3 2014-07-11 Sebastian Dröge <slomo@coaxion.net>
8 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
10 * docs/libs/gst-rtsp-server-sections.txt:
13 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
15 * gst/rtsp-server/rtsp-server.c:
16 server: implement client REMOVE filter
18 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
20 * gst/rtsp-server/rtsp-client.c:
21 * gst/rtsp-server/rtsp-client.h:
22 client: expose _close() method
23 Expose a previously internal close method to close the client
26 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
28 * gst/rtsp-server/rtsp-session-pool.c:
29 session-pool: signal session-removed outside of the lock
30 Release the lock before emiting the session-removed signal.
32 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
34 * gst/rtsp-server/rtsp-client.c:
35 * gst/rtsp-server/rtsp-server.c:
36 * gst/rtsp-server/rtsp-session-pool.c:
37 * gst/rtsp-server/rtsp-session.c:
38 * gst/rtsp-server/rtsp-stream.c:
39 filter: Release lock in filter functions
40 Release the object lock before calling the filter functions. We need to
41 keep a cookie to detect when the list changed during the filter
42 callback. We also keep a hashtable to make sure we only call the filter
43 function once for each object in case of concurrent modification.
44 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
46 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
48 * gst/rtsp-server/rtsp-client.c:
49 client: check if watch is set in handle_teardown()
50 The unit tests run without a watch
52 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
54 * tests/check/gst/client.c:
55 client tests: send teardown to cleanup session
57 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
59 * tests/check/gst/rtspserver.c:
60 server tests: send teardown to cleanup session
62 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
64 * gst/rtsp-server/rtsp-client.c:
65 client: keep ref to client for the session removed handler
66 This extra ref will be dropped when all client sessions have been
67 removed. A session is removed when a client sends teardown, closes its
68 endpoint of the TCP connection or the sessions expires.
69 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
71 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
73 * gst/rtsp-server/rtsp-client.c:
74 * gst/rtsp-server/rtsp-session.c:
75 * tests/check/gst/client.c:
76 client: manage media in session as a last step
77 Once we manage a media in a session, we can't unmanage it anymore
78 without destroying it. Therefore, first check everything before we
79 manage the media, otherwise if something is wrong we have no way to
81 If we created a new session and something went wrong, remove the session
82 again. Fixes a leak in the unit test.
84 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
86 * examples/test-mp4.c:
87 * examples/test-ogg.c:
88 examples: print 'stream ready at url' for mp4 and ogg example
90 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
92 * gst/rtsp-server/rtsp-client.c:
93 * gst/rtsp-server/rtsp-sdp.c:
94 rtsp: fix for MIKEY api change
96 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
98 * gst/rtsp-server/rtsp-client.c:
99 client: free watch context only once
100 The watch context is freed when the source is destroyed. Avoids
101 a CRITICAL when we try to unref the context twice.
103 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
105 * gst/rtsp-server/rtsp-client.c:
108 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
110 * gst/rtsp-server/rtsp-client.c:
111 client: protect sessions with lock
112 Protect the list of sessions with the lock.
113 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
115 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
117 * gst/rtsp-server/rtsp-client.c:
118 Client: keep a ref to the session
119 Don't just keep a weak ref to the session objects but use a hard ref. We
120 will be notified when a session is removed from the pool (expired) with
121 the new session-removed signal.
122 Don't automatically close the RTSP connection when all the sessions of
123 a client are removed, a client can continue to operate and it can create
124 a new session if it wants. If you want to remove the client from the
125 server, you have to use gst_rtsp_server_client_filter() now.
126 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
127 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
129 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
131 * gst/rtsp-server/rtsp-session-pool.c:
132 * gst/rtsp-server/rtsp-session-pool.h:
133 session-pool: add session-removed signal
134 Add a signal to be notified when a session is removed from the pool.
136 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
138 * gst/rtsp-server/Makefile.am:
139 * gst/rtsp-server/rtsp-server.h:
140 Make rtsp-server.h a single-include header, use it for G-I
141 https://bugzilla.gnome.org/show_bug.cgi?id=732411
143 === release 1.3.90 ===
145 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
151 * gst-rtsp-server.doap:
154 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
156 * gst/rtsp-server/rtsp-stream.c:
157 stream: crypto can be NULL
159 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
161 * gst/rtsp-server/rtsp-client.c:
162 * gst/rtsp-server/rtsp-media.c:
163 * gst/rtsp-server/rtsp-mount-points.c:
164 introspection: add missing allow-none annotations
165 https://bugzilla.gnome.org/show_bug.cgi?id=730952
167 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
169 * gst/rtsp-server/rtsp-address-pool.c:
170 * gst/rtsp-server/rtsp-media.c:
171 * gst/rtsp-server/rtsp-session-media.c:
172 * gst/rtsp-server/rtsp-session-pool.c:
173 * gst/rtsp-server/rtsp-stream-transport.c:
174 * gst/rtsp-server/rtsp-stream.c:
175 * gst/rtsp-server/rtsp-token.c:
176 introspection: add (nullable) annotations to return values
177 https://bugzilla.gnome.org/show_bug.cgi?id=730952
179 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
181 * gst/rtsp-server/rtsp-client.c:
182 * gst/rtsp-server/rtsp-stream.c:
183 gi: improve annotations
184 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
186 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
188 * gst/rtsp-server/rtsp-client.c:
189 * gst/rtsp-server/rtsp-media-factory.c:
190 * gst/rtsp-server/rtsp-media.c:
191 * gst/rtsp-server/rtsp-server.c:
192 signals: use generic marshal function
193 Use the generic C marshal function.
194 Use more explicit type instead of G_TYPE_POINTER
196 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
198 * gst/rtsp-server/rtsp-context.h:
199 context: add type macro
201 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
203 * gst/rtsp-server/rtsp-client.c:
204 * gst/rtsp-server/rtsp-sdp.c:
205 * gst/rtsp-server/rtsp-sdp.h:
206 sdp: hide key length defines
207 They don't have a namespace.
209 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
214 === release 1.3.3 ===
216 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
222 * gst-rtsp-server.doap:
225 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
227 * gst/rtsp-server/rtsp-client.c:
228 * gst/rtsp-server/rtsp-sdp.c:
229 * gst/rtsp-server/rtsp-sdp.h:
230 mikey: add different key length parameters
231 Add encryption and authentication key length parameters to MIKEY. For
232 the encoders, the key lengths are obtained from the cipher and auth
233 algorithms set in the caps. For the decoders, they are obtained while
234 parsing the key management from the client.
235 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
237 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
239 * tests/check/gst/stream.c:
240 stream tests: Make sure we get right multicast address from stream
241 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
243 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
245 * gst/rtsp-server/rtsp-client.c:
246 client: ref the context until rtsp watch is alive
247 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
249 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
251 * gst/rtsp-server/rtsp-client.c:
252 client: Destroy the rtsp watch after connection close
254 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
256 * gst/rtsp-server/rtsp-media.c:
257 media: fix confusing comment
259 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
261 * gst/rtsp-server/rtsp-session.c:
262 rtsp-session: Timeout in header.
263 Adding the possbilty to always have timout in header.
264 This is configurabe with setting "timeout-always-visible".
265 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
267 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
272 === release 1.3.2 ===
274 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
281 * gst-rtsp-server.doap:
284 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
287 Automatic update of common submodule
288 From 211fa5f to 1f5d3c3
290 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
292 * gst/rtsp-server/rtsp-client.c:
293 client: store TCP ports in transport
294 Store the TCP ports in the transport when we are doing RTSP over TCP.
295 This way, we can easily get to the ports from the transport.
296 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
298 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
300 * gst/rtsp-server/rtsp-stream.c:
301 stream: add signals for new RTP/RTCP encoders
302 New signals to allow the user to configure the dynamically created
304 https://bugzilla.gnome.org/show_bug.cgi?id=730228
306 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
308 * gst/rtsp-server/rtsp-media.c:
309 * gst/rtsp-server/rtsp-media.h:
310 media: Make suspend()/unsuspend() virtual
311 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
313 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
315 * gst/rtsp-server/rtsp-client.c:
316 client: fix send-message signal marshaller
317 Use generic marshalling for the send-message signal. It has
318 two POINTER arguments, not just one.
319 https://bugzilla.gnome.org/show_bug.cgi?id=729900
321 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
323 * tests/check/gst/media.c:
324 tests: add and remove pads only once
325 In this test we simulate a dynamic pad by watching the caps event.
326 Because of renegotiation in the base payloader now, this caps is sent
327 multiple times but we can only deal with 1 invocation, use a variable to
328 only 'add and remove' the pad once.
330 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
332 * tests/check/gst/rtspserver.c:
333 tests: add unit test for correct handling of Require headers
334 https://bugzilla.gnome.org/show_bug.cgi?id=729426
336 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
338 * gst/rtsp-server/rtsp-client.c:
339 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
340 Servers must handle Require headers and must report a failure
341 if they don't handle any of the Required options, see RFC 2326,
342 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
343 https://bugzilla.gnome.org/show_bug.cgi?id=729426
345 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
350 === release 1.3.1 ===
352 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
358 * gst-rtsp-server.doap:
361 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
364 Automatic update of common submodule
365 From bcb1518 to 211fa5f
367 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
372 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
374 * tests/check/gst/sessionmedia.c:
375 tests: fix memory leak in sessionmedia unit test
377 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
379 * gst/rtsp-server/rtsp-client.c:
380 client: emit a signal before sending a message
381 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
383 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
385 * gst/rtsp-server/rtsp-client.c:
386 client: pass context to send_message
387 Pass the current context to send_message, we will need it later.
389 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
391 * gst/rtsp-server/rtsp-client.c:
392 client: fix typo in comment
394 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
396 * gst/rtsp-server/rtsp-media.c:
397 media: Do not stop thread twice if default_prepare() fails
399 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
401 * gst/rtsp-server/rtsp-client.c:
402 client: set the watch to flushing before going to NULL
403 First set the watch to flushing so that we unblock any current and
404 future attempt to send data on the watch, Then set the pipeline to
406 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
408 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
410 * gst/rtsp-server/rtsp-session-pool.c:
411 * tests/check/gst/sessionpool.c:
412 rtsp-session-pool: Fixes annotation
413 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
414 in the sessionpool test.
415 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
417 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
419 * gst/rtsp-server/rtsp-media.c:
420 * gst/rtsp-server/rtsp-media.h:
421 media: make media_prepare virtual
422 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
424 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
426 * gst/rtsp-server/rtsp-media.c:
427 * tests/check/gst/media.c:
428 media: stop the thread in more error cases
430 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
432 * gst/rtsp-server/rtsp-media.c:
433 * tests/check/gst/media.c:
434 media: allow NULL as the thread
435 Use the default context whan passing a NULL thread.
437 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
439 * gst/rtsp-server/rtsp-client.c:
440 rtsp-client: indent cleanup
441 Coverity was moaning about unreachable code, and I think it was just
442 confused by { being before the label. We'll see if it pops up again.
445 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
447 * gst/rtsp-server/rtsp-client.c:
448 * gst/rtsp-server/rtsp-media.c:
449 client: Add drop-backlog property
450 When we have too many messages queued for a client (currently hardcoded
451 to 100) we overflow and drop the messages. Add a drop-backlog property
452 to control this behaviour. Setting this property to FALSE will retry
453 to send the messages to the client by waiting for more room in the
455 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
457 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
459 * gst/rtsp-server/rtsp-client.c:
460 client: support for POST before GET when setting up a tunnel
462 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
464 * gst/rtsp-server/rtsp-client.c:
465 client: remove watch of the second client after http tunnel setup
466 The second client will be freed after the HTTP tunnel has been set up.
467 Make sure it's RTSP watch is never dispatched again.
468 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
470 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
472 * gst/rtsp-server/rtsp-media.c:
473 * tests/check/gst/media.c:
474 media: Make media_prepare() fail if port allocation fails
475 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
477 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
479 * tests/check/gst/media.c:
480 media test: cleanup the thread pool in tests
482 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
484 * gst/rtsp-server/rtsp-media.c:
485 * tests/check/gst/media.c:
486 rtsp-media: Unblock blocked streams in unprepare
487 The streams will be blocked when a live media is prepared.
488 The streams should be unblocked in gst_rtsp_media_unprepare.
489 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
491 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
493 * gst/rtsp-server/rtsp-media.c:
494 media: release the state lock when going to NULL
495 Set our state to UNPREPARING and release the state-lock before
496 setting the pipeline to the NULL state. This way, any pad-added
497 callback will be able to take the state-lock and check that we are now
498 unpreparing instead of deadlocking.
499 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
501 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
503 * gst/rtsp-server/rtsp-media.c:
504 media: protect status with lock
505 Make sure we only update the status with the lock.
507 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
509 * gst/rtsp-server/rtsp-client.c:
510 * gst/rtsp-server/rtsp-sdp.c:
511 rtsp: update for MIKEY API changes
513 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
515 * gst/rtsp-server/rtsp-client.c:
516 client: parse the mikey response from the client
517 Parse the mikey response from the client and update the policy for
520 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
522 * gst/rtsp-server/rtsp-stream.c:
523 * gst/rtsp-server/rtsp-stream.h:
524 stream: add method to set crypto info
525 Make a method to configure the crypto information of a stream.
526 Set udpsrc in READY instead of PAUSED so that we can configure caps
529 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
531 * gst/rtsp-server/rtsp-client.c:
532 client: cleanup error paths
534 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
536 * gst/rtsp-server/rtsp-media.c:
539 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
541 * examples/test-video.c:
542 test: enable SRTP only on RTSPS
543 We only want to enable SRTP when doing rtsp over TLS so that we can
544 exchange the keys in a secure way.
546 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
548 * examples/test-video.c:
549 test: print an error on failure
551 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
554 * examples/test-video.c:
555 * gst/rtsp-server/rtsp-sdp.c:
556 * gst/rtsp-server/rtsp-stream.c:
557 * tests/check/Makefile.am:
558 stream: add SRTP support
559 Install srtp encoder and decoder elements in rtpbin
562 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
564 * tests/check/Makefile.am:
565 * tests/check/gst/sessionpool.c:
566 tests: Add unit tests for sessionpool
567 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
569 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
571 * tests/check/gst/threadpool.c:
572 tests: Improve code coverage of rtsp-threadpool tests
573 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
575 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
577 * tests/check/gst/sessionmedia.c:
578 tests: Improve code coverage for rtsp-session-media
579 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
581 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
583 gobject-introspection: Add annotations to support language bindings
584 In addition a few cosmetic changes:
585 * Adjust the order of arguments
586 * Fix typo: occured -> occurred
587 * Fix indentation after Return:-clauses
588 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
590 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
592 * gst/rtsp-server/rtsp-stream.c:
593 rtsp-stream: Don't mix IPv4 and IPv6 addresses
594 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
596 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
598 * gst/rtsp-server/rtsp-stream.c:
599 stream: take caps after the session manager
600 Take the caps for the SDP after they leave the rtpbin so that we can
601 also get the properties added by rtpbin elements.
603 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
605 * gst/rtsp-server/rtsp-stream.c:
606 stream: release lock while pushing out packets
607 Keep a cache of the transports and use this to iterate the transport
608 while pushing packets. This allows us to release the lock early.
609 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
611 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
613 * gst/rtsp-server/rtsp-client.c:
614 * gst/rtsp-server/rtsp-client.h:
615 rtsp-client: vmethod for modifying tunnel GET response
616 Add a vmethod tunnel_http_response where the response to the HTTP GET
617 for tunneled connections can be modified.
618 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
620 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
622 * gst/rtsp-server/rtsp-sdp.c:
623 sdp: make 1 media line per profile
624 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
625 line in the SDP for each profile. The client is then supposed to pick
626 one of the profiles in the SETUP request. Because the m= lines have the
627 same pt, the client also knows that only 1 option is possible.
629 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
631 * gst/rtsp-server/rtsp-media-factory.c:
632 * gst/rtsp-server/rtsp-media-factory.h:
633 * gst/rtsp-server/rtsp-media.c:
634 factory: add profile property and pass to media and streams
636 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
638 * examples/test-multicast.c:
639 * gst/rtsp-server/rtsp-sdp.c:
640 sdp: pass multicast connection for multicast-only stream
641 Pass the multicast address of the stream in the connection info in the
642 SDP so that clients try a multicast connection first.
643 Only allow multicast connections in the test-multicast example. Also
644 increase the TTL a little.
646 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
649 .gitignore: Ignore gcov intermediate files
650 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
652 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
654 * gst/rtsp-server/rtsp-stream.c:
655 stream: release some locks in error cases
657 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
659 docs: Enable and fix gtk-doc warnings
660 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
661 * addresspool/mediafactory: Add missing annotation colon
662 * stream: Annotate return value
663 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
665 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
668 Automatic update of common submodule
669 From fe1672e to bcb1518
671 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
674 Automatic update of common submodule
675 From 1a07da9 to fe1672e
677 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
679 * examples/Makefile.am:
680 examples: use LDADD for libs instead of LDFLAGS
682 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
685 configure: make sure releases are in .doap file
687 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
689 * examples/test-cgroups.c:
690 examples: test-cgroups: don't put code with side effects into g_assert()
691 The g_assert() might get compiled out with the right
692 compiler/preprocessor flags.
694 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
696 * examples/.gitignore:
697 examples: add cgroup test binary to .gitignore
699 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
701 * examples/test-cgroups.c:
702 examples: fix cgroup test build
703 Fixes build failure caused by compiler warning:
704 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
706 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
709 .gitignore: ignore temp files created in the course of 'make check'
711 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
713 * gst/rtsp-server/rtsp-media.c:
714 rtsp-media: don't loose frames handling new PLAY request
715 If client supplied a range check if the range specifies the start point.
716 If not, then do an accurate seek to the current position. If a start
717 point was specified do do a key unit seek to make sure the streaming
718 starts with decodeable frames.
719 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
721 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
723 * gst/rtsp-server/rtsp-media.c:
724 Revert "media: only flush when setting a new start position"
725 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
726 We need to do the flush in all cases, demuxer block currently for
729 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
731 * gst/rtsp-server/rtsp-media.c:
732 media: only flush when setting a new start position
733 Only flush the pipeline when we change the start position with
735 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
737 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
739 * gst/rtsp-server/rtsp-stream.c:
740 stream: set ttl-mc before adding the socket
741 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
742 never be set on socket.
743 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
745 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
747 * gst/rtsp-server/rtsp-media.c:
748 media: stop thread if media is already prepared
749 in gst_rtsp_media_prepare() the thread is not used if media is already
750 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
752 https://bugzilla.gnome.org/show_bug.cgi?id=724182
754 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
757 build: Ship gst-rtsp-server.doap file
759 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
761 * tests/check/gst/rtspserver.c:
762 tests: Fix another compiler warning with gcc
764 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
766 * gst/rtsp-server/rtsp-client.c:
767 * gst/rtsp-server/rtsp-mount-points.c:
768 * gst/rtsp-server/rtsp-stream.c:
769 * tests/check/gst/client.c:
770 rtsp-server: Fix lots of compiler warnings with clang
772 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
775 * gst-rtsp-server.doap:
777 configure: Synchronise with the configure scripts of the other modules
779 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
782 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
784 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
786 * gst/rtsp-server/rtsp-media.c:
787 * gst/rtsp-server/rtsp-stream.c:
788 Revert "rtsp-server: support build against last stable release"
789 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
790 Let us require 1.2.3 now, which is going to be released in a few
793 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
795 * gst/rtsp-server/rtsp-session-media.c:
796 * gst/rtsp-server/rtsp-stream-transport.c:
797 session: improve RTP-Info
798 Ignore streams that can't generate RTP-Info instead of failing.
799 Don't return the empty string when all streams are unconfigured but
800 return NULL so that we don't generate and empty RTP-Info header.
801 Improve docs a little.
803 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
805 * gst/rtsp-server/rtsp-session-media.c:
806 Don't free rtpinfo GString when it is NULL
807 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
809 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
811 * gst/rtsp-server/rtsp-media.c:
812 media: only set keyframe flag when modifying start
813 Only set the keyframe flag when we modify the start position. The
814 keyframe flag should probably be ignored when no change is requested but
815 until we can claim this is all documented properly and all demuxer
816 implement this, avoid setting the flag.
817 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
819 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
821 * gst/rtsp-server/rtsp-thread-pool.c:
822 thread-pool: Unref source after mainloop has quit to avoid races in GLib
823 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
825 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
827 * gst/rtsp-server/rtsp-stream.c:
828 stream: handle NULL seqnum and rtptime arguments
830 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
832 * gst/rtsp-server/rtsp-thread-pool.c:
833 * tests/check/gst/threadpool.c:
834 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
835 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
837 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
839 * gst/rtsp-server/rtsp-stream.c:
840 stream: add fallback for missing stats property
841 Use a fallback when the payloader does not have a stats property
842 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
844 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
847 Automatic update of common submodule
848 From f7bc1c3 to 1a07da9
850 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
852 * gst/rtsp-server/rtsp-stream.c:
853 stream: don't leak stats structure
854 Don't leak the stats structure and deal with NULL stats.
856 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
858 * gst/rtsp-server/rtsp-stream.c:
859 stream: Get rtpinfo properties atomically from payloader
860 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
862 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
864 * gst/rtsp-server/rtsp-media.c:
865 media: refactor state change functions and signals
866 Make functions to set the target state and the pipeline state and emit
867 the signals from those functions.
869 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
871 * gst/rtsp-server/rtsp-media.c:
872 * gst/rtsp-server/rtsp-media.h:
873 media: add signal to notify of pending state changes
875 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
877 * gst/rtsp-server/rtsp-media.c:
878 * gst/rtsp-server/rtsp-stream.c:
879 rtsp-server: support build against last stable release
880 Until 1.2.3 is out with the new get_type function and we
883 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
885 * gst/rtsp-server/rtsp-stream.c:
886 stream: fix compilation
888 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
890 * gst/rtsp-server/rtsp-media.c:
891 * gst/rtsp-server/rtsp-media.h:
892 * gst/rtsp-server/rtsp-stream.c:
893 * gst/rtsp-server/rtsp-stream.h:
894 stream: add property to configure profiles
896 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
898 * gst/rtsp-server/rtsp-client.c:
899 client: let stream check supported transport
900 Delegate the check if a transport is allowed to the stream.
901 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
903 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
905 * gst/rtsp-server/rtsp-stream.c:
906 * gst/rtsp-server/rtsp-stream.h:
907 stream: add method to check supported transport
908 Add a method to check if a transport is supported
910 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
913 configure.ac: Only check for gstreamer-check, not check
914 We include check in gstreamer-check since quite some time now.
916 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
918 * gst/rtsp-server/rtsp-session-media.c:
919 * gst/rtsp-server/rtsp-stream-transport.c:
920 * gst/rtsp-server/rtsp-stream.c:
921 * gst/rtsp-server/rtsp-stream.h:
922 stream: return clock-rate from get_rtpinfo
923 And use it to correct the rtptime to the requested start-time.
924 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
926 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
928 * gst/rtsp-server/rtsp-session-media.c:
929 * gst/rtsp-server/rtsp-stream-transport.c:
930 * gst/rtsp-server/rtsp-stream-transport.h:
931 session-media: calculate start-time
933 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
935 * gst/rtsp-server/rtsp-stream-transport.c:
936 * gst/rtsp-server/rtsp-stream.c:
937 * gst/rtsp-server/rtsp-stream.h:
938 stream: also return the running-time
939 Return the running-time in the rtpinfo as well.
941 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
943 * gst/rtsp-server/rtsp-client.c:
944 * gst/rtsp-server/rtsp-session-media.c:
945 * gst/rtsp-server/rtsp-session-media.h:
946 * gst/rtsp-server/rtsp-stream-transport.c:
947 * gst/rtsp-server/rtsp-stream-transport.h:
948 session-media: let the session-media make the RTPInfo
949 Add method to create the RTPInfo for a stream-transport.
950 Add method to create the RTPInfo for all stream-transports in a
952 Use the session-media RTPInfo code in client. This allows us to refactor
953 another method to link the TCP callbacks.
955 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
957 mount-points: sort sequence before g_sequence_lookup
958 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
959 sort sequence if dirty, otherwise lookup will fail.
960 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
962 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
965 configure: rename package from gst-rtsp to gst-rtsp-server
966 To match git module name and avoid confusion with the
967 rtsp lib in gst-plugins-base and rtsp plugin in -good.
969 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
972 configure: bump core/base/good requirement to 1.2.0
973 Bump to released stable version and make implicit
974 requirements explicit.
976 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
981 Fix broken gettext setup which is not used anyway
983 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
986 Automatic update of common submodule
987 From dbedaa0 to d48bed3
989 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
991 * gst/rtsp-server/rtsp-client.c:
992 * gst/rtsp-server/rtsp-media.c:
993 * gst/rtsp-server/rtsp-media.h:
994 media: add setup_sdp vmethod
995 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
996 gst_rtsp_media_setup_sdp.
997 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
999 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
1001 * gst/rtsp-server/rtsp-stream.c:
1002 rtsp-stream: Check return value of sscanf
1003 streamid is only valid if sscanf matched something.
1005 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
1007 * gst/rtsp-server/rtsp-client.c:
1008 rtsp-client: Fix iteration
1009 Wouldn't even enter the code block otherwise (i++ was used as the check
1010 and not the postfix).
1012 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
1014 * gst/rtsp-server/rtsp-client.c:
1015 * gst/rtsp-server/rtsp-client.h:
1016 client: add vmethod to configure media and streams
1017 Implement a vmethod that can be used to configure the media and the
1018 streams based on the current context. Handle the blocksize handling in
1019 the default handler.
1020 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
1022 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
1025 Make git ignore more unit test binaries
1027 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
1029 * gst/rtsp-server/rtsp-address-pool.h:
1030 * gst/rtsp-server/rtsp-auth.h:
1031 * gst/rtsp-server/rtsp-client.h:
1032 * gst/rtsp-server/rtsp-context.h:
1033 * gst/rtsp-server/rtsp-media-factory-uri.h:
1034 * gst/rtsp-server/rtsp-media-factory.h:
1035 * gst/rtsp-server/rtsp-media.h:
1036 * gst/rtsp-server/rtsp-mount-points.h:
1037 * gst/rtsp-server/rtsp-server.h:
1038 * gst/rtsp-server/rtsp-session-media.h:
1039 * gst/rtsp-server/rtsp-session-pool.h:
1040 * gst/rtsp-server/rtsp-session.h:
1041 * gst/rtsp-server/rtsp-stream-transport.h:
1042 * gst/rtsp-server/rtsp-stream.h:
1043 * gst/rtsp-server/rtsp-thread-pool.h:
1044 * gst/rtsp-server/rtsp-token.h:
1045 rtsp-server: add padding to many public structures
1046 Not mini objects though, since they are not subclassable
1047 anyway, nor kept on the stack or inlined in a structure.
