launch.c:
657 * examples/test-mp4.c:
658 * examples/test-ogg.c:
659 * examples/test-uri.c:
660 examples: Add command-line parsing and take a 'port' argument
661 This allows users to run multiple servers on different ports for testing.
662 Only done for examples that actually take arguments and hence are capable of
663 outputting different streams for each instance on each port.
664 https://bugzilla.gnome.org/show_bug.cgi?id=742115
666 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
668 * gst/rtsp-server/rtsp-client.c:
669 * gst/rtsp-server/rtsp-client.h:
670 rtsp-client: Add a send_message default signal handler
671 This allows subclasses to easily hook into the response sending
672 mechanism without doing everything from a signal, which seems
673 awkward from subclasses.
675 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
678 Automatic update of common submodule
679 From ef1ffdc to f2c6b95
681 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
685 configure: add --disable-examples switch
686 https://bugzilla.gnome.org/show_bug.cgi?id=741678
688 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
690 * examples/.gitignore:
691 * examples/Makefile.am:
692 * examples/test-video-rtx.c:
693 examples: add a retransmisison example implementing RFC4588
694 Currently only SSRC-multiplexed rtx streams are supported
696 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
698 * gst/rtsp-server/rtsp-stream.c:
699 rtsp-stream: Fix some minor memory leaks
701 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
703 * gst/rtsp-server/rtsp-media.c:
704 rtsp-media: Some minor cleanup
706 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
708 * gst/rtsp-server/rtsp-stream.c:
709 rtsp-stream: Fix compiler warnings
710 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
711 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
713 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
714 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
717 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
719 * docs/libs/gst-rtsp-server-sections.txt:
720 * gst/rtsp-server/rtsp-media-factory.c:
721 * gst/rtsp-server/rtsp-media-factory.h:
722 * gst/rtsp-server/rtsp-media.c:
723 * gst/rtsp-server/rtsp-media.h:
724 * gst/rtsp-server/rtsp-sdp.c:
725 * gst/rtsp-server/rtsp-stream.c:
726 * gst/rtsp-server/rtsp-stream.h:
727 media: implement ssrc-multiplexed retransmission support
728 based off RFC 4588 and the server-rtpaux example in -good
730 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
732 * gst/rtsp-server/rtsp-client.c:
733 * gst/rtsp-server/rtsp-stream-transport.c:
734 * gst/rtsp-server/rtsp-stream.c:
735 rtsp: Ref transports in hash table.
736 Also ref streams for transports.
737 This solves a crash when reciving a rtcp after teardown but before
739 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
741 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
744 Automatic update of common submodule
745 From 7bb2bce to ef1ffdc
747 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
749 * gst/rtsp-server/rtsp-client.c:
750 client: refactor cleanup of cached media
752 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
754 * tests/check/gst/client.c:
756 The session leak is now fixed, lets remove those FIXME comments.
758 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
760 * tests/check/gst/rtspserver.c:
761 tests: Test to setup two sessions on one connection
762 https://bugzilla.gnome.org/show_bug.cgi?id=739112
764 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
766 * tests/check/gst/rtspserver.c:
767 tests: Test setup with tcp transport
768 https://bugzilla.gnome.org/show_bug.cgi?id=739112
770 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
772 * gst/rtsp-server/rtsp-client.c:
773 client: Configure transport after creating session media
774 The default implementation of configure_client_transport() in
775 rtsp-client uses the session media when it chooses channels for
777 https://bugzilla.gnome.org/show_bug.cgi?id=739112
779 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
781 * gst/rtsp-server/rtsp-client.c:
782 * gst/rtsp-server/rtsp-session-media.c:
783 client: Stop caching media in client when doing setup
784 If the media has been managed by a session media, it should not be
785 cached in the client any longer. The GstRTSPSessionMedia object is now
786 responsible for unpreparing the GstRTSPMedia object using
787 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
789 https://bugzilla.gnome.org/show_bug.cgi?id=739112
791 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
793 * gst/rtsp-server/rtsp-stream.c:
794 rtsp-stream: unref srtp decoder when leaving bin
795 https://bugzilla.gnome.org/show_bug.cgi?id=739481
797 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
799 * gst/rtsp-server/rtsp-client.c:
800 rtsp-client: mikey memory leaks
801 https://bugzilla.gnome.org/show_bug.cgi?id=739383
803 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
806 Automatic update of common submodule
807 From 84d06cd to 7bb2bce
809 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
812 Parallelise 'make check-valgrind'
814 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
817 Automatic update of common submodule
818 From a8c8939 to 84d06cd
820 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
823 Automatic update of common submodule
824 From 36388a1 to a8c8939
826 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
828 * gst/rtsp-server/rtsp-media.c:
829 rtsp-media: deactivate media when shutting down from paused
830 This was only done when going directly from playing.
831 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
833 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
835 * gst/rtsp-server/rtsp-client.c:
836 * gst/rtsp-server/rtsp-context.h:
837 rtsp-client: add stream transport to context
838 We add the stream transport to the context so we can get the configured
839 client stream transport in the setup request signal.
840 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
842 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
844 * gst/rtsp-server/rtsp-stream.c:
845 stream: release lock even not all transports have been removed
846 We don't want to keep the lock even we return FALSE because not all the
847 transports have been removed. This could lead into a deadlock.
848 https://bugzilla.gnome.org/show_bug.cgi?id=737797
850 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
852 * gst/rtsp-server/rtsp-sdp.c:
853 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
854 These were renamed in GstRTPBasePayload in 1.0
856 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
858 * gst/rtsp-server/rtsp-client.c:
859 client: set session media to NULL without the lock
860 We need to set session medias to NULL without the client lock otherwise
861 we can end up in a deadlock if another thread is waiting for the lock
862 and media unprepare is also waiting for that thread to end.
863 https://bugzilla.gnome.org/show_bug.cgi?id=737690
865 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
867 * gst/rtsp-server/rtsp-media.c:
868 rtsp-media: Set state to UNPREPARING in all cases
870 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
872 * gst/rtsp-server/rtsp-media.c:
873 media: set state to unpreparing when unprepare is initiated
874 https://bugzilla.gnome.org/show_bug.cgi?id=737675
876 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
878 * gst/rtsp-server/rtsp-client.c:
879 rtsp-client: Remove backlog limit while processings requests
880 If the backlog limit is kept two cases of deadlocks may be
881 encountered when streaming over TCP. Without the backlog
882 limit this deadlocks can not happen, at the expence of
884 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
886 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
888 * gst/rtsp-server/rtsp-client.c:
889 rtsp-client: do not free main context before rtsp watch
890 https://bugzilla.gnome.org/show_bug.cgi?id=737110
892 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
894 * tests/check/gst/rtspserver.c:
895 tests: Extend unit test timeout to accomodate for valgrind
896 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
898 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
900 * gst/rtsp-server/rtsp-client.c:
901 * gst/rtsp-server/rtsp-session.c:
902 * gst/rtsp-server/rtsp-stream-transport.c:
903 rtsp-*: Treat sending packets to clients as keepalive
904 As long as gst-rtsp-server can successfully send RTP/RTCP data to
905 clients then the client must be reading. This change makes the server
906 timeout the connection if the client stops reading.
907 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
909 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
911 * gst/rtsp-server/rtsp-client.c:
912 rtsp-client: Allow backlog to grow while expiring session
913 Allow the send backlog in the RTSP watch to grow to unlimited size while
914 attempting to bring the media pipeline to NULL due to a session
915 expiring. Without this change the appsink element cannot change state
916 because it is blocked while rendering data in the new_sample callback.
917 This callback will block until it has successfully put the data into the
918 send backlog. There is a chance that the send backlog is full at this
919 point which means that the callback may block for a long time, possibly
920 forever. Therefore the media pipeline may also be prevented from
921 changing state for a long time.
922 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
924 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
926 * gst/rtsp-server/rtsp-client.c:
927 rtsp-client: Make old compilers happy
928 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
929 Just in case that guint8 doesn't fit in a pointer. Just in case ...
931 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
933 * gst/rtsp-server/rtsp-client.c:
934 client: raise the backlog limits before pausing
935 We need to raise the backlog limits before pausing the pipeline or else
936 the appsink might be blocking in the render method in wait_backlog() and
937 we would deadlock waiting for paused.
938 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
940 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
942 * gst/rtsp-server/rtsp-client.c:
943 client: make define for the WATCH_BACKLOG
944 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
946 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
948 * gst/rtsp-server/rtsp-client.c:
949 client: simplify session transport handling
950 link/unlink of the transport in a session was done to keep track of all
951 TCP transports and to send RTP/RTCP data to the streams. We can simplify
952 that by putting all the TCP transports in a hashtable indexed with the
954 We also don't need to link/unlink the transports when we pause/resume
955 the streams. The same effect is already achieved when we pause/play the
956 media. Indeed, when we pause the media, the transport is removed from
957 the media and the callbacks will not be called anymore.
958 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
960 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
962 * gst/rtsp-server/rtsp-stream-transport.c:
963 * gst/rtsp-server/rtsp-stream-transport.h:
964 stream-transport: make method to handle received data
965 Make a method to handle the data received on a channel. It sends the
966 data to the stream of the transport on the RTP or RTCP pads based on
969 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
971 * examples/test-mp4.c:
972 test: add example of dumping RTCP reports
974 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
976 * gst/rtsp-server/rtsp-media.c:
977 * gst/rtsp-server/rtsp-stream.c:
978 * gst/rtsp-server/rtsp-stream.h:
979 rtsp-media: Make sure that sequence numbers are monotonic after pause
980 The sequence number is not monotonic for RTP packets after pause. The
981 reason is basepayloader generates a randon sequence number when the
982 pipeline goes from ready to pause. With this fix generation of sequence
983 number will be monotonic when going from pause to play request.
984 https://bugzilla.gnome.org/show_bug.cgi?id=736017
986 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
988 * gst/rtsp-server/rtsp-client.c:
989 rtsp-client: Protect saved clients watch with a mutex
990 Fixes a crash when close() is called while merging clients
991 in handle_tunnel(). In that case close() would destroy the
992 watch while it is still being used in handle_tunnel().
993 https://bugzilla.gnome.org/show_bug.cgi?id=735570
995 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
997 * gst/rtsp-server/rtsp-stream.c:
998 rtsp-stream: Remove the multicast group udp sources when removing from the bin
1000 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1002 * gst/rtsp-server/rtsp-media.c:
1003 * gst/rtsp-server/rtsp-stream.c:
1004 * gst/rtsp-server/rtsp-stream.h:
1005 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
1006 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
1007 seeking and will always continue counting the time. This leads to
1008 the NPT after a backwards seek to be something completely different
1009 to the actual seek position.
1010 https://bugzilla.gnome.org/show_bug.cgi?id=732644
1012 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
1014 * examples/test-appsrc.c:
1015 examples: fix another reference leak
1016 gst_rtsp_media_get_element() returns a new ref.
1018 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1020 * examples/test-appsrc.c:
1021 examples: unref element after usage
1022 gst_bin_get_by_name_recurse_up() returns an element
1023 reference that must be unreffed after usage.
1024 https://bugzilla.gnome.org/show_bug.cgi?id=734546
1026 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
1028 * gst/rtsp-server/rtsp-media.c:
1029 signals: Fix copy-pasto in target-state signal offset
1031 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
1035 Makefile: Add usage of build-checks step
1036 Allows building checks without running them
1038 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
1040 * gst/rtsp-server/rtsp-stream.c:
1041 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
1042 When a UDP multicast transport is used it is expected that the server listens
1043 for RTP and RTCP packets on the multicast group with the corresponding port.
1044 Without this we will never get RTCP packets from clients in multicast mode.
1045 https://bugzilla.gnome.org/show_bug.cgi?id=732238
1047 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1052 === release 1.4.0 ===
1054 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1060 * gst-rtsp-server.doap:
1063 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
1065 * gst/rtsp-server/rtsp-media.h:
1066 media: correct misspelled words in description
1067 https://bugzilla.gnome.org/show_bug.cgi?id=733244
1069 === release 1.3.91 ===
1071 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1077 * gst-rtsp-server.doap:
1080 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
1082 * docs/libs/gst-rtsp-server-sections.txt:
1085 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
1087 * gst/rtsp-server/rtsp-server.c:
1088 server: implement client REMOVE filter
1090 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
1092 * gst/rtsp-server/rtsp-client.c:
1093 * gst/rtsp-server/rtsp-client.h:
1094 client: expose _close() method
1095 Expose a previously internal close method to close the client
1098 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
1100 * gst/rtsp-server/rtsp-session-pool.c:
1101 session-pool: signal session-removed outside of the lock
1102 Release the lock before emiting the session-removed signal.
1104 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
1106 * gst/rtsp-server/rtsp-client.c:
1107 * gst/rtsp-server/rtsp-server.c:
1108 * gst/rtsp-server/rtsp-session-pool.c:
1109 * gst/rtsp-server/rtsp-session.c:
1110 * gst/rtsp-server/rtsp-stream.c:
1111 filter: Release lock in filter functions
1112 Release the object lock before calling the filter functions. We need to
1113 keep a cookie to detect when the list changed during the filter
1114 callback. We also keep a hashtable to make sure we only call the filter
1115 function once for each object in case of concurrent modification.
1116 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
1118 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
1120 * gst/rtsp-server/rtsp-client.c:
1121 client: check if watch is set in handle_teardown()
1122 The unit tests run without a watch
1124 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1126 * tests/check/gst/client.c:
1127 client tests: send teardown to cleanup session
1129 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
1131 * tests/check/gst/rtspserver.c:
1132 server tests: send teardown to cleanup session
1134 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1136 * gst/rtsp-server/rtsp-client.c:
1137 client: keep ref to client for the session removed handler
1138 This extra ref will be dropped when all client sessions have been
1139 removed. A session is removed when a client sends teardown, closes its
1140 endpoint of the TCP connection or the sessions expires.
1141 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1143 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
1145 * gst/rtsp-server/rtsp-client.c:
1146 * gst/rtsp-server/rtsp-session.c:
1147 * tests/check/gst/client.c:
1148 client: manage media in session as a last step
1149 Once we manage a media in a session, we can't unmanage it anymore
1150 without destroying it. Therefore, first check everything before we
1151 manage the media, otherwise if something is wrong we have no way to
1153 If we created a new session and something went wrong, remove the session
1154 again. Fixes a leak in the unit test.
1156 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1158 * examples/test-mp4.c:
1159 * examples/test-ogg.c:
1160 examples: print 'stream ready at url' for mp4 and ogg example
1162 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
1164 * gst/rtsp-server/rtsp-client.c:
1165 * gst/rtsp-server/rtsp-sdp.c:
1166 rtsp: fix for MIKEY api change
1168 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
1170 * gst/rtsp-server/rtsp-client.c:
1171 client: free watch context only once
1172 The watch context is freed when the source is destroyed. Avoids
1173 a CRITICAL when we try to unref the context twice.
1175 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
1177 * gst/rtsp-server/rtsp-client.c:
1180 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
1182 * gst/rtsp-server/rtsp-client.c:
1183 client: protect sessions with lock
1184 Protect the list of sessions with the lock.
1185 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1187 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
1189 * gst/rtsp-server/rtsp-client.c:
1190 Client: keep a ref to the session
1191 Don't just keep a weak ref to the session objects but use a hard ref. We
1192 will be notified when a session is removed from the pool (expired) with
1193 the new session-removed signal.
1194 Don't automatically close the RTSP connection when all the sessions of
1195 a client are removed, a client can continue to operate and it can create
1196 a new session if it wants. If you want to remove the client from the
1197 server, you have to use gst_rtsp_server_client_filter() now.
1198 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
1199 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
1201 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
1203 * gst/rtsp-server/rtsp-session-pool.c:
1204 * gst/rtsp-server/rtsp-session-pool.h:
1205 session-pool: add session-removed signal
1206 Add a signal to be notified when a session is removed from the pool.
1208 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
1210 * gst/rtsp-server/Makefile.am:
1211 * gst/rtsp-server/rtsp-server.h:
1212 Make rtsp-server.h a single-include header, use it for G-I
1213 https://bugzilla.gnome.org/show_bug.cgi?id=732411
1215 === release 1.3.90 ===
1217 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1223 * gst-rtsp-server.doap:
1226 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
1228 * gst/rtsp-server/rtsp-stream.c:
1229 stream: crypto can be NULL
1231 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
1233 * gst/rtsp-server/rtsp-client.c:
1234 * gst/rtsp-server/rtsp-media.c:
1235 * gst/rtsp-server/rtsp-mount-points.c:
1236 introspection: add missing allow-none annotations
1237 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1239 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
1241 * gst/rtsp-server/rtsp-address-pool.c:
1242 * gst/rtsp-server/rtsp-media.c:
1243 * gst/rtsp-server/rtsp-session-media.c:
1244 * gst/rtsp-server/rtsp-session-pool.c:
1245 * gst/rtsp-server/rtsp-stream-transport.c:
1246 * gst/rtsp-server/rtsp-stream.c:
1247 * gst/rtsp-server/rtsp-token.c:
1248 introspection: add (nullable) annotations to return values
1249 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1251 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
1253 * gst/rtsp-server/rtsp-client.c:
1254 * gst/rtsp-server/rtsp-stream.c:
1255 gi: improve annotations
1256 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
1258 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
1260 * gst/rtsp-server/rtsp-client.c:
1261 * gst/rtsp-server/rtsp-media-factory.c:
1262 * gst/rtsp-server/rtsp-media.c:
1263 * gst/rtsp-server/rtsp-server.c:
1264 signals: use generic marshal function
1265 Use the generic C marshal function.
1266 Use more explicit type instead of G_TYPE_POINTER
1268 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
1270 * gst/rtsp-server/rtsp-context.h:
1271 context: add type macro
1273 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
1275 * gst/rtsp-server/rtsp-client.c:
1276 * gst/rtsp-server/rtsp-sdp.c:
1277 * gst/rtsp-server/rtsp-sdp.h:
1278 sdp: hide key length defines
1279 They don't have a namespace.
1281 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1286 === release 1.3.3 ===
1288 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
1294 * gst-rtsp-server.doap:
1297 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1299 * gst/rtsp-server/rtsp-client.c:
1300 * gst/rtsp-server/rtsp-sdp.c:
1301 * gst/rtsp-server/rtsp-sdp.h:
1302 mikey: add different key length parameters
1303 Add encryption and authentication key length parameters to MIKEY. For
1304 the encoders, the key lengths are obtained from the cipher and auth
1305 algorithms set in the caps. For the decoders, they are obtained while
1306 parsing the key management from the client.
1307 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
1309 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
1311 * tests/check/gst/stream.c:
1312 stream tests: Make sure we get right multicast address from stream
1313 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
1315 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
1317 * gst/rtsp-server/rtsp-client.c:
1318 client: ref the context until rtsp watch is alive
1319 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
1321 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
1323 * gst/rtsp-server/rtsp-client.c:
1324 client: Destroy the rtsp watch after connection close
1326 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
1328 * gst/rtsp-server/rtsp-media.c:
1329 media: fix confusing comment
1331 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
1333 * gst/rtsp-server/rtsp-session.c:
1334 rtsp-session: Timeout in header.
1335 Adding the possbilty to always have timout in header.
1336 This is configurabe with setting "timeout-always-visible".
1337 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
1339 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
1344 === release 1.3.2 ===
1346 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
1353 * gst-rtsp-server.doap:
1356 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1359 Automatic update of common submodule
1360 From 211fa5f to 1f5d3c3
1362 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
1364 * gst/rtsp-server/rtsp-client.c:
1365 client: store TCP ports in transport
1366 Store the TCP ports in the transport when we are doing RTSP over TCP.
1367 This way, we can easily get to the ports from the transport.
1368 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
1370 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1372 * gst/rtsp-server/rtsp-stream.c:
1373 stream: add signals for new RTP/RTCP encoders
1374 New signals to allow the user to configure the dynamically created
1376 https://bugzilla.gnome.org/show_bug.cgi?id=730228
1378 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1380 * gst/rtsp-server/rtsp-media.c:
1381 * gst/rtsp-server/rtsp-media.h:
1382 media: Make suspend()/unsuspend() virtual
1383 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
1385 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1387 * gst/rtsp-server/rtsp-client.c:
1388 client: fix send-message signal marshaller
1389 Use generic marshalling for the send-message signal. It has
1390 two POINTER arguments, not just one.
1391 https://bugzilla.gnome.org/show_bug.cgi?id=729900
1393 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
1395 * tests/check/gst/media.c:
1396 tests: add and remove pads only once
1397 In this test we simulate a dynamic pad by watching the caps event.
1398 Because of renegotiation in the base payloader now, this caps is sent
1399 multiple times but we can only deal with 1 invocation, use a variable to
1400 only 'add and remove' the pad once.
1402 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1404 * tests/check/gst/rtspserver.c:
1405 tests: add unit test for correct handling of Require headers
1406 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1408 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1410 * gst/rtsp-server/rtsp-client.c:
1411 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
1412 Servers must handle Require headers and must report a failure
1413 if they don't handle any of the Required options, see RFC 2326,
1414 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
1415 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1417 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1422 === release 1.3.1 ===
1424 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1430 * gst-rtsp-server.doap:
1433 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
1436 Automatic update of common submodule
1437 From bcb1518 to 211fa5f
1439 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
1444 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1446 * tests/check/gst/sessionmedia.c:
1447 tests: fix memory leak in sessionmedia unit test
1449 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
1451 * gst/rtsp-server/rtsp-client.c:
1452 client: emit a signal before sending a message
1453 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
1455 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
1457 * gst/rtsp-server/rtsp-client.c:
1458 client: pass context to send_message
1459 Pass the current context to send_message, we will need it later.
1461 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
1463 * gst/rtsp-server/rtsp-client.c:
1464 client: fix typo in comment
1466 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
1468 * gst/rtsp-server/rtsp-media.c:
1469 media: Do not stop thread twice if default_prepare() fails
1471 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
1473 * gst/rtsp-server/rtsp-client.c:
1474 client: set the watch to flushing before going to NULL
1475 First set the watch to flushing so that we unblock any current and
1476 future attempt to send data on the watch, Then set the pipeline to
1478 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
1480 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
1482 * gst/rtsp-server/rtsp-session-pool.c:
1483 * tests/check/gst/sessionpool.c:
1484 rtsp-session-pool: Fixes annotation
1485 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
1486 in the sessionpool test.
1487 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
1489 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
1491 * gst/rtsp-server/rtsp-media.c:
1492 * gst/rtsp-server/rtsp-media.h:
1493 media: make media_prepare virtual
1494 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
1496 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1498 * gst/rtsp-server/rtsp-media.c:
1499 * tests/check/gst/media.c:
1500 media: stop the thread in more error cases
1502 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
1504 * gst/rtsp-server/rtsp-media.c:
1505 * tests/check/gst/media.c:
1506 media: allow NULL as the thread
1507 Use the default context whan passing a NULL thread.
1509 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1511 * gst/rtsp-server/rtsp-client.c:
1512 rtsp-client: indent cleanup
1513 Coverity was moaning about unreachable code, and I think it was just
1514 confused by { being before the label. We'll see if it pops up again.
1517 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
1519 * gst/rtsp-server/rtsp-client.c:
1520 * gst/rtsp-server/rtsp-media.c:
1521 client: Add drop-backlog property
1522 When we have too many messages queued for a client (currently hardcoded
1523 to 100) we overflow and drop the messages. Add a drop-backlog property
1524 to control this behaviour. Setting this property to FALSE will retry
1525 to send the messages to the client by waiting for more room in the
1527 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
1529 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
1531 * gst/rtsp-server/rtsp-client.c:
1532 client: support for POST before GET when setting up a tunnel
1534 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
1536 * gst/rtsp-server/rtsp-client.c:
1537 client: remove watch of the second client after http tunnel setup
1538 The second client will be freed after the HTTP tunnel has been set up.
1539 Make sure it's RTSP watch is never dispatched again.
1540 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
1542 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
1544 * gst/rtsp-server/rtsp-media.c:
1545 * tests/check/gst/media.c:
1546 media: Make media_prepare() fail if port allocation fails
1547 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
1549 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
1551 * tests/check/gst/media.c:
1552 media test: cleanup the thread pool in tests
1554 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
1556 * gst/rtsp-server/rtsp-media.c:
1557 * tests/check/gst/media.c:
1558 rtsp-media: Unblock blocked streams in unprepare
1559 The streams will be blocked when a live media is prepared.
1560 The streams should be unblocked in gst_rtsp_media_unprepare.
1561 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
1563 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
1565 * gst/rtsp-server/rtsp-media.c:
1566 media: release the state lock when going to NULL
1567 Set our state to UNPREPARING and release the state-lock before
1568 setting the pipeline to the NULL state. This way, any pad-added
1569 callback will be able to take the state-lock and check that we are now
1570 unpreparing instead of deadlocking.
1571 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
1573 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
1575 * gst/rtsp-server/rtsp-media.c:
1576 media: protect status with lock
1577 Make sure we only update the status with the lock.
1579 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
1581 * gst/rtsp-server/rtsp-client.c:
1582 * gst/rtsp-server/rtsp-sdp.c:
1583 rtsp: update for MIKEY API changes
1585 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
1587 * gst/rtsp-server/rtsp-client.c:
1588 client: parse the mikey response from the client
1589 Parse the mikey response from the client and update the policy for
1592 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
1594 * gst/rtsp-server/rtsp-stream.c:
1595 * gst/rtsp-server/rtsp-stream.h:
1596 stream: add method to set crypto info
1597 Make a method to configure the crypto information of a stream.
1598 Set udpsrc in READY instead of PAUSED so that we can configure caps
1601 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
1603 * gst/rtsp-server/rtsp-client.c:
1604 client: cleanup error paths
1606 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
1608 * gst/rtsp-server/rtsp-media.c:
1611 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
1613 * examples/test-video.c:
1614 test: enable SRTP only on RTSPS
1615 We only want to enable SRTP when doing rtsp over TLS so that we can
1616 exchange the keys in a secure way.
1618 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
1620 * examples/test-video.c:
1621 test: print an error on failure
1623 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
1626 * examples/test-video.c:
1627 * gst/rtsp-server/rtsp-sdp.c:
1628 * gst/rtsp-server/rtsp-stream.c:
1629 * tests/check/Makefile.am:
1630 stream: add SRTP support
1631 Install srtp encoder and decoder elements in rtpbin
1634 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1636 * tests/check/Makefile.am:
1637 * tests/check/gst/sessionpool.c:
1638 tests: Add unit tests for sessionpool
1639 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
1641 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1643 * tests/check/gst/threadpool.c:
1644 tests: Improve code coverage of rtsp-threadpool tests
1645 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
1647 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1649 * tests/check/gst/sessionmedia.c:
1650 tests: Improve code coverage for rtsp-session-media
1651 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
1653 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1655 gobject-introspection: Add annotations to support language bindings
1656 In addition a few cosmetic changes:
1657 * Adjust the order of arguments
1658 * Fix typo: occured -> occurred
1659 * Fix indentation after Return:-clauses
1660 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
1662 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1664 * gst/rtsp-server/rtsp-stream.c:
1665 rtsp-stream: Don't mix IPv4 and IPv6 addresses
1666 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
1668 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
1670 * gst/rtsp-server/rtsp-stream.c:
1671 stream: take caps after the session manager
1672 Take the caps for the SDP after they leave the rtpbin so that we can
1673 also get the properties added by rtpbin elements.
1675 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
1677 * gst/rtsp-server/rtsp-stream.c:
1678 stream: release lock while pushing out packets
1679 Keep a cache of the transports and use this to iterate the transport
1680 while pushing packets. This allows us to release the lock early.
1681 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
1683 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
1685 * gst/rtsp-server/rtsp-client.c:
1686 * gst/rtsp-server/rtsp-client.h:
1687 rtsp-client: vmethod for modifying tunnel GET response
1688 Add a vmethod tunnel_http_response where the response to the HTTP GET
1689 for tunneled connections can be modified.
1690 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
1692 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
1694 * gst/rtsp-server/rtsp-sdp.c:
1695 sdp: make 1 media line per profile
1696 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
1697 line in the SDP for each profile. The client is then supposed to pick
1698 one of the profiles in the SETUP request. Because the m= lines have the
1699 same pt, the client also knows that only 1 option is possible.
