3 2015-09-18 Sebastian Dröge <slomo@coaxion.net>
8 2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
10 * docs/libs/gst-rtsp-server-sections.txt:
11 * gst/rtsp-server/rtsp-stream.c:
12 stream: fix docs for recently-added get/set_buffer_size API
13 https://bugzilla.gnome.org/show_bug.cgi?id=749095
15 2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
17 * gst/rtsp-server/rtsp-media.c:
18 rtsp-media: Don't crash on encrypted RTX SDP
19 In parse_keymgmt(), don't mutate the input string that's been passed
20 as const, especially since we might need the original value again if
21 the same key info applies to multiple streams (RTX, for example).
22 https://bugzilla.gnome.org/show_bug.cgi?id=754753
24 2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
26 * examples/test-mp4.c:
27 test-mp4: Support filenames with spaces in them. Error out on too few arguments
29 2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
31 * examples/test-record.c:
32 test-record: Check parameter count and print out help
33 If no launch pipeline was supplied, print out some help
35 2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
37 * gst/rtsp-server/rtsp-media.c:
38 * gst/rtsp-server/rtsp-stream.c:
39 * gst/rtsp-server/rtsp-stream.h:
40 rtsp-stream: Implement UDP buffer size setting.
41 Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
43 Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
44 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
46 2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
48 * gst/rtsp-server/rtsp-media.h:
49 rtsp-media: Fix small typo causing gtk-doc to complain
51 === release 1.5.90 ===
53 2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
59 * gst-rtsp-server.doap:
62 2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
64 * gst/rtsp-server/rtsp-media-factory.c:
65 media-factory: get port number through gst_rtsp_url_get_port
66 https://bugzilla.gnome.org/show_bug.cgi?id=753473
68 2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
70 * tests/check/gst/media.c:
71 media-test: Removing unnecessary assertion
72 https://bugzilla.gnome.org/show_bug.cgi?id=753385
74 2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
76 * gst/rtsp-server/rtsp-server.c:
77 Document that source keeps a ref on server until it's destroyed
78 https://bugzilla.gnome.org/show_bug.cgi?id=749227
80 2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
82 * tests/check/gst/media.c:
83 media-test: Test for multiple dynamic payload
84 https://bugzilla.gnome.org/show_bug.cgi?id=753385
86 2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
88 * gst/rtsp-server/rtsp-media.c:
89 media: Only add fakesink once per pipeline
90 The intention is to prevent going PLAYING state before pads are created.
91 If there was mutilple dynamic payload, it would leak few fakesink and
92 actually prevent from ever reaching playing state.
93 https://bugzilla.gnome.org/show_bug.cgi?id=753385
95 2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
97 * gst/rtsp-server/rtsp-media.c:
98 Revert "rtsp-media: Only add 1 fakesink per pipeline"
99 This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
101 2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
103 * gst/rtsp-server/rtsp-media.c:
104 rtsp-media: Only add 1 fakesink per pipeline
105 There should be only one fakesink per pipeline, not per dynpay. This
106 would lead to element naming clash.
108 2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
110 * gst/rtsp-server/rtsp-media.c:
111 rtsp-media: assertion error due to wrong condition check
112 In media to caps function, reserved_keys array is being used for variable i,
113 leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
114 changed it to variable j
115 https://bugzilla.gnome.org/show_bug.cgi?id=753009
117 2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
119 * gst/rtsp-server/rtsp-media.c:
120 rtsp-media: Strip keys from the fmtp that we use internally in our caps
121 Skip keys from the fmtp, which we already use ourselves for the
122 caps. Some software is adding random things like clock-rate into
123 the fmtp, and we would otherwise here set a string-typed clock-rate
124 in the caps... and thus fail to create valid RTP caps
125 https://bugzilla.gnome.org/show_bug.cgi?id=753009
127 2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
129 * gst/rtsp-server/rtsp-thread-pool.c:
130 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
131 https://bugzilla.gnome.org/show_bug.cgi?id=752640
133 2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
136 Automatic update of common submodule
137 From f74b2df to 9aed1d7
139 2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
144 === release 1.5.2 ===
146 2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
152 * gst-rtsp-server.doap:
155 2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
157 * gst/rtsp-server/rtsp-client.c:
158 * gst/rtsp-server/rtsp-client.h:
159 * tests/check/gst/client.c:
160 rtsp-client: allow application to decide what requirements are supported
161 Add "check-requirements" signal and vfunc to allow application
162 (and subclasses) to check the requirements.
163 Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
164 https://bugzilla.gnome.org/show_bug.cgi?id=749417
166 2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
169 Automatic update of common submodule
170 From 6015d26 to f74b2df
172 2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
174 * gst/rtsp-server/rtsp-media.c:
175 rtsp-media: Always use real payloader when creating streams
176 A bin that contains the real payloader might be used as payloader. In this
177 case we have to get the real payloader for the various properties it provides.
178 Example use cases for this are bins that payload some media and then have
179 additional elements that add metadata or RTP extension headers to the stream.
180 https://bugzilla.gnome.org/show_bug.cgi?id=750800
182 2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
184 * examples/test-netclock-client.c:
185 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
187 2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
189 * examples/test-netclock-client.c:
190 * examples/test-netclock.c:
191 test-netclock: Use new ntp-time-source property on rtpbin
192 Select the clock time to be used as NTP time source. This allows proper
193 synchronization between receivers, independent of sharing base times, and just
194 requires them to use the same clock.
196 2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
198 * examples/test-netclock-client.c:
199 * examples/test-netclock.c:
200 test-netclock: Setting the same base time on sender and receiver is not necessary
201 It's going to be fixed up by rtpbin when using ntp-sync=TRUE
203 2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
205 * gst/rtsp-server/rtsp-stream.c:
206 rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
207 https://bugzilla.gnome.org/show_bug.cgi?id=750764
209 2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
211 * docs/libs/gst-rtsp-server.types:
212 docs: add missing types
213 https://bugzilla.gnome.org/show_bug.cgi?id=750764
215 2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
217 * docs/libs/gst-rtsp-server-sections.txt:
218 docs: add missing apis
219 https://bugzilla.gnome.org/show_bug.cgi?id=750764
221 2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
223 * examples/test-netclock-client.c:
224 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
226 2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
228 * docs/libs/gst-rtsp-server-sections.txt:
229 * gst/rtsp-server/rtsp-auth.c:
230 * gst/rtsp-server/rtsp-auth.h:
231 GstRTSPAuth: Add client certificate authentication support
232 https://bugzilla.gnome.org/show_bug.cgi?id=750471
234 2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
236 * examples/test-netclock-client.c:
237 test-netclock-client: Use new GstClock API to wait for clock synchronization
239 2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
241 * examples/test-netclock-client.c:
242 test-netclock-client: Use a GMainLoop and playbin's source-setup signal
243 A mainloop is needed to get glimagesink to display something on OSX, and
244 the source-setup signal just makes things a little bit easier.
246 2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
249 Automatic update of common submodule
250 From d9a3353 to 6015d26
252 2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
255 Automatic update of common submodule
256 From d37af32 to d9a3353
258 2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
261 Automatic update of common submodule
262 From 21ba2e5 to d37af32
264 2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
267 Automatic update of common submodule
268 From c408583 to 21ba2e5
270 2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
272 * docs/libs/Makefile.am:
273 docs: remove variables that we define in the snippet from common
274 This is syncing our Makefile.am with upstream gtkdoc.
276 2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
279 Automatic update of common submodule
280 From 44a3517 to c408583
282 2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
287 === release 1.5.1 ===
289 2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
295 * gst-rtsp-server.doap:
298 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
300 * gst/rtsp-server/rtsp-client.c:
301 rtsp-client: No flush during Teardown.
302 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
303 backlog is empty it can happen that just a part of a message will be
304 sent and rest is in backlog queue. If then flush during teardown
305 just a part of message will be sent.This can lead to client miss
306 teardown response since it expect to get the last part of message.
307 The flushing during teardown was introduced to fix a deadlock that now
308 is fixed more generally in handle_request by temporary setting backlog
310 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
312 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
314 * tests/check/Makefile.am:
315 tests: Use AM_TESTS_ENVIRONMENT
316 Needed by the new automake test runner and the
317 current version of the common submodule.
319 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
321 * gst/rtsp-server/rtsp-media.h:
322 * gst/rtsp-server/rtsp-stream.h:
323 rtsp-server: Use single-include rtsp header to make sure we get all definitions
325 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
327 * gst/rtsp-server/rtsp-media.c:
328 rtsp-media: Mark some more functions static
330 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
332 * gst/rtsp-server/rtsp-media.c:
333 rtsp-media: Only unblock the media in suspend() when actually changing the state
334 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
336 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
338 * examples/test-video-rtx.c:
339 examples: Use AVPF profile for the RTX example
341 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
343 * gst/rtsp-server/rtsp-sdp.c:
344 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
346 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
348 * gst/rtsp-server/rtsp-stream.c:
349 rtsp-stream: get valid clock-rate from last-sample
350 clock-rate in last-sample's caps is integer, not unsigned.
351 To get this value properly, variable needs to be type-casted to int.
352 https://bugzilla.gnome.org/show_bug.cgi?id=747614
354 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
358 autogen.sh: only run autopoint if gettext requested in configure.ac
359 Not just because there happens to be a po directory.
360 https://bugzilla.gnome.org/show_bug.cgi?id=748058
362 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
365 Revert "configure.ac: uncomment gettext version setup"
366 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
367 We don't need a gettext setup here and there's no po
368 directory either, so no reason why autopoint would be
369 run in the first place.
370 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
372 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
374 * examples/test-multicast.c:
375 * examples/test-multicast2.c:
376 * examples/test-sdp.c:
377 * examples/test-video-rtx.c:
378 * examples/test-video.c:
379 * tests/test-cleanup.c:
380 * tests/test-reuse.c:
381 Fix timeout function signatures across tests and examples
383 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
385 * tests/check/Makefile.am:
386 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
387 Make sure the test environment is set up.
388 https://bugzilla.gnome.org//show_bug.cgi?id=747624
390 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
393 configure: bump automake requirement to 1.14 and autoconf to 2.69
394 This is only required for builds from git, people can still
395 build tarballs if they only have older autotools.
396 https://bugzilla.gnome.org//show_bug.cgi?id=747624
398 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
401 configure.ac: uncomment gettext version setup
402 Fixes autogen.sh. It would run autopoint, which would complain
403 that it could not find the gettext version in configure.ac.
404 https://bugzilla.gnome.org/show_bug.cgi?id=748058
406 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
408 * examples/test-video-rtx.c:
409 test-video-rtx: set exact payload type to PCMA payloader
410 Setting wrong payload type causes failure to do retransmission through audio stream
411 https://bugzilla.gnome.org/show_bug.cgi?id=747839
413 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
415 * gst/rtsp-server/rtsp-media.c:
416 * gst/rtsp-server/rtsp-stream.c:
417 * gst/rtsp-server/rtsp-stream.h:
418 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
419 Because of duplicated g_signal_connect for request-aux-sender signal,
420 wrong stream pointer is passed to the signal handler.
421 Instead of passing each stream, pass stream array and get the relevant stream.
422 https://bugzilla.gnome.org/show_bug.cgi?id=747839
424 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
428 Update autogen.sh to latest version from common
429 Fixes build after aclocal_check etc. helpers have been removed.
431 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
434 Automatic update of common submodule
435 From bc76a8b to c8fb372
437 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
439 * gst/rtsp-server/rtsp-stream.c:
440 rtsp-stream: Limit the queues to 1 buffer
441 We only need them to be able to pre-roll, queueing up more data here
442 is only going to harm latency and memory usage.
444 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
446 * gst/rtsp-server/rtsp-stream.c:
447 rtsp-stream: Update comment and ASCII art to the latest code
448 We have a queue in front of the udpsink too to prevent the pipeline from
451 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
453 * gst/rtsp-server/rtsp-stream.c:
454 rtsp-media: Properly return first rtptime
455 Instead we where returning first GstBuffer timestamp. This would result
456 in clock skew and unwanted behaviour in RTSP playback.
457 https://bugzilla.gnome.org/show_bug.cgi?id=746479
459 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
461 * gst/rtsp-server/rtsp-stream.c:
462 rtsp-stream: Don't leave buffer mapped
463 If the seq is NULL, the RTP buffer was left mapped. We should always
466 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
471 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
473 * gst/rtsp-server/rtsp-media-factory.c:
474 * tests/check/gst/client.c:
475 Fix double semicolons
477 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
479 * gst/rtsp-server/rtsp-stream.c:
480 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
481 This gives more accurate values than asking the payloader. There might be
482 queueing happening between the payloader and the sink.
483 https://bugzilla.gnome.org/show_bug.cgi?id=745704
485 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
487 * gst/rtsp-server/rtsp-media.c:
488 rtsp-media: Don't seek for PLAY if the position will not change
489 https://bugzilla.gnome.org/show_bug.cgi?id=745704
491 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
493 * gst/rtsp-server/rtsp-media.c:
494 rtsp-media: Don't include payload type in the caps for framesize
495 When the sdp media attribute framesize are converted to caps
496 the <payload> should not be included.
497 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
498 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
500 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
502 * gst/rtsp-server/rtsp-sdp.c:
503 rtsp-sdp: add payload type to the sdp framesize attribute
504 The sdp framesize attribute is desribed in RFC6064. It is specified
505 for payloading of H263 and has the following form
506 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
507 should be added to the caps in a payloader and the <payload type> should
508 be added by the rtsp-server.
509 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
511 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
513 * examples/test-uri.c:
514 examples: test-uri: fix tainted variable
515 Insignificant but this keeps Coverity happy.
518 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
520 * examples/.gitignore:
521 * examples/Makefile.am:
522 * examples/test-netclock-client.c:
523 * examples/test-netclock.c:
524 examples: Add a simple example of network synch for live streams.
525 An example server and client that works for synchronising live streams
526 only - as it can't support pause/play.
528 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
530 * gst/rtsp-server/rtsp-media-factory.c:
531 * gst/rtsp-server/rtsp-media-factory.h:
532 rtsp-media-factory: Add functions to set/get the media gtype
533 Allow specifying the GType of a GstRtspMedia subclass to create
534 as a simpler way to get the factory to create a custom
535 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
537 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
539 * gst/rtsp-server/rtsp-media.c:
540 rtsp-media: fix double unlock in _get_buffer_size()
541 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
542 because of double g_mutex_unlock () usage.
543 https://bugzilla.gnome.org/show_bug.cgi?id=745434
545 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
547 * gst/rtsp-server/rtsp-session-pool.c:
548 * gst/rtsp-server/rtsp-session.c:
549 * gst/rtsp-server/rtsp-session.h:
550 rtsp-session: Use monotonic time for RTSP session timeout
551 Changed RTSP session timeout handling to monotonic time
552 and deprecating the API for current system time.
553 This fixes timeouts when the system time changes.
554 https://bugzilla.gnome.org/show_bug.cgi?id=743346
556 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
558 * gst/rtsp-server/rtsp-client.c:
559 * gst/rtsp-server/rtsp-media.c:
560 rtsp-client: Only error out in PLAY if seeking actually failed
561 If the media was just not seekable, we continue from whatever position we are
562 and let the client decide if that is what is wanted or not.
563 Only if the actual seek failed, we can't really recover and should error out.
565 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
567 * gst/rtsp-server/rtsp-stream.c:
568 rtsp-stream: Add necessary queues between tee and multiudpsink
569 https://bugzilla.gnome.org/show_bug.cgi?id=744379
571 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
573 * gst/rtsp-server/rtsp-client.c:
574 * gst/rtsp-server/rtsp-media.c:
575 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
576 Instead error out properly the same way as if the SEEKING query already
579 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
581 * gst/rtsp-server/rtsp-stream.h:
582 rtsp-stream: minor code formatting fix
584 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
586 * gst/rtsp-server/rtsp-media.c:
587 rtsp-media: fix logic for collect_streams
588 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
589 all streams it knows if it got any, and can check if the transport mode is OK.
592 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
594 * gst/rtsp-server/rtsp-media.c:
595 rtsp-media: Don't set the transport mode based on what elements we find
596 Just print a warning if the one that was set before disagrees with what
597 elements we found. It must already be set to something before as this
598 function is called after we received the SDP from ANNOUNCE in RECORD mode,
599 and we would reject ANNOUNCE if the RECORD flag was not set.
601 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
603 * tests/check/gst/rtspserver.c:
604 tests: rtspserver: rename shadowed variable
605 We have two different 'sink' variables here,
606 rename one of them for clarity.
608 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
610 * gst/rtsp-server/rtsp-client.c:
611 rtsp-client: fix awkward if clause
613 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
615 * examples/test-uri.c:
616 examples: test-uri: improve uri argument handling and accept file names
617 Print an error if the argument passed is not a URI and can't
618 be converted into one, or no arguments have been provided.
620 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
622 * examples/test-uri.c:
623 examples: test-uri: don't remove mount point after 10 seconds
624 It's very irritating when trying to test stuff repeatedly
625 and serves no real purpose other than showing that it can
628 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
630 * examples/.gitignore:
631 examples: add new test-record to .gitignore
633 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
635 * examples/test-record.c:
636 * gst/rtsp-server/rtsp-client.c:
637 * gst/rtsp-server/rtsp-media-factory.c:
638 * gst/rtsp-server/rtsp-media-factory.h:
639 * gst/rtsp-server/rtsp-media.c:
640 * gst/rtsp-server/rtsp-media.h:
641 * tests/check/gst/rtspserver.c:
642 rtsp-media: Use flags to distinguish between PLAY and RECORD media
644 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
646 * examples/test-record.c:
647 test-record: Set latency for playback-style example to 2s instead of 200ms
649 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
651 * tests/check/gst/rtspserver.c:
652 tests: add some unit tests for ANNOUNCE and RECORD
653 https://bugzilla.gnome.org/show_bug.cgi?id=743175
655 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
657 * gst/rtsp-server/rtsp-client.c:
658 rtsp-client: fix a couple of leaks in handle_announce
660 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
662 * gst/rtsp-server/rtsp-media-factory.c:
663 * gst/rtsp-server/rtsp-media-factory.h:
664 * gst/rtsp-server/rtsp-media.c:
665 * gst/rtsp-server/rtsp-media.h:
666 rtsp-media: Expose latency setting for setting the rtpbin latency
668 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
670 * examples/test-record.c:
671 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
673 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
675 * gst/rtsp-server/rtsp-stream.c:
676 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
678 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
680 * examples/Makefile.am:
681 * examples/test-record.c:
682 * gst/rtsp-server/rtsp-client.c:
683 * gst/rtsp-server/rtsp-client.h:
684 * gst/rtsp-server/rtsp-media-factory.c:
685 * gst/rtsp-server/rtsp-media-factory.h:
686 * gst/rtsp-server/rtsp-media.c:
687 * gst/rtsp-server/rtsp-media.h:
688 * gst/rtsp-server/rtsp-session-media.c:
689 * gst/rtsp-server/rtsp-stream.c:
690 * gst/rtsp-server/rtsp-stream.h:
691 Add initial support for RECORD
692 We currently only support media that is RECORD or PLAY only, not both at once.
693 https://bugzilla.gnome.org/show_bug.cgi?id=743175
695 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
697 * gst/rtsp-server/rtsp-stream.c:
698 rtsp-stream: RTCP and RTP transport cache cookies seperated
699 RTCP packets were not sent because the same tr_cache_cookie was used for
700 both RTP and RTCP. So only one of the tr_cache lists were populated
701 depending on which one was sent first. If the tr_cache list is not
702 populated then no packets can be sent. Most often this happened to be
703 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
704 resulted in both the tr_cache_lists to be populated regardless of which
706 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
708 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
710 * gst/rtsp-server/rtsp-stream.c:
711 rtsp-stream: fix false compiler warning
712 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
714 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
716 * gst/rtsp-server/rtsp-client.c:
717 rtsp-client: log interleaved data received
719 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
721 * gst/rtsp-server/rtsp-client.c:
722 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
724 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
726 * gst/rtsp-server/rtsp-client.c:
727 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
729 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
731 * gst/rtsp-server/rtsp-client.c:
732 rtsp-client: Use a random session ID in the SDP
733 RFC4566 Section 5.2 says that it should make the username, session id,
734 nettype, addrtype and unicast address tuple globally unique. Always using
735 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
736 Instead let's create a 64 bit random number, which at least brings us
737 closer to the goal of global uniqueness.
738 https://tools.ietf.org/html/rfc4566#section-5.2
740 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
742 * examples/test-launch.c:
743 * examples/test-mp4.c:
744 * examples/test-ogg.c:
745 * examples/test-uri.c:
746 examples: Don't call gst_init() and gst_get_option_group()
747 The latter calls the former at the appropriate time.
749 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
751 * gst/rtsp-server/rtsp-client.c:
752 rtsp-client: Drop trailing \0 of RTSP DATA messages
753 We add a trailing \0 in GstRTSPConnection to make parsing of
754 string message bodies easier (e.g. the SDP from DESCRIBE) but
755 for actual data this means we have to drop it or otherwise
758 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
760 * gst/rtsp-server/rtsp-stream.c:
761 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
762 Fixes crash when two threads access handle_new_sample() at the same
763 time, one for RTP, one for RTCP.
764 Otherwise, when iterating over the transports cache, it might be modified by
765 another thread at the same time if the transports cookie has changed.
766 https://bugzilla.gnome.org/show_bug.cgi?id=742954
768 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
770 * gst/rtsp-server/rtsp-stream.c:
771 rtsp-stream: Set format=TIME on our app sources for TCP
773 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
775 * gst/rtsp-server/rtsp-session-pool.c:
776 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
777 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
778 RFC 2326 states that session IDs may consist of alphanumeric as well as
779 the safe characters $-_.+ -- N.B. the percent character is not allowed.
780 Previously the session ID was URI-escaped, this meant that any character
781 which was not alphanumeric or any of the characters +-._~ would be
782 percent encoded. While the RFC (surprisingly) mentions that linear white
783 space in session IDs should be URI-escaped, it does not say anything
784 about other characters. Moreover no white space is allowed in the
785 session ID. Finally the percent character which is the result of
786 URI-escaping is not allowed in a session ID.
787 So there is no reason to do any URI-escaping, and now it is removed.
788 https://bugzilla.gnome.org/show_bug.cgi?id=742869
790 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
793 Automatic update of common submodule
794 From f2c6b95 to bc76a8b
796 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
799 Fix 'make check' from top-level directory
801 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
803 * examples/test-launch.c:
804 * examples/test-mp4.c:
805 * examples/test-ogg.c:
806 * examples/test-uri.c:
807 examples: Add command-line parsing and take a 'port' argument
808 This allows users to run multiple servers on different ports for testing.
809 Only done for examples that actually take arguments and hence are capable of
810 outputting different streams for each instance on each port.
811 https://bugzilla.gnome.org/show_bug.cgi?id=742115
813 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
815 * gst/rtsp-server/rtsp-client.c:
816 * gst/rtsp-server/rtsp-client.h:
817 rtsp-client: Add a send_message default signal handler
818 This allows subclasses to easily hook into the response sending
819 mechanism without doing everything from a signal, which seems
820 awkward from subclasses.
822 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
825 Automatic update of common submodule
826 From ef1ffdc to f2c6b95
828 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
832 configure: add --disable-examples switch
833 https://bugzilla.gnome.org/show_bug.cgi?id=741678
835 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
837 * examples/.gitignore:
838 * examples/Makefile.am:
839 * examples/test-video-rtx.c:
840 examples: add a retransmisison example implementing RFC4588
841 Currently only SSRC-multiplexed rtx streams are supported
843 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
845 * gst/rtsp-server/rtsp-stream.c:
846 rtsp-stream: Fix some minor memory leaks
848 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
850 * gst/rtsp-server/rtsp-media.c:
851 rtsp-media: Some minor cleanup
853 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
855 * gst/rtsp-server/rtsp-stream.c:
856 rtsp-stream: Fix compiler warnings
857 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
858 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
860 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
861 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
864 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
866 * docs/libs/gst-rtsp-server-sections.txt:
867 * gst/rtsp-server/rtsp-media-factory.c:
868 * gst/rtsp-server/rtsp-media-factory.h:
869 * gst/rtsp-server/rtsp-media.c:
870 * gst/rtsp-server/rtsp-media.h:
871 * gst/rtsp-server/rtsp-sdp.c:
872 * gst/rtsp-server/rtsp-stream.c:
873 * gst/rtsp-server/rtsp-stream.h:
874 media: implement ssrc-multiplexed retransmission support
875 based off RFC 4588 and the server-rtpaux example in -good
877 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
879 * gst/rtsp-server/rtsp-client.c:
880 * gst/rtsp-server/rtsp-stream-transport.c:
881 * gst/rtsp-server/rtsp-stream.c:
882 rtsp: Ref transports in hash table.
883 Also ref streams for transports.
884 This solves a crash when reciving a rtcp after teardown but before
886 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
888 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
891 Automatic update of common submodule
892 From 7bb2bce to ef1ffdc
894 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
896 * gst/rtsp-server/rtsp-client.c:
897 client: refactor cleanup of cached media
899 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
901 * tests/check/gst/client.c:
903 The session leak is now fixed, lets remove those FIXME comments.
905 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
907 * tests/check/gst/rtspserver.c:
908 tests: Test to setup two sessions on one connection
909 https://bugzilla.gnome.org/show_bug.cgi?id=739112
911 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
913 * tests/check/gst/rtspserver.c:
914 tests: Test setup with tcp transport
915 https://bugzilla.gnome.org/show_bug.cgi?id=739112
917 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
919 * gst/rtsp-server/rtsp-client.c:
920 client: Configure transport after creating session media
921 The default implementation of configure_client_transport() in
922 rtsp-client uses the session media when it chooses channels for
924 https://bugzilla.gnome.org/show_bug.cgi?id=739112
926 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
928 * gst/rtsp-server/rtsp-client.c:
929 * gst/rtsp-server/rtsp-session-media.c:
930 client: Stop caching media in client when doing setup
931 If the media has been managed by a session media, it should not be
932 cached in the client any longer. The GstRTSPSessionMedia object is now
933 responsible for unpreparing the GstRTSPMedia object using
934 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
936 https://bugzilla.gnome.org/show_bug.cgi?id=739112
938 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
940 * gst/rtsp-server/rtsp-stream.c:
941 rtsp-stream: unref srtp decoder when leaving bin
942 https://bugzilla.gnome.org/show_bug.cgi?id=739481
944 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
946 * gst/rtsp-server/rtsp-client.c:
947 rtsp-client: mikey memory leaks
948 https://bugzilla.gnome.org/show_bug.cgi?id=739383
950 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
953 Automatic update of common submodule
954 From 84d06cd to 7bb2bce
956 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
959 Parallelise 'make check-valgrind'
961 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
964 Automatic update of common submodule
965 From a8c8939 to 84d06cd
967 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
970 Automatic update of common submodule
971 From 36388a1 to a8c8939
973 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
975 * gst/rtsp-server/rtsp-media.c:
976 rtsp-media: deactivate media when shutting down from paused
977 This was only done when going directly from playing.
978 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
980 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
982 * gst/rtsp-server/rtsp-client.c:
983 * gst/rtsp-server/rtsp-context.h:
984 rtsp-client: add stream transport to context
985 We add the stream transport to the context so we can get the configured
986 client stream transport in the setup request signal.
987 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
989 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
991 * gst/rtsp-server/rtsp-stream.c:
992 stream: release lock even not all transports have been removed
993 We don't want to keep the lock even we return FALSE because not all the
994 transports have been removed. This could lead into a deadlock.
995 https://bugzilla.gnome.org/show_bug.cgi?id=737797
997 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
999 * gst/rtsp-server/rtsp-sdp.c:
1000 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
1001 These were renamed in GstRTPBasePayload in 1.0
1003 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1005 * gst/rtsp-server/rtsp-client.c:
1006 client: set session media to NULL without the lock
1007 We need to set session medias to NULL without the client lock otherwise
1008 we can end up in a deadlock if another thread is waiting for the lock
1009 and media unprepare is also waiting for that thread to end.
1010 https://bugzilla.gnome.org/show_bug.cgi?id=737690
1012 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
1014 * gst/rtsp-server/rtsp-media.c:
1015 rtsp-media: Set state to UNPREPARING in all cases
1017 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
1019 * gst/rtsp-server/rtsp-media.c:
1020 media: set state to unpreparing when unprepare is initiated
1021 https://bugzilla.gnome.org/show_bug.cgi?id=737675
1023 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
1025 * gst/rtsp-server/rtsp-client.c:
1026 rtsp-client: Remove backlog limit while processings requests
1027 If the backlog limit is kept two cases of deadlocks may be
1028 encountered when streaming over TCP. Without the backlog
1029 limit this deadlocks can not happen, at the expence of
1031 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
1033 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
1035 * gst/rtsp-server/rtsp-client.c:
1036 rtsp-client: do not free main context before rtsp watch
1037 https://bugzilla.gnome.org/show_bug.cgi?id=737110
1039 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
1041 * tests/check/gst/rtspserver.c:
1042 tests: Extend unit test timeout to accomodate for valgrind
1043 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1045 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
1047 * gst/rtsp-server/rtsp-client.c:
1048 * gst/rtsp-server/rtsp-session.c:
1049 * gst/rtsp-server/rtsp-stream-transport.c:
1050 rtsp-*: Treat sending packets to clients as keepalive
1051 As long as gst-rtsp-server can successfully send RTP/RTCP data to
1052 clients then the client must be reading. This change makes the server
1053 timeout the connection if the client stops reading.
1054 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1056 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
1058 * gst/rtsp-server/rtsp-client.c:
1059 rtsp-client: Allow backlog to grow while expiring session
1060 Allow the send backlog in the RTSP watch to grow to unlimited size while
1061 attempting to bring the media pipeline to NULL due to a session
1062 expiring. Without this change the appsink element cannot change state
1063 because it is blocked while rendering data in the new_sample callback.
1064 This callback will block until it has successfully put the data into the
1065 send backlog. There is a chance that the send backlog is full at this
1066 point which means that the callback may block for a long time, possibly
1067 forever. Therefore the media pipeline may also be prevented from
1068 changing state for a long time.
1069 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
1071 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
1073 * gst/rtsp-server/rtsp-client.c:
1074 rtsp-client: Make old compilers happy
1075 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
1076 Just in case that guint8 doesn't fit in a pointer. Just in case ...
1078 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
1080 * gst/rtsp-server/rtsp-client.c:
1081 client: raise the backlog limits before pausing
1082 We need to raise the backlog limits before pausing the pipeline or else
1083 the appsink might be blocking in the render method in wait_backlog() and
1084 we would deadlock waiting for paused.
1085 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
1087 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
1089 * gst/rtsp-server/rtsp-client.c:
1090 client: make define for the WATCH_BACKLOG
1091 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
1093 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
1095 * gst/rtsp-server/rtsp-client.c:
1096 client: simplify session transport handling
1097 link/unlink of the transport in a session was done to keep track of all
1098 TCP transports and to send RTP/RTCP data to the streams. We can simplify
1099 that by putting all the TCP transports in a hashtable indexed with the
1101 We also don't need to link/unlink the transports when we pause/resume
1102 the streams. The same effect is already achieved when we pause/play the
1103 media. Indeed, when we pause the media, the transport is removed from
1104 the media and the callbacks will not be called anymore.
