3 2014-08-27 Sebastian Dröge <slomo@coaxion.net>
8 2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
10 * gst/rtsp-server/rtsp-media.c:
11 * gst/rtsp-server/rtsp-stream.c:
12 * gst/rtsp-server/rtsp-stream.h:
13 rtsp-media: Query position and stop time only on the RTP parts of the pipeline
14 The RTCP parts, in specific the RTCP udpsinks, are not flushed when
15 seeking and will always continue counting the time. This leads to
16 the NPT after a backwards seek to be something completely different
17 to the actual seek position.
18 https://bugzilla.gnome.org/show_bug.cgi?id=732644
20 2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
22 * gst/rtsp-server/rtsp-media.c:
23 signals: Fix copy-pasto in target-state signal offset
27 2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
33 * gst-rtsp-server.doap:
36 2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
38 * gst/rtsp-server/rtsp-media.h:
39 media: correct misspelled words in description
40 https://bugzilla.gnome.org/show_bug.cgi?id=733244
42 === release 1.3.91 ===
44 2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
50 * gst-rtsp-server.doap:
53 2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
55 * docs/libs/gst-rtsp-server-sections.txt:
58 2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
60 * gst/rtsp-server/rtsp-server.c:
61 server: implement client REMOVE filter
63 2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
65 * gst/rtsp-server/rtsp-client.c:
66 * gst/rtsp-server/rtsp-client.h:
67 client: expose _close() method
68 Expose a previously internal close method to close the client
71 2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
73 * gst/rtsp-server/rtsp-session-pool.c:
74 session-pool: signal session-removed outside of the lock
75 Release the lock before emiting the session-removed signal.
77 2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
79 * gst/rtsp-server/rtsp-client.c:
80 * gst/rtsp-server/rtsp-server.c:
81 * gst/rtsp-server/rtsp-session-pool.c:
82 * gst/rtsp-server/rtsp-session.c:
83 * gst/rtsp-server/rtsp-stream.c:
84 filter: Release lock in filter functions
85 Release the object lock before calling the filter functions. We need to
86 keep a cookie to detect when the list changed during the filter
87 callback. We also keep a hashtable to make sure we only call the filter
88 function once for each object in case of concurrent modification.
89 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
91 2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
93 * gst/rtsp-server/rtsp-client.c:
94 client: check if watch is set in handle_teardown()
95 The unit tests run without a watch
97 2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
99 * tests/check/gst/client.c:
100 client tests: send teardown to cleanup session
102 2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
104 * tests/check/gst/rtspserver.c:
105 server tests: send teardown to cleanup session
107 2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
109 * gst/rtsp-server/rtsp-client.c:
110 client: keep ref to client for the session removed handler
111 This extra ref will be dropped when all client sessions have been
112 removed. A session is removed when a client sends teardown, closes its
113 endpoint of the TCP connection or the sessions expires.
114 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
116 2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
118 * gst/rtsp-server/rtsp-client.c:
119 * gst/rtsp-server/rtsp-session.c:
120 * tests/check/gst/client.c:
121 client: manage media in session as a last step
122 Once we manage a media in a session, we can't unmanage it anymore
123 without destroying it. Therefore, first check everything before we
124 manage the media, otherwise if something is wrong we have no way to
126 If we created a new session and something went wrong, remove the session
127 again. Fixes a leak in the unit test.
129 2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
131 * examples/test-mp4.c:
132 * examples/test-ogg.c:
133 examples: print 'stream ready at url' for mp4 and ogg example
135 2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
137 * gst/rtsp-server/rtsp-client.c:
138 * gst/rtsp-server/rtsp-sdp.c:
139 rtsp: fix for MIKEY api change
141 2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
143 * gst/rtsp-server/rtsp-client.c:
144 client: free watch context only once
145 The watch context is freed when the source is destroyed. Avoids
146 a CRITICAL when we try to unref the context twice.
148 2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
150 * gst/rtsp-server/rtsp-client.c:
153 2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
155 * gst/rtsp-server/rtsp-client.c:
156 client: protect sessions with lock
157 Protect the list of sessions with the lock.
158 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
160 2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
162 * gst/rtsp-server/rtsp-client.c:
163 Client: keep a ref to the session
164 Don't just keep a weak ref to the session objects but use a hard ref. We
165 will be notified when a session is removed from the pool (expired) with
166 the new session-removed signal.
167 Don't automatically close the RTSP connection when all the sessions of
168 a client are removed, a client can continue to operate and it can create
169 a new session if it wants. If you want to remove the client from the
170 server, you have to use gst_rtsp_server_client_filter() now.
171 Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
172 See https://bugzilla.gnome.org/show_bug.cgi?id=732226
174 2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
176 * gst/rtsp-server/rtsp-session-pool.c:
177 * gst/rtsp-server/rtsp-session-pool.h:
178 session-pool: add session-removed signal
179 Add a signal to be notified when a session is removed from the pool.
181 2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
183 * gst/rtsp-server/Makefile.am:
184 * gst/rtsp-server/rtsp-server.h:
185 Make rtsp-server.h a single-include header, use it for G-I
186 https://bugzilla.gnome.org/show_bug.cgi?id=732411
188 === release 1.3.90 ===
190 2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
196 * gst-rtsp-server.doap:
199 2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
201 * gst/rtsp-server/rtsp-stream.c:
202 stream: crypto can be NULL
204 2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
206 * gst/rtsp-server/rtsp-client.c:
207 * gst/rtsp-server/rtsp-media.c:
208 * gst/rtsp-server/rtsp-mount-points.c:
209 introspection: add missing allow-none annotations
210 https://bugzilla.gnome.org/show_bug.cgi?id=730952
212 2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
214 * gst/rtsp-server/rtsp-address-pool.c:
215 * gst/rtsp-server/rtsp-media.c:
216 * gst/rtsp-server/rtsp-session-media.c:
217 * gst/rtsp-server/rtsp-session-pool.c:
218 * gst/rtsp-server/rtsp-stream-transport.c:
219 * gst/rtsp-server/rtsp-stream.c:
220 * gst/rtsp-server/rtsp-token.c:
221 introspection: add (nullable) annotations to return values
222 https://bugzilla.gnome.org/show_bug.cgi?id=730952
224 2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
226 * gst/rtsp-server/rtsp-client.c:
227 * gst/rtsp-server/rtsp-stream.c:
228 gi: improve annotations
229 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
231 2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
233 * gst/rtsp-server/rtsp-client.c:
234 * gst/rtsp-server/rtsp-media-factory.c:
235 * gst/rtsp-server/rtsp-media.c:
236 * gst/rtsp-server/rtsp-server.c:
237 signals: use generic marshal function
238 Use the generic C marshal function.
239 Use more explicit type instead of G_TYPE_POINTER
241 2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
243 * gst/rtsp-server/rtsp-context.h:
244 context: add type macro
246 2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
248 * gst/rtsp-server/rtsp-client.c:
249 * gst/rtsp-server/rtsp-sdp.c:
250 * gst/rtsp-server/rtsp-sdp.h:
251 sdp: hide key length defines
252 They don't have a namespace.
254 2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
259 === release 1.3.3 ===
261 2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
267 * gst-rtsp-server.doap:
270 2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
272 * gst/rtsp-server/rtsp-client.c:
273 * gst/rtsp-server/rtsp-sdp.c:
274 * gst/rtsp-server/rtsp-sdp.h:
275 mikey: add different key length parameters
276 Add encryption and authentication key length parameters to MIKEY. For
277 the encoders, the key lengths are obtained from the cipher and auth
278 algorithms set in the caps. For the decoders, they are obtained while
279 parsing the key management from the client.
280 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
282 2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
284 * tests/check/gst/stream.c:
285 stream tests: Make sure we get right multicast address from stream
286 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
288 2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
290 * gst/rtsp-server/rtsp-client.c:
291 client: ref the context until rtsp watch is alive
292 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
294 2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
296 * gst/rtsp-server/rtsp-client.c:
297 client: Destroy the rtsp watch after connection close
299 2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
301 * gst/rtsp-server/rtsp-media.c:
302 media: fix confusing comment
304 2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
306 * gst/rtsp-server/rtsp-session.c:
307 rtsp-session: Timeout in header.
308 Adding the possbilty to always have timout in header.
309 This is configurabe with setting "timeout-always-visible".
310 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
312 2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
317 === release 1.3.2 ===
319 2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
326 * gst-rtsp-server.doap:
329 2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
332 Automatic update of common submodule
333 From 211fa5f to 1f5d3c3
335 2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
337 * gst/rtsp-server/rtsp-client.c:
338 client: store TCP ports in transport
339 Store the TCP ports in the transport when we are doing RTSP over TCP.
340 This way, we can easily get to the ports from the transport.
341 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
343 2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
345 * gst/rtsp-server/rtsp-stream.c:
346 stream: add signals for new RTP/RTCP encoders
347 New signals to allow the user to configure the dynamically created
349 https://bugzilla.gnome.org/show_bug.cgi?id=730228
351 2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
353 * gst/rtsp-server/rtsp-media.c:
354 * gst/rtsp-server/rtsp-media.h:
355 media: Make suspend()/unsuspend() virtual
356 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
358 2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
360 * gst/rtsp-server/rtsp-client.c:
361 client: fix send-message signal marshaller
362 Use generic marshalling for the send-message signal. It has
363 two POINTER arguments, not just one.
364 https://bugzilla.gnome.org/show_bug.cgi?id=729900
366 2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
368 * tests/check/gst/media.c:
369 tests: add and remove pads only once
370 In this test we simulate a dynamic pad by watching the caps event.
371 Because of renegotiation in the base payloader now, this caps is sent
372 multiple times but we can only deal with 1 invocation, use a variable to
373 only 'add and remove' the pad once.
375 2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
377 * tests/check/gst/rtspserver.c:
378 tests: add unit test for correct handling of Require headers
379 https://bugzilla.gnome.org/show_bug.cgi?id=729426
381 2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
383 * gst/rtsp-server/rtsp-client.c:
384 rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
385 Servers must handle Require headers and must report a failure
386 if they don't handle any of the Required options, see RFC 2326,
387 section 12.32: https://tools.ietf.org/html/rfc2326#page-54
388 https://bugzilla.gnome.org/show_bug.cgi?id=729426
390 2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
395 === release 1.3.1 ===
397 2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
403 * gst-rtsp-server.doap:
406 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
409 Automatic update of common submodule
410 From bcb1518 to 211fa5f
412 2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
417 2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
419 * tests/check/gst/sessionmedia.c:
420 tests: fix memory leak in sessionmedia unit test
422 2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
424 * gst/rtsp-server/rtsp-client.c:
425 client: emit a signal before sending a message
426 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
428 2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
430 * gst/rtsp-server/rtsp-client.c:
431 client: pass context to send_message
432 Pass the current context to send_message, we will need it later.
434 2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
436 * gst/rtsp-server/rtsp-client.c:
437 client: fix typo in comment
439 2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
441 * gst/rtsp-server/rtsp-media.c:
442 media: Do not stop thread twice if default_prepare() fails
444 2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
446 * gst/rtsp-server/rtsp-client.c:
447 client: set the watch to flushing before going to NULL
448 First set the watch to flushing so that we unblock any current and
449 future attempt to send data on the watch, Then set the pipeline to
451 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
453 2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
455 * gst/rtsp-server/rtsp-session-pool.c:
456 * tests/check/gst/sessionpool.c:
457 rtsp-session-pool: Fixes annotation
458 Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
459 in the sessionpool test.
460 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
462 2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
464 * gst/rtsp-server/rtsp-media.c:
465 * gst/rtsp-server/rtsp-media.h:
466 media: make media_prepare virtual
467 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
469 2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
471 * gst/rtsp-server/rtsp-media.c:
472 * tests/check/gst/media.c:
473 media: stop the thread in more error cases
475 2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
477 * gst/rtsp-server/rtsp-media.c:
478 * tests/check/gst/media.c:
479 media: allow NULL as the thread
480 Use the default context whan passing a NULL thread.
482 2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
484 * gst/rtsp-server/rtsp-client.c:
485 rtsp-client: indent cleanup
486 Coverity was moaning about unreachable code, and I think it was just
487 confused by { being before the label. We'll see if it pops up again.
490 2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
492 * gst/rtsp-server/rtsp-client.c:
493 * gst/rtsp-server/rtsp-media.c:
494 client: Add drop-backlog property
495 When we have too many messages queued for a client (currently hardcoded
496 to 100) we overflow and drop the messages. Add a drop-backlog property
497 to control this behaviour. Setting this property to FALSE will retry
498 to send the messages to the client by waiting for more room in the
500 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
502 2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
504 * gst/rtsp-server/rtsp-client.c:
505 client: support for POST before GET when setting up a tunnel
507 2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
509 * gst/rtsp-server/rtsp-client.c:
510 client: remove watch of the second client after http tunnel setup
511 The second client will be freed after the HTTP tunnel has been set up.
512 Make sure it's RTSP watch is never dispatched again.
513 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
515 2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
517 * gst/rtsp-server/rtsp-media.c:
518 * tests/check/gst/media.c:
519 media: Make media_prepare() fail if port allocation fails
520 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
522 2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
524 * tests/check/gst/media.c:
525 media test: cleanup the thread pool in tests
527 2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
529 * gst/rtsp-server/rtsp-media.c:
530 * tests/check/gst/media.c:
531 rtsp-media: Unblock blocked streams in unprepare
532 The streams will be blocked when a live media is prepared.
533 The streams should be unblocked in gst_rtsp_media_unprepare.
534 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
536 2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
538 * gst/rtsp-server/rtsp-media.c:
539 media: release the state lock when going to NULL
540 Set our state to UNPREPARING and release the state-lock before
541 setting the pipeline to the NULL state. This way, any pad-added
542 callback will be able to take the state-lock and check that we are now
543 unpreparing instead of deadlocking.
544 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
546 2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
548 * gst/rtsp-server/rtsp-media.c:
549 media: protect status with lock
550 Make sure we only update the status with the lock.
552 2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
554 * gst/rtsp-server/rtsp-client.c:
555 * gst/rtsp-server/rtsp-sdp.c:
556 rtsp: update for MIKEY API changes
558 2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
560 * gst/rtsp-server/rtsp-client.c:
561 client: parse the mikey response from the client
562 Parse the mikey response from the client and update the policy for
565 2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
567 * gst/rtsp-server/rtsp-stream.c:
568 * gst/rtsp-server/rtsp-stream.h:
569 stream: add method to set crypto info
570 Make a method to configure the crypto information of a stream.
571 Set udpsrc in READY instead of PAUSED so that we can configure caps
574 2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
576 * gst/rtsp-server/rtsp-client.c:
577 client: cleanup error paths
579 2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
581 * gst/rtsp-server/rtsp-media.c:
584 2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
586 * examples/test-video.c:
587 test: enable SRTP only on RTSPS
588 We only want to enable SRTP when doing rtsp over TLS so that we can
589 exchange the keys in a secure way.
591 2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
593 * examples/test-video.c:
594 test: print an error on failure
596 2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
599 * examples/test-video.c:
600 * gst/rtsp-server/rtsp-sdp.c:
601 * gst/rtsp-server/rtsp-stream.c:
602 * tests/check/Makefile.am:
603 stream: add SRTP support
604 Install srtp encoder and decoder elements in rtpbin
607 2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
609 * tests/check/Makefile.am:
610 * tests/check/gst/sessionpool.c:
611 tests: Add unit tests for sessionpool
612 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
614 2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
616 * tests/check/gst/threadpool.c:
617 tests: Improve code coverage of rtsp-threadpool tests
618 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
620 2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
622 * tests/check/gst/sessionmedia.c:
623 tests: Improve code coverage for rtsp-session-media
624 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
626 2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
628 gobject-introspection: Add annotations to support language bindings
629 In addition a few cosmetic changes:
630 * Adjust the order of arguments
631 * Fix typo: occured -> occurred
632 * Fix indentation after Return:-clauses
633 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
635 2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
637 * gst/rtsp-server/rtsp-stream.c:
638 rtsp-stream: Don't mix IPv4 and IPv6 addresses
639 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
641 2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
643 * gst/rtsp-server/rtsp-stream.c:
644 stream: take caps after the session manager
645 Take the caps for the SDP after they leave the rtpbin so that we can
646 also get the properties added by rtpbin elements.
648 2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
650 * gst/rtsp-server/rtsp-stream.c:
651 stream: release lock while pushing out packets
652 Keep a cache of the transports and use this to iterate the transport
653 while pushing packets. This allows us to release the lock early.
654 See https://bugzilla.gnome.org/show_bug.cgi?id=725898
656 2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
658 * gst/rtsp-server/rtsp-client.c:
659 * gst/rtsp-server/rtsp-client.h:
660 rtsp-client: vmethod for modifying tunnel GET response
661 Add a vmethod tunnel_http_response where the response to the HTTP GET
662 for tunneled connections can be modified.
663 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
665 2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
667 * gst/rtsp-server/rtsp-sdp.c:
668 sdp: make 1 media line per profile
669 If we have multiple profiles (AVP or AVPF) for a stream, make one m=
670 line in the SDP for each profile. The client is then supposed to pick
671 one of the profiles in the SETUP request. Because the m= lines have the
672 same pt, the client also knows that only 1 option is possible.
674 2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
676 * gst/rtsp-server/rtsp-media-factory.c:
677 * gst/rtsp-server/rtsp-media-factory.h:
678 * gst/rtsp-server/rtsp-media.c:
679 factory: add profile property and pass to media and streams
681 2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
683 * examples/test-multicast.c:
684 * gst/rtsp-server/rtsp-sdp.c:
685 sdp: pass multicast connection for multicast-only stream
686 Pass the multicast address of the stream in the connection info in the
687 SDP so that clients try a multicast connection first.
688 Only allow multicast connections in the test-multicast example. Also
689 increase the TTL a little.
691 2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
694 .gitignore: Ignore gcov intermediate files
695 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
697 2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
699 * gst/rtsp-server/rtsp-stream.c:
700 stream: release some locks in error cases
702 2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
704 docs: Enable and fix gtk-doc warnings
705 * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
706 * addresspool/mediafactory: Add missing annotation colon
707 * stream: Annotate return value
708 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
710 2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
713 Automatic update of common submodule
714 From fe1672e to bcb1518
716 2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
719 Automatic update of common submodule
720 From 1a07da9 to fe1672e
722 2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
724 * examples/Makefile.am:
725 examples: use LDADD for libs instead of LDFLAGS
727 2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
730 configure: make sure releases are in .doap file
732 2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
734 * examples/test-cgroups.c:
735 examples: test-cgroups: don't put code with side effects into g_assert()
736 The g_assert() might get compiled out with the right
737 compiler/preprocessor flags.
739 2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
741 * examples/.gitignore:
742 examples: add cgroup test binary to .gitignore
744 2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
746 * examples/test-cgroups.c:
747 examples: fix cgroup test build
748 Fixes build failure caused by compiler warning:
749 test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
751 2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
754 .gitignore: ignore temp files created in the course of 'make check'
756 2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
758 * gst/rtsp-server/rtsp-media.c:
759 rtsp-media: don't loose frames handling new PLAY request
760 If client supplied a range check if the range specifies the start point.
761 If not, then do an accurate seek to the current position. If a start
762 point was specified do do a key unit seek to make sure the streaming
763 starts with decodeable frames.
764 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
766 2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
768 * gst/rtsp-server/rtsp-media.c:
769 Revert "media: only flush when setting a new start position"
770 This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
771 We need to do the flush in all cases, demuxer block currently for
774 2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
776 * gst/rtsp-server/rtsp-media.c:
777 media: only flush when setting a new start position
778 Only flush the pipeline when we change the start position with
780 See https://bugzilla.gnome.org/show_bug.cgi?id=724611
782 2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
784 * gst/rtsp-server/rtsp-stream.c:
785 stream: set ttl-mc before adding the socket
786 Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
787 never be set on socket.
788 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
790 2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
792 * gst/rtsp-server/rtsp-media.c:
793 media: stop thread if media is already prepared
794 in gst_rtsp_media_prepare() the thread is not used if media is already
795 prepared (e.g. media shared) so we want to stop the thread. otherwise, a
797 https://bugzilla.gnome.org/show_bug.cgi?id=724182
799 2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
802 build: Ship gst-rtsp-server.doap file
804 2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
806 * tests/check/gst/rtspserver.c:
807 tests: Fix another compiler warning with gcc
809 2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
811 * gst/rtsp-server/rtsp-client.c:
812 * gst/rtsp-server/rtsp-mount-points.c:
813 * gst/rtsp-server/rtsp-stream.c:
814 * tests/check/gst/client.c:
815 rtsp-server: Fix lots of compiler warnings with clang
817 2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
820 * gst-rtsp-server.doap:
822 configure: Synchronise with the configure scripts of the other modules
824 2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
827 configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
829 2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
831 * gst/rtsp-server/rtsp-media.c:
832 * gst/rtsp-server/rtsp-stream.c:
833 Revert "rtsp-server: support build against last stable release"
834 This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
835 Let us require 1.2.3 now, which is going to be released in a few
838 2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
840 * gst/rtsp-server/rtsp-session-media.c:
841 * gst/rtsp-server/rtsp-stream-transport.c:
842 session: improve RTP-Info
843 Ignore streams that can't generate RTP-Info instead of failing.
844 Don't return the empty string when all streams are unconfigured but
845 return NULL so that we don't generate and empty RTP-Info header.
846 Improve docs a little.
848 2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
850 * gst/rtsp-server/rtsp-session-media.c:
851 Don't free rtpinfo GString when it is NULL
852 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
854 2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
856 * gst/rtsp-server/rtsp-media.c:
857 media: only set keyframe flag when modifying start
858 Only set the keyframe flag when we modify the start position. The
859 keyframe flag should probably be ignored when no change is requested but
860 until we can claim this is all documented properly and all demuxer
861 implement this, avoid setting the flag.
862 See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
864 2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
866 * gst/rtsp-server/rtsp-thread-pool.c:
867 thread-pool: Unref source after mainloop has quit to avoid races in GLib
868 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
870 2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
872 * gst/rtsp-server/rtsp-stream.c:
873 stream: handle NULL seqnum and rtptime arguments
875 2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
877 * gst/rtsp-server/rtsp-thread-pool.c:
878 * tests/check/gst/threadpool.c:
879 thread-pool: Unref reused threads in gst_rtsp_thread_stop()
880 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
882 2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
884 * gst/rtsp-server/rtsp-stream.c:
885 stream: add fallback for missing stats property
886 Use a fallback when the payloader does not have a stats property
887 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
889 2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
892 Automatic update of common submodule
893 From f7bc1c3 to 1a07da9
895 2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
897 * gst/rtsp-server/rtsp-stream.c:
898 stream: don't leak stats structure
899 Don't leak the stats structure and deal with NULL stats.
901 2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
903 * gst/rtsp-server/rtsp-stream.c:
904 stream: Get rtpinfo properties atomically from payloader
905 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
907 2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
909 * gst/rtsp-server/rtsp-media.c:
910 media: refactor state change functions and signals
911 Make functions to set the target state and the pipeline state and emit
912 the signals from those functions.
914 2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
916 * gst/rtsp-server/rtsp-media.c:
917 * gst/rtsp-server/rtsp-media.h:
918 media: add signal to notify of pending state changes
920 2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
922 * gst/rtsp-server/rtsp-media.c:
923 * gst/rtsp-server/rtsp-stream.c:
924 rtsp-server: support build against last stable release
925 Until 1.2.3 is out with the new get_type function and we
928 2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
930 * gst/rtsp-server/rtsp-stream.c:
931 stream: fix compilation
933 2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
935 * gst/rtsp-server/rtsp-media.c:
936 * gst/rtsp-server/rtsp-media.h:
937 * gst/rtsp-server/rtsp-stream.c:
938 * gst/rtsp-server/rtsp-stream.h:
939 stream: add property to configure profiles
941 2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
943 * gst/rtsp-server/rtsp-client.c:
944 client: let stream check supported transport
945 Delegate the check if a transport is allowed to the stream.
946 See https://bugzilla.gnome.org/show_bug.cgi?id=720696
948 2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
950 * gst/rtsp-server/rtsp-stream.c:
951 * gst/rtsp-server/rtsp-stream.h:
952 stream: add method to check supported transport
953 Add a method to check if a transport is supported
955 2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
958 configure.ac: Only check for gstreamer-check, not check
959 We include check in gstreamer-check since quite some time now.
961 2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
963 * gst/rtsp-server/rtsp-session-media.c:
964 * gst/rtsp-server/rtsp-stream-transport.c:
965 * gst/rtsp-server/rtsp-stream.c:
966 * gst/rtsp-server/rtsp-stream.h:
967 stream: return clock-rate from get_rtpinfo
968 And use it to correct the rtptime to the requested start-time.
969 See https://bugzilla.gnome.org/show_bug.cgi?id=712198
971 2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
973 * gst/rtsp-server/rtsp-session-media.c:
974 * gst/rtsp-server/rtsp-stream-transport.c:
975 * gst/rtsp-server/rtsp-stream-transport.h:
976 session-media: calculate start-time
978 2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
980 * gst/rtsp-server/rtsp-stream-transport.c:
981 * gst/rtsp-server/rtsp-stream.c:
982 * gst/rtsp-server/rtsp-stream.h:
983 stream: also return the running-time
984 Return the running-time in the rtpinfo as well.
986 2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
988 * gst/rtsp-server/rtsp-client.c:
989 * gst/rtsp-server/rtsp-session-media.c:
990 * gst/rtsp-server/rtsp-session-media.h:
991 * gst/rtsp-server/rtsp-stream-transport.c:
992 * gst/rtsp-server/rtsp-stream-transport.h:
993 session-media: let the session-media make the RTPInfo
994 Add method to create the RTPInfo for a stream-transport.
995 Add method to create the RTPInfo for all stream-transports in a
997 Use the session-media RTPInfo code in client. This allows us to refactor
998 another method to link the TCP callbacks.
1000 2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
1002 mount-points: sort sequence before g_sequence_lookup
1003 * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
1004 sort sequence if dirty, otherwise lookup will fail.
1005 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
1007 2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
1010 configure: rename package from gst-rtsp to gst-rtsp-server
1011 To match git module name and avoid confusion with the
1012 rtsp lib in gst-plugins-base and rtsp plugin in -good.
1014 2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
1017 configure: bump core/base/good requirement to 1.2.0
1018 Bump to released stable version and make implicit
1019 requirements explicit.
1021 2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
1026 Fix broken gettext setup which is not used anyway
1028 2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
1031 Automatic update of common submodule
1032 From dbedaa0 to d48bed3
1034 2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
1036 * gst/rtsp-server/rtsp-client.c:
1037 * gst/rtsp-server/rtsp-media.c:
1038 * gst/rtsp-server/rtsp-media.h:
1039 media: add setup_sdp vmethod
1040 gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
1041 gst_rtsp_media_setup_sdp.
1042 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
1044 2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
1046 * gst/rtsp-server/rtsp-stream.c:
1047 rtsp-stream: Check return value of sscanf
1048 streamid is only valid if sscanf matched something.
1050 2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
1052 * gst/rtsp-server/rtsp-client.c:
1053 rtsp-client: Fix iteration
1054 Wouldn't even enter the code block otherwise (i++ was used as the check
1055 and not the postfix).