1701 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
1703 * gst/rtsp-server/rtsp-media-factory.c:
1704 * gst/rtsp-server/rtsp-media-factory.h:
1705 * gst/rtsp-server/rtsp-media.c:
1706 factory: add profile property and pass to media and streams
1708 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
1710 * examples/test-multicast.c:
1711 * gst/rtsp-server/rtsp-sdp.c:
1712 sdp: pass multicast connection for multicast-only stream
1713 Pass the multicast address of the stream in the connection info in the
1714 SDP so that clients try a multicast connection first.
1715 Only allow multicast connections in the test-multicast example. Also
1716 increase the TTL a little.
1718 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1721 .gitignore: Ignore gcov intermediate files
1722 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
1724 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
1726 * gst/rtsp-server/rtsp-stream.c:
1727 stream: release some locks in error cases
1729 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1731 docs: Enable and fix gtk-doc warnings
1732 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
1733 * addresspool/mediafactory: Add missing annotation colon
1734 * stream: Annotate return value
1735 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
1737 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1740 Automatic update of common submodule
1741 From fe1672e to bcb1518
1743 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
1746 Automatic update of common submodule
1747 From 1a07da9 to fe1672e
1749 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1751 * examples/Makefile.am:
1752 examples: use LDADD for libs instead of LDFLAGS
1754 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
1757 configure: make sure releases are in .doap file
1759 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
1761 * examples/test-cgroups.c:
1762 examples: test-cgroups: don't put code with side effects into g_assert()
1763 The g_assert() might get compiled out with the right
1764 compiler/preprocessor flags.
1766 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1768 * examples/.gitignore:
1769 examples: add cgroup test binary to .gitignore
1771 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
1773 * examples/test-cgroups.c:
1774 examples: fix cgroup test build
1775 Fixes build failure caused by compiler warning:
1776 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
1778 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
1781 .gitignore: ignore temp files created in the course of 'make check'
1783 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
1785 * gst/rtsp-server/rtsp-media.c:
1786 rtsp-media: don't loose frames handling new PLAY request
1787 If client supplied a range check if the range specifies the start point.
1788 If not, then do an accurate seek to the current position. If a start
1789 point was specified do do a key unit seek to make sure the streaming
1790 starts with decodeable frames.
1791 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
1793 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
1795 * gst/rtsp-server/rtsp-media.c:
1796 Revert "media: only flush when setting a new start position"
1797 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
1798 We need to do the flush in all cases, demuxer block currently for
1801 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
1803 * gst/rtsp-server/rtsp-media.c:
1804 media: only flush when setting a new start position
1805 Only flush the pipeline when we change the start position with
1807 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
1809 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
1811 * gst/rtsp-server/rtsp-stream.c:
1812 stream: set ttl-mc before adding the socket
1813 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
1814 never be set on socket.
1815 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
1817 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
1819 * gst/rtsp-server/rtsp-media.c:
1820 media: stop thread if media is already prepared
1821 in gst_rtsp_media_prepare() the thread is not used if media is already
1822 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
1824 https://bugzilla.gnome.org/show_bug.cgi?id=724182
1826 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
1829 build: Ship gst-rtsp-server.doap file
1831 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
1833 * tests/check/gst/rtspserver.c:
1834 tests: Fix another compiler warning with gcc
1836 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
1838 * gst/rtsp-server/rtsp-client.c:
1839 * gst/rtsp-server/rtsp-mount-points.c:
1840 * gst/rtsp-server/rtsp-stream.c:
1841 * tests/check/gst/client.c:
1842 rtsp-server: Fix lots of compiler warnings with clang
1844 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
1847 * gst-rtsp-server.doap:
1848 * tests/Makefile.am:
1849 configure: Synchronise with the configure scripts of the other modules
1851 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
1854 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
1856 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1858 * gst/rtsp-server/rtsp-media.c:
1859 * gst/rtsp-server/rtsp-stream.c:
1860 Revert "rtsp-server: support build against last stable release"
1861 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
1862 Let us require 1.2.3 now, which is going to be released in a few
1865 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
1867 * gst/rtsp-server/rtsp-session-media.c:
1868 * gst/rtsp-server/rtsp-stream-transport.c:
1869 session: improve RTP-Info
1870 Ignore streams that can't generate RTP-Info instead of failing.
1871 Don't return the empty string when all streams are unconfigured but
1872 return NULL so that we don't generate and empty RTP-Info header.
1873 Improve docs a little.
1875 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
1877 * gst/rtsp-server/rtsp-session-media.c:
1878 Don't free rtpinfo GString when it is NULL
1879 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
1881 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
1883 * gst/rtsp-server/rtsp-media.c:
1884 media: only set keyframe flag when modifying start
1885 Only set the keyframe flag when we modify the start position. The
1886 keyframe flag should probably be ignored when no change is requested but
1887 until we can claim this is all documented properly and all demuxer
1888 implement this, avoid setting the flag.
1889 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
1891 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
1893 * gst/rtsp-server/rtsp-thread-pool.c:
1894 thread-pool: Unref source after mainloop has quit to avoid races in GLib
1895 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
1897 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
1899 * gst/rtsp-server/rtsp-stream.c:
1900 stream: handle NULL seqnum and rtptime arguments
1902 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
1904 * gst/rtsp-server/rtsp-thread-pool.c:
1905 * tests/check/gst/threadpool.c:
1906 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
1907 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
1909 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
1911 * gst/rtsp-server/rtsp-stream.c:
1912 stream: add fallback for missing stats property
1913 Use a fallback when the payloader does not have a stats property
1914 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
1916 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
1919 Automatic update of common submodule
1920 From f7bc1c3 to 1a07da9
1922 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
1924 * gst/rtsp-server/rtsp-stream.c:
1925 stream: don't leak stats structure
1926 Don't leak the stats structure and deal with NULL stats.
1928 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
1930 * gst/rtsp-server/rtsp-stream.c:
1931 stream: Get rtpinfo properties atomically from payloader
1932 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
1934 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
1936 * gst/rtsp-server/rtsp-media.c:
1937 media: refactor state change functions and signals
1938 Make functions to set the target state and the pipeline state and emit
1939 the signals from those functions.
1941 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
1943 * gst/rtsp-server/rtsp-media.c:
1944 * gst/rtsp-server/rtsp-media.h:
1945 media: add signal to notify of pending state changes
1947 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1949 * gst/rtsp-server/rtsp-media.c:
1950 * gst/rtsp-server/rtsp-stream.c:
1951 rtsp-server: support build against last stable release
1952 Until 1.2.3 is out with the new get_type function and we
1955 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
1957 * gst/rtsp-server/rtsp-stream.c:
1958 stream: fix compilation
1960 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
1962 * gst/rtsp-server/rtsp-media.c:
1963 * gst/rtsp-server/rtsp-media.h:
1964 * gst/rtsp-server/rtsp-stream.c:
1965 * gst/rtsp-server/rtsp-stream.h:
1966 stream: add property to configure profiles
1968 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
1970 * gst/rtsp-server/rtsp-client.c:
1971 client: let stream check supported transport
1972 Delegate the check if a transport is allowed to the stream.
1973 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
1975 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
1977 * gst/rtsp-server/rtsp-stream.c:
1978 * gst/rtsp-server/rtsp-stream.h:
1979 stream: add method to check supported transport
1980 Add a method to check if a transport is supported
1982 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
1985 configure.ac: Only check for gstreamer-check, not check
1986 We include check in gstreamer-check since quite some time now.
1988 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
1990 * gst/rtsp-server/rtsp-session-media.c:
1991 * gst/rtsp-server/rtsp-stream-transport.c:
1992 * gst/rtsp-server/rtsp-stream.c:
1993 * gst/rtsp-server/rtsp-stream.h:
1994 stream: return clock-rate from get_rtpinfo
1995 And use it to correct the rtptime to the requested start-time.
1996 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
1998 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
2000 * gst/rtsp-server/rtsp-session-media.c:
2001 * gst/rtsp-server/rtsp-stream-transport.c:
2002 * gst/rtsp-server/rtsp-stream-transport.h:
2003 session-media: calculate start-time
2005 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
2007 * gst/rtsp-server/rtsp-stream-transport.c:
2008 * gst/rtsp-server/rtsp-stream.c:
2009 * gst/rtsp-server/rtsp-stream.h:
2010 stream: also return the running-time
2011 Return the running-time in the rtpinfo as well.
2013 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
2015 * gst/rtsp-server/rtsp-client.c:
2016 * gst/rtsp-server/rtsp-session-media.c:
2017 * gst/rtsp-server/rtsp-session-media.h:
2018 * gst/rtsp-server/rtsp-stream-transport.c:
2019 * gst/rtsp-server/rtsp-stream-transport.h:
2020 session-media: let the session-media make the RTPInfo
2021 Add method to create the RTPInfo for a stream-transport.
2022 Add method to create the RTPInfo for all stream-transports in a
2024 Use the session-media RTPInfo code in client. This allows us to refactor
2025 another method to link the TCP callbacks.
2027 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2029 mount-points: sort sequence before g_sequence_lookup
2030 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
2031 sort sequence if dirty, otherwise lookup will fail.
2032 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2034 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2037 configure: rename package from gst-rtsp to gst-rtsp-server
2038 To match git module name and avoid confusion with the
2039 rtsp lib in gst-plugins-base and rtsp plugin in -good.
2041 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
2044 configure: bump core/base/good requirement to 1.2.0
2045 Bump to released stable version and make implicit
2046 requirements explicit.
2048 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2053 Fix broken gettext setup which is not used anyway
2055 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
2058 Automatic update of common submodule
2059 From dbedaa0 to d48bed3
2061 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
2063 * gst/rtsp-server/rtsp-client.c:
2064 * gst/rtsp-server/rtsp-media.c:
2065 * gst/rtsp-server/rtsp-media.h:
2066 media: add setup_sdp vmethod
2067 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
2068 gst_rtsp_media_setup_sdp.
2069 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2071 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
2073 * gst/rtsp-server/rtsp-stream.c:
2074 rtsp-stream: Check return value of sscanf
2075 streamid is only valid if sscanf matched something.
2077 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
2079 * gst/rtsp-server/rtsp-client.c:
2080 rtsp-client: Fix iteration
2081 Wouldn't even enter the code block otherwise (i++ was used as the check
2082 and not the postfix).
2084 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
2086 * gst/rtsp-server/rtsp-client.c:
2087 * gst/rtsp-server/rtsp-client.h:
2088 client: add vmethod to configure media and streams
2089 Implement a vmethod that can be used to configure the media and the
2090 streams based on the current context. Handle the blocksize handling in
2091 the default handler.
2092 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
2094 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2097 Make git ignore more unit test binaries
2099 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2101 * gst/rtsp-server/rtsp-address-pool.h:
2102 * gst/rtsp-server/rtsp-auth.h:
2103 * gst/rtsp-server/rtsp-client.h:
2104 * gst/rtsp-server/rtsp-context.h:
2105 * gst/rtsp-server/rtsp-media-factory-uri.h:
2106 * gst/rtsp-server/rtsp-media-factory.h:
2107 * gst/rtsp-server/rtsp-media.h:
2108 * gst/rtsp-server/rtsp-mount-points.h:
2109 * gst/rtsp-server/rtsp-server.h:
2110 * gst/rtsp-server/rtsp-session-media.h:
2111 * gst/rtsp-server/rtsp-session-pool.h:
2112 * gst/rtsp-server/rtsp-session.h:
2113 * gst/rtsp-server/rtsp-stream-transport.h:
2114 * gst/rtsp-server/rtsp-stream.h:
2115 * gst/rtsp-server/rtsp-thread-pool.h:
2116 * gst/rtsp-server/rtsp-token.h:
2117 rtsp-server: add padding to many public structures
2118 Not mini objects though, since they are not subclassable
2119 anyway, nor kept on the stack or inlined in a structure.
2121 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2123 media: add new create_rtpbin vmethod
2124 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
2125 https://bugzilla.gnome.org/show_bug.cgi?id=719734
2127 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
2129 * tests/check/gst/media.c:
2130 tests: fix memory leak, free test's thread pool
2131 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
2133 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
2135 * gst/rtsp-server/rtsp-stream-transport.c:
2136 stream-transport: free url in finalize
2138 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
2140 * gst/rtsp-server/rtsp-media.c:
2141 media: also do state change in suspended state
2143 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
2145 * gst/rtsp-server/rtsp-client.c:
2146 * gst/rtsp-server/rtsp-media.c:
2147 media: also handle prepare and range in suspended state
2148 When we are suspended, we are already prepared.
2149 We can get the range in the suspended state.
2151 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
2153 * tests/check/Makefile.am:
2154 * tests/check/gst/sessionmedia.c:
2155 check: add test for uri in setup
2156 Added unit tests for the new functionality in GstRTSPStreamTransport.
2157 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2159 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
2161 * gst/rtsp-server/rtsp-client.c:
2162 client: store setup uri and use in PLAY response
2163 Store the uri used when doing the setup and use that in the PLAY
2165 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2167 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
2169 * gst/rtsp-server/rtsp-stream-transport.c:
2170 * gst/rtsp-server/rtsp-stream-transport.h:
2171 stream-transport: add method to get/set url
2173 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
2175 * gst/rtsp-server/rtsp-client.c:
2176 client: suspend after SDP and unsuspend before PLAYING
2177 Based on patches by Ognyan Tonchev <ognyan@axis.com>
2178 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
2180 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
2182 * gst/rtsp-server/rtsp-media-factory.c:
2183 * gst/rtsp-server/rtsp-media-factory.h:
2184 * gst/rtsp-server/rtsp-media.c:
2185 * gst/rtsp-server/rtsp-media.h:
2186 * gst/rtsp-server/rtsp-session-media.c:
2187 * gst/rtsp-server/rtsp-session.c:
2188 * tests/check/gst/media.c:
2189 * tests/check/gst/mediafactory.c:
2190 media: add suspend modes
2191 Add support for different suspend modes. The stream is suspended right after
2192 producing the SDP and after PAUSE. Different suspend modes are available that
2193 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
2194 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
2195 state and RESET will bring the pipeline to the NULL state.
2196 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
2197 this means that the pipeline needs to be prerolled again.
2198 Base on patches by Ognyan Tonchev <ognyan@axis.com>
2199 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2201 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
2203 * gst/rtsp-server/rtsp-media.c:
2204 media: start live streams in blocked state
2205 Start live streams in the blocked state and make them preroll using the
2206 messages. This ensure that no data is played by the sink until we explicitly
2207 unblock the stream right before going to PLAYING.
2208 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2210 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
2212 * gst/rtsp-server/rtsp-media.c:
2213 media: refactor starting and waiting for preroll
2214 Based on patches from Ognyan Tonchev <ognyan@axis.com>
2215 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2217 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
2219 * gst/rtsp-server/rtsp-stream.c:
2220 * gst/rtsp-server/rtsp-stream.h:
2221 stream: add API to block streams
2222 Add an API to block on the streams and make it post a message.
2223 Based on patch by Ognyan Tonchev <ognyan@axis.com>
2224 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2226 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
2228 * docs/libs/Makefile.am:
2229 docs: Specify the override file
2230 Even if it's empty (for now) it avoids make distcheck complaining
2232 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
2234 * gst/rtsp-server/rtsp-media.c:
2235 media: move default implementations to where they are used
2237 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
2239 * gst/rtsp-server/rtsp-media.c:
2240 media: take the right lock in gst_rtsp_media_set_pipeline_state()
2241 We need to take the state_lock when calling this method.
2243 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
2245 * gst/rtsp-server/rtsp-media.c:
2246 media: handle add-added on non-bins too
2247 Handle dynamic payloaders that are not bins, as used in the unit-test.
2249 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2251 * gst/rtsp-server/rtsp-media-factory.c:
2252 * gst/rtsp-server/rtsp-media-factory.h:
2253 * gst/rtsp-server/rtsp-media.c:
2254 rtsp-media/-factory: Fix request pad name comments
2255 These must be escaped for gtk-doc to parse the comments without warnings.
2257 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2259 rtsp-media: remove transports if media is in error status
2260 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
2261 trying to change to GST_STATE_NULL and media is in error status, we
2262 remove all transports.
2263 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
2265 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
2267 * gst/rtsp-server/rtsp-media.c:
2268 rtsp-media: use element metadata to find payloader
2269 Use the element metadata to find the payloader instead of checking
2271 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
2273 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2275 rtsp-stream: add getter for payload type
2276 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
2277 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
2278 element and create the stream with this one instead of the dynpay%d
2280 https://bugzilla.gnome.org/show_bug.cgi?id=712396
2282 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2284 * gst/rtsp-server/rtsp-client.c:
2285 * gst/rtsp-server/rtsp-context.h:
2286 * gst/rtsp-server/rtsp-media.c:
2287 * gst/rtsp-server/rtsp-mount-points.c:
2288 * gst/rtsp-server/rtsp-server.c:
2289 * gst/rtsp-server/rtsp-token.c:
2290 rtsp-*: Refer to NULL as a constant in comments
2292 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2294 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2296 rtsp-*: Fix type name typos in comments
2297 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
2298 * rtsp-auth: Refer to part of constant name as text
2299 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
2300 * rtsp-session-media: Fix GstRTSPSessionMedia typo
2301 * rtsp-stream: Fix typo when refering to GstBin
2302 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2304 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2307 * docs/libs/gst-rtsp-server-docs.sgml:
2308 * docs/libs/gst-rtsp-server-sections.txt:
2309 docs: Improve documentation
2310 * Include annotation-glossary to quiet gtk-doc
2311 * Rename remaining ClientState -> Context
2312 * Rename object hierarchy file
2313 * Remove stale chapter references
2314 * Add missing function and object references
2315 * Include missing GstRTSPAddressPoolResult
2316 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2318 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2320 * gst/rtsp-server/rtsp-client.c:
2321 * gst/rtsp-server/rtsp-server.c:
2322 * gst/rtsp-server/rtsp-session-pool.c:
2323 * gst/rtsp-server/rtsp-session.c:
2324 * gst/rtsp-server/rtsp-stream.c:
2325 rtsp-server: sprinkle some allow-none annotations for g-i
2327 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
2329 * gst/rtsp-server/rtsp-stream.c:
2330 * gst/rtsp-server/rtsp-stream.h:
2331 stream: add method to filter transports
2332 Add a method to safely iterate and collect the stream transports
2333 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2335 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
2337 * gst/rtsp-server/rtsp-client.c:
2338 * gst/rtsp-server/rtsp-server.c:
2339 * gst/rtsp-server/rtsp-session-pool.c:
2340 * gst/rtsp-server/rtsp-session.c:
2341 rtsp: allow NULL func in filters
2342 Passing a null function make the filters return a list of
2345 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
2347 * gst/rtsp-server/rtsp-address-pool.c:
2348 * tests/check/gst/addresspool.c:
2349 address-pool: fix address increment
2350 Use a guint instead of guint8 to increment the address. It's still not
2351 completely correct because a guint might not be able to hold the complete
2352 address range, but that's an enhacement for later.
2353 Add unit test to test improved behaviour.
2354 https://bugzilla.gnome.org/show_bug.cgi?id=708237
2356 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
2358 * gst/rtsp-server/rtsp-client.c:
2359 * tests/check/gst/client.c:
2360 client: allow absolute path in requests
2361 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
2363 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
2365 * gst/rtsp-server/rtsp-client.c:
2366 * gst/rtsp-server/rtsp-client.h:
2367 client: make make_path_from_uri a vmethod
2369 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2371 * docs/libs/gst-rtsp-server-sections.txt:
2372 * gst/rtsp-server/rtsp-stream.c:
2373 * gst/rtsp-server/rtsp-stream.h:
2374 * tests/check/Makefile.am:
2375 * tests/check/gst/stream.c:
2376 stream: Add functions to get rtp and rtcp sockets
2377 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2379 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2381 * gst/rtsp-server/rtsp-context.c:
2382 * gst/rtsp-server/rtsp-context.h:
2383 context: defing a GType for the context
2384 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
2386 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
2388 * gst/rtsp-server/Makefile.am:
2389 * gst/rtsp-server/rtsp-auth.c:
2390 * gst/rtsp-server/rtsp-context.c:
2391 * gst/rtsp-server/rtsp-media.c:
2392 * gst/rtsp-server/rtsp-mount-points.c:
2393 * gst/rtsp-server/rtsp-server.h:
2394 * gst/rtsp-server/rtsp-session-media.c:
2395 * gst/rtsp-server/rtsp-session.c:
2396 * gst/rtsp-server/rtsp-stream.c:
2397 Fixed several GIR warnings
2399 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
2401 * gst/rtsp-server/rtsp-auth.c:
2404 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2406 * tests/check/Makefile.am:
2407 * tests/check/gst/token.c:
2408 tests: Add unit tests for token
2409 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2411 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2413 * gst/rtsp-server/rtsp-token.c:
2414 token: Validate args for gst_rtsp_token_is_allowed
2415 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
2417 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2419 * gst/rtsp-server/rtsp-token.c:
2420 token: Fix bug when creating empty token
2421 We always want to have a valid GstStructure in the token.
2422 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2424 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2426 * gst/rtsp-server/rtsp-thread-pool.c:
2427 thread-pool: avoid race in shutdown
2428 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
2429 don't actually stop the mainloop ever. Solve this race by adding an idle source
2430 to the mainloop that calls the _quit. This way we immediately exit the mainloop
2431 if quit was called before we started it.
2433 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2435 * tests/check/Makefile.am:
2436 * tests/check/gst/permissions.c:
2437 tests: Add unit tests for permissions
2438 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
2440 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2442 * tests/check/gst/mediafactory.c:
2443 tests: Test mediafactory permissions
2444 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2446 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2448 * gst/rtsp-server/rtsp-permissions.c:
2449 permissions: Fix refcounting when adding/removing roles
2450 Previously a role that was removed was unreffed twice, and when
2451 replacing an existing role the replaced role was freed while still being
2452 referenced. Both bugs are now fixed.
2453 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2455 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2457 * tests/check/gst/media.c:
2458 * tests/check/gst/mediafactory.c:
2459 * tests/check/gst/rtspserver.c:
2460 tests: Check gst_rtsp_url_parse return value
2461 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2463 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
2466 Automatic update of common submodule
2467 From 865aa20 to dbedaa0
2469 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
2471 * gst/rtsp-server/rtsp-server.c:
2472 rtsp-server: Fix socket leak
2473 https://bugzilla.gnome.org/show_bug.cgi?id=710088
2475 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
2477 * gst/rtsp-server/rtsp-session-pool.c:
2478 rtsp-session-pool: Make sure session IDs are properly URI-escaped
2479 https://bugzilla.gnome.org/show_bug.cgi?id=643812
2481 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2483 * examples/.gitignore:
2484 * examples/test-video.c:
2485 examples: fix compilation when WITH_AUTH is defined
2486 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2488 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
2491 gitignore: Add new test binary
2493 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
2495 * tests/check/Makefile.am:
2496 * tests/check/gst/threadpool.c:
2497 thread-pool: Add unit test for the thread pools
2498 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2500 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2502 * gst/rtsp-server/rtsp-thread-pool.c:
2503 thread-pool: Fix thread leak when reusing threads
2504 https://bugzilla.gnome.org/show_bug.cgi?id=709730
2506 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
2508 * gst/rtsp-server/rtsp-server.c:
2509 * tests/check/gst/rtspserver.c:
2510 tests: fixed racy behavior in rtspserver tests
2511 https://bugzilla.gnome.org/show_bug.cgi?id=710078
2513 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2515 * tests/check/gst/addresspool.c:
2516 tests: Improve address pool unit tests
2517 Add a range with mixed IPV4 and IPV6 addresses to pool.
2518 Get an IPV4 address from an IPV6-only pool.
2519 Get an IPV6 address from an IPV4-only pool.
2520 Reserve a IPV6 address from an IPV4-only pool.
2521 Check for unicast addresses in multicast-only pool.
2522 Check for unicast addresses in uni-/multicast-mixed pool.
2523 https://bugzilla.gnome.org/show_bug.cgi?id=710128
2525 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2527 * gst/rtsp-server/rtsp-client.c:
2528 client: append query string in PAUSE/PLAY/TEARDOWN as well
2530 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
2532 * gst/rtsp-server/rtsp-client.c:
2533 client: Add query to control path
2534 If the SETUP url contains a query it must be appended to the control
2535 path so that it matches any already created stream in the media. The
2536 query will also be appended to the session media path.
2538 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2540 * gst/rtsp-server/rtsp-media.c:
2541 rtsp-media: remove old line
2543 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
2545 * gst/rtsp-server/rtsp-stream.c:
2546 stream: Correct control comparison
2547 https://bugzilla.gnome.org/show_bug.cgi?id=709176
2549 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2551 * gst/rtsp-server/rtsp-media.c:
2552 media: Check dynamically if the pipeline supports seeking
2553 We should not depend on whether or not the pipeline state change
2554 returned NO_PREROLL or not. A media could dynamically change its
2555 element and switch from seekable to non seekable so it's best to test
2556 the seekable nature of the pipeline dynamically when we try to do a seek.
2558 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2560 * gst/rtsp-server/rtsp-media.c:
2561 media: Return FALSE if seeking is not supported
2563 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2565 * gst/rtsp-server/rtsp-media.c:
2566 rtsp-media: don't seek accurate by default
2567 Accurate seeking is perhaps a little overkill in the most common situation and
2568 causes some formats (mp3) over slow media to seek extremely slowly.
2570 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
2572 * tests/check/gst/rtspserver.c:
2573 tests: fix unit test
2574 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
2576 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
2578 * gst/rtsp-server/rtsp-client.c:
2579 client: Reply 400 if media cannot be constructed
2580 Reply 400 Bad Request instead of 503 Service Unavailable if media
2581 cannot be constructed in SETUP.
2582 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
2584 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
2586 * gst/rtsp-server/rtsp-client.c:
2587 client: Send setup reply once only
2588 If find_media() failed in handle_setup_request() two replies was sent.
2589 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
2591 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
2594 Automatic update of common submodule
2595 From 6b03ba7 to 865aa20
2597 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
2599 * gst/rtsp-server/rtsp-server.c:
2600 server: Emit client-connected signal earlier
2601 Emit client-connected before the client ref is given to a GSource,
2602 otherwise client-connected can be emitted after the client object has
2605 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
2607 * gst/rtsp-server/rtsp-address-pool.c:
2608 * gst/rtsp-server/rtsp-address-pool.h:
2609 * gst/rtsp-server/rtsp-stream.c:
2610 * tests/check/gst/addresspool.c:
2611 addresspool: return reason of failure
2612 Let gst_rtsp_address_pool_reserve_address() return the reason why
2613 the address could not be reserved.
2614 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
2616 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
2619 autogen.sh: Sync behaviour with other GStreamer modules
2620 Allows building from outside of tree amongst other things
2622 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
2625 Automatic update of common submodule
2626 From b613661 to 6b03ba7
2628 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
2631 Automatic update of common submodule
2632 From 74a6857 to b613661
2634 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
2637 Automatic update of common submodule
2638 From 01a7a46 to 74a6857
2640 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
2642 * gst/rtsp-server/rtsp-client.c:
2643 client: Do not read beyond end of path string
2644 If the setup was done without a control url, make sure we don't try to read the
2645 non-existing control string and crash.
2647 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2649 * gst/rtsp-server/rtsp-client.c:
2650 client: Fix RTPInfo header
2651 Refactor the method to make the content_base.
2652 Use the content-base and the control url to construct the RTPInfo
2655 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2657 * gst/rtsp-server/rtsp-client.c:
2658 client: map url to path only in describe
2659 Only map the request url to a path in the DESCRIBE method. The SDP then
2660 contains the base and control urls that should be used to SETUP/PAUSE/
2661 PLAY/TEARDOWN the media.
2663 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2665 * gst/rtsp-server/rtsp-client.c:
2666 Revert "client: map URL to path in requests"
2667 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
2668 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
2669 contains the base and control urls which are used in the SETUP, PLAY,
2670 PAUSE and TEARDOWN requests.
2672 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2674 * gst/rtsp-server/rtsp-client.c:
2675 client: map URL to path in requests
2677 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2679 * gst/rtsp-server/rtsp-client.c:
2680 * gst/rtsp-server/rtsp-mount-points.c:
2681 * gst/rtsp-server/rtsp-mount-points.h:
2682 mount-points: make vmethod to make path from uri
2683 Make a vmethod to transform an url into a path. The path is then used to lookup
2684 the factory. This makes it possible to also use other bits of the url, such as
2685 the query parameters, to locate the factory.