1105 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
1107 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
1109 * gst/rtsp-server/rtsp-stream-transport.c:
1110 * gst/rtsp-server/rtsp-stream-transport.h:
1111 stream-transport: make method to handle received data
1112 Make a method to handle the data received on a channel. It sends the
1113 data to the stream of the transport on the RTP or RTCP pads based on
1116 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
1118 * examples/test-mp4.c:
1119 test: add example of dumping RTCP reports
1121 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
1123 * gst/rtsp-server/rtsp-media.c:
1124 * gst/rtsp-server/rtsp-stream.c:
1125 * gst/rtsp-server/rtsp-stream.h:
1126 rtsp-media: Make sure that sequence numbers are monotonic after pause
1127 The sequence number is not monotonic for RTP packets after pause. The
1128 reason is basepayloader generates a randon sequence number when the
1129 pipeline goes from ready to pause. With this fix generation of sequence
1130 number will be monotonic when going from pause to play request.
1131 https://bugzilla.gnome.org/show_bug.cgi?id=736017
1133 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
1135 * gst/rtsp-server/rtsp-client.c:
1136 rtsp-client: Protect saved clients watch with a mutex
1137 Fixes a crash when close() is called while merging clients
1138 in handle_tunnel(). In that case close() would destroy the
1139 watch while it is still being used in handle_tunnel().
1140 https://bugzilla.gnome.org/show_bug.cgi?id=735570
1142 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
1144 * gst/rtsp-server/rtsp-stream.c:
1145 rtsp-stream: Remove the multicast group udp sources when removing from the bin
1147 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
1149 * gst/rtsp-server/rtsp-media.c:
1150 * gst/rtsp-server/rtsp-stream.c:
1151 * gst/rtsp-server/rtsp-stream.h:
1152 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
1153 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
1154 seeking and will always continue counting the time. This leads to
1155 the NPT after a backwards seek to be something completely different
1156 to the actual seek position.
1157 https://bugzilla.gnome.org/show_bug.cgi?id=732644
1159 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
1161 * examples/test-appsrc.c:
1162 examples: fix another reference leak
1163 gst_rtsp_media_get_element() returns a new ref.
1165 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1167 * examples/test-appsrc.c:
1168 examples: unref element after usage
1169 gst_bin_get_by_name_recurse_up() returns an element
1170 reference that must be unreffed after usage.
1171 https://bugzilla.gnome.org/show_bug.cgi?id=734546
1173 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
1175 * gst/rtsp-server/rtsp-media.c:
1176 signals: Fix copy-pasto in target-state signal offset
1178 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
1182 Makefile: Add usage of build-checks step
1183 Allows building checks without running them
1185 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
1187 * gst/rtsp-server/rtsp-stream.c:
1188 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
1189 When a UDP multicast transport is used it is expected that the server listens
1190 for RTP and RTCP packets on the multicast group with the corresponding port.
1191 Without this we will never get RTCP packets from clients in multicast mode.
1192 https://bugzilla.gnome.org/show_bug.cgi?id=732238
1194 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
1199 === release 1.4.0 ===
1201 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1207 * gst-rtsp-server.doap:
1210 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
1212 * gst/rtsp-server/rtsp-media.h:
1213 media: correct misspelled words in description
1214 https://bugzilla.gnome.org/show_bug.cgi?id=733244
1216 === release 1.3.91 ===
1218 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
1224 * gst-rtsp-server.doap:
1227 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
1229 * docs/libs/gst-rtsp-server-sections.txt:
1232 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
1234 * gst/rtsp-server/rtsp-server.c:
1235 server: implement client REMOVE filter
1237 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
1239 * gst/rtsp-server/rtsp-client.c:
1240 * gst/rtsp-server/rtsp-client.h:
1241 client: expose _close() method
1242 Expose a previously internal close method to close the client
1245 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
1247 * gst/rtsp-server/rtsp-session-pool.c:
1248 session-pool: signal session-removed outside of the lock
1249 Release the lock before emiting the session-removed signal.
1251 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
1253 * gst/rtsp-server/rtsp-client.c:
1254 * gst/rtsp-server/rtsp-server.c:
1255 * gst/rtsp-server/rtsp-session-pool.c:
1256 * gst/rtsp-server/rtsp-session.c:
1257 * gst/rtsp-server/rtsp-stream.c:
1258 filter: Release lock in filter functions
1259 Release the object lock before calling the filter functions. We need to
1260 keep a cookie to detect when the list changed during the filter
1261 callback. We also keep a hashtable to make sure we only call the filter
1262 function once for each object in case of concurrent modification.
1263 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
1265 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
1267 * gst/rtsp-server/rtsp-client.c:
1268 client: check if watch is set in handle_teardown()
1269 The unit tests run without a watch
1271 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1273 * tests/check/gst/client.c:
1274 client tests: send teardown to cleanup session
1276 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
1278 * tests/check/gst/rtspserver.c:
1279 server tests: send teardown to cleanup session
1281 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1283 * gst/rtsp-server/rtsp-client.c:
1284 client: keep ref to client for the session removed handler
1285 This extra ref will be dropped when all client sessions have been
1286 removed. A session is removed when a client sends teardown, closes its
1287 endpoint of the TCP connection or the sessions expires.
1288 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1290 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
1292 * gst/rtsp-server/rtsp-client.c:
1293 * gst/rtsp-server/rtsp-session.c:
1294 * tests/check/gst/client.c:
1295 client: manage media in session as a last step
1296 Once we manage a media in a session, we can't unmanage it anymore
1297 without destroying it. Therefore, first check everything before we
1298 manage the media, otherwise if something is wrong we have no way to
1300 If we created a new session and something went wrong, remove the session
1301 again. Fixes a leak in the unit test.
1303 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1305 * examples/test-mp4.c:
1306 * examples/test-ogg.c:
1307 examples: print 'stream ready at url' for mp4 and ogg example
1309 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
1311 * gst/rtsp-server/rtsp-client.c:
1312 * gst/rtsp-server/rtsp-sdp.c:
1313 rtsp: fix for MIKEY api change
1315 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
1317 * gst/rtsp-server/rtsp-client.c:
1318 client: free watch context only once
1319 The watch context is freed when the source is destroyed. Avoids
1320 a CRITICAL when we try to unref the context twice.
1322 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
1324 * gst/rtsp-server/rtsp-client.c:
1327 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
1329 * gst/rtsp-server/rtsp-client.c:
1330 client: protect sessions with lock
1331 Protect the list of sessions with the lock.
1332 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1334 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
1336 * gst/rtsp-server/rtsp-client.c:
1337 Client: keep a ref to the session
1338 Don't just keep a weak ref to the session objects but use a hard ref. We
1339 will be notified when a session is removed from the pool (expired) with
1340 the new session-removed signal.
1341 Don't automatically close the RTSP connection when all the sessions of
1342 a client are removed, a client can continue to operate and it can create
1343 a new session if it wants. If you want to remove the client from the
1344 server, you have to use gst_rtsp_server_client_filter() now.
1345 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
1346 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
1348 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
1350 * gst/rtsp-server/rtsp-session-pool.c:
1351 * gst/rtsp-server/rtsp-session-pool.h:
1352 session-pool: add session-removed signal
1353 Add a signal to be notified when a session is removed from the pool.
1355 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
1357 * gst/rtsp-server/Makefile.am:
1358 * gst/rtsp-server/rtsp-server.h:
1359 Make rtsp-server.h a single-include header, use it for G-I
1360 https://bugzilla.gnome.org/show_bug.cgi?id=732411
1362 === release 1.3.90 ===
1364 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1370 * gst-rtsp-server.doap:
1373 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
1375 * gst/rtsp-server/rtsp-stream.c:
1376 stream: crypto can be NULL
1378 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
1380 * gst/rtsp-server/rtsp-client.c:
1381 * gst/rtsp-server/rtsp-media.c:
1382 * gst/rtsp-server/rtsp-mount-points.c:
1383 introspection: add missing allow-none annotations
1384 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1386 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
1388 * gst/rtsp-server/rtsp-address-pool.c:
1389 * gst/rtsp-server/rtsp-media.c:
1390 * gst/rtsp-server/rtsp-session-media.c:
1391 * gst/rtsp-server/rtsp-session-pool.c:
1392 * gst/rtsp-server/rtsp-stream-transport.c:
1393 * gst/rtsp-server/rtsp-stream.c:
1394 * gst/rtsp-server/rtsp-token.c:
1395 introspection: add (nullable) annotations to return values
1396 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1398 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
1400 * gst/rtsp-server/rtsp-client.c:
1401 * gst/rtsp-server/rtsp-stream.c:
1402 gi: improve annotations
1403 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
1405 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
1407 * gst/rtsp-server/rtsp-client.c:
1408 * gst/rtsp-server/rtsp-media-factory.c:
1409 * gst/rtsp-server/rtsp-media.c:
1410 * gst/rtsp-server/rtsp-server.c:
1411 signals: use generic marshal function
1412 Use the generic C marshal function.
1413 Use more explicit type instead of G_TYPE_POINTER
1415 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
1417 * gst/rtsp-server/rtsp-context.h:
1418 context: add type macro
1420 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
1422 * gst/rtsp-server/rtsp-client.c:
1423 * gst/rtsp-server/rtsp-sdp.c:
1424 * gst/rtsp-server/rtsp-sdp.h:
1425 sdp: hide key length defines
1426 They don't have a namespace.
1428 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1433 === release 1.3.3 ===
1435 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
1441 * gst-rtsp-server.doap:
1444 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1446 * gst/rtsp-server/rtsp-client.c:
1447 * gst/rtsp-server/rtsp-sdp.c:
1448 * gst/rtsp-server/rtsp-sdp.h:
1449 mikey: add different key length parameters
1450 Add encryption and authentication key length parameters to MIKEY. For
1451 the encoders, the key lengths are obtained from the cipher and auth
1452 algorithms set in the caps. For the decoders, they are obtained while
1453 parsing the key management from the client.
1454 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
1456 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
1458 * tests/check/gst/stream.c:
1459 stream tests: Make sure we get right multicast address from stream
1460 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
1462 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
1464 * gst/rtsp-server/rtsp-client.c:
1465 client: ref the context until rtsp watch is alive
1466 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
1468 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
1470 * gst/rtsp-server/rtsp-client.c:
1471 client: Destroy the rtsp watch after connection close
1473 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
1475 * gst/rtsp-server/rtsp-media.c:
1476 media: fix confusing comment
1478 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
1480 * gst/rtsp-server/rtsp-session.c:
1481 rtsp-session: Timeout in header.
1482 Adding the possbilty to always have timout in header.
1483 This is configurabe with setting "timeout-always-visible".
1484 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
1486 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
1491 === release 1.3.2 ===
1493 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
1500 * gst-rtsp-server.doap:
1503 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1506 Automatic update of common submodule
1507 From 211fa5f to 1f5d3c3
1509 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
1511 * gst/rtsp-server/rtsp-client.c:
1512 client: store TCP ports in transport
1513 Store the TCP ports in the transport when we are doing RTSP over TCP.
1514 This way, we can easily get to the ports from the transport.
1515 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
1517 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1519 * gst/rtsp-server/rtsp-stream.c:
1520 stream: add signals for new RTP/RTCP encoders
1521 New signals to allow the user to configure the dynamically created
1523 https://bugzilla.gnome.org/show_bug.cgi?id=730228
1525 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1527 * gst/rtsp-server/rtsp-media.c:
1528 * gst/rtsp-server/rtsp-media.h:
1529 media: Make suspend()/unsuspend() virtual
1530 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
1532 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1534 * gst/rtsp-server/rtsp-client.c:
1535 client: fix send-message signal marshaller
1536 Use generic marshalling for the send-message signal. It has
1537 two POINTER arguments, not just one.
1538 https://bugzilla.gnome.org/show_bug.cgi?id=729900
1540 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
1542 * tests/check/gst/media.c:
1543 tests: add and remove pads only once
1544 In this test we simulate a dynamic pad by watching the caps event.
1545 Because of renegotiation in the base payloader now, this caps is sent
1546 multiple times but we can only deal with 1 invocation, use a variable to
1547 only 'add and remove' the pad once.
1549 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1551 * tests/check/gst/rtspserver.c:
1552 tests: add unit test for correct handling of Require headers
1553 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1555 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1557 * gst/rtsp-server/rtsp-client.c:
1558 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
1559 Servers must handle Require headers and must report a failure
1560 if they don't handle any of the Required options, see RFC 2326,
1561 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
1562 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1564 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1569 === release 1.3.1 ===
1571 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1577 * gst-rtsp-server.doap:
1580 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
1583 Automatic update of common submodule
1584 From bcb1518 to 211fa5f
1586 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
1591 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1593 * tests/check/gst/sessionmedia.c:
1594 tests: fix memory leak in sessionmedia unit test
1596 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
1598 * gst/rtsp-server/rtsp-client.c:
1599 client: emit a signal before sending a message
1600 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
1602 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
1604 * gst/rtsp-server/rtsp-client.c:
1605 client: pass context to send_message
1606 Pass the current context to send_message, we will need it later.
1608 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
1610 * gst/rtsp-server/rtsp-client.c:
1611 client: fix typo in comment
1613 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
1615 * gst/rtsp-server/rtsp-media.c:
1616 media: Do not stop thread twice if default_prepare() fails
1618 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
1620 * gst/rtsp-server/rtsp-client.c:
1621 client: set the watch to flushing before going to NULL
1622 First set the watch to flushing so that we unblock any current and
1623 future attempt to send data on the watch, Then set the pipeline to
1625 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
1627 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
1629 * gst/rtsp-server/rtsp-session-pool.c:
1630 * tests/check/gst/sessionpool.c:
1631 rtsp-session-pool: Fixes annotation
1632 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
1633 in the sessionpool test.
1634 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
1636 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
1638 * gst/rtsp-server/rtsp-media.c:
1639 * gst/rtsp-server/rtsp-media.h:
1640 media: make media_prepare virtual
1641 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
1643 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1645 * gst/rtsp-server/rtsp-media.c:
1646 * tests/check/gst/media.c:
1647 media: stop the thread in more error cases
1649 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
1651 * gst/rtsp-server/rtsp-media.c:
1652 * tests/check/gst/media.c:
1653 media: allow NULL as the thread
1654 Use the default context whan passing a NULL thread.
1656 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1658 * gst/rtsp-server/rtsp-client.c:
1659 rtsp-client: indent cleanup
1660 Coverity was moaning about unreachable code, and I think it was just
1661 confused by { being before the label. We'll see if it pops up again.
1664 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
1666 * gst/rtsp-server/rtsp-client.c:
1667 * gst/rtsp-server/rtsp-media.c:
1668 client: Add drop-backlog property
1669 When we have too many messages queued for a client (currently hardcoded
1670 to 100) we overflow and drop the messages. Add a drop-backlog property
1671 to control this behaviour. Setting this property to FALSE will retry
1672 to send the messages to the client by waiting for more room in the
1674 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
1676 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
1678 * gst/rtsp-server/rtsp-client.c:
1679 client: support for POST before GET when setting up a tunnel
1681 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
1683 * gst/rtsp-server/rtsp-client.c:
1684 client: remove watch of the second client after http tunnel setup
1685 The second client will be freed after the HTTP tunnel has been set up.
1686 Make sure it's RTSP watch is never dispatched again.
1687 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
1689 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
1691 * gst/rtsp-server/rtsp-media.c:
1692 * tests/check/gst/media.c:
1693 media: Make media_prepare() fail if port allocation fails
1694 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
1696 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
1698 * tests/check/gst/media.c:
1699 media test: cleanup the thread pool in tests
1701 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
1703 * gst/rtsp-server/rtsp-media.c:
1704 * tests/check/gst/media.c:
1705 rtsp-media: Unblock blocked streams in unprepare
1706 The streams will be blocked when a live media is prepared.
1707 The streams should be unblocked in gst_rtsp_media_unprepare.
1708 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
1710 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
1712 * gst/rtsp-server/rtsp-media.c:
1713 media: release the state lock when going to NULL
1714 Set our state to UNPREPARING and release the state-lock before
1715 setting the pipeline to the NULL state. This way, any pad-added
1716 callback will be able to take the state-lock and check that we are now
1717 unpreparing instead of deadlocking.
1718 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
1720 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
1722 * gst/rtsp-server/rtsp-media.c:
1723 media: protect status with lock
1724 Make sure we only update the status with the lock.
1726 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
1728 * gst/rtsp-server/rtsp-client.c:
1729 * gst/rtsp-server/rtsp-sdp.c:
1730 rtsp: update for MIKEY API changes
1732 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
1734 * gst/rtsp-server/rtsp-client.c:
1735 client: parse the mikey response from the client
1736 Parse the mikey response from the client and update the policy for
1739 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
1741 * gst/rtsp-server/rtsp-stream.c:
1742 * gst/rtsp-server/rtsp-stream.h:
1743 stream: add method to set crypto info
1744 Make a method to configure the crypto information of a stream.
1745 Set udpsrc in READY instead of PAUSED so that we can configure caps
1748 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
1750 * gst/rtsp-server/rtsp-client.c:
1751 client: cleanup error paths
1753 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
1755 * gst/rtsp-server/rtsp-media.c:
1758 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
1760 * examples/test-video.c:
1761 test: enable SRTP only on RTSPS
1762 We only want to enable SRTP when doing rtsp over TLS so that we can
1763 exchange the keys in a secure way.
1765 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
1767 * examples/test-video.c:
1768 test: print an error on failure
1770 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
1773 * examples/test-video.c:
1774 * gst/rtsp-server/rtsp-sdp.c:
1775 * gst/rtsp-server/rtsp-stream.c:
1776 * tests/check/Makefile.am:
1777 stream: add SRTP support
1778 Install srtp encoder and decoder elements in rtpbin
1781 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1783 * tests/check/Makefile.am:
1784 * tests/check/gst/sessionpool.c:
1785 tests: Add unit tests for sessionpool
1786 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
1788 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1790 * tests/check/gst/threadpool.c:
1791 tests: Improve code coverage of rtsp-threadpool tests
1792 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
1794 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1796 * tests/check/gst/sessionmedia.c:
1797 tests: Improve code coverage for rtsp-session-media
1798 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
1800 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1802 gobject-introspection: Add annotations to support language bindings
1803 In addition a few cosmetic changes:
1804 * Adjust the order of arguments
1805 * Fix typo: occured -> occurred
1806 * Fix indentation after Return:-clauses
1807 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
1809 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1811 * gst/rtsp-server/rtsp-stream.c:
1812 rtsp-stream: Don't mix IPv4 and IPv6 addresses
1813 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
1815 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
1817 * gst/rtsp-server/rtsp-stream.c:
1818 stream: take caps after the session manager
1819 Take the caps for the SDP after they leave the rtpbin so that we can
1820 also get the properties added by rtpbin elements.
1822 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
1824 * gst/rtsp-server/rtsp-stream.c:
1825 stream: release lock while pushing out packets
1826 Keep a cache of the transports and use this to iterate the transport
1827 while pushing packets. This allows us to release the lock early.
1828 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
1830 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
1832 * gst/rtsp-server/rtsp-client.c:
1833 * gst/rtsp-server/rtsp-client.h:
1834 rtsp-client: vmethod for modifying tunnel GET response
1835 Add a vmethod tunnel_http_response where the response to the HTTP GET
1836 for tunneled connections can be modified.
1837 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
1839 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
1841 * gst/rtsp-server/rtsp-sdp.c:
1842 sdp: make 1 media line per profile
1843 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
1844 line in the SDP for each profile. The client is then supposed to pick
1845 one of the profiles in the SETUP request. Because the m= lines have the
1846 same pt, the client also knows that only 1 option is possible.
1848 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
1850 * gst/rtsp-server/rtsp-media-factory.c:
1851 * gst/rtsp-server/rtsp-media-factory.h:
1852 * gst/rtsp-server/rtsp-media.c:
1853 factory: add profile property and pass to media and streams
1855 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
1857 * examples/test-multicast.c:
1858 * gst/rtsp-server/rtsp-sdp.c:
1859 sdp: pass multicast connection for multicast-only stream
1860 Pass the multicast address of the stream in the connection info in the
1861 SDP so that clients try a multicast connection first.
1862 Only allow multicast connections in the test-multicast example. Also
1863 increase the TTL a little.
1865 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1868 .gitignore: Ignore gcov intermediate files
1869 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
1871 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
1873 * gst/rtsp-server/rtsp-stream.c:
1874 stream: release some locks in error cases
1876 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1878 docs: Enable and fix gtk-doc warnings
1879 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
1880 * addresspool/mediafactory: Add missing annotation colon
1881 * stream: Annotate return value
1882 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
1884 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1887 Automatic update of common submodule
1888 From fe1672e to bcb1518
1890 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
1893 Automatic update of common submodule
1894 From 1a07da9 to fe1672e
1896 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1898 * examples/Makefile.am:
1899 examples: use LDADD for libs instead of LDFLAGS
1901 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
1904 configure: make sure releases are in .doap file
1906 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
1908 * examples/test-cgroups.c:
1909 examples: test-cgroups: don't put code with side effects into g_assert()
1910 The g_assert() might get compiled out with the right
1911 compiler/preprocessor flags.
1913 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1915 * examples/.gitignore:
1916 examples: add cgroup test binary to .gitignore
1918 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
1920 * examples/test-cgroups.c:
1921 examples: fix cgroup test build
1922 Fixes build failure caused by compiler warning:
1923 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
1925 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
1928 .gitignore: ignore temp files created in the course of 'make check'
1930 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
1932 * gst/rtsp-server/rtsp-media.c:
1933 rtsp-media: don't loose frames handling new PLAY request
1934 If client supplied a range check if the range specifies the start point.
1935 If not, then do an accurate seek to the current position. If a start
1936 point was specified do do a key unit seek to make sure the streaming
1937 starts with decodeable frames.
1938 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
1940 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
1942 * gst/rtsp-server/rtsp-media.c:
1943 Revert "media: only flush when setting a new start position"
1944 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
1945 We need to do the flush in all cases, demuxer block currently for
1948 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
1950 * gst/rtsp-server/rtsp-media.c:
1951 media: only flush when setting a new start position
1952 Only flush the pipeline when we change the start position with
1954 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
1956 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
1958 * gst/rtsp-server/rtsp-stream.c:
1959 stream: set ttl-mc before adding the socket
1960 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
1961 never be set on socket.
1962 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
1964 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
1966 * gst/rtsp-server/rtsp-media.c:
1967 media: stop thread if media is already prepared
1968 in gst_rtsp_media_prepare() the thread is not used if media is already
1969 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
1971 https://bugzilla.gnome.org/show_bug.cgi?id=724182
1973 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
1976 build: Ship gst-rtsp-server.doap file
1978 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
1980 * tests/check/gst/rtspserver.c:
1981 tests: Fix another compiler warning with gcc
1983 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
1985 * gst/rtsp-server/rtsp-client.c:
1986 * gst/rtsp-server/rtsp-mount-points.c:
1987 * gst/rtsp-server/rtsp-stream.c:
1988 * tests/check/gst/client.c:
1989 rtsp-server: Fix lots of compiler warnings with clang
1991 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
1994 * gst-rtsp-server.doap:
1995 * tests/Makefile.am:
1996 configure: Synchronise with the configure scripts of the other modules
1998 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
2001 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
2003 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
2005 * gst/rtsp-server/rtsp-media.c:
2006 * gst/rtsp-server/rtsp-stream.c:
2007 Revert "rtsp-server: support build against last stable release"
2008 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
2009 Let us require 1.2.3 now, which is going to be released in a few
2012 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
2014 * gst/rtsp-server/rtsp-session-media.c:
2015 * gst/rtsp-server/rtsp-stream-transport.c:
2016 session: improve RTP-Info
2017 Ignore streams that can't generate RTP-Info instead of failing.
2018 Don't return the empty string when all streams are unconfigured but
2019 return NULL so that we don't generate and empty RTP-Info header.
2020 Improve docs a little.
2022 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
2024 * gst/rtsp-server/rtsp-session-media.c:
2025 Don't free rtpinfo GString when it is NULL
2026 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2028 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
2030 * gst/rtsp-server/rtsp-media.c:
2031 media: only set keyframe flag when modifying start
2032 Only set the keyframe flag when we modify the start position. The
2033 keyframe flag should probably be ignored when no change is requested but
2034 until we can claim this is all documented properly and all demuxer
2035 implement this, avoid setting the flag.
2036 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
2038 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
2040 * gst/rtsp-server/rtsp-thread-pool.c:
2041 thread-pool: Unref source after mainloop has quit to avoid races in GLib
2042 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
2044 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
2046 * gst/rtsp-server/rtsp-stream.c:
2047 stream: handle NULL seqnum and rtptime arguments
2049 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
2051 * gst/rtsp-server/rtsp-thread-pool.c:
2052 * tests/check/gst/threadpool.c:
2053 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
2054 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2056 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
2058 * gst/rtsp-server/rtsp-stream.c:
2059 stream: add fallback for missing stats property
2060 Use a fallback when the payloader does not have a stats property
2061 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2063 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
2066 Automatic update of common submodule
2067 From f7bc1c3 to 1a07da9
2069 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
2071 * gst/rtsp-server/rtsp-stream.c:
2072 stream: don't leak stats structure
2073 Don't leak the stats structure and deal with NULL stats.
2075 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
2077 * gst/rtsp-server/rtsp-stream.c:
2078 stream: Get rtpinfo properties atomically from payloader
2079 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2081 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
2083 * gst/rtsp-server/rtsp-media.c:
2084 media: refactor state change functions and signals
2085 Make functions to set the target state and the pipeline state and emit
2086 the signals from those functions.
2088 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
2090 * gst/rtsp-server/rtsp-media.c:
2091 * gst/rtsp-server/rtsp-media.h:
2092 media: add signal to notify of pending state changes
2094 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
2096 * gst/rtsp-server/rtsp-media.c:
2097 * gst/rtsp-server/rtsp-stream.c:
2098 rtsp-server: support build against last stable release
2099 Until 1.2.3 is out with the new get_type function and we
2102 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
2104 * gst/rtsp-server/rtsp-stream.c:
2105 stream: fix compilation
2107 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
2109 * gst/rtsp-server/rtsp-media.c:
2110 * gst/rtsp-server/rtsp-media.h:
2111 * gst/rtsp-server/rtsp-stream.c:
2112 * gst/rtsp-server/rtsp-stream.h:
2113 stream: add property to configure profiles
2115 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
2117 * gst/rtsp-server/rtsp-client.c:
2118 client: let stream check supported transport
2119 Delegate the check if a transport is allowed to the stream.
2120 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2122 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
2124 * gst/rtsp-server/rtsp-stream.c:
2125 * gst/rtsp-server/rtsp-stream.h:
2126 stream: add method to check supported transport
2127 Add a method to check if a transport is supported
2129 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
2132 configure.ac: Only check for gstreamer-check, not check
2133 We include check in gstreamer-check since quite some time now.
2135 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
2137 * gst/rtsp-server/rtsp-session-media.c:
2138 * gst/rtsp-server/rtsp-stream-transport.c:
2139 * gst/rtsp-server/rtsp-stream.c:
2140 * gst/rtsp-server/rtsp-stream.h:
2141 stream: return clock-rate from get_rtpinfo
2142 And use it to correct the rtptime to the requested start-time.
2143 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2145 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
2147 * gst/rtsp-server/rtsp-session-media.c:
2148 * gst/rtsp-server/rtsp-stream-transport.c:
2149 * gst/rtsp-server/rtsp-stream-transport.h:
2150 session-media: calculate start-time
2152 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
2154 * gst/rtsp-server/rtsp-stream-transport.c:
2155 * gst/rtsp-server/rtsp-stream.c:
2156 * gst/rtsp-server/rtsp-stream.h:
2157 stream: also return the running-time
2158 Return the running-time in the rtpinfo as well.
2160 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
2162 * gst/rtsp-server/rtsp-client.c:
2163 * gst/rtsp-server/rtsp-session-media.c:
2164 * gst/rtsp-server/rtsp-session-media.h:
2165 * gst/rtsp-server/rtsp-stream-transport.c:
2166 * gst/rtsp-server/rtsp-stream-transport.h:
2167 session-media: let the session-media make the RTPInfo
2168 Add method to create the RTPInfo for a stream-transport.
2169 Add method to create the RTPInfo for all stream-transports in a
2171 Use the session-media RTPInfo code in client. This allows us to refactor
2172 another method to link the TCP callbacks.
2174 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2176 mount-points: sort sequence before g_sequence_lookup
2177 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
2178 sort sequence if dirty, otherwise lookup will fail.
2179 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2181 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
2184 configure: rename package from gst-rtsp to gst-rtsp-server
2185 To match git module name and avoid confusion with the
2186 rtsp lib in gst-plugins-base and rtsp plugin in -good.
2188 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
2191 configure: bump core/base/good requirement to 1.2.0
2192 Bump to released stable version and make implicit
2193 requirements explicit.
2195 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
2200 Fix broken gettext setup which is not used anyway
2202 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
2205 Automatic update of common submodule
2206 From dbedaa0 to d48bed3
2208 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
2210 * gst/rtsp-server/rtsp-client.c:
2211 * gst/rtsp-server/rtsp-media.c:
2212 * gst/rtsp-server/rtsp-media.h:
2213 media: add setup_sdp vmethod
2214 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
2215 gst_rtsp_media_setup_sdp.
2216 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2218 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
2220 * gst/rtsp-server/rtsp-stream.c:
2221 rtsp-stream: Check return value of sscanf
2222 streamid is only valid if sscanf matched something.
2224 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
2226 * gst/rtsp-server/rtsp-client.c:
2227 rtsp-client: Fix iteration
2228 Wouldn't even enter the code block otherwise (i++ was used as the check
2229 and not the postfix).
2231 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
2233 * gst/rtsp-server/rtsp-client.c:
2234 * gst/rtsp-server/rtsp-client.h:
2235 client: add vmethod to configure media and streams
2236 Implement a vmethod that can be used to configure the media and the
2237 streams based on the current context. Handle the blocksize handling in
2238 the default handler.
2239 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
2241 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2244 Make git ignore more unit test binaries
2246 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
2248 * gst/rtsp-server/rtsp-address-pool.h:
2249 * gst/rtsp-server/rtsp-auth.h:
2250 * gst/rtsp-server/rtsp-client.h:
2251 * gst/rtsp-server/rtsp-context.h:
2252 * gst/rtsp-server/rtsp-media-factory-uri.h:
2253 * gst/rtsp-server/rtsp-media-factory.h:
2254 * gst/rtsp-server/rtsp-media.h:
2255 * gst/rtsp-server/rtsp-mount-points.h:
2256 * gst/rtsp-server/rtsp-server.h:
2257 * gst/rtsp-server/rtsp-session-media.h:
2258 * gst/rtsp-server/rtsp-session-pool.h:
2259 * gst/rtsp-server/rtsp-session.h:
2260 * gst/rtsp-server/rtsp-stream-transport.h:
2261 * gst/rtsp-server/rtsp-stream.h:
2262 * gst/rtsp-server/rtsp-thread-pool.h:
2263 * gst/rtsp-server/rtsp-token.h:
2264 rtsp-server: add padding to many public structures
2265 Not mini objects though, since they are not subclassable
2266 anyway, nor kept on the stack or inlined in a structure.