1049 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
1051 media: add new create_rtpbin vmethod
1052 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
1053 https://bugzilla.gnome.org/show_bug.cgi?id=719734
1055 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
1057 * tests/check/gst/media.c:
1058 tests: fix memory leak, free test's thread pool
1059 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
1061 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
1063 * gst/rtsp-server/rtsp-stream-transport.c:
1064 stream-transport: free url in finalize
1066 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
1068 * gst/rtsp-server/rtsp-media.c:
1069 media: also do state change in suspended state
1071 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
1073 * gst/rtsp-server/rtsp-client.c:
1074 * gst/rtsp-server/rtsp-media.c:
1075 media: also handle prepare and range in suspended state
1076 When we are suspended, we are already prepared.
1077 We can get the range in the suspended state.
1079 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
1081 * tests/check/Makefile.am:
1082 * tests/check/gst/sessionmedia.c:
1083 check: add test for uri in setup
1084 Added unit tests for the new functionality in GstRTSPStreamTransport.
1085 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
1087 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
1089 * gst/rtsp-server/rtsp-client.c:
1090 client: store setup uri and use in PLAY response
1091 Store the uri used when doing the setup and use that in the PLAY
1093 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
1095 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
1097 * gst/rtsp-server/rtsp-stream-transport.c:
1098 * gst/rtsp-server/rtsp-stream-transport.h:
1099 stream-transport: add method to get/set url
1101 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
1103 * gst/rtsp-server/rtsp-client.c:
1104 client: suspend after SDP and unsuspend before PLAYING
1105 Based on patches by Ognyan Tonchev <ognyan@axis.com>
1106 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
1108 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
1110 * gst/rtsp-server/rtsp-media-factory.c:
1111 * gst/rtsp-server/rtsp-media-factory.h:
1112 * gst/rtsp-server/rtsp-media.c:
1113 * gst/rtsp-server/rtsp-media.h:
1114 * gst/rtsp-server/rtsp-session-media.c:
1115 * gst/rtsp-server/rtsp-session.c:
1116 * tests/check/gst/media.c:
1117 * tests/check/gst/mediafactory.c:
1118 media: add suspend modes
1119 Add support for different suspend modes. The stream is suspended right after
1120 producing the SDP and after PAUSE. Different suspend modes are available that
1121 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
1122 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
1123 state and RESET will bring the pipeline to the NULL state.
1124 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
1125 this means that the pipeline needs to be prerolled again.
1126 Base on patches by Ognyan Tonchev <ognyan@axis.com>
1127 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
1129 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
1131 * gst/rtsp-server/rtsp-media.c:
1132 media: start live streams in blocked state
1133 Start live streams in the blocked state and make them preroll using the
1134 messages. This ensure that no data is played by the sink until we explicitly
1135 unblock the stream right before going to PLAYING.
1136 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
1138 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
1140 * gst/rtsp-server/rtsp-media.c:
1141 media: refactor starting and waiting for preroll
1142 Based on patches from Ognyan Tonchev <ognyan@axis.com>
1143 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
1145 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
1147 * gst/rtsp-server/rtsp-stream.c:
1148 * gst/rtsp-server/rtsp-stream.h:
1149 stream: add API to block streams
1150 Add an API to block on the streams and make it post a message.
1151 Based on patch by Ognyan Tonchev <ognyan@axis.com>
1152 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
1154 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
1156 * docs/libs/Makefile.am:
1157 docs: Specify the override file
1158 Even if it's empty (for now) it avoids make distcheck complaining
1160 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
1162 * gst/rtsp-server/rtsp-media.c:
1163 media: move default implementations to where they are used
1165 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
1167 * gst/rtsp-server/rtsp-media.c:
1168 media: take the right lock in gst_rtsp_media_set_pipeline_state()
1169 We need to take the state_lock when calling this method.
1171 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
1173 * gst/rtsp-server/rtsp-media.c:
1174 media: handle add-added on non-bins too
1175 Handle dynamic payloaders that are not bins, as used in the unit-test.
1177 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1179 * gst/rtsp-server/rtsp-media-factory.c:
1180 * gst/rtsp-server/rtsp-media-factory.h:
1181 * gst/rtsp-server/rtsp-media.c:
1182 rtsp-media/-factory: Fix request pad name comments
1183 These must be escaped for gtk-doc to parse the comments without warnings.
1185 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
1187 rtsp-media: remove transports if media is in error status
1188 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
1189 trying to change to GST_STATE_NULL and media is in error status, we
1190 remove all transports.
1191 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
1193 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
1195 * gst/rtsp-server/rtsp-media.c:
1196 rtsp-media: use element metadata to find payloader
1197 Use the element metadata to find the payloader instead of checking
1199 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
1201 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
1203 rtsp-stream: add getter for payload type
1204 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
1205 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
1206 element and create the stream with this one instead of the dynpay%d
1208 https://bugzilla.gnome.org/show_bug.cgi?id=712396
1210 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1212 * gst/rtsp-server/rtsp-client.c:
1213 * gst/rtsp-server/rtsp-context.h:
1214 * gst/rtsp-server/rtsp-media.c:
1215 * gst/rtsp-server/rtsp-mount-points.c:
1216 * gst/rtsp-server/rtsp-server.c:
1217 * gst/rtsp-server/rtsp-token.c:
1218 rtsp-*: Refer to NULL as a constant in comments
1220 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1222 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1224 rtsp-*: Fix type name typos in comments
1225 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
1226 * rtsp-auth: Refer to part of constant name as text
1227 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
1228 * rtsp-session-media: Fix GstRTSPSessionMedia typo
1229 * rtsp-stream: Fix typo when refering to GstBin
1230 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1232 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1235 * docs/libs/gst-rtsp-server-docs.sgml:
1236 * docs/libs/gst-rtsp-server-sections.txt:
1237 docs: Improve documentation
1238 * Include annotation-glossary to quiet gtk-doc
1239 * Rename remaining ClientState -> Context
1240 * Rename object hierarchy file
1241 * Remove stale chapter references
1242 * Add missing function and object references
1243 * Include missing GstRTSPAddressPoolResult
1244 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1246 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
1248 * gst/rtsp-server/rtsp-client.c:
1249 * gst/rtsp-server/rtsp-server.c:
1250 * gst/rtsp-server/rtsp-session-pool.c:
1251 * gst/rtsp-server/rtsp-session.c:
1252 * gst/rtsp-server/rtsp-stream.c:
1253 rtsp-server: sprinkle some allow-none annotations for g-i
1255 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
1257 * gst/rtsp-server/rtsp-stream.c:
1258 * gst/rtsp-server/rtsp-stream.h:
1259 stream: add method to filter transports
1260 Add a method to safely iterate and collect the stream transports
1261 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
1263 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
1265 * gst/rtsp-server/rtsp-client.c:
1266 * gst/rtsp-server/rtsp-server.c:
1267 * gst/rtsp-server/rtsp-session-pool.c:
1268 * gst/rtsp-server/rtsp-session.c:
1269 rtsp: allow NULL func in filters
1270 Passing a null function make the filters return a list of
1273 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
1275 * gst/rtsp-server/rtsp-address-pool.c:
1276 * tests/check/gst/addresspool.c:
1277 address-pool: fix address increment
1278 Use a guint instead of guint8 to increment the address. It's still not
1279 completely correct because a guint might not be able to hold the complete
1280 address range, but that's an enhacement for later.
1281 Add unit test to test improved behaviour.
1282 https://bugzilla.gnome.org/show_bug.cgi?id=708237
1284 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
1286 * gst/rtsp-server/rtsp-client.c:
1287 * tests/check/gst/client.c:
1288 client: allow absolute path in requests
1289 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
1291 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
1293 * gst/rtsp-server/rtsp-client.c:
1294 * gst/rtsp-server/rtsp-client.h:
1295 client: make make_path_from_uri a vmethod
1297 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1299 * docs/libs/gst-rtsp-server-sections.txt:
1300 * gst/rtsp-server/rtsp-stream.c:
1301 * gst/rtsp-server/rtsp-stream.h:
1302 * tests/check/Makefile.am:
1303 * tests/check/gst/stream.c:
1304 stream: Add functions to get rtp and rtcp sockets
1305 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
1307 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1309 * gst/rtsp-server/rtsp-context.c:
1310 * gst/rtsp-server/rtsp-context.h:
1311 context: defing a GType for the context
1312 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
1314 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
1316 * gst/rtsp-server/Makefile.am:
1317 * gst/rtsp-server/rtsp-auth.c:
1318 * gst/rtsp-server/rtsp-context.c:
1319 * gst/rtsp-server/rtsp-media.c:
1320 * gst/rtsp-server/rtsp-mount-points.c:
1321 * gst/rtsp-server/rtsp-server.h:
1322 * gst/rtsp-server/rtsp-session-media.c:
1323 * gst/rtsp-server/rtsp-session.c:
1324 * gst/rtsp-server/rtsp-stream.c:
1325 Fixed several GIR warnings
1327 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
1329 * gst/rtsp-server/rtsp-auth.c:
1332 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1334 * tests/check/Makefile.am:
1335 * tests/check/gst/token.c:
1336 tests: Add unit tests for token
1337 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
1339 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1341 * gst/rtsp-server/rtsp-token.c:
1342 token: Validate args for gst_rtsp_token_is_allowed
1343 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
1345 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1347 * gst/rtsp-server/rtsp-token.c:
1348 token: Fix bug when creating empty token
1349 We always want to have a valid GstStructure in the token.
1350 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
1352 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1354 * gst/rtsp-server/rtsp-thread-pool.c:
1355 thread-pool: avoid race in shutdown
1356 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
1357 don't actually stop the mainloop ever. Solve this race by adding an idle source
1358 to the mainloop that calls the _quit. This way we immediately exit the mainloop
1359 if quit was called before we started it.
1361 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1363 * tests/check/Makefile.am:
1364 * tests/check/gst/permissions.c:
1365 tests: Add unit tests for permissions
1366 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
1368 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1370 * tests/check/gst/mediafactory.c:
1371 tests: Test mediafactory permissions
1372 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1374 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1376 * gst/rtsp-server/rtsp-permissions.c:
1377 permissions: Fix refcounting when adding/removing roles
1378 Previously a role that was removed was unreffed twice, and when
1379 replacing an existing role the replaced role was freed while still being
1380 referenced. Both bugs are now fixed.
1381 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1383 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1385 * tests/check/gst/media.c:
1386 * tests/check/gst/mediafactory.c:
1387 * tests/check/gst/rtspserver.c:
1388 tests: Check gst_rtsp_url_parse return value
1389 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1391 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
1394 Automatic update of common submodule
1395 From 865aa20 to dbedaa0
1397 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
1399 * gst/rtsp-server/rtsp-server.c:
1400 rtsp-server: Fix socket leak
1401 https://bugzilla.gnome.org/show_bug.cgi?id=710088
1403 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
1405 * gst/rtsp-server/rtsp-session-pool.c:
1406 rtsp-session-pool: Make sure session IDs are properly URI-escaped
1407 https://bugzilla.gnome.org/show_bug.cgi?id=643812
1409 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
1411 * examples/.gitignore:
1412 * examples/test-video.c:
1413 examples: fix compilation when WITH_AUTH is defined
1414 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1416 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
1419 gitignore: Add new test binary
1421 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
1423 * tests/check/Makefile.am:
1424 * tests/check/gst/threadpool.c:
1425 thread-pool: Add unit test for the thread pools
1426 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1428 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1430 * gst/rtsp-server/rtsp-thread-pool.c:
1431 thread-pool: Fix thread leak when reusing threads
1432 https://bugzilla.gnome.org/show_bug.cgi?id=709730
1434 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
1436 * gst/rtsp-server/rtsp-server.c:
1437 * tests/check/gst/rtspserver.c:
1438 tests: fixed racy behavior in rtspserver tests
1439 https://bugzilla.gnome.org/show_bug.cgi?id=710078
1441 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1443 * tests/check/gst/addresspool.c:
1444 tests: Improve address pool unit tests
1445 Add a range with mixed IPV4 and IPV6 addresses to pool.
1446 Get an IPV4 address from an IPV6-only pool.
1447 Get an IPV6 address from an IPV4-only pool.
1448 Reserve a IPV6 address from an IPV4-only pool.
1449 Check for unicast addresses in multicast-only pool.
1450 Check for unicast addresses in uni-/multicast-mixed pool.
1451 https://bugzilla.gnome.org/show_bug.cgi?id=710128
1453 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1455 * gst/rtsp-server/rtsp-client.c:
1456 client: append query string in PAUSE/PLAY/TEARDOWN as well
1458 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
1460 * gst/rtsp-server/rtsp-client.c:
1461 client: Add query to control path
1462 If the SETUP url contains a query it must be appended to the control
1463 path so that it matches any already created stream in the media. The
1464 query will also be appended to the session media path.
1466 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1468 * gst/rtsp-server/rtsp-media.c:
1469 rtsp-media: remove old line
1471 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
1473 * gst/rtsp-server/rtsp-stream.c:
1474 stream: Correct control comparison
1475 https://bugzilla.gnome.org/show_bug.cgi?id=709176
1477 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1479 * gst/rtsp-server/rtsp-media.c:
1480 media: Check dynamically if the pipeline supports seeking
1481 We should not depend on whether or not the pipeline state change
1482 returned NO_PREROLL or not. A media could dynamically change its
1483 element and switch from seekable to non seekable so it's best to test
1484 the seekable nature of the pipeline dynamically when we try to do a seek.
1486 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1488 * gst/rtsp-server/rtsp-media.c:
1489 media: Return FALSE if seeking is not supported
1491 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1493 * gst/rtsp-server/rtsp-media.c:
1494 rtsp-media: don't seek accurate by default
1495 Accurate seeking is perhaps a little overkill in the most common situation and
1496 causes some formats (mp3) over slow media to seek extremely slowly.
1498 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
1500 * tests/check/gst/rtspserver.c:
1501 tests: fix unit test
1502 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
1504 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
1506 * gst/rtsp-server/rtsp-client.c:
1507 client: Reply 400 if media cannot be constructed
1508 Reply 400 Bad Request instead of 503 Service Unavailable if media
1509 cannot be constructed in SETUP.
1510 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
1512 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
1514 * gst/rtsp-server/rtsp-client.c:
1515 client: Send setup reply once only
1516 If find_media() failed in handle_setup_request() two replies was sent.
1517 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
1519 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
1522 Automatic update of common submodule
1523 From 6b03ba7 to 865aa20
1525 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
1527 * gst/rtsp-server/rtsp-server.c:
1528 server: Emit client-connected signal earlier
1529 Emit client-connected before the client ref is given to a GSource,
1530 otherwise client-connected can be emitted after the client object has
1533 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
1535 * gst/rtsp-server/rtsp-address-pool.c:
1536 * gst/rtsp-server/rtsp-address-pool.h:
1537 * gst/rtsp-server/rtsp-stream.c:
1538 * tests/check/gst/addresspool.c:
1539 addresspool: return reason of failure
1540 Let gst_rtsp_address_pool_reserve_address() return the reason why
1541 the address could not be reserved.
1542 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
1544 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
1547 autogen.sh: Sync behaviour with other GStreamer modules
1548 Allows building from outside of tree amongst other things
1550 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
1553 Automatic update of common submodule
1554 From b613661 to 6b03ba7
1556 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
1559 Automatic update of common submodule
1560 From 74a6857 to b613661
1562 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
1565 Automatic update of common submodule
1566 From 01a7a46 to 74a6857
1568 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
1570 * gst/rtsp-server/rtsp-client.c:
1571 client: Do not read beyond end of path string
1572 If the setup was done without a control url, make sure we don't try to read the
1573 non-existing control string and crash.
1575 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1577 * gst/rtsp-server/rtsp-client.c:
1578 client: Fix RTPInfo header
1579 Refactor the method to make the content_base.
1580 Use the content-base and the control url to construct the RTPInfo
1583 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1585 * gst/rtsp-server/rtsp-client.c:
1586 client: map url to path only in describe
1587 Only map the request url to a path in the DESCRIBE method. The SDP then
1588 contains the base and control urls that should be used to SETUP/PAUSE/
1589 PLAY/TEARDOWN the media.
1591 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1593 * gst/rtsp-server/rtsp-client.c:
1594 Revert "client: map URL to path in requests"
1595 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
1596 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
1597 contains the base and control urls which are used in the SETUP, PLAY,
1598 PAUSE and TEARDOWN requests.
1600 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1602 * gst/rtsp-server/rtsp-client.c:
1603 client: map URL to path in requests
1605 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1607 * gst/rtsp-server/rtsp-client.c:
1608 * gst/rtsp-server/rtsp-mount-points.c:
1609 * gst/rtsp-server/rtsp-mount-points.h:
1610 mount-points: make vmethod to make path from uri
1611 Make a vmethod to transform an url into a path. The path is then used to lookup
1612 the factory. This makes it possible to also use other bits of the url, such as
1613 the query parameters, to locate the factory.
1615 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
1617 * gst/rtsp-server/rtsp-thread-pool.c:
1618 * gst/rtsp-server/rtsp-thread-pool.h:
1619 thread-pool: Add cleanup to wait for the threadpool to finish
1620 Also fix race condition if two threads are asking for the first
1621 thread from the thread pool at once. This would case two internal
1622 GThreadPools to be created.
1623 https://bugzilla.gnome.org/show_bug.cgi?id=707753
1625 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
1627 * gst/rtsp-server/rtsp-client.c:
1628 * tests/check/gst/client.c:
1629 client: free threadpool
1630 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1632 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
1634 * tests/check/gst/mountpoints.c:
1635 mountpoints tests: unref matched factories
1636 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1638 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
1640 * tests/check/gst/media.c:
1641 media tests: unref thread pool and caps
1642 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1644 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
1646 * gst/rtsp-server/rtsp-auth.c:
1647 * gst/rtsp-server/rtsp-media-factory.c:
1648 * gst/rtsp-server/rtsp-media.c:
1649 auth, media, media-factory: unref permissions
1650 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1652 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1654 * examples/Makefile.am:
1655 Makefile: add rule for appsrc example
1657 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1659 * examples/test-appsrc.c:
1660 tests: add appsrc example
1661 Add an example on how to use appsrc to feed the server pipeline with data.
1663 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
1665 * gst/rtsp-server/rtsp-client.c:
1666 rtsp-client: remove query part from content-base string
1667 Make sure that after the control url has been resolved, it's
1668 not a part of the query-string.
1669 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
1671 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1673 * gst/rtsp-server/rtsp-client.c:
1674 client: don't check url in response
1675 There is no url or method in the response to check
1677 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1679 * gst/rtsp-server/rtsp-client.c:
1680 * gst/rtsp-server/rtsp-client.h:
1681 Add handle-response signal for when we receive a GET_PARAMETER response
1683 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1685 * gst/rtsp-server/rtsp-server.c:
1686 Fix gst_rtsp_server_client_filter, using wrong variable type
1688 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
1690 * gst/rtsp-server/rtsp-media-factory-uri.c:
1691 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
1692 For AAC we need to check for framed=true instead of parsed=true.
1693 https://bugzilla.gnome.org/show_bug.cgi?id=701384
1695 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1697 * gst/rtsp-server/rtsp-stream.c:
1698 stream: optimize pipeline for protocols
1699 When TCP is not an allowed protocol for the stream, avoid creating the
1700 appsrc/appsink/queue and tee elements.
1702 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1704 * gst/rtsp-server/rtsp-media.c:
1705 media: set protocols on streams
1707 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1709 * gst/rtsp-server/rtsp-client.c:
1710 client: use protocols supported by stream
1712 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1714 * gst/rtsp-server/rtsp-media-factory.c:
1715 * gst/rtsp-server/rtsp-media.c:
1716 * gst/rtsp-server/rtsp-stream.c:
1717 media-factory: allow all protocols
1719 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1721 * gst/rtsp-server/rtsp-media.c:
1722 media: configure protocols in new streams
1724 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1726 * gst/rtsp-server/rtsp-stream.c:
1727 * gst/rtsp-server/rtsp-stream.h:
1728 stream: add protocols property
1730 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1732 * gst/rtsp-server/rtsp-media.c:
1733 rtsp-media: send state in "new-state" signal
1734 https://bugzilla.gnome.org/show_bug.cgi?id=705110
1736 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
1739 build: add subdir-objects to AM_INIT_AUTOMAKE
1740 Fixes warnings with automake 1.14
1741 https://bugzilla.gnome.org/show_bug.cgi?id=705350
1743 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1745 * docs/libs/gst-rtsp-server-sections.txt:
1746 * gst/rtsp-server/rtsp-client.c:
1747 * gst/rtsp-server/rtsp-server.c:
1748 * gst/rtsp-server/rtsp-server.h:
1749 server: add method to iterate clients of server
1751 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1753 * gst/rtsp-server/rtsp-media.c:
1754 * gst/rtsp-server/rtsp-media.h:
1755 Add vmethod for rtsp-media subclass to access rtpbin
1757 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1759 * gst/rtsp-server/rtsp-client.h:
1760 small documentation fix
1762 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1764 * gst/rtsp-server/rtsp-client.c:
1765 Do not take range header if range is invalid
1767 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1769 * docs/libs/gst-rtsp-server-sections.txt:
1770 * gst/rtsp-server/rtsp-media.c:
1771 media: add docs for new method
1773 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1775 * gst/rtsp-server/rtsp-media.c:
1776 * gst/rtsp-server/rtsp-media.h:
1777 Add API to rtsp-media set the pipeline's state
1779 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1781 * gst/rtsp-server/rtsp-media.c:
1782 Update current position/duration when gst_rtsp_media_get_range_string is called
1784 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1786 * examples/test-cgroups.c:
1787 tests: add some more docs
1789 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1791 * examples/test-cgroups.c:
1792 * gst/rtsp-server/Makefile.am:
1793 * gst/rtsp-server/rtsp-auth.c:
1794 * gst/rtsp-server/rtsp-auth.h:
1795 * gst/rtsp-server/rtsp-client.c:
1796 * gst/rtsp-server/rtsp-client.h:
1797 * gst/rtsp-server/rtsp-context.c:
1798 * gst/rtsp-server/rtsp-context.h:
1799 * gst/rtsp-server/rtsp-params.c:
1800 * gst/rtsp-server/rtsp-params.h:
1801 * gst/rtsp-server/rtsp-server.c:
1802 * gst/rtsp-server/rtsp-thread-pool.c:
1803 * gst/rtsp-server/rtsp-thread-pool.h:
1804 * tests/check/gst/client.c:
1805 ClientState -> Context
1806 Rename the clientstate to context and put the code in a separate file.
1808 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1810 * examples/test-auth.c:
1811 * gst/rtsp-server/rtsp-auth.c:
1812 * gst/rtsp-server/rtsp-auth.h:
1813 auth: add support for default token
1814 The default token is used when the user is not authenticated and can be used to
1815 give minimal permissions.
1817 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1819 * examples/test-auth.c:
1820 * gst/rtsp-server/rtsp-auth.c:
1821 auth: use defines when possible
1823 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1825 * gst/rtsp-server/rtsp-address-pool.c:
1826 address-pool: improve docs
1828 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1830 * gst/rtsp-server/rtsp-permissions.c:
1831 permissions: add the role to the copy
1833 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
1835 * gst/rtsp-server/rtsp-permissions.c:
1836 permissions: Also copy the roles
1838 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
1840 * gst/rtsp-server/rtsp-permissions.c:
1841 permissions: Make it build
1843 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1845 * gst/rtsp-server/rtsp-address-pool.h:
1848 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1850 * docs/libs/gst-rtsp-server-sections.txt:
1851 * gst/rtsp-server/rtsp-auth.c:
1852 * gst/rtsp-server/rtsp-auth.h:
1853 * gst/rtsp-server/rtsp-media.c:
1854 * gst/rtsp-server/rtsp-session-media.c:
1855 * gst/rtsp-server/rtsp-stream-transport.c:
1856 * gst/rtsp-server/rtsp-stream-transport.h:
1857 * gst/rtsp-server/rtsp-stream.c:
1858 * tests/check/gst/client.c:
1861 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1863 * docs/libs/gst-rtsp-server-sections.txt:
1864 * gst/rtsp-server/rtsp-address-pool.c:
1865 * gst/rtsp-server/rtsp-address-pool.h:
1866 * tests/check/gst/addresspool.c:
1867 * tests/check/gst/rtspserver.c:
1868 address-pool: cleanups
1869 Remove redundant method, improve docs.
1871 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1873 * docs/libs/gst-rtsp-server-sections.txt:
1874 * gst/rtsp-server/rtsp-auth.h:
1875 * gst/rtsp-server/rtsp-permissions.c:
1876 * gst/rtsp-server/rtsp-permissions.h:
1877 * gst/rtsp-server/rtsp-token.c:
1880 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1882 * gst/rtsp-server/rtsp-permissions.c:
1883 permissions: implement _remove_role
1885 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1887 * gst/rtsp-server/rtsp-permissions.c:
1888 permissions: update docs
1890 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1892 * tests/check/gst/client.c:
1893 tests: simplify tests
1894 Client settings are now disabled by default so we don't need an auth
1895 module to disable them.
1897 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1899 * gst/rtsp-server/rtsp-auth.c:
1900 auth: add default authorizations
1901 When no auth module is specified, use our table of defaults to look up the
1902 default value of the check instead of always allowing everything. This was
1903 we can disallow client settings by default.
1905 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1908 README: update readme
1910 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1912 * gst/rtsp-server/rtsp-thread-pool.c:
1913 * gst/rtsp-server/rtsp-thread-pool.h:
1914 thread-pool: add more docs
1916 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1918 * gst/rtsp-server/rtsp-thread-pool.c:
1919 * gst/rtsp-server/rtsp-thread-pool.h:
1920 thread-pool: fix race in thread reuse
1921 If we try to reuse a thread right after we made it stop, we end up using a
1922 stopped thread. Catch this case and only reuse threads that are not stopping.
1924 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1926 * gst/rtsp-server/rtsp-server.c:
1927 server: add small debug
1929 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1931 * tests/check/gst/client.c:
1933 Add some permissions to media so we can use the auth and enable
1936 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1938 * gst/rtsp-server/rtsp-client.c:
1939 client: support pushed context in handle_request
1940 If we already have a pushed state, reuse it and add our own things. This makes
1941 it easier to write tests.