2687 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
2689 * gst/rtsp-server/rtsp-thread-pool.c:
2690 * gst/rtsp-server/rtsp-thread-pool.h:
2691 thread-pool: Add cleanup to wait for the threadpool to finish
2692 Also fix race condition if two threads are asking for the first
2693 thread from the thread pool at once. This would case two internal
2694 GThreadPools to be created.
2695 https://bugzilla.gnome.org/show_bug.cgi?id=707753
2697 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
2699 * gst/rtsp-server/rtsp-client.c:
2700 * tests/check/gst/client.c:
2701 client: free threadpool
2702 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2704 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
2706 * tests/check/gst/mountpoints.c:
2707 mountpoints tests: unref matched factories
2708 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2710 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
2712 * tests/check/gst/media.c:
2713 media tests: unref thread pool and caps
2714 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2716 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
2718 * gst/rtsp-server/rtsp-auth.c:
2719 * gst/rtsp-server/rtsp-media-factory.c:
2720 * gst/rtsp-server/rtsp-media.c:
2721 auth, media, media-factory: unref permissions
2722 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2724 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2726 * examples/Makefile.am:
2727 Makefile: add rule for appsrc example
2729 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2731 * examples/test-appsrc.c:
2732 tests: add appsrc example
2733 Add an example on how to use appsrc to feed the server pipeline with data.
2735 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
2737 * gst/rtsp-server/rtsp-client.c:
2738 rtsp-client: remove query part from content-base string
2739 Make sure that after the control url has been resolved, it's
2740 not a part of the query-string.
2741 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
2743 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2745 * gst/rtsp-server/rtsp-client.c:
2746 client: don't check url in response
2747 There is no url or method in the response to check
2749 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2751 * gst/rtsp-server/rtsp-client.c:
2752 * gst/rtsp-server/rtsp-client.h:
2753 Add handle-response signal for when we receive a GET_PARAMETER response
2755 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2757 * gst/rtsp-server/rtsp-server.c:
2758 Fix gst_rtsp_server_client_filter, using wrong variable type
2760 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
2762 * gst/rtsp-server/rtsp-media-factory-uri.c:
2763 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
2764 For AAC we need to check for framed=true instead of parsed=true.
2765 https://bugzilla.gnome.org/show_bug.cgi?id=701384
2767 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2769 * gst/rtsp-server/rtsp-stream.c:
2770 stream: optimize pipeline for protocols
2771 When TCP is not an allowed protocol for the stream, avoid creating the
2772 appsrc/appsink/queue and tee elements.
2774 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2776 * gst/rtsp-server/rtsp-media.c:
2777 media: set protocols on streams
2779 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2781 * gst/rtsp-server/rtsp-client.c:
2782 client: use protocols supported by stream
2784 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2786 * gst/rtsp-server/rtsp-media-factory.c:
2787 * gst/rtsp-server/rtsp-media.c:
2788 * gst/rtsp-server/rtsp-stream.c:
2789 media-factory: allow all protocols
2791 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2793 * gst/rtsp-server/rtsp-media.c:
2794 media: configure protocols in new streams
2796 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2798 * gst/rtsp-server/rtsp-stream.c:
2799 * gst/rtsp-server/rtsp-stream.h:
2800 stream: add protocols property
2802 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2804 * gst/rtsp-server/rtsp-media.c:
2805 rtsp-media: send state in "new-state" signal
2806 https://bugzilla.gnome.org/show_bug.cgi?id=705110
2808 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
2811 build: add subdir-objects to AM_INIT_AUTOMAKE
2812 Fixes warnings with automake 1.14
2813 https://bugzilla.gnome.org/show_bug.cgi?id=705350
2815 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2817 * docs/libs/gst-rtsp-server-sections.txt:
2818 * gst/rtsp-server/rtsp-client.c:
2819 * gst/rtsp-server/rtsp-server.c:
2820 * gst/rtsp-server/rtsp-server.h:
2821 server: add method to iterate clients of server
2823 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2825 * gst/rtsp-server/rtsp-media.c:
2826 * gst/rtsp-server/rtsp-media.h:
2827 Add vmethod for rtsp-media subclass to access rtpbin
2829 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2831 * gst/rtsp-server/rtsp-client.h:
2832 small documentation fix
2834 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2836 * gst/rtsp-server/rtsp-client.c:
2837 Do not take range header if range is invalid
2839 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2841 * docs/libs/gst-rtsp-server-sections.txt:
2842 * gst/rtsp-server/rtsp-media.c:
2843 media: add docs for new method
2845 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2847 * gst/rtsp-server/rtsp-media.c:
2848 * gst/rtsp-server/rtsp-media.h:
2849 Add API to rtsp-media set the pipeline's state
2851 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2853 * gst/rtsp-server/rtsp-media.c:
2854 Update current position/duration when gst_rtsp_media_get_range_string is called
2856 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2858 * examples/test-cgroups.c:
2859 tests: add some more docs
2861 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2863 * examples/test-cgroups.c:
2864 * gst/rtsp-server/Makefile.am:
2865 * gst/rtsp-server/rtsp-auth.c:
2866 * gst/rtsp-server/rtsp-auth.h:
2867 * gst/rtsp-server/rtsp-client.c:
2868 * gst/rtsp-server/rtsp-client.h:
2869 * gst/rtsp-server/rtsp-context.c:
2870 * gst/rtsp-server/rtsp-context.h:
2871 * gst/rtsp-server/rtsp-params.c:
2872 * gst/rtsp-server/rtsp-params.h:
2873 * gst/rtsp-server/rtsp-server.c:
2874 * gst/rtsp-server/rtsp-thread-pool.c:
2875 * gst/rtsp-server/rtsp-thread-pool.h:
2876 * tests/check/gst/client.c:
2877 ClientState -> Context
2878 Rename the clientstate to context and put the code in a separate file.
2880 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2882 * examples/test-auth.c:
2883 * gst/rtsp-server/rtsp-auth.c:
2884 * gst/rtsp-server/rtsp-auth.h:
2885 auth: add support for default token
2886 The default token is used when the user is not authenticated and can be used to
2887 give minimal permissions.
2889 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2891 * examples/test-auth.c:
2892 * gst/rtsp-server/rtsp-auth.c:
2893 auth: use defines when possible
2895 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2897 * gst/rtsp-server/rtsp-address-pool.c:
2898 address-pool: improve docs
2900 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2902 * gst/rtsp-server/rtsp-permissions.c:
2903 permissions: add the role to the copy
2905 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
2907 * gst/rtsp-server/rtsp-permissions.c:
2908 permissions: Also copy the roles
2910 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
2912 * gst/rtsp-server/rtsp-permissions.c:
2913 permissions: Make it build
2915 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2917 * gst/rtsp-server/rtsp-address-pool.h:
2920 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2922 * docs/libs/gst-rtsp-server-sections.txt:
2923 * gst/rtsp-server/rtsp-auth.c:
2924 * gst/rtsp-server/rtsp-auth.h:
2925 * gst/rtsp-server/rtsp-media.c:
2926 * gst/rtsp-server/rtsp-session-media.c:
2927 * gst/rtsp-server/rtsp-stream-transport.c:
2928 * gst/rtsp-server/rtsp-stream-transport.h:
2929 * gst/rtsp-server/rtsp-stream.c:
2930 * tests/check/gst/client.c:
2933 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2935 * docs/libs/gst-rtsp-server-sections.txt:
2936 * gst/rtsp-server/rtsp-address-pool.c:
2937 * gst/rtsp-server/rtsp-address-pool.h:
2938 * tests/check/gst/addresspool.c:
2939 * tests/check/gst/rtspserver.c:
2940 address-pool: cleanups
2941 Remove redundant method, improve docs.
2943 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2945 * docs/libs/gst-rtsp-server-sections.txt:
2946 * gst/rtsp-server/rtsp-auth.h:
2947 * gst/rtsp-server/rtsp-permissions.c:
2948 * gst/rtsp-server/rtsp-permissions.h:
2949 * gst/rtsp-server/rtsp-token.c:
2952 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2954 * gst/rtsp-server/rtsp-permissions.c:
2955 permissions: implement _remove_role
2957 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2959 * gst/rtsp-server/rtsp-permissions.c:
2960 permissions: update docs
2962 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2964 * tests/check/gst/client.c:
2965 tests: simplify tests
2966 Client settings are now disabled by default so we don't need an auth
2967 module to disable them.
2969 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2971 * gst/rtsp-server/rtsp-auth.c:
2972 auth: add default authorizations
2973 When no auth module is specified, use our table of defaults to look up the
2974 default value of the check instead of always allowing everything. This was
2975 we can disallow client settings by default.
2977 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2980 README: update readme
2982 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2984 * gst/rtsp-server/rtsp-thread-pool.c:
2985 * gst/rtsp-server/rtsp-thread-pool.h:
2986 thread-pool: add more docs
2988 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2990 * gst/rtsp-server/rtsp-thread-pool.c:
2991 * gst/rtsp-server/rtsp-thread-pool.h:
2992 thread-pool: fix race in thread reuse
2993 If we try to reuse a thread right after we made it stop, we end up using a
2994 stopped thread. Catch this case and only reuse threads that are not stopping.
2996 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2998 * gst/rtsp-server/rtsp-server.c:
2999 server: add small debug
3001 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3003 * tests/check/gst/client.c:
3005 Add some permissions to media so we can use the auth and enable
3008 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3010 * gst/rtsp-server/rtsp-client.c:
3011 client: support pushed context in handle_request
3012 If we already have a pushed state, reuse it and add our own things. This makes
3013 it easier to write tests.
3015 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3017 * gst/rtsp-server/rtsp-auth.c:
3018 auth: don't auth on methods
3019 Don't authorize on methods anymore but on the resources that we
3020 try to access, this is more flexible.
3021 Move the authorization checks to where they are needed and let the
3022 check return the response on error.
3024 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3026 * gst/rtsp-server/rtsp-mount-points.c:
3027 mount-points: add some debug
3029 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3031 * tests/check/gst/client.c:
3032 tests: almost fix test
3034 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3036 * gst/rtsp-server/rtsp-auth.c:
3037 * gst/rtsp-server/rtsp-auth.h:
3038 * gst/rtsp-server/rtsp-client.c:
3039 * gst/rtsp-server/rtsp-client.h:
3040 * gst/rtsp-server/rtsp-server.c:
3041 * gst/rtsp-server/rtsp-server.h:
3042 auth: let the auth module check client_settings
3043 Let the auth module decide if client settings are allowed for the
3046 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3048 * gst/rtsp-server/rtsp-token.c:
3049 * gst/rtsp-server/rtsp-token.h:
3050 token: add method to check boolean permission
3052 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3054 * examples/test-auth.c:
3055 * examples/test-cgroups.c:
3056 * gst/rtsp-server/rtsp-token.c:
3057 * gst/rtsp-server/rtsp-token.h:
3058 token: simplify token constructor
3059 Use variable arguments to make easier API.
3061 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3063 * examples/test-auth.c:
3064 * examples/test-cgroups.c:
3065 * gst/rtsp-server/rtsp-media-factory.c:
3066 * gst/rtsp-server/rtsp-media-factory.h:
3067 media-factory: add convenience API for factory
3069 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3071 * examples/test-auth.c:
3072 * examples/test-cgroups.c:
3073 * gst/rtsp-server/rtsp-permissions.c:
3074 * gst/rtsp-server/rtsp-permissions.h:
3075 permissions: simplify API a little
3076 Avoid passing GstStructure in the add_role method, use varargs instead
3077 to construct the structure behind the scenes. We can then also use the
3078 structure name as the role and simplify some more logic.
3080 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3082 * gst/rtsp-server/rtsp-auth.c:
3085 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3087 * gst/rtsp-server/rtsp-auth.c:
3088 * gst/rtsp-server/rtsp-auth.h:
3089 * gst/rtsp-server/rtsp-client.c:
3090 auth: handle unauthorized response
3091 Move handling of the unauthorized response to the auth module, it can add
3092 the appropriate headers to request authorization for the required method
3093 much better than the client.
3095 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3097 * gst/rtsp-server/rtsp-client.c:
3098 * gst/rtsp-server/rtsp-client.h:
3099 client: allow for sending any message, not only requests
3100 Change the _send_request() method to _send_message() so that we
3101 can both send requests and replies.
3103 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3105 * docs/libs/gst-rtsp-server-sections.txt:
3106 * gst/rtsp-server/rtsp-server.h:
3109 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3111 * examples/test-video.c:
3112 * gst/rtsp-server/rtsp-auth.c:
3113 * gst/rtsp-server/rtsp-auth.h:
3114 * gst/rtsp-server/rtsp-server.c:
3115 * gst/rtsp-server/rtsp-server.h:
3116 auth: move TLS handling to auth module
3117 Remove the TLS settings on the server and move it to the auth module because
3118 that is where security related bits go.
3120 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3122 * gst/rtsp-server/rtsp-client.c:
3123 * gst/rtsp-server/rtsp-client.h:
3124 client: add state push/pop
3126 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3128 * gst/rtsp-server/rtsp-client.c:
3129 * gst/rtsp-server/rtsp-client.h:
3130 client: add connection to state
3132 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3134 * gst/rtsp-server/rtsp-mount-points.c:
3135 mount-points: fix debug
3137 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3139 * tests/check/gst/media.c:
3140 tests: fix media test
3142 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3144 * gst/rtsp-server/rtsp-thread-pool.c:
3145 thread-pool: we don't require a state
3147 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3149 * gst/rtsp-server/rtsp-server.c:
3150 server: let context ref the server
3151 So that we don't risk losing the server object early anc crash.
3153 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3155 * tests/check/gst/client.c:
3156 tests: fix client test
3158 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3161 * docs/libs/gst-rtsp-server-docs.sgml:
3162 * docs/libs/gst-rtsp-server-sections.txt:
3163 * gst/rtsp-server/rtsp-address-pool.c:
3164 * gst/rtsp-server/rtsp-auth.c:
3165 * gst/rtsp-server/rtsp-client.c:
3166 * gst/rtsp-server/rtsp-client.h:
3167 * gst/rtsp-server/rtsp-media-factory-uri.c:
3168 * gst/rtsp-server/rtsp-media-factory.c:
3169 * gst/rtsp-server/rtsp-media-factory.h:
3170 * gst/rtsp-server/rtsp-media.c:
3171 * gst/rtsp-server/rtsp-mount-points.c:
3172 * gst/rtsp-server/rtsp-params.c:
3173 * gst/rtsp-server/rtsp-permissions.c:
3174 * gst/rtsp-server/rtsp-sdp.c:
3175 * gst/rtsp-server/rtsp-server.c:
3176 * gst/rtsp-server/rtsp-server.h:
3177 * gst/rtsp-server/rtsp-session-media.c:
3178 * gst/rtsp-server/rtsp-session-pool.c:
3179 * gst/rtsp-server/rtsp-session.c:
3180 * gst/rtsp-server/rtsp-stream-transport.c:
3181 * gst/rtsp-server/rtsp-stream.c:
3182 * gst/rtsp-server/rtsp-thread-pool.c:
3183 * gst/rtsp-server/rtsp-token.c:
3186 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3188 * gst/rtsp-server/rtsp-session-pool.c:
3189 * gst/rtsp-server/rtsp-session-pool.h:
3190 session-pool: make vmethod to create a session
3191 Make a vmethod to create a sessions so that subclasses can create
3192 custom session objects
3194 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3196 * gst/rtsp-server/rtsp-auth.c:
3197 * gst/rtsp-server/rtsp-media-factory.h:
3198 * gst/rtsp-server/rtsp-media.h:
3199 * gst/rtsp-server/rtsp-mount-points.h:
3200 * gst/rtsp-server/rtsp-session-pool.h:
3201 * gst/rtsp-server/rtsp-stream.h:
3204 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3206 * docs/libs/gst-rtsp-server-docs.sgml:
3207 * docs/libs/gst-rtsp-server-sections.txt:
3208 * gst/rtsp-server/rtsp-address-pool.c:
3209 * gst/rtsp-server/rtsp-address-pool.h:
3210 * gst/rtsp-server/rtsp-auth.c:
3211 * gst/rtsp-server/rtsp-client.h:
3212 * gst/rtsp-server/rtsp-media-factory.h:
3213 * gst/rtsp-server/rtsp-media.c:
3214 * gst/rtsp-server/rtsp-media.h:
3215 * gst/rtsp-server/rtsp-permissions.c:
3216 * gst/rtsp-server/rtsp-permissions.h:
3217 * gst/rtsp-server/rtsp-server.h:
3218 * gst/rtsp-server/rtsp-session-media.c:
3219 * gst/rtsp-server/rtsp-session-media.h:
3220 * gst/rtsp-server/rtsp-session-pool.h:
3221 * gst/rtsp-server/rtsp-session.h:
3222 * gst/rtsp-server/rtsp-stream-transport.h:
3223 * gst/rtsp-server/rtsp-stream.c:
3224 * gst/rtsp-server/rtsp-thread-pool.h:
3227 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3230 * examples/Makefile.am:
3231 configure: compile cgroup example conditionally
3232 Only compile the cgroup example when we have libcgroup
3234 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3237 * examples/Makefile.am:
3238 * examples/test-cgroups.c:
3239 examples: add cgroups example
3241 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3243 * tests/check/gst/rtspserver.c:
3244 tests: fix compilation
3246 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3248 * gst/rtsp-server/rtsp-thread-pool.c:
3249 thread-pool: fix vmethod invocation
3251 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3253 * gst/rtsp-server/rtsp-thread-pool.c:
3254 * gst/rtsp-server/rtsp-thread-pool.h:
3255 thread-pool: store thread type in thread
3257 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3259 * gst/rtsp-server/rtsp-client.c:
3260 client: pass thread from pool to media _prepare
3261 Get a thread from the configured threadpool and pass it to the prepare method of
3264 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3266 * gst/rtsp-server/rtsp-media.c:
3267 * gst/rtsp-server/rtsp-media.h:
3268 media: Accept a thread in _prepare
3269 Remove out own threadpool handling and use the provided thread and
3270 maincontext for the bus messages and the state changes.
3272 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3274 * gst/rtsp-server/rtsp-server.c:
3275 server: configure client thread pool
3277 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3279 * gst/rtsp-server/rtsp-client.c:
3280 * gst/rtsp-server/rtsp-client.h:
3281 client: add method to configure thread pool
3283 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3285 * gst/rtsp-server/rtsp-client.h:
3286 * gst/rtsp-server/rtsp-server.c:
3287 * gst/rtsp-server/rtsp-server.h:
3288 server: use thread pool
3289 Use the thread pool instead of doing our own thing.
3291 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3293 * gst/rtsp-server/Makefile.am:
3294 * gst/rtsp-server/rtsp-thread-pool.c:
3295 * gst/rtsp-server/rtsp-thread-pool.h:
3296 thread-pool: add object to manage threads
3297 Add an object to manage the client and media threads.
3299 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3301 * gst/rtsp-server/rtsp-auth.c:
3302 auth: debug authorization check
3304 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3306 * gst/rtsp-server/rtsp-media.c:
3307 media: start media pipeline in context
3308 Start the media pipeline in the provided context (or our default one
3309 when NULL). This makes sure that we run the bus thread in this context and that
3310 all media threads are children of this context.
3312 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3314 * gst/rtsp-server/rtsp-media-factory.c:
3315 factory: pass permissions to media by default
3317 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3319 * examples/test-auth.c:
3320 test: add permissions to auth test
3321 Ass some permissions to the media factory in the test.
3323 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3325 * gst/rtsp-server/rtsp-auth.c:
3326 * gst/rtsp-server/rtsp-auth.h:
3327 * gst/rtsp-server/rtsp-client.c:
3328 auth: simplify auth checks
3329 Remove client from methods, it's now in the state
3330 Perform the check specified by the string, use the information from the
3331 thread local context.
3333 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3335 * gst/rtsp-server/rtsp-client.c:
3336 * gst/rtsp-server/rtsp-client.h:
3337 client: add state to current thread
3338 Add the client to the ClientState object.
3339 Place the ClientState on the current thread.
3341 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3343 * gst/rtsp-server/rtsp-media-factory.c:
3344 * gst/rtsp-server/rtsp-media-factory.h:
3345 * gst/rtsp-server/rtsp-media.c:
3346 * gst/rtsp-server/rtsp-media.h:
3347 media: make it possible to set permissions
3348 Make it possible to set permissions on media and media factory objects
3350 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3352 * gst/rtsp-server/Makefile.am:
3353 * gst/rtsp-server/rtsp-permissions.c:
3354 * gst/rtsp-server/rtsp-permissions.h:
3355 permissions: add permissions object
3356 Add a mini object to store permissions based on a role.
3358 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3360 * examples/test-auth.c:
3361 * gst/rtsp-server/rtsp-auth.c:
3362 * gst/rtsp-server/rtsp-auth.h:
3363 * gst/rtsp-server/rtsp-client.c:
3364 auth: add auth checks
3365 Add an enum with auth checks and implement the checks in the auth object.
3366 Perform the checks from the client.
3368 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3370 * examples/test-auth.c:
3371 * gst/rtsp-server/rtsp-auth.c:
3372 * gst/rtsp-server/rtsp-auth.h:
3373 * gst/rtsp-server/rtsp-client.h:
3374 auth: use the token after authentication
3375 After we authenticated a user, keep the Token around in the state.
3377 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3379 * gst/rtsp-server/rtsp-client.c:
3380 * gst/rtsp-server/rtsp-media.c:
3381 * gst/rtsp-server/rtsp-media.h:
3382 * tests/check/gst/media.c:
3383 media: add optional context for bus messages
3384 Add an optional mainloop to _prepare that will handle the bus messages instead
3385 of always using the shared mainloop.
3387 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3389 * gst/rtsp-server/Makefile.am:
3390 * gst/rtsp-server/rtsp-token.c:
3391 * gst/rtsp-server/rtsp-token.h:
3392 token: add authorization token
3393 Add a simply miniobject that contains the authorizations. The object contains a
3394 GstStructure that hold all authorization fields. When a user is authenticated,
3395 the auth module will create a Token for the user. The token is then used to
3396 check what operations the user is allowed to do and various other configuration
3399 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3401 * examples/test-auth.c:
3402 * gst/rtsp-server/rtsp-auth.c:
3403 * gst/rtsp-server/rtsp-auth.h:
3404 * gst/rtsp-server/rtsp-client.c:
3405 * gst/rtsp-server/rtsp-client.h:
3406 * gst/rtsp-server/rtsp-media-factory.c:
3407 * gst/rtsp-server/rtsp-media-factory.h:
3408 * gst/rtsp-server/rtsp-media.c:
3409 * gst/rtsp-server/rtsp-media.h:
3410 auth: remove auth from media and factory
3411 Remove the auth object from media and factory. We want to have the RTSPClient
3412 authenticate and authorize resources, there is no need to place another auth
3413 manager on the media/factory.
3415 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3417 * examples/test-auth.c:
3418 * gst/rtsp-server/rtsp-auth.c:
3419 * gst/rtsp-server/rtsp-auth.h:
3420 * gst/rtsp-server/rtsp-client.h:
3421 auth: add support for multiple basic auth tokens
3422 Make it possible to add multiple basic authorisation tokens to one authorization
3423 object. Associate with each token an authorization group that will define what
3424 capabilities are allowed.
3426 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3428 * gst/rtsp-server/rtsp-client.c:
3429 client: error out on non-aggregate control
3430 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
3432 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3434 * gst/rtsp-server/rtsp-client.c:
3435 client: rework setup request a little
3436 Cache the media in DESCRIBE based on the longest matching path with the uri
3437 that we can find in the mount points.
3438 Rework the setup request a little to get the media from the session or from
3439 the longest matching path, this way we can derive the control string as
3440 everything after the path instead of hardcoding it.
3441 Find the stream based on the control string and only open a session when all
3444 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3446 * gst/rtsp-server/rtsp-media.c:
3447 * gst/rtsp-server/rtsp-media.h:
3448 media: add method to find a stream by control url
3450 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3452 * gst/rtsp-server/rtsp-stream.c:
3453 * gst/rtsp-server/rtsp-stream.h:
3454 stream: add method to check control url of stream
3456 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3458 * gst/rtsp-server/rtsp-client.c:
3459 * gst/rtsp-server/rtsp-session-media.c:
3460 * gst/rtsp-server/rtsp-session-media.h:
3461 * gst/rtsp-server/rtsp-session.c:
3462 * gst/rtsp-server/rtsp-session.h:
3463 session: use path matching for session media
3464 Use a path string instead of a uri to lookup session media in the sessions. Also
3465 use path matching to find the largest possible path that matches.
3467 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3469 * gst/rtsp-server/rtsp-client.c:
3470 * gst/rtsp-server/rtsp-mount-points.c:
3471 * gst/rtsp-server/rtsp-mount-points.h:
3472 * tests/check/gst/mountpoints.c:
3473 mount-points: remove useless vmethod
3474 Making lookups in the mount points should not be done with a URL, if there is a
3475 mapping to be done from URL to mount points, we'll need to do it somewhere
3478 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3480 * gst/rtsp-server/rtsp-mount-points.c:
3481 * gst/rtsp-server/rtsp-mount-points.h:
3482 * tests/check/gst/mountpoints.c:
3483 mount-points: improve mount point searching
3484 Use a GSequence to keep track of the mount points.
3485 Match a URL to the longest matching registered mount point. This should be the
3486 URL to perform aggreagate control and the remainder is the stream specific
3488 Add some unit tests for this.
3490 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
3492 * gst/rtsp-server/Makefile.am:
3493 rtsp-server: Allow building of static library
3495 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3497 * tests/check/gst/mediafactory.c:
3498 tests: fix compilation
3500 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3502 * gst/rtsp-server/rtsp-sdp.c:
3503 sdp: get control string from stream
3504 Use the control string as configured in the stream.
3506 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3508 * gst/rtsp-server/rtsp-stream.c:
3509 * gst/rtsp-server/rtsp-stream.h:
3510 stream: add methods and property to set control string
3512 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3514 * gst/rtsp-server/rtsp-client.c:
3516 Rename variables for clarity
3517 Keep media in state when we can
3519 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3521 * gst/rtsp-server/rtsp-client.c:
3522 * gst/rtsp-server/rtsp-stream.c:
3523 * gst/rtsp-server/rtsp-stream.h:
3524 stream: add more support for IPv6
3525 Rename _get_address to _get_multicast_address in GstRTSPStream to
3526 make it clear that this function only deals with multicast.
3527 Make it possible to have both an IPv4 and IPv6 multicast address on
3528 a stream. Give the client an IPv4 or IPv6 address depending on the
3529 address it used to connect to the server.
