3 2015-06-07 Sebastian Dröge <slomo@coaxion.net>
8 2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
10 * gst/rtsp-server/rtsp-client.c:
11 rtsp-client: No flush during Teardown.
12 When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
13 backlog is empty it can happen that just a part of a message will be
14 sent and rest is in backlog queue. If then flush during teardown
15 just a part of message will be sent.This can lead to client miss
16 teardown response since it expect to get the last part of message.
17 The flushing during teardown was introduced to fix a deadlock that now
18 is fixed more generally in handle_request by temporary setting backlog
20 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
22 2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
24 * tests/check/Makefile.am:
25 tests: Use AM_TESTS_ENVIRONMENT
26 Needed by the new automake test runner and the
27 current version of the common submodule.
29 2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
31 * gst/rtsp-server/rtsp-media.h:
32 * gst/rtsp-server/rtsp-stream.h:
33 rtsp-server: Use single-include rtsp header to make sure we get all definitions
35 2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
37 * gst/rtsp-server/rtsp-media.c:
38 rtsp-media: Mark some more functions static
40 2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
42 * gst/rtsp-server/rtsp-media.c:
43 rtsp-media: Only unblock the media in suspend() when actually changing the state
44 Otherwise we're going to lose a few packets for live streams during DESCRIBE.
46 2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
48 * examples/test-video-rtx.c:
49 examples: Use AVPF profile for the RTX example
51 2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
53 * gst/rtsp-server/rtsp-sdp.c:
54 rtsp-sdp: Only add RTX to the SDP when using a feedback profile
56 2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
58 * gst/rtsp-server/rtsp-stream.c:
59 rtsp-stream: get valid clock-rate from last-sample
60 clock-rate in last-sample's caps is integer, not unsigned.
61 To get this value properly, variable needs to be type-casted to int.
62 https://bugzilla.gnome.org/show_bug.cgi?id=747614
64 2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
68 autogen.sh: only run autopoint if gettext requested in configure.ac
69 Not just because there happens to be a po directory.
70 https://bugzilla.gnome.org/show_bug.cgi?id=748058
72 2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
75 Revert "configure.ac: uncomment gettext version setup"
76 This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
77 We don't need a gettext setup here and there's no po
78 directory either, so no reason why autopoint would be
79 run in the first place.
80 See https://bugzilla.gnome.org/show_bug.cgi?id=748058
82 2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
84 * examples/test-multicast.c:
85 * examples/test-multicast2.c:
86 * examples/test-sdp.c:
87 * examples/test-video-rtx.c:
88 * examples/test-video.c:
89 * tests/test-cleanup.c:
91 Fix timeout function signatures across tests and examples
93 2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
95 * tests/check/Makefile.am:
96 tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
97 Make sure the test environment is set up.
98 https://bugzilla.gnome.org//show_bug.cgi?id=747624
100 2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
103 configure: bump automake requirement to 1.14 and autoconf to 2.69
104 This is only required for builds from git, people can still
105 build tarballs if they only have older autotools.
106 https://bugzilla.gnome.org//show_bug.cgi?id=747624
108 2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
111 configure.ac: uncomment gettext version setup
112 Fixes autogen.sh. It would run autopoint, which would complain
113 that it could not find the gettext version in configure.ac.
114 https://bugzilla.gnome.org/show_bug.cgi?id=748058
116 2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
118 * examples/test-video-rtx.c:
119 test-video-rtx: set exact payload type to PCMA payloader
120 Setting wrong payload type causes failure to do retransmission through audio stream
121 https://bugzilla.gnome.org/show_bug.cgi?id=747839
123 2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
125 * gst/rtsp-server/rtsp-media.c:
126 * gst/rtsp-server/rtsp-stream.c:
127 * gst/rtsp-server/rtsp-stream.h:
128 rtsp-stream: fix to get valid each stream data for request-aux-sender signal
129 Because of duplicated g_signal_connect for request-aux-sender signal,
130 wrong stream pointer is passed to the signal handler.
131 Instead of passing each stream, pass stream array and get the relevant stream.
132 https://bugzilla.gnome.org/show_bug.cgi?id=747839
134 2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
138 Update autogen.sh to latest version from common
139 Fixes build after aclocal_check etc. helpers have been removed.
141 2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
144 Automatic update of common submodule
145 From bc76a8b to c8fb372
147 2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
149 * gst/rtsp-server/rtsp-stream.c:
150 rtsp-stream: Limit the queues to 1 buffer
151 We only need them to be able to pre-roll, queueing up more data here
152 is only going to harm latency and memory usage.
154 2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
156 * gst/rtsp-server/rtsp-stream.c:
157 rtsp-stream: Update comment and ASCII art to the latest code
158 We have a queue in front of the udpsink too to prevent the pipeline from
161 2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
163 * gst/rtsp-server/rtsp-stream.c:
164 rtsp-media: Properly return first rtptime
165 Instead we where returning first GstBuffer timestamp. This would result
166 in clock skew and unwanted behaviour in RTSP playback.
167 https://bugzilla.gnome.org/show_bug.cgi?id=746479
169 2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
171 * gst/rtsp-server/rtsp-stream.c:
172 rtsp-stream: Don't leave buffer mapped
173 If the seq is NULL, the RTP buffer was left mapped. We should always
176 2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
181 2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
183 * gst/rtsp-server/rtsp-media-factory.c:
184 * tests/check/gst/client.c:
185 Fix double semicolons
187 2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
189 * gst/rtsp-server/rtsp-stream.c:
190 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
191 This gives more accurate values than asking the payloader. There might be
192 queueing happening between the payloader and the sink.
193 https://bugzilla.gnome.org/show_bug.cgi?id=745704
195 2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
197 * gst/rtsp-server/rtsp-media.c:
198 rtsp-media: Don't seek for PLAY if the position will not change
199 https://bugzilla.gnome.org/show_bug.cgi?id=745704
201 2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
203 * gst/rtsp-server/rtsp-media.c:
204 rtsp-media: Don't include payload type in the caps for framesize
205 When the sdp media attribute framesize are converted to caps
206 the <payload> should not be included.
207 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
208 Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
210 2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
212 * gst/rtsp-server/rtsp-sdp.c:
213 rtsp-sdp: add payload type to the sdp framesize attribute
214 The sdp framesize attribute is desribed in RFC6064. It is specified
215 for payloading of H263 and has the following form
216 a=framesize:<payload type> <width>-<height>. The <width>-<height> part
217 should be added to the caps in a payloader and the <payload type> should
218 be added by the rtsp-server.
219 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
221 2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
223 * examples/test-uri.c:
224 examples: test-uri: fix tainted variable
225 Insignificant but this keeps Coverity happy.
228 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
230 * examples/.gitignore:
231 * examples/Makefile.am:
232 * examples/test-netclock-client.c:
233 * examples/test-netclock.c:
234 examples: Add a simple example of network synch for live streams.
235 An example server and client that works for synchronising live streams
236 only - as it can't support pause/play.
238 2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
240 * gst/rtsp-server/rtsp-media-factory.c:
241 * gst/rtsp-server/rtsp-media-factory.h:
242 rtsp-media-factory: Add functions to set/get the media gtype
243 Allow specifying the GType of a GstRtspMedia subclass to create
244 as a simpler way to get the factory to create a custom
245 GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
247 2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
249 * gst/rtsp-server/rtsp-media.c:
250 rtsp-media: fix double unlock in _get_buffer_size()
251 Fixes an abort when calling gst_rtsp_media_get_buffer_size()
252 because of double g_mutex_unlock () usage.
253 https://bugzilla.gnome.org/show_bug.cgi?id=745434
255 2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
257 * gst/rtsp-server/rtsp-session-pool.c:
258 * gst/rtsp-server/rtsp-session.c:
259 * gst/rtsp-server/rtsp-session.h:
260 rtsp-session: Use monotonic time for RTSP session timeout
261 Changed RTSP session timeout handling to monotonic time
262 and deprecating the API for current system time.
263 This fixes timeouts when the system time changes.
264 https://bugzilla.gnome.org/show_bug.cgi?id=743346
266 2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
268 * gst/rtsp-server/rtsp-client.c:
269 * gst/rtsp-server/rtsp-media.c:
270 rtsp-client: Only error out in PLAY if seeking actually failed
271 If the media was just not seekable, we continue from whatever position we are
272 and let the client decide if that is what is wanted or not.
273 Only if the actual seek failed, we can't really recover and should error out.
275 2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
277 * gst/rtsp-server/rtsp-stream.c:
278 rtsp-stream: Add necessary queues between tee and multiudpsink
279 https://bugzilla.gnome.org/show_bug.cgi?id=744379
281 2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
283 * gst/rtsp-server/rtsp-client.c:
284 * gst/rtsp-server/rtsp-media.c:
285 rtsp-media: If seeking fails, don't wait forever for the media to preroll again
286 Instead error out properly the same way as if the SEEKING query already
289 2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
291 * gst/rtsp-server/rtsp-stream.h:
292 rtsp-stream: minor code formatting fix
294 2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
296 * gst/rtsp-server/rtsp-media.c:
297 rtsp-media: fix logic for collect_streams
298 Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
299 all streams it knows if it got any, and can check if the transport mode is OK.
302 2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
304 * gst/rtsp-server/rtsp-media.c:
305 rtsp-media: Don't set the transport mode based on what elements we find
306 Just print a warning if the one that was set before disagrees with what
307 elements we found. It must already be set to something before as this
308 function is called after we received the SDP from ANNOUNCE in RECORD mode,
309 and we would reject ANNOUNCE if the RECORD flag was not set.
311 2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
313 * tests/check/gst/rtspserver.c:
314 tests: rtspserver: rename shadowed variable
315 We have two different 'sink' variables here,
316 rename one of them for clarity.
318 2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
320 * gst/rtsp-server/rtsp-client.c:
321 rtsp-client: fix awkward if clause
323 2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
325 * examples/test-uri.c:
326 examples: test-uri: improve uri argument handling and accept file names
327 Print an error if the argument passed is not a URI and can't
328 be converted into one, or no arguments have been provided.
330 2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
332 * examples/test-uri.c:
333 examples: test-uri: don't remove mount point after 10 seconds
334 It's very irritating when trying to test stuff repeatedly
335 and serves no real purpose other than showing that it can
338 2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
340 * examples/.gitignore:
341 examples: add new test-record to .gitignore
343 2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
345 * examples/test-record.c:
346 * gst/rtsp-server/rtsp-client.c:
347 * gst/rtsp-server/rtsp-media-factory.c:
348 * gst/rtsp-server/rtsp-media-factory.h:
349 * gst/rtsp-server/rtsp-media.c:
350 * gst/rtsp-server/rtsp-media.h:
351 * tests/check/gst/rtspserver.c:
352 rtsp-media: Use flags to distinguish between PLAY and RECORD media
354 2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
356 * examples/test-record.c:
357 test-record: Set latency for playback-style example to 2s instead of 200ms
359 2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
361 * tests/check/gst/rtspserver.c:
362 tests: add some unit tests for ANNOUNCE and RECORD
363 https://bugzilla.gnome.org/show_bug.cgi?id=743175
365 2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
367 * gst/rtsp-server/rtsp-client.c:
368 rtsp-client: fix a couple of leaks in handle_announce
370 2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
372 * gst/rtsp-server/rtsp-media-factory.c:
373 * gst/rtsp-server/rtsp-media-factory.h:
374 * gst/rtsp-server/rtsp-media.c:
375 * gst/rtsp-server/rtsp-media.h:
376 rtsp-media: Expose latency setting for setting the rtpbin latency
378 2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
380 * examples/test-record.c:
381 test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
383 2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
385 * gst/rtsp-server/rtsp-stream.c:
386 rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
388 2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
390 * examples/Makefile.am:
391 * examples/test-record.c:
392 * gst/rtsp-server/rtsp-client.c:
393 * gst/rtsp-server/rtsp-client.h:
394 * gst/rtsp-server/rtsp-media-factory.c:
395 * gst/rtsp-server/rtsp-media-factory.h:
396 * gst/rtsp-server/rtsp-media.c:
397 * gst/rtsp-server/rtsp-media.h:
398 * gst/rtsp-server/rtsp-session-media.c:
399 * gst/rtsp-server/rtsp-stream.c:
400 * gst/rtsp-server/rtsp-stream.h:
401 Add initial support for RECORD
402 We currently only support media that is RECORD or PLAY only, not both at once.
403 https://bugzilla.gnome.org/show_bug.cgi?id=743175
405 2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
407 * gst/rtsp-server/rtsp-stream.c:
408 rtsp-stream: RTCP and RTP transport cache cookies seperated
409 RTCP packets were not sent because the same tr_cache_cookie was used for
410 both RTP and RTCP. So only one of the tr_cache lists were populated
411 depending on which one was sent first. If the tr_cache list is not
412 populated then no packets can be sent. Most often this happened to be
413 RTCP. Now seperate RTCP and RTP transport cache cookies are added which
414 resulted in both the tr_cache_lists to be populated regardless of which
416 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
418 2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
420 * gst/rtsp-server/rtsp-stream.c:
421 rtsp-stream: fix false compiler warning
422 rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
424 2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
426 * gst/rtsp-server/rtsp-client.c:
427 rtsp-client: log interleaved data received
429 2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
431 * gst/rtsp-server/rtsp-client.c:
432 rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
434 2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
436 * gst/rtsp-server/rtsp-client.c:
437 rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
439 2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
441 * gst/rtsp-server/rtsp-client.c:
442 rtsp-client: Use a random session ID in the SDP
443 RFC4566 Section 5.2 says that it should make the username, session id,
444 nettype, addrtype and unicast address tuple globally unique. Always using
445 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
446 Instead let's create a 64 bit random number, which at least brings us
447 closer to the goal of global uniqueness.
448 https://tools.ietf.org/html/rfc4566#section-5.2
450 2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
452 * examples/test-launch.c:
453 * examples/test-mp4.c:
454 * examples/test-ogg.c:
455 * examples/test-uri.c:
456 examples: Don't call gst_init() and gst_get_option_group()
457 The latter calls the former at the appropriate time.
459 2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
461 * gst/rtsp-server/rtsp-client.c:
462 rtsp-client: Drop trailing \0 of RTSP DATA messages
463 We add a trailing \0 in GstRTSPConnection to make parsing of
464 string message bodies easier (e.g. the SDP from DESCRIBE) but
465 for actual data this means we have to drop it or otherwise
468 2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
470 * gst/rtsp-server/rtsp-stream.c:
471 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
472 Fixes crash when two threads access handle_new_sample() at the same
473 time, one for RTP, one for RTCP.
474 Otherwise, when iterating over the transports cache, it might be modified by
475 another thread at the same time if the transports cookie has changed.
476 https://bugzilla.gnome.org/show_bug.cgi?id=742954
478 2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
480 * gst/rtsp-server/rtsp-stream.c:
481 rtsp-stream: Set format=TIME on our app sources for TCP
483 2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
485 * gst/rtsp-server/rtsp-session-pool.c:
486 Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
487 This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
488 RFC 2326 states that session IDs may consist of alphanumeric as well as
489 the safe characters $-_.+ -- N.B. the percent character is not allowed.
490 Previously the session ID was URI-escaped, this meant that any character
491 which was not alphanumeric or any of the characters +-._~ would be
492 percent encoded. While the RFC (surprisingly) mentions that linear white
493 space in session IDs should be URI-escaped, it does not say anything
494 about other characters. Moreover no white space is allowed in the
495 session ID. Finally the percent character which is the result of
496 URI-escaping is not allowed in a session ID.
497 So there is no reason to do any URI-escaping, and now it is removed.
498 https://bugzilla.gnome.org/show_bug.cgi?id=742869
500 2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
503 Automatic update of common submodule
504 From f2c6b95 to bc76a8b
506 2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
509 Fix 'make check' from top-level directory
511 2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
513 * examples/test-launch.c:
514 * examples/test-mp4.c:
515 * examples/test-ogg.c:
516 * examples/test-uri.c:
517 examples: Add command-line parsing and take a 'port' argument
518 This allows users to run multiple servers on different ports for testing.
519 Only done for examples that actually take arguments and hence are capable of
520 outputting different streams for each instance on each port.
521 https://bugzilla.gnome.org/show_bug.cgi?id=742115
523 2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
525 * gst/rtsp-server/rtsp-client.c:
526 * gst/rtsp-server/rtsp-client.h:
527 rtsp-client: Add a send_message default signal handler
528 This allows subclasses to easily hook into the response sending
529 mechanism without doing everything from a signal, which seems
530 awkward from subclasses.
532 2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
535 Automatic update of common submodule
536 From ef1ffdc to f2c6b95
538 2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
542 configure: add --disable-examples switch
543 https://bugzilla.gnome.org/show_bug.cgi?id=741678
545 2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
547 * examples/.gitignore:
548 * examples/Makefile.am:
549 * examples/test-video-rtx.c:
550 examples: add a retransmisison example implementing RFC4588
551 Currently only SSRC-multiplexed rtx streams are supported
553 2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
555 * gst/rtsp-server/rtsp-stream.c:
556 rtsp-stream: Fix some minor memory leaks
558 2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
560 * gst/rtsp-server/rtsp-media.c:
561 rtsp-media: Some minor cleanup
563 2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
565 * gst/rtsp-server/rtsp-stream.c:
566 rtsp-stream: Fix compiler warnings
567 rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
568 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
570 rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
571 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
574 2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
576 * docs/libs/gst-rtsp-server-sections.txt:
577 * gst/rtsp-server/rtsp-media-factory.c:
578 * gst/rtsp-server/rtsp-media-factory.h:
579 * gst/rtsp-server/rtsp-media.c:
580 * gst/rtsp-server/rtsp-media.h:
581 * gst/rtsp-server/rtsp-sdp.c:
582 * gst/rtsp-server/rtsp-stream.c:
583 * gst/rtsp-server/rtsp-stream.h:
584 media: implement ssrc-multiplexed retransmission support
585 based off RFC 4588 and the server-rtpaux example in -good
587 2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
589 * gst/rtsp-server/rtsp-client.c:
590 * gst/rtsp-server/rtsp-stream-transport.c:
591 * gst/rtsp-server/rtsp-stream.c:
592 rtsp: Ref transports in hash table.
593 Also ref streams for transports.
594 This solves a crash when reciving a rtcp after teardown but before
596 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
598 2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
601 Automatic update of common submodule
602 From 7bb2bce to ef1ffdc
604 2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
606 * gst/rtsp-server/rtsp-client.c:
607 client: refactor cleanup of cached media
609 2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
611 * tests/check/gst/client.c:
613 The session leak is now fixed, lets remove those FIXME comments.
615 2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
617 * tests/check/gst/rtspserver.c:
618 tests: Test to setup two sessions on one connection
619 https://bugzilla.gnome.org/show_bug.cgi?id=739112
621 2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
623 * tests/check/gst/rtspserver.c:
624 tests: Test setup with tcp transport
625 https://bugzilla.gnome.org/show_bug.cgi?id=739112
627 2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
629 * gst/rtsp-server/rtsp-client.c:
630 client: Configure transport after creating session media
631 The default implementation of configure_client_transport() in
632 rtsp-client uses the session media when it chooses channels for
634 https://bugzilla.gnome.org/show_bug.cgi?id=739112
636 2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
638 * gst/rtsp-server/rtsp-client.c:
639 * gst/rtsp-server/rtsp-session-media.c:
640 client: Stop caching media in client when doing setup
641 If the media has been managed by a session media, it should not be
642 cached in the client any longer. The GstRTSPSessionMedia object is now
643 responsible for unpreparing the GstRTSPMedia object using
644 gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
646 https://bugzilla.gnome.org/show_bug.cgi?id=739112
648 2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
650 * gst/rtsp-server/rtsp-stream.c:
651 rtsp-stream: unref srtp decoder when leaving bin
652 https://bugzilla.gnome.org/show_bug.cgi?id=739481
654 2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
656 * gst/rtsp-server/rtsp-client.c:
657 rtsp-client: mikey memory leaks
658 https://bugzilla.gnome.org/show_bug.cgi?id=739383
660 2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
663 Automatic update of common submodule
664 From 84d06cd to 7bb2bce
666 2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
669 Parallelise 'make check-valgrind'
671 2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
674 Automatic update of common submodule
675 From a8c8939 to 84d06cd
677 2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
680 Automatic update of common submodule
681 From 36388a1 to a8c8939
683 2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
685 * gst/rtsp-server/rtsp-media.c:
686 rtsp-media: deactivate media when shutting down from paused
687 This was only done when going directly from playing.
688 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
690 2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
692 * gst/rtsp-server/rtsp-client.c:
693 * gst/rtsp-server/rtsp-context.h:
694 rtsp-client: add stream transport to context
695 We add the stream transport to the context so we can get the configured
696 client stream transport in the setup request signal.
697 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
699 2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
701 * gst/rtsp-server/rtsp-stream.c:
702 stream: release lock even not all transports have been removed
703 We don't want to keep the lock even we return FALSE because not all the
704 transports have been removed. This could lead into a deadlock.
705 https://bugzilla.gnome.org/show_bug.cgi?id=737797
707 2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
709 * gst/rtsp-server/rtsp-sdp.c:
710 rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
711 These were renamed in GstRTPBasePayload in 1.0
713 2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
715 * gst/rtsp-server/rtsp-client.c:
716 client: set session media to NULL without the lock
717 We need to set session medias to NULL without the client lock otherwise
718 we can end up in a deadlock if another thread is waiting for the lock
719 and media unprepare is also waiting for that thread to end.
720 https://bugzilla.gnome.org/show_bug.cgi?id=737690
722 2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
724 * gst/rtsp-server/rtsp-media.c:
725 rtsp-media: Set state to UNPREPARING in all cases
727 2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
729 * gst/rtsp-server/rtsp-media.c:
730 media: set state to unpreparing when unprepare is initiated
731 https://bugzilla.gnome.org/show_bug.cgi?id=737675
733 2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
735 * gst/rtsp-server/rtsp-client.c:
736 rtsp-client: Remove backlog limit while processings requests
737 If the backlog limit is kept two cases of deadlocks may be
738 encountered when streaming over TCP. Without the backlog
739 limit this deadlocks can not happen, at the expence of
741 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
743 2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
745 * gst/rtsp-server/rtsp-client.c:
746 rtsp-client: do not free main context before rtsp watch
747 https://bugzilla.gnome.org/show_bug.cgi?id=737110
749 2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
751 * tests/check/gst/rtspserver.c:
752 tests: Extend unit test timeout to accomodate for valgrind
753 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
755 2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
757 * gst/rtsp-server/rtsp-client.c:
758 * gst/rtsp-server/rtsp-session.c:
759 * gst/rtsp-server/rtsp-stream-transport.c:
760 rtsp-*: Treat sending packets to clients as keepalive
761 As long as gst-rtsp-server can successfully send RTP/RTCP data to
762 clients then the client must be reading. This change makes the server
763 timeout the connection if the client stops reading.
764 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
766 2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
768 * gst/rtsp-server/rtsp-client.c:
769 rtsp-client: Allow backlog to grow while expiring session
770 Allow the send backlog in the RTSP watch to grow to unlimited size while
771 attempting to bring the media pipeline to NULL due to a session
772 expiring. Without this change the appsink element cannot change state
773 because it is blocked while rendering data in the new_sample callback.
774 This callback will block until it has successfully put the data into the
775 send backlog. There is a chance that the send backlog is full at this
776 point which means that the callback may block for a long time, possibly
777 forever. Therefore the media pipeline may also be prevented from
778 changing state for a long time.
779 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
781 2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
783 * gst/rtsp-server/rtsp-client.c:
784 rtsp-client: Make old compilers happy
785 rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
786 Just in case that guint8 doesn't fit in a pointer. Just in case ...
788 2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
790 * gst/rtsp-server/rtsp-client.c:
791 client: raise the backlog limits before pausing
792 We need to raise the backlog limits before pausing the pipeline or else
793 the appsink might be blocking in the render method in wait_backlog() and
794 we would deadlock waiting for paused.
795 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
797 2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
799 * gst/rtsp-server/rtsp-client.c:
800 client: make define for the WATCH_BACKLOG
801 See https://bugzilla.gnome.org/show_bug.cgi?id=736322
803 2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
805 * gst/rtsp-server/rtsp-client.c:
806 client: simplify session transport handling
807 link/unlink of the transport in a session was done to keep track of all
808 TCP transports and to send RTP/RTCP data to the streams. We can simplify
809 that by putting all the TCP transports in a hashtable indexed with the
811 We also don't need to link/unlink the transports when we pause/resume
812 the streams. The same effect is already achieved when we pause/play the
813 media. Indeed, when we pause the media, the transport is removed from
814 the media and the callbacks will not be called anymore.
815 See https://bugzilla.gnome.org/show_bug.cgi?id=736041
817 2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
819 * gst/rtsp-server/rtsp-stream-transport.c:
820 * gst/rtsp-server/rtsp-stream-transport.h:
821 stream-transport: make method to handle received data
822 Make a method to handle the data received on a channel. It sends the
823 data to the stream of the transport on the RTP or RTCP pads based on
826 2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
828 * examples/test-mp4.c:
829 test: add example of dumping RTCP reports
831 2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
833 * gst/rtsp-server/rtsp-media.c:
834 * gst/rtsp-server/rtsp-stream.c:
835 * gst/rtsp-server/rtsp-stream.h:
836 rtsp-media: Make sure that sequence numbers are monotonic after pause
837 The sequence number is not monotonic for RTP packets after pause. The
838 reason is basepayloader generates a randon sequence number when the
839 pipeline goes from ready to pause. With this fix generation of sequence
840 number will be monotonic when going from pause to play request.
841 https://bugzilla.gnome.org/show_bug.cgi?id=736017
843 2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
845 * gst/rtsp-server/rtsp-client.c:
846 rtsp-client: Protect saved clients watch with a mutex
847 Fixes a crash when close() is called while merging clients
848 in handle_tunnel(). In that case close() would destroy the
849 watch while it is still being used in handle_tunnel().
850 https://bugzilla.gnome.org/show_bug.cgi?id=735570
852 2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
854 * gst/rtsp-server/rtsp-stream.c:
855 rtsp-stream: Remove the multicast group udp sources when removing from the bin
857 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
859 * gst/rtsp-server/rtsp-media.c:
860 * gst/rtsp-server/rtsp-stream.c:
861 * gst/rtsp-server/rtsp-stream.h:
862 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
863 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
864 seeking and will always continue counting the time. This leads to
865 the NPT after a backwards seek to be something completely different
866 to the actual seek position.
867 https://bugzilla.gnome.org/show_bug.cgi?id=732644
869 2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
871 * examples/test-appsrc.c:
872 examples: fix another reference leak
873 gst_rtsp_media_get_element() returns a new ref.
875 2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
877 * examples/test-appsrc.c:
878 examples: unref element after usage
879 gst_bin_get_by_name_recurse_up() returns an element
880 reference that must be unreffed after usage.
881 https://bugzilla.gnome.org/show_bug.cgi?id=734546
883 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
885 * gst/rtsp-server/rtsp-media.c:
886 signals: Fix copy-pasto in target-state signal offset
888 2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
892 Makefile: Add usage of build-checks step
893 Allows building checks without running them
895 2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
897 * gst/rtsp-server/rtsp-stream.c:
898 rtsp-stream: Listen on the multicast group for RTP/RTCP packets
899 When a UDP multicast transport is used it is expected that the server listens
900 for RTP and RTCP packets on the multicast group with the corresponding port.
901 Without this we will never get RTCP packets from clients in multicast mode.
902 https://bugzilla.gnome.org/show_bug.cgi?id=732238
904 2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
909 === release 1.4.0 ===
911 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
917 * gst-rtsp-server.doap:
920 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
922 * gst/rtsp-server/rtsp-media.h:
923 media: correct misspelled words in description
924 https://bugzilla.gnome.org/show_bug.cgi?id=733244
926 === release 1.3.91 ===
928 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
934 * gst-rtsp-server.doap:
937 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
939 * docs/libs/gst-rtsp-server-sections.txt:
942 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
944 * gst/rtsp-server/rtsp-server.c:
945 server: implement client REMOVE filter
947 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
949 * gst/rtsp-server/rtsp-client.c:
950 * gst/rtsp-server/rtsp-client.h:
951 client: expose _close() method
952 Expose a previously internal close method to close the client
955 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
957 * gst/rtsp-server/rtsp-session-pool.c:
958 session-pool: signal session-removed outside of the lock
959 Release the lock before emiting the session-removed signal.
961 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
963 * gst/rtsp-server/rtsp-client.c:
964 * gst/rtsp-server/rtsp-server.c:
965 * gst/rtsp-server/rtsp-session-pool.c:
966 * gst/rtsp-server/rtsp-session.c:
967 * gst/rtsp-server/rtsp-stream.c:
968 filter: Release lock in filter functions
969 Release the object lock before calling the filter functions. We need to
970 keep a cookie to detect when the list changed during the filter
971 callback. We also keep a hashtable to make sure we only call the filter
972 function once for each object in case of concurrent modification.
973 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
975 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
977 * gst/rtsp-server/rtsp-client.c:
978 client: check if watch is set in handle_teardown()
979 The unit tests run without a watch
981 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
983 * tests/check/gst/client.c:
984 client tests: send teardown to cleanup session
986 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
988 * tests/check/gst/rtspserver.c:
989 server tests: send teardown to cleanup session
991 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
993 * gst/rtsp-server/rtsp-client.c:
994 client: keep ref to client for the session removed handler
995 This extra ref will be dropped when all client sessions have been
996 removed. A session is removed when a client sends teardown, closes its
997 endpoint of the TCP connection or the sessions expires.
998 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1000 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
1002 * gst/rtsp-server/rtsp-client.c:
1003 * gst/rtsp-server/rtsp-session.c:
1004 * tests/check/gst/client.c:
1005 client: manage media in session as a last step
1006 Once we manage a media in a session, we can't unmanage it anymore
1007 without destroying it. Therefore, first check everything before we
1008 manage the media, otherwise if something is wrong we have no way to
1010 If we created a new session and something went wrong, remove the session
1011 again. Fixes a leak in the unit test.