1943 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1945 * gst/rtsp-server/rtsp-auth.c:
1946 auth: don't auth on methods
1947 Don't authorize on methods anymore but on the resources that we
1948 try to access, this is more flexible.
1949 Move the authorization checks to where they are needed and let the
1950 check return the response on error.
1952 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1954 * gst/rtsp-server/rtsp-mount-points.c:
1955 mount-points: add some debug
1957 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1959 * tests/check/gst/client.c:
1960 tests: almost fix test
1962 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1964 * gst/rtsp-server/rtsp-auth.c:
1965 * gst/rtsp-server/rtsp-auth.h:
1966 * gst/rtsp-server/rtsp-client.c:
1967 * gst/rtsp-server/rtsp-client.h:
1968 * gst/rtsp-server/rtsp-server.c:
1969 * gst/rtsp-server/rtsp-server.h:
1970 auth: let the auth module check client_settings
1971 Let the auth module decide if client settings are allowed for the
1974 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1976 * gst/rtsp-server/rtsp-token.c:
1977 * gst/rtsp-server/rtsp-token.h:
1978 token: add method to check boolean permission
1980 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1982 * examples/test-auth.c:
1983 * examples/test-cgroups.c:
1984 * gst/rtsp-server/rtsp-token.c:
1985 * gst/rtsp-server/rtsp-token.h:
1986 token: simplify token constructor
1987 Use variable arguments to make easier API.
1989 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1991 * examples/test-auth.c:
1992 * examples/test-cgroups.c:
1993 * gst/rtsp-server/rtsp-media-factory.c:
1994 * gst/rtsp-server/rtsp-media-factory.h:
1995 media-factory: add convenience API for factory
1997 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1999 * examples/test-auth.c:
2000 * examples/test-cgroups.c:
2001 * gst/rtsp-server/rtsp-permissions.c:
2002 * gst/rtsp-server/rtsp-permissions.h:
2003 permissions: simplify API a little
2004 Avoid passing GstStructure in the add_role method, use varargs instead
2005 to construct the structure behind the scenes. We can then also use the
2006 structure name as the role and simplify some more logic.
2008 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2010 * gst/rtsp-server/rtsp-auth.c:
2013 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2015 * gst/rtsp-server/rtsp-auth.c:
2016 * gst/rtsp-server/rtsp-auth.h:
2017 * gst/rtsp-server/rtsp-client.c:
2018 auth: handle unauthorized response
2019 Move handling of the unauthorized response to the auth module, it can add
2020 the appropriate headers to request authorization for the required method
2021 much better than the client.
2023 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2025 * gst/rtsp-server/rtsp-client.c:
2026 * gst/rtsp-server/rtsp-client.h:
2027 client: allow for sending any message, not only requests
2028 Change the _send_request() method to _send_message() so that we
2029 can both send requests and replies.
2031 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2033 * docs/libs/gst-rtsp-server-sections.txt:
2034 * gst/rtsp-server/rtsp-server.h:
2037 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2039 * examples/test-video.c:
2040 * gst/rtsp-server/rtsp-auth.c:
2041 * gst/rtsp-server/rtsp-auth.h:
2042 * gst/rtsp-server/rtsp-server.c:
2043 * gst/rtsp-server/rtsp-server.h:
2044 auth: move TLS handling to auth module
2045 Remove the TLS settings on the server and move it to the auth module because
2046 that is where security related bits go.
2048 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2050 * gst/rtsp-server/rtsp-client.c:
2051 * gst/rtsp-server/rtsp-client.h:
2052 client: add state push/pop
2054 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2056 * gst/rtsp-server/rtsp-client.c:
2057 * gst/rtsp-server/rtsp-client.h:
2058 client: add connection to state
2060 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2062 * gst/rtsp-server/rtsp-mount-points.c:
2063 mount-points: fix debug
2065 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2067 * tests/check/gst/media.c:
2068 tests: fix media test
2070 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2072 * gst/rtsp-server/rtsp-thread-pool.c:
2073 thread-pool: we don't require a state
2075 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2077 * gst/rtsp-server/rtsp-server.c:
2078 server: let context ref the server
2079 So that we don't risk losing the server object early anc crash.
2081 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2083 * tests/check/gst/client.c:
2084 tests: fix client test
2086 2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2089 * docs/libs/gst-rtsp-server-docs.sgml:
2090 * docs/libs/gst-rtsp-server-sections.txt:
2091 * gst/rtsp-server/rtsp-address-pool.c:
2092 * gst/rtsp-server/rtsp-auth.c:
2093 * gst/rtsp-server/rtsp-client.c:
2094 * gst/rtsp-server/rtsp-client.h:
2095 * gst/rtsp-server/rtsp-media-factory-uri.c:
2096 * gst/rtsp-server/rtsp-media-factory.c:
2097 * gst/rtsp-server/rtsp-media-factory.h:
2098 * gst/rtsp-server/rtsp-media.c:
2099 * gst/rtsp-server/rtsp-mount-points.c:
2100 * gst/rtsp-server/rtsp-params.c:
2101 * gst/rtsp-server/rtsp-permissions.c:
2102 * gst/rtsp-server/rtsp-sdp.c:
2103 * gst/rtsp-server/rtsp-server.c:
2104 * gst/rtsp-server/rtsp-server.h:
2105 * gst/rtsp-server/rtsp-session-media.c:
2106 * gst/rtsp-server/rtsp-session-pool.c:
2107 * gst/rtsp-server/rtsp-session.c:
2108 * gst/rtsp-server/rtsp-stream-transport.c:
2109 * gst/rtsp-server/rtsp-stream.c:
2110 * gst/rtsp-server/rtsp-thread-pool.c:
2111 * gst/rtsp-server/rtsp-token.c:
2114 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2116 * gst/rtsp-server/rtsp-session-pool.c:
2117 * gst/rtsp-server/rtsp-session-pool.h:
2118 session-pool: make vmethod to create a session
2119 Make a vmethod to create a sessions so that subclasses can create
2120 custom session objects
2122 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2124 * gst/rtsp-server/rtsp-auth.c:
2125 * gst/rtsp-server/rtsp-media-factory.h:
2126 * gst/rtsp-server/rtsp-media.h:
2127 * gst/rtsp-server/rtsp-mount-points.h:
2128 * gst/rtsp-server/rtsp-session-pool.h:
2129 * gst/rtsp-server/rtsp-stream.h:
2132 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2134 * docs/libs/gst-rtsp-server-docs.sgml:
2135 * docs/libs/gst-rtsp-server-sections.txt:
2136 * gst/rtsp-server/rtsp-address-pool.c:
2137 * gst/rtsp-server/rtsp-address-pool.h:
2138 * gst/rtsp-server/rtsp-auth.c:
2139 * gst/rtsp-server/rtsp-client.h:
2140 * gst/rtsp-server/rtsp-media-factory.h:
2141 * gst/rtsp-server/rtsp-media.c:
2142 * gst/rtsp-server/rtsp-media.h:
2143 * gst/rtsp-server/rtsp-permissions.c:
2144 * gst/rtsp-server/rtsp-permissions.h:
2145 * gst/rtsp-server/rtsp-server.h:
2146 * gst/rtsp-server/rtsp-session-media.c:
2147 * gst/rtsp-server/rtsp-session-media.h:
2148 * gst/rtsp-server/rtsp-session-pool.h:
2149 * gst/rtsp-server/rtsp-session.h:
2150 * gst/rtsp-server/rtsp-stream-transport.h:
2151 * gst/rtsp-server/rtsp-stream.c:
2152 * gst/rtsp-server/rtsp-thread-pool.h:
2155 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2158 * examples/Makefile.am:
2159 configure: compile cgroup example conditionally
2160 Only compile the cgroup example when we have libcgroup
2162 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2165 * examples/Makefile.am:
2166 * examples/test-cgroups.c:
2167 examples: add cgroups example
2169 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2171 * tests/check/gst/rtspserver.c:
2172 tests: fix compilation
2174 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2176 * gst/rtsp-server/rtsp-thread-pool.c:
2177 thread-pool: fix vmethod invocation
2179 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2181 * gst/rtsp-server/rtsp-thread-pool.c:
2182 * gst/rtsp-server/rtsp-thread-pool.h:
2183 thread-pool: store thread type in thread
2185 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2187 * gst/rtsp-server/rtsp-client.c:
2188 client: pass thread from pool to media _prepare
2189 Get a thread from the configured threadpool and pass it to the prepare method of
2192 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2194 * gst/rtsp-server/rtsp-media.c:
2195 * gst/rtsp-server/rtsp-media.h:
2196 media: Accept a thread in _prepare
2197 Remove out own threadpool handling and use the provided thread and
2198 maincontext for the bus messages and the state changes.
2200 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2202 * gst/rtsp-server/rtsp-server.c:
2203 server: configure client thread pool
2205 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2207 * gst/rtsp-server/rtsp-client.c:
2208 * gst/rtsp-server/rtsp-client.h:
2209 client: add method to configure thread pool
2211 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2213 * gst/rtsp-server/rtsp-client.h:
2214 * gst/rtsp-server/rtsp-server.c:
2215 * gst/rtsp-server/rtsp-server.h:
2216 server: use thread pool
2217 Use the thread pool instead of doing our own thing.
2219 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2221 * gst/rtsp-server/Makefile.am:
2222 * gst/rtsp-server/rtsp-thread-pool.c:
2223 * gst/rtsp-server/rtsp-thread-pool.h:
2224 thread-pool: add object to manage threads
2225 Add an object to manage the client and media threads.
2227 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2229 * gst/rtsp-server/rtsp-auth.c:
2230 auth: debug authorization check
2232 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2234 * gst/rtsp-server/rtsp-media.c:
2235 media: start media pipeline in context
2236 Start the media pipeline in the provided context (or our default one
2237 when NULL). This makes sure that we run the bus thread in this context and that
2238 all media threads are children of this context.
2240 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2242 * gst/rtsp-server/rtsp-media-factory.c:
2243 factory: pass permissions to media by default
2245 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2247 * examples/test-auth.c:
2248 test: add permissions to auth test
2249 Ass some permissions to the media factory in the test.
2251 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2253 * gst/rtsp-server/rtsp-auth.c:
2254 * gst/rtsp-server/rtsp-auth.h:
2255 * gst/rtsp-server/rtsp-client.c:
2256 auth: simplify auth checks
2257 Remove client from methods, it's now in the state
2258 Perform the check specified by the string, use the information from the
2259 thread local context.
2261 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2263 * gst/rtsp-server/rtsp-client.c:
2264 * gst/rtsp-server/rtsp-client.h:
2265 client: add state to current thread
2266 Add the client to the ClientState object.
2267 Place the ClientState on the current thread.
2269 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2271 * gst/rtsp-server/rtsp-media-factory.c:
2272 * gst/rtsp-server/rtsp-media-factory.h:
2273 * gst/rtsp-server/rtsp-media.c:
2274 * gst/rtsp-server/rtsp-media.h:
2275 media: make it possible to set permissions
2276 Make it possible to set permissions on media and media factory objects
2278 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2280 * gst/rtsp-server/Makefile.am:
2281 * gst/rtsp-server/rtsp-permissions.c:
2282 * gst/rtsp-server/rtsp-permissions.h:
2283 permissions: add permissions object
2284 Add a mini object to store permissions based on a role.
2286 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2288 * examples/test-auth.c:
2289 * gst/rtsp-server/rtsp-auth.c:
2290 * gst/rtsp-server/rtsp-auth.h:
2291 * gst/rtsp-server/rtsp-client.c:
2292 auth: add auth checks
2293 Add an enum with auth checks and implement the checks in the auth object.
2294 Perform the checks from the client.
2296 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2298 * examples/test-auth.c:
2299 * gst/rtsp-server/rtsp-auth.c:
2300 * gst/rtsp-server/rtsp-auth.h:
2301 * gst/rtsp-server/rtsp-client.h:
2302 auth: use the token after authentication
2303 After we authenticated a user, keep the Token around in the state.
2305 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2307 * gst/rtsp-server/rtsp-client.c:
2308 * gst/rtsp-server/rtsp-media.c:
2309 * gst/rtsp-server/rtsp-media.h:
2310 * tests/check/gst/media.c:
2311 media: add optional context for bus messages
2312 Add an optional mainloop to _prepare that will handle the bus messages instead
2313 of always using the shared mainloop.
2315 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2317 * gst/rtsp-server/Makefile.am:
2318 * gst/rtsp-server/rtsp-token.c:
2319 * gst/rtsp-server/rtsp-token.h:
2320 token: add authorization token
2321 Add a simply miniobject that contains the authorizations. The object contains a
2322 GstStructure that hold all authorization fields. When a user is authenticated,
2323 the auth module will create a Token for the user. The token is then used to
2324 check what operations the user is allowed to do and various other configuration
2327 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2329 * examples/test-auth.c:
2330 * gst/rtsp-server/rtsp-auth.c:
2331 * gst/rtsp-server/rtsp-auth.h:
2332 * gst/rtsp-server/rtsp-client.c:
2333 * gst/rtsp-server/rtsp-client.h:
2334 * gst/rtsp-server/rtsp-media-factory.c:
2335 * gst/rtsp-server/rtsp-media-factory.h:
2336 * gst/rtsp-server/rtsp-media.c:
2337 * gst/rtsp-server/rtsp-media.h:
2338 auth: remove auth from media and factory
2339 Remove the auth object from media and factory. We want to have the RTSPClient
2340 authenticate and authorize resources, there is no need to place another auth
2341 manager on the media/factory.
2343 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2345 * examples/test-auth.c:
2346 * gst/rtsp-server/rtsp-auth.c:
2347 * gst/rtsp-server/rtsp-auth.h:
2348 * gst/rtsp-server/rtsp-client.h:
2349 auth: add support for multiple basic auth tokens
2350 Make it possible to add multiple basic authorisation tokens to one authorization
2351 object. Associate with each token an authorization group that will define what
2352 capabilities are allowed.
2354 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2356 * gst/rtsp-server/rtsp-client.c:
2357 client: error out on non-aggregate control
2358 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
2360 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2362 * gst/rtsp-server/rtsp-client.c:
2363 client: rework setup request a little
2364 Cache the media in DESCRIBE based on the longest matching path with the uri
2365 that we can find in the mount points.
2366 Rework the setup request a little to get the media from the session or from
2367 the longest matching path, this way we can derive the control string as
2368 everything after the path instead of hardcoding it.
2369 Find the stream based on the control string and only open a session when all
2372 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2374 * gst/rtsp-server/rtsp-media.c:
2375 * gst/rtsp-server/rtsp-media.h:
2376 media: add method to find a stream by control url
2378 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2380 * gst/rtsp-server/rtsp-stream.c:
2381 * gst/rtsp-server/rtsp-stream.h:
2382 stream: add method to check control url of stream
2384 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2386 * gst/rtsp-server/rtsp-client.c:
2387 * gst/rtsp-server/rtsp-session-media.c:
2388 * gst/rtsp-server/rtsp-session-media.h:
2389 * gst/rtsp-server/rtsp-session.c:
2390 * gst/rtsp-server/rtsp-session.h:
2391 session: use path matching for session media
2392 Use a path string instead of a uri to lookup session media in the sessions. Also
2393 use path matching to find the largest possible path that matches.
2395 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2397 * gst/rtsp-server/rtsp-client.c:
2398 * gst/rtsp-server/rtsp-mount-points.c:
2399 * gst/rtsp-server/rtsp-mount-points.h:
2400 * tests/check/gst/mountpoints.c:
2401 mount-points: remove useless vmethod
2402 Making lookups in the mount points should not be done with a URL, if there is a
2403 mapping to be done from URL to mount points, we'll need to do it somewhere
2406 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2408 * gst/rtsp-server/rtsp-mount-points.c:
2409 * gst/rtsp-server/rtsp-mount-points.h:
2410 * tests/check/gst/mountpoints.c:
2411 mount-points: improve mount point searching
2412 Use a GSequence to keep track of the mount points.
2413 Match a URL to the longest matching registered mount point. This should be the
2414 URL to perform aggreagate control and the remainder is the stream specific
2416 Add some unit tests for this.
2418 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
2420 * gst/rtsp-server/Makefile.am:
2421 rtsp-server: Allow building of static library
2423 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2425 * tests/check/gst/mediafactory.c:
2426 tests: fix compilation
2428 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2430 * gst/rtsp-server/rtsp-sdp.c:
2431 sdp: get control string from stream
2432 Use the control string as configured in the stream.
2434 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2436 * gst/rtsp-server/rtsp-stream.c:
2437 * gst/rtsp-server/rtsp-stream.h:
2438 stream: add methods and property to set control string
2440 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2442 * gst/rtsp-server/rtsp-client.c:
2444 Rename variables for clarity
2445 Keep media in state when we can
2447 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2449 * gst/rtsp-server/rtsp-client.c:
2450 * gst/rtsp-server/rtsp-stream.c:
2451 * gst/rtsp-server/rtsp-stream.h:
2452 stream: add more support for IPv6
2453 Rename _get_address to _get_multicast_address in GstRTSPStream to
2454 make it clear that this function only deals with multicast.
2455 Make it possible to have both an IPv4 and IPv6 multicast address on
2456 a stream. Give the client an IPv4 or IPv6 address depending on the
2457 address it used to connect to the server.
2458 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
2460 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2462 * gst/rtsp-server/rtsp-client.c:
2465 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2467 * gst/rtsp-server/rtsp-stream.c:
2468 stream: handle failed port allocation
2469 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
2470 can't allocate any family at all. Also keep track of what port families we
2472 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
2474 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2476 * gst/rtsp-server/rtsp-stream.c:
2477 stream: improve docs
2479 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2481 * gst/rtsp-server/rtsp-stream-transport.c:
2482 stream-transport: remove old if 0 block
2484 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
2486 * tests/check/gst/client.c:
2488 gst_rtsp_client_get_uri() has been removed
2489 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
2491 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2493 * gst/rtsp-server/rtsp-client.c:
2494 * gst/rtsp-server/rtsp-client.h:
2495 client: add method to filter managed sessions
2496 Add a method to filter the sessions managed by this client connection.
2497 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
2499 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2501 * gst/rtsp-server/rtsp-client.c:
2502 * gst/rtsp-server/rtsp-client.h:
2503 client: remove _get_uri() method
2504 Remove the get_uri() method on the client. A client has no uri, the uri
2505 property is an internal property to manage the last cached media for
2508 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2510 * gst/rtsp-server/rtsp-media-factory.h:
2511 media-factory: fix typo
2513 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2515 * gst/rtsp-server/rtsp-media.c:
2516 rtsp-media: Do not leak the query in default_query_stop
2517 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
2519 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2521 * gst/rtsp-server/rtsp-media.c:
2522 media: don't unlock when conversion fails
2523 Don't unlock the state lock when conversion fails because it was not locked.
2525 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2527 * gst/rtsp-server/rtsp-media.c:
2528 * gst/rtsp-server/rtsp-media.h:
2529 Add query_position and query_stop vmethods to rtsp-media
2531 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2533 * gst/rtsp-server/rtsp-media.c:
2534 Fix typo in property install for rtsp-media's time-provider
2536 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2538 * gst/rtsp-server/rtsp-client.c:
2539 * gst/rtsp-server/rtsp-client.h:
2540 client: clean some variables
2541 Clean some variables and add some guards to _send_request()
2543 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2545 * gst/rtsp-server/rtsp-client.c:
2546 * gst/rtsp-server/rtsp-client.h:
2547 Add gst_rtsp_client_send_request API
2548 This makes it possible to send arbitrary messages to a client, such as
2549 SET_PARAMETER or GET_PARAMETER
2551 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2553 * gst/rtsp-server/rtsp-media.c:
2554 * gst/rtsp-server/rtsp-media.h:
2555 media: add _get_element() method
2556 Add method to get the element used when creating the media.
2557 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
2559 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2561 * gst/rtsp-server/rtsp-media.c:
2564 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2566 * gst/rtsp-server/rtsp-stream.c:
2567 * gst/rtsp-server/rtsp-stream.h:
2568 stream: allow access to the rtp session
2569 https://bugzilla.gnome.org/show_bug.cgi?id=703004
2571 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
2573 * gst/rtsp-server/rtsp-stream.c:
2574 * gst/rtsp-server/rtsp-stream.h:
2575 dscp qos support in gst-rtsp-stream
2576 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
2578 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2580 * tests/check/gst/rtspserver.c:
2582 Actually do what the comment says. Also keep the old code around, not sure what
2583 should happen when you get a 454 from a TEARDOWN, does it close the connection?
2584 it currently doesn't.
2586 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2588 * gst/rtsp-server/rtsp-client.c:
2589 client: also watch newly created session
2590 When we newly created a session, start watching it immediately instead of
2591 on the next request.
2593 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
2595 * tests/check/gst/client.c:
2596 tests: add unit test for new-session
2597 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
2599 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2601 * gst/rtsp-server/rtsp-client.c:
2602 client: emit new-session when new session is created
2603 Only emit new-session when we created a new session for a client, not when a
2604 client picked up a previous session.
2605 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
2607 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
2609 * gst/rtsp-server/rtsp-client.c:
2610 client: handle asterisk as path in requests
2611 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
2613 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2615 * gst/rtsp-server/rtsp-media.c:
2616 media: handle segment query format mismatch
2617 It's possible that the segment query returns with a different format than what
2618 we asked for, handle this case also.
2620 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
2622 * gst/rtsp-server/rtsp-media.c:
2623 media: use segment stop in collect_media_stats
2624 Use segment stop instead of duration as range end point.
2625 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
2627 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2629 * gst/rtsp-server/rtsp-media.c:
2630 * tests/check/gst/media.c:
2631 rtsp-media: Do not leak the element in take_pipeline
2632 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
2634 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
2636 * gst/rtsp-server/rtsp-client.c:
2637 * gst/rtsp-server/rtsp-client.h:
2638 rtsp-client: Make configure_client_transport virtual
2639 This patch makes configure_client_transport virtual. The functionality is
2640 needed to handle some weird clients sending multicast transport settings as url
2642 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
2644 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2646 * gst/rtsp-server/rtsp-client.c:
2647 * gst/rtsp-server/rtsp-client.h:
2648 rtsp-client: Make param_set and param_get virtual
2649 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
2651 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
2653 * gst/rtsp-server/rtsp-client.c:
2654 * gst/rtsp-server/rtsp-media.c:
2655 * gst/rtsp-server/rtsp-media.h:
2656 media: convert_range replaces get_range_times
2657 get_range_times worked for handling UTC ranges for seeks, but we also
2658 need to convert back from NPT to the requested unit in
2659 get_range_string. convert_range is now used for both.
2660 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
2662 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2664 * gst/rtsp-server/rtsp-client.c:
2665 * gst/rtsp-server/rtsp-sdp.c:
2666 * gst/rtsp-server/rtsp-sdp.h:
2667 sdp: cleanup sdp info
2668 We don't need to pass the proto, we can more easily check a boolean.
2669 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
2671 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
2673 * gst/rtsp-server/rtsp-sdp.c:
2674 use 0.0.0.0 or :: for c= line instead of server address
2676 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
2678 * gst/rtsp-server/rtsp-client.c:
2679 use local address, not remote, in SDP
2680 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
2682 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2685 Automatic update of common submodule
2686 From 098c0d7 to 01a7a46
2688 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
2690 * gst/rtsp-server/rtsp-media.c:
2691 * gst/rtsp-server/rtsp-media.h:
2692 media: possibility to override range time conversion
2693 Make it possible to override the conversion from GstRTSPTimeRange to
2694 GstClockTimes, that is done before seeking on the media
2695 pipeline. Overriding can be useful for UTC ranges, where the default
2696 conversion gives nanoseconds since 1900.
2697 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
2699 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2701 * gst/rtsp-server/rtsp-server.c:
2702 * gst/rtsp-server/rtsp-server.h:
2703 rtsp-server: Expose the use_client_settings API
2704 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
2706 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
2708 * gst/rtsp-server/rtsp-client.c:
2709 * gst/rtsp-server/rtsp-stream.c:
2710 * gst/rtsp-server/rtsp-stream.h:
2711 rtspstream: handle both ipv4 and ipv6 clients
2712 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
2714 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2716 * gst/rtsp-server/rtsp-sdp.c:
2717 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
2718 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
2719 We already have a way to place extra attributes in the SDP by using a string
2720 property with prefix x- or a- in the caps.
2722 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2724 * gst/rtsp-server/rtsp-sdp.c:
2725 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
2726 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
2727 We already have a way to place extra attributes in the SDP, just make a string
2728 property in the payloader with a- or x- prefix.
2730 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2732 * gst/rtsp-server/rtsp-sdp.c:
2733 rtsp: place a- and x- properties as attributes
2734 application/x-rtp has properties with a- and x- prefixes that should be
2735 placed as attributes in the SDP for the media instead of being added to the
2738 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2740 * examples/Makefile.am:
2741 * examples/test-video.c:
2742 example: add TLS example
2744 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2746 * gst/rtsp-server/rtsp-server.c:
2747 * gst/rtsp-server/rtsp-server.h:
2748 server: add support for TLS
2749 Add methods to set and get a TLS certificate.
2750 Add vmethod to configure a new connection. By default, configure the TLS
2751 certificate in a new connection if needed.
2753 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2755 * gst/rtsp-server/rtsp-server.c:
2756 * gst/rtsp-server/rtsp-server.h:
2757 server: remove accept_client vmethod
2758 This vmethod is not very useful so remove it.
2760 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2762 * gst/rtsp-server/rtsp-server.c:
2763 server: don't crash on NULL GError
2765 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
2767 * gst/rtsp-server/rtsp-session-pool.c:
2768 rtsp-session-pool: corrected session timeout detection
2769 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
2771 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2773 * gst/rtsp-server/rtsp-client.c:
2774 client: improve debug
2776 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2778 * gst/rtsp-server/rtsp-client.c:
2779 * gst/rtsp-server/rtsp-client.h:
2780 * gst/rtsp-server/rtsp-server.c:
2781 server: refactor connection setup
2782 Let the server accept the socket connection and construct a GstRTSPConnection
2783 from it. Remove the code from the client and let the client only deal with
2784 a fully configure GstRTSPConnection object.