3530 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
3532 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3534 * gst/rtsp-server/rtsp-client.c:
3537 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3539 * gst/rtsp-server/rtsp-stream.c:
3540 stream: handle failed port allocation
3541 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
3542 can't allocate any family at all. Also keep track of what port families we
3544 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
3546 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3548 * gst/rtsp-server/rtsp-stream.c:
3549 stream: improve docs
3551 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3553 * gst/rtsp-server/rtsp-stream-transport.c:
3554 stream-transport: remove old if 0 block
3556 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
3558 * tests/check/gst/client.c:
3560 gst_rtsp_client_get_uri() has been removed
3561 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
3563 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3565 * gst/rtsp-server/rtsp-client.c:
3566 * gst/rtsp-server/rtsp-client.h:
3567 client: add method to filter managed sessions
3568 Add a method to filter the sessions managed by this client connection.
3569 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
3571 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3573 * gst/rtsp-server/rtsp-client.c:
3574 * gst/rtsp-server/rtsp-client.h:
3575 client: remove _get_uri() method
3576 Remove the get_uri() method on the client. A client has no uri, the uri
3577 property is an internal property to manage the last cached media for
3580 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3582 * gst/rtsp-server/rtsp-media-factory.h:
3583 media-factory: fix typo
3585 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
3587 * gst/rtsp-server/rtsp-media.c:
3588 rtsp-media: Do not leak the query in default_query_stop
3589 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
3591 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3593 * gst/rtsp-server/rtsp-media.c:
3594 media: don't unlock when conversion fails
3595 Don't unlock the state lock when conversion fails because it was not locked.
3597 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3599 * gst/rtsp-server/rtsp-media.c:
3600 * gst/rtsp-server/rtsp-media.h:
3601 Add query_position and query_stop vmethods to rtsp-media
3603 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3605 * gst/rtsp-server/rtsp-media.c:
3606 Fix typo in property install for rtsp-media's time-provider
3608 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3610 * gst/rtsp-server/rtsp-client.c:
3611 * gst/rtsp-server/rtsp-client.h:
3612 client: clean some variables
3613 Clean some variables and add some guards to _send_request()
3615 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3617 * gst/rtsp-server/rtsp-client.c:
3618 * gst/rtsp-server/rtsp-client.h:
3619 Add gst_rtsp_client_send_request API
3620 This makes it possible to send arbitrary messages to a client, such as
3621 SET_PARAMETER or GET_PARAMETER
3623 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3625 * gst/rtsp-server/rtsp-media.c:
3626 * gst/rtsp-server/rtsp-media.h:
3627 media: add _get_element() method
3628 Add method to get the element used when creating the media.
3629 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
3631 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3633 * gst/rtsp-server/rtsp-media.c:
3636 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3638 * gst/rtsp-server/rtsp-stream.c:
3639 * gst/rtsp-server/rtsp-stream.h:
3640 stream: allow access to the rtp session
3641 https://bugzilla.gnome.org/show_bug.cgi?id=703004
3643 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
3645 * gst/rtsp-server/rtsp-stream.c:
3646 * gst/rtsp-server/rtsp-stream.h:
3647 dscp qos support in gst-rtsp-stream
3648 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
3650 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3652 * tests/check/gst/rtspserver.c:
3654 Actually do what the comment says. Also keep the old code around, not sure what
3655 should happen when you get a 454 from a TEARDOWN, does it close the connection?
3656 it currently doesn't.
3658 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3660 * gst/rtsp-server/rtsp-client.c:
3661 client: also watch newly created session
3662 When we newly created a session, start watching it immediately instead of
3663 on the next request.
3665 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
3667 * tests/check/gst/client.c:
3668 tests: add unit test for new-session
3669 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
3671 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3673 * gst/rtsp-server/rtsp-client.c:
3674 client: emit new-session when new session is created
3675 Only emit new-session when we created a new session for a client, not when a
3676 client picked up a previous session.
3677 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
3679 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
3681 * gst/rtsp-server/rtsp-client.c:
3682 client: handle asterisk as path in requests
3683 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
3685 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3687 * gst/rtsp-server/rtsp-media.c:
3688 media: handle segment query format mismatch
3689 It's possible that the segment query returns with a different format than what
3690 we asked for, handle this case also.
3692 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
3694 * gst/rtsp-server/rtsp-media.c:
3695 media: use segment stop in collect_media_stats
3696 Use segment stop instead of duration as range end point.
3697 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
3699 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3701 * gst/rtsp-server/rtsp-media.c:
3702 * tests/check/gst/media.c:
3703 rtsp-media: Do not leak the element in take_pipeline
3704 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
3706 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
3708 * gst/rtsp-server/rtsp-client.c:
3709 * gst/rtsp-server/rtsp-client.h:
3710 rtsp-client: Make configure_client_transport virtual
3711 This patch makes configure_client_transport virtual. The functionality is
3712 needed to handle some weird clients sending multicast transport settings as url
3714 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
3716 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3718 * gst/rtsp-server/rtsp-client.c:
3719 * gst/rtsp-server/rtsp-client.h:
3720 rtsp-client: Make param_set and param_get virtual
3721 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
3723 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
3725 * gst/rtsp-server/rtsp-client.c:
3726 * gst/rtsp-server/rtsp-media.c:
3727 * gst/rtsp-server/rtsp-media.h:
3728 media: convert_range replaces get_range_times
3729 get_range_times worked for handling UTC ranges for seeks, but we also
3730 need to convert back from NPT to the requested unit in
3731 get_range_string. convert_range is now used for both.
3732 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
3734 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3736 * gst/rtsp-server/rtsp-client.c:
3737 * gst/rtsp-server/rtsp-sdp.c:
3738 * gst/rtsp-server/rtsp-sdp.h:
3739 sdp: cleanup sdp info
3740 We don't need to pass the proto, we can more easily check a boolean.
3741 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
3743 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
3745 * gst/rtsp-server/rtsp-sdp.c:
3746 use 0.0.0.0 or :: for c= line instead of server address
3748 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
3750 * gst/rtsp-server/rtsp-client.c:
3751 use local address, not remote, in SDP
3752 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
3754 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3757 Automatic update of common submodule
3758 From 098c0d7 to 01a7a46
3760 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
3762 * gst/rtsp-server/rtsp-media.c:
3763 * gst/rtsp-server/rtsp-media.h:
3764 media: possibility to override range time conversion
3765 Make it possible to override the conversion from GstRTSPTimeRange to
3766 GstClockTimes, that is done before seeking on the media
3767 pipeline. Overriding can be useful for UTC ranges, where the default
3768 conversion gives nanoseconds since 1900.
3769 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
3771 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
3773 * gst/rtsp-server/rtsp-server.c:
3774 * gst/rtsp-server/rtsp-server.h:
3775 rtsp-server: Expose the use_client_settings API
3776 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
3778 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
3780 * gst/rtsp-server/rtsp-client.c:
3781 * gst/rtsp-server/rtsp-stream.c:
3782 * gst/rtsp-server/rtsp-stream.h:
3783 rtspstream: handle both ipv4 and ipv6 clients
3784 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
3786 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3788 * gst/rtsp-server/rtsp-sdp.c:
3789 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
3790 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
3791 We already have a way to place extra attributes in the SDP by using a string
3792 property with prefix x- or a- in the caps.
3794 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3796 * gst/rtsp-server/rtsp-sdp.c:
3797 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
3798 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
3799 We already have a way to place extra attributes in the SDP, just make a string
3800 property in the payloader with a- or x- prefix.
3802 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3804 * gst/rtsp-server/rtsp-sdp.c:
3805 rtsp: place a- and x- properties as attributes
3806 application/x-rtp has properties with a- and x- prefixes that should be
3807 placed as attributes in the SDP for the media instead of being added to the
3810 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3812 * examples/Makefile.am:
3813 * examples/test-video.c:
3814 example: add TLS example
3816 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3818 * gst/rtsp-server/rtsp-server.c:
3819 * gst/rtsp-server/rtsp-server.h:
3820 server: add support for TLS
3821 Add methods to set and get a TLS certificate.
3822 Add vmethod to configure a new connection. By default, configure the TLS
3823 certificate in a new connection if needed.
3825 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3827 * gst/rtsp-server/rtsp-server.c:
3828 * gst/rtsp-server/rtsp-server.h:
3829 server: remove accept_client vmethod
3830 This vmethod is not very useful so remove it.
3832 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3834 * gst/rtsp-server/rtsp-server.c:
3835 server: don't crash on NULL GError
3837 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
3839 * gst/rtsp-server/rtsp-session-pool.c:
3840 rtsp-session-pool: corrected session timeout detection
3841 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
3843 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3845 * gst/rtsp-server/rtsp-client.c:
3846 client: improve debug
3848 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3850 * gst/rtsp-server/rtsp-client.c:
3851 * gst/rtsp-server/rtsp-client.h:
3852 * gst/rtsp-server/rtsp-server.c:
3853 server: refactor connection setup
3854 Let the server accept the socket connection and construct a GstRTSPConnection
3855 from it. Remove the code from the client and let the client only deal with
3856 a fully configure GstRTSPConnection object.
3857 We will need this later when the server will configure the connection for
3860 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3862 * gst/rtsp-server/rtsp-stream.c:
3863 stream: keep the transport object alive
3864 Keep the transport object alive while we have it as qdata on the
3867 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
3869 * gst/rtsp-server/rtsp-client.c:
3870 * gst/rtsp-server/rtsp-server.c:
3871 rtsp-server: Do not crash on nmapping of server
3872 * generate error when gst_rtsp_connection_accept fails
3873 * do not stop accepting incoming connections because
3874 accepting a client fails
3875 https://bugzilla.gnome.org/show_bug.cgi?id=701072
3877 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
3879 * gst/rtsp-server/rtsp-client.c:
3880 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
3881 https://bugzilla.gnome.org/show_bug.cgi?id=700953
3883 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
3885 * gst/rtsp-server/rtsp-sdp.c:
3886 rtsp-sdp: Parse framerate caps field and set SDP attribute
3887 The SDP attribute and its format is described in RFC4566.
3888 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
3890 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
3892 * gst/rtsp-server/rtsp-sdp.c:
3893 rtsp-sdp: Parse width/height from caps and set SDP attribute
3894 The SDP attribute and its format is described in RFC6064.
3895 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
3897 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
3899 * gst/rtsp-server/rtsp-sdp.c:
3900 * tests/check/gst/client.c:
3901 rtsp-sdp: add bandwidth line
3902 https://bugzilla.gnome.org/show_bug.cgi?id=699220
3904 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3907 Automatic update of common submodule
3908 From 5edcd85 to 098c0d7
3910 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
3912 * tests/check/gst/media.c:
3913 tests: add dynamic payloader prepare/unprepare check
3915 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3917 * gst/rtsp-server/rtsp-media.c:
3918 media: release lock when removing fakesink
3920 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3922 * gst/rtsp-server/rtsp-stream.c:
3923 stream: set elements to NULL before removing
3924 When removing a stream, set the elements to NULL first. This avoids
3925 element-is-not-in-NULL-state errors when we dispose the elements.
3927 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
3930 Automatic update of common submodule
3931 From 3cb3d3c to 5edcd85
3933 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3935 * gst/rtsp-server/rtsp-media.c:
3936 * gst/rtsp-server/rtsp-media.h:
3937 media: listen to pad-removed signals
3938 Listen to the pad-removed signal and remove the stream associated with the
3940 Add signal to be notified of the removed pad.
3941 Remove the fakesink in unprepare()
3942 Fix signatures of the signal methods
3944 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3946 * examples/test-sdp.c:
3947 tests: add example of reusable pipelines
3949 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
3951 * gst/rtsp-server/rtsp-stream.c:
3952 * gst/rtsp-server/rtsp-stream.h:
3953 stream: add method to get the srcpad
3955 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
3957 * tests/check/gst/media.c:
3958 check: add media prepare/unprepare test
3959 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
3961 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
3963 * gst/rtsp-server/rtsp-media.c:
3964 media: disconnect from signal handlers in unprepare()
3965 We connected to the pad-added and no-more-pads signals in prepare() so
3966 we need to disconnect from them in unprepare().
3967 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
3969 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
3971 * gst/rtsp-server/rtsp-media.c:
3972 media: don't free streams array
3973 Don't free the streams array in the unprepare() method, they were not
3975 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
3977 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
3979 * gst/rtsp-server/rtsp-media.c:
3980 media: don't unref the pipeline in unprepare
3981 Unprepare() should undo what prepare() does. Because the pipeline is
3982 not created in prepare(), we should not unref it in unprepare()
3984 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
3986 * gst/rtsp-server/rtsp-stream.c:
3987 stream: clear session and caps for reuse
3988 Set the session and caps to NULL after unref otherwise we might unref
3990 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
3992 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
3994 * gst/rtsp-server/rtsp-client.c:
3995 client: send out teardown signal before tearing down
3996 The advantage is that in the signal handler you get direct access to
3997 information about what streams are about to get torn down (in the
3998 GstRTSPClientState).
3999 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
4001 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
4003 * gst/rtsp-server/rtsp-client.c:
4004 * gst/rtsp-server/rtsp-client.h:
4005 client: expose connection
4006 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
4008 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
4011 Automatic update of common submodule
4012 From aed87ae to 3cb3d3c
4014 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4016 * gst/rtsp-server/rtsp-media.c:
4017 * gst/rtsp-server/rtsp-media.h:
4018 * gst/rtsp-server/rtsp-session-media.c:
4019 * gst/rtsp-server/rtsp-session-media.h:
4020 media: add method to get the base_time of the pipeline
4021 Together with a shared clock, this base-time could eventually be sent to
4022 the client so that it can reconstruct the exact running-time of the clock
4025 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4027 * gst/rtsp-server/Makefile.am:
4028 * gst/rtsp-server/rtsp-media.c:
4029 * gst/rtsp-server/rtsp-media.h:
4030 * gst/rtsp-server/rtsp-sdp.c:
4031 media: add GstNetTimeProvider support
4032 Add a property to let the media provide a GstNetTimeProvider for its clock.
4033 Make methods to get the clock and nettimeprovider
4034 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
4035 provider and also the current time of the clock. This should make it possible
4036 for (GStreamer) clients to slave their clock to the server clock.
4038 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4041 Automatic update of common submodule
4042 From 04c7a1e to aed87ae
4044 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4046 * gst/rtsp-server/rtsp-media.c:
4047 media: wait for buffering to complete
4048 Wait for buffering to complete before changing the state to the target state.
4050 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4052 * gst/rtsp-server/rtsp-media.c:
4053 media: small cleanup
4055 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
4057 * tests/check/gst/rtspserver.c:
4058 tests: remove extra unref in test_setup_non_existing_stream
4059 The unref is not needed anymore, teardown runs without it.
4060 https://bugzilla.gnome.org/show_bug.cgi?id=696542
4062 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
4064 * tests/check/gst/rtspserver.c:
4065 tests: GSocketService cleanup in test_bind_already_in_use
4066 Use g_socket_service_stop so the rtspserver test stops listening for
4067 incoming connections in test_bind_already_in_use.
4068 https://bugzilla.gnome.org/show_bug.cgi?id=696541
4070 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
4072 * gst/rtsp-server/rtsp-media-factory.c:
4073 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
4074 Instead use a GWeakRef which is safe to use
4075 This is a known GLib bug, see:
4076 https://bugzilla.gnome.org/show_bug.cgi?id=667145
4078 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
4080 * gst/rtsp-server/rtsp-client.c:
4081 * gst/rtsp-server/rtsp-media.c:
4082 * gst/rtsp-server/rtsp-media.h:
4083 * gst/rtsp-server/rtsp-sdp.c:
4084 * tests/check/gst/media.c:
4085 * tests/check/gst/rtspserver.c:
4086 rtsp-media/client: Reply to PLAY request with same type of Range
4087 Remember the type of Range from the PLAY request and use the same type for
4090 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
4092 * gst/rtsp-server/rtsp-client.c:
4093 * gst/rtsp-server/rtsp-client.h:
4094 * tests/check/gst/client.c:
4095 rtsp-client: expose uri
4097 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
4099 * tests/check/gst/mediafactory.c:
4100 tests: Hold ref while creating second media
4101 To test if the media aren't shared, make sure we keep the first one while creating a second
4102 otherwise the same memory address may be reused.
4104 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
4107 configure: remove out-of-date comment
4109 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
4112 .gitignore: ignore more build files
4114 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
4116 * tests/check/Makefile.am:
4117 tests: use right _LIBS variable for gst-plugins-base libs
4119 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4121 * tests/check/Makefile.am:
4122 check: add librtp to libs
4124 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
4126 * tests/check/gst/rtspserver.c:
4127 tests: Add test to check selecting a port the server will send from
4129 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
4131 * tests/check/gst/rtspserver.c:
4132 tests: Make sure packets are actually received
4134 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4136 * gst/rtsp-server/rtsp-stream.c:
4137 stream: Select unicast address from pool if appropriate
4139 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
4141 * gst/rtsp-server/rtsp-stream.c:
4142 stream: Properties are always there in Gst 1.0
4144 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4146 * tests/check/gst/addresspool.c:
4147 tests: Add tests for unicast addresses in pool
4149 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
4151 * gst/rtsp-server/rtsp-address-pool.c:
4152 * tests/check/gst/addresspool.c:
4153 address-pool: Verify that multicast addresses are used for multicast and vice-versa
4155 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
4157 * docs/libs/gst-rtsp-server-sections.txt:
4158 * gst/rtsp-server/rtsp-address-pool.c:
4159 * gst/rtsp-server/rtsp-address-pool.h:
4160 * gst/rtsp-server/rtsp-stream.c:
4161 * tests/check/gst/addresspool.c:
4162 address-pool: Add unicast addresses
4164 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4167 * gst/rtsp-server/rtsp-server.c:
4168 * tests/check/gst/rtspserver.c:
4169 rtsp-server: Limit the number of threads per server instance
4170 If we exceed the maximum, just round robin the clients over the existing
4173 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
4175 * gst/rtsp-server/rtsp-server.c:
4176 rtsp-server: No need to store the GMainContext in the client context
4178 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
4180 * tests/check/gst/rtspserver.c:
4181 tests: Add test for client disconnection
4183 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4185 * tests/check/gst/rtspserver.c:
4186 tests: Test client and session timeouts with multiple threads
4188 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
4190 * gst/rtsp-server/rtsp-address-pool.c:
4191 * gst/rtsp-server/rtsp-auth.c:
4192 * gst/rtsp-server/rtsp-client.c:
4193 * gst/rtsp-server/rtsp-media-factory-uri.c:
4194 * gst/rtsp-server/rtsp-media-factory.c:
4195 * gst/rtsp-server/rtsp-media.c:
4196 * gst/rtsp-server/rtsp-mount-points.c:
4197 * gst/rtsp-server/rtsp-server.c:
4198 * gst/rtsp-server/rtsp-session-media.c:
4199 * gst/rtsp-server/rtsp-session-pool.c:
4200 * gst/rtsp-server/rtsp-session.c:
4201 Document locking and its order
4203 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
4205 * tests/check/gst/rtspserver.c:
4206 tests: Test that slow DESCRIBE don't block other clients
4208 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
4210 * tests/check/gst/client.c:
4211 tests: Add tests for client-requested multicast address
4213 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
4215 * docs/libs/gst-rtsp-server-sections.txt:
4216 docs: Put the various functions in the right sections
4218 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
4220 * docs/libs/gst-rtsp-server-docs.sgml:
4221 * docs/libs/gst-rtsp-server-sections.txt:
4222 * gst/rtsp-server/rtsp-address-pool.c:
4223 * gst/rtsp-server/rtsp-address-pool.h:
4224 docs: Generate docs for GstRTSPAddressPool
4226 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4228 * gst/rtsp-server/rtsp-client.c:
4229 * gst/rtsp-server/rtsp-stream.c:
4230 * gst/rtsp-server/rtsp-stream.h:
4231 client: Check client provided addresses against the address pool
4233 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
4235 * gst/rtsp-server/rtsp-address-pool.c:
4236 * gst/rtsp-server/rtsp-address-pool.h:
4237 * tests/check/gst/addresspool.c:
4238 address-pool: Add API to request a specific address from the pool
4239 Also add relevant unit tests.
4241 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
4243 * tests/check/gst/mediafactory.c:
4244 tests: Check the passing around of a RTSPAddressPool
4245 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
4246 way down to the stream.
4248 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
4250 * tests/check/gst/addresspool.c:
4251 tests: Add more tests for the address pool
4253 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
4255 * gst/rtsp-server/rtsp-address-pool.c:
4256 address-pool: Fix off by one error
4257 When splitting a port range, the port after a skip is not part of range.
4259 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
4262 Automatic update of common submodule
4263 From 2de221c to 04c7a1e
4265 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
4268 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
4269 AM_CONFIG_HEADER was removed in automake 1.13
4270 https://bugzilla.gnome.org/show_bug.cgi?id=693368
4272 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
4275 Automatic update of common submodule
4276 From a942293 to 2de221c
4278 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4280 * gst/rtsp-server/rtsp-client.c:
4281 client: make sure the watch exists while sending data
4282 Protect the send_func with a lock. This allows us to wait for sending
4283 to complete before changing the send_func and user_data. We add an
4284 extra ref to the watch to make sure that it remains valid during
4286 When closing the connection, set the send_func to NULL
4287 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
4289 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4291 * tests/check/Makefile.am:
4292 tests: use GST_*_1_0 environment variables everywhere
4293 The _1_0 suffixed environment variables override the
4294 non-suffixed ones, so if we're in an environment that
4295 sets the _1_0 suffixed ones, such as jhbuild, we need
4296 to set those to make sure ours actually always get
4299 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4302 Automatic update of common submodule
4303 From acb04d9 to a942293
4305 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4307 * gst/rtsp-server/rtsp-client.c:
4308 rtsp-client: set the client backlog
4309 Set the client backlog to a reasonable default
4311 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
4313 * gst/rtsp-server/rtsp-media.c:
4314 rtsp-media: Make the element a constructor parameter
4315 https://bugzilla.gnome.org/show_bug.cgi?id=689594
4317 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4319 * docs/libs/Makefile.am:
4320 docs: Link with gcov library when gcov is enabled
4321 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
4323 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4325 * gst/rtsp-server/rtsp-media.c:
4326 media: match prepare with unprepare
4327 Really unprepare when there were an equal amount of prepare calls.
4329 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4331 * gst/rtsp-server/rtsp-media.c:
4332 media: media has to be unprepared in finalize
4333 Because unprepare takes away the last ref on the media.
4335 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4337 * gst/rtsp-server/rtsp-client.c:
4338 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
4339 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
4340 We can't use the refcount to trigger unprepare because it is the unprepare call
4341 that removes the last refcount after all messages are consumed. What we should
4342 probably do is make a prepared refcount and only unprepare when the refcount
4345 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4347 * gst/rtsp-server/rtsp-media.c:
4348 media: let the source unref the last media ref
4349 the last ref to the media is held by the source so we don't need to add more ref
4350 and unrefs, we simply destroy the media when the source is gone.
4352 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4354 * gst/rtsp-server/rtsp-media.c:
4355 media: improve debug
4357 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4359 * gst/rtsp-server/rtsp-media.c:
4361 Make sure we are in the right state when collecting the position and duration.
4362 Only make ourselves PREPARED when we were previously PREPARING.
4364 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4366 * gst/rtsp-server/rtsp-media.c:
4367 media: use g_object_ref/unref for GObjects
4369 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
4371 * gst/rtsp-server/rtsp-client.c:
4372 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
4373 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
4374 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
4375 isn't being used anymore.
4377 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
4379 * gst/rtsp-server/rtsp-media.c:
4380 Fix compiler warning
4382 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
4384 * gst/rtsp-server/rtsp-media-factory-uri.c:
4385 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
4387 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4389 * gst/rtsp-server/rtsp-session-media.h:
4392 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4394 * gst/rtsp-server/rtsp-media.c:
4395 * tests/check/gst/media.c:
4396 media: avoid element leak
4398 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4400 * gst/rtsp-server/rtsp-media.c:
4401 media: require an element in media constructor
4403 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4405 * gst/rtsp-server/rtsp-client.c:
4406 Revert "client: TEARDOWN brings that state to Init again"
4407 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
4408 The object is already disposed, there is no point in setting the state.
4410 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4412 * gst/rtsp-server/rtsp-client.c:
4413 client: TEARDOWN brings that state to Init again
4415 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4417 * docs/libs/gst-rtsp-server-sections.txt:
4418 * examples/test-auth.c:
4419 * gst/rtsp-server/rtsp-auth.c:
4420 * gst/rtsp-server/rtsp-auth.h:
4421 * gst/rtsp-server/rtsp-client.c:
4422 * gst/rtsp-server/rtsp-client.h:
4423 * gst/rtsp-server/rtsp-media-factory-uri.c:
4424 * gst/rtsp-server/rtsp-media-factory-uri.h:
4425 * gst/rtsp-server/rtsp-media-factory.c:
4426 * gst/rtsp-server/rtsp-media-factory.h:
4427 * gst/rtsp-server/rtsp-media.c:
4428 * gst/rtsp-server/rtsp-media.h:
4429 * gst/rtsp-server/rtsp-mount-points.c:
4430 * gst/rtsp-server/rtsp-mount-points.h:
4431 * gst/rtsp-server/rtsp-sdp.c:
4432 * gst/rtsp-server/rtsp-server.c:
4433 * gst/rtsp-server/rtsp-server.h:
4434 * gst/rtsp-server/rtsp-session-media.c:
4435 * gst/rtsp-server/rtsp-session-media.h:
4436 * gst/rtsp-server/rtsp-session-pool.c:
4437 * gst/rtsp-server/rtsp-session-pool.h:
4438 * gst/rtsp-server/rtsp-session.c:
4439 * gst/rtsp-server/rtsp-session.h:
4440 * gst/rtsp-server/rtsp-stream-transport.c:
4441 * gst/rtsp-server/rtsp-stream-transport.h:
4442 * gst/rtsp-server/rtsp-stream.c:
4443 * gst/rtsp-server/rtsp-stream.h:
4444 * tests/check/gst/media.c:
4445 rtsp: make object details private
4446 Make all object details private
4447 Add methods to access private bits
4449 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4451 * tests/check/Makefile.am:
4452 * tests/check/gst/media.c:
4453 tests: add media tests
4455 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4457 * gst/rtsp-server/rtsp-media.c:
4458 media: check if prepared for some methods
4459 Check that the media object is prepared before doing seek and getting the
4460 current position etc.
4461 Add some g_return checks.
4463 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4465 * tests/check/Makefile.am:
4466 * tests/check/gst/mediafactory.c:
4467 tests: add mediafactory test
4469 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4471 * gst/rtsp-server/rtsp-stream.c:
4472 stream: improve debug
4474 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4476 * gst/rtsp-server/rtsp-media.c:
4477 * gst/rtsp-server/rtsp-media.h:
4478 media: unref pipeline in finalize to avoid leaking it
4480 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4482 * gst/rtsp-server/rtsp-media-factory-uri.c:
4483 * gst/rtsp-server/rtsp-media.c:
4484 rtsp: use gst_object_unref on GstObjects
4486 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4488 * gst/rtsp-server/rtsp-media-factory.c:
4489 media-factory: require an url
4491 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4493 * examples/test-uri.c:
4494 examples: fix include
4496 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4498 * gst/rtsp-server/rtsp-server.h:
4499 server: remove unused include
4501 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4503 * tests/check/Makefile.am:
4504 * tests/check/gst/mountpoints.c:
4505 tests: add test for mountpoints
4507 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4509 * gst/rtsp-server/rtsp-client.c:
4510 client: fix factory leak
4511 Keep the factory in the state object only for authorization checks and make
4512 sure we unref it on failure. Also don't keep invalid objects in the state
4515 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4517 * gst/rtsp-server/rtsp-mount-points.c:
4518 mounts: add g_return_if guards
4520 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4522 * tests/check/gst/client.c:
4523 tests: add more tests
4525 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4527 * gst/rtsp-server/rtsp-client.c:
4528 client: improve debug
4530 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4532 * gst/rtsp-server/rtsp-client.c:
4533 client: improve debug and fix leaks
4534 Cleanup the uri and session when there is a bad request.
4536 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4541 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4543 * tests/check/gst/client.c:
4544 test: add test for session in options request
4546 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4548 * gst/rtsp-server/rtsp-client.c:
4549 client: use 454 when session can't be found
4550 We should use 454 when a session can't be found because there was no session
4551 pool configured in the server. This is not a server configuration problem
4552 because the server on which the request is done might not be the same one that
4553 will keep the sessions for us and so it does not need to support sessions.
4555 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4557 * gst/rtsp-server/rtsp-client.c:
4558 client: only free connection when there is one
4559 It's possible that the client doesn't have a connection when we try to free it.
4561 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4563 * tests/check/Makefile.am:
4564 * tests/check/gst/client.c:
4565 tests: add unit test for the client object
4567 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4569 * gst/rtsp-server/rtsp-client.c:
4570 client: small cleanup
4572 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4574 * gst/rtsp-server/rtsp-client.h:
4575 client: remove unused include
4577 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4579 * gst/rtsp-server/rtsp-client.c:
4580 client: fix compilation
4582 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4584 * gst/rtsp-server/rtsp-client.c:
4585 client: call destroy without the lock
4587 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4589 * gst/rtsp-server/rtsp-client.c:
4590 * gst/rtsp-server/rtsp-client.h:
4591 client: make the client usable without a socket
4592 Make a method to let the client handle a message and a callback when the client
4593 wants us to send a response message back. This makes it possible to also use the
4594 client object without the sockets, which should make it easier to test.