1013 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
1015 * examples/test-mp4.c:
1016 * examples/test-ogg.c:
1017 examples: print 'stream ready at url' for mp4 and ogg example
1019 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
1021 * gst/rtsp-server/rtsp-client.c:
1022 * gst/rtsp-server/rtsp-sdp.c:
1023 rtsp: fix for MIKEY api change
1025 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
1027 * gst/rtsp-server/rtsp-client.c:
1028 client: free watch context only once
1029 The watch context is freed when the source is destroyed. Avoids
1030 a CRITICAL when we try to unref the context twice.
1032 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
1034 * gst/rtsp-server/rtsp-client.c:
1037 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
1039 * gst/rtsp-server/rtsp-client.c:
1040 client: protect sessions with lock
1041 Protect the list of sessions with the lock.
1042 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
1044 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
1046 * gst/rtsp-server/rtsp-client.c:
1047 Client: keep a ref to the session
1048 Don't just keep a weak ref to the session objects but use a hard ref. We
1049 will be notified when a session is removed from the pool (expired) with
1050 the new session-removed signal.
1051 Don't automatically close the RTSP connection when all the sessions of
1052 a client are removed, a client can continue to operate and it can create
1053 a new session if it wants. If you want to remove the client from the
1054 server, you have to use gst_rtsp_server_client_filter() now.
1055 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
1056 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
1058 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
1060 * gst/rtsp-server/rtsp-session-pool.c:
1061 * gst/rtsp-server/rtsp-session-pool.h:
1062 session-pool: add session-removed signal
1063 Add a signal to be notified when a session is removed from the pool.
1065 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
1067 * gst/rtsp-server/Makefile.am:
1068 * gst/rtsp-server/rtsp-server.h:
1069 Make rtsp-server.h a single-include header, use it for G-I
1070 https://bugzilla.gnome.org/show_bug.cgi?id=732411
1072 === release 1.3.90 ===
1074 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
1080 * gst-rtsp-server.doap:
1083 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
1085 * gst/rtsp-server/rtsp-stream.c:
1086 stream: crypto can be NULL
1088 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
1090 * gst/rtsp-server/rtsp-client.c:
1091 * gst/rtsp-server/rtsp-media.c:
1092 * gst/rtsp-server/rtsp-mount-points.c:
1093 introspection: add missing allow-none annotations
1094 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1096 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
1098 * gst/rtsp-server/rtsp-address-pool.c:
1099 * gst/rtsp-server/rtsp-media.c:
1100 * gst/rtsp-server/rtsp-session-media.c:
1101 * gst/rtsp-server/rtsp-session-pool.c:
1102 * gst/rtsp-server/rtsp-stream-transport.c:
1103 * gst/rtsp-server/rtsp-stream.c:
1104 * gst/rtsp-server/rtsp-token.c:
1105 introspection: add (nullable) annotations to return values
1106 https://bugzilla.gnome.org/show_bug.cgi?id=730952
1108 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
1110 * gst/rtsp-server/rtsp-client.c:
1111 * gst/rtsp-server/rtsp-stream.c:
1112 gi: improve annotations
1113 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
1115 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
1117 * gst/rtsp-server/rtsp-client.c:
1118 * gst/rtsp-server/rtsp-media-factory.c:
1119 * gst/rtsp-server/rtsp-media.c:
1120 * gst/rtsp-server/rtsp-server.c:
1121 signals: use generic marshal function
1122 Use the generic C marshal function.
1123 Use more explicit type instead of G_TYPE_POINTER
1125 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
1127 * gst/rtsp-server/rtsp-context.h:
1128 context: add type macro
1130 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
1132 * gst/rtsp-server/rtsp-client.c:
1133 * gst/rtsp-server/rtsp-sdp.c:
1134 * gst/rtsp-server/rtsp-sdp.h:
1135 sdp: hide key length defines
1136 They don't have a namespace.
1138 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
1143 === release 1.3.3 ===
1145 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
1151 * gst-rtsp-server.doap:
1154 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1156 * gst/rtsp-server/rtsp-client.c:
1157 * gst/rtsp-server/rtsp-sdp.c:
1158 * gst/rtsp-server/rtsp-sdp.h:
1159 mikey: add different key length parameters
1160 Add encryption and authentication key length parameters to MIKEY. For
1161 the encoders, the key lengths are obtained from the cipher and auth
1162 algorithms set in the caps. For the decoders, they are obtained while
1163 parsing the key management from the client.
1164 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
1166 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
1168 * tests/check/gst/stream.c:
1169 stream tests: Make sure we get right multicast address from stream
1170 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
1172 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
1174 * gst/rtsp-server/rtsp-client.c:
1175 client: ref the context until rtsp watch is alive
1176 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
1178 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
1180 * gst/rtsp-server/rtsp-client.c:
1181 client: Destroy the rtsp watch after connection close
1183 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
1185 * gst/rtsp-server/rtsp-media.c:
1186 media: fix confusing comment
1188 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
1190 * gst/rtsp-server/rtsp-session.c:
1191 rtsp-session: Timeout in header.
1192 Adding the possbilty to always have timout in header.
1193 This is configurabe with setting "timeout-always-visible".
1194 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
1196 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
1201 === release 1.3.2 ===
1203 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
1210 * gst-rtsp-server.doap:
1213 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
1216 Automatic update of common submodule
1217 From 211fa5f to 1f5d3c3
1219 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
1221 * gst/rtsp-server/rtsp-client.c:
1222 client: store TCP ports in transport
1223 Store the TCP ports in the transport when we are doing RTSP over TCP.
1224 This way, we can easily get to the ports from the transport.
1225 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
1227 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1229 * gst/rtsp-server/rtsp-stream.c:
1230 stream: add signals for new RTP/RTCP encoders
1231 New signals to allow the user to configure the dynamically created
1233 https://bugzilla.gnome.org/show_bug.cgi?id=730228
1235 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
1237 * gst/rtsp-server/rtsp-media.c:
1238 * gst/rtsp-server/rtsp-media.h:
1239 media: Make suspend()/unsuspend() virtual
1240 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
1242 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
1244 * gst/rtsp-server/rtsp-client.c:
1245 client: fix send-message signal marshaller
1246 Use generic marshalling for the send-message signal. It has
1247 two POINTER arguments, not just one.
1248 https://bugzilla.gnome.org/show_bug.cgi?id=729900
1250 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
1252 * tests/check/gst/media.c:
1253 tests: add and remove pads only once
1254 In this test we simulate a dynamic pad by watching the caps event.
1255 Because of renegotiation in the base payloader now, this caps is sent
1256 multiple times but we can only deal with 1 invocation, use a variable to
1257 only 'add and remove' the pad once.
1259 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
1261 * tests/check/gst/rtspserver.c:
1262 tests: add unit test for correct handling of Require headers
1263 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1265 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1267 * gst/rtsp-server/rtsp-client.c:
1268 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
1269 Servers must handle Require headers and must report a failure
1270 if they don't handle any of the Required options, see RFC 2326,
1271 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
1272 https://bugzilla.gnome.org/show_bug.cgi?id=729426
1274 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
1279 === release 1.3.1 ===
1281 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
1287 * gst-rtsp-server.doap:
1290 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
1293 Automatic update of common submodule
1294 From bcb1518 to 211fa5f
1296 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
1301 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
1303 * tests/check/gst/sessionmedia.c:
1304 tests: fix memory leak in sessionmedia unit test
1306 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
1308 * gst/rtsp-server/rtsp-client.c:
1309 client: emit a signal before sending a message
1310 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
1312 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
1314 * gst/rtsp-server/rtsp-client.c:
1315 client: pass context to send_message
1316 Pass the current context to send_message, we will need it later.
1318 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
1320 * gst/rtsp-server/rtsp-client.c:
1321 client: fix typo in comment
1323 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
1325 * gst/rtsp-server/rtsp-media.c:
1326 media: Do not stop thread twice if default_prepare() fails
1328 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
1330 * gst/rtsp-server/rtsp-client.c:
1331 client: set the watch to flushing before going to NULL
1332 First set the watch to flushing so that we unblock any current and
1333 future attempt to send data on the watch, Then set the pipeline to
1335 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
1337 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
1339 * gst/rtsp-server/rtsp-session-pool.c:
1340 * tests/check/gst/sessionpool.c:
1341 rtsp-session-pool: Fixes annotation
1342 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
1343 in the sessionpool test.
1344 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
1346 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
1348 * gst/rtsp-server/rtsp-media.c:
1349 * gst/rtsp-server/rtsp-media.h:
1350 media: make media_prepare virtual
1351 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
1353 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
1355 * gst/rtsp-server/rtsp-media.c:
1356 * tests/check/gst/media.c:
1357 media: stop the thread in more error cases
1359 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
1361 * gst/rtsp-server/rtsp-media.c:
1362 * tests/check/gst/media.c:
1363 media: allow NULL as the thread
1364 Use the default context whan passing a NULL thread.
1366 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
1368 * gst/rtsp-server/rtsp-client.c:
1369 rtsp-client: indent cleanup
1370 Coverity was moaning about unreachable code, and I think it was just
1371 confused by { being before the label. We'll see if it pops up again.
1374 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
1376 * gst/rtsp-server/rtsp-client.c:
1377 * gst/rtsp-server/rtsp-media.c:
1378 client: Add drop-backlog property
1379 When we have too many messages queued for a client (currently hardcoded
1380 to 100) we overflow and drop the messages. Add a drop-backlog property
1381 to control this behaviour. Setting this property to FALSE will retry
1382 to send the messages to the client by waiting for more room in the
1384 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
1386 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
1388 * gst/rtsp-server/rtsp-client.c:
1389 client: support for POST before GET when setting up a tunnel
1391 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
1393 * gst/rtsp-server/rtsp-client.c:
1394 client: remove watch of the second client after http tunnel setup
1395 The second client will be freed after the HTTP tunnel has been set up.
1396 Make sure it's RTSP watch is never dispatched again.
1397 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
1399 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
1401 * gst/rtsp-server/rtsp-media.c:
1402 * tests/check/gst/media.c:
1403 media: Make media_prepare() fail if port allocation fails
1404 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
1406 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
1408 * tests/check/gst/media.c:
1409 media test: cleanup the thread pool in tests
1411 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
1413 * gst/rtsp-server/rtsp-media.c:
1414 * tests/check/gst/media.c:
1415 rtsp-media: Unblock blocked streams in unprepare
1416 The streams will be blocked when a live media is prepared.
1417 The streams should be unblocked in gst_rtsp_media_unprepare.
1418 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
1420 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
1422 * gst/rtsp-server/rtsp-media.c:
1423 media: release the state lock when going to NULL
1424 Set our state to UNPREPARING and release the state-lock before
1425 setting the pipeline to the NULL state. This way, any pad-added
1426 callback will be able to take the state-lock and check that we are now
1427 unpreparing instead of deadlocking.
1428 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
1430 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
1432 * gst/rtsp-server/rtsp-media.c:
1433 media: protect status with lock
1434 Make sure we only update the status with the lock.
1436 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
1438 * gst/rtsp-server/rtsp-client.c:
1439 * gst/rtsp-server/rtsp-sdp.c:
1440 rtsp: update for MIKEY API changes
1442 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
1444 * gst/rtsp-server/rtsp-client.c:
1445 client: parse the mikey response from the client
1446 Parse the mikey response from the client and update the policy for
1449 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
1451 * gst/rtsp-server/rtsp-stream.c:
1452 * gst/rtsp-server/rtsp-stream.h:
1453 stream: add method to set crypto info
1454 Make a method to configure the crypto information of a stream.
1455 Set udpsrc in READY instead of PAUSED so that we can configure caps
1458 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
1460 * gst/rtsp-server/rtsp-client.c:
1461 client: cleanup error paths
1463 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
1465 * gst/rtsp-server/rtsp-media.c:
1468 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
1470 * examples/test-video.c:
1471 test: enable SRTP only on RTSPS
1472 We only want to enable SRTP when doing rtsp over TLS so that we can
1473 exchange the keys in a secure way.
1475 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
1477 * examples/test-video.c:
1478 test: print an error on failure
1480 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
1483 * examples/test-video.c:
1484 * gst/rtsp-server/rtsp-sdp.c:
1485 * gst/rtsp-server/rtsp-stream.c:
1486 * tests/check/Makefile.am:
1487 stream: add SRTP support
1488 Install srtp encoder and decoder elements in rtpbin
1491 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1493 * tests/check/Makefile.am:
1494 * tests/check/gst/sessionpool.c:
1495 tests: Add unit tests for sessionpool
1496 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
1498 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1500 * tests/check/gst/threadpool.c:
1501 tests: Improve code coverage of rtsp-threadpool tests
1502 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
1504 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1506 * tests/check/gst/sessionmedia.c:
1507 tests: Improve code coverage for rtsp-session-media
1508 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
1510 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1512 gobject-introspection: Add annotations to support language bindings
1513 In addition a few cosmetic changes:
1514 * Adjust the order of arguments
1515 * Fix typo: occured -> occurred
1516 * Fix indentation after Return:-clauses
1517 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
1519 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1521 * gst/rtsp-server/rtsp-stream.c:
1522 rtsp-stream: Don't mix IPv4 and IPv6 addresses
1523 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
1525 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
1527 * gst/rtsp-server/rtsp-stream.c:
1528 stream: take caps after the session manager
1529 Take the caps for the SDP after they leave the rtpbin so that we can
1530 also get the properties added by rtpbin elements.
1532 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
1534 * gst/rtsp-server/rtsp-stream.c:
1535 stream: release lock while pushing out packets
1536 Keep a cache of the transports and use this to iterate the transport
1537 while pushing packets. This allows us to release the lock early.
1538 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
1540 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
1542 * gst/rtsp-server/rtsp-client.c:
1543 * gst/rtsp-server/rtsp-client.h:
1544 rtsp-client: vmethod for modifying tunnel GET response
1545 Add a vmethod tunnel_http_response where the response to the HTTP GET
1546 for tunneled connections can be modified.
1547 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
1549 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
1551 * gst/rtsp-server/rtsp-sdp.c:
1552 sdp: make 1 media line per profile
1553 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
1554 line in the SDP for each profile. The client is then supposed to pick
1555 one of the profiles in the SETUP request. Because the m= lines have the
1556 same pt, the client also knows that only 1 option is possible.
1558 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
1560 * gst/rtsp-server/rtsp-media-factory.c:
1561 * gst/rtsp-server/rtsp-media-factory.h:
1562 * gst/rtsp-server/rtsp-media.c:
1563 factory: add profile property and pass to media and streams
1565 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
1567 * examples/test-multicast.c:
1568 * gst/rtsp-server/rtsp-sdp.c:
1569 sdp: pass multicast connection for multicast-only stream
1570 Pass the multicast address of the stream in the connection info in the
1571 SDP so that clients try a multicast connection first.
1572 Only allow multicast connections in the test-multicast example. Also
1573 increase the TTL a little.
1575 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1578 .gitignore: Ignore gcov intermediate files
1579 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
1581 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
1583 * gst/rtsp-server/rtsp-stream.c:
1584 stream: release some locks in error cases
1586 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1588 docs: Enable and fix gtk-doc warnings
1589 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
1590 * addresspool/mediafactory: Add missing annotation colon
1591 * stream: Annotate return value
1592 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
1594 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
1597 Automatic update of common submodule
1598 From fe1672e to bcb1518
1600 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
1603 Automatic update of common submodule
1604 From 1a07da9 to fe1672e
1606 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
1608 * examples/Makefile.am:
1609 examples: use LDADD for libs instead of LDFLAGS
1611 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
1614 configure: make sure releases are in .doap file
1616 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
1618 * examples/test-cgroups.c:
1619 examples: test-cgroups: don't put code with side effects into g_assert()
1620 The g_assert() might get compiled out with the right
1621 compiler/preprocessor flags.
1623 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
1625 * examples/.gitignore:
1626 examples: add cgroup test binary to .gitignore
1628 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
1630 * examples/test-cgroups.c:
1631 examples: fix cgroup test build
1632 Fixes build failure caused by compiler warning:
1633 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
1635 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
1638 .gitignore: ignore temp files created in the course of 'make check'
1640 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
1642 * gst/rtsp-server/rtsp-media.c:
1643 rtsp-media: don't loose frames handling new PLAY request
1644 If client supplied a range check if the range specifies the start point.
1645 If not, then do an accurate seek to the current position. If a start
1646 point was specified do do a key unit seek to make sure the streaming
1647 starts with decodeable frames.
1648 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
1650 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
1652 * gst/rtsp-server/rtsp-media.c:
1653 Revert "media: only flush when setting a new start position"
1654 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
1655 We need to do the flush in all cases, demuxer block currently for
1658 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
1660 * gst/rtsp-server/rtsp-media.c:
1661 media: only flush when setting a new start position
1662 Only flush the pipeline when we change the start position with
1664 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
1666 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
1668 * gst/rtsp-server/rtsp-stream.c:
1669 stream: set ttl-mc before adding the socket
1670 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
1671 never be set on socket.
1672 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
1674 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
1676 * gst/rtsp-server/rtsp-media.c:
1677 media: stop thread if media is already prepared
1678 in gst_rtsp_media_prepare() the thread is not used if media is already
1679 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
1681 https://bugzilla.gnome.org/show_bug.cgi?id=724182
1683 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
1686 build: Ship gst-rtsp-server.doap file
1688 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
1690 * tests/check/gst/rtspserver.c:
1691 tests: Fix another compiler warning with gcc
1693 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
1695 * gst/rtsp-server/rtsp-client.c:
1696 * gst/rtsp-server/rtsp-mount-points.c:
1697 * gst/rtsp-server/rtsp-stream.c:
1698 * tests/check/gst/client.c:
1699 rtsp-server: Fix lots of compiler warnings with clang
1701 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
1704 * gst-rtsp-server.doap:
1705 * tests/Makefile.am:
1706 configure: Synchronise with the configure scripts of the other modules
1708 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
1711 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
1713 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
1715 * gst/rtsp-server/rtsp-media.c:
1716 * gst/rtsp-server/rtsp-stream.c:
1717 Revert "rtsp-server: support build against last stable release"
1718 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
1719 Let us require 1.2.3 now, which is going to be released in a few
1722 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
1724 * gst/rtsp-server/rtsp-session-media.c:
1725 * gst/rtsp-server/rtsp-stream-transport.c:
1726 session: improve RTP-Info
1727 Ignore streams that can't generate RTP-Info instead of failing.
1728 Don't return the empty string when all streams are unconfigured but
1729 return NULL so that we don't generate and empty RTP-Info header.
1730 Improve docs a little.
1732 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
1734 * gst/rtsp-server/rtsp-session-media.c:
1735 Don't free rtpinfo GString when it is NULL
1736 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
1738 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
1740 * gst/rtsp-server/rtsp-media.c:
1741 media: only set keyframe flag when modifying start
1742 Only set the keyframe flag when we modify the start position. The
1743 keyframe flag should probably be ignored when no change is requested but
1744 until we can claim this is all documented properly and all demuxer
1745 implement this, avoid setting the flag.
1746 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
1748 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
1750 * gst/rtsp-server/rtsp-thread-pool.c:
1751 thread-pool: Unref source after mainloop has quit to avoid races in GLib
1752 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
1754 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
1756 * gst/rtsp-server/rtsp-stream.c:
1757 stream: handle NULL seqnum and rtptime arguments
1759 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
1761 * gst/rtsp-server/rtsp-thread-pool.c:
1762 * tests/check/gst/threadpool.c:
1763 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
1764 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
1766 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
1768 * gst/rtsp-server/rtsp-stream.c:
1769 stream: add fallback for missing stats property
1770 Use a fallback when the payloader does not have a stats property
1771 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
1773 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
1776 Automatic update of common submodule
1777 From f7bc1c3 to 1a07da9
1779 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
1781 * gst/rtsp-server/rtsp-stream.c:
1782 stream: don't leak stats structure
1783 Don't leak the stats structure and deal with NULL stats.
1785 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
1787 * gst/rtsp-server/rtsp-stream.c:
1788 stream: Get rtpinfo properties atomically from payloader
1789 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
1791 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
1793 * gst/rtsp-server/rtsp-media.c:
1794 media: refactor state change functions and signals
1795 Make functions to set the target state and the pipeline state and emit
1796 the signals from those functions.
1798 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
1800 * gst/rtsp-server/rtsp-media.c:
1801 * gst/rtsp-server/rtsp-media.h:
1802 media: add signal to notify of pending state changes
1804 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
1806 * gst/rtsp-server/rtsp-media.c:
1807 * gst/rtsp-server/rtsp-stream.c:
1808 rtsp-server: support build against last stable release
1809 Until 1.2.3 is out with the new get_type function and we
1812 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
1814 * gst/rtsp-server/rtsp-stream.c:
1815 stream: fix compilation
1817 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
1819 * gst/rtsp-server/rtsp-media.c:
1820 * gst/rtsp-server/rtsp-media.h:
1821 * gst/rtsp-server/rtsp-stream.c:
1822 * gst/rtsp-server/rtsp-stream.h:
1823 stream: add property to configure profiles
1825 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
1827 * gst/rtsp-server/rtsp-client.c:
1828 client: let stream check supported transport
1829 Delegate the check if a transport is allowed to the stream.
1830 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
1832 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
1834 * gst/rtsp-server/rtsp-stream.c:
1835 * gst/rtsp-server/rtsp-stream.h:
1836 stream: add method to check supported transport
1837 Add a method to check if a transport is supported
1839 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
1842 configure.ac: Only check for gstreamer-check, not check
1843 We include check in gstreamer-check since quite some time now.
1845 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
1847 * gst/rtsp-server/rtsp-session-media.c:
1848 * gst/rtsp-server/rtsp-stream-transport.c:
1849 * gst/rtsp-server/rtsp-stream.c:
1850 * gst/rtsp-server/rtsp-stream.h:
1851 stream: return clock-rate from get_rtpinfo
1852 And use it to correct the rtptime to the requested start-time.
1853 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
1855 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
1857 * gst/rtsp-server/rtsp-session-media.c:
1858 * gst/rtsp-server/rtsp-stream-transport.c:
1859 * gst/rtsp-server/rtsp-stream-transport.h:
1860 session-media: calculate start-time
1862 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
1864 * gst/rtsp-server/rtsp-stream-transport.c:
1865 * gst/rtsp-server/rtsp-stream.c:
1866 * gst/rtsp-server/rtsp-stream.h:
1867 stream: also return the running-time
1868 Return the running-time in the rtpinfo as well.
1870 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
1872 * gst/rtsp-server/rtsp-client.c:
1873 * gst/rtsp-server/rtsp-session-media.c:
1874 * gst/rtsp-server/rtsp-session-media.h:
1875 * gst/rtsp-server/rtsp-stream-transport.c:
1876 * gst/rtsp-server/rtsp-stream-transport.h:
1877 session-media: let the session-media make the RTPInfo
1878 Add method to create the RTPInfo for a stream-transport.
1879 Add method to create the RTPInfo for all stream-transports in a
1881 Use the session-media RTPInfo code in client. This allows us to refactor
1882 another method to link the TCP callbacks.
1884 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
1886 mount-points: sort sequence before g_sequence_lookup
1887 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
1888 sort sequence if dirty, otherwise lookup will fail.
1889 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
1891 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
1894 configure: rename package from gst-rtsp to gst-rtsp-server
1895 To match git module name and avoid confusion with the
1896 rtsp lib in gst-plugins-base and rtsp plugin in -good.
1898 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
1901 configure: bump core/base/good requirement to 1.2.0
1902 Bump to released stable version and make implicit
1903 requirements explicit.
1905 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
1910 Fix broken gettext setup which is not used anyway
1912 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
1915 Automatic update of common submodule
1916 From dbedaa0 to d48bed3
1918 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
1920 * gst/rtsp-server/rtsp-client.c:
1921 * gst/rtsp-server/rtsp-media.c:
1922 * gst/rtsp-server/rtsp-media.h:
1923 media: add setup_sdp vmethod
1924 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
1925 gst_rtsp_media_setup_sdp.
1926 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
1928 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
1930 * gst/rtsp-server/rtsp-stream.c:
1931 rtsp-stream: Check return value of sscanf
1932 streamid is only valid if sscanf matched something.
1934 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
1936 * gst/rtsp-server/rtsp-client.c:
1937 rtsp-client: Fix iteration
1938 Wouldn't even enter the code block otherwise (i++ was used as the check
1939 and not the postfix).
1941 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
1943 * gst/rtsp-server/rtsp-client.c:
1944 * gst/rtsp-server/rtsp-client.h:
1945 client: add vmethod to configure media and streams
1946 Implement a vmethod that can be used to configure the media and the
1947 streams based on the current context. Handle the blocksize handling in
1948 the default handler.
1949 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
1951 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
1954 Make git ignore more unit test binaries
1956 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
1958 * gst/rtsp-server/rtsp-address-pool.h:
1959 * gst/rtsp-server/rtsp-auth.h:
1960 * gst/rtsp-server/rtsp-client.h:
1961 * gst/rtsp-server/rtsp-context.h:
1962 * gst/rtsp-server/rtsp-media-factory-uri.h:
1963 * gst/rtsp-server/rtsp-media-factory.h:
1964 * gst/rtsp-server/rtsp-media.h:
1965 * gst/rtsp-server/rtsp-mount-points.h:
1966 * gst/rtsp-server/rtsp-server.h:
1967 * gst/rtsp-server/rtsp-session-media.h:
1968 * gst/rtsp-server/rtsp-session-pool.h:
1969 * gst/rtsp-server/rtsp-session.h:
1970 * gst/rtsp-server/rtsp-stream-transport.h:
1971 * gst/rtsp-server/rtsp-stream.h:
1972 * gst/rtsp-server/rtsp-thread-pool.h:
1973 * gst/rtsp-server/rtsp-token.h:
1974 rtsp-server: add padding to many public structures
1975 Not mini objects though, since they are not subclassable
1976 anyway, nor kept on the stack or inlined in a structure.
1978 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
1980 media: add new create_rtpbin vmethod
1981 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
1982 https://bugzilla.gnome.org/show_bug.cgi?id=719734
1984 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
1986 * tests/check/gst/media.c:
1987 tests: fix memory leak, free test's thread pool
1988 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
1990 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
1992 * gst/rtsp-server/rtsp-stream-transport.c:
1993 stream-transport: free url in finalize
1995 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
1997 * gst/rtsp-server/rtsp-media.c:
1998 media: also do state change in suspended state
2000 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
2002 * gst/rtsp-server/rtsp-client.c:
2003 * gst/rtsp-server/rtsp-media.c:
2004 media: also handle prepare and range in suspended state
2005 When we are suspended, we are already prepared.
2006 We can get the range in the suspended state.
2008 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
2010 * tests/check/Makefile.am:
2011 * tests/check/gst/sessionmedia.c:
2012 check: add test for uri in setup
2013 Added unit tests for the new functionality in GstRTSPStreamTransport.
2014 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2016 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
2018 * gst/rtsp-server/rtsp-client.c:
2019 client: store setup uri and use in PLAY response
2020 Store the uri used when doing the setup and use that in the PLAY
2022 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2024 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
2026 * gst/rtsp-server/rtsp-stream-transport.c:
2027 * gst/rtsp-server/rtsp-stream-transport.h:
2028 stream-transport: add method to get/set url
2030 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
2032 * gst/rtsp-server/rtsp-client.c:
2033 client: suspend after SDP and unsuspend before PLAYING
2034 Based on patches by Ognyan Tonchev <ognyan@axis.com>
2035 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
2037 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
2039 * gst/rtsp-server/rtsp-media-factory.c:
2040 * gst/rtsp-server/rtsp-media-factory.h:
2041 * gst/rtsp-server/rtsp-media.c:
2042 * gst/rtsp-server/rtsp-media.h:
2043 * gst/rtsp-server/rtsp-session-media.c:
2044 * gst/rtsp-server/rtsp-session.c:
2045 * tests/check/gst/media.c:
2046 * tests/check/gst/mediafactory.c:
2047 media: add suspend modes
2048 Add support for different suspend modes. The stream is suspended right after
2049 producing the SDP and after PAUSE. Different suspend modes are available that
2050 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
2051 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
2052 state and RESET will bring the pipeline to the NULL state.
2053 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
2054 this means that the pipeline needs to be prerolled again.
2055 Base on patches by Ognyan Tonchev <ognyan@axis.com>
2056 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2058 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
2060 * gst/rtsp-server/rtsp-media.c:
2061 media: start live streams in blocked state
2062 Start live streams in the blocked state and make them preroll using the
2063 messages. This ensure that no data is played by the sink until we explicitly
2064 unblock the stream right before going to PLAYING.
2065 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2067 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
2069 * gst/rtsp-server/rtsp-media.c:
2070 media: refactor starting and waiting for preroll
2071 Based on patches from Ognyan Tonchev <ognyan@axis.com>
2072 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2074 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
2076 * gst/rtsp-server/rtsp-stream.c:
2077 * gst/rtsp-server/rtsp-stream.h:
2078 stream: add API to block streams
2079 Add an API to block on the streams and make it post a message.
2080 Based on patch by Ognyan Tonchev <ognyan@axis.com>
2081 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2083 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
2085 * docs/libs/Makefile.am:
2086 docs: Specify the override file
2087 Even if it's empty (for now) it avoids make distcheck complaining
2089 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
2091 * gst/rtsp-server/rtsp-media.c:
2092 media: move default implementations to where they are used
2094 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
2096 * gst/rtsp-server/rtsp-media.c:
2097 media: take the right lock in gst_rtsp_media_set_pipeline_state()
2098 We need to take the state_lock when calling this method.
2100 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
2102 * gst/rtsp-server/rtsp-media.c:
2103 media: handle add-added on non-bins too
2104 Handle dynamic payloaders that are not bins, as used in the unit-test.
2106 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2108 * gst/rtsp-server/rtsp-media-factory.c:
2109 * gst/rtsp-server/rtsp-media-factory.h:
2110 * gst/rtsp-server/rtsp-media.c:
2111 rtsp-media/-factory: Fix request pad name comments
2112 These must be escaped for gtk-doc to parse the comments without warnings.
2114 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2116 rtsp-media: remove transports if media is in error status
2117 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
2118 trying to change to GST_STATE_NULL and media is in error status, we
2119 remove all transports.