2268 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
2270 media: add new create_rtpbin vmethod
2271 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
2272 https://bugzilla.gnome.org/show_bug.cgi?id=719734
2274 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
2276 * tests/check/gst/media.c:
2277 tests: fix memory leak, free test's thread pool
2278 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
2280 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
2282 * gst/rtsp-server/rtsp-stream-transport.c:
2283 stream-transport: free url in finalize
2285 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
2287 * gst/rtsp-server/rtsp-media.c:
2288 media: also do state change in suspended state
2290 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
2292 * gst/rtsp-server/rtsp-client.c:
2293 * gst/rtsp-server/rtsp-media.c:
2294 media: also handle prepare and range in suspended state
2295 When we are suspended, we are already prepared.
2296 We can get the range in the suspended state.
2298 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
2300 * tests/check/Makefile.am:
2301 * tests/check/gst/sessionmedia.c:
2302 check: add test for uri in setup
2303 Added unit tests for the new functionality in GstRTSPStreamTransport.
2304 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2306 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
2308 * gst/rtsp-server/rtsp-client.c:
2309 client: store setup uri and use in PLAY response
2310 Store the uri used when doing the setup and use that in the PLAY
2312 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2314 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
2316 * gst/rtsp-server/rtsp-stream-transport.c:
2317 * gst/rtsp-server/rtsp-stream-transport.h:
2318 stream-transport: add method to get/set url
2320 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
2322 * gst/rtsp-server/rtsp-client.c:
2323 client: suspend after SDP and unsuspend before PLAYING
2324 Based on patches by Ognyan Tonchev <ognyan@axis.com>
2325 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
2327 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
2329 * gst/rtsp-server/rtsp-media-factory.c:
2330 * gst/rtsp-server/rtsp-media-factory.h:
2331 * gst/rtsp-server/rtsp-media.c:
2332 * gst/rtsp-server/rtsp-media.h:
2333 * gst/rtsp-server/rtsp-session-media.c:
2334 * gst/rtsp-server/rtsp-session.c:
2335 * tests/check/gst/media.c:
2336 * tests/check/gst/mediafactory.c:
2337 media: add suspend modes
2338 Add support for different suspend modes. The stream is suspended right after
2339 producing the SDP and after PAUSE. Different suspend modes are available that
2340 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
2341 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
2342 state and RESET will bring the pipeline to the NULL state.
2343 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
2344 this means that the pipeline needs to be prerolled again.
2345 Base on patches by Ognyan Tonchev <ognyan@axis.com>
2346 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2348 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
2350 * gst/rtsp-server/rtsp-media.c:
2351 media: start live streams in blocked state
2352 Start live streams in the blocked state and make them preroll using the
2353 messages. This ensure that no data is played by the sink until we explicitly
2354 unblock the stream right before going to PLAYING.
2355 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2357 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
2359 * gst/rtsp-server/rtsp-media.c:
2360 media: refactor starting and waiting for preroll
2361 Based on patches from Ognyan Tonchev <ognyan@axis.com>
2362 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2364 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
2366 * gst/rtsp-server/rtsp-stream.c:
2367 * gst/rtsp-server/rtsp-stream.h:
2368 stream: add API to block streams
2369 Add an API to block on the streams and make it post a message.
2370 Based on patch by Ognyan Tonchev <ognyan@axis.com>
2371 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2373 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
2375 * docs/libs/Makefile.am:
2376 docs: Specify the override file
2377 Even if it's empty (for now) it avoids make distcheck complaining
2379 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
2381 * gst/rtsp-server/rtsp-media.c:
2382 media: move default implementations to where they are used
2384 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
2386 * gst/rtsp-server/rtsp-media.c:
2387 media: take the right lock in gst_rtsp_media_set_pipeline_state()
2388 We need to take the state_lock when calling this method.
2390 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
2392 * gst/rtsp-server/rtsp-media.c:
2393 media: handle add-added on non-bins too
2394 Handle dynamic payloaders that are not bins, as used in the unit-test.
2396 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2398 * gst/rtsp-server/rtsp-media-factory.c:
2399 * gst/rtsp-server/rtsp-media-factory.h:
2400 * gst/rtsp-server/rtsp-media.c:
2401 rtsp-media/-factory: Fix request pad name comments
2402 These must be escaped for gtk-doc to parse the comments without warnings.
2404 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2406 rtsp-media: remove transports if media is in error status
2407 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
2408 trying to change to GST_STATE_NULL and media is in error status, we
2409 remove all transports.
2410 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
2412 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
2414 * gst/rtsp-server/rtsp-media.c:
2415 rtsp-media: use element metadata to find payloader
2416 Use the element metadata to find the payloader instead of checking
2418 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
2420 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2422 rtsp-stream: add getter for payload type
2423 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
2424 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
2425 element and create the stream with this one instead of the dynpay%d
2427 https://bugzilla.gnome.org/show_bug.cgi?id=712396
2429 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2431 * gst/rtsp-server/rtsp-client.c:
2432 * gst/rtsp-server/rtsp-context.h:
2433 * gst/rtsp-server/rtsp-media.c:
2434 * gst/rtsp-server/rtsp-mount-points.c:
2435 * gst/rtsp-server/rtsp-server.c:
2436 * gst/rtsp-server/rtsp-token.c:
2437 rtsp-*: Refer to NULL as a constant in comments
2439 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2441 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2443 rtsp-*: Fix type name typos in comments
2444 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
2445 * rtsp-auth: Refer to part of constant name as text
2446 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
2447 * rtsp-session-media: Fix GstRTSPSessionMedia typo
2448 * rtsp-stream: Fix typo when refering to GstBin
2449 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2451 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2454 * docs/libs/gst-rtsp-server-docs.sgml:
2455 * docs/libs/gst-rtsp-server-sections.txt:
2456 docs: Improve documentation
2457 * Include annotation-glossary to quiet gtk-doc
2458 * Rename remaining ClientState -> Context
2459 * Rename object hierarchy file
2460 * Remove stale chapter references
2461 * Add missing function and object references
2462 * Include missing GstRTSPAddressPoolResult
2463 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2465 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2467 * gst/rtsp-server/rtsp-client.c:
2468 * gst/rtsp-server/rtsp-server.c:
2469 * gst/rtsp-server/rtsp-session-pool.c:
2470 * gst/rtsp-server/rtsp-session.c:
2471 * gst/rtsp-server/rtsp-stream.c:
2472 rtsp-server: sprinkle some allow-none annotations for g-i
2474 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
2476 * gst/rtsp-server/rtsp-stream.c:
2477 * gst/rtsp-server/rtsp-stream.h:
2478 stream: add method to filter transports
2479 Add a method to safely iterate and collect the stream transports
2480 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2482 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
2484 * gst/rtsp-server/rtsp-client.c:
2485 * gst/rtsp-server/rtsp-server.c:
2486 * gst/rtsp-server/rtsp-session-pool.c:
2487 * gst/rtsp-server/rtsp-session.c:
2488 rtsp: allow NULL func in filters
2489 Passing a null function make the filters return a list of
2492 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
2494 * gst/rtsp-server/rtsp-address-pool.c:
2495 * tests/check/gst/addresspool.c:
2496 address-pool: fix address increment
2497 Use a guint instead of guint8 to increment the address. It's still not
2498 completely correct because a guint might not be able to hold the complete
2499 address range, but that's an enhacement for later.
2500 Add unit test to test improved behaviour.
2501 https://bugzilla.gnome.org/show_bug.cgi?id=708237
2503 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
2505 * gst/rtsp-server/rtsp-client.c:
2506 * tests/check/gst/client.c:
2507 client: allow absolute path in requests
2508 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
2510 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
2512 * gst/rtsp-server/rtsp-client.c:
2513 * gst/rtsp-server/rtsp-client.h:
2514 client: make make_path_from_uri a vmethod
2516 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2518 * docs/libs/gst-rtsp-server-sections.txt:
2519 * gst/rtsp-server/rtsp-stream.c:
2520 * gst/rtsp-server/rtsp-stream.h:
2521 * tests/check/Makefile.am:
2522 * tests/check/gst/stream.c:
2523 stream: Add functions to get rtp and rtcp sockets
2524 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2526 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2528 * gst/rtsp-server/rtsp-context.c:
2529 * gst/rtsp-server/rtsp-context.h:
2530 context: defing a GType for the context
2531 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
2533 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
2535 * gst/rtsp-server/Makefile.am:
2536 * gst/rtsp-server/rtsp-auth.c:
2537 * gst/rtsp-server/rtsp-context.c:
2538 * gst/rtsp-server/rtsp-media.c:
2539 * gst/rtsp-server/rtsp-mount-points.c:
2540 * gst/rtsp-server/rtsp-server.h:
2541 * gst/rtsp-server/rtsp-session-media.c:
2542 * gst/rtsp-server/rtsp-session.c:
2543 * gst/rtsp-server/rtsp-stream.c:
2544 Fixed several GIR warnings
2546 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
2548 * gst/rtsp-server/rtsp-auth.c:
2551 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2553 * tests/check/Makefile.am:
2554 * tests/check/gst/token.c:
2555 tests: Add unit tests for token
2556 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2558 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2560 * gst/rtsp-server/rtsp-token.c:
2561 token: Validate args for gst_rtsp_token_is_allowed
2562 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
2564 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2566 * gst/rtsp-server/rtsp-token.c:
2567 token: Fix bug when creating empty token
2568 We always want to have a valid GstStructure in the token.
2569 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2571 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2573 * gst/rtsp-server/rtsp-thread-pool.c:
2574 thread-pool: avoid race in shutdown
2575 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
2576 don't actually stop the mainloop ever. Solve this race by adding an idle source
2577 to the mainloop that calls the _quit. This way we immediately exit the mainloop
2578 if quit was called before we started it.
2580 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2582 * tests/check/Makefile.am:
2583 * tests/check/gst/permissions.c:
2584 tests: Add unit tests for permissions
2585 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
2587 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2589 * tests/check/gst/mediafactory.c:
2590 tests: Test mediafactory permissions
2591 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2593 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2595 * gst/rtsp-server/rtsp-permissions.c:
2596 permissions: Fix refcounting when adding/removing roles
2597 Previously a role that was removed was unreffed twice, and when
2598 replacing an existing role the replaced role was freed while still being
2599 referenced. Both bugs are now fixed.
2600 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2602 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2604 * tests/check/gst/media.c:
2605 * tests/check/gst/mediafactory.c:
2606 * tests/check/gst/rtspserver.c:
2607 tests: Check gst_rtsp_url_parse return value
2608 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2610 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
2613 Automatic update of common submodule
2614 From 865aa20 to dbedaa0
2616 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
2618 * gst/rtsp-server/rtsp-server.c:
2619 rtsp-server: Fix socket leak
2620 https://bugzilla.gnome.org/show_bug.cgi?id=710088
2622 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
2624 * gst/rtsp-server/rtsp-session-pool.c:
2625 rtsp-session-pool: Make sure session IDs are properly URI-escaped
2626 https://bugzilla.gnome.org/show_bug.cgi?id=643812
2628 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2630 * examples/.gitignore:
2631 * examples/test-video.c:
2632 examples: fix compilation when WITH_AUTH is defined
2633 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2635 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
2638 gitignore: Add new test binary
2640 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
2642 * tests/check/Makefile.am:
2643 * tests/check/gst/threadpool.c:
2644 thread-pool: Add unit test for the thread pools
2645 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2647 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2649 * gst/rtsp-server/rtsp-thread-pool.c:
2650 thread-pool: Fix thread leak when reusing threads
2651 https://bugzilla.gnome.org/show_bug.cgi?id=709730
2653 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
2655 * gst/rtsp-server/rtsp-server.c:
2656 * tests/check/gst/rtspserver.c:
2657 tests: fixed racy behavior in rtspserver tests
2658 https://bugzilla.gnome.org/show_bug.cgi?id=710078
2660 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2662 * tests/check/gst/addresspool.c:
2663 tests: Improve address pool unit tests
2664 Add a range with mixed IPV4 and IPV6 addresses to pool.
2665 Get an IPV4 address from an IPV6-only pool.
2666 Get an IPV6 address from an IPV4-only pool.
2667 Reserve a IPV6 address from an IPV4-only pool.
2668 Check for unicast addresses in multicast-only pool.
2669 Check for unicast addresses in uni-/multicast-mixed pool.
2670 https://bugzilla.gnome.org/show_bug.cgi?id=710128
2672 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2674 * gst/rtsp-server/rtsp-client.c:
2675 client: append query string in PAUSE/PLAY/TEARDOWN as well
2677 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
2679 * gst/rtsp-server/rtsp-client.c:
2680 client: Add query to control path
2681 If the SETUP url contains a query it must be appended to the control
2682 path so that it matches any already created stream in the media. The
2683 query will also be appended to the session media path.
2685 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2687 * gst/rtsp-server/rtsp-media.c:
2688 rtsp-media: remove old line
2690 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
2692 * gst/rtsp-server/rtsp-stream.c:
2693 stream: Correct control comparison
2694 https://bugzilla.gnome.org/show_bug.cgi?id=709176
2696 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2698 * gst/rtsp-server/rtsp-media.c:
2699 media: Check dynamically if the pipeline supports seeking
2700 We should not depend on whether or not the pipeline state change
2701 returned NO_PREROLL or not. A media could dynamically change its
2702 element and switch from seekable to non seekable so it's best to test
2703 the seekable nature of the pipeline dynamically when we try to do a seek.
2705 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2707 * gst/rtsp-server/rtsp-media.c:
2708 media: Return FALSE if seeking is not supported
2710 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2712 * gst/rtsp-server/rtsp-media.c:
2713 rtsp-media: don't seek accurate by default
2714 Accurate seeking is perhaps a little overkill in the most common situation and
2715 causes some formats (mp3) over slow media to seek extremely slowly.
2717 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
2719 * tests/check/gst/rtspserver.c:
2720 tests: fix unit test
2721 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
2723 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
2725 * gst/rtsp-server/rtsp-client.c:
2726 client: Reply 400 if media cannot be constructed
2727 Reply 400 Bad Request instead of 503 Service Unavailable if media
2728 cannot be constructed in SETUP.
2729 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
2731 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
2733 * gst/rtsp-server/rtsp-client.c:
2734 client: Send setup reply once only
2735 If find_media() failed in handle_setup_request() two replies was sent.
2736 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
2738 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
2741 Automatic update of common submodule
2742 From 6b03ba7 to 865aa20
2744 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
2746 * gst/rtsp-server/rtsp-server.c:
2747 server: Emit client-connected signal earlier
2748 Emit client-connected before the client ref is given to a GSource,
2749 otherwise client-connected can be emitted after the client object has
2752 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
2754 * gst/rtsp-server/rtsp-address-pool.c:
2755 * gst/rtsp-server/rtsp-address-pool.h:
2756 * gst/rtsp-server/rtsp-stream.c:
2757 * tests/check/gst/addresspool.c:
2758 addresspool: return reason of failure
2759 Let gst_rtsp_address_pool_reserve_address() return the reason why
2760 the address could not be reserved.
2761 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
2763 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
2766 autogen.sh: Sync behaviour with other GStreamer modules
2767 Allows building from outside of tree amongst other things
2769 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
2772 Automatic update of common submodule
2773 From b613661 to 6b03ba7
2775 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
2778 Automatic update of common submodule
2779 From 74a6857 to b613661
2781 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
2784 Automatic update of common submodule
2785 From 01a7a46 to 74a6857
2787 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
2789 * gst/rtsp-server/rtsp-client.c:
2790 client: Do not read beyond end of path string
2791 If the setup was done without a control url, make sure we don't try to read the
2792 non-existing control string and crash.
2794 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2796 * gst/rtsp-server/rtsp-client.c:
2797 client: Fix RTPInfo header
2798 Refactor the method to make the content_base.
2799 Use the content-base and the control url to construct the RTPInfo
2802 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2804 * gst/rtsp-server/rtsp-client.c:
2805 client: map url to path only in describe
2806 Only map the request url to a path in the DESCRIBE method. The SDP then
2807 contains the base and control urls that should be used to SETUP/PAUSE/
2808 PLAY/TEARDOWN the media.
2810 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2812 * gst/rtsp-server/rtsp-client.c:
2813 Revert "client: map URL to path in requests"
2814 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
2815 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
2816 contains the base and control urls which are used in the SETUP, PLAY,
2817 PAUSE and TEARDOWN requests.
2819 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2821 * gst/rtsp-server/rtsp-client.c:
2822 client: map URL to path in requests
2824 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2826 * gst/rtsp-server/rtsp-client.c:
2827 * gst/rtsp-server/rtsp-mount-points.c:
2828 * gst/rtsp-server/rtsp-mount-points.h:
2829 mount-points: make vmethod to make path from uri
2830 Make a vmethod to transform an url into a path. The path is then used to lookup
2831 the factory. This makes it possible to also use other bits of the url, such as
2832 the query parameters, to locate the factory.
2834 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
2836 * gst/rtsp-server/rtsp-thread-pool.c:
2837 * gst/rtsp-server/rtsp-thread-pool.h:
2838 thread-pool: Add cleanup to wait for the threadpool to finish
2839 Also fix race condition if two threads are asking for the first
2840 thread from the thread pool at once. This would case two internal
2841 GThreadPools to be created.
2842 https://bugzilla.gnome.org/show_bug.cgi?id=707753
2844 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
2846 * gst/rtsp-server/rtsp-client.c:
2847 * tests/check/gst/client.c:
2848 client: free threadpool
2849 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2851 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
2853 * tests/check/gst/mountpoints.c:
2854 mountpoints tests: unref matched factories
2855 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2857 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
2859 * tests/check/gst/media.c:
2860 media tests: unref thread pool and caps
2861 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2863 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
2865 * gst/rtsp-server/rtsp-auth.c:
2866 * gst/rtsp-server/rtsp-media-factory.c:
2867 * gst/rtsp-server/rtsp-media.c:
2868 auth, media, media-factory: unref permissions
2869 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2871 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2873 * examples/Makefile.am:
2874 Makefile: add rule for appsrc example
2876 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2878 * examples/test-appsrc.c:
2879 tests: add appsrc example
2880 Add an example on how to use appsrc to feed the server pipeline with data.
2882 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
2884 * gst/rtsp-server/rtsp-client.c:
2885 rtsp-client: remove query part from content-base string
2886 Make sure that after the control url has been resolved, it's
2887 not a part of the query-string.
2888 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
2890 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2892 * gst/rtsp-server/rtsp-client.c:
2893 client: don't check url in response
2894 There is no url or method in the response to check
2896 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2898 * gst/rtsp-server/rtsp-client.c:
2899 * gst/rtsp-server/rtsp-client.h:
2900 Add handle-response signal for when we receive a GET_PARAMETER response
2902 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2904 * gst/rtsp-server/rtsp-server.c:
2905 Fix gst_rtsp_server_client_filter, using wrong variable type
2907 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
2909 * gst/rtsp-server/rtsp-media-factory-uri.c:
2910 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
2911 For AAC we need to check for framed=true instead of parsed=true.
2912 https://bugzilla.gnome.org/show_bug.cgi?id=701384
2914 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2916 * gst/rtsp-server/rtsp-stream.c:
2917 stream: optimize pipeline for protocols
2918 When TCP is not an allowed protocol for the stream, avoid creating the
2919 appsrc/appsink/queue and tee elements.
2921 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2923 * gst/rtsp-server/rtsp-media.c:
2924 media: set protocols on streams
2926 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2928 * gst/rtsp-server/rtsp-client.c:
2929 client: use protocols supported by stream
2931 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2933 * gst/rtsp-server/rtsp-media-factory.c:
2934 * gst/rtsp-server/rtsp-media.c:
2935 * gst/rtsp-server/rtsp-stream.c:
2936 media-factory: allow all protocols
2938 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2940 * gst/rtsp-server/rtsp-media.c:
2941 media: configure protocols in new streams
2943 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2945 * gst/rtsp-server/rtsp-stream.c:
2946 * gst/rtsp-server/rtsp-stream.h:
2947 stream: add protocols property
2949 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2951 * gst/rtsp-server/rtsp-media.c:
2952 rtsp-media: send state in "new-state" signal
2953 https://bugzilla.gnome.org/show_bug.cgi?id=705110
2955 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
2958 build: add subdir-objects to AM_INIT_AUTOMAKE
2959 Fixes warnings with automake 1.14
2960 https://bugzilla.gnome.org/show_bug.cgi?id=705350
2962 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2964 * docs/libs/gst-rtsp-server-sections.txt:
2965 * gst/rtsp-server/rtsp-client.c:
2966 * gst/rtsp-server/rtsp-server.c:
2967 * gst/rtsp-server/rtsp-server.h:
2968 server: add method to iterate clients of server
2970 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2972 * gst/rtsp-server/rtsp-media.c:
2973 * gst/rtsp-server/rtsp-media.h:
2974 Add vmethod for rtsp-media subclass to access rtpbin
2976 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2978 * gst/rtsp-server/rtsp-client.h:
2979 small documentation fix
2981 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2983 * gst/rtsp-server/rtsp-client.c:
2984 Do not take range header if range is invalid
2986 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2988 * docs/libs/gst-rtsp-server-sections.txt:
2989 * gst/rtsp-server/rtsp-media.c:
2990 media: add docs for new method
2992 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2994 * gst/rtsp-server/rtsp-media.c:
2995 * gst/rtsp-server/rtsp-media.h:
2996 Add API to rtsp-media set the pipeline's state
2998 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3000 * gst/rtsp-server/rtsp-media.c:
3001 Update current position/duration when gst_rtsp_media_get_range_string is called
3003 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3005 * examples/test-cgroups.c:
3006 tests: add some more docs
3008 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3010 * examples/test-cgroups.c:
3011 * gst/rtsp-server/Makefile.am:
3012 * gst/rtsp-server/rtsp-auth.c:
3013 * gst/rtsp-server/rtsp-auth.h:
3014 * gst/rtsp-server/rtsp-client.c:
3015 * gst/rtsp-server/rtsp-client.h:
3016 * gst/rtsp-server/rtsp-context.c:
3017 * gst/rtsp-server/rtsp-context.h:
3018 * gst/rtsp-server/rtsp-params.c:
3019 * gst/rtsp-server/rtsp-params.h:
3020 * gst/rtsp-server/rtsp-server.c:
3021 * gst/rtsp-server/rtsp-thread-pool.c:
3022 * gst/rtsp-server/rtsp-thread-pool.h:
3023 * tests/check/gst/client.c:
3024 ClientState -> Context
3025 Rename the clientstate to context and put the code in a separate file.
3027 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3029 * examples/test-auth.c:
3030 * gst/rtsp-server/rtsp-auth.c:
3031 * gst/rtsp-server/rtsp-auth.h:
3032 auth: add support for default token
3033 The default token is used when the user is not authenticated and can be used to
3034 give minimal permissions.
3036 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3038 * examples/test-auth.c:
3039 * gst/rtsp-server/rtsp-auth.c:
3040 auth: use defines when possible
3042 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3044 * gst/rtsp-server/rtsp-address-pool.c:
3045 address-pool: improve docs
3047 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3049 * gst/rtsp-server/rtsp-permissions.c:
3050 permissions: add the role to the copy
3052 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
3054 * gst/rtsp-server/rtsp-permissions.c:
3055 permissions: Also copy the roles
3057 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
3059 * gst/rtsp-server/rtsp-permissions.c:
3060 permissions: Make it build
3062 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3064 * gst/rtsp-server/rtsp-address-pool.h:
3067 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3069 * docs/libs/gst-rtsp-server-sections.txt:
3070 * gst/rtsp-server/rtsp-auth.c:
3071 * gst/rtsp-server/rtsp-auth.h:
3072 * gst/rtsp-server/rtsp-media.c:
3073 * gst/rtsp-server/rtsp-session-media.c:
3074 * gst/rtsp-server/rtsp-stream-transport.c:
3075 * gst/rtsp-server/rtsp-stream-transport.h:
3076 * gst/rtsp-server/rtsp-stream.c:
3077 * tests/check/gst/client.c:
3080 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3082 * docs/libs/gst-rtsp-server-sections.txt:
3083 * gst/rtsp-server/rtsp-address-pool.c:
3084 * gst/rtsp-server/rtsp-address-pool.h:
3085 * tests/check/gst/addresspool.c:
3086 * tests/check/gst/rtspserver.c:
3087 address-pool: cleanups
3088 Remove redundant method, improve docs.
3090 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3092 * docs/libs/gst-rtsp-server-sections.txt:
3093 * gst/rtsp-server/rtsp-auth.h:
3094 * gst/rtsp-server/rtsp-permissions.c:
3095 * gst/rtsp-server/rtsp-permissions.h:
3096 * gst/rtsp-server/rtsp-token.c:
3099 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3101 * gst/rtsp-server/rtsp-permissions.c:
3102 permissions: implement _remove_role
3104 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3106 * gst/rtsp-server/rtsp-permissions.c:
3107 permissions: update docs
3109 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3111 * tests/check/gst/client.c:
3112 tests: simplify tests
3113 Client settings are now disabled by default so we don't need an auth
3114 module to disable them.
3116 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3118 * gst/rtsp-server/rtsp-auth.c:
3119 auth: add default authorizations
3120 When no auth module is specified, use our table of defaults to look up the
3121 default value of the check instead of always allowing everything. This was
3122 we can disallow client settings by default.
3124 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3127 README: update readme
3129 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3131 * gst/rtsp-server/rtsp-thread-pool.c:
3132 * gst/rtsp-server/rtsp-thread-pool.h:
3133 thread-pool: add more docs
3135 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3137 * gst/rtsp-server/rtsp-thread-pool.c:
3138 * gst/rtsp-server/rtsp-thread-pool.h:
3139 thread-pool: fix race in thread reuse
3140 If we try to reuse a thread right after we made it stop, we end up using a
3141 stopped thread. Catch this case and only reuse threads that are not stopping.
3143 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3145 * gst/rtsp-server/rtsp-server.c:
3146 server: add small debug
3148 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3150 * tests/check/gst/client.c:
3152 Add some permissions to media so we can use the auth and enable
3155 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3157 * gst/rtsp-server/rtsp-client.c:
3158 client: support pushed context in handle_request
3159 If we already have a pushed state, reuse it and add our own things. This makes
3160 it easier to write tests.
3162 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3164 * gst/rtsp-server/rtsp-auth.c:
3165 auth: don't auth on methods
3166 Don't authorize on methods anymore but on the resources that we
3167 try to access, this is more flexible.
3168 Move the authorization checks to where they are needed and let the
3169 check return the response on error.
3171 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3173 * gst/rtsp-server/rtsp-mount-points.c:
3174 mount-points: add some debug
3176 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3178 * tests/check/gst/client.c:
3179 tests: almost fix test
3181 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3183 * gst/rtsp-server/rtsp-auth.c:
3184 * gst/rtsp-server/rtsp-auth.h:
3185 * gst/rtsp-server/rtsp-client.c:
3186 * gst/rtsp-server/rtsp-client.h:
3187 * gst/rtsp-server/rtsp-server.c:
3188 * gst/rtsp-server/rtsp-server.h:
3189 auth: let the auth module check client_settings
3190 Let the auth module decide if client settings are allowed for the
3193 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3195 * gst/rtsp-server/rtsp-token.c:
3196 * gst/rtsp-server/rtsp-token.h:
3197 token: add method to check boolean permission
3199 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3201 * examples/test-auth.c:
3202 * examples/test-cgroups.c:
3203 * gst/rtsp-server/rtsp-token.c:
3204 * gst/rtsp-server/rtsp-token.h:
3205 token: simplify token constructor
3206 Use variable arguments to make easier API.
3208 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3210 * examples/test-auth.c:
3211 * examples/test-cgroups.c:
3212 * gst/rtsp-server/rtsp-media-factory.c:
3213 * gst/rtsp-server/rtsp-media-factory.h:
3214 media-factory: add convenience API for factory
3216 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3218 * examples/test-auth.c:
3219 * examples/test-cgroups.c:
3220 * gst/rtsp-server/rtsp-permissions.c:
3221 * gst/rtsp-server/rtsp-permissions.h:
3222 permissions: simplify API a little
3223 Avoid passing GstStructure in the add_role method, use varargs instead
3224 to construct the structure behind the scenes. We can then also use the
3225 structure name as the role and simplify some more logic.
3227 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3229 * gst/rtsp-server/rtsp-auth.c:
3232 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3234 * gst/rtsp-server/rtsp-auth.c:
3235 * gst/rtsp-server/rtsp-auth.h:
3236 * gst/rtsp-server/rtsp-client.c:
3237 auth: handle unauthorized response
3238 Move handling of the unauthorized response to the auth module, it can add
3239 the appropriate headers to request authorization for the required method
3240 much better than the client.
3242 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3244 * gst/rtsp-server/rtsp-client.c:
3245 * gst/rtsp-server/rtsp-client.h:
3246 client: allow for sending any message, not only requests
3247 Change the _send_request() method to _send_message() so that we
3248 can both send requests and replies.
3250 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3252 * docs/libs/gst-rtsp-server-sections.txt:
3253 * gst/rtsp-server/rtsp-server.h:
3256 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3258 * examples/test-video.c:
3259 * gst/rtsp-server/rtsp-auth.c:
3260 * gst/rtsp-server/rtsp-auth.h:
3261 * gst/rtsp-server/rtsp-server.c:
3262 * gst/rtsp-server/rtsp-server.h:
3263 auth: move TLS handling to auth module
3264 Remove the TLS settings on the server and move it to the auth module because
3265 that is where security related bits go.