4596 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4598 * gst/rtsp-server/rtsp-client.c:
4599 * gst/rtsp-server/rtsp-client.h:
4600 client: small cleanup
4602 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4604 * docs/libs/gst-rtsp-server-sections.txt:
4605 * gst/rtsp-server/rtsp-client.c:
4606 * gst/rtsp-server/rtsp-client.h:
4607 * gst/rtsp-server/rtsp-server.c:
4608 client: remove reference to server
4609 We don't need to keep a ref to the server
4611 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4613 * gst/rtsp-server/rtsp-client.c:
4614 * gst/rtsp-server/rtsp-client.h:
4616 Also add some g_return_if()
4618 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4620 * gst/rtsp-server/rtsp-client.c:
4621 client: log more errors
4623 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4625 * gst/rtsp-server/rtsp-client.c:
4626 client: fix compilation
4628 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4630 * gst/rtsp-server/rtsp-client.c:
4631 * gst/rtsp-server/rtsp-client.h:
4632 client: add generic close-after-send support
4633 Add a property to send_response() to close the connection after the response has
4634 been sent to the client.
4636 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4639 * docs/libs/gst-rtsp-server-docs.sgml:
4640 * docs/libs/gst-rtsp-server-sections.txt:
4641 * docs/libs/gst-rtsp-server.types:
4642 * examples/test-auth.c:
4643 * examples/test-launch.c:
4644 * examples/test-mp4.c:
4645 * examples/test-multicast.c:
4646 * examples/test-multicast2.c:
4647 * examples/test-ogg.c:
4648 * examples/test-readme.c:
4649 * examples/test-sdp.c:
4650 * examples/test-uri.c:
4651 * examples/test-video.c:
4652 * gst/rtsp-server/Makefile.am:
4653 * gst/rtsp-server/rtsp-auth.h:
4654 * gst/rtsp-server/rtsp-client.c:
4655 * gst/rtsp-server/rtsp-client.h:
4656 * gst/rtsp-server/rtsp-media-mapping.c:
4657 * gst/rtsp-server/rtsp-media-mapping.h:
4658 * gst/rtsp-server/rtsp-mount-points.c:
4659 * gst/rtsp-server/rtsp-mount-points.h:
4660 * gst/rtsp-server/rtsp-server.c:
4661 * gst/rtsp-server/rtsp-server.h:
4662 * gst/rtsp-server/rtsp-session-media.c:
4663 * gst/rtsp-server/rtsp-session-pool.c:
4664 * gst/rtsp-server/rtsp-session-pool.h:
4665 * tests/check/gst/rtspserver.c:
4666 MediaMapping -> MountPoints
4667 Describes better what the object manages.
4669 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4672 configure: bump required version of -base
4674 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4676 * gst/rtsp-server/rtsp-media.c:
4679 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4681 * gst/rtsp-server/rtsp-media.c:
4682 * gst/rtsp-server/rtsp-media.h:
4683 media: support more Range formats
4684 Use the new -base methods to convert the Range string into a seek start and stop
4687 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4689 * examples/test-launch.c:
4690 examples: fix whitespace
4692 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4694 * examples/test-auth.c:
4695 test-auth: add example of how to remove sessions
4696 Add an example of the session filter api.
4698 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4700 * examples/test-uri.c:
4701 test-uri: remove mapping example
4703 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4705 * examples/test-uri.c:
4706 test-uri: fix callback signature
4708 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4710 * gst/rtsp-server/rtsp-media-factory.c:
4711 factory: keep ref to factory while media active
4712 While the media from a factory is alive, keep a ref to the factory.
4713 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
4715 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4717 * gst/rtsp-server/rtsp-media-factory-uri.c:
4718 factory-uri: add some debug
4720 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4722 * gst/rtsp-server/rtsp-stream.c:
4723 stream: set udp sources to PLAYING
4724 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
4725 so that it doesn't cause our pipeline to produce ASYNC-DONE.
4727 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4729 * gst/rtsp-server/rtsp-media-factory-uri.c:
4730 factory-uri: take ref to factory
4731 Take a ref to the factory that we place in our list.
4733 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4735 * tests/Makefile.am:
4736 * tests/test-reuse.c:
4737 test: add test for server reuse
4738 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
4740 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
4742 * gst/rtsp-server/rtsp-server.c:
4743 server: start and stop multiple times
4744 Stop listening on the RTSP port when the GSource is removed, so clients
4745 can't connect and the server can be started again.
4746 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
4748 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4750 * gst/rtsp-server/rtsp-server.c:
4751 server: fix small leak
4753 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4755 * gst/rtsp-server/rtsp-media.c:
4756 media: unref source in finish_unprepare
4757 The source is created in prepare, unref it in finish_unprepare.
4758 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
4760 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
4762 * gst/rtsp-server/rtsp-client.c:
4763 * gst/rtsp-server/rtsp-media.c:
4764 rtsp-media: remove bus watch before finalizing
4765 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
4766 * An extra media ref is added for the bus watch. This extra ref is unreffed by
4767 the GDestroyNotify function.
4768 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
4769 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
4770 gst_rtsp_media_unprepare before unreffing the media.
4771 This way, the bus watch will be removed before the media is finalized.
4772 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
4774 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
4776 * gst/rtsp-server/rtsp-client.c:
4777 * gst/rtsp-server/rtsp-client.h:
4778 client: wait until the TEARDOWN response is sent to close the connection
4779 Responses can be sent async so we need to wait until the TEARDOWN response has
4780 been written before we close the connection to the client. This avoids the risk
4781 of writing/polling closed sockets.
4782 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
4784 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
4786 * gst/rtsp-server/rtsp-stream.c:
4787 rtsp-stream: plug socket leak
4788 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
4790 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
4793 Automatic update of common submodule
4794 From 6bb6951 to a72faea
4796 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
4798 * gst/rtsp-server/rtsp-media-factory-uri.c:
4799 rtsp-server: don't use deprecated API
4801 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
4803 * gst/rtsp-server/rtsp-client.c:
4804 rtsp-client: fix unused-but-set-variable compiler warning
4805 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
4807 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4810 * docs/libs/gst-rtsp-server-sections.txt:
4811 * gst/rtsp-server/rtsp-client.c:
4814 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4816 * examples/Makefile.am:
4817 * examples/test-multicast2.c:
4818 examples: add another multicast example
4819 Add an example for how to configure separate multicast ranges for each media
4822 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4824 * examples/test-multicast.c:
4827 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4829 * gst/rtsp-server/rtsp-client.c:
4830 * gst/rtsp-server/rtsp-media.c:
4831 * gst/rtsp-server/rtsp-session-media.c:
4832 * gst/rtsp-server/rtsp-session-media.h:
4833 * gst/rtsp-server/rtsp-stream-transport.c:
4834 * gst/rtsp-server/rtsp-stream-transport.h:
4835 stream: use the address managed by the stream
4836 Use the address managed by the stream for multicast. This allows us to have 1
4837 multicast address for each stream.
4838 Because the address is now managed by the stream we don't have to pass it around
4840 Set the address pool on the streams.
4842 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4844 * gst/rtsp-server/rtsp-client.c:
4845 * gst/rtsp-server/rtsp-media.c:
4846 * gst/rtsp-server/rtsp-stream.c:
4849 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4851 * gst/rtsp-server/rtsp-media.c:
4852 * gst/rtsp-server/rtsp-media.h:
4853 media: add signal for new streams
4854 This allows applications to listen for new streams and configure properties on
4855 them, like the address pool.
4857 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4859 * gst/rtsp-server/rtsp-media.c:
4860 media: configure address pool in new streams
4862 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4864 * gst/rtsp-server/rtsp-stream.c:
4865 * gst/rtsp-server/rtsp-stream.h:
4866 stream: add methods to deal with address pool
4867 Add methods to get and set the address pool for the stream
4868 Add method to allocate and get the multicast addresses for this stream.
4870 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4872 * docs/libs/gst-rtsp-server-sections.txt:
4873 * gst/rtsp-server/rtsp-media.c:
4874 * gst/rtsp-server/rtsp-media.h:
4875 media: remove MTU property
4876 It is a stream property
4878 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4880 * gst/rtsp-server/rtsp-client.c:
4881 client: set blocksize only on stream
4882 Set the blocksize only on the current stream.
4884 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4886 * gst/rtsp-server/rtsp-stream.c:
4887 stream: share src and sink sockets
4888 the allocated socket is in the used-socket property, not socket.
4890 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4892 * gst/rtsp-server/rtsp-address-pool.c:
4893 * gst/rtsp-server/rtsp-address-pool.h:
4894 * gst/rtsp-server/rtsp-client.c:
4895 * gst/rtsp-server/rtsp-session-media.c:
4896 * gst/rtsp-server/rtsp-session-media.h:
4897 * gst/rtsp-server/rtsp-stream-transport.c:
4898 * gst/rtsp-server/rtsp-stream-transport.h:
4899 * tests/check/gst/addresspool.c:
4900 rtsp: make address-pool return an address object
4901 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
4902 store more info in the structure and allows us to more easily return the address
4903 to the right pool when no longer needed.
4904 Pass the address to the StreamTransport so that we can return it to the pool
4905 when the stream transport is freed or changed.
4907 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4909 * examples/Makefile.am:
4910 * examples/test-multicast.c:
4911 examples: add multicast example
4912 Show how to set up the multicast address pool so that media can be
4913 server with multicast.
4915 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4917 * gst/rtsp-server/rtsp-client.c:
4918 * gst/rtsp-server/rtsp-media-factory.c:
4919 * gst/rtsp-server/rtsp-media-factory.h:
4920 * gst/rtsp-server/rtsp-media.c:
4921 * gst/rtsp-server/rtsp-media.h:
4922 rtsp: use AddressPool
4923 Remove the multicast_group property.
4924 Use the configured addresspool to allocate multicast addresses.
4926 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4928 * gst/rtsp-server/rtsp-address-pool.c:
4929 * gst/rtsp-server/rtsp-address-pool.h:
4930 address-pool: add clear method
4932 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4934 * gst/rtsp-server/rtsp-address-pool.c:
4935 address-pool: small cleanups
4937 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4939 * tests/check/Makefile.am:
4940 * tests/check/gst/addresspool.c:
4941 tests: add addresspool unit test
4943 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4945 * gst/rtsp-server/Makefile.am:
4946 * gst/rtsp-server/rtsp-address-pool.c:
4947 * gst/rtsp-server/rtsp-address-pool.h:
4948 address-pool: add object to manage multicast addresses
4949 Make an object that can manage a rage of multicast addresses and ports.
4951 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4953 * gst/rtsp-server/rtsp-server.c:
4954 server: set default max-threads property
4956 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4958 * gst/rtsp-server/rtsp-media.c:
4959 media: wait for concurrent _prepare
4960 If a prepare is busy, wait for the result.
4962 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4964 * gst/rtsp-server/rtsp-media.c:
4965 media: add lock around message handler
4966 We don't want to dispatch messages while we are still processing the result of
4969 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4971 * gst/rtsp-server/rtsp-media.c:
4972 * gst/rtsp-server/rtsp-media.h:
4973 media: add lock to protect state changes
4975 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4977 * gst/rtsp-server/rtsp-stream.c:
4978 * gst/rtsp-server/rtsp-stream.h:
4981 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4983 * gst/rtsp-server/rtsp-stream-transport.c:
4984 * gst/rtsp-server/rtsp-stream-transport.h:
4985 * gst/rtsp-server/rtsp-stream.c:
4986 stream-transport: add keep-alive method
4988 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4990 * gst/rtsp-server/rtsp-stream-transport.c:
4991 * gst/rtsp-server/rtsp-stream-transport.h:
4992 * gst/rtsp-server/rtsp-stream.c:
4993 stream-transport: add method to handle RTP/RTCP
4994 Call new methods instead of poking into the structures directly.
4996 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4998 * gst/rtsp-server/rtsp-session-media.c:
4999 * gst/rtsp-server/rtsp-session-media.h:
5000 session-media: add locking
5002 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5004 * gst/rtsp-server/rtsp-session.c:
5005 * gst/rtsp-server/rtsp-session.h:
5006 session: add locking
5008 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5010 * gst/rtsp-server/rtsp-server.c:
5011 server: free old socket
5013 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5015 * gst/rtsp-server/rtsp-media-mapping.c:
5016 * gst/rtsp-server/rtsp-media-mapping.h:
5017 mapping: add locking
5019 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5021 * gst/rtsp-server/rtsp-media-factory.c:
5022 media-factory: add locking
5024 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5026 * gst/rtsp-server/rtsp-auth.c:
5027 * gst/rtsp-server/rtsp-auth.h:
5030 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5032 * gst/rtsp-server/rtsp-server.c:
5033 * gst/rtsp-server/rtsp-server.h:
5034 server: add max-thread property
5036 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5038 * gst/rtsp-server/rtsp-server.c:
5039 * gst/rtsp-server/rtsp-server.h:
5040 server: use a threadpool for the mainloops
5042 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5044 * gst/rtsp-server/rtsp-client.c:
5045 * gst/rtsp-server/rtsp-client.h:
5046 client: rename method
5047 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
5048 don't really create the client from the socket, we use the socket for the
5051 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5053 * gst/rtsp-server/rtsp-client.c:
5054 * gst/rtsp-server/rtsp-client.h:
5055 * gst/rtsp-server/rtsp-server.c:
5056 server: rework maincontext handling in clients
5057 Make a separate method to attach a client to a MainContext.
5058 Let the server decide in what GMainContext the client will operate and give this
5059 context to the client in attach. Then the server can later decide to use a
5060 separate thread for each client or just use the mainthread.
5062 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5064 * gst/rtsp-server/rtsp-client.c:
5065 * gst/rtsp-server/rtsp-session.c:
5066 * gst/rtsp-server/rtsp-session.h:
5067 session: move session header code in session object
5069 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
5073 * examples/test-auth.c:
5074 * examples/test-launch.c:
5075 * examples/test-mp4.c:
5076 * examples/test-ogg.c:
5077 * examples/test-readme.c:
5078 * examples/test-sdp.c:
5079 * examples/test-uri.c:
5080 * examples/test-video.c:
5081 * gst/rtsp-server/rtsp-auth.c:
5082 * gst/rtsp-server/rtsp-auth.h:
5083 * gst/rtsp-server/rtsp-client.c:
5084 * gst/rtsp-server/rtsp-client.h:
5085 * gst/rtsp-server/rtsp-media-factory-uri.c:
5086 * gst/rtsp-server/rtsp-media-factory-uri.h:
5087 * gst/rtsp-server/rtsp-media-factory.c:
5088 * gst/rtsp-server/rtsp-media-factory.h:
5089 * gst/rtsp-server/rtsp-media-mapping.c:
5090 * gst/rtsp-server/rtsp-media-mapping.h:
5091 * gst/rtsp-server/rtsp-media.c:
5092 * gst/rtsp-server/rtsp-media.h:
5093 * gst/rtsp-server/rtsp-params.c:
5094 * gst/rtsp-server/rtsp-params.h:
5095 * gst/rtsp-server/rtsp-sdp.c:
5096 * gst/rtsp-server/rtsp-sdp.h:
5097 * gst/rtsp-server/rtsp-server.c:
5098 * gst/rtsp-server/rtsp-server.h:
5099 * gst/rtsp-server/rtsp-session-media.c:
5100 * gst/rtsp-server/rtsp-session-media.h:
5101 * gst/rtsp-server/rtsp-session-pool.c:
5102 * gst/rtsp-server/rtsp-session-pool.h:
5103 * gst/rtsp-server/rtsp-session.c:
5104 * gst/rtsp-server/rtsp-session.h:
5105 * gst/rtsp-server/rtsp-stream-transport.c:
5106 * gst/rtsp-server/rtsp-stream-transport.h:
5107 * gst/rtsp-server/rtsp-stream.c:
5108 * gst/rtsp-server/rtsp-stream.h:
5109 * tests/check/gst/rtspserver.c:
5110 * tests/test-cleanup.c:
5113 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
5115 * gst/rtsp-server/rtsp-media.c:
5116 * gst/rtsp-server/rtsp-session-media.c:
5117 * gst/rtsp-server/rtsp-session.c:
5118 rtsp-server: added annotations to indicate type of ownership transfer of return values
5119 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5121 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5124 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
5126 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
5129 * bindings/Makefile.am:
5130 * bindings/vala/Makefile.am:
5131 * bindings/vala/gst-rtsp-server-0.10.deps:
5132 * bindings/vala/gst-rtsp-server-0.10.vapi:
5133 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
5134 * bindings/vala/packages/gst-rtsp-server-0.10.files:
5135 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5136 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5137 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
5139 bindings: remove vala bindings
5140 They'll be reunited with the other GStreamer bindings
5141 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5143 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5145 * gst/rtsp-server/rtsp-client.c:
5146 * gst/rtsp-server/rtsp-session-media.c:
5147 * gst/rtsp-server/rtsp-session-media.h:
5148 * gst/rtsp-server/rtsp-stream-transport.c:
5149 * gst/rtsp-server/rtsp-stream-transport.h:
5150 rtsp: only create transport when needed
5151 Only create the StreamTransport when configured.
5153 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5155 * gst/rtsp-server/rtsp-client.c:
5156 client: small cleanup
5158 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5160 * gst/rtsp-server/rtsp-client.c:
5161 * gst/rtsp-server/rtsp-client.h:
5162 * gst/rtsp-server/rtsp-stream-transport.c:
5163 * gst/rtsp-server/rtsp-stream-transport.h:
5164 rtsp: refactor configuration of transport
5165 Move the configuration of the transport to a place where it makes
5168 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5170 * gst/rtsp-server/rtsp-client.c:
5171 client: refactor transport parsing
5173 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5175 * gst/rtsp-server/rtsp-client.c:
5176 client: refuse to change the MTU on shared media
5177 If we change the MTU of chared media, it changes for all clients.
5178 We don't want to set the MTU to something large for clients that
5181 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5183 * examples/test-mp4.c:
5184 * gst/rtsp-server/rtsp-media.c:
5185 small fixes to docs and debug
5187 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5189 * gst/rtsp-server/rtsp-stream.c:
5190 stream: transports must already have been removed
5192 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5194 * gst/rtsp-server/rtsp-media.c:
5195 * gst/rtsp-server/rtsp-stream.c:
5196 * gst/rtsp-server/rtsp-stream.h:
5197 stream: improve join and leave of the pipeline
5199 Do the cleanup properly
5202 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5204 * gst/rtsp-server/rtsp-media.c:
5205 media: move unprepare below default implementation
5206 Makes it easier to find the default implementation
5208 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5210 * gst/rtsp-server/rtsp-media.c:
5211 media: signal unprepared when we actually finish
5213 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5215 * gst/rtsp-server/rtsp-media.c:
5216 media: no need to unlock, unprepare does that when needed
5218 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5220 * docs/libs/gst-rtsp-server-sections.txt:
5221 * gst/rtsp-server/rtsp-media-factory.h:
5222 * gst/rtsp-server/rtsp-media-mapping.c:
5223 * gst/rtsp-server/rtsp-media.h:
5224 * gst/rtsp-server/rtsp-params.c:
5225 * gst/rtsp-server/rtsp-server.c:
5226 * gst/rtsp-server/rtsp-session-pool.h:
5227 * gst/rtsp-server/rtsp-session.c:
5228 * gst/rtsp-server/rtsp-session.h:
5229 * gst/rtsp-server/rtsp-stream-transport.h:
5230 * gst/rtsp-server/rtsp-stream.h:
5233 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5235 * gst/rtsp-server/rtsp-client.c:
5236 * gst/rtsp-server/rtsp-media-mapping.h:
5237 * gst/rtsp-server/rtsp-media.c:
5238 * gst/rtsp-server/rtsp-media.h:
5239 * gst/rtsp-server/rtsp-server.h:
5240 * gst/rtsp-server/rtsp-stream.c:
5241 * gst/rtsp-server/rtsp-stream.h:
5242 rtsp: fix MTU setting
5243 Fix setting of the MTU. There is no need for a vmethod.
5245 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5250 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5253 configure: bump version number after refactoring
5255 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5257 * gst/rtsp-server/Makefile.am:
5258 * gst/rtsp-server/rtsp-client.c:
5259 * gst/rtsp-server/rtsp-client.h:
5260 * gst/rtsp-server/rtsp-media-factory-uri.c:
5261 * gst/rtsp-server/rtsp-media-factory.c:
5262 * gst/rtsp-server/rtsp-media-factory.h:
5263 * gst/rtsp-server/rtsp-media.c:
5264 * gst/rtsp-server/rtsp-media.h:
5265 * gst/rtsp-server/rtsp-sdp.c:
5266 * gst/rtsp-server/rtsp-session-media.c:
5267 * gst/rtsp-server/rtsp-session-media.h:
5268 * gst/rtsp-server/rtsp-session.c:
5269 * gst/rtsp-server/rtsp-session.h:
5270 * gst/rtsp-server/rtsp-stream-transport.c:
5271 * gst/rtsp-server/rtsp-stream-transport.h:
5272 * gst/rtsp-server/rtsp-stream.c:
5273 * gst/rtsp-server/rtsp-stream.h:
5274 rtsp: massive refactoring
5275 Make GObjects from the remaining simple structures.
5276 Remove GstRTSPSessionStream, it's not needed.
5277 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
5278 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
5279 a GstRTSPStream should be transported to a client.
5280 Rename GstRTSPMediaFactory::get_element -> create_element because that
5281 more accurately describes what it does.
5282 Make nice methods instead of poking in the structures.
5283 Move some methods inside the relevant object source code.
5284 Use GPtrArray to store objects instead of plain arrays, it is more
5285 natural and allows us to more easily clean up.
5286 Move the allocation of udp ports to the Stream object. The Stream object
5287 contains the elements needed to stream the media to a client.
5288 Improve the prepare and unprepare methods. Unprepare should now undo
5289 everything prepare did. Improve also async unprepare when doing EOS on
5290 shutdown. Make sure we always unprepare correctly.
5292 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
5294 * gst/rtsp-server/rtsp-client.c:
5295 rtsp-client: Unref server address clients connected to
5296 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
5298 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
5300 * gst/rtsp-server/rtsp-server.c:
5301 rtsp-server: don't ref server socket if it is NULL
5302 Fixes test_bind_already_in_use unit test again after commit 6a497440.
5303 https://bugzilla.gnome.org/show_bug.cgi?id=686644
5305 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
5307 * tests/check/Makefile.am:
5308 tests: Add libgio link dependency
5309 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
5311 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5313 * gst/rtsp-server/rtsp-media-mapping.c:
5314 * gst/rtsp-server/rtsp-media-mapping.h:
5315 rtsp-media-mapping: rename find_media vfunc to find_factory
5316 The virtual method and class method should have the same name
5317 so it is correctly represented in GIR file
5318 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5320 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5322 * gst/rtsp-server/rtsp-auth.c:
5323 * gst/rtsp-server/rtsp-client.c:
5324 * gst/rtsp-server/rtsp-media-factory-uri.c:
5325 * gst/rtsp-server/rtsp-media-factory.c:
5326 * gst/rtsp-server/rtsp-media-mapping.c:
5327 * gst/rtsp-server/rtsp-media.c:
5328 * gst/rtsp-server/rtsp-server.c:
5329 * gst/rtsp-server/rtsp-session-pool.c:
5330 * gst/rtsp-server/rtsp-session.c:
5331 rtsp-server: fixed comments and GIR annotations
5332 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5334 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
5336 * gst/rtsp-server/rtsp-media-mapping.c:
5337 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
5339 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
5341 * gst/rtsp-server/rtsp-server.c:
5342 rtsp-server: allow binding on port 0 (binds on a random port)
5344 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
5346 * gst/rtsp-server/rtsp-server.c:
5347 * gst/rtsp-server/rtsp-server.h:
5348 rtsp-server: add bound-port property
5349 bound-port can be used to retrieve the port number when the server is bound on
5350 port 0, which binds on a random port.
5352 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
5354 * gst/rtsp-server/rtsp-media-factory.c:
5355 * gst/rtsp-server/rtsp-media-factory.h:
5356 rtsp-media-factory: make ::get_element overridable by GI bindings
5357 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
5358 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
5359 as the invoker for ::get_element(), making it overridable by GI generated
5362 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5364 * gst/rtsp-server/rtsp-media-factory-uri.c:
5365 rtsp-media-factory-uri: don't autoplug parsers in a loop
5366 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
5369 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5371 * gst/rtsp-server/Makefile.am:
5372 Explicitly link against gio. Fix link error on mac.
5374 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5376 * gst/rtsp-server/rtsp-session.c:
5377 session: add ttl to the transport header in SETUP
5378 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
5380 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5382 * gst/rtsp-server/rtsp-client.c:
5383 * gst/rtsp-server/rtsp-client.h:
5384 * gst/rtsp-server/rtsp-media.c:
5385 client: Use client transport settings for multicast if allowed.
5386 This patch makes it possible for the client to send transport settings for
5387 multicast (destination && ttl). Client settings must be explicitly allowed or
5388 the server will use its own settings.
5389 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
5391 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
5394 Automatic update of common submodule
5395 From 6c0b52c to 6bb6951
5397 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
5399 * gst/rtsp-server/rtsp-client.c:
5400 rtsp-client: do not destroy the rtsp watch
5401 Don't destroy the client watch while dispatching. The rtsp watch is
5402 automatically destroyed after the rtsp watch function closed() has
5404 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
5406 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5409 Automatic update of common submodule
5410 From 4f962f7 to 6c0b52c
5412 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
5414 * gst/rtsp-server/rtsp-media.c:
5415 media: fix check for seekability
5417 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5419 * gst/rtsp-server/rtsp-client.c:
5420 client: use more GIO
5421 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
5423 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5425 * gst/rtsp-server/rtsp-server.c:
5426 server: remove obsolete includes
5428 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5430 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
5431 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
5432 be available in "on_new_ssrc". The transports are added in
5433 gst_rtsp_media_set_state when going to PLAYING state. However,
5434 "on_new_ssrc" might be called before this happens.
5435 https://bugzilla.gnome.org/show_bug.cgi?id=683304
5437 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5439 * gst/rtsp-server/rtsp-client.c:
5440 * gst/rtsp-server/rtsp-client.h:
5441 rtsp-client: add signals for rtsp requests (fixes #683287)
5443 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5445 * gst/rtsp-server/rtsp-client.c:
5446 * gst/rtsp-server/rtsp-client.h:
5447 add new-session signal to rtsp-client (fixes #683058)
5449 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
5452 Automatic update of common submodule
5453 From 668acee to 4f962f7
5455 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
5457 * gst/rtsp-server/rtsp-server.c:
5458 * tests/check/gst/rtspserver.c:
5459 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
5460 Do not assume that *error is set in g_socket_address_enumerator_next.
5461 Added test_bind_already_in_use unit-test.
5462 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
5464 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
5467 Automatic update of common submodule
5468 From 94ccf4c to 668acee
5470 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
5472 * gst/rtsp-server/rtsp-client.c:
5473 * gst/rtsp-server/rtsp-client.h:
5474 rtsp-client: make create_sdp virtual method
5475 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
5477 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5480 Automatic update of common submodule
5481 From 98e386f to 94ccf4c
5483 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5485 * gst/rtsp-server/rtsp-client.c:
5488 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5490 * gst/rtsp-server/rtsp-client.c:
5491 * gst/rtsp-server/rtsp-client.h:
5492 * gst/rtsp-server/rtsp-server.c:
5493 * gst/rtsp-server/rtsp-server.h:
5494 rtsp-server: use an existing socket to establish HTTP tunnel
5495 Make it possible to transfer a socket from an HTTP server to be used as
5496 an RTSP over HTTP tunnel.
5498 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
5500 * gst/rtsp-server/rtsp-client.c:
5501 * gst/rtsp-server/rtsp-media.c:
5502 * gst/rtsp-server/rtsp-media.h:
5503 rtsp: Handle the blocksize parameter
5504 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
5506 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
5508 * tests/check/Makefile.am:
5509 * tests/check/gst/rtspserver.c:
5510 Have unit test get header from source dir, not installed dir
5511 This makes compilation of unit tests work in a build directory other
5512 than the source directory.