2120 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
2122 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
2124 * gst/rtsp-server/rtsp-media.c:
2125 rtsp-media: use element metadata to find payloader
2126 Use the element metadata to find the payloader instead of checking
2128 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
2130 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
2132 rtsp-stream: add getter for payload type
2133 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
2134 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
2135 element and create the stream with this one instead of the dynpay%d
2137 https://bugzilla.gnome.org/show_bug.cgi?id=712396
2139 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2141 * gst/rtsp-server/rtsp-client.c:
2142 * gst/rtsp-server/rtsp-context.h:
2143 * gst/rtsp-server/rtsp-media.c:
2144 * gst/rtsp-server/rtsp-mount-points.c:
2145 * gst/rtsp-server/rtsp-server.c:
2146 * gst/rtsp-server/rtsp-token.c:
2147 rtsp-*: Refer to NULL as a constant in comments
2149 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2151 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2153 rtsp-*: Fix type name typos in comments
2154 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
2155 * rtsp-auth: Refer to part of constant name as text
2156 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
2157 * rtsp-session-media: Fix GstRTSPSessionMedia typo
2158 * rtsp-stream: Fix typo when refering to GstBin
2159 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2161 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
2164 * docs/libs/gst-rtsp-server-docs.sgml:
2165 * docs/libs/gst-rtsp-server-sections.txt:
2166 docs: Improve documentation
2167 * Include annotation-glossary to quiet gtk-doc
2168 * Rename remaining ClientState -> Context
2169 * Rename object hierarchy file
2170 * Remove stale chapter references
2171 * Add missing function and object references
2172 * Include missing GstRTSPAddressPoolResult
2173 https://bugzilla.gnome.org/show_bug.cgi?id=714988
2175 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
2177 * gst/rtsp-server/rtsp-client.c:
2178 * gst/rtsp-server/rtsp-server.c:
2179 * gst/rtsp-server/rtsp-session-pool.c:
2180 * gst/rtsp-server/rtsp-session.c:
2181 * gst/rtsp-server/rtsp-stream.c:
2182 rtsp-server: sprinkle some allow-none annotations for g-i
2184 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
2186 * gst/rtsp-server/rtsp-stream.c:
2187 * gst/rtsp-server/rtsp-stream.h:
2188 stream: add method to filter transports
2189 Add a method to safely iterate and collect the stream transports
2190 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2192 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
2194 * gst/rtsp-server/rtsp-client.c:
2195 * gst/rtsp-server/rtsp-server.c:
2196 * gst/rtsp-server/rtsp-session-pool.c:
2197 * gst/rtsp-server/rtsp-session.c:
2198 rtsp: allow NULL func in filters
2199 Passing a null function make the filters return a list of
2202 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
2204 * gst/rtsp-server/rtsp-address-pool.c:
2205 * tests/check/gst/addresspool.c:
2206 address-pool: fix address increment
2207 Use a guint instead of guint8 to increment the address. It's still not
2208 completely correct because a guint might not be able to hold the complete
2209 address range, but that's an enhacement for later.
2210 Add unit test to test improved behaviour.
2211 https://bugzilla.gnome.org/show_bug.cgi?id=708237
2213 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
2215 * gst/rtsp-server/rtsp-client.c:
2216 * tests/check/gst/client.c:
2217 client: allow absolute path in requests
2218 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
2220 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
2222 * gst/rtsp-server/rtsp-client.c:
2223 * gst/rtsp-server/rtsp-client.h:
2224 client: make make_path_from_uri a vmethod
2226 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2228 * docs/libs/gst-rtsp-server-sections.txt:
2229 * gst/rtsp-server/rtsp-stream.c:
2230 * gst/rtsp-server/rtsp-stream.h:
2231 * tests/check/Makefile.am:
2232 * tests/check/gst/stream.c:
2233 stream: Add functions to get rtp and rtcp sockets
2234 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2236 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2238 * gst/rtsp-server/rtsp-context.c:
2239 * gst/rtsp-server/rtsp-context.h:
2240 context: defing a GType for the context
2241 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
2243 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
2245 * gst/rtsp-server/Makefile.am:
2246 * gst/rtsp-server/rtsp-auth.c:
2247 * gst/rtsp-server/rtsp-context.c:
2248 * gst/rtsp-server/rtsp-media.c:
2249 * gst/rtsp-server/rtsp-mount-points.c:
2250 * gst/rtsp-server/rtsp-server.h:
2251 * gst/rtsp-server/rtsp-session-media.c:
2252 * gst/rtsp-server/rtsp-session.c:
2253 * gst/rtsp-server/rtsp-stream.c:
2254 Fixed several GIR warnings
2256 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
2258 * gst/rtsp-server/rtsp-auth.c:
2261 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2263 * tests/check/Makefile.am:
2264 * tests/check/gst/token.c:
2265 tests: Add unit tests for token
2266 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2268 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2270 * gst/rtsp-server/rtsp-token.c:
2271 token: Validate args for gst_rtsp_token_is_allowed
2272 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
2274 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2276 * gst/rtsp-server/rtsp-token.c:
2277 token: Fix bug when creating empty token
2278 We always want to have a valid GstStructure in the token.
2279 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2281 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
2283 * gst/rtsp-server/rtsp-thread-pool.c:
2284 thread-pool: avoid race in shutdown
2285 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
2286 don't actually stop the mainloop ever. Solve this race by adding an idle source
2287 to the mainloop that calls the _quit. This way we immediately exit the mainloop
2288 if quit was called before we started it.
2290 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2292 * tests/check/Makefile.am:
2293 * tests/check/gst/permissions.c:
2294 tests: Add unit tests for permissions
2295 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
2297 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2299 * tests/check/gst/mediafactory.c:
2300 tests: Test mediafactory permissions
2301 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2303 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2305 * gst/rtsp-server/rtsp-permissions.c:
2306 permissions: Fix refcounting when adding/removing roles
2307 Previously a role that was removed was unreffed twice, and when
2308 replacing an existing role the replaced role was freed while still being
2309 referenced. Both bugs are now fixed.
2310 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2312 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2314 * tests/check/gst/media.c:
2315 * tests/check/gst/mediafactory.c:
2316 * tests/check/gst/rtspserver.c:
2317 tests: Check gst_rtsp_url_parse return value
2318 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2320 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
2323 Automatic update of common submodule
2324 From 865aa20 to dbedaa0
2326 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
2328 * gst/rtsp-server/rtsp-server.c:
2329 rtsp-server: Fix socket leak
2330 https://bugzilla.gnome.org/show_bug.cgi?id=710088
2332 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
2334 * gst/rtsp-server/rtsp-session-pool.c:
2335 rtsp-session-pool: Make sure session IDs are properly URI-escaped
2336 https://bugzilla.gnome.org/show_bug.cgi?id=643812
2338 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2340 * examples/.gitignore:
2341 * examples/test-video.c:
2342 examples: fix compilation when WITH_AUTH is defined
2343 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2345 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
2348 gitignore: Add new test binary
2350 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
2352 * tests/check/Makefile.am:
2353 * tests/check/gst/threadpool.c:
2354 thread-pool: Add unit test for the thread pools
2355 https://bugzilla.gnome.org/show_bug.cgi?id=710228
2357 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
2359 * gst/rtsp-server/rtsp-thread-pool.c:
2360 thread-pool: Fix thread leak when reusing threads
2361 https://bugzilla.gnome.org/show_bug.cgi?id=709730
2363 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
2365 * gst/rtsp-server/rtsp-server.c:
2366 * tests/check/gst/rtspserver.c:
2367 tests: fixed racy behavior in rtspserver tests
2368 https://bugzilla.gnome.org/show_bug.cgi?id=710078
2370 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
2372 * tests/check/gst/addresspool.c:
2373 tests: Improve address pool unit tests
2374 Add a range with mixed IPV4 and IPV6 addresses to pool.
2375 Get an IPV4 address from an IPV6-only pool.
2376 Get an IPV6 address from an IPV4-only pool.
2377 Reserve a IPV6 address from an IPV4-only pool.
2378 Check for unicast addresses in multicast-only pool.
2379 Check for unicast addresses in uni-/multicast-mixed pool.
2380 https://bugzilla.gnome.org/show_bug.cgi?id=710128
2382 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2384 * gst/rtsp-server/rtsp-client.c:
2385 client: append query string in PAUSE/PLAY/TEARDOWN as well
2387 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
2389 * gst/rtsp-server/rtsp-client.c:
2390 client: Add query to control path
2391 If the SETUP url contains a query it must be appended to the control
2392 path so that it matches any already created stream in the media. The
2393 query will also be appended to the session media path.
2395 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2397 * gst/rtsp-server/rtsp-media.c:
2398 rtsp-media: remove old line
2400 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
2402 * gst/rtsp-server/rtsp-stream.c:
2403 stream: Correct control comparison
2404 https://bugzilla.gnome.org/show_bug.cgi?id=709176
2406 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2408 * gst/rtsp-server/rtsp-media.c:
2409 media: Check dynamically if the pipeline supports seeking
2410 We should not depend on whether or not the pipeline state change
2411 returned NO_PREROLL or not. A media could dynamically change its
2412 element and switch from seekable to non seekable so it's best to test
2413 the seekable nature of the pipeline dynamically when we try to do a seek.
2415 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2417 * gst/rtsp-server/rtsp-media.c:
2418 media: Return FALSE if seeking is not supported
2420 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2422 * gst/rtsp-server/rtsp-media.c:
2423 rtsp-media: don't seek accurate by default
2424 Accurate seeking is perhaps a little overkill in the most common situation and
2425 causes some formats (mp3) over slow media to seek extremely slowly.
2427 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
2429 * tests/check/gst/rtspserver.c:
2430 tests: fix unit test
2431 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
2433 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
2435 * gst/rtsp-server/rtsp-client.c:
2436 client: Reply 400 if media cannot be constructed
2437 Reply 400 Bad Request instead of 503 Service Unavailable if media
2438 cannot be constructed in SETUP.
2439 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
2441 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
2443 * gst/rtsp-server/rtsp-client.c:
2444 client: Send setup reply once only
2445 If find_media() failed in handle_setup_request() two replies was sent.
2446 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
2448 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
2451 Automatic update of common submodule
2452 From 6b03ba7 to 865aa20
2454 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
2456 * gst/rtsp-server/rtsp-server.c:
2457 server: Emit client-connected signal earlier
2458 Emit client-connected before the client ref is given to a GSource,
2459 otherwise client-connected can be emitted after the client object has
2462 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
2464 * gst/rtsp-server/rtsp-address-pool.c:
2465 * gst/rtsp-server/rtsp-address-pool.h:
2466 * gst/rtsp-server/rtsp-stream.c:
2467 * tests/check/gst/addresspool.c:
2468 addresspool: return reason of failure
2469 Let gst_rtsp_address_pool_reserve_address() return the reason why
2470 the address could not be reserved.
2471 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
2473 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
2476 autogen.sh: Sync behaviour with other GStreamer modules
2477 Allows building from outside of tree amongst other things
2479 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
2482 Automatic update of common submodule
2483 From b613661 to 6b03ba7
2485 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
2488 Automatic update of common submodule
2489 From 74a6857 to b613661
2491 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
2494 Automatic update of common submodule
2495 From 01a7a46 to 74a6857
2497 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
2499 * gst/rtsp-server/rtsp-client.c:
2500 client: Do not read beyond end of path string
2501 If the setup was done without a control url, make sure we don't try to read the
2502 non-existing control string and crash.
2504 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2506 * gst/rtsp-server/rtsp-client.c:
2507 client: Fix RTPInfo header
2508 Refactor the method to make the content_base.
2509 Use the content-base and the control url to construct the RTPInfo
2512 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2514 * gst/rtsp-server/rtsp-client.c:
2515 client: map url to path only in describe
2516 Only map the request url to a path in the DESCRIBE method. The SDP then
2517 contains the base and control urls that should be used to SETUP/PAUSE/
2518 PLAY/TEARDOWN the media.
2520 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2522 * gst/rtsp-server/rtsp-client.c:
2523 Revert "client: map URL to path in requests"
2524 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
2525 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
2526 contains the base and control urls which are used in the SETUP, PLAY,
2527 PAUSE and TEARDOWN requests.
2529 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2531 * gst/rtsp-server/rtsp-client.c:
2532 client: map URL to path in requests
2534 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2536 * gst/rtsp-server/rtsp-client.c:
2537 * gst/rtsp-server/rtsp-mount-points.c:
2538 * gst/rtsp-server/rtsp-mount-points.h:
2539 mount-points: make vmethod to make path from uri
2540 Make a vmethod to transform an url into a path. The path is then used to lookup
2541 the factory. This makes it possible to also use other bits of the url, such as
2542 the query parameters, to locate the factory.
2544 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
2546 * gst/rtsp-server/rtsp-thread-pool.c:
2547 * gst/rtsp-server/rtsp-thread-pool.h:
2548 thread-pool: Add cleanup to wait for the threadpool to finish
2549 Also fix race condition if two threads are asking for the first
2550 thread from the thread pool at once. This would case two internal
2551 GThreadPools to be created.
2552 https://bugzilla.gnome.org/show_bug.cgi?id=707753
2554 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
2556 * gst/rtsp-server/rtsp-client.c:
2557 * tests/check/gst/client.c:
2558 client: free threadpool
2559 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2561 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
2563 * tests/check/gst/mountpoints.c:
2564 mountpoints tests: unref matched factories
2565 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2567 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
2569 * tests/check/gst/media.c:
2570 media tests: unref thread pool and caps
2571 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2573 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
2575 * gst/rtsp-server/rtsp-auth.c:
2576 * gst/rtsp-server/rtsp-media-factory.c:
2577 * gst/rtsp-server/rtsp-media.c:
2578 auth, media, media-factory: unref permissions
2579 https://bugzilla.gnome.org/show_bug.cgi?id=707638
2581 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2583 * examples/Makefile.am:
2584 Makefile: add rule for appsrc example
2586 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2588 * examples/test-appsrc.c:
2589 tests: add appsrc example
2590 Add an example on how to use appsrc to feed the server pipeline with data.
2592 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
2594 * gst/rtsp-server/rtsp-client.c:
2595 rtsp-client: remove query part from content-base string
2596 Make sure that after the control url has been resolved, it's
2597 not a part of the query-string.
2598 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
2600 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2602 * gst/rtsp-server/rtsp-client.c:
2603 client: don't check url in response
2604 There is no url or method in the response to check
2606 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2608 * gst/rtsp-server/rtsp-client.c:
2609 * gst/rtsp-server/rtsp-client.h:
2610 Add handle-response signal for when we receive a GET_PARAMETER response
2612 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2614 * gst/rtsp-server/rtsp-server.c:
2615 Fix gst_rtsp_server_client_filter, using wrong variable type
2617 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
2619 * gst/rtsp-server/rtsp-media-factory-uri.c:
2620 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
2621 For AAC we need to check for framed=true instead of parsed=true.
2622 https://bugzilla.gnome.org/show_bug.cgi?id=701384
2624 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2626 * gst/rtsp-server/rtsp-stream.c:
2627 stream: optimize pipeline for protocols
2628 When TCP is not an allowed protocol for the stream, avoid creating the
2629 appsrc/appsink/queue and tee elements.
2631 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2633 * gst/rtsp-server/rtsp-media.c:
2634 media: set protocols on streams
2636 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2638 * gst/rtsp-server/rtsp-client.c:
2639 client: use protocols supported by stream
2641 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2643 * gst/rtsp-server/rtsp-media-factory.c:
2644 * gst/rtsp-server/rtsp-media.c:
2645 * gst/rtsp-server/rtsp-stream.c:
2646 media-factory: allow all protocols
2648 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2650 * gst/rtsp-server/rtsp-media.c:
2651 media: configure protocols in new streams
2653 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2655 * gst/rtsp-server/rtsp-stream.c:
2656 * gst/rtsp-server/rtsp-stream.h:
2657 stream: add protocols property
2659 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2661 * gst/rtsp-server/rtsp-media.c:
2662 rtsp-media: send state in "new-state" signal
2663 https://bugzilla.gnome.org/show_bug.cgi?id=705110
2665 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
2668 build: add subdir-objects to AM_INIT_AUTOMAKE
2669 Fixes warnings with automake 1.14
2670 https://bugzilla.gnome.org/show_bug.cgi?id=705350
2672 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2674 * docs/libs/gst-rtsp-server-sections.txt:
2675 * gst/rtsp-server/rtsp-client.c:
2676 * gst/rtsp-server/rtsp-server.c:
2677 * gst/rtsp-server/rtsp-server.h:
2678 server: add method to iterate clients of server
2680 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2682 * gst/rtsp-server/rtsp-media.c:
2683 * gst/rtsp-server/rtsp-media.h:
2684 Add vmethod for rtsp-media subclass to access rtpbin
2686 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2688 * gst/rtsp-server/rtsp-client.h:
2689 small documentation fix
2691 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2693 * gst/rtsp-server/rtsp-client.c:
2694 Do not take range header if range is invalid
2696 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2698 * docs/libs/gst-rtsp-server-sections.txt:
2699 * gst/rtsp-server/rtsp-media.c:
2700 media: add docs for new method
2702 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2704 * gst/rtsp-server/rtsp-media.c:
2705 * gst/rtsp-server/rtsp-media.h:
2706 Add API to rtsp-media set the pipeline's state
2708 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2710 * gst/rtsp-server/rtsp-media.c:
2711 Update current position/duration when gst_rtsp_media_get_range_string is called
2713 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2715 * examples/test-cgroups.c:
2716 tests: add some more docs
2718 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2720 * examples/test-cgroups.c:
2721 * gst/rtsp-server/Makefile.am:
2722 * gst/rtsp-server/rtsp-auth.c:
2723 * gst/rtsp-server/rtsp-auth.h:
2724 * gst/rtsp-server/rtsp-client.c:
2725 * gst/rtsp-server/rtsp-client.h:
2726 * gst/rtsp-server/rtsp-context.c:
2727 * gst/rtsp-server/rtsp-context.h:
2728 * gst/rtsp-server/rtsp-params.c:
2729 * gst/rtsp-server/rtsp-params.h:
2730 * gst/rtsp-server/rtsp-server.c:
2731 * gst/rtsp-server/rtsp-thread-pool.c:
2732 * gst/rtsp-server/rtsp-thread-pool.h:
2733 * tests/check/gst/client.c:
2734 ClientState -> Context
2735 Rename the clientstate to context and put the code in a separate file.
2737 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2739 * examples/test-auth.c:
2740 * gst/rtsp-server/rtsp-auth.c:
2741 * gst/rtsp-server/rtsp-auth.h:
2742 auth: add support for default token
2743 The default token is used when the user is not authenticated and can be used to
2744 give minimal permissions.
2746 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2748 * examples/test-auth.c:
2749 * gst/rtsp-server/rtsp-auth.c:
2750 auth: use defines when possible
2752 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2754 * gst/rtsp-server/rtsp-address-pool.c:
2755 address-pool: improve docs
2757 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2759 * gst/rtsp-server/rtsp-permissions.c:
2760 permissions: add the role to the copy
2762 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
2764 * gst/rtsp-server/rtsp-permissions.c:
2765 permissions: Also copy the roles
2767 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
2769 * gst/rtsp-server/rtsp-permissions.c:
2770 permissions: Make it build
2772 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2774 * gst/rtsp-server/rtsp-address-pool.h:
2777 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2779 * docs/libs/gst-rtsp-server-sections.txt:
2780 * gst/rtsp-server/rtsp-auth.c:
2781 * gst/rtsp-server/rtsp-auth.h:
2782 * gst/rtsp-server/rtsp-media.c:
2783 * gst/rtsp-server/rtsp-session-media.c:
2784 * gst/rtsp-server/rtsp-stream-transport.c:
2785 * gst/rtsp-server/rtsp-stream-transport.h:
2786 * gst/rtsp-server/rtsp-stream.c:
2787 * tests/check/gst/client.c:
2790 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2792 * docs/libs/gst-rtsp-server-sections.txt:
2793 * gst/rtsp-server/rtsp-address-pool.c:
2794 * gst/rtsp-server/rtsp-address-pool.h:
2795 * tests/check/gst/addresspool.c:
2796 * tests/check/gst/rtspserver.c:
2797 address-pool: cleanups
2798 Remove redundant method, improve docs.
2800 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2802 * docs/libs/gst-rtsp-server-sections.txt:
2803 * gst/rtsp-server/rtsp-auth.h:
2804 * gst/rtsp-server/rtsp-permissions.c:
2805 * gst/rtsp-server/rtsp-permissions.h:
2806 * gst/rtsp-server/rtsp-token.c:
2809 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2811 * gst/rtsp-server/rtsp-permissions.c:
2812 permissions: implement _remove_role
2814 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2816 * gst/rtsp-server/rtsp-permissions.c:
2817 permissions: update docs
2819 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2821 * tests/check/gst/client.c:
2822 tests: simplify tests
2823 Client settings are now disabled by default so we don't need an auth
2824 module to disable them.
2826 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2828 * gst/rtsp-server/rtsp-auth.c:
2829 auth: add default authorizations
2830 When no auth module is specified, use our table of defaults to look up the
2831 default value of the check instead of always allowing everything. This was
2832 we can disallow client settings by default.
2834 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2837 README: update readme
2839 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2841 * gst/rtsp-server/rtsp-thread-pool.c:
2842 * gst/rtsp-server/rtsp-thread-pool.h:
2843 thread-pool: add more docs
2845 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2847 * gst/rtsp-server/rtsp-thread-pool.c:
2848 * gst/rtsp-server/rtsp-thread-pool.h:
2849 thread-pool: fix race in thread reuse
2850 If we try to reuse a thread right after we made it stop, we end up using a
2851 stopped thread. Catch this case and only reuse threads that are not stopping.
2853 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2855 * gst/rtsp-server/rtsp-server.c:
2856 server: add small debug
2858 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2860 * tests/check/gst/client.c:
2862 Add some permissions to media so we can use the auth and enable
2865 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2867 * gst/rtsp-server/rtsp-client.c:
2868 client: support pushed context in handle_request
2869 If we already have a pushed state, reuse it and add our own things. This makes
2870 it easier to write tests.
2872 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2874 * gst/rtsp-server/rtsp-auth.c:
2875 auth: don't auth on methods
2876 Don't authorize on methods anymore but on the resources that we
2877 try to access, this is more flexible.
2878 Move the authorization checks to where they are needed and let the
2879 check return the response on error.
2881 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2883 * gst/rtsp-server/rtsp-mount-points.c:
2884 mount-points: add some debug
2886 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2888 * tests/check/gst/client.c:
2889 tests: almost fix test
2891 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2893 * gst/rtsp-server/rtsp-auth.c:
2894 * gst/rtsp-server/rtsp-auth.h:
2895 * gst/rtsp-server/rtsp-client.c:
2896 * gst/rtsp-server/rtsp-client.h:
2897 * gst/rtsp-server/rtsp-server.c:
2898 * gst/rtsp-server/rtsp-server.h:
2899 auth: let the auth module check client_settings
2900 Let the auth module decide if client settings are allowed for the
2903 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2905 * gst/rtsp-server/rtsp-token.c:
2906 * gst/rtsp-server/rtsp-token.h:
2907 token: add method to check boolean permission
2909 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2911 * examples/test-auth.c:
2912 * examples/test-cgroups.c:
2913 * gst/rtsp-server/rtsp-token.c:
2914 * gst/rtsp-server/rtsp-token.h:
2915 token: simplify token constructor
2916 Use variable arguments to make easier API.
2918 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2920 * examples/test-auth.c:
2921 * examples/test-cgroups.c:
2922 * gst/rtsp-server/rtsp-media-factory.c:
2923 * gst/rtsp-server/rtsp-media-factory.h:
2924 media-factory: add convenience API for factory
2926 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2928 * examples/test-auth.c:
2929 * examples/test-cgroups.c:
2930 * gst/rtsp-server/rtsp-permissions.c:
2931 * gst/rtsp-server/rtsp-permissions.h:
2932 permissions: simplify API a little
2933 Avoid passing GstStructure in the add_role method, use varargs instead
2934 to construct the structure behind the scenes. We can then also use the
2935 structure name as the role and simplify some more logic.
2937 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2939 * gst/rtsp-server/rtsp-auth.c:
2942 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2944 * gst/rtsp-server/rtsp-auth.c:
2945 * gst/rtsp-server/rtsp-auth.h:
2946 * gst/rtsp-server/rtsp-client.c:
2947 auth: handle unauthorized response
2948 Move handling of the unauthorized response to the auth module, it can add
2949 the appropriate headers to request authorization for the required method
2950 much better than the client.
2952 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2954 * gst/rtsp-server/rtsp-client.c:
2955 * gst/rtsp-server/rtsp-client.h:
2956 client: allow for sending any message, not only requests
2957 Change the _send_request() method to _send_message() so that we
2958 can both send requests and replies.
2960 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2962 * docs/libs/gst-rtsp-server-sections.txt:
2963 * gst/rtsp-server/rtsp-server.h:
2966 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2968 * examples/test-video.c:
2969 * gst/rtsp-server/rtsp-auth.c:
2970 * gst/rtsp-server/rtsp-auth.h:
2971 * gst/rtsp-server/rtsp-server.c:
2972 * gst/rtsp-server/rtsp-server.h:
2973 auth: move TLS handling to auth module
2974 Remove the TLS settings on the server and move it to the auth module because
2975 that is where security related bits go.
2977 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2979 * gst/rtsp-server/rtsp-client.c:
2980 * gst/rtsp-server/rtsp-client.h:
2981 client: add state push/pop
2983 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2985 * gst/rtsp-server/rtsp-client.c:
2986 * gst/rtsp-server/rtsp-client.h:
2987 client: add connection to state
2989 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2991 * gst/rtsp-server/rtsp-mount-points.c:
2992 mount-points: fix debug
2994 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2996 * tests/check/gst/media.c:
2997 tests: fix media test
2999 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3001 * gst/rtsp-server/rtsp-thread-pool.c:
3002 thread-pool: we don't require a state
3004 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3006 * gst/rtsp-server/rtsp-server.c:
3007 server: let context ref the server
3008 So that we don't risk losing the server object early anc crash.
3010 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3012 * tests/check/gst/client.c:
3013 tests: fix client test
3015 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3018 * docs/libs/gst-rtsp-server-docs.sgml:
3019 * docs/libs/gst-rtsp-server-sections.txt:
3020 * gst/rtsp-server/rtsp-address-pool.c:
3021 * gst/rtsp-server/rtsp-auth.c:
3022 * gst/rtsp-server/rtsp-client.c:
3023 * gst/rtsp-server/rtsp-client.h:
3024 * gst/rtsp-server/rtsp-media-factory-uri.c:
3025 * gst/rtsp-server/rtsp-media-factory.c:
3026 * gst/rtsp-server/rtsp-media-factory.h:
3027 * gst/rtsp-server/rtsp-media.c:
3028 * gst/rtsp-server/rtsp-mount-points.c:
3029 * gst/rtsp-server/rtsp-params.c:
3030 * gst/rtsp-server/rtsp-permissions.c:
3031 * gst/rtsp-server/rtsp-sdp.c:
3032 * gst/rtsp-server/rtsp-server.c:
3033 * gst/rtsp-server/rtsp-server.h:
3034 * gst/rtsp-server/rtsp-session-media.c:
3035 * gst/rtsp-server/rtsp-session-pool.c:
3036 * gst/rtsp-server/rtsp-session.c:
3037 * gst/rtsp-server/rtsp-stream-transport.c:
3038 * gst/rtsp-server/rtsp-stream.c:
3039 * gst/rtsp-server/rtsp-thread-pool.c:
3040 * gst/rtsp-server/rtsp-token.c:
3043 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3045 * gst/rtsp-server/rtsp-session-pool.c:
3046 * gst/rtsp-server/rtsp-session-pool.h:
3047 session-pool: make vmethod to create a session
3048 Make a vmethod to create a sessions so that subclasses can create
3049 custom session objects
3051 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3053 * gst/rtsp-server/rtsp-auth.c:
3054 * gst/rtsp-server/rtsp-media-factory.h:
3055 * gst/rtsp-server/rtsp-media.h:
3056 * gst/rtsp-server/rtsp-mount-points.h:
3057 * gst/rtsp-server/rtsp-session-pool.h:
3058 * gst/rtsp-server/rtsp-stream.h:
3061 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3063 * docs/libs/gst-rtsp-server-docs.sgml:
3064 * docs/libs/gst-rtsp-server-sections.txt:
3065 * gst/rtsp-server/rtsp-address-pool.c:
3066 * gst/rtsp-server/rtsp-address-pool.h:
3067 * gst/rtsp-server/rtsp-auth.c:
3068 * gst/rtsp-server/rtsp-client.h:
3069 * gst/rtsp-server/rtsp-media-factory.h:
3070 * gst/rtsp-server/rtsp-media.c:
3071 * gst/rtsp-server/rtsp-media.h:
3072 * gst/rtsp-server/rtsp-permissions.c:
3073 * gst/rtsp-server/rtsp-permissions.h:
3074 * gst/rtsp-server/rtsp-server.h:
3075 * gst/rtsp-server/rtsp-session-media.c:
3076 * gst/rtsp-server/rtsp-session-media.h:
3077 * gst/rtsp-server/rtsp-session-pool.h:
3078 * gst/rtsp-server/rtsp-session.h:
3079 * gst/rtsp-server/rtsp-stream-transport.h:
3080 * gst/rtsp-server/rtsp-stream.c:
3081 * gst/rtsp-server/rtsp-thread-pool.h:
3084 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3087 * examples/Makefile.am:
3088 configure: compile cgroup example conditionally
3089 Only compile the cgroup example when we have libcgroup
3091 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3094 * examples/Makefile.am:
3095 * examples/test-cgroups.c:
3096 examples: add cgroups example
3098 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3100 * tests/check/gst/rtspserver.c:
3101 tests: fix compilation
3103 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3105 * gst/rtsp-server/rtsp-thread-pool.c:
3106 thread-pool: fix vmethod invocation
3108 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3110 * gst/rtsp-server/rtsp-thread-pool.c:
3111 * gst/rtsp-server/rtsp-thread-pool.h:
3112 thread-pool: store thread type in thread
3114 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3116 * gst/rtsp-server/rtsp-client.c:
3117 client: pass thread from pool to media _prepare
3118 Get a thread from the configured threadpool and pass it to the prepare method of
3121 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3123 * gst/rtsp-server/rtsp-media.c:
3124 * gst/rtsp-server/rtsp-media.h:
3125 media: Accept a thread in _prepare
3126 Remove out own threadpool handling and use the provided thread and
3127 maincontext for the bus messages and the state changes.