3267 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3269 * gst/rtsp-server/rtsp-client.c:
3270 * gst/rtsp-server/rtsp-client.h:
3271 client: add state push/pop
3273 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3275 * gst/rtsp-server/rtsp-client.c:
3276 * gst/rtsp-server/rtsp-client.h:
3277 client: add connection to state
3279 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3281 * gst/rtsp-server/rtsp-mount-points.c:
3282 mount-points: fix debug
3284 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3286 * tests/check/gst/media.c:
3287 tests: fix media test
3289 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3291 * gst/rtsp-server/rtsp-thread-pool.c:
3292 thread-pool: we don't require a state
3294 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3296 * gst/rtsp-server/rtsp-server.c:
3297 server: let context ref the server
3298 So that we don't risk losing the server object early anc crash.
3300 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3302 * tests/check/gst/client.c:
3303 tests: fix client test
3305 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3308 * docs/libs/gst-rtsp-server-docs.sgml:
3309 * docs/libs/gst-rtsp-server-sections.txt:
3310 * gst/rtsp-server/rtsp-address-pool.c:
3311 * gst/rtsp-server/rtsp-auth.c:
3312 * gst/rtsp-server/rtsp-client.c:
3313 * gst/rtsp-server/rtsp-client.h:
3314 * gst/rtsp-server/rtsp-media-factory-uri.c:
3315 * gst/rtsp-server/rtsp-media-factory.c:
3316 * gst/rtsp-server/rtsp-media-factory.h:
3317 * gst/rtsp-server/rtsp-media.c:
3318 * gst/rtsp-server/rtsp-mount-points.c:
3319 * gst/rtsp-server/rtsp-params.c:
3320 * gst/rtsp-server/rtsp-permissions.c:
3321 * gst/rtsp-server/rtsp-sdp.c:
3322 * gst/rtsp-server/rtsp-server.c:
3323 * gst/rtsp-server/rtsp-server.h:
3324 * gst/rtsp-server/rtsp-session-media.c:
3325 * gst/rtsp-server/rtsp-session-pool.c:
3326 * gst/rtsp-server/rtsp-session.c:
3327 * gst/rtsp-server/rtsp-stream-transport.c:
3328 * gst/rtsp-server/rtsp-stream.c:
3329 * gst/rtsp-server/rtsp-thread-pool.c:
3330 * gst/rtsp-server/rtsp-token.c:
3333 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3335 * gst/rtsp-server/rtsp-session-pool.c:
3336 * gst/rtsp-server/rtsp-session-pool.h:
3337 session-pool: make vmethod to create a session
3338 Make a vmethod to create a sessions so that subclasses can create
3339 custom session objects
3341 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3343 * gst/rtsp-server/rtsp-auth.c:
3344 * gst/rtsp-server/rtsp-media-factory.h:
3345 * gst/rtsp-server/rtsp-media.h:
3346 * gst/rtsp-server/rtsp-mount-points.h:
3347 * gst/rtsp-server/rtsp-session-pool.h:
3348 * gst/rtsp-server/rtsp-stream.h:
3351 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3353 * docs/libs/gst-rtsp-server-docs.sgml:
3354 * docs/libs/gst-rtsp-server-sections.txt:
3355 * gst/rtsp-server/rtsp-address-pool.c:
3356 * gst/rtsp-server/rtsp-address-pool.h:
3357 * gst/rtsp-server/rtsp-auth.c:
3358 * gst/rtsp-server/rtsp-client.h:
3359 * gst/rtsp-server/rtsp-media-factory.h:
3360 * gst/rtsp-server/rtsp-media.c:
3361 * gst/rtsp-server/rtsp-media.h:
3362 * gst/rtsp-server/rtsp-permissions.c:
3363 * gst/rtsp-server/rtsp-permissions.h:
3364 * gst/rtsp-server/rtsp-server.h:
3365 * gst/rtsp-server/rtsp-session-media.c:
3366 * gst/rtsp-server/rtsp-session-media.h:
3367 * gst/rtsp-server/rtsp-session-pool.h:
3368 * gst/rtsp-server/rtsp-session.h:
3369 * gst/rtsp-server/rtsp-stream-transport.h:
3370 * gst/rtsp-server/rtsp-stream.c:
3371 * gst/rtsp-server/rtsp-thread-pool.h:
3374 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3377 * examples/Makefile.am:
3378 configure: compile cgroup example conditionally
3379 Only compile the cgroup example when we have libcgroup
3381 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3384 * examples/Makefile.am:
3385 * examples/test-cgroups.c:
3386 examples: add cgroups example
3388 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3390 * tests/check/gst/rtspserver.c:
3391 tests: fix compilation
3393 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3395 * gst/rtsp-server/rtsp-thread-pool.c:
3396 thread-pool: fix vmethod invocation
3398 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3400 * gst/rtsp-server/rtsp-thread-pool.c:
3401 * gst/rtsp-server/rtsp-thread-pool.h:
3402 thread-pool: store thread type in thread
3404 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3406 * gst/rtsp-server/rtsp-client.c:
3407 client: pass thread from pool to media _prepare
3408 Get a thread from the configured threadpool and pass it to the prepare method of
3411 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3413 * gst/rtsp-server/rtsp-media.c:
3414 * gst/rtsp-server/rtsp-media.h:
3415 media: Accept a thread in _prepare
3416 Remove out own threadpool handling and use the provided thread and
3417 maincontext for the bus messages and the state changes.
3419 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3421 * gst/rtsp-server/rtsp-server.c:
3422 server: configure client thread pool
3424 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3426 * gst/rtsp-server/rtsp-client.c:
3427 * gst/rtsp-server/rtsp-client.h:
3428 client: add method to configure thread pool
3430 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3432 * gst/rtsp-server/rtsp-client.h:
3433 * gst/rtsp-server/rtsp-server.c:
3434 * gst/rtsp-server/rtsp-server.h:
3435 server: use thread pool
3436 Use the thread pool instead of doing our own thing.
3438 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3440 * gst/rtsp-server/Makefile.am:
3441 * gst/rtsp-server/rtsp-thread-pool.c:
3442 * gst/rtsp-server/rtsp-thread-pool.h:
3443 thread-pool: add object to manage threads
3444 Add an object to manage the client and media threads.
3446 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3448 * gst/rtsp-server/rtsp-auth.c:
3449 auth: debug authorization check
3451 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3453 * gst/rtsp-server/rtsp-media.c:
3454 media: start media pipeline in context
3455 Start the media pipeline in the provided context (or our default one
3456 when NULL). This makes sure that we run the bus thread in this context and that
3457 all media threads are children of this context.
3459 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3461 * gst/rtsp-server/rtsp-media-factory.c:
3462 factory: pass permissions to media by default
3464 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3466 * examples/test-auth.c:
3467 test: add permissions to auth test
3468 Ass some permissions to the media factory in the test.
3470 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3472 * gst/rtsp-server/rtsp-auth.c:
3473 * gst/rtsp-server/rtsp-auth.h:
3474 * gst/rtsp-server/rtsp-client.c:
3475 auth: simplify auth checks
3476 Remove client from methods, it's now in the state
3477 Perform the check specified by the string, use the information from the
3478 thread local context.
3480 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3482 * gst/rtsp-server/rtsp-client.c:
3483 * gst/rtsp-server/rtsp-client.h:
3484 client: add state to current thread
3485 Add the client to the ClientState object.
3486 Place the ClientState on the current thread.
3488 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3490 * gst/rtsp-server/rtsp-media-factory.c:
3491 * gst/rtsp-server/rtsp-media-factory.h:
3492 * gst/rtsp-server/rtsp-media.c:
3493 * gst/rtsp-server/rtsp-media.h:
3494 media: make it possible to set permissions
3495 Make it possible to set permissions on media and media factory objects
3497 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3499 * gst/rtsp-server/Makefile.am:
3500 * gst/rtsp-server/rtsp-permissions.c:
3501 * gst/rtsp-server/rtsp-permissions.h:
3502 permissions: add permissions object
3503 Add a mini object to store permissions based on a role.
3505 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3507 * examples/test-auth.c:
3508 * gst/rtsp-server/rtsp-auth.c:
3509 * gst/rtsp-server/rtsp-auth.h:
3510 * gst/rtsp-server/rtsp-client.c:
3511 auth: add auth checks
3512 Add an enum with auth checks and implement the checks in the auth object.
3513 Perform the checks from the client.
3515 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3517 * examples/test-auth.c:
3518 * gst/rtsp-server/rtsp-auth.c:
3519 * gst/rtsp-server/rtsp-auth.h:
3520 * gst/rtsp-server/rtsp-client.h:
3521 auth: use the token after authentication
3522 After we authenticated a user, keep the Token around in the state.
3524 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3526 * gst/rtsp-server/rtsp-client.c:
3527 * gst/rtsp-server/rtsp-media.c:
3528 * gst/rtsp-server/rtsp-media.h:
3529 * tests/check/gst/media.c:
3530 media: add optional context for bus messages
3531 Add an optional mainloop to _prepare that will handle the bus messages instead
3532 of always using the shared mainloop.
3534 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3536 * gst/rtsp-server/Makefile.am:
3537 * gst/rtsp-server/rtsp-token.c:
3538 * gst/rtsp-server/rtsp-token.h:
3539 token: add authorization token
3540 Add a simply miniobject that contains the authorizations. The object contains a
3541 GstStructure that hold all authorization fields. When a user is authenticated,
3542 the auth module will create a Token for the user. The token is then used to
3543 check what operations the user is allowed to do and various other configuration
3546 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3548 * examples/test-auth.c:
3549 * gst/rtsp-server/rtsp-auth.c:
3550 * gst/rtsp-server/rtsp-auth.h:
3551 * gst/rtsp-server/rtsp-client.c:
3552 * gst/rtsp-server/rtsp-client.h:
3553 * gst/rtsp-server/rtsp-media-factory.c:
3554 * gst/rtsp-server/rtsp-media-factory.h:
3555 * gst/rtsp-server/rtsp-media.c:
3556 * gst/rtsp-server/rtsp-media.h:
3557 auth: remove auth from media and factory
3558 Remove the auth object from media and factory. We want to have the RTSPClient
3559 authenticate and authorize resources, there is no need to place another auth
3560 manager on the media/factory.
3562 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3564 * examples/test-auth.c:
3565 * gst/rtsp-server/rtsp-auth.c:
3566 * gst/rtsp-server/rtsp-auth.h:
3567 * gst/rtsp-server/rtsp-client.h:
3568 auth: add support for multiple basic auth tokens
3569 Make it possible to add multiple basic authorisation tokens to one authorization
3570 object. Associate with each token an authorization group that will define what
3571 capabilities are allowed.
3573 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3575 * gst/rtsp-server/rtsp-client.c:
3576 client: error out on non-aggregate control
3577 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
3579 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3581 * gst/rtsp-server/rtsp-client.c:
3582 client: rework setup request a little
3583 Cache the media in DESCRIBE based on the longest matching path with the uri
3584 that we can find in the mount points.
3585 Rework the setup request a little to get the media from the session or from
3586 the longest matching path, this way we can derive the control string as
3587 everything after the path instead of hardcoding it.
3588 Find the stream based on the control string and only open a session when all
3591 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3593 * gst/rtsp-server/rtsp-media.c:
3594 * gst/rtsp-server/rtsp-media.h:
3595 media: add method to find a stream by control url
3597 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3599 * gst/rtsp-server/rtsp-stream.c:
3600 * gst/rtsp-server/rtsp-stream.h:
3601 stream: add method to check control url of stream
3603 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3605 * gst/rtsp-server/rtsp-client.c:
3606 * gst/rtsp-server/rtsp-session-media.c:
3607 * gst/rtsp-server/rtsp-session-media.h:
3608 * gst/rtsp-server/rtsp-session.c:
3609 * gst/rtsp-server/rtsp-session.h:
3610 session: use path matching for session media
3611 Use a path string instead of a uri to lookup session media in the sessions. Also
3612 use path matching to find the largest possible path that matches.
3614 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3616 * gst/rtsp-server/rtsp-client.c:
3617 * gst/rtsp-server/rtsp-mount-points.c:
3618 * gst/rtsp-server/rtsp-mount-points.h:
3619 * tests/check/gst/mountpoints.c:
3620 mount-points: remove useless vmethod
3621 Making lookups in the mount points should not be done with a URL, if there is a
3622 mapping to be done from URL to mount points, we'll need to do it somewhere
3625 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3627 * gst/rtsp-server/rtsp-mount-points.c:
3628 * gst/rtsp-server/rtsp-mount-points.h:
3629 * tests/check/gst/mountpoints.c:
3630 mount-points: improve mount point searching
3631 Use a GSequence to keep track of the mount points.
3632 Match a URL to the longest matching registered mount point. This should be the
3633 URL to perform aggreagate control and the remainder is the stream specific
3635 Add some unit tests for this.
3637 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
3639 * gst/rtsp-server/Makefile.am:
3640 rtsp-server: Allow building of static library
3642 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3644 * tests/check/gst/mediafactory.c:
3645 tests: fix compilation
3647 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3649 * gst/rtsp-server/rtsp-sdp.c:
3650 sdp: get control string from stream
3651 Use the control string as configured in the stream.
3653 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3655 * gst/rtsp-server/rtsp-stream.c:
3656 * gst/rtsp-server/rtsp-stream.h:
3657 stream: add methods and property to set control string
3659 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3661 * gst/rtsp-server/rtsp-client.c:
3663 Rename variables for clarity
3664 Keep media in state when we can
3666 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3668 * gst/rtsp-server/rtsp-client.c:
3669 * gst/rtsp-server/rtsp-stream.c:
3670 * gst/rtsp-server/rtsp-stream.h:
3671 stream: add more support for IPv6
3672 Rename _get_address to _get_multicast_address in GstRTSPStream to
3673 make it clear that this function only deals with multicast.
3674 Make it possible to have both an IPv4 and IPv6 multicast address on
3675 a stream. Give the client an IPv4 or IPv6 address depending on the
3676 address it used to connect to the server.
3677 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
3679 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3681 * gst/rtsp-server/rtsp-client.c:
3684 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3686 * gst/rtsp-server/rtsp-stream.c:
3687 stream: handle failed port allocation
3688 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
3689 can't allocate any family at all. Also keep track of what port families we
3691 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
3693 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3695 * gst/rtsp-server/rtsp-stream.c:
3696 stream: improve docs
3698 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3700 * gst/rtsp-server/rtsp-stream-transport.c:
3701 stream-transport: remove old if 0 block
3703 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
3705 * tests/check/gst/client.c:
3707 gst_rtsp_client_get_uri() has been removed
3708 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
3710 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3712 * gst/rtsp-server/rtsp-client.c:
3713 * gst/rtsp-server/rtsp-client.h:
3714 client: add method to filter managed sessions
3715 Add a method to filter the sessions managed by this client connection.
3716 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
3718 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3720 * gst/rtsp-server/rtsp-client.c:
3721 * gst/rtsp-server/rtsp-client.h:
3722 client: remove _get_uri() method
3723 Remove the get_uri() method on the client. A client has no uri, the uri
3724 property is an internal property to manage the last cached media for
3727 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3729 * gst/rtsp-server/rtsp-media-factory.h:
3730 media-factory: fix typo
3732 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
3734 * gst/rtsp-server/rtsp-media.c:
3735 rtsp-media: Do not leak the query in default_query_stop
3736 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
3738 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3740 * gst/rtsp-server/rtsp-media.c:
3741 media: don't unlock when conversion fails
3742 Don't unlock the state lock when conversion fails because it was not locked.
3744 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3746 * gst/rtsp-server/rtsp-media.c:
3747 * gst/rtsp-server/rtsp-media.h:
3748 Add query_position and query_stop vmethods to rtsp-media
3750 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3752 * gst/rtsp-server/rtsp-media.c:
3753 Fix typo in property install for rtsp-media's time-provider
3755 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3757 * gst/rtsp-server/rtsp-client.c:
3758 * gst/rtsp-server/rtsp-client.h:
3759 client: clean some variables
3760 Clean some variables and add some guards to _send_request()
3762 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3764 * gst/rtsp-server/rtsp-client.c:
3765 * gst/rtsp-server/rtsp-client.h:
3766 Add gst_rtsp_client_send_request API
3767 This makes it possible to send arbitrary messages to a client, such as
3768 SET_PARAMETER or GET_PARAMETER
3770 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3772 * gst/rtsp-server/rtsp-media.c:
3773 * gst/rtsp-server/rtsp-media.h:
3774 media: add _get_element() method
3775 Add method to get the element used when creating the media.
3776 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
3778 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3780 * gst/rtsp-server/rtsp-media.c:
3783 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3785 * gst/rtsp-server/rtsp-stream.c:
3786 * gst/rtsp-server/rtsp-stream.h:
3787 stream: allow access to the rtp session
3788 https://bugzilla.gnome.org/show_bug.cgi?id=703004
3790 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
3792 * gst/rtsp-server/rtsp-stream.c:
3793 * gst/rtsp-server/rtsp-stream.h:
3794 dscp qos support in gst-rtsp-stream
3795 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
3797 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3799 * tests/check/gst/rtspserver.c:
3801 Actually do what the comment says. Also keep the old code around, not sure what
3802 should happen when you get a 454 from a TEARDOWN, does it close the connection?
3803 it currently doesn't.
3805 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3807 * gst/rtsp-server/rtsp-client.c:
3808 client: also watch newly created session
3809 When we newly created a session, start watching it immediately instead of
3810 on the next request.
3812 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
3814 * tests/check/gst/client.c:
3815 tests: add unit test for new-session
3816 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
3818 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3820 * gst/rtsp-server/rtsp-client.c:
3821 client: emit new-session when new session is created
3822 Only emit new-session when we created a new session for a client, not when a
3823 client picked up a previous session.
3824 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
3826 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
3828 * gst/rtsp-server/rtsp-client.c:
3829 client: handle asterisk as path in requests
3830 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
3832 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3834 * gst/rtsp-server/rtsp-media.c:
3835 media: handle segment query format mismatch
3836 It's possible that the segment query returns with a different format than what
3837 we asked for, handle this case also.
3839 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
3841 * gst/rtsp-server/rtsp-media.c:
3842 media: use segment stop in collect_media_stats
3843 Use segment stop instead of duration as range end point.
3844 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
3846 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3848 * gst/rtsp-server/rtsp-media.c:
3849 * tests/check/gst/media.c:
3850 rtsp-media: Do not leak the element in take_pipeline
3851 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
3853 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
3855 * gst/rtsp-server/rtsp-client.c:
3856 * gst/rtsp-server/rtsp-client.h:
3857 rtsp-client: Make configure_client_transport virtual
3858 This patch makes configure_client_transport virtual. The functionality is
3859 needed to handle some weird clients sending multicast transport settings as url
3861 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
3863 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3865 * gst/rtsp-server/rtsp-client.c:
3866 * gst/rtsp-server/rtsp-client.h:
3867 rtsp-client: Make param_set and param_get virtual
3868 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
3870 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
3872 * gst/rtsp-server/rtsp-client.c:
3873 * gst/rtsp-server/rtsp-media.c:
3874 * gst/rtsp-server/rtsp-media.h:
3875 media: convert_range replaces get_range_times
3876 get_range_times worked for handling UTC ranges for seeks, but we also
3877 need to convert back from NPT to the requested unit in
3878 get_range_string. convert_range is now used for both.
3879 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
3881 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3883 * gst/rtsp-server/rtsp-client.c:
3884 * gst/rtsp-server/rtsp-sdp.c:
3885 * gst/rtsp-server/rtsp-sdp.h:
3886 sdp: cleanup sdp info
3887 We don't need to pass the proto, we can more easily check a boolean.
3888 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
3890 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
3892 * gst/rtsp-server/rtsp-sdp.c:
3893 use 0.0.0.0 or :: for c= line instead of server address
3895 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
3897 * gst/rtsp-server/rtsp-client.c:
3898 use local address, not remote, in SDP
3899 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
3901 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3904 Automatic update of common submodule
3905 From 098c0d7 to 01a7a46
3907 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
3909 * gst/rtsp-server/rtsp-media.c:
3910 * gst/rtsp-server/rtsp-media.h:
3911 media: possibility to override range time conversion
3912 Make it possible to override the conversion from GstRTSPTimeRange to
3913 GstClockTimes, that is done before seeking on the media
3914 pipeline. Overriding can be useful for UTC ranges, where the default
3915 conversion gives nanoseconds since 1900.
3916 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
3918 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
3920 * gst/rtsp-server/rtsp-server.c:
3921 * gst/rtsp-server/rtsp-server.h:
3922 rtsp-server: Expose the use_client_settings API
3923 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
3925 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
3927 * gst/rtsp-server/rtsp-client.c:
3928 * gst/rtsp-server/rtsp-stream.c:
3929 * gst/rtsp-server/rtsp-stream.h:
3930 rtspstream: handle both ipv4 and ipv6 clients
3931 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
3933 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3935 * gst/rtsp-server/rtsp-sdp.c:
3936 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
3937 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
3938 We already have a way to place extra attributes in the SDP by using a string
3939 property with prefix x- or a- in the caps.
3941 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3943 * gst/rtsp-server/rtsp-sdp.c:
3944 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
3945 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
3946 We already have a way to place extra attributes in the SDP, just make a string
3947 property in the payloader with a- or x- prefix.
3949 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3951 * gst/rtsp-server/rtsp-sdp.c:
3952 rtsp: place a- and x- properties as attributes
3953 application/x-rtp has properties with a- and x- prefixes that should be
3954 placed as attributes in the SDP for the media instead of being added to the
3957 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3959 * examples/Makefile.am:
3960 * examples/test-video.c:
3961 example: add TLS example
3963 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3965 * gst/rtsp-server/rtsp-server.c:
3966 * gst/rtsp-server/rtsp-server.h:
3967 server: add support for TLS
3968 Add methods to set and get a TLS certificate.
3969 Add vmethod to configure a new connection. By default, configure the TLS
3970 certificate in a new connection if needed.
3972 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3974 * gst/rtsp-server/rtsp-server.c:
3975 * gst/rtsp-server/rtsp-server.h:
3976 server: remove accept_client vmethod
3977 This vmethod is not very useful so remove it.
3979 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3981 * gst/rtsp-server/rtsp-server.c:
3982 server: don't crash on NULL GError
3984 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
3986 * gst/rtsp-server/rtsp-session-pool.c:
3987 rtsp-session-pool: corrected session timeout detection
3988 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
3990 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3992 * gst/rtsp-server/rtsp-client.c:
3993 client: improve debug
3995 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3997 * gst/rtsp-server/rtsp-client.c:
3998 * gst/rtsp-server/rtsp-client.h:
3999 * gst/rtsp-server/rtsp-server.c:
4000 server: refactor connection setup
4001 Let the server accept the socket connection and construct a GstRTSPConnection
4002 from it. Remove the code from the client and let the client only deal with
4003 a fully configure GstRTSPConnection object.
4004 We will need this later when the server will configure the connection for
4007 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4009 * gst/rtsp-server/rtsp-stream.c:
4010 stream: keep the transport object alive
4011 Keep the transport object alive while we have it as qdata on the
4014 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
4016 * gst/rtsp-server/rtsp-client.c:
4017 * gst/rtsp-server/rtsp-server.c:
4018 rtsp-server: Do not crash on nmapping of server
4019 * generate error when gst_rtsp_connection_accept fails
4020 * do not stop accepting incoming connections because
4021 accepting a client fails
4022 https://bugzilla.gnome.org/show_bug.cgi?id=701072
4024 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
4026 * gst/rtsp-server/rtsp-client.c:
4027 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
4028 https://bugzilla.gnome.org/show_bug.cgi?id=700953
4030 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4032 * gst/rtsp-server/rtsp-sdp.c:
4033 rtsp-sdp: Parse framerate caps field and set SDP attribute
4034 The SDP attribute and its format is described in RFC4566.
4035 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4037 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
4039 * gst/rtsp-server/rtsp-sdp.c:
4040 rtsp-sdp: Parse width/height from caps and set SDP attribute
4041 The SDP attribute and its format is described in RFC6064.
4042 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
4044 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
4046 * gst/rtsp-server/rtsp-sdp.c:
4047 * tests/check/gst/client.c:
4048 rtsp-sdp: add bandwidth line
4049 https://bugzilla.gnome.org/show_bug.cgi?id=699220
4051 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4054 Automatic update of common submodule
4055 From 5edcd85 to 098c0d7
4057 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4059 * tests/check/gst/media.c:
4060 tests: add dynamic payloader prepare/unprepare check
4062 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4064 * gst/rtsp-server/rtsp-media.c:
4065 media: release lock when removing fakesink
4067 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4069 * gst/rtsp-server/rtsp-stream.c:
4070 stream: set elements to NULL before removing
4071 When removing a stream, set the elements to NULL first. This avoids
4072 element-is-not-in-NULL-state errors when we dispose the elements.
4074 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4077 Automatic update of common submodule
4078 From 3cb3d3c to 5edcd85
4080 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4082 * gst/rtsp-server/rtsp-media.c:
4083 * gst/rtsp-server/rtsp-media.h:
4084 media: listen to pad-removed signals
4085 Listen to the pad-removed signal and remove the stream associated with the
4087 Add signal to be notified of the removed pad.
4088 Remove the fakesink in unprepare()
4089 Fix signatures of the signal methods
4091 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4093 * examples/test-sdp.c:
4094 tests: add example of reusable pipelines
4096 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
4098 * gst/rtsp-server/rtsp-stream.c:
4099 * gst/rtsp-server/rtsp-stream.h:
4100 stream: add method to get the srcpad
4102 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
4104 * tests/check/gst/media.c:
4105 check: add media prepare/unprepare test
4106 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4108 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
4110 * gst/rtsp-server/rtsp-media.c:
4111 media: disconnect from signal handlers in unprepare()
4112 We connected to the pad-added and no-more-pads signals in prepare() so
4113 we need to disconnect from them in unprepare().
4114 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4116 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
4118 * gst/rtsp-server/rtsp-media.c:
4119 media: don't free streams array
4120 Don't free the streams array in the unprepare() method, they were not
4122 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4124 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
4126 * gst/rtsp-server/rtsp-media.c:
4127 media: don't unref the pipeline in unprepare
4128 Unprepare() should undo what prepare() does. Because the pipeline is
4129 not created in prepare(), we should not unref it in unprepare()
4131 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
4133 * gst/rtsp-server/rtsp-stream.c:
4134 stream: clear session and caps for reuse
4135 Set the session and caps to NULL after unref otherwise we might unref
4137 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
4139 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
4141 * gst/rtsp-server/rtsp-client.c:
4142 client: send out teardown signal before tearing down
4143 The advantage is that in the signal handler you get direct access to
4144 information about what streams are about to get torn down (in the
4145 GstRTSPClientState).
4146 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
4148 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
4150 * gst/rtsp-server/rtsp-client.c:
4151 * gst/rtsp-server/rtsp-client.h:
4152 client: expose connection
4153 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
4155 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
4158 Automatic update of common submodule
4159 From aed87ae to 3cb3d3c
4161 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4163 * gst/rtsp-server/rtsp-media.c:
4164 * gst/rtsp-server/rtsp-media.h:
4165 * gst/rtsp-server/rtsp-session-media.c:
4166 * gst/rtsp-server/rtsp-session-media.h:
4167 media: add method to get the base_time of the pipeline
4168 Together with a shared clock, this base-time could eventually be sent to
4169 the client so that it can reconstruct the exact running-time of the clock
4172 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4174 * gst/rtsp-server/Makefile.am:
4175 * gst/rtsp-server/rtsp-media.c:
4176 * gst/rtsp-server/rtsp-media.h:
4177 * gst/rtsp-server/rtsp-sdp.c:
4178 media: add GstNetTimeProvider support
4179 Add a property to let the media provide a GstNetTimeProvider for its clock.
4180 Make methods to get the clock and nettimeprovider
4181 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
4182 provider and also the current time of the clock. This should make it possible
4183 for (GStreamer) clients to slave their clock to the server clock.
4185 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
4188 Automatic update of common submodule
4189 From 04c7a1e to aed87ae
4191 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4193 * gst/rtsp-server/rtsp-media.c:
4194 media: wait for buffering to complete
4195 Wait for buffering to complete before changing the state to the target state.
4197 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4199 * gst/rtsp-server/rtsp-media.c:
4200 media: small cleanup
4202 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
4204 * tests/check/gst/rtspserver.c:
4205 tests: remove extra unref in test_setup_non_existing_stream
4206 The unref is not needed anymore, teardown runs without it.
4207 https://bugzilla.gnome.org/show_bug.cgi?id=696542
4209 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
4211 * tests/check/gst/rtspserver.c:
4212 tests: GSocketService cleanup in test_bind_already_in_use
4213 Use g_socket_service_stop so the rtspserver test stops listening for
4214 incoming connections in test_bind_already_in_use.
4215 https://bugzilla.gnome.org/show_bug.cgi?id=696541
4217 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
4219 * gst/rtsp-server/rtsp-media-factory.c:
4220 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
4221 Instead use a GWeakRef which is safe to use
4222 This is a known GLib bug, see:
4223 https://bugzilla.gnome.org/show_bug.cgi?id=667145
4225 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
4227 * gst/rtsp-server/rtsp-client.c:
4228 * gst/rtsp-server/rtsp-media.c:
4229 * gst/rtsp-server/rtsp-media.h:
4230 * gst/rtsp-server/rtsp-sdp.c:
4231 * tests/check/gst/media.c:
4232 * tests/check/gst/rtspserver.c:
4233 rtsp-media/client: Reply to PLAY request with same type of Range
4234 Remember the type of Range from the PLAY request and use the same type for
4237 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
4239 * gst/rtsp-server/rtsp-client.c:
4240 * gst/rtsp-server/rtsp-client.h:
4241 * tests/check/gst/client.c:
4242 rtsp-client: expose uri
4244 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
4246 * tests/check/gst/mediafactory.c:
4247 tests: Hold ref while creating second media
4248 To test if the media aren't shared, make sure we keep the first one while creating a second
4249 otherwise the same memory address may be reused.