5513 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
5515 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
5517 * gst/rtsp-server/rtsp-media.c:
5518 rtsp-media: update for gst_element_make_from_uri() changes
5520 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
5523 * tests/Makefile.am:
5524 * tests/check/Makefile.am:
5525 * tests/check/gst/rtspserver.c:
5527 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
5529 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
5531 * gst/rtsp-server/rtsp-media.c:
5532 rtsp-media: don't collect media stats when going to NULL
5533 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
5535 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5537 * gst/rtsp-server/rtsp-client.c:
5538 client: don't leak transports
5540 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
5542 * gst/rtsp-server/rtsp-client.c:
5543 rtsp-client: free transport on no_stream in SETUP handler
5545 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
5547 * gst/rtsp-server/rtsp-client.c:
5548 rtsp-client: changed session media iteration
5549 In client_unlink_session: now don't iterate in session->medias
5550 list where items are removed by gst_rtsp_session_release_media.
5551 Instead, repeatedly remove the first item.
5553 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
5555 * gst/rtsp-server/rtsp-client.c:
5556 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
5557 GstRTSPSessionMedia is not a GObject type. When the
5558 GstRTSPSession is freed, it will free the media.
5560 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
5562 * gst/rtsp-server/rtsp-media-factory.c:
5563 factory: plug pad leak in collect_streams
5564 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
5565 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
5566 will take one reference, and the other reference will otherwise
5569 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5572 configure: suppress some warnings when debug is disabled
5573 Warnings about unused variables should be suppressed if core has the
5574 debug system disabled.
5575 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5577 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5579 * docs/libs/Makefile.am:
5580 docs: fix build in uninstalled setup
5581 Include gst-plugins-base libs properly.
5583 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
5585 * docs/libs/gst-rtsp-server.types:
5586 docs: include headers defining rtsp-server object types
5587 Fixes compiler warnings during docs build.
5588 https://bugzilla.gnome.org/show_bug.cgi?id=676824
5590 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
5593 configure: Add warning flags for compiler when configuring
5594 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5596 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5599 Automatic update of common submodule
5600 From 03a0e57 to 98e386f
5602 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5605 Automatic update of common submodule
5606 From 1fab359 to 03a0e57
5608 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
5610 * gst/rtsp-server/rtsp-client.c:
5611 client: fix GSocketAddress leak in gst_rtsp_client_accept
5612 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
5614 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5617 Automatic update of common submodule
5618 From f1b5a96 to 1fab359
5620 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5623 Automatic update of common submodule
5624 From 92b7266 to f1b5a96
5626 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5629 Automatic update of common submodule
5630 From ec1c4a8 to 92b7266
5632 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5635 Automatic update of common submodule
5636 From 3429ba6 to ec1c4a8
5638 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
5640 * gst/rtsp-server/rtsp-auth.c:
5641 * gst/rtsp-server/rtsp-client.c:
5642 * gst/rtsp-server/rtsp-media-factory-uri.c:
5643 * gst/rtsp-server/rtsp-server.c:
5644 rtsp: fix compiler warnings
5645 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
5647 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5650 Automatic update of common submodule
5651 From dc70203 to 3429ba6
5653 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5655 * gst/rtsp-server/rtsp-client.c:
5656 * gst/rtsp-server/rtsp-media-factory.c:
5657 * gst/rtsp-server/rtsp-media-factory.h:
5658 * gst/rtsp-server/rtsp-media.c:
5659 * gst/rtsp-server/rtsp-media.h:
5660 * gst/rtsp-server/rtsp-server.c:
5661 * gst/rtsp-server/rtsp-server.h:
5662 * gst/rtsp-server/rtsp-session-pool.c:
5663 * gst/rtsp-server/rtsp-session-pool.h:
5664 rtsp-server: port to new thread API
5666 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5669 Automatic update of common submodule
5670 From 6db25be to dc70203
5672 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5674 * gst/rtsp-server/rtsp-auth.c:
5675 * gst/rtsp-server/rtsp-auth.h:
5676 * gst/rtsp-server/rtsp-client.c:
5677 rtsp-server: Fix compilation and compiler warnings
5679 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5683 * gst/rtsp-server/Makefile.am:
5684 configure: Modernize autotools setup a bit
5685 Also we now only create tar.bz2 and tar.xz tarballs.
5687 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5690 Automatic update of common submodule
5691 From 464fe15 to 6db25be
5693 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5696 Automatic update of common submodule
5697 From 7fda524 to 464fe15
5699 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5702 * docs/libs/Makefile.am:
5703 * docs/version.entities.in:
5705 * gst/rtsp-server/Makefile.am:
5706 * pkgconfig/Makefile.am:
5707 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5708 * pkgconfig/gstreamer-rtsp-server.pc.in:
5709 * tests/Makefile.am:
5710 rtsp-server: Update versioning
5712 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5714 Merge remote-tracking branch 'origin/0.10'
5716 gst/rtsp-server/rtsp-session-pool.c
5718 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5720 * gst/rtsp-server/rtsp-session-pool.c:
5721 rtsp-server: Don't use deprecated GLib API
5723 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5725 Replace master with 0.11
5727 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5729 Merge branch 'master' into 0.11
5731 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5733 Merge branch 'master' into 0.11
5735 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
5738 A couple minor typo fixes
5740 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5742 * gst/rtsp-server/rtsp-media.c:
5743 media: fix state of the appqueue
5745 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5747 * gst/rtsp-server/rtsp-media-factory-uri.c:
5748 factory: use videoconvert
5750 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5752 * gst/rtsp-server/rtsp-media-factory-uri.c:
5753 factory: change to new style caps
5755 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5757 * gst/rtsp-server/rtsp-client.c:
5758 * gst/rtsp-server/rtsp-client.h:
5759 * gst/rtsp-server/rtsp-media-factory-uri.c:
5760 * gst/rtsp-server/rtsp-media.c:
5761 * gst/rtsp-server/rtsp-server.c:
5762 * gst/rtsp-server/rtsp-server.h:
5763 * gst/rtsp-server/rtsp-session-pool.c:
5764 rtsp-server: port to GIO
5767 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5770 configure: fix build
5772 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5775 docs: fix for gst_rtsp_server_set_port() -> _set_service()
5776 https://bugzilla.gnome.org/show_bug.cgi?id=666548
5778 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5781 * examples/Makefile.am:
5782 First rule of gst-rtsp-server club: don't talk about gst-phonon
5784 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5787 * pkgconfig/Makefile.am:
5788 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
5789 * pkgconfig/gst-rtsp-server.pc.in:
5790 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5791 * pkgconfig/gstreamer-rtsp-server.pc.in:
5792 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
5793 For consistency with all other modules.
5795 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5797 * gst/rtsp-server/rtsp-client.c:
5798 rtsp-client: update for new map API
5800 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5803 * bindings/Makefile.am:
5804 * bindings/python/Makefile.am:
5805 * bindings/python/arg-types.py:
5806 * bindings/python/codegen/Makefile.am:
5807 * bindings/python/codegen/__init__.py:
5808 * bindings/python/codegen/argtypes.py:
5809 * bindings/python/codegen/code-coverage.py:
5810 * bindings/python/codegen/codegen.py:
5811 * bindings/python/codegen/definitions.py:
5812 * bindings/python/codegen/defsparser.py:
5813 * bindings/python/codegen/docextract.py:
5814 * bindings/python/codegen/docgen.py:
5815 * bindings/python/codegen/fileprefix.override:
5816 * bindings/python/codegen/fileprefixmodule.c:
5817 * bindings/python/codegen/h2def.py:
5818 * bindings/python/codegen/mergedefs.py:
5819 * bindings/python/codegen/mkskel.py:
5820 * bindings/python/codegen/override.py:
5821 * bindings/python/codegen/reversewrapper.py:
5822 * bindings/python/codegen/scmexpr.py:
5823 * bindings/python/rtspserver-types.defs:
5824 * bindings/python/rtspserver.defs:
5825 * bindings/python/rtspserver.override:
5826 * bindings/python/rtspservermodule.c:
5827 * bindings/python/test.py:
5829 python: remove pygst-based python bindings
5830 pygi is the future, apparently.
5832 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
5835 Automatic update of common submodule
5836 From c463bc0 to 7fda524
5838 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5841 Automatic update of common submodule
5842 From 2a59016 to c463bc0
5844 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5847 Automatic update of common submodule
5848 From 0807187 to 2a59016
5850 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5853 Automatic update of common submodule
5854 From 11f0cd5 to 0807187
5856 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5858 * examples/test-auth.c:
5859 example: update for new caps
5861 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5863 * examples/test-video.c:
5864 * gst/rtsp-server/rtsp-client.c:
5865 * gst/rtsp-server/rtsp-media-factory-uri.c:
5866 * gst/rtsp-server/rtsp-media.c:
5867 * gst/rtsp-server/rtsp-media.h:
5868 * gst/rtsp-server/rtsp-session.c:
5869 * gst/rtsp-server/rtsp-session.h:
5870 rtsp-server: port some more to 0.11
5872 Remove bufferlist stuff
5874 Add queue before appsink now that preroll-queue-len is gone.
5875 Update for request pad changes.
5877 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5879 Merge branch 'master' into 0.11
5881 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
5883 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5884 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
5885 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
5887 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
5889 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5890 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
5891 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
5893 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5895 Merge branch 'master' into 0.11
5897 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5899 * gst/rtsp-server/rtsp-media.c:
5900 * gst/rtsp-server/rtsp-media.h:
5901 media: add a seekable boolean
5902 Maintain the seekable state with a new variable instead of reusing the
5905 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
5907 * gst/rtsp-server/rtsp-media.c:
5908 Disallow seek in live media
5910 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5912 Merge branch 'master' into 0.11
5914 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
5916 * gst/rtsp-server/rtsp-server.c:
5917 #ifdef statements for windows socket creation were missing
5919 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
5922 Automatic update of common submodule
5923 From a39eb83 to 11f0cd5
5925 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
5928 Automatic update of common submodule
5929 From 605cd9a to a39eb83
5931 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5933 Merge branch 'master' into 0.11
5935 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5937 * gst/rtsp-server/rtsp-client.c:
5938 client: use method to access property
5940 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5942 * gst/rtsp-server/rtsp-media-factory.c:
5943 * gst/rtsp-server/rtsp-media-factory.h:
5944 media-factory: add protocols property
5945 Add a property to configure the allowed protocols in the media created from the
5948 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5950 * gst/rtsp-server/rtsp-media-factory.c:
5951 * gst/rtsp-server/rtsp-media-factory.h:
5952 media-factory: add media-configure signal
5953 Add signal to allow the application to configure the media after it was created
5956 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5958 * gst/rtsp-server/rtsp-client.c:
5959 client: use method to access property
5961 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5963 * gst/rtsp-server/rtsp-media-factory.c:
5964 * gst/rtsp-server/rtsp-media-factory.h:
5965 media-factory: add protocols property
5966 Add a property to configure the allowed protocols in the media created from the
5969 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5971 * gst/rtsp-server/rtsp-media-factory.c:
5972 * gst/rtsp-server/rtsp-media-factory.h:
5973 media-factory: add media-configure signal
5974 Add signal to allow the application to configure the media after it was created
5977 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5979 Merge branch 'master' into 0.11
5981 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5983 * gst/rtsp-server/rtsp-client.c:
5984 client: use media multicast group
5986 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5988 * gst/rtsp-server/rtsp-media-factory.h:
5989 * gst/rtsp-server/rtsp-server.h:
5990 * gst/rtsp-server/rtsp-session-pool.h:
5991 * gst/rtsp-server/rtsp-session.h:
5994 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
5996 * gst/rtsp-server/rtsp-client.c:
5997 * gst/rtsp-server/rtsp-sdp.h:
5998 sdp: copy and free the server ip address
5999 Copy and free the server ip address to make memory management easier later.
6001 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6003 * gst/rtsp-server/rtsp-media-factory.c:
6004 media-factory: configure multicast in media
6006 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6008 * gst/rtsp-server/rtsp-media.c:
6009 * gst/rtsp-server/rtsp-media.h:
6010 media: add property for multicast group
6011 Add a property to configure the multicast group in the media.
6012 Based on patches from Marc Leeman and Robert Krakora.
6014 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6016 * gst/rtsp-server/rtsp-media-factory.c:
6017 * gst/rtsp-server/rtsp-media-factory.h:
6018 media-factory: add property for multicast group
6019 Add a property to configure the multicast group in the media factory.
6020 Based on patches from Marc Leeman and Robert Krakora.
6022 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6024 * gst/rtsp-server/rtsp-client.c:
6025 client: do configuration of transport in one place
6026 Move the configuration of the transport destination address to where we also
6027 configure the other bits.
6029 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6031 * gst/rtsp-server/rtsp-client.c:
6032 client: use media multicast group
6034 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6036 * gst/rtsp-server/rtsp-media-factory.h:
6037 * gst/rtsp-server/rtsp-server.h:
6038 * gst/rtsp-server/rtsp-session-pool.h:
6039 * gst/rtsp-server/rtsp-session.h:
6042 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6044 * gst/rtsp-server/rtsp-client.c:
6045 * gst/rtsp-server/rtsp-sdp.h:
6046 sdp: copy and free the server ip address
6047 Copy and free the server ip address to make memory management easier later.
6049 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6051 * gst/rtsp-server/rtsp-media-factory.c:
6052 media-factory: configure multicast in media
6054 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6056 * gst/rtsp-server/rtsp-media.c:
6057 * gst/rtsp-server/rtsp-media.h:
6058 media: add property for multicast group
6059 Add a property to configure the multicast group in the media.
6060 Based on patches from Marc Leeman and Robert Krakora.
6062 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6064 * gst/rtsp-server/rtsp-media-factory.c:
6065 * gst/rtsp-server/rtsp-media-factory.h:
6066 media-factory: add property for multicast group
6067 Add a property to configure the multicast group in the media factory.
6068 Based on patches from Marc Leeman and Robert Krakora.
6070 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6072 * gst/rtsp-server/rtsp-client.c:
6073 client: do configuration of transport in one place
6074 Move the configuration of the transport destination address to where we also
6075 configure the other bits.
6077 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6079 Merge branch 'master' into 0.11
6081 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6083 * gst/rtsp-server/rtsp-client.c:
6084 client: destroy pipeline on client disconnect with no prior TEARDOWN.
6085 The problem occurs when the client abruptly closes the connection without
6086 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
6087 server is where the pipeline gets torn down. Since this handler is not called,
6088 the pipeline remains and is up and running. Subsequent clients get their own
6089 pipelines and if the do not issue TEARDOWNs then those pipelines will also
6090 remain up and running. This is a resource leak.
6092 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6094 Merge branch 'master' into 0.11
6096 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
6098 * gst/rtsp-server/rtsp-media-factory.c:
6099 * gst/rtsp-server/rtsp-media-factory.h:
6100 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
6101 For example, it can be used to retrieve source elements like appsrc, in a more
6102 convenient way than subclassing get_element.
6104 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6106 Merge branch 'master' into 0.11
6108 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
6110 * gst/rtsp-server/rtsp-server.c:
6111 rtsp-server: hold on to reference while using object
6113 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6115 * gst/rtsp-server/rtsp-media.c:
6118 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6121 configure: use unstable api
6123 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
6125 * gst/rtsp-server/rtsp-client.c:
6126 client: fix reference counting
6128 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
6130 * gst/rtsp-server/rtsp-client.c:
6131 * gst/rtsp-server/rtsp-media.c:
6132 fix compiler warnings about unused variables
6134 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
6136 * examples/test-launch.c:
6137 * examples/test-readme.c:
6138 * examples/test-uri.c:
6139 * examples/test-video.c:
6140 examples: tell rtsp uri when ready
6142 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
6145 Automatic update of common submodule
6146 From 69b981f to 605cd9a
6148 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6150 * gst/rtsp-server/rtsp-client.c:
6151 client: update for buffer API change
6153 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6155 * gst/rtsp-server/Makefile.am:
6156 Makefile.am: 0.10 => @GST_MAJORMINOR@
6158 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6160 * gst/rtsp-server/rtsp-media-factory-uri.c:
6161 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
6163 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6165 * gst/rtsp-server/.gitignore:
6166 .gitignore: 0.10 => 0.11
6168 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6170 * gst/rtsp-server/Makefile.am:
6171 Makefile.am: 0.10 => @GST_MAJORMINOR@
6173 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6175 Merge branch 'master' into 0.11
6177 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
6180 Automatic update of common submodule
6181 From 9e5bbd5 to 69b981f
6183 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
6186 Automatic update of common submodule
6187 From fd35073 to 9e5bbd5
6189 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
6192 Automatic update of common submodule
6193 From 46dfcea to fd35073
6195 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6197 * gst/rtsp-server/rtsp-media-factory-uri.c:
6198 * gst/rtsp-server/rtsp-media.c:
6199 media: port to new caps API
6201 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6203 Merge branch 'master' into 0.11
6205 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6207 * bindings/vala/gst-rtsp-server-0.10.vapi:
6208 Updated Vala bindings.
6209 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6211 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6213 * gst/rtsp-server/rtsp-server.c:
6214 * gst/rtsp-server/rtsp-server.h:
6215 Add a signal for newly connected clients.
6216 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6218 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6220 * bindings/python/rtspserver.override:
6221 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
6223 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6225 * gst/rtsp-server/Makefile.am:
6226 * gst/rtsp-server/rtsp-client.c:
6227 * gst/rtsp-server/rtsp-funnel.c:
6228 * gst/rtsp-server/rtsp-funnel.h:
6229 * gst/rtsp-server/rtsp-media.c:
6230 rtsp-server: port to 0.11
6232 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6237 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6239 Merge branch 'master' into 0.11
6244 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6247 Automatic update of common submodule
6248 From c3cafe1 to 46dfcea
6250 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
6252 * bindings/python/Makefile.am:
6253 * bindings/python/rtspserver.defs:
6254 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
6256 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
6258 * bindings/python/arg-types.py:
6259 python bindings: add GstRTSPUrlParam
6260 Needed to implement MediaFactory virtual proxies
6262 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
6264 * bindings/python/arg-types.py:
6265 python bindings: fix returning GstRTSPUrl types
6267 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6269 * bindings/python/arg-types.py:
6270 python bindings: add arg type for GstRTSPUrl
6272 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
6274 * bindings/python/rtspserver.defs:
6275 python bindings: fix the definition of MediaFactory.collect_stream
6277 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
6280 Automatic update of common submodule
6281 From 1ccbe09 to c3cafe1
6283 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6286 Automatic update of common submodule
6287 From 193b717 to 1ccbe09
6289 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
6292 Automatic update of common submodule
6293 From b77e2bf to 193b717
6295 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6298 build: Include lcov.mak to allow test coverage report generation
6300 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6303 Automatic update of common submodule
6304 From d8814b6 to b77e2bf
6306 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6309 Automatic update of common submodule
6310 From 6aaa286 to d8814b6
6312 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
6315 Automatic update of common submodule
6316 From 6aec6b9 to 6aaa286
6318 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
6321 autogen: wingo signed comment
6323 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
6325 * gst/rtsp-server/rtsp-session-pool.c:
6326 session: use full charset for RTSP session ID
6327 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
6328 session ID more difficult.
6329 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6331 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6333 * gst/rtsp-server/Makefile.am:
6334 rtsp-server: Don't install the funnel header
6336 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6339 Automatic update of common submodule
6340 From 1de7f6a to 6aec6b9
6342 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6345 configure: require core/base 0.10.31
6346 Needed at least for gst_plugin_feature_rank_compare_func().
6348 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
6351 Automatic update of common submodule
6352 From f94d739 to 1de7f6a
6354 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6356 * gst/rtsp-server/rtsp-media.c:
6357 media: remove more unused code
6359 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6361 * gst/rtsp-server/rtsp-media.c:
6362 * gst/rtsp-server/rtsp-media.h:
6363 media: remove duplicate filtering
6364 Remove the duplicate filtering code now that we have a released -good version.
6365 Give a warning instead.
6367 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6369 * gst/rtsp-server/rtsp-media-factory.c:
6370 * gst/rtsp-server/rtsp-media.c:
6371 media: fix default buffer size
6373 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6375 * gst/rtsp-server/rtsp-media-factory.c:
6376 * gst/rtsp-server/rtsp-media-factory.h:
6377 media-factory: add property to configure the buffer-size
6378 Add a property to configure the kernel UDP buffer size.
6380 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6382 * gst/rtsp-server/rtsp-media.c:
6383 * gst/rtsp-server/rtsp-media.h:
6384 media: add property to configure kernel buffer sizes
6385 Add a property to configure the kernel UDP buffer size.
6387 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6390 configure: set PYGOBJECT_REQ before using it
6391 https://bugzilla.gnome.org/show_bug.cgi?id=640641
6393 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6396 docs: recursive into sub-directories on 'make upload'
6398 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6400 * docs/libs/gst-rtsp-server-docs.sgml:
6401 * docs/version.entities.in:
6402 docs: mention full version these docs are for, not just major-minor
6404 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6409 === release 0.10.8 ===
6411 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6416 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6418 * gst/rtsp-server/rtsp-server.c:
6419 rtsp-server: clarify docs a little
6421 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6423 * gst/rtsp-server/rtsp-media.c:
6424 media: init debug category before starting thread
6426 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6428 * gst/rtsp-server/rtsp-auth.c:
6429 auth: add realm to make it more spec compliant
6431 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6433 * gst/rtsp-server/rtsp-server.c:
6434 * gst/rtsp-server/rtsp-server.h:
6437 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6439 * examples/test-video.c:
6440 example: improve example docs a little
6442 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6444 * gst/rtsp-server/rtsp-server.c:
6445 server: ensure the watch has a ref to the server
6447 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6449 * gst/rtsp-server/rtsp-server.c:
6450 server: simpify channel function
6452 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6454 * gst/rtsp-server/rtsp-server.c:
6455 * gst/rtsp-server/rtsp-server.h:
6456 server: simplify management of channel and source
6457 We don't need to keep around the channel and source objects. Let the mainloop
6458 and the source manage the source and channel respectively.
6460 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6466 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6469 * tests/Makefile.am:
6470 * tests/test-cleanup.c:
6471 tests: add tests directory and cleanup test
6473 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6475 * gst/rtsp-server/rtsp-media-factory-uri.c:
6476 * gst/rtsp-server/rtsp-media-factory.c:
6477 * gst/rtsp-server/rtsp-media-mapping.c:
6478 * gst/rtsp-server/rtsp-media.c:
6479 * gst/rtsp-server/rtsp-session-pool.c:
6480 * gst/rtsp-server/rtsp-session.c:
6481 server: improve debugging in various objects
6483 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6485 * gst/rtsp-server/rtsp-server.c:
6486 server: chain up to the parent finalize
6488 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
6490 * bindings/python/rtspserver-types.defs:
6491 * bindings/python/rtspserver.defs:
6492 * bindings/python/rtspserver.override:
6493 * bindings/python/test.py:
6494 gst-rtsp-server: update python bindings
6496 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6498 * gst/rtsp-server/rtsp-client.c:
6499 client: use the response from the clientstate
6500 Create the response object only once and store in the client state.
6501 Make all methods use the state response,
6503 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6505 * gst/rtsp-server/rtsp-server.c:
6506 server: use signal to keep track of clients
6507 Keep track of all the clients that the server creates and remove them when they
6508 fire the 'closed' signal.
6510 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6512 * gst/rtsp-server/rtsp-client.c:
6513 * gst/rtsp-server/rtsp-client.h:
6514 client: emit signal when closing
6516 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6518 * examples/.gitignore:
6519 * examples/Makefile.am:
6520 * examples/test-auth.c:
6521 * examples/test-video.c:
6522 * gst/rtsp-server/rtsp-auth.c:
6523 * gst/rtsp-server/rtsp-auth.h:
6524 * gst/rtsp-server/rtsp-client.c:
6525 * gst/rtsp-server/rtsp-media-factory.c:
6526 * gst/rtsp-server/rtsp-media.c:
6527 * gst/rtsp-server/rtsp-media.h:
6528 * gst/rtsp-server/rtsp-session-pool.h:
6529 * gst/rtsp-server/rtsp-session.h:
6530 media: enable per factory authorisations
6531 Allow for adding a GstRTSPAuth on the factory and media level and check
6532 permissions when accessing the factory.
6533 Add hints to the auth methods for future more fine grained authorisation.
6534 Add example application for per factory authentication.
6536 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6538 * gst/rtsp-server/rtsp-auth.c:
6539 * gst/rtsp-server/rtsp-auth.h:
6540 * gst/rtsp-server/rtsp-client.c:
6541 * gst/rtsp-server/rtsp-client.h:
6542 * gst/rtsp-server/rtsp-params.c:
6543 * gst/rtsp-server/rtsp-params.h:
6544 rtsp-server: Pass ClientState structure arround
6545 Pass the collected information for the ongoing request in a GstRTSPClientState
6546 structure that we can then pass around to simplify the method arguments. This
6547 will also be handy when we implement logging functionality.
6549 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6551 * gst/rtsp-server/rtsp-media-factory.c:
6552 * gst/rtsp-server/rtsp-media-factory.h:
6553 media-factory: add methods to configure authorisation
6555 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6557 * gst/rtsp-server/rtsp-client.c:
6558 client: unref auth in finalize
6560 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6562 * gst/rtsp-server/rtsp-server.c:
6563 server: unref auth in finalize
6565 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6567 * docs/libs/gst-rtsp-server-docs.sgml:
6568 * docs/libs/gst-rtsp-server-sections.txt:
6569 * docs/libs/gst-rtsp-server.types:
6572 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6574 * gst/rtsp-server/rtsp-server.c:
6575 * gst/rtsp-server/rtsp-server.h:
6576 server: separate create and accept
6577 Create separate create and accept methods so that subclasses can create custom
6579 Configure the server in the client object and prepare for keeping track of
6582 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6584 * gst/rtsp-server/rtsp-client.c:
6585 * gst/rtsp-server/rtsp-client.h:
6586 client: add support for setting the server.
6587 Add support for keeping a ref to the server that started this client
6590 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6592 * gst/rtsp-server/rtsp-auth.c:
6593 auth: fix memleak and add some docs
6594 Fix a memleak of the basic auth token.
6595 Add docs for the helper function
6597 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6599 * gst/rtsp-server/rtsp-auth.c:
6600 * gst/rtsp-server/rtsp-auth.h:
6601 * gst/rtsp-server/rtsp-client.c:
6602 client: delegate setup of auth to the manager
6603 Delegate the configuration of the authentication tokens to the manager object
6606 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6608 * examples/test-video.c:
6609 * gst/rtsp-server/Makefile.am:
6610 * gst/rtsp-server/rtsp-auth.c:
6611 * gst/rtsp-server/rtsp-auth.h:
6612 * gst/rtsp-server/rtsp-client.c:
6613 * gst/rtsp-server/rtsp-client.h:
6614 * gst/rtsp-server/rtsp-server.c:
6615 * gst/rtsp-server/rtsp-server.h:
6616 auth: add authentication object
6617 Add an object that can check the authorization of requests.
6618 Implement basic authentication.
6619 Add example authentication to test-video
6621 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6623 * gst/rtsp-server/rtsp-server.c:
6624 * gst/rtsp-server/rtsp-server.h:
6625 server: move includes back
6626 the includes are needed for sockaddr_in.
6628 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6630 * gst/rtsp-server/rtsp-client.c:
6631 * gst/rtsp-server/rtsp-client.h:
6632 * gst/rtsp-server/rtsp-server.c:
6633 * gst/rtsp-server/rtsp-server.h:
6634 rtsp: move network includes where they are needed
6636 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
6638 * gst/rtsp-server/rtsp-media.h:
6639 rtsp-media.h: Minor corrections in comments.