3129 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3131 * gst/rtsp-server/rtsp-server.c:
3132 server: configure client thread pool
3134 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3136 * gst/rtsp-server/rtsp-client.c:
3137 * gst/rtsp-server/rtsp-client.h:
3138 client: add method to configure thread pool
3140 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3142 * gst/rtsp-server/rtsp-client.h:
3143 * gst/rtsp-server/rtsp-server.c:
3144 * gst/rtsp-server/rtsp-server.h:
3145 server: use thread pool
3146 Use the thread pool instead of doing our own thing.
3148 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3150 * gst/rtsp-server/Makefile.am:
3151 * gst/rtsp-server/rtsp-thread-pool.c:
3152 * gst/rtsp-server/rtsp-thread-pool.h:
3153 thread-pool: add object to manage threads
3154 Add an object to manage the client and media threads.
3156 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3158 * gst/rtsp-server/rtsp-auth.c:
3159 auth: debug authorization check
3161 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3163 * gst/rtsp-server/rtsp-media.c:
3164 media: start media pipeline in context
3165 Start the media pipeline in the provided context (or our default one
3166 when NULL). This makes sure that we run the bus thread in this context and that
3167 all media threads are children of this context.
3169 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3171 * gst/rtsp-server/rtsp-media-factory.c:
3172 factory: pass permissions to media by default
3174 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3176 * examples/test-auth.c:
3177 test: add permissions to auth test
3178 Ass some permissions to the media factory in the test.
3180 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3182 * gst/rtsp-server/rtsp-auth.c:
3183 * gst/rtsp-server/rtsp-auth.h:
3184 * gst/rtsp-server/rtsp-client.c:
3185 auth: simplify auth checks
3186 Remove client from methods, it's now in the state
3187 Perform the check specified by the string, use the information from the
3188 thread local context.
3190 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3192 * gst/rtsp-server/rtsp-client.c:
3193 * gst/rtsp-server/rtsp-client.h:
3194 client: add state to current thread
3195 Add the client to the ClientState object.
3196 Place the ClientState on the current thread.
3198 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3200 * gst/rtsp-server/rtsp-media-factory.c:
3201 * gst/rtsp-server/rtsp-media-factory.h:
3202 * gst/rtsp-server/rtsp-media.c:
3203 * gst/rtsp-server/rtsp-media.h:
3204 media: make it possible to set permissions
3205 Make it possible to set permissions on media and media factory objects
3207 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3209 * gst/rtsp-server/Makefile.am:
3210 * gst/rtsp-server/rtsp-permissions.c:
3211 * gst/rtsp-server/rtsp-permissions.h:
3212 permissions: add permissions object
3213 Add a mini object to store permissions based on a role.
3215 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3217 * examples/test-auth.c:
3218 * gst/rtsp-server/rtsp-auth.c:
3219 * gst/rtsp-server/rtsp-auth.h:
3220 * gst/rtsp-server/rtsp-client.c:
3221 auth: add auth checks
3222 Add an enum with auth checks and implement the checks in the auth object.
3223 Perform the checks from the client.
3225 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3227 * examples/test-auth.c:
3228 * gst/rtsp-server/rtsp-auth.c:
3229 * gst/rtsp-server/rtsp-auth.h:
3230 * gst/rtsp-server/rtsp-client.h:
3231 auth: use the token after authentication
3232 After we authenticated a user, keep the Token around in the state.
3234 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3236 * gst/rtsp-server/rtsp-client.c:
3237 * gst/rtsp-server/rtsp-media.c:
3238 * gst/rtsp-server/rtsp-media.h:
3239 * tests/check/gst/media.c:
3240 media: add optional context for bus messages
3241 Add an optional mainloop to _prepare that will handle the bus messages instead
3242 of always using the shared mainloop.
3244 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3246 * gst/rtsp-server/Makefile.am:
3247 * gst/rtsp-server/rtsp-token.c:
3248 * gst/rtsp-server/rtsp-token.h:
3249 token: add authorization token
3250 Add a simply miniobject that contains the authorizations. The object contains a
3251 GstStructure that hold all authorization fields. When a user is authenticated,
3252 the auth module will create a Token for the user. The token is then used to
3253 check what operations the user is allowed to do and various other configuration
3256 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3258 * examples/test-auth.c:
3259 * gst/rtsp-server/rtsp-auth.c:
3260 * gst/rtsp-server/rtsp-auth.h:
3261 * gst/rtsp-server/rtsp-client.c:
3262 * gst/rtsp-server/rtsp-client.h:
3263 * gst/rtsp-server/rtsp-media-factory.c:
3264 * gst/rtsp-server/rtsp-media-factory.h:
3265 * gst/rtsp-server/rtsp-media.c:
3266 * gst/rtsp-server/rtsp-media.h:
3267 auth: remove auth from media and factory
3268 Remove the auth object from media and factory. We want to have the RTSPClient
3269 authenticate and authorize resources, there is no need to place another auth
3270 manager on the media/factory.
3272 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3274 * examples/test-auth.c:
3275 * gst/rtsp-server/rtsp-auth.c:
3276 * gst/rtsp-server/rtsp-auth.h:
3277 * gst/rtsp-server/rtsp-client.h:
3278 auth: add support for multiple basic auth tokens
3279 Make it possible to add multiple basic authorisation tokens to one authorization
3280 object. Associate with each token an authorization group that will define what
3281 capabilities are allowed.
3283 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3285 * gst/rtsp-server/rtsp-client.c:
3286 client: error out on non-aggregate control
3287 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
3289 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3291 * gst/rtsp-server/rtsp-client.c:
3292 client: rework setup request a little
3293 Cache the media in DESCRIBE based on the longest matching path with the uri
3294 that we can find in the mount points.
3295 Rework the setup request a little to get the media from the session or from
3296 the longest matching path, this way we can derive the control string as
3297 everything after the path instead of hardcoding it.
3298 Find the stream based on the control string and only open a session when all
3301 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3303 * gst/rtsp-server/rtsp-media.c:
3304 * gst/rtsp-server/rtsp-media.h:
3305 media: add method to find a stream by control url
3307 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3309 * gst/rtsp-server/rtsp-stream.c:
3310 * gst/rtsp-server/rtsp-stream.h:
3311 stream: add method to check control url of stream
3313 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3315 * gst/rtsp-server/rtsp-client.c:
3316 * gst/rtsp-server/rtsp-session-media.c:
3317 * gst/rtsp-server/rtsp-session-media.h:
3318 * gst/rtsp-server/rtsp-session.c:
3319 * gst/rtsp-server/rtsp-session.h:
3320 session: use path matching for session media
3321 Use a path string instead of a uri to lookup session media in the sessions. Also
3322 use path matching to find the largest possible path that matches.
3324 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3326 * gst/rtsp-server/rtsp-client.c:
3327 * gst/rtsp-server/rtsp-mount-points.c:
3328 * gst/rtsp-server/rtsp-mount-points.h:
3329 * tests/check/gst/mountpoints.c:
3330 mount-points: remove useless vmethod
3331 Making lookups in the mount points should not be done with a URL, if there is a
3332 mapping to be done from URL to mount points, we'll need to do it somewhere
3335 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3337 * gst/rtsp-server/rtsp-mount-points.c:
3338 * gst/rtsp-server/rtsp-mount-points.h:
3339 * tests/check/gst/mountpoints.c:
3340 mount-points: improve mount point searching
3341 Use a GSequence to keep track of the mount points.
3342 Match a URL to the longest matching registered mount point. This should be the
3343 URL to perform aggreagate control and the remainder is the stream specific
3345 Add some unit tests for this.
3347 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
3349 * gst/rtsp-server/Makefile.am:
3350 rtsp-server: Allow building of static library
3352 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3354 * tests/check/gst/mediafactory.c:
3355 tests: fix compilation
3357 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3359 * gst/rtsp-server/rtsp-sdp.c:
3360 sdp: get control string from stream
3361 Use the control string as configured in the stream.
3363 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3365 * gst/rtsp-server/rtsp-stream.c:
3366 * gst/rtsp-server/rtsp-stream.h:
3367 stream: add methods and property to set control string
3369 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3371 * gst/rtsp-server/rtsp-client.c:
3373 Rename variables for clarity
3374 Keep media in state when we can
3376 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3378 * gst/rtsp-server/rtsp-client.c:
3379 * gst/rtsp-server/rtsp-stream.c:
3380 * gst/rtsp-server/rtsp-stream.h:
3381 stream: add more support for IPv6
3382 Rename _get_address to _get_multicast_address in GstRTSPStream to
3383 make it clear that this function only deals with multicast.
3384 Make it possible to have both an IPv4 and IPv6 multicast address on
3385 a stream. Give the client an IPv4 or IPv6 address depending on the
3386 address it used to connect to the server.
3387 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
3389 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3391 * gst/rtsp-server/rtsp-client.c:
3394 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3396 * gst/rtsp-server/rtsp-stream.c:
3397 stream: handle failed port allocation
3398 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
3399 can't allocate any family at all. Also keep track of what port families we
3401 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
3403 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3405 * gst/rtsp-server/rtsp-stream.c:
3406 stream: improve docs
3408 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3410 * gst/rtsp-server/rtsp-stream-transport.c:
3411 stream-transport: remove old if 0 block
3413 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
3415 * tests/check/gst/client.c:
3417 gst_rtsp_client_get_uri() has been removed
3418 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
3420 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3422 * gst/rtsp-server/rtsp-client.c:
3423 * gst/rtsp-server/rtsp-client.h:
3424 client: add method to filter managed sessions
3425 Add a method to filter the sessions managed by this client connection.
3426 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
3428 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3430 * gst/rtsp-server/rtsp-client.c:
3431 * gst/rtsp-server/rtsp-client.h:
3432 client: remove _get_uri() method
3433 Remove the get_uri() method on the client. A client has no uri, the uri
3434 property is an internal property to manage the last cached media for
3437 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3439 * gst/rtsp-server/rtsp-media-factory.h:
3440 media-factory: fix typo
3442 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
3444 * gst/rtsp-server/rtsp-media.c:
3445 rtsp-media: Do not leak the query in default_query_stop
3446 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
3448 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3450 * gst/rtsp-server/rtsp-media.c:
3451 media: don't unlock when conversion fails
3452 Don't unlock the state lock when conversion fails because it was not locked.
3454 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3456 * gst/rtsp-server/rtsp-media.c:
3457 * gst/rtsp-server/rtsp-media.h:
3458 Add query_position and query_stop vmethods to rtsp-media
3460 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3462 * gst/rtsp-server/rtsp-media.c:
3463 Fix typo in property install for rtsp-media's time-provider
3465 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3467 * gst/rtsp-server/rtsp-client.c:
3468 * gst/rtsp-server/rtsp-client.h:
3469 client: clean some variables
3470 Clean some variables and add some guards to _send_request()
3472 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
3474 * gst/rtsp-server/rtsp-client.c:
3475 * gst/rtsp-server/rtsp-client.h:
3476 Add gst_rtsp_client_send_request API
3477 This makes it possible to send arbitrary messages to a client, such as
3478 SET_PARAMETER or GET_PARAMETER
3480 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3482 * gst/rtsp-server/rtsp-media.c:
3483 * gst/rtsp-server/rtsp-media.h:
3484 media: add _get_element() method
3485 Add method to get the element used when creating the media.
3486 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
3488 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3490 * gst/rtsp-server/rtsp-media.c:
3493 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
3495 * gst/rtsp-server/rtsp-stream.c:
3496 * gst/rtsp-server/rtsp-stream.h:
3497 stream: allow access to the rtp session
3498 https://bugzilla.gnome.org/show_bug.cgi?id=703004
3500 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
3502 * gst/rtsp-server/rtsp-stream.c:
3503 * gst/rtsp-server/rtsp-stream.h:
3504 dscp qos support in gst-rtsp-stream
3505 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
3507 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3509 * tests/check/gst/rtspserver.c:
3511 Actually do what the comment says. Also keep the old code around, not sure what
3512 should happen when you get a 454 from a TEARDOWN, does it close the connection?
3513 it currently doesn't.
3515 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3517 * gst/rtsp-server/rtsp-client.c:
3518 client: also watch newly created session
3519 When we newly created a session, start watching it immediately instead of
3520 on the next request.
3522 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
3524 * tests/check/gst/client.c:
3525 tests: add unit test for new-session
3526 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
3528 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3530 * gst/rtsp-server/rtsp-client.c:
3531 client: emit new-session when new session is created
3532 Only emit new-session when we created a new session for a client, not when a
3533 client picked up a previous session.
3534 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
3536 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
3538 * gst/rtsp-server/rtsp-client.c:
3539 client: handle asterisk as path in requests
3540 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
3542 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3544 * gst/rtsp-server/rtsp-media.c:
3545 media: handle segment query format mismatch
3546 It's possible that the segment query returns with a different format than what
3547 we asked for, handle this case also.
3549 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
3551 * gst/rtsp-server/rtsp-media.c:
3552 media: use segment stop in collect_media_stats
3553 Use segment stop instead of duration as range end point.
3554 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
3556 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3558 * gst/rtsp-server/rtsp-media.c:
3559 * tests/check/gst/media.c:
3560 rtsp-media: Do not leak the element in take_pipeline
3561 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
3563 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
3565 * gst/rtsp-server/rtsp-client.c:
3566 * gst/rtsp-server/rtsp-client.h:
3567 rtsp-client: Make configure_client_transport virtual
3568 This patch makes configure_client_transport virtual. The functionality is
3569 needed to handle some weird clients sending multicast transport settings as url
3571 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
3573 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
3575 * gst/rtsp-server/rtsp-client.c:
3576 * gst/rtsp-server/rtsp-client.h:
3577 rtsp-client: Make param_set and param_get virtual
3578 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
3580 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
3582 * gst/rtsp-server/rtsp-client.c:
3583 * gst/rtsp-server/rtsp-media.c:
3584 * gst/rtsp-server/rtsp-media.h:
3585 media: convert_range replaces get_range_times
3586 get_range_times worked for handling UTC ranges for seeks, but we also
3587 need to convert back from NPT to the requested unit in
3588 get_range_string. convert_range is now used for both.
3589 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
3591 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3593 * gst/rtsp-server/rtsp-client.c:
3594 * gst/rtsp-server/rtsp-sdp.c:
3595 * gst/rtsp-server/rtsp-sdp.h:
3596 sdp: cleanup sdp info
3597 We don't need to pass the proto, we can more easily check a boolean.
3598 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
3600 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
3602 * gst/rtsp-server/rtsp-sdp.c:
3603 use 0.0.0.0 or :: for c= line instead of server address
3605 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
3607 * gst/rtsp-server/rtsp-client.c:
3608 use local address, not remote, in SDP
3609 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
3611 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3614 Automatic update of common submodule
3615 From 098c0d7 to 01a7a46
3617 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
3619 * gst/rtsp-server/rtsp-media.c:
3620 * gst/rtsp-server/rtsp-media.h:
3621 media: possibility to override range time conversion
3622 Make it possible to override the conversion from GstRTSPTimeRange to
3623 GstClockTimes, that is done before seeking on the media
3624 pipeline. Overriding can be useful for UTC ranges, where the default
3625 conversion gives nanoseconds since 1900.
3626 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
3628 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
3630 * gst/rtsp-server/rtsp-server.c:
3631 * gst/rtsp-server/rtsp-server.h:
3632 rtsp-server: Expose the use_client_settings API
3633 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
3635 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
3637 * gst/rtsp-server/rtsp-client.c:
3638 * gst/rtsp-server/rtsp-stream.c:
3639 * gst/rtsp-server/rtsp-stream.h:
3640 rtspstream: handle both ipv4 and ipv6 clients
3641 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
3643 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3645 * gst/rtsp-server/rtsp-sdp.c:
3646 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
3647 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
3648 We already have a way to place extra attributes in the SDP by using a string
3649 property with prefix x- or a- in the caps.
3651 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3653 * gst/rtsp-server/rtsp-sdp.c:
3654 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
3655 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
3656 We already have a way to place extra attributes in the SDP, just make a string
3657 property in the payloader with a- or x- prefix.
3659 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3661 * gst/rtsp-server/rtsp-sdp.c:
3662 rtsp: place a- and x- properties as attributes
3663 application/x-rtp has properties with a- and x- prefixes that should be
3664 placed as attributes in the SDP for the media instead of being added to the
3667 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3669 * examples/Makefile.am:
3670 * examples/test-video.c:
3671 example: add TLS example
3673 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3675 * gst/rtsp-server/rtsp-server.c:
3676 * gst/rtsp-server/rtsp-server.h:
3677 server: add support for TLS
3678 Add methods to set and get a TLS certificate.
3679 Add vmethod to configure a new connection. By default, configure the TLS
3680 certificate in a new connection if needed.
3682 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3684 * gst/rtsp-server/rtsp-server.c:
3685 * gst/rtsp-server/rtsp-server.h:
3686 server: remove accept_client vmethod
3687 This vmethod is not very useful so remove it.
3689 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3691 * gst/rtsp-server/rtsp-server.c:
3692 server: don't crash on NULL GError
3694 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
3696 * gst/rtsp-server/rtsp-session-pool.c:
3697 rtsp-session-pool: corrected session timeout detection
3698 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
3700 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3702 * gst/rtsp-server/rtsp-client.c:
3703 client: improve debug
3705 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3707 * gst/rtsp-server/rtsp-client.c:
3708 * gst/rtsp-server/rtsp-client.h:
3709 * gst/rtsp-server/rtsp-server.c:
3710 server: refactor connection setup
3711 Let the server accept the socket connection and construct a GstRTSPConnection
3712 from it. Remove the code from the client and let the client only deal with
3713 a fully configure GstRTSPConnection object.
3714 We will need this later when the server will configure the connection for
3717 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3719 * gst/rtsp-server/rtsp-stream.c:
3720 stream: keep the transport object alive
3721 Keep the transport object alive while we have it as qdata on the
3724 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
3726 * gst/rtsp-server/rtsp-client.c:
3727 * gst/rtsp-server/rtsp-server.c:
3728 rtsp-server: Do not crash on nmapping of server
3729 * generate error when gst_rtsp_connection_accept fails
3730 * do not stop accepting incoming connections because
3731 accepting a client fails
3732 https://bugzilla.gnome.org/show_bug.cgi?id=701072
3734 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
3736 * gst/rtsp-server/rtsp-client.c:
3737 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
3738 https://bugzilla.gnome.org/show_bug.cgi?id=700953
3740 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
3742 * gst/rtsp-server/rtsp-sdp.c:
3743 rtsp-sdp: Parse framerate caps field and set SDP attribute
3744 The SDP attribute and its format is described in RFC4566.
3745 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
3747 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
3749 * gst/rtsp-server/rtsp-sdp.c:
3750 rtsp-sdp: Parse width/height from caps and set SDP attribute
3751 The SDP attribute and its format is described in RFC6064.
3752 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
3754 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
3756 * gst/rtsp-server/rtsp-sdp.c:
3757 * tests/check/gst/client.c:
3758 rtsp-sdp: add bandwidth line
3759 https://bugzilla.gnome.org/show_bug.cgi?id=699220
3761 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3764 Automatic update of common submodule
3765 From 5edcd85 to 098c0d7
3767 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
3769 * tests/check/gst/media.c:
3770 tests: add dynamic payloader prepare/unprepare check
3772 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3774 * gst/rtsp-server/rtsp-media.c:
3775 media: release lock when removing fakesink
3777 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3779 * gst/rtsp-server/rtsp-stream.c:
3780 stream: set elements to NULL before removing
3781 When removing a stream, set the elements to NULL first. This avoids
3782 element-is-not-in-NULL-state errors when we dispose the elements.
3784 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
3787 Automatic update of common submodule
3788 From 3cb3d3c to 5edcd85
3790 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3792 * gst/rtsp-server/rtsp-media.c:
3793 * gst/rtsp-server/rtsp-media.h:
3794 media: listen to pad-removed signals
3795 Listen to the pad-removed signal and remove the stream associated with the
3797 Add signal to be notified of the removed pad.
3798 Remove the fakesink in unprepare()
3799 Fix signatures of the signal methods
3801 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3803 * examples/test-sdp.c:
3804 tests: add example of reusable pipelines
3806 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
3808 * gst/rtsp-server/rtsp-stream.c:
3809 * gst/rtsp-server/rtsp-stream.h:
3810 stream: add method to get the srcpad
3812 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
3814 * tests/check/gst/media.c:
3815 check: add media prepare/unprepare test
3816 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
3818 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
3820 * gst/rtsp-server/rtsp-media.c:
3821 media: disconnect from signal handlers in unprepare()
3822 We connected to the pad-added and no-more-pads signals in prepare() so
3823 we need to disconnect from them in unprepare().
3824 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
3826 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
3828 * gst/rtsp-server/rtsp-media.c:
3829 media: don't free streams array
3830 Don't free the streams array in the unprepare() method, they were not
3832 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
3834 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
3836 * gst/rtsp-server/rtsp-media.c:
3837 media: don't unref the pipeline in unprepare
3838 Unprepare() should undo what prepare() does. Because the pipeline is
3839 not created in prepare(), we should not unref it in unprepare()
3841 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
3843 * gst/rtsp-server/rtsp-stream.c:
3844 stream: clear session and caps for reuse
3845 Set the session and caps to NULL after unref otherwise we might unref
3847 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
3849 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
3851 * gst/rtsp-server/rtsp-client.c:
3852 client: send out teardown signal before tearing down
3853 The advantage is that in the signal handler you get direct access to
3854 information about what streams are about to get torn down (in the
3855 GstRTSPClientState).
3856 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
3858 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
3860 * gst/rtsp-server/rtsp-client.c:
3861 * gst/rtsp-server/rtsp-client.h:
3862 client: expose connection
3863 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
3865 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
3868 Automatic update of common submodule
3869 From aed87ae to 3cb3d3c
3871 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3873 * gst/rtsp-server/rtsp-media.c:
3874 * gst/rtsp-server/rtsp-media.h:
3875 * gst/rtsp-server/rtsp-session-media.c:
3876 * gst/rtsp-server/rtsp-session-media.h:
3877 media: add method to get the base_time of the pipeline
3878 Together with a shared clock, this base-time could eventually be sent to
3879 the client so that it can reconstruct the exact running-time of the clock
3882 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3884 * gst/rtsp-server/Makefile.am:
3885 * gst/rtsp-server/rtsp-media.c:
3886 * gst/rtsp-server/rtsp-media.h:
3887 * gst/rtsp-server/rtsp-sdp.c:
3888 media: add GstNetTimeProvider support
3889 Add a property to let the media provide a GstNetTimeProvider for its clock.
3890 Make methods to get the clock and nettimeprovider
3891 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
3892 provider and also the current time of the clock. This should make it possible
3893 for (GStreamer) clients to slave their clock to the server clock.
3895 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
3898 Automatic update of common submodule
3899 From 04c7a1e to aed87ae
3901 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3903 * gst/rtsp-server/rtsp-media.c:
3904 media: wait for buffering to complete
3905 Wait for buffering to complete before changing the state to the target state.
3907 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3909 * gst/rtsp-server/rtsp-media.c:
3910 media: small cleanup
3912 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
3914 * tests/check/gst/rtspserver.c:
3915 tests: remove extra unref in test_setup_non_existing_stream
3916 The unref is not needed anymore, teardown runs without it.
3917 https://bugzilla.gnome.org/show_bug.cgi?id=696542
3919 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
3921 * tests/check/gst/rtspserver.c:
3922 tests: GSocketService cleanup in test_bind_already_in_use
3923 Use g_socket_service_stop so the rtspserver test stops listening for
3924 incoming connections in test_bind_already_in_use.
3925 https://bugzilla.gnome.org/show_bug.cgi?id=696541
3927 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
3929 * gst/rtsp-server/rtsp-media-factory.c:
3930 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
3931 Instead use a GWeakRef which is safe to use
3932 This is a known GLib bug, see:
3933 https://bugzilla.gnome.org/show_bug.cgi?id=667145
3935 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
3937 * gst/rtsp-server/rtsp-client.c:
3938 * gst/rtsp-server/rtsp-media.c:
3939 * gst/rtsp-server/rtsp-media.h:
3940 * gst/rtsp-server/rtsp-sdp.c:
3941 * tests/check/gst/media.c:
3942 * tests/check/gst/rtspserver.c:
3943 rtsp-media/client: Reply to PLAY request with same type of Range
3944 Remember the type of Range from the PLAY request and use the same type for
3947 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
3949 * gst/rtsp-server/rtsp-client.c:
3950 * gst/rtsp-server/rtsp-client.h:
3951 * tests/check/gst/client.c:
3952 rtsp-client: expose uri
3954 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
3956 * tests/check/gst/mediafactory.c:
3957 tests: Hold ref while creating second media
3958 To test if the media aren't shared, make sure we keep the first one while creating a second
3959 otherwise the same memory address may be reused.
3961 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
3964 configure: remove out-of-date comment
3966 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
3969 .gitignore: ignore more build files
3971 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
3973 * tests/check/Makefile.am:
3974 tests: use right _LIBS variable for gst-plugins-base libs
3976 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3978 * tests/check/Makefile.am:
3979 check: add librtp to libs
3981 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
3983 * tests/check/gst/rtspserver.c:
3984 tests: Add test to check selecting a port the server will send from
3986 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
3988 * tests/check/gst/rtspserver.c:
3989 tests: Make sure packets are actually received
3991 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
3993 * gst/rtsp-server/rtsp-stream.c:
3994 stream: Select unicast address from pool if appropriate
3996 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
3998 * gst/rtsp-server/rtsp-stream.c:
3999 stream: Properties are always there in Gst 1.0
4001 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4003 * tests/check/gst/addresspool.c:
4004 tests: Add tests for unicast addresses in pool
4006 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
4008 * gst/rtsp-server/rtsp-address-pool.c:
4009 * tests/check/gst/addresspool.c:
4010 address-pool: Verify that multicast addresses are used for multicast and vice-versa
4012 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
4014 * docs/libs/gst-rtsp-server-sections.txt:
4015 * gst/rtsp-server/rtsp-address-pool.c:
4016 * gst/rtsp-server/rtsp-address-pool.h:
4017 * gst/rtsp-server/rtsp-stream.c:
4018 * tests/check/gst/addresspool.c:
4019 address-pool: Add unicast addresses
4021 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4024 * gst/rtsp-server/rtsp-server.c:
4025 * tests/check/gst/rtspserver.c:
4026 rtsp-server: Limit the number of threads per server instance
4027 If we exceed the maximum, just round robin the clients over the existing
4030 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
4032 * gst/rtsp-server/rtsp-server.c:
4033 rtsp-server: No need to store the GMainContext in the client context
4035 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
4037 * tests/check/gst/rtspserver.c:
4038 tests: Add test for client disconnection
4040 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
4042 * tests/check/gst/rtspserver.c:
4043 tests: Test client and session timeouts with multiple threads
4045 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
4047 * gst/rtsp-server/rtsp-address-pool.c:
4048 * gst/rtsp-server/rtsp-auth.c:
4049 * gst/rtsp-server/rtsp-client.c:
4050 * gst/rtsp-server/rtsp-media-factory-uri.c:
4051 * gst/rtsp-server/rtsp-media-factory.c:
4052 * gst/rtsp-server/rtsp-media.c:
4053 * gst/rtsp-server/rtsp-mount-points.c:
4054 * gst/rtsp-server/rtsp-server.c:
4055 * gst/rtsp-server/rtsp-session-media.c:
4056 * gst/rtsp-server/rtsp-session-pool.c:
4057 * gst/rtsp-server/rtsp-session.c:
4058 Document locking and its order
4060 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
4062 * tests/check/gst/rtspserver.c:
4063 tests: Test that slow DESCRIBE don't block other clients
4065 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
4067 * tests/check/gst/client.c:
4068 tests: Add tests for client-requested multicast address
4070 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
4072 * docs/libs/gst-rtsp-server-sections.txt:
4073 docs: Put the various functions in the right sections
4075 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
4077 * docs/libs/gst-rtsp-server-docs.sgml:
4078 * docs/libs/gst-rtsp-server-sections.txt:
4079 * gst/rtsp-server/rtsp-address-pool.c:
4080 * gst/rtsp-server/rtsp-address-pool.h:
4081 docs: Generate docs for GstRTSPAddressPool
4083 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
4085 * gst/rtsp-server/rtsp-client.c:
4086 * gst/rtsp-server/rtsp-stream.c:
4087 * gst/rtsp-server/rtsp-stream.h:
4088 client: Check client provided addresses against the address pool
4090 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
4092 * gst/rtsp-server/rtsp-address-pool.c:
4093 * gst/rtsp-server/rtsp-address-pool.h:
4094 * tests/check/gst/addresspool.c:
4095 address-pool: Add API to request a specific address from the pool
4096 Also add relevant unit tests.
4098 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
4100 * tests/check/gst/mediafactory.c:
4101 tests: Check the passing around of a RTSPAddressPool
4102 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
4103 way down to the stream.