1057 2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
1059 * gst/rtsp-server/rtsp-client.c:
1060 * gst/rtsp-server/rtsp-client.h:
1061 client: add vmethod to configure media and streams
1062 Implement a vmethod that can be used to configure the media and the
1063 streams based on the current context. Handle the blocksize handling in
1064 the default handler.
1065 See https://bugzilla.gnome.org/show_bug.cgi?id=720667
1067 2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
1070 Make git ignore more unit test binaries
1072 2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
1074 * gst/rtsp-server/rtsp-address-pool.h:
1075 * gst/rtsp-server/rtsp-auth.h:
1076 * gst/rtsp-server/rtsp-client.h:
1077 * gst/rtsp-server/rtsp-context.h:
1078 * gst/rtsp-server/rtsp-media-factory-uri.h:
1079 * gst/rtsp-server/rtsp-media-factory.h:
1080 * gst/rtsp-server/rtsp-media.h:
1081 * gst/rtsp-server/rtsp-mount-points.h:
1082 * gst/rtsp-server/rtsp-server.h:
1083 * gst/rtsp-server/rtsp-session-media.h:
1084 * gst/rtsp-server/rtsp-session-pool.h:
1085 * gst/rtsp-server/rtsp-session.h:
1086 * gst/rtsp-server/rtsp-stream-transport.h:
1087 * gst/rtsp-server/rtsp-stream.h:
1088 * gst/rtsp-server/rtsp-thread-pool.h:
1089 * gst/rtsp-server/rtsp-token.h:
1090 rtsp-server: add padding to many public structures
1091 Not mini objects though, since they are not subclassable
1092 anyway, nor kept on the stack or inlined in a structure.
1094 2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
1096 media: add new create_rtpbin vmethod
1097 * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
1098 https://bugzilla.gnome.org/show_bug.cgi?id=719734
1100 2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
1102 * tests/check/gst/media.c:
1103 tests: fix memory leak, free test's thread pool
1104 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
1106 2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
1108 * gst/rtsp-server/rtsp-stream-transport.c:
1109 stream-transport: free url in finalize
1111 2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
1113 * gst/rtsp-server/rtsp-media.c:
1114 media: also do state change in suspended state
1116 2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
1118 * gst/rtsp-server/rtsp-client.c:
1119 * gst/rtsp-server/rtsp-media.c:
1120 media: also handle prepare and range in suspended state
1121 When we are suspended, we are already prepared.
1122 We can get the range in the suspended state.
1124 2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
1126 * tests/check/Makefile.am:
1127 * tests/check/gst/sessionmedia.c:
1128 check: add test for uri in setup
1129 Added unit tests for the new functionality in GstRTSPStreamTransport.
1130 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
1132 2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
1134 * gst/rtsp-server/rtsp-client.c:
1135 client: store setup uri and use in PLAY response
1136 Store the uri used when doing the setup and use that in the PLAY
1138 fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
1140 2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
1142 * gst/rtsp-server/rtsp-stream-transport.c:
1143 * gst/rtsp-server/rtsp-stream-transport.h:
1144 stream-transport: add method to get/set url
1146 2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
1148 * gst/rtsp-server/rtsp-client.c:
1149 client: suspend after SDP and unsuspend before PLAYING
1150 Based on patches by Ognyan Tonchev <ognyan@axis.com>
1151 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
1153 2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
1155 * gst/rtsp-server/rtsp-media-factory.c:
1156 * gst/rtsp-server/rtsp-media-factory.h:
1157 * gst/rtsp-server/rtsp-media.c:
1158 * gst/rtsp-server/rtsp-media.h:
1159 * gst/rtsp-server/rtsp-session-media.c:
1160 * gst/rtsp-server/rtsp-session.c:
1161 * tests/check/gst/media.c:
1162 * tests/check/gst/mediafactory.c:
1163 media: add suspend modes
1164 Add support for different suspend modes. The stream is suspended right after
1165 producing the SDP and after PAUSE. Different suspend modes are available that
1166 affect the state of the pipeline. NONE leaves the pipeline state unchanged and
1167 is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
1168 state and RESET will bring the pipeline to the NULL state.
1169 A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
1170 this means that the pipeline needs to be prerolled again.
1171 Base on patches by Ognyan Tonchev <ognyan@axis.com>
1172 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
1174 2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
1176 * gst/rtsp-server/rtsp-media.c:
1177 media: start live streams in blocked state
1178 Start live streams in the blocked state and make them preroll using the
1179 messages. This ensure that no data is played by the sink until we explicitly
1180 unblock the stream right before going to PLAYING.
1181 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
1183 2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
1185 * gst/rtsp-server/rtsp-media.c:
1186 media: refactor starting and waiting for preroll
1187 Based on patches from Ognyan Tonchev <ognyan@axis.com>
1188 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
1190 2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
1192 * gst/rtsp-server/rtsp-stream.c:
1193 * gst/rtsp-server/rtsp-stream.h:
1194 stream: add API to block streams
1195 Add an API to block on the streams and make it post a message.
1196 Based on patch by Ognyan Tonchev <ognyan@axis.com>
1197 See https://bugzilla.gnome.org/show_bug.cgi?id=711257
1199 2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
1201 * docs/libs/Makefile.am:
1202 docs: Specify the override file
1203 Even if it's empty (for now) it avoids make distcheck complaining
1205 2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
1207 * gst/rtsp-server/rtsp-media.c:
1208 media: move default implementations to where they are used
1210 2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
1212 * gst/rtsp-server/rtsp-media.c:
1213 media: take the right lock in gst_rtsp_media_set_pipeline_state()
1214 We need to take the state_lock when calling this method.
1216 2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
1218 * gst/rtsp-server/rtsp-media.c:
1219 media: handle add-added on non-bins too
1220 Handle dynamic payloaders that are not bins, as used in the unit-test.
1222 2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1224 * gst/rtsp-server/rtsp-media-factory.c:
1225 * gst/rtsp-server/rtsp-media-factory.h:
1226 * gst/rtsp-server/rtsp-media.c:
1227 rtsp-media/-factory: Fix request pad name comments
1228 These must be escaped for gtk-doc to parse the comments without warnings.
1230 2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
1232 rtsp-media: remove transports if media is in error status
1233 * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
1234 trying to change to GST_STATE_NULL and media is in error status, we
1235 remove all transports.
1236 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
1238 2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
1240 * gst/rtsp-server/rtsp-media.c:
1241 rtsp-media: use element metadata to find payloader
1242 Use the element metadata to find the payloader instead of checking
1244 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
1246 2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
1248 rtsp-stream: add getter for payload type
1249 * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
1250 * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
1251 element and create the stream with this one instead of the dynpay%d
1253 https://bugzilla.gnome.org/show_bug.cgi?id=712396
1255 2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1257 * gst/rtsp-server/rtsp-client.c:
1258 * gst/rtsp-server/rtsp-context.h:
1259 * gst/rtsp-server/rtsp-media.c:
1260 * gst/rtsp-server/rtsp-mount-points.c:
1261 * gst/rtsp-server/rtsp-server.c:
1262 * gst/rtsp-server/rtsp-token.c:
1263 rtsp-*: Refer to NULL as a constant in comments
1265 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1267 2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1269 rtsp-*: Fix type name typos in comments
1270 * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
1271 * rtsp-auth: Refer to part of constant name as text
1272 * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
1273 * rtsp-session-media: Fix GstRTSPSessionMedia typo
1274 * rtsp-stream: Fix typo when refering to GstBin
1275 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1277 2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
1280 * docs/libs/gst-rtsp-server-docs.sgml:
1281 * docs/libs/gst-rtsp-server-sections.txt:
1282 docs: Improve documentation
1283 * Include annotation-glossary to quiet gtk-doc
1284 * Rename remaining ClientState -> Context
1285 * Rename object hierarchy file
1286 * Remove stale chapter references
1287 * Add missing function and object references
1288 * Include missing GstRTSPAddressPoolResult
1289 https://bugzilla.gnome.org/show_bug.cgi?id=714988
1291 2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
1293 * gst/rtsp-server/rtsp-client.c:
1294 * gst/rtsp-server/rtsp-server.c:
1295 * gst/rtsp-server/rtsp-session-pool.c:
1296 * gst/rtsp-server/rtsp-session.c:
1297 * gst/rtsp-server/rtsp-stream.c:
1298 rtsp-server: sprinkle some allow-none annotations for g-i
1300 2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
1302 * gst/rtsp-server/rtsp-stream.c:
1303 * gst/rtsp-server/rtsp-stream.h:
1304 stream: add method to filter transports
1305 Add a method to safely iterate and collect the stream transports
1306 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
1308 2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
1310 * gst/rtsp-server/rtsp-client.c:
1311 * gst/rtsp-server/rtsp-server.c:
1312 * gst/rtsp-server/rtsp-session-pool.c:
1313 * gst/rtsp-server/rtsp-session.c:
1314 rtsp: allow NULL func in filters
1315 Passing a null function make the filters return a list of
1318 2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
1320 * gst/rtsp-server/rtsp-address-pool.c:
1321 * tests/check/gst/addresspool.c:
1322 address-pool: fix address increment
1323 Use a guint instead of guint8 to increment the address. It's still not
1324 completely correct because a guint might not be able to hold the complete
1325 address range, but that's an enhacement for later.
1326 Add unit test to test improved behaviour.
1327 https://bugzilla.gnome.org/show_bug.cgi?id=708237
1329 2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
1331 * gst/rtsp-server/rtsp-client.c:
1332 * tests/check/gst/client.c:
1333 client: allow absolute path in requests
1334 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
1336 2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
1338 * gst/rtsp-server/rtsp-client.c:
1339 * gst/rtsp-server/rtsp-client.h:
1340 client: make make_path_from_uri a vmethod
1342 2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1344 * docs/libs/gst-rtsp-server-sections.txt:
1345 * gst/rtsp-server/rtsp-stream.c:
1346 * gst/rtsp-server/rtsp-stream.h:
1347 * tests/check/Makefile.am:
1348 * tests/check/gst/stream.c:
1349 stream: Add functions to get rtp and rtcp sockets
1350 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
1352 2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1354 * gst/rtsp-server/rtsp-context.c:
1355 * gst/rtsp-server/rtsp-context.h:
1356 context: defing a GType for the context
1357 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
1359 2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
1361 * gst/rtsp-server/Makefile.am:
1362 * gst/rtsp-server/rtsp-auth.c:
1363 * gst/rtsp-server/rtsp-context.c:
1364 * gst/rtsp-server/rtsp-media.c:
1365 * gst/rtsp-server/rtsp-mount-points.c:
1366 * gst/rtsp-server/rtsp-server.h:
1367 * gst/rtsp-server/rtsp-session-media.c:
1368 * gst/rtsp-server/rtsp-session.c:
1369 * gst/rtsp-server/rtsp-stream.c:
1370 Fixed several GIR warnings
1372 2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
1374 * gst/rtsp-server/rtsp-auth.c:
1377 2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1379 * tests/check/Makefile.am:
1380 * tests/check/gst/token.c:
1381 tests: Add unit tests for token
1382 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
1384 2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1386 * gst/rtsp-server/rtsp-token.c:
1387 token: Validate args for gst_rtsp_token_is_allowed
1388 See https://bugzilla.gnome.org/show_bug.cgi?id=710520
1390 2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1392 * gst/rtsp-server/rtsp-token.c:
1393 token: Fix bug when creating empty token
1394 We always want to have a valid GstStructure in the token.
1395 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
1397 2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
1399 * gst/rtsp-server/rtsp-thread-pool.c:
1400 thread-pool: avoid race in shutdown
1401 If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
1402 don't actually stop the mainloop ever. Solve this race by adding an idle source
1403 to the mainloop that calls the _quit. This way we immediately exit the mainloop
1404 if quit was called before we started it.
1406 2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1408 * tests/check/Makefile.am:
1409 * tests/check/gst/permissions.c:
1410 tests: Add unit tests for permissions
1411 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
1413 2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1415 * tests/check/gst/mediafactory.c:
1416 tests: Test mediafactory permissions
1417 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1419 2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1421 * gst/rtsp-server/rtsp-permissions.c:
1422 permissions: Fix refcounting when adding/removing roles
1423 Previously a role that was removed was unreffed twice, and when
1424 replacing an existing role the replaced role was freed while still being
1425 referenced. Both bugs are now fixed.
1426 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1428 2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1430 * tests/check/gst/media.c:
1431 * tests/check/gst/mediafactory.c:
1432 * tests/check/gst/rtspserver.c:
1433 tests: Check gst_rtsp_url_parse return value
1434 See https://bugzilla.gnome.org/show_bug.cgi?id=710202
1436 2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
1439 Automatic update of common submodule
1440 From 865aa20 to dbedaa0
1442 2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
1444 * gst/rtsp-server/rtsp-server.c:
1445 rtsp-server: Fix socket leak
1446 https://bugzilla.gnome.org/show_bug.cgi?id=710088
1448 2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
1450 * gst/rtsp-server/rtsp-session-pool.c:
1451 rtsp-session-pool: Make sure session IDs are properly URI-escaped
1452 https://bugzilla.gnome.org/show_bug.cgi?id=643812
1454 2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
1456 * examples/.gitignore:
1457 * examples/test-video.c:
1458 examples: fix compilation when WITH_AUTH is defined
1459 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1461 2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
1464 gitignore: Add new test binary
1466 2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
1468 * tests/check/Makefile.am:
1469 * tests/check/gst/threadpool.c:
1470 thread-pool: Add unit test for the thread pools
1471 https://bugzilla.gnome.org/show_bug.cgi?id=710228
1473 2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
1475 * gst/rtsp-server/rtsp-thread-pool.c:
1476 thread-pool: Fix thread leak when reusing threads
1477 https://bugzilla.gnome.org/show_bug.cgi?id=709730
1479 2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
1481 * gst/rtsp-server/rtsp-server.c:
1482 * tests/check/gst/rtspserver.c:
1483 tests: fixed racy behavior in rtspserver tests
1484 https://bugzilla.gnome.org/show_bug.cgi?id=710078
1486 2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
1488 * tests/check/gst/addresspool.c:
1489 tests: Improve address pool unit tests
1490 Add a range with mixed IPV4 and IPV6 addresses to pool.
1491 Get an IPV4 address from an IPV6-only pool.
1492 Get an IPV6 address from an IPV4-only pool.
1493 Reserve a IPV6 address from an IPV4-only pool.
1494 Check for unicast addresses in multicast-only pool.
1495 Check for unicast addresses in uni-/multicast-mixed pool.
1496 https://bugzilla.gnome.org/show_bug.cgi?id=710128
1498 2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1500 * gst/rtsp-server/rtsp-client.c:
1501 client: append query string in PAUSE/PLAY/TEARDOWN as well
1503 2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
1505 * gst/rtsp-server/rtsp-client.c:
1506 client: Add query to control path
1507 If the SETUP url contains a query it must be appended to the control
1508 path so that it matches any already created stream in the media. The
1509 query will also be appended to the session media path.
1511 2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1513 * gst/rtsp-server/rtsp-media.c:
1514 rtsp-media: remove old line
1516 2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
1518 * gst/rtsp-server/rtsp-stream.c:
1519 stream: Correct control comparison
1520 https://bugzilla.gnome.org/show_bug.cgi?id=709176
1522 2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1524 * gst/rtsp-server/rtsp-media.c:
1525 media: Check dynamically if the pipeline supports seeking
1526 We should not depend on whether or not the pipeline state change
1527 returned NO_PREROLL or not. A media could dynamically change its
1528 element and switch from seekable to non seekable so it's best to test
1529 the seekable nature of the pipeline dynamically when we try to do a seek.
1531 2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1533 * gst/rtsp-server/rtsp-media.c:
1534 media: Return FALSE if seeking is not supported
1536 2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1538 * gst/rtsp-server/rtsp-media.c:
1539 rtsp-media: don't seek accurate by default
1540 Accurate seeking is perhaps a little overkill in the most common situation and
1541 causes some formats (mp3) over slow media to seek extremely slowly.
1543 2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
1545 * tests/check/gst/rtspserver.c:
1546 tests: fix unit test
1547 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
1549 2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
1551 * gst/rtsp-server/rtsp-client.c:
1552 client: Reply 400 if media cannot be constructed
1553 Reply 400 Bad Request instead of 503 Service Unavailable if media
1554 cannot be constructed in SETUP.
1555 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
1557 2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
1559 * gst/rtsp-server/rtsp-client.c:
1560 client: Send setup reply once only
1561 If find_media() failed in handle_setup_request() two replies was sent.
1562 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
1564 2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
1567 Automatic update of common submodule
1568 From 6b03ba7 to 865aa20
1570 2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
1572 * gst/rtsp-server/rtsp-server.c:
1573 server: Emit client-connected signal earlier
1574 Emit client-connected before the client ref is given to a GSource,
1575 otherwise client-connected can be emitted after the client object has
1578 2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
1580 * gst/rtsp-server/rtsp-address-pool.c:
1581 * gst/rtsp-server/rtsp-address-pool.h:
1582 * gst/rtsp-server/rtsp-stream.c:
1583 * tests/check/gst/addresspool.c:
1584 addresspool: return reason of failure
1585 Let gst_rtsp_address_pool_reserve_address() return the reason why
1586 the address could not be reserved.
1587 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
1589 2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
1592 autogen.sh: Sync behaviour with other GStreamer modules
1593 Allows building from outside of tree amongst other things
1595 2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
1598 Automatic update of common submodule
1599 From b613661 to 6b03ba7
1601 2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
1604 Automatic update of common submodule
1605 From 74a6857 to b613661
1607 2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
1610 Automatic update of common submodule
1611 From 01a7a46 to 74a6857
1613 2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
1615 * gst/rtsp-server/rtsp-client.c:
1616 client: Do not read beyond end of path string
1617 If the setup was done without a control url, make sure we don't try to read the
1618 non-existing control string and crash.
1620 2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1622 * gst/rtsp-server/rtsp-client.c:
1623 client: Fix RTPInfo header
1624 Refactor the method to make the content_base.
1625 Use the content-base and the control url to construct the RTPInfo
1628 2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1630 * gst/rtsp-server/rtsp-client.c:
1631 client: map url to path only in describe
1632 Only map the request url to a path in the DESCRIBE method. The SDP then
1633 contains the base and control urls that should be used to SETUP/PAUSE/
1634 PLAY/TEARDOWN the media.
1636 2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1638 * gst/rtsp-server/rtsp-client.c:
1639 Revert "client: map URL to path in requests"
1640 This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
1641 This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
1642 contains the base and control urls which are used in the SETUP, PLAY,
1643 PAUSE and TEARDOWN requests.
1645 2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1647 * gst/rtsp-server/rtsp-client.c:
1648 client: map URL to path in requests
1650 2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1652 * gst/rtsp-server/rtsp-client.c:
1653 * gst/rtsp-server/rtsp-mount-points.c:
1654 * gst/rtsp-server/rtsp-mount-points.h:
1655 mount-points: make vmethod to make path from uri
1656 Make a vmethod to transform an url into a path. The path is then used to lookup
1657 the factory. This makes it possible to also use other bits of the url, such as
1658 the query parameters, to locate the factory.
1660 2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
1662 * gst/rtsp-server/rtsp-thread-pool.c:
1663 * gst/rtsp-server/rtsp-thread-pool.h:
1664 thread-pool: Add cleanup to wait for the threadpool to finish
1665 Also fix race condition if two threads are asking for the first
1666 thread from the thread pool at once. This would case two internal
1667 GThreadPools to be created.
1668 https://bugzilla.gnome.org/show_bug.cgi?id=707753
1670 2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
1672 * gst/rtsp-server/rtsp-client.c:
1673 * tests/check/gst/client.c:
1674 client: free threadpool
1675 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1677 2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
1679 * tests/check/gst/mountpoints.c:
1680 mountpoints tests: unref matched factories
1681 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1683 2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
1685 * tests/check/gst/media.c:
1686 media tests: unref thread pool and caps
1687 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1689 2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
1691 * gst/rtsp-server/rtsp-auth.c:
1692 * gst/rtsp-server/rtsp-media-factory.c:
1693 * gst/rtsp-server/rtsp-media.c:
1694 auth, media, media-factory: unref permissions
1695 https://bugzilla.gnome.org/show_bug.cgi?id=707638
1697 2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1699 * examples/Makefile.am:
1700 Makefile: add rule for appsrc example
1702 2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1704 * examples/test-appsrc.c:
1705 tests: add appsrc example
1706 Add an example on how to use appsrc to feed the server pipeline with data.
1708 2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
1710 * gst/rtsp-server/rtsp-client.c:
1711 rtsp-client: remove query part from content-base string
1712 Make sure that after the control url has been resolved, it's
1713 not a part of the query-string.
1714 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
1716 2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1718 * gst/rtsp-server/rtsp-client.c:
1719 client: don't check url in response
1720 There is no url or method in the response to check
1722 2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1724 * gst/rtsp-server/rtsp-client.c:
1725 * gst/rtsp-server/rtsp-client.h:
1726 Add handle-response signal for when we receive a GET_PARAMETER response
1728 2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1730 * gst/rtsp-server/rtsp-server.c:
1731 Fix gst_rtsp_server_client_filter, using wrong variable type
1733 2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
1735 * gst/rtsp-server/rtsp-media-factory-uri.c:
1736 rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
1737 For AAC we need to check for framed=true instead of parsed=true.
1738 https://bugzilla.gnome.org/show_bug.cgi?id=701384
1740 2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1742 * gst/rtsp-server/rtsp-stream.c:
1743 stream: optimize pipeline for protocols
1744 When TCP is not an allowed protocol for the stream, avoid creating the
1745 appsrc/appsink/queue and tee elements.
1747 2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1749 * gst/rtsp-server/rtsp-media.c:
1750 media: set protocols on streams
1752 2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1754 * gst/rtsp-server/rtsp-client.c:
1755 client: use protocols supported by stream
1757 2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1759 * gst/rtsp-server/rtsp-media-factory.c:
1760 * gst/rtsp-server/rtsp-media.c:
1761 * gst/rtsp-server/rtsp-stream.c:
1762 media-factory: allow all protocols
1764 2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1766 * gst/rtsp-server/rtsp-media.c:
1767 media: configure protocols in new streams
1769 2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1771 * gst/rtsp-server/rtsp-stream.c:
1772 * gst/rtsp-server/rtsp-stream.h:
1773 stream: add protocols property
1775 2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1777 * gst/rtsp-server/rtsp-media.c:
1778 rtsp-media: send state in "new-state" signal
1779 https://bugzilla.gnome.org/show_bug.cgi?id=705110
1781 2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
1784 build: add subdir-objects to AM_INIT_AUTOMAKE
1785 Fixes warnings with automake 1.14
1786 https://bugzilla.gnome.org/show_bug.cgi?id=705350
1788 2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1790 * docs/libs/gst-rtsp-server-sections.txt:
1791 * gst/rtsp-server/rtsp-client.c:
1792 * gst/rtsp-server/rtsp-server.c:
1793 * gst/rtsp-server/rtsp-server.h:
1794 server: add method to iterate clients of server
1796 2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1798 * gst/rtsp-server/rtsp-media.c:
1799 * gst/rtsp-server/rtsp-media.h:
1800 Add vmethod for rtsp-media subclass to access rtpbin
1802 2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1804 * gst/rtsp-server/rtsp-client.h:
1805 small documentation fix
1807 2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1809 * gst/rtsp-server/rtsp-client.c:
1810 Do not take range header if range is invalid
1812 2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1814 * docs/libs/gst-rtsp-server-sections.txt:
1815 * gst/rtsp-server/rtsp-media.c:
1816 media: add docs for new method
1818 2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1820 * gst/rtsp-server/rtsp-media.c:
1821 * gst/rtsp-server/rtsp-media.h:
1822 Add API to rtsp-media set the pipeline's state
1824 2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
1826 * gst/rtsp-server/rtsp-media.c:
1827 Update current position/duration when gst_rtsp_media_get_range_string is called
1829 2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1831 * examples/test-cgroups.c:
1832 tests: add some more docs
1834 2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1836 * examples/test-cgroups.c:
1837 * gst/rtsp-server/Makefile.am:
1838 * gst/rtsp-server/rtsp-auth.c:
1839 * gst/rtsp-server/rtsp-auth.h:
1840 * gst/rtsp-server/rtsp-client.c:
1841 * gst/rtsp-server/rtsp-client.h:
1842 * gst/rtsp-server/rtsp-context.c:
1843 * gst/rtsp-server/rtsp-context.h:
1844 * gst/rtsp-server/rtsp-params.c:
1845 * gst/rtsp-server/rtsp-params.h:
1846 * gst/rtsp-server/rtsp-server.c:
1847 * gst/rtsp-server/rtsp-thread-pool.c:
1848 * gst/rtsp-server/rtsp-thread-pool.h:
1849 * tests/check/gst/client.c:
1850 ClientState -> Context
1851 Rename the clientstate to context and put the code in a separate file.
1853 2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1855 * examples/test-auth.c:
1856 * gst/rtsp-server/rtsp-auth.c:
1857 * gst/rtsp-server/rtsp-auth.h:
1858 auth: add support for default token
1859 The default token is used when the user is not authenticated and can be used to
1860 give minimal permissions.
1862 2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1864 * examples/test-auth.c:
1865 * gst/rtsp-server/rtsp-auth.c:
1866 auth: use defines when possible
1868 2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1870 * gst/rtsp-server/rtsp-address-pool.c:
1871 address-pool: improve docs
1873 2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1875 * gst/rtsp-server/rtsp-permissions.c:
1876 permissions: add the role to the copy
1878 2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
1880 * gst/rtsp-server/rtsp-permissions.c:
1881 permissions: Also copy the roles
1883 2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
1885 * gst/rtsp-server/rtsp-permissions.c:
1886 permissions: Make it build
1888 2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1890 * gst/rtsp-server/rtsp-address-pool.h:
1893 2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1895 * docs/libs/gst-rtsp-server-sections.txt:
1896 * gst/rtsp-server/rtsp-auth.c:
1897 * gst/rtsp-server/rtsp-auth.h:
1898 * gst/rtsp-server/rtsp-media.c:
1899 * gst/rtsp-server/rtsp-session-media.c:
1900 * gst/rtsp-server/rtsp-stream-transport.c:
1901 * gst/rtsp-server/rtsp-stream-transport.h:
1902 * gst/rtsp-server/rtsp-stream.c:
1903 * tests/check/gst/client.c:
1906 2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1908 * docs/libs/gst-rtsp-server-sections.txt:
1909 * gst/rtsp-server/rtsp-address-pool.c:
1910 * gst/rtsp-server/rtsp-address-pool.h:
1911 * tests/check/gst/addresspool.c:
1912 * tests/check/gst/rtspserver.c:
1913 address-pool: cleanups
1914 Remove redundant method, improve docs.
1916 2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1918 * docs/libs/gst-rtsp-server-sections.txt:
1919 * gst/rtsp-server/rtsp-auth.h:
1920 * gst/rtsp-server/rtsp-permissions.c:
1921 * gst/rtsp-server/rtsp-permissions.h:
1922 * gst/rtsp-server/rtsp-token.c:
1925 2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1927 * gst/rtsp-server/rtsp-permissions.c:
1928 permissions: implement _remove_role
1930 2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1932 * gst/rtsp-server/rtsp-permissions.c:
1933 permissions: update docs
1935 2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1937 * tests/check/gst/client.c:
1938 tests: simplify tests
1939 Client settings are now disabled by default so we don't need an auth
1940 module to disable them.