4251 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
4254 configure: remove out-of-date comment
4256 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
4259 .gitignore: ignore more build files
4261 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
4263 * tests/check/Makefile.am:
4264 tests: use right _LIBS variable for gst-plugins-base libs
4266 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4268 * tests/check/Makefile.am:
4269 check: add librtp to libs
4271 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
4273 * tests/check/gst/rtspserver.c:
4274 tests: Add test to check selecting a port the server will send from
4276 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
4278 * tests/check/gst/rtspserver.c:
4279 tests: Make sure packets are actually received
4281 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4283 * gst/rtsp-server/rtsp-stream.c:
4284 stream: Select unicast address from pool if appropriate
4286 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
4288 * gst/rtsp-server/rtsp-stream.c:
4289 stream: Properties are always there in Gst 1.0
4291 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4293 * tests/check/gst/addresspool.c:
4294 tests: Add tests for unicast addresses in pool
4296 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
4298 * gst/rtsp-server/rtsp-address-pool.c:
4299 * tests/check/gst/addresspool.c:
4300 address-pool: Verify that multicast addresses are used for multicast and vice-versa
4302 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
4304 * docs/libs/gst-rtsp-server-sections.txt:
4305 * gst/rtsp-server/rtsp-address-pool.c:
4306 * gst/rtsp-server/rtsp-address-pool.h:
4307 * gst/rtsp-server/rtsp-stream.c:
4308 * tests/check/gst/addresspool.c:
4309 address-pool: Add unicast addresses
4311 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4314 * gst/rtsp-server/rtsp-server.c:
4315 * tests/check/gst/rtspserver.c:
4316 rtsp-server: Limit the number of threads per server instance
4317 If we exceed the maximum, just round robin the clients over the existing
4320 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
4322 * gst/rtsp-server/rtsp-server.c:
4323 rtsp-server: No need to store the GMainContext in the client context
4325 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
4327 * tests/check/gst/rtspserver.c:
4328 tests: Add test for client disconnection
4330 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4332 * tests/check/gst/rtspserver.c:
4333 tests: Test client and session timeouts with multiple threads
4335 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
4337 * gst/rtsp-server/rtsp-address-pool.c:
4338 * gst/rtsp-server/rtsp-auth.c:
4339 * gst/rtsp-server/rtsp-client.c:
4340 * gst/rtsp-server/rtsp-media-factory-uri.c:
4341 * gst/rtsp-server/rtsp-media-factory.c:
4342 * gst/rtsp-server/rtsp-media.c:
4343 * gst/rtsp-server/rtsp-mount-points.c:
4344 * gst/rtsp-server/rtsp-server.c:
4345 * gst/rtsp-server/rtsp-session-media.c:
4346 * gst/rtsp-server/rtsp-session-pool.c:
4347 * gst/rtsp-server/rtsp-session.c:
4348 Document locking and its order
4350 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
4352 * tests/check/gst/rtspserver.c:
4353 tests: Test that slow DESCRIBE don't block other clients
4355 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
4357 * tests/check/gst/client.c:
4358 tests: Add tests for client-requested multicast address
4360 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
4362 * docs/libs/gst-rtsp-server-sections.txt:
4363 docs: Put the various functions in the right sections
4365 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
4367 * docs/libs/gst-rtsp-server-docs.sgml:
4368 * docs/libs/gst-rtsp-server-sections.txt:
4369 * gst/rtsp-server/rtsp-address-pool.c:
4370 * gst/rtsp-server/rtsp-address-pool.h:
4371 docs: Generate docs for GstRTSPAddressPool
4373 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4375 * gst/rtsp-server/rtsp-client.c:
4376 * gst/rtsp-server/rtsp-stream.c:
4377 * gst/rtsp-server/rtsp-stream.h:
4378 client: Check client provided addresses against the address pool
4380 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
4382 * gst/rtsp-server/rtsp-address-pool.c:
4383 * gst/rtsp-server/rtsp-address-pool.h:
4384 * tests/check/gst/addresspool.c:
4385 address-pool: Add API to request a specific address from the pool
4386 Also add relevant unit tests.
4388 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
4390 * tests/check/gst/mediafactory.c:
4391 tests: Check the passing around of a RTSPAddressPool
4392 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
4393 way down to the stream.
4395 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
4397 * tests/check/gst/addresspool.c:
4398 tests: Add more tests for the address pool
4400 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
4402 * gst/rtsp-server/rtsp-address-pool.c:
4403 address-pool: Fix off by one error
4404 When splitting a port range, the port after a skip is not part of range.
4406 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
4409 Automatic update of common submodule
4410 From 2de221c to 04c7a1e
4412 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
4415 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
4416 AM_CONFIG_HEADER was removed in automake 1.13
4417 https://bugzilla.gnome.org/show_bug.cgi?id=693368
4419 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
4422 Automatic update of common submodule
4423 From a942293 to 2de221c
4425 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4427 * gst/rtsp-server/rtsp-client.c:
4428 client: make sure the watch exists while sending data
4429 Protect the send_func with a lock. This allows us to wait for sending
4430 to complete before changing the send_func and user_data. We add an
4431 extra ref to the watch to make sure that it remains valid during
4433 When closing the connection, set the send_func to NULL
4434 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
4436 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4438 * tests/check/Makefile.am:
4439 tests: use GST_*_1_0 environment variables everywhere
4440 The _1_0 suffixed environment variables override the
4441 non-suffixed ones, so if we're in an environment that
4442 sets the _1_0 suffixed ones, such as jhbuild, we need
4443 to set those to make sure ours actually always get
4446 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4449 Automatic update of common submodule
4450 From acb04d9 to a942293
4452 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4454 * gst/rtsp-server/rtsp-client.c:
4455 rtsp-client: set the client backlog
4456 Set the client backlog to a reasonable default
4458 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
4460 * gst/rtsp-server/rtsp-media.c:
4461 rtsp-media: Make the element a constructor parameter
4462 https://bugzilla.gnome.org/show_bug.cgi?id=689594
4464 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4466 * docs/libs/Makefile.am:
4467 docs: Link with gcov library when gcov is enabled
4468 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
4470 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4472 * gst/rtsp-server/rtsp-media.c:
4473 media: match prepare with unprepare
4474 Really unprepare when there were an equal amount of prepare calls.
4476 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4478 * gst/rtsp-server/rtsp-media.c:
4479 media: media has to be unprepared in finalize
4480 Because unprepare takes away the last ref on the media.
4482 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4484 * gst/rtsp-server/rtsp-client.c:
4485 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
4486 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
4487 We can't use the refcount to trigger unprepare because it is the unprepare call
4488 that removes the last refcount after all messages are consumed. What we should
4489 probably do is make a prepared refcount and only unprepare when the refcount
4492 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4494 * gst/rtsp-server/rtsp-media.c:
4495 media: let the source unref the last media ref
4496 the last ref to the media is held by the source so we don't need to add more ref
4497 and unrefs, we simply destroy the media when the source is gone.
4499 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4501 * gst/rtsp-server/rtsp-media.c:
4502 media: improve debug
4504 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4506 * gst/rtsp-server/rtsp-media.c:
4508 Make sure we are in the right state when collecting the position and duration.
4509 Only make ourselves PREPARED when we were previously PREPARING.
4511 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4513 * gst/rtsp-server/rtsp-media.c:
4514 media: use g_object_ref/unref for GObjects
4516 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
4518 * gst/rtsp-server/rtsp-client.c:
4519 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
4520 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
4521 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
4522 isn't being used anymore.
4524 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
4526 * gst/rtsp-server/rtsp-media.c:
4527 Fix compiler warning
4529 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
4531 * gst/rtsp-server/rtsp-media-factory-uri.c:
4532 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
4534 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4536 * gst/rtsp-server/rtsp-session-media.h:
4539 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4541 * gst/rtsp-server/rtsp-media.c:
4542 * tests/check/gst/media.c:
4543 media: avoid element leak
4545 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4547 * gst/rtsp-server/rtsp-media.c:
4548 media: require an element in media constructor
4550 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4552 * gst/rtsp-server/rtsp-client.c:
4553 Revert "client: TEARDOWN brings that state to Init again"
4554 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
4555 The object is already disposed, there is no point in setting the state.
4557 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4559 * gst/rtsp-server/rtsp-client.c:
4560 client: TEARDOWN brings that state to Init again
4562 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4564 * docs/libs/gst-rtsp-server-sections.txt:
4565 * examples/test-auth.c:
4566 * gst/rtsp-server/rtsp-auth.c:
4567 * gst/rtsp-server/rtsp-auth.h:
4568 * gst/rtsp-server/rtsp-client.c:
4569 * gst/rtsp-server/rtsp-client.h:
4570 * gst/rtsp-server/rtsp-media-factory-uri.c:
4571 * gst/rtsp-server/rtsp-media-factory-uri.h:
4572 * gst/rtsp-server/rtsp-media-factory.c:
4573 * gst/rtsp-server/rtsp-media-factory.h:
4574 * gst/rtsp-server/rtsp-media.c:
4575 * gst/rtsp-server/rtsp-media.h:
4576 * gst/rtsp-server/rtsp-mount-points.c:
4577 * gst/rtsp-server/rtsp-mount-points.h:
4578 * gst/rtsp-server/rtsp-sdp.c:
4579 * gst/rtsp-server/rtsp-server.c:
4580 * gst/rtsp-server/rtsp-server.h:
4581 * gst/rtsp-server/rtsp-session-media.c:
4582 * gst/rtsp-server/rtsp-session-media.h:
4583 * gst/rtsp-server/rtsp-session-pool.c:
4584 * gst/rtsp-server/rtsp-session-pool.h:
4585 * gst/rtsp-server/rtsp-session.c:
4586 * gst/rtsp-server/rtsp-session.h:
4587 * gst/rtsp-server/rtsp-stream-transport.c:
4588 * gst/rtsp-server/rtsp-stream-transport.h:
4589 * gst/rtsp-server/rtsp-stream.c:
4590 * gst/rtsp-server/rtsp-stream.h:
4591 * tests/check/gst/media.c:
4592 rtsp: make object details private
4593 Make all object details private
4594 Add methods to access private bits
4596 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4598 * tests/check/Makefile.am:
4599 * tests/check/gst/media.c:
4600 tests: add media tests
4602 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4604 * gst/rtsp-server/rtsp-media.c:
4605 media: check if prepared for some methods
4606 Check that the media object is prepared before doing seek and getting the
4607 current position etc.
4608 Add some g_return checks.
4610 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4612 * tests/check/Makefile.am:
4613 * tests/check/gst/mediafactory.c:
4614 tests: add mediafactory test
4616 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4618 * gst/rtsp-server/rtsp-stream.c:
4619 stream: improve debug
4621 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4623 * gst/rtsp-server/rtsp-media.c:
4624 * gst/rtsp-server/rtsp-media.h:
4625 media: unref pipeline in finalize to avoid leaking it
4627 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4629 * gst/rtsp-server/rtsp-media-factory-uri.c:
4630 * gst/rtsp-server/rtsp-media.c:
4631 rtsp: use gst_object_unref on GstObjects
4633 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4635 * gst/rtsp-server/rtsp-media-factory.c:
4636 media-factory: require an url
4638 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4640 * examples/test-uri.c:
4641 examples: fix include
4643 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4645 * gst/rtsp-server/rtsp-server.h:
4646 server: remove unused include
4648 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4650 * tests/check/Makefile.am:
4651 * tests/check/gst/mountpoints.c:
4652 tests: add test for mountpoints
4654 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4656 * gst/rtsp-server/rtsp-client.c:
4657 client: fix factory leak
4658 Keep the factory in the state object only for authorization checks and make
4659 sure we unref it on failure. Also don't keep invalid objects in the state
4662 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4664 * gst/rtsp-server/rtsp-mount-points.c:
4665 mounts: add g_return_if guards
4667 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4669 * tests/check/gst/client.c:
4670 tests: add more tests
4672 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4674 * gst/rtsp-server/rtsp-client.c:
4675 client: improve debug
4677 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4679 * gst/rtsp-server/rtsp-client.c:
4680 client: improve debug and fix leaks
4681 Cleanup the uri and session when there is a bad request.
4683 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4688 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4690 * tests/check/gst/client.c:
4691 test: add test for session in options request
4693 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4695 * gst/rtsp-server/rtsp-client.c:
4696 client: use 454 when session can't be found
4697 We should use 454 when a session can't be found because there was no session
4698 pool configured in the server. This is not a server configuration problem
4699 because the server on which the request is done might not be the same one that
4700 will keep the sessions for us and so it does not need to support sessions.
4702 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4704 * gst/rtsp-server/rtsp-client.c:
4705 client: only free connection when there is one
4706 It's possible that the client doesn't have a connection when we try to free it.
4708 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4710 * tests/check/Makefile.am:
4711 * tests/check/gst/client.c:
4712 tests: add unit test for the client object
4714 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4716 * gst/rtsp-server/rtsp-client.c:
4717 client: small cleanup
4719 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4721 * gst/rtsp-server/rtsp-client.h:
4722 client: remove unused include
4724 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4726 * gst/rtsp-server/rtsp-client.c:
4727 client: fix compilation
4729 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4731 * gst/rtsp-server/rtsp-client.c:
4732 client: call destroy without the lock
4734 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4736 * gst/rtsp-server/rtsp-client.c:
4737 * gst/rtsp-server/rtsp-client.h:
4738 client: make the client usable without a socket
4739 Make a method to let the client handle a message and a callback when the client
4740 wants us to send a response message back. This makes it possible to also use the
4741 client object without the sockets, which should make it easier to test.
4743 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4745 * gst/rtsp-server/rtsp-client.c:
4746 * gst/rtsp-server/rtsp-client.h:
4747 client: small cleanup
4749 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4751 * docs/libs/gst-rtsp-server-sections.txt:
4752 * gst/rtsp-server/rtsp-client.c:
4753 * gst/rtsp-server/rtsp-client.h:
4754 * gst/rtsp-server/rtsp-server.c:
4755 client: remove reference to server
4756 We don't need to keep a ref to the server
4758 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4760 * gst/rtsp-server/rtsp-client.c:
4761 * gst/rtsp-server/rtsp-client.h:
4763 Also add some g_return_if()
4765 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4767 * gst/rtsp-server/rtsp-client.c:
4768 client: log more errors
4770 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4772 * gst/rtsp-server/rtsp-client.c:
4773 client: fix compilation
4775 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4777 * gst/rtsp-server/rtsp-client.c:
4778 * gst/rtsp-server/rtsp-client.h:
4779 client: add generic close-after-send support
4780 Add a property to send_response() to close the connection after the response has
4781 been sent to the client.
4783 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4786 * docs/libs/gst-rtsp-server-docs.sgml:
4787 * docs/libs/gst-rtsp-server-sections.txt:
4788 * docs/libs/gst-rtsp-server.types:
4789 * examples/test-auth.c:
4790 * examples/test-launch.c:
4791 * examples/test-mp4.c:
4792 * examples/test-multicast.c:
4793 * examples/test-multicast2.c:
4794 * examples/test-ogg.c:
4795 * examples/test-readme.c:
4796 * examples/test-sdp.c:
4797 * examples/test-uri.c:
4798 * examples/test-video.c:
4799 * gst/rtsp-server/Makefile.am:
4800 * gst/rtsp-server/rtsp-auth.h:
4801 * gst/rtsp-server/rtsp-client.c:
4802 * gst/rtsp-server/rtsp-client.h:
4803 * gst/rtsp-server/rtsp-media-mapping.c:
4804 * gst/rtsp-server/rtsp-media-mapping.h:
4805 * gst/rtsp-server/rtsp-mount-points.c:
4806 * gst/rtsp-server/rtsp-mount-points.h:
4807 * gst/rtsp-server/rtsp-server.c:
4808 * gst/rtsp-server/rtsp-server.h:
4809 * gst/rtsp-server/rtsp-session-media.c:
4810 * gst/rtsp-server/rtsp-session-pool.c:
4811 * gst/rtsp-server/rtsp-session-pool.h:
4812 * tests/check/gst/rtspserver.c:
4813 MediaMapping -> MountPoints
4814 Describes better what the object manages.
4816 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4819 configure: bump required version of -base
4821 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4823 * gst/rtsp-server/rtsp-media.c:
4826 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4828 * gst/rtsp-server/rtsp-media.c:
4829 * gst/rtsp-server/rtsp-media.h:
4830 media: support more Range formats
4831 Use the new -base methods to convert the Range string into a seek start and stop
4834 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4836 * examples/test-launch.c:
4837 examples: fix whitespace
4839 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4841 * examples/test-auth.c:
4842 test-auth: add example of how to remove sessions
4843 Add an example of the session filter api.
4845 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4847 * examples/test-uri.c:
4848 test-uri: remove mapping example
4850 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4852 * examples/test-uri.c:
4853 test-uri: fix callback signature
4855 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4857 * gst/rtsp-server/rtsp-media-factory.c:
4858 factory: keep ref to factory while media active
4859 While the media from a factory is alive, keep a ref to the factory.
4860 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
4862 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4864 * gst/rtsp-server/rtsp-media-factory-uri.c:
4865 factory-uri: add some debug
4867 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4869 * gst/rtsp-server/rtsp-stream.c:
4870 stream: set udp sources to PLAYING
4871 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
4872 so that it doesn't cause our pipeline to produce ASYNC-DONE.
4874 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4876 * gst/rtsp-server/rtsp-media-factory-uri.c:
4877 factory-uri: take ref to factory
4878 Take a ref to the factory that we place in our list.
4880 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4882 * tests/Makefile.am:
4883 * tests/test-reuse.c:
4884 test: add test for server reuse
4885 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
4887 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
4889 * gst/rtsp-server/rtsp-server.c:
4890 server: start and stop multiple times
4891 Stop listening on the RTSP port when the GSource is removed, so clients
4892 can't connect and the server can be started again.
4893 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
4895 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4897 * gst/rtsp-server/rtsp-server.c:
4898 server: fix small leak
4900 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4902 * gst/rtsp-server/rtsp-media.c:
4903 media: unref source in finish_unprepare
4904 The source is created in prepare, unref it in finish_unprepare.
4905 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
4907 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
4909 * gst/rtsp-server/rtsp-client.c:
4910 * gst/rtsp-server/rtsp-media.c:
4911 rtsp-media: remove bus watch before finalizing
4912 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
4913 * An extra media ref is added for the bus watch. This extra ref is unreffed by
4914 the GDestroyNotify function.
4915 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
4916 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
4917 gst_rtsp_media_unprepare before unreffing the media.
4918 This way, the bus watch will be removed before the media is finalized.
4919 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
4921 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
4923 * gst/rtsp-server/rtsp-client.c:
4924 * gst/rtsp-server/rtsp-client.h:
4925 client: wait until the TEARDOWN response is sent to close the connection
4926 Responses can be sent async so we need to wait until the TEARDOWN response has
4927 been written before we close the connection to the client. This avoids the risk
4928 of writing/polling closed sockets.
4929 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
4931 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
4933 * gst/rtsp-server/rtsp-stream.c:
4934 rtsp-stream: plug socket leak
4935 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
4937 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
4940 Automatic update of common submodule
4941 From 6bb6951 to a72faea
4943 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
4945 * gst/rtsp-server/rtsp-media-factory-uri.c:
4946 rtsp-server: don't use deprecated API
4948 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
4950 * gst/rtsp-server/rtsp-client.c:
4951 rtsp-client: fix unused-but-set-variable compiler warning
4952 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
4954 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4957 * docs/libs/gst-rtsp-server-sections.txt:
4958 * gst/rtsp-server/rtsp-client.c:
4961 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4963 * examples/Makefile.am:
4964 * examples/test-multicast2.c:
4965 examples: add another multicast example
4966 Add an example for how to configure separate multicast ranges for each media
4969 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4971 * examples/test-multicast.c:
4974 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4976 * gst/rtsp-server/rtsp-client.c:
4977 * gst/rtsp-server/rtsp-media.c:
4978 * gst/rtsp-server/rtsp-session-media.c:
4979 * gst/rtsp-server/rtsp-session-media.h:
4980 * gst/rtsp-server/rtsp-stream-transport.c:
4981 * gst/rtsp-server/rtsp-stream-transport.h:
4982 stream: use the address managed by the stream
4983 Use the address managed by the stream for multicast. This allows us to have 1
4984 multicast address for each stream.
4985 Because the address is now managed by the stream we don't have to pass it around
4987 Set the address pool on the streams.
4989 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4991 * gst/rtsp-server/rtsp-client.c:
4992 * gst/rtsp-server/rtsp-media.c:
4993 * gst/rtsp-server/rtsp-stream.c:
4996 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4998 * gst/rtsp-server/rtsp-media.c:
4999 * gst/rtsp-server/rtsp-media.h:
5000 media: add signal for new streams
5001 This allows applications to listen for new streams and configure properties on
5002 them, like the address pool.
5004 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5006 * gst/rtsp-server/rtsp-media.c:
5007 media: configure address pool in new streams
5009 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5011 * gst/rtsp-server/rtsp-stream.c:
5012 * gst/rtsp-server/rtsp-stream.h:
5013 stream: add methods to deal with address pool
5014 Add methods to get and set the address pool for the stream
5015 Add method to allocate and get the multicast addresses for this stream.
5017 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5019 * docs/libs/gst-rtsp-server-sections.txt:
5020 * gst/rtsp-server/rtsp-media.c:
5021 * gst/rtsp-server/rtsp-media.h:
5022 media: remove MTU property
5023 It is a stream property
5025 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5027 * gst/rtsp-server/rtsp-client.c:
5028 client: set blocksize only on stream
5029 Set the blocksize only on the current stream.
5031 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5033 * gst/rtsp-server/rtsp-stream.c:
5034 stream: share src and sink sockets
5035 the allocated socket is in the used-socket property, not socket.
5037 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5039 * gst/rtsp-server/rtsp-address-pool.c:
5040 * gst/rtsp-server/rtsp-address-pool.h:
5041 * gst/rtsp-server/rtsp-client.c:
5042 * gst/rtsp-server/rtsp-session-media.c:
5043 * gst/rtsp-server/rtsp-session-media.h:
5044 * gst/rtsp-server/rtsp-stream-transport.c:
5045 * gst/rtsp-server/rtsp-stream-transport.h:
5046 * tests/check/gst/addresspool.c:
5047 rtsp: make address-pool return an address object
5048 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
5049 store more info in the structure and allows us to more easily return the address
5050 to the right pool when no longer needed.
5051 Pass the address to the StreamTransport so that we can return it to the pool
5052 when the stream transport is freed or changed.
5054 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5056 * examples/Makefile.am:
5057 * examples/test-multicast.c:
5058 examples: add multicast example
5059 Show how to set up the multicast address pool so that media can be
5060 server with multicast.
5062 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5064 * gst/rtsp-server/rtsp-client.c:
5065 * gst/rtsp-server/rtsp-media-factory.c:
5066 * gst/rtsp-server/rtsp-media-factory.h:
5067 * gst/rtsp-server/rtsp-media.c:
5068 * gst/rtsp-server/rtsp-media.h:
5069 rtsp: use AddressPool
5070 Remove the multicast_group property.
5071 Use the configured addresspool to allocate multicast addresses.
5073 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5075 * gst/rtsp-server/rtsp-address-pool.c:
5076 * gst/rtsp-server/rtsp-address-pool.h:
5077 address-pool: add clear method
5079 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5081 * gst/rtsp-server/rtsp-address-pool.c:
5082 address-pool: small cleanups
5084 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5086 * tests/check/Makefile.am:
5087 * tests/check/gst/addresspool.c:
5088 tests: add addresspool unit test
5090 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5092 * gst/rtsp-server/Makefile.am:
5093 * gst/rtsp-server/rtsp-address-pool.c:
5094 * gst/rtsp-server/rtsp-address-pool.h:
5095 address-pool: add object to manage multicast addresses
5096 Make an object that can manage a rage of multicast addresses and ports.
5098 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5100 * gst/rtsp-server/rtsp-server.c:
5101 server: set default max-threads property
5103 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5105 * gst/rtsp-server/rtsp-media.c:
5106 media: wait for concurrent _prepare
5107 If a prepare is busy, wait for the result.
5109 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5111 * gst/rtsp-server/rtsp-media.c:
5112 media: add lock around message handler
5113 We don't want to dispatch messages while we are still processing the result of
5116 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5118 * gst/rtsp-server/rtsp-media.c:
5119 * gst/rtsp-server/rtsp-media.h:
5120 media: add lock to protect state changes
5122 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5124 * gst/rtsp-server/rtsp-stream.c:
5125 * gst/rtsp-server/rtsp-stream.h:
5128 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5130 * gst/rtsp-server/rtsp-stream-transport.c:
5131 * gst/rtsp-server/rtsp-stream-transport.h:
5132 * gst/rtsp-server/rtsp-stream.c:
5133 stream-transport: add keep-alive method
5135 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5137 * gst/rtsp-server/rtsp-stream-transport.c:
5138 * gst/rtsp-server/rtsp-stream-transport.h:
5139 * gst/rtsp-server/rtsp-stream.c:
5140 stream-transport: add method to handle RTP/RTCP
5141 Call new methods instead of poking into the structures directly.
5143 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5145 * gst/rtsp-server/rtsp-session-media.c:
5146 * gst/rtsp-server/rtsp-session-media.h:
5147 session-media: add locking
5149 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5151 * gst/rtsp-server/rtsp-session.c:
5152 * gst/rtsp-server/rtsp-session.h:
5153 session: add locking
5155 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5157 * gst/rtsp-server/rtsp-server.c:
5158 server: free old socket
5160 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5162 * gst/rtsp-server/rtsp-media-mapping.c:
5163 * gst/rtsp-server/rtsp-media-mapping.h:
5164 mapping: add locking
5166 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5168 * gst/rtsp-server/rtsp-media-factory.c:
5169 media-factory: add locking
5171 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5173 * gst/rtsp-server/rtsp-auth.c:
5174 * gst/rtsp-server/rtsp-auth.h:
5177 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5179 * gst/rtsp-server/rtsp-server.c:
5180 * gst/rtsp-server/rtsp-server.h:
5181 server: add max-thread property
5183 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5185 * gst/rtsp-server/rtsp-server.c:
5186 * gst/rtsp-server/rtsp-server.h:
5187 server: use a threadpool for the mainloops
5189 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5191 * gst/rtsp-server/rtsp-client.c:
5192 * gst/rtsp-server/rtsp-client.h:
5193 client: rename method
5194 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
5195 don't really create the client from the socket, we use the socket for the
5198 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5200 * gst/rtsp-server/rtsp-client.c:
5201 * gst/rtsp-server/rtsp-client.h:
5202 * gst/rtsp-server/rtsp-server.c:
5203 server: rework maincontext handling in clients
5204 Make a separate method to attach a client to a MainContext.
5205 Let the server decide in what GMainContext the client will operate and give this
5206 context to the client in attach. Then the server can later decide to use a
5207 separate thread for each client or just use the mainthread.
5209 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5211 * gst/rtsp-server/rtsp-client.c:
5212 * gst/rtsp-server/rtsp-session.c:
5213 * gst/rtsp-server/rtsp-session.h:
5214 session: move session header code in session object
5216 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
5220 * examples/test-auth.c:
5221 * examples/test-launch.c:
5222 * examples/test-mp4.c:
5223 * examples/test-ogg.c:
5224 * examples/test-readme.c:
5225 * examples/test-sdp.c:
5226 * examples/test-uri.c:
5227 * examples/test-video.c:
5228 * gst/rtsp-server/rtsp-auth.c:
5229 * gst/rtsp-server/rtsp-auth.h:
5230 * gst/rtsp-server/rtsp-client.c:
5231 * gst/rtsp-server/rtsp-client.h:
5232 * gst/rtsp-server/rtsp-media-factory-uri.c:
5233 * gst/rtsp-server/rtsp-media-factory-uri.h:
5234 * gst/rtsp-server/rtsp-media-factory.c:
5235 * gst/rtsp-server/rtsp-media-factory.h:
5236 * gst/rtsp-server/rtsp-media-mapping.c:
5237 * gst/rtsp-server/rtsp-media-mapping.h:
5238 * gst/rtsp-server/rtsp-media.c:
5239 * gst/rtsp-server/rtsp-media.h:
5240 * gst/rtsp-server/rtsp-params.c:
5241 * gst/rtsp-server/rtsp-params.h:
5242 * gst/rtsp-server/rtsp-sdp.c:
5243 * gst/rtsp-server/rtsp-sdp.h:
5244 * gst/rtsp-server/rtsp-server.c:
5245 * gst/rtsp-server/rtsp-server.h:
5246 * gst/rtsp-server/rtsp-session-media.c:
5247 * gst/rtsp-server/rtsp-session-media.h:
5248 * gst/rtsp-server/rtsp-session-pool.c:
5249 * gst/rtsp-server/rtsp-session-pool.h:
5250 * gst/rtsp-server/rtsp-session.c:
5251 * gst/rtsp-server/rtsp-session.h:
5252 * gst/rtsp-server/rtsp-stream-transport.c:
5253 * gst/rtsp-server/rtsp-stream-transport.h:
5254 * gst/rtsp-server/rtsp-stream.c:
5255 * gst/rtsp-server/rtsp-stream.h:
5256 * tests/check/gst/rtspserver.c:
5257 * tests/test-cleanup.c:
5260 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
5262 * gst/rtsp-server/rtsp-media.c:
5263 * gst/rtsp-server/rtsp-session-media.c:
5264 * gst/rtsp-server/rtsp-session.c:
5265 rtsp-server: added annotations to indicate type of ownership transfer of return values
5266 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5268 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
5271 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
5273 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
5276 * bindings/Makefile.am:
5277 * bindings/vala/Makefile.am:
5278 * bindings/vala/gst-rtsp-server-0.10.deps:
5279 * bindings/vala/gst-rtsp-server-0.10.vapi:
5280 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
5281 * bindings/vala/packages/gst-rtsp-server-0.10.files:
5282 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
5283 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5284 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
5286 bindings: remove vala bindings
5287 They'll be reunited with the other GStreamer bindings
5288 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5290 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5292 * gst/rtsp-server/rtsp-client.c:
5293 * gst/rtsp-server/rtsp-session-media.c:
5294 * gst/rtsp-server/rtsp-session-media.h:
5295 * gst/rtsp-server/rtsp-stream-transport.c:
5296 * gst/rtsp-server/rtsp-stream-transport.h:
5297 rtsp: only create transport when needed
5298 Only create the StreamTransport when configured.
5300 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5302 * gst/rtsp-server/rtsp-client.c:
5303 client: small cleanup
5305 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5307 * gst/rtsp-server/rtsp-client.c:
5308 * gst/rtsp-server/rtsp-client.h:
5309 * gst/rtsp-server/rtsp-stream-transport.c:
5310 * gst/rtsp-server/rtsp-stream-transport.h:
5311 rtsp: refactor configuration of transport
5312 Move the configuration of the transport to a place where it makes
5315 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5317 * gst/rtsp-server/rtsp-client.c:
5318 client: refactor transport parsing
5320 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5322 * gst/rtsp-server/rtsp-client.c:
5323 client: refuse to change the MTU on shared media
5324 If we change the MTU of chared media, it changes for all clients.
5325 We don't want to set the MTU to something large for clients that
5328 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5330 * examples/test-mp4.c:
5331 * gst/rtsp-server/rtsp-media.c:
5332 small fixes to docs and debug
5334 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5336 * gst/rtsp-server/rtsp-stream.c:
5337 stream: transports must already have been removed
5339 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5341 * gst/rtsp-server/rtsp-media.c:
5342 * gst/rtsp-server/rtsp-stream.c:
5343 * gst/rtsp-server/rtsp-stream.h:
5344 stream: improve join and leave of the pipeline
5346 Do the cleanup properly
5349 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5351 * gst/rtsp-server/rtsp-media.c:
5352 media: move unprepare below default implementation
5353 Makes it easier to find the default implementation
5355 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5357 * gst/rtsp-server/rtsp-media.c:
5358 media: signal unprepared when we actually finish
5360 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5362 * gst/rtsp-server/rtsp-media.c:
5363 media: no need to unlock, unprepare does that when needed
5365 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5367 * docs/libs/gst-rtsp-server-sections.txt:
5368 * gst/rtsp-server/rtsp-media-factory.h:
5369 * gst/rtsp-server/rtsp-media-mapping.c:
5370 * gst/rtsp-server/rtsp-media.h:
5371 * gst/rtsp-server/rtsp-params.c:
5372 * gst/rtsp-server/rtsp-server.c:
5373 * gst/rtsp-server/rtsp-session-pool.h:
5374 * gst/rtsp-server/rtsp-session.c:
5375 * gst/rtsp-server/rtsp-session.h:
5376 * gst/rtsp-server/rtsp-stream-transport.h:
5377 * gst/rtsp-server/rtsp-stream.h:
5380 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5382 * gst/rtsp-server/rtsp-client.c:
5383 * gst/rtsp-server/rtsp-media-mapping.h:
5384 * gst/rtsp-server/rtsp-media.c:
5385 * gst/rtsp-server/rtsp-media.h:
5386 * gst/rtsp-server/rtsp-server.h:
5387 * gst/rtsp-server/rtsp-stream.c:
5388 * gst/rtsp-server/rtsp-stream.h:
5389 rtsp: fix MTU setting
5390 Fix setting of the MTU. There is no need for a vmethod.
5392 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5397 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5400 configure: bump version number after refactoring
5402 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5404 * gst/rtsp-server/Makefile.am:
5405 * gst/rtsp-server/rtsp-client.c:
5406 * gst/rtsp-server/rtsp-client.h:
5407 * gst/rtsp-server/rtsp-media-factory-uri.c:
5408 * gst/rtsp-server/rtsp-media-factory.c:
5409 * gst/rtsp-server/rtsp-media-factory.h:
5410 * gst/rtsp-server/rtsp-media.c:
5411 * gst/rtsp-server/rtsp-media.h:
5412 * gst/rtsp-server/rtsp-sdp.c:
5413 * gst/rtsp-server/rtsp-session-media.c:
5414 * gst/rtsp-server/rtsp-session-media.h:
5415 * gst/rtsp-server/rtsp-session.c:
5416 * gst/rtsp-server/rtsp-session.h:
5417 * gst/rtsp-server/rtsp-stream-transport.c:
5418 * gst/rtsp-server/rtsp-stream-transport.h:
5419 * gst/rtsp-server/rtsp-stream.c:
5420 * gst/rtsp-server/rtsp-stream.h:
5421 rtsp: massive refactoring
5422 Make GObjects from the remaining simple structures.
5423 Remove GstRTSPSessionStream, it's not needed.
5424 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
5425 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
5426 a GstRTSPStream should be transported to a client.