2785 We will need this later when the server will configure the connection for
2788 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2790 * gst/rtsp-server/rtsp-stream.c:
2791 stream: keep the transport object alive
2792 Keep the transport object alive while we have it as qdata on the
2795 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
2797 * gst/rtsp-server/rtsp-client.c:
2798 * gst/rtsp-server/rtsp-server.c:
2799 rtsp-server: Do not crash on nmapping of server
2800 * generate error when gst_rtsp_connection_accept fails
2801 * do not stop accepting incoming connections because
2802 accepting a client fails
2803 https://bugzilla.gnome.org/show_bug.cgi?id=701072
2805 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
2807 * gst/rtsp-server/rtsp-client.c:
2808 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
2809 https://bugzilla.gnome.org/show_bug.cgi?id=700953
2811 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
2813 * gst/rtsp-server/rtsp-sdp.c:
2814 rtsp-sdp: Parse framerate caps field and set SDP attribute
2815 The SDP attribute and its format is described in RFC4566.
2816 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2818 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
2820 * gst/rtsp-server/rtsp-sdp.c:
2821 rtsp-sdp: Parse width/height from caps and set SDP attribute
2822 The SDP attribute and its format is described in RFC6064.
2823 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2825 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
2827 * gst/rtsp-server/rtsp-sdp.c:
2828 * tests/check/gst/client.c:
2829 rtsp-sdp: add bandwidth line
2830 https://bugzilla.gnome.org/show_bug.cgi?id=699220
2832 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2835 Automatic update of common submodule
2836 From 5edcd85 to 098c0d7
2838 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2840 * tests/check/gst/media.c:
2841 tests: add dynamic payloader prepare/unprepare check
2843 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2845 * gst/rtsp-server/rtsp-media.c:
2846 media: release lock when removing fakesink
2848 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2850 * gst/rtsp-server/rtsp-stream.c:
2851 stream: set elements to NULL before removing
2852 When removing a stream, set the elements to NULL first. This avoids
2853 element-is-not-in-NULL-state errors when we dispose the elements.
2855 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
2858 Automatic update of common submodule
2859 From 3cb3d3c to 5edcd85
2861 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2863 * gst/rtsp-server/rtsp-media.c:
2864 * gst/rtsp-server/rtsp-media.h:
2865 media: listen to pad-removed signals
2866 Listen to the pad-removed signal and remove the stream associated with the
2868 Add signal to be notified of the removed pad.
2869 Remove the fakesink in unprepare()
2870 Fix signatures of the signal methods
2872 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2874 * examples/test-sdp.c:
2875 tests: add example of reusable pipelines
2877 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2879 * gst/rtsp-server/rtsp-stream.c:
2880 * gst/rtsp-server/rtsp-stream.h:
2881 stream: add method to get the srcpad
2883 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2885 * tests/check/gst/media.c:
2886 check: add media prepare/unprepare test
2887 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2889 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
2891 * gst/rtsp-server/rtsp-media.c:
2892 media: disconnect from signal handlers in unprepare()
2893 We connected to the pad-added and no-more-pads signals in prepare() so
2894 we need to disconnect from them in unprepare().
2895 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2897 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2899 * gst/rtsp-server/rtsp-media.c:
2900 media: don't free streams array
2901 Don't free the streams array in the unprepare() method, they were not
2903 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2905 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
2907 * gst/rtsp-server/rtsp-media.c:
2908 media: don't unref the pipeline in unprepare
2909 Unprepare() should undo what prepare() does. Because the pipeline is
2910 not created in prepare(), we should not unref it in unprepare()
2912 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
2914 * gst/rtsp-server/rtsp-stream.c:
2915 stream: clear session and caps for reuse
2916 Set the session and caps to NULL after unref otherwise we might unref
2918 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2920 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
2922 * gst/rtsp-server/rtsp-client.c:
2923 client: send out teardown signal before tearing down
2924 The advantage is that in the signal handler you get direct access to
2925 information about what streams are about to get torn down (in the
2926 GstRTSPClientState).
2927 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
2929 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
2931 * gst/rtsp-server/rtsp-client.c:
2932 * gst/rtsp-server/rtsp-client.h:
2933 client: expose connection
2934 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
2936 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
2939 Automatic update of common submodule
2940 From aed87ae to 3cb3d3c
2942 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2944 * gst/rtsp-server/rtsp-media.c:
2945 * gst/rtsp-server/rtsp-media.h:
2946 * gst/rtsp-server/rtsp-session-media.c:
2947 * gst/rtsp-server/rtsp-session-media.h:
2948 media: add method to get the base_time of the pipeline
2949 Together with a shared clock, this base-time could eventually be sent to
2950 the client so that it can reconstruct the exact running-time of the clock
2953 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2955 * gst/rtsp-server/Makefile.am:
2956 * gst/rtsp-server/rtsp-media.c:
2957 * gst/rtsp-server/rtsp-media.h:
2958 * gst/rtsp-server/rtsp-sdp.c:
2959 media: add GstNetTimeProvider support
2960 Add a property to let the media provide a GstNetTimeProvider for its clock.
2961 Make methods to get the clock and nettimeprovider
2962 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
2963 provider and also the current time of the clock. This should make it possible
2964 for (GStreamer) clients to slave their clock to the server clock.
2966 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
2969 Automatic update of common submodule
2970 From 04c7a1e to aed87ae
2972 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2974 * gst/rtsp-server/rtsp-media.c:
2975 media: wait for buffering to complete
2976 Wait for buffering to complete before changing the state to the target state.
2978 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2980 * gst/rtsp-server/rtsp-media.c:
2981 media: small cleanup
2983 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
2985 * tests/check/gst/rtspserver.c:
2986 tests: remove extra unref in test_setup_non_existing_stream
2987 The unref is not needed anymore, teardown runs without it.
2988 https://bugzilla.gnome.org/show_bug.cgi?id=696542
2990 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
2992 * tests/check/gst/rtspserver.c:
2993 tests: GSocketService cleanup in test_bind_already_in_use
2994 Use g_socket_service_stop so the rtspserver test stops listening for
2995 incoming connections in test_bind_already_in_use.
2996 https://bugzilla.gnome.org/show_bug.cgi?id=696541
2998 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
3000 * gst/rtsp-server/rtsp-media-factory.c:
3001 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
3002 Instead use a GWeakRef which is safe to use
3003 This is a known GLib bug, see:
3004 https://bugzilla.gnome.org/show_bug.cgi?id=667145
3006 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
3008 * gst/rtsp-server/rtsp-client.c:
3009 * gst/rtsp-server/rtsp-media.c:
3010 * gst/rtsp-server/rtsp-media.h:
3011 * gst/rtsp-server/rtsp-sdp.c:
3012 * tests/check/gst/media.c:
3013 * tests/check/gst/rtspserver.c:
3014 rtsp-media/client: Reply to PLAY request with same type of Range
3015 Remember the type of Range from the PLAY request and use the same type for
3018 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
3020 * gst/rtsp-server/rtsp-client.c:
3021 * gst/rtsp-server/rtsp-client.h:
3022 * tests/check/gst/client.c:
3023 rtsp-client: expose uri
3025 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
3027 * tests/check/gst/mediafactory.c:
3028 tests: Hold ref while creating second media
3029 To test if the media aren't shared, make sure we keep the first one while creating a second
3030 otherwise the same memory address may be reused.
3032 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
3035 configure: remove out-of-date comment
3037 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
3040 .gitignore: ignore more build files
3042 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
3044 * tests/check/Makefile.am:
3045 tests: use right _LIBS variable for gst-plugins-base libs
3047 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3049 * tests/check/Makefile.am:
3050 check: add librtp to libs
3052 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
3054 * tests/check/gst/rtspserver.c:
3055 tests: Add test to check selecting a port the server will send from
3057 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
3059 * tests/check/gst/rtspserver.c:
3060 tests: Make sure packets are actually received
3062 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
3064 * gst/rtsp-server/rtsp-stream.c:
3065 stream: Select unicast address from pool if appropriate
3067 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
3069 * gst/rtsp-server/rtsp-stream.c:
3070 stream: Properties are always there in Gst 1.0
3072 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
3074 * tests/check/gst/addresspool.c:
3075 tests: Add tests for unicast addresses in pool
3077 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
3079 * gst/rtsp-server/rtsp-address-pool.c:
3080 * tests/check/gst/addresspool.c:
3081 address-pool: Verify that multicast addresses are used for multicast and vice-versa
3083 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
3085 * docs/libs/gst-rtsp-server-sections.txt:
3086 * gst/rtsp-server/rtsp-address-pool.c:
3087 * gst/rtsp-server/rtsp-address-pool.h:
3088 * gst/rtsp-server/rtsp-stream.c:
3089 * tests/check/gst/addresspool.c:
3090 address-pool: Add unicast addresses
3092 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
3095 * gst/rtsp-server/rtsp-server.c:
3096 * tests/check/gst/rtspserver.c:
3097 rtsp-server: Limit the number of threads per server instance
3098 If we exceed the maximum, just round robin the clients over the existing
3101 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
3103 * gst/rtsp-server/rtsp-server.c:
3104 rtsp-server: No need to store the GMainContext in the client context
3106 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
3108 * tests/check/gst/rtspserver.c:
3109 tests: Add test for client disconnection
3111 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
3113 * tests/check/gst/rtspserver.c:
3114 tests: Test client and session timeouts with multiple threads
3116 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
3118 * gst/rtsp-server/rtsp-address-pool.c:
3119 * gst/rtsp-server/rtsp-auth.c:
3120 * gst/rtsp-server/rtsp-client.c:
3121 * gst/rtsp-server/rtsp-media-factory-uri.c:
3122 * gst/rtsp-server/rtsp-media-factory.c:
3123 * gst/rtsp-server/rtsp-media.c:
3124 * gst/rtsp-server/rtsp-mount-points.c:
3125 * gst/rtsp-server/rtsp-server.c:
3126 * gst/rtsp-server/rtsp-session-media.c:
3127 * gst/rtsp-server/rtsp-session-pool.c:
3128 * gst/rtsp-server/rtsp-session.c:
3129 Document locking and its order
3131 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
3133 * tests/check/gst/rtspserver.c:
3134 tests: Test that slow DESCRIBE don't block other clients
3136 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
3138 * tests/check/gst/client.c:
3139 tests: Add tests for client-requested multicast address
3141 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
3143 * docs/libs/gst-rtsp-server-sections.txt:
3144 docs: Put the various functions in the right sections
3146 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
3148 * docs/libs/gst-rtsp-server-docs.sgml:
3149 * docs/libs/gst-rtsp-server-sections.txt:
3150 * gst/rtsp-server/rtsp-address-pool.c:
3151 * gst/rtsp-server/rtsp-address-pool.h:
3152 docs: Generate docs for GstRTSPAddressPool
3154 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
3156 * gst/rtsp-server/rtsp-client.c:
3157 * gst/rtsp-server/rtsp-stream.c:
3158 * gst/rtsp-server/rtsp-stream.h:
3159 client: Check client provided addresses against the address pool
3161 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
3163 * gst/rtsp-server/rtsp-address-pool.c:
3164 * gst/rtsp-server/rtsp-address-pool.h:
3165 * tests/check/gst/addresspool.c:
3166 address-pool: Add API to request a specific address from the pool
3167 Also add relevant unit tests.
3169 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
3171 * tests/check/gst/mediafactory.c:
3172 tests: Check the passing around of a RTSPAddressPool
3173 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
3174 way down to the stream.
3176 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
3178 * tests/check/gst/addresspool.c:
3179 tests: Add more tests for the address pool
3181 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
3183 * gst/rtsp-server/rtsp-address-pool.c:
3184 address-pool: Fix off by one error
3185 When splitting a port range, the port after a skip is not part of range.
3187 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
3190 Automatic update of common submodule
3191 From 2de221c to 04c7a1e
3193 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
3196 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
3197 AM_CONFIG_HEADER was removed in automake 1.13
3198 https://bugzilla.gnome.org/show_bug.cgi?id=693368
3200 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
3203 Automatic update of common submodule
3204 From a942293 to 2de221c
3206 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3208 * gst/rtsp-server/rtsp-client.c:
3209 client: make sure the watch exists while sending data
3210 Protect the send_func with a lock. This allows us to wait for sending
3211 to complete before changing the send_func and user_data. We add an
3212 extra ref to the watch to make sure that it remains valid during
3214 When closing the connection, set the send_func to NULL
3215 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
3217 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3219 * tests/check/Makefile.am:
3220 tests: use GST_*_1_0 environment variables everywhere
3221 The _1_0 suffixed environment variables override the
3222 non-suffixed ones, so if we're in an environment that
3223 sets the _1_0 suffixed ones, such as jhbuild, we need
3224 to set those to make sure ours actually always get
3227 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3230 Automatic update of common submodule
3231 From acb04d9 to a942293
3233 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3235 * gst/rtsp-server/rtsp-client.c:
3236 rtsp-client: set the client backlog
3237 Set the client backlog to a reasonable default
3239 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
3241 * gst/rtsp-server/rtsp-media.c:
3242 rtsp-media: Make the element a constructor parameter
3243 https://bugzilla.gnome.org/show_bug.cgi?id=689594
3245 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3247 * docs/libs/Makefile.am:
3248 docs: Link with gcov library when gcov is enabled
3249 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
3251 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3253 * gst/rtsp-server/rtsp-media.c:
3254 media: match prepare with unprepare
3255 Really unprepare when there were an equal amount of prepare calls.
3257 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3259 * gst/rtsp-server/rtsp-media.c:
3260 media: media has to be unprepared in finalize
3261 Because unprepare takes away the last ref on the media.
3263 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3265 * gst/rtsp-server/rtsp-client.c:
3266 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
3267 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
3268 We can't use the refcount to trigger unprepare because it is the unprepare call
3269 that removes the last refcount after all messages are consumed. What we should
3270 probably do is make a prepared refcount and only unprepare when the refcount
3273 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3275 * gst/rtsp-server/rtsp-media.c:
3276 media: let the source unref the last media ref
3277 the last ref to the media is held by the source so we don't need to add more ref
3278 and unrefs, we simply destroy the media when the source is gone.
3280 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3282 * gst/rtsp-server/rtsp-media.c:
3283 media: improve debug
3285 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3287 * gst/rtsp-server/rtsp-media.c:
3289 Make sure we are in the right state when collecting the position and duration.
3290 Only make ourselves PREPARED when we were previously PREPARING.
3292 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3294 * gst/rtsp-server/rtsp-media.c:
3295 media: use g_object_ref/unref for GObjects
3297 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
3299 * gst/rtsp-server/rtsp-client.c:
3300 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
3301 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
3302 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
3303 isn't being used anymore.
3305 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
3307 * gst/rtsp-server/rtsp-media.c:
3308 Fix compiler warning
3310 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
3312 * gst/rtsp-server/rtsp-media-factory-uri.c:
3313 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
3315 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3317 * gst/rtsp-server/rtsp-session-media.h:
3320 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3322 * gst/rtsp-server/rtsp-media.c:
3323 * tests/check/gst/media.c:
3324 media: avoid element leak
3326 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3328 * gst/rtsp-server/rtsp-media.c:
3329 media: require an element in media constructor
3331 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3333 * gst/rtsp-server/rtsp-client.c:
3334 Revert "client: TEARDOWN brings that state to Init again"
3335 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
3336 The object is already disposed, there is no point in setting the state.
3338 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3340 * gst/rtsp-server/rtsp-client.c:
3341 client: TEARDOWN brings that state to Init again
3343 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3345 * docs/libs/gst-rtsp-server-sections.txt:
3346 * examples/test-auth.c:
3347 * gst/rtsp-server/rtsp-auth.c:
3348 * gst/rtsp-server/rtsp-auth.h:
3349 * gst/rtsp-server/rtsp-client.c:
3350 * gst/rtsp-server/rtsp-client.h:
3351 * gst/rtsp-server/rtsp-media-factory-uri.c:
3352 * gst/rtsp-server/rtsp-media-factory-uri.h:
3353 * gst/rtsp-server/rtsp-media-factory.c:
3354 * gst/rtsp-server/rtsp-media-factory.h:
3355 * gst/rtsp-server/rtsp-media.c:
3356 * gst/rtsp-server/rtsp-media.h:
3357 * gst/rtsp-server/rtsp-mount-points.c:
3358 * gst/rtsp-server/rtsp-mount-points.h:
3359 * gst/rtsp-server/rtsp-sdp.c:
3360 * gst/rtsp-server/rtsp-server.c:
3361 * gst/rtsp-server/rtsp-server.h:
3362 * gst/rtsp-server/rtsp-session-media.c:
3363 * gst/rtsp-server/rtsp-session-media.h:
3364 * gst/rtsp-server/rtsp-session-pool.c:
3365 * gst/rtsp-server/rtsp-session-pool.h:
3366 * gst/rtsp-server/rtsp-session.c:
3367 * gst/rtsp-server/rtsp-session.h:
3368 * gst/rtsp-server/rtsp-stream-transport.c:
3369 * gst/rtsp-server/rtsp-stream-transport.h:
3370 * gst/rtsp-server/rtsp-stream.c:
3371 * gst/rtsp-server/rtsp-stream.h:
3372 * tests/check/gst/media.c:
3373 rtsp: make object details private
3374 Make all object details private
3375 Add methods to access private bits
3377 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3379 * tests/check/Makefile.am:
3380 * tests/check/gst/media.c:
3381 tests: add media tests
3383 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3385 * gst/rtsp-server/rtsp-media.c:
3386 media: check if prepared for some methods
3387 Check that the media object is prepared before doing seek and getting the
3388 current position etc.
3389 Add some g_return checks.
3391 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3393 * tests/check/Makefile.am:
3394 * tests/check/gst/mediafactory.c:
3395 tests: add mediafactory test
3397 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3399 * gst/rtsp-server/rtsp-stream.c:
3400 stream: improve debug
3402 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3404 * gst/rtsp-server/rtsp-media.c:
3405 * gst/rtsp-server/rtsp-media.h:
3406 media: unref pipeline in finalize to avoid leaking it
3408 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3410 * gst/rtsp-server/rtsp-media-factory-uri.c:
3411 * gst/rtsp-server/rtsp-media.c:
3412 rtsp: use gst_object_unref on GstObjects
3414 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3416 * gst/rtsp-server/rtsp-media-factory.c:
3417 media-factory: require an url
3419 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3421 * examples/test-uri.c:
3422 examples: fix include
3424 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3426 * gst/rtsp-server/rtsp-server.h:
3427 server: remove unused include
3429 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3431 * tests/check/Makefile.am:
3432 * tests/check/gst/mountpoints.c:
3433 tests: add test for mountpoints
3435 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3437 * gst/rtsp-server/rtsp-client.c:
3438 client: fix factory leak
3439 Keep the factory in the state object only for authorization checks and make
3440 sure we unref it on failure. Also don't keep invalid objects in the state
3443 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3445 * gst/rtsp-server/rtsp-mount-points.c:
3446 mounts: add g_return_if guards
3448 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3450 * tests/check/gst/client.c:
3451 tests: add more tests
3453 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3455 * gst/rtsp-server/rtsp-client.c:
3456 client: improve debug
3458 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3460 * gst/rtsp-server/rtsp-client.c:
3461 client: improve debug and fix leaks
3462 Cleanup the uri and session when there is a bad request.
3464 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3469 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3471 * tests/check/gst/client.c:
3472 test: add test for session in options request
3474 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3476 * gst/rtsp-server/rtsp-client.c:
3477 client: use 454 when session can't be found
3478 We should use 454 when a session can't be found because there was no session
3479 pool configured in the server. This is not a server configuration problem
3480 because the server on which the request is done might not be the same one that
3481 will keep the sessions for us and so it does not need to support sessions.
3483 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3485 * gst/rtsp-server/rtsp-client.c:
3486 client: only free connection when there is one
3487 It's possible that the client doesn't have a connection when we try to free it.
3489 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3491 * tests/check/Makefile.am:
3492 * tests/check/gst/client.c:
3493 tests: add unit test for the client object
3495 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3497 * gst/rtsp-server/rtsp-client.c:
3498 client: small cleanup
3500 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3502 * gst/rtsp-server/rtsp-client.h:
3503 client: remove unused include
3505 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3507 * gst/rtsp-server/rtsp-client.c:
3508 client: fix compilation
3510 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3512 * gst/rtsp-server/rtsp-client.c:
3513 client: call destroy without the lock
3515 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3517 * gst/rtsp-server/rtsp-client.c:
3518 * gst/rtsp-server/rtsp-client.h:
3519 client: make the client usable without a socket
3520 Make a method to let the client handle a message and a callback when the client
3521 wants us to send a response message back. This makes it possible to also use the
3522 client object without the sockets, which should make it easier to test.
3524 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3526 * gst/rtsp-server/rtsp-client.c:
3527 * gst/rtsp-server/rtsp-client.h:
3528 client: small cleanup
3530 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3532 * docs/libs/gst-rtsp-server-sections.txt:
3533 * gst/rtsp-server/rtsp-client.c:
3534 * gst/rtsp-server/rtsp-client.h:
3535 * gst/rtsp-server/rtsp-server.c:
3536 client: remove reference to server
3537 We don't need to keep a ref to the server
3539 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3541 * gst/rtsp-server/rtsp-client.c:
3542 * gst/rtsp-server/rtsp-client.h:
3544 Also add some g_return_if()
3546 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3548 * gst/rtsp-server/rtsp-client.c:
3549 client: log more errors
3551 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3553 * gst/rtsp-server/rtsp-client.c:
3554 client: fix compilation
3556 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3558 * gst/rtsp-server/rtsp-client.c:
3559 * gst/rtsp-server/rtsp-client.h:
3560 client: add generic close-after-send support
3561 Add a property to send_response() to close the connection after the response has
3562 been sent to the client.
3564 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3567 * docs/libs/gst-rtsp-server-docs.sgml:
3568 * docs/libs/gst-rtsp-server-sections.txt:
3569 * docs/libs/gst-rtsp-server.types:
3570 * examples/test-auth.c:
3571 * examples/test-launch.c:
3572 * examples/test-mp4.c:
3573 * examples/test-multicast.c:
3574 * examples/test-multicast2.c:
3575 * examples/test-ogg.c:
3576 * examples/test-readme.c:
3577 * examples/test-sdp.c:
3578 * examples/test-uri.c:
3579 * examples/test-video.c:
3580 * gst/rtsp-server/Makefile.am:
3581 * gst/rtsp-server/rtsp-auth.h:
3582 * gst/rtsp-server/rtsp-client.c:
3583 * gst/rtsp-server/rtsp-client.h:
3584 * gst/rtsp-server/rtsp-media-mapping.c:
3585 * gst/rtsp-server/rtsp-media-mapping.h:
3586 * gst/rtsp-server/rtsp-mount-points.c:
3587 * gst/rtsp-server/rtsp-mount-points.h:
3588 * gst/rtsp-server/rtsp-server.c:
3589 * gst/rtsp-server/rtsp-server.h:
3590 * gst/rtsp-server/rtsp-session-media.c:
3591 * gst/rtsp-server/rtsp-session-pool.c:
3592 * gst/rtsp-server/rtsp-session-pool.h:
3593 * tests/check/gst/rtspserver.c:
3594 MediaMapping -> MountPoints
3595 Describes better what the object manages.
3597 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3600 configure: bump required version of -base
3602 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3604 * gst/rtsp-server/rtsp-media.c:
3607 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3609 * gst/rtsp-server/rtsp-media.c:
3610 * gst/rtsp-server/rtsp-media.h:
3611 media: support more Range formats
3612 Use the new -base methods to convert the Range string into a seek start and stop
3615 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3617 * examples/test-launch.c:
3618 examples: fix whitespace
3620 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3622 * examples/test-auth.c:
3623 test-auth: add example of how to remove sessions
3624 Add an example of the session filter api.
3626 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3628 * examples/test-uri.c:
3629 test-uri: remove mapping example
3631 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3633 * examples/test-uri.c:
3634 test-uri: fix callback signature
3636 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3638 * gst/rtsp-server/rtsp-media-factory.c:
3639 factory: keep ref to factory while media active
3640 While the media from a factory is alive, keep a ref to the factory.
3641 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
3643 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3645 * gst/rtsp-server/rtsp-media-factory-uri.c:
3646 factory-uri: add some debug
3648 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3650 * gst/rtsp-server/rtsp-stream.c:
3651 stream: set udp sources to PLAYING
3652 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
3653 so that it doesn't cause our pipeline to produce ASYNC-DONE.
3655 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3657 * gst/rtsp-server/rtsp-media-factory-uri.c:
3658 factory-uri: take ref to factory
3659 Take a ref to the factory that we place in our list.
3661 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3663 * tests/Makefile.am:
3664 * tests/test-reuse.c:
3665 test: add test for server reuse
3666 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
3668 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
3670 * gst/rtsp-server/rtsp-server.c:
3671 server: start and stop multiple times
3672 Stop listening on the RTSP port when the GSource is removed, so clients
3673 can't connect and the server can be started again.
3674 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
3676 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3678 * gst/rtsp-server/rtsp-server.c:
3679 server: fix small leak
3681 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3683 * gst/rtsp-server/rtsp-media.c:
3684 media: unref source in finish_unprepare
3685 The source is created in prepare, unref it in finish_unprepare.
3686 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
3688 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
3690 * gst/rtsp-server/rtsp-client.c:
3691 * gst/rtsp-server/rtsp-media.c:
3692 rtsp-media: remove bus watch before finalizing
3693 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
3694 * An extra media ref is added for the bus watch. This extra ref is unreffed by
3695 the GDestroyNotify function.
3696 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
3697 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
3698 gst_rtsp_media_unprepare before unreffing the media.
3699 This way, the bus watch will be removed before the media is finalized.
3700 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
3702 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
3704 * gst/rtsp-server/rtsp-client.c:
3705 * gst/rtsp-server/rtsp-client.h:
3706 client: wait until the TEARDOWN response is sent to close the connection
3707 Responses can be sent async so we need to wait until the TEARDOWN response has
3708 been written before we close the connection to the client. This avoids the risk
3709 of writing/polling closed sockets.
3710 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
3712 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
3714 * gst/rtsp-server/rtsp-stream.c:
3715 rtsp-stream: plug socket leak
3716 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
3718 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
3721 Automatic update of common submodule
3722 From 6bb6951 to a72faea
3724 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
3726 * gst/rtsp-server/rtsp-media-factory-uri.c:
3727 rtsp-server: don't use deprecated API
3729 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
3731 * gst/rtsp-server/rtsp-client.c:
3732 rtsp-client: fix unused-but-set-variable compiler warning
3733 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
3735 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3738 * docs/libs/gst-rtsp-server-sections.txt:
3739 * gst/rtsp-server/rtsp-client.c:
3742 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3744 * examples/Makefile.am:
3745 * examples/test-multicast2.c:
3746 examples: add another multicast example
3747 Add an example for how to configure separate multicast ranges for each media
3750 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3752 * examples/test-multicast.c:
3755 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3757 * gst/rtsp-server/rtsp-client.c:
3758 * gst/rtsp-server/rtsp-media.c:
3759 * gst/rtsp-server/rtsp-session-media.c:
3760 * gst/rtsp-server/rtsp-session-media.h:
3761 * gst/rtsp-server/rtsp-stream-transport.c:
3762 * gst/rtsp-server/rtsp-stream-transport.h:
3763 stream: use the address managed by the stream
3764 Use the address managed by the stream for multicast. This allows us to have 1
3765 multicast address for each stream.