1942 2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1944 * gst/rtsp-server/rtsp-auth.c:
1945 auth: add default authorizations
1946 When no auth module is specified, use our table of defaults to look up the
1947 default value of the check instead of always allowing everything. This was
1948 we can disallow client settings by default.
1950 2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1953 README: update readme
1955 2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1957 * gst/rtsp-server/rtsp-thread-pool.c:
1958 * gst/rtsp-server/rtsp-thread-pool.h:
1959 thread-pool: add more docs
1961 2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1963 * gst/rtsp-server/rtsp-thread-pool.c:
1964 * gst/rtsp-server/rtsp-thread-pool.h:
1965 thread-pool: fix race in thread reuse
1966 If we try to reuse a thread right after we made it stop, we end up using a
1967 stopped thread. Catch this case and only reuse threads that are not stopping.
1969 2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1971 * gst/rtsp-server/rtsp-server.c:
1972 server: add small debug
1974 2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1976 * tests/check/gst/client.c:
1978 Add some permissions to media so we can use the auth and enable
1981 2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1983 * gst/rtsp-server/rtsp-client.c:
1984 client: support pushed context in handle_request
1985 If we already have a pushed state, reuse it and add our own things. This makes
1986 it easier to write tests.
1988 2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1990 * gst/rtsp-server/rtsp-auth.c:
1991 auth: don't auth on methods
1992 Don't authorize on methods anymore but on the resources that we
1993 try to access, this is more flexible.
1994 Move the authorization checks to where they are needed and let the
1995 check return the response on error.
1997 2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1999 * gst/rtsp-server/rtsp-mount-points.c:
2000 mount-points: add some debug
2002 2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2004 * tests/check/gst/client.c:
2005 tests: almost fix test
2007 2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2009 * gst/rtsp-server/rtsp-auth.c:
2010 * gst/rtsp-server/rtsp-auth.h:
2011 * gst/rtsp-server/rtsp-client.c:
2012 * gst/rtsp-server/rtsp-client.h:
2013 * gst/rtsp-server/rtsp-server.c:
2014 * gst/rtsp-server/rtsp-server.h:
2015 auth: let the auth module check client_settings
2016 Let the auth module decide if client settings are allowed for the
2019 2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2021 * gst/rtsp-server/rtsp-token.c:
2022 * gst/rtsp-server/rtsp-token.h:
2023 token: add method to check boolean permission
2025 2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2027 * examples/test-auth.c:
2028 * examples/test-cgroups.c:
2029 * gst/rtsp-server/rtsp-token.c:
2030 * gst/rtsp-server/rtsp-token.h:
2031 token: simplify token constructor
2032 Use variable arguments to make easier API.
2034 2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2036 * examples/test-auth.c:
2037 * examples/test-cgroups.c:
2038 * gst/rtsp-server/rtsp-media-factory.c:
2039 * gst/rtsp-server/rtsp-media-factory.h:
2040 media-factory: add convenience API for factory
2042 2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2044 * examples/test-auth.c:
2045 * examples/test-cgroups.c:
2046 * gst/rtsp-server/rtsp-permissions.c:
2047 * gst/rtsp-server/rtsp-permissions.h:
2048 permissions: simplify API a little
2049 Avoid passing GstStructure in the add_role method, use varargs instead
2050 to construct the structure behind the scenes. We can then also use the
2051 structure name as the role and simplify some more logic.
2053 2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2055 * gst/rtsp-server/rtsp-auth.c:
2058 2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2060 * gst/rtsp-server/rtsp-auth.c:
2061 * gst/rtsp-server/rtsp-auth.h:
2062 * gst/rtsp-server/rtsp-client.c:
2063 auth: handle unauthorized response
2064 Move handling of the unauthorized response to the auth module, it can add
2065 the appropriate headers to request authorization for the required method
2066 much better than the client.
2068 2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2070 * gst/rtsp-server/rtsp-client.c:
2071 * gst/rtsp-server/rtsp-client.h:
2072 client: allow for sending any message, not only requests
2073 Change the _send_request() method to _send_message() so that we
2074 can both send requests and replies.
2076 2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2078 * docs/libs/gst-rtsp-server-sections.txt:
2079 * gst/rtsp-server/rtsp-server.h:
2082 2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2084 * examples/test-video.c:
2085 * gst/rtsp-server/rtsp-auth.c:
2086 * gst/rtsp-server/rtsp-auth.h:
2087 * gst/rtsp-server/rtsp-server.c:
2088 * gst/rtsp-server/rtsp-server.h:
2089 auth: move TLS handling to auth module
2090 Remove the TLS settings on the server and move it to the auth module because
2091 that is where security related bits go.
2093 2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2095 * gst/rtsp-server/rtsp-client.c:
2096 * gst/rtsp-server/rtsp-client.h:
2097 client: add state push/pop
2099 2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2101 * gst/rtsp-server/rtsp-client.c:
2102 * gst/rtsp-server/rtsp-client.h:
2103 client: add connection to state
2105 2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2107 * gst/rtsp-server/rtsp-mount-points.c:
2108 mount-points: fix debug
2110 2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2112 * tests/check/gst/media.c:
2113 tests: fix media test
2115 2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2117 * gst/rtsp-server/rtsp-thread-pool.c:
2118 thread-pool: we don't require a state
2120 2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2122 * gst/rtsp-server/rtsp-server.c:
2123 server: let context ref the server
2124 So that we don't risk losing the server object early anc crash.
2126 2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2128 * tests/check/gst/client.c:
2129 tests: fix client test
2131 2013-07-11 16:57:14 +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-auth.c:
2138 * gst/rtsp-server/rtsp-client.c:
2139 * gst/rtsp-server/rtsp-client.h:
2140 * gst/rtsp-server/rtsp-media-factory-uri.c:
2141 * gst/rtsp-server/rtsp-media-factory.c:
2142 * gst/rtsp-server/rtsp-media-factory.h:
2143 * gst/rtsp-server/rtsp-media.c:
2144 * gst/rtsp-server/rtsp-mount-points.c:
2145 * gst/rtsp-server/rtsp-params.c:
2146 * gst/rtsp-server/rtsp-permissions.c:
2147 * gst/rtsp-server/rtsp-sdp.c:
2148 * gst/rtsp-server/rtsp-server.c:
2149 * gst/rtsp-server/rtsp-server.h:
2150 * gst/rtsp-server/rtsp-session-media.c:
2151 * gst/rtsp-server/rtsp-session-pool.c:
2152 * gst/rtsp-server/rtsp-session.c:
2153 * gst/rtsp-server/rtsp-stream-transport.c:
2154 * gst/rtsp-server/rtsp-stream.c:
2155 * gst/rtsp-server/rtsp-thread-pool.c:
2156 * gst/rtsp-server/rtsp-token.c:
2159 2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2161 * gst/rtsp-server/rtsp-session-pool.c:
2162 * gst/rtsp-server/rtsp-session-pool.h:
2163 session-pool: make vmethod to create a session
2164 Make a vmethod to create a sessions so that subclasses can create
2165 custom session objects
2167 2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2169 * gst/rtsp-server/rtsp-auth.c:
2170 * gst/rtsp-server/rtsp-media-factory.h:
2171 * gst/rtsp-server/rtsp-media.h:
2172 * gst/rtsp-server/rtsp-mount-points.h:
2173 * gst/rtsp-server/rtsp-session-pool.h:
2174 * gst/rtsp-server/rtsp-stream.h:
2177 2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2179 * docs/libs/gst-rtsp-server-docs.sgml:
2180 * docs/libs/gst-rtsp-server-sections.txt:
2181 * gst/rtsp-server/rtsp-address-pool.c:
2182 * gst/rtsp-server/rtsp-address-pool.h:
2183 * gst/rtsp-server/rtsp-auth.c:
2184 * gst/rtsp-server/rtsp-client.h:
2185 * gst/rtsp-server/rtsp-media-factory.h:
2186 * gst/rtsp-server/rtsp-media.c:
2187 * gst/rtsp-server/rtsp-media.h:
2188 * gst/rtsp-server/rtsp-permissions.c:
2189 * gst/rtsp-server/rtsp-permissions.h:
2190 * gst/rtsp-server/rtsp-server.h:
2191 * gst/rtsp-server/rtsp-session-media.c:
2192 * gst/rtsp-server/rtsp-session-media.h:
2193 * gst/rtsp-server/rtsp-session-pool.h:
2194 * gst/rtsp-server/rtsp-session.h:
2195 * gst/rtsp-server/rtsp-stream-transport.h:
2196 * gst/rtsp-server/rtsp-stream.c:
2197 * gst/rtsp-server/rtsp-thread-pool.h:
2200 2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2203 * examples/Makefile.am:
2204 configure: compile cgroup example conditionally
2205 Only compile the cgroup example when we have libcgroup
2207 2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2210 * examples/Makefile.am:
2211 * examples/test-cgroups.c:
2212 examples: add cgroups example
2214 2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2216 * tests/check/gst/rtspserver.c:
2217 tests: fix compilation
2219 2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2221 * gst/rtsp-server/rtsp-thread-pool.c:
2222 thread-pool: fix vmethod invocation
2224 2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2226 * gst/rtsp-server/rtsp-thread-pool.c:
2227 * gst/rtsp-server/rtsp-thread-pool.h:
2228 thread-pool: store thread type in thread
2230 2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2232 * gst/rtsp-server/rtsp-client.c:
2233 client: pass thread from pool to media _prepare
2234 Get a thread from the configured threadpool and pass it to the prepare method of
2237 2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2239 * gst/rtsp-server/rtsp-media.c:
2240 * gst/rtsp-server/rtsp-media.h:
2241 media: Accept a thread in _prepare
2242 Remove out own threadpool handling and use the provided thread and
2243 maincontext for the bus messages and the state changes.
2245 2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2247 * gst/rtsp-server/rtsp-server.c:
2248 server: configure client thread pool
2250 2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2252 * gst/rtsp-server/rtsp-client.c:
2253 * gst/rtsp-server/rtsp-client.h:
2254 client: add method to configure thread pool
2256 2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2258 * gst/rtsp-server/rtsp-client.h:
2259 * gst/rtsp-server/rtsp-server.c:
2260 * gst/rtsp-server/rtsp-server.h:
2261 server: use thread pool
2262 Use the thread pool instead of doing our own thing.
2264 2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2266 * gst/rtsp-server/Makefile.am:
2267 * gst/rtsp-server/rtsp-thread-pool.c:
2268 * gst/rtsp-server/rtsp-thread-pool.h:
2269 thread-pool: add object to manage threads
2270 Add an object to manage the client and media threads.
2272 2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2274 * gst/rtsp-server/rtsp-auth.c:
2275 auth: debug authorization check
2277 2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2279 * gst/rtsp-server/rtsp-media.c:
2280 media: start media pipeline in context
2281 Start the media pipeline in the provided context (or our default one
2282 when NULL). This makes sure that we run the bus thread in this context and that
2283 all media threads are children of this context.
2285 2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2287 * gst/rtsp-server/rtsp-media-factory.c:
2288 factory: pass permissions to media by default
2290 2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2292 * examples/test-auth.c:
2293 test: add permissions to auth test
2294 Ass some permissions to the media factory in the test.
2296 2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2298 * gst/rtsp-server/rtsp-auth.c:
2299 * gst/rtsp-server/rtsp-auth.h:
2300 * gst/rtsp-server/rtsp-client.c:
2301 auth: simplify auth checks
2302 Remove client from methods, it's now in the state
2303 Perform the check specified by the string, use the information from the
2304 thread local context.
2306 2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2308 * gst/rtsp-server/rtsp-client.c:
2309 * gst/rtsp-server/rtsp-client.h:
2310 client: add state to current thread
2311 Add the client to the ClientState object.
2312 Place the ClientState on the current thread.
2314 2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2316 * gst/rtsp-server/rtsp-media-factory.c:
2317 * gst/rtsp-server/rtsp-media-factory.h:
2318 * gst/rtsp-server/rtsp-media.c:
2319 * gst/rtsp-server/rtsp-media.h:
2320 media: make it possible to set permissions
2321 Make it possible to set permissions on media and media factory objects
2323 2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2325 * gst/rtsp-server/Makefile.am:
2326 * gst/rtsp-server/rtsp-permissions.c:
2327 * gst/rtsp-server/rtsp-permissions.h:
2328 permissions: add permissions object
2329 Add a mini object to store permissions based on a role.
2331 2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2333 * examples/test-auth.c:
2334 * gst/rtsp-server/rtsp-auth.c:
2335 * gst/rtsp-server/rtsp-auth.h:
2336 * gst/rtsp-server/rtsp-client.c:
2337 auth: add auth checks
2338 Add an enum with auth checks and implement the checks in the auth object.
2339 Perform the checks from the client.
2341 2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2343 * examples/test-auth.c:
2344 * gst/rtsp-server/rtsp-auth.c:
2345 * gst/rtsp-server/rtsp-auth.h:
2346 * gst/rtsp-server/rtsp-client.h:
2347 auth: use the token after authentication
2348 After we authenticated a user, keep the Token around in the state.
2350 2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2352 * gst/rtsp-server/rtsp-client.c:
2353 * gst/rtsp-server/rtsp-media.c:
2354 * gst/rtsp-server/rtsp-media.h:
2355 * tests/check/gst/media.c:
2356 media: add optional context for bus messages
2357 Add an optional mainloop to _prepare that will handle the bus messages instead
2358 of always using the shared mainloop.
2360 2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2362 * gst/rtsp-server/Makefile.am:
2363 * gst/rtsp-server/rtsp-token.c:
2364 * gst/rtsp-server/rtsp-token.h:
2365 token: add authorization token
2366 Add a simply miniobject that contains the authorizations. The object contains a
2367 GstStructure that hold all authorization fields. When a user is authenticated,
2368 the auth module will create a Token for the user. The token is then used to
2369 check what operations the user is allowed to do and various other configuration
2372 2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2374 * examples/test-auth.c:
2375 * gst/rtsp-server/rtsp-auth.c:
2376 * gst/rtsp-server/rtsp-auth.h:
2377 * gst/rtsp-server/rtsp-client.c:
2378 * gst/rtsp-server/rtsp-client.h:
2379 * gst/rtsp-server/rtsp-media-factory.c:
2380 * gst/rtsp-server/rtsp-media-factory.h:
2381 * gst/rtsp-server/rtsp-media.c:
2382 * gst/rtsp-server/rtsp-media.h:
2383 auth: remove auth from media and factory
2384 Remove the auth object from media and factory. We want to have the RTSPClient
2385 authenticate and authorize resources, there is no need to place another auth
2386 manager on the media/factory.
2388 2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2390 * examples/test-auth.c:
2391 * gst/rtsp-server/rtsp-auth.c:
2392 * gst/rtsp-server/rtsp-auth.h:
2393 * gst/rtsp-server/rtsp-client.h:
2394 auth: add support for multiple basic auth tokens
2395 Make it possible to add multiple basic authorisation tokens to one authorization
2396 object. Associate with each token an authorization group that will define what
2397 capabilities are allowed.
2399 2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2401 * gst/rtsp-server/rtsp-client.c:
2402 client: error out on non-aggregate control
2403 We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
2405 2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2407 * gst/rtsp-server/rtsp-client.c:
2408 client: rework setup request a little
2409 Cache the media in DESCRIBE based on the longest matching path with the uri
2410 that we can find in the mount points.
2411 Rework the setup request a little to get the media from the session or from
2412 the longest matching path, this way we can derive the control string as
2413 everything after the path instead of hardcoding it.
2414 Find the stream based on the control string and only open a session when all
2417 2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2419 * gst/rtsp-server/rtsp-media.c:
2420 * gst/rtsp-server/rtsp-media.h:
2421 media: add method to find a stream by control url
2423 2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2425 * gst/rtsp-server/rtsp-stream.c:
2426 * gst/rtsp-server/rtsp-stream.h:
2427 stream: add method to check control url of stream
2429 2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2431 * gst/rtsp-server/rtsp-client.c:
2432 * gst/rtsp-server/rtsp-session-media.c:
2433 * gst/rtsp-server/rtsp-session-media.h:
2434 * gst/rtsp-server/rtsp-session.c:
2435 * gst/rtsp-server/rtsp-session.h:
2436 session: use path matching for session media
2437 Use a path string instead of a uri to lookup session media in the sessions. Also
2438 use path matching to find the largest possible path that matches.
2440 2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2442 * gst/rtsp-server/rtsp-client.c:
2443 * gst/rtsp-server/rtsp-mount-points.c:
2444 * gst/rtsp-server/rtsp-mount-points.h:
2445 * tests/check/gst/mountpoints.c:
2446 mount-points: remove useless vmethod
2447 Making lookups in the mount points should not be done with a URL, if there is a
2448 mapping to be done from URL to mount points, we'll need to do it somewhere
2451 2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2453 * gst/rtsp-server/rtsp-mount-points.c:
2454 * gst/rtsp-server/rtsp-mount-points.h:
2455 * tests/check/gst/mountpoints.c:
2456 mount-points: improve mount point searching
2457 Use a GSequence to keep track of the mount points.
2458 Match a URL to the longest matching registered mount point. This should be the
2459 URL to perform aggreagate control and the remainder is the stream specific
2461 Add some unit tests for this.
2463 2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
2465 * gst/rtsp-server/Makefile.am:
2466 rtsp-server: Allow building of static library
2468 2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2470 * tests/check/gst/mediafactory.c:
2471 tests: fix compilation
2473 2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2475 * gst/rtsp-server/rtsp-sdp.c:
2476 sdp: get control string from stream
2477 Use the control string as configured in the stream.
2479 2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2481 * gst/rtsp-server/rtsp-stream.c:
2482 * gst/rtsp-server/rtsp-stream.h:
2483 stream: add methods and property to set control string
2485 2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2487 * gst/rtsp-server/rtsp-client.c:
2489 Rename variables for clarity
2490 Keep media in state when we can
2492 2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2494 * gst/rtsp-server/rtsp-client.c:
2495 * gst/rtsp-server/rtsp-stream.c:
2496 * gst/rtsp-server/rtsp-stream.h:
2497 stream: add more support for IPv6
2498 Rename _get_address to _get_multicast_address in GstRTSPStream to
2499 make it clear that this function only deals with multicast.
2500 Make it possible to have both an IPv4 and IPv6 multicast address on
2501 a stream. Give the client an IPv4 or IPv6 address depending on the
2502 address it used to connect to the server.
2503 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
2505 2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2507 * gst/rtsp-server/rtsp-client.c:
2510 2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2512 * gst/rtsp-server/rtsp-stream.c:
2513 stream: handle failed port allocation
2514 Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
2515 can't allocate any family at all. Also keep track of what port families we
2517 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
2519 2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2521 * gst/rtsp-server/rtsp-stream.c:
2522 stream: improve docs
2524 2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2526 * gst/rtsp-server/rtsp-stream-transport.c:
2527 stream-transport: remove old if 0 block
2529 2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
2531 * tests/check/gst/client.c:
2533 gst_rtsp_client_get_uri() has been removed
2534 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
2536 2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2538 * gst/rtsp-server/rtsp-client.c:
2539 * gst/rtsp-server/rtsp-client.h:
2540 client: add method to filter managed sessions
2541 Add a method to filter the sessions managed by this client connection.
2542 See https://bugzilla.gnome.org/show_bug.cgi?id=703016
2544 2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2546 * gst/rtsp-server/rtsp-client.c:
2547 * gst/rtsp-server/rtsp-client.h:
2548 client: remove _get_uri() method
2549 Remove the get_uri() method on the client. A client has no uri, the uri
2550 property is an internal property to manage the last cached media for
2553 2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2555 * gst/rtsp-server/rtsp-media-factory.h:
2556 media-factory: fix typo
2558 2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
2560 * gst/rtsp-server/rtsp-media.c:
2561 rtsp-media: Do not leak the query in default_query_stop
2562 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
2564 2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2566 * gst/rtsp-server/rtsp-media.c:
2567 media: don't unlock when conversion fails
2568 Don't unlock the state lock when conversion fails because it was not locked.
2570 2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2572 * gst/rtsp-server/rtsp-media.c:
2573 * gst/rtsp-server/rtsp-media.h:
2574 Add query_position and query_stop vmethods to rtsp-media
2576 2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2578 * gst/rtsp-server/rtsp-media.c:
2579 Fix typo in property install for rtsp-media's time-provider
2581 2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2583 * gst/rtsp-server/rtsp-client.c:
2584 * gst/rtsp-server/rtsp-client.h:
2585 client: clean some variables
2586 Clean some variables and add some guards to _send_request()
2588 2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
2590 * gst/rtsp-server/rtsp-client.c:
2591 * gst/rtsp-server/rtsp-client.h:
2592 Add gst_rtsp_client_send_request API
2593 This makes it possible to send arbitrary messages to a client, such as
2594 SET_PARAMETER or GET_PARAMETER
2596 2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2598 * gst/rtsp-server/rtsp-media.c:
2599 * gst/rtsp-server/rtsp-media.h:
2600 media: add _get_element() method
2601 Add method to get the element used when creating the media.
2602 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
2604 2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2606 * gst/rtsp-server/rtsp-media.c:
2609 2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
2611 * gst/rtsp-server/rtsp-stream.c:
2612 * gst/rtsp-server/rtsp-stream.h:
2613 stream: allow access to the rtp session
2614 https://bugzilla.gnome.org/show_bug.cgi?id=703004
2616 2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
2618 * gst/rtsp-server/rtsp-stream.c:
2619 * gst/rtsp-server/rtsp-stream.h:
2620 dscp qos support in gst-rtsp-stream
2621 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
2623 2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2625 * tests/check/gst/rtspserver.c:
2627 Actually do what the comment says. Also keep the old code around, not sure what
2628 should happen when you get a 454 from a TEARDOWN, does it close the connection?
2629 it currently doesn't.
2631 2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2633 * gst/rtsp-server/rtsp-client.c:
2634 client: also watch newly created session
2635 When we newly created a session, start watching it immediately instead of
2636 on the next request.
2638 2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
2640 * tests/check/gst/client.c:
2641 tests: add unit test for new-session
2642 See https://bugzilla.gnome.org/show_bug.cgi?id=701587
2644 2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2646 * gst/rtsp-server/rtsp-client.c:
2647 client: emit new-session when new session is created
2648 Only emit new-session when we created a new session for a client, not when a
2649 client picked up a previous session.
2650 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
2652 2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
2654 * gst/rtsp-server/rtsp-client.c:
2655 client: handle asterisk as path in requests
2656 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
2658 2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2660 * gst/rtsp-server/rtsp-media.c:
2661 media: handle segment query format mismatch
2662 It's possible that the segment query returns with a different format than what
2663 we asked for, handle this case also.
2665 2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
2667 * gst/rtsp-server/rtsp-media.c:
2668 media: use segment stop in collect_media_stats
2669 Use segment stop instead of duration as range end point.
2670 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
2672 2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2674 * gst/rtsp-server/rtsp-media.c:
2675 * tests/check/gst/media.c:
2676 rtsp-media: Do not leak the element in take_pipeline
2677 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
2679 2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
2681 * gst/rtsp-server/rtsp-client.c:
2682 * gst/rtsp-server/rtsp-client.h:
2683 rtsp-client: Make configure_client_transport virtual
2684 This patch makes configure_client_transport virtual. The functionality is
2685 needed to handle some weird clients sending multicast transport settings as url
2687 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
2689 2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
2691 * gst/rtsp-server/rtsp-client.c:
2692 * gst/rtsp-server/rtsp-client.h:
2693 rtsp-client: Make param_set and param_get virtual
2694 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
2696 2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
2698 * gst/rtsp-server/rtsp-client.c:
2699 * gst/rtsp-server/rtsp-media.c:
2700 * gst/rtsp-server/rtsp-media.h:
2701 media: convert_range replaces get_range_times
2702 get_range_times worked for handling UTC ranges for seeks, but we also
2703 need to convert back from NPT to the requested unit in
2704 get_range_string. convert_range is now used for both.
2705 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
2707 2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2709 * gst/rtsp-server/rtsp-client.c:
2710 * gst/rtsp-server/rtsp-sdp.c:
2711 * gst/rtsp-server/rtsp-sdp.h:
2712 sdp: cleanup sdp info
2713 We don't need to pass the proto, we can more easily check a boolean.
2714 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
2716 2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
2718 * gst/rtsp-server/rtsp-sdp.c:
2719 use 0.0.0.0 or :: for c= line instead of server address
2721 2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
2723 * gst/rtsp-server/rtsp-client.c:
2724 use local address, not remote, in SDP
2725 See https://bugzilla.gnome.org/show_bug.cgi?id=702063
2727 2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2730 Automatic update of common submodule
2731 From 098c0d7 to 01a7a46
2733 2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
2735 * gst/rtsp-server/rtsp-media.c:
2736 * gst/rtsp-server/rtsp-media.h:
2737 media: possibility to override range time conversion
2738 Make it possible to override the conversion from GstRTSPTimeRange to
2739 GstClockTimes, that is done before seeking on the media
2740 pipeline. Overriding can be useful for UTC ranges, where the default
2741 conversion gives nanoseconds since 1900.
2742 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
2744 2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
2746 * gst/rtsp-server/rtsp-server.c:
2747 * gst/rtsp-server/rtsp-server.h:
2748 rtsp-server: Expose the use_client_settings API
2749 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
2751 2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
2753 * gst/rtsp-server/rtsp-client.c:
2754 * gst/rtsp-server/rtsp-stream.c:
2755 * gst/rtsp-server/rtsp-stream.h:
2756 rtspstream: handle both ipv4 and ipv6 clients
2757 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
2759 2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2761 * gst/rtsp-server/rtsp-sdp.c:
2762 Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
2763 This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
2764 We already have a way to place extra attributes in the SDP by using a string
2765 property with prefix x- or a- in the caps.
2767 2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2769 * gst/rtsp-server/rtsp-sdp.c:
2770 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
2771 This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
2772 We already have a way to place extra attributes in the SDP, just make a string
2773 property in the payloader with a- or x- prefix.
2775 2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2777 * gst/rtsp-server/rtsp-sdp.c:
2778 rtsp: place a- and x- properties as attributes
2779 application/x-rtp has properties with a- and x- prefixes that should be
2780 placed as attributes in the SDP for the media instead of being added to the
2783 2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2785 * examples/Makefile.am:
2786 * examples/test-video.c:
2787 example: add TLS example
2789 2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2791 * gst/rtsp-server/rtsp-server.c:
2792 * gst/rtsp-server/rtsp-server.h:
2793 server: add support for TLS
2794 Add methods to set and get a TLS certificate.
2795 Add vmethod to configure a new connection. By default, configure the TLS
2796 certificate in a new connection if needed.
2798 2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2800 * gst/rtsp-server/rtsp-server.c:
2801 * gst/rtsp-server/rtsp-server.h:
2802 server: remove accept_client vmethod
2803 This vmethod is not very useful so remove it.
2805 2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2807 * gst/rtsp-server/rtsp-server.c:
2808 server: don't crash on NULL GError
2810 2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
2812 * gst/rtsp-server/rtsp-session-pool.c:
2813 rtsp-session-pool: corrected session timeout detection
2814 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
2816 2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2818 * gst/rtsp-server/rtsp-client.c:
2819 client: improve debug
2821 2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2823 * gst/rtsp-server/rtsp-client.c:
2824 * gst/rtsp-server/rtsp-client.h:
2825 * gst/rtsp-server/rtsp-server.c:
2826 server: refactor connection setup
2827 Let the server accept the socket connection and construct a GstRTSPConnection
2828 from it. Remove the code from the client and let the client only deal with
2829 a fully configure GstRTSPConnection object.