4105 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
4107 * tests/check/gst/addresspool.c:
4108 tests: Add more tests for the address pool
4110 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
4112 * gst/rtsp-server/rtsp-address-pool.c:
4113 address-pool: Fix off by one error
4114 When splitting a port range, the port after a skip is not part of range.
4116 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
4119 Automatic update of common submodule
4120 From 2de221c to 04c7a1e
4122 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
4125 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
4126 AM_CONFIG_HEADER was removed in automake 1.13
4127 https://bugzilla.gnome.org/show_bug.cgi?id=693368
4129 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
4132 Automatic update of common submodule
4133 From a942293 to 2de221c
4135 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4137 * gst/rtsp-server/rtsp-client.c:
4138 client: make sure the watch exists while sending data
4139 Protect the send_func with a lock. This allows us to wait for sending
4140 to complete before changing the send_func and user_data. We add an
4141 extra ref to the watch to make sure that it remains valid during
4143 When closing the connection, set the send_func to NULL
4144 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
4146 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4148 * tests/check/Makefile.am:
4149 tests: use GST_*_1_0 environment variables everywhere
4150 The _1_0 suffixed environment variables override the
4151 non-suffixed ones, so if we're in an environment that
4152 sets the _1_0 suffixed ones, such as jhbuild, we need
4153 to set those to make sure ours actually always get
4156 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4159 Automatic update of common submodule
4160 From acb04d9 to a942293
4162 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4164 * gst/rtsp-server/rtsp-client.c:
4165 rtsp-client: set the client backlog
4166 Set the client backlog to a reasonable default
4168 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
4170 * gst/rtsp-server/rtsp-media.c:
4171 rtsp-media: Make the element a constructor parameter
4172 https://bugzilla.gnome.org/show_bug.cgi?id=689594
4174 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
4176 * docs/libs/Makefile.am:
4177 docs: Link with gcov library when gcov is enabled
4178 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
4180 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4182 * gst/rtsp-server/rtsp-media.c:
4183 media: match prepare with unprepare
4184 Really unprepare when there were an equal amount of prepare calls.
4186 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4188 * gst/rtsp-server/rtsp-media.c:
4189 media: media has to be unprepared in finalize
4190 Because unprepare takes away the last ref on the media.
4192 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4194 * gst/rtsp-server/rtsp-client.c:
4195 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
4196 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
4197 We can't use the refcount to trigger unprepare because it is the unprepare call
4198 that removes the last refcount after all messages are consumed. What we should
4199 probably do is make a prepared refcount and only unprepare when the refcount
4202 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4204 * gst/rtsp-server/rtsp-media.c:
4205 media: let the source unref the last media ref
4206 the last ref to the media is held by the source so we don't need to add more ref
4207 and unrefs, we simply destroy the media when the source is gone.
4209 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4211 * gst/rtsp-server/rtsp-media.c:
4212 media: improve debug
4214 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4216 * gst/rtsp-server/rtsp-media.c:
4218 Make sure we are in the right state when collecting the position and duration.
4219 Only make ourselves PREPARED when we were previously PREPARING.
4221 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4223 * gst/rtsp-server/rtsp-media.c:
4224 media: use g_object_ref/unref for GObjects
4226 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
4228 * gst/rtsp-server/rtsp-client.c:
4229 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
4230 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
4231 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
4232 isn't being used anymore.
4234 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
4236 * gst/rtsp-server/rtsp-media.c:
4237 Fix compiler warning
4239 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
4241 * gst/rtsp-server/rtsp-media-factory-uri.c:
4242 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
4244 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4246 * gst/rtsp-server/rtsp-session-media.h:
4249 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4251 * gst/rtsp-server/rtsp-media.c:
4252 * tests/check/gst/media.c:
4253 media: avoid element leak
4255 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4257 * gst/rtsp-server/rtsp-media.c:
4258 media: require an element in media constructor
4260 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4262 * gst/rtsp-server/rtsp-client.c:
4263 Revert "client: TEARDOWN brings that state to Init again"
4264 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
4265 The object is already disposed, there is no point in setting the state.
4267 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4269 * gst/rtsp-server/rtsp-client.c:
4270 client: TEARDOWN brings that state to Init again
4272 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4274 * docs/libs/gst-rtsp-server-sections.txt:
4275 * examples/test-auth.c:
4276 * gst/rtsp-server/rtsp-auth.c:
4277 * gst/rtsp-server/rtsp-auth.h:
4278 * gst/rtsp-server/rtsp-client.c:
4279 * gst/rtsp-server/rtsp-client.h:
4280 * gst/rtsp-server/rtsp-media-factory-uri.c:
4281 * gst/rtsp-server/rtsp-media-factory-uri.h:
4282 * gst/rtsp-server/rtsp-media-factory.c:
4283 * gst/rtsp-server/rtsp-media-factory.h:
4284 * gst/rtsp-server/rtsp-media.c:
4285 * gst/rtsp-server/rtsp-media.h:
4286 * gst/rtsp-server/rtsp-mount-points.c:
4287 * gst/rtsp-server/rtsp-mount-points.h:
4288 * gst/rtsp-server/rtsp-sdp.c:
4289 * gst/rtsp-server/rtsp-server.c:
4290 * gst/rtsp-server/rtsp-server.h:
4291 * gst/rtsp-server/rtsp-session-media.c:
4292 * gst/rtsp-server/rtsp-session-media.h:
4293 * gst/rtsp-server/rtsp-session-pool.c:
4294 * gst/rtsp-server/rtsp-session-pool.h:
4295 * gst/rtsp-server/rtsp-session.c:
4296 * gst/rtsp-server/rtsp-session.h:
4297 * gst/rtsp-server/rtsp-stream-transport.c:
4298 * gst/rtsp-server/rtsp-stream-transport.h:
4299 * gst/rtsp-server/rtsp-stream.c:
4300 * gst/rtsp-server/rtsp-stream.h:
4301 * tests/check/gst/media.c:
4302 rtsp: make object details private
4303 Make all object details private
4304 Add methods to access private bits
4306 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4308 * tests/check/Makefile.am:
4309 * tests/check/gst/media.c:
4310 tests: add media tests
4312 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4314 * gst/rtsp-server/rtsp-media.c:
4315 media: check if prepared for some methods
4316 Check that the media object is prepared before doing seek and getting the
4317 current position etc.
4318 Add some g_return checks.
4320 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4322 * tests/check/Makefile.am:
4323 * tests/check/gst/mediafactory.c:
4324 tests: add mediafactory test
4326 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4328 * gst/rtsp-server/rtsp-stream.c:
4329 stream: improve debug
4331 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4333 * gst/rtsp-server/rtsp-media.c:
4334 * gst/rtsp-server/rtsp-media.h:
4335 media: unref pipeline in finalize to avoid leaking it
4337 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4339 * gst/rtsp-server/rtsp-media-factory-uri.c:
4340 * gst/rtsp-server/rtsp-media.c:
4341 rtsp: use gst_object_unref on GstObjects
4343 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4345 * gst/rtsp-server/rtsp-media-factory.c:
4346 media-factory: require an url
4348 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4350 * examples/test-uri.c:
4351 examples: fix include
4353 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4355 * gst/rtsp-server/rtsp-server.h:
4356 server: remove unused include
4358 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4360 * tests/check/Makefile.am:
4361 * tests/check/gst/mountpoints.c:
4362 tests: add test for mountpoints
4364 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4366 * gst/rtsp-server/rtsp-client.c:
4367 client: fix factory leak
4368 Keep the factory in the state object only for authorization checks and make
4369 sure we unref it on failure. Also don't keep invalid objects in the state
4372 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4374 * gst/rtsp-server/rtsp-mount-points.c:
4375 mounts: add g_return_if guards
4377 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4379 * tests/check/gst/client.c:
4380 tests: add more tests
4382 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4384 * gst/rtsp-server/rtsp-client.c:
4385 client: improve debug
4387 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4389 * gst/rtsp-server/rtsp-client.c:
4390 client: improve debug and fix leaks
4391 Cleanup the uri and session when there is a bad request.
4393 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4398 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4400 * tests/check/gst/client.c:
4401 test: add test for session in options request
4403 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4405 * gst/rtsp-server/rtsp-client.c:
4406 client: use 454 when session can't be found
4407 We should use 454 when a session can't be found because there was no session
4408 pool configured in the server. This is not a server configuration problem
4409 because the server on which the request is done might not be the same one that
4410 will keep the sessions for us and so it does not need to support sessions.
4412 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4414 * gst/rtsp-server/rtsp-client.c:
4415 client: only free connection when there is one
4416 It's possible that the client doesn't have a connection when we try to free it.
4418 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4420 * tests/check/Makefile.am:
4421 * tests/check/gst/client.c:
4422 tests: add unit test for the client object
4424 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4426 * gst/rtsp-server/rtsp-client.c:
4427 client: small cleanup
4429 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4431 * gst/rtsp-server/rtsp-client.h:
4432 client: remove unused include
4434 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4436 * gst/rtsp-server/rtsp-client.c:
4437 client: fix compilation
4439 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4441 * gst/rtsp-server/rtsp-client.c:
4442 client: call destroy without the lock
4444 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4446 * gst/rtsp-server/rtsp-client.c:
4447 * gst/rtsp-server/rtsp-client.h:
4448 client: make the client usable without a socket
4449 Make a method to let the client handle a message and a callback when the client
4450 wants us to send a response message back. This makes it possible to also use the
4451 client object without the sockets, which should make it easier to test.
4453 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4455 * gst/rtsp-server/rtsp-client.c:
4456 * gst/rtsp-server/rtsp-client.h:
4457 client: small cleanup
4459 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4461 * docs/libs/gst-rtsp-server-sections.txt:
4462 * gst/rtsp-server/rtsp-client.c:
4463 * gst/rtsp-server/rtsp-client.h:
4464 * gst/rtsp-server/rtsp-server.c:
4465 client: remove reference to server
4466 We don't need to keep a ref to the server
4468 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4470 * gst/rtsp-server/rtsp-client.c:
4471 * gst/rtsp-server/rtsp-client.h:
4473 Also add some g_return_if()
4475 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4477 * gst/rtsp-server/rtsp-client.c:
4478 client: log more errors
4480 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4482 * gst/rtsp-server/rtsp-client.c:
4483 client: fix compilation
4485 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4487 * gst/rtsp-server/rtsp-client.c:
4488 * gst/rtsp-server/rtsp-client.h:
4489 client: add generic close-after-send support
4490 Add a property to send_response() to close the connection after the response has
4491 been sent to the client.
4493 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4496 * docs/libs/gst-rtsp-server-docs.sgml:
4497 * docs/libs/gst-rtsp-server-sections.txt:
4498 * docs/libs/gst-rtsp-server.types:
4499 * examples/test-auth.c:
4500 * examples/test-launch.c:
4501 * examples/test-mp4.c:
4502 * examples/test-multicast.c:
4503 * examples/test-multicast2.c:
4504 * examples/test-ogg.c:
4505 * examples/test-readme.c:
4506 * examples/test-sdp.c:
4507 * examples/test-uri.c:
4508 * examples/test-video.c:
4509 * gst/rtsp-server/Makefile.am:
4510 * gst/rtsp-server/rtsp-auth.h:
4511 * gst/rtsp-server/rtsp-client.c:
4512 * gst/rtsp-server/rtsp-client.h:
4513 * gst/rtsp-server/rtsp-media-mapping.c:
4514 * gst/rtsp-server/rtsp-media-mapping.h:
4515 * gst/rtsp-server/rtsp-mount-points.c:
4516 * gst/rtsp-server/rtsp-mount-points.h:
4517 * gst/rtsp-server/rtsp-server.c:
4518 * gst/rtsp-server/rtsp-server.h:
4519 * gst/rtsp-server/rtsp-session-media.c:
4520 * gst/rtsp-server/rtsp-session-pool.c:
4521 * gst/rtsp-server/rtsp-session-pool.h:
4522 * tests/check/gst/rtspserver.c:
4523 MediaMapping -> MountPoints
4524 Describes better what the object manages.
4526 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4529 configure: bump required version of -base
4531 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4533 * gst/rtsp-server/rtsp-media.c:
4536 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4538 * gst/rtsp-server/rtsp-media.c:
4539 * gst/rtsp-server/rtsp-media.h:
4540 media: support more Range formats
4541 Use the new -base methods to convert the Range string into a seek start and stop
4544 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4546 * examples/test-launch.c:
4547 examples: fix whitespace
4549 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4551 * examples/test-auth.c:
4552 test-auth: add example of how to remove sessions
4553 Add an example of the session filter api.
4555 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4557 * examples/test-uri.c:
4558 test-uri: remove mapping example
4560 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4562 * examples/test-uri.c:
4563 test-uri: fix callback signature
4565 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4567 * gst/rtsp-server/rtsp-media-factory.c:
4568 factory: keep ref to factory while media active
4569 While the media from a factory is alive, keep a ref to the factory.
4570 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
4572 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4574 * gst/rtsp-server/rtsp-media-factory-uri.c:
4575 factory-uri: add some debug
4577 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4579 * gst/rtsp-server/rtsp-stream.c:
4580 stream: set udp sources to PLAYING
4581 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
4582 so that it doesn't cause our pipeline to produce ASYNC-DONE.
4584 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4586 * gst/rtsp-server/rtsp-media-factory-uri.c:
4587 factory-uri: take ref to factory
4588 Take a ref to the factory that we place in our list.
4590 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4592 * tests/Makefile.am:
4593 * tests/test-reuse.c:
4594 test: add test for server reuse
4595 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
4597 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
4599 * gst/rtsp-server/rtsp-server.c:
4600 server: start and stop multiple times
4601 Stop listening on the RTSP port when the GSource is removed, so clients
4602 can't connect and the server can be started again.
4603 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
4605 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4607 * gst/rtsp-server/rtsp-server.c:
4608 server: fix small leak
4610 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4612 * gst/rtsp-server/rtsp-media.c:
4613 media: unref source in finish_unprepare
4614 The source is created in prepare, unref it in finish_unprepare.
4615 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
4617 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
4619 * gst/rtsp-server/rtsp-client.c:
4620 * gst/rtsp-server/rtsp-media.c:
4621 rtsp-media: remove bus watch before finalizing
4622 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
4623 * An extra media ref is added for the bus watch. This extra ref is unreffed by
4624 the GDestroyNotify function.
4625 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
4626 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
4627 gst_rtsp_media_unprepare before unreffing the media.
4628 This way, the bus watch will be removed before the media is finalized.
4629 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
4631 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
4633 * gst/rtsp-server/rtsp-client.c:
4634 * gst/rtsp-server/rtsp-client.h:
4635 client: wait until the TEARDOWN response is sent to close the connection
4636 Responses can be sent async so we need to wait until the TEARDOWN response has
4637 been written before we close the connection to the client. This avoids the risk
4638 of writing/polling closed sockets.
4639 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
4641 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
4643 * gst/rtsp-server/rtsp-stream.c:
4644 rtsp-stream: plug socket leak
4645 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
4647 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
4650 Automatic update of common submodule
4651 From 6bb6951 to a72faea
4653 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
4655 * gst/rtsp-server/rtsp-media-factory-uri.c:
4656 rtsp-server: don't use deprecated API
4658 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
4660 * gst/rtsp-server/rtsp-client.c:
4661 rtsp-client: fix unused-but-set-variable compiler warning
4662 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
4664 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4667 * docs/libs/gst-rtsp-server-sections.txt:
4668 * gst/rtsp-server/rtsp-client.c:
4671 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4673 * examples/Makefile.am:
4674 * examples/test-multicast2.c:
4675 examples: add another multicast example
4676 Add an example for how to configure separate multicast ranges for each media
4679 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4681 * examples/test-multicast.c:
4684 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4686 * gst/rtsp-server/rtsp-client.c:
4687 * gst/rtsp-server/rtsp-media.c:
4688 * gst/rtsp-server/rtsp-session-media.c:
4689 * gst/rtsp-server/rtsp-session-media.h:
4690 * gst/rtsp-server/rtsp-stream-transport.c:
4691 * gst/rtsp-server/rtsp-stream-transport.h:
4692 stream: use the address managed by the stream
4693 Use the address managed by the stream for multicast. This allows us to have 1
4694 multicast address for each stream.
4695 Because the address is now managed by the stream we don't have to pass it around
4697 Set the address pool on the streams.
4699 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4701 * gst/rtsp-server/rtsp-client.c:
4702 * gst/rtsp-server/rtsp-media.c:
4703 * gst/rtsp-server/rtsp-stream.c:
4706 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4708 * gst/rtsp-server/rtsp-media.c:
4709 * gst/rtsp-server/rtsp-media.h:
4710 media: add signal for new streams
4711 This allows applications to listen for new streams and configure properties on
4712 them, like the address pool.
4714 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4716 * gst/rtsp-server/rtsp-media.c:
4717 media: configure address pool in new streams
4719 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4721 * gst/rtsp-server/rtsp-stream.c:
4722 * gst/rtsp-server/rtsp-stream.h:
4723 stream: add methods to deal with address pool
4724 Add methods to get and set the address pool for the stream
4725 Add method to allocate and get the multicast addresses for this stream.
4727 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4729 * docs/libs/gst-rtsp-server-sections.txt:
4730 * gst/rtsp-server/rtsp-media.c:
4731 * gst/rtsp-server/rtsp-media.h:
4732 media: remove MTU property
4733 It is a stream property
4735 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4737 * gst/rtsp-server/rtsp-client.c:
4738 client: set blocksize only on stream
4739 Set the blocksize only on the current stream.
4741 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4743 * gst/rtsp-server/rtsp-stream.c:
4744 stream: share src and sink sockets
4745 the allocated socket is in the used-socket property, not socket.
4747 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4749 * gst/rtsp-server/rtsp-address-pool.c:
4750 * gst/rtsp-server/rtsp-address-pool.h:
4751 * gst/rtsp-server/rtsp-client.c:
4752 * gst/rtsp-server/rtsp-session-media.c:
4753 * gst/rtsp-server/rtsp-session-media.h:
4754 * gst/rtsp-server/rtsp-stream-transport.c:
4755 * gst/rtsp-server/rtsp-stream-transport.h:
4756 * tests/check/gst/addresspool.c:
4757 rtsp: make address-pool return an address object
4758 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
4759 store more info in the structure and allows us to more easily return the address
4760 to the right pool when no longer needed.
4761 Pass the address to the StreamTransport so that we can return it to the pool
4762 when the stream transport is freed or changed.
4764 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4766 * examples/Makefile.am:
4767 * examples/test-multicast.c:
4768 examples: add multicast example
4769 Show how to set up the multicast address pool so that media can be
4770 server with multicast.
4772 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4774 * gst/rtsp-server/rtsp-client.c:
4775 * gst/rtsp-server/rtsp-media-factory.c:
4776 * gst/rtsp-server/rtsp-media-factory.h:
4777 * gst/rtsp-server/rtsp-media.c:
4778 * gst/rtsp-server/rtsp-media.h:
4779 rtsp: use AddressPool
4780 Remove the multicast_group property.
4781 Use the configured addresspool to allocate multicast addresses.
4783 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4785 * gst/rtsp-server/rtsp-address-pool.c:
4786 * gst/rtsp-server/rtsp-address-pool.h:
4787 address-pool: add clear method
4789 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4791 * gst/rtsp-server/rtsp-address-pool.c:
4792 address-pool: small cleanups
4794 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4796 * tests/check/Makefile.am:
4797 * tests/check/gst/addresspool.c:
4798 tests: add addresspool unit test
4800 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4802 * gst/rtsp-server/Makefile.am:
4803 * gst/rtsp-server/rtsp-address-pool.c:
4804 * gst/rtsp-server/rtsp-address-pool.h:
4805 address-pool: add object to manage multicast addresses
4806 Make an object that can manage a rage of multicast addresses and ports.
4808 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4810 * gst/rtsp-server/rtsp-server.c:
4811 server: set default max-threads property
4813 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4815 * gst/rtsp-server/rtsp-media.c:
4816 media: wait for concurrent _prepare
4817 If a prepare is busy, wait for the result.
4819 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4821 * gst/rtsp-server/rtsp-media.c:
4822 media: add lock around message handler
4823 We don't want to dispatch messages while we are still processing the result of
4826 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4828 * gst/rtsp-server/rtsp-media.c:
4829 * gst/rtsp-server/rtsp-media.h:
4830 media: add lock to protect state changes
4832 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4834 * gst/rtsp-server/rtsp-stream.c:
4835 * gst/rtsp-server/rtsp-stream.h:
4838 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4840 * gst/rtsp-server/rtsp-stream-transport.c:
4841 * gst/rtsp-server/rtsp-stream-transport.h:
4842 * gst/rtsp-server/rtsp-stream.c:
4843 stream-transport: add keep-alive method
4845 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4847 * gst/rtsp-server/rtsp-stream-transport.c:
4848 * gst/rtsp-server/rtsp-stream-transport.h:
4849 * gst/rtsp-server/rtsp-stream.c:
4850 stream-transport: add method to handle RTP/RTCP
4851 Call new methods instead of poking into the structures directly.
4853 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4855 * gst/rtsp-server/rtsp-session-media.c:
4856 * gst/rtsp-server/rtsp-session-media.h:
4857 session-media: add locking
4859 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4861 * gst/rtsp-server/rtsp-session.c:
4862 * gst/rtsp-server/rtsp-session.h:
4863 session: add locking
4865 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4867 * gst/rtsp-server/rtsp-server.c:
4868 server: free old socket
4870 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4872 * gst/rtsp-server/rtsp-media-mapping.c:
4873 * gst/rtsp-server/rtsp-media-mapping.h:
4874 mapping: add locking
4876 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4878 * gst/rtsp-server/rtsp-media-factory.c:
4879 media-factory: add locking
4881 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4883 * gst/rtsp-server/rtsp-auth.c:
4884 * gst/rtsp-server/rtsp-auth.h:
4887 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4889 * gst/rtsp-server/rtsp-server.c:
4890 * gst/rtsp-server/rtsp-server.h:
4891 server: add max-thread property
4893 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4895 * gst/rtsp-server/rtsp-server.c:
4896 * gst/rtsp-server/rtsp-server.h:
4897 server: use a threadpool for the mainloops
4899 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4901 * gst/rtsp-server/rtsp-client.c:
4902 * gst/rtsp-server/rtsp-client.h:
4903 client: rename method
4904 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
4905 don't really create the client from the socket, we use the socket for the
4908 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4910 * gst/rtsp-server/rtsp-client.c:
4911 * gst/rtsp-server/rtsp-client.h:
4912 * gst/rtsp-server/rtsp-server.c:
4913 server: rework maincontext handling in clients
4914 Make a separate method to attach a client to a MainContext.
4915 Let the server decide in what GMainContext the client will operate and give this
4916 context to the client in attach. Then the server can later decide to use a
4917 separate thread for each client or just use the mainthread.
4919 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4921 * gst/rtsp-server/rtsp-client.c:
4922 * gst/rtsp-server/rtsp-session.c:
4923 * gst/rtsp-server/rtsp-session.h:
4924 session: move session header code in session object
4926 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
4930 * examples/test-auth.c:
4931 * examples/test-launch.c:
4932 * examples/test-mp4.c:
4933 * examples/test-ogg.c:
4934 * examples/test-readme.c:
4935 * examples/test-sdp.c:
4936 * examples/test-uri.c:
4937 * examples/test-video.c:
4938 * gst/rtsp-server/rtsp-auth.c:
4939 * gst/rtsp-server/rtsp-auth.h:
4940 * gst/rtsp-server/rtsp-client.c:
4941 * gst/rtsp-server/rtsp-client.h:
4942 * gst/rtsp-server/rtsp-media-factory-uri.c:
4943 * gst/rtsp-server/rtsp-media-factory-uri.h:
4944 * gst/rtsp-server/rtsp-media-factory.c:
4945 * gst/rtsp-server/rtsp-media-factory.h:
4946 * gst/rtsp-server/rtsp-media-mapping.c:
4947 * gst/rtsp-server/rtsp-media-mapping.h:
4948 * gst/rtsp-server/rtsp-media.c:
4949 * gst/rtsp-server/rtsp-media.h:
4950 * gst/rtsp-server/rtsp-params.c:
4951 * gst/rtsp-server/rtsp-params.h:
4952 * gst/rtsp-server/rtsp-sdp.c:
4953 * gst/rtsp-server/rtsp-sdp.h:
4954 * gst/rtsp-server/rtsp-server.c:
4955 * gst/rtsp-server/rtsp-server.h:
4956 * gst/rtsp-server/rtsp-session-media.c:
4957 * gst/rtsp-server/rtsp-session-media.h:
4958 * gst/rtsp-server/rtsp-session-pool.c:
4959 * gst/rtsp-server/rtsp-session-pool.h:
4960 * gst/rtsp-server/rtsp-session.c:
4961 * gst/rtsp-server/rtsp-session.h:
4962 * gst/rtsp-server/rtsp-stream-transport.c:
4963 * gst/rtsp-server/rtsp-stream-transport.h:
4964 * gst/rtsp-server/rtsp-stream.c:
4965 * gst/rtsp-server/rtsp-stream.h:
4966 * tests/check/gst/rtspserver.c:
4967 * tests/test-cleanup.c:
4970 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
4972 * gst/rtsp-server/rtsp-media.c:
4973 * gst/rtsp-server/rtsp-session-media.c:
4974 * gst/rtsp-server/rtsp-session.c:
4975 rtsp-server: added annotations to indicate type of ownership transfer of return values
4976 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4978 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
4981 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
4983 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
4986 * bindings/Makefile.am:
4987 * bindings/vala/Makefile.am:
4988 * bindings/vala/gst-rtsp-server-0.10.deps:
4989 * bindings/vala/gst-rtsp-server-0.10.vapi:
4990 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
4991 * bindings/vala/packages/gst-rtsp-server-0.10.files:
4992 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
4993 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4994 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
4996 bindings: remove vala bindings
4997 They'll be reunited with the other GStreamer bindings
4998 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5000 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5002 * gst/rtsp-server/rtsp-client.c:
5003 * gst/rtsp-server/rtsp-session-media.c:
5004 * gst/rtsp-server/rtsp-session-media.h:
5005 * gst/rtsp-server/rtsp-stream-transport.c:
5006 * gst/rtsp-server/rtsp-stream-transport.h:
5007 rtsp: only create transport when needed
5008 Only create the StreamTransport when configured.
5010 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5012 * gst/rtsp-server/rtsp-client.c:
5013 client: small cleanup
5015 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5017 * gst/rtsp-server/rtsp-client.c:
5018 * gst/rtsp-server/rtsp-client.h:
5019 * gst/rtsp-server/rtsp-stream-transport.c:
5020 * gst/rtsp-server/rtsp-stream-transport.h:
5021 rtsp: refactor configuration of transport
5022 Move the configuration of the transport to a place where it makes
5025 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5027 * gst/rtsp-server/rtsp-client.c:
5028 client: refactor transport parsing
5030 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5032 * gst/rtsp-server/rtsp-client.c:
5033 client: refuse to change the MTU on shared media
5034 If we change the MTU of chared media, it changes for all clients.
5035 We don't want to set the MTU to something large for clients that
5038 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5040 * examples/test-mp4.c:
5041 * gst/rtsp-server/rtsp-media.c:
5042 small fixes to docs and debug
5044 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5046 * gst/rtsp-server/rtsp-stream.c:
5047 stream: transports must already have been removed
5049 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5051 * gst/rtsp-server/rtsp-media.c:
5052 * gst/rtsp-server/rtsp-stream.c:
5053 * gst/rtsp-server/rtsp-stream.h:
5054 stream: improve join and leave of the pipeline
5056 Do the cleanup properly
5059 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5061 * gst/rtsp-server/rtsp-media.c:
5062 media: move unprepare below default implementation
5063 Makes it easier to find the default implementation
5065 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5067 * gst/rtsp-server/rtsp-media.c:
5068 media: signal unprepared when we actually finish
5070 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5072 * gst/rtsp-server/rtsp-media.c:
5073 media: no need to unlock, unprepare does that when needed
5075 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5077 * docs/libs/gst-rtsp-server-sections.txt:
5078 * gst/rtsp-server/rtsp-media-factory.h:
5079 * gst/rtsp-server/rtsp-media-mapping.c:
5080 * gst/rtsp-server/rtsp-media.h:
5081 * gst/rtsp-server/rtsp-params.c:
5082 * gst/rtsp-server/rtsp-server.c:
5083 * gst/rtsp-server/rtsp-session-pool.h:
5084 * gst/rtsp-server/rtsp-session.c:
5085 * gst/rtsp-server/rtsp-session.h:
5086 * gst/rtsp-server/rtsp-stream-transport.h:
5087 * gst/rtsp-server/rtsp-stream.h:
5090 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5092 * gst/rtsp-server/rtsp-client.c:
5093 * gst/rtsp-server/rtsp-media-mapping.h:
5094 * gst/rtsp-server/rtsp-media.c:
5095 * gst/rtsp-server/rtsp-media.h:
5096 * gst/rtsp-server/rtsp-server.h:
5097 * gst/rtsp-server/rtsp-stream.c:
5098 * gst/rtsp-server/rtsp-stream.h:
5099 rtsp: fix MTU setting
5100 Fix setting of the MTU. There is no need for a vmethod.
5102 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5107 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5110 configure: bump version number after refactoring
5112 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5114 * gst/rtsp-server/Makefile.am:
5115 * gst/rtsp-server/rtsp-client.c:
5116 * gst/rtsp-server/rtsp-client.h:
5117 * gst/rtsp-server/rtsp-media-factory-uri.c:
5118 * gst/rtsp-server/rtsp-media-factory.c:
5119 * gst/rtsp-server/rtsp-media-factory.h:
5120 * gst/rtsp-server/rtsp-media.c:
5121 * gst/rtsp-server/rtsp-media.h:
5122 * gst/rtsp-server/rtsp-sdp.c:
5123 * gst/rtsp-server/rtsp-session-media.c:
5124 * gst/rtsp-server/rtsp-session-media.h:
5125 * gst/rtsp-server/rtsp-session.c:
5126 * gst/rtsp-server/rtsp-session.h:
5127 * gst/rtsp-server/rtsp-stream-transport.c:
5128 * gst/rtsp-server/rtsp-stream-transport.h:
5129 * gst/rtsp-server/rtsp-stream.c:
5130 * gst/rtsp-server/rtsp-stream.h:
5131 rtsp: massive refactoring
5132 Make GObjects from the remaining simple structures.
5133 Remove GstRTSPSessionStream, it's not needed.
5134 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
5135 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
5136 a GstRTSPStream should be transported to a client.
5137 Rename GstRTSPMediaFactory::get_element -> create_element because that
5138 more accurately describes what it does.
5139 Make nice methods instead of poking in the structures.
5140 Move some methods inside the relevant object source code.
5141 Use GPtrArray to store objects instead of plain arrays, it is more
5142 natural and allows us to more easily clean up.
5143 Move the allocation of udp ports to the Stream object. The Stream object
5144 contains the elements needed to stream the media to a client.
5145 Improve the prepare and unprepare methods. Unprepare should now undo
5146 everything prepare did. Improve also async unprepare when doing EOS on
5147 shutdown. Make sure we always unprepare correctly.