5427 Rename GstRTSPMediaFactory::get_element -> create_element because that
5428 more accurately describes what it does.
5429 Make nice methods instead of poking in the structures.
5430 Move some methods inside the relevant object source code.
5431 Use GPtrArray to store objects instead of plain arrays, it is more
5432 natural and allows us to more easily clean up.
5433 Move the allocation of udp ports to the Stream object. The Stream object
5434 contains the elements needed to stream the media to a client.
5435 Improve the prepare and unprepare methods. Unprepare should now undo
5436 everything prepare did. Improve also async unprepare when doing EOS on
5437 shutdown. Make sure we always unprepare correctly.
5439 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
5441 * gst/rtsp-server/rtsp-client.c:
5442 rtsp-client: Unref server address clients connected to
5443 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
5445 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
5447 * gst/rtsp-server/rtsp-server.c:
5448 rtsp-server: don't ref server socket if it is NULL
5449 Fixes test_bind_already_in_use unit test again after commit 6a497440.
5450 https://bugzilla.gnome.org/show_bug.cgi?id=686644
5452 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
5454 * tests/check/Makefile.am:
5455 tests: Add libgio link dependency
5456 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
5458 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5460 * gst/rtsp-server/rtsp-media-mapping.c:
5461 * gst/rtsp-server/rtsp-media-mapping.h:
5462 rtsp-media-mapping: rename find_media vfunc to find_factory
5463 The virtual method and class method should have the same name
5464 so it is correctly represented in GIR file
5465 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5467 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5469 * gst/rtsp-server/rtsp-auth.c:
5470 * gst/rtsp-server/rtsp-client.c:
5471 * gst/rtsp-server/rtsp-media-factory-uri.c:
5472 * gst/rtsp-server/rtsp-media-factory.c:
5473 * gst/rtsp-server/rtsp-media-mapping.c:
5474 * gst/rtsp-server/rtsp-media.c:
5475 * gst/rtsp-server/rtsp-server.c:
5476 * gst/rtsp-server/rtsp-session-pool.c:
5477 * gst/rtsp-server/rtsp-session.c:
5478 rtsp-server: fixed comments and GIR annotations
5479 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5481 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
5483 * gst/rtsp-server/rtsp-media-mapping.c:
5484 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
5486 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
5488 * gst/rtsp-server/rtsp-server.c:
5489 rtsp-server: allow binding on port 0 (binds on a random port)
5491 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
5493 * gst/rtsp-server/rtsp-server.c:
5494 * gst/rtsp-server/rtsp-server.h:
5495 rtsp-server: add bound-port property
5496 bound-port can be used to retrieve the port number when the server is bound on
5497 port 0, which binds on a random port.
5499 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
5501 * gst/rtsp-server/rtsp-media-factory.c:
5502 * gst/rtsp-server/rtsp-media-factory.h:
5503 rtsp-media-factory: make ::get_element overridable by GI bindings
5504 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
5505 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
5506 as the invoker for ::get_element(), making it overridable by GI generated
5509 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5511 * gst/rtsp-server/rtsp-media-factory-uri.c:
5512 rtsp-media-factory-uri: don't autoplug parsers in a loop
5513 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
5516 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5518 * gst/rtsp-server/Makefile.am:
5519 Explicitly link against gio. Fix link error on mac.
5521 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5523 * gst/rtsp-server/rtsp-session.c:
5524 session: add ttl to the transport header in SETUP
5525 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
5527 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5529 * gst/rtsp-server/rtsp-client.c:
5530 * gst/rtsp-server/rtsp-client.h:
5531 * gst/rtsp-server/rtsp-media.c:
5532 client: Use client transport settings for multicast if allowed.
5533 This patch makes it possible for the client to send transport settings for
5534 multicast (destination && ttl). Client settings must be explicitly allowed or
5535 the server will use its own settings.
5536 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
5538 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
5541 Automatic update of common submodule
5542 From 6c0b52c to 6bb6951
5544 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
5546 * gst/rtsp-server/rtsp-client.c:
5547 rtsp-client: do not destroy the rtsp watch
5548 Don't destroy the client watch while dispatching. The rtsp watch is
5549 automatically destroyed after the rtsp watch function closed() has
5551 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
5553 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5556 Automatic update of common submodule
5557 From 4f962f7 to 6c0b52c
5559 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
5561 * gst/rtsp-server/rtsp-media.c:
5562 media: fix check for seekability
5564 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5566 * gst/rtsp-server/rtsp-client.c:
5567 client: use more GIO
5568 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
5570 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5572 * gst/rtsp-server/rtsp-server.c:
5573 server: remove obsolete includes
5575 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5577 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
5578 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
5579 be available in "on_new_ssrc". The transports are added in
5580 gst_rtsp_media_set_state when going to PLAYING state. However,
5581 "on_new_ssrc" might be called before this happens.
5582 https://bugzilla.gnome.org/show_bug.cgi?id=683304
5584 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5586 * gst/rtsp-server/rtsp-client.c:
5587 * gst/rtsp-server/rtsp-client.h:
5588 rtsp-client: add signals for rtsp requests (fixes #683287)
5590 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5592 * gst/rtsp-server/rtsp-client.c:
5593 * gst/rtsp-server/rtsp-client.h:
5594 add new-session signal to rtsp-client (fixes #683058)
5596 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
5599 Automatic update of common submodule
5600 From 668acee to 4f962f7
5602 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
5604 * gst/rtsp-server/rtsp-server.c:
5605 * tests/check/gst/rtspserver.c:
5606 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
5607 Do not assume that *error is set in g_socket_address_enumerator_next.
5608 Added test_bind_already_in_use unit-test.
5609 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
5611 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
5614 Automatic update of common submodule
5615 From 94ccf4c to 668acee
5617 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
5619 * gst/rtsp-server/rtsp-client.c:
5620 * gst/rtsp-server/rtsp-client.h:
5621 rtsp-client: make create_sdp virtual method
5622 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
5624 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5627 Automatic update of common submodule
5628 From 98e386f to 94ccf4c
5630 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5632 * gst/rtsp-server/rtsp-client.c:
5635 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5637 * gst/rtsp-server/rtsp-client.c:
5638 * gst/rtsp-server/rtsp-client.h:
5639 * gst/rtsp-server/rtsp-server.c:
5640 * gst/rtsp-server/rtsp-server.h:
5641 rtsp-server: use an existing socket to establish HTTP tunnel
5642 Make it possible to transfer a socket from an HTTP server to be used as
5643 an RTSP over HTTP tunnel.
5645 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
5647 * gst/rtsp-server/rtsp-client.c:
5648 * gst/rtsp-server/rtsp-media.c:
5649 * gst/rtsp-server/rtsp-media.h:
5650 rtsp: Handle the blocksize parameter
5651 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
5653 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
5655 * tests/check/Makefile.am:
5656 * tests/check/gst/rtspserver.c:
5657 Have unit test get header from source dir, not installed dir
5658 This makes compilation of unit tests work in a build directory other
5659 than the source directory.
5660 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
5662 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
5664 * gst/rtsp-server/rtsp-media.c:
5665 rtsp-media: update for gst_element_make_from_uri() changes
5667 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
5670 * tests/Makefile.am:
5671 * tests/check/Makefile.am:
5672 * tests/check/gst/rtspserver.c:
5674 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
5676 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
5678 * gst/rtsp-server/rtsp-media.c:
5679 rtsp-media: don't collect media stats when going to NULL
5680 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
5682 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5684 * gst/rtsp-server/rtsp-client.c:
5685 client: don't leak transports
5687 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
5689 * gst/rtsp-server/rtsp-client.c:
5690 rtsp-client: free transport on no_stream in SETUP handler
5692 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
5694 * gst/rtsp-server/rtsp-client.c:
5695 rtsp-client: changed session media iteration
5696 In client_unlink_session: now don't iterate in session->medias
5697 list where items are removed by gst_rtsp_session_release_media.
5698 Instead, repeatedly remove the first item.
5700 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
5702 * gst/rtsp-server/rtsp-client.c:
5703 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
5704 GstRTSPSessionMedia is not a GObject type. When the
5705 GstRTSPSession is freed, it will free the media.
5707 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
5709 * gst/rtsp-server/rtsp-media-factory.c:
5710 factory: plug pad leak in collect_streams
5711 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
5712 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
5713 will take one reference, and the other reference will otherwise
5716 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5719 configure: suppress some warnings when debug is disabled
5720 Warnings about unused variables should be suppressed if core has the
5721 debug system disabled.
5722 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5724 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5726 * docs/libs/Makefile.am:
5727 docs: fix build in uninstalled setup
5728 Include gst-plugins-base libs properly.
5730 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
5732 * docs/libs/gst-rtsp-server.types:
5733 docs: include headers defining rtsp-server object types
5734 Fixes compiler warnings during docs build.
5735 https://bugzilla.gnome.org/show_bug.cgi?id=676824
5737 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
5740 configure: Add warning flags for compiler when configuring
5741 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5743 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5746 Automatic update of common submodule
5747 From 03a0e57 to 98e386f
5749 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5752 Automatic update of common submodule
5753 From 1fab359 to 03a0e57
5755 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
5757 * gst/rtsp-server/rtsp-client.c:
5758 client: fix GSocketAddress leak in gst_rtsp_client_accept
5759 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
5761 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5764 Automatic update of common submodule
5765 From f1b5a96 to 1fab359
5767 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5770 Automatic update of common submodule
5771 From 92b7266 to f1b5a96
5773 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5776 Automatic update of common submodule
5777 From ec1c4a8 to 92b7266
5779 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5782 Automatic update of common submodule
5783 From 3429ba6 to ec1c4a8
5785 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
5787 * gst/rtsp-server/rtsp-auth.c:
5788 * gst/rtsp-server/rtsp-client.c:
5789 * gst/rtsp-server/rtsp-media-factory-uri.c:
5790 * gst/rtsp-server/rtsp-server.c:
5791 rtsp: fix compiler warnings
5792 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
5794 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5797 Automatic update of common submodule
5798 From dc70203 to 3429ba6
5800 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5802 * gst/rtsp-server/rtsp-client.c:
5803 * gst/rtsp-server/rtsp-media-factory.c:
5804 * gst/rtsp-server/rtsp-media-factory.h:
5805 * gst/rtsp-server/rtsp-media.c:
5806 * gst/rtsp-server/rtsp-media.h:
5807 * gst/rtsp-server/rtsp-server.c:
5808 * gst/rtsp-server/rtsp-server.h:
5809 * gst/rtsp-server/rtsp-session-pool.c:
5810 * gst/rtsp-server/rtsp-session-pool.h:
5811 rtsp-server: port to new thread API
5813 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5816 Automatic update of common submodule
5817 From 6db25be to dc70203
5819 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5821 * gst/rtsp-server/rtsp-auth.c:
5822 * gst/rtsp-server/rtsp-auth.h:
5823 * gst/rtsp-server/rtsp-client.c:
5824 rtsp-server: Fix compilation and compiler warnings
5826 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5830 * gst/rtsp-server/Makefile.am:
5831 configure: Modernize autotools setup a bit
5832 Also we now only create tar.bz2 and tar.xz tarballs.
5834 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5837 Automatic update of common submodule
5838 From 464fe15 to 6db25be
5840 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5843 Automatic update of common submodule
5844 From 7fda524 to 464fe15
5846 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5849 * docs/libs/Makefile.am:
5850 * docs/version.entities.in:
5852 * gst/rtsp-server/Makefile.am:
5853 * pkgconfig/Makefile.am:
5854 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5855 * pkgconfig/gstreamer-rtsp-server.pc.in:
5856 * tests/Makefile.am:
5857 rtsp-server: Update versioning
5859 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5861 Merge remote-tracking branch 'origin/0.10'
5863 gst/rtsp-server/rtsp-session-pool.c
5865 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5867 * gst/rtsp-server/rtsp-session-pool.c:
5868 rtsp-server: Don't use deprecated GLib API
5870 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5872 Replace master with 0.11
5874 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5876 Merge branch 'master' into 0.11
5878 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5880 Merge branch 'master' into 0.11
5882 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
5885 A couple minor typo fixes
5887 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5889 * gst/rtsp-server/rtsp-media.c:
5890 media: fix state of the appqueue
5892 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5894 * gst/rtsp-server/rtsp-media-factory-uri.c:
5895 factory: use videoconvert
5897 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5899 * gst/rtsp-server/rtsp-media-factory-uri.c:
5900 factory: change to new style caps
5902 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5904 * gst/rtsp-server/rtsp-client.c:
5905 * gst/rtsp-server/rtsp-client.h:
5906 * gst/rtsp-server/rtsp-media-factory-uri.c:
5907 * gst/rtsp-server/rtsp-media.c:
5908 * gst/rtsp-server/rtsp-server.c:
5909 * gst/rtsp-server/rtsp-server.h:
5910 * gst/rtsp-server/rtsp-session-pool.c:
5911 rtsp-server: port to GIO
5914 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5917 configure: fix build
5919 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5922 docs: fix for gst_rtsp_server_set_port() -> _set_service()
5923 https://bugzilla.gnome.org/show_bug.cgi?id=666548
5925 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5928 * examples/Makefile.am:
5929 First rule of gst-rtsp-server club: don't talk about gst-phonon
5931 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5934 * pkgconfig/Makefile.am:
5935 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
5936 * pkgconfig/gst-rtsp-server.pc.in:
5937 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5938 * pkgconfig/gstreamer-rtsp-server.pc.in:
5939 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
5940 For consistency with all other modules.
5942 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5944 * gst/rtsp-server/rtsp-client.c:
5945 rtsp-client: update for new map API
5947 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5950 * bindings/Makefile.am:
5951 * bindings/python/Makefile.am:
5952 * bindings/python/arg-types.py:
5953 * bindings/python/codegen/Makefile.am:
5954 * bindings/python/codegen/__init__.py:
5955 * bindings/python/codegen/argtypes.py:
5956 * bindings/python/codegen/code-coverage.py:
5957 * bindings/python/codegen/codegen.py:
5958 * bindings/python/codegen/definitions.py:
5959 * bindings/python/codegen/defsparser.py:
5960 * bindings/python/codegen/docextract.py:
5961 * bindings/python/codegen/docgen.py:
5962 * bindings/python/codegen/fileprefix.override:
5963 * bindings/python/codegen/fileprefixmodule.c:
5964 * bindings/python/codegen/h2def.py:
5965 * bindings/python/codegen/mergedefs.py:
5966 * bindings/python/codegen/mkskel.py:
5967 * bindings/python/codegen/override.py:
5968 * bindings/python/codegen/reversewrapper.py:
5969 * bindings/python/codegen/scmexpr.py:
5970 * bindings/python/rtspserver-types.defs:
5971 * bindings/python/rtspserver.defs:
5972 * bindings/python/rtspserver.override:
5973 * bindings/python/rtspservermodule.c:
5974 * bindings/python/test.py:
5976 python: remove pygst-based python bindings
5977 pygi is the future, apparently.
5979 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
5982 Automatic update of common submodule
5983 From c463bc0 to 7fda524
5985 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5988 Automatic update of common submodule
5989 From 2a59016 to c463bc0
5991 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5994 Automatic update of common submodule
5995 From 0807187 to 2a59016
5997 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6000 Automatic update of common submodule
6001 From 11f0cd5 to 0807187
6003 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6005 * examples/test-auth.c:
6006 example: update for new caps
6008 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6010 * examples/test-video.c:
6011 * gst/rtsp-server/rtsp-client.c:
6012 * gst/rtsp-server/rtsp-media-factory-uri.c:
6013 * gst/rtsp-server/rtsp-media.c:
6014 * gst/rtsp-server/rtsp-media.h:
6015 * gst/rtsp-server/rtsp-session.c:
6016 * gst/rtsp-server/rtsp-session.h:
6017 rtsp-server: port some more to 0.11
6019 Remove bufferlist stuff
6021 Add queue before appsink now that preroll-queue-len is gone.
6022 Update for request pad changes.
6024 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6026 Merge branch 'master' into 0.11
6028 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6030 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6031 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6032 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6034 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
6036 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6037 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
6038 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6040 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6042 Merge branch 'master' into 0.11
6044 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6046 * gst/rtsp-server/rtsp-media.c:
6047 * gst/rtsp-server/rtsp-media.h:
6048 media: add a seekable boolean
6049 Maintain the seekable state with a new variable instead of reusing the
6052 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
6054 * gst/rtsp-server/rtsp-media.c:
6055 Disallow seek in live media
6057 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6059 Merge branch 'master' into 0.11
6061 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
6063 * gst/rtsp-server/rtsp-server.c:
6064 #ifdef statements for windows socket creation were missing
6066 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
6069 Automatic update of common submodule
6070 From a39eb83 to 11f0cd5
6072 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
6075 Automatic update of common submodule
6076 From 605cd9a to a39eb83
6078 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6080 Merge branch 'master' into 0.11
6082 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6084 * gst/rtsp-server/rtsp-client.c:
6085 client: use method to access property
6087 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6089 * gst/rtsp-server/rtsp-media-factory.c:
6090 * gst/rtsp-server/rtsp-media-factory.h:
6091 media-factory: add protocols property
6092 Add a property to configure the allowed protocols in the media created from the
6095 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6097 * gst/rtsp-server/rtsp-media-factory.c:
6098 * gst/rtsp-server/rtsp-media-factory.h:
6099 media-factory: add media-configure signal
6100 Add signal to allow the application to configure the media after it was created
6103 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6105 * gst/rtsp-server/rtsp-client.c:
6106 client: use method to access property
6108 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6110 * gst/rtsp-server/rtsp-media-factory.c:
6111 * gst/rtsp-server/rtsp-media-factory.h:
6112 media-factory: add protocols property
6113 Add a property to configure the allowed protocols in the media created from the
6116 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6118 * gst/rtsp-server/rtsp-media-factory.c:
6119 * gst/rtsp-server/rtsp-media-factory.h:
6120 media-factory: add media-configure signal
6121 Add signal to allow the application to configure the media after it was created
6124 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6126 Merge branch 'master' into 0.11
6128 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6130 * gst/rtsp-server/rtsp-client.c:
6131 client: use media multicast group
6133 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6135 * gst/rtsp-server/rtsp-media-factory.h:
6136 * gst/rtsp-server/rtsp-server.h:
6137 * gst/rtsp-server/rtsp-session-pool.h:
6138 * gst/rtsp-server/rtsp-session.h:
6141 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6143 * gst/rtsp-server/rtsp-client.c:
6144 * gst/rtsp-server/rtsp-sdp.h:
6145 sdp: copy and free the server ip address
6146 Copy and free the server ip address to make memory management easier later.
6148 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6150 * gst/rtsp-server/rtsp-media-factory.c:
6151 media-factory: configure multicast in media
6153 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6155 * gst/rtsp-server/rtsp-media.c:
6156 * gst/rtsp-server/rtsp-media.h:
6157 media: add property for multicast group
6158 Add a property to configure the multicast group in the media.
6159 Based on patches from Marc Leeman and Robert Krakora.
6161 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6163 * gst/rtsp-server/rtsp-media-factory.c:
6164 * gst/rtsp-server/rtsp-media-factory.h:
6165 media-factory: add property for multicast group
6166 Add a property to configure the multicast group in the media factory.
6167 Based on patches from Marc Leeman and Robert Krakora.
6169 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6171 * gst/rtsp-server/rtsp-client.c:
6172 client: do configuration of transport in one place
6173 Move the configuration of the transport destination address to where we also
6174 configure the other bits.
6176 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6178 * gst/rtsp-server/rtsp-client.c:
6179 client: use media multicast group
6181 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6183 * gst/rtsp-server/rtsp-media-factory.h:
6184 * gst/rtsp-server/rtsp-server.h:
6185 * gst/rtsp-server/rtsp-session-pool.h:
6186 * gst/rtsp-server/rtsp-session.h:
6189 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6191 * gst/rtsp-server/rtsp-client.c:
6192 * gst/rtsp-server/rtsp-sdp.h:
6193 sdp: copy and free the server ip address
6194 Copy and free the server ip address to make memory management easier later.
6196 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6198 * gst/rtsp-server/rtsp-media-factory.c:
6199 media-factory: configure multicast in media
6201 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6203 * gst/rtsp-server/rtsp-media.c:
6204 * gst/rtsp-server/rtsp-media.h:
6205 media: add property for multicast group
6206 Add a property to configure the multicast group in the media.
6207 Based on patches from Marc Leeman and Robert Krakora.
6209 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6211 * gst/rtsp-server/rtsp-media-factory.c:
6212 * gst/rtsp-server/rtsp-media-factory.h:
6213 media-factory: add property for multicast group
6214 Add a property to configure the multicast group in the media factory.
6215 Based on patches from Marc Leeman and Robert Krakora.
6217 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6219 * gst/rtsp-server/rtsp-client.c:
6220 client: do configuration of transport in one place
6221 Move the configuration of the transport destination address to where we also
6222 configure the other bits.
6224 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6226 Merge branch 'master' into 0.11
6228 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6230 * gst/rtsp-server/rtsp-client.c:
6231 client: destroy pipeline on client disconnect with no prior TEARDOWN.
6232 The problem occurs when the client abruptly closes the connection without
6233 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
6234 server is where the pipeline gets torn down. Since this handler is not called,
6235 the pipeline remains and is up and running. Subsequent clients get their own
6236 pipelines and if the do not issue TEARDOWNs then those pipelines will also
6237 remain up and running. This is a resource leak.
6239 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6241 Merge branch 'master' into 0.11
6243 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
6245 * gst/rtsp-server/rtsp-media-factory.c:
6246 * gst/rtsp-server/rtsp-media-factory.h:
6247 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
6248 For example, it can be used to retrieve source elements like appsrc, in a more
6249 convenient way than subclassing get_element.
6251 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6253 Merge branch 'master' into 0.11
6255 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
6257 * gst/rtsp-server/rtsp-server.c:
6258 rtsp-server: hold on to reference while using object
6260 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6262 * gst/rtsp-server/rtsp-media.c:
6265 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6268 configure: use unstable api
6270 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
6272 * gst/rtsp-server/rtsp-client.c:
6273 client: fix reference counting
6275 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
6277 * gst/rtsp-server/rtsp-client.c:
6278 * gst/rtsp-server/rtsp-media.c:
6279 fix compiler warnings about unused variables
6281 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
6283 * examples/test-launch.c:
6284 * examples/test-readme.c:
6285 * examples/test-uri.c:
6286 * examples/test-video.c:
6287 examples: tell rtsp uri when ready
6289 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
6292 Automatic update of common submodule
6293 From 69b981f to 605cd9a
6295 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6297 * gst/rtsp-server/rtsp-client.c:
6298 client: update for buffer API change
6300 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6302 * gst/rtsp-server/Makefile.am:
6303 Makefile.am: 0.10 => @GST_MAJORMINOR@
6305 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6307 * gst/rtsp-server/rtsp-media-factory-uri.c:
6308 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
6310 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6312 * gst/rtsp-server/.gitignore:
6313 .gitignore: 0.10 => 0.11
6315 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6317 * gst/rtsp-server/Makefile.am:
6318 Makefile.am: 0.10 => @GST_MAJORMINOR@
6320 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6322 Merge branch 'master' into 0.11
6324 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
6327 Automatic update of common submodule
6328 From 9e5bbd5 to 69b981f
6330 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
6333 Automatic update of common submodule
6334 From fd35073 to 9e5bbd5
6336 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
6339 Automatic update of common submodule
6340 From 46dfcea to fd35073
6342 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6344 * gst/rtsp-server/rtsp-media-factory-uri.c:
6345 * gst/rtsp-server/rtsp-media.c:
6346 media: port to new caps API
6348 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6350 Merge branch 'master' into 0.11
6352 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6354 * bindings/vala/gst-rtsp-server-0.10.vapi:
6355 Updated Vala bindings.
6356 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6358 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6360 * gst/rtsp-server/rtsp-server.c:
6361 * gst/rtsp-server/rtsp-server.h:
6362 Add a signal for newly connected clients.
6363 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6365 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6367 * bindings/python/rtspserver.override:
6368 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
6370 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6372 * gst/rtsp-server/Makefile.am:
6373 * gst/rtsp-server/rtsp-client.c:
6374 * gst/rtsp-server/rtsp-funnel.c:
6375 * gst/rtsp-server/rtsp-funnel.h:
6376 * gst/rtsp-server/rtsp-media.c:
6377 rtsp-server: port to 0.11
6379 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6384 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6386 Merge branch 'master' into 0.11
6391 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6394 Automatic update of common submodule
6395 From c3cafe1 to 46dfcea
6397 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
6399 * bindings/python/Makefile.am:
6400 * bindings/python/rtspserver.defs:
6401 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
6403 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
6405 * bindings/python/arg-types.py:
6406 python bindings: add GstRTSPUrlParam
6407 Needed to implement MediaFactory virtual proxies
6409 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
6411 * bindings/python/arg-types.py:
6412 python bindings: fix returning GstRTSPUrl types
6414 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6416 * bindings/python/arg-types.py:
6417 python bindings: add arg type for GstRTSPUrl
6419 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
6421 * bindings/python/rtspserver.defs:
6422 python bindings: fix the definition of MediaFactory.collect_stream
6424 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
6427 Automatic update of common submodule
6428 From 1ccbe09 to c3cafe1
6430 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6433 Automatic update of common submodule
6434 From 193b717 to 1ccbe09
6436 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
6439 Automatic update of common submodule
6440 From b77e2bf to 193b717
6442 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6445 build: Include lcov.mak to allow test coverage report generation
6447 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6450 Automatic update of common submodule
6451 From d8814b6 to b77e2bf
6453 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6456 Automatic update of common submodule
6457 From 6aaa286 to d8814b6
6459 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
6462 Automatic update of common submodule
6463 From 6aec6b9 to 6aaa286
6465 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
6468 autogen: wingo signed comment
6470 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
6472 * gst/rtsp-server/rtsp-session-pool.c:
6473 session: use full charset for RTSP session ID
6474 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
6475 session ID more difficult.
6476 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6478 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6480 * gst/rtsp-server/Makefile.am:
6481 rtsp-server: Don't install the funnel header
6483 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6486 Automatic update of common submodule
6487 From 1de7f6a to 6aec6b9
6489 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6492 configure: require core/base 0.10.31
6493 Needed at least for gst_plugin_feature_rank_compare_func().
6495 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
6498 Automatic update of common submodule
6499 From f94d739 to 1de7f6a
6501 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6503 * gst/rtsp-server/rtsp-media.c:
6504 media: remove more unused code
6506 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6508 * gst/rtsp-server/rtsp-media.c:
6509 * gst/rtsp-server/rtsp-media.h:
6510 media: remove duplicate filtering
6511 Remove the duplicate filtering code now that we have a released -good version.
6512 Give a warning instead.
6514 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6516 * gst/rtsp-server/rtsp-media-factory.c:
6517 * gst/rtsp-server/rtsp-media.c:
6518 media: fix default buffer size
6520 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6522 * gst/rtsp-server/rtsp-media-factory.c:
6523 * gst/rtsp-server/rtsp-media-factory.h:
6524 media-factory: add property to configure the buffer-size
6525 Add a property to configure the kernel UDP buffer size.
6527 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6529 * gst/rtsp-server/rtsp-media.c:
6530 * gst/rtsp-server/rtsp-media.h:
6531 media: add property to configure kernel buffer sizes
6532 Add a property to configure the kernel UDP buffer size.
6534 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6537 configure: set PYGOBJECT_REQ before using it
6538 https://bugzilla.gnome.org/show_bug.cgi?id=640641
6540 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6543 docs: recursive into sub-directories on 'make upload'
6545 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6547 * docs/libs/gst-rtsp-server-docs.sgml:
6548 * docs/version.entities.in:
6549 docs: mention full version these docs are for, not just major-minor
6551 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6556 === release 0.10.8 ===
6558 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6563 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6565 * gst/rtsp-server/rtsp-server.c:
6566 rtsp-server: clarify docs a little
6568 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6570 * gst/rtsp-server/rtsp-media.c:
6571 media: init debug category before starting thread
6573 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6575 * gst/rtsp-server/rtsp-auth.c:
6576 auth: add realm to make it more spec compliant
6578 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6580 * gst/rtsp-server/rtsp-server.c:
6581 * gst/rtsp-server/rtsp-server.h:
6584 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6586 * examples/test-video.c:
6587 example: improve example docs a little
6589 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6591 * gst/rtsp-server/rtsp-server.c:
6592 server: ensure the watch has a ref to the server
6594 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6596 * gst/rtsp-server/rtsp-server.c:
6597 server: simpify channel function
6599 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6601 * gst/rtsp-server/rtsp-server.c:
6602 * gst/rtsp-server/rtsp-server.h:
6603 server: simplify management of channel and source
6604 We don't need to keep around the channel and source objects. Let the mainloop
6605 and the source manage the source and channel respectively.