2830 We will need this later when the server will configure the connection for
2833 2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2835 * gst/rtsp-server/rtsp-stream.c:
2836 stream: keep the transport object alive
2837 Keep the transport object alive while we have it as qdata on the
2840 2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
2842 * gst/rtsp-server/rtsp-client.c:
2843 * gst/rtsp-server/rtsp-server.c:
2844 rtsp-server: Do not crash on nmapping of server
2845 * generate error when gst_rtsp_connection_accept fails
2846 * do not stop accepting incoming connections because
2847 accepting a client fails
2848 https://bugzilla.gnome.org/show_bug.cgi?id=701072
2850 2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
2852 * gst/rtsp-server/rtsp-client.c:
2853 rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
2854 https://bugzilla.gnome.org/show_bug.cgi?id=700953
2856 2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
2858 * gst/rtsp-server/rtsp-sdp.c:
2859 rtsp-sdp: Parse framerate caps field and set SDP attribute
2860 The SDP attribute and its format is described in RFC4566.
2861 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2863 2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
2865 * gst/rtsp-server/rtsp-sdp.c:
2866 rtsp-sdp: Parse width/height from caps and set SDP attribute
2867 The SDP attribute and its format is described in RFC6064.
2868 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2870 2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
2872 * gst/rtsp-server/rtsp-sdp.c:
2873 * tests/check/gst/client.c:
2874 rtsp-sdp: add bandwidth line
2875 https://bugzilla.gnome.org/show_bug.cgi?id=699220
2877 2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2880 Automatic update of common submodule
2881 From 5edcd85 to 098c0d7
2883 2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2885 * tests/check/gst/media.c:
2886 tests: add dynamic payloader prepare/unprepare check
2888 2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2890 * gst/rtsp-server/rtsp-media.c:
2891 media: release lock when removing fakesink
2893 2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2895 * gst/rtsp-server/rtsp-stream.c:
2896 stream: set elements to NULL before removing
2897 When removing a stream, set the elements to NULL first. This avoids
2898 element-is-not-in-NULL-state errors when we dispose the elements.
2900 2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
2903 Automatic update of common submodule
2904 From 3cb3d3c to 5edcd85
2906 2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2908 * gst/rtsp-server/rtsp-media.c:
2909 * gst/rtsp-server/rtsp-media.h:
2910 media: listen to pad-removed signals
2911 Listen to the pad-removed signal and remove the stream associated with the
2913 Add signal to be notified of the removed pad.
2914 Remove the fakesink in unprepare()
2915 Fix signatures of the signal methods
2917 2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2919 * examples/test-sdp.c:
2920 tests: add example of reusable pipelines
2922 2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
2924 * gst/rtsp-server/rtsp-stream.c:
2925 * gst/rtsp-server/rtsp-stream.h:
2926 stream: add method to get the srcpad
2928 2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
2930 * tests/check/gst/media.c:
2931 check: add media prepare/unprepare test
2932 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2934 2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
2936 * gst/rtsp-server/rtsp-media.c:
2937 media: disconnect from signal handlers in unprepare()
2938 We connected to the pad-added and no-more-pads signals in prepare() so
2939 we need to disconnect from them in unprepare().
2940 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2942 2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
2944 * gst/rtsp-server/rtsp-media.c:
2945 media: don't free streams array
2946 Don't free the streams array in the unprepare() method, they were not
2948 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2950 2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
2952 * gst/rtsp-server/rtsp-media.c:
2953 media: don't unref the pipeline in unprepare
2954 Unprepare() should undo what prepare() does. Because the pipeline is
2955 not created in prepare(), we should not unref it in unprepare()
2957 2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
2959 * gst/rtsp-server/rtsp-stream.c:
2960 stream: clear session and caps for reuse
2961 Set the session and caps to NULL after unref otherwise we might unref
2963 See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2965 2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
2967 * gst/rtsp-server/rtsp-client.c:
2968 client: send out teardown signal before tearing down
2969 The advantage is that in the signal handler you get direct access to
2970 information about what streams are about to get torn down (in the
2971 GstRTSPClientState).
2972 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
2974 2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
2976 * gst/rtsp-server/rtsp-client.c:
2977 * gst/rtsp-server/rtsp-client.h:
2978 client: expose connection
2979 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
2981 2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
2984 Automatic update of common submodule
2985 From aed87ae to 3cb3d3c
2987 2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2989 * gst/rtsp-server/rtsp-media.c:
2990 * gst/rtsp-server/rtsp-media.h:
2991 * gst/rtsp-server/rtsp-session-media.c:
2992 * gst/rtsp-server/rtsp-session-media.h:
2993 media: add method to get the base_time of the pipeline
2994 Together with a shared clock, this base-time could eventually be sent to
2995 the client so that it can reconstruct the exact running-time of the clock
2998 2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3000 * gst/rtsp-server/Makefile.am:
3001 * gst/rtsp-server/rtsp-media.c:
3002 * gst/rtsp-server/rtsp-media.h:
3003 * gst/rtsp-server/rtsp-sdp.c:
3004 media: add GstNetTimeProvider support
3005 Add a property to let the media provide a GstNetTimeProvider for its clock.
3006 Make methods to get the clock and nettimeprovider
3007 Add a x-gst-clock property to the SDP with the IP and port number of the nettime
3008 provider and also the current time of the clock. This should make it possible
3009 for (GStreamer) clients to slave their clock to the server clock.
3011 2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
3014 Automatic update of common submodule
3015 From 04c7a1e to aed87ae
3017 2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3019 * gst/rtsp-server/rtsp-media.c:
3020 media: wait for buffering to complete
3021 Wait for buffering to complete before changing the state to the target state.
3023 2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3025 * gst/rtsp-server/rtsp-media.c:
3026 media: small cleanup
3028 2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
3030 * tests/check/gst/rtspserver.c:
3031 tests: remove extra unref in test_setup_non_existing_stream
3032 The unref is not needed anymore, teardown runs without it.
3033 https://bugzilla.gnome.org/show_bug.cgi?id=696542
3035 2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
3037 * tests/check/gst/rtspserver.c:
3038 tests: GSocketService cleanup in test_bind_already_in_use
3039 Use g_socket_service_stop so the rtspserver test stops listening for
3040 incoming connections in test_bind_already_in_use.
3041 https://bugzilla.gnome.org/show_bug.cgi?id=696541
3043 2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
3045 * gst/rtsp-server/rtsp-media-factory.c:
3046 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
3047 Instead use a GWeakRef which is safe to use
3048 This is a known GLib bug, see:
3049 https://bugzilla.gnome.org/show_bug.cgi?id=667145
3051 2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
3053 * gst/rtsp-server/rtsp-client.c:
3054 * gst/rtsp-server/rtsp-media.c:
3055 * gst/rtsp-server/rtsp-media.h:
3056 * gst/rtsp-server/rtsp-sdp.c:
3057 * tests/check/gst/media.c:
3058 * tests/check/gst/rtspserver.c:
3059 rtsp-media/client: Reply to PLAY request with same type of Range
3060 Remember the type of Range from the PLAY request and use the same type for
3063 2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
3065 * gst/rtsp-server/rtsp-client.c:
3066 * gst/rtsp-server/rtsp-client.h:
3067 * tests/check/gst/client.c:
3068 rtsp-client: expose uri
3070 2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
3072 * tests/check/gst/mediafactory.c:
3073 tests: Hold ref while creating second media
3074 To test if the media aren't shared, make sure we keep the first one while creating a second
3075 otherwise the same memory address may be reused.
3077 2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
3080 configure: remove out-of-date comment
3082 2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
3085 .gitignore: ignore more build files
3087 2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
3089 * tests/check/Makefile.am:
3090 tests: use right _LIBS variable for gst-plugins-base libs
3092 2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3094 * tests/check/Makefile.am:
3095 check: add librtp to libs
3097 2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
3099 * tests/check/gst/rtspserver.c:
3100 tests: Add test to check selecting a port the server will send from
3102 2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
3104 * tests/check/gst/rtspserver.c:
3105 tests: Make sure packets are actually received
3107 2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
3109 * gst/rtsp-server/rtsp-stream.c:
3110 stream: Select unicast address from pool if appropriate
3112 2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
3114 * gst/rtsp-server/rtsp-stream.c:
3115 stream: Properties are always there in Gst 1.0
3117 2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
3119 * tests/check/gst/addresspool.c:
3120 tests: Add tests for unicast addresses in pool
3122 2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
3124 * gst/rtsp-server/rtsp-address-pool.c:
3125 * tests/check/gst/addresspool.c:
3126 address-pool: Verify that multicast addresses are used for multicast and vice-versa
3128 2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
3130 * docs/libs/gst-rtsp-server-sections.txt:
3131 * gst/rtsp-server/rtsp-address-pool.c:
3132 * gst/rtsp-server/rtsp-address-pool.h:
3133 * gst/rtsp-server/rtsp-stream.c:
3134 * tests/check/gst/addresspool.c:
3135 address-pool: Add unicast addresses
3137 2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
3140 * gst/rtsp-server/rtsp-server.c:
3141 * tests/check/gst/rtspserver.c:
3142 rtsp-server: Limit the number of threads per server instance
3143 If we exceed the maximum, just round robin the clients over the existing
3146 2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
3148 * gst/rtsp-server/rtsp-server.c:
3149 rtsp-server: No need to store the GMainContext in the client context
3151 2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
3153 * tests/check/gst/rtspserver.c:
3154 tests: Add test for client disconnection
3156 2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
3158 * tests/check/gst/rtspserver.c:
3159 tests: Test client and session timeouts with multiple threads
3161 2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
3163 * gst/rtsp-server/rtsp-address-pool.c:
3164 * gst/rtsp-server/rtsp-auth.c:
3165 * gst/rtsp-server/rtsp-client.c:
3166 * gst/rtsp-server/rtsp-media-factory-uri.c:
3167 * gst/rtsp-server/rtsp-media-factory.c:
3168 * gst/rtsp-server/rtsp-media.c:
3169 * gst/rtsp-server/rtsp-mount-points.c:
3170 * gst/rtsp-server/rtsp-server.c:
3171 * gst/rtsp-server/rtsp-session-media.c:
3172 * gst/rtsp-server/rtsp-session-pool.c:
3173 * gst/rtsp-server/rtsp-session.c:
3174 Document locking and its order
3176 2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
3178 * tests/check/gst/rtspserver.c:
3179 tests: Test that slow DESCRIBE don't block other clients
3181 2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
3183 * tests/check/gst/client.c:
3184 tests: Add tests for client-requested multicast address
3186 2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
3188 * docs/libs/gst-rtsp-server-sections.txt:
3189 docs: Put the various functions in the right sections
3191 2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
3193 * docs/libs/gst-rtsp-server-docs.sgml:
3194 * docs/libs/gst-rtsp-server-sections.txt:
3195 * gst/rtsp-server/rtsp-address-pool.c:
3196 * gst/rtsp-server/rtsp-address-pool.h:
3197 docs: Generate docs for GstRTSPAddressPool
3199 2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
3201 * gst/rtsp-server/rtsp-client.c:
3202 * gst/rtsp-server/rtsp-stream.c:
3203 * gst/rtsp-server/rtsp-stream.h:
3204 client: Check client provided addresses against the address pool
3206 2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
3208 * gst/rtsp-server/rtsp-address-pool.c:
3209 * gst/rtsp-server/rtsp-address-pool.h:
3210 * tests/check/gst/addresspool.c:
3211 address-pool: Add API to request a specific address from the pool
3212 Also add relevant unit tests.
3214 2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
3216 * tests/check/gst/mediafactory.c:
3217 tests: Check the passing around of a RTSPAddressPool
3218 Make sure the RTSPAddressPool is propagated from the MediaFactory all the
3219 way down to the stream.
3221 2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
3223 * tests/check/gst/addresspool.c:
3224 tests: Add more tests for the address pool
3226 2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
3228 * gst/rtsp-server/rtsp-address-pool.c:
3229 address-pool: Fix off by one error
3230 When splitting a port range, the port after a skip is not part of range.
3232 2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
3235 Automatic update of common submodule
3236 From 2de221c to 04c7a1e
3238 2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
3241 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
3242 AM_CONFIG_HEADER was removed in automake 1.13
3243 https://bugzilla.gnome.org/show_bug.cgi?id=693368
3245 2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
3248 Automatic update of common submodule
3249 From a942293 to 2de221c
3251 2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3253 * gst/rtsp-server/rtsp-client.c:
3254 client: make sure the watch exists while sending data
3255 Protect the send_func with a lock. This allows us to wait for sending
3256 to complete before changing the send_func and user_data. We add an
3257 extra ref to the watch to make sure that it remains valid during
3259 When closing the connection, set the send_func to NULL
3260 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
3262 2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3264 * tests/check/Makefile.am:
3265 tests: use GST_*_1_0 environment variables everywhere
3266 The _1_0 suffixed environment variables override the
3267 non-suffixed ones, so if we're in an environment that
3268 sets the _1_0 suffixed ones, such as jhbuild, we need
3269 to set those to make sure ours actually always get
3272 2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3275 Automatic update of common submodule
3276 From acb04d9 to a942293
3278 2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3280 * gst/rtsp-server/rtsp-client.c:
3281 rtsp-client: set the client backlog
3282 Set the client backlog to a reasonable default
3284 2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
3286 * gst/rtsp-server/rtsp-media.c:
3287 rtsp-media: Make the element a constructor parameter
3288 https://bugzilla.gnome.org/show_bug.cgi?id=689594
3290 2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
3292 * docs/libs/Makefile.am:
3293 docs: Link with gcov library when gcov is enabled
3294 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
3296 2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3298 * gst/rtsp-server/rtsp-media.c:
3299 media: match prepare with unprepare
3300 Really unprepare when there were an equal amount of prepare calls.
3302 2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3304 * gst/rtsp-server/rtsp-media.c:
3305 media: media has to be unprepared in finalize
3306 Because unprepare takes away the last ref on the media.
3308 2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3310 * gst/rtsp-server/rtsp-client.c:
3311 Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
3312 This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
3313 We can't use the refcount to trigger unprepare because it is the unprepare call
3314 that removes the last refcount after all messages are consumed. What we should
3315 probably do is make a prepared refcount and only unprepare when the refcount
3318 2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3320 * gst/rtsp-server/rtsp-media.c:
3321 media: let the source unref the last media ref
3322 the last ref to the media is held by the source so we don't need to add more ref
3323 and unrefs, we simply destroy the media when the source is gone.
3325 2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3327 * gst/rtsp-server/rtsp-media.c:
3328 media: improve debug
3330 2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3332 * gst/rtsp-server/rtsp-media.c:
3334 Make sure we are in the right state when collecting the position and duration.
3335 Only make ourselves PREPARED when we were previously PREPARING.
3337 2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3339 * gst/rtsp-server/rtsp-media.c:
3340 media: use g_object_ref/unref for GObjects
3342 2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
3344 * gst/rtsp-server/rtsp-client.c:
3345 client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
3346 Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
3347 GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
3348 isn't being used anymore.
3350 2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
3352 * gst/rtsp-server/rtsp-media.c:
3353 Fix compiler warning
3355 2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
3357 * gst/rtsp-server/rtsp-media-factory-uri.c:
3358 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
3360 2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3362 * gst/rtsp-server/rtsp-session-media.h:
3365 2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3367 * gst/rtsp-server/rtsp-media.c:
3368 * tests/check/gst/media.c:
3369 media: avoid element leak
3371 2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3373 * gst/rtsp-server/rtsp-media.c:
3374 media: require an element in media constructor
3376 2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3378 * gst/rtsp-server/rtsp-client.c:
3379 Revert "client: TEARDOWN brings that state to Init again"
3380 This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
3381 The object is already disposed, there is no point in setting the state.
3383 2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3385 * gst/rtsp-server/rtsp-client.c:
3386 client: TEARDOWN brings that state to Init again
3388 2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3390 * docs/libs/gst-rtsp-server-sections.txt:
3391 * examples/test-auth.c:
3392 * gst/rtsp-server/rtsp-auth.c:
3393 * gst/rtsp-server/rtsp-auth.h:
3394 * gst/rtsp-server/rtsp-client.c:
3395 * gst/rtsp-server/rtsp-client.h:
3396 * gst/rtsp-server/rtsp-media-factory-uri.c:
3397 * gst/rtsp-server/rtsp-media-factory-uri.h:
3398 * gst/rtsp-server/rtsp-media-factory.c:
3399 * gst/rtsp-server/rtsp-media-factory.h:
3400 * gst/rtsp-server/rtsp-media.c:
3401 * gst/rtsp-server/rtsp-media.h:
3402 * gst/rtsp-server/rtsp-mount-points.c:
3403 * gst/rtsp-server/rtsp-mount-points.h:
3404 * gst/rtsp-server/rtsp-sdp.c:
3405 * gst/rtsp-server/rtsp-server.c:
3406 * gst/rtsp-server/rtsp-server.h:
3407 * gst/rtsp-server/rtsp-session-media.c:
3408 * gst/rtsp-server/rtsp-session-media.h:
3409 * gst/rtsp-server/rtsp-session-pool.c:
3410 * gst/rtsp-server/rtsp-session-pool.h:
3411 * gst/rtsp-server/rtsp-session.c:
3412 * gst/rtsp-server/rtsp-session.h:
3413 * gst/rtsp-server/rtsp-stream-transport.c:
3414 * gst/rtsp-server/rtsp-stream-transport.h:
3415 * gst/rtsp-server/rtsp-stream.c:
3416 * gst/rtsp-server/rtsp-stream.h:
3417 * tests/check/gst/media.c:
3418 rtsp: make object details private
3419 Make all object details private
3420 Add methods to access private bits
3422 2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3424 * tests/check/Makefile.am:
3425 * tests/check/gst/media.c:
3426 tests: add media tests
3428 2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3430 * gst/rtsp-server/rtsp-media.c:
3431 media: check if prepared for some methods
3432 Check that the media object is prepared before doing seek and getting the
3433 current position etc.
3434 Add some g_return checks.
3436 2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3438 * tests/check/Makefile.am:
3439 * tests/check/gst/mediafactory.c:
3440 tests: add mediafactory test
3442 2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3444 * gst/rtsp-server/rtsp-stream.c:
3445 stream: improve debug
3447 2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3449 * gst/rtsp-server/rtsp-media.c:
3450 * gst/rtsp-server/rtsp-media.h:
3451 media: unref pipeline in finalize to avoid leaking it
3453 2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3455 * gst/rtsp-server/rtsp-media-factory-uri.c:
3456 * gst/rtsp-server/rtsp-media.c:
3457 rtsp: use gst_object_unref on GstObjects
3459 2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3461 * gst/rtsp-server/rtsp-media-factory.c:
3462 media-factory: require an url
3464 2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3466 * examples/test-uri.c:
3467 examples: fix include
3469 2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3471 * gst/rtsp-server/rtsp-server.h:
3472 server: remove unused include
3474 2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3476 * tests/check/Makefile.am:
3477 * tests/check/gst/mountpoints.c:
3478 tests: add test for mountpoints
3480 2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3482 * gst/rtsp-server/rtsp-client.c:
3483 client: fix factory leak
3484 Keep the factory in the state object only for authorization checks and make
3485 sure we unref it on failure. Also don't keep invalid objects in the state
3488 2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3490 * gst/rtsp-server/rtsp-mount-points.c:
3491 mounts: add g_return_if guards
3493 2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3495 * tests/check/gst/client.c:
3496 tests: add more tests
3498 2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3500 * gst/rtsp-server/rtsp-client.c:
3501 client: improve debug
3503 2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3505 * gst/rtsp-server/rtsp-client.c:
3506 client: improve debug and fix leaks
3507 Cleanup the uri and session when there is a bad request.
3509 2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3514 2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3516 * tests/check/gst/client.c:
3517 test: add test for session in options request
3519 2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3521 * gst/rtsp-server/rtsp-client.c:
3522 client: use 454 when session can't be found
3523 We should use 454 when a session can't be found because there was no session
3524 pool configured in the server. This is not a server configuration problem
3525 because the server on which the request is done might not be the same one that
3526 will keep the sessions for us and so it does not need to support sessions.
3528 2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3530 * gst/rtsp-server/rtsp-client.c:
3531 client: only free connection when there is one
3532 It's possible that the client doesn't have a connection when we try to free it.
3534 2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3536 * tests/check/Makefile.am:
3537 * tests/check/gst/client.c:
3538 tests: add unit test for the client object
3540 2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3542 * gst/rtsp-server/rtsp-client.c:
3543 client: small cleanup
3545 2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3547 * gst/rtsp-server/rtsp-client.h:
3548 client: remove unused include
3550 2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3552 * gst/rtsp-server/rtsp-client.c:
3553 client: fix compilation
3555 2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3557 * gst/rtsp-server/rtsp-client.c:
3558 client: call destroy without the lock
3560 2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3562 * gst/rtsp-server/rtsp-client.c:
3563 * gst/rtsp-server/rtsp-client.h:
3564 client: make the client usable without a socket
3565 Make a method to let the client handle a message and a callback when the client
3566 wants us to send a response message back. This makes it possible to also use the
3567 client object without the sockets, which should make it easier to test.
3569 2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3571 * gst/rtsp-server/rtsp-client.c:
3572 * gst/rtsp-server/rtsp-client.h:
3573 client: small cleanup
3575 2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3577 * docs/libs/gst-rtsp-server-sections.txt:
3578 * gst/rtsp-server/rtsp-client.c:
3579 * gst/rtsp-server/rtsp-client.h:
3580 * gst/rtsp-server/rtsp-server.c:
3581 client: remove reference to server
3582 We don't need to keep a ref to the server
3584 2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3586 * gst/rtsp-server/rtsp-client.c:
3587 * gst/rtsp-server/rtsp-client.h:
3589 Also add some g_return_if()
3591 2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3593 * gst/rtsp-server/rtsp-client.c:
3594 client: log more errors
3596 2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3598 * gst/rtsp-server/rtsp-client.c:
3599 client: fix compilation
3601 2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3603 * gst/rtsp-server/rtsp-client.c:
3604 * gst/rtsp-server/rtsp-client.h:
3605 client: add generic close-after-send support
3606 Add a property to send_response() to close the connection after the response has
3607 been sent to the client.
3609 2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3612 * docs/libs/gst-rtsp-server-docs.sgml:
3613 * docs/libs/gst-rtsp-server-sections.txt:
3614 * docs/libs/gst-rtsp-server.types:
3615 * examples/test-auth.c:
3616 * examples/test-launch.c:
3617 * examples/test-mp4.c:
3618 * examples/test-multicast.c:
3619 * examples/test-multicast2.c:
3620 * examples/test-ogg.c:
3621 * examples/test-readme.c:
3622 * examples/test-sdp.c:
3623 * examples/test-uri.c:
3624 * examples/test-video.c:
3625 * gst/rtsp-server/Makefile.am:
3626 * gst/rtsp-server/rtsp-auth.h:
3627 * gst/rtsp-server/rtsp-client.c:
3628 * gst/rtsp-server/rtsp-client.h:
3629 * gst/rtsp-server/rtsp-media-mapping.c:
3630 * gst/rtsp-server/rtsp-media-mapping.h:
3631 * gst/rtsp-server/rtsp-mount-points.c:
3632 * gst/rtsp-server/rtsp-mount-points.h:
3633 * gst/rtsp-server/rtsp-server.c:
3634 * gst/rtsp-server/rtsp-server.h:
3635 * gst/rtsp-server/rtsp-session-media.c:
3636 * gst/rtsp-server/rtsp-session-pool.c:
3637 * gst/rtsp-server/rtsp-session-pool.h:
3638 * tests/check/gst/rtspserver.c:
3639 MediaMapping -> MountPoints
3640 Describes better what the object manages.
3642 2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3645 configure: bump required version of -base
3647 2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3649 * gst/rtsp-server/rtsp-media.c:
3652 2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3654 * gst/rtsp-server/rtsp-media.c:
3655 * gst/rtsp-server/rtsp-media.h:
3656 media: support more Range formats
3657 Use the new -base methods to convert the Range string into a seek start and stop
3660 2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3662 * examples/test-launch.c:
3663 examples: fix whitespace
3665 2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3667 * examples/test-auth.c:
3668 test-auth: add example of how to remove sessions
3669 Add an example of the session filter api.
3671 2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3673 * examples/test-uri.c:
3674 test-uri: remove mapping example
3676 2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3678 * examples/test-uri.c:
3679 test-uri: fix callback signature
3681 2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3683 * gst/rtsp-server/rtsp-media-factory.c:
3684 factory: keep ref to factory while media active
3685 While the media from a factory is alive, keep a ref to the factory.
3686 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
3688 2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3690 * gst/rtsp-server/rtsp-media-factory-uri.c:
3691 factory-uri: add some debug
3693 2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3695 * gst/rtsp-server/rtsp-stream.c:
3696 stream: set udp sources to PLAYING
3697 Set the UDP sources to PLAYING and locked state before we add it to the pipeline
3698 so that it doesn't cause our pipeline to produce ASYNC-DONE.
3700 2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3702 * gst/rtsp-server/rtsp-media-factory-uri.c:
3703 factory-uri: take ref to factory
3704 Take a ref to the factory that we place in our list.
3706 2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3708 * tests/Makefile.am:
3709 * tests/test-reuse.c:
3710 test: add test for server reuse
3711 See https://bugzilla.gnome.org/show_bug.cgi?id=688395
3713 2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
3715 * gst/rtsp-server/rtsp-server.c:
3716 server: start and stop multiple times
3717 Stop listening on the RTSP port when the GSource is removed, so clients
3718 can't connect and the server can be started again.
3719 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
3721 2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3723 * gst/rtsp-server/rtsp-server.c:
3724 server: fix small leak
3726 2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3728 * gst/rtsp-server/rtsp-media.c:
3729 media: unref source in finish_unprepare
3730 The source is created in prepare, unref it in finish_unprepare.
3731 See https://bugzilla.gnome.org/show_bug.cgi?id=688707
3733 2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
3735 * gst/rtsp-server/rtsp-client.c:
3736 * gst/rtsp-server/rtsp-media.c:
3737 rtsp-media: remove bus watch before finalizing
3738 * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
3739 * An extra media ref is added for the bus watch. This extra ref is unreffed by
3740 the GDestroyNotify function.
3741 * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
3742 * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
3743 gst_rtsp_media_unprepare before unreffing the media.
3744 This way, the bus watch will be removed before the media is finalized.
3745 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
3747 2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
3749 * gst/rtsp-server/rtsp-client.c:
3750 * gst/rtsp-server/rtsp-client.h:
3751 client: wait until the TEARDOWN response is sent to close the connection
3752 Responses can be sent async so we need to wait until the TEARDOWN response has
3753 been written before we close the connection to the client. This avoids the risk
3754 of writing/polling closed sockets.