5149 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
5151 * gst/rtsp-server/rtsp-client.c:
5152 rtsp-client: Unref server address clients connected to
5153 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
5155 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
5157 * gst/rtsp-server/rtsp-server.c:
5158 rtsp-server: don't ref server socket if it is NULL
5159 Fixes test_bind_already_in_use unit test again after commit 6a497440.
5160 https://bugzilla.gnome.org/show_bug.cgi?id=686644
5162 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
5164 * tests/check/Makefile.am:
5165 tests: Add libgio link dependency
5166 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
5168 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5170 * gst/rtsp-server/rtsp-media-mapping.c:
5171 * gst/rtsp-server/rtsp-media-mapping.h:
5172 rtsp-media-mapping: rename find_media vfunc to find_factory
5173 The virtual method and class method should have the same name
5174 so it is correctly represented in GIR file
5175 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5177 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
5179 * gst/rtsp-server/rtsp-auth.c:
5180 * gst/rtsp-server/rtsp-client.c:
5181 * gst/rtsp-server/rtsp-media-factory-uri.c:
5182 * gst/rtsp-server/rtsp-media-factory.c:
5183 * gst/rtsp-server/rtsp-media-mapping.c:
5184 * gst/rtsp-server/rtsp-media.c:
5185 * gst/rtsp-server/rtsp-server.c:
5186 * gst/rtsp-server/rtsp-session-pool.c:
5187 * gst/rtsp-server/rtsp-session.c:
5188 rtsp-server: fixed comments and GIR annotations
5189 https://bugzilla.gnome.org/show_bug.cgi?id=680777
5191 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
5193 * gst/rtsp-server/rtsp-media-mapping.c:
5194 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
5196 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
5198 * gst/rtsp-server/rtsp-server.c:
5199 rtsp-server: allow binding on port 0 (binds on a random port)
5201 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
5203 * gst/rtsp-server/rtsp-server.c:
5204 * gst/rtsp-server/rtsp-server.h:
5205 rtsp-server: add bound-port property
5206 bound-port can be used to retrieve the port number when the server is bound on
5207 port 0, which binds on a random port.
5209 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
5211 * gst/rtsp-server/rtsp-media-factory.c:
5212 * gst/rtsp-server/rtsp-media-factory.h:
5213 rtsp-media-factory: make ::get_element overridable by GI bindings
5214 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
5215 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
5216 as the invoker for ::get_element(), making it overridable by GI generated
5219 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5221 * gst/rtsp-server/rtsp-media-factory-uri.c:
5222 rtsp-media-factory-uri: don't autoplug parsers in a loop
5223 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
5226 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5228 * gst/rtsp-server/Makefile.am:
5229 Explicitly link against gio. Fix link error on mac.
5231 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5233 * gst/rtsp-server/rtsp-session.c:
5234 session: add ttl to the transport header in SETUP
5235 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
5237 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
5239 * gst/rtsp-server/rtsp-client.c:
5240 * gst/rtsp-server/rtsp-client.h:
5241 * gst/rtsp-server/rtsp-media.c:
5242 client: Use client transport settings for multicast if allowed.
5243 This patch makes it possible for the client to send transport settings for
5244 multicast (destination && ttl). Client settings must be explicitly allowed or
5245 the server will use its own settings.
5246 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
5248 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
5251 Automatic update of common submodule
5252 From 6c0b52c to 6bb6951
5254 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
5256 * gst/rtsp-server/rtsp-client.c:
5257 rtsp-client: do not destroy the rtsp watch
5258 Don't destroy the client watch while dispatching. The rtsp watch is
5259 automatically destroyed after the rtsp watch function closed() has
5261 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
5263 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
5266 Automatic update of common submodule
5267 From 4f962f7 to 6c0b52c
5269 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
5271 * gst/rtsp-server/rtsp-media.c:
5272 media: fix check for seekability
5274 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5276 * gst/rtsp-server/rtsp-client.c:
5277 client: use more GIO
5278 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
5280 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5282 * gst/rtsp-server/rtsp-server.c:
5283 server: remove obsolete includes
5285 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5287 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
5288 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
5289 be available in "on_new_ssrc". The transports are added in
5290 gst_rtsp_media_set_state when going to PLAYING state. However,
5291 "on_new_ssrc" might be called before this happens.
5292 https://bugzilla.gnome.org/show_bug.cgi?id=683304
5294 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5296 * gst/rtsp-server/rtsp-client.c:
5297 * gst/rtsp-server/rtsp-client.h:
5298 rtsp-client: add signals for rtsp requests (fixes #683287)
5300 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
5302 * gst/rtsp-server/rtsp-client.c:
5303 * gst/rtsp-server/rtsp-client.h:
5304 add new-session signal to rtsp-client (fixes #683058)
5306 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
5309 Automatic update of common submodule
5310 From 668acee to 4f962f7
5312 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
5314 * gst/rtsp-server/rtsp-server.c:
5315 * tests/check/gst/rtspserver.c:
5316 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
5317 Do not assume that *error is set in g_socket_address_enumerator_next.
5318 Added test_bind_already_in_use unit-test.
5319 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
5321 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
5324 Automatic update of common submodule
5325 From 94ccf4c to 668acee
5327 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
5329 * gst/rtsp-server/rtsp-client.c:
5330 * gst/rtsp-server/rtsp-client.h:
5331 rtsp-client: make create_sdp virtual method
5332 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
5334 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5337 Automatic update of common submodule
5338 From 98e386f to 94ccf4c
5340 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5342 * gst/rtsp-server/rtsp-client.c:
5345 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
5347 * gst/rtsp-server/rtsp-client.c:
5348 * gst/rtsp-server/rtsp-client.h:
5349 * gst/rtsp-server/rtsp-server.c:
5350 * gst/rtsp-server/rtsp-server.h:
5351 rtsp-server: use an existing socket to establish HTTP tunnel
5352 Make it possible to transfer a socket from an HTTP server to be used as
5353 an RTSP over HTTP tunnel.
5355 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
5357 * gst/rtsp-server/rtsp-client.c:
5358 * gst/rtsp-server/rtsp-media.c:
5359 * gst/rtsp-server/rtsp-media.h:
5360 rtsp: Handle the blocksize parameter
5361 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
5363 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
5365 * tests/check/Makefile.am:
5366 * tests/check/gst/rtspserver.c:
5367 Have unit test get header from source dir, not installed dir
5368 This makes compilation of unit tests work in a build directory other
5369 than the source directory.
5370 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
5372 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
5374 * gst/rtsp-server/rtsp-media.c:
5375 rtsp-media: update for gst_element_make_from_uri() changes
5377 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
5380 * tests/Makefile.am:
5381 * tests/check/Makefile.am:
5382 * tests/check/gst/rtspserver.c:
5384 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
5386 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
5388 * gst/rtsp-server/rtsp-media.c:
5389 rtsp-media: don't collect media stats when going to NULL
5390 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
5392 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5394 * gst/rtsp-server/rtsp-client.c:
5395 client: don't leak transports
5397 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
5399 * gst/rtsp-server/rtsp-client.c:
5400 rtsp-client: free transport on no_stream in SETUP handler
5402 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
5404 * gst/rtsp-server/rtsp-client.c:
5405 rtsp-client: changed session media iteration
5406 In client_unlink_session: now don't iterate in session->medias
5407 list where items are removed by gst_rtsp_session_release_media.
5408 Instead, repeatedly remove the first item.
5410 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
5412 * gst/rtsp-server/rtsp-client.c:
5413 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
5414 GstRTSPSessionMedia is not a GObject type. When the
5415 GstRTSPSession is freed, it will free the media.
5417 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
5419 * gst/rtsp-server/rtsp-media-factory.c:
5420 factory: plug pad leak in collect_streams
5421 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
5422 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
5423 will take one reference, and the other reference will otherwise
5426 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
5429 configure: suppress some warnings when debug is disabled
5430 Warnings about unused variables should be suppressed if core has the
5431 debug system disabled.
5432 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5434 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5436 * docs/libs/Makefile.am:
5437 docs: fix build in uninstalled setup
5438 Include gst-plugins-base libs properly.
5440 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
5442 * docs/libs/gst-rtsp-server.types:
5443 docs: include headers defining rtsp-server object types
5444 Fixes compiler warnings during docs build.
5445 https://bugzilla.gnome.org/show_bug.cgi?id=676824
5447 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
5450 configure: Add warning flags for compiler when configuring
5451 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
5453 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5456 Automatic update of common submodule
5457 From 03a0e57 to 98e386f
5459 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5462 Automatic update of common submodule
5463 From 1fab359 to 03a0e57
5465 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
5467 * gst/rtsp-server/rtsp-client.c:
5468 client: fix GSocketAddress leak in gst_rtsp_client_accept
5469 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
5471 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5474 Automatic update of common submodule
5475 From f1b5a96 to 1fab359
5477 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5480 Automatic update of common submodule
5481 From 92b7266 to f1b5a96
5483 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5486 Automatic update of common submodule
5487 From ec1c4a8 to 92b7266
5489 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5492 Automatic update of common submodule
5493 From 3429ba6 to ec1c4a8
5495 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
5497 * gst/rtsp-server/rtsp-auth.c:
5498 * gst/rtsp-server/rtsp-client.c:
5499 * gst/rtsp-server/rtsp-media-factory-uri.c:
5500 * gst/rtsp-server/rtsp-server.c:
5501 rtsp: fix compiler warnings
5502 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
5504 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5507 Automatic update of common submodule
5508 From dc70203 to 3429ba6
5510 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5512 * gst/rtsp-server/rtsp-client.c:
5513 * gst/rtsp-server/rtsp-media-factory.c:
5514 * gst/rtsp-server/rtsp-media-factory.h:
5515 * gst/rtsp-server/rtsp-media.c:
5516 * gst/rtsp-server/rtsp-media.h:
5517 * gst/rtsp-server/rtsp-server.c:
5518 * gst/rtsp-server/rtsp-server.h:
5519 * gst/rtsp-server/rtsp-session-pool.c:
5520 * gst/rtsp-server/rtsp-session-pool.h:
5521 rtsp-server: port to new thread API
5523 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5526 Automatic update of common submodule
5527 From 6db25be to dc70203
5529 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5531 * gst/rtsp-server/rtsp-auth.c:
5532 * gst/rtsp-server/rtsp-auth.h:
5533 * gst/rtsp-server/rtsp-client.c:
5534 rtsp-server: Fix compilation and compiler warnings
5536 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5540 * gst/rtsp-server/Makefile.am:
5541 configure: Modernize autotools setup a bit
5542 Also we now only create tar.bz2 and tar.xz tarballs.
5544 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5547 Automatic update of common submodule
5548 From 464fe15 to 6db25be
5550 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5553 Automatic update of common submodule
5554 From 7fda524 to 464fe15
5556 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5559 * docs/libs/Makefile.am:
5560 * docs/version.entities.in:
5562 * gst/rtsp-server/Makefile.am:
5563 * pkgconfig/Makefile.am:
5564 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5565 * pkgconfig/gstreamer-rtsp-server.pc.in:
5566 * tests/Makefile.am:
5567 rtsp-server: Update versioning
5569 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5571 Merge remote-tracking branch 'origin/0.10'
5573 gst/rtsp-server/rtsp-session-pool.c
5575 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5577 * gst/rtsp-server/rtsp-session-pool.c:
5578 rtsp-server: Don't use deprecated GLib API
5580 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5582 Replace master with 0.11
5584 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5586 Merge branch 'master' into 0.11
5588 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5590 Merge branch 'master' into 0.11
5592 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
5595 A couple minor typo fixes
5597 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5599 * gst/rtsp-server/rtsp-media.c:
5600 media: fix state of the appqueue
5602 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5604 * gst/rtsp-server/rtsp-media-factory-uri.c:
5605 factory: use videoconvert
5607 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5609 * gst/rtsp-server/rtsp-media-factory-uri.c:
5610 factory: change to new style caps
5612 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5614 * gst/rtsp-server/rtsp-client.c:
5615 * gst/rtsp-server/rtsp-client.h:
5616 * gst/rtsp-server/rtsp-media-factory-uri.c:
5617 * gst/rtsp-server/rtsp-media.c:
5618 * gst/rtsp-server/rtsp-server.c:
5619 * gst/rtsp-server/rtsp-server.h:
5620 * gst/rtsp-server/rtsp-session-pool.c:
5621 rtsp-server: port to GIO
5624 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5627 configure: fix build
5629 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5632 docs: fix for gst_rtsp_server_set_port() -> _set_service()
5633 https://bugzilla.gnome.org/show_bug.cgi?id=666548
5635 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5638 * examples/Makefile.am:
5639 First rule of gst-rtsp-server club: don't talk about gst-phonon
5641 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5644 * pkgconfig/Makefile.am:
5645 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
5646 * pkgconfig/gst-rtsp-server.pc.in:
5647 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
5648 * pkgconfig/gstreamer-rtsp-server.pc.in:
5649 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
5650 For consistency with all other modules.
5652 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5654 * gst/rtsp-server/rtsp-client.c:
5655 rtsp-client: update for new map API
5657 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5660 * bindings/Makefile.am:
5661 * bindings/python/Makefile.am:
5662 * bindings/python/arg-types.py:
5663 * bindings/python/codegen/Makefile.am:
5664 * bindings/python/codegen/__init__.py:
5665 * bindings/python/codegen/argtypes.py:
5666 * bindings/python/codegen/code-coverage.py:
5667 * bindings/python/codegen/codegen.py:
5668 * bindings/python/codegen/definitions.py:
5669 * bindings/python/codegen/defsparser.py:
5670 * bindings/python/codegen/docextract.py:
5671 * bindings/python/codegen/docgen.py:
5672 * bindings/python/codegen/fileprefix.override:
5673 * bindings/python/codegen/fileprefixmodule.c:
5674 * bindings/python/codegen/h2def.py:
5675 * bindings/python/codegen/mergedefs.py:
5676 * bindings/python/codegen/mkskel.py:
5677 * bindings/python/codegen/override.py:
5678 * bindings/python/codegen/reversewrapper.py:
5679 * bindings/python/codegen/scmexpr.py:
5680 * bindings/python/rtspserver-types.defs:
5681 * bindings/python/rtspserver.defs:
5682 * bindings/python/rtspserver.override:
5683 * bindings/python/rtspservermodule.c:
5684 * bindings/python/test.py:
5686 python: remove pygst-based python bindings
5687 pygi is the future, apparently.
5689 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
5692 Automatic update of common submodule
5693 From c463bc0 to 7fda524
5695 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5698 Automatic update of common submodule
5699 From 2a59016 to c463bc0
5701 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5704 Automatic update of common submodule
5705 From 0807187 to 2a59016
5707 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5710 Automatic update of common submodule
5711 From 11f0cd5 to 0807187
5713 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5715 * examples/test-auth.c:
5716 example: update for new caps
5718 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5720 * examples/test-video.c:
5721 * gst/rtsp-server/rtsp-client.c:
5722 * gst/rtsp-server/rtsp-media-factory-uri.c:
5723 * gst/rtsp-server/rtsp-media.c:
5724 * gst/rtsp-server/rtsp-media.h:
5725 * gst/rtsp-server/rtsp-session.c:
5726 * gst/rtsp-server/rtsp-session.h:
5727 rtsp-server: port some more to 0.11
5729 Remove bufferlist stuff
5731 Add queue before appsink now that preroll-queue-len is gone.
5732 Update for request pad changes.
5734 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5736 Merge branch 'master' into 0.11
5738 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
5740 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5741 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
5742 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
5744 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
5746 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
5747 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
5748 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
5750 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5752 Merge branch 'master' into 0.11
5754 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5756 * gst/rtsp-server/rtsp-media.c:
5757 * gst/rtsp-server/rtsp-media.h:
5758 media: add a seekable boolean
5759 Maintain the seekable state with a new variable instead of reusing the
5762 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
5764 * gst/rtsp-server/rtsp-media.c:
5765 Disallow seek in live media
5767 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5769 Merge branch 'master' into 0.11
5771 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
5773 * gst/rtsp-server/rtsp-server.c:
5774 #ifdef statements for windows socket creation were missing
5776 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
5779 Automatic update of common submodule
5780 From a39eb83 to 11f0cd5
5782 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
5785 Automatic update of common submodule
5786 From 605cd9a to a39eb83
5788 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5790 Merge branch 'master' into 0.11
5792 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5794 * gst/rtsp-server/rtsp-client.c:
5795 client: use method to access property
5797 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5799 * gst/rtsp-server/rtsp-media-factory.c:
5800 * gst/rtsp-server/rtsp-media-factory.h:
5801 media-factory: add protocols property
5802 Add a property to configure the allowed protocols in the media created from the
5805 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5807 * gst/rtsp-server/rtsp-media-factory.c:
5808 * gst/rtsp-server/rtsp-media-factory.h:
5809 media-factory: add media-configure signal
5810 Add signal to allow the application to configure the media after it was created
5813 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5815 * gst/rtsp-server/rtsp-client.c:
5816 client: use method to access property
5818 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5820 * gst/rtsp-server/rtsp-media-factory.c:
5821 * gst/rtsp-server/rtsp-media-factory.h:
5822 media-factory: add protocols property
5823 Add a property to configure the allowed protocols in the media created from the
5826 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5828 * gst/rtsp-server/rtsp-media-factory.c:
5829 * gst/rtsp-server/rtsp-media-factory.h:
5830 media-factory: add media-configure signal
5831 Add signal to allow the application to configure the media after it was created
5834 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5836 Merge branch 'master' into 0.11
5838 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5840 * gst/rtsp-server/rtsp-client.c:
5841 client: use media multicast group
5843 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5845 * gst/rtsp-server/rtsp-media-factory.h:
5846 * gst/rtsp-server/rtsp-server.h:
5847 * gst/rtsp-server/rtsp-session-pool.h:
5848 * gst/rtsp-server/rtsp-session.h:
5851 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
5853 * gst/rtsp-server/rtsp-client.c:
5854 * gst/rtsp-server/rtsp-sdp.h:
5855 sdp: copy and free the server ip address
5856 Copy and free the server ip address to make memory management easier later.
5858 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5860 * gst/rtsp-server/rtsp-media-factory.c:
5861 media-factory: configure multicast in media
5863 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5865 * gst/rtsp-server/rtsp-media.c:
5866 * gst/rtsp-server/rtsp-media.h:
5867 media: add property for multicast group
5868 Add a property to configure the multicast group in the media.
5869 Based on patches from Marc Leeman and Robert Krakora.
5871 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5873 * gst/rtsp-server/rtsp-media-factory.c:
5874 * gst/rtsp-server/rtsp-media-factory.h:
5875 media-factory: add property for multicast group
5876 Add a property to configure the multicast group in the media factory.
5877 Based on patches from Marc Leeman and Robert Krakora.
5879 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5881 * gst/rtsp-server/rtsp-client.c:
5882 client: do configuration of transport in one place
5883 Move the configuration of the transport destination address to where we also
5884 configure the other bits.
5886 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5888 * gst/rtsp-server/rtsp-client.c:
5889 client: use media multicast group
5891 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5893 * gst/rtsp-server/rtsp-media-factory.h:
5894 * gst/rtsp-server/rtsp-server.h:
5895 * gst/rtsp-server/rtsp-session-pool.h:
5896 * gst/rtsp-server/rtsp-session.h:
5899 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
5901 * gst/rtsp-server/rtsp-client.c:
5902 * gst/rtsp-server/rtsp-sdp.h:
5903 sdp: copy and free the server ip address
5904 Copy and free the server ip address to make memory management easier later.
5906 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5908 * gst/rtsp-server/rtsp-media-factory.c:
5909 media-factory: configure multicast in media
5911 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5913 * gst/rtsp-server/rtsp-media.c:
5914 * gst/rtsp-server/rtsp-media.h:
5915 media: add property for multicast group
5916 Add a property to configure the multicast group in the media.
5917 Based on patches from Marc Leeman and Robert Krakora.
5919 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5921 * gst/rtsp-server/rtsp-media-factory.c:
5922 * gst/rtsp-server/rtsp-media-factory.h:
5923 media-factory: add property for multicast group
5924 Add a property to configure the multicast group in the media factory.
5925 Based on patches from Marc Leeman and Robert Krakora.
5927 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5929 * gst/rtsp-server/rtsp-client.c:
5930 client: do configuration of transport in one place
5931 Move the configuration of the transport destination address to where we also
5932 configure the other bits.
5934 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5936 Merge branch 'master' into 0.11
5938 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
5940 * gst/rtsp-server/rtsp-client.c:
5941 client: destroy pipeline on client disconnect with no prior TEARDOWN.
5942 The problem occurs when the client abruptly closes the connection without
5943 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
5944 server is where the pipeline gets torn down. Since this handler is not called,
5945 the pipeline remains and is up and running. Subsequent clients get their own
5946 pipelines and if the do not issue TEARDOWNs then those pipelines will also
5947 remain up and running. This is a resource leak.
5949 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5951 Merge branch 'master' into 0.11
5953 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
5955 * gst/rtsp-server/rtsp-media-factory.c:
5956 * gst/rtsp-server/rtsp-media-factory.h:
5957 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
5958 For example, it can be used to retrieve source elements like appsrc, in a more
5959 convenient way than subclassing get_element.
5961 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5963 Merge branch 'master' into 0.11
5965 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
5967 * gst/rtsp-server/rtsp-server.c:
5968 rtsp-server: hold on to reference while using object
5970 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5972 * gst/rtsp-server/rtsp-media.c:
5975 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5978 configure: use unstable api
5980 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
5982 * gst/rtsp-server/rtsp-client.c:
5983 client: fix reference counting
5985 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
5987 * gst/rtsp-server/rtsp-client.c:
5988 * gst/rtsp-server/rtsp-media.c:
5989 fix compiler warnings about unused variables
5991 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
5993 * examples/test-launch.c:
5994 * examples/test-readme.c:
5995 * examples/test-uri.c:
5996 * examples/test-video.c:
5997 examples: tell rtsp uri when ready
5999 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
6002 Automatic update of common submodule
6003 From 69b981f to 605cd9a
6005 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6007 * gst/rtsp-server/rtsp-client.c:
6008 client: update for buffer API change
6010 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6012 * gst/rtsp-server/Makefile.am:
6013 Makefile.am: 0.10 => @GST_MAJORMINOR@
6015 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6017 * gst/rtsp-server/rtsp-media-factory-uri.c:
6018 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
6020 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6022 * gst/rtsp-server/.gitignore:
6023 .gitignore: 0.10 => 0.11
6025 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
6027 * gst/rtsp-server/Makefile.am:
6028 Makefile.am: 0.10 => @GST_MAJORMINOR@
6030 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6032 Merge branch 'master' into 0.11
6034 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
6037 Automatic update of common submodule
6038 From 9e5bbd5 to 69b981f
6040 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
6043 Automatic update of common submodule
6044 From fd35073 to 9e5bbd5
6046 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
6049 Automatic update of common submodule
6050 From 46dfcea to fd35073
6052 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6054 * gst/rtsp-server/rtsp-media-factory-uri.c:
6055 * gst/rtsp-server/rtsp-media.c:
6056 media: port to new caps API
6058 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6060 Merge branch 'master' into 0.11
6062 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6064 * bindings/vala/gst-rtsp-server-0.10.vapi:
6065 Updated Vala bindings.
6066 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6068 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
6070 * gst/rtsp-server/rtsp-server.c:
6071 * gst/rtsp-server/rtsp-server.h:
6072 Add a signal for newly connected clients.
6073 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
6075 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
6077 * bindings/python/rtspserver.override:
6078 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
6080 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6082 * gst/rtsp-server/Makefile.am:
6083 * gst/rtsp-server/rtsp-client.c:
6084 * gst/rtsp-server/rtsp-funnel.c:
6085 * gst/rtsp-server/rtsp-funnel.h:
6086 * gst/rtsp-server/rtsp-media.c:
6087 rtsp-server: port to 0.11
6089 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6094 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6096 Merge branch 'master' into 0.11
6101 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6104 Automatic update of common submodule
6105 From c3cafe1 to 46dfcea
6107 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
6109 * bindings/python/Makefile.am:
6110 * bindings/python/rtspserver.defs:
6111 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
6113 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
6115 * bindings/python/arg-types.py:
6116 python bindings: add GstRTSPUrlParam
6117 Needed to implement MediaFactory virtual proxies
6119 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
6121 * bindings/python/arg-types.py:
6122 python bindings: fix returning GstRTSPUrl types
6124 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
6126 * bindings/python/arg-types.py:
6127 python bindings: add arg type for GstRTSPUrl
6129 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
6131 * bindings/python/rtspserver.defs:
6132 python bindings: fix the definition of MediaFactory.collect_stream
6134 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
6137 Automatic update of common submodule
6138 From 1ccbe09 to c3cafe1
6140 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6143 Automatic update of common submodule
6144 From 193b717 to 1ccbe09
6146 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
6149 Automatic update of common submodule
6150 From b77e2bf to 193b717
6152 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6155 build: Include lcov.mak to allow test coverage report generation
6157 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6160 Automatic update of common submodule
6161 From d8814b6 to b77e2bf
6163 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6166 Automatic update of common submodule
6167 From 6aaa286 to d8814b6
6169 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
6172 Automatic update of common submodule
6173 From 6aec6b9 to 6aaa286
6175 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
6178 autogen: wingo signed comment
6180 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
6182 * gst/rtsp-server/rtsp-session-pool.c:
6183 session: use full charset for RTSP session ID
6184 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
6185 session ID more difficult.
6186 https://bugzilla.gnome.org/show_bug.cgi?id=643812
6188 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6190 * gst/rtsp-server/Makefile.am:
6191 rtsp-server: Don't install the funnel header
6193 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6196 Automatic update of common submodule
6197 From 1de7f6a to 6aec6b9
6199 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6202 configure: require core/base 0.10.31
6203 Needed at least for gst_plugin_feature_rank_compare_func().
6205 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
6208 Automatic update of common submodule
6209 From f94d739 to 1de7f6a
6211 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6213 * gst/rtsp-server/rtsp-media.c:
6214 media: remove more unused code
6216 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6218 * gst/rtsp-server/rtsp-media.c:
6219 * gst/rtsp-server/rtsp-media.h:
6220 media: remove duplicate filtering
6221 Remove the duplicate filtering code now that we have a released -good version.
6222 Give a warning instead.
6224 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6226 * gst/rtsp-server/rtsp-media-factory.c:
6227 * gst/rtsp-server/rtsp-media.c:
6228 media: fix default buffer size
6230 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6232 * gst/rtsp-server/rtsp-media-factory.c:
6233 * gst/rtsp-server/rtsp-media-factory.h:
6234 media-factory: add property to configure the buffer-size
6235 Add a property to configure the kernel UDP buffer size.
6237 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6239 * gst/rtsp-server/rtsp-media.c:
6240 * gst/rtsp-server/rtsp-media.h:
6241 media: add property to configure kernel buffer sizes
6242 Add a property to configure the kernel UDP buffer size.
6244 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6247 configure: set PYGOBJECT_REQ before using it
6248 https://bugzilla.gnome.org/show_bug.cgi?id=640641
6250 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6253 docs: recursive into sub-directories on 'make upload'
6255 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6257 * docs/libs/gst-rtsp-server-docs.sgml:
6258 * docs/version.entities.in:
6259 docs: mention full version these docs are for, not just major-minor
6261 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6266 === release 0.10.8 ===
6268 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6273 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6275 * gst/rtsp-server/rtsp-server.c:
6276 rtsp-server: clarify docs a little
6278 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6280 * gst/rtsp-server/rtsp-media.c:
6281 media: init debug category before starting thread
6283 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6285 * gst/rtsp-server/rtsp-auth.c:
6286 auth: add realm to make it more spec compliant
6288 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6290 * gst/rtsp-server/rtsp-server.c:
6291 * gst/rtsp-server/rtsp-server.h:
6294 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6296 * examples/test-video.c:
6297 example: improve example docs a little
6299 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6301 * gst/rtsp-server/rtsp-server.c:
6302 server: ensure the watch has a ref to the server
6304 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6306 * gst/rtsp-server/rtsp-server.c:
6307 server: simpify channel function
6309 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6311 * gst/rtsp-server/rtsp-server.c:
6312 * gst/rtsp-server/rtsp-server.h:
6313 server: simplify management of channel and source
6314 We don't need to keep around the channel and source objects. Let the mainloop
6315 and the source manage the source and channel respectively.
6317 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6323 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6326 * tests/Makefile.am:
6327 * tests/test-cleanup.c:
6328 tests: add tests directory and cleanup test
6330 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6332 * gst/rtsp-server/rtsp-media-factory-uri.c:
6333 * gst/rtsp-server/rtsp-media-factory.c:
6334 * gst/rtsp-server/rtsp-media-mapping.c:
6335 * gst/rtsp-server/rtsp-media.c:
6336 * gst/rtsp-server/rtsp-session-pool.c:
6337 * gst/rtsp-server/rtsp-session.c:
6338 server: improve debugging in various objects
6340 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6342 * gst/rtsp-server/rtsp-server.c:
6343 server: chain up to the parent finalize
6345 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
6347 * bindings/python/rtspserver-types.defs:
6348 * bindings/python/rtspserver.defs:
6349 * bindings/python/rtspserver.override:
6350 * bindings/python/test.py:
6351 gst-rtsp-server: update python bindings
6353 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6355 * gst/rtsp-server/rtsp-client.c:
6356 client: use the response from the clientstate
6357 Create the response object only once and store in the client state.
6358 Make all methods use the state response,
6360 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6362 * gst/rtsp-server/rtsp-server.c:
6363 server: use signal to keep track of clients
6364 Keep track of all the clients that the server creates and remove them when they
6365 fire the 'closed' signal.
6367 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6369 * gst/rtsp-server/rtsp-client.c:
6370 * gst/rtsp-server/rtsp-client.h:
6371 client: emit signal when closing
6373 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6375 * examples/.gitignore:
6376 * examples/Makefile.am:
6377 * examples/test-auth.c:
6378 * examples/test-video.c:
6379 * gst/rtsp-server/rtsp-auth.c:
6380 * gst/rtsp-server/rtsp-auth.h:
6381 * gst/rtsp-server/rtsp-client.c:
6382 * gst/rtsp-server/rtsp-media-factory.c:
6383 * gst/rtsp-server/rtsp-media.c:
6384 * gst/rtsp-server/rtsp-media.h:
6385 * gst/rtsp-server/rtsp-session-pool.h:
6386 * gst/rtsp-server/rtsp-session.h:
6387 media: enable per factory authorisations
6388 Allow for adding a GstRTSPAuth on the factory and media level and check
6389 permissions when accessing the factory.