6642 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
6645 Automatic update of common submodule
6646 From e572c87 to f94d739
6648 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6652 * docs/libs/.gitignore:
6653 * examples/.gitignore:
6654 * gst/rtsp-server/.gitignore:
6657 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6659 * docs/libs/Makefile.am:
6660 docs: We don't build ps/pdf for API reference docs
6662 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6665 Automatic update of common submodule
6666 From ccbaa85 to e572c87
6668 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6671 Automatic update of common submodule
6672 From 46445ad to ccbaa85
6674 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6676 * gst/rtsp-server/Makefile.am:
6677 * gst/rtsp-server/fs-funnel.c:
6678 * gst/rtsp-server/fs-funnel.h:
6679 * gst/rtsp-server/rtsp-funnel.c:
6680 * gst/rtsp-server/rtsp-funnel.h:
6681 * gst/rtsp-server/rtsp-media.c:
6682 funnel: rename fsfunnel to rtspfunnel
6683 Rename the funnel to avoid conflicts with the farsight one.
6685 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6687 * gst/rtsp-server/Makefile.am:
6688 * gst/rtsp-server/fs-funnel.c:
6689 * gst/rtsp-server/fs-funnel.h:
6690 * gst/rtsp-server/rtsp-media.c:
6691 rtsp-media: add and use fsfunnel
6692 Add a copy of fsfunnel to the build because input-selector removed the (broken)
6693 select-all property that we need.
6695 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6697 * gst/rtsp-server/Makefile.am:
6698 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
6699 Use PKG_CONFIG_PATH specified at configure time (if any) as well
6700 for the g-ir-compiler, rather than just assuming the env var has
6703 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6710 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
6712 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6715 * gst/rtsp-server/Makefile.am:
6716 gobject-introspection: fix g-i build for uninstalled setup
6717 Requires gst-plugins-base git (> 0.10.31.2).
6719 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6721 * examples/test-uri.c:
6722 examples: add some more options and comments
6724 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6726 * gst/rtsp-server/rtsp-media-factory-uri.c:
6727 factory-uri: use right property type
6729 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6731 * gst/rtsp-server/rtsp-media-factory-uri.c:
6732 factory-uri: attempt to configure buffer-lists
6733 Attempt to configure buffer lists in the payloader for improved performance.
6735 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6737 * gst/rtsp-server/rtsp-media.c:
6738 media: attempt to configure bigger UDP buffers
6739 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
6740 send buffers with high bitrate streams.
6742 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
6744 * gst/rtsp-server/rtsp-client.c:
6745 client: use the socket length from getsockname
6746 Use the length returned by getsockname to perform the getnameinfo call because
6747 the size can depend on the socket type and platform.
6750 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6752 * docs/libs/gst-rtsp-server-docs.sgml:
6753 * docs/libs/gst-rtsp-server-sections.txt:
6754 docs: add uri factory to the docs
6756 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6758 * gst/rtsp-server/rtsp-client.c:
6759 * gst/rtsp-server/rtsp-media.h:
6762 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6764 * gst/rtsp-server/rtsp-client.c:
6765 * gst/rtsp-server/rtsp-media.c:
6766 * gst/rtsp-server/rtsp-media.h:
6767 * gst/rtsp-server/rtsp-session.c:
6768 * gst/rtsp-server/rtsp-session.h:
6769 rtsp-server: add support for buffer lists
6770 Add support for sending bufferlists received from appsink.
6773 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6775 * gst/rtsp-server/rtsp-client.c:
6776 * gst/rtsp-server/rtsp-media.c:
6777 * gst/rtsp-server/rtsp-media.h:
6778 * gst/rtsp-server/rtsp-sdp.c:
6779 media: make method to retrieve the play range
6780 Make a method to retrieve the playback range so that we can conditionally create
6781 a different range for the SDP and the PLAY requests.
6783 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6785 * gst/rtsp-server/rtsp-media.c:
6786 * gst/rtsp-server/rtsp-media.h:
6787 media: add signal to notify of state changes
6789 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6791 * gst/rtsp-server/rtsp-client.h:
6792 client: cleanup headers
6794 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6796 * gst/rtsp-server/rtsp-client.c:
6799 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6801 * gst/rtsp-server/rtsp-media-factory-uri.c:
6802 * gst/rtsp-server/rtsp-media-factory-uri.h:
6803 factory-uri: add support for gstpay
6804 Add an option to prefer gstpay over decoder + raw payloader.
6806 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6808 * gst/rtsp-server/rtsp-media-factory-uri.c:
6809 * gst/rtsp-server/rtsp-media-factory-uri.h:
6810 factory-uri: rework the autoplugger.
6811 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
6814 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6816 * gst/rtsp-server/rtsp-media-factory-uri.c:
6817 factory-uri: use better factory filter
6818 Make better payloader filter based on autoplug rank and RTP use case.
6820 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6823 Automatic update of common submodule
6824 From 169462a to 46445ad
6826 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6828 * gst/rtsp-server/rtsp-server.c:
6829 server: set SO_REUSEADDR before bind
6830 Set the SO_REUSEADDR _before_ bind() to make it actually work.
6832 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6834 * gst/rtsp-server/rtsp-media.c:
6835 * gst/rtsp-server/rtsp-media.h:
6836 media: emit prepared signal when prepared
6837 Make a 'prepared' signal and emit it when we successfully prepared the element.
6838 This signal can be used to configure the media object after it has been prepared
6841 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
6844 Automatic update of common submodule
6845 From 011bcc8 to 169462a
6847 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
6849 python an optional dependency
6850 * configure.ac: Move up valgrind and g-i checks. Make the python
6851 dependency optional, as it was before.
6853 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6855 Merge branch 'master' into 0.11
6860 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6862 * gst/rtsp-server/rtsp-media.c:
6863 media: update range when active clients changed
6864 When we changed the number of active clients, update the current range
6865 information because we want the second client connecting to a shared resource
6866 continue from where the stream currently.
6868 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6870 * gst/rtsp-server/rtsp-media-factory-uri.c:
6871 * gst/rtsp-server/rtsp-media-factory-uri.h:
6872 factory-uri: add colorspace and fix pt
6873 Rework the way we pass data to the autoplugger.
6874 When we have raw caps, plug a converter element to make pluggin to raw
6875 payloaders more successful.
6876 Make sure all dynamically plugged payloaders have a unique payload types.
6878 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6880 * examples/Makefile.am:
6881 * examples/test-uri.c:
6882 example: add example of the uri factory
6884 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6886 * gst/rtsp-server/Makefile.am:
6887 * gst/rtsp-server/rtsp-media-factory-uri.c:
6888 * gst/rtsp-server/rtsp-media-factory-uri.h:
6889 * gst/rtsp-server/rtsp-server.h:
6890 factory-uri: add a factory to stream any URI
6891 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
6894 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6896 * gst/rtsp-server/rtsp-media.c:
6897 * gst/rtsp-server/rtsp-media.h:
6898 media: ignore spurious ASYNC_DONE messages
6899 When we are dynamically adding pads, the addition of the udpsrc elements will
6900 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
6901 the real ASYNC_DONE when everything is prerolled.
6903 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6905 * gst/rtsp-server/rtsp-media-factory.c:
6906 * gst/rtsp-server/rtsp-media-factory.h:
6907 media-factory: make lock macro
6909 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
6911 * gst/rtsp-server/rtsp-client.c:
6912 rtsp-server: Remove unused variable and dead assignment
6914 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
6916 * examples/test-launch.c:
6917 * examples/test-mp4.c:
6918 * examples/test-ogg.c:
6919 * examples/test-readme.c:
6920 * examples/test-sdp.c:
6921 * examples/test-video.c:
6922 examples: Run gst-indent
6924 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
6926 * gst/rtsp-server/rtsp-client.c:
6927 * gst/rtsp-server/rtsp-media-factory.c:
6928 * gst/rtsp-server/rtsp-media-mapping.c:
6929 * gst/rtsp-server/rtsp-media.c:
6930 * gst/rtsp-server/rtsp-params.c:
6931 * gst/rtsp-server/rtsp-sdp.c:
6932 * gst/rtsp-server/rtsp-server.c:
6933 * gst/rtsp-server/rtsp-session-pool.c:
6934 * gst/rtsp-server/rtsp-session.c:
6935 rtsp-server: Run gst-indent
6936 Since it wasn't using the upstream common previously, there was no
6937 indentation check before commiting.
6939 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
6941 * gst/rtsp-server/rtsp-media-mapping.h:
6942 * gst/rtsp-server/rtsp-media.c:
6943 * gst/rtsp-server/rtsp-media.h:
6944 * gst/rtsp-server/rtsp-sdp.c:
6945 * gst/rtsp-server/rtsp-session-pool.h:
6946 * gst/rtsp-server/rtsp-session.c:
6947 * gst/rtsp-server/rtsp-session.h:
6948 rtsp-server: Some more doc fixups
6950 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6953 Makefile: Add cruft-cleaning support
6955 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6960 * docs/libs/Makefile.am:
6961 * docs/libs/gst-rtsp-server-docs.sgml:
6962 * docs/libs/gst-rtsp-server-sections.txt:
6963 * docs/libs/gst-rtsp-server.types:
6964 * docs/version.entities.in:
6965 docs: Add gtk-doc build system
6967 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6969 * gst/rtsp-server/Makefile.am:
6970 Makefile.am: Use standard GIR make behaviour
6972 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6976 autogen/configure: Bring more in sync to standard gst module behaviour
6978 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6980 * gst/rtsp-server/rtsp-media.c:
6981 media: warn and fail when gstrtpbin is not found
6983 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6986 configure: open 0.11 branch
6988 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
6992 Add common submodule
6994 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
6997 * common/Makefile.am:
6998 * common/c-to-xml.py:
7000 * common/coverage/coverage-report-entry.pl:
7001 * common/coverage/coverage-report.pl:
7002 * common/coverage/coverage-report.xsl:
7003 * common/coverage/lcov.mak:
7004 * common/gettext.patch:
7005 * common/glib-gen.mak:
7006 * common/gst-autogen.sh:
7007 * common/gst-xmlinspect.py:
7009 * common/gstdoc-scangobj:
7010 * common/gtk-doc-plugins.mak:
7011 * common/gtk-doc.mak:
7012 * common/m4/.gitignore:
7013 * common/m4/Makefile.am:
7015 * common/m4/as-ac-expand.m4:
7016 * common/m4/as-auto-alt.m4:
7017 * common/m4/as-compiler-flag.m4:
7018 * common/m4/as-compiler.m4:
7019 * common/m4/as-docbook.m4:
7020 * common/m4/as-libtool-tags.m4:
7021 * common/m4/as-libtool.m4:
7022 * common/m4/as-python.m4:
7023 * common/m4/as-scrub-include.m4:
7024 * common/m4/as-version.m4:
7025 * common/m4/ax_create_stdint_h.m4:
7026 * common/m4/check.m4:
7027 * common/m4/glib-gettext.m4:
7028 * common/m4/gst-arch.m4:
7029 * common/m4/gst-args.m4:
7030 * common/m4/gst-check.m4:
7031 * common/m4/gst-debuginfo.m4:
7032 * common/m4/gst-default.m4:
7033 * common/m4/gst-doc.m4:
7034 * common/m4/gst-error.m4:
7035 * common/m4/gst-feature.m4:
7036 * common/m4/gst-function.m4:
7037 * common/m4/gst-gettext.m4:
7038 * common/m4/gst-glib2.m4:
7039 * common/m4/gst-libxml2.m4:
7040 * common/m4/gst-plugindir.m4:
7041 * common/m4/gst-valgrind.m4:
7042 * common/m4/gtk-doc.m4:
7043 * common/m4/introspection.m4:
7045 * common/mangle-tmpl.py:
7046 * common/plugins.xsl:
7048 * common/release.mak:
7049 * common/scangobj-merge.py:
7050 * common/upload.mak:
7051 common: Remove static version
7053 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
7055 * common/m4/introspection.m4:
7056 Update introspection.m4 to match usage
7058 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7062 Remove old stuff from the README
7064 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7069 === release 0.10.7 ===
7071 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7076 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7078 * examples/test-ogg.c:
7079 test-ogg: remove parsers
7080 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
7081 buffers with timestamps. Using the parsers also seems to break things.
7083 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7085 * bindings/vala/gst-rtsp-server-0.10.vapi:
7086 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7087 Updated Vala bindings
7089 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7091 * common/m4/introspection.m4:
7093 * gst/rtsp-server/Makefile.am:
7094 Added initial gobject-introspection support
7096 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7098 * gst/rtsp-server/rtsp-media-factory.c:
7099 media-factory: don't use host for shared hash key
7100 When we generate the key to share made between connections, don't include the
7101 host used to connect so that we can share media even if between clients that
7102 connected with localhost and ones with the ip address.
7104 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7106 * bindings/vala/Makefile.am:
7107 build: fix distcheck
7109 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7111 * bindings/vala/gst-rtsp-server-0.10.vapi:
7112 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7113 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7114 Update Vala bindings
7116 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7118 * bindings/vala/Makefile.am:
7120 Fix configure checks and installation location for Vala bindings
7123 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7128 === release 0.10.6 ===
7130 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7133 configure: release 0.10.6
7135 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7137 * gst/rtsp-server/rtsp-media.c:
7138 media: help the compiler a little
7140 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7142 * gst/rtsp-server/rtsp-media.c:
7143 * gst/rtsp-server/rtsp-media.h:
7144 * gst/rtsp-server/rtsp-session.c:
7145 media: cleanup media transport before freeing
7146 Cleanup the media transport data before freeing. In particular, remove the qdata
7147 from the rtpsource object.
7149 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7151 * gst/rtsp-server/rtsp-media-factory.c:
7152 * gst/rtsp-server/rtsp-media-factory.h:
7153 * gst/rtsp-server/rtsp-media.c:
7154 * gst/rtsp-server/rtsp-media.h:
7155 media-factory: add eos-shutdown property
7156 Add an eos-shutdown property that will send an EOS to the pipeline before
7157 shutting it down. This allows for nice cleanup in case of a muxer.
7160 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7162 * gst/rtsp-server/rtsp-media.c:
7163 * gst/rtsp-server/rtsp-media.h:
7164 media: use multiudpsink send-duplicates when we can
7165 If we have a new enough multiudpsink with the send-duplicates property, use this
7166 instead of doing our own filtering. Our custom filtering code should eventually
7167 be removed when we can depend on a released -good.
7169 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7171 * gst/rtsp-server/rtsp-media.c:
7172 media: don't leak destinations
7173 Refactor and cleanup the destinations array when the stream is destroyed.
7175 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7177 * gst/rtsp-server/rtsp-media.c:
7178 * gst/rtsp-server/rtsp-media.h:
7179 media: don't add udp addresses multiple times
7180 Keep track of the udp addresses we added to udpsink and never add the same udp
7181 destination twice. This avoids duplicate packets when using multicast.
7183 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7185 * gst/rtsp-server/rtsp-server.c:
7186 server: disable use of SO_LINGER
7187 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
7188 server close()s the connection.
7190 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7192 * gst/rtsp-server/rtsp-server.c:
7193 server: use 5 second linger period in SO_LINGER
7194 Wait 5 seconds before clearing the send buffers and reseting the connection with
7195 the client when we do a close. This should be enough time to get the message to
7199 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7201 * gst/rtsp-server/rtsp-server.c:
7202 server: use SO_LINGER
7203 SO_LINGER on the socket will make sure that any pending data on the socket is
7204 flushed ASAP and that the socket connection is reset. This makes sure that the
7205 socket can be reused immediately.
7208 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7211 README: add blurb about shared media factories
7213 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
7215 * gst/rtsp-server/rtsp-media.c:
7216 Add stdlib.h for atoi()
7218 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7220 * bindings/python/Makefile.am:
7221 * bindings/vala/Makefile.am:
7222 build: distcheck fixes
7223 Fix 'make distcheck', somewhat (it still fails because it tries to
7224 install files into /usr/share/vala/vapi/ irrespective of the
7227 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7230 configure: bump core/base requirements to released version
7231 Makes things less confusing for people.
7233 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7236 configure: fail if GStreamer core/base requirements are not met
7238 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7240 * gst/rtsp-server/rtsp-client.c:
7241 client: improve client cleanups
7242 Make sure the session does not timeout when using TCP. We need to do this
7243 because quicktime player does not send RTCP for some reason in tunneled
7245 Refactor some cleanup code.
7248 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7250 * gst/rtsp-server/rtsp-session.c:
7251 * gst/rtsp-server/rtsp-session.h:
7252 session: add support for prevent session timeouts
7253 Add an atomix counter to prevent session timeouts when we are, for example,
7256 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7258 * gst/rtsp-server/rtsp-client.c:
7259 client: fix unlink on session timeouts
7260 When our session times out, make sure we unlink all streams in this
7262 Remove the tunnelid when closing the connection.
7264 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7266 * gst/rtsp-server/rtsp-session.c:
7267 session: small cleanups
7269 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7271 * gst/rtsp-server/rtsp-client.c:
7272 client: handle lost_tunnel callbacks
7273 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
7274 hashtable so that we can reuse it for when the client reopens the POST
7276 Close the connection after a TEARDOWN.
7277 Make sure or watchid is cleared when the watch is removed.
7280 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7282 * gst/rtsp-server/rtsp-client.c:
7283 * gst/rtsp-server/rtsp-media.c:
7284 * gst/rtsp-server/rtsp-sdp.c:
7285 rtsp-server: add more support for multicast
7287 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7290 * gst/rtsp-server/rtsp-media.c:
7291 * gst/rtsp-server/rtsp-media.h:
7292 media: allow configuration of allowed lower transport
7294 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7296 * gst/rtsp-server/rtsp-client.h:
7297 * gst/rtsp-server/rtsp-media.c:
7298 * gst/rtsp-server/rtsp-media.h:
7299 * gst/rtsp-server/rtsp-sdp.c:
7300 * gst/rtsp-server/rtsp-sdp.h:
7301 * gst/rtsp-server/rtsp-server.c:
7302 rtsp: keep track of server ip and ipv6
7303 Keep track of how the client connected to the server and setup the udp ports
7304 with the same protocol.
7305 Copy the server ip address in the SDP so that clients can send RTCP back to
7308 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7310 * gst/rtsp-server/rtsp-session.c:
7313 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7315 * gst/rtsp-server/rtsp-client.c:
7316 client: use right size for malloc
7318 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7320 * gst/rtsp-server/rtsp-server.c:
7321 server: comment ipv6 server listening address
7323 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7325 * gst/rtsp-server/rtsp-media.c:
7326 media: allow for ipv6 sockets
7328 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7330 * gst/rtsp-server/rtsp-server.c:
7331 * gst/rtsp-server/rtsp-server.h:
7332 server: rework server part
7333 Allow setting a bind address, make sure we can deal with ipv6.
7334 Remove the port property and change with the service property.
7336 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7338 * gst/rtsp-server/rtsp-media.h:
7339 media: update comments a little
7341 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7343 * gst/rtsp-server/rtsp-client.c:
7344 client: make content-base better
7345 Use the URI formatting functions to make a content-base. Also make sure that
7346 there is a trailing / at the end.
7348 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7350 * gst/rtsp-server/rtsp-client.c:
7351 client: guard against invalid paths
7353 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7355 * examples/test-video.c:
7356 test: catch server bind errors
7358 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
7360 * gst/rtsp-server/rtsp-media.c:
7361 rtspmedia: emit "unprepared" if _prepare fails.
7362 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
7363 media object is removed from its factory's cache.
7365 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7367 * gst/rtsp-server/rtsp-media.c:
7368 media: collect media position when seek completes
7370 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
7372 * gst/rtsp-server/rtsp-client.c:
7373 client: call unlink_streams in client finalize
7376 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7378 * gst/rtsp-server/rtsp-media.c:
7379 media: limit the time to wait to something huge
7380 Avoid waiting forever but limit the timeout to 20 seconds.
7382 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7384 * gst/rtsp-server/rtsp-sdp.c:
7385 sdp: reindent and check for prepared status
7387 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7389 * gst/rtsp-server/rtsp-media.c:
7390 * gst/rtsp-server/rtsp-media.h:
7391 * gst/rtsp-server/rtsp-session.c:
7392 media: avoid doing _get_state() for state changes
7393 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
7394 until the media is prerolled or in error. This avoids doing a blocking call of
7395 gst_element_get_state() that can cause lockups when there is an error.
7398 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7400 * gst/rtsp-server/rtsp-media.c:
7403 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7405 * gst/rtsp-server/rtsp-media-factory.c:
7406 media-factory: better error handling
7407 Improve the error handling a bit.
7409 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7411 * gst/rtsp-server/rtsp-client.c:
7412 client: rework transport parsing
7413 Rework the transport parsing code so that we can ignore transports we don't
7414 support instead of just picking the first one we can parse.
7415 Configure a (for now hardcoded) destination for multicast transports.
7417 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7419 * gst/rtsp-server/rtsp-media.c:
7420 media: set multicast sink parameters
7421 Disable loop and automatic multicast join on the udpsink elements.
7422 Add some more debug info.
7423 Reset some state variables in the right place.
7424 Use the right port numbers for multicast.
7426 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7428 * gst/rtsp-server/rtsp-session.c:
7429 session: handle transport setup correctly
7430 Handle UDP, MCAST and TCP transport negotiation more correctly.
7431 Store the server session SSRC in the transport.
7433 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7435 * gst/rtsp-server/rtsp-client.c:
7436 rtsp-client: implement error_full
7437 Implement error_full to avoid some segfaults when the rtspconnection calls it.
7440 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7443 * gst/rtsp-server/rtsp-client.c:
7444 * gst/rtsp-server/rtsp-server.c:
7445 docs: update docs and comments
7447 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
7449 * gst/rtsp-server/rtsp-sdp.c:
7450 sdp: make server work better when behind a proxy
7452 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7454 * gst/rtsp-server/rtsp-client.c:
7455 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
7457 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7459 * gst/rtsp-server/rtsp-client.c:
7460 * gst/rtsp-server/rtsp-media-factory.c:
7461 * gst/rtsp-server/rtsp-media-mapping.c:
7462 * gst/rtsp-server/rtsp-media.c:
7463 * gst/rtsp-server/rtsp-server.c:
7464 * gst/rtsp-server/rtsp-session-pool.c:
7465 * gst/rtsp-server/rtsp-session.c:
7466 Use GStreamer's debugging subsystem
7468 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7470 * gst/rtsp-server/rtsp-media-factory.c:
7471 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
7473 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7478 === release 0.10.5 ===
7480 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7485 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7488 configure: bump required versions
7490 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
7492 * gst/rtsp-server/rtsp-client.c:
7493 client: call weak-unref on client->sessions from finalize
7496 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7498 * gst/rtsp-server/rtsp-media.c:
7499 media: Fixed crasher where caps got unref'ed too often
7501 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7504 * pkgconfig/.gitignore:
7505 * pkgconfig/Makefile.am:
7506 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
7507 Added pkg-config file to use gst-rtsp-server uninstalled
7509 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7511 * gst/rtsp-server/rtsp-media.c:
7512 media: add some docs
7514 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
7516 * gst/rtsp-server/rtsp-client.c:
7517 rtsp: Use gst_rtsp_watch_send_message().
7518 Use gst_rtsp_watch_send_message() since the old API which used
7519 gst_rtsp_watch_queue_message() has been deprecated.
7521 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7526 === release 0.10.4 ===
7528 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7533 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7535 * gst/rtsp-server/rtsp-client.c:
7536 * gst/rtsp-server/rtsp-session.c:
7537 * gst/rtsp-server/rtsp-session.h:
7538 rtsp: allocate channels in TCP mode
7539 When the client does not provide us with channels in TCP mode, allocate channels
7542 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7544 * gst/rtsp-server/rtsp-client.c:
7545 client: don't crash when tunnelid is missing
7546 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
7547 don't crash but return an error response to the client.
7550 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7552 * bindings/vala/gst-rtsp-server-0.10.vapi:
7553 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7554 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7555 bindings: update vala bindings with new method
7557 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7559 * gst/rtsp-server/rtsp-session-pool.c:
7560 * gst/rtsp-server/rtsp-session-pool.h:
7561 sessionpool: add function to filter sessions
7562 Add generic function to retrieve/remove sessions.
7564 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7567 configure: bump core/base requirements to release
7569 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7571 * gst/rtsp-server/rtsp-media.c:
7572 media: fix indentation
7574 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7576 * gst/rtsp-server/rtsp-media.c:
7577 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
7579 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7581 * gst/rtsp-server/rtsp-media.c:
7582 set state and remove elements of media in for loop
7584 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
7586 * bindings/vala/gst-rtsp-server-0.10.vapi:
7587 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7588 Added gst_rtsp_media_remove_elements function to Vala bindings
7590 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
7592 * gst/rtsp-server/rtsp-media.c:
7593 * gst/rtsp-server/rtsp-media.h:
7594 Added gst_rtsp_media_remove_elements function
7596 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
7598 * gst/rtsp-server/rtsp-media.c:
7599 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
7601 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7603 * bindings/vala/gst-rtsp-server-0.10.vapi:
7604 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7605 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7606 Updated Vala bindings
7608 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7610 * gst/rtsp-server/rtsp-media.c:
7611 * gst/rtsp-server/rtsp-media.h:
7612 Added vmethod unprepare to GstRTSPMedia
7613 The default implementation sets the state of the pipeline to GST_STATE_NULL
7615 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7617 * gst/rtsp-server/rtsp-media-factory.c:
7618 * gst/rtsp-server/rtsp-media-factory.h:
7619 Made collect_streams function public
7621 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7623 * gst/rtsp-server/rtsp-media-factory.c:
7624 * gst/rtsp-server/rtsp-media-factory.h:
7625 * gst/rtsp-server/rtsp-media.c:
7626 Added vmethod create_pipeline to GstRTSPMediaFactory
7627 The pipeline is created in this method and the GstRTSPMedia's element is added to it
7629 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7631 * gst/rtsp-server/rtsp-client.c:
7632 client: use g_source_destroy()
7633 We need to use g_source_destroy() because we might have added the source to a
7634 different main context than the default one.
7636 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7638 * gst/rtsp-server/Makefile.am:
7639 * gst/rtsp-server/rtsp-client.c:
7640 * gst/rtsp-server/rtsp-params.c:
7641 * gst/rtsp-server/rtsp-params.h:
7642 rtsp: prepare for handling GET/SET_PARAMETER
7643 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
7645 Fix return codes of handlers.
7647 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7649 * gst/rtsp-server/rtsp-media.c:
7650 media: don't leak session pads
7652 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7654 * gst/rtsp-server/rtsp-media.c:
7655 media: clean up the messages a bit
7657 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7659 * gst/rtsp-server/rtsp-sdp.c:
7660 sdp: warn and skip streams without media
7662 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7664 * bindings/vala/gst-rtsp-server-0.10.vapi:
7665 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7666 vala: Fixed typo in header file of RTSPMediaStream
7668 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7670 * gst/rtsp-server/rtsp-media.c:
7673 Make dumping RTCP stats configurable
7675 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7677 * gst/rtsp-server/rtsp-media.c:
7678 media: be less verbose and leak less
7680 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7682 * gst/rtsp-server/rtsp-media.c:
7683 media: don't leak the destination address
7685 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7687 * gst/rtsp-server/rtsp-client.c:
7688 * gst/rtsp-server/rtsp-media.c:
7689 * gst/rtsp-server/rtsp-media.h:
7690 * gst/rtsp-server/rtsp-session.c:
7691 * gst/rtsp-server/rtsp-session.h:
7692 rtsp: use RTCP to keep the session alive
7693 Use the RTCP rtcp-from stats field to find the associated session and use this
7694 to keep the session alive.
7696 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7698 * gst/rtsp-server/rtsp-session.c:
7699 session: add 5sec to the real session timeout
7700 Allow the session to live 5sec longer before really timing out. This should give
7701 clients some extra time to keep the session active.
7703 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7705 * gst/rtsp-server/rtsp-client.c:
7706 client: replay OK to GET/SET_PARAMETER
7707 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
7708 so that we return OK for those requests.