3766 Because the address is now managed by the stream we don't have to pass it around
3768 Set the address pool on the streams.
3770 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3772 * gst/rtsp-server/rtsp-client.c:
3773 * gst/rtsp-server/rtsp-media.c:
3774 * gst/rtsp-server/rtsp-stream.c:
3777 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3779 * gst/rtsp-server/rtsp-media.c:
3780 * gst/rtsp-server/rtsp-media.h:
3781 media: add signal for new streams
3782 This allows applications to listen for new streams and configure properties on
3783 them, like the address pool.
3785 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3787 * gst/rtsp-server/rtsp-media.c:
3788 media: configure address pool in new streams
3790 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3792 * gst/rtsp-server/rtsp-stream.c:
3793 * gst/rtsp-server/rtsp-stream.h:
3794 stream: add methods to deal with address pool
3795 Add methods to get and set the address pool for the stream
3796 Add method to allocate and get the multicast addresses for this stream.
3798 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3800 * docs/libs/gst-rtsp-server-sections.txt:
3801 * gst/rtsp-server/rtsp-media.c:
3802 * gst/rtsp-server/rtsp-media.h:
3803 media: remove MTU property
3804 It is a stream property
3806 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3808 * gst/rtsp-server/rtsp-client.c:
3809 client: set blocksize only on stream
3810 Set the blocksize only on the current stream.
3812 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3814 * gst/rtsp-server/rtsp-stream.c:
3815 stream: share src and sink sockets
3816 the allocated socket is in the used-socket property, not socket.
3818 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3820 * gst/rtsp-server/rtsp-address-pool.c:
3821 * gst/rtsp-server/rtsp-address-pool.h:
3822 * gst/rtsp-server/rtsp-client.c:
3823 * gst/rtsp-server/rtsp-session-media.c:
3824 * gst/rtsp-server/rtsp-session-media.h:
3825 * gst/rtsp-server/rtsp-stream-transport.c:
3826 * gst/rtsp-server/rtsp-stream-transport.h:
3827 * tests/check/gst/addresspool.c:
3828 rtsp: make address-pool return an address object
3829 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
3830 store more info in the structure and allows us to more easily return the address
3831 to the right pool when no longer needed.
3832 Pass the address to the StreamTransport so that we can return it to the pool
3833 when the stream transport is freed or changed.
3835 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3837 * examples/Makefile.am:
3838 * examples/test-multicast.c:
3839 examples: add multicast example
3840 Show how to set up the multicast address pool so that media can be
3841 server with multicast.
3843 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3845 * gst/rtsp-server/rtsp-client.c:
3846 * gst/rtsp-server/rtsp-media-factory.c:
3847 * gst/rtsp-server/rtsp-media-factory.h:
3848 * gst/rtsp-server/rtsp-media.c:
3849 * gst/rtsp-server/rtsp-media.h:
3850 rtsp: use AddressPool
3851 Remove the multicast_group property.
3852 Use the configured addresspool to allocate multicast addresses.
3854 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3856 * gst/rtsp-server/rtsp-address-pool.c:
3857 * gst/rtsp-server/rtsp-address-pool.h:
3858 address-pool: add clear method
3860 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3862 * gst/rtsp-server/rtsp-address-pool.c:
3863 address-pool: small cleanups
3865 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3867 * tests/check/Makefile.am:
3868 * tests/check/gst/addresspool.c:
3869 tests: add addresspool unit test
3871 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3873 * gst/rtsp-server/Makefile.am:
3874 * gst/rtsp-server/rtsp-address-pool.c:
3875 * gst/rtsp-server/rtsp-address-pool.h:
3876 address-pool: add object to manage multicast addresses
3877 Make an object that can manage a rage of multicast addresses and ports.
3879 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3881 * gst/rtsp-server/rtsp-server.c:
3882 server: set default max-threads property
3884 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3886 * gst/rtsp-server/rtsp-media.c:
3887 media: wait for concurrent _prepare
3888 If a prepare is busy, wait for the result.
3890 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3892 * gst/rtsp-server/rtsp-media.c:
3893 media: add lock around message handler
3894 We don't want to dispatch messages while we are still processing the result of
3897 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3899 * gst/rtsp-server/rtsp-media.c:
3900 * gst/rtsp-server/rtsp-media.h:
3901 media: add lock to protect state changes
3903 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3905 * gst/rtsp-server/rtsp-stream.c:
3906 * gst/rtsp-server/rtsp-stream.h:
3909 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3911 * gst/rtsp-server/rtsp-stream-transport.c:
3912 * gst/rtsp-server/rtsp-stream-transport.h:
3913 * gst/rtsp-server/rtsp-stream.c:
3914 stream-transport: add keep-alive method
3916 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3918 * gst/rtsp-server/rtsp-stream-transport.c:
3919 * gst/rtsp-server/rtsp-stream-transport.h:
3920 * gst/rtsp-server/rtsp-stream.c:
3921 stream-transport: add method to handle RTP/RTCP
3922 Call new methods instead of poking into the structures directly.
3924 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3926 * gst/rtsp-server/rtsp-session-media.c:
3927 * gst/rtsp-server/rtsp-session-media.h:
3928 session-media: add locking
3930 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3932 * gst/rtsp-server/rtsp-session.c:
3933 * gst/rtsp-server/rtsp-session.h:
3934 session: add locking
3936 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3938 * gst/rtsp-server/rtsp-server.c:
3939 server: free old socket
3941 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3943 * gst/rtsp-server/rtsp-media-mapping.c:
3944 * gst/rtsp-server/rtsp-media-mapping.h:
3945 mapping: add locking
3947 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3949 * gst/rtsp-server/rtsp-media-factory.c:
3950 media-factory: add locking
3952 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3954 * gst/rtsp-server/rtsp-auth.c:
3955 * gst/rtsp-server/rtsp-auth.h:
3958 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3960 * gst/rtsp-server/rtsp-server.c:
3961 * gst/rtsp-server/rtsp-server.h:
3962 server: add max-thread property
3964 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3966 * gst/rtsp-server/rtsp-server.c:
3967 * gst/rtsp-server/rtsp-server.h:
3968 server: use a threadpool for the mainloops
3970 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3972 * gst/rtsp-server/rtsp-client.c:
3973 * gst/rtsp-server/rtsp-client.h:
3974 client: rename method
3975 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
3976 don't really create the client from the socket, we use the socket for the
3979 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3981 * gst/rtsp-server/rtsp-client.c:
3982 * gst/rtsp-server/rtsp-client.h:
3983 * gst/rtsp-server/rtsp-server.c:
3984 server: rework maincontext handling in clients
3985 Make a separate method to attach a client to a MainContext.
3986 Let the server decide in what GMainContext the client will operate and give this
3987 context to the client in attach. Then the server can later decide to use a
3988 separate thread for each client or just use the mainthread.
3990 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3992 * gst/rtsp-server/rtsp-client.c:
3993 * gst/rtsp-server/rtsp-session.c:
3994 * gst/rtsp-server/rtsp-session.h:
3995 session: move session header code in session object
3997 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
4001 * examples/test-auth.c:
4002 * examples/test-launch.c:
4003 * examples/test-mp4.c:
4004 * examples/test-ogg.c:
4005 * examples/test-readme.c:
4006 * examples/test-sdp.c:
4007 * examples/test-uri.c:
4008 * examples/test-video.c:
4009 * gst/rtsp-server/rtsp-auth.c:
4010 * gst/rtsp-server/rtsp-auth.h:
4011 * gst/rtsp-server/rtsp-client.c:
4012 * gst/rtsp-server/rtsp-client.h:
4013 * gst/rtsp-server/rtsp-media-factory-uri.c:
4014 * gst/rtsp-server/rtsp-media-factory-uri.h:
4015 * gst/rtsp-server/rtsp-media-factory.c:
4016 * gst/rtsp-server/rtsp-media-factory.h:
4017 * gst/rtsp-server/rtsp-media-mapping.c:
4018 * gst/rtsp-server/rtsp-media-mapping.h:
4019 * gst/rtsp-server/rtsp-media.c:
4020 * gst/rtsp-server/rtsp-media.h:
4021 * gst/rtsp-server/rtsp-params.c:
4022 * gst/rtsp-server/rtsp-params.h:
4023 * gst/rtsp-server/rtsp-sdp.c:
4024 * gst/rtsp-server/rtsp-sdp.h:
4025 * gst/rtsp-server/rtsp-server.c:
4026 * gst/rtsp-server/rtsp-server.h:
4027 * gst/rtsp-server/rtsp-session-media.c:
4028 * gst/rtsp-server/rtsp-session-media.h:
4029 * gst/rtsp-server/rtsp-session-pool.c:
4030 * gst/rtsp-server/rtsp-session-pool.h:
4031 * gst/rtsp-server/rtsp-session.c:
4032 * gst/rtsp-server/rtsp-session.h:
4033 * gst/rtsp-server/rtsp-stream-transport.c:
4034 * gst/rtsp-server/rtsp-stream-transport.h:
4035 * gst/rtsp-server/rtsp-stream.c:
4036 * gst/rtsp-server/rtsp-stream.h:
4037 * tests/check/gst/rtspserver.c:
4038 * tests/test-cleanup.c:
4041 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
4043 * gst/rtsp-server/rtsp-media.c:
4044 * gst/rtsp-server/rtsp-session-media.c:
4045 * gst/rtsp-server/rtsp-session.c:
4046 rtsp-server: added annotations to indicate type of ownership transfer of return values
4047 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4049 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
4052 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
4054 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
4057 * bindings/Makefile.am:
4058 * bindings/vala/Makefile.am:
4059 * bindings/vala/gst-rtsp-server-0.10.deps:
4060 * bindings/vala/gst-rtsp-server-0.10.vapi:
4061 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
4062 * bindings/vala/packages/gst-rtsp-server-0.10.files:
4063 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
4064 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4065 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
4067 bindings: remove vala bindings
4068 They'll be reunited with the other GStreamer bindings
4069 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4071 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4073 * gst/rtsp-server/rtsp-client.c:
4074 * gst/rtsp-server/rtsp-session-media.c:
4075 * gst/rtsp-server/rtsp-session-media.h:
4076 * gst/rtsp-server/rtsp-stream-transport.c:
4077 * gst/rtsp-server/rtsp-stream-transport.h:
4078 rtsp: only create transport when needed
4079 Only create the StreamTransport when configured.
4081 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4083 * gst/rtsp-server/rtsp-client.c:
4084 client: small cleanup
4086 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4088 * gst/rtsp-server/rtsp-client.c:
4089 * gst/rtsp-server/rtsp-client.h:
4090 * gst/rtsp-server/rtsp-stream-transport.c:
4091 * gst/rtsp-server/rtsp-stream-transport.h:
4092 rtsp: refactor configuration of transport
4093 Move the configuration of the transport to a place where it makes
4096 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4098 * gst/rtsp-server/rtsp-client.c:
4099 client: refactor transport parsing
4101 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4103 * gst/rtsp-server/rtsp-client.c:
4104 client: refuse to change the MTU on shared media
4105 If we change the MTU of chared media, it changes for all clients.
4106 We don't want to set the MTU to something large for clients that
4109 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4111 * examples/test-mp4.c:
4112 * gst/rtsp-server/rtsp-media.c:
4113 small fixes to docs and debug
4115 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4117 * gst/rtsp-server/rtsp-stream.c:
4118 stream: transports must already have been removed
4120 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4122 * gst/rtsp-server/rtsp-media.c:
4123 * gst/rtsp-server/rtsp-stream.c:
4124 * gst/rtsp-server/rtsp-stream.h:
4125 stream: improve join and leave of the pipeline
4127 Do the cleanup properly
4130 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4132 * gst/rtsp-server/rtsp-media.c:
4133 media: move unprepare below default implementation
4134 Makes it easier to find the default implementation
4136 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4138 * gst/rtsp-server/rtsp-media.c:
4139 media: signal unprepared when we actually finish
4141 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4143 * gst/rtsp-server/rtsp-media.c:
4144 media: no need to unlock, unprepare does that when needed
4146 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4148 * docs/libs/gst-rtsp-server-sections.txt:
4149 * gst/rtsp-server/rtsp-media-factory.h:
4150 * gst/rtsp-server/rtsp-media-mapping.c:
4151 * gst/rtsp-server/rtsp-media.h:
4152 * gst/rtsp-server/rtsp-params.c:
4153 * gst/rtsp-server/rtsp-server.c:
4154 * gst/rtsp-server/rtsp-session-pool.h:
4155 * gst/rtsp-server/rtsp-session.c:
4156 * gst/rtsp-server/rtsp-session.h:
4157 * gst/rtsp-server/rtsp-stream-transport.h:
4158 * gst/rtsp-server/rtsp-stream.h:
4161 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4163 * gst/rtsp-server/rtsp-client.c:
4164 * gst/rtsp-server/rtsp-media-mapping.h:
4165 * gst/rtsp-server/rtsp-media.c:
4166 * gst/rtsp-server/rtsp-media.h:
4167 * gst/rtsp-server/rtsp-server.h:
4168 * gst/rtsp-server/rtsp-stream.c:
4169 * gst/rtsp-server/rtsp-stream.h:
4170 rtsp: fix MTU setting
4171 Fix setting of the MTU. There is no need for a vmethod.
4173 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4178 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4181 configure: bump version number after refactoring
4183 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4185 * gst/rtsp-server/Makefile.am:
4186 * gst/rtsp-server/rtsp-client.c:
4187 * gst/rtsp-server/rtsp-client.h:
4188 * gst/rtsp-server/rtsp-media-factory-uri.c:
4189 * gst/rtsp-server/rtsp-media-factory.c:
4190 * gst/rtsp-server/rtsp-media-factory.h:
4191 * gst/rtsp-server/rtsp-media.c:
4192 * gst/rtsp-server/rtsp-media.h:
4193 * gst/rtsp-server/rtsp-sdp.c:
4194 * gst/rtsp-server/rtsp-session-media.c:
4195 * gst/rtsp-server/rtsp-session-media.h:
4196 * gst/rtsp-server/rtsp-session.c:
4197 * gst/rtsp-server/rtsp-session.h:
4198 * gst/rtsp-server/rtsp-stream-transport.c:
4199 * gst/rtsp-server/rtsp-stream-transport.h:
4200 * gst/rtsp-server/rtsp-stream.c:
4201 * gst/rtsp-server/rtsp-stream.h:
4202 rtsp: massive refactoring
4203 Make GObjects from the remaining simple structures.
4204 Remove GstRTSPSessionStream, it's not needed.
4205 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
4206 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
4207 a GstRTSPStream should be transported to a client.
4208 Rename GstRTSPMediaFactory::get_element -> create_element because that
4209 more accurately describes what it does.
4210 Make nice methods instead of poking in the structures.
4211 Move some methods inside the relevant object source code.
4212 Use GPtrArray to store objects instead of plain arrays, it is more
4213 natural and allows us to more easily clean up.
4214 Move the allocation of udp ports to the Stream object. The Stream object
4215 contains the elements needed to stream the media to a client.
4216 Improve the prepare and unprepare methods. Unprepare should now undo
4217 everything prepare did. Improve also async unprepare when doing EOS on
4218 shutdown. Make sure we always unprepare correctly.
4220 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
4222 * gst/rtsp-server/rtsp-client.c:
4223 rtsp-client: Unref server address clients connected to
4224 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
4226 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
4228 * gst/rtsp-server/rtsp-server.c:
4229 rtsp-server: don't ref server socket if it is NULL
4230 Fixes test_bind_already_in_use unit test again after commit 6a497440.
4231 https://bugzilla.gnome.org/show_bug.cgi?id=686644
4233 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
4235 * tests/check/Makefile.am:
4236 tests: Add libgio link dependency
4237 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
4239 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
4241 * gst/rtsp-server/rtsp-media-mapping.c:
4242 * gst/rtsp-server/rtsp-media-mapping.h:
4243 rtsp-media-mapping: rename find_media vfunc to find_factory
4244 The virtual method and class method should have the same name
4245 so it is correctly represented in GIR file
4246 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4248 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
4250 * gst/rtsp-server/rtsp-auth.c:
4251 * gst/rtsp-server/rtsp-client.c:
4252 * gst/rtsp-server/rtsp-media-factory-uri.c:
4253 * gst/rtsp-server/rtsp-media-factory.c:
4254 * gst/rtsp-server/rtsp-media-mapping.c:
4255 * gst/rtsp-server/rtsp-media.c:
4256 * gst/rtsp-server/rtsp-server.c:
4257 * gst/rtsp-server/rtsp-session-pool.c:
4258 * gst/rtsp-server/rtsp-session.c:
4259 rtsp-server: fixed comments and GIR annotations
4260 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4262 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
4264 * gst/rtsp-server/rtsp-media-mapping.c:
4265 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
4267 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
4269 * gst/rtsp-server/rtsp-server.c:
4270 rtsp-server: allow binding on port 0 (binds on a random port)
4272 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
4274 * gst/rtsp-server/rtsp-server.c:
4275 * gst/rtsp-server/rtsp-server.h:
4276 rtsp-server: add bound-port property
4277 bound-port can be used to retrieve the port number when the server is bound on
4278 port 0, which binds on a random port.
4280 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
4282 * gst/rtsp-server/rtsp-media-factory.c:
4283 * gst/rtsp-server/rtsp-media-factory.h:
4284 rtsp-media-factory: make ::get_element overridable by GI bindings
4285 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
4286 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
4287 as the invoker for ::get_element(), making it overridable by GI generated
4290 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4292 * gst/rtsp-server/rtsp-media-factory-uri.c:
4293 rtsp-media-factory-uri: don't autoplug parsers in a loop
4294 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
4297 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4299 * gst/rtsp-server/Makefile.am:
4300 Explicitly link against gio. Fix link error on mac.
4302 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
4304 * gst/rtsp-server/rtsp-session.c:
4305 session: add ttl to the transport header in SETUP
4306 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
4308 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
4310 * gst/rtsp-server/rtsp-client.c:
4311 * gst/rtsp-server/rtsp-client.h:
4312 * gst/rtsp-server/rtsp-media.c:
4313 client: Use client transport settings for multicast if allowed.
4314 This patch makes it possible for the client to send transport settings for
4315 multicast (destination && ttl). Client settings must be explicitly allowed or
4316 the server will use its own settings.
4317 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
4319 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
4322 Automatic update of common submodule
4323 From 6c0b52c to 6bb6951
4325 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
4327 * gst/rtsp-server/rtsp-client.c:
4328 rtsp-client: do not destroy the rtsp watch
4329 Don't destroy the client watch while dispatching. The rtsp watch is
4330 automatically destroyed after the rtsp watch function closed() has
4332 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
4334 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4337 Automatic update of common submodule
4338 From 4f962f7 to 6c0b52c
4340 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
4342 * gst/rtsp-server/rtsp-media.c:
4343 media: fix check for seekability
4345 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4347 * gst/rtsp-server/rtsp-client.c:
4348 client: use more GIO
4349 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
4351 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4353 * gst/rtsp-server/rtsp-server.c:
4354 server: remove obsolete includes
4356 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4358 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
4359 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
4360 be available in "on_new_ssrc". The transports are added in
4361 gst_rtsp_media_set_state when going to PLAYING state. However,
4362 "on_new_ssrc" might be called before this happens.
4363 https://bugzilla.gnome.org/show_bug.cgi?id=683304
4365 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4367 * gst/rtsp-server/rtsp-client.c:
4368 * gst/rtsp-server/rtsp-client.h:
4369 rtsp-client: add signals for rtsp requests (fixes #683287)
4371 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4373 * gst/rtsp-server/rtsp-client.c:
4374 * gst/rtsp-server/rtsp-client.h:
4375 add new-session signal to rtsp-client (fixes #683058)
4377 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
4380 Automatic update of common submodule
4381 From 668acee to 4f962f7
4383 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
4385 * gst/rtsp-server/rtsp-server.c:
4386 * tests/check/gst/rtspserver.c:
4387 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
4388 Do not assume that *error is set in g_socket_address_enumerator_next.
4389 Added test_bind_already_in_use unit-test.
4390 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
4392 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
4395 Automatic update of common submodule
4396 From 94ccf4c to 668acee
4398 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
4400 * gst/rtsp-server/rtsp-client.c:
4401 * gst/rtsp-server/rtsp-client.h:
4402 rtsp-client: make create_sdp virtual method
4403 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
4405 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4408 Automatic update of common submodule
4409 From 98e386f to 94ccf4c
4411 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4413 * gst/rtsp-server/rtsp-client.c:
4416 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
4418 * gst/rtsp-server/rtsp-client.c:
4419 * gst/rtsp-server/rtsp-client.h:
4420 * gst/rtsp-server/rtsp-server.c:
4421 * gst/rtsp-server/rtsp-server.h:
4422 rtsp-server: use an existing socket to establish HTTP tunnel
4423 Make it possible to transfer a socket from an HTTP server to be used as
4424 an RTSP over HTTP tunnel.
4426 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
4428 * gst/rtsp-server/rtsp-client.c:
4429 * gst/rtsp-server/rtsp-media.c:
4430 * gst/rtsp-server/rtsp-media.h:
4431 rtsp: Handle the blocksize parameter
4432 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
4434 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
4436 * tests/check/Makefile.am:
4437 * tests/check/gst/rtspserver.c:
4438 Have unit test get header from source dir, not installed dir
4439 This makes compilation of unit tests work in a build directory other
4440 than the source directory.
4441 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
4443 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
4445 * gst/rtsp-server/rtsp-media.c:
4446 rtsp-media: update for gst_element_make_from_uri() changes
4448 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
4451 * tests/Makefile.am:
4452 * tests/check/Makefile.am:
4453 * tests/check/gst/rtspserver.c:
4455 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
4457 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
4459 * gst/rtsp-server/rtsp-media.c:
4460 rtsp-media: don't collect media stats when going to NULL
4461 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
4463 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4465 * gst/rtsp-server/rtsp-client.c:
4466 client: don't leak transports
4468 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
4470 * gst/rtsp-server/rtsp-client.c:
4471 rtsp-client: free transport on no_stream in SETUP handler
4473 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
4475 * gst/rtsp-server/rtsp-client.c:
4476 rtsp-client: changed session media iteration
4477 In client_unlink_session: now don't iterate in session->medias
4478 list where items are removed by gst_rtsp_session_release_media.
4479 Instead, repeatedly remove the first item.
4481 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
4483 * gst/rtsp-server/rtsp-client.c:
4484 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
4485 GstRTSPSessionMedia is not a GObject type. When the
4486 GstRTSPSession is freed, it will free the media.
4488 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
4490 * gst/rtsp-server/rtsp-media-factory.c:
4491 factory: plug pad leak in collect_streams
4492 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
4493 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
4494 will take one reference, and the other reference will otherwise
4497 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4500 configure: suppress some warnings when debug is disabled
4501 Warnings about unused variables should be suppressed if core has the
4502 debug system disabled.
4503 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4505 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4507 * docs/libs/Makefile.am:
4508 docs: fix build in uninstalled setup
4509 Include gst-plugins-base libs properly.
4511 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
4513 * docs/libs/gst-rtsp-server.types:
4514 docs: include headers defining rtsp-server object types
4515 Fixes compiler warnings during docs build.
4516 https://bugzilla.gnome.org/show_bug.cgi?id=676824
4518 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
4521 configure: Add warning flags for compiler when configuring
4522 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4524 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4527 Automatic update of common submodule
4528 From 03a0e57 to 98e386f
4530 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4533 Automatic update of common submodule
4534 From 1fab359 to 03a0e57
4536 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
4538 * gst/rtsp-server/rtsp-client.c:
4539 client: fix GSocketAddress leak in gst_rtsp_client_accept
4540 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
4542 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4545 Automatic update of common submodule
4546 From f1b5a96 to 1fab359
4548 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4551 Automatic update of common submodule
4552 From 92b7266 to f1b5a96
4554 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4557 Automatic update of common submodule
4558 From ec1c4a8 to 92b7266
4560 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4563 Automatic update of common submodule
4564 From 3429ba6 to ec1c4a8
4566 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
4568 * gst/rtsp-server/rtsp-auth.c:
4569 * gst/rtsp-server/rtsp-client.c:
4570 * gst/rtsp-server/rtsp-media-factory-uri.c:
4571 * gst/rtsp-server/rtsp-server.c:
4572 rtsp: fix compiler warnings
4573 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
4575 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4578 Automatic update of common submodule
4579 From dc70203 to 3429ba6
4581 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4583 * gst/rtsp-server/rtsp-client.c:
4584 * gst/rtsp-server/rtsp-media-factory.c:
4585 * gst/rtsp-server/rtsp-media-factory.h:
4586 * gst/rtsp-server/rtsp-media.c:
4587 * gst/rtsp-server/rtsp-media.h:
4588 * gst/rtsp-server/rtsp-server.c:
4589 * gst/rtsp-server/rtsp-server.h:
4590 * gst/rtsp-server/rtsp-session-pool.c:
4591 * gst/rtsp-server/rtsp-session-pool.h:
4592 rtsp-server: port to new thread API
4594 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4597 Automatic update of common submodule
4598 From 6db25be to dc70203
4600 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4602 * gst/rtsp-server/rtsp-auth.c:
4603 * gst/rtsp-server/rtsp-auth.h:
4604 * gst/rtsp-server/rtsp-client.c:
4605 rtsp-server: Fix compilation and compiler warnings
4607 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4611 * gst/rtsp-server/Makefile.am:
4612 configure: Modernize autotools setup a bit
4613 Also we now only create tar.bz2 and tar.xz tarballs.
4615 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4618 Automatic update of common submodule
4619 From 464fe15 to 6db25be
4621 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4624 Automatic update of common submodule
4625 From 7fda524 to 464fe15
4627 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4630 * docs/libs/Makefile.am:
4631 * docs/version.entities.in:
4633 * gst/rtsp-server/Makefile.am:
4634 * pkgconfig/Makefile.am:
4635 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4636 * pkgconfig/gstreamer-rtsp-server.pc.in:
4637 * tests/Makefile.am:
4638 rtsp-server: Update versioning
4640 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4642 Merge remote-tracking branch 'origin/0.10'
4644 gst/rtsp-server/rtsp-session-pool.c
4646 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4648 * gst/rtsp-server/rtsp-session-pool.c:
4649 rtsp-server: Don't use deprecated GLib API
4651 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4653 Replace master with 0.11
4655 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4657 Merge branch 'master' into 0.11
4659 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4661 Merge branch 'master' into 0.11
4663 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4666 A couple minor typo fixes
4668 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4670 * gst/rtsp-server/rtsp-media.c:
4671 media: fix state of the appqueue
4673 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4675 * gst/rtsp-server/rtsp-media-factory-uri.c:
4676 factory: use videoconvert
4678 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4680 * gst/rtsp-server/rtsp-media-factory-uri.c:
4681 factory: change to new style caps
4683 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4685 * gst/rtsp-server/rtsp-client.c:
4686 * gst/rtsp-server/rtsp-client.h:
4687 * gst/rtsp-server/rtsp-media-factory-uri.c:
4688 * gst/rtsp-server/rtsp-media.c:
4689 * gst/rtsp-server/rtsp-server.c:
4690 * gst/rtsp-server/rtsp-server.h:
4691 * gst/rtsp-server/rtsp-session-pool.c:
4692 rtsp-server: port to GIO
4695 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4698 configure: fix build
4700 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4703 docs: fix for gst_rtsp_server_set_port() -> _set_service()
4704 https://bugzilla.gnome.org/show_bug.cgi?id=666548
4706 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4709 * examples/Makefile.am:
4710 First rule of gst-rtsp-server club: don't talk about gst-phonon
4712 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4715 * pkgconfig/Makefile.am:
4716 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
4717 * pkgconfig/gst-rtsp-server.pc.in:
4718 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4719 * pkgconfig/gstreamer-rtsp-server.pc.in:
4720 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
4721 For consistency with all other modules.
4723 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4725 * gst/rtsp-server/rtsp-client.c:
4726 rtsp-client: update for new map API
4728 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4731 * bindings/Makefile.am:
4732 * bindings/python/Makefile.am:
4733 * bindings/python/arg-types.py:
4734 * bindings/python/codegen/Makefile.am:
4735 * bindings/python/codegen/__init__.py:
4736 * bindings/python/codegen/argtypes.py:
4737 * bindings/python/codegen/code-coverage.py:
4738 * bindings/python/codegen/codegen.py:
4739 * bindings/python/codegen/definitions.py:
4740 * bindings/python/codegen/defsparser.py:
4741 * bindings/python/codegen/docextract.py:
4742 * bindings/python/codegen/docgen.py:
4743 * bindings/python/codegen/fileprefix.override:
4744 * bindings/python/codegen/fileprefixmodule.c:
4745 * bindings/python/codegen/h2def.py:
4746 * bindings/python/codegen/mergedefs.py:
4747 * bindings/python/codegen/mkskel.py:
4748 * bindings/python/codegen/override.py:
4749 * bindings/python/codegen/reversewrapper.py:
4750 * bindings/python/codegen/scmexpr.py:
4751 * bindings/python/rtspserver-types.defs:
4752 * bindings/python/rtspserver.defs:
4753 * bindings/python/rtspserver.override:
4754 * bindings/python/rtspservermodule.c:
4755 * bindings/python/test.py:
4757 python: remove pygst-based python bindings
4758 pygi is the future, apparently.