3755 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
3757 2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
3759 * gst/rtsp-server/rtsp-stream.c:
3760 rtsp-stream: plug socket leak
3761 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
3763 2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
3766 Automatic update of common submodule
3767 From 6bb6951 to a72faea
3769 2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
3771 * gst/rtsp-server/rtsp-media-factory-uri.c:
3772 rtsp-server: don't use deprecated API
3774 2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
3776 * gst/rtsp-server/rtsp-client.c:
3777 rtsp-client: fix unused-but-set-variable compiler warning
3778 rtsp-client.c:1260:21: error: variable 'protocols' set but not used
3780 2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3783 * docs/libs/gst-rtsp-server-sections.txt:
3784 * gst/rtsp-server/rtsp-client.c:
3787 2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3789 * examples/Makefile.am:
3790 * examples/test-multicast2.c:
3791 examples: add another multicast example
3792 Add an example for how to configure separate multicast ranges for each media
3795 2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3797 * examples/test-multicast.c:
3800 2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3802 * gst/rtsp-server/rtsp-client.c:
3803 * gst/rtsp-server/rtsp-media.c:
3804 * gst/rtsp-server/rtsp-session-media.c:
3805 * gst/rtsp-server/rtsp-session-media.h:
3806 * gst/rtsp-server/rtsp-stream-transport.c:
3807 * gst/rtsp-server/rtsp-stream-transport.h:
3808 stream: use the address managed by the stream
3809 Use the address managed by the stream for multicast. This allows us to have 1
3810 multicast address for each stream.
3811 Because the address is now managed by the stream we don't have to pass it around
3813 Set the address pool on the streams.
3815 2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3817 * gst/rtsp-server/rtsp-client.c:
3818 * gst/rtsp-server/rtsp-media.c:
3819 * gst/rtsp-server/rtsp-stream.c:
3822 2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3824 * gst/rtsp-server/rtsp-media.c:
3825 * gst/rtsp-server/rtsp-media.h:
3826 media: add signal for new streams
3827 This allows applications to listen for new streams and configure properties on
3828 them, like the address pool.
3830 2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3832 * gst/rtsp-server/rtsp-media.c:
3833 media: configure address pool in new streams
3835 2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3837 * gst/rtsp-server/rtsp-stream.c:
3838 * gst/rtsp-server/rtsp-stream.h:
3839 stream: add methods to deal with address pool
3840 Add methods to get and set the address pool for the stream
3841 Add method to allocate and get the multicast addresses for this stream.
3843 2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3845 * docs/libs/gst-rtsp-server-sections.txt:
3846 * gst/rtsp-server/rtsp-media.c:
3847 * gst/rtsp-server/rtsp-media.h:
3848 media: remove MTU property
3849 It is a stream property
3851 2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3853 * gst/rtsp-server/rtsp-client.c:
3854 client: set blocksize only on stream
3855 Set the blocksize only on the current stream.
3857 2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3859 * gst/rtsp-server/rtsp-stream.c:
3860 stream: share src and sink sockets
3861 the allocated socket is in the used-socket property, not socket.
3863 2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3865 * gst/rtsp-server/rtsp-address-pool.c:
3866 * gst/rtsp-server/rtsp-address-pool.h:
3867 * gst/rtsp-server/rtsp-client.c:
3868 * gst/rtsp-server/rtsp-session-media.c:
3869 * gst/rtsp-server/rtsp-session-media.h:
3870 * gst/rtsp-server/rtsp-stream-transport.c:
3871 * gst/rtsp-server/rtsp-stream-transport.h:
3872 * tests/check/gst/addresspool.c:
3873 rtsp: make address-pool return an address object
3874 Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
3875 store more info in the structure and allows us to more easily return the address
3876 to the right pool when no longer needed.
3877 Pass the address to the StreamTransport so that we can return it to the pool
3878 when the stream transport is freed or changed.
3880 2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3882 * examples/Makefile.am:
3883 * examples/test-multicast.c:
3884 examples: add multicast example
3885 Show how to set up the multicast address pool so that media can be
3886 server with multicast.
3888 2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3890 * gst/rtsp-server/rtsp-client.c:
3891 * gst/rtsp-server/rtsp-media-factory.c:
3892 * gst/rtsp-server/rtsp-media-factory.h:
3893 * gst/rtsp-server/rtsp-media.c:
3894 * gst/rtsp-server/rtsp-media.h:
3895 rtsp: use AddressPool
3896 Remove the multicast_group property.
3897 Use the configured addresspool to allocate multicast addresses.
3899 2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3901 * gst/rtsp-server/rtsp-address-pool.c:
3902 * gst/rtsp-server/rtsp-address-pool.h:
3903 address-pool: add clear method
3905 2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3907 * gst/rtsp-server/rtsp-address-pool.c:
3908 address-pool: small cleanups
3910 2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3912 * tests/check/Makefile.am:
3913 * tests/check/gst/addresspool.c:
3914 tests: add addresspool unit test
3916 2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3918 * gst/rtsp-server/Makefile.am:
3919 * gst/rtsp-server/rtsp-address-pool.c:
3920 * gst/rtsp-server/rtsp-address-pool.h:
3921 address-pool: add object to manage multicast addresses
3922 Make an object that can manage a rage of multicast addresses and ports.
3924 2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3926 * gst/rtsp-server/rtsp-server.c:
3927 server: set default max-threads property
3929 2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3931 * gst/rtsp-server/rtsp-media.c:
3932 media: wait for concurrent _prepare
3933 If a prepare is busy, wait for the result.
3935 2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3937 * gst/rtsp-server/rtsp-media.c:
3938 media: add lock around message handler
3939 We don't want to dispatch messages while we are still processing the result of
3942 2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3944 * gst/rtsp-server/rtsp-media.c:
3945 * gst/rtsp-server/rtsp-media.h:
3946 media: add lock to protect state changes
3948 2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3950 * gst/rtsp-server/rtsp-stream.c:
3951 * gst/rtsp-server/rtsp-stream.h:
3954 2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3956 * gst/rtsp-server/rtsp-stream-transport.c:
3957 * gst/rtsp-server/rtsp-stream-transport.h:
3958 * gst/rtsp-server/rtsp-stream.c:
3959 stream-transport: add keep-alive method
3961 2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3963 * gst/rtsp-server/rtsp-stream-transport.c:
3964 * gst/rtsp-server/rtsp-stream-transport.h:
3965 * gst/rtsp-server/rtsp-stream.c:
3966 stream-transport: add method to handle RTP/RTCP
3967 Call new methods instead of poking into the structures directly.
3969 2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3971 * gst/rtsp-server/rtsp-session-media.c:
3972 * gst/rtsp-server/rtsp-session-media.h:
3973 session-media: add locking
3975 2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3977 * gst/rtsp-server/rtsp-session.c:
3978 * gst/rtsp-server/rtsp-session.h:
3979 session: add locking
3981 2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3983 * gst/rtsp-server/rtsp-server.c:
3984 server: free old socket
3986 2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3988 * gst/rtsp-server/rtsp-media-mapping.c:
3989 * gst/rtsp-server/rtsp-media-mapping.h:
3990 mapping: add locking
3992 2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3994 * gst/rtsp-server/rtsp-media-factory.c:
3995 media-factory: add locking
3997 2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
3999 * gst/rtsp-server/rtsp-auth.c:
4000 * gst/rtsp-server/rtsp-auth.h:
4003 2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4005 * gst/rtsp-server/rtsp-server.c:
4006 * gst/rtsp-server/rtsp-server.h:
4007 server: add max-thread property
4009 2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4011 * gst/rtsp-server/rtsp-server.c:
4012 * gst/rtsp-server/rtsp-server.h:
4013 server: use a threadpool for the mainloops
4015 2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4017 * gst/rtsp-server/rtsp-client.c:
4018 * gst/rtsp-server/rtsp-client.h:
4019 client: rename method
4020 gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
4021 don't really create the client from the socket, we use the socket for the
4024 2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4026 * gst/rtsp-server/rtsp-client.c:
4027 * gst/rtsp-server/rtsp-client.h:
4028 * gst/rtsp-server/rtsp-server.c:
4029 server: rework maincontext handling in clients
4030 Make a separate method to attach a client to a MainContext.
4031 Let the server decide in what GMainContext the client will operate and give this
4032 context to the client in attach. Then the server can later decide to use a
4033 separate thread for each client or just use the mainthread.
4035 2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4037 * gst/rtsp-server/rtsp-client.c:
4038 * gst/rtsp-server/rtsp-session.c:
4039 * gst/rtsp-server/rtsp-session.h:
4040 session: move session header code in session object
4042 2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
4046 * examples/test-auth.c:
4047 * examples/test-launch.c:
4048 * examples/test-mp4.c:
4049 * examples/test-ogg.c:
4050 * examples/test-readme.c:
4051 * examples/test-sdp.c:
4052 * examples/test-uri.c:
4053 * examples/test-video.c:
4054 * gst/rtsp-server/rtsp-auth.c:
4055 * gst/rtsp-server/rtsp-auth.h:
4056 * gst/rtsp-server/rtsp-client.c:
4057 * gst/rtsp-server/rtsp-client.h:
4058 * gst/rtsp-server/rtsp-media-factory-uri.c:
4059 * gst/rtsp-server/rtsp-media-factory-uri.h:
4060 * gst/rtsp-server/rtsp-media-factory.c:
4061 * gst/rtsp-server/rtsp-media-factory.h:
4062 * gst/rtsp-server/rtsp-media-mapping.c:
4063 * gst/rtsp-server/rtsp-media-mapping.h:
4064 * gst/rtsp-server/rtsp-media.c:
4065 * gst/rtsp-server/rtsp-media.h:
4066 * gst/rtsp-server/rtsp-params.c:
4067 * gst/rtsp-server/rtsp-params.h:
4068 * gst/rtsp-server/rtsp-sdp.c:
4069 * gst/rtsp-server/rtsp-sdp.h:
4070 * gst/rtsp-server/rtsp-server.c:
4071 * gst/rtsp-server/rtsp-server.h:
4072 * gst/rtsp-server/rtsp-session-media.c:
4073 * gst/rtsp-server/rtsp-session-media.h:
4074 * gst/rtsp-server/rtsp-session-pool.c:
4075 * gst/rtsp-server/rtsp-session-pool.h:
4076 * gst/rtsp-server/rtsp-session.c:
4077 * gst/rtsp-server/rtsp-session.h:
4078 * gst/rtsp-server/rtsp-stream-transport.c:
4079 * gst/rtsp-server/rtsp-stream-transport.h:
4080 * gst/rtsp-server/rtsp-stream.c:
4081 * gst/rtsp-server/rtsp-stream.h:
4082 * tests/check/gst/rtspserver.c:
4083 * tests/test-cleanup.c:
4086 2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
4088 * gst/rtsp-server/rtsp-media.c:
4089 * gst/rtsp-server/rtsp-session-media.c:
4090 * gst/rtsp-server/rtsp-session.c:
4091 rtsp-server: added annotations to indicate type of ownership transfer of return values
4092 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4094 2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
4097 No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
4099 2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
4102 * bindings/Makefile.am:
4103 * bindings/vala/Makefile.am:
4104 * bindings/vala/gst-rtsp-server-0.10.deps:
4105 * bindings/vala/gst-rtsp-server-0.10.vapi:
4106 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
4107 * bindings/vala/packages/gst-rtsp-server-0.10.files:
4108 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
4109 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4110 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
4112 bindings: remove vala bindings
4113 They'll be reunited with the other GStreamer bindings
4114 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4116 2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4118 * gst/rtsp-server/rtsp-client.c:
4119 * gst/rtsp-server/rtsp-session-media.c:
4120 * gst/rtsp-server/rtsp-session-media.h:
4121 * gst/rtsp-server/rtsp-stream-transport.c:
4122 * gst/rtsp-server/rtsp-stream-transport.h:
4123 rtsp: only create transport when needed
4124 Only create the StreamTransport when configured.
4126 2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4128 * gst/rtsp-server/rtsp-client.c:
4129 client: small cleanup
4131 2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4133 * gst/rtsp-server/rtsp-client.c:
4134 * gst/rtsp-server/rtsp-client.h:
4135 * gst/rtsp-server/rtsp-stream-transport.c:
4136 * gst/rtsp-server/rtsp-stream-transport.h:
4137 rtsp: refactor configuration of transport
4138 Move the configuration of the transport to a place where it makes
4141 2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4143 * gst/rtsp-server/rtsp-client.c:
4144 client: refactor transport parsing
4146 2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4148 * gst/rtsp-server/rtsp-client.c:
4149 client: refuse to change the MTU on shared media
4150 If we change the MTU of chared media, it changes for all clients.
4151 We don't want to set the MTU to something large for clients that
4154 2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4156 * examples/test-mp4.c:
4157 * gst/rtsp-server/rtsp-media.c:
4158 small fixes to docs and debug
4160 2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4162 * gst/rtsp-server/rtsp-stream.c:
4163 stream: transports must already have been removed
4165 2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4167 * gst/rtsp-server/rtsp-media.c:
4168 * gst/rtsp-server/rtsp-stream.c:
4169 * gst/rtsp-server/rtsp-stream.h:
4170 stream: improve join and leave of the pipeline
4172 Do the cleanup properly
4175 2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4177 * gst/rtsp-server/rtsp-media.c:
4178 media: move unprepare below default implementation
4179 Makes it easier to find the default implementation
4181 2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4183 * gst/rtsp-server/rtsp-media.c:
4184 media: signal unprepared when we actually finish
4186 2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4188 * gst/rtsp-server/rtsp-media.c:
4189 media: no need to unlock, unprepare does that when needed
4191 2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4193 * docs/libs/gst-rtsp-server-sections.txt:
4194 * gst/rtsp-server/rtsp-media-factory.h:
4195 * gst/rtsp-server/rtsp-media-mapping.c:
4196 * gst/rtsp-server/rtsp-media.h:
4197 * gst/rtsp-server/rtsp-params.c:
4198 * gst/rtsp-server/rtsp-server.c:
4199 * gst/rtsp-server/rtsp-session-pool.h:
4200 * gst/rtsp-server/rtsp-session.c:
4201 * gst/rtsp-server/rtsp-session.h:
4202 * gst/rtsp-server/rtsp-stream-transport.h:
4203 * gst/rtsp-server/rtsp-stream.h:
4206 2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4208 * gst/rtsp-server/rtsp-client.c:
4209 * gst/rtsp-server/rtsp-media-mapping.h:
4210 * gst/rtsp-server/rtsp-media.c:
4211 * gst/rtsp-server/rtsp-media.h:
4212 * gst/rtsp-server/rtsp-server.h:
4213 * gst/rtsp-server/rtsp-stream.c:
4214 * gst/rtsp-server/rtsp-stream.h:
4215 rtsp: fix MTU setting
4216 Fix setting of the MTU. There is no need for a vmethod.
4218 2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4223 2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4226 configure: bump version number after refactoring
4228 2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4230 * gst/rtsp-server/Makefile.am:
4231 * gst/rtsp-server/rtsp-client.c:
4232 * gst/rtsp-server/rtsp-client.h:
4233 * gst/rtsp-server/rtsp-media-factory-uri.c:
4234 * gst/rtsp-server/rtsp-media-factory.c:
4235 * gst/rtsp-server/rtsp-media-factory.h:
4236 * gst/rtsp-server/rtsp-media.c:
4237 * gst/rtsp-server/rtsp-media.h:
4238 * gst/rtsp-server/rtsp-sdp.c:
4239 * gst/rtsp-server/rtsp-session-media.c:
4240 * gst/rtsp-server/rtsp-session-media.h:
4241 * gst/rtsp-server/rtsp-session.c:
4242 * gst/rtsp-server/rtsp-session.h:
4243 * gst/rtsp-server/rtsp-stream-transport.c:
4244 * gst/rtsp-server/rtsp-stream-transport.h:
4245 * gst/rtsp-server/rtsp-stream.c:
4246 * gst/rtsp-server/rtsp-stream.h:
4247 rtsp: massive refactoring
4248 Make GObjects from the remaining simple structures.
4249 Remove GstRTSPSessionStream, it's not needed.
4250 Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
4251 Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
4252 a GstRTSPStream should be transported to a client.
4253 Rename GstRTSPMediaFactory::get_element -> create_element because that
4254 more accurately describes what it does.
4255 Make nice methods instead of poking in the structures.
4256 Move some methods inside the relevant object source code.
4257 Use GPtrArray to store objects instead of plain arrays, it is more
4258 natural and allows us to more easily clean up.
4259 Move the allocation of udp ports to the Stream object. The Stream object
4260 contains the elements needed to stream the media to a client.
4261 Improve the prepare and unprepare methods. Unprepare should now undo
4262 everything prepare did. Improve also async unprepare when doing EOS on
4263 shutdown. Make sure we always unprepare correctly.
4265 2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
4267 * gst/rtsp-server/rtsp-client.c:
4268 rtsp-client: Unref server address clients connected to
4269 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
4271 2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
4273 * gst/rtsp-server/rtsp-server.c:
4274 rtsp-server: don't ref server socket if it is NULL
4275 Fixes test_bind_already_in_use unit test again after commit 6a497440.
4276 https://bugzilla.gnome.org/show_bug.cgi?id=686644
4278 2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
4280 * tests/check/Makefile.am:
4281 tests: Add libgio link dependency
4282 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
4284 2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
4286 * gst/rtsp-server/rtsp-media-mapping.c:
4287 * gst/rtsp-server/rtsp-media-mapping.h:
4288 rtsp-media-mapping: rename find_media vfunc to find_factory
4289 The virtual method and class method should have the same name
4290 so it is correctly represented in GIR file
4291 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4293 2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
4295 * gst/rtsp-server/rtsp-auth.c:
4296 * gst/rtsp-server/rtsp-client.c:
4297 * gst/rtsp-server/rtsp-media-factory-uri.c:
4298 * gst/rtsp-server/rtsp-media-factory.c:
4299 * gst/rtsp-server/rtsp-media-mapping.c:
4300 * gst/rtsp-server/rtsp-media.c:
4301 * gst/rtsp-server/rtsp-server.c:
4302 * gst/rtsp-server/rtsp-session-pool.c:
4303 * gst/rtsp-server/rtsp-session.c:
4304 rtsp-server: fixed comments and GIR annotations
4305 https://bugzilla.gnome.org/show_bug.cgi?id=680777
4307 2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
4309 * gst/rtsp-server/rtsp-media-mapping.c:
4310 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
4312 2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
4314 * gst/rtsp-server/rtsp-server.c:
4315 rtsp-server: allow binding on port 0 (binds on a random port)
4317 2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
4319 * gst/rtsp-server/rtsp-server.c:
4320 * gst/rtsp-server/rtsp-server.h:
4321 rtsp-server: add bound-port property
4322 bound-port can be used to retrieve the port number when the server is bound on
4323 port 0, which binds on a random port.
4325 2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
4327 * gst/rtsp-server/rtsp-media-factory.c:
4328 * gst/rtsp-server/rtsp-media-factory.h:
4329 rtsp-media-factory: make ::get_element overridable by GI bindings
4330 The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
4331 for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
4332 as the invoker for ::get_element(), making it overridable by GI generated
4335 2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4337 * gst/rtsp-server/rtsp-media-factory-uri.c:
4338 rtsp-media-factory-uri: don't autoplug parsers in a loop
4339 Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
4342 2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
4344 * gst/rtsp-server/Makefile.am:
4345 Explicitly link against gio. Fix link error on mac.
4347 2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
4349 * gst/rtsp-server/rtsp-session.c:
4350 session: add ttl to the transport header in SETUP
4351 See https://bugzilla.gnome.org/show_bug.cgi?id=685561
4353 2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
4355 * gst/rtsp-server/rtsp-client.c:
4356 * gst/rtsp-server/rtsp-client.h:
4357 * gst/rtsp-server/rtsp-media.c:
4358 client: Use client transport settings for multicast if allowed.
4359 This patch makes it possible for the client to send transport settings for
4360 multicast (destination && ttl). Client settings must be explicitly allowed or
4361 the server will use its own settings.
4362 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
4364 2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
4367 Automatic update of common submodule
4368 From 6c0b52c to 6bb6951
4370 2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
4372 * gst/rtsp-server/rtsp-client.c:
4373 rtsp-client: do not destroy the rtsp watch
4374 Don't destroy the client watch while dispatching. The rtsp watch is
4375 automatically destroyed after the rtsp watch function closed() has
4377 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
4379 2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
4382 Automatic update of common submodule
4383 From 4f962f7 to 6c0b52c
4385 2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
4387 * gst/rtsp-server/rtsp-media.c:
4388 media: fix check for seekability
4390 2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4392 * gst/rtsp-server/rtsp-client.c:
4393 client: use more GIO
4394 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
4396 2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4398 * gst/rtsp-server/rtsp-server.c:
4399 server: remove obsolete includes
4401 2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4403 rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
4404 * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
4405 be available in "on_new_ssrc". The transports are added in
4406 gst_rtsp_media_set_state when going to PLAYING state. However,
4407 "on_new_ssrc" might be called before this happens.
4408 https://bugzilla.gnome.org/show_bug.cgi?id=683304
4410 2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4412 * gst/rtsp-server/rtsp-client.c:
4413 * gst/rtsp-server/rtsp-client.h:
4414 rtsp-client: add signals for rtsp requests (fixes #683287)
4416 2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
4418 * gst/rtsp-server/rtsp-client.c:
4419 * gst/rtsp-server/rtsp-client.h:
4420 add new-session signal to rtsp-client (fixes #683058)
4422 2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
4425 Automatic update of common submodule
4426 From 668acee to 4f962f7
4428 2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
4430 * gst/rtsp-server/rtsp-server.c:
4431 * tests/check/gst/rtspserver.c:
4432 rtsp-server: fixed segfault in gst_rtsp_server_create_socket
4433 Do not assume that *error is set in g_socket_address_enumerator_next.
4434 Added test_bind_already_in_use unit-test.
4435 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
4437 2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
4440 Automatic update of common submodule
4441 From 94ccf4c to 668acee
4443 2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
4445 * gst/rtsp-server/rtsp-client.c:
4446 * gst/rtsp-server/rtsp-client.h:
4447 rtsp-client: make create_sdp virtual method
4448 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
4450 2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4453 Automatic update of common submodule
4454 From 98e386f to 94ccf4c
4456 2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4458 * gst/rtsp-server/rtsp-client.c:
4461 2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
4463 * gst/rtsp-server/rtsp-client.c:
4464 * gst/rtsp-server/rtsp-client.h:
4465 * gst/rtsp-server/rtsp-server.c:
4466 * gst/rtsp-server/rtsp-server.h:
4467 rtsp-server: use an existing socket to establish HTTP tunnel
4468 Make it possible to transfer a socket from an HTTP server to be used as
4469 an RTSP over HTTP tunnel.
4471 2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
4473 * gst/rtsp-server/rtsp-client.c:
4474 * gst/rtsp-server/rtsp-media.c:
4475 * gst/rtsp-server/rtsp-media.h:
4476 rtsp: Handle the blocksize parameter
4477 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
4479 2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
4481 * tests/check/Makefile.am:
4482 * tests/check/gst/rtspserver.c:
4483 Have unit test get header from source dir, not installed dir
4484 This makes compilation of unit tests work in a build directory other
4485 than the source directory.
4486 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
4488 2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
4490 * gst/rtsp-server/rtsp-media.c:
4491 rtsp-media: update for gst_element_make_from_uri() changes
4493 2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
4496 * tests/Makefile.am:
4497 * tests/check/Makefile.am:
4498 * tests/check/gst/rtspserver.c:
4500 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
4502 2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
4504 * gst/rtsp-server/rtsp-media.c:
4505 rtsp-media: don't collect media stats when going to NULL
4506 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
4508 2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4510 * gst/rtsp-server/rtsp-client.c:
4511 client: don't leak transports
4513 2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
4515 * gst/rtsp-server/rtsp-client.c:
4516 rtsp-client: free transport on no_stream in SETUP handler
4518 2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
4520 * gst/rtsp-server/rtsp-client.c:
4521 rtsp-client: changed session media iteration
4522 In client_unlink_session: now don't iterate in session->medias
4523 list where items are removed by gst_rtsp_session_release_media.
4524 Instead, repeatedly remove the first item.
4526 2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
4528 * gst/rtsp-server/rtsp-client.c:
4529 rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
4530 GstRTSPSessionMedia is not a GObject type. When the
4531 GstRTSPSession is freed, it will free the media.
4533 2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
4535 * gst/rtsp-server/rtsp-media-factory.c:
4536 factory: plug pad leak in collect_streams
4537 In gst_rtsp_media_factory_collect_streams: unref the srcpad that
4538 was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
4539 will take one reference, and the other reference will otherwise
4542 2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
4545 configure: suppress some warnings when debug is disabled
4546 Warnings about unused variables should be suppressed if core has the
4547 debug system disabled.