6607 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6613 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6616 * tests/Makefile.am:
6617 * tests/test-cleanup.c:
6618 tests: add tests directory and cleanup test
6620 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6622 * gst/rtsp-server/rtsp-media-factory-uri.c:
6623 * gst/rtsp-server/rtsp-media-factory.c:
6624 * gst/rtsp-server/rtsp-media-mapping.c:
6625 * gst/rtsp-server/rtsp-media.c:
6626 * gst/rtsp-server/rtsp-session-pool.c:
6627 * gst/rtsp-server/rtsp-session.c:
6628 server: improve debugging in various objects
6630 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6632 * gst/rtsp-server/rtsp-server.c:
6633 server: chain up to the parent finalize
6635 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
6637 * bindings/python/rtspserver-types.defs:
6638 * bindings/python/rtspserver.defs:
6639 * bindings/python/rtspserver.override:
6640 * bindings/python/test.py:
6641 gst-rtsp-server: update python bindings
6643 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6645 * gst/rtsp-server/rtsp-client.c:
6646 client: use the response from the clientstate
6647 Create the response object only once and store in the client state.
6648 Make all methods use the state response,
6650 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6652 * gst/rtsp-server/rtsp-server.c:
6653 server: use signal to keep track of clients
6654 Keep track of all the clients that the server creates and remove them when they
6655 fire the 'closed' signal.
6657 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6659 * gst/rtsp-server/rtsp-client.c:
6660 * gst/rtsp-server/rtsp-client.h:
6661 client: emit signal when closing
6663 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6665 * examples/.gitignore:
6666 * examples/Makefile.am:
6667 * examples/test-auth.c:
6668 * examples/test-video.c:
6669 * gst/rtsp-server/rtsp-auth.c:
6670 * gst/rtsp-server/rtsp-auth.h:
6671 * gst/rtsp-server/rtsp-client.c:
6672 * gst/rtsp-server/rtsp-media-factory.c:
6673 * gst/rtsp-server/rtsp-media.c:
6674 * gst/rtsp-server/rtsp-media.h:
6675 * gst/rtsp-server/rtsp-session-pool.h:
6676 * gst/rtsp-server/rtsp-session.h:
6677 media: enable per factory authorisations
6678 Allow for adding a GstRTSPAuth on the factory and media level and check
6679 permissions when accessing the factory.
6680 Add hints to the auth methods for future more fine grained authorisation.
6681 Add example application for per factory authentication.
6683 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6685 * gst/rtsp-server/rtsp-auth.c:
6686 * gst/rtsp-server/rtsp-auth.h:
6687 * gst/rtsp-server/rtsp-client.c:
6688 * gst/rtsp-server/rtsp-client.h:
6689 * gst/rtsp-server/rtsp-params.c:
6690 * gst/rtsp-server/rtsp-params.h:
6691 rtsp-server: Pass ClientState structure arround
6692 Pass the collected information for the ongoing request in a GstRTSPClientState
6693 structure that we can then pass around to simplify the method arguments. This
6694 will also be handy when we implement logging functionality.
6696 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6698 * gst/rtsp-server/rtsp-media-factory.c:
6699 * gst/rtsp-server/rtsp-media-factory.h:
6700 media-factory: add methods to configure authorisation
6702 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6704 * gst/rtsp-server/rtsp-client.c:
6705 client: unref auth in finalize
6707 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6709 * gst/rtsp-server/rtsp-server.c:
6710 server: unref auth in finalize
6712 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6714 * docs/libs/gst-rtsp-server-docs.sgml:
6715 * docs/libs/gst-rtsp-server-sections.txt:
6716 * docs/libs/gst-rtsp-server.types:
6719 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6721 * gst/rtsp-server/rtsp-server.c:
6722 * gst/rtsp-server/rtsp-server.h:
6723 server: separate create and accept
6724 Create separate create and accept methods so that subclasses can create custom
6726 Configure the server in the client object and prepare for keeping track of
6729 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6731 * gst/rtsp-server/rtsp-client.c:
6732 * gst/rtsp-server/rtsp-client.h:
6733 client: add support for setting the server.
6734 Add support for keeping a ref to the server that started this client
6737 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6739 * gst/rtsp-server/rtsp-auth.c:
6740 auth: fix memleak and add some docs
6741 Fix a memleak of the basic auth token.
6742 Add docs for the helper function
6744 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6746 * gst/rtsp-server/rtsp-auth.c:
6747 * gst/rtsp-server/rtsp-auth.h:
6748 * gst/rtsp-server/rtsp-client.c:
6749 client: delegate setup of auth to the manager
6750 Delegate the configuration of the authentication tokens to the manager object
6753 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6755 * examples/test-video.c:
6756 * gst/rtsp-server/Makefile.am:
6757 * gst/rtsp-server/rtsp-auth.c:
6758 * gst/rtsp-server/rtsp-auth.h:
6759 * gst/rtsp-server/rtsp-client.c:
6760 * gst/rtsp-server/rtsp-client.h:
6761 * gst/rtsp-server/rtsp-server.c:
6762 * gst/rtsp-server/rtsp-server.h:
6763 auth: add authentication object
6764 Add an object that can check the authorization of requests.
6765 Implement basic authentication.
6766 Add example authentication to test-video
6768 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6770 * gst/rtsp-server/rtsp-server.c:
6771 * gst/rtsp-server/rtsp-server.h:
6772 server: move includes back
6773 the includes are needed for sockaddr_in.
6775 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6777 * gst/rtsp-server/rtsp-client.c:
6778 * gst/rtsp-server/rtsp-client.h:
6779 * gst/rtsp-server/rtsp-server.c:
6780 * gst/rtsp-server/rtsp-server.h:
6781 rtsp: move network includes where they are needed
6783 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
6785 * gst/rtsp-server/rtsp-media.h:
6786 rtsp-media.h: Minor corrections in comments.
6789 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
6792 Automatic update of common submodule
6793 From e572c87 to f94d739
6795 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6799 * docs/libs/.gitignore:
6800 * examples/.gitignore:
6801 * gst/rtsp-server/.gitignore:
6804 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6806 * docs/libs/Makefile.am:
6807 docs: We don't build ps/pdf for API reference docs
6809 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6812 Automatic update of common submodule
6813 From ccbaa85 to e572c87
6815 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6818 Automatic update of common submodule
6819 From 46445ad to ccbaa85
6821 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6823 * gst/rtsp-server/Makefile.am:
6824 * gst/rtsp-server/fs-funnel.c:
6825 * gst/rtsp-server/fs-funnel.h:
6826 * gst/rtsp-server/rtsp-funnel.c:
6827 * gst/rtsp-server/rtsp-funnel.h:
6828 * gst/rtsp-server/rtsp-media.c:
6829 funnel: rename fsfunnel to rtspfunnel
6830 Rename the funnel to avoid conflicts with the farsight one.
6832 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6834 * gst/rtsp-server/Makefile.am:
6835 * gst/rtsp-server/fs-funnel.c:
6836 * gst/rtsp-server/fs-funnel.h:
6837 * gst/rtsp-server/rtsp-media.c:
6838 rtsp-media: add and use fsfunnel
6839 Add a copy of fsfunnel to the build because input-selector removed the (broken)
6840 select-all property that we need.
6842 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6844 * gst/rtsp-server/Makefile.am:
6845 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
6846 Use PKG_CONFIG_PATH specified at configure time (if any) as well
6847 for the g-ir-compiler, rather than just assuming the env var has
6850 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6857 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
6859 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6862 * gst/rtsp-server/Makefile.am:
6863 gobject-introspection: fix g-i build for uninstalled setup
6864 Requires gst-plugins-base git (> 0.10.31.2).
6866 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6868 * examples/test-uri.c:
6869 examples: add some more options and comments
6871 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6873 * gst/rtsp-server/rtsp-media-factory-uri.c:
6874 factory-uri: use right property type
6876 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6878 * gst/rtsp-server/rtsp-media-factory-uri.c:
6879 factory-uri: attempt to configure buffer-lists
6880 Attempt to configure buffer lists in the payloader for improved performance.
6882 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6884 * gst/rtsp-server/rtsp-media.c:
6885 media: attempt to configure bigger UDP buffers
6886 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
6887 send buffers with high bitrate streams.
6889 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
6891 * gst/rtsp-server/rtsp-client.c:
6892 client: use the socket length from getsockname
6893 Use the length returned by getsockname to perform the getnameinfo call because
6894 the size can depend on the socket type and platform.
6897 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6899 * docs/libs/gst-rtsp-server-docs.sgml:
6900 * docs/libs/gst-rtsp-server-sections.txt:
6901 docs: add uri factory to the docs
6903 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6905 * gst/rtsp-server/rtsp-client.c:
6906 * gst/rtsp-server/rtsp-media.h:
6909 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6911 * gst/rtsp-server/rtsp-client.c:
6912 * gst/rtsp-server/rtsp-media.c:
6913 * gst/rtsp-server/rtsp-media.h:
6914 * gst/rtsp-server/rtsp-session.c:
6915 * gst/rtsp-server/rtsp-session.h:
6916 rtsp-server: add support for buffer lists
6917 Add support for sending bufferlists received from appsink.
6920 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6922 * gst/rtsp-server/rtsp-client.c:
6923 * gst/rtsp-server/rtsp-media.c:
6924 * gst/rtsp-server/rtsp-media.h:
6925 * gst/rtsp-server/rtsp-sdp.c:
6926 media: make method to retrieve the play range
6927 Make a method to retrieve the playback range so that we can conditionally create
6928 a different range for the SDP and the PLAY requests.
6930 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6932 * gst/rtsp-server/rtsp-media.c:
6933 * gst/rtsp-server/rtsp-media.h:
6934 media: add signal to notify of state changes
6936 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6938 * gst/rtsp-server/rtsp-client.h:
6939 client: cleanup headers
6941 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6943 * gst/rtsp-server/rtsp-client.c:
6946 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6948 * gst/rtsp-server/rtsp-media-factory-uri.c:
6949 * gst/rtsp-server/rtsp-media-factory-uri.h:
6950 factory-uri: add support for gstpay
6951 Add an option to prefer gstpay over decoder + raw payloader.
6953 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6955 * gst/rtsp-server/rtsp-media-factory-uri.c:
6956 * gst/rtsp-server/rtsp-media-factory-uri.h:
6957 factory-uri: rework the autoplugger.
6958 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
6961 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6963 * gst/rtsp-server/rtsp-media-factory-uri.c:
6964 factory-uri: use better factory filter
6965 Make better payloader filter based on autoplug rank and RTP use case.
6967 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6970 Automatic update of common submodule
6971 From 169462a to 46445ad
6973 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6975 * gst/rtsp-server/rtsp-server.c:
6976 server: set SO_REUSEADDR before bind
6977 Set the SO_REUSEADDR _before_ bind() to make it actually work.
6979 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6981 * gst/rtsp-server/rtsp-media.c:
6982 * gst/rtsp-server/rtsp-media.h:
6983 media: emit prepared signal when prepared
6984 Make a 'prepared' signal and emit it when we successfully prepared the element.
6985 This signal can be used to configure the media object after it has been prepared
6988 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
6991 Automatic update of common submodule
6992 From 011bcc8 to 169462a
6994 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
6996 python an optional dependency
6997 * configure.ac: Move up valgrind and g-i checks. Make the python
6998 dependency optional, as it was before.
7000 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7002 Merge branch 'master' into 0.11
7007 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7009 * gst/rtsp-server/rtsp-media.c:
7010 media: update range when active clients changed
7011 When we changed the number of active clients, update the current range
7012 information because we want the second client connecting to a shared resource
7013 continue from where the stream currently.
7015 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7017 * gst/rtsp-server/rtsp-media-factory-uri.c:
7018 * gst/rtsp-server/rtsp-media-factory-uri.h:
7019 factory-uri: add colorspace and fix pt
7020 Rework the way we pass data to the autoplugger.
7021 When we have raw caps, plug a converter element to make pluggin to raw
7022 payloaders more successful.
7023 Make sure all dynamically plugged payloaders have a unique payload types.
7025 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7027 * examples/Makefile.am:
7028 * examples/test-uri.c:
7029 example: add example of the uri factory
7031 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7033 * gst/rtsp-server/Makefile.am:
7034 * gst/rtsp-server/rtsp-media-factory-uri.c:
7035 * gst/rtsp-server/rtsp-media-factory-uri.h:
7036 * gst/rtsp-server/rtsp-server.h:
7037 factory-uri: add a factory to stream any URI
7038 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
7041 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7043 * gst/rtsp-server/rtsp-media.c:
7044 * gst/rtsp-server/rtsp-media.h:
7045 media: ignore spurious ASYNC_DONE messages
7046 When we are dynamically adding pads, the addition of the udpsrc elements will
7047 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
7048 the real ASYNC_DONE when everything is prerolled.
7050 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7052 * gst/rtsp-server/rtsp-media-factory.c:
7053 * gst/rtsp-server/rtsp-media-factory.h:
7054 media-factory: make lock macro
7056 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
7058 * gst/rtsp-server/rtsp-client.c:
7059 rtsp-server: Remove unused variable and dead assignment
7061 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
7063 * examples/test-launch.c:
7064 * examples/test-mp4.c:
7065 * examples/test-ogg.c:
7066 * examples/test-readme.c:
7067 * examples/test-sdp.c:
7068 * examples/test-video.c:
7069 examples: Run gst-indent
7071 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
7073 * gst/rtsp-server/rtsp-client.c:
7074 * gst/rtsp-server/rtsp-media-factory.c:
7075 * gst/rtsp-server/rtsp-media-mapping.c:
7076 * gst/rtsp-server/rtsp-media.c:
7077 * gst/rtsp-server/rtsp-params.c:
7078 * gst/rtsp-server/rtsp-sdp.c:
7079 * gst/rtsp-server/rtsp-server.c:
7080 * gst/rtsp-server/rtsp-session-pool.c:
7081 * gst/rtsp-server/rtsp-session.c:
7082 rtsp-server: Run gst-indent
7083 Since it wasn't using the upstream common previously, there was no
7084 indentation check before commiting.
7086 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
7088 * gst/rtsp-server/rtsp-media-mapping.h:
7089 * gst/rtsp-server/rtsp-media.c:
7090 * gst/rtsp-server/rtsp-media.h:
7091 * gst/rtsp-server/rtsp-sdp.c:
7092 * gst/rtsp-server/rtsp-session-pool.h:
7093 * gst/rtsp-server/rtsp-session.c:
7094 * gst/rtsp-server/rtsp-session.h:
7095 rtsp-server: Some more doc fixups
7097 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7100 Makefile: Add cruft-cleaning support
7102 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7107 * docs/libs/Makefile.am:
7108 * docs/libs/gst-rtsp-server-docs.sgml:
7109 * docs/libs/gst-rtsp-server-sections.txt:
7110 * docs/libs/gst-rtsp-server.types:
7111 * docs/version.entities.in:
7112 docs: Add gtk-doc build system
7114 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7116 * gst/rtsp-server/Makefile.am:
7117 Makefile.am: Use standard GIR make behaviour
7119 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
7123 autogen/configure: Bring more in sync to standard gst module behaviour
7125 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7127 * gst/rtsp-server/rtsp-media.c:
7128 media: warn and fail when gstrtpbin is not found
7130 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7133 configure: open 0.11 branch
7135 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
7139 Add common submodule
7141 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
7144 * common/Makefile.am:
7145 * common/c-to-xml.py:
7147 * common/coverage/coverage-report-entry.pl:
7148 * common/coverage/coverage-report.pl:
7149 * common/coverage/coverage-report.xsl:
7150 * common/coverage/lcov.mak:
7151 * common/gettext.patch:
7152 * common/glib-gen.mak:
7153 * common/gst-autogen.sh:
7154 * common/gst-xmlinspect.py:
7156 * common/gstdoc-scangobj:
7157 * common/gtk-doc-plugins.mak:
7158 * common/gtk-doc.mak:
7159 * common/m4/.gitignore:
7160 * common/m4/Makefile.am:
7162 * common/m4/as-ac-expand.m4:
7163 * common/m4/as-auto-alt.m4:
7164 * common/m4/as-compiler-flag.m4:
7165 * common/m4/as-compiler.m4:
7166 * common/m4/as-docbook.m4:
7167 * common/m4/as-libtool-tags.m4:
7168 * common/m4/as-libtool.m4:
7169 * common/m4/as-python.m4:
7170 * common/m4/as-scrub-include.m4:
7171 * common/m4/as-version.m4:
7172 * common/m4/ax_create_stdint_h.m4:
7173 * common/m4/check.m4:
7174 * common/m4/glib-gettext.m4:
7175 * common/m4/gst-arch.m4:
7176 * common/m4/gst-args.m4:
7177 * common/m4/gst-check.m4:
7178 * common/m4/gst-debuginfo.m4:
7179 * common/m4/gst-default.m4:
7180 * common/m4/gst-doc.m4:
7181 * common/m4/gst-error.m4:
7182 * common/m4/gst-feature.m4:
7183 * common/m4/gst-function.m4:
7184 * common/m4/gst-gettext.m4:
7185 * common/m4/gst-glib2.m4:
7186 * common/m4/gst-libxml2.m4:
7187 * common/m4/gst-plugindir.m4:
7188 * common/m4/gst-valgrind.m4:
7189 * common/m4/gtk-doc.m4:
7190 * common/m4/introspection.m4:
7192 * common/mangle-tmpl.py:
7193 * common/plugins.xsl:
7195 * common/release.mak:
7196 * common/scangobj-merge.py:
7197 * common/upload.mak:
7198 common: Remove static version
7200 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
7202 * common/m4/introspection.m4:
7203 Update introspection.m4 to match usage
7205 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7209 Remove old stuff from the README
7211 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7216 === release 0.10.7 ===
7218 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7223 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7225 * examples/test-ogg.c:
7226 test-ogg: remove parsers
7227 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
7228 buffers with timestamps. Using the parsers also seems to break things.
7230 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7232 * bindings/vala/gst-rtsp-server-0.10.vapi:
7233 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7234 Updated Vala bindings
7236 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7238 * common/m4/introspection.m4:
7240 * gst/rtsp-server/Makefile.am:
7241 Added initial gobject-introspection support
7243 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7245 * gst/rtsp-server/rtsp-media-factory.c:
7246 media-factory: don't use host for shared hash key
7247 When we generate the key to share made between connections, don't include the
7248 host used to connect so that we can share media even if between clients that
7249 connected with localhost and ones with the ip address.
7251 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7253 * bindings/vala/Makefile.am:
7254 build: fix distcheck
7256 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7258 * bindings/vala/gst-rtsp-server-0.10.vapi:
7259 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7260 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7261 Update Vala bindings
7263 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
7265 * bindings/vala/Makefile.am:
7267 Fix configure checks and installation location for Vala bindings
7270 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7275 === release 0.10.6 ===
7277 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7280 configure: release 0.10.6
7282 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7284 * gst/rtsp-server/rtsp-media.c:
7285 media: help the compiler a little
7287 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7289 * gst/rtsp-server/rtsp-media.c:
7290 * gst/rtsp-server/rtsp-media.h:
7291 * gst/rtsp-server/rtsp-session.c:
7292 media: cleanup media transport before freeing
7293 Cleanup the media transport data before freeing. In particular, remove the qdata
7294 from the rtpsource object.
7296 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7298 * gst/rtsp-server/rtsp-media-factory.c:
7299 * gst/rtsp-server/rtsp-media-factory.h:
7300 * gst/rtsp-server/rtsp-media.c:
7301 * gst/rtsp-server/rtsp-media.h:
7302 media-factory: add eos-shutdown property
7303 Add an eos-shutdown property that will send an EOS to the pipeline before
7304 shutting it down. This allows for nice cleanup in case of a muxer.
7307 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7309 * gst/rtsp-server/rtsp-media.c:
7310 * gst/rtsp-server/rtsp-media.h:
7311 media: use multiudpsink send-duplicates when we can
7312 If we have a new enough multiudpsink with the send-duplicates property, use this
7313 instead of doing our own filtering. Our custom filtering code should eventually
7314 be removed when we can depend on a released -good.
7316 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7318 * gst/rtsp-server/rtsp-media.c:
7319 media: don't leak destinations
7320 Refactor and cleanup the destinations array when the stream is destroyed.
7322 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7324 * gst/rtsp-server/rtsp-media.c:
7325 * gst/rtsp-server/rtsp-media.h:
7326 media: don't add udp addresses multiple times
7327 Keep track of the udp addresses we added to udpsink and never add the same udp
7328 destination twice. This avoids duplicate packets when using multicast.
7330 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7332 * gst/rtsp-server/rtsp-server.c:
7333 server: disable use of SO_LINGER
7334 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
7335 server close()s the connection.
7337 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7339 * gst/rtsp-server/rtsp-server.c:
7340 server: use 5 second linger period in SO_LINGER
7341 Wait 5 seconds before clearing the send buffers and reseting the connection with
7342 the client when we do a close. This should be enough time to get the message to
7346 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7348 * gst/rtsp-server/rtsp-server.c:
7349 server: use SO_LINGER
7350 SO_LINGER on the socket will make sure that any pending data on the socket is
7351 flushed ASAP and that the socket connection is reset. This makes sure that the
7352 socket can be reused immediately.
7355 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7358 README: add blurb about shared media factories
7360 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
7362 * gst/rtsp-server/rtsp-media.c:
7363 Add stdlib.h for atoi()
7365 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7367 * bindings/python/Makefile.am:
7368 * bindings/vala/Makefile.am:
7369 build: distcheck fixes
7370 Fix 'make distcheck', somewhat (it still fails because it tries to
7371 install files into /usr/share/vala/vapi/ irrespective of the
7374 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7377 configure: bump core/base requirements to released version
7378 Makes things less confusing for people.
7380 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7383 configure: fail if GStreamer core/base requirements are not met
7385 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7387 * gst/rtsp-server/rtsp-client.c:
7388 client: improve client cleanups
7389 Make sure the session does not timeout when using TCP. We need to do this
7390 because quicktime player does not send RTCP for some reason in tunneled
7392 Refactor some cleanup code.
7395 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7397 * gst/rtsp-server/rtsp-session.c:
7398 * gst/rtsp-server/rtsp-session.h:
7399 session: add support for prevent session timeouts
7400 Add an atomix counter to prevent session timeouts when we are, for example,
7403 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7405 * gst/rtsp-server/rtsp-client.c:
7406 client: fix unlink on session timeouts
7407 When our session times out, make sure we unlink all streams in this
7409 Remove the tunnelid when closing the connection.
7411 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7413 * gst/rtsp-server/rtsp-session.c:
7414 session: small cleanups
7416 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7418 * gst/rtsp-server/rtsp-client.c:
7419 client: handle lost_tunnel callbacks
7420 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
7421 hashtable so that we can reuse it for when the client reopens the POST
7423 Close the connection after a TEARDOWN.
7424 Make sure or watchid is cleared when the watch is removed.
7427 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7429 * gst/rtsp-server/rtsp-client.c:
7430 * gst/rtsp-server/rtsp-media.c:
7431 * gst/rtsp-server/rtsp-sdp.c:
7432 rtsp-server: add more support for multicast
7434 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7437 * gst/rtsp-server/rtsp-media.c:
7438 * gst/rtsp-server/rtsp-media.h:
7439 media: allow configuration of allowed lower transport
7441 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7443 * gst/rtsp-server/rtsp-client.h:
7444 * gst/rtsp-server/rtsp-media.c:
7445 * gst/rtsp-server/rtsp-media.h:
7446 * gst/rtsp-server/rtsp-sdp.c:
7447 * gst/rtsp-server/rtsp-sdp.h:
7448 * gst/rtsp-server/rtsp-server.c:
7449 rtsp: keep track of server ip and ipv6
7450 Keep track of how the client connected to the server and setup the udp ports
7451 with the same protocol.
7452 Copy the server ip address in the SDP so that clients can send RTCP back to
7455 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7457 * gst/rtsp-server/rtsp-session.c:
7460 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7462 * gst/rtsp-server/rtsp-client.c:
7463 client: use right size for malloc
7465 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7467 * gst/rtsp-server/rtsp-server.c:
7468 server: comment ipv6 server listening address
7470 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7472 * gst/rtsp-server/rtsp-media.c:
7473 media: allow for ipv6 sockets
7475 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7477 * gst/rtsp-server/rtsp-server.c:
7478 * gst/rtsp-server/rtsp-server.h:
7479 server: rework server part
7480 Allow setting a bind address, make sure we can deal with ipv6.
7481 Remove the port property and change with the service property.
7483 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7485 * gst/rtsp-server/rtsp-media.h:
7486 media: update comments a little
7488 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7490 * gst/rtsp-server/rtsp-client.c:
7491 client: make content-base better
7492 Use the URI formatting functions to make a content-base. Also make sure that
7493 there is a trailing / at the end.
7495 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7497 * gst/rtsp-server/rtsp-client.c:
7498 client: guard against invalid paths
7500 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7502 * examples/test-video.c:
7503 test: catch server bind errors
7505 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
7507 * gst/rtsp-server/rtsp-media.c:
7508 rtspmedia: emit "unprepared" if _prepare fails.
7509 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
7510 media object is removed from its factory's cache.
7512 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7514 * gst/rtsp-server/rtsp-media.c:
7515 media: collect media position when seek completes
7517 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
7519 * gst/rtsp-server/rtsp-client.c:
7520 client: call unlink_streams in client finalize
7523 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7525 * gst/rtsp-server/rtsp-media.c:
7526 media: limit the time to wait to something huge
7527 Avoid waiting forever but limit the timeout to 20 seconds.
7529 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7531 * gst/rtsp-server/rtsp-sdp.c:
7532 sdp: reindent and check for prepared status
7534 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7536 * gst/rtsp-server/rtsp-media.c:
7537 * gst/rtsp-server/rtsp-media.h:
7538 * gst/rtsp-server/rtsp-session.c:
7539 media: avoid doing _get_state() for state changes
7540 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
7541 until the media is prerolled or in error. This avoids doing a blocking call of
7542 gst_element_get_state() that can cause lockups when there is an error.
7545 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7547 * gst/rtsp-server/rtsp-media.c:
7550 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7552 * gst/rtsp-server/rtsp-media-factory.c:
7553 media-factory: better error handling
7554 Improve the error handling a bit.
7556 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7558 * gst/rtsp-server/rtsp-client.c:
7559 client: rework transport parsing
7560 Rework the transport parsing code so that we can ignore transports we don't
7561 support instead of just picking the first one we can parse.
7562 Configure a (for now hardcoded) destination for multicast transports.
7564 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7566 * gst/rtsp-server/rtsp-media.c:
7567 media: set multicast sink parameters
7568 Disable loop and automatic multicast join on the udpsink elements.
7569 Add some more debug info.
7570 Reset some state variables in the right place.
7571 Use the right port numbers for multicast.
7573 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7575 * gst/rtsp-server/rtsp-session.c:
7576 session: handle transport setup correctly
7577 Handle UDP, MCAST and TCP transport negotiation more correctly.
7578 Store the server session SSRC in the transport.
7580 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7582 * gst/rtsp-server/rtsp-client.c:
7583 rtsp-client: implement error_full
7584 Implement error_full to avoid some segfaults when the rtspconnection calls it.
7587 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7590 * gst/rtsp-server/rtsp-client.c:
7591 * gst/rtsp-server/rtsp-server.c:
7592 docs: update docs and comments
7594 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
7596 * gst/rtsp-server/rtsp-sdp.c:
7597 sdp: make server work better when behind a proxy
7599 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7601 * gst/rtsp-server/rtsp-client.c:
7602 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
7604 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7606 * gst/rtsp-server/rtsp-client.c:
7607 * gst/rtsp-server/rtsp-media-factory.c:
7608 * gst/rtsp-server/rtsp-media-mapping.c:
7609 * gst/rtsp-server/rtsp-media.c:
7610 * gst/rtsp-server/rtsp-server.c:
7611 * gst/rtsp-server/rtsp-session-pool.c:
7612 * gst/rtsp-server/rtsp-session.c:
7613 Use GStreamer's debugging subsystem
7615 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7617 * gst/rtsp-server/rtsp-media-factory.c:
7618 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
7620 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7625 === release 0.10.5 ===
7627 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7632 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7635 configure: bump required versions
7637 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
7639 * gst/rtsp-server/rtsp-client.c:
7640 client: call weak-unref on client->sessions from finalize
7643 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7645 * gst/rtsp-server/rtsp-media.c:
7646 media: Fixed crasher where caps got unref'ed too often
7648 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7651 * pkgconfig/.gitignore:
7652 * pkgconfig/Makefile.am:
7653 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
7654 Added pkg-config file to use gst-rtsp-server uninstalled
7656 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7658 * gst/rtsp-server/rtsp-media.c:
7659 media: add some docs
7661 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
7663 * gst/rtsp-server/rtsp-client.c:
7664 rtsp: Use gst_rtsp_watch_send_message().
7665 Use gst_rtsp_watch_send_message() since the old API which used
7666 gst_rtsp_watch_queue_message() has been deprecated.
7668 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7673 === release 0.10.4 ===
7675 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7680 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7682 * gst/rtsp-server/rtsp-client.c:
7683 * gst/rtsp-server/rtsp-session.c:
7684 * gst/rtsp-server/rtsp-session.h:
7685 rtsp: allocate channels in TCP mode
7686 When the client does not provide us with channels in TCP mode, allocate channels
7689 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7691 * gst/rtsp-server/rtsp-client.c:
7692 client: don't crash when tunnelid is missing
7693 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
7694 don't crash but return an error response to the client.
7697 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7699 * bindings/vala/gst-rtsp-server-0.10.vapi:
7700 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7701 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7702 bindings: update vala bindings with new method
7704 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7706 * gst/rtsp-server/rtsp-session-pool.c:
7707 * gst/rtsp-server/rtsp-session-pool.h:
7708 sessionpool: add function to filter sessions
7709 Add generic function to retrieve/remove sessions.
7711 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7714 configure: bump core/base requirements to release
7716 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7718 * gst/rtsp-server/rtsp-media.c:
7719 media: fix indentation
7721 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7723 * gst/rtsp-server/rtsp-media.c:
7724 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
7726 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7728 * gst/rtsp-server/rtsp-media.c:
7729 set state and remove elements of media in for loop
7731 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
7733 * bindings/vala/gst-rtsp-server-0.10.vapi:
7734 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7735 Added gst_rtsp_media_remove_elements function to Vala bindings
7737 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
7739 * gst/rtsp-server/rtsp-media.c:
7740 * gst/rtsp-server/rtsp-media.h:
7741 Added gst_rtsp_media_remove_elements function
7743 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
7745 * gst/rtsp-server/rtsp-media.c:
7746 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
7748 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7750 * bindings/vala/gst-rtsp-server-0.10.vapi:
7751 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7752 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7753 Updated Vala bindings
7755 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7757 * gst/rtsp-server/rtsp-media.c:
7758 * gst/rtsp-server/rtsp-media.h:
7759 Added vmethod unprepare to GstRTSPMedia
7760 The default implementation sets the state of the pipeline to GST_STATE_NULL
7762 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7764 * gst/rtsp-server/rtsp-media-factory.c:
7765 * gst/rtsp-server/rtsp-media-factory.h:
7766 Made collect_streams function public
7768 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7770 * gst/rtsp-server/rtsp-media-factory.c:
7771 * gst/rtsp-server/rtsp-media-factory.h:
7772 * gst/rtsp-server/rtsp-media.c:
7773 Added vmethod create_pipeline to GstRTSPMediaFactory
7774 The pipeline is created in this method and the GstRTSPMedia's element is added to it
7776 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7778 * gst/rtsp-server/rtsp-client.c:
7779 client: use g_source_destroy()
7780 We need to use g_source_destroy() because we might have added the source to a
7781 different main context than the default one.