4760 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
4763 Automatic update of common submodule
4764 From c463bc0 to 7fda524
4766 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4769 Automatic update of common submodule
4770 From 2a59016 to c463bc0
4772 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4775 Automatic update of common submodule
4776 From 0807187 to 2a59016
4778 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4781 Automatic update of common submodule
4782 From 11f0cd5 to 0807187
4784 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4786 * examples/test-auth.c:
4787 example: update for new caps
4789 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4791 * examples/test-video.c:
4792 * gst/rtsp-server/rtsp-client.c:
4793 * gst/rtsp-server/rtsp-media-factory-uri.c:
4794 * gst/rtsp-server/rtsp-media.c:
4795 * gst/rtsp-server/rtsp-media.h:
4796 * gst/rtsp-server/rtsp-session.c:
4797 * gst/rtsp-server/rtsp-session.h:
4798 rtsp-server: port some more to 0.11
4800 Remove bufferlist stuff
4802 Add queue before appsink now that preroll-queue-len is gone.
4803 Update for request pad changes.
4805 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4807 Merge branch 'master' into 0.11
4809 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4811 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4812 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4813 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4815 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4817 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4818 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4819 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4821 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4823 Merge branch 'master' into 0.11
4825 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4827 * gst/rtsp-server/rtsp-media.c:
4828 * gst/rtsp-server/rtsp-media.h:
4829 media: add a seekable boolean
4830 Maintain the seekable state with a new variable instead of reusing the
4833 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
4835 * gst/rtsp-server/rtsp-media.c:
4836 Disallow seek in live media
4838 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4840 Merge branch 'master' into 0.11
4842 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
4844 * gst/rtsp-server/rtsp-server.c:
4845 #ifdef statements for windows socket creation were missing
4847 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
4850 Automatic update of common submodule
4851 From a39eb83 to 11f0cd5
4853 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
4856 Automatic update of common submodule
4857 From 605cd9a to a39eb83
4859 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4861 Merge branch 'master' into 0.11
4863 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4865 * gst/rtsp-server/rtsp-client.c:
4866 client: use method to access property
4868 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4870 * gst/rtsp-server/rtsp-media-factory.c:
4871 * gst/rtsp-server/rtsp-media-factory.h:
4872 media-factory: add protocols property
4873 Add a property to configure the allowed protocols in the media created from the
4876 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4878 * gst/rtsp-server/rtsp-media-factory.c:
4879 * gst/rtsp-server/rtsp-media-factory.h:
4880 media-factory: add media-configure signal
4881 Add signal to allow the application to configure the media after it was created
4884 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4886 * gst/rtsp-server/rtsp-client.c:
4887 client: use method to access property
4889 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4891 * gst/rtsp-server/rtsp-media-factory.c:
4892 * gst/rtsp-server/rtsp-media-factory.h:
4893 media-factory: add protocols property
4894 Add a property to configure the allowed protocols in the media created from the
4897 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4899 * gst/rtsp-server/rtsp-media-factory.c:
4900 * gst/rtsp-server/rtsp-media-factory.h:
4901 media-factory: add media-configure signal
4902 Add signal to allow the application to configure the media after it was created
4905 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4907 Merge branch 'master' into 0.11
4909 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4911 * gst/rtsp-server/rtsp-client.c:
4912 client: use media multicast group
4914 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4916 * gst/rtsp-server/rtsp-media-factory.h:
4917 * gst/rtsp-server/rtsp-server.h:
4918 * gst/rtsp-server/rtsp-session-pool.h:
4919 * gst/rtsp-server/rtsp-session.h:
4922 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4924 * gst/rtsp-server/rtsp-client.c:
4925 * gst/rtsp-server/rtsp-sdp.h:
4926 sdp: copy and free the server ip address
4927 Copy and free the server ip address to make memory management easier later.
4929 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4931 * gst/rtsp-server/rtsp-media-factory.c:
4932 media-factory: configure multicast in media
4934 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4936 * gst/rtsp-server/rtsp-media.c:
4937 * gst/rtsp-server/rtsp-media.h:
4938 media: add property for multicast group
4939 Add a property to configure the multicast group in the media.
4940 Based on patches from Marc Leeman and Robert Krakora.
4942 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4944 * gst/rtsp-server/rtsp-media-factory.c:
4945 * gst/rtsp-server/rtsp-media-factory.h:
4946 media-factory: add property for multicast group
4947 Add a property to configure the multicast group in the media factory.
4948 Based on patches from Marc Leeman and Robert Krakora.
4950 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4952 * gst/rtsp-server/rtsp-client.c:
4953 client: do configuration of transport in one place
4954 Move the configuration of the transport destination address to where we also
4955 configure the other bits.
4957 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4959 * gst/rtsp-server/rtsp-client.c:
4960 client: use media multicast group
4962 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4964 * gst/rtsp-server/rtsp-media-factory.h:
4965 * gst/rtsp-server/rtsp-server.h:
4966 * gst/rtsp-server/rtsp-session-pool.h:
4967 * gst/rtsp-server/rtsp-session.h:
4970 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4972 * gst/rtsp-server/rtsp-client.c:
4973 * gst/rtsp-server/rtsp-sdp.h:
4974 sdp: copy and free the server ip address
4975 Copy and free the server ip address to make memory management easier later.
4977 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4979 * gst/rtsp-server/rtsp-media-factory.c:
4980 media-factory: configure multicast in media
4982 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4984 * gst/rtsp-server/rtsp-media.c:
4985 * gst/rtsp-server/rtsp-media.h:
4986 media: add property for multicast group
4987 Add a property to configure the multicast group in the media.
4988 Based on patches from Marc Leeman and Robert Krakora.
4990 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4992 * gst/rtsp-server/rtsp-media-factory.c:
4993 * gst/rtsp-server/rtsp-media-factory.h:
4994 media-factory: add property for multicast group
4995 Add a property to configure the multicast group in the media factory.
4996 Based on patches from Marc Leeman and Robert Krakora.
4998 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5000 * gst/rtsp-server/rtsp-client.c:
5001 client: do configuration of transport in one place
5002 Move the configuration of the transport destination address to where we also
5003 configure the other bits.
5005 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5007 Merge branch 'master' into 0.11
5009 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
5011 * gst/rtsp-server/rtsp-client.c:
5012 client: destroy pipeline on client disconnect with no prior TEARDOWN.
5013 The problem occurs when the client abruptly closes the connection without
5014 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
5015 server is where the pipeline gets torn down. Since this handler is not called,
5016 the pipeline remains and is up and running. Subsequent clients get their own
5017 pipelines and if the do not issue TEARDOWNs then those pipelines will also
5018 remain up and running. This is a resource leak.
5020 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5022 Merge branch 'master' into 0.11
5024 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
5026 * gst/rtsp-server/rtsp-media-factory.c:
5027 * gst/rtsp-server/rtsp-media-factory.h:
5028 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
5029 For example, it can be used to retrieve source elements like appsrc, in a more
5030 convenient way than subclassing get_element.
5032 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5034 Merge branch 'master' into 0.11
5036 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
5038 * gst/rtsp-server/rtsp-server.c:
5039 rtsp-server: hold on to reference while using object
5041 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5043 * gst/rtsp-server/rtsp-media.c:
5046 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5049 configure: use unstable api
5051 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
5053 * gst/rtsp-server/rtsp-client.c:
5054 client: fix reference counting
5056 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
5058 * gst/rtsp-server/rtsp-client.c:
5059 * gst/rtsp-server/rtsp-media.c:
5060 fix compiler warnings about unused variables
5062 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
5064 * examples/test-launch.c:
5065 * examples/test-readme.c:
5066 * examples/test-uri.c:
5067 * examples/test-video.c:
5068 examples: tell rtsp uri when ready
5070 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
5073 Automatic update of common submodule
5074 From 69b981f to 605cd9a
5076 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5078 * gst/rtsp-server/rtsp-client.c:
5079 client: update for buffer API change
5081 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5083 * gst/rtsp-server/Makefile.am:
5084 Makefile.am: 0.10 => @GST_MAJORMINOR@
5086 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5088 * gst/rtsp-server/rtsp-media-factory-uri.c:
5089 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
5091 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5093 * gst/rtsp-server/.gitignore:
5094 .gitignore: 0.10 => 0.11
5096 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5098 * gst/rtsp-server/Makefile.am:
5099 Makefile.am: 0.10 => @GST_MAJORMINOR@
5101 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5103 Merge branch 'master' into 0.11
5105 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
5108 Automatic update of common submodule
5109 From 9e5bbd5 to 69b981f
5111 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
5114 Automatic update of common submodule
5115 From fd35073 to 9e5bbd5
5117 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
5120 Automatic update of common submodule
5121 From 46dfcea to fd35073
5123 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5125 * gst/rtsp-server/rtsp-media-factory-uri.c:
5126 * gst/rtsp-server/rtsp-media.c:
5127 media: port to new caps API
5129 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5131 Merge branch 'master' into 0.11
5133 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
5135 * bindings/vala/gst-rtsp-server-0.10.vapi:
5136 Updated Vala bindings.
5137 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
5139 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
5141 * gst/rtsp-server/rtsp-server.c:
5142 * gst/rtsp-server/rtsp-server.h:
5143 Add a signal for newly connected clients.
5144 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
5146 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
5148 * bindings/python/rtspserver.override:
5149 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
5151 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5153 * gst/rtsp-server/Makefile.am:
5154 * gst/rtsp-server/rtsp-client.c:
5155 * gst/rtsp-server/rtsp-funnel.c:
5156 * gst/rtsp-server/rtsp-funnel.h:
5157 * gst/rtsp-server/rtsp-media.c:
5158 rtsp-server: port to 0.11
5160 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5165 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5167 Merge branch 'master' into 0.11
5172 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5175 Automatic update of common submodule
5176 From c3cafe1 to 46dfcea
5178 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
5180 * bindings/python/Makefile.am:
5181 * bindings/python/rtspserver.defs:
5182 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
5184 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
5186 * bindings/python/arg-types.py:
5187 python bindings: add GstRTSPUrlParam
5188 Needed to implement MediaFactory virtual proxies
5190 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
5192 * bindings/python/arg-types.py:
5193 python bindings: fix returning GstRTSPUrl types
5195 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5197 * bindings/python/arg-types.py:
5198 python bindings: add arg type for GstRTSPUrl
5200 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
5202 * bindings/python/rtspserver.defs:
5203 python bindings: fix the definition of MediaFactory.collect_stream
5205 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
5208 Automatic update of common submodule
5209 From 1ccbe09 to c3cafe1
5211 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5214 Automatic update of common submodule
5215 From 193b717 to 1ccbe09
5217 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
5220 Automatic update of common submodule
5221 From b77e2bf to 193b717
5223 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5226 build: Include lcov.mak to allow test coverage report generation
5228 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5231 Automatic update of common submodule
5232 From d8814b6 to b77e2bf
5234 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5237 Automatic update of common submodule
5238 From 6aaa286 to d8814b6
5240 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
5243 Automatic update of common submodule
5244 From 6aec6b9 to 6aaa286
5246 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
5249 autogen: wingo signed comment
5251 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
5253 * gst/rtsp-server/rtsp-session-pool.c:
5254 session: use full charset for RTSP session ID
5255 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
5256 session ID more difficult.
5257 https://bugzilla.gnome.org/show_bug.cgi?id=643812
5259 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5261 * gst/rtsp-server/Makefile.am:
5262 rtsp-server: Don't install the funnel header
5264 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
5267 Automatic update of common submodule
5268 From 1de7f6a to 6aec6b9
5270 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5273 configure: require core/base 0.10.31
5274 Needed at least for gst_plugin_feature_rank_compare_func().
5276 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
5279 Automatic update of common submodule
5280 From f94d739 to 1de7f6a
5282 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5284 * gst/rtsp-server/rtsp-media.c:
5285 media: remove more unused code
5287 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5289 * gst/rtsp-server/rtsp-media.c:
5290 * gst/rtsp-server/rtsp-media.h:
5291 media: remove duplicate filtering
5292 Remove the duplicate filtering code now that we have a released -good version.
5293 Give a warning instead.
5295 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5297 * gst/rtsp-server/rtsp-media-factory.c:
5298 * gst/rtsp-server/rtsp-media.c:
5299 media: fix default buffer size
5301 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5303 * gst/rtsp-server/rtsp-media-factory.c:
5304 * gst/rtsp-server/rtsp-media-factory.h:
5305 media-factory: add property to configure the buffer-size
5306 Add a property to configure the kernel UDP buffer size.
5308 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5310 * gst/rtsp-server/rtsp-media.c:
5311 * gst/rtsp-server/rtsp-media.h:
5312 media: add property to configure kernel buffer sizes
5313 Add a property to configure the kernel UDP buffer size.
5315 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5318 configure: set PYGOBJECT_REQ before using it
5319 https://bugzilla.gnome.org/show_bug.cgi?id=640641
5321 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5324 docs: recursive into sub-directories on 'make upload'
5326 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5328 * docs/libs/gst-rtsp-server-docs.sgml:
5329 * docs/version.entities.in:
5330 docs: mention full version these docs are for, not just major-minor
5332 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5337 === release 0.10.8 ===
5339 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5344 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5346 * gst/rtsp-server/rtsp-server.c:
5347 rtsp-server: clarify docs a little
5349 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5351 * gst/rtsp-server/rtsp-media.c:
5352 media: init debug category before starting thread
5354 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5356 * gst/rtsp-server/rtsp-auth.c:
5357 auth: add realm to make it more spec compliant
5359 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5361 * gst/rtsp-server/rtsp-server.c:
5362 * gst/rtsp-server/rtsp-server.h:
5365 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5367 * examples/test-video.c:
5368 example: improve example docs a little
5370 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5372 * gst/rtsp-server/rtsp-server.c:
5373 server: ensure the watch has a ref to the server
5375 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5377 * gst/rtsp-server/rtsp-server.c:
5378 server: simpify channel function
5380 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5382 * gst/rtsp-server/rtsp-server.c:
5383 * gst/rtsp-server/rtsp-server.h:
5384 server: simplify management of channel and source
5385 We don't need to keep around the channel and source objects. Let the mainloop
5386 and the source manage the source and channel respectively.
5388 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5394 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5397 * tests/Makefile.am:
5398 * tests/test-cleanup.c:
5399 tests: add tests directory and cleanup test
5401 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5403 * gst/rtsp-server/rtsp-media-factory-uri.c:
5404 * gst/rtsp-server/rtsp-media-factory.c:
5405 * gst/rtsp-server/rtsp-media-mapping.c:
5406 * gst/rtsp-server/rtsp-media.c:
5407 * gst/rtsp-server/rtsp-session-pool.c:
5408 * gst/rtsp-server/rtsp-session.c:
5409 server: improve debugging in various objects
5411 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5413 * gst/rtsp-server/rtsp-server.c:
5414 server: chain up to the parent finalize
5416 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
5418 * bindings/python/rtspserver-types.defs:
5419 * bindings/python/rtspserver.defs:
5420 * bindings/python/rtspserver.override:
5421 * bindings/python/test.py:
5422 gst-rtsp-server: update python bindings
5424 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5426 * gst/rtsp-server/rtsp-client.c:
5427 client: use the response from the clientstate
5428 Create the response object only once and store in the client state.
5429 Make all methods use the state response,
5431 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5433 * gst/rtsp-server/rtsp-server.c:
5434 server: use signal to keep track of clients
5435 Keep track of all the clients that the server creates and remove them when they
5436 fire the 'closed' signal.
5438 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5440 * gst/rtsp-server/rtsp-client.c:
5441 * gst/rtsp-server/rtsp-client.h:
5442 client: emit signal when closing
5444 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5446 * examples/.gitignore:
5447 * examples/Makefile.am:
5448 * examples/test-auth.c:
5449 * examples/test-video.c:
5450 * gst/rtsp-server/rtsp-auth.c:
5451 * gst/rtsp-server/rtsp-auth.h:
5452 * gst/rtsp-server/rtsp-client.c:
5453 * gst/rtsp-server/rtsp-media-factory.c:
5454 * gst/rtsp-server/rtsp-media.c:
5455 * gst/rtsp-server/rtsp-media.h:
5456 * gst/rtsp-server/rtsp-session-pool.h:
5457 * gst/rtsp-server/rtsp-session.h:
5458 media: enable per factory authorisations
5459 Allow for adding a GstRTSPAuth on the factory and media level and check
5460 permissions when accessing the factory.
5461 Add hints to the auth methods for future more fine grained authorisation.
5462 Add example application for per factory authentication.
5464 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5466 * gst/rtsp-server/rtsp-auth.c:
5467 * gst/rtsp-server/rtsp-auth.h:
5468 * gst/rtsp-server/rtsp-client.c:
5469 * gst/rtsp-server/rtsp-client.h:
5470 * gst/rtsp-server/rtsp-params.c:
5471 * gst/rtsp-server/rtsp-params.h:
5472 rtsp-server: Pass ClientState structure arround
5473 Pass the collected information for the ongoing request in a GstRTSPClientState
5474 structure that we can then pass around to simplify the method arguments. This
5475 will also be handy when we implement logging functionality.
5477 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5479 * gst/rtsp-server/rtsp-media-factory.c:
5480 * gst/rtsp-server/rtsp-media-factory.h:
5481 media-factory: add methods to configure authorisation
5483 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5485 * gst/rtsp-server/rtsp-client.c:
5486 client: unref auth in finalize
5488 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5490 * gst/rtsp-server/rtsp-server.c:
5491 server: unref auth in finalize
5493 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5495 * docs/libs/gst-rtsp-server-docs.sgml:
5496 * docs/libs/gst-rtsp-server-sections.txt:
5497 * docs/libs/gst-rtsp-server.types:
5500 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5502 * gst/rtsp-server/rtsp-server.c:
5503 * gst/rtsp-server/rtsp-server.h:
5504 server: separate create and accept
5505 Create separate create and accept methods so that subclasses can create custom
5507 Configure the server in the client object and prepare for keeping track of
5510 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5512 * gst/rtsp-server/rtsp-client.c:
5513 * gst/rtsp-server/rtsp-client.h:
5514 client: add support for setting the server.
5515 Add support for keeping a ref to the server that started this client
5518 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5520 * gst/rtsp-server/rtsp-auth.c:
5521 auth: fix memleak and add some docs
5522 Fix a memleak of the basic auth token.
5523 Add docs for the helper function
5525 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5527 * gst/rtsp-server/rtsp-auth.c:
5528 * gst/rtsp-server/rtsp-auth.h:
5529 * gst/rtsp-server/rtsp-client.c:
5530 client: delegate setup of auth to the manager
5531 Delegate the configuration of the authentication tokens to the manager object
5534 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5536 * examples/test-video.c:
5537 * gst/rtsp-server/Makefile.am:
5538 * gst/rtsp-server/rtsp-auth.c:
5539 * gst/rtsp-server/rtsp-auth.h:
5540 * gst/rtsp-server/rtsp-client.c:
5541 * gst/rtsp-server/rtsp-client.h:
5542 * gst/rtsp-server/rtsp-server.c:
5543 * gst/rtsp-server/rtsp-server.h:
5544 auth: add authentication object
5545 Add an object that can check the authorization of requests.
5546 Implement basic authentication.
5547 Add example authentication to test-video
5549 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5551 * gst/rtsp-server/rtsp-server.c:
5552 * gst/rtsp-server/rtsp-server.h:
5553 server: move includes back
5554 the includes are needed for sockaddr_in.
5556 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5558 * gst/rtsp-server/rtsp-client.c:
5559 * gst/rtsp-server/rtsp-client.h:
5560 * gst/rtsp-server/rtsp-server.c:
5561 * gst/rtsp-server/rtsp-server.h:
5562 rtsp: move network includes where they are needed
5564 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
5566 * gst/rtsp-server/rtsp-media.h:
5567 rtsp-media.h: Minor corrections in comments.
5570 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
5573 Automatic update of common submodule
5574 From e572c87 to f94d739
5576 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5580 * docs/libs/.gitignore:
5581 * examples/.gitignore:
5582 * gst/rtsp-server/.gitignore:
5585 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5587 * docs/libs/Makefile.am:
5588 docs: We don't build ps/pdf for API reference docs
5590 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5593 Automatic update of common submodule
5594 From ccbaa85 to e572c87
5596 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5599 Automatic update of common submodule
5600 From 46445ad to ccbaa85
5602 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5604 * gst/rtsp-server/Makefile.am:
5605 * gst/rtsp-server/fs-funnel.c:
5606 * gst/rtsp-server/fs-funnel.h:
5607 * gst/rtsp-server/rtsp-funnel.c:
5608 * gst/rtsp-server/rtsp-funnel.h:
5609 * gst/rtsp-server/rtsp-media.c:
5610 funnel: rename fsfunnel to rtspfunnel
5611 Rename the funnel to avoid conflicts with the farsight one.
5613 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5615 * gst/rtsp-server/Makefile.am:
5616 * gst/rtsp-server/fs-funnel.c:
5617 * gst/rtsp-server/fs-funnel.h:
5618 * gst/rtsp-server/rtsp-media.c:
5619 rtsp-media: add and use fsfunnel
5620 Add a copy of fsfunnel to the build because input-selector removed the (broken)
5621 select-all property that we need.
5623 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5625 * gst/rtsp-server/Makefile.am:
5626 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
5627 Use PKG_CONFIG_PATH specified at configure time (if any) as well
5628 for the g-ir-compiler, rather than just assuming the env var has
5631 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5638 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
5640 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5643 * gst/rtsp-server/Makefile.am:
5644 gobject-introspection: fix g-i build for uninstalled setup
5645 Requires gst-plugins-base git (> 0.10.31.2).
5647 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5649 * examples/test-uri.c:
5650 examples: add some more options and comments
5652 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5654 * gst/rtsp-server/rtsp-media-factory-uri.c:
5655 factory-uri: use right property type
5657 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5659 * gst/rtsp-server/rtsp-media-factory-uri.c:
5660 factory-uri: attempt to configure buffer-lists
5661 Attempt to configure buffer lists in the payloader for improved performance.
5663 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5665 * gst/rtsp-server/rtsp-media.c:
5666 media: attempt to configure bigger UDP buffers
5667 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
5668 send buffers with high bitrate streams.
5670 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
5672 * gst/rtsp-server/rtsp-client.c:
5673 client: use the socket length from getsockname
5674 Use the length returned by getsockname to perform the getnameinfo call because
5675 the size can depend on the socket type and platform.
5678 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5680 * docs/libs/gst-rtsp-server-docs.sgml:
5681 * docs/libs/gst-rtsp-server-sections.txt:
5682 docs: add uri factory to the docs
5684 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5686 * gst/rtsp-server/rtsp-client.c:
5687 * gst/rtsp-server/rtsp-media.h:
5690 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5692 * gst/rtsp-server/rtsp-client.c:
5693 * gst/rtsp-server/rtsp-media.c:
5694 * gst/rtsp-server/rtsp-media.h:
5695 * gst/rtsp-server/rtsp-session.c:
5696 * gst/rtsp-server/rtsp-session.h:
5697 rtsp-server: add support for buffer lists
5698 Add support for sending bufferlists received from appsink.
5701 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5703 * gst/rtsp-server/rtsp-client.c:
5704 * gst/rtsp-server/rtsp-media.c:
5705 * gst/rtsp-server/rtsp-media.h:
5706 * gst/rtsp-server/rtsp-sdp.c:
5707 media: make method to retrieve the play range
5708 Make a method to retrieve the playback range so that we can conditionally create
5709 a different range for the SDP and the PLAY requests.