4548 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4550 2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4552 * docs/libs/Makefile.am:
4553 docs: fix build in uninstalled setup
4554 Include gst-plugins-base libs properly.
4556 2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
4558 * docs/libs/gst-rtsp-server.types:
4559 docs: include headers defining rtsp-server object types
4560 Fixes compiler warnings during docs build.
4561 https://bugzilla.gnome.org/show_bug.cgi?id=676824
4563 2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
4566 configure: Add warning flags for compiler when configuring
4567 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
4569 2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4572 Automatic update of common submodule
4573 From 03a0e57 to 98e386f
4575 2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4578 Automatic update of common submodule
4579 From 1fab359 to 03a0e57
4581 2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
4583 * gst/rtsp-server/rtsp-client.c:
4584 client: fix GSocketAddress leak in gst_rtsp_client_accept
4585 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
4587 2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
4590 Automatic update of common submodule
4591 From f1b5a96 to 1fab359
4593 2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4596 Automatic update of common submodule
4597 From 92b7266 to f1b5a96
4599 2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4602 Automatic update of common submodule
4603 From ec1c4a8 to 92b7266
4605 2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4608 Automatic update of common submodule
4609 From 3429ba6 to ec1c4a8
4611 2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
4613 * gst/rtsp-server/rtsp-auth.c:
4614 * gst/rtsp-server/rtsp-client.c:
4615 * gst/rtsp-server/rtsp-media-factory-uri.c:
4616 * gst/rtsp-server/rtsp-server.c:
4617 rtsp: fix compiler warnings
4618 Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
4620 2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4623 Automatic update of common submodule
4624 From dc70203 to 3429ba6
4626 2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4628 * gst/rtsp-server/rtsp-client.c:
4629 * gst/rtsp-server/rtsp-media-factory.c:
4630 * gst/rtsp-server/rtsp-media-factory.h:
4631 * gst/rtsp-server/rtsp-media.c:
4632 * gst/rtsp-server/rtsp-media.h:
4633 * gst/rtsp-server/rtsp-server.c:
4634 * gst/rtsp-server/rtsp-server.h:
4635 * gst/rtsp-server/rtsp-session-pool.c:
4636 * gst/rtsp-server/rtsp-session-pool.h:
4637 rtsp-server: port to new thread API
4639 2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4642 Automatic update of common submodule
4643 From 6db25be to dc70203
4645 2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4647 * gst/rtsp-server/rtsp-auth.c:
4648 * gst/rtsp-server/rtsp-auth.h:
4649 * gst/rtsp-server/rtsp-client.c:
4650 rtsp-server: Fix compilation and compiler warnings
4652 2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4656 * gst/rtsp-server/Makefile.am:
4657 configure: Modernize autotools setup a bit
4658 Also we now only create tar.bz2 and tar.xz tarballs.
4660 2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4663 Automatic update of common submodule
4664 From 464fe15 to 6db25be
4666 2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4669 Automatic update of common submodule
4670 From 7fda524 to 464fe15
4672 2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4675 * docs/libs/Makefile.am:
4676 * docs/version.entities.in:
4678 * gst/rtsp-server/Makefile.am:
4679 * pkgconfig/Makefile.am:
4680 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4681 * pkgconfig/gstreamer-rtsp-server.pc.in:
4682 * tests/Makefile.am:
4683 rtsp-server: Update versioning
4685 2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4687 Merge remote-tracking branch 'origin/0.10'
4689 gst/rtsp-server/rtsp-session-pool.c
4691 2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4693 * gst/rtsp-server/rtsp-session-pool.c:
4694 rtsp-server: Don't use deprecated GLib API
4696 2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4698 Replace master with 0.11
4700 2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4702 Merge branch 'master' into 0.11
4704 2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4706 Merge branch 'master' into 0.11
4708 2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
4711 A couple minor typo fixes
4713 2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4715 * gst/rtsp-server/rtsp-media.c:
4716 media: fix state of the appqueue
4718 2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4720 * gst/rtsp-server/rtsp-media-factory-uri.c:
4721 factory: use videoconvert
4723 2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4725 * gst/rtsp-server/rtsp-media-factory-uri.c:
4726 factory: change to new style caps
4728 2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4730 * gst/rtsp-server/rtsp-client.c:
4731 * gst/rtsp-server/rtsp-client.h:
4732 * gst/rtsp-server/rtsp-media-factory-uri.c:
4733 * gst/rtsp-server/rtsp-media.c:
4734 * gst/rtsp-server/rtsp-server.c:
4735 * gst/rtsp-server/rtsp-server.h:
4736 * gst/rtsp-server/rtsp-session-pool.c:
4737 rtsp-server: port to GIO
4740 2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4743 configure: fix build
4745 2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4748 docs: fix for gst_rtsp_server_set_port() -> _set_service()
4749 https://bugzilla.gnome.org/show_bug.cgi?id=666548
4751 2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4754 * examples/Makefile.am:
4755 First rule of gst-rtsp-server club: don't talk about gst-phonon
4757 2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4760 * pkgconfig/Makefile.am:
4761 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
4762 * pkgconfig/gst-rtsp-server.pc.in:
4763 * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
4764 * pkgconfig/gstreamer-rtsp-server.pc.in:
4765 pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
4766 For consistency with all other modules.
4768 2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4770 * gst/rtsp-server/rtsp-client.c:
4771 rtsp-client: update for new map API
4773 2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4776 * bindings/Makefile.am:
4777 * bindings/python/Makefile.am:
4778 * bindings/python/arg-types.py:
4779 * bindings/python/codegen/Makefile.am:
4780 * bindings/python/codegen/__init__.py:
4781 * bindings/python/codegen/argtypes.py:
4782 * bindings/python/codegen/code-coverage.py:
4783 * bindings/python/codegen/codegen.py:
4784 * bindings/python/codegen/definitions.py:
4785 * bindings/python/codegen/defsparser.py:
4786 * bindings/python/codegen/docextract.py:
4787 * bindings/python/codegen/docgen.py:
4788 * bindings/python/codegen/fileprefix.override:
4789 * bindings/python/codegen/fileprefixmodule.c:
4790 * bindings/python/codegen/h2def.py:
4791 * bindings/python/codegen/mergedefs.py:
4792 * bindings/python/codegen/mkskel.py:
4793 * bindings/python/codegen/override.py:
4794 * bindings/python/codegen/reversewrapper.py:
4795 * bindings/python/codegen/scmexpr.py:
4796 * bindings/python/rtspserver-types.defs:
4797 * bindings/python/rtspserver.defs:
4798 * bindings/python/rtspserver.override:
4799 * bindings/python/rtspservermodule.c:
4800 * bindings/python/test.py:
4802 python: remove pygst-based python bindings
4803 pygi is the future, apparently.
4805 2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
4808 Automatic update of common submodule
4809 From c463bc0 to 7fda524
4811 2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4814 Automatic update of common submodule
4815 From 2a59016 to c463bc0
4817 2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4820 Automatic update of common submodule
4821 From 0807187 to 2a59016
4823 2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4826 Automatic update of common submodule
4827 From 11f0cd5 to 0807187
4829 2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4831 * examples/test-auth.c:
4832 example: update for new caps
4834 2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4836 * examples/test-video.c:
4837 * gst/rtsp-server/rtsp-client.c:
4838 * gst/rtsp-server/rtsp-media-factory-uri.c:
4839 * gst/rtsp-server/rtsp-media.c:
4840 * gst/rtsp-server/rtsp-media.h:
4841 * gst/rtsp-server/rtsp-session.c:
4842 * gst/rtsp-server/rtsp-session.h:
4843 rtsp-server: port some more to 0.11
4845 Remove bufferlist stuff
4847 Add queue before appsink now that preroll-queue-len is gone.
4848 Update for request pad changes.
4850 2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4852 Merge branch 'master' into 0.11
4854 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4856 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4857 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4858 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4860 2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
4862 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
4863 bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
4864 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
4866 2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4868 Merge branch 'master' into 0.11
4870 2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4872 * gst/rtsp-server/rtsp-media.c:
4873 * gst/rtsp-server/rtsp-media.h:
4874 media: add a seekable boolean
4875 Maintain the seekable state with a new variable instead of reusing the
4878 2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
4880 * gst/rtsp-server/rtsp-media.c:
4881 Disallow seek in live media
4883 2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
4885 Merge branch 'master' into 0.11
4887 2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
4889 * gst/rtsp-server/rtsp-server.c:
4890 #ifdef statements for windows socket creation were missing
4892 2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
4895 Automatic update of common submodule
4896 From a39eb83 to 11f0cd5
4898 2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
4901 Automatic update of common submodule
4902 From 605cd9a to a39eb83
4904 2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4906 Merge branch 'master' into 0.11
4908 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4910 * gst/rtsp-server/rtsp-client.c:
4911 client: use method to access property
4913 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4915 * gst/rtsp-server/rtsp-media-factory.c:
4916 * gst/rtsp-server/rtsp-media-factory.h:
4917 media-factory: add protocols property
4918 Add a property to configure the allowed protocols in the media created from the
4921 2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4923 * gst/rtsp-server/rtsp-media-factory.c:
4924 * gst/rtsp-server/rtsp-media-factory.h:
4925 media-factory: add media-configure signal
4926 Add signal to allow the application to configure the media after it was created
4929 2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4931 * gst/rtsp-server/rtsp-client.c:
4932 client: use method to access property
4934 2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4936 * gst/rtsp-server/rtsp-media-factory.c:
4937 * gst/rtsp-server/rtsp-media-factory.h:
4938 media-factory: add protocols property
4939 Add a property to configure the allowed protocols in the media created from the
4942 2011-08-16 15:03:06 +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 media-configure signal
4947 Add signal to allow the application to configure the media after it was created
4950 2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4952 Merge branch 'master' into 0.11
4954 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4956 * gst/rtsp-server/rtsp-client.c:
4957 client: use media multicast group
4959 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4961 * gst/rtsp-server/rtsp-media-factory.h:
4962 * gst/rtsp-server/rtsp-server.h:
4963 * gst/rtsp-server/rtsp-session-pool.h:
4964 * gst/rtsp-server/rtsp-session.h:
4967 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
4969 * gst/rtsp-server/rtsp-client.c:
4970 * gst/rtsp-server/rtsp-sdp.h:
4971 sdp: copy and free the server ip address
4972 Copy and free the server ip address to make memory management easier later.
4974 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4976 * gst/rtsp-server/rtsp-media-factory.c:
4977 media-factory: configure multicast in media
4979 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4981 * gst/rtsp-server/rtsp-media.c:
4982 * gst/rtsp-server/rtsp-media.h:
4983 media: add property for multicast group
4984 Add a property to configure the multicast group in the media.
4985 Based on patches from Marc Leeman and Robert Krakora.
4987 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4989 * gst/rtsp-server/rtsp-media-factory.c:
4990 * gst/rtsp-server/rtsp-media-factory.h:
4991 media-factory: add property for multicast group
4992 Add a property to configure the multicast group in the media factory.
4993 Based on patches from Marc Leeman and Robert Krakora.
4995 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4997 * gst/rtsp-server/rtsp-client.c:
4998 client: do configuration of transport in one place
4999 Move the configuration of the transport destination address to where we also
5000 configure the other bits.
5002 2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5004 * gst/rtsp-server/rtsp-client.c:
5005 client: use media multicast group
5007 2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5009 * gst/rtsp-server/rtsp-media-factory.h:
5010 * gst/rtsp-server/rtsp-server.h:
5011 * gst/rtsp-server/rtsp-session-pool.h:
5012 * gst/rtsp-server/rtsp-session.h:
5015 2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
5017 * gst/rtsp-server/rtsp-client.c:
5018 * gst/rtsp-server/rtsp-sdp.h:
5019 sdp: copy and free the server ip address
5020 Copy and free the server ip address to make memory management easier later.
5022 2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5024 * gst/rtsp-server/rtsp-media-factory.c:
5025 media-factory: configure multicast in media
5027 2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5029 * gst/rtsp-server/rtsp-media.c:
5030 * gst/rtsp-server/rtsp-media.h:
5031 media: add property for multicast group
5032 Add a property to configure the multicast group in the media.
5033 Based on patches from Marc Leeman and Robert Krakora.
5035 2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5037 * gst/rtsp-server/rtsp-media-factory.c:
5038 * gst/rtsp-server/rtsp-media-factory.h:
5039 media-factory: add property for multicast group
5040 Add a property to configure the multicast group in the media factory.
5041 Based on patches from Marc Leeman and Robert Krakora.
5043 2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5045 * gst/rtsp-server/rtsp-client.c:
5046 client: do configuration of transport in one place
5047 Move the configuration of the transport destination address to where we also
5048 configure the other bits.
5050 2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5052 Merge branch 'master' into 0.11
5054 2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
5056 * gst/rtsp-server/rtsp-client.c:
5057 client: destroy pipeline on client disconnect with no prior TEARDOWN.
5058 The problem occurs when the client abruptly closes the connection without
5059 issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
5060 server is where the pipeline gets torn down. Since this handler is not called,
5061 the pipeline remains and is up and running. Subsequent clients get their own
5062 pipelines and if the do not issue TEARDOWNs then those pipelines will also
5063 remain up and running. This is a resource leak.
5065 2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5067 Merge branch 'master' into 0.11
5069 2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
5071 * gst/rtsp-server/rtsp-media-factory.c:
5072 * gst/rtsp-server/rtsp-media-factory.h:
5073 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
5074 For example, it can be used to retrieve source elements like appsrc, in a more
5075 convenient way than subclassing get_element.
5077 2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5079 Merge branch 'master' into 0.11
5081 2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
5083 * gst/rtsp-server/rtsp-server.c:
5084 rtsp-server: hold on to reference while using object
5086 2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5088 * gst/rtsp-server/rtsp-media.c:
5091 2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5094 configure: use unstable api
5096 2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
5098 * gst/rtsp-server/rtsp-client.c:
5099 client: fix reference counting
5101 2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
5103 * gst/rtsp-server/rtsp-client.c:
5104 * gst/rtsp-server/rtsp-media.c:
5105 fix compiler warnings about unused variables
5107 2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
5109 * examples/test-launch.c:
5110 * examples/test-readme.c:
5111 * examples/test-uri.c:
5112 * examples/test-video.c:
5113 examples: tell rtsp uri when ready
5115 2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
5118 Automatic update of common submodule
5119 From 69b981f to 605cd9a
5121 2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5123 * gst/rtsp-server/rtsp-client.c:
5124 client: update for buffer API change
5126 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5128 * gst/rtsp-server/Makefile.am:
5129 Makefile.am: 0.10 => @GST_MAJORMINOR@
5131 2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5133 * gst/rtsp-server/rtsp-media-factory-uri.c:
5134 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
5136 2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5138 * gst/rtsp-server/.gitignore:
5139 .gitignore: 0.10 => 0.11
5141 2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
5143 * gst/rtsp-server/Makefile.am:
5144 Makefile.am: 0.10 => @GST_MAJORMINOR@
5146 2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5148 Merge branch 'master' into 0.11
5150 2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
5153 Automatic update of common submodule
5154 From 9e5bbd5 to 69b981f
5156 2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
5159 Automatic update of common submodule
5160 From fd35073 to 9e5bbd5
5162 2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
5165 Automatic update of common submodule
5166 From 46dfcea to fd35073
5168 2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5170 * gst/rtsp-server/rtsp-media-factory-uri.c:
5171 * gst/rtsp-server/rtsp-media.c:
5172 media: port to new caps API
5174 2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5176 Merge branch 'master' into 0.11
5178 2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
5180 * bindings/vala/gst-rtsp-server-0.10.vapi:
5181 Updated Vala bindings.
5182 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
5184 2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
5186 * gst/rtsp-server/rtsp-server.c:
5187 * gst/rtsp-server/rtsp-server.h:
5188 Add a signal for newly connected clients.
5189 Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
5191 2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
5193 * bindings/python/rtspserver.override:
5194 python: override gst_rtsp_media_mapping_add_factory to fix refcounting
5196 2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5198 * gst/rtsp-server/Makefile.am:
5199 * gst/rtsp-server/rtsp-client.c:
5200 * gst/rtsp-server/rtsp-funnel.c:
5201 * gst/rtsp-server/rtsp-funnel.h:
5202 * gst/rtsp-server/rtsp-media.c:
5203 rtsp-server: port to 0.11
5205 2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5210 2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5212 Merge branch 'master' into 0.11
5217 2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5220 Automatic update of common submodule
5221 From c3cafe1 to 46dfcea
5223 2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
5225 * bindings/python/Makefile.am:
5226 * bindings/python/rtspserver.defs:
5227 python bindings: wrap GstRTSPMediaFactoryClass vfuncs
5229 2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
5231 * bindings/python/arg-types.py:
5232 python bindings: add GstRTSPUrlParam
5233 Needed to implement MediaFactory virtual proxies
5235 2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
5237 * bindings/python/arg-types.py:
5238 python bindings: fix returning GstRTSPUrl types
5240 2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
5242 * bindings/python/arg-types.py:
5243 python bindings: add arg type for GstRTSPUrl
5245 2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
5247 * bindings/python/rtspserver.defs:
5248 python bindings: fix the definition of MediaFactory.collect_stream
5250 2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
5253 Automatic update of common submodule
5254 From 1ccbe09 to c3cafe1
5256 2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5259 Automatic update of common submodule
5260 From 193b717 to 1ccbe09
5262 2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
5265 Automatic update of common submodule
5266 From b77e2bf to 193b717
5268 2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5271 build: Include lcov.mak to allow test coverage report generation
5273 2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5276 Automatic update of common submodule
5277 From d8814b6 to b77e2bf
5279 2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5282 Automatic update of common submodule
5283 From 6aaa286 to d8814b6
5285 2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
5288 Automatic update of common submodule
5289 From 6aec6b9 to 6aaa286
5291 2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
5294 autogen: wingo signed comment
5296 2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
5298 * gst/rtsp-server/rtsp-session-pool.c:
5299 session: use full charset for RTSP session ID
5300 As specified in RFC 2326 section 3.4 use full valid charset to make guessing
5301 session ID more difficult.
5302 https://bugzilla.gnome.org/show_bug.cgi?id=643812
5304 2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5306 * gst/rtsp-server/Makefile.am:
5307 rtsp-server: Don't install the funnel header
5309 2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
5312 Automatic update of common submodule
5313 From 1de7f6a to 6aec6b9
5315 2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5318 configure: require core/base 0.10.31
5319 Needed at least for gst_plugin_feature_rank_compare_func().
5321 2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
5324 Automatic update of common submodule
5325 From f94d739 to 1de7f6a
5327 2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5329 * gst/rtsp-server/rtsp-media.c:
5330 media: remove more unused code
5332 2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5334 * gst/rtsp-server/rtsp-media.c:
5335 * gst/rtsp-server/rtsp-media.h:
5336 media: remove duplicate filtering
5337 Remove the duplicate filtering code now that we have a released -good version.
5338 Give a warning instead.
5340 2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5342 * gst/rtsp-server/rtsp-media-factory.c:
5343 * gst/rtsp-server/rtsp-media.c:
5344 media: fix default buffer size
5346 2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5348 * gst/rtsp-server/rtsp-media-factory.c:
5349 * gst/rtsp-server/rtsp-media-factory.h:
5350 media-factory: add property to configure the buffer-size
5351 Add a property to configure the kernel UDP buffer size.
5353 2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5355 * gst/rtsp-server/rtsp-media.c:
5356 * gst/rtsp-server/rtsp-media.h:
5357 media: add property to configure kernel buffer sizes
5358 Add a property to configure the kernel UDP buffer size.
5360 2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5363 configure: set PYGOBJECT_REQ before using it
5364 https://bugzilla.gnome.org/show_bug.cgi?id=640641
5366 2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5369 docs: recursive into sub-directories on 'make upload'
5371 2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5373 * docs/libs/gst-rtsp-server-docs.sgml:
5374 * docs/version.entities.in:
5375 docs: mention full version these docs are for, not just major-minor
5377 2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5382 === release 0.10.8 ===
5384 2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5389 2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5391 * gst/rtsp-server/rtsp-server.c:
5392 rtsp-server: clarify docs a little
5394 2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5396 * gst/rtsp-server/rtsp-media.c:
5397 media: init debug category before starting thread
5399 2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5401 * gst/rtsp-server/rtsp-auth.c:
5402 auth: add realm to make it more spec compliant
5404 2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5406 * gst/rtsp-server/rtsp-server.c:
5407 * gst/rtsp-server/rtsp-server.h:
5410 2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5412 * examples/test-video.c:
5413 example: improve example docs a little
5415 2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5417 * gst/rtsp-server/rtsp-server.c:
5418 server: ensure the watch has a ref to the server
5420 2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5422 * gst/rtsp-server/rtsp-server.c:
5423 server: simpify channel function
5425 2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5427 * gst/rtsp-server/rtsp-server.c:
5428 * gst/rtsp-server/rtsp-server.h:
5429 server: simplify management of channel and source
5430 We don't need to keep around the channel and source objects. Let the mainloop
5431 and the source manage the source and channel respectively.
5433 2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5439 2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5442 * tests/Makefile.am:
5443 * tests/test-cleanup.c:
5444 tests: add tests directory and cleanup test
5446 2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5448 * gst/rtsp-server/rtsp-media-factory-uri.c:
5449 * gst/rtsp-server/rtsp-media-factory.c:
5450 * gst/rtsp-server/rtsp-media-mapping.c:
5451 * gst/rtsp-server/rtsp-media.c:
5452 * gst/rtsp-server/rtsp-session-pool.c:
5453 * gst/rtsp-server/rtsp-session.c:
5454 server: improve debugging in various objects
5456 2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5458 * gst/rtsp-server/rtsp-server.c:
5459 server: chain up to the parent finalize
5461 2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
5463 * bindings/python/rtspserver-types.defs:
5464 * bindings/python/rtspserver.defs:
5465 * bindings/python/rtspserver.override:
5466 * bindings/python/test.py:
5467 gst-rtsp-server: update python bindings
5469 2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5471 * gst/rtsp-server/rtsp-client.c:
5472 client: use the response from the clientstate
5473 Create the response object only once and store in the client state.
5474 Make all methods use the state response,
5476 2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5478 * gst/rtsp-server/rtsp-server.c:
5479 server: use signal to keep track of clients
5480 Keep track of all the clients that the server creates and remove them when they
5481 fire the 'closed' signal.
5483 2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5485 * gst/rtsp-server/rtsp-client.c:
5486 * gst/rtsp-server/rtsp-client.h:
5487 client: emit signal when closing
5489 2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5491 * examples/.gitignore:
5492 * examples/Makefile.am:
5493 * examples/test-auth.c:
5494 * examples/test-video.c:
5495 * gst/rtsp-server/rtsp-auth.c:
5496 * gst/rtsp-server/rtsp-auth.h:
5497 * gst/rtsp-server/rtsp-client.c:
5498 * gst/rtsp-server/rtsp-media-factory.c:
5499 * gst/rtsp-server/rtsp-media.c:
5500 * gst/rtsp-server/rtsp-media.h:
5501 * gst/rtsp-server/rtsp-session-pool.h:
5502 * gst/rtsp-server/rtsp-session.h:
5503 media: enable per factory authorisations
5504 Allow for adding a GstRTSPAuth on the factory and media level and check
5505 permissions when accessing the factory.
5506 Add hints to the auth methods for future more fine grained authorisation.
5507 Add example application for per factory authentication.
5509 2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5511 * gst/rtsp-server/rtsp-auth.c:
5512 * gst/rtsp-server/rtsp-auth.h:
5513 * gst/rtsp-server/rtsp-client.c:
5514 * gst/rtsp-server/rtsp-client.h:
5515 * gst/rtsp-server/rtsp-params.c:
5516 * gst/rtsp-server/rtsp-params.h:
5517 rtsp-server: Pass ClientState structure arround
5518 Pass the collected information for the ongoing request in a GstRTSPClientState
5519 structure that we can then pass around to simplify the method arguments. This
5520 will also be handy when we implement logging functionality.
5522 2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5524 * gst/rtsp-server/rtsp-media-factory.c:
5525 * gst/rtsp-server/rtsp-media-factory.h:
5526 media-factory: add methods to configure authorisation
5528 2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5530 * gst/rtsp-server/rtsp-client.c:
5531 client: unref auth in finalize
5533 2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5535 * gst/rtsp-server/rtsp-server.c:
5536 server: unref auth in finalize
5538 2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5540 * docs/libs/gst-rtsp-server-docs.sgml:
5541 * docs/libs/gst-rtsp-server-sections.txt:
5542 * docs/libs/gst-rtsp-server.types:
5545 2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5547 * gst/rtsp-server/rtsp-server.c:
5548 * gst/rtsp-server/rtsp-server.h:
5549 server: separate create and accept
5550 Create separate create and accept methods so that subclasses can create custom
5552 Configure the server in the client object and prepare for keeping track of
5555 2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5557 * gst/rtsp-server/rtsp-client.c:
5558 * gst/rtsp-server/rtsp-client.h:
5559 client: add support for setting the server.
5560 Add support for keeping a ref to the server that started this client
5563 2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5565 * gst/rtsp-server/rtsp-auth.c:
5566 auth: fix memleak and add some docs
5567 Fix a memleak of the basic auth token.
5568 Add docs for the helper function
5570 2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5572 * gst/rtsp-server/rtsp-auth.c:
5573 * gst/rtsp-server/rtsp-auth.h:
5574 * gst/rtsp-server/rtsp-client.c:
5575 client: delegate setup of auth to the manager
5576 Delegate the configuration of the authentication tokens to the manager object
5579 2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5581 * examples/test-video.c:
5582 * gst/rtsp-server/Makefile.am:
5583 * gst/rtsp-server/rtsp-auth.c:
5584 * gst/rtsp-server/rtsp-auth.h:
5585 * gst/rtsp-server/rtsp-client.c:
5586 * gst/rtsp-server/rtsp-client.h:
5587 * gst/rtsp-server/rtsp-server.c:
5588 * gst/rtsp-server/rtsp-server.h:
5589 auth: add authentication object
5590 Add an object that can check the authorization of requests.
5591 Implement basic authentication.
5592 Add example authentication to test-video
5594 2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5596 * gst/rtsp-server/rtsp-server.c:
5597 * gst/rtsp-server/rtsp-server.h:
5598 server: move includes back
5599 the includes are needed for sockaddr_in.
5601 2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5603 * gst/rtsp-server/rtsp-client.c:
5604 * gst/rtsp-server/rtsp-client.h:
5605 * gst/rtsp-server/rtsp-server.c:
5606 * gst/rtsp-server/rtsp-server.h:
5607 rtsp: move network includes where they are needed
5609 2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
5611 * gst/rtsp-server/rtsp-media.h:
5612 rtsp-media.h: Minor corrections in comments.
5615 2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
5618 Automatic update of common submodule
5619 From e572c87 to f94d739
5621 2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5625 * docs/libs/.gitignore:
5626 * examples/.gitignore:
5627 * gst/rtsp-server/.gitignore:
5630 2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5632 * docs/libs/Makefile.am:
5633 docs: We don't build ps/pdf for API reference docs
5635 2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5638 Automatic update of common submodule
5639 From ccbaa85 to e572c87
5641 2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5644 Automatic update of common submodule
5645 From 46445ad to ccbaa85
5647 2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5649 * gst/rtsp-server/Makefile.am:
5650 * gst/rtsp-server/fs-funnel.c:
5651 * gst/rtsp-server/fs-funnel.h:
5652 * gst/rtsp-server/rtsp-funnel.c:
5653 * gst/rtsp-server/rtsp-funnel.h:
5654 * gst/rtsp-server/rtsp-media.c:
5655 funnel: rename fsfunnel to rtspfunnel
5656 Rename the funnel to avoid conflicts with the farsight one.
5658 2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5660 * gst/rtsp-server/Makefile.am:
5661 * gst/rtsp-server/fs-funnel.c:
5662 * gst/rtsp-server/fs-funnel.h:
5663 * gst/rtsp-server/rtsp-media.c:
5664 rtsp-media: add and use fsfunnel
5665 Add a copy of fsfunnel to the build because input-selector removed the (broken)
5666 select-all property that we need.
5668 2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5670 * gst/rtsp-server/Makefile.am:
5671 gobject-introspection: use PKG_CONFIG_PATH specified at configure time
5672 Use PKG_CONFIG_PATH specified at configure time (if any) as well
5673 for the g-ir-compiler, rather than just assuming the env var has
5676 2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5683 build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
5685 2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5688 * gst/rtsp-server/Makefile.am:
5689 gobject-introspection: fix g-i build for uninstalled setup
5690 Requires gst-plugins-base git (> 0.10.31.2).
5692 2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5694 * examples/test-uri.c:
5695 examples: add some more options and comments
5697 2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5699 * gst/rtsp-server/rtsp-media-factory-uri.c:
5700 factory-uri: use right property type
5702 2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5704 * gst/rtsp-server/rtsp-media-factory-uri.c:
5705 factory-uri: attempt to configure buffer-lists
5706 Attempt to configure buffer lists in the payloader for improved performance.
5708 2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5710 * gst/rtsp-server/rtsp-media.c:
5711 media: attempt to configure bigger UDP buffers
5712 Attempt to configure bigger udp kernel send buffers to avoid overflowing the
5713 send buffers with high bitrate streams.
5715 2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
5717 * gst/rtsp-server/rtsp-client.c:
5718 client: use the socket length from getsockname
5719 Use the length returned by getsockname to perform the getnameinfo call because
5720 the size can depend on the socket type and platform.
5723 2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5725 * docs/libs/gst-rtsp-server-docs.sgml:
5726 * docs/libs/gst-rtsp-server-sections.txt:
5727 docs: add uri factory to the docs
5729 2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5731 * gst/rtsp-server/rtsp-client.c:
5732 * gst/rtsp-server/rtsp-media.h:
5735 2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5737 * gst/rtsp-server/rtsp-client.c:
5738 * gst/rtsp-server/rtsp-media.c:
5739 * gst/rtsp-server/rtsp-media.h:
5740 * gst/rtsp-server/rtsp-session.c:
5741 * gst/rtsp-server/rtsp-session.h:
5742 rtsp-server: add support for buffer lists
5743 Add support for sending bufferlists received from appsink.
5746 2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5748 * gst/rtsp-server/rtsp-client.c:
5749 * gst/rtsp-server/rtsp-media.c:
5750 * gst/rtsp-server/rtsp-media.h:
5751 * gst/rtsp-server/rtsp-sdp.c:
5752 media: make method to retrieve the play range
5753 Make a method to retrieve the playback range so that we can conditionally create
5754 a different range for the SDP and the PLAY requests.