7783 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7785 * gst/rtsp-server/Makefile.am:
7786 * gst/rtsp-server/rtsp-client.c:
7787 * gst/rtsp-server/rtsp-params.c:
7788 * gst/rtsp-server/rtsp-params.h:
7789 rtsp: prepare for handling GET/SET_PARAMETER
7790 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
7792 Fix return codes of handlers.
7794 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7796 * gst/rtsp-server/rtsp-media.c:
7797 media: don't leak session pads
7799 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7801 * gst/rtsp-server/rtsp-media.c:
7802 media: clean up the messages a bit
7804 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7806 * gst/rtsp-server/rtsp-sdp.c:
7807 sdp: warn and skip streams without media
7809 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7811 * bindings/vala/gst-rtsp-server-0.10.vapi:
7812 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7813 vala: Fixed typo in header file of RTSPMediaStream
7815 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7817 * gst/rtsp-server/rtsp-media.c:
7820 Make dumping RTCP stats configurable
7822 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7824 * gst/rtsp-server/rtsp-media.c:
7825 media: be less verbose and leak less
7827 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7829 * gst/rtsp-server/rtsp-media.c:
7830 media: don't leak the destination address
7832 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7834 * gst/rtsp-server/rtsp-client.c:
7835 * gst/rtsp-server/rtsp-media.c:
7836 * gst/rtsp-server/rtsp-media.h:
7837 * gst/rtsp-server/rtsp-session.c:
7838 * gst/rtsp-server/rtsp-session.h:
7839 rtsp: use RTCP to keep the session alive
7840 Use the RTCP rtcp-from stats field to find the associated session and use this
7841 to keep the session alive.
7843 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7845 * gst/rtsp-server/rtsp-session.c:
7846 session: add 5sec to the real session timeout
7847 Allow the session to live 5sec longer before really timing out. This should give
7848 clients some extra time to keep the session active.
7850 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7852 * gst/rtsp-server/rtsp-client.c:
7853 client: replay OK to GET/SET_PARAMETER
7854 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
7855 so that we return OK for those requests.
7857 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7859 * gst/rtsp-server/rtsp-media.c:
7860 * gst/rtsp-server/rtsp-media.h:
7861 media: keep track of active transports
7862 Keep track of which transport is active to avoid closing the connection too
7864 Remove the destination transport also when going to NULL.
7865 Print some stats about the SDES and other RTCP messages we receive from the
7868 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7870 * examples/.gitignore:
7871 * examples/Makefile.am:
7872 * examples/test-sdp.c:
7873 example: add SDP relay example
7875 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7877 * gst/rtsp-server/rtsp-media.c:
7878 media: also count active TCP connections
7880 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7882 * gst/rtsp-server/rtsp-media-factory.c:
7883 * gst/rtsp-server/rtsp-media.c:
7884 * gst/rtsp-server/rtsp-media.h:
7885 rtsp: add support for dynamic elements
7886 Add support for dynamic elements.
7887 Don't set live pipelines back to paused.
7889 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7891 * gst/rtsp-server/rtsp-sdp.c:
7892 sdp: don't add encoding name when absent in caps
7894 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7896 * gst/rtsp-server/rtsp-client.c:
7897 client: warn when we can't do RTP-Info
7899 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7901 * gst/rtsp-server/rtsp-media-factory.c:
7902 factory: factor out the stream construction
7904 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7906 * gst/rtsp-server/rtsp-client.c:
7907 client: only add RTP-Info when we have the info
7908 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
7911 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7916 === release 0.10.3 ===
7918 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7922 - Fixes a bug where it put the wrong verion in pkgconfig
7923 - Link RTP and RTCP sources
7925 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7927 * gst/rtsp-server/rtsp-media.c:
7928 * gst/rtsp-server/rtsp-media.h:
7929 media: link the RTP udpsrc to the session manager
7930 Link the RTP udpsrc and the appsrc to the session manager so that they don't
7931 shut down when the client sends a packet to open firewalls.
7933 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7935 * pkgconfig/gst-rtsp-server.pc.in:
7936 Don't use hard-coded version number in pkg-config file
7938 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7943 === release 0.10.2 ===
7945 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7950 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7953 * common/m4/.gitignore:
7954 * examples/.gitignore:
7955 * pkgconfig/.gitignore:
7956 add some .gitignore files
7958 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7960 * gst/rtsp-server/rtsp-media.c:
7961 media: seek to key frames
7963 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7965 * gst/rtsp-server/rtsp-media.c:
7966 media: emit the unprepared signal by id
7967 Emit the unprepared signal by id instead of name and set the media as
7970 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7972 * gst/rtsp-server/rtsp-media.c:
7973 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
7975 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7977 * gst/rtsp-server/rtsp-server.c:
7978 Added finalize function to GstRTPSPServer to unref session pool and media mapping
7980 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7982 * bindings/vala/gst-rtsp-server-0.10.vapi:
7983 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7984 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7985 Updated vala bindings
7987 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7989 * gst/rtsp-server/Makefile.am:
7990 * gst/rtsp-server/rtsp-client.c:
7991 * gst/rtsp-server/rtsp-media.c:
7992 server: use appsink and appsrc with the API
7993 Use the appsink/appsrc API instead of the signals for higher
7996 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7998 * examples/test-ogg.c:
7999 tests: set the payload type correctly
8001 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8003 * gst/rtsp-server/rtsp-media-factory.c:
8004 factory: connect to the unprepare signal
8005 Connect to the unprepare signal for non-reusable media so that we can remove
8006 them from the cache.
8008 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8010 * gst/rtsp-server/rtsp-media.c:
8011 * gst/rtsp-server/rtsp-media.h:
8012 media: add signal to notify of unprepare
8014 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8016 * gst/rtsp-server/rtsp-media.c:
8017 * gst/rtsp-server/rtsp-media.h:
8018 media: more work on making the media shared
8019 Add a reusable flag to medias, indicating that they can be reused after a state
8023 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8025 * examples/test-readme.c:
8026 examples: mark the example as shared for testing
8028 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8030 * gst/rtsp-server/rtsp-media.c:
8031 * gst/rtsp-server/rtsp-media.h:
8032 client: support shared media
8033 Always perform the state actions even if the target state of the pipeline is
8034 already correct, we still want to add/remove the transports when we are dealing
8036 Keep a counter of the number of active transports for a media so that we can use
8037 this to perform a state change when needed.
8038 Perform a state change of the pipeline only when the first transport was added
8039 or when there are no active transports.
8041 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
8043 * gst/rtsp-server/rtsp-client.c:
8044 client: fix refcounting crasher
8045 Don't need to remove the weak refs in the finalize methods, they are already
8046 removed in the dispose.
8047 Don't register the callback with a DestroyNofity.
8049 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8051 * gst/rtsp-server/rtsp-client.c:
8052 Fix rtsp client refcount management in TCP mode.
8053 Don't unref a client ref we never had. Fixes an unref
8054 of an already-free client object after a client
8055 teardown request for me.
8057 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
8059 * gst/rtsp-server/rtsp-session.c:
8060 docs: fix typo in API docs
8062 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8064 * gst/rtsp-server/rtsp-media.c:
8066 Keep the udp sources in playing even if we go to paused. unlock the sources when
8068 Add some more debug info.
8069 Only seek when we need to.
8070 Keep track of the position when we go to paused.
8072 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8074 * gst/rtsp-server/rtsp-client.c:
8075 * gst/rtsp-server/rtsp-media.c:
8076 * gst/rtsp-server/rtsp-media.h:
8077 Add beginnings of seeking.
8078 Parse the Range header and perform a seek on the pipeline for the requested
8079 position. It's disabled currently until I figure out what's going wrong.
8081 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8083 * gst/rtsp-server/rtsp-client.c:
8084 allow pause requests for now.
8087 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8089 * gst/rtsp-server/rtsp-client.c:
8090 Remove weak ref on the session in teardown
8091 We need to remove our weakref from the session when we do a teardown because
8092 else we close the TCP connection prematurely.
8094 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8096 * gst/rtsp-server/rtsp-client.c:
8097 * gst/rtsp-server/rtsp-client.h:
8098 * gst/rtsp-server/rtsp-session-pool.c:
8099 Do some more session cleanup
8100 Make session timeout kill the TCP connection that currently watches the
8102 Remove the client timeout property.
8104 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8106 * gst/rtsp-server/rtsp-client.c:
8107 * gst/rtsp-server/rtsp-client.h:
8108 * gst/rtsp-server/rtsp-media.c:
8109 * gst/rtsp-server/rtsp-media.h:
8110 * gst/rtsp-server/rtsp-server.c:
8111 * gst/rtsp-server/rtsp-session.c:
8112 * gst/rtsp-server/rtsp-session.h:
8114 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
8117 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8119 * examples/Makefile.am:
8120 * examples/test-launch.c:
8121 Add example server that takes launch lines
8122 Add an example server that streams any -launch line.
8124 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8126 * examples/test-readme.c:
8127 * gst/rtsp-server/rtsp-client.c:
8128 * gst/rtsp-server/rtsp-media.c:
8129 * gst/rtsp-server/rtsp-media.h:
8130 Add support for live streams
8131 Add support for live streams and ranges
8132 Start on handling TCP data transfer.
8134 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8136 * gst/rtsp-server/rtsp-media.c:
8137 Free the pipeline before other things
8140 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8142 * gst/rtsp-server/rtsp-client.c:
8143 Only free the pending tunnel if there is one
8146 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8148 * gst/rtsp-server/rtsp-client.c:
8149 * gst/rtsp-server/rtsp-client.h:
8150 * gst/rtsp-server/rtsp-media.c:
8151 rtsp-server: Add support for tunneling
8152 Add support for tunneling over HTTP.
8153 Use new connection methods to retrieve the url.
8154 Dispatch messages based on the message type instead of blindly
8155 assuming it's always a request.
8156 Keep track of the watch id so that we can remove it later.
8157 Set the media pipeline to NULL before unreffing the pipeline.
8159 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8161 * gst/rtsp-server/rtsp-client.c:
8162 * gst/rtsp-server/rtsp-client.h:
8163 Fix for channel -> watch rename in gstreamer
8164 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
8166 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8168 * gst/rtsp-server/rtsp-client.c:
8169 * gst/rtsp-server/rtsp-client.h:
8171 Use the async RTSP channels instead of spawning a new thread for each client.
8172 If a sessionid is specified in a request, fail if we don't have the session.
8174 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8176 * gst/rtsp-server/rtsp-media.c:
8177 Add better debug info
8178 Add some better debug info.
8180 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8182 * examples/test-video.c:
8184 Add support for session timeouts in the example.
8186 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8188 * gst/rtsp-server/rtsp-session-pool.c:
8189 * gst/rtsp-server/rtsp-session-pool.h:
8190 Pass GTimeVal around for performance reasons
8191 Get the current time only once and pass it around so that sessions don't have to
8192 get the current time anymore.
8193 Add experimental support for a GSource that dispatches when the session needs to
8196 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8198 * gst/rtsp-server/rtsp-session.c:
8199 * gst/rtsp-server/rtsp-session.h:
8200 Add better support for session timeouts
8201 Add a method to request the number of milliseconds when a session will timeout.
8203 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8205 * gst/rtsp-server/rtsp-media.c:
8206 * gst/rtsp-server/rtsp-media.h:
8207 Add suport for RTP manager monitoring
8208 Add the first stage in monitoring the rtp manager.
8209 Make sure we don't update the state to something we don't want.
8211 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8213 * gst/rtsp-server/rtsp-client.c:
8214 Add support for session keepalive
8215 Get and update the session timeout for all requests. get the session as early as
8218 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8220 * gst/rtsp-server/rtsp-media-factory.h:
8221 * gst/rtsp-server/rtsp-media.c:
8222 * gst/rtsp-server/rtsp-media.h:
8223 Handle media bus messages
8224 Handle media bus messages in a custom mainloop and dispatch them to the
8225 RTSPMedia objects. Let the default implementation handle some common messages.
8227 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8229 * gst/rtsp-server/rtsp-client.c:
8230 * gst/rtsp-server/rtsp-session-pool.c:
8231 * gst/rtsp-server/rtsp-session.c:
8232 Some more session timeout handling
8233 Move the session header setting code to a central place so that we always add
8234 the timeout parameter too.
8235 Handle timeouts by running the session cleanup code.
8236 Stop media before cleaning up.
8238 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8240 * gst/rtsp-server/rtsp-client.c:
8241 * gst/rtsp-server/rtsp-client.h:
8242 Add timeout property
8243 Add a timeout property ot the client and make the other properties into GObject
8246 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8248 * gst/rtsp-server/rtsp-session-pool.c:
8249 Use getters and setters in property code
8250 Use the getters and setters for the timeout property instead of locking
8253 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8255 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
8257 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8259 * gst/rtsp-server/rtsp-session-pool.c:
8260 * gst/rtsp-server/rtsp-session-pool.h:
8261 * gst/rtsp-server/rtsp-session.c:
8262 * gst/rtsp-server/rtsp-session.h:
8263 Add more timeout stuff
8264 Add method to check if a session is expired.
8265 Add method to perform cleanup on a session pool.
8267 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8269 * gst/rtsp-server/rtsp-client.c:
8270 * gst/rtsp-server/rtsp-session-pool.c:
8271 * gst/rtsp-server/rtsp-session-pool.h:
8272 * gst/rtsp-server/rtsp-session.c:
8273 * gst/rtsp-server/rtsp-session.h:
8274 Add beginnings of session timeouts and limits
8275 Add the timeout value to the Session header for unusual timeout values.
8276 Allow us to configure a limit to the amount of active sessions in a pool. Set a
8277 limit on the amount of retry we do after a sessionid collision.
8278 Add properties to the sessionid and the timeout of a session. Keep track of
8279 creation time and last access time for sessions.
8281 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8283 * gst/rtsp-server/rtsp-client.c:
8284 * gst/rtsp-server/rtsp-media.c:
8285 * gst/rtsp-server/rtsp-media.h:
8286 * gst/rtsp-server/rtsp-sdp.c:
8287 * gst/rtsp-server/rtsp-session-pool.c:
8288 * gst/rtsp-server/rtsp-session.c:
8289 * gst/rtsp-server/rtsp-session.h:
8290 Cleanup of sessions and more
8291 Fix the refcounting of media and sessions in the client. Properly clean up the
8292 session data when the client performs a teardown.
8293 Add Server header to responses.
8294 Allow for multiple uri setups in one session.
8295 Add Range header to the PLAY response and add the range attribute to the SDP
8297 Fix the session pool remove method, it used the wrong key in the hashtable. Also
8298 give the ownership of the sessionid to the session object.
8300 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8302 * gst/rtsp-server/rtsp-server.c:
8303 * gst/rtsp-server/rtsp-server.h:
8305 Rename the 'server_port' variable to simply 'port'.
8307 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8310 * gst/rtsp-server/rtsp-client.c:
8311 * gst/rtsp-server/rtsp-media.c:
8312 * gst/rtsp-server/rtsp-media.h:
8313 * gst/rtsp-server/rtsp-session.c:
8314 * gst/rtsp-server/rtsp-session.h:
8315 Rework the way we handle transports for streams
8316 Make the media accept an array of transports for the streams that we have
8317 configured for the play/pause requests.
8318 Implement server states for a client and its media.
8319 Require 0.10.22.1 (git HEAD) of gstreamer.
8321 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8323 * gst/rtsp-server/rtsp-client.c:
8324 * gst/rtsp-server/rtsp-media-factory.c:
8325 Drop const from functions dealing with urls
8326 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
8327 have the right const in them.
8329 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8331 * gst/rtsp-server/rtsp-client.c:
8332 * gst/rtsp-server/rtsp-media.c:
8333 * gst/rtsp-server/rtsp-sdp.c:
8337 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8339 * gst/rtsp-server/rtsp-client.c:
8340 * gst/rtsp-server/rtsp-media-factory.c:
8341 * gst/rtsp-server/rtsp-media.c:
8342 * gst/rtsp-server/rtsp-media.h:
8344 Don't keep a reference to the GstRTSPMedia in the stream.
8345 Free more things when freeing the GstRTSPMedia.
8347 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8350 * gst/rtsp-server/rtsp-media-factory.c:
8351 * gst/rtsp-server/rtsp-media-factory.h:
8352 * gst/rtsp-server/rtsp-media.c:
8353 * gst/rtsp-server/rtsp-media.h:
8354 * gst/rtsp-server/rtsp-server.c:
8355 * gst/rtsp-server/rtsp-server.h:
8356 More docs and small cleanups
8357 Add some more docs and update the README
8358 Cleanup some method names.
8359 Remove an unneeded idx field in the GstRTSPMediaStream
8361 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8364 * examples/Makefile.am:
8365 * examples/test-readme.c:
8366 Add a README and more example code
8367 Add a README file that contains a small introduction on how to use the server
8368 along with the example code explained in the readme.
8370 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8372 * gst/rtsp-server/rtsp-media.c:
8373 * gst/rtsp-server/rtsp-server.c:
8374 Fix some leaks and change default port
8375 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
8376 we finished the initial preroll. If we keep them locked, setting the pipeline to
8377 NULL will not stop and clean up the sources correctly.
8378 Change the default RTSP port to 8554 aka the official alternative RTSP port.
8380 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8382 * gst/rtsp-server/rtsp-session.c:
8383 * gst/rtsp-server/rtsp-session.h:
8384 Cleanups to the session object
8385 Remove some unneeded variables in the session state of a stream such as the
8386 owner media and the server transport.
8387 Get the configuration of a media stream in a session based on the media_stream
8388 in the original object instead of our cached index.
8389 Free more data in the finalize method.
8391 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8393 * gst/rtsp-server/rtsp-client.c:
8394 * gst/rtsp-server/rtsp-client.h:
8395 Cleanups and reuse media from DESCRIBE
8396 Handle thread create errors.
8397 Rename some internal methods to better match what they actually do.
8398 Handle misconfiguration of session_pool and media_mapping gracefully.
8399 Cache the DESCRIBE media and uri in the client connection and reuse them when
8400 we receive a SETUP request in the same connection for the same uri.
8401 Cleanup the client connection object.
8403 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8405 * gst/rtsp-server/rtsp-media-factory.c:
8406 * gst/rtsp-server/rtsp-media-factory.h:
8407 * gst/rtsp-server/rtsp-media.c:
8408 * gst/rtsp-server/rtsp-media.h:
8409 Add shared properties to media and factory
8410 Add the shared property to media.
8411 Implement some simple caching in the factory depending on if the media is shared
8414 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8416 * gst/rtsp-server/rtsp-client.c:
8417 Add a little comment
8418 Add some comment about the content-base header.
8420 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8422 * examples/Makefile.am:
8424 * examples/test-mp4.c:
8425 * examples/test-ogg.c:
8426 * examples/test-video.c:
8427 * gst/rtsp-server/Makefile.am:
8428 * gst/rtsp-server/rtsp-client.c:
8429 * gst/rtsp-server/rtsp-client.h:
8430 * gst/rtsp-server/rtsp-media-factory.c:
8431 * gst/rtsp-server/rtsp-media-factory.h:
8432 * gst/rtsp-server/rtsp-media.c:
8433 * gst/rtsp-server/rtsp-media.h:
8434 * gst/rtsp-server/rtsp-sdp.c:
8435 * gst/rtsp-server/rtsp-sdp.h:
8436 * gst/rtsp-server/rtsp-server.c:
8437 * gst/rtsp-server/rtsp-server.h:
8438 * gst/rtsp-server/rtsp-session.c:
8439 * gst/rtsp-server/rtsp-session.h:
8440 Reorganize things, prepare for media sharing
8441 Added various other test server examples
8442 Move the SDP message generation to a separate helper.
8443 Refactor common code for finding the session.
8444 Add content-base for realplayer compatibility
8445 Clean up request uris before processing for better vlc compatibility.
8446 Move prerolling and pipeline construction to the RTSPMedia object.
8447 Use multiudpsink for future pipeline reuse.
8449 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8455 === release 0.10.1 ===
8457 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8463 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8465 * bindings/vala/Makefile.am:
8467 Add more directories and files to the dist.
8469 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8471 * bindings/python/Makefile.am:
8472 * bindings/python/rtspserver.override:
8473 Fixed compile error of python bindings
8475 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8477 * bindings/vala/gst-rtsp-server-0.10.vapi:
8478 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8479 Marked values as nullable accordingly
8481 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8483 * bindings/vala/gst-rtsp-server-0.10.vapi:
8484 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8485 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8486 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8487 Updated Vala bindings
8489 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8491 * gst/rtsp-server/rtsp-client.c:
8492 * gst/rtsp-server/rtsp-media-mapping.c:
8493 * gst/rtsp-server/rtsp-media-mapping.h:
8494 * gst/rtsp-server/rtsp-media.h:
8495 * gst/rtsp-server/rtsp-session-pool.h:
8496 Cleanups and doc updates
8497 Add some more documentation and do some minor cleanups here and there.
8499 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8501 * gst/rtsp-server/rtsp-client.c:
8502 * gst/rtsp-server/rtsp-media-factory.c:
8503 * gst/rtsp-server/rtsp-media-factory.h:
8504 * gst/rtsp-server/rtsp-media.c:
8505 * gst/rtsp-server/rtsp-media.h:
8506 * gst/rtsp-server/rtsp-session.c:
8507 * gst/rtsp-server/rtsp-session.h:
8509 Rename GstRTSPMediaBin to GstRTSPMedia
8510 Parse the request url into a GstRTSPUri object and pass this object to the
8511 various handlers and methods that require the uri.
8513 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8517 Add some more docs and remove some old code from the example.
8519 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8521 * gst/rtsp-server/rtsp-client.c:
8522 Handle state change failures better
8523 Handle state change failures better when changing the state of the pipeline to
8526 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8528 * gst/rtsp-server/rtsp-media-factory.c:
8529 * gst/rtsp-server/rtsp-media-factory.h:
8530 Make element creation more extendible
8531 Add get_element vmethod to the default MediaFactory so that subclasses can just
8532 override that method and still use the default logic for making a MediaBin from
8535 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8538 * gst/rtsp-server/Makefile.am:
8539 * gst/rtsp-server/rtsp-client.c:
8540 * gst/rtsp-server/rtsp-client.h:
8541 * gst/rtsp-server/rtsp-media-factory.c:
8542 * gst/rtsp-server/rtsp-media-factory.h:
8543 * gst/rtsp-server/rtsp-media-mapping.c:
8544 * gst/rtsp-server/rtsp-media-mapping.h:
8545 * gst/rtsp-server/rtsp-media.c:
8546 * gst/rtsp-server/rtsp-media.h:
8547 * gst/rtsp-server/rtsp-server.c:
8548 * gst/rtsp-server/rtsp-server.h:
8549 * gst/rtsp-server/rtsp-session.c:
8550 * gst/rtsp-server/rtsp-session.h:
8551 Make the server handle arbitrary pipelines
8552 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
8553 The GstMediaBin object has a handle to a bin with elements and to a list of
8554 GstMediaStream objects that this bin produces.
8555 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
8556 with methods to register and remove those mappings.
8557 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
8558 used by the server instance.
8559 Modify the example application so that it shows how to create custom pipelines
8560 attached to a specific mount point.
8561 Various misc cleanps.
8563 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8565 * gst/rtsp-server/rtsp-server.c:
8566 * gst/rtsp-server/rtsp-server.h:
8567 Allow setting a custom media factory for a server
8569 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8571 * gst/rtsp-server/rtsp-client.c:
8572 * gst/rtsp-server/rtsp-client.h:
8573 Allow setting a custom media factory for a client.
8575 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8577 * gst/rtsp-server/Makefile.am:
8578 Add Makefile entry for the media factory
8580 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8582 * gst/rtsp-server/rtsp-media-factory.c:
8583 * gst/rtsp-server/rtsp-media-factory.h:
8584 Add media factory to map urls to media pipeline objects.
8586 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8588 * gst/rtsp-server/rtsp-media.c:
8589 * gst/rtsp-server/rtsp-media.h:
8590 Add comments. Remove unused field
8592 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8594 * gst/rtsp-server/rtsp-session-pool.c:
8595 * gst/rtsp-server/rtsp-session-pool.h:
8596 Allow custom session pools to override the session id allocation algorithms Add some comments.
8598 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8600 * gst/rtsp-server/rtsp-session.h:
8603 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8605 * gst/rtsp-server/rtsp-client.c:
8606 * gst/rtsp-server/rtsp-client.h:
8607 Move the connection code in one place Add some comments
8609 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8611 * gst/rtsp-server/rtsp-server.c:
8612 * gst/rtsp-server/rtsp-server.h:
8613 Make vmethod to create and accept new clients. Add some docs.
8615 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8617 * gst/rtsp-server/rtsp-server.c:
8618 * gst/rtsp-server/rtsp-server.h:
8619 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
8621 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8623 * gst/rtsp-server/rtsp-client.c:
8624 * gst/rtsp-server/rtsp-client.h:
8625 Name the parameters more appropriately.
8627 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8629 * gst/rtsp-server/rtsp-session-pool.c:
8630 Do some more cleanup of the session pool.
8632 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8634 * gst/rtsp-server/Makefile.am:
8635 * gst/rtsp-server/rtsp-client.c:
8636 Check if return value of gst_rtsp_session_get_media is not NULL
8638 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8640 * gst/rtsp-server/Makefile.am:
8641 Install rtsp-session and rtsp-session-pool headers
8643 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8648 * bindings/python/Makefile.am:
8649 * bindings/python/arg-types.py:
8650 * bindings/python/codegen/Makefile.am:
8651 * bindings/python/codegen/__init__.py:
8652 * bindings/python/codegen/argtypes.py:
8653 * bindings/python/codegen/code-coverage.py:
8654 * bindings/python/codegen/codegen.py:
8655 * bindings/python/codegen/definitions.py:
8656 * bindings/python/codegen/defsparser.py:
8657 * bindings/python/codegen/docextract.py:
8658 * bindings/python/codegen/docgen.py:
8659 * bindings/python/codegen/fileprefix.override:
8660 * bindings/python/codegen/fileprefixmodule.c:
8661 * bindings/python/codegen/h2def.py:
8662 * bindings/python/codegen/mergedefs.py:
8663 * bindings/python/codegen/mkskel.py:
8664 * bindings/python/codegen/override.py:
8665 * bindings/python/codegen/reversewrapper.py:
8666 * bindings/python/codegen/scmexpr.py:
8667 * bindings/python/rtspserver-types.defs:
8668 * bindings/python/rtspserver.defs:
8669 * bindings/python/rtspserver.override:
8670 * bindings/python/rtspservermodule.c:
8672 Add python bindings.
8674 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8676 * bindings/Makefile.am:
8678 Don't go into python dir when requirements for python bindings are missing
8680 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8682 * bindings/Makefile.am:
8683 * bindings/vala/Makefile.am:
8685 Install Vala bindings if vala is available
8687 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8689 * bindings/vala/gst-rtsp-server-0.10.deps:
8690 * bindings/vala/gst-rtsp-server-0.10.vapi:
8691 * bindings/vala/gst-rtsp-server.vapi:
8692 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
8693 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8694 * bindings/vala/packages/gst-rtsp-server-0.10.files:
8695 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8696 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8697 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
8698 * bindings/vala/packages/gst-rtsp-server.deps:
8699 * bindings/vala/packages/gst-rtsp-server.excludes:
8700 * bindings/vala/packages/gst-rtsp-server.files:
8701 * bindings/vala/packages/gst-rtsp-server.gi:
8702 * bindings/vala/packages/gst-rtsp-server.metadata:
8703 * bindings/vala/packages/gst-rtsp-server.namespace:
8704 Regenerated Vala bindings
8706 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8708 * bindings/vala/gst-rtsp-server.vapi:
8709 * bindings/vala/packages/gst-rtsp-server.metadata:
8710 Fixed typo in included headers for vala bindings
8712 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8716 * pkgconfig/Makefile.am:
8717 * pkgconfig/gst-rtsp-server.pc.in:
8718 Added pkgconfig file
8720 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8722 * bindings/vala/gst-rtsp-server.vapi:
8723 * bindings/vala/packages/gst-rtsp-server.excludes:
8724 * bindings/vala/packages/gst-rtsp-server.gi:
8725 * bindings/vala/packages/gst-rtsp-server.metadata:
8726 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
8728 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8730 * bindings/vala/gst-rtsp-server.vapi:
8731 * bindings/vala/packages/gst-rtsp-server.deps:
8732 * bindings/vala/packages/gst-rtsp-server.files:
8733 * bindings/vala/packages/gst-rtsp-server.gi:
8734 * bindings/vala/packages/gst-rtsp-server.metadata:
8735 * bindings/vala/packages/gst-rtsp-server.namespace:
8738 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
8740 * gst/rtsp-server/rtsp-session.c:
8741 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
8743 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8745 * examples/Makefile.am:
8746 * gst/rtsp-server/Makefile.am:
8747 Put GStreamer version in library name
8749 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8751 * examples/Makefile.am:
8752 * gst/rtsp-server/Makefile.am:
8753 Fix some issues to pass distcheck
8755 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8757 * gst/rtsp-server/rtsp-server.c:
8758 Added port property to GstRTSPServer class.
8760 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8765 * examples/Makefile.am:
8768 * gst/rtsp-server/Makefile.am:
8769 * gst/rtsp-server/rtsp-client.c:
8770 * gst/rtsp-server/rtsp-client.h:
8771 * gst/rtsp-server/rtsp-media.c:
8772 * gst/rtsp-server/rtsp-media.h:
8773 * gst/rtsp-server/rtsp-server.c:
8774 * gst/rtsp-server/rtsp-server.h:
8775 * gst/rtsp-server/rtsp-session-pool.c:
8776 * gst/rtsp-server/rtsp-session-pool.h:
8777 * gst/rtsp-server/rtsp-session.c:
8778 * gst/rtsp-server/rtsp-session.h:
8781 * src/rtsp-client.c:
8782 * src/rtsp-client.h:
8785 * src/rtsp-server.c:
8786 * src/rtsp-server.h:
8787 * src/rtsp-session-pool.c:
8788 * src/rtsp-session-pool.h:
8789 * src/rtsp-session.c:
8790 * src/rtsp-session.h:
8791 Split in library and example program
8793 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8795 * src/rtsp-client.h:
8796 Removed obsolete variable
8798 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8800 * src/rtsp-client.c:
8801 * src/rtsp-client.h:
8802 Removed pipeline variable GstRTSPClient, because it's only used in one function
8804 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8807 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
8809 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
8811 * src/rtsp-session.c:
8812 Initialize some more vars.
8814 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
8816 * src/rtsp-session.c:
8817 Initialize variable to avoid compiler warning.
8819 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
8822 Add a reasonable generic .gitignore