5711 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5713 * gst/rtsp-server/rtsp-media.c:
5714 * gst/rtsp-server/rtsp-media.h:
5715 media: add signal to notify of state changes
5717 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5719 * gst/rtsp-server/rtsp-client.h:
5720 client: cleanup headers
5722 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5724 * gst/rtsp-server/rtsp-client.c:
5727 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5729 * gst/rtsp-server/rtsp-media-factory-uri.c:
5730 * gst/rtsp-server/rtsp-media-factory-uri.h:
5731 factory-uri: add support for gstpay
5732 Add an option to prefer gstpay over decoder + raw payloader.
5734 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5736 * gst/rtsp-server/rtsp-media-factory-uri.c:
5737 * gst/rtsp-server/rtsp-media-factory-uri.h:
5738 factory-uri: rework the autoplugger.
5739 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
5742 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5744 * gst/rtsp-server/rtsp-media-factory-uri.c:
5745 factory-uri: use better factory filter
5746 Make better payloader filter based on autoplug rank and RTP use case.
5748 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5751 Automatic update of common submodule
5752 From 169462a to 46445ad
5754 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5756 * gst/rtsp-server/rtsp-server.c:
5757 server: set SO_REUSEADDR before bind
5758 Set the SO_REUSEADDR _before_ bind() to make it actually work.
5760 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5762 * gst/rtsp-server/rtsp-media.c:
5763 * gst/rtsp-server/rtsp-media.h:
5764 media: emit prepared signal when prepared
5765 Make a 'prepared' signal and emit it when we successfully prepared the element.
5766 This signal can be used to configure the media object after it has been prepared
5769 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
5772 Automatic update of common submodule
5773 From 011bcc8 to 169462a
5775 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
5777 python an optional dependency
5778 * configure.ac: Move up valgrind and g-i checks. Make the python
5779 dependency optional, as it was before.
5781 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5783 Merge branch 'master' into 0.11
5788 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5790 * gst/rtsp-server/rtsp-media.c:
5791 media: update range when active clients changed
5792 When we changed the number of active clients, update the current range
5793 information because we want the second client connecting to a shared resource
5794 continue from where the stream currently.
5796 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5798 * gst/rtsp-server/rtsp-media-factory-uri.c:
5799 * gst/rtsp-server/rtsp-media-factory-uri.h:
5800 factory-uri: add colorspace and fix pt
5801 Rework the way we pass data to the autoplugger.
5802 When we have raw caps, plug a converter element to make pluggin to raw
5803 payloaders more successful.
5804 Make sure all dynamically plugged payloaders have a unique payload types.
5806 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5808 * examples/Makefile.am:
5809 * examples/test-uri.c:
5810 example: add example of the uri factory
5812 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5814 * gst/rtsp-server/Makefile.am:
5815 * gst/rtsp-server/rtsp-media-factory-uri.c:
5816 * gst/rtsp-server/rtsp-media-factory-uri.h:
5817 * gst/rtsp-server/rtsp-server.h:
5818 factory-uri: add a factory to stream any URI
5819 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
5822 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5824 * gst/rtsp-server/rtsp-media.c:
5825 * gst/rtsp-server/rtsp-media.h:
5826 media: ignore spurious ASYNC_DONE messages
5827 When we are dynamically adding pads, the addition of the udpsrc elements will
5828 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
5829 the real ASYNC_DONE when everything is prerolled.
5831 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5833 * gst/rtsp-server/rtsp-media-factory.c:
5834 * gst/rtsp-server/rtsp-media-factory.h:
5835 media-factory: make lock macro
5837 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
5839 * gst/rtsp-server/rtsp-client.c:
5840 rtsp-server: Remove unused variable and dead assignment
5842 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
5844 * examples/test-launch.c:
5845 * examples/test-mp4.c:
5846 * examples/test-ogg.c:
5847 * examples/test-readme.c:
5848 * examples/test-sdp.c:
5849 * examples/test-video.c:
5850 examples: Run gst-indent
5852 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
5854 * gst/rtsp-server/rtsp-client.c:
5855 * gst/rtsp-server/rtsp-media-factory.c:
5856 * gst/rtsp-server/rtsp-media-mapping.c:
5857 * gst/rtsp-server/rtsp-media.c:
5858 * gst/rtsp-server/rtsp-params.c:
5859 * gst/rtsp-server/rtsp-sdp.c:
5860 * gst/rtsp-server/rtsp-server.c:
5861 * gst/rtsp-server/rtsp-session-pool.c:
5862 * gst/rtsp-server/rtsp-session.c:
5863 rtsp-server: Run gst-indent
5864 Since it wasn't using the upstream common previously, there was no
5865 indentation check before commiting.
5867 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
5869 * gst/rtsp-server/rtsp-media-mapping.h:
5870 * gst/rtsp-server/rtsp-media.c:
5871 * gst/rtsp-server/rtsp-media.h:
5872 * gst/rtsp-server/rtsp-sdp.c:
5873 * gst/rtsp-server/rtsp-session-pool.h:
5874 * gst/rtsp-server/rtsp-session.c:
5875 * gst/rtsp-server/rtsp-session.h:
5876 rtsp-server: Some more doc fixups
5878 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5881 Makefile: Add cruft-cleaning support
5883 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5888 * docs/libs/Makefile.am:
5889 * docs/libs/gst-rtsp-server-docs.sgml:
5890 * docs/libs/gst-rtsp-server-sections.txt:
5891 * docs/libs/gst-rtsp-server.types:
5892 * docs/version.entities.in:
5893 docs: Add gtk-doc build system
5895 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5897 * gst/rtsp-server/Makefile.am:
5898 Makefile.am: Use standard GIR make behaviour
5900 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5904 autogen/configure: Bring more in sync to standard gst module behaviour
5906 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5908 * gst/rtsp-server/rtsp-media.c:
5909 media: warn and fail when gstrtpbin is not found
5911 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5914 configure: open 0.11 branch
5916 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
5920 Add common submodule
5922 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
5925 * common/Makefile.am:
5926 * common/c-to-xml.py:
5928 * common/coverage/coverage-report-entry.pl:
5929 * common/coverage/coverage-report.pl:
5930 * common/coverage/coverage-report.xsl:
5931 * common/coverage/lcov.mak:
5932 * common/gettext.patch:
5933 * common/glib-gen.mak:
5934 * common/gst-autogen.sh:
5935 * common/gst-xmlinspect.py:
5937 * common/gstdoc-scangobj:
5938 * common/gtk-doc-plugins.mak:
5939 * common/gtk-doc.mak:
5940 * common/m4/.gitignore:
5941 * common/m4/Makefile.am:
5943 * common/m4/as-ac-expand.m4:
5944 * common/m4/as-auto-alt.m4:
5945 * common/m4/as-compiler-flag.m4:
5946 * common/m4/as-compiler.m4:
5947 * common/m4/as-docbook.m4:
5948 * common/m4/as-libtool-tags.m4:
5949 * common/m4/as-libtool.m4:
5950 * common/m4/as-python.m4:
5951 * common/m4/as-scrub-include.m4:
5952 * common/m4/as-version.m4:
5953 * common/m4/ax_create_stdint_h.m4:
5954 * common/m4/check.m4:
5955 * common/m4/glib-gettext.m4:
5956 * common/m4/gst-arch.m4:
5957 * common/m4/gst-args.m4:
5958 * common/m4/gst-check.m4:
5959 * common/m4/gst-debuginfo.m4:
5960 * common/m4/gst-default.m4:
5961 * common/m4/gst-doc.m4:
5962 * common/m4/gst-error.m4:
5963 * common/m4/gst-feature.m4:
5964 * common/m4/gst-function.m4:
5965 * common/m4/gst-gettext.m4:
5966 * common/m4/gst-glib2.m4:
5967 * common/m4/gst-libxml2.m4:
5968 * common/m4/gst-plugindir.m4:
5969 * common/m4/gst-valgrind.m4:
5970 * common/m4/gtk-doc.m4:
5971 * common/m4/introspection.m4:
5973 * common/mangle-tmpl.py:
5974 * common/plugins.xsl:
5976 * common/release.mak:
5977 * common/scangobj-merge.py:
5978 * common/upload.mak:
5979 common: Remove static version
5981 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
5983 * common/m4/introspection.m4:
5984 Update introspection.m4 to match usage
5986 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5990 Remove old stuff from the README
5992 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5997 === release 0.10.7 ===
5999 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6004 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6006 * examples/test-ogg.c:
6007 test-ogg: remove parsers
6008 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
6009 buffers with timestamps. Using the parsers also seems to break things.
6011 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6013 * bindings/vala/gst-rtsp-server-0.10.vapi:
6014 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6015 Updated Vala bindings
6017 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6019 * common/m4/introspection.m4:
6021 * gst/rtsp-server/Makefile.am:
6022 Added initial gobject-introspection support
6024 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6026 * gst/rtsp-server/rtsp-media-factory.c:
6027 media-factory: don't use host for shared hash key
6028 When we generate the key to share made between connections, don't include the
6029 host used to connect so that we can share media even if between clients that
6030 connected with localhost and ones with the ip address.
6032 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6034 * bindings/vala/Makefile.am:
6035 build: fix distcheck
6037 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6039 * bindings/vala/gst-rtsp-server-0.10.vapi:
6040 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6041 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6042 Update Vala bindings
6044 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6046 * bindings/vala/Makefile.am:
6048 Fix configure checks and installation location for Vala bindings
6051 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6056 === release 0.10.6 ===
6058 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6061 configure: release 0.10.6
6063 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6065 * gst/rtsp-server/rtsp-media.c:
6066 media: help the compiler a little
6068 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6070 * gst/rtsp-server/rtsp-media.c:
6071 * gst/rtsp-server/rtsp-media.h:
6072 * gst/rtsp-server/rtsp-session.c:
6073 media: cleanup media transport before freeing
6074 Cleanup the media transport data before freeing. In particular, remove the qdata
6075 from the rtpsource object.
6077 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6079 * gst/rtsp-server/rtsp-media-factory.c:
6080 * gst/rtsp-server/rtsp-media-factory.h:
6081 * gst/rtsp-server/rtsp-media.c:
6082 * gst/rtsp-server/rtsp-media.h:
6083 media-factory: add eos-shutdown property
6084 Add an eos-shutdown property that will send an EOS to the pipeline before
6085 shutting it down. This allows for nice cleanup in case of a muxer.
6088 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6090 * gst/rtsp-server/rtsp-media.c:
6091 * gst/rtsp-server/rtsp-media.h:
6092 media: use multiudpsink send-duplicates when we can
6093 If we have a new enough multiudpsink with the send-duplicates property, use this
6094 instead of doing our own filtering. Our custom filtering code should eventually
6095 be removed when we can depend on a released -good.
6097 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6099 * gst/rtsp-server/rtsp-media.c:
6100 media: don't leak destinations
6101 Refactor and cleanup the destinations array when the stream is destroyed.
6103 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6105 * gst/rtsp-server/rtsp-media.c:
6106 * gst/rtsp-server/rtsp-media.h:
6107 media: don't add udp addresses multiple times
6108 Keep track of the udp addresses we added to udpsink and never add the same udp
6109 destination twice. This avoids duplicate packets when using multicast.
6111 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6113 * gst/rtsp-server/rtsp-server.c:
6114 server: disable use of SO_LINGER
6115 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
6116 server close()s the connection.
6118 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6120 * gst/rtsp-server/rtsp-server.c:
6121 server: use 5 second linger period in SO_LINGER
6122 Wait 5 seconds before clearing the send buffers and reseting the connection with
6123 the client when we do a close. This should be enough time to get the message to
6127 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6129 * gst/rtsp-server/rtsp-server.c:
6130 server: use SO_LINGER
6131 SO_LINGER on the socket will make sure that any pending data on the socket is
6132 flushed ASAP and that the socket connection is reset. This makes sure that the
6133 socket can be reused immediately.
6136 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6139 README: add blurb about shared media factories
6141 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
6143 * gst/rtsp-server/rtsp-media.c:
6144 Add stdlib.h for atoi()
6146 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6148 * bindings/python/Makefile.am:
6149 * bindings/vala/Makefile.am:
6150 build: distcheck fixes
6151 Fix 'make distcheck', somewhat (it still fails because it tries to
6152 install files into /usr/share/vala/vapi/ irrespective of the
6155 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6158 configure: bump core/base requirements to released version
6159 Makes things less confusing for people.
6161 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6164 configure: fail if GStreamer core/base requirements are not met
6166 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6168 * gst/rtsp-server/rtsp-client.c:
6169 client: improve client cleanups
6170 Make sure the session does not timeout when using TCP. We need to do this
6171 because quicktime player does not send RTCP for some reason in tunneled
6173 Refactor some cleanup code.
6176 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6178 * gst/rtsp-server/rtsp-session.c:
6179 * gst/rtsp-server/rtsp-session.h:
6180 session: add support for prevent session timeouts
6181 Add an atomix counter to prevent session timeouts when we are, for example,
6184 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6186 * gst/rtsp-server/rtsp-client.c:
6187 client: fix unlink on session timeouts
6188 When our session times out, make sure we unlink all streams in this
6190 Remove the tunnelid when closing the connection.
6192 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6194 * gst/rtsp-server/rtsp-session.c:
6195 session: small cleanups
6197 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6199 * gst/rtsp-server/rtsp-client.c:
6200 client: handle lost_tunnel callbacks
6201 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
6202 hashtable so that we can reuse it for when the client reopens the POST
6204 Close the connection after a TEARDOWN.
6205 Make sure or watchid is cleared when the watch is removed.
6208 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6210 * gst/rtsp-server/rtsp-client.c:
6211 * gst/rtsp-server/rtsp-media.c:
6212 * gst/rtsp-server/rtsp-sdp.c:
6213 rtsp-server: add more support for multicast
6215 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6218 * gst/rtsp-server/rtsp-media.c:
6219 * gst/rtsp-server/rtsp-media.h:
6220 media: allow configuration of allowed lower transport
6222 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6224 * gst/rtsp-server/rtsp-client.h:
6225 * gst/rtsp-server/rtsp-media.c:
6226 * gst/rtsp-server/rtsp-media.h:
6227 * gst/rtsp-server/rtsp-sdp.c:
6228 * gst/rtsp-server/rtsp-sdp.h:
6229 * gst/rtsp-server/rtsp-server.c:
6230 rtsp: keep track of server ip and ipv6
6231 Keep track of how the client connected to the server and setup the udp ports
6232 with the same protocol.
6233 Copy the server ip address in the SDP so that clients can send RTCP back to
6236 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6238 * gst/rtsp-server/rtsp-session.c:
6241 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6243 * gst/rtsp-server/rtsp-client.c:
6244 client: use right size for malloc
6246 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6248 * gst/rtsp-server/rtsp-server.c:
6249 server: comment ipv6 server listening address
6251 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6253 * gst/rtsp-server/rtsp-media.c:
6254 media: allow for ipv6 sockets
6256 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6258 * gst/rtsp-server/rtsp-server.c:
6259 * gst/rtsp-server/rtsp-server.h:
6260 server: rework server part
6261 Allow setting a bind address, make sure we can deal with ipv6.
6262 Remove the port property and change with the service property.
6264 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6266 * gst/rtsp-server/rtsp-media.h:
6267 media: update comments a little
6269 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6271 * gst/rtsp-server/rtsp-client.c:
6272 client: make content-base better
6273 Use the URI formatting functions to make a content-base. Also make sure that
6274 there is a trailing / at the end.
6276 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6278 * gst/rtsp-server/rtsp-client.c:
6279 client: guard against invalid paths
6281 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6283 * examples/test-video.c:
6284 test: catch server bind errors
6286 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
6288 * gst/rtsp-server/rtsp-media.c:
6289 rtspmedia: emit "unprepared" if _prepare fails.
6290 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
6291 media object is removed from its factory's cache.
6293 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6295 * gst/rtsp-server/rtsp-media.c:
6296 media: collect media position when seek completes
6298 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
6300 * gst/rtsp-server/rtsp-client.c:
6301 client: call unlink_streams in client finalize
6304 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6306 * gst/rtsp-server/rtsp-media.c:
6307 media: limit the time to wait to something huge
6308 Avoid waiting forever but limit the timeout to 20 seconds.
6310 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6312 * gst/rtsp-server/rtsp-sdp.c:
6313 sdp: reindent and check for prepared status
6315 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6317 * gst/rtsp-server/rtsp-media.c:
6318 * gst/rtsp-server/rtsp-media.h:
6319 * gst/rtsp-server/rtsp-session.c:
6320 media: avoid doing _get_state() for state changes
6321 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
6322 until the media is prerolled or in error. This avoids doing a blocking call of
6323 gst_element_get_state() that can cause lockups when there is an error.
6326 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6328 * gst/rtsp-server/rtsp-media.c:
6331 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6333 * gst/rtsp-server/rtsp-media-factory.c:
6334 media-factory: better error handling
6335 Improve the error handling a bit.
6337 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6339 * gst/rtsp-server/rtsp-client.c:
6340 client: rework transport parsing
6341 Rework the transport parsing code so that we can ignore transports we don't
6342 support instead of just picking the first one we can parse.
6343 Configure a (for now hardcoded) destination for multicast transports.
6345 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6347 * gst/rtsp-server/rtsp-media.c:
6348 media: set multicast sink parameters
6349 Disable loop and automatic multicast join on the udpsink elements.
6350 Add some more debug info.
6351 Reset some state variables in the right place.
6352 Use the right port numbers for multicast.
6354 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6356 * gst/rtsp-server/rtsp-session.c:
6357 session: handle transport setup correctly
6358 Handle UDP, MCAST and TCP transport negotiation more correctly.
6359 Store the server session SSRC in the transport.
6361 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6363 * gst/rtsp-server/rtsp-client.c:
6364 rtsp-client: implement error_full
6365 Implement error_full to avoid some segfaults when the rtspconnection calls it.
6368 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6371 * gst/rtsp-server/rtsp-client.c:
6372 * gst/rtsp-server/rtsp-server.c:
6373 docs: update docs and comments
6375 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
6377 * gst/rtsp-server/rtsp-sdp.c:
6378 sdp: make server work better when behind a proxy
6380 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6382 * gst/rtsp-server/rtsp-client.c:
6383 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
6385 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6387 * gst/rtsp-server/rtsp-client.c:
6388 * gst/rtsp-server/rtsp-media-factory.c:
6389 * gst/rtsp-server/rtsp-media-mapping.c:
6390 * gst/rtsp-server/rtsp-media.c:
6391 * gst/rtsp-server/rtsp-server.c:
6392 * gst/rtsp-server/rtsp-session-pool.c:
6393 * gst/rtsp-server/rtsp-session.c:
6394 Use GStreamer's debugging subsystem
6396 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6398 * gst/rtsp-server/rtsp-media-factory.c:
6399 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
6401 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6406 === release 0.10.5 ===
6408 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6413 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6416 configure: bump required versions
6418 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
6420 * gst/rtsp-server/rtsp-client.c:
6421 client: call weak-unref on client->sessions from finalize
6424 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6426 * gst/rtsp-server/rtsp-media.c:
6427 media: Fixed crasher where caps got unref'ed too often
6429 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6432 * pkgconfig/.gitignore:
6433 * pkgconfig/Makefile.am:
6434 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6435 Added pkg-config file to use gst-rtsp-server uninstalled
6437 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6439 * gst/rtsp-server/rtsp-media.c:
6440 media: add some docs
6442 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
6444 * gst/rtsp-server/rtsp-client.c:
6445 rtsp: Use gst_rtsp_watch_send_message().
6446 Use gst_rtsp_watch_send_message() since the old API which used
6447 gst_rtsp_watch_queue_message() has been deprecated.
6449 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6454 === release 0.10.4 ===
6456 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6461 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6463 * gst/rtsp-server/rtsp-client.c:
6464 * gst/rtsp-server/rtsp-session.c:
6465 * gst/rtsp-server/rtsp-session.h:
6466 rtsp: allocate channels in TCP mode
6467 When the client does not provide us with channels in TCP mode, allocate channels
6470 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6472 * gst/rtsp-server/rtsp-client.c:
6473 client: don't crash when tunnelid is missing
6474 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
6475 don't crash but return an error response to the client.
6478 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6480 * bindings/vala/gst-rtsp-server-0.10.vapi:
6481 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6482 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6483 bindings: update vala bindings with new method
6485 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6487 * gst/rtsp-server/rtsp-session-pool.c:
6488 * gst/rtsp-server/rtsp-session-pool.h:
6489 sessionpool: add function to filter sessions
6490 Add generic function to retrieve/remove sessions.
6492 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6495 configure: bump core/base requirements to release
6497 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6499 * gst/rtsp-server/rtsp-media.c:
6500 media: fix indentation
6502 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6504 * gst/rtsp-server/rtsp-media.c:
6505 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
6507 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6509 * gst/rtsp-server/rtsp-media.c:
6510 set state and remove elements of media in for loop
6512 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
6514 * bindings/vala/gst-rtsp-server-0.10.vapi:
6515 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6516 Added gst_rtsp_media_remove_elements function to Vala bindings
6518 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
6520 * gst/rtsp-server/rtsp-media.c:
6521 * gst/rtsp-server/rtsp-media.h:
6522 Added gst_rtsp_media_remove_elements function
6524 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
6526 * gst/rtsp-server/rtsp-media.c:
6527 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
6529 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6531 * bindings/vala/gst-rtsp-server-0.10.vapi:
6532 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6533 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6534 Updated Vala bindings
6536 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6538 * gst/rtsp-server/rtsp-media.c:
6539 * gst/rtsp-server/rtsp-media.h:
6540 Added vmethod unprepare to GstRTSPMedia
6541 The default implementation sets the state of the pipeline to GST_STATE_NULL
6543 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6545 * gst/rtsp-server/rtsp-media-factory.c:
6546 * gst/rtsp-server/rtsp-media-factory.h:
6547 Made collect_streams function public
6549 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6551 * gst/rtsp-server/rtsp-media-factory.c:
6552 * gst/rtsp-server/rtsp-media-factory.h:
6553 * gst/rtsp-server/rtsp-media.c:
6554 Added vmethod create_pipeline to GstRTSPMediaFactory
6555 The pipeline is created in this method and the GstRTSPMedia's element is added to it
6557 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6559 * gst/rtsp-server/rtsp-client.c:
6560 client: use g_source_destroy()
6561 We need to use g_source_destroy() because we might have added the source to a
6562 different main context than the default one.
6564 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6566 * gst/rtsp-server/Makefile.am:
6567 * gst/rtsp-server/rtsp-client.c:
6568 * gst/rtsp-server/rtsp-params.c:
6569 * gst/rtsp-server/rtsp-params.h:
6570 rtsp: prepare for handling GET/SET_PARAMETER
6571 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
6573 Fix return codes of handlers.
6575 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6577 * gst/rtsp-server/rtsp-media.c:
6578 media: don't leak session pads
6580 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6582 * gst/rtsp-server/rtsp-media.c:
6583 media: clean up the messages a bit
6585 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6587 * gst/rtsp-server/rtsp-sdp.c:
6588 sdp: warn and skip streams without media
6590 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6592 * bindings/vala/gst-rtsp-server-0.10.vapi:
6593 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6594 vala: Fixed typo in header file of RTSPMediaStream
6596 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6598 * gst/rtsp-server/rtsp-media.c:
6601 Make dumping RTCP stats configurable
6603 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6605 * gst/rtsp-server/rtsp-media.c:
6606 media: be less verbose and leak less
6608 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6610 * gst/rtsp-server/rtsp-media.c:
6611 media: don't leak the destination address
6613 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6615 * gst/rtsp-server/rtsp-client.c:
6616 * gst/rtsp-server/rtsp-media.c:
6617 * gst/rtsp-server/rtsp-media.h:
6618 * gst/rtsp-server/rtsp-session.c:
6619 * gst/rtsp-server/rtsp-session.h:
6620 rtsp: use RTCP to keep the session alive
6621 Use the RTCP rtcp-from stats field to find the associated session and use this
6622 to keep the session alive.
6624 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6626 * gst/rtsp-server/rtsp-session.c:
6627 session: add 5sec to the real session timeout
6628 Allow the session to live 5sec longer before really timing out. This should give
6629 clients some extra time to keep the session active.
6631 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6633 * gst/rtsp-server/rtsp-client.c:
6634 client: replay OK to GET/SET_PARAMETER
6635 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
6636 so that we return OK for those requests.
6638 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6640 * gst/rtsp-server/rtsp-media.c:
6641 * gst/rtsp-server/rtsp-media.h:
6642 media: keep track of active transports
6643 Keep track of which transport is active to avoid closing the connection too
6645 Remove the destination transport also when going to NULL.
6646 Print some stats about the SDES and other RTCP messages we receive from the
6649 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6651 * examples/.gitignore:
6652 * examples/Makefile.am:
6653 * examples/test-sdp.c:
6654 example: add SDP relay example
6656 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6658 * gst/rtsp-server/rtsp-media.c:
6659 media: also count active TCP connections
6661 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6663 * gst/rtsp-server/rtsp-media-factory.c:
6664 * gst/rtsp-server/rtsp-media.c:
6665 * gst/rtsp-server/rtsp-media.h:
6666 rtsp: add support for dynamic elements
6667 Add support for dynamic elements.
6668 Don't set live pipelines back to paused.
6670 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6672 * gst/rtsp-server/rtsp-sdp.c:
6673 sdp: don't add encoding name when absent in caps
6675 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6677 * gst/rtsp-server/rtsp-client.c:
6678 client: warn when we can't do RTP-Info
6680 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6682 * gst/rtsp-server/rtsp-media-factory.c:
6683 factory: factor out the stream construction
6685 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6687 * gst/rtsp-server/rtsp-client.c:
6688 client: only add RTP-Info when we have the info
6689 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
6692 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6697 === release 0.10.3 ===
6699 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6703 - Fixes a bug where it put the wrong verion in pkgconfig
6704 - Link RTP and RTCP sources
6706 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6708 * gst/rtsp-server/rtsp-media.c:
6709 * gst/rtsp-server/rtsp-media.h:
6710 media: link the RTP udpsrc to the session manager
6711 Link the RTP udpsrc and the appsrc to the session manager so that they don't
6712 shut down when the client sends a packet to open firewalls.