6390 Add hints to the auth methods for future more fine grained authorisation.
6391 Add example application for per factory authentication.
6393 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6395 * gst/rtsp-server/rtsp-auth.c:
6396 * gst/rtsp-server/rtsp-auth.h:
6397 * gst/rtsp-server/rtsp-client.c:
6398 * gst/rtsp-server/rtsp-client.h:
6399 * gst/rtsp-server/rtsp-params.c:
6400 * gst/rtsp-server/rtsp-params.h:
6401 rtsp-server: Pass ClientState structure arround
6402 Pass the collected information for the ongoing request in a GstRTSPClientState
6403 structure that we can then pass around to simplify the method arguments. This
6404 will also be handy when we implement logging functionality.
6406 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6408 * gst/rtsp-server/rtsp-media-factory.c:
6409 * gst/rtsp-server/rtsp-media-factory.h:
6410 media-factory: add methods to configure authorisation
6412 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6414 * gst/rtsp-server/rtsp-client.c:
6415 client: unref auth in finalize
6417 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6419 * gst/rtsp-server/rtsp-server.c:
6420 server: unref auth in finalize
6422 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6424 * docs/libs/gst-rtsp-server-docs.sgml:
6425 * docs/libs/gst-rtsp-server-sections.txt:
6426 * docs/libs/gst-rtsp-server.types:
6429 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6431 * gst/rtsp-server/rtsp-server.c:
6432 * gst/rtsp-server/rtsp-server.h:
6433 server: separate create and accept
6434 Create separate create and accept methods so that subclasses can create custom
6436 Configure the server in the client object and prepare for keeping track of
6439 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6441 * gst/rtsp-server/rtsp-client.c:
6442 * gst/rtsp-server/rtsp-client.h:
6443 client: add support for setting the server.
6444 Add support for keeping a ref to the server that started this client
6447 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6449 * gst/rtsp-server/rtsp-auth.c:
6450 auth: fix memleak and add some docs
6451 Fix a memleak of the basic auth token.
6452 Add docs for the helper function
6454 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6456 * gst/rtsp-server/rtsp-auth.c:
6457 * gst/rtsp-server/rtsp-auth.h:
6458 * gst/rtsp-server/rtsp-client.c:
6459 client: delegate setup of auth to the manager
6460 Delegate the configuration of the authentication tokens to the manager object
6463 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6465 * examples/test-video.c:
6466 * gst/rtsp-server/Makefile.am:
6467 * gst/rtsp-server/rtsp-auth.c:
6468 * gst/rtsp-server/rtsp-auth.h:
6469 * gst/rtsp-server/rtsp-client.c:
6470 * gst/rtsp-server/rtsp-client.h:
6471 * gst/rtsp-server/rtsp-server.c:
6472 * gst/rtsp-server/rtsp-server.h:
6473 auth: add authentication object
6474 Add an object that can check the authorization of requests.
6475 Implement basic authentication.
6476 Add example authentication to test-video
6478 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6480 * gst/rtsp-server/rtsp-server.c:
6481 * gst/rtsp-server/rtsp-server.h:
6482 server: move includes back
6483 the includes are needed for sockaddr_in.
6485 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6487 * gst/rtsp-server/rtsp-client.c:
6488 * gst/rtsp-server/rtsp-client.h:
6489 * gst/rtsp-server/rtsp-server.c:
6490 * gst/rtsp-server/rtsp-server.h:
6491 rtsp: move network includes where they are needed
6493 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
6495 * gst/rtsp-server/rtsp-media.h:
6496 rtsp-media.h: Minor corrections in comments.
6499 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
6502 Automatic update of common submodule
6503 From e572c87 to f94d739
6505 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6509 * docs/libs/.gitignore:
6510 * examples/.gitignore:
6511 * gst/rtsp-server/.gitignore:
6514 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6516 * docs/libs/Makefile.am:
6517 docs: We don't build ps/pdf for API reference docs
6519 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6522 Automatic update of common submodule
6523 From ccbaa85 to e572c87
6525 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6528 Automatic update of common submodule
6529 From 46445ad to ccbaa85
6531 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6533 * gst/rtsp-server/Makefile.am:
6534 * gst/rtsp-server/fs-funnel.c:
6535 * gst/rtsp-server/fs-funnel.h:
6536 * gst/rtsp-server/rtsp-funnel.c:
6537 * gst/rtsp-server/rtsp-funnel.h:
6538 * gst/rtsp-server/rtsp-media.c:
6539 funnel: rename fsfunnel to rtspfunnel
6540 Rename the funnel to avoid conflicts with the farsight one.
6542 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6544 * gst/rtsp-server/Makefile.am:
6545 * gst/rtsp-server/fs-funnel.c:
6546 * gst/rtsp-server/fs-funnel.h:
6547 * gst/rtsp-server/rtsp-media.c:
6548 rtsp-media: add and use fsfunnel
6549 Add a copy of fsfunnel to the build because input-selector removed the (broken)
6550 select-all property that we need.
6552 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6554 * gst/rtsp-server/Makefile.am:
6555 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
6556 Use PKG_CONFIG_PATH specified at configure time (if any) as well
6557 for the g-ir-compiler, rather than just assuming the env var has
6560 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6567 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
6569 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6572 * gst/rtsp-server/Makefile.am:
6573 gobject-introspection: fix g-i build for uninstalled setup
6574 Requires gst-plugins-base git (> 0.10.31.2).
6576 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6578 * examples/test-uri.c:
6579 examples: add some more options and comments
6581 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6583 * gst/rtsp-server/rtsp-media-factory-uri.c:
6584 factory-uri: use right property type
6586 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6588 * gst/rtsp-server/rtsp-media-factory-uri.c:
6589 factory-uri: attempt to configure buffer-lists
6590 Attempt to configure buffer lists in the payloader for improved performance.
6592 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6594 * gst/rtsp-server/rtsp-media.c:
6595 media: attempt to configure bigger UDP buffers
6596 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
6597 send buffers with high bitrate streams.
6599 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
6601 * gst/rtsp-server/rtsp-client.c:
6602 client: use the socket length from getsockname
6603 Use the length returned by getsockname to perform the getnameinfo call because
6604 the size can depend on the socket type and platform.
6607 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6609 * docs/libs/gst-rtsp-server-docs.sgml:
6610 * docs/libs/gst-rtsp-server-sections.txt:
6611 docs: add uri factory to the docs
6613 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6615 * gst/rtsp-server/rtsp-client.c:
6616 * gst/rtsp-server/rtsp-media.h:
6619 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6621 * gst/rtsp-server/rtsp-client.c:
6622 * gst/rtsp-server/rtsp-media.c:
6623 * gst/rtsp-server/rtsp-media.h:
6624 * gst/rtsp-server/rtsp-session.c:
6625 * gst/rtsp-server/rtsp-session.h:
6626 rtsp-server: add support for buffer lists
6627 Add support for sending bufferlists received from appsink.
6630 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6632 * gst/rtsp-server/rtsp-client.c:
6633 * gst/rtsp-server/rtsp-media.c:
6634 * gst/rtsp-server/rtsp-media.h:
6635 * gst/rtsp-server/rtsp-sdp.c:
6636 media: make method to retrieve the play range
6637 Make a method to retrieve the playback range so that we can conditionally create
6638 a different range for the SDP and the PLAY requests.
6640 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6642 * gst/rtsp-server/rtsp-media.c:
6643 * gst/rtsp-server/rtsp-media.h:
6644 media: add signal to notify of state changes
6646 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6648 * gst/rtsp-server/rtsp-client.h:
6649 client: cleanup headers
6651 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6653 * gst/rtsp-server/rtsp-client.c:
6656 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6658 * gst/rtsp-server/rtsp-media-factory-uri.c:
6659 * gst/rtsp-server/rtsp-media-factory-uri.h:
6660 factory-uri: add support for gstpay
6661 Add an option to prefer gstpay over decoder + raw payloader.
6663 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6665 * gst/rtsp-server/rtsp-media-factory-uri.c:
6666 * gst/rtsp-server/rtsp-media-factory-uri.h:
6667 factory-uri: rework the autoplugger.
6668 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
6671 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6673 * gst/rtsp-server/rtsp-media-factory-uri.c:
6674 factory-uri: use better factory filter
6675 Make better payloader filter based on autoplug rank and RTP use case.
6677 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6680 Automatic update of common submodule
6681 From 169462a to 46445ad
6683 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6685 * gst/rtsp-server/rtsp-server.c:
6686 server: set SO_REUSEADDR before bind
6687 Set the SO_REUSEADDR _before_ bind() to make it actually work.
6689 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6691 * gst/rtsp-server/rtsp-media.c:
6692 * gst/rtsp-server/rtsp-media.h:
6693 media: emit prepared signal when prepared
6694 Make a 'prepared' signal and emit it when we successfully prepared the element.
6695 This signal can be used to configure the media object after it has been prepared
6698 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
6701 Automatic update of common submodule
6702 From 011bcc8 to 169462a
6704 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
6706 python an optional dependency
6707 * configure.ac: Move up valgrind and g-i checks. Make the python
6708 dependency optional, as it was before.
6710 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6712 Merge branch 'master' into 0.11
6717 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6719 * gst/rtsp-server/rtsp-media.c:
6720 media: update range when active clients changed
6721 When we changed the number of active clients, update the current range
6722 information because we want the second client connecting to a shared resource
6723 continue from where the stream currently.
6725 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6727 * gst/rtsp-server/rtsp-media-factory-uri.c:
6728 * gst/rtsp-server/rtsp-media-factory-uri.h:
6729 factory-uri: add colorspace and fix pt
6730 Rework the way we pass data to the autoplugger.
6731 When we have raw caps, plug a converter element to make pluggin to raw
6732 payloaders more successful.
6733 Make sure all dynamically plugged payloaders have a unique payload types.
6735 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6737 * examples/Makefile.am:
6738 * examples/test-uri.c:
6739 example: add example of the uri factory
6741 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6743 * gst/rtsp-server/Makefile.am:
6744 * gst/rtsp-server/rtsp-media-factory-uri.c:
6745 * gst/rtsp-server/rtsp-media-factory-uri.h:
6746 * gst/rtsp-server/rtsp-server.h:
6747 factory-uri: add a factory to stream any URI
6748 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
6751 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6753 * gst/rtsp-server/rtsp-media.c:
6754 * gst/rtsp-server/rtsp-media.h:
6755 media: ignore spurious ASYNC_DONE messages
6756 When we are dynamically adding pads, the addition of the udpsrc elements will
6757 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
6758 the real ASYNC_DONE when everything is prerolled.
6760 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6762 * gst/rtsp-server/rtsp-media-factory.c:
6763 * gst/rtsp-server/rtsp-media-factory.h:
6764 media-factory: make lock macro
6766 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
6768 * gst/rtsp-server/rtsp-client.c:
6769 rtsp-server: Remove unused variable and dead assignment
6771 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
6773 * examples/test-launch.c:
6774 * examples/test-mp4.c:
6775 * examples/test-ogg.c:
6776 * examples/test-readme.c:
6777 * examples/test-sdp.c:
6778 * examples/test-video.c:
6779 examples: Run gst-indent
6781 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
6783 * gst/rtsp-server/rtsp-client.c:
6784 * gst/rtsp-server/rtsp-media-factory.c:
6785 * gst/rtsp-server/rtsp-media-mapping.c:
6786 * gst/rtsp-server/rtsp-media.c:
6787 * gst/rtsp-server/rtsp-params.c:
6788 * gst/rtsp-server/rtsp-sdp.c:
6789 * gst/rtsp-server/rtsp-server.c:
6790 * gst/rtsp-server/rtsp-session-pool.c:
6791 * gst/rtsp-server/rtsp-session.c:
6792 rtsp-server: Run gst-indent
6793 Since it wasn't using the upstream common previously, there was no
6794 indentation check before commiting.
6796 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
6798 * gst/rtsp-server/rtsp-media-mapping.h:
6799 * gst/rtsp-server/rtsp-media.c:
6800 * gst/rtsp-server/rtsp-media.h:
6801 * gst/rtsp-server/rtsp-sdp.c:
6802 * gst/rtsp-server/rtsp-session-pool.h:
6803 * gst/rtsp-server/rtsp-session.c:
6804 * gst/rtsp-server/rtsp-session.h:
6805 rtsp-server: Some more doc fixups
6807 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6810 Makefile: Add cruft-cleaning support
6812 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6817 * docs/libs/Makefile.am:
6818 * docs/libs/gst-rtsp-server-docs.sgml:
6819 * docs/libs/gst-rtsp-server-sections.txt:
6820 * docs/libs/gst-rtsp-server.types:
6821 * docs/version.entities.in:
6822 docs: Add gtk-doc build system
6824 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6826 * gst/rtsp-server/Makefile.am:
6827 Makefile.am: Use standard GIR make behaviour
6829 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
6833 autogen/configure: Bring more in sync to standard gst module behaviour
6835 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6837 * gst/rtsp-server/rtsp-media.c:
6838 media: warn and fail when gstrtpbin is not found
6840 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6843 configure: open 0.11 branch
6845 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
6849 Add common submodule
6851 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
6854 * common/Makefile.am:
6855 * common/c-to-xml.py:
6857 * common/coverage/coverage-report-entry.pl:
6858 * common/coverage/coverage-report.pl:
6859 * common/coverage/coverage-report.xsl:
6860 * common/coverage/lcov.mak:
6861 * common/gettext.patch:
6862 * common/glib-gen.mak:
6863 * common/gst-autogen.sh:
6864 * common/gst-xmlinspect.py:
6866 * common/gstdoc-scangobj:
6867 * common/gtk-doc-plugins.mak:
6868 * common/gtk-doc.mak:
6869 * common/m4/.gitignore:
6870 * common/m4/Makefile.am:
6872 * common/m4/as-ac-expand.m4:
6873 * common/m4/as-auto-alt.m4:
6874 * common/m4/as-compiler-flag.m4:
6875 * common/m4/as-compiler.m4:
6876 * common/m4/as-docbook.m4:
6877 * common/m4/as-libtool-tags.m4:
6878 * common/m4/as-libtool.m4:
6879 * common/m4/as-python.m4:
6880 * common/m4/as-scrub-include.m4:
6881 * common/m4/as-version.m4:
6882 * common/m4/ax_create_stdint_h.m4:
6883 * common/m4/check.m4:
6884 * common/m4/glib-gettext.m4:
6885 * common/m4/gst-arch.m4:
6886 * common/m4/gst-args.m4:
6887 * common/m4/gst-check.m4:
6888 * common/m4/gst-debuginfo.m4:
6889 * common/m4/gst-default.m4:
6890 * common/m4/gst-doc.m4:
6891 * common/m4/gst-error.m4:
6892 * common/m4/gst-feature.m4:
6893 * common/m4/gst-function.m4:
6894 * common/m4/gst-gettext.m4:
6895 * common/m4/gst-glib2.m4:
6896 * common/m4/gst-libxml2.m4:
6897 * common/m4/gst-plugindir.m4:
6898 * common/m4/gst-valgrind.m4:
6899 * common/m4/gtk-doc.m4:
6900 * common/m4/introspection.m4:
6902 * common/mangle-tmpl.py:
6903 * common/plugins.xsl:
6905 * common/release.mak:
6906 * common/scangobj-merge.py:
6907 * common/upload.mak:
6908 common: Remove static version
6910 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
6912 * common/m4/introspection.m4:
6913 Update introspection.m4 to match usage
6915 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6919 Remove old stuff from the README
6921 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6926 === release 0.10.7 ===
6928 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6933 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6935 * examples/test-ogg.c:
6936 test-ogg: remove parsers
6937 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
6938 buffers with timestamps. Using the parsers also seems to break things.
6940 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6942 * bindings/vala/gst-rtsp-server-0.10.vapi:
6943 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6944 Updated Vala bindings
6946 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6948 * common/m4/introspection.m4:
6950 * gst/rtsp-server/Makefile.am:
6951 Added initial gobject-introspection support
6953 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6955 * gst/rtsp-server/rtsp-media-factory.c:
6956 media-factory: don't use host for shared hash key
6957 When we generate the key to share made between connections, don't include the
6958 host used to connect so that we can share media even if between clients that
6959 connected with localhost and ones with the ip address.
6961 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6963 * bindings/vala/Makefile.am:
6964 build: fix distcheck
6966 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6968 * bindings/vala/gst-rtsp-server-0.10.vapi:
6969 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6970 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6971 Update Vala bindings
6973 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6975 * bindings/vala/Makefile.am:
6977 Fix configure checks and installation location for Vala bindings
6980 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6985 === release 0.10.6 ===
6987 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6990 configure: release 0.10.6
6992 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6994 * gst/rtsp-server/rtsp-media.c:
6995 media: help the compiler a little
6997 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6999 * gst/rtsp-server/rtsp-media.c:
7000 * gst/rtsp-server/rtsp-media.h:
7001 * gst/rtsp-server/rtsp-session.c:
7002 media: cleanup media transport before freeing
7003 Cleanup the media transport data before freeing. In particular, remove the qdata
7004 from the rtpsource object.
7006 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7008 * gst/rtsp-server/rtsp-media-factory.c:
7009 * gst/rtsp-server/rtsp-media-factory.h:
7010 * gst/rtsp-server/rtsp-media.c:
7011 * gst/rtsp-server/rtsp-media.h:
7012 media-factory: add eos-shutdown property
7013 Add an eos-shutdown property that will send an EOS to the pipeline before
7014 shutting it down. This allows for nice cleanup in case of a muxer.
7017 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7019 * gst/rtsp-server/rtsp-media.c:
7020 * gst/rtsp-server/rtsp-media.h:
7021 media: use multiudpsink send-duplicates when we can
7022 If we have a new enough multiudpsink with the send-duplicates property, use this
7023 instead of doing our own filtering. Our custom filtering code should eventually
7024 be removed when we can depend on a released -good.
7026 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7028 * gst/rtsp-server/rtsp-media.c:
7029 media: don't leak destinations
7030 Refactor and cleanup the destinations array when the stream is destroyed.
7032 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7034 * gst/rtsp-server/rtsp-media.c:
7035 * gst/rtsp-server/rtsp-media.h:
7036 media: don't add udp addresses multiple times
7037 Keep track of the udp addresses we added to udpsink and never add the same udp
7038 destination twice. This avoids duplicate packets when using multicast.
7040 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7042 * gst/rtsp-server/rtsp-server.c:
7043 server: disable use of SO_LINGER
7044 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
7045 server close()s the connection.
7047 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7049 * gst/rtsp-server/rtsp-server.c:
7050 server: use 5 second linger period in SO_LINGER
7051 Wait 5 seconds before clearing the send buffers and reseting the connection with
7052 the client when we do a close. This should be enough time to get the message to
7056 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
7058 * gst/rtsp-server/rtsp-server.c:
7059 server: use SO_LINGER
7060 SO_LINGER on the socket will make sure that any pending data on the socket is
7061 flushed ASAP and that the socket connection is reset. This makes sure that the
7062 socket can be reused immediately.
7065 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7068 README: add blurb about shared media factories
7070 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
7072 * gst/rtsp-server/rtsp-media.c:
7073 Add stdlib.h for atoi()
7075 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7077 * bindings/python/Makefile.am:
7078 * bindings/vala/Makefile.am:
7079 build: distcheck fixes
7080 Fix 'make distcheck', somewhat (it still fails because it tries to
7081 install files into /usr/share/vala/vapi/ irrespective of the
7084 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7087 configure: bump core/base requirements to released version
7088 Makes things less confusing for people.
7090 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7093 configure: fail if GStreamer core/base requirements are not met
7095 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7097 * gst/rtsp-server/rtsp-client.c:
7098 client: improve client cleanups
7099 Make sure the session does not timeout when using TCP. We need to do this
7100 because quicktime player does not send RTCP for some reason in tunneled
7102 Refactor some cleanup code.
7105 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7107 * gst/rtsp-server/rtsp-session.c:
7108 * gst/rtsp-server/rtsp-session.h:
7109 session: add support for prevent session timeouts
7110 Add an atomix counter to prevent session timeouts when we are, for example,
7113 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7115 * gst/rtsp-server/rtsp-client.c:
7116 client: fix unlink on session timeouts
7117 When our session times out, make sure we unlink all streams in this
7119 Remove the tunnelid when closing the connection.
7121 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7123 * gst/rtsp-server/rtsp-session.c:
7124 session: small cleanups
7126 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7128 * gst/rtsp-server/rtsp-client.c:
7129 client: handle lost_tunnel callbacks
7130 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
7131 hashtable so that we can reuse it for when the client reopens the POST
7133 Close the connection after a TEARDOWN.
7134 Make sure or watchid is cleared when the watch is removed.
7137 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7139 * gst/rtsp-server/rtsp-client.c:
7140 * gst/rtsp-server/rtsp-media.c:
7141 * gst/rtsp-server/rtsp-sdp.c:
7142 rtsp-server: add more support for multicast
7144 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7147 * gst/rtsp-server/rtsp-media.c:
7148 * gst/rtsp-server/rtsp-media.h:
7149 media: allow configuration of allowed lower transport
7151 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7153 * gst/rtsp-server/rtsp-client.h:
7154 * gst/rtsp-server/rtsp-media.c:
7155 * gst/rtsp-server/rtsp-media.h:
7156 * gst/rtsp-server/rtsp-sdp.c:
7157 * gst/rtsp-server/rtsp-sdp.h:
7158 * gst/rtsp-server/rtsp-server.c:
7159 rtsp: keep track of server ip and ipv6
7160 Keep track of how the client connected to the server and setup the udp ports
7161 with the same protocol.
7162 Copy the server ip address in the SDP so that clients can send RTCP back to
7165 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7167 * gst/rtsp-server/rtsp-session.c:
7170 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7172 * gst/rtsp-server/rtsp-client.c:
7173 client: use right size for malloc
7175 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7177 * gst/rtsp-server/rtsp-server.c:
7178 server: comment ipv6 server listening address
7180 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7182 * gst/rtsp-server/rtsp-media.c:
7183 media: allow for ipv6 sockets
7185 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7187 * gst/rtsp-server/rtsp-server.c:
7188 * gst/rtsp-server/rtsp-server.h:
7189 server: rework server part
7190 Allow setting a bind address, make sure we can deal with ipv6.
7191 Remove the port property and change with the service property.
7193 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7195 * gst/rtsp-server/rtsp-media.h:
7196 media: update comments a little
7198 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7200 * gst/rtsp-server/rtsp-client.c:
7201 client: make content-base better
7202 Use the URI formatting functions to make a content-base. Also make sure that
7203 there is a trailing / at the end.
7205 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7207 * gst/rtsp-server/rtsp-client.c:
7208 client: guard against invalid paths
7210 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7212 * examples/test-video.c:
7213 test: catch server bind errors
7215 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
7217 * gst/rtsp-server/rtsp-media.c:
7218 rtspmedia: emit "unprepared" if _prepare fails.
7219 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
7220 media object is removed from its factory's cache.
7222 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7224 * gst/rtsp-server/rtsp-media.c:
7225 media: collect media position when seek completes
7227 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
7229 * gst/rtsp-server/rtsp-client.c:
7230 client: call unlink_streams in client finalize
7233 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7235 * gst/rtsp-server/rtsp-media.c:
7236 media: limit the time to wait to something huge
7237 Avoid waiting forever but limit the timeout to 20 seconds.
7239 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7241 * gst/rtsp-server/rtsp-sdp.c:
7242 sdp: reindent and check for prepared status
7244 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7246 * gst/rtsp-server/rtsp-media.c:
7247 * gst/rtsp-server/rtsp-media.h:
7248 * gst/rtsp-server/rtsp-session.c:
7249 media: avoid doing _get_state() for state changes
7250 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
7251 until the media is prerolled or in error. This avoids doing a blocking call of
7252 gst_element_get_state() that can cause lockups when there is an error.
7255 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7257 * gst/rtsp-server/rtsp-media.c:
7260 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7262 * gst/rtsp-server/rtsp-media-factory.c:
7263 media-factory: better error handling
7264 Improve the error handling a bit.
7266 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7268 * gst/rtsp-server/rtsp-client.c:
7269 client: rework transport parsing
7270 Rework the transport parsing code so that we can ignore transports we don't
7271 support instead of just picking the first one we can parse.
7272 Configure a (for now hardcoded) destination for multicast transports.
7274 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7276 * gst/rtsp-server/rtsp-media.c:
7277 media: set multicast sink parameters
7278 Disable loop and automatic multicast join on the udpsink elements.
7279 Add some more debug info.
7280 Reset some state variables in the right place.
7281 Use the right port numbers for multicast.
7283 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7285 * gst/rtsp-server/rtsp-session.c:
7286 session: handle transport setup correctly
7287 Handle UDP, MCAST and TCP transport negotiation more correctly.
7288 Store the server session SSRC in the transport.
7290 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7292 * gst/rtsp-server/rtsp-client.c:
7293 rtsp-client: implement error_full
7294 Implement error_full to avoid some segfaults when the rtspconnection calls it.
7297 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7300 * gst/rtsp-server/rtsp-client.c:
7301 * gst/rtsp-server/rtsp-server.c:
7302 docs: update docs and comments
7304 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
7306 * gst/rtsp-server/rtsp-sdp.c:
7307 sdp: make server work better when behind a proxy
7309 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7311 * gst/rtsp-server/rtsp-client.c:
7312 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
7314 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7316 * gst/rtsp-server/rtsp-client.c:
7317 * gst/rtsp-server/rtsp-media-factory.c:
7318 * gst/rtsp-server/rtsp-media-mapping.c:
7319 * gst/rtsp-server/rtsp-media.c:
7320 * gst/rtsp-server/rtsp-server.c:
7321 * gst/rtsp-server/rtsp-session-pool.c:
7322 * gst/rtsp-server/rtsp-session.c:
7323 Use GStreamer's debugging subsystem
7325 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7327 * gst/rtsp-server/rtsp-media-factory.c:
7328 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
7330 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7335 === release 0.10.5 ===
7337 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7342 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7345 configure: bump required versions
7347 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
7349 * gst/rtsp-server/rtsp-client.c:
7350 client: call weak-unref on client->sessions from finalize
7353 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7355 * gst/rtsp-server/rtsp-media.c:
7356 media: Fixed crasher where caps got unref'ed too often
7358 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7361 * pkgconfig/.gitignore:
7362 * pkgconfig/Makefile.am:
7363 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
7364 Added pkg-config file to use gst-rtsp-server uninstalled
7366 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7368 * gst/rtsp-server/rtsp-media.c:
7369 media: add some docs
7371 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
7373 * gst/rtsp-server/rtsp-client.c:
7374 rtsp: Use gst_rtsp_watch_send_message().
7375 Use gst_rtsp_watch_send_message() since the old API which used
7376 gst_rtsp_watch_queue_message() has been deprecated.
7378 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7383 === release 0.10.4 ===
7385 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7390 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7392 * gst/rtsp-server/rtsp-client.c:
7393 * gst/rtsp-server/rtsp-session.c:
7394 * gst/rtsp-server/rtsp-session.h:
7395 rtsp: allocate channels in TCP mode
7396 When the client does not provide us with channels in TCP mode, allocate channels
7399 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7401 * gst/rtsp-server/rtsp-client.c:
7402 client: don't crash when tunnelid is missing
7403 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
7404 don't crash but return an error response to the client.
7407 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7409 * bindings/vala/gst-rtsp-server-0.10.vapi:
7410 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7411 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7412 bindings: update vala bindings with new method
7414 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7416 * gst/rtsp-server/rtsp-session-pool.c:
7417 * gst/rtsp-server/rtsp-session-pool.h:
7418 sessionpool: add function to filter sessions
7419 Add generic function to retrieve/remove sessions.
7421 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7424 configure: bump core/base requirements to release
7426 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7428 * gst/rtsp-server/rtsp-media.c:
7429 media: fix indentation
7431 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7433 * gst/rtsp-server/rtsp-media.c:
7434 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
7436 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7438 * gst/rtsp-server/rtsp-media.c:
7439 set state and remove elements of media in for loop
7441 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
7443 * bindings/vala/gst-rtsp-server-0.10.vapi:
7444 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7445 Added gst_rtsp_media_remove_elements function to Vala bindings
7447 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
7449 * gst/rtsp-server/rtsp-media.c:
7450 * gst/rtsp-server/rtsp-media.h:
7451 Added gst_rtsp_media_remove_elements function
7453 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
7455 * gst/rtsp-server/rtsp-media.c:
7456 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
7458 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7460 * bindings/vala/gst-rtsp-server-0.10.vapi:
7461 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7462 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7463 Updated Vala bindings
7465 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7467 * gst/rtsp-server/rtsp-media.c:
7468 * gst/rtsp-server/rtsp-media.h:
7469 Added vmethod unprepare to GstRTSPMedia
7470 The default implementation sets the state of the pipeline to GST_STATE_NULL
7472 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7474 * gst/rtsp-server/rtsp-media-factory.c:
7475 * gst/rtsp-server/rtsp-media-factory.h:
7476 Made collect_streams function public
7478 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7480 * gst/rtsp-server/rtsp-media-factory.c:
7481 * gst/rtsp-server/rtsp-media-factory.h:
7482 * gst/rtsp-server/rtsp-media.c:
7483 Added vmethod create_pipeline to GstRTSPMediaFactory
7484 The pipeline is created in this method and the GstRTSPMedia's element is added to it
7486 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7488 * gst/rtsp-server/rtsp-client.c:
7489 client: use g_source_destroy()
7490 We need to use g_source_destroy() because we might have added the source to a
7491 different main context than the default one.