5756 2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5758 * gst/rtsp-server/rtsp-media.c:
5759 * gst/rtsp-server/rtsp-media.h:
5760 media: add signal to notify of state changes
5762 2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5764 * gst/rtsp-server/rtsp-client.h:
5765 client: cleanup headers
5767 2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5769 * gst/rtsp-server/rtsp-client.c:
5772 2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5774 * gst/rtsp-server/rtsp-media-factory-uri.c:
5775 * gst/rtsp-server/rtsp-media-factory-uri.h:
5776 factory-uri: add support for gstpay
5777 Add an option to prefer gstpay over decoder + raw payloader.
5779 2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5781 * gst/rtsp-server/rtsp-media-factory-uri.c:
5782 * gst/rtsp-server/rtsp-media-factory-uri.h:
5783 factory-uri: rework the autoplugger.
5784 Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
5787 2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5789 * gst/rtsp-server/rtsp-media-factory-uri.c:
5790 factory-uri: use better factory filter
5791 Make better payloader filter based on autoplug rank and RTP use case.
5793 2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5796 Automatic update of common submodule
5797 From 169462a to 46445ad
5799 2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5801 * gst/rtsp-server/rtsp-server.c:
5802 server: set SO_REUSEADDR before bind
5803 Set the SO_REUSEADDR _before_ bind() to make it actually work.
5805 2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5807 * gst/rtsp-server/rtsp-media.c:
5808 * gst/rtsp-server/rtsp-media.h:
5809 media: emit prepared signal when prepared
5810 Make a 'prepared' signal and emit it when we successfully prepared the element.
5811 This signal can be used to configure the media object after it has been prepared
5814 2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
5817 Automatic update of common submodule
5818 From 011bcc8 to 169462a
5820 2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
5822 python an optional dependency
5823 * configure.ac: Move up valgrind and g-i checks. Make the python
5824 dependency optional, as it was before.
5826 2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5828 Merge branch 'master' into 0.11
5833 2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5835 * gst/rtsp-server/rtsp-media.c:
5836 media: update range when active clients changed
5837 When we changed the number of active clients, update the current range
5838 information because we want the second client connecting to a shared resource
5839 continue from where the stream currently.
5841 2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5843 * gst/rtsp-server/rtsp-media-factory-uri.c:
5844 * gst/rtsp-server/rtsp-media-factory-uri.h:
5845 factory-uri: add colorspace and fix pt
5846 Rework the way we pass data to the autoplugger.
5847 When we have raw caps, plug a converter element to make pluggin to raw
5848 payloaders more successful.
5849 Make sure all dynamically plugged payloaders have a unique payload types.
5851 2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5853 * examples/Makefile.am:
5854 * examples/test-uri.c:
5855 example: add example of the uri factory
5857 2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5859 * gst/rtsp-server/Makefile.am:
5860 * gst/rtsp-server/rtsp-media-factory-uri.c:
5861 * gst/rtsp-server/rtsp-media-factory-uri.h:
5862 * gst/rtsp-server/rtsp-server.h:
5863 factory-uri: add a factory to stream any URI
5864 Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
5867 2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5869 * gst/rtsp-server/rtsp-media.c:
5870 * gst/rtsp-server/rtsp-media.h:
5871 media: ignore spurious ASYNC_DONE messages
5872 When we are dynamically adding pads, the addition of the udpsrc elements will
5873 trigger an ASYNC_DONE. We have to ignore this because we only want to react to
5874 the real ASYNC_DONE when everything is prerolled.
5876 2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5878 * gst/rtsp-server/rtsp-media-factory.c:
5879 * gst/rtsp-server/rtsp-media-factory.h:
5880 media-factory: make lock macro
5882 2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
5884 * gst/rtsp-server/rtsp-client.c:
5885 rtsp-server: Remove unused variable and dead assignment
5887 2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
5889 * examples/test-launch.c:
5890 * examples/test-mp4.c:
5891 * examples/test-ogg.c:
5892 * examples/test-readme.c:
5893 * examples/test-sdp.c:
5894 * examples/test-video.c:
5895 examples: Run gst-indent
5897 2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
5899 * gst/rtsp-server/rtsp-client.c:
5900 * gst/rtsp-server/rtsp-media-factory.c:
5901 * gst/rtsp-server/rtsp-media-mapping.c:
5902 * gst/rtsp-server/rtsp-media.c:
5903 * gst/rtsp-server/rtsp-params.c:
5904 * gst/rtsp-server/rtsp-sdp.c:
5905 * gst/rtsp-server/rtsp-server.c:
5906 * gst/rtsp-server/rtsp-session-pool.c:
5907 * gst/rtsp-server/rtsp-session.c:
5908 rtsp-server: Run gst-indent
5909 Since it wasn't using the upstream common previously, there was no
5910 indentation check before commiting.
5912 2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
5914 * gst/rtsp-server/rtsp-media-mapping.h:
5915 * gst/rtsp-server/rtsp-media.c:
5916 * gst/rtsp-server/rtsp-media.h:
5917 * gst/rtsp-server/rtsp-sdp.c:
5918 * gst/rtsp-server/rtsp-session-pool.h:
5919 * gst/rtsp-server/rtsp-session.c:
5920 * gst/rtsp-server/rtsp-session.h:
5921 rtsp-server: Some more doc fixups
5923 2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5926 Makefile: Add cruft-cleaning support
5928 2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5933 * docs/libs/Makefile.am:
5934 * docs/libs/gst-rtsp-server-docs.sgml:
5935 * docs/libs/gst-rtsp-server-sections.txt:
5936 * docs/libs/gst-rtsp-server.types:
5937 * docs/version.entities.in:
5938 docs: Add gtk-doc build system
5940 2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5942 * gst/rtsp-server/Makefile.am:
5943 Makefile.am: Use standard GIR make behaviour
5945 2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
5949 autogen/configure: Bring more in sync to standard gst module behaviour
5951 2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5953 * gst/rtsp-server/rtsp-media.c:
5954 media: warn and fail when gstrtpbin is not found
5956 2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5959 configure: open 0.11 branch
5961 2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
5965 Add common submodule
5967 2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
5970 * common/Makefile.am:
5971 * common/c-to-xml.py:
5973 * common/coverage/coverage-report-entry.pl:
5974 * common/coverage/coverage-report.pl:
5975 * common/coverage/coverage-report.xsl:
5976 * common/coverage/lcov.mak:
5977 * common/gettext.patch:
5978 * common/glib-gen.mak:
5979 * common/gst-autogen.sh:
5980 * common/gst-xmlinspect.py:
5982 * common/gstdoc-scangobj:
5983 * common/gtk-doc-plugins.mak:
5984 * common/gtk-doc.mak:
5985 * common/m4/.gitignore:
5986 * common/m4/Makefile.am:
5988 * common/m4/as-ac-expand.m4:
5989 * common/m4/as-auto-alt.m4:
5990 * common/m4/as-compiler-flag.m4:
5991 * common/m4/as-compiler.m4:
5992 * common/m4/as-docbook.m4:
5993 * common/m4/as-libtool-tags.m4:
5994 * common/m4/as-libtool.m4:
5995 * common/m4/as-python.m4:
5996 * common/m4/as-scrub-include.m4:
5997 * common/m4/as-version.m4:
5998 * common/m4/ax_create_stdint_h.m4:
5999 * common/m4/check.m4:
6000 * common/m4/glib-gettext.m4:
6001 * common/m4/gst-arch.m4:
6002 * common/m4/gst-args.m4:
6003 * common/m4/gst-check.m4:
6004 * common/m4/gst-debuginfo.m4:
6005 * common/m4/gst-default.m4:
6006 * common/m4/gst-doc.m4:
6007 * common/m4/gst-error.m4:
6008 * common/m4/gst-feature.m4:
6009 * common/m4/gst-function.m4:
6010 * common/m4/gst-gettext.m4:
6011 * common/m4/gst-glib2.m4:
6012 * common/m4/gst-libxml2.m4:
6013 * common/m4/gst-plugindir.m4:
6014 * common/m4/gst-valgrind.m4:
6015 * common/m4/gtk-doc.m4:
6016 * common/m4/introspection.m4:
6018 * common/mangle-tmpl.py:
6019 * common/plugins.xsl:
6021 * common/release.mak:
6022 * common/scangobj-merge.py:
6023 * common/upload.mak:
6024 common: Remove static version
6026 2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
6028 * common/m4/introspection.m4:
6029 Update introspection.m4 to match usage
6031 2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6035 Remove old stuff from the README
6037 2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6042 === release 0.10.7 ===
6044 2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6049 2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6051 * examples/test-ogg.c:
6052 test-ogg: remove parsers
6053 Remove the parsers, they are not needed anymore as oggdemux now outputs normal
6054 buffers with timestamps. Using the parsers also seems to break things.
6056 2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6058 * bindings/vala/gst-rtsp-server-0.10.vapi:
6059 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6060 Updated Vala bindings
6062 2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6064 * common/m4/introspection.m4:
6066 * gst/rtsp-server/Makefile.am:
6067 Added initial gobject-introspection support
6069 2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6071 * gst/rtsp-server/rtsp-media-factory.c:
6072 media-factory: don't use host for shared hash key
6073 When we generate the key to share made between connections, don't include the
6074 host used to connect so that we can share media even if between clients that
6075 connected with localhost and ones with the ip address.
6077 2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6079 * bindings/vala/Makefile.am:
6080 build: fix distcheck
6082 2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6084 * bindings/vala/gst-rtsp-server-0.10.vapi:
6085 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6086 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6087 Update Vala bindings
6089 2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6091 * bindings/vala/Makefile.am:
6093 Fix configure checks and installation location for Vala bindings
6096 2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6101 === release 0.10.6 ===
6103 2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6106 configure: release 0.10.6
6108 2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6110 * gst/rtsp-server/rtsp-media.c:
6111 media: help the compiler a little
6113 2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6115 * gst/rtsp-server/rtsp-media.c:
6116 * gst/rtsp-server/rtsp-media.h:
6117 * gst/rtsp-server/rtsp-session.c:
6118 media: cleanup media transport before freeing
6119 Cleanup the media transport data before freeing. In particular, remove the qdata
6120 from the rtpsource object.
6122 2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6124 * gst/rtsp-server/rtsp-media-factory.c:
6125 * gst/rtsp-server/rtsp-media-factory.h:
6126 * gst/rtsp-server/rtsp-media.c:
6127 * gst/rtsp-server/rtsp-media.h:
6128 media-factory: add eos-shutdown property
6129 Add an eos-shutdown property that will send an EOS to the pipeline before
6130 shutting it down. This allows for nice cleanup in case of a muxer.
6133 2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6135 * gst/rtsp-server/rtsp-media.c:
6136 * gst/rtsp-server/rtsp-media.h:
6137 media: use multiudpsink send-duplicates when we can
6138 If we have a new enough multiudpsink with the send-duplicates property, use this
6139 instead of doing our own filtering. Our custom filtering code should eventually
6140 be removed when we can depend on a released -good.
6142 2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6144 * gst/rtsp-server/rtsp-media.c:
6145 media: don't leak destinations
6146 Refactor and cleanup the destinations array when the stream is destroyed.
6148 2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6150 * gst/rtsp-server/rtsp-media.c:
6151 * gst/rtsp-server/rtsp-media.h:
6152 media: don't add udp addresses multiple times
6153 Keep track of the udp addresses we added to udpsink and never add the same udp
6154 destination twice. This avoids duplicate packets when using multicast.
6156 2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6158 * gst/rtsp-server/rtsp-server.c:
6159 server: disable use of SO_LINGER
6160 SO_LINGER cause the client to fail to receive a TEARDOWN message because the
6161 server close()s the connection.
6163 2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6165 * gst/rtsp-server/rtsp-server.c:
6166 server: use 5 second linger period in SO_LINGER
6167 Wait 5 seconds before clearing the send buffers and reseting the connection with
6168 the client when we do a close. This should be enough time to get the message to
6172 2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
6174 * gst/rtsp-server/rtsp-server.c:
6175 server: use SO_LINGER
6176 SO_LINGER on the socket will make sure that any pending data on the socket is
6177 flushed ASAP and that the socket connection is reset. This makes sure that the
6178 socket can be reused immediately.
6181 2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6184 README: add blurb about shared media factories
6186 2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
6188 * gst/rtsp-server/rtsp-media.c:
6189 Add stdlib.h for atoi()
6191 2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6193 * bindings/python/Makefile.am:
6194 * bindings/vala/Makefile.am:
6195 build: distcheck fixes
6196 Fix 'make distcheck', somewhat (it still fails because it tries to
6197 install files into /usr/share/vala/vapi/ irrespective of the
6200 2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6203 configure: bump core/base requirements to released version
6204 Makes things less confusing for people.
6206 2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6209 configure: fail if GStreamer core/base requirements are not met
6211 2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6213 * gst/rtsp-server/rtsp-client.c:
6214 client: improve client cleanups
6215 Make sure the session does not timeout when using TCP. We need to do this
6216 because quicktime player does not send RTCP for some reason in tunneled
6218 Refactor some cleanup code.
6221 2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6223 * gst/rtsp-server/rtsp-session.c:
6224 * gst/rtsp-server/rtsp-session.h:
6225 session: add support for prevent session timeouts
6226 Add an atomix counter to prevent session timeouts when we are, for example,
6229 2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6231 * gst/rtsp-server/rtsp-client.c:
6232 client: fix unlink on session timeouts
6233 When our session times out, make sure we unlink all streams in this
6235 Remove the tunnelid when closing the connection.
6237 2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6239 * gst/rtsp-server/rtsp-session.c:
6240 session: small cleanups
6242 2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6244 * gst/rtsp-server/rtsp-client.c:
6245 client: handle lost_tunnel callbacks
6246 Handle lost_tunnel callbacks and use it to store the tunnelid back into the
6247 hashtable so that we can reuse it for when the client reopens the POST
6249 Close the connection after a TEARDOWN.
6250 Make sure or watchid is cleared when the watch is removed.
6253 2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6255 * gst/rtsp-server/rtsp-client.c:
6256 * gst/rtsp-server/rtsp-media.c:
6257 * gst/rtsp-server/rtsp-sdp.c:
6258 rtsp-server: add more support for multicast
6260 2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6263 * gst/rtsp-server/rtsp-media.c:
6264 * gst/rtsp-server/rtsp-media.h:
6265 media: allow configuration of allowed lower transport
6267 2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6269 * gst/rtsp-server/rtsp-client.h:
6270 * gst/rtsp-server/rtsp-media.c:
6271 * gst/rtsp-server/rtsp-media.h:
6272 * gst/rtsp-server/rtsp-sdp.c:
6273 * gst/rtsp-server/rtsp-sdp.h:
6274 * gst/rtsp-server/rtsp-server.c:
6275 rtsp: keep track of server ip and ipv6
6276 Keep track of how the client connected to the server and setup the udp ports
6277 with the same protocol.
6278 Copy the server ip address in the SDP so that clients can send RTCP back to
6281 2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6283 * gst/rtsp-server/rtsp-session.c:
6286 2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6288 * gst/rtsp-server/rtsp-client.c:
6289 client: use right size for malloc
6291 2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6293 * gst/rtsp-server/rtsp-server.c:
6294 server: comment ipv6 server listening address
6296 2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6298 * gst/rtsp-server/rtsp-media.c:
6299 media: allow for ipv6 sockets
6301 2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6303 * gst/rtsp-server/rtsp-server.c:
6304 * gst/rtsp-server/rtsp-server.h:
6305 server: rework server part
6306 Allow setting a bind address, make sure we can deal with ipv6.
6307 Remove the port property and change with the service property.
6309 2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6311 * gst/rtsp-server/rtsp-media.h:
6312 media: update comments a little
6314 2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6316 * gst/rtsp-server/rtsp-client.c:
6317 client: make content-base better
6318 Use the URI formatting functions to make a content-base. Also make sure that
6319 there is a trailing / at the end.
6321 2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6323 * gst/rtsp-server/rtsp-client.c:
6324 client: guard against invalid paths
6326 2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6328 * examples/test-video.c:
6329 test: catch server bind errors
6331 2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
6333 * gst/rtsp-server/rtsp-media.c:
6334 rtspmedia: emit "unprepared" if _prepare fails.
6335 Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
6336 media object is removed from its factory's cache.
6338 2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6340 * gst/rtsp-server/rtsp-media.c:
6341 media: collect media position when seek completes
6343 2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
6345 * gst/rtsp-server/rtsp-client.c:
6346 client: call unlink_streams in client finalize
6349 2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6351 * gst/rtsp-server/rtsp-media.c:
6352 media: limit the time to wait to something huge
6353 Avoid waiting forever but limit the timeout to 20 seconds.
6355 2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6357 * gst/rtsp-server/rtsp-sdp.c:
6358 sdp: reindent and check for prepared status
6360 2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6362 * gst/rtsp-server/rtsp-media.c:
6363 * gst/rtsp-server/rtsp-media.h:
6364 * gst/rtsp-server/rtsp-session.c:
6365 media: avoid doing _get_state() for state changes
6366 When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
6367 until the media is prerolled or in error. This avoids doing a blocking call of
6368 gst_element_get_state() that can cause lockups when there is an error.
6371 2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6373 * gst/rtsp-server/rtsp-media.c:
6376 2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6378 * gst/rtsp-server/rtsp-media-factory.c:
6379 media-factory: better error handling
6380 Improve the error handling a bit.
6382 2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6384 * gst/rtsp-server/rtsp-client.c:
6385 client: rework transport parsing
6386 Rework the transport parsing code so that we can ignore transports we don't
6387 support instead of just picking the first one we can parse.
6388 Configure a (for now hardcoded) destination for multicast transports.
6390 2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6392 * gst/rtsp-server/rtsp-media.c:
6393 media: set multicast sink parameters
6394 Disable loop and automatic multicast join on the udpsink elements.
6395 Add some more debug info.
6396 Reset some state variables in the right place.
6397 Use the right port numbers for multicast.
6399 2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6401 * gst/rtsp-server/rtsp-session.c:
6402 session: handle transport setup correctly
6403 Handle UDP, MCAST and TCP transport negotiation more correctly.
6404 Store the server session SSRC in the transport.
6406 2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6408 * gst/rtsp-server/rtsp-client.c:
6409 rtsp-client: implement error_full
6410 Implement error_full to avoid some segfaults when the rtspconnection calls it.
6413 2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6416 * gst/rtsp-server/rtsp-client.c:
6417 * gst/rtsp-server/rtsp-server.c:
6418 docs: update docs and comments
6420 2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
6422 * gst/rtsp-server/rtsp-sdp.c:
6423 sdp: make server work better when behind a proxy
6425 2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6427 * gst/rtsp-server/rtsp-client.c:
6428 client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
6430 2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6432 * gst/rtsp-server/rtsp-client.c:
6433 * gst/rtsp-server/rtsp-media-factory.c:
6434 * gst/rtsp-server/rtsp-media-mapping.c:
6435 * gst/rtsp-server/rtsp-media.c:
6436 * gst/rtsp-server/rtsp-server.c:
6437 * gst/rtsp-server/rtsp-session-pool.c:
6438 * gst/rtsp-server/rtsp-session.c:
6439 Use GStreamer's debugging subsystem
6441 2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
6443 * gst/rtsp-server/rtsp-media-factory.c:
6444 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
6446 2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6451 === release 0.10.5 ===
6453 2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6458 2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6461 configure: bump required versions
6463 2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
6465 * gst/rtsp-server/rtsp-client.c:
6466 client: call weak-unref on client->sessions from finalize
6469 2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6471 * gst/rtsp-server/rtsp-media.c:
6472 media: Fixed crasher where caps got unref'ed too often
6474 2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6477 * pkgconfig/.gitignore:
6478 * pkgconfig/Makefile.am:
6479 * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
6480 Added pkg-config file to use gst-rtsp-server uninstalled
6482 2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6484 * gst/rtsp-server/rtsp-media.c:
6485 media: add some docs
6487 2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
6489 * gst/rtsp-server/rtsp-client.c:
6490 rtsp: Use gst_rtsp_watch_send_message().
6491 Use gst_rtsp_watch_send_message() since the old API which used
6492 gst_rtsp_watch_queue_message() has been deprecated.
6494 2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6499 === release 0.10.4 ===
6501 2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6506 2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6508 * gst/rtsp-server/rtsp-client.c:
6509 * gst/rtsp-server/rtsp-session.c:
6510 * gst/rtsp-server/rtsp-session.h:
6511 rtsp: allocate channels in TCP mode
6512 When the client does not provide us with channels in TCP mode, allocate channels
6515 2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6517 * gst/rtsp-server/rtsp-client.c:
6518 client: don't crash when tunnelid is missing
6519 When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
6520 don't crash but return an error response to the client.
6523 2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6525 * bindings/vala/gst-rtsp-server-0.10.vapi:
6526 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6527 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6528 bindings: update vala bindings with new method
6530 2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6532 * gst/rtsp-server/rtsp-session-pool.c:
6533 * gst/rtsp-server/rtsp-session-pool.h:
6534 sessionpool: add function to filter sessions
6535 Add generic function to retrieve/remove sessions.
6537 2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6540 configure: bump core/base requirements to release
6542 2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6544 * gst/rtsp-server/rtsp-media.c:
6545 media: fix indentation
6547 2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6549 * gst/rtsp-server/rtsp-media.c:
6550 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
6552 2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6554 * gst/rtsp-server/rtsp-media.c:
6555 set state and remove elements of media in for loop
6557 2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
6559 * bindings/vala/gst-rtsp-server-0.10.vapi:
6560 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6561 Added gst_rtsp_media_remove_elements function to Vala bindings
6563 2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
6565 * gst/rtsp-server/rtsp-media.c:
6566 * gst/rtsp-server/rtsp-media.h:
6567 Added gst_rtsp_media_remove_elements function
6569 2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
6571 * gst/rtsp-server/rtsp-media.c:
6572 Don't use name for gstrtpbin so we can add multiple instances to the pipeline
6574 2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6576 * bindings/vala/gst-rtsp-server-0.10.vapi:
6577 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6578 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6579 Updated Vala bindings
6581 2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6583 * gst/rtsp-server/rtsp-media.c:
6584 * gst/rtsp-server/rtsp-media.h:
6585 Added vmethod unprepare to GstRTSPMedia
6586 The default implementation sets the state of the pipeline to GST_STATE_NULL
6588 2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6590 * gst/rtsp-server/rtsp-media-factory.c:
6591 * gst/rtsp-server/rtsp-media-factory.h:
6592 Made collect_streams function public
6594 2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6596 * gst/rtsp-server/rtsp-media-factory.c:
6597 * gst/rtsp-server/rtsp-media-factory.h:
6598 * gst/rtsp-server/rtsp-media.c:
6599 Added vmethod create_pipeline to GstRTSPMediaFactory
6600 The pipeline is created in this method and the GstRTSPMedia's element is added to it
6602 2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6604 * gst/rtsp-server/rtsp-client.c:
6605 client: use g_source_destroy()
6606 We need to use g_source_destroy() because we might have added the source to a
6607 different main context than the default one.
6609 2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6611 * gst/rtsp-server/Makefile.am:
6612 * gst/rtsp-server/rtsp-client.c:
6613 * gst/rtsp-server/rtsp-params.c:
6614 * gst/rtsp-server/rtsp-params.h:
6615 rtsp: prepare for handling GET/SET_PARAMETER
6616 Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
6618 Fix return codes of handlers.
6620 2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6622 * gst/rtsp-server/rtsp-media.c:
6623 media: don't leak session pads
6625 2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6627 * gst/rtsp-server/rtsp-media.c:
6628 media: clean up the messages a bit
6630 2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6632 * gst/rtsp-server/rtsp-sdp.c:
6633 sdp: warn and skip streams without media
6635 2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6637 * bindings/vala/gst-rtsp-server-0.10.vapi:
6638 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6639 vala: Fixed typo in header file of RTSPMediaStream
6641 2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6643 * gst/rtsp-server/rtsp-media.c:
6646 Make dumping RTCP stats configurable
6648 2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6650 * gst/rtsp-server/rtsp-media.c:
6651 media: be less verbose and leak less
6653 2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6655 * gst/rtsp-server/rtsp-media.c:
6656 media: don't leak the destination address
6658 2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6660 * gst/rtsp-server/rtsp-client.c:
6661 * gst/rtsp-server/rtsp-media.c:
6662 * gst/rtsp-server/rtsp-media.h:
6663 * gst/rtsp-server/rtsp-session.c:
6664 * gst/rtsp-server/rtsp-session.h:
6665 rtsp: use RTCP to keep the session alive
6666 Use the RTCP rtcp-from stats field to find the associated session and use this
6667 to keep the session alive.
6669 2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6671 * gst/rtsp-server/rtsp-session.c:
6672 session: add 5sec to the real session timeout
6673 Allow the session to live 5sec longer before really timing out. This should give
6674 clients some extra time to keep the session active.
6676 2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6678 * gst/rtsp-server/rtsp-client.c:
6679 client: replay OK to GET/SET_PARAMETER
6680 Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
6681 so that we return OK for those requests.
6683 2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6685 * gst/rtsp-server/rtsp-media.c:
6686 * gst/rtsp-server/rtsp-media.h:
6687 media: keep track of active transports
6688 Keep track of which transport is active to avoid closing the connection too
6690 Remove the destination transport also when going to NULL.
6691 Print some stats about the SDES and other RTCP messages we receive from the
6694 2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6696 * examples/.gitignore:
6697 * examples/Makefile.am:
6698 * examples/test-sdp.c:
6699 example: add SDP relay example
6701 2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6703 * gst/rtsp-server/rtsp-media.c:
6704 media: also count active TCP connections
6706 2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6708 * gst/rtsp-server/rtsp-media-factory.c:
6709 * gst/rtsp-server/rtsp-media.c:
6710 * gst/rtsp-server/rtsp-media.h:
6711 rtsp: add support for dynamic elements
6712 Add support for dynamic elements.
6713 Don't set live pipelines back to paused.
6715 2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6717 * gst/rtsp-server/rtsp-sdp.c:
6718 sdp: don't add encoding name when absent in caps
6720 2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6722 * gst/rtsp-server/rtsp-client.c:
6723 client: warn when we can't do RTP-Info
6725 2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6727 * gst/rtsp-server/rtsp-media-factory.c:
6728 factory: factor out the stream construction
6730 2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6732 * gst/rtsp-server/rtsp-client.c:
6733 client: only add RTP-Info when we have the info
6734 Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
6737 2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6742 === release 0.10.3 ===
6744 2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6748 - Fixes a bug where it put the wrong verion in pkgconfig
6749 - Link RTP and RTCP sources
6751 2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6753 * gst/rtsp-server/rtsp-media.c:
6754 * gst/rtsp-server/rtsp-media.h:
6755 media: link the RTP udpsrc to the session manager
6756 Link the RTP udpsrc and the appsrc to the session manager so that they don't
6757 shut down when the client sends a packet to open firewalls.