7710 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7712 * gst/rtsp-server/rtsp-media.c:
7713 * gst/rtsp-server/rtsp-media.h:
7714 media: keep track of active transports
7715 Keep track of which transport is active to avoid closing the connection too
7717 Remove the destination transport also when going to NULL.
7718 Print some stats about the SDES and other RTCP messages we receive from the
7721 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7723 * examples/.gitignore:
7724 * examples/Makefile.am:
7725 * examples/test-sdp.c:
7726 example: add SDP relay example
7728 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7730 * gst/rtsp-server/rtsp-media.c:
7731 media: also count active TCP connections
7733 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7735 * gst/rtsp-server/rtsp-media-factory.c:
7736 * gst/rtsp-server/rtsp-media.c:
7737 * gst/rtsp-server/rtsp-media.h:
7738 rtsp: add support for dynamic elements
7739 Add support for dynamic elements.
7740 Don't set live pipelines back to paused.
7742 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7744 * gst/rtsp-server/rtsp-sdp.c:
7745 sdp: don't add encoding name when absent in caps
7747 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7749 * gst/rtsp-server/rtsp-client.c:
7750 client: warn when we can't do RTP-Info
7752 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7754 * gst/rtsp-server/rtsp-media-factory.c:
7755 factory: factor out the stream construction
7757 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7759 * gst/rtsp-server/rtsp-client.c:
7760 client: only add RTP-Info when we have the info
7761 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
7764 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7769 === release 0.10.3 ===
7771 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7775 - Fixes a bug where it put the wrong verion in pkgconfig
7776 - Link RTP and RTCP sources
7778 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7780 * gst/rtsp-server/rtsp-media.c:
7781 * gst/rtsp-server/rtsp-media.h:
7782 media: link the RTP udpsrc to the session manager
7783 Link the RTP udpsrc and the appsrc to the session manager so that they don't
7784 shut down when the client sends a packet to open firewalls.
7786 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7788 * pkgconfig/gst-rtsp-server.pc.in:
7789 Don't use hard-coded version number in pkg-config file
7791 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7796 === release 0.10.2 ===
7798 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7803 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7806 * common/m4/.gitignore:
7807 * examples/.gitignore:
7808 * pkgconfig/.gitignore:
7809 add some .gitignore files
7811 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7813 * gst/rtsp-server/rtsp-media.c:
7814 media: seek to key frames
7816 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7818 * gst/rtsp-server/rtsp-media.c:
7819 media: emit the unprepared signal by id
7820 Emit the unprepared signal by id instead of name and set the media as
7823 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7825 * gst/rtsp-server/rtsp-media.c:
7826 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
7828 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7830 * gst/rtsp-server/rtsp-server.c:
7831 Added finalize function to GstRTPSPServer to unref session pool and media mapping
7833 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7835 * bindings/vala/gst-rtsp-server-0.10.vapi:
7836 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7837 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7838 Updated vala bindings
7840 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7842 * gst/rtsp-server/Makefile.am:
7843 * gst/rtsp-server/rtsp-client.c:
7844 * gst/rtsp-server/rtsp-media.c:
7845 server: use appsink and appsrc with the API
7846 Use the appsink/appsrc API instead of the signals for higher
7849 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7851 * examples/test-ogg.c:
7852 tests: set the payload type correctly
7854 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7856 * gst/rtsp-server/rtsp-media-factory.c:
7857 factory: connect to the unprepare signal
7858 Connect to the unprepare signal for non-reusable media so that we can remove
7859 them from the cache.
7861 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7863 * gst/rtsp-server/rtsp-media.c:
7864 * gst/rtsp-server/rtsp-media.h:
7865 media: add signal to notify of unprepare
7867 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7869 * gst/rtsp-server/rtsp-media.c:
7870 * gst/rtsp-server/rtsp-media.h:
7871 media: more work on making the media shared
7872 Add a reusable flag to medias, indicating that they can be reused after a state
7876 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7878 * examples/test-readme.c:
7879 examples: mark the example as shared for testing
7881 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7883 * gst/rtsp-server/rtsp-media.c:
7884 * gst/rtsp-server/rtsp-media.h:
7885 client: support shared media
7886 Always perform the state actions even if the target state of the pipeline is
7887 already correct, we still want to add/remove the transports when we are dealing
7889 Keep a counter of the number of active transports for a media so that we can use
7890 this to perform a state change when needed.
7891 Perform a state change of the pipeline only when the first transport was added
7892 or when there are no active transports.
7894 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7896 * gst/rtsp-server/rtsp-client.c:
7897 client: fix refcounting crasher
7898 Don't need to remove the weak refs in the finalize methods, they are already
7899 removed in the dispose.
7900 Don't register the callback with a DestroyNofity.
7902 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7904 * gst/rtsp-server/rtsp-client.c:
7905 Fix rtsp client refcount management in TCP mode.
7906 Don't unref a client ref we never had. Fixes an unref
7907 of an already-free client object after a client
7908 teardown request for me.
7910 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7912 * gst/rtsp-server/rtsp-session.c:
7913 docs: fix typo in API docs
7915 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7917 * gst/rtsp-server/rtsp-media.c:
7919 Keep the udp sources in playing even if we go to paused. unlock the sources when
7921 Add some more debug info.
7922 Only seek when we need to.
7923 Keep track of the position when we go to paused.
7925 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7927 * gst/rtsp-server/rtsp-client.c:
7928 * gst/rtsp-server/rtsp-media.c:
7929 * gst/rtsp-server/rtsp-media.h:
7930 Add beginnings of seeking.
7931 Parse the Range header and perform a seek on the pipeline for the requested
7932 position. It's disabled currently until I figure out what's going wrong.
7934 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7936 * gst/rtsp-server/rtsp-client.c:
7937 allow pause requests for now.
7940 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7942 * gst/rtsp-server/rtsp-client.c:
7943 Remove weak ref on the session in teardown
7944 We need to remove our weakref from the session when we do a teardown because
7945 else we close the TCP connection prematurely.
7947 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7949 * gst/rtsp-server/rtsp-client.c:
7950 * gst/rtsp-server/rtsp-client.h:
7951 * gst/rtsp-server/rtsp-session-pool.c:
7952 Do some more session cleanup
7953 Make session timeout kill the TCP connection that currently watches the
7955 Remove the client timeout property.
7957 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7959 * gst/rtsp-server/rtsp-client.c:
7960 * gst/rtsp-server/rtsp-client.h:
7961 * gst/rtsp-server/rtsp-media.c:
7962 * gst/rtsp-server/rtsp-media.h:
7963 * gst/rtsp-server/rtsp-server.c:
7964 * gst/rtsp-server/rtsp-session.c:
7965 * gst/rtsp-server/rtsp-session.h:
7967 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
7970 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7972 * examples/Makefile.am:
7973 * examples/test-launch.c:
7974 Add example server that takes launch lines
7975 Add an example server that streams any -launch line.
7977 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7979 * examples/test-readme.c:
7980 * gst/rtsp-server/rtsp-client.c:
7981 * gst/rtsp-server/rtsp-media.c:
7982 * gst/rtsp-server/rtsp-media.h:
7983 Add support for live streams
7984 Add support for live streams and ranges
7985 Start on handling TCP data transfer.
7987 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7989 * gst/rtsp-server/rtsp-media.c:
7990 Free the pipeline before other things
7993 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7995 * gst/rtsp-server/rtsp-client.c:
7996 Only free the pending tunnel if there is one
7999 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8001 * gst/rtsp-server/rtsp-client.c:
8002 * gst/rtsp-server/rtsp-client.h:
8003 * gst/rtsp-server/rtsp-media.c:
8004 rtsp-server: Add support for tunneling
8005 Add support for tunneling over HTTP.
8006 Use new connection methods to retrieve the url.
8007 Dispatch messages based on the message type instead of blindly
8008 assuming it's always a request.
8009 Keep track of the watch id so that we can remove it later.
8010 Set the media pipeline to NULL before unreffing the pipeline.
8012 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8014 * gst/rtsp-server/rtsp-client.c:
8015 * gst/rtsp-server/rtsp-client.h:
8016 Fix for channel -> watch rename in gstreamer
8017 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
8019 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8021 * gst/rtsp-server/rtsp-client.c:
8022 * gst/rtsp-server/rtsp-client.h:
8024 Use the async RTSP channels instead of spawning a new thread for each client.
8025 If a sessionid is specified in a request, fail if we don't have the session.
8027 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8029 * gst/rtsp-server/rtsp-media.c:
8030 Add better debug info
8031 Add some better debug info.
8033 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8035 * examples/test-video.c:
8037 Add support for session timeouts in the example.
8039 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8041 * gst/rtsp-server/rtsp-session-pool.c:
8042 * gst/rtsp-server/rtsp-session-pool.h:
8043 Pass GTimeVal around for performance reasons
8044 Get the current time only once and pass it around so that sessions don't have to
8045 get the current time anymore.
8046 Add experimental support for a GSource that dispatches when the session needs to
8049 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8051 * gst/rtsp-server/rtsp-session.c:
8052 * gst/rtsp-server/rtsp-session.h:
8053 Add better support for session timeouts
8054 Add a method to request the number of milliseconds when a session will timeout.
8056 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8058 * gst/rtsp-server/rtsp-media.c:
8059 * gst/rtsp-server/rtsp-media.h:
8060 Add suport for RTP manager monitoring
8061 Add the first stage in monitoring the rtp manager.
8062 Make sure we don't update the state to something we don't want.
8064 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8066 * gst/rtsp-server/rtsp-client.c:
8067 Add support for session keepalive
8068 Get and update the session timeout for all requests. get the session as early as
8071 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8073 * gst/rtsp-server/rtsp-media-factory.h:
8074 * gst/rtsp-server/rtsp-media.c:
8075 * gst/rtsp-server/rtsp-media.h:
8076 Handle media bus messages
8077 Handle media bus messages in a custom mainloop and dispatch them to the
8078 RTSPMedia objects. Let the default implementation handle some common messages.
8080 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8082 * gst/rtsp-server/rtsp-client.c:
8083 * gst/rtsp-server/rtsp-session-pool.c:
8084 * gst/rtsp-server/rtsp-session.c:
8085 Some more session timeout handling
8086 Move the session header setting code to a central place so that we always add
8087 the timeout parameter too.
8088 Handle timeouts by running the session cleanup code.
8089 Stop media before cleaning up.
8091 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8093 * gst/rtsp-server/rtsp-client.c:
8094 * gst/rtsp-server/rtsp-client.h:
8095 Add timeout property
8096 Add a timeout property ot the client and make the other properties into GObject
8099 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8101 * gst/rtsp-server/rtsp-session-pool.c:
8102 Use getters and setters in property code
8103 Use the getters and setters for the timeout property instead of locking
8106 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8108 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
8110 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8112 * gst/rtsp-server/rtsp-session-pool.c:
8113 * gst/rtsp-server/rtsp-session-pool.h:
8114 * gst/rtsp-server/rtsp-session.c:
8115 * gst/rtsp-server/rtsp-session.h:
8116 Add more timeout stuff
8117 Add method to check if a session is expired.
8118 Add method to perform cleanup on a session pool.
8120 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8122 * gst/rtsp-server/rtsp-client.c:
8123 * gst/rtsp-server/rtsp-session-pool.c:
8124 * gst/rtsp-server/rtsp-session-pool.h:
8125 * gst/rtsp-server/rtsp-session.c:
8126 * gst/rtsp-server/rtsp-session.h:
8127 Add beginnings of session timeouts and limits
8128 Add the timeout value to the Session header for unusual timeout values.
8129 Allow us to configure a limit to the amount of active sessions in a pool. Set a
8130 limit on the amount of retry we do after a sessionid collision.
8131 Add properties to the sessionid and the timeout of a session. Keep track of
8132 creation time and last access time for sessions.
8134 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8136 * gst/rtsp-server/rtsp-client.c:
8137 * gst/rtsp-server/rtsp-media.c:
8138 * gst/rtsp-server/rtsp-media.h:
8139 * gst/rtsp-server/rtsp-sdp.c:
8140 * gst/rtsp-server/rtsp-session-pool.c:
8141 * gst/rtsp-server/rtsp-session.c:
8142 * gst/rtsp-server/rtsp-session.h:
8143 Cleanup of sessions and more
8144 Fix the refcounting of media and sessions in the client. Properly clean up the
8145 session data when the client performs a teardown.
8146 Add Server header to responses.
8147 Allow for multiple uri setups in one session.
8148 Add Range header to the PLAY response and add the range attribute to the SDP
8150 Fix the session pool remove method, it used the wrong key in the hashtable. Also
8151 give the ownership of the sessionid to the session object.
8153 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8155 * gst/rtsp-server/rtsp-server.c:
8156 * gst/rtsp-server/rtsp-server.h:
8158 Rename the 'server_port' variable to simply 'port'.
8160 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8163 * gst/rtsp-server/rtsp-client.c:
8164 * gst/rtsp-server/rtsp-media.c:
8165 * gst/rtsp-server/rtsp-media.h:
8166 * gst/rtsp-server/rtsp-session.c:
8167 * gst/rtsp-server/rtsp-session.h:
8168 Rework the way we handle transports for streams
8169 Make the media accept an array of transports for the streams that we have
8170 configured for the play/pause requests.
8171 Implement server states for a client and its media.
8172 Require 0.10.22.1 (git HEAD) of gstreamer.
8174 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8176 * gst/rtsp-server/rtsp-client.c:
8177 * gst/rtsp-server/rtsp-media-factory.c:
8178 Drop const from functions dealing with urls
8179 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
8180 have the right const in them.
8182 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8184 * gst/rtsp-server/rtsp-client.c:
8185 * gst/rtsp-server/rtsp-media.c:
8186 * gst/rtsp-server/rtsp-sdp.c:
8190 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8192 * gst/rtsp-server/rtsp-client.c:
8193 * gst/rtsp-server/rtsp-media-factory.c:
8194 * gst/rtsp-server/rtsp-media.c:
8195 * gst/rtsp-server/rtsp-media.h:
8197 Don't keep a reference to the GstRTSPMedia in the stream.
8198 Free more things when freeing the GstRTSPMedia.
8200 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8203 * gst/rtsp-server/rtsp-media-factory.c:
8204 * gst/rtsp-server/rtsp-media-factory.h:
8205 * gst/rtsp-server/rtsp-media.c:
8206 * gst/rtsp-server/rtsp-media.h:
8207 * gst/rtsp-server/rtsp-server.c:
8208 * gst/rtsp-server/rtsp-server.h:
8209 More docs and small cleanups
8210 Add some more docs and update the README
8211 Cleanup some method names.
8212 Remove an unneeded idx field in the GstRTSPMediaStream
8214 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8217 * examples/Makefile.am:
8218 * examples/test-readme.c:
8219 Add a README and more example code
8220 Add a README file that contains a small introduction on how to use the server
8221 along with the example code explained in the readme.
8223 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8225 * gst/rtsp-server/rtsp-media.c:
8226 * gst/rtsp-server/rtsp-server.c:
8227 Fix some leaks and change default port
8228 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
8229 we finished the initial preroll. If we keep them locked, setting the pipeline to
8230 NULL will not stop and clean up the sources correctly.
8231 Change the default RTSP port to 8554 aka the official alternative RTSP port.
8233 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8235 * gst/rtsp-server/rtsp-session.c:
8236 * gst/rtsp-server/rtsp-session.h:
8237 Cleanups to the session object
8238 Remove some unneeded variables in the session state of a stream such as the
8239 owner media and the server transport.
8240 Get the configuration of a media stream in a session based on the media_stream
8241 in the original object instead of our cached index.
8242 Free more data in the finalize method.
8244 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8246 * gst/rtsp-server/rtsp-client.c:
8247 * gst/rtsp-server/rtsp-client.h:
8248 Cleanups and reuse media from DESCRIBE
8249 Handle thread create errors.
8250 Rename some internal methods to better match what they actually do.
8251 Handle misconfiguration of session_pool and media_mapping gracefully.
8252 Cache the DESCRIBE media and uri in the client connection and reuse them when
8253 we receive a SETUP request in the same connection for the same uri.
8254 Cleanup the client connection object.
8256 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8258 * gst/rtsp-server/rtsp-media-factory.c:
8259 * gst/rtsp-server/rtsp-media-factory.h:
8260 * gst/rtsp-server/rtsp-media.c:
8261 * gst/rtsp-server/rtsp-media.h:
8262 Add shared properties to media and factory
8263 Add the shared property to media.
8264 Implement some simple caching in the factory depending on if the media is shared
8267 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8269 * gst/rtsp-server/rtsp-client.c:
8270 Add a little comment
8271 Add some comment about the content-base header.
8273 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8275 * examples/Makefile.am:
8277 * examples/test-mp4.c:
8278 * examples/test-ogg.c:
8279 * examples/test-video.c:
8280 * gst/rtsp-server/Makefile.am:
8281 * gst/rtsp-server/rtsp-client.c:
8282 * gst/rtsp-server/rtsp-client.h:
8283 * gst/rtsp-server/rtsp-media-factory.c:
8284 * gst/rtsp-server/rtsp-media-factory.h:
8285 * gst/rtsp-server/rtsp-media.c:
8286 * gst/rtsp-server/rtsp-media.h:
8287 * gst/rtsp-server/rtsp-sdp.c:
8288 * gst/rtsp-server/rtsp-sdp.h:
8289 * gst/rtsp-server/rtsp-server.c:
8290 * gst/rtsp-server/rtsp-server.h:
8291 * gst/rtsp-server/rtsp-session.c:
8292 * gst/rtsp-server/rtsp-session.h:
8293 Reorganize things, prepare for media sharing
8294 Added various other test server examples
8295 Move the SDP message generation to a separate helper.
8296 Refactor common code for finding the session.
8297 Add content-base for realplayer compatibility
8298 Clean up request uris before processing for better vlc compatibility.
8299 Move prerolling and pipeline construction to the RTSPMedia object.
8300 Use multiudpsink for future pipeline reuse.
8302 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8308 === release 0.10.1 ===
8310 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8316 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8318 * bindings/vala/Makefile.am:
8320 Add more directories and files to the dist.
8322 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8324 * bindings/python/Makefile.am:
8325 * bindings/python/rtspserver.override:
8326 Fixed compile error of python bindings
8328 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8330 * bindings/vala/gst-rtsp-server-0.10.vapi:
8331 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8332 Marked values as nullable accordingly
8334 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8336 * bindings/vala/gst-rtsp-server-0.10.vapi:
8337 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8338 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8339 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8340 Updated Vala bindings
8342 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8344 * gst/rtsp-server/rtsp-client.c:
8345 * gst/rtsp-server/rtsp-media-mapping.c:
8346 * gst/rtsp-server/rtsp-media-mapping.h:
8347 * gst/rtsp-server/rtsp-media.h:
8348 * gst/rtsp-server/rtsp-session-pool.h:
8349 Cleanups and doc updates
8350 Add some more documentation and do some minor cleanups here and there.
8352 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8354 * gst/rtsp-server/rtsp-client.c:
8355 * gst/rtsp-server/rtsp-media-factory.c:
8356 * gst/rtsp-server/rtsp-media-factory.h:
8357 * gst/rtsp-server/rtsp-media.c:
8358 * gst/rtsp-server/rtsp-media.h:
8359 * gst/rtsp-server/rtsp-session.c:
8360 * gst/rtsp-server/rtsp-session.h:
8362 Rename GstRTSPMediaBin to GstRTSPMedia
8363 Parse the request url into a GstRTSPUri object and pass this object to the
8364 various handlers and methods that require the uri.
8366 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8370 Add some more docs and remove some old code from the example.
8372 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8374 * gst/rtsp-server/rtsp-client.c:
8375 Handle state change failures better
8376 Handle state change failures better when changing the state of the pipeline to
8379 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8381 * gst/rtsp-server/rtsp-media-factory.c:
8382 * gst/rtsp-server/rtsp-media-factory.h:
8383 Make element creation more extendible
8384 Add get_element vmethod to the default MediaFactory so that subclasses can just
8385 override that method and still use the default logic for making a MediaBin from
8388 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8391 * gst/rtsp-server/Makefile.am:
8392 * gst/rtsp-server/rtsp-client.c:
8393 * gst/rtsp-server/rtsp-client.h:
8394 * gst/rtsp-server/rtsp-media-factory.c:
8395 * gst/rtsp-server/rtsp-media-factory.h:
8396 * gst/rtsp-server/rtsp-media-mapping.c:
8397 * gst/rtsp-server/rtsp-media-mapping.h:
8398 * gst/rtsp-server/rtsp-media.c:
8399 * gst/rtsp-server/rtsp-media.h:
8400 * gst/rtsp-server/rtsp-server.c:
8401 * gst/rtsp-server/rtsp-server.h:
8402 * gst/rtsp-server/rtsp-session.c:
8403 * gst/rtsp-server/rtsp-session.h:
8404 Make the server handle arbitrary pipelines
8405 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
8406 The GstMediaBin object has a handle to a bin with elements and to a list of
8407 GstMediaStream objects that this bin produces.
8408 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
8409 with methods to register and remove those mappings.
8410 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
8411 used by the server instance.
8412 Modify the example application so that it shows how to create custom pipelines
8413 attached to a specific mount point.
8414 Various misc cleanps.
8416 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8418 * gst/rtsp-server/rtsp-server.c:
8419 * gst/rtsp-server/rtsp-server.h:
8420 Allow setting a custom media factory for a server
8422 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8424 * gst/rtsp-server/rtsp-client.c:
8425 * gst/rtsp-server/rtsp-client.h:
8426 Allow setting a custom media factory for a client.
8428 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8430 * gst/rtsp-server/Makefile.am:
8431 Add Makefile entry for the media factory
8433 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8435 * gst/rtsp-server/rtsp-media-factory.c:
8436 * gst/rtsp-server/rtsp-media-factory.h:
8437 Add media factory to map urls to media pipeline objects.
8439 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8441 * gst/rtsp-server/rtsp-media.c:
8442 * gst/rtsp-server/rtsp-media.h:
8443 Add comments. Remove unused field
8445 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8447 * gst/rtsp-server/rtsp-session-pool.c:
8448 * gst/rtsp-server/rtsp-session-pool.h:
8449 Allow custom session pools to override the session id allocation algorithms Add some comments.
8451 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8453 * gst/rtsp-server/rtsp-session.h:
8456 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8458 * gst/rtsp-server/rtsp-client.c:
8459 * gst/rtsp-server/rtsp-client.h:
8460 Move the connection code in one place Add some comments
8462 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8464 * gst/rtsp-server/rtsp-server.c:
8465 * gst/rtsp-server/rtsp-server.h:
8466 Make vmethod to create and accept new clients. Add some docs.
8468 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8470 * gst/rtsp-server/rtsp-server.c:
8471 * gst/rtsp-server/rtsp-server.h:
8472 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
8474 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8476 * gst/rtsp-server/rtsp-client.c:
8477 * gst/rtsp-server/rtsp-client.h:
8478 Name the parameters more appropriately.
8480 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8482 * gst/rtsp-server/rtsp-session-pool.c:
8483 Do some more cleanup of the session pool.
8485 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8487 * gst/rtsp-server/Makefile.am:
8488 * gst/rtsp-server/rtsp-client.c:
8489 Check if return value of gst_rtsp_session_get_media is not NULL
8491 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8493 * gst/rtsp-server/Makefile.am:
8494 Install rtsp-session and rtsp-session-pool headers
8496 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8501 * bindings/python/Makefile.am:
8502 * bindings/python/arg-types.py:
8503 * bindings/python/codegen/Makefile.am:
8504 * bindings/python/codegen/__init__.py:
8505 * bindings/python/codegen/argtypes.py:
8506 * bindings/python/codegen/code-coverage.py:
8507 * bindings/python/codegen/codegen.py:
8508 * bindings/python/codegen/definitions.py:
8509 * bindings/python/codegen/defsparser.py:
8510 * bindings/python/codegen/docextract.py:
8511 * bindings/python/codegen/docgen.py:
8512 * bindings/python/codegen/fileprefix.override:
8513 * bindings/python/codegen/fileprefixmodule.c:
8514 * bindings/python/codegen/h2def.py:
8515 * bindings/python/codegen/mergedefs.py:
8516 * bindings/python/codegen/mkskel.py:
8517 * bindings/python/codegen/override.py:
8518 * bindings/python/codegen/reversewrapper.py:
8519 * bindings/python/codegen/scmexpr.py:
8520 * bindings/python/rtspserver-types.defs:
8521 * bindings/python/rtspserver.defs:
8522 * bindings/python/rtspserver.override:
8523 * bindings/python/rtspservermodule.c:
8525 Add python bindings.
8527 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8529 * bindings/Makefile.am:
8531 Don't go into python dir when requirements for python bindings are missing
8533 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8535 * bindings/Makefile.am:
8536 * bindings/vala/Makefile.am:
8538 Install Vala bindings if vala is available
8540 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8542 * bindings/vala/gst-rtsp-server-0.10.deps:
8543 * bindings/vala/gst-rtsp-server-0.10.vapi:
8544 * bindings/vala/gst-rtsp-server.vapi:
8545 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
8546 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8547 * bindings/vala/packages/gst-rtsp-server-0.10.files:
8548 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8549 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8550 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
8551 * bindings/vala/packages/gst-rtsp-server.deps:
8552 * bindings/vala/packages/gst-rtsp-server.excludes:
8553 * bindings/vala/packages/gst-rtsp-server.files:
8554 * bindings/vala/packages/gst-rtsp-server.gi:
8555 * bindings/vala/packages/gst-rtsp-server.metadata:
8556 * bindings/vala/packages/gst-rtsp-server.namespace:
8557 Regenerated Vala bindings
8559 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8561 * bindings/vala/gst-rtsp-server.vapi:
8562 * bindings/vala/packages/gst-rtsp-server.metadata:
8563 Fixed typo in included headers for vala bindings
8565 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8569 * pkgconfig/Makefile.am:
8570 * pkgconfig/gst-rtsp-server.pc.in:
8571 Added pkgconfig file
8573 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8575 * bindings/vala/gst-rtsp-server.vapi:
8576 * bindings/vala/packages/gst-rtsp-server.excludes:
8577 * bindings/vala/packages/gst-rtsp-server.gi:
8578 * bindings/vala/packages/gst-rtsp-server.metadata:
8579 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
8581 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8583 * bindings/vala/gst-rtsp-server.vapi:
8584 * bindings/vala/packages/gst-rtsp-server.deps:
8585 * bindings/vala/packages/gst-rtsp-server.files:
8586 * bindings/vala/packages/gst-rtsp-server.gi:
8587 * bindings/vala/packages/gst-rtsp-server.metadata:
8588 * bindings/vala/packages/gst-rtsp-server.namespace:
8591 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
8593 * gst/rtsp-server/rtsp-session.c:
8594 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
8596 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8598 * examples/Makefile.am:
8599 * gst/rtsp-server/Makefile.am:
8600 Put GStreamer version in library name
8602 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8604 * examples/Makefile.am:
8605 * gst/rtsp-server/Makefile.am:
8606 Fix some issues to pass distcheck
8608 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8610 * gst/rtsp-server/rtsp-server.c:
8611 Added port property to GstRTSPServer class.
8613 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8618 * examples/Makefile.am:
8621 * gst/rtsp-server/Makefile.am:
8622 * gst/rtsp-server/rtsp-client.c:
8623 * gst/rtsp-server/rtsp-client.h:
8624 * gst/rtsp-server/rtsp-media.c:
8625 * gst/rtsp-server/rtsp-media.h:
8626 * gst/rtsp-server/rtsp-server.c:
8627 * gst/rtsp-server/rtsp-server.h:
8628 * gst/rtsp-server/rtsp-session-pool.c:
8629 * gst/rtsp-server/rtsp-session-pool.h:
8630 * gst/rtsp-server/rtsp-session.c:
8631 * gst/rtsp-server/rtsp-session.h:
8634 * src/rtsp-client.c:
8635 * src/rtsp-client.h:
8638 * src/rtsp-server.c:
8639 * src/rtsp-server.h:
8640 * src/rtsp-session-pool.c:
8641 * src/rtsp-session-pool.h:
8642 * src/rtsp-session.c:
8643 * src/rtsp-session.h:
8644 Split in library and example program
8646 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8648 * src/rtsp-client.h:
8649 Removed obsolete variable
8651 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8653 * src/rtsp-client.c:
8654 * src/rtsp-client.h:
8655 Removed pipeline variable GstRTSPClient, because it's only used in one function
8657 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8660 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
8662 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
8664 * src/rtsp-session.c:
8665 Initialize some more vars.
8667 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
8669 * src/rtsp-session.c:
8670 Initialize variable to avoid compiler warning.
8672 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
8675 Add a reasonable generic .gitignore
projects / platform / upstream / gstreamer.git / blob