6714 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6716 * pkgconfig/gst-rtsp-server.pc.in:
6717 Don't use hard-coded version number in pkg-config file
6719 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6724 === release 0.10.2 ===
6726 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6731 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6734 * common/m4/.gitignore:
6735 * examples/.gitignore:
6736 * pkgconfig/.gitignore:
6737 add some .gitignore files
6739 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6741 * gst/rtsp-server/rtsp-media.c:
6742 media: seek to key frames
6744 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6746 * gst/rtsp-server/rtsp-media.c:
6747 media: emit the unprepared signal by id
6748 Emit the unprepared signal by id instead of name and set the media as
6751 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6753 * gst/rtsp-server/rtsp-media.c:
6754 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
6756 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6758 * gst/rtsp-server/rtsp-server.c:
6759 Added finalize function to GstRTPSPServer to unref session pool and media mapping
6761 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6763 * bindings/vala/gst-rtsp-server-0.10.vapi:
6764 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6765 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6766 Updated vala bindings
6768 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6770 * gst/rtsp-server/Makefile.am:
6771 * gst/rtsp-server/rtsp-client.c:
6772 * gst/rtsp-server/rtsp-media.c:
6773 server: use appsink and appsrc with the API
6774 Use the appsink/appsrc API instead of the signals for higher
6777 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6779 * examples/test-ogg.c:
6780 tests: set the payload type correctly
6782 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6784 * gst/rtsp-server/rtsp-media-factory.c:
6785 factory: connect to the unprepare signal
6786 Connect to the unprepare signal for non-reusable media so that we can remove
6787 them from the cache.
6789 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6791 * gst/rtsp-server/rtsp-media.c:
6792 * gst/rtsp-server/rtsp-media.h:
6793 media: add signal to notify of unprepare
6795 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6797 * gst/rtsp-server/rtsp-media.c:
6798 * gst/rtsp-server/rtsp-media.h:
6799 media: more work on making the media shared
6800 Add a reusable flag to medias, indicating that they can be reused after a state
6804 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6806 * examples/test-readme.c:
6807 examples: mark the example as shared for testing
6809 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6811 * gst/rtsp-server/rtsp-media.c:
6812 * gst/rtsp-server/rtsp-media.h:
6813 client: support shared media
6814 Always perform the state actions even if the target state of the pipeline is
6815 already correct, we still want to add/remove the transports when we are dealing
6817 Keep a counter of the number of active transports for a media so that we can use
6818 this to perform a state change when needed.
6819 Perform a state change of the pipeline only when the first transport was added
6820 or when there are no active transports.
6822 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6824 * gst/rtsp-server/rtsp-client.c:
6825 client: fix refcounting crasher
6826 Don't need to remove the weak refs in the finalize methods, they are already
6827 removed in the dispose.
6828 Don't register the callback with a DestroyNofity.
6830 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6832 * gst/rtsp-server/rtsp-client.c:
6833 Fix rtsp client refcount management in TCP mode.
6834 Don't unref a client ref we never had. Fixes an unref
6835 of an already-free client object after a client
6836 teardown request for me.
6838 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6840 * gst/rtsp-server/rtsp-session.c:
6841 docs: fix typo in API docs
6843 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6845 * gst/rtsp-server/rtsp-media.c:
6847 Keep the udp sources in playing even if we go to paused. unlock the sources when
6849 Add some more debug info.
6850 Only seek when we need to.
6851 Keep track of the position when we go to paused.
6853 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6855 * gst/rtsp-server/rtsp-client.c:
6856 * gst/rtsp-server/rtsp-media.c:
6857 * gst/rtsp-server/rtsp-media.h:
6858 Add beginnings of seeking.
6859 Parse the Range header and perform a seek on the pipeline for the requested
6860 position. It's disabled currently until I figure out what's going wrong.
6862 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6864 * gst/rtsp-server/rtsp-client.c:
6865 allow pause requests for now.
6868 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6870 * gst/rtsp-server/rtsp-client.c:
6871 Remove weak ref on the session in teardown
6872 We need to remove our weakref from the session when we do a teardown because
6873 else we close the TCP connection prematurely.
6875 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6877 * gst/rtsp-server/rtsp-client.c:
6878 * gst/rtsp-server/rtsp-client.h:
6879 * gst/rtsp-server/rtsp-session-pool.c:
6880 Do some more session cleanup
6881 Make session timeout kill the TCP connection that currently watches the
6883 Remove the client timeout property.
6885 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6887 * gst/rtsp-server/rtsp-client.c:
6888 * gst/rtsp-server/rtsp-client.h:
6889 * gst/rtsp-server/rtsp-media.c:
6890 * gst/rtsp-server/rtsp-media.h:
6891 * gst/rtsp-server/rtsp-server.c:
6892 * gst/rtsp-server/rtsp-session.c:
6893 * gst/rtsp-server/rtsp-session.h:
6895 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
6898 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6900 * examples/Makefile.am:
6901 * examples/test-launch.c:
6902 Add example server that takes launch lines
6903 Add an example server that streams any -launch line.
6905 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6907 * examples/test-readme.c:
6908 * gst/rtsp-server/rtsp-client.c:
6909 * gst/rtsp-server/rtsp-media.c:
6910 * gst/rtsp-server/rtsp-media.h:
6911 Add support for live streams
6912 Add support for live streams and ranges
6913 Start on handling TCP data transfer.
6915 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6917 * gst/rtsp-server/rtsp-media.c:
6918 Free the pipeline before other things
6921 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6923 * gst/rtsp-server/rtsp-client.c:
6924 Only free the pending tunnel if there is one
6927 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6929 * gst/rtsp-server/rtsp-client.c:
6930 * gst/rtsp-server/rtsp-client.h:
6931 * gst/rtsp-server/rtsp-media.c:
6932 rtsp-server: Add support for tunneling
6933 Add support for tunneling over HTTP.
6934 Use new connection methods to retrieve the url.
6935 Dispatch messages based on the message type instead of blindly
6936 assuming it's always a request.
6937 Keep track of the watch id so that we can remove it later.
6938 Set the media pipeline to NULL before unreffing the pipeline.
6940 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6942 * gst/rtsp-server/rtsp-client.c:
6943 * gst/rtsp-server/rtsp-client.h:
6944 Fix for channel -> watch rename in gstreamer
6945 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
6947 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6949 * gst/rtsp-server/rtsp-client.c:
6950 * gst/rtsp-server/rtsp-client.h:
6952 Use the async RTSP channels instead of spawning a new thread for each client.
6953 If a sessionid is specified in a request, fail if we don't have the session.
6955 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6957 * gst/rtsp-server/rtsp-media.c:
6958 Add better debug info
6959 Add some better debug info.
6961 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6963 * examples/test-video.c:
6965 Add support for session timeouts in the example.
6967 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6969 * gst/rtsp-server/rtsp-session-pool.c:
6970 * gst/rtsp-server/rtsp-session-pool.h:
6971 Pass GTimeVal around for performance reasons
6972 Get the current time only once and pass it around so that sessions don't have to
6973 get the current time anymore.
6974 Add experimental support for a GSource that dispatches when the session needs to
6977 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6979 * gst/rtsp-server/rtsp-session.c:
6980 * gst/rtsp-server/rtsp-session.h:
6981 Add better support for session timeouts
6982 Add a method to request the number of milliseconds when a session will timeout.
6984 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6986 * gst/rtsp-server/rtsp-media.c:
6987 * gst/rtsp-server/rtsp-media.h:
6988 Add suport for RTP manager monitoring
6989 Add the first stage in monitoring the rtp manager.
6990 Make sure we don't update the state to something we don't want.
6992 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6994 * gst/rtsp-server/rtsp-client.c:
6995 Add support for session keepalive
6996 Get and update the session timeout for all requests. get the session as early as
6999 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7001 * gst/rtsp-server/rtsp-media-factory.h:
7002 * gst/rtsp-server/rtsp-media.c:
7003 * gst/rtsp-server/rtsp-media.h:
7004 Handle media bus messages
7005 Handle media bus messages in a custom mainloop and dispatch them to the
7006 RTSPMedia objects. Let the default implementation handle some common messages.
7008 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7010 * gst/rtsp-server/rtsp-client.c:
7011 * gst/rtsp-server/rtsp-session-pool.c:
7012 * gst/rtsp-server/rtsp-session.c:
7013 Some more session timeout handling
7014 Move the session header setting code to a central place so that we always add
7015 the timeout parameter too.
7016 Handle timeouts by running the session cleanup code.
7017 Stop media before cleaning up.
7019 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7021 * gst/rtsp-server/rtsp-client.c:
7022 * gst/rtsp-server/rtsp-client.h:
7023 Add timeout property
7024 Add a timeout property ot the client and make the other properties into GObject
7027 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7029 * gst/rtsp-server/rtsp-session-pool.c:
7030 Use getters and setters in property code
7031 Use the getters and setters for the timeout property instead of locking
7034 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7036 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
7038 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7040 * gst/rtsp-server/rtsp-session-pool.c:
7041 * gst/rtsp-server/rtsp-session-pool.h:
7042 * gst/rtsp-server/rtsp-session.c:
7043 * gst/rtsp-server/rtsp-session.h:
7044 Add more timeout stuff
7045 Add method to check if a session is expired.
7046 Add method to perform cleanup on a session pool.
7048 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7050 * gst/rtsp-server/rtsp-client.c:
7051 * gst/rtsp-server/rtsp-session-pool.c:
7052 * gst/rtsp-server/rtsp-session-pool.h:
7053 * gst/rtsp-server/rtsp-session.c:
7054 * gst/rtsp-server/rtsp-session.h:
7055 Add beginnings of session timeouts and limits
7056 Add the timeout value to the Session header for unusual timeout values.
7057 Allow us to configure a limit to the amount of active sessions in a pool. Set a
7058 limit on the amount of retry we do after a sessionid collision.
7059 Add properties to the sessionid and the timeout of a session. Keep track of
7060 creation time and last access time for sessions.
7062 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7064 * gst/rtsp-server/rtsp-client.c:
7065 * gst/rtsp-server/rtsp-media.c:
7066 * gst/rtsp-server/rtsp-media.h:
7067 * gst/rtsp-server/rtsp-sdp.c:
7068 * gst/rtsp-server/rtsp-session-pool.c:
7069 * gst/rtsp-server/rtsp-session.c:
7070 * gst/rtsp-server/rtsp-session.h:
7071 Cleanup of sessions and more
7072 Fix the refcounting of media and sessions in the client. Properly clean up the
7073 session data when the client performs a teardown.
7074 Add Server header to responses.
7075 Allow for multiple uri setups in one session.
7076 Add Range header to the PLAY response and add the range attribute to the SDP
7078 Fix the session pool remove method, it used the wrong key in the hashtable. Also
7079 give the ownership of the sessionid to the session object.
7081 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7083 * gst/rtsp-server/rtsp-server.c:
7084 * gst/rtsp-server/rtsp-server.h:
7086 Rename the 'server_port' variable to simply 'port'.
7088 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7091 * gst/rtsp-server/rtsp-client.c:
7092 * gst/rtsp-server/rtsp-media.c:
7093 * gst/rtsp-server/rtsp-media.h:
7094 * gst/rtsp-server/rtsp-session.c:
7095 * gst/rtsp-server/rtsp-session.h:
7096 Rework the way we handle transports for streams
7097 Make the media accept an array of transports for the streams that we have
7098 configured for the play/pause requests.
7099 Implement server states for a client and its media.
7100 Require 0.10.22.1 (git HEAD) of gstreamer.
7102 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7104 * gst/rtsp-server/rtsp-client.c:
7105 * gst/rtsp-server/rtsp-media-factory.c:
7106 Drop const from functions dealing with urls
7107 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
7108 have the right const in them.
7110 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7112 * gst/rtsp-server/rtsp-client.c:
7113 * gst/rtsp-server/rtsp-media.c:
7114 * gst/rtsp-server/rtsp-sdp.c:
7118 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7120 * gst/rtsp-server/rtsp-client.c:
7121 * gst/rtsp-server/rtsp-media-factory.c:
7122 * gst/rtsp-server/rtsp-media.c:
7123 * gst/rtsp-server/rtsp-media.h:
7125 Don't keep a reference to the GstRTSPMedia in the stream.
7126 Free more things when freeing the GstRTSPMedia.
7128 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7131 * gst/rtsp-server/rtsp-media-factory.c:
7132 * gst/rtsp-server/rtsp-media-factory.h:
7133 * gst/rtsp-server/rtsp-media.c:
7134 * gst/rtsp-server/rtsp-media.h:
7135 * gst/rtsp-server/rtsp-server.c:
7136 * gst/rtsp-server/rtsp-server.h:
7137 More docs and small cleanups
7138 Add some more docs and update the README
7139 Cleanup some method names.
7140 Remove an unneeded idx field in the GstRTSPMediaStream
7142 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7145 * examples/Makefile.am:
7146 * examples/test-readme.c:
7147 Add a README and more example code
7148 Add a README file that contains a small introduction on how to use the server
7149 along with the example code explained in the readme.
7151 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7153 * gst/rtsp-server/rtsp-media.c:
7154 * gst/rtsp-server/rtsp-server.c:
7155 Fix some leaks and change default port
7156 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
7157 we finished the initial preroll. If we keep them locked, setting the pipeline to
7158 NULL will not stop and clean up the sources correctly.
7159 Change the default RTSP port to 8554 aka the official alternative RTSP port.
7161 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7163 * gst/rtsp-server/rtsp-session.c:
7164 * gst/rtsp-server/rtsp-session.h:
7165 Cleanups to the session object
7166 Remove some unneeded variables in the session state of a stream such as the
7167 owner media and the server transport.
7168 Get the configuration of a media stream in a session based on the media_stream
7169 in the original object instead of our cached index.
7170 Free more data in the finalize method.
7172 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7174 * gst/rtsp-server/rtsp-client.c:
7175 * gst/rtsp-server/rtsp-client.h:
7176 Cleanups and reuse media from DESCRIBE
7177 Handle thread create errors.
7178 Rename some internal methods to better match what they actually do.
7179 Handle misconfiguration of session_pool and media_mapping gracefully.
7180 Cache the DESCRIBE media and uri in the client connection and reuse them when
7181 we receive a SETUP request in the same connection for the same uri.
7182 Cleanup the client connection object.
7184 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7186 * gst/rtsp-server/rtsp-media-factory.c:
7187 * gst/rtsp-server/rtsp-media-factory.h:
7188 * gst/rtsp-server/rtsp-media.c:
7189 * gst/rtsp-server/rtsp-media.h:
7190 Add shared properties to media and factory
7191 Add the shared property to media.
7192 Implement some simple caching in the factory depending on if the media is shared
7195 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7197 * gst/rtsp-server/rtsp-client.c:
7198 Add a little comment
7199 Add some comment about the content-base header.
7201 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7203 * examples/Makefile.am:
7205 * examples/test-mp4.c:
7206 * examples/test-ogg.c:
7207 * examples/test-video.c:
7208 * gst/rtsp-server/Makefile.am:
7209 * gst/rtsp-server/rtsp-client.c:
7210 * gst/rtsp-server/rtsp-client.h:
7211 * gst/rtsp-server/rtsp-media-factory.c:
7212 * gst/rtsp-server/rtsp-media-factory.h:
7213 * gst/rtsp-server/rtsp-media.c:
7214 * gst/rtsp-server/rtsp-media.h:
7215 * gst/rtsp-server/rtsp-sdp.c:
7216 * gst/rtsp-server/rtsp-sdp.h:
7217 * gst/rtsp-server/rtsp-server.c:
7218 * gst/rtsp-server/rtsp-server.h:
7219 * gst/rtsp-server/rtsp-session.c:
7220 * gst/rtsp-server/rtsp-session.h:
7221 Reorganize things, prepare for media sharing
7222 Added various other test server examples
7223 Move the SDP message generation to a separate helper.
7224 Refactor common code for finding the session.
7225 Add content-base for realplayer compatibility
7226 Clean up request uris before processing for better vlc compatibility.
7227 Move prerolling and pipeline construction to the RTSPMedia object.
7228 Use multiudpsink for future pipeline reuse.
7230 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7236 === release 0.10.1 ===
7238 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7244 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7246 * bindings/vala/Makefile.am:
7248 Add more directories and files to the dist.
7250 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7252 * bindings/python/Makefile.am:
7253 * bindings/python/rtspserver.override:
7254 Fixed compile error of python bindings
7256 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7258 * bindings/vala/gst-rtsp-server-0.10.vapi:
7259 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7260 Marked values as nullable accordingly
7262 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7264 * bindings/vala/gst-rtsp-server-0.10.vapi:
7265 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
7266 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7267 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7268 Updated Vala bindings
7270 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7272 * gst/rtsp-server/rtsp-client.c:
7273 * gst/rtsp-server/rtsp-media-mapping.c:
7274 * gst/rtsp-server/rtsp-media-mapping.h:
7275 * gst/rtsp-server/rtsp-media.h:
7276 * gst/rtsp-server/rtsp-session-pool.h:
7277 Cleanups and doc updates
7278 Add some more documentation and do some minor cleanups here and there.
7280 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7282 * gst/rtsp-server/rtsp-client.c:
7283 * gst/rtsp-server/rtsp-media-factory.c:
7284 * gst/rtsp-server/rtsp-media-factory.h:
7285 * gst/rtsp-server/rtsp-media.c:
7286 * gst/rtsp-server/rtsp-media.h:
7287 * gst/rtsp-server/rtsp-session.c:
7288 * gst/rtsp-server/rtsp-session.h:
7290 Rename GstRTSPMediaBin to GstRTSPMedia
7291 Parse the request url into a GstRTSPUri object and pass this object to the
7292 various handlers and methods that require the uri.
7294 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7298 Add some more docs and remove some old code from the example.
7300 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7302 * gst/rtsp-server/rtsp-client.c:
7303 Handle state change failures better
7304 Handle state change failures better when changing the state of the pipeline to
7307 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7309 * gst/rtsp-server/rtsp-media-factory.c:
7310 * gst/rtsp-server/rtsp-media-factory.h:
7311 Make element creation more extendible
7312 Add get_element vmethod to the default MediaFactory so that subclasses can just
7313 override that method and still use the default logic for making a MediaBin from
7316 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7319 * gst/rtsp-server/Makefile.am:
7320 * gst/rtsp-server/rtsp-client.c:
7321 * gst/rtsp-server/rtsp-client.h:
7322 * gst/rtsp-server/rtsp-media-factory.c:
7323 * gst/rtsp-server/rtsp-media-factory.h:
7324 * gst/rtsp-server/rtsp-media-mapping.c:
7325 * gst/rtsp-server/rtsp-media-mapping.h:
7326 * gst/rtsp-server/rtsp-media.c:
7327 * gst/rtsp-server/rtsp-media.h:
7328 * gst/rtsp-server/rtsp-server.c:
7329 * gst/rtsp-server/rtsp-server.h:
7330 * gst/rtsp-server/rtsp-session.c:
7331 * gst/rtsp-server/rtsp-session.h:
7332 Make the server handle arbitrary pipelines
7333 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
7334 The GstMediaBin object has a handle to a bin with elements and to a list of
7335 GstMediaStream objects that this bin produces.
7336 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
7337 with methods to register and remove those mappings.
7338 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
7339 used by the server instance.
7340 Modify the example application so that it shows how to create custom pipelines
7341 attached to a specific mount point.
7342 Various misc cleanps.
7344 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7346 * gst/rtsp-server/rtsp-server.c:
7347 * gst/rtsp-server/rtsp-server.h:
7348 Allow setting a custom media factory for a server
7350 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7352 * gst/rtsp-server/rtsp-client.c:
7353 * gst/rtsp-server/rtsp-client.h:
7354 Allow setting a custom media factory for a client.
7356 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7358 * gst/rtsp-server/Makefile.am:
7359 Add Makefile entry for the media factory
7361 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7363 * gst/rtsp-server/rtsp-media-factory.c:
7364 * gst/rtsp-server/rtsp-media-factory.h:
7365 Add media factory to map urls to media pipeline objects.
7367 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7369 * gst/rtsp-server/rtsp-media.c:
7370 * gst/rtsp-server/rtsp-media.h:
7371 Add comments. Remove unused field
7373 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7375 * gst/rtsp-server/rtsp-session-pool.c:
7376 * gst/rtsp-server/rtsp-session-pool.h:
7377 Allow custom session pools to override the session id allocation algorithms Add some comments.
7379 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7381 * gst/rtsp-server/rtsp-session.h:
7384 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7386 * gst/rtsp-server/rtsp-client.c:
7387 * gst/rtsp-server/rtsp-client.h:
7388 Move the connection code in one place Add some comments
7390 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7392 * gst/rtsp-server/rtsp-server.c:
7393 * gst/rtsp-server/rtsp-server.h:
7394 Make vmethod to create and accept new clients. Add some docs.
7396 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7398 * gst/rtsp-server/rtsp-server.c:
7399 * gst/rtsp-server/rtsp-server.h:
7400 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
7402 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7404 * gst/rtsp-server/rtsp-client.c:
7405 * gst/rtsp-server/rtsp-client.h:
7406 Name the parameters more appropriately.
7408 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7410 * gst/rtsp-server/rtsp-session-pool.c:
7411 Do some more cleanup of the session pool.
7413 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7415 * gst/rtsp-server/Makefile.am:
7416 * gst/rtsp-server/rtsp-client.c:
7417 Check if return value of gst_rtsp_session_get_media is not NULL
7419 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7421 * gst/rtsp-server/Makefile.am:
7422 Install rtsp-session and rtsp-session-pool headers
7424 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7429 * bindings/python/Makefile.am:
7430 * bindings/python/arg-types.py:
7431 * bindings/python/codegen/Makefile.am:
7432 * bindings/python/codegen/__init__.py:
7433 * bindings/python/codegen/argtypes.py:
7434 * bindings/python/codegen/code-coverage.py:
7435 * bindings/python/codegen/codegen.py:
7436 * bindings/python/codegen/definitions.py:
7437 * bindings/python/codegen/defsparser.py:
7438 * bindings/python/codegen/docextract.py:
7439 * bindings/python/codegen/docgen.py:
7440 * bindings/python/codegen/fileprefix.override:
7441 * bindings/python/codegen/fileprefixmodule.c:
7442 * bindings/python/codegen/h2def.py:
7443 * bindings/python/codegen/mergedefs.py:
7444 * bindings/python/codegen/mkskel.py:
7445 * bindings/python/codegen/override.py:
7446 * bindings/python/codegen/reversewrapper.py:
7447 * bindings/python/codegen/scmexpr.py:
7448 * bindings/python/rtspserver-types.defs:
7449 * bindings/python/rtspserver.defs:
7450 * bindings/python/rtspserver.override:
7451 * bindings/python/rtspservermodule.c:
7453 Add python bindings.
7455 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7457 * bindings/Makefile.am:
7459 Don't go into python dir when requirements for python bindings are missing
7461 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7463 * bindings/Makefile.am:
7464 * bindings/vala/Makefile.am:
7466 Install Vala bindings if vala is available
7468 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7470 * bindings/vala/gst-rtsp-server-0.10.deps:
7471 * bindings/vala/gst-rtsp-server-0.10.vapi:
7472 * bindings/vala/gst-rtsp-server.vapi:
7473 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
7474 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
7475 * bindings/vala/packages/gst-rtsp-server-0.10.files:
7476 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7477 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7478 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
7479 * bindings/vala/packages/gst-rtsp-server.deps:
7480 * bindings/vala/packages/gst-rtsp-server.excludes:
7481 * bindings/vala/packages/gst-rtsp-server.files:
7482 * bindings/vala/packages/gst-rtsp-server.gi:
7483 * bindings/vala/packages/gst-rtsp-server.metadata:
7484 * bindings/vala/packages/gst-rtsp-server.namespace:
7485 Regenerated Vala bindings
7487 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7489 * bindings/vala/gst-rtsp-server.vapi:
7490 * bindings/vala/packages/gst-rtsp-server.metadata:
7491 Fixed typo in included headers for vala bindings
7493 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7497 * pkgconfig/Makefile.am:
7498 * pkgconfig/gst-rtsp-server.pc.in:
7499 Added pkgconfig file
7501 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7503 * bindings/vala/gst-rtsp-server.vapi:
7504 * bindings/vala/packages/gst-rtsp-server.excludes:
7505 * bindings/vala/packages/gst-rtsp-server.gi:
7506 * bindings/vala/packages/gst-rtsp-server.metadata:
7507 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
7509 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7511 * bindings/vala/gst-rtsp-server.vapi:
7512 * bindings/vala/packages/gst-rtsp-server.deps:
7513 * bindings/vala/packages/gst-rtsp-server.files:
7514 * bindings/vala/packages/gst-rtsp-server.gi:
7515 * bindings/vala/packages/gst-rtsp-server.metadata:
7516 * bindings/vala/packages/gst-rtsp-server.namespace:
7519 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
7521 * gst/rtsp-server/rtsp-session.c:
7522 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
7524 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7526 * examples/Makefile.am:
7527 * gst/rtsp-server/Makefile.am:
7528 Put GStreamer version in library name
7530 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7532 * examples/Makefile.am:
7533 * gst/rtsp-server/Makefile.am:
7534 Fix some issues to pass distcheck
7536 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7538 * gst/rtsp-server/rtsp-server.c:
7539 Added port property to GstRTSPServer class.
7541 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7546 * examples/Makefile.am:
7549 * gst/rtsp-server/Makefile.am:
7550 * gst/rtsp-server/rtsp-client.c:
7551 * gst/rtsp-server/rtsp-client.h:
7552 * gst/rtsp-server/rtsp-media.c:
7553 * gst/rtsp-server/rtsp-media.h:
7554 * gst/rtsp-server/rtsp-server.c:
7555 * gst/rtsp-server/rtsp-server.h:
7556 * gst/rtsp-server/rtsp-session-pool.c:
7557 * gst/rtsp-server/rtsp-session-pool.h:
7558 * gst/rtsp-server/rtsp-session.c:
7559 * gst/rtsp-server/rtsp-session.h:
7562 * src/rtsp-client.c:
7563 * src/rtsp-client.h:
7566 * src/rtsp-server.c:
7567 * src/rtsp-server.h:
7568 * src/rtsp-session-pool.c:
7569 * src/rtsp-session-pool.h:
7570 * src/rtsp-session.c:
7571 * src/rtsp-session.h:
7572 Split in library and example program
7574 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7576 * src/rtsp-client.h:
7577 Removed obsolete variable
7579 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7581 * src/rtsp-client.c:
7582 * src/rtsp-client.h:
7583 Removed pipeline variable GstRTSPClient, because it's only used in one function
7585 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7588 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
7590 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
7592 * src/rtsp-session.c:
7593 Initialize some more vars.
7595 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
7597 * src/rtsp-session.c:
7598 Initialize variable to avoid compiler warning.
7600 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
7603 Add a reasonable generic .gitignore