7493 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7495 * gst/rtsp-server/Makefile.am:
7496 * gst/rtsp-server/rtsp-client.c:
7497 * gst/rtsp-server/rtsp-params.c:
7498 * gst/rtsp-server/rtsp-params.h:
7499 rtsp: prepare for handling GET/SET_PARAMETER
7500 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
7502 Fix return codes of handlers.
7504 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7506 * gst/rtsp-server/rtsp-media.c:
7507 media: don't leak session pads
7509 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7511 * gst/rtsp-server/rtsp-media.c:
7512 media: clean up the messages a bit
7514 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7516 * gst/rtsp-server/rtsp-sdp.c:
7517 sdp: warn and skip streams without media
7519 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7521 * bindings/vala/gst-rtsp-server-0.10.vapi:
7522 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7523 vala: Fixed typo in header file of RTSPMediaStream
7525 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7527 * gst/rtsp-server/rtsp-media.c:
7530 Make dumping RTCP stats configurable
7532 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7534 * gst/rtsp-server/rtsp-media.c:
7535 media: be less verbose and leak less
7537 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7539 * gst/rtsp-server/rtsp-media.c:
7540 media: don't leak the destination address
7542 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7544 * gst/rtsp-server/rtsp-client.c:
7545 * gst/rtsp-server/rtsp-media.c:
7546 * gst/rtsp-server/rtsp-media.h:
7547 * gst/rtsp-server/rtsp-session.c:
7548 * gst/rtsp-server/rtsp-session.h:
7549 rtsp: use RTCP to keep the session alive
7550 Use the RTCP rtcp-from stats field to find the associated session and use this
7551 to keep the session alive.
7553 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7555 * gst/rtsp-server/rtsp-session.c:
7556 session: add 5sec to the real session timeout
7557 Allow the session to live 5sec longer before really timing out. This should give
7558 clients some extra time to keep the session active.
7560 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7562 * gst/rtsp-server/rtsp-client.c:
7563 client: replay OK to GET/SET_PARAMETER
7564 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
7565 so that we return OK for those requests.
7567 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7569 * gst/rtsp-server/rtsp-media.c:
7570 * gst/rtsp-server/rtsp-media.h:
7571 media: keep track of active transports
7572 Keep track of which transport is active to avoid closing the connection too
7574 Remove the destination transport also when going to NULL.
7575 Print some stats about the SDES and other RTCP messages we receive from the
7578 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7580 * examples/.gitignore:
7581 * examples/Makefile.am:
7582 * examples/test-sdp.c:
7583 example: add SDP relay example
7585 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7587 * gst/rtsp-server/rtsp-media.c:
7588 media: also count active TCP connections
7590 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7592 * gst/rtsp-server/rtsp-media-factory.c:
7593 * gst/rtsp-server/rtsp-media.c:
7594 * gst/rtsp-server/rtsp-media.h:
7595 rtsp: add support for dynamic elements
7596 Add support for dynamic elements.
7597 Don't set live pipelines back to paused.
7599 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7601 * gst/rtsp-server/rtsp-sdp.c:
7602 sdp: don't add encoding name when absent in caps
7604 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7606 * gst/rtsp-server/rtsp-client.c:
7607 client: warn when we can't do RTP-Info
7609 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7611 * gst/rtsp-server/rtsp-media-factory.c:
7612 factory: factor out the stream construction
7614 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7616 * gst/rtsp-server/rtsp-client.c:
7617 client: only add RTP-Info when we have the info
7618 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
7621 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7626 === release 0.10.3 ===
7628 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7632 - Fixes a bug where it put the wrong verion in pkgconfig
7633 - Link RTP and RTCP sources
7635 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7637 * gst/rtsp-server/rtsp-media.c:
7638 * gst/rtsp-server/rtsp-media.h:
7639 media: link the RTP udpsrc to the session manager
7640 Link the RTP udpsrc and the appsrc to the session manager so that they don't
7641 shut down when the client sends a packet to open firewalls.
7643 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7645 * pkgconfig/gst-rtsp-server.pc.in:
7646 Don't use hard-coded version number in pkg-config file
7648 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7653 === release 0.10.2 ===
7655 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7660 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7663 * common/m4/.gitignore:
7664 * examples/.gitignore:
7665 * pkgconfig/.gitignore:
7666 add some .gitignore files
7668 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7670 * gst/rtsp-server/rtsp-media.c:
7671 media: seek to key frames
7673 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7675 * gst/rtsp-server/rtsp-media.c:
7676 media: emit the unprepared signal by id
7677 Emit the unprepared signal by id instead of name and set the media as
7680 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7682 * gst/rtsp-server/rtsp-media.c:
7683 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
7685 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7687 * gst/rtsp-server/rtsp-server.c:
7688 Added finalize function to GstRTPSPServer to unref session pool and media mapping
7690 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
7692 * bindings/vala/gst-rtsp-server-0.10.vapi:
7693 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7694 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7695 Updated vala bindings
7697 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7699 * gst/rtsp-server/Makefile.am:
7700 * gst/rtsp-server/rtsp-client.c:
7701 * gst/rtsp-server/rtsp-media.c:
7702 server: use appsink and appsrc with the API
7703 Use the appsink/appsrc API instead of the signals for higher
7706 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7708 * examples/test-ogg.c:
7709 tests: set the payload type correctly
7711 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7713 * gst/rtsp-server/rtsp-media-factory.c:
7714 factory: connect to the unprepare signal
7715 Connect to the unprepare signal for non-reusable media so that we can remove
7716 them from the cache.
7718 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7720 * gst/rtsp-server/rtsp-media.c:
7721 * gst/rtsp-server/rtsp-media.h:
7722 media: add signal to notify of unprepare
7724 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7726 * gst/rtsp-server/rtsp-media.c:
7727 * gst/rtsp-server/rtsp-media.h:
7728 media: more work on making the media shared
7729 Add a reusable flag to medias, indicating that they can be reused after a state
7733 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7735 * examples/test-readme.c:
7736 examples: mark the example as shared for testing
7738 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7740 * gst/rtsp-server/rtsp-media.c:
7741 * gst/rtsp-server/rtsp-media.h:
7742 client: support shared media
7743 Always perform the state actions even if the target state of the pipeline is
7744 already correct, we still want to add/remove the transports when we are dealing
7746 Keep a counter of the number of active transports for a media so that we can use
7747 this to perform a state change when needed.
7748 Perform a state change of the pipeline only when the first transport was added
7749 or when there are no active transports.
7751 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
7753 * gst/rtsp-server/rtsp-client.c:
7754 client: fix refcounting crasher
7755 Don't need to remove the weak refs in the finalize methods, they are already
7756 removed in the dispose.
7757 Don't register the callback with a DestroyNofity.
7759 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7761 * gst/rtsp-server/rtsp-client.c:
7762 Fix rtsp client refcount management in TCP mode.
7763 Don't unref a client ref we never had. Fixes an unref
7764 of an already-free client object after a client
7765 teardown request for me.
7767 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
7769 * gst/rtsp-server/rtsp-session.c:
7770 docs: fix typo in API docs
7772 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7774 * gst/rtsp-server/rtsp-media.c:
7776 Keep the udp sources in playing even if we go to paused. unlock the sources when
7778 Add some more debug info.
7779 Only seek when we need to.
7780 Keep track of the position when we go to paused.
7782 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7784 * gst/rtsp-server/rtsp-client.c:
7785 * gst/rtsp-server/rtsp-media.c:
7786 * gst/rtsp-server/rtsp-media.h:
7787 Add beginnings of seeking.
7788 Parse the Range header and perform a seek on the pipeline for the requested
7789 position. It's disabled currently until I figure out what's going wrong.
7791 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7793 * gst/rtsp-server/rtsp-client.c:
7794 allow pause requests for now.
7797 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7799 * gst/rtsp-server/rtsp-client.c:
7800 Remove weak ref on the session in teardown
7801 We need to remove our weakref from the session when we do a teardown because
7802 else we close the TCP connection prematurely.
7804 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7806 * gst/rtsp-server/rtsp-client.c:
7807 * gst/rtsp-server/rtsp-client.h:
7808 * gst/rtsp-server/rtsp-session-pool.c:
7809 Do some more session cleanup
7810 Make session timeout kill the TCP connection that currently watches the
7812 Remove the client timeout property.
7814 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7816 * gst/rtsp-server/rtsp-client.c:
7817 * gst/rtsp-server/rtsp-client.h:
7818 * gst/rtsp-server/rtsp-media.c:
7819 * gst/rtsp-server/rtsp-media.h:
7820 * gst/rtsp-server/rtsp-server.c:
7821 * gst/rtsp-server/rtsp-session.c:
7822 * gst/rtsp-server/rtsp-session.h:
7824 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
7827 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7829 * examples/Makefile.am:
7830 * examples/test-launch.c:
7831 Add example server that takes launch lines
7832 Add an example server that streams any -launch line.
7834 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7836 * examples/test-readme.c:
7837 * gst/rtsp-server/rtsp-client.c:
7838 * gst/rtsp-server/rtsp-media.c:
7839 * gst/rtsp-server/rtsp-media.h:
7840 Add support for live streams
7841 Add support for live streams and ranges
7842 Start on handling TCP data transfer.
7844 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7846 * gst/rtsp-server/rtsp-media.c:
7847 Free the pipeline before other things
7850 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7852 * gst/rtsp-server/rtsp-client.c:
7853 Only free the pending tunnel if there is one
7856 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7858 * gst/rtsp-server/rtsp-client.c:
7859 * gst/rtsp-server/rtsp-client.h:
7860 * gst/rtsp-server/rtsp-media.c:
7861 rtsp-server: Add support for tunneling
7862 Add support for tunneling over HTTP.
7863 Use new connection methods to retrieve the url.
7864 Dispatch messages based on the message type instead of blindly
7865 assuming it's always a request.
7866 Keep track of the watch id so that we can remove it later.
7867 Set the media pipeline to NULL before unreffing the pipeline.
7869 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7871 * gst/rtsp-server/rtsp-client.c:
7872 * gst/rtsp-server/rtsp-client.h:
7873 Fix for channel -> watch rename in gstreamer
7874 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
7876 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7878 * gst/rtsp-server/rtsp-client.c:
7879 * gst/rtsp-server/rtsp-client.h:
7881 Use the async RTSP channels instead of spawning a new thread for each client.
7882 If a sessionid is specified in a request, fail if we don't have the session.
7884 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7886 * gst/rtsp-server/rtsp-media.c:
7887 Add better debug info
7888 Add some better debug info.
7890 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7892 * examples/test-video.c:
7894 Add support for session timeouts in the example.
7896 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7898 * gst/rtsp-server/rtsp-session-pool.c:
7899 * gst/rtsp-server/rtsp-session-pool.h:
7900 Pass GTimeVal around for performance reasons
7901 Get the current time only once and pass it around so that sessions don't have to
7902 get the current time anymore.
7903 Add experimental support for a GSource that dispatches when the session needs to
7906 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7908 * gst/rtsp-server/rtsp-session.c:
7909 * gst/rtsp-server/rtsp-session.h:
7910 Add better support for session timeouts
7911 Add a method to request the number of milliseconds when a session will timeout.
7913 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7915 * gst/rtsp-server/rtsp-media.c:
7916 * gst/rtsp-server/rtsp-media.h:
7917 Add suport for RTP manager monitoring
7918 Add the first stage in monitoring the rtp manager.
7919 Make sure we don't update the state to something we don't want.
7921 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7923 * gst/rtsp-server/rtsp-client.c:
7924 Add support for session keepalive
7925 Get and update the session timeout for all requests. get the session as early as
7928 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7930 * gst/rtsp-server/rtsp-media-factory.h:
7931 * gst/rtsp-server/rtsp-media.c:
7932 * gst/rtsp-server/rtsp-media.h:
7933 Handle media bus messages
7934 Handle media bus messages in a custom mainloop and dispatch them to the
7935 RTSPMedia objects. Let the default implementation handle some common messages.
7937 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7939 * gst/rtsp-server/rtsp-client.c:
7940 * gst/rtsp-server/rtsp-session-pool.c:
7941 * gst/rtsp-server/rtsp-session.c:
7942 Some more session timeout handling
7943 Move the session header setting code to a central place so that we always add
7944 the timeout parameter too.
7945 Handle timeouts by running the session cleanup code.
7946 Stop media before cleaning up.
7948 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7950 * gst/rtsp-server/rtsp-client.c:
7951 * gst/rtsp-server/rtsp-client.h:
7952 Add timeout property
7953 Add a timeout property ot the client and make the other properties into GObject
7956 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7958 * gst/rtsp-server/rtsp-session-pool.c:
7959 Use getters and setters in property code
7960 Use the getters and setters for the timeout property instead of locking
7963 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7965 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
7967 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7969 * gst/rtsp-server/rtsp-session-pool.c:
7970 * gst/rtsp-server/rtsp-session-pool.h:
7971 * gst/rtsp-server/rtsp-session.c:
7972 * gst/rtsp-server/rtsp-session.h:
7973 Add more timeout stuff
7974 Add method to check if a session is expired.
7975 Add method to perform cleanup on a session pool.
7977 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7979 * gst/rtsp-server/rtsp-client.c:
7980 * gst/rtsp-server/rtsp-session-pool.c:
7981 * gst/rtsp-server/rtsp-session-pool.h:
7982 * gst/rtsp-server/rtsp-session.c:
7983 * gst/rtsp-server/rtsp-session.h:
7984 Add beginnings of session timeouts and limits
7985 Add the timeout value to the Session header for unusual timeout values.
7986 Allow us to configure a limit to the amount of active sessions in a pool. Set a
7987 limit on the amount of retry we do after a sessionid collision.
7988 Add properties to the sessionid and the timeout of a session. Keep track of
7989 creation time and last access time for sessions.
7991 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7993 * gst/rtsp-server/rtsp-client.c:
7994 * gst/rtsp-server/rtsp-media.c:
7995 * gst/rtsp-server/rtsp-media.h:
7996 * gst/rtsp-server/rtsp-sdp.c:
7997 * gst/rtsp-server/rtsp-session-pool.c:
7998 * gst/rtsp-server/rtsp-session.c:
7999 * gst/rtsp-server/rtsp-session.h:
8000 Cleanup of sessions and more
8001 Fix the refcounting of media and sessions in the client. Properly clean up the
8002 session data when the client performs a teardown.
8003 Add Server header to responses.
8004 Allow for multiple uri setups in one session.
8005 Add Range header to the PLAY response and add the range attribute to the SDP
8007 Fix the session pool remove method, it used the wrong key in the hashtable. Also
8008 give the ownership of the sessionid to the session object.
8010 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8012 * gst/rtsp-server/rtsp-server.c:
8013 * gst/rtsp-server/rtsp-server.h:
8015 Rename the 'server_port' variable to simply 'port'.
8017 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8020 * gst/rtsp-server/rtsp-client.c:
8021 * gst/rtsp-server/rtsp-media.c:
8022 * gst/rtsp-server/rtsp-media.h:
8023 * gst/rtsp-server/rtsp-session.c:
8024 * gst/rtsp-server/rtsp-session.h:
8025 Rework the way we handle transports for streams
8026 Make the media accept an array of transports for the streams that we have
8027 configured for the play/pause requests.
8028 Implement server states for a client and its media.
8029 Require 0.10.22.1 (git HEAD) of gstreamer.
8031 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8033 * gst/rtsp-server/rtsp-client.c:
8034 * gst/rtsp-server/rtsp-media-factory.c:
8035 Drop const from functions dealing with urls
8036 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
8037 have the right const in them.
8039 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8041 * gst/rtsp-server/rtsp-client.c:
8042 * gst/rtsp-server/rtsp-media.c:
8043 * gst/rtsp-server/rtsp-sdp.c:
8047 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8049 * gst/rtsp-server/rtsp-client.c:
8050 * gst/rtsp-server/rtsp-media-factory.c:
8051 * gst/rtsp-server/rtsp-media.c:
8052 * gst/rtsp-server/rtsp-media.h:
8054 Don't keep a reference to the GstRTSPMedia in the stream.
8055 Free more things when freeing the GstRTSPMedia.
8057 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8060 * gst/rtsp-server/rtsp-media-factory.c:
8061 * gst/rtsp-server/rtsp-media-factory.h:
8062 * gst/rtsp-server/rtsp-media.c:
8063 * gst/rtsp-server/rtsp-media.h:
8064 * gst/rtsp-server/rtsp-server.c:
8065 * gst/rtsp-server/rtsp-server.h:
8066 More docs and small cleanups
8067 Add some more docs and update the README
8068 Cleanup some method names.
8069 Remove an unneeded idx field in the GstRTSPMediaStream
8071 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8074 * examples/Makefile.am:
8075 * examples/test-readme.c:
8076 Add a README and more example code
8077 Add a README file that contains a small introduction on how to use the server
8078 along with the example code explained in the readme.
8080 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8082 * gst/rtsp-server/rtsp-media.c:
8083 * gst/rtsp-server/rtsp-server.c:
8084 Fix some leaks and change default port
8085 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
8086 we finished the initial preroll. If we keep them locked, setting the pipeline to
8087 NULL will not stop and clean up the sources correctly.
8088 Change the default RTSP port to 8554 aka the official alternative RTSP port.
8090 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8092 * gst/rtsp-server/rtsp-session.c:
8093 * gst/rtsp-server/rtsp-session.h:
8094 Cleanups to the session object
8095 Remove some unneeded variables in the session state of a stream such as the
8096 owner media and the server transport.
8097 Get the configuration of a media stream in a session based on the media_stream
8098 in the original object instead of our cached index.
8099 Free more data in the finalize method.
8101 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8103 * gst/rtsp-server/rtsp-client.c:
8104 * gst/rtsp-server/rtsp-client.h:
8105 Cleanups and reuse media from DESCRIBE
8106 Handle thread create errors.
8107 Rename some internal methods to better match what they actually do.
8108 Handle misconfiguration of session_pool and media_mapping gracefully.
8109 Cache the DESCRIBE media and uri in the client connection and reuse them when
8110 we receive a SETUP request in the same connection for the same uri.
8111 Cleanup the client connection object.
8113 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8115 * gst/rtsp-server/rtsp-media-factory.c:
8116 * gst/rtsp-server/rtsp-media-factory.h:
8117 * gst/rtsp-server/rtsp-media.c:
8118 * gst/rtsp-server/rtsp-media.h:
8119 Add shared properties to media and factory
8120 Add the shared property to media.
8121 Implement some simple caching in the factory depending on if the media is shared
8124 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8126 * gst/rtsp-server/rtsp-client.c:
8127 Add a little comment
8128 Add some comment about the content-base header.
8130 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8132 * examples/Makefile.am:
8134 * examples/test-mp4.c:
8135 * examples/test-ogg.c:
8136 * examples/test-video.c:
8137 * gst/rtsp-server/Makefile.am:
8138 * gst/rtsp-server/rtsp-client.c:
8139 * gst/rtsp-server/rtsp-client.h:
8140 * gst/rtsp-server/rtsp-media-factory.c:
8141 * gst/rtsp-server/rtsp-media-factory.h:
8142 * gst/rtsp-server/rtsp-media.c:
8143 * gst/rtsp-server/rtsp-media.h:
8144 * gst/rtsp-server/rtsp-sdp.c:
8145 * gst/rtsp-server/rtsp-sdp.h:
8146 * gst/rtsp-server/rtsp-server.c:
8147 * gst/rtsp-server/rtsp-server.h:
8148 * gst/rtsp-server/rtsp-session.c:
8149 * gst/rtsp-server/rtsp-session.h:
8150 Reorganize things, prepare for media sharing
8151 Added various other test server examples
8152 Move the SDP message generation to a separate helper.
8153 Refactor common code for finding the session.
8154 Add content-base for realplayer compatibility
8155 Clean up request uris before processing for better vlc compatibility.
8156 Move prerolling and pipeline construction to the RTSPMedia object.
8157 Use multiudpsink for future pipeline reuse.
8159 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8165 === release 0.10.1 ===
8167 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8173 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8175 * bindings/vala/Makefile.am:
8177 Add more directories and files to the dist.
8179 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8181 * bindings/python/Makefile.am:
8182 * bindings/python/rtspserver.override:
8183 Fixed compile error of python bindings
8185 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8187 * bindings/vala/gst-rtsp-server-0.10.vapi:
8188 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8189 Marked values as nullable accordingly
8191 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8193 * bindings/vala/gst-rtsp-server-0.10.vapi:
8194 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8195 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8196 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8197 Updated Vala bindings
8199 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8201 * gst/rtsp-server/rtsp-client.c:
8202 * gst/rtsp-server/rtsp-media-mapping.c:
8203 * gst/rtsp-server/rtsp-media-mapping.h:
8204 * gst/rtsp-server/rtsp-media.h:
8205 * gst/rtsp-server/rtsp-session-pool.h:
8206 Cleanups and doc updates
8207 Add some more documentation and do some minor cleanups here and there.
8209 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8211 * gst/rtsp-server/rtsp-client.c:
8212 * gst/rtsp-server/rtsp-media-factory.c:
8213 * gst/rtsp-server/rtsp-media-factory.h:
8214 * gst/rtsp-server/rtsp-media.c:
8215 * gst/rtsp-server/rtsp-media.h:
8216 * gst/rtsp-server/rtsp-session.c:
8217 * gst/rtsp-server/rtsp-session.h:
8219 Rename GstRTSPMediaBin to GstRTSPMedia
8220 Parse the request url into a GstRTSPUri object and pass this object to the
8221 various handlers and methods that require the uri.
8223 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8227 Add some more docs and remove some old code from the example.
8229 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8231 * gst/rtsp-server/rtsp-client.c:
8232 Handle state change failures better
8233 Handle state change failures better when changing the state of the pipeline to
8236 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8238 * gst/rtsp-server/rtsp-media-factory.c:
8239 * gst/rtsp-server/rtsp-media-factory.h:
8240 Make element creation more extendible
8241 Add get_element vmethod to the default MediaFactory so that subclasses can just
8242 override that method and still use the default logic for making a MediaBin from
8245 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8248 * gst/rtsp-server/Makefile.am:
8249 * gst/rtsp-server/rtsp-client.c:
8250 * gst/rtsp-server/rtsp-client.h:
8251 * gst/rtsp-server/rtsp-media-factory.c:
8252 * gst/rtsp-server/rtsp-media-factory.h:
8253 * gst/rtsp-server/rtsp-media-mapping.c:
8254 * gst/rtsp-server/rtsp-media-mapping.h:
8255 * gst/rtsp-server/rtsp-media.c:
8256 * gst/rtsp-server/rtsp-media.h:
8257 * gst/rtsp-server/rtsp-server.c:
8258 * gst/rtsp-server/rtsp-server.h:
8259 * gst/rtsp-server/rtsp-session.c:
8260 * gst/rtsp-server/rtsp-session.h:
8261 Make the server handle arbitrary pipelines
8262 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
8263 The GstMediaBin object has a handle to a bin with elements and to a list of
8264 GstMediaStream objects that this bin produces.
8265 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
8266 with methods to register and remove those mappings.
8267 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
8268 used by the server instance.
8269 Modify the example application so that it shows how to create custom pipelines
8270 attached to a specific mount point.
8271 Various misc cleanps.
8273 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8275 * gst/rtsp-server/rtsp-server.c:
8276 * gst/rtsp-server/rtsp-server.h:
8277 Allow setting a custom media factory for a server
8279 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8281 * gst/rtsp-server/rtsp-client.c:
8282 * gst/rtsp-server/rtsp-client.h:
8283 Allow setting a custom media factory for a client.
8285 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8287 * gst/rtsp-server/Makefile.am:
8288 Add Makefile entry for the media factory
8290 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8292 * gst/rtsp-server/rtsp-media-factory.c:
8293 * gst/rtsp-server/rtsp-media-factory.h:
8294 Add media factory to map urls to media pipeline objects.
8296 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8298 * gst/rtsp-server/rtsp-media.c:
8299 * gst/rtsp-server/rtsp-media.h:
8300 Add comments. Remove unused field
8302 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8304 * gst/rtsp-server/rtsp-session-pool.c:
8305 * gst/rtsp-server/rtsp-session-pool.h:
8306 Allow custom session pools to override the session id allocation algorithms Add some comments.
8308 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8310 * gst/rtsp-server/rtsp-session.h:
8313 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8315 * gst/rtsp-server/rtsp-client.c:
8316 * gst/rtsp-server/rtsp-client.h:
8317 Move the connection code in one place Add some comments
8319 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8321 * gst/rtsp-server/rtsp-server.c:
8322 * gst/rtsp-server/rtsp-server.h:
8323 Make vmethod to create and accept new clients. Add some docs.
8325 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8327 * gst/rtsp-server/rtsp-server.c:
8328 * gst/rtsp-server/rtsp-server.h:
8329 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
8331 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8333 * gst/rtsp-server/rtsp-client.c:
8334 * gst/rtsp-server/rtsp-client.h:
8335 Name the parameters more appropriately.
8337 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8339 * gst/rtsp-server/rtsp-session-pool.c:
8340 Do some more cleanup of the session pool.
8342 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8344 * gst/rtsp-server/Makefile.am:
8345 * gst/rtsp-server/rtsp-client.c:
8346 Check if return value of gst_rtsp_session_get_media is not NULL
8348 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8350 * gst/rtsp-server/Makefile.am:
8351 Install rtsp-session and rtsp-session-pool headers
8353 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8358 * bindings/python/Makefile.am:
8359 * bindings/python/arg-types.py:
8360 * bindings/python/codegen/Makefile.am:
8361 * bindings/python/codegen/__init__.py:
8362 * bindings/python/codegen/argtypes.py:
8363 * bindings/python/codegen/code-coverage.py:
8364 * bindings/python/codegen/codegen.py:
8365 * bindings/python/codegen/definitions.py:
8366 * bindings/python/codegen/defsparser.py:
8367 * bindings/python/codegen/docextract.py:
8368 * bindings/python/codegen/docgen.py:
8369 * bindings/python/codegen/fileprefix.override:
8370 * bindings/python/codegen/fileprefixmodule.c:
8371 * bindings/python/codegen/h2def.py:
8372 * bindings/python/codegen/mergedefs.py:
8373 * bindings/python/codegen/mkskel.py:
8374 * bindings/python/codegen/override.py:
8375 * bindings/python/codegen/reversewrapper.py:
8376 * bindings/python/codegen/scmexpr.py:
8377 * bindings/python/rtspserver-types.defs:
8378 * bindings/python/rtspserver.defs:
8379 * bindings/python/rtspserver.override:
8380 * bindings/python/rtspservermodule.c:
8382 Add python bindings.
8384 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8386 * bindings/Makefile.am:
8388 Don't go into python dir when requirements for python bindings are missing
8390 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8392 * bindings/Makefile.am:
8393 * bindings/vala/Makefile.am:
8395 Install Vala bindings if vala is available
8397 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8399 * bindings/vala/gst-rtsp-server-0.10.deps:
8400 * bindings/vala/gst-rtsp-server-0.10.vapi:
8401 * bindings/vala/gst-rtsp-server.vapi:
8402 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
8403 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
8404 * bindings/vala/packages/gst-rtsp-server-0.10.files:
8405 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
8406 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
8407 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
8408 * bindings/vala/packages/gst-rtsp-server.deps:
8409 * bindings/vala/packages/gst-rtsp-server.excludes:
8410 * bindings/vala/packages/gst-rtsp-server.files:
8411 * bindings/vala/packages/gst-rtsp-server.gi:
8412 * bindings/vala/packages/gst-rtsp-server.metadata:
8413 * bindings/vala/packages/gst-rtsp-server.namespace:
8414 Regenerated Vala bindings
8416 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
8418 * bindings/vala/gst-rtsp-server.vapi:
8419 * bindings/vala/packages/gst-rtsp-server.metadata:
8420 Fixed typo in included headers for vala bindings
8422 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8426 * pkgconfig/Makefile.am:
8427 * pkgconfig/gst-rtsp-server.pc.in:
8428 Added pkgconfig file
8430 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8432 * bindings/vala/gst-rtsp-server.vapi:
8433 * bindings/vala/packages/gst-rtsp-server.excludes:
8434 * bindings/vala/packages/gst-rtsp-server.gi:
8435 * bindings/vala/packages/gst-rtsp-server.metadata:
8436 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
8438 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
8440 * bindings/vala/gst-rtsp-server.vapi:
8441 * bindings/vala/packages/gst-rtsp-server.deps:
8442 * bindings/vala/packages/gst-rtsp-server.files:
8443 * bindings/vala/packages/gst-rtsp-server.gi:
8444 * bindings/vala/packages/gst-rtsp-server.metadata:
8445 * bindings/vala/packages/gst-rtsp-server.namespace:
8448 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
8450 * gst/rtsp-server/rtsp-session.c:
8451 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
8453 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8455 * examples/Makefile.am:
8456 * gst/rtsp-server/Makefile.am:
8457 Put GStreamer version in library name
8459 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8461 * examples/Makefile.am:
8462 * gst/rtsp-server/Makefile.am:
8463 Fix some issues to pass distcheck
8465 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8467 * gst/rtsp-server/rtsp-server.c:
8468 Added port property to GstRTSPServer class.
8470 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8475 * examples/Makefile.am:
8478 * gst/rtsp-server/Makefile.am:
8479 * gst/rtsp-server/rtsp-client.c:
8480 * gst/rtsp-server/rtsp-client.h:
8481 * gst/rtsp-server/rtsp-media.c:
8482 * gst/rtsp-server/rtsp-media.h:
8483 * gst/rtsp-server/rtsp-server.c:
8484 * gst/rtsp-server/rtsp-server.h:
8485 * gst/rtsp-server/rtsp-session-pool.c:
8486 * gst/rtsp-server/rtsp-session-pool.h:
8487 * gst/rtsp-server/rtsp-session.c:
8488 * gst/rtsp-server/rtsp-session.h:
8491 * src/rtsp-client.c:
8492 * src/rtsp-client.h:
8495 * src/rtsp-server.c:
8496 * src/rtsp-server.h:
8497 * src/rtsp-session-pool.c:
8498 * src/rtsp-session-pool.h:
8499 * src/rtsp-session.c:
8500 * src/rtsp-session.h:
8501 Split in library and example program
8503 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8505 * src/rtsp-client.h:
8506 Removed obsolete variable
8508 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
8510 * src/rtsp-client.c:
8511 * src/rtsp-client.h:
8512 Removed pipeline variable GstRTSPClient, because it's only used in one function
8514 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
8517 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
8519 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
8521 * src/rtsp-session.c:
8522 Initialize some more vars.
8524 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
8526 * src/rtsp-session.c:
8527 Initialize variable to avoid compiler warning.
8529 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
8532 Add a reasonable generic .gitignore