6759 2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6761 * pkgconfig/gst-rtsp-server.pc.in:
6762 Don't use hard-coded version number in pkg-config file
6764 2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6769 === release 0.10.2 ===
6771 2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6776 2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6779 * common/m4/.gitignore:
6780 * examples/.gitignore:
6781 * pkgconfig/.gitignore:
6782 add some .gitignore files
6784 2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6786 * gst/rtsp-server/rtsp-media.c:
6787 media: seek to key frames
6789 2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6791 * gst/rtsp-server/rtsp-media.c:
6792 media: emit the unprepared signal by id
6793 Emit the unprepared signal by id instead of name and set the media as
6796 2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6798 * gst/rtsp-server/rtsp-media.c:
6799 Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
6801 2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6803 * gst/rtsp-server/rtsp-server.c:
6804 Added finalize function to GstRTPSPServer to unref session pool and media mapping
6806 2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
6808 * bindings/vala/gst-rtsp-server-0.10.vapi:
6809 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
6810 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
6811 Updated vala bindings
6813 2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6815 * gst/rtsp-server/Makefile.am:
6816 * gst/rtsp-server/rtsp-client.c:
6817 * gst/rtsp-server/rtsp-media.c:
6818 server: use appsink and appsrc with the API
6819 Use the appsink/appsrc API instead of the signals for higher
6822 2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6824 * examples/test-ogg.c:
6825 tests: set the payload type correctly
6827 2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6829 * gst/rtsp-server/rtsp-media-factory.c:
6830 factory: connect to the unprepare signal
6831 Connect to the unprepare signal for non-reusable media so that we can remove
6832 them from the cache.
6834 2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6836 * gst/rtsp-server/rtsp-media.c:
6837 * gst/rtsp-server/rtsp-media.h:
6838 media: add signal to notify of unprepare
6840 2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6842 * gst/rtsp-server/rtsp-media.c:
6843 * gst/rtsp-server/rtsp-media.h:
6844 media: more work on making the media shared
6845 Add a reusable flag to medias, indicating that they can be reused after a state
6849 2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6851 * examples/test-readme.c:
6852 examples: mark the example as shared for testing
6854 2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6856 * gst/rtsp-server/rtsp-media.c:
6857 * gst/rtsp-server/rtsp-media.h:
6858 client: support shared media
6859 Always perform the state actions even if the target state of the pipeline is
6860 already correct, we still want to add/remove the transports when we are dealing
6862 Keep a counter of the number of active transports for a media so that we can use
6863 this to perform a state change when needed.
6864 Perform a state change of the pipeline only when the first transport was added
6865 or when there are no active transports.
6867 2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
6869 * gst/rtsp-server/rtsp-client.c:
6870 client: fix refcounting crasher
6871 Don't need to remove the weak refs in the finalize methods, they are already
6872 removed in the dispose.
6873 Don't register the callback with a DestroyNofity.
6875 2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6877 * gst/rtsp-server/rtsp-client.c:
6878 Fix rtsp client refcount management in TCP mode.
6879 Don't unref a client ref we never had. Fixes an unref
6880 of an already-free client object after a client
6881 teardown request for me.
6883 2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6885 * gst/rtsp-server/rtsp-session.c:
6886 docs: fix typo in API docs
6888 2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6890 * gst/rtsp-server/rtsp-media.c:
6892 Keep the udp sources in playing even if we go to paused. unlock the sources when
6894 Add some more debug info.
6895 Only seek when we need to.
6896 Keep track of the position when we go to paused.
6898 2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6900 * gst/rtsp-server/rtsp-client.c:
6901 * gst/rtsp-server/rtsp-media.c:
6902 * gst/rtsp-server/rtsp-media.h:
6903 Add beginnings of seeking.
6904 Parse the Range header and perform a seek on the pipeline for the requested
6905 position. It's disabled currently until I figure out what's going wrong.
6907 2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6909 * gst/rtsp-server/rtsp-client.c:
6910 allow pause requests for now.
6913 2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6915 * gst/rtsp-server/rtsp-client.c:
6916 Remove weak ref on the session in teardown
6917 We need to remove our weakref from the session when we do a teardown because
6918 else we close the TCP connection prematurely.
6920 2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6922 * gst/rtsp-server/rtsp-client.c:
6923 * gst/rtsp-server/rtsp-client.h:
6924 * gst/rtsp-server/rtsp-session-pool.c:
6925 Do some more session cleanup
6926 Make session timeout kill the TCP connection that currently watches the
6928 Remove the client timeout property.
6930 2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6932 * gst/rtsp-server/rtsp-client.c:
6933 * gst/rtsp-server/rtsp-client.h:
6934 * gst/rtsp-server/rtsp-media.c:
6935 * gst/rtsp-server/rtsp-media.h:
6936 * gst/rtsp-server/rtsp-server.c:
6937 * gst/rtsp-server/rtsp-session.c:
6938 * gst/rtsp-server/rtsp-session.h:
6940 Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
6943 2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6945 * examples/Makefile.am:
6946 * examples/test-launch.c:
6947 Add example server that takes launch lines
6948 Add an example server that streams any -launch line.
6950 2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6952 * examples/test-readme.c:
6953 * gst/rtsp-server/rtsp-client.c:
6954 * gst/rtsp-server/rtsp-media.c:
6955 * gst/rtsp-server/rtsp-media.h:
6956 Add support for live streams
6957 Add support for live streams and ranges
6958 Start on handling TCP data transfer.
6960 2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6962 * gst/rtsp-server/rtsp-media.c:
6963 Free the pipeline before other things
6966 2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6968 * gst/rtsp-server/rtsp-client.c:
6969 Only free the pending tunnel if there is one
6972 2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6974 * gst/rtsp-server/rtsp-client.c:
6975 * gst/rtsp-server/rtsp-client.h:
6976 * gst/rtsp-server/rtsp-media.c:
6977 rtsp-server: Add support for tunneling
6978 Add support for tunneling over HTTP.
6979 Use new connection methods to retrieve the url.
6980 Dispatch messages based on the message type instead of blindly
6981 assuming it's always a request.
6982 Keep track of the watch id so that we can remove it later.
6983 Set the media pipeline to NULL before unreffing the pipeline.
6985 2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6987 * gst/rtsp-server/rtsp-client.c:
6988 * gst/rtsp-server/rtsp-client.h:
6989 Fix for channel -> watch rename in gstreamer
6990 Rename the RTSPChannel to RTSPWatch and remove an unused variable.
6992 2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6994 * gst/rtsp-server/rtsp-client.c:
6995 * gst/rtsp-server/rtsp-client.h:
6997 Use the async RTSP channels instead of spawning a new thread for each client.
6998 If a sessionid is specified in a request, fail if we don't have the session.
7000 2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7002 * gst/rtsp-server/rtsp-media.c:
7003 Add better debug info
7004 Add some better debug info.
7006 2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7008 * examples/test-video.c:
7010 Add support for session timeouts in the example.
7012 2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7014 * gst/rtsp-server/rtsp-session-pool.c:
7015 * gst/rtsp-server/rtsp-session-pool.h:
7016 Pass GTimeVal around for performance reasons
7017 Get the current time only once and pass it around so that sessions don't have to
7018 get the current time anymore.
7019 Add experimental support for a GSource that dispatches when the session needs to
7022 2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7024 * gst/rtsp-server/rtsp-session.c:
7025 * gst/rtsp-server/rtsp-session.h:
7026 Add better support for session timeouts
7027 Add a method to request the number of milliseconds when a session will timeout.
7029 2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7031 * gst/rtsp-server/rtsp-media.c:
7032 * gst/rtsp-server/rtsp-media.h:
7033 Add suport for RTP manager monitoring
7034 Add the first stage in monitoring the rtp manager.
7035 Make sure we don't update the state to something we don't want.
7037 2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7039 * gst/rtsp-server/rtsp-client.c:
7040 Add support for session keepalive
7041 Get and update the session timeout for all requests. get the session as early as
7044 2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7046 * gst/rtsp-server/rtsp-media-factory.h:
7047 * gst/rtsp-server/rtsp-media.c:
7048 * gst/rtsp-server/rtsp-media.h:
7049 Handle media bus messages
7050 Handle media bus messages in a custom mainloop and dispatch them to the
7051 RTSPMedia objects. Let the default implementation handle some common messages.
7053 2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7055 * gst/rtsp-server/rtsp-client.c:
7056 * gst/rtsp-server/rtsp-session-pool.c:
7057 * gst/rtsp-server/rtsp-session.c:
7058 Some more session timeout handling
7059 Move the session header setting code to a central place so that we always add
7060 the timeout parameter too.
7061 Handle timeouts by running the session cleanup code.
7062 Stop media before cleaning up.
7064 2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7066 * gst/rtsp-server/rtsp-client.c:
7067 * gst/rtsp-server/rtsp-client.h:
7068 Add timeout property
7069 Add a timeout property ot the client and make the other properties into GObject
7072 2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7074 * gst/rtsp-server/rtsp-session-pool.c:
7075 Use getters and setters in property code
7076 Use the getters and setters for the timeout property instead of locking
7079 2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7081 Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
7083 2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7085 * gst/rtsp-server/rtsp-session-pool.c:
7086 * gst/rtsp-server/rtsp-session-pool.h:
7087 * gst/rtsp-server/rtsp-session.c:
7088 * gst/rtsp-server/rtsp-session.h:
7089 Add more timeout stuff
7090 Add method to check if a session is expired.
7091 Add method to perform cleanup on a session pool.
7093 2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7095 * gst/rtsp-server/rtsp-client.c:
7096 * gst/rtsp-server/rtsp-session-pool.c:
7097 * gst/rtsp-server/rtsp-session-pool.h:
7098 * gst/rtsp-server/rtsp-session.c:
7099 * gst/rtsp-server/rtsp-session.h:
7100 Add beginnings of session timeouts and limits
7101 Add the timeout value to the Session header for unusual timeout values.
7102 Allow us to configure a limit to the amount of active sessions in a pool. Set a
7103 limit on the amount of retry we do after a sessionid collision.
7104 Add properties to the sessionid and the timeout of a session. Keep track of
7105 creation time and last access time for sessions.
7107 2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7109 * gst/rtsp-server/rtsp-client.c:
7110 * gst/rtsp-server/rtsp-media.c:
7111 * gst/rtsp-server/rtsp-media.h:
7112 * gst/rtsp-server/rtsp-sdp.c:
7113 * gst/rtsp-server/rtsp-session-pool.c:
7114 * gst/rtsp-server/rtsp-session.c:
7115 * gst/rtsp-server/rtsp-session.h:
7116 Cleanup of sessions and more
7117 Fix the refcounting of media and sessions in the client. Properly clean up the
7118 session data when the client performs a teardown.
7119 Add Server header to responses.
7120 Allow for multiple uri setups in one session.
7121 Add Range header to the PLAY response and add the range attribute to the SDP
7123 Fix the session pool remove method, it used the wrong key in the hashtable. Also
7124 give the ownership of the sessionid to the session object.
7126 2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7128 * gst/rtsp-server/rtsp-server.c:
7129 * gst/rtsp-server/rtsp-server.h:
7131 Rename the 'server_port' variable to simply 'port'.
7133 2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7136 * gst/rtsp-server/rtsp-client.c:
7137 * gst/rtsp-server/rtsp-media.c:
7138 * gst/rtsp-server/rtsp-media.h:
7139 * gst/rtsp-server/rtsp-session.c:
7140 * gst/rtsp-server/rtsp-session.h:
7141 Rework the way we handle transports for streams
7142 Make the media accept an array of transports for the streams that we have
7143 configured for the play/pause requests.
7144 Implement server states for a client and its media.
7145 Require 0.10.22.1 (git HEAD) of gstreamer.
7147 2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7149 * gst/rtsp-server/rtsp-client.c:
7150 * gst/rtsp-server/rtsp-media-factory.c:
7151 Drop const from functions dealing with urls
7152 Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
7153 have the right const in them.
7155 2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7157 * gst/rtsp-server/rtsp-client.c:
7158 * gst/rtsp-server/rtsp-media.c:
7159 * gst/rtsp-server/rtsp-sdp.c:
7163 2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7165 * gst/rtsp-server/rtsp-client.c:
7166 * gst/rtsp-server/rtsp-media-factory.c:
7167 * gst/rtsp-server/rtsp-media.c:
7168 * gst/rtsp-server/rtsp-media.h:
7170 Don't keep a reference to the GstRTSPMedia in the stream.
7171 Free more things when freeing the GstRTSPMedia.
7173 2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7176 * gst/rtsp-server/rtsp-media-factory.c:
7177 * gst/rtsp-server/rtsp-media-factory.h:
7178 * gst/rtsp-server/rtsp-media.c:
7179 * gst/rtsp-server/rtsp-media.h:
7180 * gst/rtsp-server/rtsp-server.c:
7181 * gst/rtsp-server/rtsp-server.h:
7182 More docs and small cleanups
7183 Add some more docs and update the README
7184 Cleanup some method names.
7185 Remove an unneeded idx field in the GstRTSPMediaStream
7187 2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7190 * examples/Makefile.am:
7191 * examples/test-readme.c:
7192 Add a README and more example code
7193 Add a README file that contains a small introduction on how to use the server
7194 along with the example code explained in the readme.
7196 2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7198 * gst/rtsp-server/rtsp-media.c:
7199 * gst/rtsp-server/rtsp-server.c:
7200 Fix some leaks and change default port
7201 Fix some memory leaks by setting the udpsrc elements to the unlocked state after
7202 we finished the initial preroll. If we keep them locked, setting the pipeline to
7203 NULL will not stop and clean up the sources correctly.
7204 Change the default RTSP port to 8554 aka the official alternative RTSP port.
7206 2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7208 * gst/rtsp-server/rtsp-session.c:
7209 * gst/rtsp-server/rtsp-session.h:
7210 Cleanups to the session object
7211 Remove some unneeded variables in the session state of a stream such as the
7212 owner media and the server transport.
7213 Get the configuration of a media stream in a session based on the media_stream
7214 in the original object instead of our cached index.
7215 Free more data in the finalize method.
7217 2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7219 * gst/rtsp-server/rtsp-client.c:
7220 * gst/rtsp-server/rtsp-client.h:
7221 Cleanups and reuse media from DESCRIBE
7222 Handle thread create errors.
7223 Rename some internal methods to better match what they actually do.
7224 Handle misconfiguration of session_pool and media_mapping gracefully.
7225 Cache the DESCRIBE media and uri in the client connection and reuse them when
7226 we receive a SETUP request in the same connection for the same uri.
7227 Cleanup the client connection object.
7229 2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7231 * gst/rtsp-server/rtsp-media-factory.c:
7232 * gst/rtsp-server/rtsp-media-factory.h:
7233 * gst/rtsp-server/rtsp-media.c:
7234 * gst/rtsp-server/rtsp-media.h:
7235 Add shared properties to media and factory
7236 Add the shared property to media.
7237 Implement some simple caching in the factory depending on if the media is shared
7240 2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7242 * gst/rtsp-server/rtsp-client.c:
7243 Add a little comment
7244 Add some comment about the content-base header.
7246 2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7248 * examples/Makefile.am:
7250 * examples/test-mp4.c:
7251 * examples/test-ogg.c:
7252 * examples/test-video.c:
7253 * gst/rtsp-server/Makefile.am:
7254 * gst/rtsp-server/rtsp-client.c:
7255 * gst/rtsp-server/rtsp-client.h:
7256 * gst/rtsp-server/rtsp-media-factory.c:
7257 * gst/rtsp-server/rtsp-media-factory.h:
7258 * gst/rtsp-server/rtsp-media.c:
7259 * gst/rtsp-server/rtsp-media.h:
7260 * gst/rtsp-server/rtsp-sdp.c:
7261 * gst/rtsp-server/rtsp-sdp.h:
7262 * gst/rtsp-server/rtsp-server.c:
7263 * gst/rtsp-server/rtsp-server.h:
7264 * gst/rtsp-server/rtsp-session.c:
7265 * gst/rtsp-server/rtsp-session.h:
7266 Reorganize things, prepare for media sharing
7267 Added various other test server examples
7268 Move the SDP message generation to a separate helper.
7269 Refactor common code for finding the session.
7270 Add content-base for realplayer compatibility
7271 Clean up request uris before processing for better vlc compatibility.
7272 Move prerolling and pipeline construction to the RTSPMedia object.
7273 Use multiudpsink for future pipeline reuse.
7275 2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7281 === release 0.10.1 ===
7283 2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7289 2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7291 * bindings/vala/Makefile.am:
7293 Add more directories and files to the dist.
7295 2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7297 * bindings/python/Makefile.am:
7298 * bindings/python/rtspserver.override:
7299 Fixed compile error of python bindings
7301 2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7303 * bindings/vala/gst-rtsp-server-0.10.vapi:
7304 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7305 Marked values as nullable accordingly
7307 2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7309 * bindings/vala/gst-rtsp-server-0.10.vapi:
7310 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
7311 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7312 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7313 Updated Vala bindings
7315 2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7317 * gst/rtsp-server/rtsp-client.c:
7318 * gst/rtsp-server/rtsp-media-mapping.c:
7319 * gst/rtsp-server/rtsp-media-mapping.h:
7320 * gst/rtsp-server/rtsp-media.h:
7321 * gst/rtsp-server/rtsp-session-pool.h:
7322 Cleanups and doc updates
7323 Add some more documentation and do some minor cleanups here and there.
7325 2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7327 * gst/rtsp-server/rtsp-client.c:
7328 * gst/rtsp-server/rtsp-media-factory.c:
7329 * gst/rtsp-server/rtsp-media-factory.h:
7330 * gst/rtsp-server/rtsp-media.c:
7331 * gst/rtsp-server/rtsp-media.h:
7332 * gst/rtsp-server/rtsp-session.c:
7333 * gst/rtsp-server/rtsp-session.h:
7335 Rename GstRTSPMediaBin to GstRTSPMedia
7336 Parse the request url into a GstRTSPUri object and pass this object to the
7337 various handlers and methods that require the uri.
7339 2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7343 Add some more docs and remove some old code from the example.
7345 2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7347 * gst/rtsp-server/rtsp-client.c:
7348 Handle state change failures better
7349 Handle state change failures better when changing the state of the pipeline to
7352 2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7354 * gst/rtsp-server/rtsp-media-factory.c:
7355 * gst/rtsp-server/rtsp-media-factory.h:
7356 Make element creation more extendible
7357 Add get_element vmethod to the default MediaFactory so that subclasses can just
7358 override that method and still use the default logic for making a MediaBin from
7361 2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7364 * gst/rtsp-server/Makefile.am:
7365 * gst/rtsp-server/rtsp-client.c:
7366 * gst/rtsp-server/rtsp-client.h:
7367 * gst/rtsp-server/rtsp-media-factory.c:
7368 * gst/rtsp-server/rtsp-media-factory.h:
7369 * gst/rtsp-server/rtsp-media-mapping.c:
7370 * gst/rtsp-server/rtsp-media-mapping.h:
7371 * gst/rtsp-server/rtsp-media.c:
7372 * gst/rtsp-server/rtsp-media.h:
7373 * gst/rtsp-server/rtsp-server.c:
7374 * gst/rtsp-server/rtsp-server.h:
7375 * gst/rtsp-server/rtsp-session.c:
7376 * gst/rtsp-server/rtsp-session.h:
7377 Make the server handle arbitrary pipelines
7378 Make GstMediaFactory an object that can instantiate GstMediaBin objects.
7379 The GstMediaBin object has a handle to a bin with elements and to a list of
7380 GstMediaStream objects that this bin produces.
7381 Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
7382 with methods to register and remove those mappings.
7383 Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
7384 used by the server instance.
7385 Modify the example application so that it shows how to create custom pipelines
7386 attached to a specific mount point.
7387 Various misc cleanps.
7389 2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7391 * gst/rtsp-server/rtsp-server.c:
7392 * gst/rtsp-server/rtsp-server.h:
7393 Allow setting a custom media factory for a server
7395 2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7397 * gst/rtsp-server/rtsp-client.c:
7398 * gst/rtsp-server/rtsp-client.h:
7399 Allow setting a custom media factory for a client.
7401 2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7403 * gst/rtsp-server/Makefile.am:
7404 Add Makefile entry for the media factory
7406 2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7408 * gst/rtsp-server/rtsp-media-factory.c:
7409 * gst/rtsp-server/rtsp-media-factory.h:
7410 Add media factory to map urls to media pipeline objects.
7412 2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7414 * gst/rtsp-server/rtsp-media.c:
7415 * gst/rtsp-server/rtsp-media.h:
7416 Add comments. Remove unused field
7418 2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7420 * gst/rtsp-server/rtsp-session-pool.c:
7421 * gst/rtsp-server/rtsp-session-pool.h:
7422 Allow custom session pools to override the session id allocation algorithms Add some comments.
7424 2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7426 * gst/rtsp-server/rtsp-session.h:
7429 2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7431 * gst/rtsp-server/rtsp-client.c:
7432 * gst/rtsp-server/rtsp-client.h:
7433 Move the connection code in one place Add some comments
7435 2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7437 * gst/rtsp-server/rtsp-server.c:
7438 * gst/rtsp-server/rtsp-server.h:
7439 Make vmethod to create and accept new clients. Add some docs.
7441 2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7443 * gst/rtsp-server/rtsp-server.c:
7444 * gst/rtsp-server/rtsp-server.h:
7445 Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
7447 2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7449 * gst/rtsp-server/rtsp-client.c:
7450 * gst/rtsp-server/rtsp-client.h:
7451 Name the parameters more appropriately.
7453 2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7455 * gst/rtsp-server/rtsp-session-pool.c:
7456 Do some more cleanup of the session pool.
7458 2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7460 * gst/rtsp-server/Makefile.am:
7461 * gst/rtsp-server/rtsp-client.c:
7462 Check if return value of gst_rtsp_session_get_media is not NULL
7464 2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7466 * gst/rtsp-server/Makefile.am:
7467 Install rtsp-session and rtsp-session-pool headers
7469 2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7474 * bindings/python/Makefile.am:
7475 * bindings/python/arg-types.py:
7476 * bindings/python/codegen/Makefile.am:
7477 * bindings/python/codegen/__init__.py:
7478 * bindings/python/codegen/argtypes.py:
7479 * bindings/python/codegen/code-coverage.py:
7480 * bindings/python/codegen/codegen.py:
7481 * bindings/python/codegen/definitions.py:
7482 * bindings/python/codegen/defsparser.py:
7483 * bindings/python/codegen/docextract.py:
7484 * bindings/python/codegen/docgen.py:
7485 * bindings/python/codegen/fileprefix.override:
7486 * bindings/python/codegen/fileprefixmodule.c:
7487 * bindings/python/codegen/h2def.py:
7488 * bindings/python/codegen/mergedefs.py:
7489 * bindings/python/codegen/mkskel.py:
7490 * bindings/python/codegen/override.py:
7491 * bindings/python/codegen/reversewrapper.py:
7492 * bindings/python/codegen/scmexpr.py:
7493 * bindings/python/rtspserver-types.defs:
7494 * bindings/python/rtspserver.defs:
7495 * bindings/python/rtspserver.override:
7496 * bindings/python/rtspservermodule.c:
7498 Add python bindings.
7500 2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7502 * bindings/Makefile.am:
7504 Don't go into python dir when requirements for python bindings are missing
7506 2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7508 * bindings/Makefile.am:
7509 * bindings/vala/Makefile.am:
7511 Install Vala bindings if vala is available
7513 2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7515 * bindings/vala/gst-rtsp-server-0.10.deps:
7516 * bindings/vala/gst-rtsp-server-0.10.vapi:
7517 * bindings/vala/gst-rtsp-server.vapi:
7518 * bindings/vala/packages/gst-rtsp-server-0.10.deps:
7519 * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
7520 * bindings/vala/packages/gst-rtsp-server-0.10.files:
7521 * bindings/vala/packages/gst-rtsp-server-0.10.gi:
7522 * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
7523 * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
7524 * bindings/vala/packages/gst-rtsp-server.deps:
7525 * bindings/vala/packages/gst-rtsp-server.excludes:
7526 * bindings/vala/packages/gst-rtsp-server.files:
7527 * bindings/vala/packages/gst-rtsp-server.gi:
7528 * bindings/vala/packages/gst-rtsp-server.metadata:
7529 * bindings/vala/packages/gst-rtsp-server.namespace:
7530 Regenerated Vala bindings
7532 2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
7534 * bindings/vala/gst-rtsp-server.vapi:
7535 * bindings/vala/packages/gst-rtsp-server.metadata:
7536 Fixed typo in included headers for vala bindings
7538 2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7542 * pkgconfig/Makefile.am:
7543 * pkgconfig/gst-rtsp-server.pc.in:
7544 Added pkgconfig file
7546 2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7548 * bindings/vala/gst-rtsp-server.vapi:
7549 * bindings/vala/packages/gst-rtsp-server.excludes:
7550 * bindings/vala/packages/gst-rtsp-server.gi:
7551 * bindings/vala/packages/gst-rtsp-server.metadata:
7552 Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
7554 2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
7556 * bindings/vala/gst-rtsp-server.vapi:
7557 * bindings/vala/packages/gst-rtsp-server.deps:
7558 * bindings/vala/packages/gst-rtsp-server.files:
7559 * bindings/vala/packages/gst-rtsp-server.gi:
7560 * bindings/vala/packages/gst-rtsp-server.metadata:
7561 * bindings/vala/packages/gst-rtsp-server.namespace:
7564 2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
7566 * gst/rtsp-server/rtsp-session.c:
7567 Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
7569 2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7571 * examples/Makefile.am:
7572 * gst/rtsp-server/Makefile.am:
7573 Put GStreamer version in library name
7575 2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7577 * examples/Makefile.am:
7578 * gst/rtsp-server/Makefile.am:
7579 Fix some issues to pass distcheck
7581 2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7583 * gst/rtsp-server/rtsp-server.c:
7584 Added port property to GstRTSPServer class.
7586 2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7591 * examples/Makefile.am:
7594 * gst/rtsp-server/Makefile.am:
7595 * gst/rtsp-server/rtsp-client.c:
7596 * gst/rtsp-server/rtsp-client.h:
7597 * gst/rtsp-server/rtsp-media.c:
7598 * gst/rtsp-server/rtsp-media.h:
7599 * gst/rtsp-server/rtsp-server.c:
7600 * gst/rtsp-server/rtsp-server.h:
7601 * gst/rtsp-server/rtsp-session-pool.c:
7602 * gst/rtsp-server/rtsp-session-pool.h:
7603 * gst/rtsp-server/rtsp-session.c:
7604 * gst/rtsp-server/rtsp-session.h:
7607 * src/rtsp-client.c:
7608 * src/rtsp-client.h:
7611 * src/rtsp-server.c:
7612 * src/rtsp-server.h:
7613 * src/rtsp-session-pool.c:
7614 * src/rtsp-session-pool.h:
7615 * src/rtsp-session.c:
7616 * src/rtsp-session.h:
7617 Split in library and example program
7619 2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7621 * src/rtsp-client.h:
7622 Removed obsolete variable
7624 2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
7626 * src/rtsp-client.c:
7627 * src/rtsp-client.h:
7628 Removed pipeline variable GstRTSPClient, because it's only used in one function
7630 2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7633 Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
7635 2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
7637 * src/rtsp-session.c:
7638 Initialize some more vars.
7640 2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
7642 * src/rtsp-session.c:
7643 Initialize variable to avoid compiler warning.
7645 2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
7648 Add a reasonable generic .gitignore